From <EMAIL-ADDRESS-HIDDEN> Fri Oct 13 03:41:45 2000
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Subject: [Live-devel] LiveMedia on windows
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Hello! 
 I am a PhD student from University of Genova, Italy. My lab is 
working on an MPEG-4 compliant Facial Animation Engine. 
Specifically, my job is interfacing the animation engine with the 
Web. I have developed a netscape plugin which downloads 
animations from an httpd server. I am now starting to work on 
MPEG-4 animation bitstream streaming with RTP. 

I was looking for a c/C++ RTP library, and found live.com's 
liveMedia. 

I succesfully compiled the lib on windows, but cannot compile the 
samples, because the BasicUsage Environment does not compile 
on windows. Does this mean that it is not possible to use the lib on 
windows by now, or that there are alternative ways of using it? 

Can you help me with some hints/sample source code?

Carlo

Carlo Bonamico
PhD student 
DSP - Lab, 
DIST, University of Genova
Via Opera Pia 13, 16145 GENOVA, ITALY

From <EMAIL-ADDRESS-HIDDEN> Fri Oct 13 15:19:15 2000
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Date: Fri, 13 Oct 2000 15:17:03 -0700
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At 03:41 AM 10/13/00, charlieb(at)dist.unige.it wrote:
>Hello!
>  I am a PhD student from University of Genova, Italy. My lab is
>working on an MPEG-4 compliant Facial Animation Engine.
>Specifically, my job is interfacing the animation engine with the
>Web. I have developed a netscape plugin which downloads
>animations from an httpd server. I am now starting to work on
>MPEG-4 animation bitstream streaming with RTP.

Wow!  This sounds very interesting.

>I was looking for a c/C++ RTP library, and found live.com's
>liveMedia.
>
>I succesfully compiled the lib on windows, but cannot compile the
>samples, because the BasicUsage Environment does not compile
>on windows. Does this mean that it is not possible to use the lib on
>windows by now, or that there are alternative ways of using it?

The BasicUsageEnvironment class currently doesn't compile on Windows, 
mainly because Windows doesn't provide the setitimer() interval timer 
function that's present on Unix.  Windows, does - apparently - provide a 
"SetTimer()" function that's similar, so it should be possible to (using 
"#ifdef"s) get BasicUsageEnvironment to compile and run under both Unix and 
Windows.

Alternatively, one could develop their own subclass of UsageEnvironment 
specifically for Windows, or for some common usage environment that itself 
is portable to both Unix and Windows.  For instance, my "liveCaster" tool - 
<http://www.live.com/liveCaster/> - uses this library in combination with 
"TclUsageEnvironment": a subclass of UsageEnvironment developed 
specifically to run within the Tcl scripting language environment.  (At 
some point in the future I'll probably include TclUsageEnvironment in 
"LIVE.COM Streaming Media" along with BasicUsageEnvironment, but it's not 
quite ready to release just yet.

Anyway, good luck on your work with MPEG-4.  Should you develop MPEG-4 
specific subclasses of "liveMedia", I hope you'll consider contributing 
them to the library.

         Ross.


From <EMAIL-ADDRESS-HIDDEN> Thu Oct 26 20:38:16 2000
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Subject: [Live-devel] New implementation of BasicUsageEnvironment - now portable to
 Windows
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FYI, I have released a new version (2000.10.26) of "LIVE.COM Streaming 
Media" that improves the implementation of BasicUsageEnvironment to make it 
portable to Windows as well as Unix.  (The previous implementation was for 
Unix only.)

Because of this, the test server and client in "testProgs" can also now be 
built for Windows as well as Unix.

(Thanks to Carlo Bonamico for inspiring this.)

	Ross.

From <EMAIL-ADDRESS-HIDDEN> Fri Dec 29 14:37:09 2000
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Status: O

FYI, as of the most recent release (2000.12.28) of the "LIVE.COM Streaming 
Media" source code <http://live.sourceforge.net/>, ".cpp" is used as the 
suffix for C++ source files (rather than the previous suffix, ".C").

This means that the code will now compile directly on Windows (Visual C++), 
without having to first rename the files.

	Ross.



From <EMAIL-ADDRESS-HIDDEN> Tue Jan 09 14:02:49 2001
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Subject: [Live-devel] New "LIVE.COM Streaming Media" version
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Status: O

FYI, I have just released a new version (2001.01.09) of the "LIVE.COM 
Streaming Media" source code that fixes a bug noted by Roland Paviainen: 
The previous versions were incorrectly including (& checking for) a 4-byte 
MPEG audio header at the beginning of the RTP payload, even if the stream 
was not a RFC2250 stream.

         Ross.




From <EMAIL-ADDRESS-HIDDEN> Fri May 18 03:07:51 2001
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Subject: [Live-devel] MPEG-1 video
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Hello,

has anybody used this library to packetize MPEG-1 video?

Thanks

	Francisco J. Cabello



From <EMAIL-ADDRESS-HIDDEN> Tue Jun 19 02:48:33 2001
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Date: Tue, 19 Jun 2001 02:47:23 -0700
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Status: O

At 12:54 AM 6/19/01, you wrote:
>hi,
>
>to build live under OpenBSD I needed to add the following to
>config.openbsd:
>
>         .SUFFIXES: .cpp

Jakob (cc. "live-devel"),

Thanks for the 'heads up'.  I'll add that line to "config.openbsd" in 
future releases.


>perhaps this is needed on more platforms?

Possibly, although I haven't heard any other report that it's needed.

Thanks again for the report.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Mon Jun 25 23:12:36 2001
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Subject: [Live-devel] New support for MPEG video streaming
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Status: O

The "LIVE.COM Streaming Media" libraries <http://live.sourceforge.net/> now 
includes support for MPEG (1 or2) video streaming, in addition to MPEG 
audio (e.g., MP3) streaming.  The MPEG streaming implementation conforms to 
the RTP payload format described in RFC 2250.

There are also three new test programs: "testMPEGVideoStreamer", 
"testMPEGVideoReceiver", and "testMPEGSplitter".  (The latter program 
splits a MPEG Program Stream into its consitituent audio and video 
Elementary Streams.)

I have also renamed the MP3 audio test programs, to be consistent with the 
names of the video test programs.  The audio test programs are now called 
"testMP3Streamer" and "testMP3Receiver".

	Ross.



From <EMAIL-ADDRESS-HIDDEN> Mon Jun 25 23:33:11 2001
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Status: O

I'm curious as to how many of you on this mailing list are using (or 
planning to use) the "LIVE.COM Streaming Media" libraries 
<http://live.sourceforge.net/> in your own application(s).  If you're 
willing to talk about your plans for using (and/or adding to) the 
libraries, feel free to reply (either to me, or to the mailing list).

         Ross.




From <EMAIL-ADDRESS-HIDDEN> Tue Jun 26 03:10:23 2001
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 .fr>
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Status: O

At 01:21 AM 6/26/01, you wrote:
>There isn't any file for
>downloading in the web
>
>         http://sourceforge.net/projects/live
>
>         only the notes and the release name (2001.06.25)

There seems to be a problem at SourceForge.net; the release file (.tar.gz) 
is not showing up on their web site.

Until they fix this problem is fixed, you can get the latest version of the 
file from
         http://www.live.com/liveMedia/

     Ross.



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Status: O

---------------------- Forwarded by Anna Harsha Vardhan/India/VJIL on
01/21/2001 05:00 PM ---------------------------


live-devel-request(at)lists.sourceforge.net(at)lists.sourceforge.net on
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From <EMAIL-ADDRESS-HIDDEN> Thu Jun 28 02:57:56 2001
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Status: O

At 08:59 AM 6/27/01, de la Fuente Conde, Jose Antonio wrote:

>         Hi!
>         It's me another time. I'm traying your library. It seems that it
>works fine, congratulation. I have some questions to do to you.

Jose,

I'm answering your questions on the developers' mailing list, as they are 
likely to be of general interest.  (Follow-up discussion should probably 
take place on the mailing list also.)

>         1. testMPEGVideoStreamer and testMPEGVideoReceiver takes an
>test.mpg file, then it's converting in an Elementary Stream file and send.
>What's happening with the audio? aren't there Audio Elementary Stream? For
>streaming video and audio is it necesary to use firt testSpliter and then
>testMP3Streamer and testMPEGVideoStreamer?

"testMPEGVideoStreamer" - as written - assumes that the input file 
("test.mpg") is a MPEG Video Elementary Stream - i.e., without any 
audio.  If, however, the input file is a MPEG *Program* stream - i.e., 
containing multiplexed audio and video - then you would need to use a 
demultiplexor to extract the video stream from this before streaming 
it.  In "testMPEGVideoStreamer.cpp" I show how you could do this (look for 
"#ifdef SOURCE_IS_PROGRAM_STREAM").

What these test programs don't do, however, is extract and stream *both* 
audio and video streams from a Program Stream.  To do this, you would need 
to write a program that combined "testMPEGSplitter", 
"testMPEGVideoStreamer", and "testMP3Streamer".  I.e., the program would 
need to first use a "MPEGDemux" to break the Program Stream up into its 
constituent audio and video Elementary Streams, and then use:
         - a "MPEGVideoStreamFramer" plus a "MPEGVideoRTPSink"
           for each video stream, and
         - a "MPEGAudioStreamFramer" (to be written) plus a
           "MPEGAudioRTPSink" for each audio stream
Someday I may write a test program like this myself, and add it to the test 
suite.  For now, though, I'll leave it as an exercise for others :-)


>         2. How do you probe it? (the video streamer) What player are you
>using? I have try Mplayer, but it can not read form the stdin (I
>think)(The testMPEGVideoReceiver give me the receiver paquets to the
>stdout) and also with xine (it can read, but it doesn't go very well).
>Now, i'm doing it like that
>         testMPEGVideoReceiver > aux.mpg
>         mplayer aux.mpg (when the transmission start)

The best way to receive and play the stream is to use a tool that can 
handle MPEG video/RTP directly, without having to use 
"testMPEGVideoReceiver" at all.  The tool that I prefer is Apple's 
QuickTime 5 Player (which has a version for Windows as well as MacOS).  To 
view the stream using QuickTime 5 Player, just open the file 
"testMPEGVideo.sdp" (that's included in the "testProgs" directory).  You 
can get QuickTime from <http://quicktime.apple.com/>

If you're using a system that doesn't run QuickTime 5 Player, then one 
possible player is "mim": <http://videolab.uoregon.edu/mim/>

As a last resort, you can run "testMPEGVideoReceiver", and use a regular 
MPEG player (e.g., Windows Media Player) to read from 
"testMPEGVideoReceiver's" built-in web server.  To do this, open the URL
<http://localhost:10006/dummy.mpg>.  (Of course, if "testMPEGVideoReceiver" 
is running on a different machine than the MPEG player tool, then you'll 
need to use its host name, rather than "localhost".)


>         3. I have seen that for using MPEG TS or PS, it's neccesary to
>change a variable (SOURCE_IS_PROGRAM_STREAM). Does can it make
>automatically?

You could probably do this by writing a new "FramedFilter" that would 
inspect the MPEG stream, and automatically create and insert appropriate 
new filters, depending on the type of MPEG data that it saw.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Thu Jun 28 12:19:20 2001
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Date: 28 Jun 2001 13:19:15 MDT
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Status: O

Hello Sir/Mamdam
  We are from india having a web development team.
We would like to telecast a church feast live on =

the web please inform me some particulars to do it.
Hope u will do the best for us and we would be
grateful to u.

Regards,
Janarth.S
Web Master
(Spiderboys)



From <EMAIL-ADDRESS-HIDDEN> Thu Jun 28 13:13:46 2001
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Status: O

At 12:19 PM 6/28/01, Spider Boys wrote:
>Hello Sir/Mamdam
>   We are from india having a web development team.
>We would like to telecast a church feast live on
>the web please inform me some particulars to do it.
>Hope u will do the best for us and we would be
>grateful to u.

Janarth,

You are sending your message to the wrong mailing list.  This mailing list 
is for technical discussion related to the development of certain software 
libraries for streaming media.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Thu Jul 12 07:57:49 2001
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Status: O

At 05:11 AM 7/12/01, Francisco Javier Cabello wrote:
>How do you connect QT with your test programs?
>have you implemented a RTSP server?
>
>I don't know the way to tell QT to receive RTP/MPEG-1 video with sending RTSP
>commands.


Yes, Apple's QuickTime Player version 5.0 (*not* any earlier version) can 
be used to receive and play the video streamed by the 
"testMPEGVideoStreamer" tool (or the audio streamed by the 
"testMP3Streamer" tool).  However, you can't (yet) use RTSP to do this, as 
the streaming server code doesn't yet include an implementation of a RTSP 
server.  Instead, have QuickTime Player open the appropriate ".sdp" file 
(that you'll find in the "testProgs" directory).

I.e., to receive/play MPEG video streamed by "testMPEGVideoStreamer", have 
QuickTime Player open the file "testMPEGVideo.sdp".  (Note that QuickTime 
Player currently only handles MPEG-1, even though "testMPEGVideoStreamer" 
is also capable of streaming MPEG-2.)  To receive/play MP3 audio streamed 
by "testMP3Streamer", have QuickTime Player open the file 
"testMP3.sdp".  (RealNetworks' RealPlayer will also understand this file, 
although it currently can't play streamed MPEG video.)

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Thu Jul 12 08:14:19 2001
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Status: O

hi all,

We have implemented a RTSP client and Test the control flow alone.
Is there any server available on the net for testing the control and the 
out of  band data
flow coming over RTP/AVP/UDP transport.

Thanx.
Palani.



From <EMAIL-ADDRESS-HIDDEN> Thu Jul 12 14:41:15 2001
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I've had a report that the Windows Makefiles (produced by running 
"genWindowsMakefiles") do not work with Microsoft Visual C++ version 
6.0.  (So far I've personally used them only with VC version 5.0.)

Has anyone gotten these Makefiles to work with VC version 6.0?  Is it just 
a matter of changing
         !IF "$(MSVC_VERSION)" == "5.0"
in "win32config" to
         !IF "$(MSVC_VERSION)" == "6.0"
??

         Ross (who doesn't want to upgrade his own VC to version 6.0, 
because he's tired of giving Bill Gates money)




From <EMAIL-ADDRESS-HIDDEN> Fri Jul 20 10:34:27 2001
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FYI, Francisco Cabello noted that "gcc" version 2.96 generated many warning 
messages when compiling the "LIVE.COM Streaming Media" libraries 
(complaining about things like tags at the end of "#endif"s), so I've just 
installed a new version of the code to fix this problem.

I've also updated the RTCP implementation (in "liveMedia") to include 
complete reception statistics in outgoing "RR" RTCP packets.

	Ross.



From <EMAIL-ADDRESS-HIDDEN> Mon Jul 23 18:39:11 2001
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Subject: [Live-devel] Questions from Francisco Javier
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At 03:00 PM 7/23/01, Francisco Javier Cabello Torres wrote:
>I'm thinking about use your code in my current project. I have my own
>implementation of RTP, but I haven't implemented RTCP, and I won't have 
>time to update code as the same way as live.com will be.

Yes, this is a good idea.  You'll be saving a lot of work.  Plus, you'll 
get a RTCP implementation 'for free'.

>I would like, if it's possible, to evaluate the cost of this change, because
>of that I'll ask you some questions.
>how difficult is to connect your library to a video server that capture
>frames from a video board (Bttv) and send it by RTP?

This should be straightforward - you need only write your own subclass of 
"FramedSource" that implements "doGetNextFrame()", by scheduling the 
capture of a frame from your video board.  For an illustration of how to do 
this, see "MPEGVideoDeviceSource.{hh,cpp}", which is the outline for a 
class that encapsulates a video (e.g.) device.  (Actually, "MPEG" and 
"Video" should not really have been part of the name, because this class 
structure could be used for any device source.)

For example, your class's implementation of "doGetNextFrame()" could set up 
the device with a call-back function to be invoked when the next frame of 
data is available.  This call-back function would call ->deliverFrame(), 
and then deliverFrame() would do the actual data delivery (to the class 
member vars).  Note that you want the call-back function to be executed 
*only* after a previous call to "doGetNextFrame()".  I.e., for every call 
to "doGetNextFrame()", there should (later) be exactly one call to your 
call-back function.

Your "deliverFrame()" member function would do the actual delivery of data 
to the invoking reader.  E.g., suppose that your call-back function tells 
you that it has a frame of length  "len" bytes, at address "addr".  Then, 
in deliverFrame(), you would do something like:

         if (len > fMaxSize) len = fMaxSize;
         memmove(fTo, addr, len);
         fFrameSize = len;
         fPresentationTime = ... ; // this frame's timestamp (seconds, 
useconds)

(and then call "scheduleDelayedTask" (with a 'delay' of zero, as I note in 
"MPEGVideoDeviceSource.cpp")

Note that you may need to add parameters to deliverFrame() to pass the 
"len" and "addr" parameters from the call-back function.

To write a server that uses your device as a source, I suggest that you use 
"testMPEGVideoStreamer.cpp" as a model.  You would replace the call to 
"ByteStreamFileSource::createNew()" with a similar function to instantiate 
your new 'video device' class.

You will probably also need to implement your own "RTP sink" class, as a 
subclass of "MultiFramedRTPSink" (unless, of course, you are planning to 
stream MPEG-1 or 2, in which case you can use the "MPEGVideoRTPSink" class 
that already exists.  Note that "MultiFramedRTPSink" already does most of 
the work for you; implementing a new RTP payload format is just a matter of 
redefining some virtual functions in your subclass.

>Now, I have functions to init RTP session, and other to send one frame. When
>I send a frame I specify the timestamp, because of that RTP library won't
>need to compute it, only will have to code this timestamp in all packets from
>the same frame.

Note that "MultiFramedRTPSink" (and thus, any class that you derive from 
this) automatically takes care of this.  If a frame gets fragmented over 
more than one RTP packet, then the same timestamp will be used for each 
packet.  (Also, if more than one frame appears in a packet, then (by 
default) the first frame's timestamp will be the one that's used in the 
outgoing RTP packet.)  See the implementation "MultiFramedRTPSink", and its 
two existing subclasses "MPEGVideoRTPSink" and "MPEGAudioRTPSink".

>Other problems are 'sleeps'. When you read from a file, you have to introduce
>'waits' for synchronize server and client, but when you are capturing from a
>video board, the synchronize goes from the video board. Do you understand
>what I mean? Do you thing will be problems with this?

No, that shouldn't be a problem.  The implementation of 
"MultiFramedRTPSink" ensures that outgoing packets are not sent any faster 
than whatever rate is natural for the codec.  If a live video capture 
device is used as the data source, then this test will usually be 
unnecessary, but shouldn't cause any problems.

         Ross.



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Status: O

At 04:44 PM 7/26/01, Joe Chen wrote:
>   I'm trying out the library / test programs on Linux (Redhat 7.1).
>Compilation had few warnings in, but was successful.
>
>   I tried running testMP3Streamer on Linux, and was able to play it on
>RealPlayer 8 on Win 2000, when using the .sdp file.
>
>   But running QT 5 on Win 2000 resulted in no sound, although I
>believe it was streaming - It said "buffering....%" and than "Live
>Broadcast."   Nothing was in the display area.

I'm not sure if QuickTime Player supports multicast MP3 audio properly.  I, 
too, have had inconsistent results when trying to play this.

For playing multicast MP3 streams, I suggest running FreeAmp 
<http://www.freeamp.org/>, or Winamp <http://www.winamp.com/> using 
the  "in_rtp.dll" MP3/RTP plugin: 
<http://www.live.com/multikit/winamp-plugin.html>.   (BTW, this plugin is 
implemented using the "LIVE.COM Streaming Media" libraries, using code 
similar to "testMP3Receiver".)  In each case, you receive the MP3/RTP 
stream by telling the player to open the URL
         rtp://<multicast-address>:<port>
E.g., to receive the multicast stream output by "testMP3Streamer", give the 
player (Freeamp or Winamp+in_rtp.dll) the URL:
         rtp://239.255.42.42:8888

>   I tried running testMPEGVideoStreamer on Linux.  First with
>MPEG2 file.   I get continous message "MPEGVideoStreamParser::
>parseSlice(): Saw unexpected code 0x00000xxx" where the hex is
>rotating between 1ba, 1e0, 1ba, 1c0...  These files played fine when
>loaded directly into any Win 2000 player.

OK, the problem here is that "testMPEGVideoStreamer" - by default - assumes 
that its input file is a MPEG Video *Elementary* Stream - i.e., without any 
audio.  Your file was a MPEG *Program* stream - i.e., containing 
multiplexed video *and* audio.  To stream video from such a stream, you 
need to use a demultiplexor to extract the video stream from this before 
streaming it.  In "testMPEGVideoStreamer.cpp" I show how you could do this 
(look for "#ifdef SOURCE_IS_PROGRAM_STREAM").  If you uncomment the line
         //#define SOURCE_IS_PROGRAM_STREAM 1
and recompile the program, you should be able to stream the video.

Note, though, that QuickTime Player 5 won't be able to receive/play this 
stream, because it supports only MPEG-1 video streams - not MPEG-2.

I don't know of any MPEG-*2* player that handles RTP-streamed data 
directly.  (It's possible that Cisco's IP/TV viewer handles this, but I'm 
not sure.)  If you have a MPEG-2 player tool that can read its data from 
'stdin', or from a HTTP server, then you can use the "playRTPMPEG" tool 
(with the "-d" flag) to receive the MPEG-2/RTP data.  See 
<http://www.live.com/multikit/playRTPMPEG.html> for details.  (Note that 
"playRTPMPEG"  is also implemented using the "LIVE.COM Streaming Media" 
libraries.)

(For more information about streaming/receiving MPEG video, see the 
archived email message at 
<http://www.geocrawler.com/archives/3/6637/2001/6/0/6061946/>)


>   I tried feeding it a MPEG-1 file (I think).  No complaints from the
>testMPEGVideoStreamer program.  But no picture either on QT 5, as
>this was the same with MPEG-2 and MP3 feeds.   QT5 tried to buffer
>and than display "Live Broadcast" in all cases...

That's strange.  Because you got no complaints from 
"testMPEGVideoStreamer", this suggests that your MPEG-1 file was (unlike 
the MPEG-2 file that you tried earlier) a proper Video Elementary 
Stream.  So it's strange that QuickTime Player 5 didn't display the 
stream.  Perhaps your file was really a MPEG-*2* Video Elementary Stream???

I hope this helps.  Please let us know how things work out.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Sun Jul 29 14:35:07 2001
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Date: Sun, 29 Jul 2001 14:34:28 -0700
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Status: O

At 12:37 PM 7/29/01, Christian Cier-Zniewski wrote:
>I successfully compiled your libs under MSVC++ 6.0 with SP5.
>I only had to change the TOOLS32 variable to point to my VS
>installation.
>
>After starting ie. testmp3streamer or testmpegvideostreamer I get the
>following error message: (so does testmp3receiver or
>testmpegvideoreceiver as well)
>
>21:05:49 Groupsock(108: 239.255.42.42, 8888, 7)
>t(IP_ADD_MEMBERSHIP) error: Unknown error
>Unable to determine our source address: setsock
>nown error

Christian,

Other people have also reported problems like this if the code is built 
using MSVC++ version 6.0.  (MSVC+ version 5.0 doesn't have any 
problems.)  The problem seems to be that - for some reason - with MSC++ 
version 6.0 the code is being linked with the "ws2_32.lib" library; 
instead, it should be linked with the "wsock32.lib" library.

I'd like to try to track this problem down, but unfortunately I don't have 
MSVC++ version 6.0 handy.  So could you please mail me a copy of the file 
"ntwin32.mak" (this is the file that gets included by each of our 
Makefiles).  You should find this file in your MSVC++ "include" directory.

If you find that "ntwin32.mak" consists solely of another !include line, e.g.
         !include <win32.mak>
then could you also mail me a copy of *that* included file.  Thanks.

(I have cc'd this message to the developers' mailing list, in case someone 
else has already worked out a fix.)

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Sun Jul 29 22:47:26 2001
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Status: O

FYI,

At 07:03 PM 7/29/01, Christian Cier-Zniewski wrote:
>What is the difference between wsock32.lib and w2_32.lib? Is it
>Winsock1 and Winsock2?

Yes, apparently "wsock32.lib" is winsock1, and "ws2_32.lib" is 
winsock2.  However, for some reason (still unknown to me), the multicast 
code in the libraries seems to work only when linked with 
"wsock32.lib".  I'm not sure why this is - perhaps there are some 
incompatibilities with winsock2's implementation of the standard Berkeley 
socket calls??

Anyway, you should be able to fix this problem by changing
         TARGETOS=BOTH
in each of the Makefiles to
         TARGETOS=WIN95
(If you created your makefiles by running "genWindowsMakefiles" on a Unix 
system, then you can make this change in the "win32config" file, and then 
re-running "genWindowsMakefiles".)

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Mon Aug 06 15:56:50 2001
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Date: Mon, 6 Aug 2001 17:57:48 -0500 (CDT)
From: Omar Antonio Ponce Carrillo <EMAIL-ADDRESS-HIDDEN>
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Subject: [Live-devel] LIVE COMPILING ERROR (PLEASE HELP!!!)
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Status: O

We are planning some serious develpment using this libraries but we 
cannot ever start!

the problem maybe has a quick answer, if you can help us please let us know!

we want to compile it on solaris 2.6 after we did the ./genMakefiles solaris

we run the make and got:

cadaplic:TU JEFA >make
cd liveMedia; make
make: Fatal error: Don't know how to make target `Media.o'
Current working directory /develop/users/ua012771/desarrollo/live/liveMedia
*** Error code 1
make: Fatal error: Command failed for target `liveMedia/libliveMedia.a'
acadaplic:TU JEFA >

why is that? are we missing some libraries? (*.a and/or *.o) if so where 
can we get them?

if somebody knows why we are having this  please let us know!! we are 
getting crazy!

					thanks in advance
					Omar POnce Carrillo

					ua012771(at)mail.udlap.mx
					tujefa80(at)yahoo.com.mx





From <EMAIL-ADDRESS-HIDDEN> Mon Aug 06 16:20:07 2001
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Status: O

At 03:57 PM 8/6/01, Omar Antonio Ponce Carrillo wrote:
>We are planning some serious develpment using this libraries but we
>cannot ever start!
>
>the problem maybe has a quick answer, if you can help us please let us know!
>
>we want to compile it on solaris 2.6 after we did the ./genMakefiles solaris
>
>we run the make and got:
>
>cadaplic:TU JEFA >make
>cd liveMedia; make
>make: Fatal error: Don't know how to make target `Media.o'

Omar,

I'm not sure what might be going wrong here; the libraries compile OK on a 
Solaris guest account that I'm using (this is version "SunOS 5.8  Generic 
February 2000").  (I'm not sure how this version relates to "Solaris 2.6".)

In your "liveMedia" directory, are the files "Media.hh" and "Media.cpp" 
present?  (They should be.)  Do you get the same error if you try building 
"Media.o" directly - i.e., by running
         make Media.o
?

It sounds like there's something different about your environment.  If so, 
you may end up having to make small changes to your "config.solaris" file, 
and then rerun "genMakefiles solaris".  If you have any other C++ code that 
compiles OK on your system, you might want to take a look at *its* 
Makefile, to see what might be different.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Tue Aug 07 03:43:35 2001
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Subject: [Live-devel] RTP/MPEG problem
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Status: O

Hi,

I think there is a problem with the field 'picture type' when live libraries 
code it.
My test was:
*QT as client.
*testMPEGVideoStreamer  as video server.

With QT I open a SDP file, and QT wait to receive video from specefied ports.
After, I run 'testMPEGVideoStreamer' and all things seems to be ok. QT draws 
video sent by testMPEGVideoStreamer, but I you take a network analyzer 
(sniffer) and look for rtp packets you'll see errors.

I use ethereal sniffer to do this work.  If you see 'picture type' field, 
you'll see errors. The first packets is coded with 0 value (forbidden). A lot 
of packets after, this value change to 2 (P frame).
I suppose that the problems was when this value should be 1 (I frame), 
however live code as 0.

	Bye.








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Status: O

At 03:40 AM 8/7/01, Francisco Javier Cabello wrote:
>I think there is a problem with the field 'picture type' when live libraries
>code it.
>My test was:
>*QT as client.
>*testMPEGVideoStreamer  as video server.
>
>With QT I open a SDP file, and QT wait to receive video from specefied ports.
>After, I run 'testMPEGVideoStreamer' and all things seems to be ok. QT draws
>video sent by testMPEGVideoStreamer, but I you take a network analyzer
>(sniffer) and look for rtp packets you'll see errors.
>
>I use ethereal sniffer to do this work.

Is this sniffer looking at the 4-byte "video-specific header" (as defined 
in RFC2250) in the RTP packet?

>   If you see 'picture type' field,
>you'll see errors. The first packets is coded with 0 value (forbidden). A lot
>of packets after, this value change to 2 (P frame).
>I suppose that the problems was when this value should be 1 (I frame),
>however live code as 0.

It's strange that your sniffer should be reporting problems in the RTP 
packets, yet QuickTime Player seems to receive and play them OK.  This 
suggests that perhaps your sniffer is looking at the wrong field in the RTP 
packet??

Anyway, could you send me the URL for this file (if it's on a web server), 
or else send me a < 1 Mbyte portion that illustrates the problem.  Thanks.

         Ross.



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From: Omar Antonio Ponce Carrillo <EMAIL-ADDRESS-HIDDEN>
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Subject: [Live-devel] Re: LIVE COMPILING ERROR (PLEASE HELP!!!) (fwd)
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Status: O

Hello again Ross.. i did investigate what was happening and i did notice 
that the make installed on the machine i log  was not accurate... i 
downloaded the GNU make and i had no problems with the compiling!

By the way do you have any kind of tutorial or guide to start from?

				thanks regards O.

---------- Forwarded message ----------
Date: Mon, 06 Aug 2001 16:19:40 -0700
From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
To: Omar Antonio Ponce Carrillo <EMAIL-ADDRESS-HIDDEN>
Cc: <EMAIL-ADDRESS-HIDDEN>
Subject: [Live-devel] Re: LIVE COMPILING ERROR (PLEASE HELP!!!)

At 03:57 PM 8/6/01, Omar Antonio Ponce Carrillo wrote:
>We are planning some serious develpment using this libraries but we
>cannot ever start!
>
>the problem maybe has a quick answer, if you can help us please let us know!
>
>we want to compile it on solaris 2.6 after we did the ./genMakefiles solaris
>
>we run the make and got:
>
>cadaplic:TU JEFA >make
>cd liveMedia; make
>make: Fatal error: Don't know how to make target `Media.o'

Omar,

I'm not sure what might be going wrong here; the libraries compile OK on a 
Solaris guest account that I'm using (this is version "SunOS 5.8  Generic 
February 2000").  (I'm not sure how this version relates to "Solaris 2.6".)

In your "liveMedia" directory, are the files "Media.hh" and "Media.cpp" 
present?  (They should be.)  Do you get the same error if you try building 
"Media.o" directly - i.e., by running
         make Media.o
?

It sounds like there's something different about your environment.  If so, 
you may end up having to make small changes to your "config.solaris" file, 
and then rerun "genMakefiles solaris".  If you have any other C++ code that 
compiles OK on your system, you might want to take a look at *its* 
Makefile, to see what might be different.

         Ross.


_______________________________________________
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Status: O

At 11:12 AM 8/7/01, Omar Antonio Ponce Carrillo wrote:

>Hello again Ross.. i did investigate what was happening and i did notice
>that the make installed on the machine i log  was not accurate... i
>downloaded the GNU make and i had no problems with the compiling!

OK, that's good to hear.

>By the way do you have any kind of tutorial or guide to start from?

No, not right now.  The only documentation right now is
         - the web page at <http://live.sourceforge.net/>
         - the code itself
         - the test programs in the "testProgs" directory, which provide
            examples of how to use the libraries
Plus, of course, there's this mailing list.  If you have any questions 
about how to use the libraries, feel free to post them here.

         Ross.



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Subject: [Live-devel] where do you print it???
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Status: O

Hello Ross.. ive been looking through the code for the testMP3Streamer  
and the testMP3Receiver and i wonder where do you print the mp3 you are 
recieving and how you do that as well?


					thanks O. 



From <EMAIL-ADDRESS-HIDDEN> Wed Aug 08 11:44:13 2001
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Subject: Re: [Live-devel] where do you print it???
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Status: O

At 11:19 AM 8/8/01, Omar Antonio Ponce Carrillo wrote:

>Hello Ross.. ive been looking through the code for the testMP3Streamer
>and the testMP3Receiver and i wonder where do you print the mp3 you are
>recieving and how you do that as well?

Omar,

Please note the "testProgs" section of <http://live.sourceforge.net/>  for 
a description of each of the test programs.

To summarize:

- "testMP3Receiver" outputs its received MP3 data to "stdout" (but note 
that you can easily change the code to output to an actual file, by 
replacing "stdout" with the file name (in "testMP3Receiver.cpp")).

- "testMPEGVideoReceiver" sets up an internal HTTP server (on port 10006) 
to serve the MPEG video data that it receives.  (You can access this data 
by pointing a MPEG video player tool at "http://localhost:10006")

If you prefer, you can easily change "testMPEGVideoReceiver" to output to 
"stdout" (or a file) instead.  See ""testMPEGVideoReceiver.cpp" for details.

         Ross.



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Date: Thu, 9 Aug 2001 11:24:04 -0500 (CDT)
From: Omar Antonio Ponce Carrillo <EMAIL-ADDRESS-HIDDEN>
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Subject: [Live-devel] TOO BIG FOR IT?
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Status: O

Hello there, ive been streaming mpegs lower than 1 meg with no problem 
but when i tried to stream and store a 40 megs file it writes the file 
with 0 size.

Is there a limit for transmiting mpegs? whats happening?

					regards O. 

it cant be the version (1 or 2 right?) cause the JMF plays the original 
file and it only accepts 1 type.



From <EMAIL-ADDRESS-HIDDEN> Thu Aug 09 09:38:14 2001
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Status: O

At 09:24 AM 8/9/01, Omar Antonio Ponce Carrillo wrote:

>Hello there, ive been streaming mpegs lower than 1 meg with no problem
>but when i tried to stream and store a 40 megs file it writes the file
>with 0 size.
>
>Is there a limit for transmiting mpegs? whats happening?

The *size* of an MPEG input file shouldn't be a problem - but perhaps there 
are some other attributes of the stream that are causing problems??

If you try to stream just a small portion of the original file (i.e., chop 
off < 1 MByte of the original file, and try steaming that), does that work OK?

Could you put the problem file on a web server and send me the URL, so I 
could take a look at it?

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Thu Aug 09 10:25:43 2001
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Date: Thu, 9 Aug 2001 12:26:52 -0500 (CDT)
From: Omar Antonio Ponce Carrillo <EMAIL-ADDRESS-HIDDEN>
To: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
cc: live-devel(at)lists.sourceforge.net
Subject: Re: [Live-devel] Re: TOO BIG FOR IT?
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Status: O

Here you have the file... its a music video... please chech it and tell 
me what could it be i was not able to chop it i dont have the necessary 
tools )=

http://mailweb.udlap.mx/~ua012771/bad.mpg


ill try with other big videos... cause we are planning to use this as a 
transmisor for really big files.

						regards O.

On Thu, 9 Aug 2001, Ross Finlayson wrote:

> At 09:24 AM 8/9/01, Omar Antonio Ponce Carrillo wrote:
> 
> >Hello there, ive been streaming mpegs lower than 1 meg with no problem
> >but when i tried to stream and store a 40 megs file it writes the file
> >with 0 size.
> >
> >Is there a limit for transmiting mpegs? whats happening?
> 
> The *size* of an MPEG input file shouldn't be a problem - but perhaps there 
> are some other attributes of the stream that are causing problems??
> 
> If you try to stream just a small portion of the original file (i.e., chop 
> off < 1 MByte of the original file, and try steaming that), does that work OK?
> 
> Could you put the problem file on a web server and send me the URL, so I 
> could take a look at it?
> 
>          Ross.
> 
> 
> _______________________________________________
> Live-devel mailing list
> Live-devel(at)lists.sourceforge.net
> http://lists.sourceforge.net/lists/listinfo/live-devel
> 


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Date: Thu, 9 Aug 2001 13:20:32 -0500 (CDT)
From: Omar Antonio Ponce Carrillo <EMAIL-ADDRESS-HIDDEN>
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Subject: [Live-devel] STREAMING PROBLEMS...
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Status: O

Hello again Ross, look:

There are other "normal" mpeg files :

http://mailweb.udlap.mx/~ua012771/Nike_90Fin.mpg
http://mailweb.udlap.mx/~ua012771/FVL.mpg

that when transmited i got the message:


MPEGVideoStreamParser::parseSlice(): Saw unexpected code 0x000001ba
MPEGVideoStreamParser::parseSlice(): Saw unexpected code 0x000001e0
MPEGVideoStreamParser::parseSlice(): Saw unexpected code 0x000001ba
MPEGVideoStreamParser::parseSlice(): Saw unexpected code 0x000001c0
MPEGVideoStreamParser::parseSlice(): Saw unexpected code 0x000001e0
MPEGVideoStreamParser::parseSlice(): Saw unexpected code 0x000001ba


during the trasmition, and when i open it if using real i saw nothing and 
using JMF it says theres a lot of bad data.

but when played the original file i got no problems... 

i wonder whats the streamer saying? does he found some bad code? but if 
so how can i saw it originally with no problems?


					regards O.

On Thu, 9 Aug 2001, Omar Antonio Ponce Carrillo wrote:

> 
> Here you have the file... its a music video... please chech it and tell 
> me what could it be i was not able to chop it i dont have the necessary 
> tools )=
> 
> http://mailweb.udlap.mx/~ua012771/bad.mpg
> 
> 
> ill try with other big videos... cause we are planning to use this as a 
> transmisor for really big files.
> 
> 						regards O.
> 
> On Thu, 9 Aug 2001, Ross Finlayson wrote:
> 
> > At 09:24 AM 8/9/01, Omar Antonio Ponce Carrillo wrote:
> > 
> > >Hello there, ive been streaming mpegs lower than 1 meg with no problem
> > >but when i tried to stream and store a 40 megs file it writes the file
> > >with 0 size.
> > >
> > >Is there a limit for transmiting mpegs? whats happening?
> > 
> > The *size* of an MPEG input file shouldn't be a problem - but perhaps there 
> > are some other attributes of the stream that are causing problems??
> > 
> > If you try to stream just a small portion of the original file (i.e., chop 
> > off < 1 MByte of the original file, and try steaming that), does that work OK?
> > 
> > Could you put the problem file on a web server and send me the URL, so I 
> > could take a look at it?
> > 
> >          Ross.
> > 
> > 
> > _______________________________________________
> > Live-devel mailing list
> > Live-devel(at)lists.sourceforge.net
> > http://lists.sourceforge.net/lists/listinfo/live-devel
> > 
> 
> _______________________________________________
> Live-devel mailing list
> Live-devel(at)lists.sourceforge.net
> http://lists.sourceforge.net/lists/listinfo/live-devel
> 


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Status: O

At 11:20 AM 8/9/01, Omar Antonio Ponce Carrillo wrote:
>There are other "normal" mpeg files :
>
>http://mailweb.udlap.mx/~ua012771/Nike_90Fin.mpg
>http://mailweb.udlap.mx/~ua012771/FVL.mpg
>
>that when transmited i got the message:
>
>
>MPEGVideoStreamParser::parseSlice(): Saw unexpected code 0x000001ba

[etc...]

Omar,

By default, the "testMPEGVideoStreamer" test program assumes that its input 
is a MPEG Video Elementary Stream.  The files that you tried to stream 
(producing the error messages) are MPEG *Program Stream* files. To stream 
video from files like this, uncomment the line
         //#define SOURCE_IS_PROGRAM_STREAM 1
in testMPEGVideoStreamer.cpp, and recompile.

         Ross.



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From: Omar Antonio Ponce Carrillo <EMAIL-ADDRESS-HIDDEN>
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Subject: [Live-devel] JUST MPEG VIDEO???
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Status: O

hello again!

i was looking to the streamed file obtained from both programs 
(testMPEGVideoReceiver.cpp and testMPEGVideoStreamer.cpp) and it seems to 
be all video with no sound. i suppose theres only exemplified the video 
sending nor the sound right?.

Is there any example showing both sendings to obtain an audio/video file?

if not, how can i implement the audio sending on the current program to 
have it?

					thanks O.




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Status: O

At 03:26 PM 8/9/01, Omar Antonio Ponce Carrillo wrote:
>i was looking to the streamed file obtained from both programs
>(testMPEGVideoReceiver.cpp and testMPEGVideoStreamer.cpp) and it seems to
>be all video with no sound. i suppose theres only exemplified the video
>sending nor the sound right?.
>
>Is there any example showing both sendings to obtain an audio/video file?
>
>if not, how can i implement the audio sending on the current program to
>have it?

Omar,

Last month I posted a message about this to the mailing list.  Here's a copy:

-----
What these test programs don't do, however, is extract and stream *both* 
audio and video streams from a Program Stream.  To do this, you would need 
to write a program that combined "testMPEGSplitter", 
"testMPEGVideoStreamer", and "testMP3Streamer".  I.e., the program would 
need to first use a "MPEGDemux" to break the Program Stream up into its 
constituent audio and video Elementary Streams, and then use:
         - a "MPEGVideoStreamFramer" plus a "MPEGVideoRTPSink"
           for each video stream, and
         - a "MPEGAudioStreamFramer" (to be written) plus a
           "MPEGAudioRTPSink" for each audio stream
Someday I may write a test program like this myself, and add it to the test 
suite.  For now, though, I'll leave it as an exercise for others :-)



From <EMAIL-ADDRESS-HIDDEN> Tue Aug 14 14:24:12 2001
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Status: O


hello again Ross! hope you are not tired of me already (=

look! im really interested on having a full mpeg transmited... im doing 
some tests due to my lack of knowledge about handling mpegs.

i tried to send just the audio part using the MPEGVideoStreamFramer and 
obviously i got a segmentation fault...can you recommend me some info 
sources  
regarding the way to frame the audio part the same way you did with 
the video or at least some specification about how to deal with this issue? 

is there a chance to send the audio part as an mp3 file or something? 
cause the audio obtained from the testMPEGSplitter is not mp3 format 
right? i tried to send the audio obtained and it clearly failed (funny 
cause if i rename the obtained file as an mp3 it plays with no problem)


						regards
						O.






From <EMAIL-ADDRESS-HIDDEN> Wed Aug 15 00:42:53 2001
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Status: O

At 02:25 PM 8/14/01, Omar Antonio Ponce Carrillo wrote:
>can you recommend me some info
>sources
>regarding the way to frame the audio part the same way you did with
>the video or at least some specification about how to deal with this issue?

Each MPEG audio frame begins with a 'sync' code, consisting of 12 bits all 
set to 1.  I.e., each such frame begins with a byte 0xFF, followed by a 
byte 0xF<something>.  You can break a MPEG audio stream into frames by 
looking for this this sequence of 12 1-bits.

>is there a chance to send the audio part as an mp3 file or something?
>cause the audio obtained from the testMPEGSplitter is not mp3 format
>right?

It might be.  "MP3" means "MPEG-1 or 2, layer III audio".  More likely, 
though, the audio that you extract from MPEG Program Streams will be layer 
I or layer II - i.e., not really MP3.

>  i tried to send the audio obtained and it clearly failed

That's because an audio stream that's read from a "MPEGDemux" hasn't yet 
been broken into separate frames.  That needs to be done before the audio 
can be sent in RTP packets.

>(funny
>cause if i rename the obtained file as an mp3 it plays with no problem)

Most 'MP3' players are also capable of playing MPEG layer I or layer II 
streams as well.

         Ross.



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Hello again Ross!

could you tell me the differences between a MPEG audio and video stream? 
do they only differ on the 'sync' code at the beginning of each frame?


						thanks in advance O.






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Status: O

At 08:58 AM 8/16/01, Omar Antonio Ponce Carrillo wrote:

>Hello again Ross!
>
>could you tell me the differences between a MPEG audio and video stream?
>do they only differ on the 'sync' code at the beginning of each frame?

No, there are considerable differences between MPEG audio and video 
elementary streams.

         Ross.



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------=_NextPart_000_0014_01C12AA6.ACCA6E90
Content-Type: text/plain;
	charset="iso-8859-1"
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Hi

I've being using jmf to transmit mpeg via RTP.  I'm trying exactly the =
same with ur live.com lib.  When I'm transmitting to jstudio (an app =
that can tx and rx mpeg via RTP)  using ur testprog examples the update =
rate of the mpeg is terrible. =20

My question is, am I doing something incorrect, does some settings need =
to be set, or are there some difference.

Thanks
  John.

------=_NextPart_000_0014_01C12AA6.ACCA6E90
Content-Type: text/html;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content=3D"text/html; charset=3Diso-8859-1" =
http-equiv=3DContent-Type>
<META content=3D"MSHTML 5.00.3315.2870" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Hi</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>I've being using jmf to transmit mpeg =
via=20
RTP.&nbsp; I'm trying exactly the same with ur live.com lib.&nbsp; When =
I'm=20
transmitting to jstudio (an app that can tx and rx mpeg via RTP)&nbsp; =
using ur=20
testprog examples the update rate of the mpeg is terrible.&nbsp; =
</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>My question is,&nbsp;am I doing =
something=20
incorrect, does some settings need to be set, or are there some=20
difference.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Thanks</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp; John.</FONT></DIV></BODY></HTML>

------=_NextPart_000_0014_01C12AA6.ACCA6E90--



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Subject: [Live-devel] again rtp question
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Hi

When I changed the address of the multicaster from 239.255.42.42:8080:7 =
(ur default) to any other the jstudio app could not connect. Why not?

Thanks
  John

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content=3D"text/html; charset=3Diso-8859-1" =
http-equiv=3DContent-Type>
<META content=3D"MSHTML 5.00.3315.2870" name=3DGENERATOR>
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<DIV><FONT face=3DArial size=3D2>Hi</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>When I&nbsp;changed the address of the =
multicaster=20
from 239.255.42.42:8080:7 (ur default) to any other the jstudio app =
could not=20
connect. Why not?</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Thanks</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp; John</FONT></DIV></BODY></HTML>

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From <EMAIL-ADDRESS-HIDDEN> Tue Aug 21 16:29:15 2001
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To: Willard <EMAIL-ADDRESS-HIDDEN>
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Subject: Re: [Live-devel] RTP with live to/from jmf?
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At 04:06 PM 8/21/01, Willard wrote:
>I've being using jmf to transmit mpeg via RTP.  I'm trying exactly the 
>same with ur live.com lib.  When I'm transmitting to jstudio (an app that 
>can tx and rx mpeg via RTP)  using ur testprog examples the update rate of 
>the mpeg is terrible.
>
>My question is, am I doing something incorrect, does some settings need to 
>be set, or are there some difference.

John,

Can you say some more about the problems you're seeing?  What kind of MPEG 
file are you transmitting (using "testMPEGVideoStreamer")?  Is it a Program 
Stream file, or a Video Elementary Stream file?  What is its bit rate?  Do 
you see any error/warning messages coming from "testMPEGVideoStreamer"?

"testMPEGVideoStreamer" transmits all frame data that it sees, at the 
stream's natural bit rate, so if you're seeing slow frame display rate in 
your receiving tool, then it could be for one of the following reasons:
1/ Something about the input file is confusing "testMPEGVideoStreamer", so 
that it's transmitting only some of the video data.  (However, if you're 
not seeing any error/warning messages from "testMPEGVideoStreamer", then 
this is unlikely.)
2/ Your receiving tool doesn't have enough horsepower to handle/display the 
incoming MPEG/RTP data.  (Have you successfully using this same tool to 
receive/play MPEG/RTP data streamed from another tool?)

To help figure out what the problem is, you could also try 
receiving/playing the stream using Apple's QuickTime 5.0 player (assuming 
that the stream is MPEG-1; QT player doesn't yet handle MPEG-2).  You could 
also try changing "testMPEGVideoReceiver" to output to a file (see 
"testMPEGVideoReceiver.cpp" for details on how to do this), and then use 
this to record the incoming stream to a file.  You can then try playing the 
received file later (to see if you still see a low fdisplayed rame rate).

         Ross.



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Status: O

At 04:24 PM 8/21/01, Willard wrote:
>Hi
>
>When I changed the address of the multicaster from 239.255.42.42:8080:7 
>(ur default) to any other the jstudio app could not connect. Why not?


How is the 'jstudio app' told the multicast address and port that it needs 
to use?  If this is done on the command line, then this should work 
OK.  If, however, it gets this information from the ".sdp" file 
("testMPEGVideo.sdp"), then you will need to change the multicast address 
and/or port in that file as well.

         Ross.

ps. the port that is used (by default) in the test programs is 8888, not 8080



From <EMAIL-ADDRESS-HIDDEN> Tue Aug 21 16:35:24 2001
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Subject: [Live-devel] java vs C
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Hello everybody... one general question..... should both implementations 
(rtp on java and C) work with the same speed factor?

would a C rtp implementation  be faster than a java one because it is 
compiled and not interpreted?

					thanks O.


From <EMAIL-ADDRESS-HIDDEN> Wed Aug 22 01:15:48 2001
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From: Simon Magnusson <EMAIL-ADDRESS-HIDDEN>
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Subject: [Live-devel] Using the libraries with Mpeg-4 ?
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Hi, me and a friend are planning on a master thieses project where we will
use RTP to stream mpeg4 to an iPaq with Windows CE. We have found 3:rd part
mpeg4-player for the iPaq that says it uses the open standard RTP protocols
for streaming. Our task is to build the server side, and now I'm wondering
if the LIVE libraries can be used for streaming mpeg-4??

Simon


From <EMAIL-ADDRESS-HIDDEN> Wed Aug 22 03:36:05 2001
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Status: O

At 01:12 AM 8/22/01, Simon Magnusson wrote:
>Hi, me and a friend are planning on a master thieses project where we will
>use RTP to stream mpeg4 to an iPaq with Windows CE. We have found 3:rd part
>mpeg4-player for the iPaq that says it uses the open standard RTP protocols
>for streaming. Our task is to build the server side, and now I'm wondering
>if the LIVE libraries can be used for streaming mpeg-4??

Simon,

Yes, with a little bit of work you should be able to use these libraries to 
stream MPEG-4 over RTP.  You should be able to do this by writing new 'RTP 
sink' classes for MPEG-4 audio and video, similar to the existing 
"MPEGAudioRTPSink" and "MPEGVideoRTPSink" classes that are defined in the 
"liveMedia" library.  (The names of these classes are a little misleading, 
because they implement only MPEG-1 or MPEG-2.)

I'm a little sceptical about your 3rd party's claim that their MPEG-4 
player tool supports "the open standard RTP protocols", because the RTP 
payload formats for MPEG-4 are still moving through the IETF, and have not 
been fully standardized just yet.  If possible, you should try to find out 
as much as you can about the details of whatever payload format that they 
claim to implement, to assist you with developing a server implementation.

         Ross




From <EMAIL-ADDRESS-HIDDEN> Wed Aug 22 05:38:28 2001
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Status: O

Hi again,
this is a part from the white papers of the packet-video system that we plan
to use.
PacketVideo has a complete system for packing, streaming from server and
playing mpeg4 on the PDA. And since the player is free we thought we'd use
that. The only thing I can get from this is that it uses RTP and RTSP. Is
RTP different depending on what is streamed?

"
2.4 IETF Streaming Protocols
The IETF is the Internet Engineering Task Force. This voluntary standards
body is dedicated to
making recommendations for how to communicate information over the IP
networks that are the
basis for the Internet. The IETF has a number of activities related to the
delivery of multimedia
information in a packet-based environment. PacketVideo currently uses IETF
recommended
methods based on RTP and RTSP for the delivery of video bitstreams with
synchronized audio
from a server to a terminal. The protocol stack for the delivery of
multimedia data is shown in
Figure 3.
This idea behind RTP, which stands for Real Time Protocol, is that certain
data needs to be
delivered from a server to a client in a real time manner. Multimedia data
such as synchronized
audio and video falls into this category. Guaranteed delivery transport
protocols, such as TCP
(Transport Control Protocol) add significant delay by retransmitting data
packets until they are
acknowledged as correctly received by a client. RTP is an application layer
component that
utilizes UDP (User Datagram Protocol) as a transport mechanism. UDP data is
not guaranteed to
arrive at a client, but is rather a "best-effort", connectionless protocol.
It is therefore suitable for
delivery data that must arrive without delay. RTP headers consist primarily
of sequence
numbers, timestamps, and payload type bits. RTP enables a client application
to monitor the loss
of packets, and to "re-order" those packets that arrive out of order at the
client.
RTP includes a sub-component known as RTCP, or Real Time Control Protocol.
RTCP is used to
control performance information between a server and a client. PacketVideo
technology uses
RTCP to send reports between the client and server to indicate information
such as the
percentage of RTP packet loss during a video session. This information is
crucial to managing
the quality and throughput of the video data from the server.
RTSP stands for Real Time Streaming Protocol. This is a session-oriented
protocol that is
transported over TCP between server and client. The purpose of RTSP is to
provide a language
for communicating standard video-on-demand requests. PV uses RTSP to control
the server and
allow tracking of the stream session status as a video is being served.
"


Simon


-----Original Message-----
From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
Sent: den 22 augusti 2001 12:35
To: Simon Magnusson
Cc: <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] Using the libraries with Mpeg-4 ?


At 01:12 AM 8/22/01, Simon Magnusson wrote:
>Hi, me and a friend are planning on a master thieses project where we will
>use RTP to stream mpeg4 to an iPaq with Windows CE. We have found 3:rd part
>mpeg4-player for the iPaq that says it uses the open standard RTP protocols
>for streaming. Our task is to build the server side, and now I'm wondering
>if the LIVE libraries can be used for streaming mpeg-4??

Simon,

Yes, with a little bit of work you should be able to use these libraries to 
stream MPEG-4 over RTP.  You should be able to do this by writing new 'RTP 
sink' classes for MPEG-4 audio and video, similar to the existing 
"MPEGAudioRTPSink" and "MPEGVideoRTPSink" classes that are defined in the 
"liveMedia" library.  (The names of these classes are a little misleading, 
because they implement only MPEG-1 or MPEG-2.)

I'm a little sceptical about your 3rd party's claim that their MPEG-4 
player tool supports "the open standard RTP protocols", because the RTP 
payload formats for MPEG-4 are still moving through the IETF, and have not 
been fully standardized just yet.  If possible, you should try to find out 
as much as you can about the details of whatever payload format that they 
claim to implement, to assist you with developing a server implementation.

         Ross



From <EMAIL-ADDRESS-HIDDEN> Wed Aug 22 06:14:55 2001
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Status: O

Hi, me and a friend are planning on a master thieses project where we will
use RTP to stream mpeg4 to an iPaq with Windows CE. We have found 3:rd part
mpeg4-player for the iPaq that says it uses the open standard RTP protocols
for streaming. Our task is to build the server side, and now I'm wondering
if the LIVE libraries can be used for streaming mpeg-4??

Simon


From <EMAIL-ADDRESS-HIDDEN> Wed Aug 22 06:24:09 2001
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From: Simon Magnusson <EMAIL-ADDRESS-HIDDEN>
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Subject: [Live-devel] Using the libraries with Mpeg-4 ?
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Status: O

Hi again,
this is a part from the white papers of the packet-video system that we plan
to use.
PacketVideo has a complete system for packing, streaming from server and
playing mpeg4 on the PDA. And since the player is free we thought we'd use
that. The only thing I can get from this is that it uses RTP and RTSP. Is
RTP different depending on what is streamed?

"
2.4 IETF Streaming Protocols
The IETF is the Internet Engineering Task Force. This voluntary standards
body is dedicated to
making recommendations for how to communicate information over the IP
networks that are the
basis for the Internet. The IETF has a number of activities related to the
delivery of multimedia
information in a packet-based environment. PacketVideo currently uses IETF
recommended
methods based on RTP and RTSP for the delivery of video bitstreams with
synchronized audio
from a server to a terminal. The protocol stack for the delivery of
multimedia data is shown in
Figure 3.
This idea behind RTP, which stands for Real Time Protocol, is that certain
data needs to be
delivered from a server to a client in a real time manner. Multimedia data
such as synchronized
audio and video falls into this category. Guaranteed delivery transport
protocols, such as TCP
(Transport Control Protocol) add significant delay by retransmitting data
packets until they are
acknowledged as correctly received by a client. RTP is an application layer
component that
utilizes UDP (User Datagram Protocol) as a transport mechanism. UDP data is
not guaranteed to
arrive at a client, but is rather a "best-effort", connectionless protocol.
It is therefore suitable for
delivery data that must arrive without delay. RTP headers consist primarily
of sequence
numbers, timestamps, and payload type bits. RTP enables a client application
to monitor the loss
of packets, and to "re-order" those packets that arrive out of order at the
client.
RTP includes a sub-component known as RTCP, or Real Time Control Protocol.
RTCP is used to
control performance information between a server and a client. PacketVideo
technology uses
RTCP to send reports between the client and server to indicate information
such as the
percentage of RTP packet loss during a video session. This information is
crucial to managing
the quality and throughput of the video data from the server.
RTSP stands for Real Time Streaming Protocol. This is a session-oriented
protocol that is
transported over TCP between server and client. The purpose of RTSP is to
provide a language
for communicating standard video-on-demand requests. PV uses RTSP to control
the server and
allow tracking of the stream session status as a video is being served.
"


Simon


-----Original Message-----
From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
Sent: den 22 augusti 2001 12:35
To: Simon Magnusson
Cc: <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] Using the libraries with Mpeg-4 ?


At 01:12 AM 8/22/01, Simon Magnusson wrote:
>Hi, me and a friend are planning on a master thieses project where we will
>use RTP to stream mpeg4 to an iPaq with Windows CE. We have found 3:rd part
>mpeg4-player for the iPaq that says it uses the open standard RTP protocols
>for streaming. Our task is to build the server side, and now I'm wondering
>if the LIVE libraries can be used for streaming mpeg-4??

Simon,

Yes, with a little bit of work you should be able to use these libraries to 
stream MPEG-4 over RTP.  You should be able to do this by writing new 'RTP 
sink' classes for MPEG-4 audio and video, similar to the existing 
"MPEGAudioRTPSink" and "MPEGVideoRTPSink" classes that are defined in the 
"liveMedia" library.  (The names of these classes are a little misleading, 
because they implement only MPEG-1 or MPEG-2.)

I'm a little sceptical about your 3rd party's claim that their MPEG-4 
player tool supports "the open standard RTP protocols", because the RTP 
payload formats for MPEG-4 are still moving through the IETF, and have not 
been fully standardized just yet.  If possible, you should try to find out 
as much as you can about the details of whatever payload format that they 
claim to implement, to assist you with developing a server implementation.

         Ross



From <EMAIL-ADDRESS-HIDDEN> Wed Aug 22 09:14:23 2001
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Status: O

At 05:27 AM 8/22/01, Simon Magnusson wrote:
>Hi again,
>this is a part from the white papers of the packet-video system that we plan
>to use.
>PacketVideo has a complete system for packing, streaming from server and
>playing mpeg4 on the PDA. And since the player is free we thought we'd use
>that. The only thing I can get from this is that it uses RTP and RTSP. Is
>RTP different depending on what is streamed?

Yes, sort of.  Because RTP is a packet-based protocol (as opposed to a 
stream-based protocol like TCP), there are rules for how each type of media 
is packetized into RTP packets.  In particular, there are rules - for each 
type of media - about whether or not multiple 'frames' can be packed into a 
single RTP packet, and/or whether/how large 'frames' can be fragmented over 
more than one RTP packet.  Depending on the media type, there may also be 
special, media-specific headers in each RTP packet (i.e., in addition to 
the common RTP header).

These rules - for each media type - are called a "RTP Payload Format" for 
that media type.  In the "LIVE.COM Streaming Media" libraries, a "RTP 
Payload Format" is implemented by subclassing "RTPSink" (in the case of a 
transmitter) or "RTPSource" (in the case of a receiver).

For example, The RTP payload formats for MPEG-1 and MPEG-2 (audio and 
video) are defined by RFC 2250 <ftp://ftp.isi.edu/in-notes/rfc2250.txt>, 
and the libraries implement this using the classes "MPEGVideoRTPSink", 
"MPEGAudioRTPSink", "MPEGVideoRTPSource", and "MPEGAudioRTPSource".

In the case of MPEG-4, there may end up being more than one packet format 
defined, depending on which features of MPEG-4 are being used.  However, my 
reading of the "PacketVideo" whitepaper that you quoted 
<http://www.packetvideo.com/pdf/pv_whitepaper.pdf>, as well as the 
"Wireless Multimedia Forum" technical framework document that it references 
<http://www.wmmforum.com/Documents/WMF_RTFD_v1.pdf>, suggests that the 
PacketVideo Player probably uses the RTP Payload Format defined in RFC 
3016: <ftp://ftp.isi.edu/in-notes/rfc3016.txt>

So, if you were to define subclasses of "RTPSink" that implement this 
payload format, you could probably build a server that could stream MPEG-4 
data that your PacketVideo Player could receive and play.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Wed Aug 22 09:35:12 2001
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From: john spurling <EMAIL-ADDRESS-HIDDEN>
To: Simon Magnusson <EMAIL-ADDRESS-HIDDEN>
cc: live-devel(at)lists.sourceforge.net
Subject: RE: [Live-devel] Using the libraries with Mpeg-4 ?
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Status: O

Simon,

you also might look at an open source mpeg-4 player written by some folks
at cisco. it's already been shown to be interoperable with a variety of
mpeg-4 implementations and is following the ietf standards closely. its
homepage is: http://mpeg4ip.sourceforge.net

cheers,
john spurling

--
"not only does the english language borrow from other languages, it
sometimes chases them through dark alleys, hits them over the head,
and goes through their pockets."  -- eddy






On Wed, 22 Aug 2001, Ross Finlayson wrote:

> At 05:27 AM 8/22/01, Simon Magnusson wrote:
> >Hi again,
> >this is a part from the white papers of the packet-video system that we plan
> >to use.
> >PacketVideo has a complete system for packing, streaming from server and
> >playing mpeg4 on the PDA. And since the player is free we thought we'd use
> >that. The only thing I can get from this is that it uses RTP and RTSP. Is
> >RTP different depending on what is streamed?
> 
> Yes, sort of.  Because RTP is a packet-based protocol (as opposed to a 
> stream-based protocol like TCP), there are rules for how each type of media 
> is packetized into RTP packets.  In particular, there are rules - for each 
> type of media - about whether or not multiple 'frames' can be packed into a 
> single RTP packet, and/or whether/how large 'frames' can be fragmented over 
> more than one RTP packet.  Depending on the media type, there may also be 
> special, media-specific headers in each RTP packet (i.e., in addition to 
> the common RTP header).
> 
> These rules - for each media type - are called a "RTP Payload Format" for 
> that media type.  In the "LIVE.COM Streaming Media" libraries, a "RTP 
> Payload Format" is implemented by subclassing "RTPSink" (in the case of a 
> transmitter) or "RTPSource" (in the case of a receiver).
> 
> For example, The RTP payload formats for MPEG-1 and MPEG-2 (audio and 
> video) are defined by RFC 2250 <ftp://ftp.isi.edu/in-notes/rfc2250.txt>, 
> and the libraries implement this using the classes "MPEGVideoRTPSink", 
> "MPEGAudioRTPSink", "MPEGVideoRTPSource", and "MPEGAudioRTPSource".
> 
> In the case of MPEG-4, there may end up being more than one packet format 
> defined, depending on which features of MPEG-4 are being used.  However, my 
> reading of the "PacketVideo" whitepaper that you quoted 
> <http://www.packetvideo.com/pdf/pv_whitepaper.pdf>, as well as the 
> "Wireless Multimedia Forum" technical framework document that it references 
> <http://www.wmmforum.com/Documents/WMF_RTFD_v1.pdf>, suggests that the 
> PacketVideo Player probably uses the RTP Payload Format defined in RFC 
> 3016: <ftp://ftp.isi.edu/in-notes/rfc3016.txt>
> 
> So, if you were to define subclasses of "RTPSink" that implement this 
> payload format, you could probably build a server that could stream MPEG-4 
> data that your PacketVideo Player could receive and play.
> 
>          Ross.
> 
> 
> _______________________________________________
> Live-devel mailing list
> Live-devel(at)lists.sourceforge.net
> http://lists.sourceforge.net/lists/listinfo/live-devel
> 



From <EMAIL-ADDRESS-HIDDEN> Wed Aug 22 11:57:56 2001
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Status: O

At 10:42 AM 8/22/01, you wrote:
>I modified testMPEGVideoReceiver to save a file transmitted by 
>testMPEGVideoTransmitter.  It takes about 150 seconds to transmit a 7 
>second file (308kb MPEG 1).  When I replay the saved mpeg, it replays at a 
>perfect rate.  Why is it taking so long to transmit the file??

I'm not sure.  It's possible that there's a bug in the library code (where 
it calculates how long to delay between sending each packet).  Could you 
send me a copy of the 308 kByte MPEG-1 file so I can take a look at 
it?  (Or even better, put it on a web server somewhere and send me the URL?)

>  How do I use Apple Quick Time 5.0 to receive RTP stream transmitted by 
> testMPEGVideoTransmitter??

(You mean "testMPEGVideoStreamer".)  To receive/play the stream using 
QuickTime Player, have it open the file "testMPEGVideo.sdp".  (Note that if 
you change the IP address or port that "testMPEGVideoStreamer" uses, you 
will also need to change the corresponding fields in the 
"testMPEGVideo.sdp" file.)

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Wed Aug 22 12:00:32 2001
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Status: O

At 10:50 AM 8/22/01, Willard wrote:
>Hi
>
>How do I set-up live such that it uses normal IP address instead of the 
>multicast address.  Or how do I setup the multicast address such that it 
>transmits to a single pc only??

I think the "testMPEGVideoStreamer" code will still work if you change the 
"multicastAddressStr" variable to be a unicast address (the address of the 
intended destination).

However, there's really no reason not to use multicast, unless you plan to 
stream over a network that doesn't support IP multicast routing.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Wed Aug 22 16:52:41 2001
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Subject: [Live-devel] Understanding Library
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Hello,
I downloaded this library in an attempt to use it to stream a bmpeg (bundled 
mpeg) program stream rather than using the existing code to separate out the 
audio and video elementary streams.

In order to do this I am looking at all the source code to see how they are 
demuxing the program stream.  Other than going through the code does anybody 
have like an api for all the classes and methods or is there a tool that 
will create one for me.

Also if anybody knows about using this RTP Payload and has any experience 
please let me know, it might be worth your while.

Thanks.

_________________________________________________________________
Get your FREE download of MSN Explorer at http://explorer.msn.com/intl.asp



From <EMAIL-ADDRESS-HIDDEN> Thu Aug 23 02:10:20 2001
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To: {Spike 12123} <EMAIL-ADDRESS-HIDDEN>
From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
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At 04:52 PM 8/22/01, Spike 12123 wrote:
>Hello,
>I downloaded this library in an attempt to use it to stream a bmpeg 
>(bundled mpeg) program stream rather than using the existing code to 
>separate out the audio and video elementary streams.

Are you interested using the RTP payload format for MPEG System (Program) 
streams, as defined by section 2 of RFC 2250, or are you interested in 
using the RTP payload format for 'Bundled MPEG', as defined in  the 
(Experimental) RFC 2343?

If it's the former, then implementing the RTP payload format using the 
library would be very easy, as there are no packetization 
restrictions.  You could do this by defining a very simple subclass of 
"MultiFramedRTPSink", and feeding this from a "ByteStreamFileSource" 
(assuming that the MPEG Program Stream input comes from a file).

If it's the latter, then implementing the RTP payload format should still 
be possible, but more difficult.  Again, you would need to define a 
subclass of "MultiFramedRTPSink", and feed this from both a video and an 
audio elementary stream.  These elementary streams could be generated by a 
pair of "MPEGDemuxedElementaryStream"s, produced by a "MPEGDemux".  (See 
the code for the "testMPEGSplitter" application (in "testProgs") for an 
illustration of how to do this.)

>In order to do this I am looking at all the source code to see how they 
>are demuxing the program stream.  Other than going through the code does 
>anybody have like an api for all the classes and methods or is there a 
>tool that will create one for me.

Right now the best description of the API is the header files themselves - 
in particular, the header files that are included by "liveMedia.hh".  Also, 
the code for the various test programs provides an illustration of how to 
use the libraries.

(Sometime in the future I plan to write up more detailed documentation of 
the library APIs, but I'm waiting for the code to stabilize more before I 
start doing this.)

         Ross.





From <EMAIL-ADDRESS-HIDDEN> Thu Aug 23 02:26:33 2001
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From: Simon Magnusson <EMAIL-ADDRESS-HIDDEN>
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Subject: RE: [Live-devel] Using the libraries with Mpeg-4 ?
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Status: O

Thanks for all the info! It really helped me understanding things.
I also looked at the Mpeg4IP at sourceforge, but since we don't have the
need to encode nor play the clips, this libraray seems more suiteble. And
besides that we are currently planning on using Windows.
Now I'm wondering if there is any functions for dealing with the RTSP in
this libraray?

Thanks
/Simon


-----Original Message-----
From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
Sent: den 22 augusti 2001 18:14
To: Simon Magnusson
Cc: <EMAIL-ADDRESS-HIDDEN>
Subject: RE: [Live-devel] Using the libraries with Mpeg-4 ?


At 05:27 AM 8/22/01, Simon Magnusson wrote:
>Hi again,
>this is a part from the white papers of the packet-video system that we
plan
>to use.
>PacketVideo has a complete system for packing, streaming from server and
>playing mpeg4 on the PDA. And since the player is free we thought we'd use
>that. The only thing I can get from this is that it uses RTP and RTSP. Is
>RTP different depending on what is streamed?

Yes, sort of.  Because RTP is a packet-based protocol (as opposed to a 
stream-based protocol like TCP), there are rules for how each type of media 
is packetized into RTP packets.  In particular, there are rules - for each 
type of media - about whether or not multiple 'frames' can be packed into a 
single RTP packet, and/or whether/how large 'frames' can be fragmented over 
more than one RTP packet.  Depending on the media type, there may also be 
special, media-specific headers in each RTP packet (i.e., in addition to 
the common RTP header).

These rules - for each media type - are called a "RTP Payload Format" for 
that media type.  In the "LIVE.COM Streaming Media" libraries, a "RTP 
Payload Format" is implemented by subclassing "RTPSink" (in the case of a 
transmitter) or "RTPSource" (in the case of a receiver).

For example, The RTP payload formats for MPEG-1 and MPEG-2 (audio and 
video) are defined by RFC 2250 <ftp://ftp.isi.edu/in-notes/rfc2250.txt>, 
and the libraries implement this using the classes "MPEGVideoRTPSink", 
"MPEGAudioRTPSink", "MPEGVideoRTPSource", and "MPEGAudioRTPSource".

In the case of MPEG-4, there may end up being more than one packet format 
defined, depending on which features of MPEG-4 are being used.  However, my 
reading of the "PacketVideo" whitepaper that you quoted 
<http://www.packetvideo.com/pdf/pv_whitepaper.pdf>, as well as the 
"Wireless Multimedia Forum" technical framework document that it references 
<http://www.wmmforum.com/Documents/WMF_RTFD_v1.pdf>, suggests that the 
PacketVideo Player probably uses the RTP Payload Format defined in RFC 
3016: <ftp://ftp.isi.edu/in-notes/rfc3016.txt>

So, if you were to define subclasses of "RTPSink" that implement this 
payload format, you could probably build a server that could stream MPEG-4 
data that your PacketVideo Player could receive and play.

         Ross.


From <EMAIL-ADDRESS-HIDDEN> Thu Aug 23 08:58:32 2001
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Date: Thu, 23 Aug 2001 08:58:23 -0700
To: Simon Magnusson <EMAIL-ADDRESS-HIDDEN>
From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
Subject: RE: [Live-devel] Using the libraries with Mpeg-4 ?
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Status: O

At 02:20 AM 8/23/01, Simon Magnusson wrote:
>Now I'm wondering if there is any functions for dealing with the RTSP in
>this libraray?

I am currently adding some basic RTSP server support to the "liveMedia" 
library.  This will enable a RTSP client (such as RealPlayer or QuickTime 
Player) to query and play from a streaming application using a RTSP URL, 
rather than having to explicitly open a ".sdp" file.  (Later I also hope to 
add support for seeking/pausing within a (prerecorded) stream.)

         Ross.



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Subject: [Live-devel] Debugging
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Status: O

Ross,

I am still working on modifications to stream MPEG System streams as 
described in RFC 2250. I am planning on creating a subclass as you described 
for MultiFramedRTPSink. Right now I am trying to set up a debugging 
enviornment. I have the code compiled for Visual C++ on Windows but I don't 
know how to use the debugger with your code. I suppose its more a Visual C++ 
question, but since I am using the *.mak files to build my projects there is 
no debugging information. Do you know how i can add the debugging 
information? Is it just another argument I can add to the *.mak files?
Thanks,
Todd

_________________________________________________________________
Get your FREE download of MSN Explorer at http://explorer.msn.com/intl.asp



From <EMAIL-ADDRESS-HIDDEN> Mon Aug 27 12:39:49 2001
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Status: O

At 12:25 PM 8/27/01, Spike 12123 wrote:
>I am still working on modifications to stream MPEG System streams as 
>described in RFC 2250. I am planning on creating a subclass as you 
>described for MultiFramedRTPSink. Right now I am trying to set up a 
>debugging enviornment. I have the code compiled for Visual C++ on Windows 
>but I don't know how to use the debugger with your code. I suppose its 
>more a Visual C++ question, but since I am using the *.mak files to build 
>my projects there is no debugging information. Do you know how i can add 
>the debugging information? Is it just another argument I can add to the 
>*.mak files?

Todd,

If you comment out the line
         NODEBUG=1
that's in each Makefile (or, equivalently, comment out this line in 
"win32config" and then re-run "genWindowsMakefiles"), and then rebuild 
everything, you should get a debuggable version of the Windows code.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Mon Aug 27 16:34:04 2001
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Subject: [Live-devel] Exceptions and other issues
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Status: O

Ross,

I took your advice on how to get the debugger working in Windows and it 
works great. I am having some other issues I am hoping you can help me iron 
out.

1) When running testMPEGVideoStreamer I get the following msdos console 
messages:
======================================================================
Unable to determine our source address: setsockopt(IP_ADD_MEMBERSHIP) error: 
Unk
nown error
Unable to determine our source address: setsockopt(IP_ADD_MEMBERSHIP) error: 
Unk
nown error
Beginning streaming...
Beginning to read from file...
...done reading from file
Beginning to read from file...
======================================================================
I am trying to understand if it is working correctly. The file will loop 
through and appears to stream indefinitely as scheduled. Can you explain 
what the code is doing? It seems to error when the function 
-ourSourceAddressForMulticast(env) - is called in Groupsock.cpp.

2) Perhaps this is related to the errors in question 1, but I cannot view 
anything in Quicktime. I am using the sdp file identical to the one you 
distributed except I changed the ip address to my local address for o and c 
(corresponds to change of - multicastAddressStr - in 
testMPEGVideoStreamer.cpp). I also uncommented the - #define 
SOURCE_IS_PROGRAM_STREAM 1 - because the MPEG1 file is a program stream 
containing audio and video. My only success so far was when I sent a video 
elementary stream MPEG1 file and was able to receive it using JMStudio (this 
of course with - #define SOURCE_IS_PROGRAM_STREAM 1 - commented out).

Could the problem be that I am doing the streaming and receiving on the same 
PC, possibly creating an issue with multiple connections to the same port?

3) Again, could be related to previous questions, when I run in debug mode 
the MS Visual C++ debug window will repeatedly display the message:
First-chance exception in testMPEGVideoStreamer.exe (KERNEL32.DLL): 
0xE06D7363: Microsoft C++ Exception.
Is this a major problem or just a minor bug I don't care about for now?

4)I tried running testMPEGVideoReceiver yet but I am unclear on what it is 
supposed to do. Comments say it "outputs the resulting MPEG file stream 
using a built-in HTTP server." Where does it output the stream? The console 
message is says:
=======================================================================
Unable to determine our source address: setsockopt(IP_ADD_MEMBERSHIP) error: 
Unk
nown error
Unable to determine our source address: setsockopt(IP_ADD_MEMBERSHIP) error: 
Unk
nown error
Beginning receiving multicast stream...
=======================================================================
It just stops right there even with the testMPEGVideoStreamer running along. 
I suppose if I get issues with my first few questions cleared up then it 
might work. I get the impression the stream is redirected from RTP to HTTP 
but if so then how can it be viewed? Is this meant to just receive 
bitstreams and print them out?


Thats it for now. Thanks for response. Last time you were fast to reply and 
I appreciated that. Hope these questions sound familiar and there is 
something real elementary I am not doing. If you need I can email you my 
MPEG file but I don't have a web site I can post it to.

- Todd

_________________________________________________________________
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From <EMAIL-ADDRESS-HIDDEN> Mon Aug 27 18:47:14 2001
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Status: O

At 04:33 PM 8/27/01, Spike 12123 wrote:
>I took your advice on how to get the debugger working in Windows and it 
>works great. I am having some other issues I am hoping you can help me 
>iron out.
>
>1) When running testMPEGVideoStreamer I get the following msdos console 
>messages:
>======================================================================
>Unable to determine our source address: setsockopt(IP_ADD_MEMBERSHIP) 
>error: Unknown error
>Unable to determine our source address: setsockopt(IP_ADD_MEMBERSHIP) 
>error: Unknown error

Do these errors occur only with the debuggable version, or with the earlier 
version that you built without debugging symbols, or both?  Also, could you 
please download and run the version at 
<http://www.live.com/testMPEGVideoStreamer.exe>.  Let me know if you also 
see the error messages with that, pre-built version.


>2) Perhaps this is related to the errors in question 1, but I cannot view 
>anything in Quicktime. I am using the sdp file identical to the one you 
>distributed except I changed the ip address to my local address for o and 
>c (corresponds to change of - multicastAddressStr - in 
>testMPEGVideoStreamer.cpp).

I suggest that you make sure that the original, multicast-based version is 
working OK (using the originally supplied .sdp file, and the the original 
"multicastAddressStr").  Only when you've confirmed that the multicast 
version is working OK should you try changing the program.

>Could the problem be that I am doing the streaming and receiving on the 
>same PC, possibly creating an issue with multiple connections to the same port?

This shouldn't be a problem if you're running the original, multicast 
version.  That's why it's important to get that working first.

(As a general rule, there's no reason not to use multicast, unless you're 
streaming over a network that doesn't support multicast routing.)


>3) Again, could be related to previous questions, when I run in debug mode 
>the MS Visual C++ debug window will repeatedly display the message:
>First-chance exception in testMPEGVideoStreamer.exe (KERNEL32.DLL): 
>0xE06D7363: Microsoft C++ Exception.
>Is this a major problem or just a minor bug I don't care about for now?

I'm not sure.  Note that the routines called by 
MPEGVideoStreamParser::parse() occasionally throw exceptions - as part of 
their normal operation - but MPEGVideoStreamParser::parse() contains a 
"catch" statement that should catch and handle these.  I.e., these 
exceptions do not indicate an error, but you shouldn't be seeing them.

However, if you *are* seeing these exceptions, then there may be a problem 
that I'll need to look into.


>4)I tried running testMPEGVideoReceiver yet but I am unclear on what it is 
>supposed to do. Comments say it "outputs the resulting MPEG file stream 
>using a built-in HTTP server." Where does it output the stream?

"testMPEGVideoReceiver" has a built-in HTTP server, running on port 
10006.  (This port number can be changed.)  If your player tool can read 
data from "http://" URLs, you can read/play the data received by 
"testMPEGVideoReceiver" by pointing your player tool at the URL 
"http://<computer-thats-running-testMPEGVideoReceiver>:10006/"

Alternatively, you can change the code so that "testMPEGVideoReceiver" 
outputs to a file (possible "stdout"), instead of using a built-in HTTP 
server.  (To do this, see the line containing "FileSink::createNew".)

But in any case, you'll need to get rid of the "Unknown error" error 
messages first, before you can expect anything else to work.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Mon Aug 27 23:56:22 2001
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Status: O

Dear Ross,
 My name is Carlos Rivera I am a workmate of Paquito whom you may
know from your mailing list.  At our company we are working on a video server
of recorded MPEG files which accepts RTSP commands, I have just finished
implementing a very rough first version of it. The RTSP parser and server
are written in Python but the RTP/RTCP module is a shared library version
of your code.  I have learned how to use your code mainly from the 
testMPEGVideoStreamer example, and made minor modifications so that I 
could obtain things like RTP time stamp and sequence number (which are 
private members of RTPSink If I remember correctly.) Also I did modify the 
BasicTaskScheduler so that it doesn't block and I have a multithreaded 
server which can accept many requests simultaneously.

        However my knowledge of the inner workings of the library is still
very poor, and I would like to be able to do a couple of things which
require a better understanding than I have. I would like to be able to play
from a given 'time position' (in seconds) since the beginning of the MPEG 
file (not just always from the beginning.)
I have implemented the RTSP PAUSE request, but when I next get a PLAY
request I am not able to start from the exact point that the client wants,
My only 'rough' control is simply a pair of function calls of the form:
videoSink->stopPlaying() and videoSink->startPlaying and implementing it
this way makes the client loose images (the images sent in the time between
when the client send the PAUSE request and when my server actually pauses.)
        The second thing that I would like to do would be to interpret the
statistics from the RTCP packets so that I could change the playing rate
'on the fly', I guess the interpretation is easy the real problem is how
to dynamically adjust the rate.

        Well I guess my message is already long enough, please could you
hint me as to how I could proceed?   Also I have read in previous messages
that you are working on an RTSP server, you may be interested in my
Python code which I imagine it will have GPL licence, although I would
have to ask my boss just to make sure.

Thanks for your time.

Carlos.


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Status: O

Carlos,

Thanks for the note.  It's always great to hear from people who are making 
good use of the libraries.

>I have learned how to use your code mainly from the
>testMPEGVideoStreamer example, and made minor modifications so that I
>could obtain things like RTP time stamp and sequence number (which are
>private members of RTPSink If I remember correctly.) Also I did modify the
>BasicTaskScheduler so that it doesn't block and I have a multithreaded
>server which can accept many requests simultaneously.

If you'd like to mail me the files that you're updated (or 'patch' files), 
I'll see if I can incorporate the changes into future releases of the 
library code, so that you don't have to keep patching your own copy each time.


>        However my knowledge of the inner workings of the library is still
>very poor, and I would like to be able to do a couple of things which
>require a better understanding than I have. I would like to be able to play
>from a given 'time position' (in seconds) since the beginning of the MPEG
>file (not just always from the beginning.)

Yes, this has been a popular request :-)  To support this, I will need to 
update the "liveMedia" library so that certain 'sources' can handle a 
'seek-by-time' operation.  I will also see if I can improve the 
implementation of "stopPlaying()"/"startPlaying()" for the various media 
types, so that you can more effectively use these functions to implement a 
'pause' command (as you have been trying to do).

These changes to the library are on my "to do" list, and will happen at 
some point in the future.  In general, though, the library 
changes/additions that I give highest priority to are those for which we 
(LIVE.COM) have received financial support.  (E.g., the recent support for 
MPEG video came about as a result of financial support by a company that's 
developing IP-based security cameras.)  If your company is interested in 
providing financial support to help add features to the library (either 
general purposes changes, such as the support for seeking-by-time, or 
special-purpose or proprietary additions that can be linked with the 
libraries), please let me know.


>         The second thing that I would like to do would be to interpret the
>statistics from the RTCP packets so that I could change the playing rate
>'on the fly', I guess the interpretation is easy the real problem is how
>to dynamically adjust the rate.

One relatively simple way to do this would be to add a 'filter' component 
that selectively adjusts the output frame rate by dropping certain frames.


>Also I have read in previous messages
>that you are working on an RTSP server

Yes, I have recently implemented a very simple RTSP server that supports 
just the OPTIONS, DESCRIBE, SETUP, TEARDOWN, PLAY methods (& doesn't handle 
seeking-by-time).  This hasn't been well tested yet, but I'll be adding it 
to the library (& test programs) soon.


>you may be interested in my
>Python code which I imagine it will have GPL licence, although I would
>have to ask my boss just to make sure.

Yes, I'd be happy to accept new code contributions like this (non-C/C++ 
code like this would end up going in a separate directory), as long as the 
license is the LGPL, rather than the GPL.  (I don't want to disallow people 
from building proprietary applications that make use of the libraries, or 
from linking the libraries with their own, proprietary source/sink 
components, if they choose.  The LGPL allows people to do this, but the GPL 
doesn't.)

         Ross Finlayson
         LIVE.COM



From <EMAIL-ADDRESS-HIDDEN> Tue Aug 28 13:06:41 2001
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Subject: [Live-devel] Groupsock Errors
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Status: O

Ross,

I took your advice and downloaded the test program from 
<http://www.live.com/testMPEGVideoStreamer.exe>. It worked perfectly. I used 
quicktime and the exisitng sdp and saw the video just fine. So for testing 
on I reinstalled all the files on my Linux machine (just in case windows had 
introduced some strange errors) and recompiled and linked everything using 
the config.linux as described on your website.
So running this produced the groupsock error again (shown below).

-------------
12:55:45 Groupsock(8: 239.255.42.42, 8888, 7): failed to join group: 
setsockopt(
IP_ADD_MEMBERSHIP) error: Unknown error
Unable to determine our source address: setsockopt(IP_ADD_MEMBERSHIP) error: 
Unknown error
12:56:10 Groupsock(56: 239.255.42.42, 8889, 7): failed to join group: 
setsockopt (IP_ADD_MEMBERSHIP) error: Unknown error
-------------


Since I cannot get the orginal multicast version working okay I don't know 
if there is internal code I should be debugging or if maybe the config.linux 
is incorrect or something else I should be lookgin at.  I am determined to 
make this work because this software seems to contain many of the components 
I need for a streaming MPEG/RTP project I am working on.

Any more advice or things to try I would appreciate.

Thanks,
Todd


_________________________________________________________________
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From <EMAIL-ADDRESS-HIDDEN> Tue Aug 28 13:21:24 2001
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Status: O

At 01:06 PM 8/28/01, Spike 12123 wrote:
>I took your advice and downloaded the test program from 
><http://www.live.com/testMPEGVideoStreamer.exe>. It worked perfectly. I 
>used quicktime and the exisitng sdp and saw the video just fine. So for 
>testing on I reinstalled all the files on my Linux machine (just in case 
>windows had introduced some strange errors) and recompiled and linked 
>everything using the config.linux as described on your website.
>So running this produced the groupsock error again (shown below).

Now I'm confused.  Are you seeing these error messages when running the 
software on Windows, or on Linux???  In your last message, you implied that 
you were running the software on Windows.  Now you imply that you're 
running the software on Linux.  Which is it?

If you're seeing these error messages on Linux, then it's probably because 
your Linux kernel isn't configured for IP multicast.  To fix this, you'll 
need to set
         CONFIG_IP_MULTICAST=y
in your Linux kernel config file, and rebuild your kernel.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Tue Aug 28 14:04:42 2001
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Status: O

Sorry to confuse you. I am using windows and since I have been getting the 
errors on Windows I thought I would attempt to run the software on Linux and 
see if the results are any different. I get the same errors on both windows 
and linux. At this point I would be happy if I could get the original 
multicast test program working in either machine. I checked my kernal and 
CONFIG_IP_MULTICAST is set to y. Even if this were the problem why would I 
be getting the same error on Windows? I appreciate your help and if you can 
think of something to try let me know.
Thanks.
Todd



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From <EMAIL-ADDRESS-HIDDEN> Tue Aug 28 14:13:12 2001
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Status: O

At 02:04 PM 8/28/01, Spike 12123 wrote:
>Sorry to confuse you. I am using windows and since I have been getting the 
>errors on Windows I thought I would attempt to run the software on Linux 
>and see if the results are any different. I get the same errors on both 
>windows and linux. At this point I would be happy if I could get the 
>original multicast test program working in either machine. I checked my 
>kernal and CONFIG_IP_MULTICAST is set to y. Even if this were the problem 
>why would I be getting the same error on Windows?

Because sometimes the same error message can be triggered by different 
problems :-)

If multicast is configured in your Linux kernel, then I don't understand 
why you would be seeing these errors when you run the Linux version of the 
software.  (I'm assuming here that you really are running the Linux version 
of the software, and are not trying to do something bizarre like run the 
Windows version on Linux using WINE :-)

Could you mail me the output of running
         netstat -r
on your Linux machine?

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Wed Aug 29 10:09:49 2001
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Status: O

Hi Ross,

> Carlos,
>
> Thanks for the note.  It's always great to hear from people who are making
> good use of the libraries.
>
> >I have learned how to use your code mainly from the
> >testMPEGVideoStreamer example, and made minor modifications so that I
> >could obtain things like RTP time stamp and sequence number (which are
> >private members of RTPSink If I remember correctly.) Also I did modify the
> >BasicTaskScheduler so that it doesn't block and I have a multithreaded
> >server which can accept many requests simultaneously.
>
> If you'd like to mail me the files that you're updated (or 'patch' files),
> I'll see if I can incorporate the changes into future releases of the
> library code, so that you don't have to keep patching your own copy each
> time.

Ok, although I am not sure if the changes I made break the philosophical 
design of the library,  I needed quick access to a member so I made
it public before thinking it twice.   Also I run into a problem which I don't 
understand, I get a segmentation fault because the function 
MultiFramedRTPSink::packFrame() gets called after the sink is closed
and fSource has become a null pointer (fSource gets used inside of 
packFrame.)
This is what I do, I run startPlaying and then launch on a separate
thread the function scheduler->BlockMyself().   Inside of the loop of
BlockMyself I have changed the while(1) by while(m_block), where
m_block is a new member variable.  When I want to stop playing 
I set m_block to false -from a thread different than where BlockMyself
is running-.  Then i use pthread_join( )  to wait for the thread to finish
and then I do call stopPlaying.   Later on I run startPlaying again and
launch a new thread where the function BlockMyself runs.
This seems to work until the MPEG gets to the end, then the source
gets closed and I get the segfault I mentioned.
I have solved it by checking fSource in packFrame before it gets
used and that seems to work, but since I do not understand it it is
an ugly fix.  Also I sometimes do not get the segfault but get an
error message from getNextFrame that complains that I am attempting
to read a frame twice, whether the former or the later happen seems
completely random.

> >        However my knowledge of the inner workings of the library is still
> >very poor, and I would like to be able to do a couple of things which
> >require a better understanding than I have. I would like to be able to
> > play from a given 'time position' (in seconds) since the beginning of the
> > MPEG file (not just always from the beginning.)
>
> Yes, this has been a popular request :-)  To support this, I will need to
> update the "liveMedia" library so that certain 'sources' can handle a
> 'seek-by-time' operation.  I will also see if I can improve the
> implementation of "stopPlaying()"/"startPlaying()" for the various media
> types, so that you can more effectively use these functions to implement a
> 'pause' command (as you have been trying to do).
>
> These changes to the library are on my "to do" list, and will happen at
> some point in the future.  In general, though, the library
> changes/additions that I give highest priority to are those for which we
> (LIVE.COM) have received financial support.  (E.g., the recent support for
> MPEG video came about as a result of financial support by a company that's
> developing IP-based security cameras.)  If your company is interested in
> providing financial support to help add features to the library (either
> general purposes changes, such as the support for seeking-by-time, or
> special-purpose or proprietary additions that can be linked with the
> libraries), please let me know.

For the moment I don't think we are interested and we would like to work on 
the modifications ourselves, but I would let you know if that changed.

> >         The second thing that I would like to do would be to interpret
> > the statistics from the RTCP packets so that I could change the playing
> > rate 'on the fly', I guess the interpretation is easy the real problem is
> > how to dynamically adjust the rate.
>
> One relatively simple way to do this would be to add a 'filter' component
> that selectively adjusts the output frame rate by dropping certain frames.
>

That would lower the rate, would it be possible to increase it? what would 
happen to the normal playing time of the client?  

> >Also I have read in previous messages
> >that you are working on an RTSP server
>
> Yes, I have recently implemented a very simple RTSP server that supports
> just the OPTIONS, DESCRIBE, SETUP, TEARDOWN, PLAY methods (& doesn't handle
> seeking-by-time).  This hasn't been well tested yet, but I'll be adding it
> to the library (& test programs) soon.
>
> >you may be interested in my
> >Python code which I imagine it will have GPL licence, although I would
> >have to ask my boss just to make sure.
>
> Yes, I'd be happy to accept new code contributions like this (non-C/C++
> code like this would end up going in a separate directory), as long as the
> license is the LGPL, rather than the GPL.  (I don't want to disallow people
> from building proprietary applications that make use of the libraries, or
> from linking the libraries with their own, proprietary source/sink
> components, if they choose.  The LGPL allows people to do this, but the GPL
> doesn't.)

Okey, I will look into that.

Thanks.
Carlos.



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From: {Spike 12123} <EMAIL-ADDRESS-HIDDEN>
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Subject: [Live-devel] Video Players
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Status: O

Hello,
In order to use testMPEGVideoReceiver with a player, the player must support 
urls of the form http://<host-address>:10006 . I tried this on Quicktime, 
RealPlayer and MediaPlayer 6.4 and they rejected the format. Do you know 
which players will support this http stream?

Related question, do you know how to get RealPlayer and Windows Media Player 
to play RTP streams such as the one from testMPEGVideoStreamer (assuming the 
payload type is supported, ie. MPEG1)?

Thanks very much.
Todd

_________________________________________________________________
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From <EMAIL-ADDRESS-HIDDEN> Wed Aug 29 17:01:51 2001
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Date: Wed, 29 Aug 2001 17:01:36 -0700
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Subject: [Live-devel] Re: Video Players
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Status: O

At 12:41 PM 8/29/01, Spike 12123 wrote:
>In order to use testMPEGVideoReceiver with a player, the player must 
>support urls of the form http://<host-address>:10006 . I tried this on 
>Quicktime, RealPlayer and MediaPlayer 6.4 and they rejected the format.

Sorry, I should have said:
         http://<host-address>:10006/dummy.mpg

The "/dummy.mpg" part at the end is important, for RealPlayer at least, 
otherwise it gets confused.


>  Do you know which players will support this http stream?

I've demonstrated that both QuickTime Player version 5.0, and Windows Media 
Player (I'm running version 6.4.07.1112) will play MPEG-1 video (*not* 
MPEG-2 video) from a HTTP server.  For some strange reason, I haven't been 
able to get RealPlayer to play MPEG-1 video from a HTTP server, even though 
it can play MPEG-1 video *files* without problems.


>Related question, do you know how to get RealPlayer and Windows Media 
>Player to play RTP streams such as the one from testMPEGVideoStreamer 
>(assuming the payload type is supported, ie. MPEG1)?

I've been told (by contacts at Real Networks) that RealPlayer currently 
can't play MPEG video/RTP streams (only MPEG audio/RTP), but that some 
future version might support this.

As far as I know, Windows Media Player can't receive RTP streams at all 
(presumably because RTP is a standard that M$ can't control :-).

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Wed Aug 29 18:58:20 2001
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Date: Wed, 29 Aug 2001 18:53:14 -0700
To: Carlos Rivera <EMAIL-ADDRESS-HIDDEN>
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Subject: Re: [Live-devel] Re: Re: pausing and flow adjustment
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Status: O

At 10:09 AM 8/29/01, Carlos Rivera wrote:
>Ok, although I am not sure if the changes I made break the philosophical
>design of the library,  I needed quick access to a member so I made
>it public before thinking it twice.

Let me know which member variables you made public, so that I can do the 
same (or, equivalently, define accessor functions) in future releases of 
the code.

>    Also I run into a problem which I don't
>understand, I get a segmentation fault because the function
>MultiFramedRTPSink::packFrame() gets called after the sink is closed
>and fSource has become a null pointer

I think this is happening because you're trying to execute library code 
from more than one thread.  You shouldn't be doing this - the library code 
is not designed to be multithreaded.  Instead, it is designed to be run 
from a single-threaded event loop (like the one that implements 
"BasicUsageEnvironment::BlockMyself()").

If you want to control streaming that's already underway, then you should 
do so using the event loop.  E.g., to stop playing a stream, you could 
modify the event loop in some way, so that it recognizes when a request has 
been made to stop playing the stream.  The event loop could then call 
"stopPlaying()".  In this way, all of the library code gets called from 
within a single thread.

One simple way to do this would be to have a global 'command' variable 
that's checked within each iteration of the "BlockMyself()" event loop, and 
set from another thread (e.g., whichever thread is running your 
GUI).  However, while I think this would probably let you stop a stream, 
I'm not sure whether you'd be able to start it up again (because, with no 
stream currently running, the event loop will likely be blocked in the 
select() call).  So, an alternative (perhaps better) way to do this would 
be to create a new socket especially for controlling the library, and write 
to this socket whenever you wish to (e.g.) stop the stream.  If you did 
this, you could register your own command handler function using 
"turnOnBackgroundReadHandling()", and you wouldn't have to modify the 
"BlockMyself()" event loop at all!

Yet another approach would be to integrate the library's event loop with 
your GUI's event loop, so that you don't need separate threads for the GUI 
and for the library.  Instead, there would be just a single threaded event 
loop that handles both GUI operations and streaming library operations.

In any case, if you choose to make a change to the event loop, you should 
probably do so by defining your own subclass of "TaskScheduler" rather than 
modifying the "BasicTaskScheduler" code.  That way, your changes won't get 
overwritten whenever you download an updated version of the library code.


> > some point in the future.  In general, though, the library
> > changes/additions that I give highest priority to are those for which we
> > (LIVE.COM) have received financial support.  (E.g., the recent support for
> > MPEG video came about as a result of financial support by a company that's
> > developing IP-based security cameras.)  If your company is interested in
> > providing financial support to help add features to the library (either
> > general purposes changes, such as the support for seeking-by-time, or
> > special-purpose or proprietary additions that can be linked with the
> > libraries), please let me know.
>
>For the moment I don't think we are interested and we would like to work on
>the modifications ourselves, but I would let you know if that changed.

If time-to-market of your product is an important goal for you, then you 
may find it worthwhile to consider dividing up the work so that you folks 
can concentrate on code that's specific to your product, with others (like 
me) focusing on the generic libray code that's underneath.


> > >         The second thing that I would like to do would be to interpret
> > > the statistics from the RTCP packets so that I could change the playing
> > > rate 'on the fly', I guess the interpretation is easy the real problem is
> > > how to dynamically adjust the rate.
> >
> > One relatively simple way to do this would be to add a 'filter' component
> > that selectively adjusts the output frame rate by dropping certain frames.
>
>That would lower the rate, would it be possible to increase it?

There's no (easy) way to increase the frame rate of a MPEG stream that's 
already been encoded, so one thing you could do is encode the stream at the 
highest frame rate that you are likely to need, and have your control 
filter alway stream frames at <= that rate.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Thu Aug 30 02:13:28 2001
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Subject: [Live-devel] New library release, including initial RTSP server
 implementation
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Status: O

FYI, I have just released a new version (2001.08.30) of the "LIVE.COM 
Streaming Media" libraries.

The major change this time is the addition (to the "liveMedia" library) of 
an initial, bare-bones, implementation of a RTSP server.  (See also 
"testMPEGVideoStreamer.cpp" and "testMP3Streamer.cpp" (in "testProgs"0 for 
examples of how to create a RTSP server for an outgoing stream.)

This initial RTSP server implementation is not particularly useful just yet 
- you can 'play' an existing, ongoing media stream, but you can't yet pause 
or seek within a stream.  (These are additions that I hope to make to the 
library in the future.)

Unfortunately RealPlayer has a bug (that I've reported to RealNetworks) 
that prevents it from playing a multicast MP3 session using a RTSP 
URL.  Also, QuickTime player is currently having some trouble playing a 
MPEG video session using a RTSP URL.  So the bottom line is that this RTSP 
server code is still 'work in progress'...

	Ross.



From <EMAIL-ADDRESS-HIDDEN> Fri Aug 31 16:43:00 2001
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Subject: [Live-devel] MPEG system streams
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Status: O

Ross,
I am about to begin writing a subclass for MultiFramedRTPSink so that I can 
implement the MPEG System streams as defined by sec 2 of RFC 2250. You 
recommended this would be easy because there are no packetization 
restrictions. I wanted to clarify with you my understanding of RFC 2250 sec 
2. Your code very clearly builds an MPEG Video-specific header for the 
MPEGVideoRTPSink. Is it your understanding that for sec 2 (system streams) 
that only the RTP fixed data packet header is required? The video specific 
header is NOT needed? Also, in the RTP packet header the Payload type, M 
bit, and timestamp are specific to the MPEG system stream. If this sounds 
correct so far, do you know what Payload type should be used for systems 
streams? According to the documents I have read 33 is the static payload 
type for MP2T (transport streams) and 32 is specifically for elementary mpeg 
video streams.  Will I have to use a dynamic payload type (96 - 127)? Your 
advice so far as been invaluable. Thanks.
Todd



_________________________________________________________________
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From <EMAIL-ADDRESS-HIDDEN> Fri Aug 31 18:10:26 2001
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From: {Dan Phan} <EMAIL-ADDRESS-HIDDEN>
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Subject: [Live-devel] RTSP server
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Status: O

I just downloaded version 8/30/01 and did a quick test on RTSP.
After making a few minor changes on TestMP3Streamer to enable RTSP, I use
QuickTime player 4 with the URL "rtsp:mylinuxbox:554". I could see the
TestMP3Streamer program accepted the connection on this port, however an
error code of 400 is then seen on the Quicktime side.  After a little bit of
digging, I could see that the function parseRequestString in RTSPServer.cpp
returned with a failure code.  It look like it was looking for the string
"CSeq:" immediately after a command (e.g., "GET").  I could see QuickTime
was sending GET HTTP/1.0 but nothing about "CSeq".  As far as I know the
draft RFC for RTSP by Schulzrinne has no mentioning of "CSeq".

Ross, since you mentioned in the last announcement that the RTSP
implementation is a work in progress, I'm just curious about the methods you
used for testing.

Dan



From <EMAIL-ADDRESS-HIDDEN> Sat Sep 01 06:27:29 2001
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Subject: Re: [Live-devel] RTSP server
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Date: Sat Sep  1 06:28:07 2001
X-Original-Date: Sat, 1 Sep 2001 15:29:34 +0200
Status: O

Hi,

> I just downloaded version 8/30/01 and did a quick test on RTSP.
> After making a few minor changes on TestMP3Streamer to enable RTSP, I u=
se
> QuickTime player 4 with the URL "rtsp:mylinuxbox:554". I could see the
> TestMP3Streamer program accepted the connection on this port, however a=
n
> error code of 400 is then seen on the Quicktime side.  After a little b=
it
> of digging, I could see that the function parseRequestString in
> RTSPServer.cpp returned with a failure code.  It look like it was looki=
ng
> for the string "CSeq:" immediately after a command (e.g., "GET").  I co=
uld
> see QuickTime was sending GET HTTP/1.0 but nothing about "CSeq".  As fa=
r as
> I know the draft RFC for RTSP by Schulzrinne has no mentioning of "CSeq=
".

Your problem is the QuickTime settings. During QT instalation process, it=
=20
tests your internet connection to know what kind of streaming protocol it=
=20
will use before. For QT there are two possibilities to receive video=20
streaming, the first one is by RTSP/RTP over UDP, and the second one by=20
HTTP/RTP (something like 'interleaved RTSP', but it isn't the same ) .  Q=
T=20
select the second way when it cann't receive RTP/UDP packets from a video=
=20
server. For example, if you are behind a ADSL router (NAT), you cann't=20
receive RTP packets.=20
When QT uses 'interleaved RTSP', it doesn't do as RFC describes, because =
it=20
send a GET command, instead of a RTSP command.
You can select streaming way from QT menu:
Go to  Edit / Preferences / QuickTime Preferences / Streaming Transport
and set 'Streaming transport' to RTSP/RTP

=09Francisco J. Cabello


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From: Francisco Javier Cabello Torres <EMAIL-ADDRESS-HIDDEN>
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Date: Sat Sep  1 06:29:02 2001
X-Original-Date: Sat, 1 Sep 2001 15:30:43 +0200
Status: O

Hi,

> I just downloaded version 8/30/01 and did a quick test on RTSP.
> After making a few minor changes on TestMP3Streamer to enable RTSP, I u=
se
> QuickTime player 4 with the URL "rtsp:mylinuxbox:554". I could see the
> TestMP3Streamer program accepted the connection on this port, however a=
n
> error code of 400 is then seen on the Quicktime side.  After a little b=
it
> of digging, I could see that the function parseRequestString in
> RTSPServer.cpp returned with a failure code.  It look like it was looki=
ng
> for the string "CSeq:" immediately after a command (e.g., "GET").  I co=
uld
> see QuickTime was sending GET HTTP/1.0 but nothing about "CSeq".  As fa=
r as
> I know the draft RFC for RTSP by Schulzrinne has no mentioning of "CSeq=
".


Your problem is the QuickTime settings. During QT instalation process, it=
=20
tests your internet connection to know what kind of streaming protocol it=
=20
will use before. For QT there are two possibilities to receive video=20
streaming, the first one is by RTSP/RTP over UDP, and the second one by=20
HTTP/RTP (something like 'interleaved RTSP', but it isn't the same ) .  Q=
T=20
select the second way when it cann't receive RTP/UDP packets from a video=
=20
server. For example, if you are behind a ADSL router (NAT), you cann't=20
receive RTP packets.=20
When QT uses 'interleaved RTSP', it doesn't do as RFC describes, because =
it=20
send a GET command, instead of a RTSP command.
You can select streaming way from QT menu:
Go to  Edit / Preferences / QuickTime Preferences / Streaming Transport
and set 'Streaming transport' to RTSP/RTP

         Francisco J. Cabello


From <EMAIL-ADDRESS-HIDDEN> Mon Sep 03 20:54:07 2001
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To: {Spike 12123} <EMAIL-ADDRESS-HIDDEN>
From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] MPEG system streams
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Date: Mon Sep  3 20:55:02 2001
X-Original-Date: Mon, 03 Sep 2001 20:51:01 -0700
Status: O

At 04:42 PM 8/31/01, Spike 12123 wrote:
>Ross,
>I am about to begin writing a subclass for MultiFramedRTPSink so that I 
>can implement the MPEG System streams as defined by sec 2 of RFC 2250. You 
>recommended this would be easy because there are no packetization 
>restrictions. I wanted to clarify with you my understanding of RFC 2250 
>sec 2. Your code very clearly builds an MPEG Video-specific header for the 
>MPEGVideoRTPSink. Is it your understanding that for sec 2 (system streams) 
>that only the RTP fixed data packet header is required? The video specific 
>header is NOT needed?

That's correct.  Unfortunately RFC 2250 can be a little confusing about 
this, but if for streaming MPEG system streams, the *only* relevant section 
of this RFC is section 2.  So, section 2 is the only section that you need 
read..  (Section 3 applies *only* to elementary (video or audio) streams.)

>  Also, in the RTP packet header the Payload type, M bit, and timestamp 
> are specific to the MPEG system stream. If this sounds correct so far, do 
> you know what Payload type should be used for systems streams? According 
> to the documents I have read 33 is the static payload type for MP2T 
> (transport streams) and 32 is specifically for elementary mpeg video 
> streams.  Will I have to use a dynamic payload type (96 - 127)?

You should note
<ftp://ftp.ietf.org/internet-drafts/draft-ietf-avt-profile-new-11.txt>

This document notes that yes, 33 is the (static) payload format for MP2T, 
so that is the payload format number that you should use if you're 
streaming MPEG-2 Transport streams.

However, no static payload format numbers have been defined for streaming 
MPEG-2 Program streams or MPEG-1 System streams, so if you're streaming 
these (as opposed to MPEG-2 Transport streams), then you'll need to use a 
dynamic payload type number (one that's in the range [96,127])

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Tue Sep 04 00:36:30 2001
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Date: Tue Sep  4 00:37:02 2001
X-Original-Date: Tue, 04 Sep 2001 00:32:38 -0700
Status: O

At 06:09 PM 8/31/01, Dan Phan wrote:
>I just downloaded version 8/30/01 and did a quick test on RTSP.
>After making a few minor changes on TestMP3Streamer to enable RTSP, I use
>QuickTime player 4 with the URL "rtsp:mylinuxbox:554".

Dan,

(Following up on Francisco's good advice.)

Please also try using the most recent version - version 5 - of the 
QuickTime Player.  I have been doing all of my testing with this version, 
and so I am more confident in QuickTime Player version 5.

I also recommend that you upgrade to the newest version of the LIVE.COM 
Streaming Media libraries - version 2001.09.03 - as this contains some 
minor improvements to the RTSP server implementation.

Unfortunately I have not yet gotten QuickTime Player to work properly with 
the RTSP server implementation.  For some reason - still unknown to me - 
the player is not happy with the way my server responds to a DESCRIBE 
request.  (The player displays "-5402 - Bad Data".)  I have asked a contact 
at Apple for help in trying to figure out what the QuickTime Player thinks 
is wrong with the server's response.  (If anyone on this list can figure 
this out, please let me know.)


>Ross, since you mentioned in the last announcement that the RTSP
>implementation is a work in progress, I'm just curious about the methods you
>used for testing.

The server currently implements OPTIONS, DESCRIBE, SETUP, PLAY, and 
TEARDOWN, although (because of the QuickTime Player problem) I have not yet 
tested PLAY and TEARDOWN.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Tue Sep 04 14:04:57 2001
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Date: Tue Sep  4 14:05:11 2001
X-Original-Date: Tue, 4 Sep 2001 14:04:18 -0700
Status: O

Ross, Francisco:

Thanks for your reply messages.  Following up on your advice, I ran QT 5 and
got the same 5402 error code as described by Ross.  One possibility that
might help resolve this mystery is to look into Apple's Darwin Streaming
Server, which is an open source project.  I was told that DSS responded to
rtsp command with QT 5.

Dan

> -----Original Message-----
> From: live-devel-admin(at)lists.sourceforge.net
> [mailto:live-devel-admin(at)lists.sourceforge.net]On Behalf Of Ross
> Finlayson
> Sent: Tuesday, September 04, 2001 12:33 AM
> To: dphan(at)arch.sel.sony.com
> Cc: live-devel(at)lists.sourceforge.net
> Subject: Re: [Live-devel] RTSP server
>
>
> At 06:09 PM 8/31/01, Dan Phan wrote:
> >I just downloaded version 8/30/01 and did a quick test on RTSP.
> >After making a few minor changes on TestMP3Streamer to enable RTSP, I use
> >QuickTime player 4 with the URL "rtsp:mylinuxbox:554".
>
> Dan,
>
> (Following up on Francisco's good advice.)
>
> Please also try using the most recent version - version 5 - of the
> QuickTime Player.  I have been doing all of my testing with this version,
> and so I am more confident in QuickTime Player version 5.
>
> I also recommend that you upgrade to the newest version of the LIVE.COM
> Streaming Media libraries - version 2001.09.03 - as this contains some
> minor improvements to the RTSP server implementation.
>
> Unfortunately I have not yet gotten QuickTime Player to work
> properly with
> the RTSP server implementation.  For some reason - still unknown to me -
> the player is not happy with the way my server responds to a DESCRIBE
> request.  (The player displays "-5402 - Bad Data".)  I have asked
> a contact
> at Apple for help in trying to figure out what the QuickTime
> Player thinks
> is wrong with the server's response.  (If anyone on this list can figure
> this out, please let me know.)
>
>
> >Ross, since you mentioned in the last announcement that the RTSP
> >implementation is a work in progress, I'm just curious about the
> methods you
> >used for testing.
>
> The server currently implements OPTIONS, DESCRIBE, SETUP, PLAY, and
> TEARDOWN, although (because of the QuickTime Player problem) I
> have not yet
> tested PLAY and TEARDOWN.
>
>          Ross.
>
>
> _______________________________________________
> Live-devel mailing list
> Live-devel(at)lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/live-devel
>



From <EMAIL-ADDRESS-HIDDEN> Tue Sep 04 23:11:40 2001
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Date: Tue Sep  4 23:12:02 2001
X-Original-Date: Tue, 04 Sep 2001 23:11:16 -0700
Status: O

At 02:04 PM 9/4/01, Dan Phan wrote:
>Thanks for your reply messages.  Following up on your advice, I ran QT 5 and
>got the same 5402 error code as described by Ross.

FYI, I was able to find out why QT 5 Player was complaining - it was 
because my RTSP server implementation wasn't including a
         a=control:...
line in the SDP data returned in response to a DESCRIBE request.

I have fixed this now, and installed a new version of the library (2001.09.04).

Unfortunately, another problem has shown up: a bug in QT 5 Player.  When it 
subsequently does a SETUP, it tries to do so for a unicast stream, even 
though the actual stream is multicast.  Apparently the people who 
programmed this player thought that RTSP would be used only to play a 
unicast stream, and didn't realize that it's perfectly legitimate to use 
RTSP to play a multicast stream.

Strangely enough, RealPlayer has the same problem.  (I've reported these 
bugs to Apple and RealNetworks respectively.)

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Thu Sep 06 00:30:34 2001
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Date: Thu Sep  6 00:31:02 2001
X-Original-Date: Thu, 6 Sep 2001 09:30:28 +0200
Status: O

> Message: 2
> Date: Tue, 04 Sep 2001 23:11:16 -0700
> To: <dphan(at)arch.sel.sony.com>
> Subject: RE: [Live-devel] RTSP server
> Cc: <live-devel(at)lists.sourceforge.net>
>
> At 02:04 PM 9/4/01, Dan Phan wrote:
> >Thanks for your reply messages.  Following up on your advice, I ran QT 5
> > and got the same 5402 error code as described by Ross.
>
> FYI, I was able to find out why QT 5 Player was complaining - it was
> because my RTSP server implementation wasn't including a
>          a=control:...
> line in the SDP data returned in response to a DESCRIBE request.
>
> I have fixed this now, and installed a new version of the library
> (2001.09.04).

Hi Ross,  
 Since you brought that up,  do you use a=control:track='something' ?
Do you know if that is QuickTime specific? what should the 'something' be?
In my implementation of RTSP server in Python I use track=2 and it works
but I am not sure why,  I haven't managed to make my server work with
Java MFStudio but I think it is for another reason (since it does actually 
send me a PLAY request.)

> Unfortunately, another problem has shown up: a bug in QT 5 Player.  When it
> subsequently does a SETUP, it tries to do so for a unicast stream, even
> though the actual stream is multicast.  Apparently the people who
> programmed this player thought that RTSP would be used only to play a
> unicast stream, and didn't realize that it's perfectly legitimate to use
> RTSP to play a multicast stream.
>
> Strangely enough, RealPlayer has the same problem.  (I've reported these
> bugs to Apple and RealNetworks respectively.)

That's strange, how do you tell Quicktime that you are going to send a 
multicast stream?

Carlos


From <EMAIL-ADDRESS-HIDDEN> Thu Sep 06 10:00:30 2001
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Date: Thu Sep  6 10:01:04 2001
X-Original-Date: Thu, 06 Sep 2001 10:00:08 -0700
Status: O

At 12:30 AM 9/6/01, Carlos Rivera wrote:
>  Since you brought that up,  do you use a=control:track='something' ?

The RTSP server implementation currently sends back
         a=control:video
(or "a=control:audio" if its an audio stream).  This is just to make the 
QuickTime Player happy; I think it will accept just about anything after 
the "a=control:" - it will just get used in the URL that gets sent in the 
subsequent SETUP.


> > Unfortunately, another problem has shown up: a bug in QT 5 Player.  When it
> > subsequently does a SETUP, it tries to do so for a unicast stream, even
> > though the actual stream is multicast.  Apparently the people who
> > programmed this player thought that RTSP would be used only to play a
> > unicast stream, and didn't realize that it's perfectly legitimate to use
> > RTSP to play a multicast stream.
> >
> > Strangely enough, RealPlayer has the same problem.  (I've reported these
> > bugs to Apple and RealNetworks respectively.)
>
>That's strange, how do you tell Quicktime that you are going to send a
>multicast stream?

Right now, unfortunately, you can't tell QuickTime Player (or RealPlayer) 
to receive/play a multicast stream using RTSP.  Instead, you have to feed 
it a SDP file directly - either 'by hand', or from a HTTP server.

I have told both Apple and RealNetworks about this bug in their players, so 
(I hope) future versions of the players will fix this problem.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Mon Sep 17 01:20:40 2001
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Date: Mon Sep 17 01:21:02 2001
X-Original-Date: Mon, 17 Sep 2001 10:20:38 +0200
Status: O

Hi Ross,
  Please consider initiating fCurrentTimestamp member of RTPSink to 
fTimestampBase, I think that is more useful than a bogus value since
then you can obtain the first timestamp without starting to play.
Just to remind you this is the constructor I am refering to:


RTPSink::RTPSink(UsageEnvironment& env,
		 Groupsock* RTPgs, unsigned char RTPPayloadType)
  : MediaSink(env), fRTPgs(RTPgs), fRTPPayloadType(RTPPayloadType),
    fPacketCount(0), fOctetCount(0) {
  gettimeofday(&fCreationTime, &Idunno);

  fSeqNo = (unsigned)our_random();
  fSSRC = (unsigned)our_random();
  fTimestampBase = (unsigned)our_random();
  fTimestampFrequency = 90000; // by default; subclasses can change it
  fCurrentTimestamp = 0xDEADBEEF;  // bogus value until setTimestamp() called
}


Thanks
Carlos


From <EMAIL-ADDRESS-HIDDEN> Mon Sep 17 03:33:18 2001
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Date: Mon Sep 17 03:34:03 2001
X-Original-Date: Mon, 17 Sep 2001 03:33:05 -0700
Status: O

At 01:20 AM 9/17/01, you wrote:
>Hi Ross,
>   Please consider initiating fCurrentTimestamp member of RTPSink to
>fTimestampBase


OK, thanks for the suggestion; I'll make this change in the next release.


>I think that is more useful than a bogus value since
>then you can obtain the first timestamp without starting to play.

(Note, though, that this is true only if the very first 'presentation time' 
(struct timeval) is [0,0] - i.e., 0 seconds, 0 microseconds.)

         Ross.




From <EMAIL-ADDRESS-HIDDEN> Sat Sep 29 12:39:15 2001
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Subject: [Live-devel] Streaming Media Lib not compatible with Win XP
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Date: Sat Sep 29 12:40:01 2001
X-Original-Date: Sat, 29 Sep 2001 12:38:33 -0700
Status: O

On Win XP (I'm using a beta copy, build 2600), it looks like the socket
number returned by socket() is a large number (e.g., 1852) thus causing
access violation when the array fHandlerProc is accessed in
BasicTaskScheduler::turnOnBackgroundReadHandling().  The size of this table
is set to FD_SETSIZE (which is defined as 64 in winsock.h and 1024 in
Linux's equiv).

FDSETSIZE is defined as 64 even the latest Windows platform SDK.  Any idea
anyone?  A possible hack is to bump the array FHandlerProc and fClientData
to some big number, but how big?

Dan



From <EMAIL-ADDRESS-HIDDEN> Sat Sep 29 16:16:07 2001
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Date: Sat Sep 29 16:17:01 2001
X-Original-Date: Sat, 29 Sep 2001 16:15:43 -0700
Status: O

At 12:38 PM 9/29/01, you wrote:
>On Win XP (I'm using a beta copy, build 2600), it looks like the socket
>number returned by socket() is a large number (e.g., 1852) thus causing
>access violation when the array fHandlerProc is accessed in
>BasicTaskScheduler::turnOnBackgroundReadHandling().  The size of this table
>is set to FD_SETSIZE (which is defined as 64 in winsock.h and 1024 in
>Linux's equiv).
>
>FDSETSIZE is defined as 64 even the latest Windows platform SDK.  Any idea
>anyone?

Dan,

Yes, I believe the Windows header files are broken when they define 
FD_SETSIZE to be too low a value.  (This is apparently a problem with WinNT 
4.0 and probably Win2k as well, not just XP.)  I fixed this problem in the 
most recent library release (2001.09.25) by redefining FD_SETSIZE to be 
1024.  This is done in "BasicUsageEnvironment/BasicUsageEnvironment.cpp", 
before it does any #includes.

So, please download the latest version (2001.09.25) of the libraries.  If, 
for some reason, you still have problems even with this version, then let 
me know.

         Ross.




From <EMAIL-ADDRESS-HIDDEN> Sun Sep 30 19:06:28 2001
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Subject: [Live-devel] Any Vorbis Development
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Date: Sun Sep 30 19:07:01 2001
X-Original-Date: Sun, 30 Sep 2001 22:11:20 -0400
Status: O

Ross;

   Hello. Has there been any work on integrating the
vorbis format into the Live.com code ?

                                     Regards
                                     Marshall Eubanks


Marshall Eubanks

tme(at)21rst-century.com


From <EMAIL-ADDRESS-HIDDEN> Mon Oct 01 03:00:22 2001
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Date: Mon Oct  1 03:01:03 2001
X-Original-Date: Mon, 01 Oct 2001 02:56:04 -0700
Status: O

At 07:11 PM 9/30/01, Marshall Eubanks wrote:
>Ross;
>
>    Hello. Has there been any work on integrating the
>vorbis format into the Live.com code ?

Not yet, because a RTP payload format for Vorbis hasn't yet been nailed 
down.  There has been some discussion of this in the IETF, but the biggest 
stumbling block for streaming Vorbis in RTP is that each Vorbis stream can 
(potentially) use a different 'codebook' (a large data structure used for 
encoding/decoding), and these codebooks (unlike the rest of the Vorbis 
stream) would need to be transferred reliably from the transmitter to the 
(possibly many) receivers.

So, the streaming of a Vorbis stream will need to entail both (i) the 
reliable transfer (using some as-yet-unspecified protocol(s)) of the 
stream's codebook, and (ii) a RTP session (using some to-be-determined 
payload format).

(I've offered to help the Vorbis folks out with this, but haven't yet heard 
back from them.)

rOSS.



From <EMAIL-ADDRESS-HIDDEN> Mon Oct 01 03:45:48 2001
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Subject: Re: [Live-devel] Re: Any Vorbis Development
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Date: Mon Oct  1 03:46:01 2001
X-Original-Date: Mon, 1 Oct 2001 06:50:50 -0400
Status: O

>At 07:11 PM 9/30/01, Marshall Eubanks wrote:
>>Ross;
>>
>>    Hello. Has there been any work on integrating the
>>vorbis format into the Live.com code ?
>
>Not yet, because a RTP payload format for Vorbis hasn't yet been nailed 
>down.  There has been some discussion of this in the IETF, but the biggest

>stumbling block for streaming Vorbis in RTP is that each Vorbis stream can

>(potentially) use a different 'codebook' (a large data structure used for 

>encoding/decoding), and these codebooks (unlike the rest of the Vorbis 
>stream) would need to be transferred reliably from the transmitter to the 

>(possibly many) receivers.
>
>So, the streaming of a Vorbis stream will need to entail both (i) the 
>reliable transfer (using some as-yet-unspecified protocol(s)) of the 
>stream's codebook, and (ii) a RTP session (using some to-be-determined 
>payload format).
>
Dear Ross;

Thanks for the info. I have also been talking to Jack and Monte 
about this, apparently along similar lines.

Some background & info :

Vorbis uses a vector encoding in the fourier domain, 
as specified in the code book,
which can be thought of as a sort of truncated principal component analysis
of a sample stream. The code book
can be tuned, in principle, for any sound file and
any listener or set of listeners and any desired bit rate /
quality level. Vorbis also supports bit rate peeling, by which
a lower quality and bit rate can be derived from an
existing encoding by shortening the frame. 
Tests indicate that this is only
efficient over a fairly narrow bit rate range (order 20%);
if you really want to (say) halve the bit rate, it's better 
to use a different code-book tuned for the lower bit rate.

Vorbis codebooks can be 10-12 kilobytes in size, much bigger than the < 500
byte size 
of the typical vorbis frame. But the codebooks are themselves also vorbis frames,

and can be sent at any time. This is, for the naked format at present, literally
true;
each audio frame could in principle have its own unique codebook. 
If you are streaming files gathered from elsewhere, each song could easily have
a separate codebook.
Nothing stops you from re-encoding a file in your favorite codebook, though.


Vorbis is intended to be carried in the Ogg data format, which in
some ways has the functionality of / is a competitor for RTP.

So, as I see it, the outstanding issues are

1.) Is Vorbis to be streamed as ((Vorbis in Ogg) in RTP)
or as (Vorbis in RTP) ?  (It could be sent as (Vorbis in OGG), and there is

code for this for http streaming I believe). 
The simplist from the point of view of using existing code is the ((Vorbis in
Ogg) in RTP) solution.

2.) As you say, how are the codebooks to be maintained ?
Obviously, having a 10 kilobyte codebook sent even once per second means that
the bit rate is 80 kilobits 
per second before you send any actual data. My solution for this is just to
say that codebooks are always 
sent out of band. I think that each streaming broadcaster would set a series
of code-books that you would 
download (say with the preferred player), and would encode or re-encode everything
with those codebooks. 
From the standpoint of the user, codebooks could be treated like plug-ins -
a dialog box pops up if you 
need one you don't have, and asks if you want to download it.

I really think that the RTP spec could be forthcoming in short order if someone
wants to push it, and I 
know some people who are interested; I just wanted to see if there had been
any work done on it on this 
end.

                        Thanks
                        Marshall

>(I've offered to help the Vorbis folks out with this, but haven't yet heard

>back from them.)
>
>rOSS.
>
>
>_______________________________________________
>Live-devel mailing list
>Live-devel(at)lists.sourceforge.net
>https://lists.sourceforge.net/lists/listinfo/live-devel
>
>

Marshall Eubanks

tme(at)21rst-century.com


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Date: Tue Oct  2 19:43:02 2001
X-Original-Date: Tue, 2 Oct 2001 22:42:14 -0400
Status: O

--Apple-Mail-1-147414189
Content-Transfer-Encoding: 7bit
Content-Type: text/plain;
	charset=US-ASCII;
	format=flowed

Attached config.macosx, BasicUsageEnvironment/DelayQueue.hh

modifications to config.macosx from config.freebsd
line 1
	SOCKLEN_T=int
line 9
	add space after -o
line 12
	add space after -o
line 13
	removed -Bdynamic

Other source modifications

BasicUsageEnvironment/DelayQueue.hh
line 84
	cast return to (time_base)
line 87
	cast return to (time_base)

I'm working on a ProjectBuilder file to build the live library
as a MacOSX framework.  As soon as its ready I'll post it
here as well.

Marty Schoch
<mschoch(at)multicasttech.com>

--Apple-Mail-1-147414189
Content-Disposition: attachment;
	filename=config.macosx
Content-Transfer-Encoding: 7bit
Content-Type: application/octet-stream;
	x-unix-mode=0644;
	name="config.macosx"

COMPILE_OPTS =		-I. -DBSD=1 -O -DSOCKLEN_T=int
C =			c
C_COMPILER =		cc
C_FLAGS =		$(COMPILE_OPTS)
CPP =			cpp
CPLUSPLUS_COMPILER =	c++
CPLUSPLUS_FLAGS =	$(COMPILE_OPTS) -Wall
OBJ =			o
LINK =			c++ -o 
LINK_OPTS =		-L.
CONSOLE_LINK_OPTS =	$(LINK_OPTS)
LIBRARY_LINK =		ld -o 
LIBRARY_LINK_OPTS =	$(LINK_OPTS) -r 
LIB_SUFFIX =			a
LIBS_FOR_CONSOLE_APPLICATION =
LIBS_FOR_GUI_APPLICATION =
EXE =

--Apple-Mail-1-147414189
Content-Disposition: attachment;
	filename=DelayQueue.hh
Content-Transfer-Encoding: 7bit
Content-Type: application/octet-stream;
	x-unix-mode=0644;
	name="DelayQueue.hh"

/**********
This library is free software; you can redistribute it and/or modify it under
the terms of the GNU Lesser General Public License as published by the
Free Software Foundation; either version 2.1 of the License, or (at your
option) any later version. (See <http://www.gnu.org/copyleft/lesser.html>.)

This library is distributed in the hope that it will be useful, but WITHOUT
ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS
FOR A PARTICULAR PURPOSE.  See the GNU Lesser General Public License for
more details.

You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
**********/
 // Copyright (c) 1996-2000, Live Networks, Inc.  All rights reserved
// Delay queue
// C++ header

#ifndef _DELAY_QUEUE_HH
#define _DELAY_QUEUE_HH

#ifndef _LOCK_HH
#include "Lock.hh"
#endif

#if defined(__WIN32__) || defined(_WIN32)
#include <windows.h>
#else
#include <sys/time.h>
#endif

#ifdef TIME_BASE
typedef TIME_BASE time_base;
#else
typedef long time_base;
#endif

///// A "Timeval" can be either an absolute time, or a time interval /////

class Timeval {
public:
  time_base seconds() const {
    return fTv.tv_sec;
  }
  time_base seconds() {
    return fTv.tv_sec;
  }
  time_base useconds() const {
    return fTv.tv_usec;
  }
  time_base useconds() {
    return fTv.tv_usec;
  }
  
  int operator>=(Timeval const& arg2) const;
  int operator<=(Timeval const& arg2) const {
    return arg2 >= *this;
  }
  int operator<(Timeval const& arg2) const {
    return !(*this >= arg2);
  }
  int operator>(Timeval const& arg2) const {
    return arg2 < *this;
  }
  int operator==(Timeval const& arg2) const {
    return *this >= arg2 && arg2 >= *this;
  }
  int operator!=(Timeval const& arg2) const {
    return !(*this == arg2);
  }
  
  void operator+=(class DelayInterval const& arg2);
  void operator-=(class DelayInterval const& arg2);
  // returns ZERO iff arg2 >= arg1
  
protected:
  Timeval(time_base seconds, time_base useconds) {
    fTv.tv_sec = seconds; fTv.tv_usec = useconds;
  }
  
private:
  time_base& secs() {
    return (time_base)fTv.tv_sec;
  }
  time_base& usecs() {
    return (time_base)fTv.tv_usec;
  }
  
  struct timeval fTv;
};

#ifndef max
inline Timeval max(Timeval const& arg1, Timeval const& arg2) {
  return arg1 >= arg2 ? arg1 : arg2;
}
#endif
#ifndef min
inline Timeval min(Timeval const& arg1, Timeval const& arg2) {
  return arg1 <= arg2 ? arg1 : arg2;
}
#endif

class DelayInterval operator-(Timeval const& arg1, Timeval const& arg2);
// returns ZERO iff arg2 >= arg1


///// DelayInterval /////

class DelayInterval: public Timeval {
public:
  DelayInterval(time_base seconds, time_base useconds)
    : Timeval(seconds, useconds) {}
};

DelayInterval operator*(short arg1, DelayInterval const& arg2);

extern DelayInterval const ZERO;
extern DelayInterval const SECOND;
DelayInterval const MINUTE = 60*SECOND;
DelayInterval const HOUR = 60*MINUTE;
DelayInterval const DAY = 24*HOUR;

///// EventTime /////

class EventTime: public Timeval {
public:
  EventTime(unsigned secondsSinceEpoch = 0,
	    unsigned usecondsSinceEpoch = 0)
    // We use the Unix standard epoch: January 1, 1970
    : Timeval(secondsSinceEpoch, usecondsSinceEpoch) {}
};

EventTime TimeNow();

DelayInterval TimeRemainingUntil(EventTime const& futureEvent);
// Returns ZERO if "futureEvent" has already occurred.

extern EventTime const THE_END_OF_TIME;


///// DelayQueueEntry /////

class DelayQueueEntry {
public:
  int token() {
    return fToken;
  }

protected: // abstract base class
  DelayQueueEntry(DelayInterval delay);
  
  virtual void handleTimeout();
  
private:
  friend class DelayQueue;
  DelayQueueEntry* fNext;
  DelayQueueEntry* fPrev;
  DelayInterval fDeltaTimeRemaining;

  int fToken;
  static int tokenCounter;
};

///// DelayQueue /////

class DelayQueue: public DelayQueueEntry {
public:
  DelayQueue();
  virtual ~DelayQueue();
  
  void addEntry(DelayQueueEntry* newEntry); // returns a token for the entry
  void updateEntry(DelayQueueEntry* entry, DelayInterval newDelay);
  void updateEntry(int tokenToFind, DelayInterval newDelay);
  void removeEntry(DelayQueueEntry* entry); // but doesn't delete it
  DelayQueueEntry* removeEntry(int tokenToFind); // but doesn't delete it
  
  DelayInterval& timeToNextAlarm() {return head()->fDeltaTimeRemaining;}
  void handleAlarm();

private:
  DelayQueueEntry* head() const {
    return fNext;
  }
  DelayQueueEntry* findEntryByToken(int token);
  void addEntry1(DelayQueueEntry* newEntry);
  void removeEntry1(DelayQueueEntry* entry);
  
  EventTime fLastAlarmTime;
  RWLock fLock; // to protect against concurrent access
};

#endif

--Apple-Mail-1-147414189--



From <EMAIL-ADDRESS-HIDDEN> Wed Oct 03 03:41:55 2001
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Date: Wed Oct  3 03:42:03 2001
X-Original-Date: Wed, 03 Oct 2001 03:41:51 -0700
Status: O

Thanks Marty,

I've now installed a new version of the libraries that includes these 
changes (among others): 2001.10.03.  I'm not sure if it's shown up on 
SourceForge yet; if not, you can get it from <http://www.live.com/liveMedia/>


>Other source modifications
>
>BasicUsageEnvironment/DelayQueue.hh
>line 84
>         cast return to (time_base)
>line 87
>         cast return to (time_base)

Actually, these needed to be cast to (time_base&) - at least, in order for 
this to compile on my system.


>I'm working on a ProjectBuilder file to build the live library
>as a MacOSX framework.  As soon as its ready I'll post it
>here as well.

Will this file just be able to use the already-generated Makefile, or will 
this be an independent file that would need to be changed whenever the 
Makefile would have changed (ugh)?  (I'm not familiar with how 
ProjectBuilder works.)

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Wed Oct 03 14:02:53 2001
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Date: Wed Oct  3 14:03:02 2001
X-Original-Date: 03 Oct 2001 17:02:50 -0400
Status: O

On Wed, 2001-10-03 at 06:41, Ross Finlayson wrote:
 
> Actually, these needed to be cast to (time_base&) - at least, in order for 
> this to compile on my system.

Yes, this makes sense.
 
> >I'm working on a ProjectBuilder file to build the live library
> >as a MacOSX framework.  As soon as its ready I'll post it
> >here as well.
> 
> Will this file just be able to use the already-generated Makefile, or will 
> this be an independent file that would need to be changed whenever the 
> Makefile would have changed (ugh)?  (I'm not familiar with how 
> ProjectBuilder works.)

Unfortunately not.  Probably you won't want to role it into the
distribution.  But I may maintain it as best I can, and if the live
code reaches any milestone release it could be included.

....

I just built the 2001.10.03 release and it seems to build fine on os X.
I did run into a problem with the MediaSession.hh file not being copied
to the testProgs directory.  Easily fixed, but this leads me to the more
interesting quetion of why you copy the files in the first place?  As
opposed to adding something like -I../liveMedia to the build commands.

Marty Schoch
<mschoch(at)multicasttech.com>



From <EMAIL-ADDRESS-HIDDEN> Wed Oct 03 14:46:29 2001
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Date: Wed Oct  3 14:47:01 2001
X-Original-Date: Wed, 03 Oct 2001 14:43:14 -0700
Status: O

At 02:02 PM 10/3/01, Marty Schoch wrote:
>I just built the 2001.10.03 release and it seems to build fine on os X.
>I did run into a problem with the MediaSession.hh file not being copied
>to the testProgs directory.

Thanks for noting this; I've fixed it now.

>  Easily fixed, but this leads me to the more
>interesting quetion of why you copy the files in the first place?  As
>opposed to adding something like -I../liveMedia to the build commands.

Basically, because not all of the header files in the "liveMedia/" 
directory are intended to be usable by applications - some of them are just 
internal interfaces that are used within the "liveMedia" library only.  The 
header files that are copied (actually symlin ked) are those that represent 
the external interfaces.

(FYI, the "MediaSession" stuff is part of a RTSP client implementation, 
still in progress...)

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Mon Oct 08 12:04:15 2001
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Subject: [Live-devel] general development
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Date: Mon Oct  8 12:05:51 2001
X-Original-Date: 08 Oct 2001 15:03:52 -0400
Status: O


The RTSP / MediaSession stuff is very exciting.  In fact I'd like to
help out on some of this so it can become usable ASAP.  However, I have
general questions about coordinating development.  Since there is no CVS
tree, I'm reluctant to start coding without knowing what you're working
on.

I was looking at how the PriorityStreamSelector could be triggered by
an attribute in the SDP description...this part looks easy.  However,
I'm not exactly sure (since a lot of this code is skeletal) how the
multiple streams  (with one priority stream selector) will fit into the
linked list of substreams.  I have more questions, but I'll wait till I
hear back on this.

Another general question I have is what the policy/plans are for linking
to external libraries.  It looks to be fairly trivial to add support for
OggVorbis local files / http / rtp sink and source.  But since the ogg
format is a little bit more sophisticated than the basic mp3 file
format, it would be easiest to implement this by linking to libogg and
libvorbis.  This leads to many side issues such as licensing, cross
platform make files, etc.  Any thoughts on this?

Marty Schoch
<mschoch(at)multicasttech.com>



From <EMAIL-ADDRESS-HIDDEN> Mon Oct 08 14:06:33 2001
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Date: Mon Oct  8 14:07:27 2001
X-Original-Date: Mon, 08 Oct 2001 14:06:25 -0700
Status: O

At 12:03 PM 10/8/01, Marty Schoch wrote:
>The RTSP / MediaSession stuff is very exciting.  In fact I'd like to
>help out on some of this so it can become usable ASAP.  However, I have
>general questions about coordinating development.  Since there is no CVS
>tree, I'm reluctant to start coding without knowing what you're working
>on.

The "MediaSession" structure is intended to be a structure that represents 
a composite multimedia session, consisting (in general) of more than one 
subsession (e.g., audio + video).  It is initialized from a SDP description 
(and so contains a SDP parser).

I'm currently using this in combination with a RTSP client implementation 
("RTSPClient").  Together, these classes can be used to read a SDP 
description from a RTSP server, and then (using this) set up "RTPSource"s 
to receive the data streams.

Within a couple of days I'll be releasing (probably as one of the 
"testProgs") a RTSP client test program that opens and reads a "rtsp://..." 
URL, and dumps the received data to stdout (or to multiple output files if 
there's more than one stream).  This code will give you a clearer idea as 
to how all this stuff fits together, so you'll probably want to wait until 
then before hacking too much on this.


>I was looking at how the PriorityStreamSelector could be triggered by
>an attribute in the SDP description...this part looks easy.  However,
>I'm not exactly sure (since a lot of this code is skeletal) how the
>multiple streams  (with one priority stream selector) will fit into the
>linked list of substreams.

Probably the best way to handle this would be to let the MediaSession first 
construct a set of  multiple 'subsession's from the SDP description.  Then, 
once this is done, these multiple subsessions could be plugged into a 
"PrioritizedRTPStreamSelector", which would then replace the multiple 
subsessions in the linked list.


>Another general question I have is what the policy/plans are for linking
>to external libraries.  It looks to be fairly trivial to add support for
>OggVorbis local files / http / rtp sink and source.  But since the ogg
>format is a little bit more sophisticated than the basic mp3 file
>format, it would be easiest to implement this by linking to libogg and
>libvorbis.  This leads to many side issues such as licensing, cross
>platform make files, etc.  Any thoughts on this?

The "LIVE.COM Streaming Media" libraries are under the LGPL.  I want to 
keep the distributed library code as LGPL, so any future code that's 
bundled with the library distribution should also be LGPL.

However, because the code is LGPL, people can pretty much link the 
libraries with whatever code they like.  So, this doesn't preclude the 
library including some stubs for code that doesn't appear within the 
library code itself, but instead is implemented by some external library 
(which might have some other license, or may even be closed-source).  This 
is probably the best way to deal with Vorbis code, since parts of it are 
distributed under a BSD-like license.

         Ross.





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Subject: Re: [Live-devel] general development
From: Marty Schoch <EMAIL-ADDRESS-HIDDEN>
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Date: Tue Oct  9 12:59:06 2001
X-Original-Date: 09 Oct 2001 15:58:04 -0400
Status: O


I suppose my question was meant to be more generic.  Such as,
I want to hack on piece X, but don't want to unless I'm sure
that no one else is working on it.  In this specific case since
the code to parse lines in an SDP file was incomplete, I thought
about working on that.  But I thought there was a high likelihood
that you already had this code further along than the current release.

I guess without CVS this list is as good as any other method to perform
such coordination.

As for the external libraries, lets say we have some OggVorbisFile.h and
OggVorbisFile.cpp which reference the external libogg/libvorbis.  I
don't know of any mechanism in the current build process to
include/exclude parts of the package.  Obviously you want people to be
able to build the rest of the library even if they don't have libogg
installed.

I'm just curious how this is going to be handled before I write any such
code.

Marty Schoch
<mschoch(at)multicasttech.com>



From <EMAIL-ADDRESS-HIDDEN> Wed Oct 10 02:54:47 2001
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Date: Wed Oct 10 02:55:10 2001
X-Original-Date: Wed, 10 Oct 2001 02:52:12 -0700
Status: O

At 12:58 PM 10/9/01, Marty Schoch wrote:
>As for the external libraries, lets say we have some OggVorbisFile.h and
>OggVorbisFile.cpp which reference the external libogg/libvorbis.  I
>don't know of any mechanism in the current build process to
>include/exclude parts of the package.  Obviously you want people to be
>able to build the rest of the library even if they don't have libogg
>installed.

If there are any modules like this that rely upon external libraries, then 
perhaps the best thing to do with them would be to move them to a separate, 
optional library.  E.g., we could have a separate "liveMedia-vorbis" 
library that contains whatever new source/sink/filter modules rely upon 
external Vorbis libraries.  Only those people who choose to use these new 
modules would find they'll also need to link with the external libraries.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Wed Oct 10 10:29:39 2001
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Subject: [Live-devel] A new RTSP client test program: "openRTSP"
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Date: Wed Oct 10 10:30:10 2001
X-Original-Date: Wed, 10 Oct 2001 10:29:31 -0700
Status: O

I'm pleased to announce the availability of a new RTSP client test program 
called "openRTSP".  The source code for this program is now bundled as one 
of the "testProgs" in the LIVE.COM Streaming Media distribution: 
<http://live.sourceforge.net/>.  (For the next few days, I'll also have a 
pre-built Linux/x86 binary available online: <http://www.live.com/openRTSP>)

I will shortly be putting up a web page that describes how to use a the 
program; here's a brief overview:

You can run this program as follows:
	openRTSP <url>

where <url> is the RTSP URL to open (i.e., beginning with 
"rtsp://...").  The program will open the give URL (using RTSP's "DESCRIBE" 
command), and then, for each audio/video subsession whose RTP payload 
format it understands, "SETUP" and "PLAY" the subsession.  The extracted 
data for each subsession is written into a separate output file.  E.g., if 
the session contains a MPEG audio subsession (RTP payload type 14) - e.g., 
MP3, then this will be extracted and written to a file named "audio-MPA".

By default, the program will ignore any subsession whose payload format it 
doesn't understand (because if it doesn't know the RTP payload format, it 
doesn't know how to extract data from the incoming RTP stream).  However, 
if you want the program to *play* the RTSP session, but not actually 
*receive* it, you can do so by giving the program the "-r" ('don't 
receive') flag.  This is useful if you have some other application (on the 
same machine) that actually receives the incoming media session(s).

If you use the "-r" option, you'll probably also want to use the "-p 
<startingPortNumber>" option.  This tells the program which client port 
numbers to use in the RTSP "SETUP" commands - i.e., which ports the server 
should send to.  (Without the "-r" option, the program receives the streams 
itself, and uses its own ephemeral port numbers for 
this.)  <startingPortNumber> must be an even number.

E.g., if the RTSP session consists of an audio and a video subsession 
(listed in that order in the returned SDP description), then "-p 6666" will 
cause ports 6666 and 6667 to be used for the audio subsession, and ports 
6668 and 6669 to be used for the video subsession.

You can also give the program the "-V" option, which will give you more 
debugging output (you'll get to see each of the RTSP requests and responses 
in detail).

Limitations (these will all be fixed soon):
- The program currently works only for unicast sessions, not multicast.
	(If anyone has a "rtsp://..." URL for a (publically available) multicast
	 session, please let me know, so I can use this for testing.)
- If the program is receiving the media streams itself (i.e., you don't use 
the "-r" option), then it doesn't send back (or listen to) RTCP reports.
- The program stays running forever, even if the media session(s) run for 
only a finite length of time (e.g., if the server is playing a fixed-length 
audio or video clip).  (If you kill the running program with "kill -1" 
(SIGHUP), it will cleanly shut down each media subsession  - using the RTSP 
"TEARDOWN" command.)

Also, if you see a RTP payload format that the program doesn't support, but 
that you'd like it to, let me know.  (Support for simple RTP payload 
formats - that don't have funky extra headers, or do interleaving, etc. - 
can be added easily.)

The program makes use of the new "RTSPClient" and "MediaSession" modules - 
available in the "liveMedia" library.  The "openRTSP" code (in "testProgs") 
gives a good illustration of how to ue these modules.

	Ross. 



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Date: Sat Oct 13 04:03:05 2001
X-Original-Date: Sat, 13 Oct 2001 19:05:23 +0800
Status: O

All,

Is there any documentation of the API for this library?
I have seen whats at live.sourceforge.net.

Ideally what Im looking for is the type of documentation
produced by using tools such as doxygen or doc++.


From <EMAIL-ADDRESS-HIDDEN> Sat Oct 13 08:41:45 2001
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Date: Sat Oct 13 08:42:06 2001
X-Original-Date: Sat, 13 Oct 2001 08:41:12 -0700
Status: O

At 04:05 AM 10/13/01, Chad Williams wrote:
>Is there any documentation of the API for this library?
>I have seen whats at live.sourceforge.net.
>
>Ideally what Im looking for is the type of documentation
>produced by using tools such as doxygen or doc++.

Chad,

At some point (in the not too distant future) I definitely want to produce 
some real documentation for the libraries (especially now that the APIs 
seem to be stabilizing), but right now my first priority is improving the 
libraries' functionality (e.g., by supporting more codecs/payload formats, 
and by improving the RTSP implementation).

For now, the best way to understand how to use the libraries is to (i) 
study the example programs in the "testProgs" directory, and especially 
(ii) ask questions on this mailing list.

You noted a couple of C/C++ documentation systems ("doxygen" and "doc++") 
that might be useful to use.  I haven't used either of these, so I'd be 
interested to hear opinions from the list as to the relative merits of each 
(and other relevant documentation systems).

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>




From <EMAIL-ADDRESS-HIDDEN> Mon Oct 22 11:15:52 2001
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From: Don Molaro <EMAIL-ADDRESS-HIDDEN>
Organization: Sony Electronics Inc.
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Date: Mon Oct 22 11:16:20 2001
X-Original-Date: Mon, 22 Oct 2001 11:17:33 -0700
Status: O

While the multicast stuff is very cool we have a requirement to receive
a RTP MP3 session that has been unicast rather than multicast.
Does anyone have an example (or documentation) that shows this ?

--
Donald Molaro
Interactive Television Development, Sony Electronics Inc.
email:dmolaro(at)itvd.sel.sony.com     voice: 408.955.4936




From <EMAIL-ADDRESS-HIDDEN> Mon Oct 22 12:19:56 2001
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Date: Mon Oct 22 12:20:27 2001
X-Original-Date: Mon, 22 Oct 2001 12:19:27 -0700
Status: O

At 11:17 AM 10/22/01, Don Molaro wrote:
>While the multicast stuff is very cool we have a requirement to receive
>a RTP MP3 session that has been unicast rather than multicast.
>Does anyone have an example (or documentation) that shows this ?

Donald,

Although the code was designed initially for multicast streams, it will 
work just as well for unicast RTP streams.

E.g., in the code for "testMP3Receiver" 
("testProgs/testMP3Receiver.cpp"),  you can do the following:
- set "rtpPortNum" to be the number of the port on which you wish to 
*receive* RTP (in the example code, it's 8888)
- set "multicastAddressStr" to "0.0.0.0"
         (because you're not joining a multicast group in this case)

This will let you receive a unicast RTP stream sent to port 8888.  You will 
also receive RTCP reports (on port 8889).

Now, if you also want to send RTCP packets (e.g., RTCP Receiver Reports) 
*back* to the streaming server, then you will also need to do the 
following.  (In this example, suppose that the streaming server has IP 
address "10.20.30.40" and uses port 6667 for RTCP.):

     struct in_addr serverAddress;
     serverAddress.s_addr = our_inet_addr("10.20.30.40");
     rtcpGroupsock.changeDestinationParameters(serverAddress, 6667, 255);

(Note that you'll need the latest version of the library code to use 
"changeDestinationParameters()")

Please let me know if this works OK for you.

         Ross.

ps. You can also use the "playRTPMPEG" tool 
<http://www.live.com/multikit/playRTPMPEG.html> to receive MP3/RTP streams 
(multicast *or* unicast).  (This application was built from the "LIVE.COM 
Streaming Media" libraries.)

e.g., to receive a MP3/RTP unicast session on port 8888, you can run
         playRTPMPEG - 0.0.0.0/8888 255
This will output the received MP3 data to 'stdout'.
(Unfortunately you can't currently won't be able to send back unicast RTCP 
reports using "playRTPMPEG".)



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Date: Mon Oct 22 12:34:08 2001
X-Original-Date: Mon, 22 Oct 2001 12:34:05 -0700
Status: O

One thing to consider for unicast.  In the case of unicast, the receiver 
needs to be the one in charge of RTP port selection.  I imagine most RTSP 
player implementations will fail if the server tried to mandate a port.

At 12:19 PM 10/22/01 -0700, Ross Finlayson wrote:
>At 11:17 AM 10/22/01, Don Molaro wrote:
>>While the multicast stuff is very cool we have a requirement to receive
>>a RTP MP3 session that has been unicast rather than multicast.
>>Does anyone have an example (or documentation) that shows this ?
>
>Donald,
>
>Although the code was designed initially for multicast streams, it will 
>work just as well for unicast RTP streams.
>
>E.g., in the code for "testMP3Receiver" 
>("testProgs/testMP3Receiver.cpp"),  you can do the following:
>- set "rtpPortNum" to be the number of the port on which you wish to 
>*receive* RTP (in the example code, it's 8888)
>- set "multicastAddressStr" to "0.0.0.0"
>         (because you're not joining a multicast group in this case)
>
>This will let you receive a unicast RTP stream sent to port 8888.  You 
>will also receive RTCP reports (on port 8889).
>
>Now, if you also want to send RTCP packets (e.g., RTCP Receiver Reports) 
>*back* to the streaming server, then you will also need to do the 
>following.  (In this example, suppose that the streaming server has IP 
>address "10.20.30.40" and uses port 6667 for RTCP.):
>
>     struct in_addr serverAddress;
>     serverAddress.s_addr = our_inet_addr("10.20.30.40");
>     rtcpGroupsock.changeDestinationParameters(serverAddress, 6667, 255);
>
>(Note that you'll need the latest version of the library code to use 
>"changeDestinationParameters()")
>
>Please let me know if this works OK for you.
>
>         Ross.
>
>ps. You can also use the "playRTPMPEG" tool 
><http://www.live.com/multikit/playRTPMPEG.html> to receive MP3/RTP streams 
>(multicast *or* unicast).  (This application was built from the "LIVE.COM 
>Streaming Media" libraries.)
>
>e.g., to receive a MP3/RTP unicast session on port 8888, you can run
>         playRTPMPEG - 0.0.0.0/8888 255
>This will output the received MP3 data to 'stdout'.
>(Unfortunately you can't currently won't be able to send back unicast RTCP 
>reports using "playRTPMPEG".)
>
>
>_______________________________________________
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Subject: [Live-devel] Problem in reciving stream from realserver
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Date: Mon Oct 22 13:55:07 2001
X-Original-Date: Mon, 22 Oct 2001 13:56:03 -0700
Status: O

I've got a problem with receiving MP3 stream from realserver with test
programs.
Ok I've got the unicast issue worked out. The issue now is getting the receiver
object to actually process the data. Using the openRTSP program to setup the
stream and using a hacked up version of the testMP3Receiver program set for
unicast (thanks Ross) I get the following behavior out of the system:

What it looks like to my untrained eye is that the liveMedia library does not
understand the mime type that comes back from the server (which is
audio/X-MP3-DRAFT-00) it looks like it is expecting audio/mpeg.

I set the debugging flag in the groupsock library so I know that data is coming
back from the server but it does not seem to get through some part of library.
Any clues would be appreciated. Can I convince the library that it really wants
to understand (audio/X-MP3-DRAFT-00) or am I headed off in the wrong
direction..

 Don.
------------------------------

[don(at)kirkland testProgs]$ ./openRTSP -r -V -p 8888
rtsp://43.134.39.230/realmp3.mp3
Sending request: DESCRIBE rtsp://43.134.39.230/realmp3.mp3 RTSP/1.0
CSeq: 1
Accept: application/sdp


Received DESCRIBE response: RTSP/1.0 200 OK
CSeq: 1
Date: Mon, 22 Oct 2001 19:24:28 GMT
Set-Cookie:
cbid=jfdgihelhjjkgldmeoprmppqmojrktlufkfgkielmjgkjiplosmrmpqqnrrsdtlugfejdhcl;path=/;expires=Thu,31-Dec-2037
23:59:59 GMT
vsrc:
http://43.134.39.230:8080/viewsource/template.html?nuyhtggaysz6nc6vsffjsqt69pmc20710000400000nvww370Cjx20buu7se

Last-Modified: Thu, 18 Oct 2001 20:31:40 GMT
Content-base: rtsp://43.134.39.230/realmp3.mp3/
Content-type: application/sdp
Content-length: 651

v=0
o=- 1003433500 1003433500 IN IP4 43.134.39.230
s=realmp3
i=<No author> <No copyright>
c=IN IP4 0.0.0.0
t=0 0
a=SdpplinVersion:1610643188
a=StreamCount:integer;1
a=Title:buffer;"cmVhbG1wMwA="
a=range:npt=0-39.471000
m=audio 0 RTP/AVP 101
b=AS:128
a=control:streamid=0
a=range:npt=0-39.471000
a=length:npt=39.471000
a=rtpmap:101 X-MP3-draft-00
a=mimetype:string;"audio/X-MP3-draft-00"
a=StartTime:integer;0
a=AvgBitRate:integer;128000
a=SampleRate:integer;44100
a=AvgPacketSize:integer;417
a=Preroll:integer;1000
a=NumChannels:integer;2
a=MaxPacketSize:integer;1024
a=ASMRuleBook:string;"AverageBandwidth=128000, AverageBandwidthStd=0,
Priority=9;"

Opened URL "rtsp://43.134.39.230/realmp3.mp3", returning a SDP description:
v=0
o=- 1003433500 1003433500 IN IP4 43.134.39.230
s=realmp3
i=<No author> <No copyright>
c=IN IP4 0.0.0.0
t=0 0
a=SdpplinVersion:1610643188
a=StreamCount:integer;1
a=Title:buffer;"cmVhbG1wMwA="
a=range:npt=0-39.471000
m=audio 0 RTP/AVP 101
b=AS:128
a=control:streamid=0
a=range:npt=0-39.471000
a=length:npt=39.471000
a=rtpmap:101 X-MP3-draft-00
a=mimetype:string;"audio/X-MP3-draft-00"
a=StartTime:integer;0
a=AvgBitRate:integer;128000
a=SampleRate:integer;44100
a=AvgPacketSize:integer;417
a=Preroll:integer;1000
a=NumChannels:integer;2
a=MaxPacketSize:integer;1024
a=ASMRuleBook:string;"AverageBandwidth=128000, AverageBandwidthStd=0,
Priority=9;"

Sending request: SETUP rtsp://43.134.39.230/realmp3.mp3/streamid=0 RTSP/1.0
CSeq: 2
Transport: RTP/AVP;unicast;client_port=8888-8889


Received SETUP response: RTSP/1.0 200 OK
CSeq: 2
Date: Mon, 22 Oct 2001 19:24:28 GMT
Session: 29562-2
RDTFeatureLevel: 2
Transport: rtp/avp;unicast;client_port=8888-8889;server_port=30614-30615


Issued RTSP "SETUP" on "audio/X-MP3-DRAFT-00" subsession (ports 8888-8889)
Sending request: PLAY rtsp://43.134.39.230/realmp3.mp3/streamid=0 RTSP/1.0
CSeq: 3
Session: 29562-2
Range: npt=0-


Received PLAY response: RTSP/1.0 200 OK
CSeq: 3
Date: Mon, 22 Oct 2001 19:24:28 GMT
RTP-Info: url=rtsp://43.134.39.230/realmp3.mp3/streamid=0;seq=0;rtptime=0
Issued RTSP "PLAY" command on "audio/X-MP3-DRAFT-00" subsession
Data is being streamed (for 44.5 seconds)...

--------------------------------------------
..............Data shows up in the streamReceiver program
here...................
-------------------------------------------
13:37:05 Groupsock(3: 0.0.0.0, 8888, 1): created
13:37:05 Groupsock(5: 0.0.0.0, 8889, 1): created
13:37:08 Groupsock(5: 0.0.0.0, 8889, 1): wrote 28 bytes, ttl 1
13:37:08 Groupsock(5: 0.0.0.0, 8889, 1): read 28 bytes from 127.0.0.1
13:37:08 Groupsock(3: 0.0.0.0, 8888, 1): read 421 bytes from 43.134.39.230
13:37:08 Groupsock(3: 0.0.0.0, 8888, 1): read 416 bytes from 43.134.39.230
13:37:08 Groupsock(5: 0.0.0.0, 8889, 1): read 48 bytes from 43.134.39.230
13:37:11 Groupsock(3: 0.0.0.0, 8888, 1): read 570 bytes from 43.134.39.230
............... Lot's o stuff deleted......................
13:37:14 Groupsock(5: 0.0.0.0, 8889, 1): read 28 bytes from 127.0.0.1
13:37:14 Groupsock(3: 0.0.0.0, 8888, 1): read 453 bytes from 43.134.39.230
13:37:14 Groupsock(3: 0.0.0.0, 8888, 1): read 485 bytes from 43.134.39.230
13:37:54 Groupsock(5: 0.0.0.0, 8889, 1): wrote 28 bytes, ttl 1
13:37:54 Groupsock(5: 0.0.0.0, 8889, 1): read 28 bytes from 127.0.0.1
13:37:59 Groupsock(5: 0.0.0.0, 8889, 1): wrote 28 bytes, ttl 1
13:37:59 Groupsock(5: 0.0.0.0, 8889, 1): read 28 bytes from 127.0.0.1
------------------------------------------------------------------
Closing "audio/X-MP3-DRAFT-00" subsession
Sending request: TEARDOWN rtsp://43.134.39.230/realmp3.mp3/streamid=0 RTSP/1.0
CSeq: 4
Session: 29562-2


[don(at)kirkland testProgs]$
--
Donald Molaro
Interactive Television Development, Sony Electronics Inc.
email:dmolaro(at)itvd.sel.sony.com     voice: 408.955.4936




From <EMAIL-ADDRESS-HIDDEN> Mon Oct 22 14:19:29 2001
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Date: Mon Oct 22 14:20:07 2001
X-Original-Date: Mon, 22 Oct 2001 14:18:51 -0700
Status: O

At 01:56 PM 10/22/01, Don Molaro wrote:
>I've got a problem with receiving MP3 stream from realserver with test
>programs.
>Ok I've got the unicast issue worked out. The issue now is getting the 
>receiver
>object to actually process the data. Using the openRTSP program to setup the
>stream and using a hacked up version of the testMP3Receiver program set for
>unicast (thanks Ross) I get the following behavior out of the system:
>
>What it looks like to my untrained eye is that the liveMedia library does not
>understand the mime type that comes back from the server (which is
>audio/X-MP3-DRAFT-00) it looks like it is expecting audio/mpeg.

Don,

Unfortunately the MIME type - "audio/X-MP3-DRAFT-00" - that your RTSP 
server is using for its 'MP3' stream is an undocumented payload format 
that's used only by RealNetworks.  (I've been told that it's an early 
version of the "audio/MPA-ROBUST" format that I define in RFC 3119, but I 
don't think it's compatible with that.)  So, I don't know how to deal with 
this data.

Is there any way to tell your RTSP server to use the "audio/MPEG" MIME type 
- i.e., the RTP payload format defined in RFC 2250?  If so, then you will 
be able to receive this stream not only with "testMP3Receiver", but also 
with "openRTSP" itself (without the "-r" flag).


>I set the debugging flag in the groupsock library so I know that data is 
>coming
>back from the server but it does not seem to get through some part of library.

That's because the incoming RTP stream is using a payload format that the 
library doesn't understand.  ("testMP3Reciever" uses a "MPEGAudioRTPSource" 
object, which expects a payload format number of 14, formatted as described 
in RFC 2250.)

>Any clues would be appreciated. Can I convince the library that it really 
>wants
>to understand (audio/X-MP3-DRAFT-00)

No, not unless RealNetworks were to release documentation that describes 
this (non-standard) payload format.  As noted above, the best solution is 
to convince the server - if you can - to use "audio/MPEG" (or 
"audio/MPA-ROBUST").

         Ross.



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Date: Tue Oct 23 00:01:06 2001
X-Original-Date: Tue, 23 Oct 2001 00:00:25 -0700 (PDT)
Status: O

On Mon, 22 Oct 2001, Ross Finlayson wrote:
> Is there any way to tell your RTSP server to use the "audio/MPEG" MIME type
> - i.e., the RTP payload format defined in RFC 2250?  If so, then you will
> be able to receive this stream not only with "testMP3Receiver", but also
> with "openRTSP" itself (without the "-r" flag).

Not possible with the current version of RealServer.

> >Any clues would be appreciated. Can I convince the library that it really
> >wants to understand (audio/X-MP3-DRAFT-00)
>
> No, not unless RealNetworks were to release documentation that describes
> this (non-standard) payload format.

I spoke with the developer on our current solution, who provided some
details.  It's actually not based off of draft-00, as I once thought, and
as the name implies (the name is an historical indication of where we
started).

It's just one ADU frame per packet.  That's it.  If you want to use the
Live SDK to decode these, it is just like non-interleaved 3119 without
fragmentation or headers.  The continuation flag is never set, and the
header is unnecessary - ADU size is always just the packet data size.

Hope this helps,
Rob



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Date: Tue Oct 23 01:29:06 2001
X-Original-Date: Tue, 23 Oct 2001 01:28:03 -0700
Status: O

At 12:00 AM 10/23/01, Rob Lanphier wrote:
> > >Any clues would be appreciated. Can I convince the library that it really
> > >wants to understand (audio/X-MP3-DRAFT-00)
> >
> > No, not unless RealNetworks were to release documentation that describes
> > this (non-standard) payload format.
>
>I spoke with the developer on our current solution, who provided some
>details.  It's actually not based off of draft-00, as I once thought, and
>as the name implies (the name is an historical indication of where we
>started).
>
>It's just one ADU frame per packet.  That's it.  If you want to use the
>Live SDK to decode these, it is just like non-interleaved 3119 without
>fragmentation or headers.  The continuation flag is never set, and the
>header is unnecessary - ADU size is always just the packet data size.

OK, if that's the case then I should be able to support this payload format 
very easily, as it's just a simplified version of the "audio/mpa-robust" 
(RFC 3119) payload format that I already implement.

Can anyone point me at a publically-accessible RTSP server (serving 
"audio/X-MP3-DRAFT-00") that I can use for testing?

         Ross.



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Date: Tue Oct 23 09:09:01 2001
X-Original-Date: Tue, 23 Oct 2001 09:08:12 -0700 (PDT)
Status: O

On Tue, 23 Oct 2001, Ross Finlayson wrote:
> OK, if that's the case then I should be able to support this payload format
> very easily, as it's just a simplified version of the "audio/mpa-robust"
> (RFC 3119) payload format that I already implement.
>
> Can anyone point me at a publically-accessible RTSP server (serving
> "audio/X-MP3-DRAFT-00") that I can use for testing?

I'll see what I can do (I don't know of any that are out there on our site
right now).  In the meantime, you can download a free server here:

http://proforma.real.com/rnforms/products/servers/eval/?ulf=bas

Alternately, the file "realmp3.mp3" exists as a default download on our
servers, and so a number of sites never delete this.  Here's one:

rtsp://media.schreiner.edu/realmp3.mp3

Hope this helps,
Rob




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Date: Tue Oct 23 19:55:06 2001
X-Original-Date: Tue, 23 Oct 2001 19:52:13 -0700
Status: O

Thanks to Rob Lanphier for his help in explaining the mystery of 
"audio/X-MP3-DRAFT-00".

FYI, I have now implemented this payload type in the "openRTSP" client, so 
- for example - you can now run:
	openRTSP -a rtsp://media.schreiner.edu/realmp3.mp3 > foo.mp3

This reads the stream from the specified RTSP URL, and outputs the 
resulting MP3 data to the file "foo.mp3".

The "-a" flag to openRTSP means "output the audio stream only, to 
stdout".  (I'll shortly be putting up a web page that describes all of 
"openRTSP"s flags in detail.)

	Ross.



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Date: Tue Oct 23 20:32:01 2001
X-Original-Date: Tue, 23 Oct 2001 20:29:24 -0700
Status: O

One more thing about this...

As I noted earlier, you can now use "openRTSP" to receive 
"audio/X-MP3-DRAFT-00" data directly.

If, however, you still want to use the "testMP3Receiver" test program to do 
this, then you can do so by making the following modifications to 
"testMP3Receiver.cpp":

1/ *Un*comment line 34, to get:
	#define STREAM_USING_ADUS 1
2/ Change line 102 from
	= MP3ADURTPSource::createNew(*env, &rtpGroupsock, rtpPayloadFormat);
to
	= SimpleRTPSource::createNew(*env, &rtpGroupsock, rtpPayloadFormat,
						"audio/mpa-robust" /*hack*/);
3/ Delete lines 119 through 125 (the stuff beginning "Add a filter that 
deinterleaves...")
4/ Change line 129 from
	= MP3FromADUSource::createNew(*env, sessionState.source);
to
	= MP3FromADUSource::createNew(*env, sessionState.source, False);

Of course, you will probably also need to change the RTP payload format 
number at line 100.

	Ross. 



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Date: Wed Oct 24 10:27:03 2001
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Status: O

This is a multi-part message in MIME format.

------=_NextPart_000_0007_01C15C8F.3AB7B040
Content-Type: text/plain;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

I am testing the program on windows 2000, with test programs for audio
Mp3 files.For some reason the client program goes into a loop.I haven't
changes anything from the downloaded version.
=20
Thanks
Mahua

------=_NextPart_000_0007_01C15C8F.3AB7B040
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	charset="iso-8859-1"
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>

<META content=3D"MSHTML 5.50.4134.600" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>I am testing the program on windows =
2000, with test=20
programs for audio Mp3 files.For some reason the client program goes =
into a=20
loop.I haven't changes anything from the downloaded =
version.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Thanks</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Mahua</FONT></DIV></BODY></HTML>

------=_NextPart_000_0007_01C15C8F.3AB7B040--


From <EMAIL-ADDRESS-HIDDEN> Wed Oct 24 10:51:16 2001
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Date: Wed Oct 24 10:52:02 2001
X-Original-Date: Wed, 24 Oct 2001 10:46:20 -0700
Status: O

At 10:24 AM 10/24/01, Mahua De wrote:
>I am testing the program on windows 2000, with test programs for audio Mp3 
>files.For some reason the client program goes into a loop.I haven't 
>changes anything from the downloaded version.

Mahua,

Thanks for the report.

Which particular test program are you running, and where is it going into a 
loop?  (What is the program outputting?)

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Fri Oct 26 03:20:47 2001
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Date: Fri Oct 26 03:21:02 2001
X-Original-Date: Fri, 26 Oct 2001 03:19:32 -0700
Status: O

FYI, I have now put up a web site with full documentation of the "openRTSP" 
command-line RTSP client tool:
	<http://www.live.com/openRTSP/>
This program lets you open, stream, receive, and (optionally) record media 
streams that are specified by a "rtsp://" URL.

The source code for this program can be found as "openRTSP.cpp" in the 
"testProgs" directory.

(Many thanks to HorizonLive, Plustream, and Multicast Technologies for 
contributing funding towards the development of this software.)

	Ross.



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Date: Mon Oct 29 15:14:01 2001
X-Original-Date: Mon, 29 Oct 2001 18:26:19 -0500
Status: O

I am having problem trying to run the Test programs testMP3Streamer
,testMP3Receiver on different host machines within an intranet network.The
testMP3Receiver is able to receive only the first packet and after that the
wasLoopedBackFromUs() in GroupSock.cpp returns true always.

However, if run both the test programs testMP3Streamer and testMP3Receiver
on the same host machine, the testMP3Receiver  program successfully receives
all the packets.

Thanks 



From <EMAIL-ADDRESS-HIDDEN> Tue Oct 30 01:15:01 2001
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Date: Tue Oct 30 01:15:04 2001
X-Original-Date: Tue, 30 Oct 2001 01:11:39 -0800
Status: O

At 03:26 PM 10/29/01, Mahua De wrote:
>I am having problem trying to run the Test programs testMP3Streamer
>,testMP3Receiver on different host machines within an intranet network.The
>testMP3Receiver is able to receive only the first packet and after that the
>wasLoopedBackFromUs() in GroupSock.cpp returns true always.
>
>However, if run both the test programs testMP3Streamer and testMP3Receiver
>on the same host machine, the testMP3Receiver  program successfully receives
>all the packets.

Mahua,

No, I haven't seen this problem.  What operating systems are you running on 
each machine?  If you're running different OSs on each machine, does it 
make a difference if you reverse which one runs "testMP3Streamer" and which 
one runs "testMP3Receiver"?

You might also try increasing the TTL of outgoing multicast packets (in 
"testMP3Streamer"), in case that has something to do with the problem.  On 
line 114 of "testMP3Streamer.cpp", where it says
         const unsigned char ttl = 1
try changing 1 to 255.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Tue Oct 30 16:10:35 2001
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Date: Tue Oct 30 16:11:02 2001
X-Original-Date: Tue, 30 Oct 2001 16:10:33 -0800
Status: O

I downloaded the source from http://live.sourceforge.net/ today but got 
the following error when building on MacOSX:

% make
cd liveMedia; make
...
c++ -c -I. -DBSD=1 -O -DSOCKLEN_T=int -Wall 
PrioritizedRTPStreamSelector.cpp
c++: PrioritizedRTPStreamSelector.cpp: No such file or directory
c++: No input files
make[1]: *** [PrioritizedRTPStreamSelector.o] Error 1
make: *** [liveMedia/libliveMedia.a] Error 2




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Date: Tue Oct 30 16:44:01 2001
X-Original-Date: Tue, 30 Oct 2001 16:42:58 -0800
Status: O

At 04:10 PM 10/30/01, Steve Dekorte wrote:

>I downloaded the source from http://live.sourceforge.net/ today but got 
>the following error when building on MacOSX:
>
>% make
>cd liveMedia; make
>...
>c++ -c -I. -DBSD=1 -O -DSOCKLEN_T=int -Wall PrioritizedRTPStreamSelector.cpp
>c++: PrioritizedRTPStreamSelector.cpp: No such file or directory
>c++: No input files
>make[1]: *** [PrioritizedRTPStreamSelector.o] Error 1
>make: *** [liveMedia/libliveMedia.a] Error 2

Steve,

That's strange - the "liveMedia" directory *should* contain a files named 
"PrioritizedRTPStreamSelector.cpp" and "PrioritizedRTPStreamSelector.hh" - 
they're included in the "tar" file.

However, the tar file also includes two (unused) files named 
"PrioritizedRTPStreamSelector.cpp.save" and 
"PrioritizedRTPStreamSelector.hh.save".  Perhaps - for some reason - when 
you extracted the tar file these two '.save' files ended up replacing the 
good '.cpp' and '.hh' files?  Could this be some MacOS X filename 
extension-related issue?

Anyway, I have left copies of "PrioritizedRTPStreamSelector.cpp" and 
"PrioritizedRTPStreamSelector.hh" online at 
<http://www.live.com/liveMedia/>, so you can get them from there if you 
don't have them.

         Ross Finlayson
         LIVE.COM



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Subject: [Live-devel] PrioritizedRTPStreamSelector.cp?
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Date: Tue Oct 30 17:19:02 2001
X-Original-Date: Tue, 30 Oct 2001 17:18:47 -0800
Status: O

Ross Finlayson wrote:
> Steve,
>
> That's strange - the "liveMedia" directory *should* contain a files 
> named "PrioritizedRTPStreamSelector.cpp" and 
> "PrioritizedRTPStreamSelector.hh" - they're included in the "tar" file.

I took a look and noticed it had the wrong extension:

% ls PrioritizedRTPStreamSelector*
PrioritizedRTPStreamSelector.cp                
PrioritizedRTPStreamSelector.hh

After changing it to cpp the compile finished successfully.

Thanks for the quick response,
Steve




From <EMAIL-ADDRESS-HIDDEN> Tue Oct 30 19:35:58 2001
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Date: Tue Oct 30 19:36:01 2001
X-Original-Date: Tue, 30 Oct 2001 22:35:45 -0500
Status: O

I also encountered this problem when first porting the software to OSX.
Turns out Stuffit Expander does not always do things properly for tar.gz 
files.

Ever since I've used gnu tar/gz and that has solved everything.

Marty Schoch
<mschoch(at)multicasttech.com>

On Tuesday, October 30, 2001, at 08:18 PM, Steve Dekorte wrote:

>
> Ross Finlayson wrote:
>> Steve,
>>
>> That's strange - the "liveMedia" directory *should* contain a files 
>> named "PrioritizedRTPStreamSelector.cpp" and 
>> "PrioritizedRTPStreamSelector.hh" - they're included in the "tar" file.
>
> I took a look and noticed it had the wrong extension:
>
> % ls PrioritizedRTPStreamSelector*
> PrioritizedRTPStreamSelector.cp                
> PrioritizedRTPStreamSelector.hh
>
> After changing it to cpp the compile finished successfully.
>
> Thanks for the quick response,
> Steve
>
>
>
> _______________________________________________
> Live-devel mailing list
> Live-devel(at)lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/live-devel



From <EMAIL-ADDRESS-HIDDEN> Wed Oct 31 02:17:17 2001
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Date: Wed Oct 31 02:18:02 2001
X-Original-Date: Wed, 31 Oct 2001 02:15:10 -0800
Status: O

I have just released a new version of the code, with the following (minor) 
changes:

1/ In the "TaskScheduler" class (in the "UsageEnvironment" library), the 
signature of the "blockMyself()" function is now:
	virtual void blockMyself(char* watchVariable = NULL)
"watchVariable" - if not NULL - specifies the address of a byte whose value 
is checked within the event loop.  If this byte gets set to anything other 
than zero, blockMyself() returns.  This gives programmers more control over 
when/how to exit the event loop.

2/ I made the SDP parsing code in "liveMedia"/"MediaSession" more robust to 
SDP descriptions that (erroneously) don't specify a RTP timestamp frequency 
in "rtpmap" attributes.  (Both RealServer and (at least older versions of) 
the Darwin server have this problem.)

I have also released new "openRTSP" binaries that include this change.  If 
you're using openRTSP to generate QuickTime files, then you should upgrade 
to the newest version.

3/  I made some cosmetic changes to eliminate the annoying compiler 
warnings about "private destructors and no friends".

	Ross.



From <EMAIL-ADDRESS-HIDDEN> Wed Oct 31 08:39:40 2001
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Date: Wed Oct 31 08:40:03 2001
X-Original-Date: Wed, 31 Oct 2001 11:52:49 -0500
Status: O

I am running both testMP3Streamer, testMP3Receiver on windows 2000 operating
system.
I tried changing the const unsigned char ttl from 1 to 255 ,but didn't help
much , some times i am able to receive two data packets in testMP3Receiver
.Do you think it has some thing to do with network problems?

I wonder if this has anything to do with the CNAME(host name ) passed to
RTCPInstance::createNew in both the testMP3Streamer and testMP3Receiver .

Thanks
Mahua









At 03:26 PM 10/29/01, Mahua De wrote:
>I am having problem trying to run the Test programs testMP3Streamer
>,testMP3Receiver on different host machines within an intranet network.The
>testMP3Receiver is able to receive only the first packet and after that the
>wasLoopedBackFromUs() in GroupSock.cpp returns true always.
>
>However, if run both the test programs testMP3Streamer and testMP3Receiver
>on the same host machine, the testMP3Receiver  program successfully
receives
>all the packets.

Mahua,

No, I haven't seen this problem.  What operating systems are you running on 
each machine?  If you're running different OSs on each machine, does it 
make a difference if you reverse which one runs "testMP3Streamer" and which 
one runs "testMP3Receiver"?

You might also try increasing the TTL of outgoing multicast packets (in 
"testMP3Streamer"), in case that has something to do with the problem.  On 
line 114 of "testMP3Streamer.cpp", where it says
         const unsigned char ttl = 1
try changing 1 to 255.

         Ross.




From <EMAIL-ADDRESS-HIDDEN> Wed Oct 31 10:46:11 2001
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Date: Wed Oct 31 10:47:02 2001
X-Original-Date: Wed, 31 Oct 2001 10:45:54 -0800
Status: O

At 08:52 AM 10/31/01, Mahua De wrote:
>I am running both testMP3Streamer, testMP3Receiver on windows 2000 
>operating system.
>I tried changing the const unsigned char ttl from 1 to 255 ,but didn't help
>much , some times i am able to receive two data packets in testMP3Receiver
>.Do you think it has some thing to do with network problems?

I think the problem is caused by some (unknown) inconsistency in your use 
of the Windows networking ("winsock") libraries.

Could you try running the pre-built Windows binaries that you'll find in 
<http://www.live.com/windows/>, instead of the versions that you 
built?  Please let me know whether or not things work OK if you use these 
pre-built binaries.

When you compiled the code on Windows, did you have to make *any* changes 
at all to your Makefiles, or were you able to build the code using the 
Makefiles generated by "genWindowsMakefiles" 'as is'?  If you needed to 
change your Makefiles to get the code to compile/link, then I need to know 
what changes you made, as this might have something to do with the problem.

>I wonder if this has anything to do with the CNAME(host name ) passed to
>RTCPInstance::createNew in both the testMP3Streamer and testMP3Receiver .

No, the problem will have nothing to do with this.

         Ross.



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Date: Tue Nov  6 10:16:02 2001
X-Original-Date: Tue, 6 Nov 2001 10:19:23 -0800
Status: O

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------=_NextPart_000_0006_01C166AC.81BB0510
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Hi, all:
I read a little bit about the live library.
According to my understanding, container files
have both audio and video streams, each of which takes
a separate RTP stream. This will be shown in the
SDP file. (with two "m="s)
  I'm just wondering whether there is any SDP
file for the MPEG2 when I download a mpeg2 file
from rtsp server? If there is, how many streams 
are in the SDP file?
 
 Thank you so much!
 
Sincerely, Yanito
 


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From <EMAIL-ADDRESS-HIDDEN> Tue Nov 06 12:26:49 2001
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Date: Tue Nov  6 12:27:18 2001
X-Original-Date: Tue, 06 Nov 2001 12:26:10 -0800
Status: O

At 10:19 AM 11/6/01, Yun Yan wrote:
>Hi, all:
>I read a little bit about the live library.
>According to my understanding, container files
>have both audio and video streams, each of which takes
>a separate RTP stream. This will be shown in the
>SDP file. (with two "m="s)
>   I'm just wondering whether there is any SDP
>file for the MPEG2 when I download a mpeg2 file
>from rtsp server? If there is, how many streams
>are in the SDP file?

Yanito,

If you already have a RTSP server that's serving up a MPEG-2 session, then 
you can test this out right now by running the "openRTSP" program 
<http://www.live.com/openRTSP/> on this session's rtsp:// URL.

What you'll probably find is that each MPEG elementary stream (audio and 
video) will be given its own RTP stream, each with its own "m=" line in the 
SDP description.  So, there will be two "m=" lines in the SDP description.

It is also possible for MPEG audio and video to be combined in a single RTP 
stream - either as a MPEG transport stream, or using the experimental (and 
rarely implemented) payload format for bundled audio+video described in RFC 
2343.

Note that the "testMPEGVideoStreamer" test program (included with the 
LIVE.COM Streaming Media libraries) currently streams just a video 
elementary stream (and so, when you enable its built-in RTSP server, the 
SDP description will contain just a single "m=" line).

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Tue Nov 06 13:38:05 2001
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Date: Tue Nov  6 13:39:03 2001
X-Original-Date: Tue, 6 Nov 2001 13:42:13 -0800
Status: O

Hi, Ross:
  Thank you for the reply.
  I have another question
>It is also possible for MPEG audio and video to be combined in a single RTP
>stream - either as a MPEG transport stream, or using the experimental (and
>rarely implemented) payload format for bundled audio+video described in RFC
>2343.
  Is there any "m=" line in the SDP file for the single RTP stream (as a
MPEG
transport stream)? If there is, what should be medium name for it?

Sincerely, Yanito





From <EMAIL-ADDRESS-HIDDEN> Tue Nov 06 14:10:39 2001
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Date: Tue Nov  6 14:11:03 2001
X-Original-Date: Tue, 06 Nov 2001 14:10:20 -0800
Status: O

At 01:42 PM 11/6/01, you wrote:
> >It is also possible for MPEG audio and video to be combined in a single RTP
> >stream - either as a MPEG transport stream, or using the experimental (and
> >rarely implemented) payload format for bundled audio+video described in RFC
> >2343.
>   Is there any "m=" line in the SDP file for the single RTP stream (as a
>MPEG transport stream)?

Yes, all RTP sessions have a corresponding "m=" line in the SDP description.

Note that - according to <draft-ietf-avt-profile-new-11.txt> - MPEG-2 
Transport Streams use a static payload format of 33.  (In constrast, MPEG 
audio (elementary) streams use a static payload format of 14, and MPEG 
video (elementary) streams use a static payload format of 32.)

>If there is, what should be medium name for it?

That's a good question - I don't know.  "video" would probably be 
appropriate (even though it also includes audio).

In any case, I don't recommend streaming a MPEG transport stream as a 
single RTP session.  This doesn't give you the flexibility to handle the 
audio and video streams in a different way, if you wish to do this.  Also, 
I don't know how many tools exist that can receive this payload 
format.  (In particular, the "liveMedia" library doesn't currently support 
this.)

Do you have any specific questions about how to make use of the libraries?

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Wed Nov 07 05:57:13 2001
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Date: Wed Nov  7 05:58:02 2001
X-Original-Date: Wed, 7 Nov 2001 08:36:39 -0500
Status: O

Hello

I have recently been looking at the Live libraries and our company is
interested in streaming MPEG.  I have been able to build and run the
testMP3Streamer from Windows 2000 and Redhat 6.2 -- My problem comes when I
receive the stream using the real player or winamp. The audio frequently
pauses and rebuffers but it does not seem to be dropping anything - What
should I check?

Thanks

Bruce Wilber


From <EMAIL-ADDRESS-HIDDEN> Wed Nov 07 06:40:59 2001
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Date: Wed Nov  7 06:41:08 2001
X-Original-Date: Wed, 7 Nov 2001 10:46:17 -0500
Status: O

>Hello
>
>I have recently been looking at the Live libraries and our 
company is
>interested in streaming MPEG.  I have been able to build 
and run the
>testMP3Streamer from Windows 2000 and Redhat 6.2 -- 
My problem comes when I
>receive the stream using the real player or winamp. The 
audio frequently
>pauses and rebuffers but it does not seem to be dropping 
anything - What
>should I check?
>

I do not know much about winamp, but the Real player is
very sensitive to dropped or out of order packets. It seems
to rebuffer more than is necessary. 

Our player (the MCTPlayer - www.mctplayer.com - which is 
soon to start using the Live.com RTP library) uses a
fairly large buffer size for streaming MP3 playback, which
helps, but a really good quality non-interrupted stream 
really requires some sort of FEC (which we also offer and
our player supports). This functionality is soon to be 
incorporated into the Live.com Winamp plug-in

Regards
Marshall Eubanks
Multicast Technologies, Inc.


>Thanks
>
>Bruce Wilber
>
>
______________________________________________
_
>Live-devel mailing list
>Live-devel(at)lists.sourceforge.net
>https://lists.sourceforge.net/lists/listinfo/live-devel
>
>




Marshall Eubanks

tme(at)21rst-century.com


From <EMAIL-ADDRESS-HIDDEN> Wed Nov 07 09:19:19 2001
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Date: Wed Nov  7 09:20:17 2001
X-Original-Date: Wed, 07 Nov 2001 09:18:29 -0800
Status: O

At 05:36 AM 11/7/01, bwilber(at)vela.com wrote:
>I have recently been looking at the Live libraries and our company is
>interested in streaming MPEG.  I have been able to build and run the
>testMP3Streamer from Windows 2000 and Redhat 6.2 -- My problem comes when I
>receive the stream using the real player or winamp. The audio frequently
>pauses and rebuffers but it does not seem to be dropping anything - What
>should I check?

Bruce,

This problem is caused by insufficient buffering in the receiving client 
application; you can fix it by increasing the amount of buffering that the 
client performs on incoming streams.

For RealPlayer, you can do this using the 
"Preferences"/"Connection"/"Buffered play" option.  (Personally, though, 
this hasn't worked for me, for some reason...)

For Winamp (with the "in_rtp.dll" plugin, of course), you can do this using 
"Preferences" (<control>p)/"Plug-ins"/"Input"/"Nullsoft MPEG Audio 
Decoder"/"Configure"/"Streaming"

         Ross.




From <EMAIL-ADDRESS-HIDDEN> Wed Nov 07 09:53:32 2001
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Date: Wed Nov  7 09:54:03 2001
X-Original-Date: Wed, 7 Nov 2001 12:32:59 -0500
Status: O

Thank you Marshall and Ross

Both the WinAmp plug-in (with increased buffering) and the MCT Player work
great. 
For me - increasing Real's buffer (which I had tried before) made no
difference. Real also drops the connection not to be regained when the
streamer loops - both MCT and WinAmp keep the connection.

I knew that "Live" probably worked better than I was experiencing.  Now I am
going to look at the video streamer - any clients other than Real for this?

Bruce


From <EMAIL-ADDRESS-HIDDEN> Wed Nov 07 10:30:43 2001
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Date: Wed Nov  7 10:31:03 2001
X-Original-Date: Wed, 07 Nov 2001 13:30:17 -0500
Status: O

bwilber(at)vela.com wrote:

>Thank you Marshall and Ross
>
>Both the WinAmp plug-in (with increased buffering) and the MCT Player work
>great. 
>For me - increasing Real's buffer (which I had tried before) made no
>difference. Real also drops the connection not to be regained when the
>streamer loops - both MCT and WinAmp keep the connection.
>
>I knew that "Live" probably worked better than I was experiencing.  Now I am
>going to look at the video streamer - any clients other than Real for this?
>
Real does not do MPEG video streaming (at least for multicast). It does 
do H.261

Quicktime does a good job of it (as long as the audio is Layer II). If 
you want
to multicast TV on a Mac, there is MacTV(which runs QT).

On Unix platforms there is MIM, which is OK
On Windows, there is Cisco IPTV, which works well.
vBrick also makes a MPEG player for multicast, and there may be others.

See
http://videolab.uoregon.edu/tools/multicast_tools.html
for more info

All of this is for MPEG-1. MPEG-2 is expensive as there is a license fee 
even
for free players, so there are no free players AFAIK that support it.

Regards

Marshall Eubanks

>
>
>Bruce
>
>_______________________________________________
>Live-devel mailing list
>Live-devel(at)lists.sourceforge.net
>https://lists.sourceforge.net/lists/listinfo/live-devel
>
>


-- 
                                 Regards
                                 Marshall Eubanks

This e-mail may contain confidential and proprietary information of 
Multicast Technologies, Inc, subject to Non-Disclosure Agreements


T.M. Eubanks
Multicast Technologies, Inc
10301 Democracy Lane, Suite 410
Fairfax, Virginia 22030
Phone : 703-293-9624       Fax     : 703-293-9609
e-mail : tme(at)multicasttech.com
http://www.on-the-i.com  

Test your network for multicast : http://www.multicasttech.com/mt/
 Check the status of multicast in real time :
 http://www.multicasttech.com/status/index.html





From <EMAIL-ADDRESS-HIDDEN> Wed Nov 07 11:26:20 2001
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Subject: mpeg2 software was Re: [Live-devel] Real buffering
From: Marty Schoch <EMAIL-ADDRESS-HIDDEN>
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Date: Wed Nov  7 11:27:04 2001
X-Original-Date: 07 Nov 2001 14:26:15 -0500
Status: O

Through the Livid project there are open implementations of 
mpeg2 decoders and players (mainly geared to DVD).  Not sure
of the legal/licensing fee issues with actually using this
software.

http://www.linuxvideo.org

Marty Schoch
<mschoch(at)multicasttech.com>

On Wed, 2001-11-07 at 13:30, Marshall Eubanks wrote:
> bwilber(at)vela.com wrote:
> 
> >Thank you Marshall and Ross
> >
> >Both the WinAmp plug-in (with increased buffering) and the MCT Player work
> >great. 
> >For me - increasing Real's buffer (which I had tried before) made no
> >difference. Real also drops the connection not to be regained when the
> >streamer loops - both MCT and WinAmp keep the connection.
> >
> >I knew that "Live" probably worked better than I was experiencing.  Now I am
> >going to look at the video streamer - any clients other than Real for this?
> >
> Real does not do MPEG video streaming (at least for multicast). It does 
> do H.261
> 
> Quicktime does a good job of it (as long as the audio is Layer II). If 
> you want
> to multicast TV on a Mac, there is MacTV(which runs QT).
> 
> On Unix platforms there is MIM, which is OK
> On Windows, there is Cisco IPTV, which works well.
> vBrick also makes a MPEG player for multicast, and there may be others.
> 
> See
> http://videolab.uoregon.edu/tools/multicast_tools.html
> for more info
> 
> All of this is for MPEG-1. MPEG-2 is expensive as there is a license fee 
> even
> for free players, so there are no free players AFAIK that support it.
> 
> Regards
> 
> Marshall Eubanks
> 
> >
> >
> >Bruce
> >
> >_______________________________________________
> >Live-devel mailing list
> >Live-devel(at)lists.sourceforge.net
> >https://lists.sourceforge.net/lists/listinfo/live-devel
> >
> >
> 
> 
> -- 
>                                  Regards
>                                  Marshall Eubanks
> 
> This e-mail may contain confidential and proprietary information of 
> Multicast Technologies, Inc, subject to Non-Disclosure Agreements
> 
> 
> T.M. Eubanks
> Multicast Technologies, Inc
> 10301 Democracy Lane, Suite 410
> Fairfax, Virginia 22030
> Phone : 703-293-9624       Fax     : 703-293-9609
> e-mail : tme(at)multicasttech.com
> http://www.on-the-i.com  
> 
> Test your network for multicast : http://www.multicasttech.com/mt/
>  Check the status of multicast in real time :
>  http://www.multicasttech.com/status/index.html
> 
> 
> 
> 
> _______________________________________________
> Live-devel mailing list
> Live-devel(at)lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/live-devel




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Date: Thu Nov  8 11:27:01 2001
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Status: O

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Hi, all:
  
  When I run openRTSP to download a mp3 file from 
RTSP server, I got the file "AUDIO-X-MP3-DRAFT-00". 
I found this file is different from the file on server. 
( both the size and hexdumps are a lot of different)
I also got the error message "Can't rewind stream by 
53 bits" when I run "mpg123 AUDIO-X-MP3-DRAFT-00".
  Any hint about this?
  I appreciate your help.

Sincerely, Yanito


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From <EMAIL-ADDRESS-HIDDEN> Thu Nov 08 12:11:08 2001
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Date: Thu Nov  8 12:16:04 2001
X-Original-Date: Thu, 08 Nov 2001 12:10:47 -0800
Status: O

At 11:30 AM 11/8/01, Yun Yan wrote:
>   When I run openRTSP to download a mp3 file from
>RTSP server, I got the file "AUDIO-X-MP3-DRAFT-00".
>I found this file is different from the file on server.
>( both the size and hexdumps are a lot of different)
>I also got the error message "Can't rewind stream by
>53 bits" when I run "mpg123 AUDIO-X-MP3-DRAFT-00".

Yanito,

If possible, could you make this RTSP publically accessible, and let me 
know the rtsp:// URL, so I can try this out for myself to figure out what's 
going wrong?  Also, could you put the original file on a public HTTP server?

Also, please try running openRTSP on the URL
         rtsp://media.schreiner.edu/realmp3.mp3
and check whether the resulting output is a valid MP3 file.  (It should be.)

         Ross.



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Subject: [Live-devel] Unicast streams and firewalls
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Date: Thu Nov  8 14:11:02 2001
X-Original-Date: Thu, 08 Nov 2001 23:10:18 +0100
Status: O

I have changed the testMP3Streamer to stream to a unicast address, and is 
able to receive the stream on my client. This works perfectly in my test 
environment, but the real world is not so simple.

I would like to stream a unicast stream from a server with no firewall, or a 
firewall I can poke holes in as I wish, to a client inside a firewall, that 
I have no control over. Is that possible with the LIVE.COM Streaming Media? 
Maybe by a client initiated session?

Thanks,
Kenneth Andreasen

_________________________________________________________________
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From <EMAIL-ADDRESS-HIDDEN> Thu Nov 08 16:06:41 2001
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Date: Thu Nov  8 16:07:02 2001
X-Original-Date: Thu, 08 Nov 2001 16:05:54 -0800
Status: O

At 02:10 PM 11/8/01, Kenneth Andreasen wrote:
>I would like to stream a unicast stream from a server with no firewall, or 
>a firewall I can poke holes in as I wish, to a client inside a firewall, 
>that I have no control over. Is that possible with the LIVE.COM Streaming 
>Media?

Kenneth,

If your client is behind a firewall that blocks all UDP, then there's not 
much you can do.  If, however, the firewall allows certain UDP ports, or 
can be reconfigured to allow certain UDP ports, then you could have your 
stream use one of those ports (provided, of course, that this port is not 
being used by some other application).

If your client's firewall does not allow any UDP through at all, then 
you'll probably have to resort to streaming over HTTP - e.g., using an 
"icecast" server and a client such as Winamp or Freeamp.

         Ross.



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Date: Mon Nov 12 22:11:01 2001
X-Original-Date: Tue, 13 Nov 2001 11:09:12 +0500
Status: O

Hi Guys

I looked into the source code . Hats off to the wonderful work u people are
doing . I am very much new to this field and very far
from u all guys from technologival point of view .I am trying to understand
the realm of streaming media before embarking on the next
destination of MPEG-4 DMIF .Alot of information is already present inside
the code but it would be a lot helpful if some body guide
me a little and tell me how to setup a client and server on My PC ? I feel
it would be easy to bisect and examine the protocols if
we have some running infrastructure.    I have also tried to run the
openrtsp client .Anbody help me about some vaild rtsp urls to
run it successfully?

Any help would be highly appreciated

Regards




From <EMAIL-ADDRESS-HIDDEN> Tue Nov 13 00:25:17 2001
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Date: Tue Nov 13 00:26:02 2001
X-Original-Date: Tue, 13 Nov 2001 00:24:11 -0800
Status: O

>I have also tried to run the
>openrtsp client .Anbody help me about some vaild rtsp urls to
>run it successfully?

Here's one that you can try.  If you run:

         openRTSP -a rtsp://media.schreiner.edu/realmp3.mp3 > test.mp3

you'll end up with a MP3 file "test.mp3".  (If you like, you can then use 
"testMP3Streamer" to stream this MP3 file via multicast.)

         Ross.



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Subject: [Live-devel] Problem in running openrtsp
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Date: Tue Nov 13 01:18:01 2001
X-Original-Date: Tue, 13 Nov 2001 14:16:20 +0500
Status: O

I tried to run the openrtsp by using the command shown below.It seems that
things are fine as far as log shows
but the output file (test.mp3) is always of 0 bytes. Why is it so?



-------------------COMMAND ISSUED---------------------

openrtsp -a rtsp://media.schreiner.edu/realmp3.mp3 >test.mp3


-----------------------------------LOG-----------------
Opened URL "rtsp://media.schreiner.edu/realmp3.mp3", returning a SDP
description
:
v=0
o=- 979010679 979010679 IN IP4 207.71.37.7
s=realmp3
i=<No author> <No copyright>
c=IN IP4 0.0.0.0
t=0 0
a=SdpplinVersion:1610642970
a=StreamCount:integer;1
a=Title:buffer;"cmVhbG1wMwA="
a=range:npt=0-39.471000
m=audio 0 RTP/AVP 101
b=AS:128
a=control:streamid=0
a=range:npt=0-39.471000
a=length:npt=39.471000
a=rtpmap:101 X-MP3-draft-00
a=mimetype:string;"audio/X-MP3-draft-00"
a=StartTime:integer;0
a=AvgBitRate:integer;128000
a=SampleRate:integer;44100
a=AvgPacketSize:integer;417
a=Preroll:integer;1000
a=NumChannels:integer;2
a=MaxPacketSize:integer;1024
a=ASMRuleBook:string;"AverageBandwidth=128000, AverageBandwidthStd=0,
Priority=9
;"

Created receiver for "audio/X-MP3-DRAFT-00" subsession (client ports
2560-2561)
Issued RTSP "SETUP" on "audio/X-MP3-DRAFT-00" subsession (client ports
2560-2561
)
Outputting data from the "audio/X-MP3-DRAFT-00" subsession to 'stdout'
Issued RTSP "PLAY" command on "audio/X-MP3-DRAFT-00" subsession
Receiving streamed data (for up to 44.5 seconds)...
Closing "audio/X-MP3-DRAFT-00" subsession





From <EMAIL-ADDRESS-HIDDEN> Tue Nov 13 02:04:48 2001
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Date: Tue Nov 13 02:05:02 2001
X-Original-Date: Tue, 13 Nov 2001 02:03:15 -0800
Status: O

At 01:16 AM 11/13/01, Mumtaz Ahmad wrote:
>I tried to run the openrtsp by using the command shown below.It seems that
>things are fine as far as log shows
>but the output file (test.mp3) is always of 0 bytes. Why is it so?

Mumtaz,

It appears that your PC is not receiving RTP (and RTCP) packets that come 
from the server, for some reason.  Do you have a firewall of some sort 
that's blocking incoming UDP packets?  If so, you will need to reconfigure 
it to allow at least a single even/odd pair of client port numbers (for 
incoming RTP and RTCP packets).  (You can then use the
         -p <startingPort>
option to "openRTSP" to tell it to use these ports.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Tue Nov 13 05:15:34 2001
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Date: Tue Nov 13 05:16:02 2001
X-Original-Date: Tue, 13 Nov 2001 14:13:32 +0100
Status: O

hello,

Does your client send RTCP packets? I know you can get it.

BRs,

Jerry



From <EMAIL-ADDRESS-HIDDEN> Tue Nov 13 08:00:10 2001
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Date: Tue Nov 13 08:01:03 2001
X-Original-Date: Tue, 13 Nov 2001 07:59:45 -0800
Status: O

At 05:13 AM 11/13/01, Jerry wrote:
>Does your client send RTCP packets? I know you can get it.

Jerry,

Yes, the "liveMedia" library contains a RTCP implementation, which both 
sends and receives RTCP reports.  RTP senders (such as the 
"testMP3Streamer" and "testMPEGVideoStreamer" test program, and 
"liveCaster") send RTCP Sender Reports.  RTP receivers (such as "openRTSP", 
the "testMP3Receiver" and "testMPEGVideoReceiver" test programs, and 
"playRTPMPEG") send RTCP Reception Reports.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Thu Nov 15 04:23:16 2001
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Subject: [Live-devel] about scheduler
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Date: Thu Nov 15 04:24:04 2001
X-Original-Date: Thu, 15 Nov 2001 13:20:51 +0100
Status: O

hello,

I have a question about class TaskScheduler, in which file you implement it?
I am new to your software.

Thanks,

Jerry


From <EMAIL-ADDRESS-HIDDEN> Thu Nov 15 06:25:00 2001
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Subject: [Live-devel] Minimum delay for packet send
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Date: Thu Nov 15 06:25:11 2001
X-Original-Date: Thu, 15 Nov 2001 15:24:33 +0100
Status: O

I have questions, concerning the TaskScheduler and DelayQueue. First I'll 
describe the application.
I'm using the LIVE.COM library in a Windows-App for (at the moment unicast) 
streaming of MP3 files to embedded targets. These targets are some kind of 
slow, thus they only can deal with packets with a delay (or better a time 
distance between the packets) of at least 10ms. But the Windows-App sends 
some packets with (for example) a distance of 1ms, although the delays in 
the DelayQueue should be even greater than 10ms (for a 128kbps MP3). But 
that could be a sideeffect of the windows taskprocessing scheme. Actually 
the 1ms shouldn't be a problem, since a normal network interface should 
handle this. But with my embedded target I lost packets.
Now I made a hack in the library. In the DelayQueue::handleAlarm() method I 
inserted an absolute waiting time of 20ms (sleep(20)) inside the alarm 
handling loop. This causes that all packets will be send with a delay of at 
least 20ms. And it worked fine. I didn't lose any packet.
Now I wanted to stream to several destinations at the same time, and of 
course I lost packets again. Because _all_ packets are delayed by 20ms. I 
limited my output bandwidth. Thus this is not the solution.
My question is now. Is it possible to set a minimum delay for all packets 
of one stream? Packets from different streams can (actually must) be send 
with less delay (even 1ms).
Since the Events in the DelayQueue are scheduled right (with a sufficient 
delay) one has to handle this somewhere else. If it's possible at all, 
where can I do this in the code?
Do I have to implement a new UsageEnvironment with a new TaskScheduler for 
Windows?
Does someone have any idea?
Many thanks in advance.

Michael

P.S. Sorry that I might have chosen the wrong words in some cases. It's not 
my native language.



From <EMAIL-ADDRESS-HIDDEN> Thu Nov 15 09:21:55 2001
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Date: Thu Nov 15 09:22:02 2001
X-Original-Date: Thu, 15 Nov 2001 09:17:46 -0800
Status: O

At 04:20 AM 11/15/01, Jerry wrote:
>hello,
>
>I have a question about class TaskScheduler, in which file you implement it?

The "TaskScheduler" class  is implemented in the "UsageEnvironment" 
library, in the file "UsageEnvironment.cpp".

However, "TaskScheduler" is an abstract base class, which needs to be 
subclassed to form an actual implementation that you can use in a 
program.  One possible subclass is "BasicTaskScheduler", which is 
implemented in the "BasicUsageEnvironment" library (in the file 
"BasicUsageEnvironment.cpp").

Depending on your environment, you may wish to replace 
"BasicUsageEnvironment" with your own implementation.  (For example, you 
might want to do this if you are using the LIVE.COM libraries within an 
application that already has its own event loop mechanism.)

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Thu Nov 15 09:58:24 2001
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   <live-devel(at)lists.sourceforge.net>
Subject: RE: [Live-devel] Minimum delay for packet send
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Date: Thu Nov 15 09:59:04 2001
X-Original-Date: Thu, 15 Nov 2001 09:58:09 -0800
Status: O

Mike,

Your message brings an interesting perspective to the usage of this library
in the sense that sometimes an asymmetry in the computing power of the
server machine and the client machines would bring out a new issue.  In my
playing with this library in the past several months, both my server and
client machines are Pentium class PC.

Now to your issue, I'm surprised that the packets are not scheduled with the
right interval.  For MP3, the packets are scheduled to be sent in the method
MultiFramedRTPSink::sendPacketIfNecessary() in the file
MultiFramedRTPSink.cpp.  The interval is the timeToNextPacket variable which
is from fSource->getPlayTime(), which in turn comes from
MP3StreamState::getPlayTime(), which in turn comes from
MP3StreamState::currentFramePlayTime().  This is in the file
MP3StreamState.cpp.

Since I've not experienced the problem you brought up and I've not seriously
studied currentFramePlayTime(), may be you can investigate and let us know
what you find out.

Dan



> -----Original Message-----
> From: live-devel-admin(at)lists.sourceforge.net
> [mailto:live-devel-admin(at)lists.sourceforge.net]On Behalf Of Michael
> Sommer
> Sent: Thursday, November 15, 2001 6:25 AM
> To: live-devel(at)lists.sourceforge.net
> Subject: [Live-devel] Minimum delay for packet send
>
>
> I have questions, concerning the TaskScheduler and DelayQueue. First I'll
> describe the application.
> I'm using the LIVE.COM library in a Windows-App for (at the
> moment unicast)
> streaming of MP3 files to embedded targets. These targets are
> some kind of
> slow, thus they only can deal with packets with a delay (or better a time
> distance between the packets) of at least 10ms. But the Windows-App sends
> some packets with (for example) a distance of 1ms, although the delays in
> the DelayQueue should be even greater than 10ms (for a 128kbps MP3). But
> that could be a sideeffect of the windows taskprocessing scheme. Actually
> the 1ms shouldn't be a problem, since a normal network interface should
> handle this. But with my embedded target I lost packets.
> Now I made a hack in the library. In the
> DelayQueue::handleAlarm() method I
> inserted an absolute waiting time of 20ms (sleep(20)) inside the alarm
> handling loop. This causes that all packets will be send with a
> delay of at
> least 20ms. And it worked fine. I didn't lose any packet.
> Now I wanted to stream to several destinations at the same time, and of
> course I lost packets again. Because _all_ packets are delayed by 20ms. I
> limited my output bandwidth. Thus this is not the solution.
> My question is now. Is it possible to set a minimum delay for all packets
> of one stream? Packets from different streams can (actually must) be send
> with less delay (even 1ms).
> Since the Events in the DelayQueue are scheduled right (with a sufficient
> delay) one has to handle this somewhere else. If it's possible at all,
> where can I do this in the code?
> Do I have to implement a new UsageEnvironment with a new
> TaskScheduler for
> Windows?
> Does someone have any idea?
> Many thanks in advance.
>
> Michael
>
> P.S. Sorry that I might have chosen the wrong words in some
> cases. It's not
> my native language.
>
>
> _______________________________________________
> Live-devel mailing list
> Live-devel(at)lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/live-devel
>



From <EMAIL-ADDRESS-HIDDEN> Thu Nov 15 14:42:30 2001
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Date: Thu Nov 15 14:43:06 2001
X-Original-Date: Thu, 15 Nov 2001 14:40:56 -0800
Status: O

At 06:24 AM 11/15/01, Michael Sommer wrote:
>I'm using the LIVE.COM library in a Windows-App for (at the moment 
>unicast) streaming of MP3 files to embedded targets. These targets are 
>some kind of slow, thus they only can deal with packets with a delay (or 
>better a time distance between the packets) of at least 10ms. But the 
>Windows-App sends some packets with (for example) a distance of 1ms, 
>although the delays in the DelayQueue should be even greater than 10ms 
>(for a 128kbps MP3). But that could be a sideeffect of the windows 
>taskprocessing scheme.

Yes, I believe so.  For a 128 kbps, 44.1 kHz MP3 stream, with one frame per 
packet, the interval between each successive packet should be about 26 
ms.  If you're occasionally seeing much shorter intervals between packets, 
then this is probably caused by the application trying to 'catch up', to 
overcome Windows' crappy task scheduling.

>Now I made a hack in the library. In the DelayQueue::handleAlarm() method 
>I inserted an absolute waiting time of 20ms (sleep(20)) inside the alarm 
>handling loop.

Ugh!  This is a bad idea, because the delay queue can be used for more than 
just inter-packet delays.  Instead, if you want to set a minimum 
inter-packet delay, then you should do so at the point where this gap is 
computed.

As Dan Phan noted, the place where this is computed is in the file 
"liveMedia/MultiFramedRTPSink.cpp".  You should be able to set a minimum 
inter-packet delay of 10 ms (for example), by adding the following code at 
line 289:

         if (uSecondsToGo < 10000) uSecondsToGo = 10000;

I hope this gives you the results that you're looking for.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Thu Nov 15 16:17:03 2001
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Date: Thu Nov 15 16:18:01 2001
X-Original-Date: Thu, 15 Nov 2001 16:15:56 -0800
Status: O

If the observed effect is perhaps due to the "crappy" Windows task scheduler
as suggested by Ross, has anyone tried setting the process priority higher
than normal using Win32 SetPriorityClass()?

Dan

> -----Original Message-----
> From: live-devel-admin(at)lists.sourceforge.net
> [mailto:live-devel-admin(at)lists.sourceforge.net]On Behalf Of Ross
> Finlayson
> Sent: Thursday, November 15, 2001 2:41 PM
> To: Michael Sommer
> Cc: live-devel(at)lists.sourceforge.net
> Subject: Re: [Live-devel] Minimum delay for packet send
>
>
> At 06:24 AM 11/15/01, Michael Sommer wrote:
> >I'm using the LIVE.COM library in a Windows-App for (at the moment
> >unicast) streaming of MP3 files to embedded targets. These targets are
> >some kind of slow, thus they only can deal with packets with a delay (or
> >better a time distance between the packets) of at least 10ms. But the
> >Windows-App sends some packets with (for example) a distance of 1ms,
> >although the delays in the DelayQueue should be even greater than 10ms
> >(for a 128kbps MP3). But that could be a sideeffect of the windows
> >taskprocessing scheme.
>
> Yes, I believe so.  For a 128 kbps, 44.1 kHz MP3 stream, with one
> frame per
> packet, the interval between each successive packet should be about 26
> ms.  If you're occasionally seeing much shorter intervals between
> packets,
> then this is probably caused by the application trying to 'catch up', to
> overcome Windows' crappy task scheduling.
>
> >Now I made a hack in the library. In the
> DelayQueue::handleAlarm() method
> >I inserted an absolute waiting time of 20ms (sleep(20)) inside the alarm
> >handling loop.
>
> Ugh!  This is a bad idea, because the delay queue can be used for
> more than
> just inter-packet delays.  Instead, if you want to set a minimum
> inter-packet delay, then you should do so at the point where this gap is
> computed.
>
> As Dan Phan noted, the place where this is computed is in the file
> "liveMedia/MultiFramedRTPSink.cpp".  You should be able to set a minimum
> inter-packet delay of 10 ms (for example), by adding the
> following code at
> line 289:
>
>          if (uSecondsToGo < 10000) uSecondsToGo = 10000;
>
> I hope this gives you the results that you're looking for.
>
>          Ross.
>
>
> _______________________________________________
> Live-devel mailing list
> Live-devel(at)lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/live-devel
>



From <EMAIL-ADDRESS-HIDDEN> Fri Nov 16 02:03:39 2001
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Date: Fri Nov 16 02:04:02 2001
X-Original-Date: Fri, 16 Nov 2001 11:03:06 +0100
Status: O

Thanks Ross and Dan for your response.

At 16:15 15.11.01 -0800, Dan Phan wrote:
>If the observed effect is perhaps due to the "crappy" Windows task scheduler
>as suggested by Ross, has anyone tried setting the process priority higher
>than normal using Win32 SetPriorityClass()?
I already tried that. I run two processes in my Windows-App, one with the 
GUI and the other one with the blockMyself() loop (both running at 
THREAD_PRIORITY_NORMAL). That works quite nice. When I set the priority of 
the packet scheduling process to THREAD_PRIORITY_ABOVE_NORMAL, the GUI is 
getting very slow. With THREAD_PRIORITY_HIGHEST you can't work anymore. So, 
this seems to be no solution. But I think it could be that the 
BasicTaskScheduler is not the right implementation for Windows GUI 
(multithreaded) applications.

> > >Now I made a hack in the library. In the
> > DelayQueue::handleAlarm() method
> > >I inserted an absolute waiting time of 20ms (sleep(20)) inside the alarm
> > >handling loop.
> >
> > Ugh!  This is a bad idea, because the delay queue can be used for
> > more than
> > just inter-packet delays.

I know that it was bad idea, but it helped me to find the problem and to 
solve it in some way. ;o>

>  Instead, if you want to set a minimum
> > inter-packet delay, then you should do so at the point where this gap is
> > computed.
> >
> > As Dan Phan noted, the place where this is computed is in the file
> > "liveMedia/MultiFramedRTPSink.cpp".  You should be able to set a minimum
> > inter-packet delay of 10 ms (for example), by adding the
> > following code at
> > line 289:
> >
> >          if (uSecondsToGo < 10000) uSecondsToGo = 10000;

Sorry Ross, but unfortunately it didn't work. To show you the effect I made 
a network sniffing and captured all RTP packets send from my maschine (I'm 
doing unicast, 128kbps MP3). Here it is:

PNr | Time (h:m:s.us) | Network                         | Description
170 | 09:37:22.388263 | IP:10.0.10.25=>10.0.10.113(1298)| 
UDP:Length=1278,Port(8888=>8888)|
       00:00:00.109956
171 | 09:37:22.498219 | IP:10.0.10.25=>10.0.10.113(1298)| 
UDP:Length=1278,Port(8888=>8888)|
       00:00:00.054897
172 | 09:37:22.553116 | IP:10.0.10.25=>10.0.10.113(1298)| 
UDP:Length=1278,Port(8888=>8888)|
       00:00:00.111174
173 | 09:37:22.664290 | IP:10.0.10.25=>10.0.10.113(1297)| 
UDP:Length=1277,Port(8888=>8888)|
       00:00:00.000301
174 | 09:37:22.664591 | IP:10.0.10.25=>10.0.10.113(1298)| 
UDP:Length=1278,Port(8888=>8888)| <= lost packet
       00:00:00.163557
175 | 09:37:22.828148 | IP:10.0.10.25=>10.0.10.113(1298)| 
UDP:Length=1278,Port(8888=>8888)|

The packet nr.174 get lost. And if you look at the delta time between 173 
and 174, these are only 301us which is far to less for my embedded target.
I don't know yet (maybe I' getting into it the next time) what happens 
'behind the scences' of the library. I think the scheduling of the packets 
is right, but because of the (as you said 'crappy') Windows task management 
the alarmHandler() missed the deadline of PNr.173 and 174 and sends them 
now immediately one after the other. Maybe you could explain in short how 
the scheduling works. Will the next (packet send) event be scheduled while 
the current packet is send? Or does it schedule more than one event in advance?

Michael



From <EMAIL-ADDRESS-HIDDEN> Fri Nov 16 03:25:27 2001
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Date: Fri Nov 16 03:26:02 2001
X-Original-Date: Fri, 16 Nov 2001 03:22:42 -0800
Status: O

>> > You should be able to set a minimum
>> > inter-packet delay of 10 ms (for example), by adding the
>> > following code at
>> > line 289:
>> >
>> >          if (uSecondsToGo < 10000) uSecondsToGo = 10000;
>
>Sorry Ross, but unfortunately it didn't work.

Hmm...  This should definitely cause a >=10ms delay between successive 
packet sends (using sendto()).  Because you're occasionally seeing smaller 
inter-packet gaps as packets get sent on the network, it seems that the 
underlying OS (Windows) - or perhaps your network interface - is 
occasionally delaying (bunching up) packets before sending them out on the 
network.

Which version of Windows are you running?  98?  If so, do you have any 
better luck running NT, 2000, or XP?

Anyway, I wonder if perhaps you're looking at this problem the wrong 
way.  An inter-packet gap of 1 ms is quite large in this day and age, so 
it's a bit surprising that your receiver - no matter how limited - cannot 
handle this.  Is there any way that your receiver can be made to handle 
this better - e.g., by buffering incoming packets before processing them?

Alternatively, if this packet loss occurs infrequently, then mayzbe this is 
not worth worrying about.  (After all, IP is a datagram protocol, and some 
packet loss should always be anticipated.)


>Maybe you could explain in short how the scheduling works.

In "BasicUsageEnvironment" - the particular implementation (subclass) of 
"UsageEnvironment" that's shipped with the libraries - the scheduler is 
implemented as a simple priority queue, ordered by 
time-to-next-event.  (See the implementation of "DelayQueue" and then 
"BasicTaskScheduler" in the "BasicUsageEnvironment" directory.)


>Will the next (packet send) event be scheduled while the current packet is 
>send? Or does it schedule more than one event in advance?

No, the "sendNext()" routine will be called no sooner than "uSecondsToGo" 
microseconds after the call to "scheduleDelayedTask()".

         Ross.



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Date: Fri Nov 16 06:56:08 2001
X-Original-Date: Fri, 16 Nov 2001 15:55:08 +0100
Status: O

Ross,

I have surprising news. I got the system running without packet loss and 
even without your suggested enhancement for inserting defined gaps between 
the packets. The library works quite right as it is.

>Anyway, I wonder if perhaps you're looking at this problem the wrong 
>way.  An inter-packet gap of 1 ms is quite large in this day and age, so 
>it's a bit surprising that your receiver - no matter how limited - cannot 
>handle this.  Is there any way that your receiver can be made to handle 
>this better - e.g., by buffering incoming packets before processing them?
You were right. I looked a bit closer at my receiver and found out that the 
buffer size of the receiver was limited to 2k Byte for UDP. Thus it would 
only fit one packet in it. Now I set the size to 8k and it runs very well.
Thanks Ross and Dan

Michael



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From: Vesselin Kostadinov <EMAIL-ADDRESS-HIDDEN>
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Subject: [Live-devel] openRTSP problem/setting up MPEG video sender-receiver
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Date: Mon Nov 19 17:23:03 2001
X-Original-Date: Tue, 20 Nov 2001 11:53:05 +1030
Status: O

Hi,

I need to set-up a video broadcaster and a couple of receivers and I decided
to try Live library. I managed to compile it with Borland C++ Builder but
when running openRTSP I encountered a problem - the client was not receiving
any data.

My OS is Windows 2000, SP2. On the same PC I can run Real player, point it
to rtsp://media.schreiner.edu/realmp3.mp3 and listen to the music.

Here is the output from openRTSP: 

-------------------------------------------------------------------------
>openRTSP -a -V -n rtsp://media.schreiner.edu/realmp3.mp3 >test.mp3
Sending request: DESCRIBE rtsp://media.schreiner.edu/realmp3.mp3 RTSP/1.0
CSeq: 1
Accept: application/sdp


Received DESCRIBE response: RTSP/1.0 200 OK
CSeq: 1
Date: Tue, 20 Nov 2001 00:14:25 GMT
Set-Cookie:
cbid=hfhjcmiihjhkjllmeomrqugqrojrktlufkfgkihligjkhiplrnqopppqnrksptk
uekcjehil;path=/;expires=Thu,31-Dec-2037 23:59:59 GMT
X-TSPort: 7802
vsrc:
http://media.schreiner.edu:8080/viewsource/template.html?nuyhtggaysz6nc6vs
ffjsqt6725n10A1v1yeC3d4ngEt5o5gwuw4t6DC4jbh
Last-Modified: Tue, 09 Jan 2001 03:24:39 GMT
Content-base: rtsp://media.schreiner.edu/realmp3.mp3/
Content-type: application/sdp
Content-length: 647

v=0
o=- 979010679 979010679 IN IP4 207.71.37.7
s=realmp3
i=<No author> <No copyright>
c=IN IP4 0.0.0.0
t=0 0
a=SdpplinVersion:1610642970
a=StreamCount:integer;1
a=Title:buffer;"cmVhbG1wMwA="
a=range:npt=0-39.471000
m=audio 0 RTP/AVP 101
b=AS:128
a=control:streamid=0
a=range:npt=0-39.471000
a=length:npt=39.471000
a=rtpmap:101 X-MP3-draft-00
a=mimetype:string;"audio/X-MP3-draft-00"
a=StartTime:integer;0
a=AvgBitRate:integer;128000
a=SampleRate:integer;44100
a=AvgPacketSize:integer;417
a=Preroll:integer;1000
a=NumChannels:integer;2
a=MaxPacketSize:integer;1024
a=ASMRuleBook:string;"AverageBandwidth=128000, AverageBandwidthStd=0,
Priority=9
;"

Opened URL "rtsp://media.schreiner.edu/realmp3.mp3", returning a SDP
description
:
v=0
o=- 979010679 979010679 IN IP4 207.71.37.7
s=realmp3
i=<No author> <No copyright>
c=IN IP4 0.0.0.0
t=0 0
a=SdpplinVersion:1610642970
a=StreamCount:integer;1
a=Title:buffer;"cmVhbG1wMwA="
a=range:npt=0-39.471000
m=audio 0 RTP/AVP 101
b=AS:128
a=control:streamid=0
a=range:npt=0-39.471000
a=length:npt=39.471000
a=rtpmap:101 X-MP3-draft-00
a=mimetype:string;"audio/X-MP3-draft-00"
a=StartTime:integer;0
a=AvgBitRate:integer;128000
a=SampleRate:integer;44100
a=AvgPacketSize:integer;417
a=Preroll:integer;1000
a=NumChannels:integer;2
a=MaxPacketSize:integer;1024
a=ASMRuleBook:string;"AverageBandwidth=128000, AverageBandwidthStd=0,
Priority=9
;"

Created receiver for "audio/X-MP3-DRAFT-00" subsession (client ports
2032-2033)
Sending request: SETUP rtsp://media.schreiner.edu/realmp3.mp3/streamid=0
RTSP/1.
0
CSeq: 2
Transport: RTP/AVP;unicast;client_port=2032-2033


Received SETUP response: RTSP/1.0 200 OK
CSeq: 2
Date: Tue, 20 Nov 2001 00:14:25 GMT
Session: 26878-2
RDTFeatureLevel: 2
Transport: rtp/avp;unicast;client_port=2032-2033;server_port=14558-14559


Issued RTSP "SETUP" on "audio/X-MP3-DRAFT-00" subsession (client ports
2032-2033
)
Outputting data from the "audio/X-MP3-DRAFT-00" subsession to 'stdout'
Sending request: PLAY rtsp://media.schreiner.edu/realmp3.mp3/streamid=0
RTSP/1.0

CSeq: 3
Session: 26878-2
Range: npt=0-


Received PLAY response: RTSP/1.0 200 OK
CSeq: 3
Date: Tue, 20 Nov 2001 00:14:26 GMT
RTP-Info:
url=rtsp://media.schreiner.edu/realmp3.mp3/streamid=0;seq=0;rtptime=0


Issued RTSP "PLAY" command on "audio/X-MP3-DRAFT-00" subsession
Receiving streamed data (for up to 44.5 seconds)...
Closing "audio/X-MP3-DRAFT-00" subsession
Sending request: TEARDOWN rtsp://media.schreiner.edu/realmp3.mp3/streamid=0
RTSP
/1.0
CSeq: 4
Session: 26878-2
-------------------------------------------------------------------------

A new file - test.mp3 was created but it's size was 0.

Then I tried without the -a switch and the result was an empty file named
audio-X-MP3-DRAFT-00.

I placed a breakpoint in file openRTSP.CPP, function checkForPacketArrival.
It is called quite often by AlarmHandler::handleTimeout (wich in term was
called by DelayQueue::handleAlarm()). Sometimes [line 566] ((subsession =
iter.next()) != NULL) was true and an RTPSource object was created but the
second condition - (src != NULL &&
src->receptionStatsDB().numActiveSourcesSinceLastReset() > 0) was always
false.

Can anyone point me to possible reasons for not receiving the data?


Next question: the documentation says that testMPEGVideoStreamer streams
MPEG video to a multicast group 239.255.42.42, port 8888. How do I create
multicast group/port in our office network? I want to use the
testMPEGVideoStreamer - testMPEGVideoReceiver pair.



Thanks in advance

Vesselin


From <EMAIL-ADDRESS-HIDDEN> Mon Nov 19 19:07:05 2001
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Subject: Re: [Live-devel] openRTSP problem/setting up MPEG video
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Date: Mon Nov 19 19:08:02 2001
X-Original-Date: Mon, 19 Nov 2001 19:06:41 -0800
Status: O

At 05:23 PM 11/19/01, Vesselin Kostadinov wrote:
>I need to set-up a video broadcaster and a couple of receivers and I decided
>to try Live library. I managed to compile it with Borland C++ Builder but
>when running openRTSP I encountered a problem - the client was not receiving
>any data.

Vesselin,

Could you try running this same test
         openRTSP -a -V -n rtsp://media.schreiner.edu/realmp3.mp3 > test.mp3
using the pre-built version of "openRTSP" that you'll find in 
<http://www.live.com/openRTSP/binaries/windows/>, and please let me know 
the result.  This will help narrow down what the problem might be.

You might also try running pre-built Windows versions of some other test 
programs; you can find these at <http://www.live.com/windows/>


>Next question: the documentation says that testMPEGVideoStreamer streams
>MPEG video to a multicast group 239.255.42.42, port 8888. How do I create
>multicast group/port in our office network? I want to use the
>testMPEGVideoStreamer - testMPEGVideoReceiver pair.

You don't really 'create' a multicast group/port; instead, you choose 
one.  Multicast group addresses beginning with 239.255 (like the address 
239.255.42.42 that the test programs use) are reserved for use on local 
networks, so you should be able to choose any such address.  (You can also 
use any port number that's not being used by any other application or service.)

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Mon Nov 19 19:28:29 2001
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From: Vesselin Kostadinov <EMAIL-ADDRESS-HIDDEN>
To: 'Ross Finlayson' <EMAIL-ADDRESS-HIDDEN>
Cc: <EMAIL-ADDRESS-HIDDEN>
	 <live-devel(at)lists.sourceforge.net>
Subject: RE: [Live-devel] openRTSP problem/setting up MPEG video sender-re
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Date: Mon Nov 19 19:29:02 2001
X-Original-Date: Tue, 20 Nov 2001 13:58:57 +1030
Status: O

This is a multi-part message in MIME format.

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Hi Ross,

> Could you try running this same test
>          openRTSP -a -V -n rtsp://media.schreiner.edu/realmp3.mp3 >
test.mp3
> using the pre-built version of "openRTSP" that you'll find in 

The results with the pre-built openRTSP are pretty much the same:
test.mp3 with length 0. Perhaps I am having some sort of network
problem. We have an ADSL connection and our router is Cisco 800 series.
It's installer warned us that some UDP based games may not work.

But then why Real player can deliver the music?

openRTSP output:
--------------------------------------------------------
>openRTSP -a -V -n rtsp://media.schreiner.edu/realmp3.mp3 > test.mp3
Sending request: DESCRIBE rtsp://media.schreiner.edu/realmp3.mp3
RTSP/1.0
CSeq: 1
Accept: application/sdp


Received DESCRIBE response: RTSP/1.0 200 OK
CSeq: 1
Date: Tue, 20 Nov 2001 03:16:13 GMT
Set-Cookie:
cbid=hfhjcmiihjhkilemeomrqugqrojrktlufkfgkihligjkhiplrnqopppqnrksptk
uekcjehil;path=/;expires=Thu,31-Dec-2037 23:59:59 GMT
X-TSPort: 7802
vsrc:
http://media.schreiner.edu:8080/viewsource/template.html?nuyhtggaysz6nc6
vs
ffjsqt6725n10A1v1yeC3d4ngEt5o5gwuw4t6DC4jbh
Last-Modified: Tue, 09 Jan 2001 03:24:39 GMT
Content-base: rtsp://media.schreiner.edu/realmp3.mp3/
Content-type: application/sdp
Content-length: 647

v=0
o=- 979010679 979010679 IN IP4 207.71.37.7
s=realmp3
i=<No author> <No copyright>
c=IN IP4 0.0.0.0
t=0 0
a=SdpplinVersion:1610642970
a=StreamCount:integer;1
a=Title:buffer;"cmVhbG1wMwA="
a=range:npt=0-39.471000
m=audio 0 RTP/AVP 101
b=AS:128
a=control:streamid=0
a=range:npt=0-39.471000
a=length:npt=39.471000
a=rtpmap:101 X-MP3-draft-00
a=mimetype:string;"audio/X-MP3-draft-00"
a=StartTime:integer;0
a=AvgBitRate:integer;128000
a=SampleRate:integer;44100
a=AvgPacketSize:integer;417
a=Preroll:integer;1000
a=NumChannels:integer;2
a=MaxPacketSize:integer;1024
a=ASMRuleBook:string;"AverageBandwidth=128000, AverageBandwidthStd=0,
Priority=9
;"

Opened URL "rtsp://media.schreiner.edu/realmp3.mp3", returning a SDP
description
:
v=0
o=- 979010679 979010679 IN IP4 207.71.37.7
s=realmp3
i=<No author> <No copyright>
c=IN IP4 0.0.0.0
t=0 0
a=SdpplinVersion:1610642970
a=StreamCount:integer;1
a=Title:buffer;"cmVhbG1wMwA="
a=range:npt=0-39.471000
m=audio 0 RTP/AVP 101
b=AS:128
a=control:streamid=0
a=range:npt=0-39.471000
a=length:npt=39.471000
a=rtpmap:101 X-MP3-draft-00
a=mimetype:string;"audio/X-MP3-draft-00"
a=StartTime:integer;0
a=AvgBitRate:integer;128000
a=SampleRate:integer;44100
a=AvgPacketSize:integer;417
a=Preroll:integer;1000
a=NumChannels:integer;2
a=MaxPacketSize:integer;1024
a=ASMRuleBook:string;"AverageBandwidth=128000, AverageBandwidthStd=0,
Priority=9
;"

Created receiver for "audio/X-MP3-DRAFT-00" subsession (client ports
1298-1299)
Sending request: SETUP rtsp://media.schreiner.edu/realmp3.mp3/streamid=0
RTSP/1.
0
CSeq: 2
Transport: RTP/AVP;unicast;client_port=1298-1299


Received SETUP response: RTSP/1.0 200 OK
CSeq: 2
Date: Tue, 20 Nov 2001 03:16:13 GMT
Session: 26896-2
RDTFeatureLevel: 2
Transport: rtp/avp;unicast;client_port=1298-1299;server_port=28500-28501


Issued RTSP "SETUP" on "audio/X-MP3-DRAFT-00" subsession (client ports
1298-1299
)
Outputting data from the "audio/X-MP3-DRAFT-00" subsession to 'stdout'
Sending request: PLAY rtsp://media.schreiner.edu/realmp3.mp3/streamid=0
RTSP/1.0

CSeq: 3
Session: 26896-2
Range: npt=0-


Received PLAY response: RTSP/1.0 200 OK
CSeq: 3
Date: Tue, 20 Nov 2001 03:16:14 GMT
RTP-Info:
url=rtsp://media.schreiner.edu/realmp3.mp3/streamid=0;seq=0;rtptime=0


Issued RTSP "PLAY" command on "audio/X-MP3-DRAFT-00" subsession
Receiving streamed data (for up to 44.5 seconds)...
Closing "audio/X-MP3-DRAFT-00" subsession
Sending request: TEARDOWN
rtsp://media.schreiner.edu/realmp3.mp3/streamid=0 RTSP
/1.0
CSeq: 4
Session: 26896-2
--------------------------------------------------------



> You don't really 'create' a multicast group/port; instead, you choose 
> one.  Multicast group addresses beginning with 239.255 (like 
> the address 
> 239.255.42.42 that the test programs use) are reserved for 
> use on local 
> networks, so you should be able to choose any such address.  
> (You can also 
> use any port number that's not being used by any other 
> application or service.)

Thanks for that, I should have searched the newsgroups before bothering
you.

By the way, testMPEGVideoStreamer works fine as it is here, compiled
with Borland tools. I managed to view a short movie using Quicktime.


Regards

Vesselin

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From <EMAIL-ADDRESS-HIDDEN> Mon Nov 19 19:59:20 2001
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From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
Subject: RE: [Live-devel] openRTSP problem/setting up MPEG video
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Date: Mon Nov 19 20:00:01 2001
X-Original-Date: Mon, 19 Nov 2001 19:59:06 -0800
Status: O

At 07:28 PM 11/19/01, Vesselin Kostadinov wrote:
>Hi Ross,
>
> > Could you try running this same test
> >          openRTSP -a -V -n rtsp://media.schreiner.edu/realmp3.mp3 >
>test.mp3
> > using the pre-built version of "openRTSP" that you'll find in
>
>The results with the pre-built openRTSP are pretty much the same:
>test.mp3 with length 0. Perhaps I am having some sort of network
>problem. We have an ADSL connection and our router is Cisco 800 series.
>It's installer warned us that some UDP based games may not work.

Yes, I suspect that you have a firewall that is blocking UDP packets.  If 
so, you will need to reconfigure it to allow at least a single even/odd 
pair of client port numbers (for incoming RTP and RTCP packets).  (You can 
then use the
         -p <startingPort>
option to "openRTSP" to tell it to use these ports.


>But then why Real player can deliver the music?

Perhaps because Real Player is smart enough to ask to receive the streamed 
data via TCP, if UDP doesn't work.  "openRTSP" isn't (yet) smart enough to 
do this.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Tue Nov 20 00:02:54 2001
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Subject: [Live-devel] VC++6.0 Make file prob...Please First solve this.
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Date: Tue Nov 20 00:03:02 2001
X-Original-Date: 20 Nov 2001 08:02:40 -0000
Status: O

hi,=0D=0A=0D=0A  I alaready worked some devlopement on streaming media over=
 internet..we Chose JMF is ultimate plateform.. but when we come into perfo=
rmance i think native code is the best..boz JMF takes lot of system resourc=
es  and is not fast enough..This is the only site i found for VC streaming =
(Others are just frame transmission).I just test the executable's .=0D=0A=
=0D=0AtestMP3Receiver=0D=0AtestMP3Streamer=0D=0A=0D=0AI am able to stream t=
he mp3 file.. but not able to receive it..boz that receiver exe gives some =
error..=0D=0A=0D=0Ai tried using jmf for receiver end..it's quit well to st=
ream from one(testMP3Streamer)  and receive from other (JMF).it worked well=
.=0D=0A=0D=0A=0D=0Abut when i download the code .. and extract in linux for=
 getting make file ..it's not worked with my vc++6.0..first it asks some mi=
ssing file.. i just put all the file in current path.. then atlast it gives=
 me the below error..=0D=0A=0D=0A=0D=0A------------------------------------=
-------------------=0D=0A C:\Program Files\Microsoft Visual Studio\VC98\bin=
\cl -c -Ox -c -W3 -DCRTAPI1=3D_cdecl -DCRTAPI2=3D_cdecl -nologo -D_X86_=3D1=
 -D_WIN95 -D_WIN32_WINDOWS=3D0x0400 -D_WIN32_IE=3D0x0300 -DWINVER=3D0x0400 =
-DWIN32  -D_WIN32 -D_MT -D_DLL -MD -I. -IC:\Program Files\Mic=0D=0Arosoft V=
isual Studio\VC98\include BasicUsageEnvironment.cpp=0D=0ANMAKE : fatal erro=
r U1045: spawn failed : Invalid argument=0D=0AStop.=0D=0AError executing NM=
AKE.=0D=0A=0D=0ABasicUsageEnvironment1.exe - 1 error(s), 0 warning(s)   =0D=
=0A-------------------------------------------------------=0D=0A=0D=0Ai did=
't see any line like=0D=0A=0D=0AFROM ARCHIVE:=0D=0A=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=0D=0Afrom author.=
.=0D=0A=0D=0AHas anyone gotten these Makefiles to work with VC version 6.0?=
  Is it just =0D=0Aa matter of changing ( I  DID'T SEE ANYTHING LIKE THE BE=
LOW LINE IN WIN32CONFIG MAKE FILE)=0D=0A=0D=0A         !IF "$(MSVC_VERSION)=
" =3D=3D "5.0"=0D=0Ain "win32config" to=0D=0A         !IF "$(MSVC_VERSION)"=
 =3D=3D "6.0"=0D=0A=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=0D=0A=0D=0A=0D=0A=0D=0A=0D=0Ai saw in archives.=
 i found one answer from author about the documentation and stablized versi=
on of this code.. if he is not solving the basic prob.(compiler problems). =
Then how people will test your software.. so first solve this iniconsistenc=
y on latest version of VC++6.0.if you solve then people will test the softw=
are and will tell you the enhancement and other good ideas..=0D=0A =0A


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From: Michael Sommer <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] VC++6.0 Make file prob...Please First solve
  this.
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Date: Tue Nov 20 00:36:02 2001
X-Original-Date: Tue, 20 Nov 2001 09:35:00 +0100
Status: O

Hi,

I can't give you an answer to your questions but I can tell you my=20
experience with VC6.
I didn't had a problem with generating the LIVE library BUT I created a new=
=20
project within the Developer Studio and added all the source files. For the=
=20
test programs you have to create separate (console application) projects=20
where you have to add the 'Wsock32.lib' (Properties/Linker). I even linked=
=20
it with the MFC libraries to use the LIVE library with MFC applications.=20
And it works very well.
Some small hint: Did you set the TOOLS32 variable inside the makefile to=20
your VC6 installation? The first time I compiled the library I used the=20
delivered makefile (with my own settings of TOOL32) and it worked as well.

Michael

At 08:02 20.11.01 +0000, baskaran  veerapathiran wrote:
>Has anyone gotten these Makefiles to work with VC version 6.0?  Is it just
>a matter of changing ( I  DID'T SEE ANYTHING LIKE THE BELOW LINE IN=20
>WIN32CONFIG MAKE FILE)
>
>          !IF "$(MSVC_VERSION)" =3D=3D "5.0"
>in "win32config" to
>          !IF "$(MSVC_VERSION)" =3D=3D "6.0"
>=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D
>
>
>
>
>i saw in archives. i found one answer from author about the documentation=
=20
>and stablized version of this code.. if he is not solving the basic=20
>prob.(compiler problems). Then how people will test your software.. so=20
>first solve this iniconsistency on latest version of VC++6.0.if you solve=
=20
>then people will test the software and will tell you the enhancement and=20
>other good ideas..
>
>
>_______________________________________________
>Live-devel mailing list
>Live-devel(at)lists.sourceforge.net
>https://lists.sourceforge.net/lists/listinfo/live-devel

Dipl.-Inf. (FH)
Michael Sommer - Software Development
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Subject: [Live-devel] Re: Need  help..(was 'VC++6.0 Make file prob...Please First
 solve this.')
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Date: Tue Nov 20 06:50:10 2001
X-Original-Date: Tue, 20 Nov 2001 15:49:03 +0100
Status: O

Hi Baskaran,

It seems like it couldn't link the LIVE library. Here's a small checklist 
for you.
I assume you created a new project from the library source files (all 
directories without 'testProgs') (static library).
Lock if all files of the library are compiled and linked together to one 
library file (e.g. live.lib)
If you create new projects for each test program you have to add the 
'live.lib' to the library path of each project as well (additional to 
'wsock32.lib'). Otherwise the linker can't find and link the library to 
your application.
Try these things first and tell me what happend.

Michael

Please send your questions to the mailing-list to get response from other 
people as well. (Replay to all - cc:live-devel(at)lists.sourceforge.net)
At 14:27 20.11.01 +0000, baskaran  veerapathiran wrote:

>hi,
>   i tried to open new winconsole app for testMP3Streamer.cpp but it gives 
> some linker error..
>i think it requires some lib files.. but which one .. when i try to create 
>basicusageenvironment it gives fatal error1010: unexpected end of line. { 
>when i try to create it using the make file this is an application type so 
>bit confused.) ..
>
>what are the lib this  testMP3Streamer.cpp need to compile. (But for Jpeg 
>compression exe. we need to add some lib to project ..but i just add 
>needed source file it created the exe without asking lib..is there is any 
>setting for that.. without creating lib files.)



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Date: Tue Nov 20 14:18:24 2001
X-Original-Date: Tue, 20 Nov 2001 14:17:15 -0800
Status: O

At 06:49 AM 11/20/01, Michael Sommer wrote:
>I assume you created a new project from the library source files (all 
>directories without 'testProgs')

I would like to discourage people from doing this.  Instead, please try to 
use the Makefiles that have been provided, and build the code as separate 
projects ("UsageEnvironment", "groupsock", "liveMedia", 
"BasicUsageEnvironment").

There are two main reasons for doing this:
1/ Ease of support.  If people are having problems building the code on 
Windows, it's hard (if not impossible) for me to figure out what's going 
wrong it they're not using the Makefiles and projects that have been 
provided, but are instead trying to 'roll their own'.
2/ Flexibility.  Having separate projects makes it easier to modify or 
replace parts of the code.  For instance, you may wish to replace 
"BasicUsageEnvironment" with your own subclasses (e.g., to hook into an 
existing GUI's event handling mechanism).

If people are having problems building on Windows from the Makefiles that 
have been provided, then I'd really like to know about this.

One difficulty that I have been running into is that Unix symbolic links 
don't work in Windows, so if you begin by unpacking the source code on a 
Unix machine (which is what I recommend), then you will need to dereference 
all of the various symbolic links (to header files), before you copy the 
source tree over to a Windows machine.

Now, I *thought* that the "-h" (or "--derefence") option to "tar" would 
automatically derefence symbolic links - at least, that's what the man page 
seems to say.  However, I have never gotten this to work for me.  Does 
anyone know of a reliable way to dereference symbolic links when extracting 
a "tar" file??

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Tue Nov 20 14:42:07 2001
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Date: Tue Nov 20 14:43:01 2001
X-Original-Date: Tue, 20 Nov 2001 14:42:02 -0800
Status: O

Following up on my earlier message...

On reflection, I realize that the various symbolic links (to header files) 
in the code are causing so much grief (especially when trying to move the 
code to Windows) that it'd be better off if I got rid of them, and instead 
just relied upon the "-I" compiler flag to do the right thing.

So, sometime soon I'll probably generate a new version of the source code 
that eliminates the symbolic links.

	Ross.



From <EMAIL-ADDRESS-HIDDEN> Wed Nov 21 04:16:49 2001
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Subject: [Live-devel] working from mak files is good..but
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Date: Wed Nov 21 04:17:10 2001
X-Original-Date: 21 Nov 2001 12:17:06 -0000
Status: O

i tried every method but all are failed .. but from   Michael Sommer  way i=
 got everything worked fine..=0D=0A=0D=0ABut as you told this is not the ri=
ght way(boz it will loss it modularization ..) so it's good not doing like =
taking all the files into single lib..=0D=0A=0D=0ABut your make files are n=
ot working that's why people use their own ways..is it? ..( nobody will got=
 himself into trouble without using your mak files that's what these concep=
t are coming -> portabiliy )=0D=0A=0D=0AOK LET WE WILL FIND THE PROB.. AND =
TRY TO SOLVE THIS..i think you know that apache is running 60% of www as ht=
tp server boz they(archive) will solve each and every small prob. (if it's =
true ).=0D=0A=0D=0A=0D=0A=0D=0A=0D=0A=0D=0AI TRIED BELOW METHODS BEFORE I S=
ENT THE MAIL TO HERE THIS ARCHIVE..=0D=0A( i think that symblic link prob..=
 for sys/mman.h and features.h is it? ) i also tried by removeing sys from =
the include directives.. nothing happed).=0D=0A=0D=0A=0D=0A1.=0D=0A********=
***=0D=0A i am not have linux access last week ..so i tried the last method=
 you told.. by concatinating. ( .head win32config then .trail ) just for th=
at basicusage environement. but it gives the below error..=0D=0A=0D=0A=0D=
=0AAt compile time..=0D=0A=0D=0A(FROM THIS BELOW PATH YOU WILL KNOW THAT I =
CHANGED THE V.STUDIO PATH TO VC6.O INSTALLED ) IN TOOLS32)=0D=0ATOOLS32	=3D=
c:\Program Files\Microsoft Visual Studio\VC98=0D=0A=0D=0A=0D=0Ac:\Program F=
iles\Microsoft Visual Studio\VC98\bin\cl -c -Ox -c -W3 -DCRTAPI1=3D_cdecl -=
DCRTAPI2=3D_cdecl -nologo -D_X86_=3D1 -D_WIN95 -D_WIN32_WINDOWS=3D0x0400 -D=
_WIN32_IE=3D0x0300 -DWINVER=3D0x0400 -DWIN32  -D_WIN32 -D_MT -D_DLL -MD -I.=
 BasicUsageEnvironment.=0D=0Acpp=0D=0ANMAKE : fatal error U1045: spawn fail=
ed : Invalid argument=0D=0AStop.=0D=0AError executing NMAKE.=0D=0A=0D=0A=0D=
=0ABUT WHEN I JUST CUT AND PASTE THE LINE TO CMD PROMPT  IT GIVES THE BELOW=
 RESULT WHAT PROB. IN GUI BASED COMPI. SETTING ? IT PRODUCES .OBJ FILE.=0D=
=0A=0D=0AC:\Program Files\Microsoft Visual Studio\VC98\Bin>cl -c -Ox -c -W3=
 -DCRTAPI1=3D_cd=0D=0Aecl -DCRTAPI2=3D_cdecl -nologo -D_X86_=3D1 -D_WIN95 -=
D_WIN32_WINDOWS=3D0x0400 -D_WIN32=0D=0A_IE=3D0x0300 -DWINVER=3D0x0400 -DWIN=
32  -D_WIN32 -D_MT -D_DLL -MD -I. BasicUsageEnvironment.cpp=0D=0A=0D=0ABasi=
cUsageEnvironment.cpp=0D=0ABasicUsageEnvironment.cpp(173) : warning C4018: =
'=3D=3D' : signed/unsigned mismatch=0D=0A=0D=0AC:\Program Files\Microsoft V=
isual Studio\VC98\Bin>=0D=0ADirectory of C:\Program Files\Microsoft Visual =
Studio\VC98\Bin=0D=0A=0D=0A11/21/2001  05:17p               7,459 BasicUsag=
eEnvironment.obj=0D=0A=0D=0AOK JUST TELL ME WHERE IS THE PROB.. ANY PATH SE=
TTING..=0D=0A***********=0D=0A2. I ALSO TRIED BY EXTRACING THE FILES IN LIN=
UX ( AFTER MY LINUX ADM.. AVAILABILITY)..=0D=0A=0D=0AI GOT THE SAME THING..=
 BOZ I SAW BOTH ARE SAME.. THAT'S GENWINDOWMAKEFILE IS  AUTOMATED ..THIRD M=
ETHOD IS MANUAL CONCA.( HEAD WIN32 TRIAL)..THAT'S ALL..I GOT THE SAME ERROR=
 MSG. .=0D=0A=0D=0A=0D=0AWAITING FOR YOUR RESPONSE..( MEANWHILE I WILL TRY =
 TO GET VC5.0 BUT IT'S HARD TO GET OLD VERSIONS )..=0D=0A=0D=0A=0D=0ABY =0D=
=0ABASKARAN=0D=0A=0D=0A =0A


From <EMAIL-ADDRESS-HIDDEN> Wed Nov 21 09:56:53 2001
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Date: Wed Nov 21 09:57:06 2001
X-Original-Date: Wed, 21 Nov 2001 09:52:45 -0800
Status: O

At 04:17 AM 11/21/01, baskaran  veerapathiran wrote:
>But your make files are not working that's why people use their own 
>ways..is it?

OK, so let's see if we can figure out why the Makefiles are not working for 
some people.


>(FROM THIS BELOW PATH YOU WILL KNOW THAT I CHANGED THE V.STUDIO PATH TO 
>VC6.O INSTALLED ) IN TOOLS32))
>TOOLS32 =c:\Program Files\Microsoft Visual Studio\VC988

I wonder if perhaps the VC++ tools are getting confused by the fact that 
your TOOLS32 path has spaces in directory names.  Please try changing this to
         TOOLS32 = "c:\Program Files\Microsoft Visual Studio\VC988"
(i.e., putting quotes around the path) and see if that makes any difference.


>c:\Program Files\Microsoft Visual Studio\VC98\bin\cl -c -Ox -c -W3 
>-DCRTAPI1=_cdecl -DCRTAPI2=_cdecl -nologo -D_X86_=1 -D_WIN95 
>-D_WIN32_WINDOWS=0x0400 -D_WIN32_IE=0x0300 -DWINVER=0x0400 
>-DWIN32  -D_WIN32 -D_MT -D_DLL -MD -I. BasicUsageEnvironment..
>cppp
>NMAKE : fatal error U1045: spawn failed : Invalid argument

Does anyone know what this error means?


>BUT WHEN I JUST CUT AND PASTE THE LINE TO CMD PROMPT  IT GIVES THE BELOW 
>RESULT WHAT PROB. IN GUI BASED COMPI. SETTING ? IT PRODUCES .OBJ FILE..
  ..
>WAITING FOR YOUR RESPONSE..( MEANWHILE I WILL TRY  TO GET VC5.0 BUT IT'S 
>HARD TO GET OLD VERSIONS )...

Although I have been doing all of my own testing using VC5.0, I have heard 
of other people succeeding in building the code - using the provided 
Makefiles - using VC6.0.

Can anyone help figure out Baskaran's problem?

         Ross.





From <EMAIL-ADDRESS-HIDDEN> Wed Nov 21 11:47:38 2001
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   "baskaran  veerapathiran" <v_baskaran_2000(at)rediffmail.com>
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Date: Wed Nov 21 11:48:01 2001
X-Original-Date: Wed, 21 Nov 2001 11:46:58 -0800
Status: O

First of all a short message to Baskaran: your posting is quite hard to read
since all the lines seems to lump together.  Is there an issue with your
mail program causing linefeed and carriage return characters to be dropped?

Now to Baskaran's problem.  I have no problem running the makefile with VC6.
I did do one thing different though, which is to use the following in the
makefile:

TOOLS32 =	c:\PROGRA~1\MICROS~2\VC98

Of course, the 8.3 version of your directory/file name will differ. So run
"dir" to see the proper ones on your machine.  My machine runs WinME.  I
believe it's the same for 95/98.  If you are on XP or NT, I suspect you
won't have the same issue.

I also suspect the "spawn" error message is an indication that the main "CL"
program cannot run the subprocesses such as preprocessor, linker, etc.

I hope this helps,

Dan

> -----Original Message-----
> From: live-devel-admin(at)lists.sourceforge.net
> [mailto:live-devel-admin(at)lists.sourceforge.net]On Behalf Of Ross
> Finlayson
> Sent: Wednesday, November 21, 2001 9:53 AM
> To: baskaran veerapathiran
> Cc: live-devel(at)lists.sourceforge.net
> Subject: Re: [Live-devel] working from mak files is good..but
>
>
> At 04:17 AM 11/21/01, baskaran  veerapathiran wrote:
> >But your make files are not working that's why people use their own
> >ways..is it?
>
> OK, so let's see if we can figure out why the Makefiles are not
> working for
> some people.
>
>
> >(FROM THIS BELOW PATH YOU WILL KNOW THAT I CHANGED THE V.STUDIO PATH TO
> >VC6.O INSTALLED ) IN TOOLS32))
> >TOOLS32 =c:\Program Files\Microsoft Visual Studio\VC988
>
> I wonder if perhaps the VC++ tools are getting confused by the fact that
> your TOOLS32 path has spaces in directory names.  Please try
> changing this to
>          TOOLS32 = "c:\Program Files\Microsoft Visual Studio\VC988"
> (i.e., putting quotes around the path) and see if that makes any
> difference.
>
>
> >c:\Program Files\Microsoft Visual Studio\VC98\bin\cl -c -Ox -c -W3
> >-DCRTAPI1=_cdecl -DCRTAPI2=_cdecl -nologo -D_X86_=1 -D_WIN95
> >-D_WIN32_WINDOWS=0x0400 -D_WIN32_IE=0x0300 -DWINVER=0x0400
> >-DWIN32  -D_WIN32 -D_MT -D_DLL -MD -I. BasicUsageEnvironment..
> >cppp
> >NMAKE : fatal error U1045: spawn failed : Invalid argument
>
> Does anyone know what this error means?
>
>
> >BUT WHEN I JUST CUT AND PASTE THE LINE TO CMD PROMPT  IT GIVES THE BELOW
> >RESULT WHAT PROB. IN GUI BASED COMPI. SETTING ? IT PRODUCES .OBJ FILE..
>   ..
> >WAITING FOR YOUR RESPONSE..( MEANWHILE I WILL TRY  TO GET VC5.0 BUT IT'S
> >HARD TO GET OLD VERSIONS )...
>
> Although I have been doing all of my own testing using VC5.0, I
> have heard
> of other people succeeding in building the code - using the provided
> Makefiles - using VC6.0.
>
> Can anyone help figure out Baskaran's problem?
>
>          Ross.
>
>
>
>
> _______________________________________________
> Live-devel mailing list
> Live-devel(at)lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/live-devel
>



From <EMAIL-ADDRESS-HIDDEN> Thu Nov 22 13:18:13 2001
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Date: Thu Nov 22 13:19:01 2001
X-Original-Date: Thu, 22 Nov 2001 13:14:49 -0800
Status: O

Baskaran,

In the future, please try to use the mailing list to ask questions about 
the library code.  (For one-on-one email, I generally give priority to 
consulting clients.)


>  1.(I am able to  TRANSMIT MP3 file using testMP3streamer(win2000) and 
> received using JMFSTUDIO(win98/ME/win2000))

That's great to hear.

>  2. testMp3receiver is not working on any machine.(any winsock problem 
> 32bit and older version conflict?)..
>what is solution for these prob...
>(on win2000 it gives some strange output in console then it gives 
>conti..beep ))

That's because "testMP3Receiver" is outputting to stdout.  If you run
         testMP3Receiver > foo.mp3
you'll get a MP3 file that you can play using a MP3 player.


>  3. For that testMPEGVideoStreamer.exe. i searched net and got some 
> elementary mpeg mpeg-1) but i am not able to receive by 
> testMpegvideoreceiver.exe and JMStudio..)any format prob..sending is working
>  Then i got following  o/pp
>(for certain mpg files))
>Beginning streaming....
>Beginning to read from file....
>MPEGVideoStreamParser::parseSlice(): Saw unexpected code 0x000001baa
>MPEGVideoStreamParser::parseSlice(): Saw unexpected code 0x000001e00

That's probably because the MPEG video file that you're using is a MPEG 
Program Stream (i.e., with an audio track as well as a video track).  By 
default, "testMPEGVideoStreamer" assumes that the input is a MPEG Video 
Elementary Stream. To change this, uncomment the line
         //#define SOURCE_IS_PROGRAM_STREAM 1
in "testMPEGVideoStreamer.cpp"


>Then how i can view the Mpeg video files that are retransmitted using 
>testMPEGVideoReceiver.exe to some viewable form without using any viewer

To actually view the received MPEG video data, you'll need a MPEG video 
decoder.  The LIVE.COM libraries don't (yet) include any video (or audio) 
decoding code - that's why you need a MPEG video player (and, for MP3, a 
MP3 player) to actually play the received data.

         Ross.





From <EMAIL-ADDRESS-HIDDEN> Mon Nov 26 01:31:22 2001
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Date: Mon Nov 26 01:32:01 2001
X-Original-Date: Mon, 26 Nov 2001 01:30:24 -0800
Status: O

As promised, I have now released a new version of the "LIVE.COM Streaming 
Media" code that eliminates the widely hated symbolic links to header 
files.  Instead, each header file appears in only one place (an "include/" 
subdirectory for each library source tree), and the "-I" compiler flag is 
used to locate the various public header files.

This should make it easier to copy the source code to a Windows development 
environment.  (There should no longer be any excuse for people to try to 
build all of the code together as a single project.)

You can get the new code release from <http://www.live.com/liveMedia/> (for 
some reason new releases take a while to show up in SourceForge).

Also, thanks to a contribution by Vesselin Kostadinov, there is now support 
for the Borland C++ development environment on Windows (as an alternative 
to Microsoft Visual C++).  For instructions on how to use this, see 
<http://live.sourceforge.net/#config-windows>.  (I have not tested this 
myself, so if there are any problems with this please let me know.)

	Ross.



From <EMAIL-ADDRESS-HIDDEN> Mon Nov 26 06:14:39 2001
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Subject: [Live-devel] current play time
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Date: Mon Nov 26 06:15:08 2001
X-Original-Date: Mon, 26 Nov 2001 15:14:13 +0100
Status: O

Hello,

I need some help again.
How do I get the current play time (current position) of a streaming file? 
The 'getPlayTime()' method of e.g. class 'MP3FileSource' needs a number of 
frames. Where do I get the number of already sent frames?
Thanks

Michael



From <EMAIL-ADDRESS-HIDDEN> Mon Nov 26 10:17:04 2001
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Date: Mon Nov 26 10:18:01 2001
X-Original-Date: Mon, 26 Nov 2001 10:15:56 -0800
Status: O

At 06:14 AM 11/26/01, Michael Sommer wrote:
>How do I get the current play time (current position) of a streaming file? 
>The 'getPlayTime()' method of e.g. class 'MP3FileSource' needs a number of 
>frames. Where do I get the number of already sent frames?

Unfortunately there isn't any way right now to get this information 
(without modifying the library code).  Instead, I suggest that you just 
keep track of the current play time directly - from 'wall clock' 
time.  I.e., make a note of the time when you started playing, and then - 
whenever you want to know the current play time within the file - just 
subtract this start time from the current time.  This works because the 
streaming happens at close to 'real time' (at least, from a human's 
perspective).

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Tue Nov 27 03:06:58 2001
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Date: Tue Nov 27 03:07:03 2001
X-Original-Date: Tue, 27 Nov 2001 03:06:45 -0800
Status: O

Thanks to Bob Nadler, who was able to get the library code (& test 
programs) running on the QNX4 embedded OS (using the Watcom compiler).

Bob has contributed a "config.qnx4" file.  This is contained in a new 
library release (2001.11.27).

	Ross.



From <EMAIL-ADDRESS-HIDDEN> Tue Nov 27 13:55:19 2001
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Date: Tue Nov 27 13:56:01 2001
X-Original-Date: Tue, 27 Nov 2001 22:05:51 +0000
Status: O

Sorry to bore you if it all sounds too basic, but info seems hard to get.
Am I wrong in thinking that most if not ALL streaming mp3 on the net is 
done with http?
Even live365.com's stations?
If you have the same bandwith and dodgy net conditions would a rtsp/rtp 
client have less re-buffering or better quality than an http equiv?
Is there any more basic info on the header files? I can disect the code 
eventually, but as I'm new to sockets and net coding, its slow work and I'm 
busy on degree stuff.(computing!).
Any good (and recent) books on the streaming technology?




From <EMAIL-ADDRESS-HIDDEN> Tue Nov 27 14:22:24 2001
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Date: Tue Nov 27 14:23:02 2001
X-Original-Date: Tue, 27 Nov 2001 14:21:06 -0800
Status: O

At 02:05 PM 11/27/01, iainhaywood(at)madasafish.com wrote:
>Am I wrong in thinking that most if not ALL streaming mp3 on the net is 
>done with http?

Yes, most streaming MP3 is being done with HTTP, but certainly not 'all' 
(otherwise this mailing list, and lots of other groups, wouldn't exist :-)

>If you have the same bandwith and dodgy net conditions would a rtsp/rtp 
>client have less re-buffering or better quality than an http equiv?

It depends on what you mean by "quality".  Because HTTP runs on top of TCP, 
which is a reliable transport protocol, if you send a stream using HTTP, 
then all of the data will be received, without loss.  However, the cost of 
this is that the retransmissions necessary to ensure reliability can cause 
large delays before the receiver gets all of the data that it needs to 
play.  That's why HTTP streaming players usually do quite a lot of 
buffering of incoming data before they start to play it.

RTP, on the other hand, runs on top of UDP, which is a datagram protocol, 
so some data will be lost if the connection is lossy.  However, RTP is 
better if you want close to 'real time' response (note that "RTP" stands 
for "Real-Time Protocol").  Also, if you're streaming using multicast, then 
you have to use RTP, because multicast doesn't work over TCP.

>Is there any more basic info on the header files?

Not really.  The best way to learn about how the code works is to build and 
run the test programs, and then look at how they work.  Of course, feel 
free to use this mailing list if you have any questions.

For more information about RTP itself, see the latest version of the RTP 
spec, which is online at 
<http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-new-10.txt>. 
Also, for information about RTSP (a control protocol that can be used to 
start/stop RTP streams), see <http://www.rtsp.org/>

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Thu Nov 29 11:39:32 2001
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Date: Thu Nov 29 11:40:02 2001
X-Original-Date: Thu, 29 Nov 2001 11:36:51 -0800
Status: O

At 10:18 AM 11/29/01, Robert Nadler wrote:
>I'm trying to get RealPlayer (8 Basic) to run a live stream via your RTSP
>server. The same RTP stream works fine with QuickTime and JMStudio.
>RealPlayer sends OPTIONS, DESCRIBE, and SETUP messages which appear to be
>responded to correctly. It then just hangs until it timesout (a TEARDOWN is
>sent).

This is due to a bug in RealPlayer - for some reason it can't play 
multicast sessions using RTSP.  (If you look at the "SETUP" command that it 
sends to the RTSP server, you'll note that "RTP/AVP;multicast" is not 
listed as one of the "Transport:" options.)

I've reported this bug to RealNetworks.  One thing you *can* do, however, 
is take the SDP information that the server returns in response to the 
"DESCRIBE" command (everything from "v=0" onwards), put this in a '.sdp' 
file, and open that file with RealPlayer.  (However, I'm not sure if 
RealPlayer can play MPEG *video* multicast streams.)

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Thu Nov 29 12:29:29 2001
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Date: Thu Nov 29 12:30:07 2001
X-Original-Date: Thu, 29 Nov 2001 12:30:19 -0800
Status: O

At 11:36 AM 11/29/01 -0800, Ross Finlayson wrote:
>At 10:18 AM 11/29/01, Robert Nadler wrote:
>>I'm trying to get RealPlayer (8 Basic) to run a live stream via your RTSP
>>server. The same RTP stream works fine with QuickTime and JMStudio.
>>RealPlayer sends OPTIONS, DESCRIBE, and SETUP messages which appear to be
>>responded to correctly. It then just hangs until it timesout (a TEARDOWN is
>>sent).
>
>This is due to a bug in RealPlayer - for some reason it can't play 
>multicast sessions using RTSP.  (If you look at the "SETUP" command that 
>it sends to the RTSP server, you'll note that "RTP/AVP;multicast" is not 
>listed as one of the "Transport:" options.)

Yup.  I've been pestering our dev staff to get this fixed.  No promises, 
though.

>I've reported this bug to RealNetworks.  One thing you *can* do, however, 
>is take the SDP information that the server returns in response to the 
>"DESCRIBE" command (everything from "v=0" onwards), put this in a '.sdp' 
>file, and open that file with RealPlayer.  (However, I'm not sure if 
>RealPlayer can play MPEG *video* multicast streams.)

The answer depends on the player version.  RealPlayer 8 supports MPEG Video 
via a plugin developed by Digital Bitcasting (now part of EMC).  That 
plugin does not support MPEG via RTP.

The RealOne Player (our latest version) supports MPEG via a plugin 
developed here at RealNetworks.  The new plugin supports MPEG via RTP, but 
is still a little fragile in its current release (as I remember, the video 
ends up looking pretty garbled from the Live.com server).  Keep an eye out 
for the next release of the player, and give it a whirl.

Rob



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Date: Mon Dec  3 15:31:01 2001
X-Original-Date: Mon, 3 Dec 2001 15:34:25 -0800
Status: O

Hi, all:

   After I read the openRTSP.cpp source code, I am
wondering if there is any other way beside exit(0) to 
break the while(1) loop in the blockMyself()?

   Thanks in advance.

Sincerely, Yanito



From <EMAIL-ADDRESS-HIDDEN> Mon Dec 03 16:40:14 2001
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Date: Mon Dec  3 16:41:04 2001
X-Original-Date: Mon, 03 Dec 2001 16:38:13 -0800
Status: O

At 03:34 PM 12/3/01, Yun Yan wrote:
>Hi, all:
>
>    After I read the openRTSP.cpp source code, I am
>wondering if there is any other way beside exit(0) to
>break the while(1) loop in the blockMyself()?

Yanito,

Yes, there are a couple of ways that you can do this.

The simplest way is to use the (optional) "watchVariable" argument to 
blockMyself().  Note the definition of blockMyself() (in 
"UsageEnvironment/include/UsageEnvironment.hh"):

   virtual void blockMyself(char* watchVariable = NULL) = 0;
         // Stops the current thread of control from proceeding,
         // but allows delayed tasks (and/or background I/O handling)
         // to proceed.
         // (If "watchVariable" is not NULL, then we return from this
         // routine when *watchVariable != 0)

So, you can do the following:
char watchVariable = 0;
...
blockMyself(&watchVariable);

and then, if you want to exit from this (e.g., from one of the data 
handlers), just do:
         extern char watchVariable;
         watchVariable = 1;

Of course, another way to break from blockMyself() is to write your own 
subclass of "TaskScheduler" that implements blockMyself() however you 
like.  (The "BasicTaskScheduler" class that's provided in the 
"BasicUsageEnvironment" library is just one possible implementation of 
"TaskScheduler".)

         Ross.





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Date: Mon Dec  3 19:34:05 2001
X-Original-Date: Mon, 3 Dec 2001 19:38:20 -0800
Status: O

Thank you very much. 
I didn't notice there is a newer version of library.

-----Original Message-----
From: <EMAIL-ADDRESS-HIDDEN>
[mailto:live-devel-admin(at)lists.sourceforge.net]On Behalf Of Ross
Finlayson
Sent: Monday, December 03, 2001 4:38 PM
To: <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] (no subject)


At 03:34 PM 12/3/01, Yun Yan wrote:
>Hi, all:
>
>    After I read the openRTSP.cpp source code, I am
>wondering if there is any other way beside exit(0) to
>break the while(1) loop in the blockMyself()?

Yanito,

Yes, there are a couple of ways that you can do this.

The simplest way is to use the (optional) "watchVariable" argument to 
blockMyself().  Note the definition of blockMyself() (in 
"UsageEnvironment/include/UsageEnvironment.hh"):

   virtual void blockMyself(char* watchVariable = NULL) = 0;
         // Stops the current thread of control from proceeding,
         // but allows delayed tasks (and/or background I/O handling)
         // to proceed.
         // (If "watchVariable" is not NULL, then we return from this
         // routine when *watchVariable != 0)

So, you can do the following:
char watchVariable = 0;
...
blockMyself(&watchVariable);

and then, if you want to exit from this (e.g., from one of the data 
handlers), just do:
         extern char watchVariable;
         watchVariable = 1;

Of course, another way to break from blockMyself() is to write your own 
subclass of "TaskScheduler" that implements blockMyself() however you 
like.  (The "BasicTaskScheduler" class that's provided in the 
"BasicUsageEnvironment" library is just one possible implementation of 
"TaskScheduler".)

         Ross.




_______________________________________________
Live-devel mailing list
Live-devel(at)lists.sourceforge.net
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From <EMAIL-ADDRESS-HIDDEN> Tue Dec 04 16:10:13 2001
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Subject: [Live-devel] connecting to several RTP servers
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Date: Tue Dec  4 16:11:01 2001
X-Original-Date: Wed, 5 Dec 2001 10:40:18 +1030
Status: O

Hi

I am trying to connect 1 RTP client to 2 RTP servers.

My client program uses a subclass of MultiFramedRTPSource and my server
programs use a subclass of MultiFramedRTPSink. 

In the client program I call MultiFramedRTPSource::networkReadHandler for
both instances of the MultiFramedRTPSource subclass.

The problem is that the 2 servers multicast the data at different rates, but
the client reads the data at the rate of the slowest server. I believe that
the reason for this is the blocking read from the sockets - the program
first reads ~1500 bytes from socket 1, then the same amount of data from
socket 2, then again and again. The final result is that the read rate
(bytes per second) is the same for all instances of MultiFramedRTPSource.
And because the servers' rates are different the packets from the faster
server are being buffered somewhere in a Windows internal buffer - until
that buffer is overflown, parts of the data are lost and all frames become
corrupted.

I would be happier if I am able to drop a frame or two from the faster
server but avoid such a condition.

At the moment when I receive a corrupted frame I do a couple of extra reads
from the socket in question. It sort of works but not as good as it should.

Anyone with an idea how to tackle this?

Thanks in advance

Vesselin


From <EMAIL-ADDRESS-HIDDEN> Tue Dec 04 17:00:34 2001
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Date: Tue Dec  4 17:01:09 2001
X-Original-Date: Tue, 04 Dec 2001 16:56:50 -0800
Status: O

At 04:10 PM 12/4/01, Vesselin Kostadinov wrote:
>I am trying to connect 1 RTP client to 2 RTP servers.
>
>My client program uses a subclass of MultiFramedRTPSource and my server
>programs use a subclass of MultiFramedRTPSink.
>
>In the client program I call MultiFramedRTPSource::networkReadHandler for
>both instances of the MultiFramedRTPSource subclass.

Vesselin,

First off, I'm curious as to why you (apparently) needed to redefine 
doGetNextFrame() when you subclassed "MultiFramedRTPSource"?  If you look 
at the other subclasses of this that are already defined in the library - 
e.g., MPEGAudioRTPSource, MPEGVideoRTPSource, SimpleRTPSource - you'll see 
that these don't need to redefine doGetNextFrame(); instead, they just use 
the doGetNextFrame() that's already implemented in MultiFramedRTPSource.

Following on from this - if your RTP payload format is very simple - 
without any special header following the RTP header - then perhaps you can 
just use "SimpleRTPSource", without having to define your own 
subclass?  Out of curiosity - which RTP payload format are you implementing?


>The problem is that the 2 servers multicast the data at different rates, but
>the client reads the data at the rate of the slowest server. I believe that
>the reason for this is the blocking read from the sockets

Note that the socket read operation that occurs in 
MultiFramedRTPSource::networkReadHandler() (when it calls 
"source->RTPgs()->handleRead()") isn't *really* blocking, because 
"networkReadHandler()" should have been called only when data is available 
to be read on the socket.  (This is assuming that you called 
"turnOnBackgroundReadHandling()" with the correct socket number and handler 
function as arguments.  Once again, the existing doGetNextFrame() 
implementation already does this, so you should look into whether you can 
get away without reimplementing it.)


>And because the servers' rates are different the packets from the faster
>server are being buffered somewhere in a Windows internal buffer - until
>that buffer is overflown, parts of the data are lost and all frames become
>corrupted.

One thing you can try doing is increasing the size of the OS's internal 
buffer that it uses for receiving data on each socket.  E.g., note the 
calls to "increaseReceiveBufferTo()" in the implementations of several of 
the RTP source classes.  (Again, if you're able to use "SimpleRTPSource", 
then this will be done for you automatically.)

If this doesn't work, then unfortunately I'm not sure what might be wrong - 
maybe some Windoze brain damage...

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Tue Dec 04 21:04:30 2001
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To: <EMAIL-ADDRESS-HIDDEN>
Subject: RE: [Live-devel] connecting to several RTP servers
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Date: Tue Dec  4 21:05:07 2001
X-Original-Date: Wed, 5 Dec 2001 15:34:31 +1030
Status: O

> First off, I'm curious as to why you (apparently) needed to redefine 
> doGetNextFrame() when you subclassed "MultiFramedRTPSource"?  

Ross,

In fact I do not redefine doGetNextFrame()

> Following on from this - if your RTP payload format is very simple - 
> without any special header following the RTP header - then 
> perhaps you can 
> just use "SimpleRTPSource", without having to define your own 
> subclass?

Did that and got the same result, with out-of-the-box SimpleRTPSource.


> Out of curiosity - which RTP payload format are you implementing?

A sequence of (variable sized) structures with fixed size header (size,
timestamp and few other things) and a JPEG image. I used payload type 26
(JPEG) in the constructor of the SimpleRTPSource.


> Note that the socket read operation that occurs in 
> MultiFramedRTPSource::networkReadHandler() (when it calls 
> "source->RTPgs()->handleRead()") isn't *really* blocking, because 
> "networkReadHandler()" should have been called only when data 
> is available to be read on the socket.  

I did some statistical analysis. In GroupsockHelper.cpp I added:

static int socketReads[1024];
static int socketTotalData[1024];
...

		bytesRead = recvfrom(socket, (char*)buffer, bufferSize, 0,
							 (struct
sockaddr*)&fromAddress,
							 &addressSize);
            socketReads[socket]++;
            socketTotalData[socket] += bytesRead;

I placed a breakpoint soon after the program started (while there were still
no data distortions) and I got the following results:

socketReads[SocketToTheFASTERServer] = 126
socketReads[SocketToTheSLOWERServer] = 127
socketTotalData[SocketToTheFASTERServer] = 157196
socketTotalData[SocketToTheSLOWERServer] = 172804

Average packet size from the faster server = 1248
Average packet size from the slower server = 1361

Then I restarted the program, let it run for a while (with noticeable data
loss) and again placed a breakpoint. Results:

socketReads[SocketToTheFASTERServer] = 9280
socketReads[SocketToTheSLOWERServer] = 9215
socketTotalData[SocketToTheFASTERServer] = 11283392
socketTotalData[SocketToTheSLOWERServer] = 12531744

Average packet size from the faster server = 1216
Average packet size from the slower server = 1359

Conclusions:

The number of packets read from the faster server is bigger than the number
of packets read from the slower server. This means that the read from the
socket is not a blocking read - as you mentioned.

The average packet size from the faster server becomes smaller - perhaps
because some data has been lost. In fact each server multicasts its own
sequence of 4 static images so the average packet size should stay constant.
What is worse, this data loss occurs before the read from the socket and I
have no idea how to prevent it.


> One thing you can try doing is increasing the size of the 
> OS's internal 
> buffer that it uses for receiving data on each socket.  E.g., 
> note the 
> calls to "increaseReceiveBufferTo()" in the implementations 
> of several of 
> the RTP source classes.  (Again, if you're able to use 
> "SimpleRTPSource", 
> then this will be done for you automatically.)

I increased that size from 50 to 250 KB and as a result the "playback" ran
Ok for longer. After that I've got the same problem. 

Have you got function checkTheDataSizeInTheReceiveBuffer() handy?



> maybe some Windoze brain damage...

Yeah, sometimes it is just too easy to put "Windoze" and "brain damage" in
one sentence.


Anyway, thanks for your help.


Vesselin


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Date: Wed Dec  5 01:31:10 2001
X-Original-Date: Wed, 5 Dec 2001 14:30:19 +0500
Status: O

Hi
I have downloaded the new source .Code is compiled successfully but when i
run openRTSP client on unix it say that
"This version of "./openRTSP" is out-of-date.  To download an up-to-date
version, visit <http://www.live.com/openRTSP/>, or build a new version from
the "LIVE.COM Streaming Media" source code at
<http://live.sourceforge.net/>"


Am i doing something wrong or some error in new release?

Regards




From <EMAIL-ADDRESS-HIDDEN> Wed Dec 05 01:42:26 2001
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Date: Wed Dec  5 01:43:02 2001
X-Original-Date: Wed, 5 Dec 2001 14:42:07 +0500
Status: O

why the   testMPEGVideoServer  is not built?
is it a makefile problem or is it deliberately not compiled?

Regards




From <EMAIL-ADDRESS-HIDDEN> Wed Dec 05 01:47:30 2001
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Date: Wed Dec  5 01:48:05 2001
X-Original-Date: Wed, 05 Dec 2001 01:47:18 -0800
Status: O

At 01:30 AM 12/5/01, Mumtaz Ahmad wrote:
>I have downloaded the new source .Code is compiled successfully but when i
>run openRTSP client on unix it say that
>"This version of "./openRTSP" is out-of-date.  To download an up-to-date
>version, visit <http://www.live.com/openRTSP/>, or build a new version from
>the "LIVE.COM Streaming Media" source code at
><http://live.sourceforge.net/>"
>
>Am i doing something wrong or some error in new release?

Near the beginning of the file "openRTSP.cpp", you'll see a line beginning
         #define EXPIRATION ...
If you comment out or delete this line, and recompile the program, it will 
no longer complain about being 'out-of-date'.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Wed Dec 05 01:56:16 2001
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Date: Wed Dec  5 01:57:04 2001
X-Original-Date: Wed, 05 Dec 2001 01:56:13 -0800
Status: O

At 01:42 AM 12/5/01, Mumtaz Ahmad wrote:
>why the   testMPEGVideoServer  is not built?
>is it a makefile problem or is it deliberately not compiled?

"testMPEGVideoServer" is deliberately not built in the Makefile, because 
this is not a complete program.  (It relies upon a - currently 
unimplemented - source object called "MPEGVideoDeviceSource" (e.g., 
encapsulating a MPEG encoder) before it can work.)

         Ross.

ps. In the future, please add subject lines to your messages.



From <EMAIL-ADDRESS-HIDDEN> Wed Dec 05 12:31:14 2001
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Date: Wed Dec  5 12:32:07 2001
X-Original-Date: Wed, 05 Dec 2001 12:26:27 -0800
Status: O

At 09:04 PM 12/4/01, Vesselin Kostadinov wrote:
>The average packet size from the faster server becomes smaller - perhaps
>because some data has been lost.

If any data is getting lost (either on the network, or within the OS), then 
this loss should be complete packets, not partial packets.  So, the average 
size of received packets should not be changing.  Or are you seeing any 
incomplete and/or malformed packets, rather than just missing packets?

>I increased that size from 50 to 250 KB and as a result the "playback" ran
>Ok for longer. After that I've got the same problem.

OK, this suggests that the process that's reading the incoming packets 
(i.e., using the LIVE.COM library code) is not able to keep up with the 
arrival of incoming network packets.  I.e., incoming network packets fill 
up the OS's internal buffer at a faster rate than they can be read, and 
eventually this buffer fills up and packets start getting lost.  I suspect 
that this has to do the brain-damaged task scheduling that's in Windoze 9x 
- apparently it takes a ridiculous amount of time to reschedule a task 
after it calls select() - even if select() is able to return immediately 
(e.g., because data is already available on the socket(s) that it's reading 
from).

Do you find that this problem occurs on Windows NT, 2000, or XP also - or 
just on Windoze 9x?  I think it's only 9x that has the brain-damaged task 
scheduling.

In any case, I think there's some OS call you can make in Windoze to 
increase the scheduling priority of you application.  You may find that 
this overcomes the problem.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Wed Dec 05 18:09:33 2001
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Date: Wed Dec  5 18:10:03 2001
X-Original-Date: Thu, 6 Dec 2001 12:39:32 +1030
Status: O

> If any data is getting lost (either on the network, or within the OS),
then 
> this loss should be complete packets, not partial packets. 

Well, we do not know what Windows is doing. If there are only, say, 1000
bytes available in the receive buffer and a new packet containing 1480 bytes
is coming then Windows may decide to get whatever it can (only 1000 bytes)
and drop the remaining 480 bytes. This may explain the lower average packet
size.


>... Or are you seeing any 
> incomplete and/or malformed packets, rather than just missing packets?

I just see invalid data at the end. But because of the way I calculate the
average packet size (totalling bytesRead straight after the call to
recvfrom...) and I know that the source supplies data packets with constant
average size I believe that quite a few incomplete packets are being
received by recvfrom().

> >I increased that size from 50 to 250 KB and as a result the "playback"
ran
> >Ok for longer. After that I've got the same problem.
> 
> OK, this suggests that the process that's reading the incoming packets 
> (i.e., using the LIVE.COM library code) is not able to keep up with the 
> arrival of incoming network packets. 

Windows reported less than 40% cpu usage during playback from both sources.
So if the CPU is not the bottleneck then perhaps the task scheduler is the
bad guy.

> Do you find that this problem occurs on Windows NT, 2000, ...

It is Windows 2000 Pro, SP2 


> In any case, I think there's some OS call you can make in Windoze to 
> increase the scheduling priority of you application.  You may 
> find that this overcomes the problem.

A call to SetPriorityClass(GetCurrentProcess(), HIGH_PRIORITY_CLASS) did not
result in a noticeable improvement.

I thing I will stick to the original hack - reading more packets when I
detect corrupted data. This leaves the user with the illusion that the
system is running fine. Perhaps this is as good as one can get under
windows.

Once again - thanks for your help

Regards


Vesselin


From <EMAIL-ADDRESS-HIDDEN> Wed Dec 05 20:56:40 2001
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Date: Wed Dec  5 20:57:04 2001
X-Original-Date: Thu, 6 Dec 2001 07:33:49 +0530 (IST)
Status: O

Hi,
 i have installed the ffmpeg server and for my work just i have stream
only stored file, in the ffserver.conf file i have given the path as
/tmp/clock.avi. but i am unable to play this avi from any of the system in
the network, and also i have one doubt of whether the avi file has to be
converted into .asm file manually .

thanking you

with regards
M.Balathandayuthapani




From <EMAIL-ADDRESS-HIDDEN> Thu Dec 06 00:24:28 2001
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Date: Thu Dec  6 00:25:04 2001
X-Original-Date: Thu, 06 Dec 2001 00:23:56 -0800
Status: O

At 06:03 PM 12/5/01, M BALATHANDAYUTHAPANI wrote:
>Hi,
>  i have installed the ffmpeg server and for my work just i have stream
>only stored file, in the ffserver.conf file i have given the path as
>/tmp/clock.avi. but i am unable to play this avi from any of the system in
>the network, and also i have one doubt of whether the avi file has to be
>converted into .asm file manually .

Are you sure you have the right email address?  There's nothing called 
"ffmpeg server" or "ffserver.conf" in the LIVE.COM Streaming Media code.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Fri Dec 07 00:34:28 2001
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Date: Fri Dec  7 00:35:02 2001
X-Original-Date: Fri, 7 Dec 2001 12:01:51 +0530 (IST)
Status: O

Hi
 Thanks Mr. Rose,
 It is not in Live.com,it is related with ffmepg site, in Live.com also i
got probelm with running client side, is there any RTP support client
running under linux or in windows.
In my system(linux) server is ruuning fine(mp3 streaming server program),
in the client side i am getting the streams in "stdout" but i want to
redirect directly into some player.i need help regarding this.

thanking you 
M.Bala




From <EMAIL-ADDRESS-HIDDEN> Fri Dec 07 08:22:17 2001
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Date: Fri Dec  7 08:23:02 2001
X-Original-Date: Fri, 07 Dec 2001 08:18:46 -0800
Status: O

At 10:31 PM 12/6/01, M BALATHANDAYUTHAPANI wrote:
>got probelm with running client side, is there any RTP support client
>running under linux or in windows.

Yes, the library, and all of the client test applications, will compile for 
both Linux and Windows (and FreeBSD, Solaris, MacOS X, QNX, etc., etc).

>In my system(linux) server is ruuning fine(mp3 streaming server program),
>in the client side i am getting the streams in "stdout" but i want to
>redirect directly into some player.i need help regarding this.

Please see <http://www.live.com/liveCaster/receiving.html> for a list of 
MP3 players (or plugins to MP3 players) that will let you receive and play 
the multicast MP3 stream directly.  (If you use one of these, you won't 
need to use "testMP3Receiver".)

         Ross.



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Subject: [Live-devel] Re: OpenRTSP and quicktime streams (sorenson codec)
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Date: Mon Dec 10 08:40:05 2001
X-Original-Date: Mon, 10 Dec 2001 09:39:28 -0700
Status: O

At 08:08 AM 12/10/01, you wrote:
>I've tried to download the following movie with your great openRTSP,
>but....it doesn't work!
>(It's a quicktime streams about WebObjects & Mac OS X)
[...]
>What I get is:
[...]
>Unable to create receiver for "audio/X-QDM" subsession: RTP payload format
>unknown or not supported
>Unable to create receiver for "video/X-SORENSONVIDEO" subsession: RTP
>payload format unknown or not supported

Zeno,

The problem here is that both of these RTP payload formats ("X-QDM" for 
audio; "X-SORENSONVIDEO" for video) are proprietary and undocumented, so I 
don't know how data for these sessions are packed into RTP packets, so 
"openRTSP" can't extract the media data from these sessions.

One thing you can try is use the "-s 0" option.  This will extract all of 
the data in each packet, following the RTP header.  I'm not sure whether or 
not this will give you useful data (but I note that you said that "-s 0" 
didn't work).

         Ross.
  



From <EMAIL-ADDRESS-HIDDEN> Wed Dec 12 19:09:19 2001
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Subject: [Live-devel] Bug in MultiFramedRTPSink::sendPacketIfNecessary() ?
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Date: Wed Dec 12 19:10:01 2001
X-Original-Date: Thu, 13 Dec 2001 13:38:58 +1030
Status: O

I have a subclass of FramedSource that does not know exactly when the next
frame is due, so my implementation of "virtual float getPlayTime(unsigned
numFrames)" returns 0.0. I also have a subclass of MultiFramedRTPSink that
streams the data to the network.
The system had quite a weird behaviour: it worked fine for ~36 minutes, then
stopped for another ~36 minutes, then started again and so on. It seems that
the reason is in MultiFramedRTPSink::sendPacketIfNecessary(). Take a look at
the following code:

    float timeToNextPacket = fSource == NULL ? (float)(0.0)
		: fSource->getPlayTime(fNumFramesUsedSoFar);

    unsigned secs = (unsigned)timeToNextPacket;
    unsigned usecs = (unsigned)((timeToNextPacket-secs)*1000000.0);
        // we don't round up here, to avoid creeping delays
    fNextSendTime.tv_usec += usecs;
    fNextSendTime.tv_sec += secs + fNextSendTime.tv_usec/1000000;
    fNextSendTime.tv_usec %= 1000000;

    // Figure out how long we have until this time occurs:
    struct timeval timeNow;
    gettimeofday(&timeNow, &Idunno);
    int uSecondsToGo = (fNextSendTime.tv_sec - timeNow.tv_sec)*1000000
      + (fNextSendTime.tv_usec - timeNow.tv_usec);

    // Delay this amount of time:
    nextTask() = envir().taskScheduler().scheduleDelayedTask(uSecondsToGo,
						(TaskProc*)sendNext, this);


fNextSendTime is initialised with the call to continuePlaying(). If
timeToNextPacket == 0.0 fNextSendTime will never change! uSecondsToGo will
become negative and 36 minutes later the expression (fNextSendTime.tv_sec -
timeNow.tv_sec)*1000000 will become < -2147483648, resulting in integer
underflow and causing the next "sendNext" to be scheduled in another ~36
minutes.


I placed a check for such a condition between the last 2 statements:

    int uSecondsToGo = (fNextSendTime.tv_sec - timeNow.tv_sec)*1000000
      + (fNextSendTime.tv_usec - timeNow.tv_usec);

    if (uSecondsToGo < 0) 
      fNextSendTime = timeNow;

    // Delay this amount of time:
    nextTask() = envir().taskScheduler().scheduleDelayedTask(uSecondsToGo,
						(TaskProc*)sendNext, this);

This keeps fNextSendTime closer to the current time, avoiding risk of
underflows. After that the process worked fine for me. I hope there will not
be any side effects. However if the user changes the PC clock he could
experience similar behaviour!

Regards

Vesselin


From <EMAIL-ADDRESS-HIDDEN> Wed Dec 12 19:58:08 2001
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Date: Wed Dec 12 19:59:02 2001
X-Original-Date: Wed, 12 Dec 2001 20:55:40 -0700
Status: O

Vesselin,

Thanks for the bug report.

>I placed a check for such a condition between the last 2 statements:
[...]
>    if (uSecondsToGo < 0)
>       fNextSendTime = timeNow;
[...]
>This keeps fNextSendTime closer to the current time

Actually, I'd prefer "fNextSendTime" to remain "the time at which the next 
packet will be (or should have been) sent", based on the value returned by 
getPlayTime().  Note that it is possible for "fNextSendTime" to fall behind 
real time for a variety of reasons, but we'd like it to remain accurate in 
case we later catch up.

So, the fix I made was to replace the statement
     int uSecondsToGo = (fNextSendTime.tv_sec - timeNow.tv_sec)*1000000
            + (fNextSendTime.tv_usec - timeNow.tv_usec);
with
     int uSecondsToGo;
     if (fNextSendTime.tv_sec < timeNow.tv_sec) {
       uSecondsToGo = 0; // prevents integer underflow if too far behind
     } else {
       uSecondsToGo = (fNextSendTime.tv_sec - timeNow.tv_sec)*1000000
         + (fNextSendTime.tv_usec - timeNow.tv_usec);
     }

(Note that - in the subsequent call to "scheduleDelayedTask()", having a 
delay value of  0 has the same effect as having a delay value <0.)

This fix will prevent "uSecondsToGo" from underflowing, while keeping 
"fNextSendTime" at its correct value.

This fix will appear in the next release of the library.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Thu Dec 13 18:46:48 2001
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Date: Thu Dec 13 18:47:03 2001
X-Original-Date: Fri, 14 Dec 2001 06:37:33 +0530 (IST)
Status: O

Hi Ross,
 I have gone through the some of the source code of test programs, it is
understandable but i want to know is there any documentation available to
start our own simple programs. 

with regards
M.Balathandayuthapani




From <EMAIL-ADDRESS-HIDDEN> Thu Dec 13 20:45:16 2001
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Date: Thu Dec 13 20:46:01 2001
X-Original-Date: Thu, 13 Dec 2001 21:41:12 -0700
Status: O

At 06:07 PM 12/13/01, M BALATHANDAYUTHAPANI wrote:
>  I have gone through the some of the source code of test programs, it is
>understandable but i want to know is there any documentation available to
>start our own simple programs.

For now, the best way to understand how to use the libraries is to (i) 
study the example programs in the "testProgs" directory, and (especially) 
(ii) ask questions on this mailing list.

At some point (in the not too distant future) I definitely want to produce 
some real documentation for the libraries (especially now that the APIs 
seem to be stabilizing), but right now I have to focus on activities that 
pay the bills.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Sun Dec 16 19:00:23 2001
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Subject: [Live-devel] suggestions
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Date: Sun Dec 16 19:01:02 2001
X-Original-Date: Mon, 17 Dec 2001 13:29:53 +1030
Status: O

This is a multi-part message in MIME format.

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Ross,

Could you consider following changes for the next release of the
library:

1. In Groupsock.cpp replace:

  int numBytes = readSocket(env(), socketNum(),
			    encapsulatedData, maxBytesToRead,
			    fromAddress);

with:

  struct timeval timeout;
  timeout.tv_sec = 0;
  timeout.tv_usec = 0;

  int numBytes = readSocket(env(), socketNum(),
			    encapsulatedData, maxBytesToRead,
			    fromAddress, &timeout);

This will disable the blocking call to select() in readSocket
(GroupsockHelper.cpp). You would not want blocking calls in a
"real-time" library, would you?


2. In BasicHashTable.hh could you replace:

#define SILLY_ARRAY_SIZE 10

with:

#define SILLY_ARRAY_SIZE 100

I found an overflow of fSillyArray to be pretty hard to debug...


Regards

Vesselin

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From <EMAIL-ADDRESS-HIDDEN> Sun Dec 16 22:03:08 2001
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Date: Sun Dec 16 22:04:02 2001
X-Original-Date: Sun, 16 Dec 2001 22:01:08 -0800
Status: O

At 06:59 PM 12/16/01, Vesselin Kostadinov wrote:
>This will disable the blocking call to select() in readSocket
>(GroupsockHelper.cpp). You would not want blocking calls in a
>"real-time" library, would you?

Actually, yes.  Reading from a 'groupsock' object is meant to be blocking - 
i.e., until data arrives on its socket.  That way, the caller always has 
data available when it returns from the call (to Groupsock::handleRead()) - 
unless, of course the socket is no longer open.  This makes the calling 
code simple, and doesn't require wasting any CPU time by having to sit in a 
loop 'polling' the socket.

If you want to read from a 'groupsock' asynchronously, then the way to do 
this is to create a handler routine, and use 
"TaskScheduler::turnOnBackgroundReadHandling()" to call this handler 
routine whenever data is available on the socket.  The handler routine can 
then call Groupsock::handleRead(), knowing that this will not block 
(despite the fact that it calls select()).


>2. In BasicHashTable.hh could you replace:
>
>#define SILLY_ARRAY_SIZE 10
>
>with:
>
>#define SILLY_ARRAY_SIZE 100
>
>I found an overflow of fSillyArray to be pretty hard to debug...

OK, I'll do this.  Of course, the real fix would be for me to implement a 
real hash table, rather than the linear array hack that's there now...

Thanks for the suggestions.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Sun Dec 16 23:27:11 2001
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From: Vesselin Kostadinov <EMAIL-ADDRESS-HIDDEN>
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Date: Sun Dec 16 23:28:02 2001
X-Original-Date: Mon, 17 Dec 2001 17:56:43 +1030
Status: O

Hi Ross,

> >.... You would not want blocking calls in a
> >"real-time" library, would you?
> 
> Actually, yes.  Reading from a 'groupsock' object is meant to be blocking
- 
> i.e., until data arrives on its socket....

Well, a real-time program (including one that is using RTSP - _real_time_
streaming protocol) should be designed to respond to events as soon as they
occur. What is supposed to happen with the event occurred while the program
makes blocking calls to select()?

This is not just a theoretical question. Windose sometimes freezes for over
6 seconds during a blocking call to select(). Needless to say neither the
user interface nor the other RTSP clients within the program work during
such a call.

Also note that this particular socket has already been select()ed by the
BasicTaskScheduler, so there should be data available in it. The blocking
select() in readSocket looks redundant to me, especially when recvfrom()
works fine.

To make both of us happy would you consider adding another boolean property
of Groupsock - something like fDisableBlockingSelects, defaulted to True and
controlling the call to readSocket? I can write the code if you agree to
include it in the library...


By the way I solved the problem with the simultaneous connections to several
RTSP servers. It was my lace implementation - I was creating 1
BasicUsageEnvironment and 1 SimpleRTPSource for each RTSP server, so I had
several BasicUsageEnvironment-s at a time (shame on me!). It worked fine
when I created one shared BasicUsageEnvironment and used it for all
SimpleRTPSource-s and sinks.



> If you want to read from a 'groupsock' asynchronously, then 
> the way to do this is to create a handler routine, and use 
> "TaskScheduler::turnOnBackgroundReadHandling()" to call this handler 
> routine whenever data is available on the socket.  The handler routine can

> then call Groupsock::handleRead(), knowing that this will not block 
> (despite the fact that it calls select()).

This is more or less exactly what I am doing. I am using an instance of
SimpleRTPSource whose parent class - MultiFramedRTPSource - calls
turnOnBackgroundReadHandling() in doGetNextFrame1(). Regardless of that
during high CPU & network load up to 30% of the time is wasted in blocking
calls to select(), regardless of the fact that there is data in the socket.
Note that there is absolutely no problem with the call to recvfrom(), it
never blocked for over 22 milliseconds! Perhaps this is another feature of
Windows...


Regards

Vesselin


 


From <EMAIL-ADDRESS-HIDDEN> Mon Dec 17 01:41:57 2001
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Date: Mon Dec 17 01:42:03 2001
X-Original-Date: Mon, 17 Dec 2001 01:41:53 -0800
Status: O

> > If you want to read from a 'groupsock' asynchronously, then
> > the way to do this is to create a handler routine, and use
> > "TaskScheduler::turnOnBackgroundReadHandling()" to call this handler
> > routine whenever data is available on the socket.  The handler routine can
> > then call Groupsock::handleRead(), knowing that this will not block
> > (despite the fact that it calls select()).
>
>This is more or less exactly what I am doing. I am using an instance of
>SimpleRTPSource whose parent class - MultiFramedRTPSource - calls
>turnOnBackgroundReadHandling() in doGetNextFrame1(). Regardless of that
>during high CPU & network load up to 30% of the time is wasted in blocking
>calls to select(), regardless of the fact that there is data in the socket.

If there is data in the socket (which there must be, otherwise you would 
not have returned from the select() loop in BasicUsageEnvironment and your 
handler function would not have been called), then the additional call to 
select() in the "readSocket()" function (defined in "GroupsockHelper.cpp") 
*should* always return immediately.  Are you really sure that you are 
seeing this select() call block?  Are you doing anything that might cause 
it to block - e.g., by calling Groupsock::handleRead() more than once from 
inside a handler function?  (However, if you're just using SimpleRTPSource, 
then that won't be happening...)

But anyway, you're correct that the call to select() within "readSocket()" 
is redundant, especially since the subsequent call to recvfrom() is 
supposed to have the same effect - blocking if no data is present on the 
socket.  So, that call to select() can probably go away - it's there for 
historical reasons; I seem to have some recollection of it being needed 
years ago for things to work properly - but perhaps not...

So, anyway, could you try removing the call to select() from within 
"readSocket()" (except in the case where the "timeout" parameter is 
non-NULL), and let me know if this improves your program's performance on 
Windows (or if it causes problems of some sort).  It may be that this is 
enough to overcome the problems that you're seeing.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Mon Dec 17 20:08:40 2001
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> ... then the  additional call to 
> select() in the "readSocket()" function (defined in
"GroupsockHelper.cpp") 
> *should* always return immediately. Are you really sure that you are 
> seeing this select() call block?


Here is how I measured it (code in GroupsockHelper.cpp, explanation
follows):


const int MaxSlots = 2;
const int histSize = 30;
const int TimeProcessIsConsideredTooLong = 500;

timeb astarts[MaxSlots];
int maxMSecs[MaxSlots];
int histptr = 0;
struct hist_t {timeb TS; int Id; int OldMax; int NewMax;int CurBigVal;}
  hist[histSize];

void setStart(int Id) {
  ftime(&(astarts[Id]));
}

void setEnd(int Id){
  timeb endtime;
  ftime(&endtime);
  int msecs = (endtime.time - astarts[Id].time) * 1000 +
    (endtime.millitm - astarts[Id].millitm);
  if (msecs > maxMSecs[Id]) {
    hist[histptr].TS = endtime;
    hist[histptr].Id = Id;
    hist[histptr].OldMax = maxMSecs[Id];
    hist[histptr].NewMax = msecs;
    if (histptr < histSize)
      histptr++;
    maxMSecs[Id] = msecs;
  } else {
    if (msecs > TimeProcessIsConsideredTooLong) {
      hist[histptr].TS = endtime;
      hist[histptr].Id = Id;
      hist[histptr].CurBigVal = msecs;
      if (histptr < histSize)
        histptr++;
    }
  }
}

int readSocket(UsageEnvironment& env,
	       int socket, unsigned char* buffer, unsigned bufferSize,
	       struct sockaddr_in& fromAddress,
	       struct timeval* timeout) {
	int bytesRead = -1;
	do {
		fd_set rd_set;
		FD_ZERO(&rd_set);
		FD_SET(((unsigned int)socket), &rd_set);
		const unsigned numFds = socket+1;

            setStart(0);
		int selectResult
			= select(numFds, &rd_set, NULL, NULL, timeout);
            setEnd(0);

		if (selectResult == 0 && timeout != NULL) {
			bytesRead = 0;
			break;
		} else if (selectResult <= 0) {
			socketErr(env, "select() error: ");
			break;
		}

		if (!FD_ISSET(socket, &rd_set)) {
			socketErr(env, "select() error - !FD_ISSET");
			break;
		}

		SOCKLEN_T addressSize = sizeof fromAddress;
            setStart(1);
		bytesRead = recvfrom(socket, (char*)buffer, bufferSize,
0,
							 (struct
sockaddr*)&fromAddress,
							 &addressSize);
            setEnd(1);

.....


Basically with setStart I record a timestamp and with setEnd I calculate
the time since setStart. Then if that time is higher that the previous
maximum I record an event into hist and update the current maximum.
Event is also recorded when the time is > 0.5 sec. After ~2 minutes the
contents of the structures is:

maxMSecs: { 6079, 11 }

hist: { 
{ { 1008649562, 992, -570, 0 }, 0, 0, 10, 0 }, 
{ { 1008649563, 392, -570, 0 }, 1, 0, 10, 0 }, 
{ { 1008649571, 624, -570, 0 }, 0, 10, 20, 0 }, 
{ { 1008649573, 827, -570, 0 }, 0, 20, 40, 0 }, 
{ { 1008649574, 67, -570, 0 } , 0, 40, 80, 0 }, 
{ { 1008649585, 975, -570, 0 }, 0, 80, 5378, 0 }, 
{ { 1008649600, 596, -570, 0 }, 0, 0, 0, 4076 }, 
{ { 1008649606, 815, -570, 0 }, 0, 0, 0, 2264 }, 
{ { 1008649610, 530, -570, 0 }, 0, 0, 0, 3695 }, 
{ { 1008649616, 298, -570, 0 }, 0, 0, 0, 791 }, 
{ { 1008649618, 872, -570, 0 }, 0, 0, 0, 2544 },
{ { 1008649625, 1, -570, 0 }  , 0, 0, 0, 4537 }, 
{ { 1008649634, 524, -570, 0 }, 0, 5378, 5477, 0 }, 
{ { 1008649640, 603, -570, 0 }, 0, 5477, 6079, 0 }, 
{ { 1008649652, 30, -570, 0 } , 0, 0, 0, 3936 }, 
{ { 1008649661, 904, -570, 0 }, 0, 0, 0, 5939 }, 
{ { 1008649667, 782, -570, 0 }, 0, 0, 0, 5878 }, 
{ { 1008649690, 926, -570, 0 }, 1, 10, 11, 0 }, {... nulls
                                           ^ time, if > 0.5 sec
                                        ^  new maximum time
                                    ^  old maximum time
                                ^  call being measured, 0: select(), 1:
recvfrom 
     ^  timestamp                   
         
so the calls to select() recorded a total blocking time of over 50
seconds (out of 128). The calls to recvfrom() are Ok
                       



> So, anyway, could you try removing the call to select() from within 
> "readSocket()" (except in the case where the "timeout" parameter is 
> non-NULL), and let me know if this improves your program's 
> performance on 
> Windows (or if it causes problems of some sort). 

I tried that piece of code:

		const unsigned numFds = socket+1;

            setStart(0);
		int selectResult = (timeout == NULL) ? 1 /*Is this
OK?*/:
			select(numFds, &rd_set, NULL, NULL, timeout);
            setEnd(0);
		if (selectResult == 0 && timeout != NULL) {


Result:

maxMSecs: { 10, 6079 }
hist: { 
{ { 1008651274, 703, -570, 0 }, 1, 0, 10, 0 }, 
{ { 1008651276, 265, -570, 0 }, 1, 10, 80, 0 }, 
{ { 1008651278, 488, -570, 0 }, 1, 80, 801, 0 }, 
{ { 1008651280, 992, -570, 0 }, 0, 0, 10, 0 }, 
{ { 1008651288, 763, -570, 0 }, 1, 801, 6079, 0 }, 
{ { 1008651293, 900, -570, 0 }, 1, 0, 0, 5107 }, 
{ { 1008651306, 969, -570, 0 }, 1, 0, 0, 4847 }, 
{ { 1008651318, 936, -570, 0 }, 1, 0, 0, 1872 }, 
{ { 1008651323, 603, -570, 0 }, 1, 0, 0, 4657 }, 
{ { 1008651326, 628, -570, 0 }, 1, 0, 0, 3005 }, 
{ { 1008651330, 203, -570, 0 }, 1, 0, 0, 1833 }, 
{ { 1008651337, 744, -570, 0 }, 1, 0, 0, 4587 }, 
{ { 1008651346, 206, -570, 0 }, 1, 0, 0, 5869 }, 
{ { 1008651351, 273, -570, 0 }, 1, 0, 0, 5047 }, 
{ { 1008651361, 908, -570, 0 }, 1, 0, 0, 3715 }, 
{ { 1008651365, 664, -570, 0 }, 1, 0, 0, 3726 }, 
{ { 1008651374, 617, -570, 0 }, 1, 0, 0, 5809 },  {...nulls

So now the calls to recvfrom() are blocking, with total block time 57
out of 100 seconds. It is getting worse.

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From <EMAIL-ADDRESS-HIDDEN> Mon Dec 17 20:35:26 2001
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Date: Mon Dec 17 20:36:02 2001
X-Original-Date: Tue, 18 Dec 2001 08:06:05 +0530 (IST)
Status: O

Hi Ross
 Thanks for your reply, in my previous mail i asked about, how to play the
received stream instead of stroing in one file, i couldn't find the mp3
players for that, please once again give me the detailed reply.

with regards
M.Bala




From <EMAIL-ADDRESS-HIDDEN> Tue Dec 18 00:04:27 2001
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Date: Tue Dec 18 00:05:03 2001
X-Original-Date: Tue, 18 Dec 2001 09:04:20 +0100
Status: O

Hello,

I'm trying live.com libraries with Windows98 and I have seen OpenRTSP 
crashes. The first time I tried with binary of OpenRTSP you can download from
web page, but it crashed. Now, I have downloaded the sources and compiled it
with Visual Studio with the same result. OpenRTSP crashes after prints SDP 
information.

Any ideas?
Thanks.

	Francisco J. Cabello 

-- 
    
 In a world without fences, who needs Gates.
  8:21am  up 36 days, 44 min,  5 users,  load average: 0.08, 0.02, 0.01



From <EMAIL-ADDRESS-HIDDEN> Tue Dec 18 00:27:47 2001
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Date: Tue Dec 18 00:28:07 2001
X-Original-Date: Tue, 18 Dec 2001 00:25:43 -0800
Status: O

At 12:04 AM 12/18/01, Francisco Javier Cabello wrote:
>I'm trying live.com libraries with Windows98 and I have seen OpenRTSP
>crashes. The first time I tried with binary of OpenRTSP you can download from
>web page, but it crashed. Now, I have downloaded the sources and compiled it
>with Visual Studio with the same result. OpenRTSP crashes after prints SDP
>information.

Francisco,

Thanks for the report.  Can you tell me a (publically accessible) "rtsp://" 
URL that illustrates this problem?

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Tue Dec 18 00:38:29 2001
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Date: Tue Dec 18 00:39:03 2001
X-Original-Date: Tue, 18 Dec 2001 00:32:53 -0800
Status: O

At 06:36 PM 12/17/01, M BALATHANDAYUTHAPANI wrote:
>Hi Ross
>  Thanks for your reply, in my previous mail i asked about, how to play the
>received stream instead of stroing in one file, i couldn't find the mp3
>players for that, please once again give me the detailed reply.

Again, please see <http://www.live.com/liveCaster/receiving.html> for a 
list of MP3 players (or plugins to MP3 players) that will let you receive 
and play incoming multicast MP3 streams.  (If you use one of these, you 
won't need to use "testMP3Receiver".)

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Tue Dec 18 01:02:26 2001
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Date: Tue Dec 18 01:03:02 2001
X-Original-Date: Tue, 18 Dec 2001 01:01:41 -0800
Status: O

Vesselin,

Thanks for the report.  It's astonishing that Windows is so brain damaged 
here; none of those calls to select() or recvfrom() should be blocking, as 
they're being called from within a read handler function.

Anyway, I'll try taking the same measurements on my own Windows systems (NT 
& 95), to see if I see the same thing.

	Ross.



From <EMAIL-ADDRESS-HIDDEN> Tue Dec 18 03:39:53 2001
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Date: Tue Dec 18 03:40:12 2001
X-Original-Date: Tue, 18 Dec 2001 12:39:45 +0100
Status: O

Ross,

I haven't any server accessible from internet, sorry. I'll try to debug it 
with Visual Studio. 
OpenRTSP compiled an runned in a Linux machine goes well, but the same source 
compiled with Visual Studio an runned in a Win98 machine crashes when try to 
receive video.
My server is a own RTSP/RTP video server. If you want, I will sent you RTSP 
dialogue between openRTSP and RTSP server.
My feeling is that error arrive before starting to receive RTP packets.

Regards,

	Francisco J. Cabello.


On Tuesday 18 December 2001 09:25, Ross Finlayson wrote:
> At 12:04 AM 12/18/01, Francisco Javier Cabello wrote:
> >I'm trying live.com libraries with Windows98 and I have seen OpenRTSP
> >crashes. The first time I tried with binary of OpenRTSP you can download
> > from web page, but it crashed. Now, I have downloaded the sources and
> > compiled it with Visual Studio with the same result. OpenRTSP crashes
> > after prints SDP information.
>
> Francisco,
>
> Thanks for the report.  Can you tell me a (publically accessible) "rtsp://"
> URL that illustrates this problem?
>
>          Ross.

-- 
    
 In a world without fences, who needs Gates.
 12:26pm  up 36 days,  4:49,  5 users,  load average: 1.13, 1.03, 0.91



From <EMAIL-ADDRESS-HIDDEN> Tue Dec 18 05:53:19 2001
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Date: Tue Dec 18 05:54:08 2001
X-Original-Date: Tue, 18 Dec 2001 05:53:15 -0800
Status: O

On 2001.12.18 03:39:45 -0800 Francisco Javier Cabello wrote:
> If you want, I will sent you RTSP 
> dialogue between openRTSP and RTSP server.

Yes, please run the program with the "-V" option, and send the output.

> My feeling is that error arrive before starting to receive RTP packets.

One way to test this is to also use the "-n" (notify the user when RTP
packets arrive) option.

	Ross.



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Date: Tue Dec 18 23:44:04 2001
X-Original-Date: Wed, 19 Dec 2001 09:19:44 +0530 (IST)
Status: O

Hi
 Thanks for your reply, yesterday i have got one more problem, while
reciecving the streming data, if i am running both server and client on
same machine it is getting the stremed data into file, if server is
running in some other machine it is not receving data, i have used 
url format of query like

openRTSP rtsp://ourservername:portnumber
it is starting the streaming but it is not receiving data.what  coule be
wrong.

with regards
M.Bala



From <EMAIL-ADDRESS-HIDDEN> Wed Dec 19 00:33:10 2001
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Date: Wed Dec 19 00:34:02 2001
X-Original-Date: Wed, 19 Dec 2001 00:30:03 -0800
Status: O

At 07:49 PM 12/18/01, you wrote:
>if i am running both server and client on
>same machine it is getting the stremed data into file, if server is
>running in some other machine it is not receving data, i have used
>url format of query like
>
>openRTSP rtsp://ourservername:portnumber
>it is starting the streaming but it is not receiving data.what  coule be
>wrong.

Maybe there is a firewall somewhere between the server and client machines 
that is blocking UDP packets?

Alternatively, if the session is multicast (rather than unicast), then the 
problem might be that there's no multicast routing between the server and 
client machines.

         Ross.



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Date: Thu Dec 20 02:37:01 2001
X-Original-Date: Thu, 20 Dec 2001 13:18:18 +0530 (IST)
Status: O

Hi
 Thanks for your reply,

 I am testing the client and server within our intranet, our intranet
is physically separated from internet, so we don't have firewall.
and client system ipaddress is 201.123.80.100 and server system ipaddress
is 201.123.80.112.

In testmp3streamer.cpp 
   you have assigned some ip address, in that what part i have to change
for unicast transmission and for multicast transmission.


with regards
M.Bala


> Maybe there is a firewall somewhere between the server and client machines 
> that is blocking UDP packets?
> 
> Alternatively, if the session is multicast (rather than unicast), then the 
> problem might be that there's no multicast routing between the server and 
> client machines.
> 
>          Ross.
> 



From <EMAIL-ADDRESS-HIDDEN> Thu Dec 20 10:33:04 2001
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Date: Thu Dec 20 10:34:03 2001
X-Original-Date: Thu, 20 Dec 2001 10:31:34 -0800
Status: O

At 11:48 PM 12/19/01, you wrote:
>  I am testing the client and server within our intranet, our intranet
>is physically separated from internet, so we don't have firewall.

Are you trying to use "openRTSP" to receive a *multicast* session (e.g., 
from "testMP3Streamer" with the built-in RTSP server enabled)?  If so, 
please try using the newest binary version of "openRTSP", which is 
available from <http://www.live.com/openRTSP/binaries/>.  This version 
fixes a bug that could prevent multicast sessions from being 
received.  (The installed library source code does not yet include this bug 
fix, so for now you'll need to use one of the pre-built binary versions of 
openRTSP.)


>In testmp3streamer.cpp
>    you have assigned some ip address, in that what part i have to change
>for unicast transmission and for multicast transmission.

Simply change "multicastAddressStr" from a multicast address to a unicast 
address (the IP address of the intended destination (i.e, receiving 
client)).  The port number should be the port on which this client receives 
the stream.

Note, however, that you currently won't be able to receive this stream 
using "openRTSP", even if you've enabled the RTSP server in 
"testMP3Streamer".  (This is because of a limitation in the library's RTSP 
server implementation; this will get fixed soon.)

However, you will be able to receive the unicast stream using 
"playRTPMPEG", or using Winamp with the "in_rtp.dll" plugin (see 
<http://www.live.com/multikit/winamp-plugin.html>.  In Winamp, use a URL of 
the form
         rtp://<unicast-address-of-sender>:<port-number>

I hope this helps.

     Ross.



From <EMAIL-ADDRESS-HIDDEN> Fri Dec 21 10:38:04 2001
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Date: Fri Dec 21 10:39:01 2001
X-Original-Date: Fri, 21 Dec 2001 10:35:30 -0800
Status: O

First, *please* put Subject: lines on your email messages!

>Today we have tested the playRTPmpeg it is running fine in unicast mode,
>if i want connect more than one client with same server, what changes i
>have to do in the server side,(is it possible to assign more than one
>address for multicassAddrStr).

Ugh.  If you want to send to more than one client, then you should use 
multicast.  That's what it's for.

If you *really* want to do this using 'multiple unicast', then the simplest 
way to do this - using the test programs (testMP3Streamer or 
testMPEGVideoStreamer) that are available in the "testProgs" directory - is 
to run more than one copy of the same program, with each copy sending to a 
different unicast address/port.

>  we have given the ipaddress of client
>machine in the server machine. Also we found that it is receiving and
>playing only video it is not giving any sound during the video play.what
>could be wrong.

Which server program are you using?  Please be more specific in your 
questions.  If you're using "testMPEGVideoStreamer", note that it transmits 
a MPEG video elementary stream (using the payload format defined in RFC 
2250) only - not audio.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Fri Dec 21 17:58:04 2001
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Date: Fri Dec 21 17:59:02 2001
X-Original-Date: Sat, 22 Dec 2001 05:53:16 +0530 (IST)
Status: O

Hi
Yes we have tested with "testMPEGVideoStreamer" and we are getting
only video, from your reply i got an information that it only send video,
ok, if i want to tranmit a movie file which contain both audio and video,
what is the solution for tranmitting as well as for receiving.

with regards
M.Bala

On Fri, 21 Dec 2001, Ross Finlayson wrote:

> First, *please* put Subject: lines on your email messages!
> 
> >Today we have tested the playRTPmpeg it is running fine in unicast mode,
> >if i want connect more than one client with same server, what changes i
> >have to do in the server side,(is it possible to assign more than one
> >address for multicassAddrStr).
> 
> Ugh.  If you want to send to more than one client, then you should use 
> multicast.  That's what it's for.
> 
> If you *really* want to do this using 'multiple unicast', then the simplest 
> way to do this - using the test programs (testMP3Streamer or 
> testMPEGVideoStreamer) that are available in the "testProgs" directory - is 
> to run more than one copy of the same program, with each copy sending to a 
> different unicast address/port.
> 
> >  we have given the ipaddress of client
> >machine in the server machine. Also we found that it is receiving and
> >playing only video it is not giving any sound during the video play.what
> >could be wrong.
> 
> Which server program are you using?  Please be more specific in your 
> questions.  If you're using "testMPEGVideoStreamer", note that it transmits 
> a MPEG video elementary stream (using the payload format defined in RFC 
> 2250) only - not audio.
> 
>          Ross.
> 



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Date: Fri Dec 21 22:57:45 2001
X-Original-Date: Fri, 21 Dec 2001 14:47:42 +0530 (IST)
Status: O

Hi
 Thanks for your reply.

Today we have tested the playRTPmpeg it is running fine in unicast mode,
if i want connect more than one client with same server, what changes i
have to do in the server side,(is it possible to assign more than one
address for multicassAddrStr). we have given the ipaddress of client 
machine in the server machine. Also we found that it is receiving and
playing only video it is not giving any sound during the video play.what
could be wrong.

with regards
M.Bala




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Subject: [Live-devel] openRTSP, how to turn the data into sound ?
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Date: Sat Dec 22 15:05:01 2001
X-Original-Date: Sun, 23 Dec 2001 00:03:50 +0100 (MET)
Status: O

Hi, I am wanting to use openRTSP to listen to live streamed audio, I try
this command line:
openRTSP rtsp://203.167.246.143/coolrdu.sdp
and it creates an audio-X-QT file, xmms and mpg123 see this as a mpeg2 layer
2 file, but xmms wont play it, and mpg123 makes a squeak, and then exits.
I have also tried splay.  I have tried using the -a command line flag to
make it decode audio only, and piped that into splay, but once again
no sound...

I notice when I use the a flag, and put it into a file, one time xmsm
identified it as mpg 2.5 layer 1, and the other players didn't recognise it.

Another time, even xmms didn't recognise it as an mpg file.

What method to you people use to hear the output of openRTSP please?

Thanks

Kurt

-- 
Kurt Häusler

GMX - Die Kommunikationsplattform im Internet.
http://www.gmx.net



From <EMAIL-ADDRESS-HIDDEN> Sat Dec 22 13:51:51 2001
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To: M BALATHANDAYUTHAPANI <EMAIL-ADDRESS-HIDDEN>
From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] how to transmit and receive both audio and
  video streams
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Date: Sun Dec 23 11:54:30 2001
X-Original-Date: Sat, 22 Dec 2001 13:49:41 -0800
Status: O

At 04:23 PM 12/21/01, M BALATHANDAYUTHAPANI wrote:
>Yes we have tested with "testMPEGVideoStreamer" and we are getting
>only video, from your reply i got an information that it only send video,
>ok, if i want to tranmit a movie file which contain both audio and video,
>what is the solution for tranmitting as well as for receiving.

Yes, it is possible to extract and stream both audio and video streams from 
a MPEG Program Stream, but it will require some programming work.  To do 
this, you would need to write a program that combined "testMPEGSplitter", 
"testMPEGVideoStreamer", and "testMP3Streamer".  I.e., the program would 
need to first use a "MPEGDemux" to break the Program Stream up into its 
constituent audio and video Elementary Streams, and then use:
         - a "MPEGVideoStreamFramer" plus a "MPEGVideoRTPSink"
           for each video stream, and
         - a "MPEGAudioStreamFramer" (to be written) plus a
           "MPEGAudioRTPSink" for each audio stream

Someday I may write a test program like this myself, and add it to the test 
suite.  Please note, though, that the programs in "testProgs" are just that 
- test programs.  Apart from "openRTSP" these are not intended to be 
complete, fully functional application.

         Ross.



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Date: Sun Dec 23 12:05:59 2001
X-Original-Date: Sat, 22 Dec 2001 16:56:06 -0800
Status: O

At 03:03 PM 12/22/01, Kurt Hausler wrote:
>Hi, I am wanting to use openRTSP to listen to live streamed audio, I try
>this command line:
>openRTSP rtsp://203.167.246.143/coolrdu.sdp
>and it creates an audio-X-QT file, xmms and mpg123 see this as a mpeg2 layer
>2 file, but xmms wont play it, and mpg123 makes a squeak, and then exits.

Kurt,

FYI. I was able to open and play this URL in QuickTime Player.  It 
indicated that this audio stream was encoded in the "QDesign Music 2" 
format.  I don't know anything about this format, but it's definitely not 
MPEG audio, so that's why you weren't able to play it using xmms, etc.

So unfortunately I don't know of any player that accepts this audio format 
- except QuickTime Player, which unfortunately runs only on Mac OS and Windows.

(If you wanted to record this stream to a ".mov" file for later playback 
(using QuickTime Player), then
         openRTSP -q rtsp://203.167.246.143/coolrdu.sdp > foo.mov
*almost* works.  This will create a file that you can open in QuickTime 
Player, but unfortunately it won't play properly, because I don't have 
enough knowledge about this particular codec to properly set up all of the 
necessary parameters in the resulting .mov file.)

>What method to you people use to hear the output of openRTSP please?

It depends on what what format (codec) the stream is coded in.  Even though 
"openRTSP" can (usually) receive a data stream successfully, you still need 
an appropriate player tool (that understands the codec) to be able to play 
this data.

         Ross.



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Date: Sun Dec 23 12:44:51 2001
X-Original-Date: Sun, 23 Dec 2001 20:07:09 +0800
Status: O

This is a multi-part message in MIME format.

------=_NextPart_000_0007_01C18BED.674FD640
Content-Type: text/plain;
	charset="big5"
Content-Transfer-Encoding: quoted-printable

Hello : )
I want to build the code on Windows (live.2001.11.27a.tar.gz)
but when I compile it in Visual C++, the error message is as followed :

    NMAKE : fatal error U1073: don't know how to make =
'../liveMedia/libliveMedia.lib'

and also I can't find the *.lib .
Please help me to fix the problem.
Thank you very mucu.


------=_NextPart_000_0007_01C18BED.674FD640
Content-Type: text/html;
	charset="big5"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; charset=3Dbig5">
<META content=3D"MSHTML 6.00.2600.0" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT size=3D2>Hello : )</FONT></DIV>
<DIV><FONT size=3D2>I want to build the code on Windows=20
(live.2001.11.27a.tar.gz)</FONT></DIV>
<DIV><FONT size=3D2>but when I compile it in Visual C++, the error =
message is as=20
followed :</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>&nbsp;&nbsp;&nbsp; NMAKE : fatal error U1073: don't =
know how=20
to make '../liveMedia/libliveMedia.lib'</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>and also I can't find the *.lib .</FONT></DIV>
<DIV><FONT size=3D2>Please help me to fix the problem.</FONT></DIV>
<DIV><FONT size=3D2>Thank you very mucu.</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV></BODY></HTML>

------=_NextPart_000_0007_01C18BED.674FD640--




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Date: Sun Dec 23 13:39:04 2001
X-Original-Date: Sun, 23 Dec 2001 13:34:55 -0800
Status: O

At 04:07 AM 12/23/01, ARMSTRONG wrote:
>Hello : )
>I want to build the code on Windows (live.2001.11.27a.tar.gz)
>but when I compile it in Visual C++, the error message is as followed :
>
>     NMAKE : fatal error U1073: don't know how to make 
> '../liveMedia/libliveMedia.lib'

You need to build the "UsageEnvironment", "BasicUsageEnvironment", 
"groupsock", and "liveMedia" projects first, *before* you build the 
"testProgs" project.  (Sorry that the instructions on the web page didn't 
make that clear.)

I hope this helps.

         Ross.



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Date: Sun Dec 23 17:24:01 2001
X-Original-Date: Sun, 23 Dec 2001 17:21:53 -0800
Status: O

At 05:24 PM 12/23/01, iainhaywood(at)madasafish.com wrote:
>Ant hints on compiling yor source with Borland C Builder5?    *.mak files 
>dinna work! Dont have microsoft visC++.

Yes, please see the section titled "If you're using the Borland C++ 
compiler" in <http://live.sourceforge.net/#config-windows>

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Mon Dec 24 09:23:07 2001
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Subject: [Live-devel] Updated library release - now handles out-of-order packets
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Date: Mon Dec 24 09:24:02 2001
X-Original-Date: Mon, 24 Dec 2001 09:22:17 -0800
Status: O

I have released a new version of the library code (available at 
<http://www.live.com/liveMedia/>) that makes significant changes to 
"MultiFramedRTPSource" - the code that handles incoming RTP packets.

Previously, this code did not handle out-of-order RTP packets (based on 
their RTP sequence number) at all; if a packet arrived with a RTP sequence 
number less than those that had already been received, then it was simply 
discarded.

Normally, this didn't matter much, because misordered packets are supposed 
to be a rare occurrence.  However, I've discovered that Apple's 'Darwin 
Streaming Server' frequently misorders its outgoing RTP packets.

Now, an incoming out-of-order packet will be accepted (and delivered to the 
caller in the correct order) as long as it is received within a certain 
time threshold.  By default, this threshold is 100ms, but it can be set 
(using a new member function RTPSource::setPacketReorderingThresholdTime()) 
to any value, including 0 (which means: don't accept out-of-order packets 
at all).

The "openRTSP" client application - because it records its incoming data to 
a file rather than trying to play it live - sets this threshold to 1 
second, so "openRTSP" will now handle virtually all packet reordering,

	Ross.



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Date: Tue Dec 25 03:02:02 2001
X-Original-Date: Tue, 25 Dec 2001 18:59:27 +0800
Status: O

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>Note, however, that even when the RTSP server is working at =
61.153.19.89,=20
>you might not be able to use openRTSP to access this stream, because =
it's a=20
>RealNetworks stream, and RealNetworks' RTSP server often uses an=20
>undocumented, proprietary protocol, that openRTSP does not understand.

What kind of streams that I can use the openRTSP to access ?
(what streaming server)

Thanks in advance.

            Armstrong



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<META http-equiv=3DContent-Type content=3D"text/html; charset=3Dbig5">
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<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT size=3D2><FONT size=3D3>&gt;Note, however, that even when the =
RTSP server=20
is working at 61.153.19.89, <BR>&gt;you might not be able to use =
openRTSP to=20
access this stream, because it's a <BR>&gt;RealNetworks stream, and=20
RealNetworks' RTSP server often uses an <BR>&gt;undocumented, =
proprietary=20
protocol, that openRTSP does not understand.</FONT></FONT></DIV>
<DIV><FONT size=3D2><FONT size=3D3></FONT><FONT =
size=3D3></FONT></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2><FONT size=3D3>What kind of streams that I can use =
the openRTSP=20
to access ?</FONT></FONT></DIV>
<DIV><FONT size=3D2><FONT size=3D3>(what streaming =
server)</FONT></FONT></DIV>
<DIV><FONT size=3D2><FONT size=3D3></FONT></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2><FONT size=3D3>Thanks in =
advance.</FONT></FONT></DIV>
<DIV><FONT size=3D2><FONT size=3D3></FONT></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;=20
Armstrong</DIV>
<DIV><BR></DIV></FONT></BODY></HTML>

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From <EMAIL-ADDRESS-HIDDEN> Tue Dec 25 03:52:36 2001
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Date: Tue Dec 25 03:53:02 2001
X-Original-Date: Tue, 25 Dec 2001 19:49:58 +0800
Status: O

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>Note, however, that even when the RTSP server is working at =
61.153.19.89,=20
>you might not be able to use openRTSP to access this stream, because =
it's a=20
>RealNetworks stream, and RealNetworks' RTSP server often uses an=20
>undocumented, proprietary protocol, that openRTSP does not understand.

What kind of streams that I can use the openRTSP to access ?
(what streaming server)

Thanks in advance.

            Armstrong

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	charset="big5"
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<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; charset=3Dbig5">
<META content=3D"MSHTML 6.00.2600.0" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT size=3D2>
<DIV><FONT size=3D2><FONT size=3D3>&gt;Note, however, that even when the =
RTSP server=20
is working at 61.153.19.89, <BR>&gt;you might not be able to use =
openRTSP to=20
access this stream, because it's a <BR>&gt;RealNetworks stream, and=20
RealNetworks' RTSP server often uses an <BR>&gt;undocumented, =
proprietary=20
protocol, that openRTSP does not understand.</FONT></FONT></DIV>
<DIV><FONT size=3D2><FONT size=3D3></FONT><FONT =
size=3D3></FONT></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2><FONT size=3D3>What kind of streams that I can use =
the openRTSP=20
to access ?</FONT></FONT></DIV>
<DIV><FONT size=3D2><FONT size=3D3>(what streaming =
server)</FONT></FONT></DIV>
<DIV><FONT size=3D2><FONT size=3D3></FONT></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2><FONT size=3D3>Thanks in =
advance.</FONT></FONT></DIV>
<DIV><FONT size=3D2><FONT size=3D3></FONT></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;=20
Armstrong</DIV></FONT></FONT></DIV></BODY></HTML>

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Date: Tue Dec 25 18:21:02 2001
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>Note, however, that even when the RTSP server is working at =
61.153.19.89,=20
>you might not be able to use openRTSP to access this stream, because =
it's a=20
>RealNetworks stream, and RealNetworks' RTSP server often uses an=20
>undocumented, proprietary protocol, that openRTSP does not understand.

What kind of streams that I can use the openRTSP to access ?
(what streaming server)

Thanks in advance.

            Armstrong

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<HTML><HEAD>
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<META content=3D"MSHTML 6.00.2600.0" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><BR></DIV>
<DIV><FONT size=3D2>
<DIV><FONT size=3D2><FONT size=3D3>&gt;Note, however, that even when the =
RTSP server=20
is working at 61.153.19.89, <BR>&gt;you might not be able to use =
openRTSP to=20
access this stream, because it's a <BR>&gt;RealNetworks stream, and=20
RealNetworks' RTSP server often uses an <BR>&gt;undocumented, =
proprietary=20
protocol, that openRTSP does not understand.</FONT></FONT></DIV>
<DIV><FONT size=3D2><FONT size=3D3></FONT><FONT =
size=3D3></FONT></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2><FONT size=3D3>What kind of streams that I can use =
the openRTSP=20
to access ?</FONT></FONT></DIV>
<DIV><FONT size=3D2><FONT size=3D3>(what streaming =
server)</FONT></FONT></DIV>
<DIV><FONT size=3D2><FONT size=3D3></FONT></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2><FONT size=3D3>Thanks in =
advance.</FONT></FONT></DIV>
<DIV><FONT size=3D2><FONT size=3D3></FONT></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;=20
Armstrong</DIV></FONT></FONT></DIV></BODY></HTML>

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From <EMAIL-ADDRESS-HIDDEN> Tue Dec 25 18:35:03 2001
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Date: Tue Dec 25 18:36:09 2001
X-Original-Date: Tue, 25 Dec 2001 18:34:57 -0800
Status: O

At 02:59 AM 12/25/01, ARMSTRONG wrote:
> >Note, however, that even when the RTSP server is working at 61.153.19.89,
> >you might not be able to use openRTSP to access this stream, because it's a
> >RealNetworks stream, and RealNetworks' RTSP server often uses an
> >undocumented, proprietary protocol, that openRTSP does not understand.
>
>What kind of streams that I can use the openRTSP to access ?
>(what streaming server)

This depends not so much on the type of server, but more on the type of 
*stream*.

"openRTSP" can be used to *play* any "rtsp://.." stream that uses RTP for 
transport.  Most "rtsp://" streams fall into this category, but many 
RealNetworks streams don't, because they use their own proprietary protocol 
(called RDP or RDT or something) instead of RTP.

If you also want openRTSP to *receive* the streamed data (i.e., you don't 
use the "-r" option), then the RTP stream needs to use a payload format 
that openRTSP understands.  (A "payload format" describes the way that 
stream data is packed into RTP packets.)  If you see an error message "RTP 
payload format unknown or not supported", then this means that openRTSP 
doesn't have enough knowledge to extract the stream data from the RTP stream.

But anyway, here are a couple of "rtsp://" URLs that will work with openRTSP:
- rtsp://media.schreiner.edu/realmp3.mp3
         This will produce a MP3 file.  If you rename it with a ".mp3" file 
extension, you can play this with a MP3 player.
- rtsp://darwin.horizonlive.com:554/finlayson/sample-h263-ulaw-8000.mov
         This produces a H.263+ video stream, and a PCM u-law audio 
stream.  The best way to receive these is to combine them in a QuickTime 
movie file.  Use the "-q" option, and redirect the output to a ".mov" file.

         Ross.



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Date: Tue Dec 25 20:26:01 2001
X-Original-Date: Wed, 26 Dec 2001 12:24:14 +0800
Status: O

> But anyway, here are a couple of "rtsp://" URLs that will work with
openRTSP:
> - rtsp://media.schreiner.edu/realmp3.mp3
>          This will produce a MP3 file.  If you rename it with a ".mp3"
file
> extension, you can play this with a MP3 player.
> - rtsp://darwin.horizonlive.com:554/finlayson/sample-h263-ulaw-8000.mov
>          This produces a H.263+ video stream, and a PCM u-law audio
> stream.  The best way to receive these is to combine them in a QuickTime
> movie file.  Use the "-q" option, and redirect the output to a ".mov"
file.
>
>          Ross.

Thanks for your detailed anwers with patience.
I am sorry that I give you so many troubles.
Because I need the openRTSP, I will probably ask a lot of questions. : )
I try the two URLs, but they still don't work.
Maybe I make something wrong
   >openRTSP.exe  -q
rtsp://darwin.horizonlive.com:554/finlayson/sample-h263-ulaw-8000.mov
   >openRTSP.exe rtsp://media.schreiner.edu/realmp3.mp3
After printing the information of SDP, it pop up a windows message box, says
"The openRTSP.exe has some problem, it have to be closed. "

Execution result :
1. rtsp://darwin.horizonlive.com:554/finlayson/sample-h263-ulaw-8000.mov
Opened URL
"rtsp://darwin.horizonlive.com:554/finlayson/sample-h263-ulaw-8000.mo
v", returning a SDP description:
v=0
s=/finlayson/sample-h263-ulaw-8000.mov
u=http:///
e=admin(at)
c=IN IP4 208.185.32.22
a=control:*
b=AS:528
a=range:npt=0-  37.40333
m=video 0 RTP/AVP 96
b=AS:464
a=rtpmap:96 H263-1998/90000
a=control:trackID=3
a=cliprect:0,0,180,240
m=audio 0 RTP/AVP 0
b=AS:63
a=control:trackID=4

2. rtsp://media.schreiner.edu/realmp3.mp3
Opened URL "rtsp://media.schreiner.edu/realmp3.mp3", returning a SDP
description
:
v=0
o=- 979010679 979010679 IN IP4 207.71.37.7
s=realmp3
i=<No author> <No copyright>
c=IN IP4 0.0.0.0
t=0 0
a=SdpplinVersion:1610642970
a=StreamCount:integer;1
a=Title:buffer;"cmVhbG1wMwA="
a=range:npt=0-39.471000
m=audio 0 RTP/AVP 101
b=AS:128
a=control:streamid=0
a=range:npt=0-39.471000
a=length:npt=39.471000
a=rtpmap:101 X-MP3-draft-00
a=mimetype:string;"audio/X-MP3-draft-00"
a=StartTime:integer;0
a=AvgBitRate:integer;128000
a=SampleRate:integer;44100
a=AvgPacketSize:integer;417
a=Preroll:integer;1000
a=NumChannels:integer;2
a=MaxPacketSize:integer;1024
a=ASMRuleBook:string;"AverageBandwidth=128000, AverageBandwidthStd=0,
Priority=9
;"

Could you please tell me what the problem is ?
Thanks in advance.

                Armstrong




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Date: Tue Dec 25 20:37:02 2001
X-Original-Date: Tue, 25 Dec 2001 20:36:19 -0800
Status: O

At 08:24 PM 12/25/01, ARMSTRONG wrote:
>I try the two URLs, but they still don't work.
>Maybe I make something wrong
>    >openRTSP.exe  -q
>rtsp://darwin.horizonlive.com:554/finlayson/sample-h263-ulaw-8000.mov
>    >openRTSP.exe rtsp://media.schreiner.edu/realmp3.mp3
>After printing the information of SDP, it pop up a windows message box, says
>"The openRTSP.exe has some problem, it have to be closed. "

Does the same problem occur if you run the pre-built binary version from 
<http://www.live.com/openRTSP/binaries/windows/>?

         Ross.



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From: ARMSTRONG <EMAIL-ADDRESS-HIDDEN>
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Subject: Re: [Live-devel] openRTSP
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Date: Tue Dec 25 21:23:01 2001
X-Original-Date: Wed, 26 Dec 2001 13:07:13 +0800
Status: O

> >I try the two URLs, but they still don't work.
> >Maybe I make something wrong
> >    >openRTSP.exe  -q
> >rtsp://darwin.horizonlive.com:554/finlayson/sample-h263-ulaw-8000.mov
> >    >openRTSP.exe rtsp://media.schreiner.edu/realmp3.mp3
> >After printing the information of SDP, it pop up a windows message box,
says
> >"The openRTSP.exe has some problem, it have to be closed. "
>
> Does the same problem occur if you run the pre-built binary version from
> <http://www.live.com/openRTSP/binaries/windows/>?
>
>          Ross.
    Yes !

            Armstrong




From <EMAIL-ADDRESS-HIDDEN> Tue Dec 25 23:52:00 2001
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Subject: Re: [Live-devel] openRTSP
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Date: Tue Dec 25 23:52:02 2001
X-Original-Date: Tue, 25 Dec 2001 23:50:23 -0800
Status: O

At 09:07 PM 12/25/01, ARMSTRONG wrote:
> > >After printing the information of SDP, it pop up a windows message box,
>says
> > >"The openRTSP.exe has some problem, it have to be closed. "
> >
> > Does the same problem occur if you run the pre-built binary version from
> > <http://www.live.com/openRTSP/binaries/windows/>?
> >
> >          Ross.
>     Yes !

OK, so there may be some problem with your Windows computer's network 
configuration.

Are you able to succesfully do either of the following (on the same Windows 
machine on which you're having problems with "openRTSP")?

1/ Open rtsp://media.schreiner.edu/realmp3.mp3 from "RealPlayer" ?

2/ Open 
rtsp://darwin.horizonlive.com:554/finlayson/sample-h263-ulaw-8000.mov from 
"QuickTime Player" ?

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Wed Dec 26 01:08:09 2001
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Subject: Re: [Live-devel] openRTSP
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Date: Wed Dec 26 01:09:02 2001
X-Original-Date: Wed, 26 Dec 2001 17:02:25 +0800
Status: O

> OK, so there may be some problem with your Windows computer's network
> configuration.
>
> Are you able to succesfully do either of the following (on the same
Windows
> machine on which you're having problems with "openRTSP")?
>
> 1/ Open rtsp://media.schreiner.edu/realmp3.mp3 from "RealPlayer" ?
>
> 2/ Open
> rtsp://darwin.horizonlive.com:554/finlayson/sample-h263-ulaw-8000.mov from
> "QuickTime Player" ?
>
>          Ross.
>

1/ It is still buffering (can't play)
2/ The playing bar is moving from the start to the end, but no movie and no
sound.

            Armstrong




From <EMAIL-ADDRESS-HIDDEN> Wed Dec 26 01:44:37 2001
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Subject: Re: [Live-devel] openRTSP
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Date: Wed Dec 26 01:45:03 2001
X-Original-Date: Wed, 26 Dec 2001 17:41:57 +0800
Status: O

> OK, so there may be some problem with your Windows computer's network
> configuration.
>
> Are you able to succesfully do either of the following (on the same
Windows
> machine on which you're having problems with "openRTSP")?
>
> 1/ Open rtsp://media.schreiner.edu/realmp3.mp3 from "RealPlayer" ?
>
> 2/ Open
> rtsp://darwin.horizonlive.com:554/finlayson/sample-h263-ulaw-8000.mov from
> "QuickTime Player" ?
>
>          Ross.

I update my words as followed :
1/ About waiting for few minutes, it can play.
2/ The playing bar is moving from the start to the end, but no movie and no
sound.





From <EMAIL-ADDRESS-HIDDEN> Wed Dec 26 02:38:13 2001
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Date: Wed Dec 26 02:39:02 2001
X-Original-Date: Wed, 26 Dec 2001 02:37:28 -0800
Status: O

At 01:02 AM 12/26/01, ARMSTRONG wrote:
> > 1/ Open rtsp://media.schreiner.edu/realmp3.mp3 from "RealPlayer" ?
> >
> > 2/ Open
> > rtsp://darwin.horizonlive.com:554/finlayson/sample-h263-ulaw-8000.mov from
> > "QuickTime Player" ?
>
>1/ It is still buffering (can't play)
>2/ The playing bar is moving from the start to the end, but no movie and no
>sound.

OK, you'll first need to figure out what is wrong with your computer that 
is preventing these players from working.  Only after you get these players 
to work can you expect "openRTSP" to work.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Wed Dec 26 07:51:51 2001
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Subject: Fw: [Live-devel] openRTSP
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Date: Wed Dec 26 07:52:02 2001
X-Original-Date: Wed, 26 Dec 2001 15:38:13 +0800
Status: O

 > Does the same problem occur if you run the pre-built binary version from
  > <http://www.live.com/openRTSP/binaries/windows/>?
  >
  >          Ross.
 
 It looks like something wrong in this line.
 In RTCP.cpp (Ln 142)
  // Arrange to handle incoming reports from others:
 env.taskScheduler().turnOnBackgroundReadHandling(RTCPgs->socketNum(),
 (TaskScheduler::BackgroundHandlerProc*)&incomingReportHandler,this);
 
             Armstrong
 




From <EMAIL-ADDRESS-HIDDEN> Wed Dec 26 10:20:59 2001
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Date: Wed Dec 26 10:21:02 2001
X-Original-Date: Wed, 26 Dec 2001 10:20:49 -0800
Status: O

At 11:38 PM 12/25/01, ARMSTRONG wrote:
>  It looks like something wrong in this line.
>  In RTCP.cpp (Ln 142)

Why do you think this?

Your main task now should be to figure out why RealPlayer and QuickTime 
Player are not working properly on your machine.

         Ross.



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Subject: [Live-devel] streaming using openRTSP
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Date: Wed Jan  2 12:20:53 2002
X-Original-Date: Wed, 2 Jan 2002 12:15:13 -0800
Status: O

Hi,  HAPPY new year!
Here is a newbie question:
I have been trying to experiment with streaming (RTSP) MPEG-1 video (not
multiplexed stream) and/or MP3  using openRTSP from my
linux machine to a PC running Real player (or quicktime).
I have been running into the following problem:

I changed the definition to IMPLEMENT_RTSP_SERVER. I changed the client port
number to match the response from the client (in testMP3.sdp)
and used my PCs destination IP address, and "multicast" to "unicast".
I run testMP3Streamer and connect using rtsp://...:554/testMP3.sdp
Real One player seems to quit with "connection to server has been lost"...
(i changed "preferences" to increase network timeout values)
On the server side, it seems to receive
PLAY rtsp://...:554/testMP3.sdp RTSP/1.0
followed by some SET_PARAMETER commands
and the server appears to be still functional.
Similar problems seem to occur with MPEG-1 video with testMPEGVideo.

Is there anything wrong in my attempts, or is there a better means to get
MP3 (to begin with) to stream using RTSP from openRTSP to Real or Quicktime
player?
Thank you very much for your time and response in advance,

thanks
pramila




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Date: Wed Jan  2 13:45:05 2002
X-Original-Date: Wed, 02 Jan 2002 13:44:35 -0800
Status: O

Pramila,

Thanks for the note.

>I have been trying to experiment with streaming (RTSP) MPEG-1 video (not
>multiplexed stream) and/or MP3  using openRTSP

I think you mean "using testMPEGVideoStreamer (or testMP3Streamer)".  These 
are applications that stream video (or audio).  In contrast, "openRTSP" is 
a client application that *receives* RTSP-initiated sessions.


>Is there anything wrong in my attempts, or is there a better means to get
>MP3 (to begin with) to stream using RTSP from openRTSP to Real or Quicktime
>player?

First, the built-in RTSP server implementation that's used (optionally) in 
"testMPEGVideoStreamer" and "testMP3Streamer" doesn't yet work properly for 
unicast sessions - only for multicast sessions.  So, for now, if you want 
to use the built-in RTSP server in these test programs, you should leave 
the destination address as multicast.  (If you're streaming on a LAN, then 
this should be fine, as virtually all LANs support multicast.)

Second, RealPlayer and QuickTime Player each have some issues that can trip 
you up.  RealPlayer version 8 has a bug that prevents RTSP-initiated MPEG-1 
video sessions from working properly (although I've heard that the new 
"RealOne" player does not have this problem).  Also, QuickTime Player 
currently does not support RTP-streamed MP3 (i.e., layer III) audio - only 
layer I or layer II audio.

So, to summarize:
1/ If you want to use the RTSP server that's built in to 
"testMPEGVideoStreamer" or "testMP3Streamer", then you should leave the 
destination address as multicast, rather than unicast.  (I'll be fixing 
this limitation soon.)
2/ To receive a RTSP-initiated MPEG-1 video stream, use QuickTime Player 
(version 5 or greater)
3/ To receive a RTSP-initiated MP3 audio stream, use RealPlayer.

I hope this helps.

         Ross.
  



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Date: Thu Jan  3 02:03:06 2002
X-Original-Date: Thu, 3 Jan 2002 11:02:55 +0100
Status: O

Ross,

is openrtsp capable to get video from a RealVideo Server?
Some time ago, I was testing it, and I could that real guys has a lot 
proprietary extension in the RTSP negotiation. 

Thanks.

	Francisco J. Cabello


From <EMAIL-ADDRESS-HIDDEN> Thu Jan 03 02:59:09 2002
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Date: Thu Jan  3 03:00:02 2002
X-Original-Date: Thu, 03 Jan 2002 02:57:55 -0800
Status: O

At 02:02 AM 1/3/02, Francisco Javier Cabello wrote:
>is openrtsp capable to get video from a RealVideo Server?

Only if the video is something *other than* RealVideo (e.g., it's MPEG).

Unfortunately, RealVideo (and RealAudio) sessions do not use RTP for 
transport.  (Instead, these sessions use RealNetworks' proprietary (and 
undocumented) "RDT" protocol.)

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Mon Jan 07 14:06:26 2002
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Date: Mon Jan  7 14:07:01 2002
X-Original-Date: Mon, 07 Jan 2002 17:06:22 -0500
Status: O

I think there is a problem with the 2002.01.02 release.
It shows no files.

Marty Schoch
<mschoch(at)multicasttech.com>



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Date: Mon Jan  7 14:11:02 2002
X-Original-Date: Mon, 07 Jan 2002 14:10:20 -0800
Status: O

At 02:06 PM 1/7/02, Marty Schoch wrote:
>I think there is a problem with the 2002.01.02 release.
>It shows no files.

Actually, this is a problem with SourceForge - it often takes a long time 
for uploaded files to show up there.

You can always get the most up-to-date release from 
<http://www.live.com/liveMedia/>

         Ross.



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From <EMAIL-ADDRESS-HIDDEN> Thu Jan 10 10:10:50 2002
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Date: Thu Jan 10 10:11:09 2002
X-Original-Date: Thu, 10 Jan 2002 10:10:41 -0800
Status: O

At 06:54 AM 1/10/02, ligrax(at)libero.it wrote:
>hello!
>i'm trying to use testVideoStreamer program.
>I can see that it transmits only a video stream;
>if I uncomment the first #define, it is able to
>read a mpeg video and audio file, performs a demultiplexing
>and then transmits only the video stream.
>Is it possible transmitting ordinary mpeg video-audio file,
>with this approach, or I must find out a solution?
>I'm writing a thesis on multimedia streaming based on RTP.

This is becoming a frequently-asked question :-)

Yes, it is possible to extract and stream both audio and video streams from 
a MPEG Program Stream, but it will require some programming work.  To do 
this, you would need to write a program that combines "testMPEGSplitter", 
"testMPEGVideoStreamer", and "testMP3Streamer".  I.e., the program would 
need to first use a "MPEGDemux" to break the Program Stream up into its 
constituent audio and video Elementary Streams, and then use:
         - a "MPEGVideoStreamFramer" plus a "MPEGVideoRTPSink"
           for each video stream, and
         - a "MPEGAudioStreamFramer" (to be written) plus a
           "MPEGAudioRTPSink" for each audio stream

So many people have been asking for this that I'm probably going to have to 
write  a test program like this myself, and add it to the test 
suite.  (This may not happen for a few weeks, though.)

>I it's not possible, where can I find a software for
>streaming and receiving video and audio based on RTP?

RealNetworks' and Apple QuickTime software will do this, as will Cisco's 
IP/TV software <http://www.cisco.com/warp/public/cc/pd/mxsv/iptv3400/>

         Ross.





From <EMAIL-ADDRESS-HIDDEN> Thu Jan 10 10:26:56 2002
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Date: Thu Jan 10 10:27:05 2002
X-Original-Date: Thu, 10 Jan 2002 12:29:50 -0600 (CST)
Status: O

great !! when you have a test version for full transmission (video &
audio) let us know Ross!! i think this will be the first step for 
developing new projects on our own.

						regards O.


On Thu, 10 Jan 2002, Ross Finlayson wrote:

> At 06:54 AM 1/10/02, ligrax(at)libero.it wrote:
> >hello!
> >i'm trying to use testVideoStreamer program.
> >I can see that it transmits only a video stream;
> >if I uncomment the first #define, it is able to
> >read a mpeg video and audio file, performs a demultiplexing
> >and then transmits only the video stream.
> >Is it possible transmitting ordinary mpeg video-audio file,
> >with this approach, or I must find out a solution?
> >I'm writing a thesis on multimedia streaming based on RTP.
> 
> This is becoming a frequently-asked question :-)
> 
> Yes, it is possible to extract and stream both audio and video streams from 
> a MPEG Program Stream, but it will require some programming work.  To do 
> this, you would need to write a program that combines "testMPEGSplitter", 
> "testMPEGVideoStreamer", and "testMP3Streamer".  I.e., the program would 
> need to first use a "MPEGDemux" to break the Program Stream up into its 
> constituent audio and video Elementary Streams, and then use:
>          - a "MPEGVideoStreamFramer" plus a "MPEGVideoRTPSink"
>            for each video stream, and
>          - a "MPEGAudioStreamFramer" (to be written) plus a
>            "MPEGAudioRTPSink" for each audio stream
> 
> So many people have been asking for this that I'm probably going to have to 
> write  a test program like this myself, and add it to the test 
> suite.  (This may not happen for a few weeks, though.)
> 
> >I it's not possible, where can I find a software for
> >streaming and receiving video and audio based on RTP?
> 
> RealNetworks' and Apple QuickTime software will do this, as will Cisco's 
> IP/TV software <http://www.cisco.com/warp/public/cc/pd/mxsv/iptv3400/>
> 
>          Ross.
> 
> 
> 
> 
> _______________________________________________
> Live-devel mailing list
> Live-devel(at)lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/live-devel
> 


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Date: Sat Jan 12 01:57:02 2002
X-Original-Date: Sat, 12 Jan 2002 01:54:18 -0800
Status: O

FYI, I have just installed a new version of the LIVE.COM Streaming Media 
libraries that improves the implementation of MPEG video streaming:
- Fixed a bug that could cause very large MPEG frames to be truncated
- Fixed a timestamp-related bug that could cause video player tools (such 
as QuickTime player) to get confused when "testMPEGVideoStreamer" started 
repeating its input file.
- Fixed a bug that could cause "testMPEGVideoStreamer" to crash when 
reading MPEG Program Streams.
- Made the video streaming code slightly more efficient.

If you're using the libraries to do MPEG video streaming, you should 
upgrade to this new version of the code.

	Ross.



From <EMAIL-ADDRESS-HIDDEN> Tue Jan 15 02:43:34 2002
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Subject: [Live-devel] Re: testRelay
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Date: Tue Jan 15 02:44:03 2002
X-Original-Date: Tue, 15 Jan 2002 02:40:08 -0800
Status: O

At 02:12 AM 1/15/02, mimmo wrote:
>I need to write a code that's similar to your testRelay.cpp;
>
>the difference is that it must receive from a unicast IP address,port,
>
>and then relay the stream to a multicast IP address,port.
>
>Can  I substitute the unicast address with your multicast
>
>predefined for the input

Yes, that will work.  You can do this by replacing the input 'multicast' 
address 239.255.42.42 with *any* unicast address (even 0.0.0.0 is 
OK).  Note that the sender of the packet doesn't matter.  As long as the 
packet is sent to port 8888, "testRelay" will relay it to multicast address 
239.255.43.43, port 6666.  (Of course, you can change these values.)

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Fri Jan 18 22:56:07 2002
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Subject: [Live-devel] Re: how to receive the streamed audio data
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Date: Fri Jan 18 22:57:02 2002
X-Original-Date: Fri, 18 Jan 2002 22:51:51 -0800
Status: O

At 12:40 AM 1/18/02, leonheat99 wrote:
>     I am a college student of China. I saw your project liveMedia on the 
> Internet occasionally . I admire and am very interested in  your work.
>    I also have some questions about it and I hope you can help me with them..
>    What development tools are you using? I mean if you used some 
> analysing  and designing  tools to make your program structural..

I do most of my development on a FreeBSD Unix machine, using emacs (for 
editing), and g++ (etc.) for compiling.  (When I build/test the code on 
Windows, I use Microsoft Visual C++)

>    The other questions is that which part of your program is resposible 
> for the receiving of streamed data after the PLAY request of RTSP has 
> response correctly.

"openRTSP" creates a set of 'RTPSource' objects (one for each media 
subsession that is being received), and a set of corresponding 'FileSink' 
objects.  It then calls the 'startPlaying()' operation on each FileSink 
(using the corresponding RTPSource as the data source).

>I have used your program openRTSP.exe to play the url 
>rtsp://media.schreiner.edu/realmp3.mp3  . I could see that the 
>DESCRIBE,SETUP,PLAY request success,but I could not receive the later 
>streamed media data from the server.

This is probably because you have a firewall (somewhere) that is blocking 
UDP packets from reaching you.  (The RTSP commands (DESCRIBE, SETUP, PLAY) 
and responses use TCP, but the actual data stream uses RTP/UDP.)

You should ask your college's network administrator if your college is 
running a firewall, and, if so, whether it can be reconfigured to allow UDP 
packets to pass through.

>But the Realplayer could play it.I don't know why.

Apparently RealPlayer has a mechanism that allows it to request that the 
data stream be carried over the RTSP (TCP) connection, if it fails to 
receive any UDP packets.  Someday I may update "openRTSP" to support this 
(as an option), but this will be non-trivial to do, because the current 
implementation of the 'RTPSource' objects assumes that their input source 
is a RTP/UDP socket onlly.

         Ross.



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Subject: [Live-devel] H.263 Streamer
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Date: Mon Jan 21 03:31:07 2002
X-Original-Date: Mon, 21 Jan 2002 13:30:31 +0200
Status: O

This is a multi-part message in MIME format.

------=_NextPart_000_0054_01C1A27F.CC3B01D0
Content-Type: text/plain;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

Hi!

I'm trying to develop streaming application (based on =
testMPEGVideoStreamer) that uses H.263 for video compression. Can I use =
ByteStreamFileSource as a MediaSource for H263plusVideoRTPSink or do I =
have to implement "Framer" that brakes H.263 stream into frames like =
MPEGVideoStreamFramer? The files that I'm trying to stream are =
.h263-files.

tmk


------=_NextPart_000_0054_01C1A27F.CC3B01D0
Content-Type: text/html;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2600.0" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Hi!</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>I'm trying to develop streaming =
application (based=20
on testMPEGVideoStreamer) that uses H.263 for video compression. Can I =
use=20
ByteStreamFileSource as a MediaSource for H263plusVideoRTPSink or do I =
have to=20
implement "Framer" that brakes H.263 stream into frames like=20
MPEGVideoStreamFramer? The files that I'm trying to stream are=20
.h263-files.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>tmk</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV></BODY></HTML>

------=_NextPart_000_0054_01C1A27F.CC3B01D0--



From <EMAIL-ADDRESS-HIDDEN> Mon Jan 21 10:54:41 2002
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Date: Mon Jan 21 10:55:13 2002
X-Original-Date: Mon, 21 Jan 2002 10:54:38 -0800 (PST)
Status: O

Hi,

I tried to download openRTSP client pre-built
binaries for Solaris-Sparc but got permission
denied after click link openRTSP under 
/openRTSP/binaries/solaris-sparc

Please advise.

Pius

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From <EMAIL-ADDRESS-HIDDEN> Mon Jan 21 10:59:16 2002
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Date: Mon Jan 21 11:00:02 2002
X-Original-Date: Mon, 21 Jan 2002 10:56:18 -0800
Status: O

At 03:30 AM 1/21/02, Tomi Korhonen wrote:
>I'm trying to develop streaming application (based on 
>testMPEGVideoStreamer) that uses H.263 for video compression. Can I use 
>ByteStreamFileSource as a MediaSource for H263plusVideoRTPSink or do I 
>have to implement "Framer" that brakes H.263 stream into frames like 
>MPEGVideoStreamFramer?

Tom,

The "H263plusVideoRTPSink" class expects to receive a complete H.263+ 
(i.e., H.263-1998) frame from it's source whenever it (actually, its parent 
class "MultiFramedRTPSink") calls "getNextFrame()".

So, yes, you will need a 'framer' source object (similar to 
"MPEGVideoStreamFramer") that breaks the input stream up into discrete 
H.263+ frames.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Mon Jan 21 11:01:28 2002
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Date: Mon Jan 21 11:02:03 2002
X-Original-Date: Mon, 21 Jan 2002 11:01:20 -0800
Status: O

At 10:54 AM 1/21/02, Pius Ng wrote:
>I tried to download openRTSP client pre-built
>binaries for Solaris-Sparc but got permission
>denied after click link openRTSP under
>/openRTSP/binaries/solaris-sparc

Oops, the file permissions were wrong on that file.  Thanks for reporting 
this; you should be able to download it now.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Thu Jan 24 09:16:43 2002
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Subject: [Live-devel] where to read the data before playing
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Date: Thu Jan 24 09:17:11 2002
X-Original-Date: Thu, 24 Jan 2002 09:16:41 -0800 (PST)
Status: O

hi,
 I am trying to read the data before playing into a
ring buffer. After the call to the 'startPlaying()'
,which part of your program is resposible 
 for  reading the  data from the network ? is it in
the function 'afterGettingFrame()' of the FileSink.cpp

  i changed the fwrite() function to do the ring
buffer write but surprisingly i am unable to see the
code flow entering the 'afterGettingFrame()' function.
  please can u suggest if this is the place where the
changes are to be made or in any other class?
 Your help would be great.
thank you
 surya

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From <EMAIL-ADDRESS-HIDDEN> Thu Jan 24 12:12:52 2002
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Date: Thu Jan 24 12:13:02 2002
X-Original-Date: Thu, 24 Jan 2002 12:11:39 -0800
Status: O

At 09:16 AM 1/24/02, suryakumari sagiraju wrote:
>hi,
>  I am trying to read the data before playing into a
>ring buffer. After the call to the 'startPlaying()'
>,which part of your program is resposible
>  for  reading the  data from the network ? is it in
>the function 'afterGettingFrame()' of the FileSink.cpp

The sink's data source is responsible for getting it the data that it 
wants.  Note (in FileSink::continuePlaying()) the call to "getNextFrame()":

   fSource->getNextFrame(fBuffer, sizeof fBuffer,
                         afterGettingFrame, this,
                         onSourceClosure, this);

This tells the data source to get up to "sizeof fBuffer" bytes of data, and 
store it at "fBuffer".  After doing this, the data source is to call 
"afterGettingFrame()", with "this" as argument.  (If, for some reason, the 
data source ends, then "onSourceClosure()" is to be called instead.)

So, if you want to change the place where the incoming data is delivered, 
then the best way to do this is to change the first parameter to the 
"getNextFrame()" call.

>   i changed the fwrite() function to do the ring
>buffer write but surprisingly i am unable to see the
>code flow entering the 'afterGettingFrame()' function.

You haven't said what object(s) you're using as a data source, but if it's 
successfully getting data from the network, then the "afterGettingFrame()" 
function should be getting called.

I suggest that you start with one of the "test...Receiver" programs - 
making sure that it receives and delivers data OK - before you start 
modifying it to change where the incoming data gets stored.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Wed Jan 30 12:29:15 2002
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Subject: [Live-devel] Audio/Video Mux and Demux
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Date: Wed Jan 30 12:30:03 2002
X-Original-Date: Wed, 30 Jan 2002 14:29:12 -0600
Status: O

Hi,

I was able to use the examples to write a streaming server that will demux
an MPEG-1 clip and send the audio and video streams. At the receiver side,
beoth streams are retrieved.

Is there a library for muxing back the audio and video streams so that I can
play the clip via a command line MPEG player?

If not, does this mean that in order to playback the MPEG clip, the reciever
has to playback audio and video streams individually? How does the player
synchronize the playback of the two streams?

I'm just starting with RTCP, so any hints will be highly appreciated.

Hazem



From <EMAIL-ADDRESS-HIDDEN> Wed Jan 30 12:49:44 2002
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Subject: Re: [Live-devel] Audio/Video Mux and Demux
From: Marty Schoch <EMAIL-ADDRESS-HIDDEN>
To: Hazem Hamed <EMAIL-ADDRESS-HIDDEN>
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Date: Wed Jan 30 12:50:07 2002
X-Original-Date: 30 Jan 2002 15:49:12 -0500
Status: O

I am by no means an expert on this subject, though I'm trying to learn.

It would seem that the quick-and-dirty way to do this would be to take
the RTP timestamp information, and recreate a MPEG1 program stream. 
This would allow you to pipe this stream into almost any existing
player.  There is software out there to mux elementary streams back
together.  However, none of these seem to have hooks in place for
syncing them with an external clock (RTP timestamps).  Definately a
possibility, and I might be interested in helping someone with this
effort.

The other situation, which seems cleaner to me is for a player to be
able to synchronize playback directly with the RTP timestamps.  This
would allow you to do things like syncrhonize a Sorenson-encoded QT
video stream with a MP3 audio stream.  Note this wouldn't be possible
with the first solution.  The only problem with this is that all the
open source players I'm aware of treat RTP playback as a hack.

Marty Schoch
<mschoch(at)multicasttech.com>

On Wed, 2002-01-30 at 15:29, Hazem Hamed wrote:
> Hi,
> 
> I was able to use the examples to write a streaming server that will demux
> an MPEG-1 clip and send the audio and video streams. At the receiver side,
> beoth streams are retrieved.
> 
> Is there a library for muxing back the audio and video streams so that I can
> play the clip via a command line MPEG player?
> 
> If not, does this mean that in order to playback the MPEG clip, the reciever
> has to playback audio and video streams individually? How does the player
> synchronize the playback of the two streams?
> 
> I'm just starting with RTCP, so any hints will be highly appreciated.
> 
> Hazem
> 
> 
> _______________________________________________
> Live-devel mailing list
> Live-devel(at)lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/live-devel




From <EMAIL-ADDRESS-HIDDEN> Wed Jan 30 13:04:49 2002
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Date: Wed Jan 30 13:05:04 2002
X-Original-Date: Wed, 30 Jan 2002 13:03:24 -0800
Status: O

At 12:29 PM 1/30/02, Hazem Hamed wrote:
>I was able to use the examples to write a streaming server that will demux
>an MPEG-1 clip and send the audio and video streams. At the receiver side,
>beoth streams are retrieved.
>
>Is there a library for muxing back the audio and video streams so that I can
>play the clip via a command line MPEG player?

No, there's no such code in the library at present.  Because the streams 
are being sent using RTP, you should use a RTP-compliant tool to receive 
and play them.  If you do this, then you won't need to combine the audio 
and video back together, as this would be done within the player tool itself.

For example, both RealPlayer and QuickTime Player should be able to receive 
and play MPEG/RTP audio and video streams (although QuickTime Player can 
currently only handle MPEG layer II audio, not layer III (aka. MP3)).  If 
you create a single SDP file that describes both the audio and video RTP 
streams, either of these players should be able to receive and play them.

(Another possible player tool that you can use is Cisco's "IP/TV Player".)

Take, for example, the following SDP description:
         v=0
         o=- 49451 3 IN IP4 127.0.0.1
         s=Test MPEG Audio/Video session
         i=Test MPEG Audio/Video session
         t=0 0
         m=audio 6666 RTP/AVP 14
         c=IN IP4 239.255.42.42/127
         m=video 8888 RTP/AVP 32
         c=IN IP4 239.255.42.42/127

This describes a MPEG audio stream that's being sent to 239.255.42.42, port 
6666, and a MPEG video stream that's being sent to 239.255.42.42, port 8888.

>If not, does this mean that in order to playback the MPEG clip, the reciever
>has to playback audio and video streams individually?

Yes, that's what RTP-compliant audio/video player tools do.

>  How does the player
>synchronize the playback of the two streams?

Using the timestamps in the individual RTP streams, along with the 
(NTP-based) timestamps in the corresponding RTCP "Sender Reports".

         Ross.



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Subject: [Live-devel] How can i develop my streaming application?
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Date: Fri Feb  1 07:17:13 2002
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------=_NextPart_000_0029_01C1AB3B.E593E6A0
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Hi!

My name is Pablo Espada and i'm a spanish student.=20
I'm gonna tell you my problem. I have to develop a streaming application =
similar to a telephony over IP application. To make it easier, it will =
be only half duplex (one person speaking and another one listening).
Now, i capture PCM sound from the soundcard (using Alsa drivers in =
Linux) and i send it to receiver application using raw UDP, and i would =
like to use RTSP or RTP instead.

I would like to use "live" but i don't know which modules shoud i use =
for this purpose. Should i use file source and sink and adapt them in =
order to take/put data from/in soundcard buffer on time? Perhaps another =
module?

I think I've also read in this list that test programs only work with =
multicast IP's. =BFis it true?

I hope someone could help me. Thank you !!!!
Pablo Espada.

------=_NextPart_000_0029_01C1AB3B.E593E6A0
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	charset="iso-8859-1"
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2600.0" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Hi!</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>My name is Pablo Espada and i'm a =
spanish student.=20
</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>I'm gonna tell you my problem. I have =
to develop a=20
streaming application similar to a telephony over IP application. To =
make it=20
easier, it will be only half duplex (one person speaking and another one =

listening).</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Now, i capture PCM sound from the =
soundcard (using=20
Alsa drivers in Linux) and i send it to receiver application using raw =
UDP, and=20
i would like to use RTSP or RTP instead.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>I would like to use "live" but i don't =
know which=20
modules shoud i use for this purpose. Should i use file source and sink =
and=20
adapt them in order to take/put data from/in soundcard buffer on time? =
Perhaps=20
another module?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>I think I've also read in this list =
that test=20
programs only work with multicast IP's. =BFis it true?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>I hope someone could help me. Thank you =

!!!!</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Pablo =
Espada.</FONT></DIV></BODY></HTML>

------=_NextPart_000_0029_01C1AB3B.E593E6A0--



From <EMAIL-ADDRESS-HIDDEN> Fri Feb 01 10:27:14 2002
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To: {Pablo Espada} <EMAIL-ADDRESS-HIDDEN>
From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
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Date: Fri Feb  1 10:29:50 2002
X-Original-Date: Fri, 01 Feb 2002 10:26:13 -0800
Status: O

At 07:17 AM 2/1/02, Pablo Espada wrote:
>I'm gonna tell you my problem. I have to develop a streaming application=20
>similar to a telephony over IP application. To make it easier, it will be=
=20
>only half duplex (one person speaking and another one listening).
>Now, i capture PCM sound from the soundcard (using Alsa drivers in Linux)=
=20
>and i send it to receiver application using raw UDP, and i would like to=20
>use RTSP or RTP instead.

This is a good idea, because you will be able to test your implementation=20
(both sending and receiving) using open standard tools such as "rat".

The RTP payload format for PCM is very very simple.  See=20
<http://www.ietf.org/internet-drafts/draft-ietf-avt-profile-new-12.txt>,=20
section 4.5.14.  PCM (u-law or a-law) samples are simply packed=20
sequentially into a RTP packet - as many as can fit.  There are also static=
=20
RTP payload types defined for PCM: 0 for PCMU (u-law), and 8 for PCMA=
 (a-law).

>  I would like to use "live" but i don't know which modules shoud i use=20
> for this purpose. Should i use file source and sink and adapt them in=20
> order to take/put data from/in soundcard buffer on time?

Yes, this will probably work.  You will also need specialized RTP sources=20
and sinks for PCM (instead of the "MPEGAudioRTPSink/Source" that=20
"testMP3Streamer/Receiver" use), but fortunately they will be very simple,=
=20
because the RTP payload format for PCM is so simple.

For a RTP source (i.e., for use by your receiving implementation), you can=
=20
simply use a "SimpleRTPSource" - i.e., call
         rtpSource =3D SimpleRTPSource::createNew(*env, &rtpGroupsock, 0,=20
8000, "audio/pcmu");
(If you're using a-law instead of u-law ,then substitute 8 for 0, and=20
"pcma" for "pcmu".)

For a RTP sink, there's currently no corresponding "SimpleRTPSink" (perhaps=
=20
I'll write one), but in the meantime you can define a very simple subclass=
=20
of "MultiFramedRTPSink" that does nothing other than define the RTP payload=
=20
type (0 or 8), and the RTP timestamp frequency (8000).


>I think I've also read in this list that test programs only work with=20
>multicast IP's. =BFis it true?

No, these programs can also use unicast addresses instead of multicast=20
addresses.  In the "...Streamer" application, you can use the unicast=20
address of the (single) destination.  In the "...Receiver" application, you=
=20
can use the unicast address of the sender (this is so that RTCP Reception=20
Reports go back to the sender).

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Fri Feb 01 21:30:49 2002
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Subject: [Live-devel] Help ! Modify the openRTSP
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Date: Fri Feb  1 21:31:29 2002
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Hello !
I have some problem of modifying the openRTSP.cpp .
I hope that after playing the *.mov file, it does not directyly exit .
In shutdown() :
   exit(exitCode);

So I try to command the line
  // exit(exitCode);
and add=20
   return 0;

But I found it will go to the loop blockMySelf(), so I add a new =
variable,
and modify the line
    exit(exitCode) =3D>=20
    watch =3D 0;

   env->taskScheduler().blockMyself(); // does not return   =3D>
   env->taskScheduler().blockMyself(&watch); // does not return

Although it can run out of the loop blockMySelf(),but it sometimes run =
into the
RTCP.cpp Schedule(double nextTime, event e)
and hang on an access exception.

How should I modify the code properly ?
Would you please do me a favor ?

Thank you very much!

        Armstrong






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	charset="big5"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content=3D"text/html; charset=3Dbig5" http-equiv=3DContent-Type>
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</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT size=3D2>Hello !</FONT></DIV>
<DIV><FONT size=3D2>I have some problem of modifying the openRTSP.cpp=20
.</FONT></DIV>
<DIV><FONT size=3D2>I hope that <FONT color=3D#0000ff>after playing the =
*.mov file,=20
it does not directyly exit </FONT>.</FONT></DIV>
<DIV><FONT size=3D2>In shutdown() :</FONT></DIV>
<DIV><FONT size=3D2>&nbsp;&nbsp; exit(exitCode);</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT size=3D2>So I try to command the line</FONT></DIV>
<DIV><FONT size=3D2>&nbsp; // exit(exitCode);</FONT></DIV>
<DIV><FONT size=3D2>and add </FONT></DIV>
<DIV><FONT size=3D2>&nbsp;&nbsp; return 0;</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT size=3D2>But I found it will go to the loop blockMySelf(), so =
I <FONT=20
color=3D#0000ff>add a new variable,</FONT></FONT></DIV>
<DIV><FONT size=3D2>and modify the line</FONT></DIV>
<DIV><FONT size=3D2>&nbsp;&nbsp;&nbsp; exit(exitCode) =3D&gt; =
</FONT></DIV>
<DIV><FONT size=3D2>&nbsp;&nbsp;&nbsp; <FONT color=3D#008080>watch =3D=20
0;</FONT></FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT size=3D2>&nbsp;&nbsp; env-&gt;taskScheduler().blockMyself(); =
// does=20
not return&nbsp;&nbsp; =3D&gt;</FONT></DIV>
<DIV><FONT size=3D2>&nbsp;&nbsp; <FONT=20
color=3D#008080>env-&gt;taskScheduler().blockMyself(&amp;watch); // does =
not=20
return</FONT></FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT size=3D2>Although it can run out of the loop =
blockMySelf(),but it=20
sometimes run into the</FONT></DIV>
<DIV><FONT size=3D2>RTCP.cpp&nbsp;<FONT color=3D#0000ff>Schedule(double =
nextTime,=20
event e)</FONT></FONT></DIV>
<DIV><FONT size=3D2>and&nbsp;hang on an <FONT color=3D#800000>access=20
e</FONT></FONT><FONT size=3D2><FONT =
color=3D#800000>xception.</FONT></FONT></DIV>
<DIV><FONT size=3D2><FONT color=3D#800000></FONT></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>How should I modify the code properly ?</FONT></DIV>
<DIV><FONT size=3D2>Would you please do me a favor ?</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>Thank you very much!</FONT><FONT =
size=3D2><BR></FONT></DIV>
<DIV><FONT size=3D2>&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; =
Armstrong</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2><BR>&nbsp;</DIV></FONT>
<DIV>&nbsp;</DIV></BODY></HTML>

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From <EMAIL-ADDRESS-HIDDEN> Sun Feb 03 08:22:46 2002
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Subject: [Live-devel] both video & audio
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Date: Sun Feb  3 08:23:16 2002
X-Original-Date: Sun, 3 Feb 2002 17:18:21 +0100
Status: O

I'm trying to stream both video and audio since several weeks.

I thought to combine both testMPEGvideoStreamer.cpp and
testMP3Streamer.cpp : it's quite difficult because I don't know how
to handle the audio stream produced by  mpegDemux.
I red in the newsgroup that a solution would be employing
a new module named MPEGAudioStreamFramer plus 
MPEGAudioRTPSink  for the audio stream.
Unfortunately I don't have the right knowledge to develop
by myself such module.

I'll be grateful to anyone would be so polite 
to let me solving this stuff.

thanks,
Domenico (mliguori(at)inwind.it)



From <EMAIL-ADDRESS-HIDDEN> Mon Feb 04 00:48:03 2002
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Subject: [Live-devel] I want to develop streaming video application VOD!
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Date: Mon Feb  4 00:49:02 2002
X-Original-Date: Mon, 4 Feb 2002 09:48:01 +0100 (CET)
Status: O

Hi 
 My name is GAZDAR Achraf and I am a student. To
achieve my Master degre I have to develop a
video-on-demand prototype allowing interactivity over
protocols such as Interactive Pagoda Broadcasting
Protocol. I want to use standards like RTP/RTCP RTSP.
I have downloaded live.com application but I can't
identify classes or function responsible of:
    1. retrieving data from video file
    2. sending video/audio on RTP format
    3. the codec and the player to use
so I will be very thankful if one can help me about
live.com architecture or give to me some advices.
thanks 
GAZDAR Achraf  
   
  

___________________________________________________________
Do You Yahoo!? -- Une adresse (at)yahoo.fr gratuite et en français !
Yahoo! Mail : http://fr.mail.yahoo.fr


From <EMAIL-ADDRESS-HIDDEN> Mon Feb 04 01:19:57 2002
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Subject: Re: [Live-devel] How can i develop my streaming application?
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Date: Mon Feb  4 01:20:04 2002
X-Original-Date: Mon, 04 Feb 2002 01:15:24 -0800
Status: O

Following up on my earlier response to Pablo Espada's message - I have now 
released a new version of the "liveMedia" library code that includes a new 
RTP 'sink' class "SimpleRTPSink".  This can be used for simple RTP payload 
formats that just send a single frame (or an unstructured set of bytes) at 
a time, without any fragmentation or special headers.  This class can be 
used to stream PCM audio over RTP.  E.g.,
	sink = SimpleRTPSink::createNew(*env, &rtpGroupsock, 0, 8000, "audio);
(This particular sink can be played from an unstructured byte-stream source.)

The new source code can be found (as usual) at <http://www.live.com/liveMedia/>

	Ross.



From <EMAIL-ADDRESS-HIDDEN> Mon Feb 04 01:28:32 2002
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Date: Mon Feb  4 01:29:03 2002
X-Original-Date: Mon, 04 Feb 2002 01:23:02 -0800
Status: O

Armstrong,

I think the modifications that you're trying to make are basically correct.

>Although it can run out of the loop blockMySelf(),but it sometimes run 
>into the
>RTCP.cpp Schedule(double nextTime, event e)
>and hang on an access exception.

Ahh, you found a bug in the "openRTSP" source code - it wasn't closing down 
it's "MediaSession" object when it was exiting.  (Properly closing this 
object is a good idea, because it will cause RTCP "BYE"s to get sent.)

You should be able to fix the bug by adding the following line to the end 
of the shutdown() routine:

           Medium::close(session);

(I have also fixed this in the most recent source code release.)

     Ross.



From <EMAIL-ADDRESS-HIDDEN> Mon Feb 04 01:32:44 2002
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Date: Mon Feb  4 01:33:05 2002
X-Original-Date: Mon, 04 Feb 2002 01:31:40 -0800
Status: O

At 08:18 AM 2/3/02, mimmo wrote:
>I'm trying to stream both video and audio since several weeks.

So many people have been asking for this recently (a test program that will 
stream both video and audio from a MPEG program stream file) that I'll 
probably go ahead and do this sometime later this week.  I'll send email to 
this list when it's ready to try out.

         Ross.



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  VOD!
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Date: Mon Feb  4 01:40:12 2002
X-Original-Date: Mon, 04 Feb 2002 01:39:00 -0800
Status: O

At 12:48 AM 2/4/02, achraf_g(at)yahoo.fr wrote:
>I have downloaded live.com application but I can't
>identify classes or function responsible of:
>     1. retrieving data from video file
>     2. sending video/audio on RTP format
>     3. the codec and the player to use

Please build, run and inspect the test programs "testMPEGVideoStreamer" and 
"testMPEGVideoReceiver".  (Similarly, "testMP3Streamer" and 
"testMP3Receiver" for audio).  You should also be familiar with RTP (as 
described in the Internet Draft 
<http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-new-11.txt>)

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Mon Feb 04 20:35:41 2002
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Date: Mon Feb  4 20:36:03 2002
X-Original-Date: Mon, 04 Feb 2002 20:33:09 -0800
Status: O

At 08:13 PM 2/4/02, Hazem Hamed wrote:
>We are unable to stream out MP2 using testMP3Streamer. It looks that all the
>liveMedia classes only support MP3. Is there a way around this so that we
>can send MP2?

Hazem,

Unfortunately the implementation of the "MP3FileSource" class (which 
"testMP3Streamer" uses as a source) currently has some code that checks 
specifically for layer III audio (i.e., MP3), which is why streaming MP2 
files didn't work for you.  However, you *may* be able to remove this 
restriction by making the following changes to 
"liveMedia/MP3StreamState.cpp".  (I haven't tested this myself, though.)

1/ Comment out line 227:
         || (fr().hdr & 0x00060000) != 0x00020000 /* test for layer 3 */

2/ Remove the "if" statement at line 325 that begins:
     if (fr().layer != 3) { ...


>If no, you think we can use this SimpleRTPSink for this
>purpose?

No, because streaming MPEG audio in RTP (using either RFC 2250 or RFC 3119) 
requires special headers, in addition to the MPEG audio frame(s).  But in 
any case, the problem is *not* the "RTPSink" class; "MPEGAudioRTPSink" 
should work just fine for you.  The problem is the source object that feeds 
the RTPSink; this object is responsible for parsing the input MPEG audio 
data into distinct frames (for consumption by the RTPSink object).  At 
present, the only source object (in the library) that does this is 
"MP3FileSource".

         Ross.

ps. For any follow-up questions, please use the mailing list 
<live-devel(at)lists.sourceforge.net>.  I prefer to reserve one-on-one email 
for consulting clients.



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Date: Tue Feb  5 10:30:08 2002
X-Original-Date: Tue, 5 Feb 2002 12:27:55 -0600
Status: O

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Hi Ross,

Thanks for the hint. I tried to disable the MP3 check, and MP2 audio was
sent, but still it seems there's a problem cause the transmission stops in
the middle of the track.

One other problem. I extended testMPEGSplitter.cpp to be able to split and
stream MPEG movies. This file, testMPEGStreamer.cpp is attached.
It seems that it cannot stream BOTH audio and video simultaneously. If I
comment out audioSink->startPlaying(), video is transmitted fine. When
videoSink->startPlaying is commented, audio is transmitting too. But when
both are active, nothing is being transmitted. I tested the outgoing
multicast packets and only 4-5 packets are transmitted, then nothing.

Ah, I appologize for sending a personal e-mail. I just blindly reply to your
e-mail expecting that the reply-to address is by default the mailing list
address. If you change your list-server config setting reply-to to the list
address, you'll get rid of such a problem.

Thanks and best regards,
Hazem

-----Original Message-----
From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
To: <EMAIL-ADDRESS-HIDDEN> <EMAIL-ADDRESS-HIDDEN>
Date: Monday, February 04, 2002 10:36 PM
Subject: Re: [Live-devel] How can i develop my streaming application?


>At 08:13 PM 2/4/02, Hazem Hamed wrote:
>>We are unable to stream out MP2 using testMP3Streamer. It looks that all
the
>>liveMedia classes only support MP3. Is there a way around this so that we
>>can send MP2?
>
>Hazem,
>
>Unfortunately the implementation of the "MP3FileSource" class (which
>"testMP3Streamer" uses as a source) currently has some code that checks
>specifically for layer III audio (i.e., MP3), which is why streaming MP2
>files didn't work for you.  However, you *may* be able to remove this
>restriction by making the following changes to
>"liveMedia/MP3StreamState.cpp".  (I haven't tested this myself, though.)
>
>1/ Comment out line 227:
>         || (fr().hdr & 0x00060000) != 0x00020000 /* test for layer 3 */
>
>2/ Remove the "if" statement at line 325 that begins:
>     if (fr().layer != 3) { ...
>
>
>>If no, you think we can use this SimpleRTPSink for this
>>purpose?
>
>No, because streaming MPEG audio in RTP (using either RFC 2250 or RFC 3119)
>requires special headers, in addition to the MPEG audio frame(s).  But in
>any case, the problem is *not* the "RTPSink" class; "MPEGAudioRTPSink"
>should work just fine for you.  The problem is the source object that feeds
>the RTPSink; this object is responsible for parsing the input MPEG audio
>data into distinct frames (for consumption by the RTPSink object).  At
>present, the only source object (in the library) that does this is
>"MP3FileSource".
>
>         Ross.
>
>ps. For any follow-up questions, please use the mailing list
><live-devel(at)lists.sourceforge.net>.  I prefer to reserve one-on-one email
>for consulting clients.
>
>
>_______________________________________________
>Live-devel mailing list
>Live-devel(at)lists.sourceforge.net
>https://lists.sourceforge.net/lists/listinfo/live-devel
>

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From <EMAIL-ADDRESS-HIDDEN> Tue Feb 05 13:09:56 2002
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From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] How can i develop my streaming application?
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Date: Tue Feb  5 13:10:09 2002
X-Original-Date: Tue, 05 Feb 2002 13:06:41 -0800
Status: O

At 10:27 AM 2/5/02, Hazem Hamed wrote:
>One other problem. I extended testMPEGSplitter.cpp to be able to split and
>stream MPEG movies. This file, testMPEGStreamer.cpp is attached.
>It seems that it cannot stream BOTH audio and video simultaneously. If I
>comment out audioSink->startPlaying(), video is transmitted fine. When
>videoSink->startPlaying is commented, audio is transmitting too. But when
>both are active, nothing is being transmitted. I tested the outgoing
>multicast packets and only 4-5 packets are transmitted, then nothing.

I'll take a look at this.  What you're doing looks correct (except that 
there's no "MPEGAudioStreamFramer" filter), so I'm yet not sure why it 
wouldn't be working.

         Ross.



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Date: Wed Feb  6 15:21:13 2002
X-Original-Date: Wed, 6 Feb 2002 15:20:55 -0800 (PST)
Status: O

Hi,

In my application I'm grabbing audio from the sound card and using LAME to
encode the data into MP3. I'd like to send the data across the network
using liveMedia's interleaved ADU's, but I can't figure out a good way to
do this.

Right now I am just taking each MP3 frame that returns from LAME and
putting it into an RTP packet and sending it... How can I convert this raw
MP3 frame into an ADU using the library? How can I use the ADUFromMP3Source
structures?

Thanks for your help!

Michael



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Date: Wed Feb  6 15:39:04 2002
X-Original-Date: Wed, 06 Feb 2002 15:38:14 -0800
Status: O

At 03:20 PM 2/6/02, Michael Kang wrote:
>In my application I'm grabbing audio from the sound card and using LAME to
>encode the data into MP3. I'd like to send the data across the network
>using liveMedia's interleaved ADU's, but I can't figure out a good way to
>do this.

Michael,

Note the "testMP3Streamer" and "testMP3Receiver" test programs that come 
bundled with the libraries (in the "testProgs" directory).  These already 
do most of what you want.  Note that you'll need to uncomment the line
         #define STREAM_USING_ADUS 1
in each case.

Also, in "testMP3Streamer.cpp", you can change the "inputFileName" string from
         "test.mp3"
to
         "stdin"
If you do this, you should just be able to pipe your LAME output into 
"testMP3Streamer".

Of course, you may also wish to change "multicastAddressStr", "rtpPortNum", 
"rtpPayloadFormat", and, if you choose to do interleaving, 
"interleaveCycle".  Ditto for the receiving application "testMP3Receiver" 
(which already outputs to stdout).

         Ross.



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Date: Thu Feb  7 07:05:08 2002
X-Original-Date: Thu, 07 Feb 2002 15:02:02 +0000
Status: O

Hi everyone,

I am new to this library and was wondering whether anyone had written any 
documentation for this library.  I noticed that none of the files have 
doc++/doxygen style comments is this in the future roadmap?

look forward to using the library

Cheers
Dan

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Subject: [Live-devel] I get error message when I test  :tesMPEGvideoStreamer!
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Date: Thu Feb  7 07:55:04 2002
X-Original-Date: Thu, 7 Feb 2002 16:54:14 +0100 (CET)
Status: O

Hi Ross,
I get this messege when I try to test the
testMPEGVideoStreamer :
MPEGVideoStreamParser::parseSlice(): Saw unexpected
code 0x000001ba
I have an MPEG1 File that we have changed its name to
"test.mpg".
Finally I want to know if there's any documentations
about this labrary
and haw to use it in VC++ ver 6.0.
I want to know how ton include this library in my
project. because the compilar display this message
"couldn' open file"   .h".
thanks
Achraf

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Date: Thu Feb  7 10:19:07 2002
X-Original-Date: Thu, 07 Feb 2002 10:15:03 -0800
Status: O

At 07:54 AM 2/7/02, achraf_g(at)yahoo.fr wrote:
>Hi Ross,
>I get this messege when I try to test the
>testMPEGVideoStreamer :
>MPEGVideoStreamParser::parseSlice(): Saw unexpected
>code 0x000001ba

By default, the "testMPEGVideoStreamer" test program assumes that its input 
is a MPEG Video Elementary Stream.  The files that you tried to stream 
(producing the error messages) are MPEG *Program Stream* files. To stream 
video from files like this, uncomment the line
         //#define SOURCE_IS_PROGRAM_STREAM 1
in testMPEGVideoStreamer.cpp, and recompile.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Thu Feb 07 10:23:11 2002
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Date: Thu Feb  7 10:24:03 2002
X-Original-Date: Thu, 07 Feb 2002 10:21:06 -0800
Status: O

At 07:02 AM 2/7/02, Daniel Laird wrote:
>I am new to this library and was wondering whether anyone had written any 
>documentation for this library.

Daniel,

For now, the best way to understand how to use the libraries is to (i) 
study the example programs in the "testProgs" directory, and (especially) 
(ii) ask questions on this mailing list.

At some point (in the not too distant future) I definitely want to produce 
some real documentation for the libraries (especially now that the APIs 
seem to be stabilizing), but right now I have to focus on activities that 
pay the bills.


>I noticed that none of the files have doc++/doxygen style comments is this 
>in the future roadmap?

I haven't used either of these, so I'd be interested to hear opinions from 
the list as to the relative merits of each (and other relevant 
documentation systems).

         Ross.



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Subject: [Live-devel] H.263 Video and GSM Audio
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Date: Thu Feb  7 11:42:03 2002
X-Original-Date: Thu, 7 Feb 2002 13:41:08 -0600
Status: O

Hi,

I'm currently developing a media streaming server based on liveMedia
library. I've started with the existing classes, and found that MPEG video
and MP3 audio are the supported formats. Since MPEG requires a relatively
high network bandwidth, I'm thinking of using H.263 video and GSM audio.

Is it possible to re-use the liveMedia class library to implement classes
for streaming H.263 video and GSM audio? Any hints are highly appreciated.

Thanks and best regards.
Hazem



From <EMAIL-ADDRESS-HIDDEN> Thu Feb 07 12:56:41 2002
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Date: Thu Feb  7 12:57:05 2002
X-Original-Date: Thu, 07 Feb 2002 12:56:38 -0800
Status: O

At 11:41 AM 2/7/02, you wrote:
>Is it possible to re-use the liveMedia class library to implement classes
>for streaming H.263 video and GSM audio? Any hints are highly appreciated.

Hazem,

You're in luck - the library already contains RTP 'sink' classes for this: 
"H263plusVideoRTPSink" and "GSMAudioRTPSink".  You can use these in place 
of "MPEGVideoRTPSink" and "MPEGAudioRTPSink" (respectively).  Note that, in 
each case, these classes will need to be fed discrete frames, so if you're 
reading the input data from a byte stream, it will need to be framed 
appropriately.

Fortunately, GSM audio frames are always 33 bytes in length, so you can 
easily read GSM frames using a "ByteStreamFileSource", as follows:
         ByteStreamFileSource::createNew(*env, inputFileName, 33, 0.02);
(The "33" tells the ByteStreamFileSource to return 33 bytes of data each 
time it's called.  The "0.02" indicates that each frame is timed 20 ms apart.)

Unfortunately, H.263+ frames vary in size, so if you're reading H.263+ data 
from a byte stream, you will need to implement a separate 'framer' filter 
that feeds into the "H263plusVideoRTPSink".


At the receiving end:
- For H.263+:
There's a "H263plusVideoRTPSource" class that can be used just like 
"MPEGVideoRTPSource".

- For GSM:
Because the GSM RTP payload format is so simple, you can just create a 
"SimpleRTPSource" to receive this data - using 
"SimpleRTPSource::createNew().  (The "rtpPayloadFormat" parameter will be 
3; the "rtpTimestampFrequency" parameter will be 8000; the "mimeType" 
parameter will be "audio/gsm"; the "offset" parameter will be 0.)

I hope this helps.

         Ross.

         



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Subject: Re: [Live-devel] H.263 Video and GSM Audio
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Date: Thu Feb  7 13:53:10 2002
X-Original-Date: Thu, 7 Feb 2002 15:43:06 -0600
Status: O

That's great... Thanks Ross... I'll keep you updated with my progress.

Hazem

----- Original Message -----
From: {Ross Finlayson} <EMAIL-ADDRESS-HIDDEN>
To: <EMAIL-ADDRESS-HIDDEN>
Sent: Thursday, February 07, 2002 2:56 PM
Subject: Re: [Live-devel] H.263 Video and GSM Audio


> At 11:41 AM 2/7/02, you wrote:
> >Is it possible to re-use the liveMedia class library to implement classes
> >for streaming H.263 video and GSM audio? Any hints are highly
appreciated.
>
> Hazem,
>
> You're in luck - the library already contains RTP 'sink' classes for this:
> "H263plusVideoRTPSink" and "GSMAudioRTPSink".  You can use these in place
> of "MPEGVideoRTPSink" and "MPEGAudioRTPSink" (respectively).  Note that,
in
> each case, these classes will need to be fed discrete frames, so if you're
> reading the input data from a byte stream, it will need to be framed
> appropriately.
>
> Fortunately, GSM audio frames are always 33 bytes in length, so you can
> easily read GSM frames using a "ByteStreamFileSource", as follows:
>          ByteStreamFileSource::createNew(*env, inputFileName, 33, 0.02);
> (The "33" tells the ByteStreamFileSource to return 33 bytes of data each
> time it's called.  The "0.02" indicates that each frame is timed 20 ms
apart.)
>
> Unfortunately, H.263+ frames vary in size, so if you're reading H.263+
data
> from a byte stream, you will need to implement a separate 'framer' filter
> that feeds into the "H263plusVideoRTPSink".
>
>
> At the receiving end:
> - For H.263+:
> There's a "H263plusVideoRTPSource" class that can be used just like
> "MPEGVideoRTPSource".
>
> - For GSM:
> Because the GSM RTP payload format is so simple, you can just create a
> "SimpleRTPSource" to receive this data - using
> "SimpleRTPSource::createNew().  (The "rtpPayloadFormat" parameter will be
> 3; the "rtpTimestampFrequency" parameter will be 8000; the "mimeType"
> parameter will be "audio/gsm"; the "offset" parameter will be 0.)
>
> I hope this helps.
>
>          Ross.
>
>
>
>
> _______________________________________________
> Live-devel mailing list
> Live-devel(at)lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/live-devel



From <EMAIL-ADDRESS-HIDDEN> Mon Feb 11 10:43:07 2002
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Date: Mon Feb 11 10:44:06 2002
X-Original-Date: Mon, 11 Feb 2002 10:42:58 -0800
Status: O

At 10:27 AM 2/5/02, Hazem Hamed wrote:
>One other problem. I extended testMPEGSplitter.cpp to be able to split and
>stream MPEG movies. This file, testMPEGStreamer.cpp is attached.
>It seems that it cannot stream BOTH audio and video simultaneously. If I
>comment out audioSink->startPlaying(), video is transmitted fine. When
>videoSink->startPlaying is commented, audio is transmitting too. But when
>both are active, nothing is being transmitted. I tested the outgoing
>multicast packets and only 4-5 packets are transmitted, then nothing.

FYI, I've just found a bug (in my MPEG stream parsing code) that was 
causing this problem.  I'll be releasing a fix shortly, along with a test 
program that will stream both audio and video from a MPEG Program Stream 
file.  So, please stay tuned...

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Mon Feb 11 17:14:39 2002
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Date: Mon Feb 11 17:15:13 2002
X-Original-Date: Mon, 11 Feb 2002 17:14:06 -0800
Status: O

By popular demand, I have now released a new test program 
"testMPEGAudioVideoStreamer" that demonstrates how to stream *both* audio 
and video (from a MPEG 1 or 2 Program Stream source).  Note the new source 
file "testMPEGAudioVideoStreamer.cpp".

The SDP file "testMPEGAudioVideoStreamer.sdp" can be used to view the 
audio+video streams from this test program.  QuickTime Player 5 works 
well.  You can also use RealNetworks' "RealOne" player.  (Neither of these 
players support MPEG-2 streams, however.  To play MPEG-2/RTP video streams, 
I suggest Cisco's "IP/TV" player.)

To make audio+video streaming work, I also needed to make some minor 
additions and fixes to the "liveMedia" library, so you should update this also.

The new source code is currently available at 
<http://www.live.com/liveMedia/>  (The Sourceforge.net repository is 
currently offline)

	Ross.



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Date: Tue Feb 12 04:35:10 2002
X-Original-Date: Tue, 12 Feb 2002 13:32:42 +0100
Status: O

This is a multi-part message in MIME format.

------=_NextPart_000_001D_01C1B3C9.BF38BE90
Content-Type: text/plain;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

Hi Ross!

I want to say thanks for your work;

sincerely

Domenico


------=_NextPart_000_001D_01C1B3C9.BF38BE90
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http-equiv=3DContent-Type>
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<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2><FONT size=3D2>
<P>Hi Ross!</P>
<P>I want to say thanks for your work;</P>
<P>sincerely</P>
<P>Domenico</P></FONT></FONT></DIV></BODY></HTML>

------=_NextPart_000_001D_01C1B3C9.BF38BE90--



From <EMAIL-ADDRESS-HIDDEN> Tue Feb 12 05:04:05 2002
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Subject: [Live-devel] Mpeg2 Transport Streams
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Date: Tue Feb 12 05:05:04 2002
X-Original-Date: Tue, 12 Feb 2002 13:03:49 +0000
Status: O

Hi there,

I am doing an MPEG2 related project and am lookinh into the possibility of 
getting a MPEG2 Transport Stream around using RTP as a way to do this.

The first question is that have you implmented this.
The second question is that rfc 2250 is very vague does anyone have any good 
thoughts on What i should include in the profile specific headers - as the 
RFC only defines Elementary Streams?

Look forward to hearing from you
Cheers
Dan

_________________________________________________________________
Get your FREE download of MSN Explorer at http://explorer.msn.com/intl.asp.



From <EMAIL-ADDRESS-HIDDEN> Tue Feb 12 11:08:57 2002
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Date: Tue Feb 12 11:09:33 2002
X-Original-Date: Tue, 12 Feb 2002 11:08:52 -0800
Status: O

At 05:03 AM 2/12/02, Daniel Laird wrote:
>The second question is that rfc 2250 is very vague does anyone have any 
>good thoughts on What i should include in the profile specific headers - 
>as the RFC only defines Elementary Streams?

RFC 2250 can be a bit confusing, but it does defines a RTP payload format 
for Transport Streams (and Program Streams) - in section 2.  (The rest of 
the RFC applies only to Elementary Streams).  In other words, you should 
read just section 2 of RFC 2250, and ignore the rest.


>The first question is that have you implmented this.

Yes, sort of.  The RTP payload for MPEG Transport Streams is so simple that 
it can be implemented (for transmitting) using a "SimpleRTPSink" - e.g.,
         SimpleRTPSink::createNew(*env, rtpGS, 33, 90000, "application");
(Note that the static payload 33 is defined for MPEG Transport Streams.)

You will also need a source that 'frames' the MPEG Transport Stream into 
discrete (188 byte?) frames.

At the receiving end, you can use a "SimpleRTPSource" - e.g.,
         SimpleRTPSource::createNew(*env, rtpGS, 33, 90000, 
"application/mp2t");

I hope this helps.

     Ross.



From <EMAIL-ADDRESS-HIDDEN> Thu Feb 14 05:50:09 2002
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Subject: [Live-devel] IPv6 support to live.com?
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Date: Thu Feb 14 05:51:02 2002
X-Original-Date: Thu, 14 Feb 2002 15:49:55 +0200
Status: O

Hi!
Is there any schedule when IPv6 support will be added to live.com?

I have tried to do the changes by myself, but I have discovered some
problems.
If someone could answer.

1. I have discovered, that in IPv4 sockets, in struct sockaddr_in, the
variable in_addr.sin_addr; seems to be unsigned int, and in IPv6 the same
variable ...sin6_addr seems to unsigned char[16] ? Is that true? At least my
compiler says so...

2. If the following is true, it seems that  everywhere in code, int
variables have been used, so they all have to be changed to char[16], and
there is no change to use same code in both applications, in IPv4 and IPv6?

Thanks
Tomi



From <EMAIL-ADDRESS-HIDDEN> Thu Feb 14 10:25:26 2002
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Date: Thu Feb 14 10:26:06 2002
X-Original-Date: Thu, 14 Feb 2002 10:24:27 -0800
Status: O

Tomi,

Unfortunately (as you noticed) the code assumes, in several places, that IP 
addresses will fit in 4-byte words.

The good news, however, is that most of these uses of IP addresses are 
contained within the "groupsock" library, so that's the only part of the 
code that really needs to be fixed to support IPv6.

This is on my "to do" list, but I don't yet have a timeframe for when it 
will be done. John Spurling (a subscriber to this mailing list) also tells 
me that he is taking a look at this.

	Ross.



From <EMAIL-ADDRESS-HIDDEN> Thu Feb 14 10:36:51 2002
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Date: Thu Feb 14 10:37:07 2002
X-Original-Date: Thu, 14 Feb 2002 18:36:48 +0000
Status: O

Tomi et al.,

i'm more or less in the same place you are: changing libgroupsock to
not rely on 32 bit addresses. i hope to finish up by this weekend and
start testing. if you're in a big hurry, feel free to go ahead. if
not, i should have some patches for Ross fairly soon.

-john


On Thu, Feb 14, 2002 at 10:24:27AM -0800, Ross Finlayson wrote:
> Tomi,
> 
> Unfortunately (as you noticed) the code assumes, in several places, that IP 
> addresses will fit in 4-byte words.
> 
> The good news, however, is that most of these uses of IP addresses are 
> contained within the "groupsock" library, so that's the only part of the 
> code that really needs to be fixed to support IPv6.
> 
> This is on my "to do" list, but I don't yet have a timeframe for when it 
> will be done. John Spurling (a subscriber to this mailing list) also tells 
> me that he is taking a look at this.
> 
> 	Ross.
> 
> 
> _______________________________________________
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From <EMAIL-ADDRESS-HIDDEN> Mon Feb 18 00:32:08 2002
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Subject: [Live-devel] problems when compiling with VC++ 6.0!!
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Date: Mon Feb 18 00:33:03 2002
X-Original-Date: Mon, 18 Feb 2002 09:32:07 +0100 (CET)
Status: O

Hi
 I try to re-use the test  
"programtestMPEGAudioVideoStreamer.cpp" in my code but
I face this compilation-message: 
  "error C2664: 'startPlaying' : cannot convert
parameter 2 from 'void (void *)' to 'void (__cdecl
*)(void *)'
        None of the functions with this name in scope
match the target type"
 I will be grateful if one can help me!!
Achraf  

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From <EMAIL-ADDRESS-HIDDEN> Mon Feb 18 03:09:46 2002
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Date: Mon Feb 18 03:10:12 2002
X-Original-Date: Mon, 18 Feb 2002 03:08:26 -0800
Status: O

At 12:32 AM 2/18/02, achraf_g(at)yahoo.fr wrote:
>Hi
>  I try to re-use the test
>"programtestMPEGAudioVideoStreamer.cpp" in my code but
>I face this compilation-message:
>   "error C2664: 'startPlaying' : cannot convert
>parameter 2 from 'void (void *)' to 'void (__cdecl
>*)(void *)'

That's odd.  Do you see the same error when you build 
"testMPEGVideoStreamer"? Both programs call "startPlaying()" in the same 
way, so it would be strange if the error occurred only for one of these 
programs.

In the calls to "startPlaying()", try replacing
         afterPlaying
with
         (afterPlayingFunc)afterPlaying
or
         (afterPlayingFunc)&afterPlaying
and see if that makes the error go away.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Mon Feb 18 06:11:58 2002
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Date: Mon Feb 18 06:12:12 2002
X-Original-Date: Mon, 18 Feb 2002 15:11:57 +0100 (CET)
Status: O

 Hi
I have made chages but I get the same error-message.
Note  that I face thoses problem not when bilding
live.com but when I try to reproduce the same code as
"testMPEGAudioVideoStreamer.cpp" in my "start button"
function in my application.
the error message is "cannot convert parameter 2 from
'void (void *)' to 'void (__cdecl *)(void *)'
        None of the functions with this name in scope
match the target type"
thanks 
 

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From <EMAIL-ADDRESS-HIDDEN> Mon Feb 18 10:03:35 2002
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Date: Mon Feb 18 10:04:04 2002
X-Original-Date: Mon, 18 Feb 2002 10:02:12 -0800
Status: O

At 06:11 AM 2/18/02, achraf_g(at)yahoo.fr wrote:
>I have made chages but I get the same error-message.
>Note  that I face thoses problem not when bilding
>live.com but when I try to reproduce the same code as
>"testMPEGAudioVideoStreamer.cpp" in my "start button"
>function in my application.

OK, I understand now.  So if the test programs compile OK for you, you 
should make sure that your own code calls "startPlaying()" in the same way.

In particular, the second parameter to "startPlaying()" - i.e., a function 
that gets called when the input source closes - should be declared like:
         void afterPlaying(void* clientData) {
         ...
         }

Note that if you use a class's member function for this, then it *must* be 
static.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Mon Feb 18 14:53:47 2002
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Subject: [Live-devel] trying to play .mpg file
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Date: Mon Feb 18 14:54:03 2002
X-Original-Date: Mon, 18 Feb 2002 14:53:46 -0800 (PST)
Status: O

hi Ross,
i am trying to play a .mpg file using
testMPEGVideoStreamer test program and i am getting a
message called 
" MPEGVideoStreamerParser::parseSlice(): saw
unexpected code "

why am i getting that error? can u please advise me
what to do?

thank you
surya

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From <EMAIL-ADDRESS-HIDDEN> Mon Feb 18 15:17:32 2002
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Date: Mon Feb 18 15:18:03 2002
X-Original-Date: Mon, 18 Feb 2002 15:15:52 -0800
Status: O

At 02:53 PM 2/18/02, you wrote:
>hi Ross,
>i am trying to play a .mpg file using
>testMPEGVideoStreamer test program and i am getting a
>message called
>" MPEGVideoStreamerParser::parseSlice(): saw
>unexpected code "
>
>why am i getting that error? can u please advise me
>what to do?

By default, the "testMPEGVideoStreamer" test program assumes that its input 
is a MPEG Video Elementary Stream.  The files that you tried to stream 
(producing the error messages) was probably a MPEG *Program Stream* file. 
To stream video from files like this, uncomment the line
         //#define SOURCE_IS_PROGRAM_STREAM 1
in testMPEGVideoStreamer.cpp, and recompile.

Alternatively, you can run the "testMPEGAudioVideoStreamer" test program, 
which will stream both audio and video from a Program Stream file.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Tue Feb 19 02:35:28 2002
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Date: Tue Feb 19 02:36:02 2002
X-Original-Date: Tue, 19 Feb 2002 11:35:28 +0100 (CET)
Status: O

Hi,
 To resolve the problem I change StartPlaying function
with only two parametres. the new problem arrearing
now is a LINK one. I get those error when linking:
(ServerVod.obj : error LNK2001: unresolved external
symbol "public: virtual __thiscall
BasicTaskScheduler::~BasicTaskScheduler(void)"
(??1BasicTaskScheduler(at)(at)UAE(at)XZ)).
I think that this problem is dealing with build and
link configuration but I don't Know the right one.
So can one tell me how to overcome thoses problems??
Thanks
Achraf, 


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From <EMAIL-ADDRESS-HIDDEN> Wed Feb 20 03:32:54 2002
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Date: Wed Feb 20 03:33:02 2002
X-Original-Date: Wed, 20 Feb 2002 12:32:53 +0100 (CET)
Status: O

 Hi
When I have build live.com over Windows2000 server I
haven't face problems. but when I have trayed to bring
it at Home on my Win98 Os, I face problesms dealing
with connection to "groupsocks".
How to overcome this. Note that I have'nt network
adapter at home!
Thanks
Achraf.
 

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From <EMAIL-ADDRESS-HIDDEN> Wed Feb 20 04:05:46 2002
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Subject: [Live-devel] Problems with unicast
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Date: Wed Feb 20 04:06:08 2002
X-Original-Date: Wed, 20 Feb 2002 14:05:38 +0200
Status: O

Hi!

I have made an application that sends h.263 streams using unicast addresses.
It works fine on normal linux distribution, but when I try to run that
application on iPaq handheld that runs on familiar distribution the
application fails to stream. It starts to send video stream, but destination
address is always 236.216.127.191 which I believe is reserved for multicast.
I'm using H263plusVideoRTPSink for transmission. Any suggestions what's
wrong?

Thanks
Tomi



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Date: Wed Feb 20 13:02:14 2002
X-Original-Date: Wed, 20 Feb 2002 11:46:41 -0800
Status: O

At 02:35 AM 2/19/02, achraf_g(at)yahoo.fr wrote:
>I get those error when linking:
>(ServerVod.obj : error LNK2001: unresolved external
>symbol "public: virtual __thiscall
>BasicTaskScheduler::~BasicTaskScheduler(void)"
>(??1BasicTaskScheduler(at)(at)UAE(at)XZ)).

It sounds like you're not linking in the "BasicUsageEnvironment" library.

I suggest using the (working) "testProgs" project as a model for your new 
project.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Wed Feb 20 12:59:23 2002
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Date: Wed Feb 20 13:02:16 2002
X-Original-Date: Wed, 20 Feb 2002 10:19:48 -0800
Status: O

At 03:43 PM 2/19/02, john spurling wrote:
>two questions:
>    -what is source specific multicast? i don't see anything about it
>in stevens.

"Source Specific Multicast" (SSM) is a recently-developed form of multicast 
where the 'group address' (as used by receivers) consists of both a 
traditional multicast address, *and* the source address of the sender.  It 
can be used as an alternative to the traditional multicast model ("Any 
Source Multicast", or "ASM" for short) for the common case where all of the 
multicast data is being sent by a single source.

For more information, see 
<http://www.ietf.org/internet-drafts/draft-ietf-ssm-arch-00.txt>


>    -what is the GroupEId class really used for? it seems like it's
>only really used for holding onto addresses in Groupsock.

The "GroupEId" class is intended to encapsulate all of the state of a 
multicast endpoint.  It includes not only the group address (ASM or SSM), 
but also the TTL ('time to live') that will be used in outgoing packets 
(sent to the multicast group).  It also includes a field to indicate an 
'encryption key' to be used for encrypted data, but to date this is not 
used anywhere.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Thu Feb 21 08:36:48 2002
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Date: Thu Feb 21 08:37:09 2002
X-Original-Date: Thu, 21 Feb 2002 03:50:35 -0800
Status: O

At 03:32 AM 2/20/02, you wrote:
>  Hi
>When I have build live.com over Windows2000 server I
>haven't face problems. but when I have trayed to bring
>it at Home on my Win98 Os, I face problesms dealing
>with connection to "groupsocks".
>How to overcome this. Note that I have'nt network
>adapter at home!

That's the problem.  On Windows (some versions, at least) socket operations 
don't work properly if there's no network interface.

What is the point of running streaming sofrware on a computer without any 
network interfaces??

         Ross.



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Date: Thu Feb 21 08:37:12 2002
X-Original-Date: Thu, 21 Feb 2002 03:58:15 -0800
Status: O

At 04:05 AM 2/20/02, you wrote:
>I have made an application that sends h.263 streams using unicast addresses.
>It works fine on normal linux distribution, but when I try to run that
>application on iPaq handheld that runs on familiar distribution

That's interesting.  This is the first time I've heard of someone compiling 
the libraries for a StrongArm.


>  the
>application fails to stream. It starts to send video stream, but destination
>address is always 236.216.127.191 which I believe is reserved for multicast.

That's really odd.  Do the "test*" programs (e.g., testMPEGVideoStreamer) 
work properly on the iPaq?  If so, do they still work OK if you change the 
multicast address string to a unicast address?

I suggest first making sure that the "test*" programs work OK on the iPaq, 
before trying to figure out why your new code is not working.

         Ross.



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Date: Thu Feb 21 14:04:05 2002
X-Original-Date: Thu, 21 Feb 2002 14:03:33 -0800 (PST)
Status: O

hi Ross,
 i am trying to learn about the MPEG strems and i am
trying to stream the MPEG1 streams using rtsp.
 can i use openRTSP to do it? what are the
modifications to be done ? 
 when i tried to do it, i am getting a 404 file not
found error. i am using the real server as a streaming
server.
please can u advise me how to do?
thank you
surya

__________________________________________________
Do You Yahoo!?
Yahoo! Sports - Coverage of the 2002 Olympic Games
http://sports.yahoo.com


From <EMAIL-ADDRESS-HIDDEN> Thu Feb 21 14:07:04 2002
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Date: Thu Feb 21 14:08:03 2002
X-Original-Date: Thu, 21 Feb 2002 14:07:02 -0800 (PST)
Status: O

hi Ross,
 i am trying to learn about MPEG streaming .and i am
trying to stream MPEG1 streams using rtsp. 
can i use openRTSP for it? what are the necessary
modifications that need to be done?
  when i tried it i ma getting 404 file not found
error.i am using real server as the streaming server.
please can u advise me how to do?
thank you
surya


__________________________________________________
Do You Yahoo!?
Yahoo! Sports - Coverage of the 2002 Olympic Games
http://sports.yahoo.com


From <EMAIL-ADDRESS-HIDDEN> Fri Feb 22 13:33:51 2002
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Date: Fri Feb 22 13:34:06 2002
X-Original-Date: Thu, 21 Feb 2002 22:07:41 -0800
Status: O

At 02:07 PM 2/21/02, suryakumari sagiraju wrote:
>  i am trying to learn about MPEG streaming .and i am
>trying to stream MPEG1 streams using rtsp.
>can i use openRTSP for it?

Yes, if you have a RTSP server that streams MPEG-1 using RTP, then you can 
use the "openRTSP" client program to setup, play, and (optionally) receive 
the streamed data.

>  what are the necessary
>modifications that need to be done?

None.

>   when i tried it i ma getting 404 file not found
>error.i am using real server as the streaming server.
>please can u advise me how to do?

Unfortunately I can't help you with setting up Real Server.  But you should 
make sure that te MPEG data is being streamed using *RTP*.  (RealNetworks 
sometimes uses their own, proprietary protocol - rather than RTP - to 
stream data.)

You could also try using Apple's "QuickTime Streaming Server".  This is 
quite easy to set up.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Wed Feb 27 16:40:52 2002
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Date: Wed Feb 27 16:41:06 2002
X-Original-Date: Wed, 27 Feb 2002 16:40:46 -0800
Status: O

At 02:57 PM 2/26/02, Sandeep wrote:
>Hi,
>    Do you have an example of server? As you mentioned in
>testprogams, in order to run those client programs we need
>RealNetwork or Quicktime. How can we run those client
>programs with Quicktime? Does it mean that I have to write
>RTP server in order to run programs in testporgram dir? Do
>you have an example of those server program?

Sandeep,

"testMP3Streamer", "testMPEGVideoServer", and "testMPEGAudioVideoServer" 
are all 'server' programs (i.e., they send RTP streams, rather than receive 
them).

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Wed Feb 27 22:33:11 2002
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Date: Wed Feb 27 22:34:03 2002
X-Original-Date: Thu, 28 Feb 2002 16:47:51 +1030
Status: O

Ross,

I suggest the following change in DelayQueue.cpp:


Current code:

DelayQueue::~DelayQueue() {
}


New code:

DelayQueue::~DelayQueue() {
  while (fNext != this)
    removeEntry1(fNext);
}


or perhaps:

DelayQueue::~DelayQueue() {
  ScopedRWLock l(fLock);
  while (fNext != this)
    removeEntry1(fNext);
}


Reason: If the program is supposed to stop broadcasting and the delay queue
is to be deleted, the memory taken for handlers of any outstanding delayed
tasks will be freed properly.


Regards

Vesselin


From <EMAIL-ADDRESS-HIDDEN> Thu Feb 28 00:06:22 2002
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Date: Thu Feb 28 00:07:02 2002
X-Original-Date: Thu, 28 Feb 2002 00:03:35 -0800
Status: O

At 09:18 PM 2/27/02, you wrote:
>Hi Ross,
>     Where is test.mpg file located?

(In the future, could you please use the developers' mailing list 
<live-devel(at)lists.sourceforge.net>.  For one-on-one email, I generally give 
priority to paying consulting clients.)

The file "test.mpg" (or "test.mp3" for "testMP3Streamer") is a MPEG file 
that you provide yourself.

         Ross.



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Subject: [Live-devel] testMP3Streamer Problem
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Date: Thu Feb 28 11:35:06 2002
X-Original-Date: Thu, 28 Feb 2002 13:34:44 -0600
Status: O

Hi,

When I run "testMP3Streamer" on Solaris 8 with the latest liveMedia library
release, I get the following error:

Beginning streaming...
...done streaming
Beginning streaming...
...done streaming
Beginning streaming...
...done streaming
.
.
19:28:10 Groupsock(-1: 239.255.42.42, 8888, 1): failed to join group:
setsockopt(IP_ADD_MEMBERSHIP) error: Bad file number
19:28:10 Groupsock(-1: 239.255.42.42, 8889, 1): failed to join group:
setsockopt(IP_ADD_MEMBERSHIP) error: Bad file number
Segmentation fault (core dumped)

The program used to run properly in the earlier realease of liveMedia. I
also tried "testMPEGAudioVideoStreamer" and it worked fine.

Hazem



From <EMAIL-ADDRESS-HIDDEN> Thu Feb 28 14:39:19 2002
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Date: Thu Feb 28 14:40:04 2002
X-Original-Date: Thu, 28 Feb 2002 14:34:59 -0800
Status: O

Vesselin,

Thanks for the fix.  It will appear in the next source code release.

	Ross.



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Date: Thu Feb 28 14:48:16 2002
X-Original-Date: Thu, 28 Feb 2002 14:41:06 -0800
Status: O

At 11:34 AM 2/28/02, Hazem Hamed wrote:
>setsockopt(IP_ADD_MEMBERSHIP) error: Bad file number
>19:28:10 Groupsock(-1: 239.255.42.42, 8889, 1): failed to join group:
>setsockopt(IP_ADD_MEMBERSHIP) error: Bad file number
>Segmentation fault (core dumped)
>
>The program used to run properly in the earlier realease of liveMedia. I
>also tried "testMPEGAudioVideoStreamer" and it worked fine.

Hazem,

That's strange, because "testMPEGAudioVideoStreamer" and "testMP3Streamer" 
use the same networking code, and the same multicast address.  (Only the 
RTP payload format is different.)  Also, "testMP3Streamer" uses the same 
port number (8888) that "testMPEGAudioVideoStreamer" uses for video.

Are you trying to run "testMP3Streamer" at the same time as one of the 
other "test*" programs?  (That shouldn't be a problem, but might help 
explain what's going wrong.)

If you recompile everything from scratch, and run each program 
one-at-a-time, is it still the case that "testMPEGAudioVideoStreamer" works 
OK, but "testMP3Streamer" doesn't?

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Thu Feb 28 15:34:04 2002
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Date: Thu Feb 28 15:35:02 2002
X-Original-Date: Thu, 28 Feb 2002 17:34:02 -0600
Status: O

I'm running testMP3Streamer only. I did make clean and a fresh build, but
now I get a different error:

Beginning streaming...
...done streaming
Beginning streaming...
...done streaming
Beginning streaming...
...done streaming
.
.
Failed to create RTSP server: unable to create stream socket: Too many open
files

The first two lines (Beginning streaming...) are repeated 29 times before
the last error is reported. It seems that the streaming is started several
times before it crashes. In the old liveMedia, testMP3Streamer prints only
the first line (Beginning streaming...) and keeps streaming fine.

Hazem

----- Original Message -----
From: {Ross Finlayson} <EMAIL-ADDRESS-HIDDEN>
To: <EMAIL-ADDRESS-HIDDEN>
Sent: Thursday, February 28, 2002 4:41 PM
Subject: Re: [Live-devel] testMP3Streamer Problem


> At 11:34 AM 2/28/02, Hazem Hamed wrote:
> >setsockopt(IP_ADD_MEMBERSHIP) error: Bad file number
> >19:28:10 Groupsock(-1: 239.255.42.42, 8889, 1): failed to join group:
> >setsockopt(IP_ADD_MEMBERSHIP) error: Bad file number
> >Segmentation fault (core dumped)
> >
> >The program used to run properly in the earlier realease of liveMedia. I
> >also tried "testMPEGAudioVideoStreamer" and it worked fine.
>
> Hazem,
>
> That's strange, because "testMPEGAudioVideoStreamer" and "testMP3Streamer"
> use the same networking code, and the same multicast address.  (Only the
> RTP payload format is different.)  Also, "testMP3Streamer" uses the same
> port number (8888) that "testMPEGAudioVideoStreamer" uses for video.
>
> Are you trying to run "testMP3Streamer" at the same time as one of the
> other "test*" programs?  (That shouldn't be a problem, but might help
> explain what's going wrong.)
>
> If you recompile everything from scratch, and run each program
> one-at-a-time, is it still the case that "testMPEGAudioVideoStreamer"
works
> OK, but "testMP3Streamer" doesn't?
>
>          Ross.
>
>
> _______________________________________________
> Live-devel mailing list
> Live-devel(at)lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/live-devel



From <EMAIL-ADDRESS-HIDDEN> Thu Feb 28 15:44:59 2002
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Date: Thu Feb 28 15:45:04 2002
X-Original-Date: Thu, 28 Feb 2002 18:44:41 -0500
Status: O

Ross,

When I looked at testMP3Streamer.cpp I found there's RTSP stuff in it (#ifdef IMPLEMENT_RTSP_SERVER). However the program doesn't seem to be really using RTSP and it just streams the test file anyway without caring about what's happening with RTSP.

I'm wondering somehow streaming needs to be activated when RTSP command PLAY is being handled; but handleCmd_PLAY() in class RTSPSession does nothing other than sending back a response. And based on the way RTSPServer and RTSPServer::Session are defined, it seems to me I can't override its behaviour. Is there a way around or is it not yet a complete implementation? 

cheers,
Luca

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From <EMAIL-ADDRESS-HIDDEN> Thu Feb 28 15:54:37 2002
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Date: Thu Feb 28 15:55:06 2002
X-Original-Date: Thu, 28 Feb 2002 15:52:49 -0800
Status: O

At 03:34 PM 2/28/02, you wrote:
>I'm running testMP3Streamer only. I did make clean and a fresh build, but
>now I get a different error:
...
>Failed to create RTSP server: unable to create stream socket: Too many open
>files

OK, it's possible that the RTSP server code's socket is not getting closed 
when the server is closed.  I'll take a look at this.  In the meantime, you 
may wish to comment out the line:
         #define IMPLEMENT_RTSP_SERVER 1
once again.

You should also check to see whether any zombie "test*" processes are still 
running (& if so, kill them).  If, after this, you still get the "too many 
open files" error, then there is some brain damage in your OS, and you'll 
need to reboot.


>The first two lines (Beginning streaming...) are repeated 29 times before
>the last error is reported. It seems that the streaming is started several
>times before it crashes. In the old liveMedia, testMP3Streamer prints only
>the first line (Beginning streaming...) and keeps streaming fine.

If "testMP3Streamer" is working correctly, then each "Beginning 
streaming...", "...done streaming" pair should last exactly as long as the 
playing length of the "test.mp3" input file.  If these pairs are printed 
more (or less) frequently than this, then something is going wrong.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Thu Feb 28 16:03:21 2002
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Date: Thu Feb 28 16:04:06 2002
X-Original-Date: Thu, 28 Feb 2002 16:03:19 -0800
Status: O

At 03:44 PM 2/28/02, rdczhuh(at)netscape.net wrote:
>When I looked at testMP3Streamer.cpp I found there's RTSP stuff in it 
>(#ifdef IMPLEMENT_RTSP_SERVER). However the program doesn't seem to be 
>really using RTSP and it just streams the test file anyway without caring 
>about what's happening with RTSP.

That's correct.  The current RTSP server implementation basically just 
implements the "DESCRIBE" command, which allows a RTSP client to obtain the 
SDP description for an *ongoing* session, and tune into it.  (As you noted, 
the implementation of the "SETUP" and "PLAY" commands are basically no-ops.)

The code does not (yet) implement audio/video streaming on demand, in which 
a RTSP client would use the "SETUP" and "PLAY" commands to start playing a 
stream (usually unicast) from the beginning, rather than tuning into an 
existing stream (usually multicast).


>Is there a way around or is it not yet a complete implementation?

The latter - the implementation is still incomplete in this respect.

         Ross.



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Date: Thu Feb 28 16:09:01 2002
X-Original-Date: Thu, 28 Feb 102 18:00:50 +0000
Status: O

Hi Ross,
	All the programs in testprog directory use the "test.mp3" as an input 
file. Do you have sample program which can show the creation of test.mp3? 

Thanks,

Sandeep
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At 10:00 AM 2/28/02, sandeep(at)ccm.net wrote:
>         All the programs in testprog directory use the "test.mp3" as an 
> input
>file.

No, only "testMP3Streamer" uses "test.mp3".  The other test programs use 
"test.mpg".

>  Do you have sample program which can show the creation of test.mp3?

For "test.mp3", any MP3 file should work.

The "test.mpg" file used by "testMPEGAudioVideo" streamer is a MPEG-1 (or 
2) Program Stream file.

         Ross.



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Hi Ross,
    I know I am asking pretty dumb questions but believe me I pick up =
pretty quick.  Now I am trying to compile livemedia library in my Window =
98 computer.

So I successfully compiled groupsock, livemedia, BasicUsageEnvironment =
and UsageEnvirnoment. (Cool)

Now in order to compile testMPEGVideoStream.cpp file, I did the =
following:

1) Opened a new project as console program
2) Added the testMPEGVideoStream.cpp
3) And link the libgroupsock.lib, liblivemedia.lib, =
libBasicUsuageEnvironment, and libUsageEnvironment.lib

I was able to successfully compile the testMPEGVideoStream.cpp file but =
I got the following Link Error:

Deleting intermediate files and output files for project 'Multimedia - =
Win32 Debug'.
--------------------Configuration: Multimedia - Win32 =
Debug--------------------
Compiling...
testMPEGVideoStreamer.cpp
Linking...
MSVCRT.lib(MSVCRT.dll) : error LNK2005: _exit already defined in =
LIBCMTD.lib(crt0dat.obj)
MSVCRT.lib(MSVCRT.dll) : error LNK2005: __errno already defined in =
LIBCMTD.lib(dosmap.obj)
MSVCRT.lib(MSVCRT.dll) : error LNK2005: _sprintf already defined in =
LIBCMTD.lib(sprintf.obj)
MSVCRT.lib(MSVCRT.dll) : error LNK2005: _fprintf already defined in =
LIBCMTD.lib(fprintf.obj)
MSVCRT.lib(MSVCRT.dll) : error LNK2005: _strncmp already defined in =
LIBCMTD.lib(strncmp.obj)
MSVCRT.lib(MSVCRT.dll) : error LNK2005: _free already defined in =
LIBCMTD.lib(dbgheap.obj)
MSVCRT.lib(MSVCRT.dll) : error LNK2005: _memmove already defined in =
LIBCMTD.lib(memmove.obj)
MSVCRT.lib(MSVCRT.dll) : error LNK2005: _fclose already defined in =
LIBCMTD.lib(fclose.obj)
MSVCRT.lib(MSVCRT.dll) : error LNK2005: _fflush already defined in =
LIBCMTD.lib(fflush.obj)
LINK : warning LNK4098: defaultlib "MSVCRT" conflicts with use of other =
libs; use /NODEFAULTLIB:library
testMPEGVideoStreamer.obj : error LNK2001: unresolved external symbol =
_gethostname(at)8
libgroupsock.lib(GroupsockHelper.obj) : error LNK2001: unresolved =
external symbol _gethostname(at)8
libBasicUsageEnvironment.lib(BasicUsageEnvironment.obj) : error LNK2001: =
unresolved external symbol ___WSAFDIsSet(at)8
libgroupsock.lib(GroupsockHelper.obj) : error LNK2001: unresolved =
external symbol ___WSAFDIsSet(at)8
libBasicUsageEnvironment.lib(BasicUsageEnvironment.obj) : error LNK2001: =
unresolved external symbol _WSAGetLastError(at)0
libgroupsock.lib(GroupsockHelper.obj) : error LNK2001: unresolved =
external symbol _WSAGetLastError(at)0
libBasicUsageEnvironment.lib(BasicUsageEnvironment.obj) : error LNK2001: =
unresolved external symbol _select(at)20
libgroupsock.lib(GroupsockHelper.obj) : error LNK2001: unresolved =
external symbol _select(at)20
libgroupsock.lib(NetAddress.obj) : error LNK2001: unresolved external =
symbol _htons(at)4
libgroupsock.lib(NetAddress.obj) : error LNK2001: unresolved external =
symbol _ntohs(at)4
libgroupsock.lib(GroupsockHelper.obj) : error LNK2001: unresolved =
external symbol _ntohs(at)4
libgroupsock.lib(NetAddress.obj) : error LNK2001: unresolved external =
symbol _ntohl(at)4
libgroupsock.lib(GroupsockHelper.obj) : error LNK2001: unresolved =
external symbol _ntohl(at)4
libliveMedia.lib(RTCP.obj) : error LNK2001: unresolved external symbol =
_ntohl(at)4
libliveMedia.lib(MediaSink.obj) : error LNK2001: unresolved external =
symbol _ntohl(at)4
libgroupsock.lib(inet.obj) : error LNK2001: unresolved external symbol =
_inet_addr(at)4
libgroupsock.lib(inet.obj) : error LNK2001: unresolved external symbol =
_inet_ntoa(at)4
libgroupsock.lib(inet.obj) : error LNK2001: unresolved external symbol =
_WSACleanup(at)0
libgroupsock.lib(inet.obj) : error LNK2001: unresolved external symbol =
_WSAStartup(at)8
libgroupsock.lib(inet.obj) : error LNK2001: unresolved external symbol =
_gethostbyname(at)4
libgroupsock.lib(GroupsockHelper.obj) : error LNK2001: unresolved =
external symbol _gethostbyname(at)4
libgroupsock.lib(GroupsockHelper.obj) : error LNK2001: unresolved =
external symbol _bind(at)12
libgroupsock.lib(GroupsockHelper.obj) : error LNK2001: unresolved =
external symbol _closesocket(at)4
libgroupsock.lib(NetInterface.obj) : error LNK2001: unresolved external =
symbol _closesocket(at)4
libgroupsock.lib(GroupsockHelper.obj) : error LNK2001: unresolved =
external symbol _setsockopt(at)20
libgroupsock.lib(GroupsockHelper.obj) : error LNK2001: unresolved =
external symbol _socket(at)12
libgroupsock.lib(GroupsockHelper.obj) : error LNK2001: unresolved =
external symbol _ioctlsocket(at)12
libgroupsock.lib(GroupsockHelper.obj) : error LNK2001: unresolved =
external symbol _recvfrom(at)24
libgroupsock.lib(GroupsockHelper.obj) : error LNK2001: unresolved =
external symbol _sendto(at)24
libgroupsock.lib(GroupsockHelper.obj) : error LNK2001: unresolved =
external symbol _getsockopt(at)20
libgroupsock.lib(GroupsockHelper.obj) : error LNK2001: unresolved =
external symbol _getsockname(at)12
libliveMedia.lib(MediaSink.obj) : error LNK2001: unresolved external =
symbol _htonl(at)4
Debug/Multimedia.exe : fatal error LNK1120: 22 unresolved externals
Error executing link.exe.

Multimedia.exe - 42 error(s), 1 warning(s)


Any ideas?

Sandeep

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
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<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Hi Ross,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp;&nbsp; I know I am asking =
pretty dumb=20
questions but believe me I pick up pretty quick.&nbsp; Now I am trying =
to=20
compile livemedia library in my Window 98 computer.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>So I successfully compiled groupsock, =
livemedia,=20
BasicUsageEnvironment and UsageEnvirnoment. (Cool)</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Now in order to compile =
testMPEGVideoStream.cpp=20
file, I did the following:</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>1) Opened a new project as console=20
program</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>2) Added the =
testMPEGVideoStream.cpp</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>3) And link the libgroupsock.lib, =
liblivemedia.lib,=20
libBasicUsuageEnvironment, and libUsageEnvironment.lib</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>I was able to successfully compile the=20
testMPEGVideoStream.cpp file but I got the following Link =
Error:</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Deleting intermediate files and output =
files for=20
project 'Multimedia - Win32 =
Debug'.<BR>--------------------Configuration:=20
Multimedia - Win32=20
Debug--------------------<BR>Compiling...<BR>testMPEGVideoStreamer.cpp<BR=
>Linking...<BR>MSVCRT.lib(MSVCRT.dll)=20
: error LNK2005: _exit already defined in=20
LIBCMTD.lib(crt0dat.obj)<BR>MSVCRT.lib(MSVCRT.dll) : error LNK2005: =
__errno=20
already defined in LIBCMTD.lib(dosmap.obj)<BR>MSVCRT.lib(MSVCRT.dll) : =
error=20
LNK2005: _sprintf already defined in=20
LIBCMTD.lib(sprintf.obj)<BR>MSVCRT.lib(MSVCRT.dll) : error LNK2005: =
_fprintf=20
already defined in LIBCMTD.lib(fprintf.obj)<BR>MSVCRT.lib(MSVCRT.dll) : =
error=20
LNK2005: _strncmp already defined in=20
LIBCMTD.lib(strncmp.obj)<BR>MSVCRT.lib(MSVCRT.dll) : error LNK2005: =
_free=20
already defined in LIBCMTD.lib(dbgheap.obj)<BR>MSVCRT.lib(MSVCRT.dll) : =
error=20
LNK2005: _memmove already defined in=20
LIBCMTD.lib(memmove.obj)<BR>MSVCRT.lib(MSVCRT.dll) : error LNK2005: =
_fclose=20
already defined in LIBCMTD.lib(fclose.obj)<BR>MSVCRT.lib(MSVCRT.dll) : =
error=20
LNK2005: _fflush already defined in LIBCMTD.lib(fflush.obj)<BR>LINK : =
warning=20
LNK4098: defaultlib "MSVCRT" conflicts with use of other libs; use=20
/NODEFAULTLIB:library<BR>testMPEGVideoStreamer.obj : error LNK2001: =
unresolved=20
external symbol <A=20
href=3D"mailto:_gethostname(at)8">_gethostname(at)8</A><BR>libgroupsock.lib(Gro=
upsockHelper.obj)=20
: error LNK2001: unresolved external symbol <A=20
href=3D"mailto:_gethostname(at)8">_gethostname(at)8</A><BR>libBasicUsageEnviron=
ment.lib(BasicUsageEnvironment.obj)=20
: error LNK2001: unresolved external symbol <A=20
href=3D"mailto:___WSAFDIsSet(at)8">___WSAFDIsSet(at)8</A><BR>libgroupsock.lib(G=
roupsockHelper.obj)=20
: error LNK2001: unresolved external symbol <A=20
href=3D"mailto:___WSAFDIsSet(at)8">___WSAFDIsSet(at)8</A><BR>libBasicUsageEnvir=
onment.lib(BasicUsageEnvironment.obj)=20
: error LNK2001: unresolved external symbol <A=20
href=3D"mailto:_WSAGetLastError(at)0">_WSAGetLastError(at)0</A><BR>libgroupsock=
.lib(GroupsockHelper.obj)=20
: error LNK2001: unresolved external symbol <A=20
href=3D"mailto:_WSAGetLastError(at)0">_WSAGetLastError(at)0</A><BR>libBasicUsag=
eEnvironment.lib(BasicUsageEnvironment.obj)=20
: error LNK2001: unresolved external symbol <A=20
href=3D"mailto:_select(at)20">_select(at)20</A><BR>libgroupsock.lib(GroupsockHe=
lper.obj)=20
: error LNK2001: unresolved external symbol <A=20
href=3D"mailto:_select(at)20">_select(at)20</A><BR>libgroupsock.lib(NetAddress.=
obj) :=20
error LNK2001: unresolved external symbol <A=20
href=3D"mailto:_htons(at)4">_htons(at)4</A><BR>libgroupsock.lib(NetAddress.obj)=
 : error=20
LNK2001: unresolved external symbol <A=20
href=3D"mailto:_ntohs(at)4">_ntohs(at)4</A><BR>libgroupsock.lib(GroupsockHelper=
.obj) :=20
error LNK2001: unresolved external symbol <A=20
href=3D"mailto:_ntohs(at)4">_ntohs(at)4</A><BR>libgroupsock.lib(NetAddress.obj)=
 : error=20
LNK2001: unresolved external symbol <A=20
href=3D"mailto:_ntohl(at)4">_ntohl(at)4</A><BR>libgroupsock.lib(GroupsockHelper=
.obj) :=20
error LNK2001: unresolved external symbol <A=20
href=3D"mailto:_ntohl(at)4">_ntohl(at)4</A><BR>libliveMedia.lib(RTCP.obj) : =
error=20
LNK2001: unresolved external symbol <A=20
href=3D"mailto:_ntohl(at)4">_ntohl(at)4</A><BR>libliveMedia.lib(MediaSink.obj) =
: error=20
LNK2001: unresolved external symbol <A=20
href=3D"mailto:_ntohl(at)4">_ntohl(at)4</A><BR>libgroupsock.lib(inet.obj) : =
error=20
LNK2001: unresolved external symbol <A=20
href=3D"mailto:_inet_addr(at)4">_inet_addr(at)4</A><BR>libgroupsock.lib(inet.ob=
j) :=20
error LNK2001: unresolved external symbol <A=20
href=3D"mailto:_inet_ntoa(at)4">_inet_ntoa(at)4</A><BR>libgroupsock.lib(inet.ob=
j) :=20
error LNK2001: unresolved external symbol <A=20
href=3D"mailto:_WSACleanup(at)0">_WSACleanup(at)0</A><BR>libgroupsock.lib(inet.=
obj) :=20
error LNK2001: unresolved external symbol <A=20
href=3D"mailto:_WSAStartup(at)8">_WSAStartup(at)8</A><BR>libgroupsock.lib(inet.=
obj) :=20
error LNK2001: unresolved external symbol <A=20
href=3D"mailto:_gethostbyname(at)4">_gethostbyname(at)4</A><BR>libgroupsock.lib=
(GroupsockHelper.obj)=20
: error LNK2001: unresolved external symbol <A=20
href=3D"mailto:_gethostbyname(at)4">_gethostbyname(at)4</A><BR>libgroupsock.lib=
(GroupsockHelper.obj)=20
: error LNK2001: unresolved external symbol <A=20
href=3D"mailto:_bind(at)12">_bind(at)12</A><BR>libgroupsock.lib(GroupsockHelper=
.obj) :=20
error LNK2001: unresolved external symbol <A=20
href=3D"mailto:_closesocket(at)4">_closesocket(at)4</A><BR>libgroupsock.lib(Net=
Interface.obj)=20
: error LNK2001: unresolved external symbol <A=20
href=3D"mailto:_closesocket(at)4">_closesocket(at)4</A><BR>libgroupsock.lib(Gro=
upsockHelper.obj)=20
: error LNK2001: unresolved external symbol <A=20
href=3D"mailto:_setsockopt(at)20">_setsockopt(at)20</A><BR>libgroupsock.lib(Gro=
upsockHelper.obj)=20
: error LNK2001: unresolved external symbol <A=20
href=3D"mailto:_socket(at)12">_socket(at)12</A><BR>libgroupsock.lib(GroupsockHe=
lper.obj)=20
: error LNK2001: unresolved external symbol <A=20
href=3D"mailto:_ioctlsocket(at)12">_ioctlsocket(at)12</A><BR>libgroupsock.lib(G=
roupsockHelper.obj)=20
: error LNK2001: unresolved external symbol <A=20
href=3D"mailto:_recvfrom(at)24">_recvfrom(at)24</A><BR>libgroupsock.lib(Groupso=
ckHelper.obj)=20
: error LNK2001: unresolved external symbol <A=20
href=3D"mailto:_sendto(at)24">_sendto(at)24</A><BR>libgroupsock.lib(GroupsockHe=
lper.obj)=20
: error LNK2001: unresolved external symbol <A=20
href=3D"mailto:_getsockopt(at)20">_getsockopt(at)20</A><BR>libgroupsock.lib(Gro=
upsockHelper.obj)=20
: error LNK2001: unresolved external symbol <A=20
href=3D"mailto:_getsockname(at)12">_getsockname(at)12</A><BR>libliveMedia.lib(M=
ediaSink.obj)=20
: error LNK2001: unresolved external symbol <A=20
href=3D"mailto:_htonl(at)4">_htonl(at)4</A><BR>Debug/Multimedia.exe : fatal =
error=20
LNK1120: 22 unresolved externals<BR>Error executing =
link.exe.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Multimedia.exe - 42 error(s), 1=20
warning(s)<BR></FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Any ideas?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Sandeep</DIV></FONT></BODY></HTML>

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Date: Thu Feb 28 21:27:02 2002
X-Original-Date: Thu, 28 Feb 2002 21:22:32 -0800
Status: O

At 09:08 PM 2/28/02, Sandeep Kinra wrote:
>Now in order to compile testMPEGVideoStream.cpp file, I did the following:
>
>1) Opened a new project as console program
>2) Added the testMPEGVideoStream.cpp
>3) And link the libgroupsock.lib, liblivemedia.lib, 
>libBasicUsuageEnvironment, and libUsageEnvironment.lib

Instead of doing this, I suggest using the ".mak" files (Makefiles), as 
described in <http://live.sourceforge.net/#config-windows>.  This way, 
you'll be sure to generate projects that link all of the required libraries.

         Ross.



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Date: Fri Mar  1 00:59:10 2002
X-Original-Date: Fri, 01 Mar 2002 09:58:14 +0100
Status: O

At 21:22 28.02.02 -0800, Ross Finlayson wrote:
>At 09:08 PM 2/28/02, Sandeep Kinra wrote:
>>Now in order to compile testMPEGVideoStream.cpp file, I did the following:
>>
>>1) Opened a new project as console program
>>2) Added the testMPEGVideoStream.cpp
>>3) And link the libgroupsock.lib, liblivemedia.lib, 
>>libBasicUsuageEnvironment, and libUsageEnvironment.lib
>
>Instead of doing this, I suggest using the ".mak" files (Makefiles), as 
>described in <http://live.sourceforge.net/#config-windows>.  This way, 
>you'll be sure to generate projects that link all of the required libraries.
This is working fine as long as you only want to build your test programs. 
The mentioned problem is a general problem when using your lib with windows 
programs (not using your Makefile). It depends on which run-time library 
you linked the live.com library with and which run-time library you suppose 
to use to link your application. A good source for that is to look in the 
MSDN Library under the topic '/MD, /ML, /MT, /LD (Use Run-Time Library)'. 
This is explaining the different run-time libraries.
Although you will get even more problems if you want to use the live libs 
with a WindowsGUI application using MFC and multithreads. In this case you 
have to link the live libs with the correct run-time library.
The second part of the error message, looks like that the application is 
using the wrong library for socket functions. It's necessary to link with 
'Wsock32.lib'.

Michael



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Date: Fri Mar  1 10:38:48 2002
X-Original-Date: Fri, 01 Mar 2002 10:36:23 -0800
Status: O

FYI, I have now installed a new version (2002.03.01) of the source code 
that should fix the problems with "testMP3Streamer" (running out of open 
files) that Hazem encountered.

	Ross.



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Date: Fri Mar  1 10:56:37 2002
X-Original-Date: Fri, 1 Mar 2002 19:51:54 +0100
Status: O

This is a multi-part message in MIME format.

------=_NextPart_000_001E_01C1C15A.89FB26C0
Content-Type: text/plain;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

   hello!

 I have several questions about live streaming media.

- what is the rule of audiosink, videosink, MPEGaudioRTPsink and =
MPEGvideoRTPsink in testMpegAudioVideoStreamer.cpp ?

- Is the server testMpegAudioVideoStreamer.cpp  able to receive the =
Receive Report RTCP from a client application ?

-testMpegAudioVideoStreamer.cpp is able to stream mpeg video format such =
as mpeg 1 and mpeg 2 (other?).  In order to stream this data
=20
  the media is splitted into the audio and video elementary streams. Do =
you know the payload type code in RTP packet related to

  this streams  ?   (video and audio)

- also, do you know the payload type related to quicktime movie stream ? =
 ( I refer to quicktime streaming server )

- to perform a stream of a stereo audio signal do I have to use separate =
RTP sessions for each audio channel ?

 ( this could mean that if there is a multimedia that comprises a stream =
video, we need three RTP sessions)


about sychronisation:

the timestamp NTP is stored in RTCP packets for both video and audio =
sessions. Is this value obtained from the local clock system?


about testRelay.cpp

I use this software to receive rtp packets in unicast mode and than =
forward in multicast.

-like testMpegAudioVideoStreamer.cpp, is the testrelay able to receive =
the RTCP receiver reports from a client ?

-In the configuration I' ve described, do I need an independent software =
to receive the RR report and the return it to the sender ?


I've made so many questions. I hope someone has the goodness to send me =
some replies.
thank you for all.   :-)

Domenico=20

=20






------=_NextPart_000_001E_01C1C15A.89FB26C0
Content-Type: text/html;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content=3D"text/html; charset=3Diso-8859-1" =
http-equiv=3DContent-Type>
<META content=3D"MSHTML 5.00.2920.0" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp; hello!</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;I have several questions about =
live streaming=20
media.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>- what is the rule of audiosink, =
videosink,=20
MPEGaudioRTPsink and MPEGvideoRTPsink in testMpegAudioVideoStreamer.cpp=20
?</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>- Is the server=20
testMpegAudioVideoStreamer.cpp&nbsp; able to receive the Receive Report =
RTCP=20
from a client application ?</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>-testMpegAudioVideoStreamer.cpp&nbsp;is =
able to=20
stream mpeg video format such as mpeg 1 and mpeg 2 (other?).&nbsp; In =
order to=20
stream this data</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp; the media is splitted into the =
audio and=20
video elementary streams. Do you know the payload type code in RTP =
packet=20
related to</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp; this streams&nbsp; ?&nbsp;&nbsp; =
(video and=20
audio)</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>- also, do you know the payload type =
related to=20
quicktime movie stream ?&nbsp; ( I refer to quicktime streaming server=20
)</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>- to perform a stream of a stereo audio =
signal do I=20
have to use separate RTP sessions for each audio channel ?</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;(&nbsp;this could mean that if =
there is a=20
multimedia that comprises a stream video,&nbsp;we need three RTP=20
sessions)</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>about sychronisation:</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>the timestamp NTP is stored in RTCP =
packets for=20
both video and audio sessions. Is this value obtained from the local =
clock=20
system?</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>about testRelay.cpp</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>I use this software to receive rtp =
packets in=20
unicast mode and than forward in multicast.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>-like testMpegAudioVideoStreamer.cpp, =
is the=20
testrelay able to receive the RTCP receiver reports from a client =
?</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>-In the configuration I' ve described, =
do I need an=20
independent software to receive the RR report and the return it to the =
sender=20
?</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>I've made so many questions. I hope =
someone has the=20
goodness to send me some replies.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>thank you for all.&nbsp;  =
:-)</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Domenico </FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV></BODY></HTML>

------=_NextPart_000_001E_01C1C15A.89FB26C0--



From <EMAIL-ADDRESS-HIDDEN> Fri Mar 01 13:27:50 2002
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Date: Fri Mar  1 13:34:41 2002
X-Original-Date: Fri, 01 Mar 2002 13:22:46 -0800
Status: O

At 12:58 AM 3/1/02, Michael Sommer wrote:
>>>1) Opened a new project as console program
>>>2) Added the testMPEGVideoStream.cpp
>>>3) And link the libgroupsock.lib, liblivemedia.lib, 
>>>libBasicUsuageEnvironment, and libUsageEnvironment.lib
>>
>>Instead of doing this, I suggest using the ".mak" files (Makefiles), as 
>>described in <http://live.sourceforge.net/#config-windows>.  This way, 
>>you'll be sure to generate projects that link all of the required libraries.
>This is working fine as long as you only want to build your test programs. 
>The mentioned problem is a general problem when using your lib with 
>windows programs (not using your Makefile). It depends on which run-time 
>library you linked the live.com library with and which run-time library 
>you suppose to use to link your application.

Yes, that's correct.  However, you may still find it useful to use the test 
programs as guidelines for figuring out how to create new VC++ 'projects' 
(for other programs).  I.e., when you use VC++ to open a workspace from a 
test program's '.mak' file, you will end up creating a proper VC++ 
'project', which you can then examine to see what libraries (etc.) are used 
to build the project.  You should then be able to create new projects, 
using it as a model.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Fri Mar 01 15:18:46 2002
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Date: Fri Mar  1 15:23:32 2002
X-Original-Date: Fri, 01 Mar 2002 15:15:42 -0800
Status: O

Domenico,

Excellent questions.


>  - what is the rule of audiosink, videosink, MPEGaudioRTPsink and 
> MPEGvideoRTPsink in testMpegAudioVideoStreamer.cpp ?

In general, the role of a 'sink' object (an instance of a subclass of 
"MediaSink") is to act as the destination for a data stream.  Each such 
object is 'driven' by calling the "startPlaying()" member function, taking 
a corresponding 'source' object (a subclass of "MediaSource") as parameter.

In particular, for a network streaming server (like 
"testMPEGAudioVideoStreamer"), each 'sink' object ("audioSink" and 
"videoSink") represents a network destination (that uses RTP), and the 
corresponding 'source' object ("audioSource" or "videoSource") represents 
the (demultiplexed) input data that is to be streamed.

(Note that, in "testMPEGAudioVideoStreamer", "audioSink" is an instance of 
the class "MPEGAudioRTPSink", and "videoSink" is an instance of the class 
"MPEGVideoRTPSink".  Both of these classes are defined in the"liveMedia" 
library.)


>  - Is the server testMpegAudioVideoStreamer.cpp  able to receive the 
> Receive Report RTCP from a client application ?

Yes, each "RTCPInstance" object will receive each RTCP Reception Report 
(for its RTP session) that is sent by clients.  At present, however, the 
implementation does not do much with these reports, other than recording 
which clients (based on SSRC) have sent such reports, and when.  However, 
you can call the "numMembers()" member function (for each "RTCPInstance") 
to find out how many clients are currently active.


>  -testMpegAudioVideoStreamer.cpp is able to stream mpeg video format such 
> as mpeg 1 and mpeg 2 (other?).

At present, just MPEG-1 or MPEG-2.  (MPEG-4 is on my 'to do' list, but will 
require the use of a new RTP payload format - incompatible with that used 
by MPEG-1 or 2.)

>   In order to stream this data
>
>   the media is splitted into the audio and video elementary streams. Do 
> you know the payload type code in RTP packet related to
>
>   this streams  ?   (video and audio)

Yes, the audio stream uses the RTP static payload type 14; the video stream 
uses the RTP static payload type 32.  (To see this, note the 
implementations of "MPEGAudioRTPSink" and "MPEGVideoRTPSink".)  FYI, these 
payload types are defined in 
<http://www.ietf.org/internet-drafts/draft-ietf-avt-profile-new-12.txt>


>  - also, do you know the payload type related to quicktime movie stream 
> ?  ( I refer to quicktime streaming server )

"QuickTime Streaming Server" streams using standard RTP/RTCP, so it will 
use whatever RTP payload types are appropriate for the particular codecs 
that it is streaming.  (This is why the "QuickTime Player" can be used to 
view the audio/video streams sent by "testMPEGAudioVideoStreamer" - they 
look identical to streams that would be sent by a "QuickTime Streaming 
Server".)

(When streaming a codec - such as Sorenson Video - for which an IETF 
standard RTP payload format is not defined, "QuickTime Streaming Server" 
may instead use an Apple-specific RTP payload format, as defined in 
<http://developer.apple.com/quicktime/icefloe/dispatch026.html>.)


>  - to perform a stream of a stereo audio signal do I have to use separate 
> RTP sessions for each audio channel ?

In general, no.  For example, in MPEG audio, the stereo channel separation 
info is embedded within the (single) audio stream, which is carried in a 
single RTP session.

However, separate RTP sessions would probably be used to carry separate 
audio tracks (e.g., recorded in different languages).


>about sychronisation:
>
>the timestamp NTP is stored in RTCP packets for both video and audio 
>sessions. Is this value obtained from the local clock system?

Yes.  See RTCP.cpp, line 476


>  about testRelay.cpp
>
>I use this software to receive rtp packets in unicast mode and than 
>forward in multicast.
>
>-like testMpegAudioVideoStreamer.cpp, is the testrelay able to receive the 
>RTCP receiver reports from a client ?

No - "testRelay", as currently written, relays a single stream, and in only 
one direction.

>  -In the configuration I' ve described, do I need an independent software 
> to receive the RR report and the return it to the sender ?

Yes, that would work.  Don't forget that RTCP uses a different port number 
than RTP (the RTP port number, plus 1).

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>



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Date: Sun Mar  3 01:12:02 2002
X-Original-Date: Sun, 3 Mar 2002 09:51:27 +0100
Status: O

This is a multi-part message in MIME format.

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   Hello Ross,=20

thanks for the really clear answers to my questions.


about testrelay I'm a bit confused:

testrelay is able to read input UDP packets from an IP address + port =
number,=20

(independently from the payload carried) and than forward them to=20

a different output IP address + port


I've a doubt: according to the experiments I've done,
I've always used two testrelay to forward two RTP sessions=20
related to audio and video streams.  This streams are well synchronised
on the player side.=20
So I deduced that each testrelay is able to read from a specified IP =
address, Port
and at the same time reads form the odd port, Port+1 for RTCP packets.

When I gave a look at the source code, I was not able to see this =
capacity
to read also from the odd port, so I got quite confused.

Can you tell me the right behaviour of testrelay ?


I've a last question:

testrealy forward UDP packets making no changes to the payload.
This means that in the case of RTP, SSRC field is unchanged.

On the client side, are the RR RTCP packets  sent to the relaying host
or to the source one?

How can I detect the port on the relaying host or source host that =
receives the RR RTCP packets?

Finally, if I relays in a configuration like unicast input, multicast =
output, on the same relaying host,
can I use another testrelay to receive the RR packets form multicast and =
than return to the sender?

wich address and port for the input and output must I specify for this =
"feedback relay" ?

thanks for the patience,

I'll be obliged for all.

Domenico :)






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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content=3D"text/html; charset=3Diso-8859-1" =
http-equiv=3DContent-Type>
<META content=3D"MSHTML 5.00.2920.0" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp; Hello Ross, </FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>thanks for the really clear answers to =
my=20
questions.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>about testrelay I'm&nbsp;a bit=20
confused:</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>testrelay is able to read input UDP =
packets from an=20
IP address + port number, </FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>(independently from the payload =
carried)=20
</FONT><FONT face=3DArial size=3D2>and than forward them =
to&nbsp;</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>a different&nbsp;output IP address +=20
port</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>I've a doubt: according to the =
experiments I've=20
done,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>I've always used two testrelay to =
forward two RTP=20
sessions </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>related to audio and video =
streams.&nbsp; This=20
streams are well synchronised</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>on the player side. </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>So I deduced that each testrelay is =
able to read=20
from a specified IP address, Port</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>and at the same time reads form the odd =
port,=20
Port+1 for RTCP packets.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>When I gave a look at the source code, =
I was not=20
able to see this capacity</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>to read also from the odd port, so I =
got quite=20
confused.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Can you tell me the right behaviour of =
testrelay=20
?</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>I've a last question:</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>testrealy forward UDP packets making no =

changes&nbsp;to the payload.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>This means that in the case of RTP, =
SSRC field is=20
unchanged.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>On the client side, are the RR RTCP =
packets&nbsp;=20
sent to the relaying host</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>or to the source one?</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>How can I detect the port on the =
relaying host or=20
source host that receives the RR RTCP packets?</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Finally, if I relays in a configuration =
like=20
unicast input, multicast output, on the same relaying host,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>can I use&nbsp;another testrelay to =
receive the RR=20
packets form multicast and than return to the sender?</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>wich address and port for the input and =
output must=20
I specify for this "feedback relay" ?</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>thanks for the patience,</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>I'll be obliged for all.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Domenico :)</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV></BODY></HTML>

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From <EMAIL-ADDRESS-HIDDEN> Sun Mar 03 11:17:33 2002
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From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
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Date: Sun Mar  3 11:18:19 2002
X-Original-Date: Sun, 03 Mar 2002 11:13:40 -0800
Status: O

At 12:51 AM 3/3/02, mimmo wrote:
>about testrelay I'm a bit confused:
>
>testrelay is able to read input UDP packets from an IP address + port number,
>  (independently from the payload carried) and than forward them to
>  a different output IP address + port

That's correct - it's just a simple UDP relay.  (Note that if you look at 
"testRelay.cpp", you won't see any mention of "RTP" or "RTCP".)


>I've a doubt: according to the experiments I've done,
>I've always used two testrelay to forward two RTP sessions
>related to audio and video streams.  This streams are well synchronised
>on the player side.

This is probably just a coincidence.  If the streams start out 
synchronized, and are sent only over a LAN, and the computer running the 
"testRelay" programs is not overloaded, then the streams will probably stay 
more or less synchronized when they reach the player.


>So I deduced that each testrelay is able to read from a specified IP 
>address, Port
>and at the same time reads form the odd port, Port+1 for RTCP packets.

No, the RTCP port won't be forwarded unless you explicitly ask for this 
(e.g., by running another copy of "testRelay" to forward that port).


>testrealy forward UDP packets making no changes to the payload.
>This means that in the case of RTP, SSRC field is unchanged.

Correct.


>  On the client side, are the RR RTCP packets  sent to the relaying host
>or to the source one?

If the client (player?) is receiving the RTP packets on a multicast 
address, then it will send its RTCP RR packets to the same multicast 
address (but with port number +1).

If, however, the client is receiving unicast packets, then it's less clear 
where RTCP packets are sent - this can depend on the particular client 
application.  If a unicast address is specified (e.g., in a SDP file), then 
RTCP RRs might get sent to that address.  Or it's possible that no RTCP RR 
packets will get sent at all.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Sun Mar 03 19:11:46 2002
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Date: Sun Mar  3 19:12:04 2002
X-Original-Date: Sun, 03 Mar 2002 22:11:27 -0500
Status: O

Hi,

This confused me when I tried to run testMP3Streamer/Receiver for unicast. If the streamer starts first, all goes well. However if the receiver starts before streamer does, then the receiver doesn't receive anything. (This happens with unicast only, for multicast it works fine for me.)

Any idea?

Thanks and best regards,
luca
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Subject: [Live-devel] some question about testRelay.cpp
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Date: Mon Mar  4 06:16:09 2002
X-Original-Date: Mon, 4 Mar 2002 15:15:32 +0100 (CET)
Status: O

Hello
 I want to know more about the call-function
"taskScheduler().turnOnBackgroundReadHandling(inputGroupsock.socketNum(),
						    handleRead, &inputGroupsock);" in
"testRelay.cpp". I know the main scope of this test
program (tesRelay) but I can't identify it through the
function indicated above.

Thanks Achraf    

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Subject: [Live-devel] some question about testRelay.cpp
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Date: Mon Mar  4 06:18:07 2002
X-Original-Date: Mon, 4 Mar 2002 15:17:18 +0100 (CET)
Status: O

Hello
 I want to know more about the call-function
"taskScheduler().turnOnBackgroundReadHandling(inputGroupsock.socketNum(),
						    handleRead, &inputGroupsock);" in
"testRelay.cpp". I know the main scope of this test
program (tesRelay) but I can't identify it through the
function indicated above.

Thanks Achraf    

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From <EMAIL-ADDRESS-HIDDEN> Mon Mar 04 11:34:22 2002
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From: <EMAIL-ADDRESS-HIDDEN> (luca)
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Subject: RE: [Live-devel] some question about testRelay.cpp
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Date: Mon Mar  4 11:35:11 2002
X-Original-Date: Mon, 04 Mar 2002 14:34:07 -0500
Status: O

Achraf,

As far as what I know, turnOnBackgroundReadHandling() is used for non-blocking i/o. It adds the given socket to the monitored set of reading descriptors, so later on select() can be used to test if there is any data coming in on that socket. Not sure if that is your question though.

/luca


"The number of Unix installation has grown to 10, with more expected."
-- The Unix Programmer's Manual, 2nd Edition, June 1972

  <achraf_g(at)yahoo.fr> wrote:

>Hello
> I want to know more about the call-function
>"taskScheduler().turnOnBackgroundReadHandling(inputGroupsock.socketNum(),
>                            handleRead, &inputGroupsock);" in
>"testRelay.cpp". I know the main scope of this test
>program (tesRelay) but I can't identify it through the
>function indicated above.
>
>Thanks Achraf    
>
>___________________________________________________________
>Do You Yahoo!? -- Une adresse (at)yahoo.fr gratuite et en français !
>Yahoo! Mail : http://fr.mail.yahoo.com
>
>_______________________________________________
>Live-devel mailing list
>Live-devel(at)lists.sourceforge.net
>https://lists.sourceforge.net/lists/listinfo/live-devel
>
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From <EMAIL-ADDRESS-HIDDEN> Mon Mar 04 15:00:27 2002
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Subject: RE: [Live-devel] some question about testRelay.cpp
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Date: Mon Mar  4 15:01:02 2002
X-Original-Date: Mon, 04 Mar 2002 14:56:15 -0800
Status: O

At 11:34 AM 3/4/02, luca wrote:
>As far as what I know, turnOnBackgroundReadHandling() is used for 
>non-blocking i/o.

That's correct.  This function (which is a member function of the 
"TaskScheduler" base class defined in 
"UsageEnvironment/include/UsageEnvironment.hh") allows the called to 
specify a 'handler' routine to be called whenever incoming network data 
arrives on a 'groupsock'.

>It adds the given socket to the monitored set of reading descriptors, so 
>later on select() can be used to test if there is any data coming in on 
>that socket.

True, but note that this is just the particular implementation used by the 
"BasicTaskScheduler" (defined in the "BasicUsageEnvironment" library) - 
which is just one possible implementation of "TaskScheduler".

         Ross.



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Subject: [Live-devel] Online directory of products using these libraries
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Date: Mon Mar  4 20:32:11 2002
X-Original-Date: Mon, 04 Mar 2002 20:31:20 -0800
Status: O

Now that several companies have begun using the "LIVE.COM Streaming Media" 
libraries in actual products, I'm planning to create (probably on the 
"live.sourceforge.net" web page) a table that links to some of these products.

If you have developed - or are currently developing - a product that uses 
these libraries, and would like to see a link added to your product's (or 
company's) web site, then please reply to this message, letting me know:
	- The URL that you'd like me to add to the table
	- A brief description of your product
	- (in general terms) What parts of the libraries the product uses

Think of this as the opportunity for some free advertising for your product 
and/or company.

	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



From <EMAIL-ADDRESS-HIDDEN> Tue Mar 05 01:01:27 2002
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Subject: [Live-devel] problem with "testMPEGVideoStreamer"!! + RTSP & interactivity
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Date: Tue Mar  5 01:02:08 2002
X-Original-Date: Tue, 5 Mar 2002 10:01:25 +0100 (CET)
Status: O

hello 
 When I try the test program "testMPEGVideoStreamer"
using an elementary MPEG stream I get no error. the
problem is faced with MPEG Program Stream, although I
uncomment the ligne "#define SOURCE_IS_PROGRAM_STREAM
1".
 last question:
I want to know if the rtsp client/server can handle
user interactivity such as (fast forwarding,
pause/resume, jump forward/backward...etc)?
 Any Idea
Thanks Achraf 

___________________________________________________________
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Yahoo! Mail : http://fr.mail.yahoo.com


From <EMAIL-ADDRESS-HIDDEN> Tue Mar 05 02:09:08 2002
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Date: Tue Mar  5 02:10:02 2002
X-Original-Date: Tue, 05 Mar 2002 02:04:33 -0800
Status: O

At 01:01 AM 3/5/02, achraf_g(at)yahoo.fr wrote:
>hello
>  When I try the test program "testMPEGVideoStreamer"
>using an elementary MPEG stream I get no error. the
>problem is faced with MPEG Program Stream, although I
>uncomment the ligne "#define SOURCE_IS_PROGRAM_STREAM 1".

What problem do you see??


>  last question:
>I want to know if the rtsp client/server can handle
>user interactivity such as (fast forwarding,
>pause/resume, jump forward/backward...etc)?

No, not yet.  I plan to support these features sometime in the future.  (As 
always, I can give this higher priority if someone were to contribute funding.)

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>




From <EMAIL-ADDRESS-HIDDEN> Tue Mar 05 02:25:09 2002
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Subject: [Live-devel] I have resolved the link problem in MFC App (VC++)
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Date: Tue Mar  5 02:26:07 2002
X-Original-Date: Tue, 5 Mar 2002 11:25:06 +0100 (CET)
Status: O

Hello
 To avercome linking problem when using MFC
Application
#include "liveMedia.hh"
#include "BasicUsageEnvironment.hh"
#include "GroupsockHelper.hh"
must be added on "stdfx.h"
on the top.
but I get new error wich is
"fatal error C1189: #error :  WINDOWS.H already
included.  MFC apps must not #include <windows.h>"
I want to know if it will be necessary to delete
#include <windows.h> in "FramedSource.hh".?
I have changed the place of the includes statements
after line "#include <afxwin.h> " in "stdfx.h" but I
get new error "error C2079: 'fPresentationTime' uses
undefined struct 'timeval'". Is 'timeval' defined in
"windows.h" ???
Thanks
Achraf





___________________________________________________________
Do You Yahoo!? -- Une adresse (at)yahoo.fr gratuite et en français !
Yahoo! Mail : http://fr.mail.yahoo.com


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Date: Tue Mar  5 10:25:07 2002
X-Original-Date: Tue, 05 Mar 2002 13:23:52 -0500
Status: O

Hi,

I found that testMP3Receiver receives nothing if it is started before testMP3Streamer kicks off. (ie, the receiver can receive an ongoing stream but is not able to wait there for a stream to come). This problem occurs after I modified both testMP3Streamer and testMP3Receiver to make them unicast.

I'll appreciate it if anyone can give me a hint.

Thanks,
luca

-- 




__________________________________________________________________
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From <EMAIL-ADDRESS-HIDDEN> Tue Mar 05 13:05:00 2002
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Date: Tue Mar  5 13:21:41 2002
X-Original-Date: Tue, 05 Mar 2002 12:59:55 -0800
Status: O

At 10:23 AM 3/5/02, you wrote:
>I found that testMP3Receiver receives nothing if it is started before 
>testMP3Streamer kicks off. (ie, the receiver can receive an ongoing stream 
>but is not able to wait there for a stream to come). This problem occurs 
>after I modified both testMP3Streamer and testMP3Receiver to make them unicast.

Luca,

Thanks again for the report.  I plan to investigate this myself when I get 
time.

In your copy of "testMP3Receiver", what IP address are you using (i.e. for 
"multicastAddressStr")?  Are you using the IP address of the sender, the IP 
address of the receiver, or something else (like 0.0.0.0)?

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Tue Mar 05 20:18:33 2002
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Date: Tue Mar  5 20:19:02 2002
X-Original-Date: Tue, 05 Mar 2002 23:18:22 -0500
Status: O

Hi Ross,

>In your copy of "testMP3Receiver", what IP address are you using (i.e. for 
>"multicastAddressStr")?  Are you using the IP address of the sender, the IP 
>address of the receiver, or something else (like 0.0.0.0)?

Streamer and receiver were running on the same machine. I used "0.0.0.0" with Receiver, my own IP and "127.0.0.1" with Streamer. Maybe I should try test them on different boxes.

Regards,
luca



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__________________________________________________________________
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From <EMAIL-ADDRESS-HIDDEN> Wed Mar 06 07:49:33 2002
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Subject: [Live-devel] about the use of unicast!!!??
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Date: Wed Mar  6 07:50:03 2002
X-Original-Date: Wed, 6 Mar 2002 16:48:33 +0100 (CET)
Status: O

 hello,
In order to handle user interactivity I want to open
at the same time unicast and multicast streams. 
the idea is to split the user who initiate
interactivity from the cuurrent session and second we
must open a unicast stream to him. 
I begin by changing "testMP3Streamer.cpp". I have
replaced the multicast adress "239.255.42.42" by the
unicast one ( the adress of the client machine". but
the client player can't play the audio stream.
what changes are to add ?
Thanks
Achraf

  
 

___________________________________________________________
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From <EMAIL-ADDRESS-HIDDEN> Wed Mar 06 13:21:10 2002
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Date: Wed Mar  6 13:24:38 2002
X-Original-Date: Wed, 06 Mar 2002 13:20:59 -0800
Status: O

At 07:48 AM 3/6/02, achraf_g(at)yahoo.fr wrote:
>I begin by changing "testMP3Streamer.cpp". I have
>replaced the multicast adress "239.255.42.42" by the
>unicast one ( the adress of the client machine". but
>the client player can't play the audio stream.

What client player tool are you using?  And what IP address are you giving 
*it*?

(Another list member,  Luca, has also reported problems with unicast 
streaming with "testMP3Streamer", but I have yet to check this out.)

         Ross.



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Date: Thu Mar  7 03:35:04 2002
X-Original-Date: Thu, 7 Mar 2002 12:34:15 +0100 (CET)
Status: O

Hello
 I still have link problem when I try to use live.com
libraries in my "personal" project. after adding
libraries to the project and rebuilding the project, I
get error messages such as:
"Linking...
MSVCRT.lib(MSVCRT.dll) : error LNK2005: _exit already
defined in LIBCD.lib(crt0dat.obj)
MSVCRT.lib(MSVCRT.dll) : error LNK2005: _sprintf
already defined in LIBCD.lib(sprintf.obj)"

and other kind of error :

"testMP3Streamer.obj : error LNK2001: unresolved
external symbol _gethostname(at)8
libgroupsock.lib(GroupsockHelper.obj) : error LNK2001:
unresolved external symbol _gethostname(at)8
libBasicUsageEnvironment.lib(BasicUsageEnvironment.obj)
: error LNK2001: unresolved external symbol
___WSAFDIsSet(at)8
libgroupsock.lib(GroupsockHelper.obj) : error LNK2001:
unresolved external symbol ___WSAFDIsSet(at)8
"

any Idea 
Thanks
Achraf

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Date: Thu Mar  7 04:22:06 2002
X-Original-Date: Thu, 07 Mar 2002 13:20:50 +0100
Status: O

Hello Achraf,

I guess your problem is that you are trying to use different run-time 
library versions. But you can't mix them up in the same application. It 
seems to me that the live library was linked with LIBCD.lib (static library 
for debug single-threaded applications) and you are trying to link your 
actual application with MSVCRT.lib (dynamic library for multi-threaded 
applications). You can solve this if you specify the same run-time library 
for both projects (live-lib and you application). Either you link the 
live-lib with MSVCRT.lib or your application with LIBCD.lib. If you are 
using MFC-DLL in your application you have to take the first alternativ. A 
good source for this topic is to look in the MSDN Library under the topic 
'/MD, /ML, /MT, /LD (Use Run-Time Library)'. This is explaining the 
different run-time libraries.

At 12:34 07.03.02 +0100, achraf_g(at)yahoo.fr wrote:
>Hello
>  I still have link problem when I try to use live.com
>libraries in my "personal" project. after adding
>libraries to the project and rebuilding the project, I
>get error messages such as:
>"Linking...
>MSVCRT.lib(MSVCRT.dll) : error LNK2005: _exit already
>defined in LIBCD.lib(crt0dat.obj)
>MSVCRT.lib(MSVCRT.dll) : error LNK2005: _sprintf
>already defined in LIBCD.lib(sprintf.obj)"
>
>and other kind of error :
I'm not sure with this error, but it seems that you are using the wrong 
library for socket functions. It's necessary to link with 'Wsock32.lib'.

>"testMP3Streamer.obj : error LNK2001: unresolved
>external symbol _gethostname(at)8
>libgroupsock.lib(GroupsockHelper.obj) : error LNK2001:
>unresolved external symbol _gethostname(at)8
>libBasicUsageEnvironment.lib(BasicUsageEnvironment.obj)
>: error LNK2001: unresolved external symbol
>___WSAFDIsSet(at)8
>libgroupsock.lib(GroupsockHelper.obj) : error LNK2001:
>unresolved external symbol ___WSAFDIsSet(at)8
>"

Hope I could help.
Michael




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Date: Fri Mar  8 00:19:08 2002
X-Original-Date: Fri, 8 Mar 2002 09:18:13 +0100 (CET)
Status: O

Hello
 first I want to thank you, Michael for the help.
Although I still Have problem on how to link live-lib
with MSVCRT.lib.
 Second, and about the use of the unicast in
"testMP3Streamer". I use RealPlayer on the client
side. Our LAN IP address are formed as follow :
172.16.x.x
(My server machine has the IP address 172.16.1.113 and
the client one has the 172.16.1.112 address.")
So what changes are to do in "testMP3Streamer.cpp" as
information indicated above.
Thanks  
   Achraf   

___________________________________________________________
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Date: Fri Mar  8 03:06:12 2002
X-Original-Date: Fri, 08 Mar 2002 12:05:17 +0100
Status: O

Hello Achraf,

Here is some more detailed help for you. You have to configure your=20
projects in the following way. Since I'm using a german version of VisualC=
=20
I only can guess the names of the english menu entries, thus I won't use=20
the correct names.
In 'Project Settings' set in 'General' 'Use MFC in shared DLL'. At tab=20
'C/C++' under 'Code Generation' under 'Runtime-Library' set 'Multithreaded=
=20
DLL'.
In your application setting add under tab 'Linker'=20
'Object-/Library-Modules' the file 'Wsock32.lib' and all the live.com=20
libraries.
I know this sounded horrible for people using makefiles but this is the way=
=20
it works with VisualC++ IDE.
If this still doesn't push you further, don't hesitate to ask the mailing=20
list again.

Michael

At 09:18 08.03.02 +0100, achraf_g(at)yahoo.fr wrote:
>Hello
>  first I want to thank you, Michael for the help.
>Although I still Have problem on how to link live-lib
>with MSVCRT.lib.
>  Second, and about the use of the unicast in
>"testMP3Streamer". I use RealPlayer on the client
>side. Our LAN IP address are formed as follow :
>172.16.x.x
>(My server machine has the IP address 172.16.1.113 and
>the client one has the 172.16.1.112 address.")
>So what changes are to do in "testMP3Streamer.cpp" as
>information indicated above.
>Thanks
>    Achraf
>
>___________________________________________________________
>Do You Yahoo!? -- Une adresse (at)yahoo.fr gratuite et en fran=E7ais !
>Yahoo! Mail : http://fr.mail.yahoo.com
>
>_______________________________________________
>Live-devel mailing list
>Live-devel(at)lists.sourceforge.net
>https://lists.sourceforge.net/lists/listinfo/live-devel

Dipl.-Inf. (FH)
Michael Sommer - Software Development
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Subject: [Live-devel] compilation error with VC++ 6.0
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Date: Fri Mar  8 06:45:22 2002
X-Original-Date: Fri, 8 Mar 2002 15:44:45 +0100 (CET)
Status: O

hello
Thank you Michael, I make change on VC++ and this work
well. The new problem now is a compilation one as I
get this message :" error C2664: 'startPlaying' :
cannot convert parameter 2 from 'void (void *)' to
'void (__cdecl *)(void *)'
        None of the functions with this name in scope
match the target type"
Any idea
thanks
Achraf
  

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Date: Fri Mar  8 07:10:13 2002
X-Original-Date: Fri, 08 Mar 2002 16:09:07 +0100
Status: O

Achraf,

I'm not sure with this error. Could you please specify the file where this=
=20
error happend.
Do you use any C file in your project (instead of CPP files)?
The second parameter of this function (class MediaSink) is the afterPlaying=
=20
function. Did you define this function like in the original testMP3Streamer?
If this error occured in your application try (for first) to set the input=
=20
parameters (afterPlaying function and data) of startPlaying to NULL. If you=
=20
doing so you stop with playing after running the MP3 file once.
Michael

At 15:44 08.03.02 +0100, achraf_g(at)yahoo.fr wrote:
>hello
>Thank you Michael, I make change on VC++ and this work
>well. The new problem now is a compilation one as I
>get this message :" error C2664: 'startPlaying' :
>cannot convert parameter 2 from 'void (void *)' to
>'void (__cdecl *)(void *)'
>         None of the functions with this name in scope
>match the target type"
>Any idea
>thanks
>Achraf
>
>
>___________________________________________________________
>Do You Yahoo!? -- Une adresse (at)yahoo.fr gratuite et en fran=E7ais !
>Yahoo! Mail : http://fr.mail.yahoo.com
>
>_______________________________________________
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>Live-devel(at)lists.sourceforge.net
>https://lists.sourceforge.net/lists/listinfo/live-devel



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Date: Fri Mar  8 11:16:07 2002
X-Original-Date: Fri, 08 Mar 2002 11:10:38 -0800
Status: O

At 12:18 AM 3/8/02, achraf_g(at)yahoo.fr wrote:
>  Second, and about the use of the unicast in
>"testMP3Streamer". I use RealPlayer on the client
>side. Our LAN IP address are formed as follow :
>172.16.x.x
>(My server machine has the IP address 172.16.1.113 and
>the client one has the 172.16.1.112 address.")
>So what changes are to do in "testMP3Streamer.cpp" as
>information indicated above.

In  "testMP3Streamer.cpp", change "multicastAddressStr" to "172.16.1.112".

Also, in the file "testMP3.sdp", change the "c=" line to:
         c=IN IP4 172.16.1.113/127

*However*, it's possible that the problem is the fault of RealPlayer - it 
sometimes has problems playing unicast RTP streams.

Instead, I suggest that you try running "Winamp" <http://www.winamp.com/>, 
using the "in_rtp.dll" input plugin 
<http://www.live.com/multikit/winamp-plugin.html>.  If you do this, then 
you should be able to receive the MP3/RTP stream either by reading the 
"testMP3.sdp" file (modified as noted above), or by simply opening the 
'pseudo-URL'
         rtp://172.16.1.113:8888

I hope this helps.

         Ross.



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Date: Mon Mar 11 01:41:02 2002
X-Original-Date: Mon, 11 Mar 2002 10:40:04 +0100 (CET)
Status: O

Hello,
 when I have made changes indicated by Ross about
unicast, the tesMP3Streamer work well.
 the problem faced now is when traying to execute "my
personnel appliaction" wich use live-lib. Thank to
Michael  build-link problems are avercomed, but I get
this message when I debug the application :"Invalid
Address specified to RtlValidateHeap( 350000, 2f3c40
)"
although I have copied the "testMP3Streamer.cpp" as it
is, (in my application)
Any Idea
Thanks 
Achraf 

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Subject: [Live-devel] bit rate and testMPEGAudioVideoStreamer
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Date: Mon Mar 11 16:56:03 2002
X-Original-Date: Tue, 12 Mar 2002 00:55:31
Status: O

Hi,
i'm looking at the live library and the test program 
"testMPEGAudioVideoStreamer". In there, it looks like the bit rate for video 
to send at is hardcoded to 4.5M/s. (line#85).  What if the mpeg2 file was 
encoded at a different bit rate?  Is it possible to read the encoding rate 
from the file itself?  I see that this is being passed to RTCPInstance so is 
it just for reporting purposes?

Is my understanding correct?

THank you.



Ramana Murthy Suryam



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From <EMAIL-ADDRESS-HIDDEN> Mon Mar 11 17:36:49 2002
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Date: Mon Mar 11 17:37:06 2002
X-Original-Date: Mon, 11 Mar 2002 17:36:32 -0800
Status: O

At 04:55 PM 3/11/02, ramana murthy wrote:
>  I see that this is being passed to RTCPInstance so is it just for 
> reporting purposes?

That's correct - this number (4.5 Mbps in this case) is just an estimate of 
the session bandwidth, and is used only for RTCP (to help it compute the 
average time between RTCP reports).

The actual media output rate (for RTP packets) is computed from the MPEG 
data source itself, so it should always be correct.

         Ross.



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Date: Mon Mar 11 19:15:02 2002
X-Original-Date: Tue, 12 Mar 2002 03:13:46
Status: O

Hi,
thanks for your quick response to my last posting.

One other question i have, is there a library function call that, given a 
filename will tell me if its content is a program or elementry stream?  I 
noticed that all the test programs knew before hand what type of file they 
were dealing with.

Thank you.

Ramana Murthy Suryam



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From <EMAIL-ADDRESS-HIDDEN> Tue Mar 12 02:47:21 2002
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Date: Tue Mar 12 02:48:03 2002
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Status: O

At 07:13 PM 3/11/02, ramana murthy wrote:
>One other question i have, is there a library function call that, given a 
>filename will tell me if its content is a program or elementry stream?

Ramana,

There's no library routine that does this, but you can test this yourself 
by looking at the first 4 bytes of the file.  For a Program Stream, the 
first 4 bytes should be
         0x00 0x00 0x01 0xBA (a "Pack Start Code")

     Ross.



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Date: Tue Mar 12 07:57:09 2002
X-Original-Date: Tue, 12 Mar 2002 17:46:13 +0000
Status: O

hello,
i would be very pleased if you send me QuickTime 5 Player or other play=
ers that work on RED HAT 7.1 environment or other linux operating system=
 (mandrake, freebsd...)
looking forward to your reply, i thank you in advance.

Joignez le JUG ARABICA-TN
_______________________________________________________________________=
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From <EMAIL-ADDRESS-HIDDEN> Tue Mar 12 09:08:45 2002
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Date: Tue Mar 12 09:09:06 2002
X-Original-Date: Tue, 12 Mar 2002 09:07:19 -0800
Status: O

At 09:46 AM 3/12/02, cyrine lisou wrote:
>hello,
>i would be very pleased if you send me QuickTime 5 Player or other players 
>that work on RED HAT 7.1 environment or other linux operating system 
>(mandrake, freebsd...)

QuickTime Player is available only for Mac OS and Windows; not for Linux or 
any other Unix operating system.  (By the way, "FreeBSD" is Unix, but not 
Linux.)

I you want to receive and view a MPEG/RTP video stream on Linux (or other 
forms of Unix), there are two ways that I know of to do this:

1/ Run "playRTPMPEG" <http://www.live.com/multikit/playRTPMPEG.html> (with 
the "-" and "-d" flags), and pipe the resulting MPEG video stream into a 
separate MPEG video player (such as "mplayer" and "xine").

2/ Run the "mim" (& "mtvp") applications, available from the University of 
Oregon <http://videolab.uoregon.edu/mim/>.  These can supposedly receive 
and play MPEG-1 (but not MPEG-2) RTP streams.  (I have not tried this 
myself, though.)

(FYI, I am currently working on adding RTP & RTSP support to the "mplayer" 
application.  I'll be posting to the list when this is available.)

If you just want to receive/play MPEG *audio* (e.g., MP3) RTP streams on 
Linux, then you can use "Freeamp" <http://www.freeamp.org/>, or "playRTPMPEG".

         Ross.



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Date: Tue Mar 12 10:22:17 2002
X-Original-Date: Tue, 12 Mar 2002 19:20:55 +0100
Status: O

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LjEsIGJ1dCBpIGZpbmQgbWFueSBwcm9ibGVtcywNCnJlcXVlc3QgZm9yIGhlbHANCg0KX19f
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From <EMAIL-ADDRESS-HIDDEN> Tue Mar 12 11:12:41 2002
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Date: Tue Mar 12 11:13:04 2002
X-Original-Date: Tue, 12 Mar 2002 11:12:25 -0800
Status: O

At 09:50 AM 3/12/02, kalif wrote:
>i'm trying to install this soft on Red Hat environment but
>when i tried ./genMakefiles
>:../config.:Aucun fichier ou repertoire de ce typee
>:../config.:Aucun fichier ou repertoire de ce typee
>:../config.:Aucun fichier ou repertoire de ce typee
>:../config.:Aucun fichier ou repertoire de ce typee
>:../config.:Aucun fichier ou repertoire de ce typee
>can you help me

You need to run
         ./genMakefiles linux
(not just "./genMakefiles")

         Ross.



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Date: Tue Mar 12 11:23:03 2002
X-Original-Date: Tue, 12 Mar 2002 11:22:13 -0800
Status: O

At 09:52 AM 3/12/02, Tamer Demir wrote:
>Hello,
>In the Linux I have changed the testMP3Streamer.cpp and testMP3.sdp 
>inorder to stream to a unicast IP. but although The server sends data to 
>the client (according to tcpdump). The Winamp client crashes and the 
>Realplayer gives error.

RealPlayer has problems receiving a unicast RTP stream.  (Multicast works 
OK, though.)

For Winamp (with the "in_rtp.dll" plugin):
- In the file "testMP3.sdp", try changing the "c=" line to:
         c=IN IP4 0.0.0.0/127
rather than
         c=IN IP4 172.16.1.113/127

Alternatively, rather than using a ".sdp" file, try opening the 'pseudo-URL'
         rtp://0.0.0.0:8888
or
         rtp://172.16.1.113:8888

     Ross.

ps. If you plan to stream over a local-area network only, then you should 
just use multicast, rather than trying to change the programs to use 
unicast.  (The advantage of multicast is that it will let you have more 
than one receiver listening to the same stream.)



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Subject: [Live-devel] determining if ES is audio or video
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Date: Tue Mar 12 16:50:03 2002
X-Original-Date: Wed, 13 Mar 2002 00:49:08
Status: O

Hi,
from your previous response, i can determine if my file is a Program stream 
or not.  If it is not, how do i know if the file is an audio or video 
elementry stream.  I think that a video ES has 0x00 0x00 0x01 0xB3 but what 
would an audio ES have?  I'd like to determine for sure its audio instead of 
asuming if its not video, it must be audio.

Thanks for you help.


Ramana Murthy Suryam



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Date: Tue Mar 12 20:06:24 2002
X-Original-Date: Tue, 12 Mar 2002 19:19:32 +0200
Status: O

I think there is no QuickTime Player for linux..

Tamer Demir

At 19:46 12.03.2002, cyrine lisou wrote:
>hello,
>i would be very pleased if you send me QuickTime 5 Player or other players=
=20
>that work on RED HAT 7.1 environment or other linux operating system=20
>(mandrake, freebsd...)
>looking forward to your reply, i thank you in advance.
>
>Joignez le JUG ARABICA-TN
>___________________________________________________________________________=
_______
>Constituez votre propre service d'e-mail sur le Web =E0 l'adresse=20
>http://www.zzn.com
>
>_______________________________________________
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Date: Tue Mar 12 20:23:02 2002
X-Original-Date: Wed, 13 Mar 2002 00:52:37 +0200
Status: O

Hello,
I want to measure the %packet loss in a unicast RTP MP3 or video stream,
How can I do this or which CPP files I must concanterate on?

Thank you for your help,
Sincerely,

Tamer Demir



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Date: Tue Mar 12 20:56:04 2002
X-Original-Date: Tue, 12 Mar 2002 19:52:38 +0200
Status: O

Hello,
In the Linux I have changed the testMP3Streamer.cpp and testMP3.sdp inorder 
to stream to a unicast IP. but although The server sends data to the client 
(according to tcpdump). The Winamp client crashes and the Realplayer gives 
error.

Do you have any idea??

Sincerely,

Tamer Demir

At 21:10 08.03.2002, Ross Finlayson wrote:
>At 12:18 AM 3/8/02, achraf_g(at)yahoo.fr wrote:
>>  Second, and about the use of the unicast in
>>"testMP3Streamer". I use RealPlayer on the client
>>side. Our LAN IP address are formed as follow :
>>172.16.x.x
>>(My server machine has the IP address 172.16.1.113 and
>>the client one has the 172.16.1.112 address.")
>>So what changes are to do in "testMP3Streamer.cpp" as
>>information indicated above.
>
>In  "testMP3Streamer.cpp", change "multicastAddressStr" to "172.16.1.112".
>
>Also, in the file "testMP3.sdp", change the "c=" line to:
>         c=IN IP4 172.16.1.113/127
>
>*However*, it's possible that the problem is the fault of RealPlayer - it 
>sometimes has problems playing unicast RTP streams.
>
>Instead, I suggest that you try running "Winamp" <http://www.winamp.com/>, 
>using the "in_rtp.dll" input plugin 
><http://www.live.com/multikit/winamp-plugin.html>.  If you do this, then 
>you should be able to receive the MP3/RTP stream either by reading the 
>"testMP3.sdp" file (modified as noted above), or by simply opening the 
>'pseudo-URL'
>         rtp://172.16.1.113:8888
>
>I hope this helps.
>
>         Ross.
>
>
>_______________________________________________
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Subject: [Live-devel] No Video using testMPEGVideoStreamer
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Date: Wed Mar 13 00:58:02 2002
X-Original-Date: Wed, 13 Mar 2002 08:56:44 +0000
Status: O

Hi

I am have trouble trying to view an MPEG1 stream using Quicktime 5.02 
running on a Win 98 machine. I am using Redhat 7.1 to run 
testMPEGVideoStreamer. I have enabled the RTSP define in the cpp file. Both 
computers are connected via a hub with the server IP 192.168.1.10 and the 
client 192.168.1.11

Without any other modification to the cpp file or the sdp file and typing 
rtsp://239.255.42.42:testMPEGVideo.sdp in the Quicktime URL field, QT first 
says connecting and then gives the number 10049.

Please can you tell me what I am doing wrong in this multicast case ?

I then tried the following: In the cpp file I changed the multicast address 
to the clients IP address. In the SDP file I changed the multicast address 
to the server IP address. In the cpp file I enabled the RTSP server define. 
In QT I typed rtsp://192.168.1.10:testMPEGVideo.sdp
QT then says connecting followed by getting info. When I look at the file 
properties for QT it gives me the correct values for my test.mpg file, 
however I do not see any video. I know the file is an MPEG1 elementry stream 
and I can play it as a file from both the Win 98 machine and the Linux 
machine.

Please can you tell me what I am doing wrong in this unicast case ?

Thanks in advance for you help.

George

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From <EMAIL-ADDRESS-HIDDEN> Wed Mar 13 01:44:00 2002
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Date: Wed Mar 13 01:44:04 2002
X-Original-Date: Wed, 13 Mar 2002 01:42:17 -0800
Status: O

At 04:49 PM 3/12/02, ramana murthy wrote:
>Hi,
>from your previous response, i can determine if my file is a Program 
>stream or not.  If it is not, how do i know if the file is an audio or 
>video elementry stream.  I think that a video ES has 0x00 0x00 0x01 0xB3 
>but what would an audio ES have?

A MPEG (1 or 2) audio elementary stream starts with a 4-byte 'syncword', 
which has the first 11 bits all 1.

See, for example, "MPEGAudioStreamFramer.cpp", line 170.

         Ross.



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Date: Wed Mar 13 01:52:03 2002
X-Original-Date: Wed, 13 Mar 2002 01:51:33 -0800
Status: O

At 12:56 AM 3/13/02, you wrote:
>Hi
>
>I am have trouble trying to view an MPEG1 stream using Quicktime 5.02 
>running on a Win 98 machine. I am using Redhat 7.1 to run 
>testMPEGVideoStreamer. I have enabled the RTSP define in the cpp file. 
>Both computers are connected via a hub with the server IP 192.168.1.10 and 
>the client 192.168.1.11
>
>Without any other modification to the cpp file or the sdp file and typing 
>rtsp://239.255.42.42:testMPEGVideo.sdp in the Quicktime URL field, QT 
>first says connecting and then gives the number 10049.
>
>Please can you tell me what I am doing wrong in this multicast case ?

The "rtsp://" URL that you gave above is incorrect.  It should have been:
         rtsp://39.255.42.42/testMPEGVideo.sdp
(note: "/" instead of ":")

Another thing you can do is have QT Player open the "testMPEGVideo.sdp" 
file, rather than opening a "rtsp://" URL.

If you're still having problems, then you should first make sure that 
streaming works when both the streamer and receiver are running on the same 
computer.  Then, you can move them to separate computers.

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>




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Date: Wed Mar 13 02:06:02 2002
X-Original-Date: Wed, 13 Mar 2002 02:05:00 -0800
Status: O

At 02:52 PM 3/12/02, Tamer Demir wrote:
>Hello,
>I want to measure the %packet loss in a unicast RTP MP3 or video stream,
>How can I do this or which CPP files I must concanterate on?

Tamer,

The easiest way to get this information is to use RTCP.  (This requires 
that the receiver be set up to send periodic RTCP "Reception Report" 
packets back to the sender.)

The sender can then get packet loss statistics by examining the incoming 
"Reception Report" packets (see "RTCP.cpp", line 286).

At the receiver end, you can record the packet loss statistics at the point 
where "Reception Reports" are generated (see "RTCP.cpp", line 566).

For more information about RTCP, see the latest version of the RTP/.RTCP 
specification: 
<http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-new-11.txt>

         Ross.




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Subject: [Live-devel] recombining ES
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Date: Wed Mar 13 16:24:02 2002
X-Original-Date: Thu, 14 Mar 2002 00:23:02
Status: O

Hi,
Thanks for your previous replies. You've been very helpfull.

Now i have a program stream broken into 2 elementry streams (audio and 
video) and am RTP'ing them out.  My question is how to combine them on the 
receiving side.

I see that testMPEGVideoReciver takes a video ES and resends it out on HTTP. 
But what about the audio ES?  I didn't see how the audio stream would be 
recombined with the audio for playback?

Thank you

Ramana Murthy Suryam



_________________________________________________________________
Get your FREE download of MSN Explorer at http://explorer.msn.com/intl.asp.



From <EMAIL-ADDRESS-HIDDEN> Wed Mar 13 18:45:29 2002
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Date: Wed Mar 13 18:46:03 2002
X-Original-Date: Wed, 13 Mar 2002 18:45:02 -0800
Status: O

At 04:23 PM 3/13/02, ramana murthy wrote:
>Now i have a program stream broken into 2 elementry streams (audio and 
>video) and am RTP'ing them out.  My question is how to combine them on the 
>receiving side.

Receiving implementations usually don't try to combine incoming MPEG 
Elementary Streams back into a single Program Stream.  Instead, they play 
each Elementary Stream (audio, video) separately, using either separate 
applications (one for audio; one for video), or using separate playback 
modules inside the same application.  They also need to deal with 
synchronizing the audio and video streams (using the RTP timestamps, and 
the NTP timestamps in RTCP SR packets).

As you can imagine, this is rather difficult to do, which is why the 
"LIVE.COM Streaming Media" code currently doesn't include any code for 
doing this.  Instead, it relies upon existing player implementations (e.g., 
QuickTime Player and RealPlayer) to do this.

In principle it would be possible to combine incoming MPEG/RTP audio and 
video streams into a single MPEG Program Stream.  However, this would be 
rather inflexible, as it would work only for MPEG codecs.

         Ross.




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Date: Thu Mar 14 10:10:05 2002
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Status: O

At 07:40 AM 3/14/02, =?utf-8?Q?k.alouini(at)voila.fr?= wrote:

>i made the change on testMPEGVideoStreamer ( multicastAdresseStr="127.0.0.1")
>and testMPEGVideoReceiver (multicastAdresseStr="127.0.0.1")

Of course, this will work only if "testMPEGVideoStreamer" and 
"testMPEGVideoReceiver" are running on the same computer.  However, if this 
is the case, then there is *no reason* not to continue to use multicast 
addresses.

Suggestion: Please do not change the IP addresses in the test programs from 
multicast to unicast, unless you are *sure* that you want to the stream to 
be sent between two computers that do not have multicast routing between 
them.  In particular, if you are running the programs over a local-area 
network (e.g., Ethernet), then you should continue to use multicast.

I don't understand why so many people are eager to use unicast instead of 
multicast for streaming.  Multicast is your friend :-)


>the server works perfectualy, (begennig to read from file ... done reading 
>from file)but the testMPEGVideoReceiver is blocked and it said the message 
>(beginning receiving multicast stream...)

"testMPEGVideoReceiver" may still be working correctly.  Note that - unlike 
"testMP3Receiver", which outputs to stdout - "testMPEGVideoReceiver" works 
by setting up an internal HTTP server (on port 10006) for the received MPEG 
video.

To play the MPEG video received by "testMPEGVideoReceiver", you start a 
MPEG video player (such as QuickTime Player or RealOne Player) and point it 
at the URL
         http://<localhost>:10006/video

Alternatively, you can change "testMPEGVideoReceiver.cpp" to output to 
'stdout' (instead of using an internal HTTP server), and pipe its output to 
your MPEG video player (if it supports reading from 'stdin').

         Ross.




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Subject: Re: [Live-devel] problem:with testMPEGVideoReceiver
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Date: Fri Mar 15 18:33:03 2002
X-Original-Date: Sat, 16 Mar 2002 04:31:22 +0200
Status: O

>I don't understand why so many people are eager to use unicast instead of 
>multicast for streaming.  Multicast is your friend :-)

Everything is of course very good in multicast, but as far as I know 
someone should develop MBone in LAN (although some switches or routers are 
capable of understanding multicast addresses, but this is not desired), and 
I think this is not so easy. If it is not the way I think (means that MBone 
development is easy), could you please refer me the links,manuals or your 
experiences for developing MBone in a LAN.

Thanks for any help,
Sincerely

Tamer Demir



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Date: Sun Mar 17 03:54:07 2002
X-Original-Date: Sun, 17 Mar 2002 12:50:23 +0300
Status: O

i would be   very pleasant if there is any documentation about how these=
 libs encapsulate  the MPEG Video stream in RTP playload and the way the=
 IP packets were generated.(wicth class is responsible of generation of =
ip packet)
thanks

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Date: Sun Mar 17 17:27:10 2002
X-Original-Date: Mon, 18 Mar 2002 01:25:50 +0000
Status: O



>Hi
>
>I am have trouble trying to view an MPEG1 stream using Quicktime 5.02
>running on a Win 98 machine. I am using Redhat 7.1 to run
>testMPEGVideoStreamer. I have enabled the RTSP define in the cpp file.
>Both computers are connected via a hub with the server IP 192.168.1.10 and
>the client 192.168.1.11
>
>Without any other modification to the cpp file or the sdp file and typing
>rtsp://239.255.42.42:testMPEGVideo.sdp in the Quicktime URL field, QT
>first says connecting and then gives the number 10049.
>
>Please can you tell me what I am doing wrong in this multicast case ?

The "rtsp://" URL that you gave above is incorrect.  It should have been:
         rtsp://39.255.42.42/testMPEGVideo.sdp
(note: "/" instead of ":")

I have tried using:
rtsp://239.255.42.42/testMPEGVideo.sdp

And QT now gives me the following message: "Connecting", then "requested 
data" and it just stays like that with no video.


	Another thing you can do is have QT Player open the 			"testMPEGVideo.sdp" 
file, rather than opening a "rtsp://" URL.

I copied the .sdp file to my QT machine and tried to open it. QT now gives 
me the following messages: "Live Broadcast", then "Connecting", then "ready" 
and just stays like that with no video.

If you're still having problems, then you should first make sure that
streaming works when both the streamer and receiver are running on the same 
computer.  Then, you can move them to separate computers.

I ran testMPEGVideoStreamer and testMPEGVideoReceiver on the same Linux 
machine. I then could open and play the file from a browser.

I also ran testMPEGVideoStreamer and openRTSP and when I stopped the 
openRTSP I could open and play the video-MPV file.

I did change the 127.0.0.1 in the sdp file to my "server" machines IP 
address when I tried playing from QT.

Please could you tell me what I am doing wrong. Am I correct in assuming 
that I do not need to run the testMPEGVideoReceiver on my Windows machine 
with Quicktime ? From my understanding QT should be able to play the MPEG1 
file directly from testMPEGVideoStreamer.

Thanks
George





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Date: Sun Mar 17 17:27:20 2002
X-Original-Date: Mon, 18 Mar 2002 01:26:38 +0000
Status: O



>Hi
>
>I am have trouble trying to view an MPEG1 stream using Quicktime 5.02
>running on a Win 98 machine. I am using Redhat 7.1 to run
>testMPEGVideoStreamer. I have enabled the RTSP define in the cpp file.
>Both computers are connected via a hub with the server IP 192.168.1.10 and
>the client 192.168.1.11
>
>Without any other modification to the cpp file or the sdp file and typing
>rtsp://239.255.42.42:testMPEGVideo.sdp in the Quicktime URL field, QT
>first says connecting and then gives the number 10049.
>
>Please can you tell me what I am doing wrong in this multicast case ?

The "rtsp://" URL that you gave above is incorrect.  It should have been:
         rtsp://39.255.42.42/testMPEGVideo.sdp
(note: "/" instead of ":")

I have tried using:
rtsp://239.255.42.42/testMPEGVideo.sdp

And QT now gives me the following message: "Connecting", then "requested 
data" and it just stays like that with no video.


	Another thing you can do is have QT Player open the 			"testMPEGVideo.sdp" 
file, rather than opening a "rtsp://" URL.

I copied the .sdp file to my QT machine and tried to open it. QT now gives 
me the following messages: "Live Broadcast", then "Connecting", then "ready" 
and just stays like that with no video.

If you're still having problems, then you should first make sure that
streaming works when both the streamer and receiver are running on the same 
computer.  Then, you can move them to separate computers.

I ran testMPEGVideoStreamer and testMPEGVideoReceiver on the same Linux 
machine. I then could open and play the file from a browser.

I also ran testMPEGVideoStreamer and openRTSP and when I stopped the 
openRTSP I could open and play the video-MPV file.

I did change the 127.0.0.1 in the sdp file to my "server" machines IP 
address when I tried playing from QT.

Please could you tell me what I am doing wrong. Am I correct in assuming 
that I do not need to run the testMPEGVideoReceiver on my Windows machine 
with Quicktime ? From my understanding QT should be able to play the MPEG1 
file directly from testMPEGVideoStreamer.

Thanks
George





_________________________________________________________________
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Date: Mon Mar 18 07:47:02 2002
X-Original-Date: Mon, 18 Mar 2002 17:57:58 +0100
Status: O

hello,
what is the rule of ReordingPacketBuffer class and the method MultiFram=
edRTPSource::doGetNextFrame1()
i don't understand the way that MPEGVideoRTPSink is encapsulating the M=
PEGVideo Stream  
thank you

Joignez le JUG ARABICA-TN
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Date: Mon Mar 18 08:47:01 2002
X-Original-Date: Mon, 18 Mar 2002 16:43:36 -0000
Status: O

	Greetings everyone
	I'd like to set up an RTSP server when running testMPEGAudioVideoStreamer.
How do I accomplish this?
	I looked into the bit of code that does just that in testMPEGVideoStreamer
but the createNew() function accepts either the VideoSink or the AudioSink
(not both). In fact, I tried copy/pasting that bit of code into the
testMPEGAudioVideoStreamer.cpp file and (not surpringly) only managed to
receive either sound or pictures (depending on what 'Sink I defined).

Any help is very much appreciated
Best regards,
David Pontes



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Date: Mon Mar 18 08:51:06 2002
X-Original-Date: Mon, 18 Mar 2002 17:50:26 +0100 (CET)
Status: O

hello,
I want to know if the following sentences are true or
not:"testMPEGVideoStreamer"

to send via network a MPEGVideStream,the
MPEgVideoRTPSource read an ES Video source from
"test.mpg" and gives a frame to MPEGVideoRTPSink witch

takes the responsability of encapsulation these frames
into RTP packet and then sends them into the network


thanks


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From <EMAIL-ADDRESS-HIDDEN> Tue Mar 19 01:29:32 2002
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Date: Tue Mar 19 01:30:08 2002
X-Original-Date: Tue, 19 Mar 2002 01:27:04 -0800
Status: O

At 09:34 AM 3/15/02, you wrote:
>Hello
>
>I test TestMpegVideoStream and TestMpegAudioVideoStream samples in Borland 
>C++ builder 6(win 2000 professional),
>however, I got a Exception Address Error, I debug, found that RWLock seem 
>having problem.

I haven't seen this yet.

If you can get more details about this error (i.e., about exactly where/how 
it occurs), please let me know.

         Ross.



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Date: Tue Mar 19 01:30:11 2002
X-Original-Date: Tue, 19 Mar 2002 01:25:10 -0800
Status: O

At 08:50 AM 3/18/02, you wrote:
>hello,
>I want to know if the following sentences are true or
>not:"testMPEGVideoStreamer"
>
>to send via network a MPEGVideStream,the
>MPEgVideoRTPSource read an ES Video source from
>"test.mpg" and gives a frame to MPEGVideoRTPSink witch
>
>takes the responsability of encapsulation these frames
>into RTP packet and then sends them into the network

No, the actual data chain is:

         ByteStreamFileSource -> MPEGVideoStreamFramer -> MPEGVideoRTPSink

Note that a "MPEGVideoRTPSource" does not appear at all in 
"testMPEGVideoStreamer.cpp".  This class is used only at the *receiving* 
end (e.g., by "testMPEGVideoReceiver".

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Tue Mar 19 01:32:35 2002
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Date: Tue Mar 19 01:33:16 2002
X-Original-Date: Tue, 19 Mar 2002 01:29:00 -0800
Status: O

At 06:31 PM 3/15/02, you wrote:

>>I don't understand why so many people are eager to use unicast instead of 
>>multicast for streaming.  Multicast is your friend :-)
>
>Everything is of course very good in multicast, but as far as I know 
>someone should develop MBone in LAN (although some switches or routers are 
>capable of understanding multicast addresses, but this is not desired), 
>and I think this is not so easy. If it is not the way I think (means that 
>MBone development is easy), could you please refer me the links,manuals or 
>your experiences for developing MBone in a LAN.

Almost every local-area network (e.g., Ethernet) supports multicast 'as 
is'.  However, once you start linking multiple local-area networks together 
(using routers), you will need to make sure that these routers also have 
multicast routing turned on.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Tue Mar 19 01:42:37 2002
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Date: Tue Mar 19 01:43:06 2002
X-Original-Date: Tue, 19 Mar 2002 01:39:39 -0800
Status: O

At 08:43 AM 3/18/02, Grupo Olympic - David Pontes wrote:

>         Greetings everyone
>         I'd like to set up an RTSP server when running 
> testMPEGAudioVideoStreamer.
>How do I accomplish this?
>         I looked into the bit of code that does just that in 
> testMPEGVideoStreamer
>but the createNew() function accepts either the VideoSink or the AudioSink
>(not both). In fact, I tried copy/pasting that bit of code into the
>testMPEGAudioVideoStreamer.cpp file and (not surpringly) only managed to
>receive either sound or pictures (depending on what 'Sink I defined).

Right now the "RTSPServer" implementation takes only one 'RTPSink' object 
as parameter.  To support both audio and video, it would need to be changed 
to take multiple 'RTPSink' objects as parameter.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Tue Mar 19 05:44:42 2002
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From: {Grupo Olympic - David Pontes} <EMAIL-ADDRESS-HIDDEN>
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Subject: RE: [Live-devel] RTSP with MPEGAudioVideo
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Date: Tue Mar 19 05:45:12 2002
X-Original-Date: Tue, 19 Mar 2002 13:42:17 -0000
Status: O

> >         Greetings everyone
> >         I'd like to set up an RTSP server when running
> > testMPEGAudioVideoStreamer.
> >How do I accomplish this?
> >         I looked into the bit of code that does just that in
> > testMPEGVideoStreamer
> >but the createNew() function accepts either the VideoSink or the
> AudioSink
> >(not both). In fact, I tried copy/pasting that bit of code into the
> >testMPEGAudioVideoStreamer.cpp file and (not surpringly) only managed to
> >receive either sound or pictures (depending on what 'Sink I defined).
>
> Right now the "RTSPServer" implementation takes only one 'RTPSink' object
> as parameter.  To support both audio and video, it would need to
> be changed to take multiple 'RTPSink' objects as parameter.

	Is there any work right now towards that end? If so, can you give me some
general info on how that work is progressing (ie. when can we more or less
expect to see another version with that changed)?

Thank you for the quick response
Best regards
David Pontes



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Date: Tue Mar 19 06:23:09 2002
X-Original-Date: Tue, 19 Mar 2002 15:22:04 +0100
Status: O

hello,
i see the vitual method:
void doGetNextFrame()=3D0; //called by getNextFrame()
in the class FramedSource.cpp
can you say me where is the code of this method? where is the class whi=
ch contain this method?
thank you

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Date: Tue Mar 19 07:53:07 2002
X-Original-Date: Tue, 19 Mar 2002 16:48:34 +0100 (CET)
Status: O

please , is there any documentation for the rule of
each class and about the encapsulation process of the
MPEG Video (1-2)
thank you for your help

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From <EMAIL-ADDRESS-HIDDEN> Tue Mar 19 09:49:45 2002
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Date: Tue Mar 19 09:50:04 2002
X-Original-Date: Tue, 19 Mar 2002 09:49:35 -0800
Status: O

At 05:25 PM 3/17/02, George Smith wrote:
>I did change the 127.0.0.1 in the sdp file to my "server" machines IP 
>address when I tried playing from QT.
>
>Please could you tell me what I am doing wrong. Am I correct in assuming 
>that I do not need to run the testMPEGVideoReceiver on my Windows machine 
>with Quicktime ? From my understanding QT should be able to play the MPEG1 
>file directly from testMPEGVideoStreamer.

That's correct.  The problem you are having seems to be in your choice of 
IP addresses (in "testMPEGVideoStreamer" and/or your "testMPEGVideo.sdp" file).

First, if you are streaming over a single local-area network, then 
multicast should work OK, so you *do not need* to change the IP address (in 
either "testMPEGVideoStreamer" or "testMPEGVideo.sdp") to be a unicast address.

If you insist on replacing multicast streaming with unicast (and, once 
again, your only reason for wanting to do this should be that the source 
and destination are on different networks without multicast routing between 
them), then you should change "testMPEGVideoStreamer" and 
"testMPEGVideo.sdp" as follows:

- In "testMPEGVideoStreamer", change "multicastAddressStr" to be the 
(unicast) IP address of the intended destination.
- In "testMPEGVideo.sdp" change the string "239.255.42.42" to the IP 
address of the sender.  (You can also change it to "0.0.0.0", but if you do 
this, no RTCP Reception Reports will get sent back to the sender.)

If you do this, you will be able to watch the stream using "QuickTime 
Player" (by opening the (modified) "testMPEGVideo.sdp" file).  Note, 
however, that you will *not* be able to view the stream using the "RealOne 
Player", as this seems to be able to handle multicast streams only.

         Ross.



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Date: Tue Mar 19 09:55:07 2002
X-Original-Date: Tue, 19 Mar 2002 09:53:51 -0800
Status: O

At 07:48 AM 3/19/02, alouini khalif wrote:
>please , is there any documentation for the rule of
>each class and about the encapsulation process of the
>MPEG Video (1-2)

At present, the only documentation is (i) the source code itself; in 
particular, the source code for "liveMedia" (the streaming libraries), and 
"testProgs" (which provide real, working examples of how to use the 
libraries), and  (ii) this mailing list.  (Note also this mailing list's 
archives, which are available at 
<http://www.geocrawler.com/lists/3/SourceForge/6637/0/>.)

         Ross.



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Date: Tue Mar 19 10:10:08 2002
X-Original-Date: Tue, 19 Mar 2002 10:07:22 -0800
Status: O

At 06:22 AM 3/19/02, cyrine lisou wrote:
>hello,
>i see the vitual method:
>void doGetNextFrame()=0; //called by getNextFrame()
>in the class FramedSource.cpp
>can you say me where is the code of this method? where is the class which 
>contain this method?

(Note that you can usually find answers to questions like this by searching 
through the source code of the "liveMedia" libraries.)

"doGetNextFrame()" is a virtual function that is is defined by each 
subclass of "FramedSource".  It is the routine that implements the 
"getNextFrame()" operation.  I.e., each particular 'data source' object 
defines "doGetNextFrame()" so as to deliver its data.

Some sources - e.g., "ByteStreamFileSource" - are pure sources; they 
implement "doGetNextFrame()" by generating their own data.  (See the 
implementation of "ByteStreamFileSource::doGetNextFrame()" for an example 
of this.)

Other sources - e.g., "MPEGVideoStreamFramer" - are *filters* (i.e., 
subclasses of "FramedFilter").  They produce their data by first reading 
data from another source (i.e., by calling this other source's 
"getNextFrame()" routine), and then transforming or framing it in some 
way.  (See the implementation of "MPEGVideoStreamFramer::doGetNextFrame()" 
for an example of this.)

         Ross.



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Date: Tue Mar 19 10:31:02 2002
X-Original-Date: Tue, 19 Mar 2002 10:26:46 -0800
Status: O

At 08:57 AM 3/18/02, you wrote:
>hello,
>what is the rule of ReordingPacketBuffer class

Because RTP uses UDP - an unreliable datagram protocol, it is possible for 
RTP packets to arrive out-of-order.  If this happens, the RTP "sequence 
number" field can be used to put the incoming packets back in the correct 
order.  This is the job of the "ReorderingPacketBuffer" class.

>  and the method MultiFramedRTPSource::doGetNextFrame1()

Note that the call
         BufferedPacket* nextPacket
                 = fReorderingBuffer->getNextAvailablePacket();
will return the contents of the next incoming RTP packet (excluding the RTP 
header itself) in the sequence.

The code then skips over any special header (that occurs after the RTP 
header).  The virtual function "processSpecialHeader()" is used to test for 
such a header.  (Note, in particular, the default implementation of this 
function (in "MultiFramedRTPSource.cpp"), and the special implementation 
for MPEG video (in "MPEGVideoRTPSource.cpp").)

The routine "BufferedPacket::use()" is then called to deliver the remaining 
data (i.e., after the RTP header and any special header) to the destination 
(that called "getNextFrame()").  By default (and this is the case for MPEG 
video streams) the entire remaining data is delivered to the 
destination.  However, in the case where several frames are packed into a 
single RTP packet, then the implementation can redefine 
"nextEnclosedFrameSize()" so as to deliver only individual frames - on at a 
time - to the destination.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Tue Mar 19 10:48:52 2002
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Date: Tue Mar 19 10:49:06 2002
X-Original-Date: Tue, 19 Mar 2002 10:48:32 -0800
Status: O

At 05:42 AM 3/19/02, you wrote:
> > Right now the "RTSPServer" implementation takes only one 'RTPSink' object
> > as parameter.  To support both audio and video, it would need to
> > be changed to take multiple 'RTPSink' objects as parameter.
>
>         Is there any work right now towards that end? If so, can you give 
> me some
>general info on how that work is progressing (ie. when can we more or less
>expect to see another version with that changed)?

David,

This is on my "to do" list, but I'm not sure yet when it is likely to get 
done.  (I give priority to work funded by paying consulting clients.)

(If your organization is interested in providing funding for this task, 
then please let me know (via private email).)

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>




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Date: Tue Mar 19 13:02:14 2002
X-Original-Date: Tue, 19 Mar 2002 15:02:12 -0600
Status: O

This is a multi-part message in MIME format.

------=_NextPart_000_000F_01C1CF57.0CCC3B20
Content-Type: text/plain;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

Hi,
   After successfully compiled and linked, when I debug the code line by =
line in
program "testMPEGAudioVideostreamer" . But at the following:

RTCPInstance::createNew(*env, &rtcpGroupsockAudio,
     totalSessionBandwidthAudio, CNAME,
     audioSink, NULL)=20

I get Unhandled exception Error at testMPEGAudioVideoStreamer : =
0XC000005: Access Violation.

Why and how to fix it.

------=_NextPart_000_000F_01C1CF57.0CCC3B20
Content-Type: text/html;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 5.50.4522.1800" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT size=3D2>Hi,</FONT></DIV>
<DIV><FONT size=3D2>&nbsp;&nbsp; After successfully compiled and linked, =
when I=20
debug the code line by line in</FONT></DIV>
<DIV><FONT size=3D2>program "testMPEGAudioVideostreamer" . But at the=20
following:</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>RTCPInstance::createNew(*env,=20
&amp;rtcpGroupsockAudio,<BR>&nbsp;&nbsp;&nbsp;&nbsp; =
totalSessionBandwidthAudio,=20
CNAME,<BR>&nbsp;&nbsp;&nbsp;&nbsp; audioSink, NULL) </FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>I get Unhandled exception Error at =
testMPEGAudioVideoStreamer=20
: 0XC000005: Access Violation.</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>Why and how to fix it.</FONT></DIV></BODY></HTML>

------=_NextPart_000_000F_01C1CF57.0CCC3B20--



From <EMAIL-ADDRESS-HIDDEN> Tue Mar 19 17:58:07 2002
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Subject: [Live-devel] About testMpegAudioVideoStream Exception in c++ Builder 6
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Date: Tue Mar 19 17:59:01 2002
X-Original-Date: Thu, 21 Mar 2002 09:59:39 +0800
Status: O

Ross

    Thanks your reply!

    The error prompt is "Project testMpegAudioVideoStream raised a =
exception class int with message 'Exception Object =
Address:0x9785AE'.Process stopped. Use Step or Run to continue."
   =20
    The cursor stop in the line "throw NO_MORE_BUFFERED_INPUT;" of =
StreamParser.cpp file.
   =20
    The mpeg file using for test is water.mpg in MS DirectXSDK8.1.


    Thank you again!
                                                                         =
   jhshao
                                                                         =
  2002.3.21  


From <EMAIL-ADDRESS-HIDDEN> Tue Mar 19 20:24:30 2002
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Date: Tue Mar 19 20:25:02 2002
X-Original-Date: Tue, 19 Mar 2002 20:23:47 -0800
Status: O

At 05:59 PM 3/20/02, Shao J.H wrote:
>    The error prompt is "Project testMpegAudioVideoStream raised a 
> exception class int with message 'Exception Object 
> Address:0x9785AE'.Process stopped. Use Step or Run to continue."
>
>    The cursor stop in the line "throw NO_MORE_BUFFERED_INPUT;" of 
> StreamParser.cpp file.

But that's not an error.  This code is *supposed* to be throwing a (C++) 
exception when it fills up an input buffer - this how the code handles this 
special case.  Note that this is a C++ language exception (not a runtime 
error like 'invalid memory access' or 'divide by zero').

Note that these exceptions are supposed to be caught by the "try ... catch" 
statement in "MPEGVideoStreamParser::parse()" (see 
"MPEGVideoStreamFramer.cpp").  If, for some reason, this is not happening, 
then there may be a problem in the way that your development environment 
deals with C++ exceptions.  (Perhaps there's some flag that you can give to 
your compiler to make it do the right thing with C++ exceptions?)

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Tue Mar 19 20:42:40 2002
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Date: Tue Mar 19 20:43:03 2002
X-Original-Date: Wed, 20 Mar 2002 15:11:49 +1030
Status: O

Yes, this is a feature of BCB. Go to Tools | Debugger Options, tab "Language
Exceptions" and remove the tick from "Stop on C++ exceptions" (RTFM). What
happens is Borland IDE traps the exceptions as they occur regardless of the
fact that this may happen within try {} catch statement.

If you run the compiled .exe it should run without this error message. 

I have BCB5 but I assume it is the same with BCB6.

Regards

Vesselin

> 
> At 05:59 PM 3/20/02, Shao J.H wrote:
> >    The error prompt is "Project testMpegAudioVideoStream raised a 
> > exception class int with message 'Exception Object 
> > Address:0x9785AE'.Process stopped. Use Step or Run to continue."
> >
> >    The cursor stop in the line "throw NO_MORE_BUFFERED_INPUT;" of 
> > StreamParser.cpp file.
> 
> But that's not an error.  This code is *supposed* to be 
> throwing a (C++) exception when it fills up an input buffer - this how the

(snip)


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From: {Chad Williams} <EMAIL-ADDRESS-HIDDEN>
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Date: Thu Mar 21 10:12:12 2002
X-Original-Date: Fri, 22 Mar 2002 02:15:19 -0800
Status: O

This is a multi-part message in MIME format.

------=_NextPart_000_0012_01C1D147.6A23D160
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	charset="iso-8859-1"
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Hi all,

  Im trying to recieve the stream output of testmp3streamer on=20
  an embedded system and I have been able to recieve the=20
  udp packets with the rtcp info but not the rtp packets.

  I was wondering with multicast operation could there
  be an issue with ip settings for multicast not being handled
  by my embedded system and hence preventing the rtp packets
  from being passed on to my app.

  Is there a way I can convert testMP3 streamer to unicast ?

thanks

Chad


------=_NextPart_000_0012_01C1D147.6A23D160
Content-Type: text/html;
	charset="iso-8859-1"
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 5.50.4913.1100" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Hi all,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp; Im trying to recieve the stream =
output of=20
testmp3streamer on </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp; an embedded system and I have =
been able to=20
recieve the </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp; udp packets with the rtcp info =
but not the=20
rtp packets.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp; I was wondering with multicast =
operation=20
could there</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp; be an issue with ip settings for =
multicast=20
not being handled</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp; by my embedded system and hence =
preventing=20
the rtp packets</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp; from being passed on to my =
app.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp; Is there a way I can convert =
testMP3=20
streamer to unicast ?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>thanks</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Chad</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV></BODY></HTML>

------=_NextPart_000_0012_01C1D147.6A23D160--



From <EMAIL-ADDRESS-HIDDEN> Thu Mar 21 12:35:22 2002
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Subject: Re: [Live-devel] testMP3streamer unicast operation
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Date: Thu Mar 21 12:43:01 2002
X-Original-Date: Thu, 21 Mar 2002 12:34:06 -0800
Status: O

At 02:15 AM 3/22/02, Chad Williams wrote:
>   Im trying to recieve the stream output of testmp3streamer on
>   an embedded system and I have been able to recieve the
>   udp packets with the rtcp info but not the rtp packets.

That's odd.  Just to be clear: Are you saying that your receiving software 
- when it joined the multicast address (239.255.42.42) was able to receive 
RTCP packets (on port 8889), but received nothing on port 8888?


>    I was wondering with multicast operation could there
>   be an issue with ip settings for multicast not being handled
>   by my embedded system and hence preventing the rtp packets
>   from being passed on to my app.

Out of curiosity - what kind of embedded system is your receiving software 
running, and how is it connected (physically) to the system that's running 
"testMP3Streamer"?  If the connection is using something like Ethernet, 
then multicast should work, although it's conceivable - depending on the 
nature of your receiving system - that it is not implementing multicast 
reception properly.


>   Is there a way I can convert testMP3 streamer to unicast ?

Yes - you can just replace "239.255.42.42" with a unicast IP address (the 
address of the intended destination).

If possible, though, you should continue to use multicast, as it's more 
flexible (it would allow you to have more than one receiver on the same 
network, and would not require you to 'hard wire' the destination IP 
address into the sender).

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Thu Mar 21 14:44:08 2002
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Subject: [Live-devel] testMPEGVideoStreaming HELP!!!
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Date: Thu Mar 21 14:45:02 2002
X-Original-Date: Thu, 21 Mar 2002 16:46:29 -0600
Status: O

Hi,
     I am trying to run testMPEGVideoStreaming.cpp on Window 98. I made the
following changes:

    - change value of the inputFileName from test.mpeg to "car.mpeg"

    - I change the IP address(239.255.42.42) to the IP address of my
machine.

Then I tried to execute this program the following error:
  Unhandled exception Error at testMPEGAudioVideoStreamer : =
  0XC000005: Access Violation.

Upon further invesigation, it was specifically crashing at
RTCPInstance::createNew(*env, &rtcpGroupsock,
			  totalSessionBandwidth, CNAME,
			  videoSink, NULL )

More specifically:
  void BasicTaskScheduler::turnOnBackgroundReadHandling(int socketNum,
				BackgroundHandlerProc* handlerProc,
				void* clientData) {
  FD_SET(socketNum, &fReadSet);
  fHandlerProc[socketNum] = handlerProc;
  fClientData[socketNum] = clientData;

  if (socketNum+1 > fMaxNumSockets) {
    fMaxNumSockets = socketNum+1;
  }
}

In this above code, when handlerProc is being copied into
fhandlerProc[socketNum], then it crash.

PLEASE HELP!!!



From <EMAIL-ADDRESS-HIDDEN> Thu Mar 21 15:05:21 2002
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Date: Thu Mar 21 15:06:23 2002
X-Original-Date: Thu, 21 Mar 2002 15:05:13 -0800
Status: O

At 02:46 PM 3/21/02, Sandeep Kinra wrote:
>    - I change the IP address(239.255.42.42) to the IP address of my
>machine.

That was the mistake.  You should either leave the IP address as is (if 
"testMP3Streamer" and the receiving application are running on the same 
machine, or are both on a multicast-capable network (e.g., an 
Ethernet)).  Alternatively, you can change it to be the IP address of the 
*receiver*.


>More specifically:
>   void BasicTaskScheduler::turnOnBackgroundReadHandling(int socketNum,
>                                 BackgroundHandlerProc* handlerProc,
>                                 void* clientData) {
>   FD_SET(socketNum, &fReadSet);
>   fHandlerProc[socketNum] = handlerProc;
>   fClientData[socketNum] = clientData;
>
>   if (socketNum+1 > fMaxNumSockets) {
>     fMaxNumSockets = socketNum+1;
>   }
>}
>
>In this above code, when handlerProc is being copied into
>fhandlerProc[socketNum], then it crash.

Yes, this is happening because "socketNum" in this case is "-1" (because 
the socket was not created correctly).  In the next released version of the 
library, I'll try to make this code handle this error without crashing.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Thu Mar 21 15:16:19 2002
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From: Vesselin Kostadinov <EMAIL-ADDRESS-HIDDEN>
To: <EMAIL-ADDRESS-HIDDEN>
Subject: RE: [Live-devel] testMPEGVideoStreaming HELP!!!
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Date: Thu Mar 21 15:17:04 2002
X-Original-Date: Fri, 22 Mar 2002 09:45:24 +1030
Status: O

Quick fix: 

#define FD_SETSIZE 16384


(search for FD_SETSIZE and change the number)


Explanation:

Perhaps this is somewhat a result of a "quick 'n' durty" approach in
implementing the BasicTaskScheduler. FD_SETSIZE defines the maximum count of
sockets that the application use simultaneously, so an application that uses
sockets NoNo 680, 720, 840 and 1050 should work fine with FD_SETSIZE 4. 

However FD_SETSIZE is used (not quite correct IMHO) to allocate the size of
2 arrays:

BackgroundHandlerProc* fHandlerProc[FD_SETSIZE];
void* fClientData[FD_SETSIZE];

Those arrays are indexed by the socket number (wich in turn is generated by
the OS) and could well be greater than the predefined size of FD_SETSIZE.



Hope this helps.

Vesselin


> -----Original Message-----
> From: Sandeep Kinra [mailto:kinra(at)endinfosys.com]
> Sent: Friday, 22 March 2002 9:16
> To: live-devel(at)lists.sourceforge.net
> Subject: [Live-devel] testMPEGVideoStreaming HELP!!!
> 
> 
> Hi,
>      I am trying to run testMPEGVideoStreaming.cpp on Window 
> 98. I made the
> following changes:
> 
>     - change value of the inputFileName from test.mpeg to "car.mpeg"
> 
>     - I change the IP address(239.255.42.42) to the IP address of my
> machine.
> 
> Then I tried to execute this program the following error:
>   Unhandled exception Error at testMPEGAudioVideoStreamer : =
>   0XC000005: Access Violation.
> 
> Upon further invesigation, it was specifically crashing at
> RTCPInstance::createNew(*env, &rtcpGroupsock,
> 			  totalSessionBandwidth, CNAME,
> 			  videoSink, NULL )
> 
> More specifically:
>   void BasicTaskScheduler::turnOnBackgroundReadHandling(int socketNum,
> 				BackgroundHandlerProc* handlerProc,
> 				void* clientData) {
>   FD_SET(socketNum, &fReadSet);
>   fHandlerProc[socketNum] = handlerProc;
>   fClientData[socketNum] = clientData;
> 
>   if (socketNum+1 > fMaxNumSockets) {
>     fMaxNumSockets = socketNum+1;
>   }
> }
> 
> In this above code, when handlerProc is being copied into
> fhandlerProc[socketNum], then it crash.
> 
> PLEASE HELP!!!
> 
> 
> _______________________________________________
> Live-devel mailing list
> Live-devel(at)lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/live-devel
> 


From <EMAIL-ADDRESS-HIDDEN> Thu Mar 21 15:44:39 2002
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Subject: RE: [Live-devel] testMPEGVideoStreaming HELP!!!
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Date: Thu Mar 21 15:45:03 2002
X-Original-Date: Thu, 21 Mar 2002 15:44:01 -0800
Status: O

At 03:15 PM 3/21/02, Vesselin Kostadinov wrote:
>  FD_SETSIZE defines the maximum count of
>sockets that the application use simultaneously, so an application that uses
>sockets NoNo 680, 720, 840 and 1050 should work fine with FD_SETSIZE 4.

I don't think so, because if you look at the definitions of the macros 
FD_SET, FD_CLR, FD_ISSET etc. (usually in "/usr/include/sys/types.h"), then 
you set that these macros - which take the socket number as parameter - are 
implemented by indexing into a bit array (of size FD_SETSIZE bits).  So, 
for these macros to work correctly, "FD_SETSIZE" has to be the greatest 
possible socket number (less one) - not just the greatest number of sockets 
in use simultaneously.

I suspect that the original poster's problem was caused by trying to index 
one of these arrays with an index of -1.

         Ross.



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From: {Sandeep Kinra} <EMAIL-ADDRESS-HIDDEN>
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Subject: RE: [Live-devel] testMPEGVideoStreaming HELP!!!
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Date: Thu Mar 21 15:53:04 2002
X-Original-Date: Thu, 21 Mar 2002 17:55:04 -0600
Status: O

Hi.
   Thanks for the response. It did not work.

All I am trying to do is run sample testProgram testMPEGAudioVideoStreamer
which will stream my "car.mpeg" file rather than "test.mpeg". And run
testMPEGAudioVideoReceiver to receive the content of "car.mpeg".

Once I compile both program successfully I tried to first execute
testMPEGAudioVideoStreamer program it gives me the following error:

"This program has performed an illegal operation and will shut down"

When I debug it I gave the Unexception error as mentioned in previous
E-mail.

And socknum is not -1 but 48.

Am I missing anything? Do I have to run anything else?  Please help.

Thanks,

-----Original Message-----
From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
Sent: Thursday, March 21, 2002 5:05 PM
To: <EMAIL-ADDRESS-HIDDEN>
Cc: <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] testMPEGVideoStreaming HELP!!!


At 02:46 PM 3/21/02, Sandeep Kinra wrote:
>    - I change the IP address(239.255.42.42) to the IP address of my
>machine.

That was the mistake.  You should either leave the IP address as is (if
"testMP3Streamer" and the receiving application are running on the same
machine, or are both on a multicast-capable network (e.g., an
Ethernet)).  Alternatively, you can change it to be the IP address of the
*receiver*.


>More specifically:
>   void BasicTaskScheduler::turnOnBackgroundReadHandling(int socketNum,
>                                 BackgroundHandlerProc* handlerProc,
>                                 void* clientData) {
>   FD_SET(socketNum, &fReadSet);
>   fHandlerProc[socketNum] = handlerProc;
>   fClientData[socketNum] = clientData;
>
>   if (socketNum+1 > fMaxNumSockets) {
>     fMaxNumSockets = socketNum+1;
>   }
>}
>
>In this above code, when handlerProc is being copied into
>fhandlerProc[socketNum], then it crash.

Yes, this is happening because "socketNum" in this case is "-1" (because
the socket was not created correctly).  In the next released version of the
library, I'll try to make this code handle this error without crashing.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Thu Mar 21 16:08:09 2002
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Date: Thu Mar 21 16:09:05 2002
X-Original-Date: Thu, 21 Mar 2002 16:08:04 -0800
Status: O

At 03:55 PM 3/21/02, Sandeep Kinra wrote:
>   Thanks for the response. It did not work.
>
>All I am trying to do is run sample testProgram testMPEGAudioVideoStreamer
>which will stream my "car.mpeg" file rather than "test.mpeg". And run
>testMPEGAudioVideoReceiver to receive the content of "car.mpeg".

Do you mean "testMPEGVideoReceiver"?  (There's no 
"testMPEGAudioVideoReceiver" program.)

>And socknum is not -1 but 48.

OK, I stand corrected.

What operating system are you running "testMPEGAudioVideoStreamer" on, and 
what development tools did you use to build it?

         Ross.

ps. You should really join the mailing list, so you can be sure that all of 
the messages in this thread will get to you.



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Date: Thu Mar 21 16:23:02 2002
X-Original-Date: Fri, 22 Mar 2002 10:51:33 +1030
Status: O


>... because if you look at the definitions of the macros 
> FD_SET, FD_CLR, FD_ISSET etc. (usually in "/usr/include/sys/types.h"),

Ok, we may be facing a fundamental difference between Unix and Windoze.

On BCB5 these are defined in C:\Program
Files\Borland\CBuilder5\Include\winsock.h:

typedef struct fd_set {
        u_int   fd_count;               /* how many are SET? */
        SOCKET  fd_array[FD_SETSIZE];   /* an array of SOCKETs */
} fd_set;

//.....

#define FD_ZERO(set) (((fd_set FAR *)(set))->fd_count=0)

//....

#define FD_SET(fd, set) do { \
    if (((fd_set FAR *)(set))->fd_count < FD_SETSIZE) \
        ((fd_set FAR *)(set))->fd_array[((fd_set FAR
*)(set))->fd_count++]=(fd);\
} while(0)



further comments follow.



>... then you set that these macros - which take the socket number as
parameter - ...

I agree


>... are implemented by indexing into a bit array (of size FD_SETSIZE bits).


Yes, but the index is NOT the socket number, it is rather a number between 0
and fd_set.fd_count

As you can see FD_ZERO just sets fd_count to 0

Then calling FD_SET(680, set):
1. checks if set.fd_count < FD_SETSIZE (to prevent overflow)
2. Sets set.fd_array[0] to 680
3. Sets set.fd_count to 1

After another call FD_SET(720, set) structure set is:

fd_count = 2
fd_array = (680, 720, /* other (FD_SETSIZE - 2) irrelevant numbers*/)



> So, for these macros to work correctly, "FD_SETSIZE" has to be the
greatest 
> possible socket number (less one) - not just the greatest number of
sockets 
> in use simultaneously.

Well, not in Windoze/BCB5



From <EMAIL-ADDRESS-HIDDEN> Thu Mar 21 16:51:59 2002
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Date: Thu Mar 21 16:52:03 2002
X-Original-Date: Thu, 21 Mar 2002 16:51:51 -0800
Status: O

At 04:21 PM 3/21/02, you wrote:
> > So, for these macros to work correctly, "FD_SETSIZE" has to be the
>greatest
> > possible socket number (less one) - not just the greatest number of
>sockets
> > in use simultaneously.
>
>Well, not in Windoze/BCB5

Ahh, you're correct.  It turns out that this is also the case for Visual C++.

My implementation of "BasicTaskScheduler" assumed that FD_SETSIZE was 
always used the Unix way.

Fortunately, I also do the following, which happens to make the code work 
on Windows OK:

#if defined(__WIN32__) || defined(_WIN32)
#define FD_SETSIZE 1024
#endif

So, this won't explain the original poster's problem, unless perhaps he was 
compiling for some non-Windows environment that happens to use FD_SETSIZE 
in the same way as Windows.

The proper solution will be for me to reimplement "BasicTaskScheduler" to 
not use FD_SETSIZE at all.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Thu Mar 21 17:03:44 2002
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Date: Thu Mar 21 17:04:16 2002
X-Original-Date: Thu, 21 Mar 2002 19:06:00 -0600
Status: O

Hi,
  Sorry for misunderstanding. The programs I am trying to run are
testMPEGVideoStreaming and testMPEGVideoReceiver.

Operating System: Window 98. I use VC++ 6.0

Sandeep

-----Original Message-----
From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
Sent: Thursday, March 21, 2002 6:08 PM
To: <EMAIL-ADDRESS-HIDDEN>
Cc: <EMAIL-ADDRESS-HIDDEN>
Subject: RE: [Live-devel] testMPEGVideoStreaming HELP!!!


At 03:55 PM 3/21/02, Sandeep Kinra wrote:
>   Thanks for the response. It did not work.
>
>All I am trying to do is run sample testProgram testMPEGAudioVideoStreamer
>which will stream my "car.mpeg" file rather than "test.mpeg". And run
>testMPEGAudioVideoReceiver to receive the content of "car.mpeg".

Do you mean "testMPEGVideoReceiver"?  (There's no
"testMPEGAudioVideoReceiver" program.)

>And socknum is not -1 but 48.

OK, I stand corrected.

What operating system are you running "testMPEGAudioVideoStreamer" on, and
what development tools did you use to build it?

         Ross.

ps. You should really join the mailing list, so you can be sure that all of
the messages in this thread will get to you.



From <EMAIL-ADDRESS-HIDDEN> Thu Mar 21 17:29:39 2002
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Date: Thu Mar 21 17:30:13 2002
X-Original-Date: Thu, 21 Mar 2002 17:29:34 -0800
Status: O

At 05:06 PM 3/21/02, Sandeep Kinra wrote:
>Hi,
>   Sorry for misunderstanding. The programs I am trying to run are
>testMPEGVideoStreaming and testMPEGVideoReceiver.
>
>Operating System: Window 98. I use VC++ 6.0

Note the code in lines 20 through 24 of 
"BasicUsageEnvironment/BasicUsageEnvironment.cpp":

         #if defined(__WIN32__) || defined(_WIN32)
         // For some reason Windoze doesn't define FD_SETSIZE large enough
         // (go figure), so we redefine it here, before we #include anything.
         #define FD_SETSIZE 1024
         #endif

This redefinition of "FD_SETSIZE" should be happening for you (because 
you're building for Windows).  But please try removing the
         #if defined ...
and
         #endif
lines, and recompile, to see if that makes any difference.  You could also 
try changing the (re)definition of FD_SETSIZE from 1024 to 16384.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Thu Mar 21 19:36:52 2002
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Subject: Re: [Live-devel] testMP3streamer unicast operation
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Date: Thu Mar 21 19:37:03 2002
X-Original-Date: Fri, 22 Mar 2002 11:40:46 -0800
Status: O

Actually this was due to the the MTU in my recieving software being set at
600
 and as a result it discarded the frames containing rtp packets.. with no
error generated.
Setting MTU to 1500 fixed the problem.

My embedded system doesn't implement IGMP so perhaps multicast operation
isn't possible.  FYI the system is a module (RCM2100) from www.zworld.com

Cheers

----- Original Message -----
From: {Ross Finlayson} <EMAIL-ADDRESS-HIDDEN>
To: <EMAIL-ADDRESS-HIDDEN>
Sent: Thursday, March 21, 2002 12:34 PM
Subject: Re: [Live-devel] testMP3streamer unicast operation


> At 02:15 AM 3/22/02, Chad Williams wrote:
> >   Im trying to recieve the stream output of testmp3streamer on
> >   an embedded system and I have been able to recieve the
> >   udp packets with the rtcp info but not the rtp packets.
>
> That's odd.  Just to be clear: Are you saying that your receiving software
> - when it joined the multicast address (239.255.42.42) was able to receive
> RTCP packets (on port 8889), but received nothing on port 8888?
>
>
> >    I was wondering with multicast operation could there
> >   be an issue with ip settings for multicast not being handled
> >   by my embedded system and hence preventing the rtp packets
> >   from being passed on to my app.
>
> Out of curiosity - what kind of embedded system is your receiving software
> running, and how is it connected (physically) to the system that's running
> "testMP3Streamer"?  If the connection is using something like Ethernet,
> then multicast should work, although it's conceivable - depending on the
> nature of your receiving system - that it is not implementing multicast
> reception properly.
>
>
> >   Is there a way I can convert testMP3 streamer to unicast ?
>
> Yes - you can just replace "239.255.42.42" with a unicast IP address (the
> address of the intended destination).
>
> If possible, though, you should continue to use multicast, as it's more
> flexible (it would allow you to have more than one receiver on the same
> network, and would not require you to 'hard wire' the destination IP
> address into the sender).
>
>          Ross.
>
>
> _______________________________________________
> Live-devel mailing list
> Live-devel(at)lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/live-devel



From <EMAIL-ADDRESS-HIDDEN> Thu Mar 21 20:03:07 2002
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Date: Thu Mar 21 20:04:02 2002
X-Original-Date: Thu, 21 Mar 2002 20:02:35 -0800
Status: O

At 11:40 AM 3/22/02, you wrote:
>Actually this was due to the the MTU in my recieving software being set at
>600
>  and as a result it discarded the frames containing rtp packets.. with no
>error generated.
>Setting MTU to 1500 fixed the problem.

That's great to hear.

FYI, it would also have been possible to reduce the sender's packet size by 
changing the definitions of "maxPacketSize" and "preferredPacketSize" in 
"liveMedia/MultiFramedRTPSink.cpp".  I don't recommend doing this, though, 
as it's more efficient to do what you did: Upgrade your receiver to accept 
larger packets.  (Also, if you made the packet size too small, you'd find 
that single MP3 frames would not fit, and the implementation doesn't yet 
implement RFC 2250 fragmentation properly.)


>My embedded system doesn't implement IGMP so perhaps multicast operation
>isn't possible.  FYI the system is a module (RCM2100) from www.zworld.com

I notice (from <http://www.zworld.com/products/rcm2100/>) that this module 
comes with its own TCP/IP stack, with source code.  You might find it 
worthwhile to run "grep -i multicast" on this source code.  By the way, 
IGMP is a protocol that hosts use to signal changes in IP multicast 
memberships to *routers or switches*.  If your network is just a single 
Ethernet segment (or uses a simple Ethernet hub, rather than a more complex 
Ethernet 'switch'), then you might not need IGMP at all.

Anyway, do you now find that unicast streaming works OK in your 
environment, but that multicast streaming doesn't?

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Thu Mar 21 20:19:01 2002
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Date: Thu Mar 21 20:19:04 2002
X-Original-Date: Fri, 22 Mar 2002 12:22:54 -0800
Status: O

Ross,

  Yes the unicast streaming is working fine.
   As I will not be needing to look at multicast, for now I will
   concentrate on getting the other parts of the system working.
Thanks

Chad

----- Original Message -----
From: {Ross Finlayson} <EMAIL-ADDRESS-HIDDEN>
To: <EMAIL-ADDRESS-HIDDEN>
Sent: Thursday, March 21, 2002 8:02 PM
Subject: Re: [Live-devel] testMP3streamer unicast operation


> At 11:40 AM 3/22/02, you wrote:
> >Actually this was due to the the MTU in my recieving software being set
at
> >600
> >  and as a result it discarded the frames containing rtp packets.. with
no
> >error generated.
> >Setting MTU to 1500 fixed the problem.
>
> That's great to hear.
>
> FYI, it would also have been possible to reduce the sender's packet size
by
> changing the definitions of "maxPacketSize" and "preferredPacketSize" in
> "liveMedia/MultiFramedRTPSink.cpp".  I don't recommend doing this, though,
> as it's more efficient to do what you did: Upgrade your receiver to accept
> larger packets.  (Also, if you made the packet size too small, you'd find
> that single MP3 frames would not fit, and the implementation doesn't yet
> implement RFC 2250 fragmentation properly.)
>
>
> >My embedded system doesn't implement IGMP so perhaps multicast operation
> >isn't possible.  FYI the system is a module (RCM2100) from www.zworld.com
>
> I notice (from <http://www.zworld.com/products/rcm2100/>) that this module
> comes with its own TCP/IP stack, with source code.  You might find it
> worthwhile to run "grep -i multicast" on this source code.  By the way,
> IGMP is a protocol that hosts use to signal changes in IP multicast
> memberships to *routers or switches*.  If your network is just a single
> Ethernet segment (or uses a simple Ethernet hub, rather than a more
complex
> Ethernet 'switch'), then you might not need IGMP at all.
>
> Anyway, do you now find that unicast streaming works OK in your
> environment, but that multicast streaming doesn't?
>
>          Ross.
>
>
> _______________________________________________
> Live-devel mailing list
> Live-devel(at)lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/live-devel



From <EMAIL-ADDRESS-HIDDEN> Thu Mar 21 20:32:40 2002
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Date: Thu Mar 21 20:33:04 2002
X-Original-Date: Thu, 21 Mar 2002 20:31:52 -0800
Status: O

At 12:22 PM 3/22/02, Chad Williams wrote:
>   Yes the unicast streaming is working fine.
>    As I will not be needing to look at multicast, for now I will
>    concentrate on getting the other parts of the system working.

Chad,

Out of curiosity - how easy (or difficult) was it for you to get this code 
built and running on your (embedded system module) receiver?  What OS is it 
running?  Did you have to make many changes to the code (e.g., to modify or 
redefine "BasicUsageEnvironment")?

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Fri Mar 22 07:04:58 2002
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Subject: RE: [Live-devel] testMPEGVideoStreaming HELP!!!
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Date: Fri Mar 22 07:05:09 2002
X-Original-Date: Fri, 22 Mar 2002 09:07:24 -0600
Status: O

Hi,
Try it by removing #if defined(__WIN32__) || defined(_WIN32)
			 and
			 #endif


and setting FD_SETSIZE 16384

Still no luck!!!  What else I need to do? Any alternative?

Thanks,

Sandeep

-----Original Message-----
From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
Sent: Thursday, March 21, 2002 7:30 PM
To: <EMAIL-ADDRESS-HIDDEN>
Cc: <EMAIL-ADDRESS-HIDDEN>
Subject: RE: [Live-devel] testMPEGVideoStreaming HELP!!!


At 05:06 PM 3/21/02, Sandeep Kinra wrote:
>Hi,
>   Sorry for misunderstanding. The programs I am trying to run are
>testMPEGVideoStreaming and testMPEGVideoReceiver.
>
>Operating System: Window 98. I use VC++ 6.0

Note the code in lines 20 through 24 of
"BasicUsageEnvironment/BasicUsageEnvironment.cpp":

         #if defined(__WIN32__) || defined(_WIN32)
         // For some reason Windoze doesn't define FD_SETSIZE large enough
         // (go figure), so we redefine it here, before we #include
anything.
         #define FD_SETSIZE 1024
         #endif

This redefinition of "FD_SETSIZE" should be happening for you (because
you're building for Windows).  But please try removing the
         #if defined ...
and
         #endif
lines, and recompile, to see if that makes any difference.  You could also
try changing the (re)definition of  from 1024 to 16384.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Fri Mar 22 09:07:27 2002
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This is a multi-part message in MIME format.

------=_NextPart_000_0005_01C1D1CC.445D8E00
Content-Type: text/plain;
	charset="iso-8859-2"
Content-Transfer-Encoding: quoted-printable

I get Unhandled exception Error at testMP3Streamer : =3D
0XC000005: Access Violation.

Why and how to fix it.



------=_NextPart_000_0005_01C1D1CC.445D8E00
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content=3D"text/html; charset=3Diso-8859-2" =
http-equiv=3DContent-Type>
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<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3D"Arial CE" size=3D2>I get Unhandled exception Error at =

testMP3Streamer : =3D<BR>0XC000005: Access Violation.<BR><BR>Why and how =
to fix=20
it.<BR><BR></FONT></DIV></BODY></HTML>

------=_NextPart_000_0005_01C1D1CC.445D8E00--



From <EMAIL-ADDRESS-HIDDEN> Fri Mar 22 10:21:35 2002
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Date: Fri Mar 22 10:22:10 2002
X-Original-Date: Fri, 22 Mar 2002 10:20:02 -0800
Status: O

At 07:07 AM 3/22/02, Sandeep Kinra wrote:
>Still no luck!!!  What else I need to do? Any alternative?

Later today I'll be releasing a new version of the code (with an improved 
implementation of "BasicUsageEnvironment") that, I hope, will fix the 
problem that you're seeing.

(Note that when I release the new version, I'll be announcing it only to 
the developers' mailing list (live-devel(at)lists.sourceforge.net), so you 
should join this mailing list.)

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Fri Mar 22 10:22:08 2002
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Date: Fri Mar 22 10:23:02 2002
X-Original-Date: Fri, 22 Mar 2002 10:21:56 -0800
Status: O

At 09:06 AM 3/22/02, Marek Nowacki wrote:
>I get Unhandled exception Error at testMP3Streamer : =
>0XC000005: Access Violation.
>
>Why and how to fix it.

This is odd - you're the second person in two days to see an error like 
this (but the code has not changed in quite a long time).

Later today I'll be releasing a new version of the code that, I hope, will 
fix the problem that you're seeing.

(Note that when I release the new version, I'll be announcing it only to 
the developers' mailing list (live-devel(at)lists.sourceforge.net), so you 
should join this mailing list.)

         Ross.



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Date: Sat Mar 23 00:13:02 2002
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From <EMAIL-ADDRESS-HIDDEN> Sat Mar 23 03:49:20 2002
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Subject: [Live-devel] New library version - fixes Windows read handling problem
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Date: Sat Mar 23 03:50:01 2002
X-Original-Date: Sat, 23 Mar 2002 03:43:24 -0800
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I have installed a new version (2002.03.23) of the library that 
reimplements the handling of socket read events within "BasicTaskScheduler" 
(in the "BasicUsageEnvironment" library), so that it no longer assumes the 
Unix definition of the "FD_SETSIZE" macro.

(Thanks to Vesselin Kostadinov for pointing out this problem.)

This may overcome the crashes that people have recently been seeing on Windows.

	Ross.



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Date: Sun Mar 24 20:14:01 2002
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Ross,

I suggest the following updates:


1. As Borland already has Boolean type leaving the one in the library
creates ambiguity. (Boolean.hh.diff)


2. Class BasicTaskScheduler will be more reusable if you move most of the
processing from  blockMyself into a separate public method (suggested name:
singleStep()). This way the main program could actually do some work and
call SingleStep() when idle (which usually is often enough), rather than
blocking it's execution in BasicTaskScheduler::blockMyself. Method
blockMyself could still be there in the form of:

void BasicTaskScheduler::blockMyself(char* watchVariable) {
  while (1) {
    if (watchVariable != NULL && *watchVariable != 0) break;
    singleStep();
  }
}

(see file BasicTaskScheduler.cpp.diff) 


3. GroupSock - this was an old discussion from December last year. I would
prefer the call to readSocket to be with a pointer to a timeval structure
with values = 0 (GroupSock.cpp.diff)


4. Compiler warnings:

4.1. RTCP.hh.diff: I moved reapOldMembers() outside the class definition and
got rid of warning "Functions containing do are not expanded inline"

4.2. Lots of other warning busters

4.3. If anyone wants to get rid of the warnings in rtcp_from_spec he/she
should define USE_PROTOTYPE while compiling the library




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Subject: [Live-devel] testMPEGVideoStreaming Help -- Important
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Hi,
     Thanks Ross, for fixing the FD_SETSIZE problem in
BasicTaskSchedular. Now I can execute the testMPEGVideoStreaming without =
getting Illegal Exception error.

Okay, can you tell me the order to execute the programs to view the =
test.mpg file.

First I execute the testMPEGVideoReciever, then I execute the =
testMPEGVideoStreaming, then What?=20

Does it means that next I should open Quicktime player? If yes what =
rtp:// I should put in?

Thanks,

P.S.  Running in window 98.

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charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2713.1100" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Hi,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp;&nbsp;&nbsp; Thanks Ross, =
for fixing=20
the FD_SETSIZE problem in</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>BasicTaskSchedular. Now I can execute =
the=20
testMPEGVideoStreaming without getting Illegal Exception =
error.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Okay, can you tell me the order to =
execute the=20
programs to view the test.mpg file.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>First I execute the =
testMPEGVideoReciever, then I=20
execute the testMPEGVideoStreaming, then What? </FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Does it means that next I should open =
Quicktime=20
player? If yes what rtp:// I should put in?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Thanks,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>P.S.&nbsp; Running in window=20
98.</FONT></DIV></BODY></HTML>

------=_NextPart_000_000D_01C1D3C4.68778A80--



From <EMAIL-ADDRESS-HIDDEN> Mon Mar 25 10:28:26 2002
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Date: Mon Mar 25 10:29:06 2002
X-Original-Date: Mon, 25 Mar 2002 10:28:22 -0800
Status: O

At 04:15 AM 3/25/02, Sandeep Kinra wrote:
>Okay, can you tell me the order to execute the programs to view the 
>test.mpg file.
>
>First I execute the testMPEGVideoReciever, then I execute the 
>testMPEGVideoStreaming, then What?
>
>Does it means that next I should open Quicktime player? If yes what rtp:// 
>I should put in?

If you have QuickTime Player (version 5 or later), then you don't need 
"testMPEGVideoReceiver".  Instead, you can  just run QuickTime Player, and 
have it open the file "testMPEGVideo.sdp" (or "testMPEGAudioVideo" if 
you're running "testMPEGAudioVideoStreamer".  (You will find a copy of this 
in the "testProgs" directory.)

You can run "testMPEGVideoStreamer" (or "testMPEGAudioVideoStreamer") and 
"QuickTime Player" in either order.

         Ross.



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Date: Mon Mar 25 10:32:54 2002
X-Original-Date: Mon, 25 Mar 2002 10:31:11 -0800
Status: O

Vesselin,

Thanks for the suggestions.  I'll try to include as many of these as I can 
in the next release.

	Ross.



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Subject: [Live-devel] RE:testMPEGVideoStreaming Help -- Important
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Date: Mon Mar 25 21:22:01 2002
X-Original-Date: Mon, 25 Mar 2002 23:20:46 -0600
Status: O

Hi Ross,
    I tried your way. I ran testMPEGVideoStreamer without any changes.
Then I ran Quicktime(Version 5) with testMPEGVideo.sdp.

I get connecting.. then ready but no video except Big Q sign.

When I ran testMPEGVideoStreamer, it ran quickly and then it was finished 
pretty quickly. But no video.


Another question: in testMPEGVideo.sdp file:

v=0
o=- 49451 3 IN IP4 127.0.0.1
s=Test MPEG Video session
i=Parameters for the session streamed by "testMPEGVideoStreamer"
t=0 0
a=tool:testMPEGVideoStreamer
a=type:broadcast
m=video 8888 RTP/AVP 32
c=IN IP4 239.255.42.42/127

what is the o represent compare to c?  If I decide to use unicast, for
instance, 10.120.1.254, I have to make changes in testMPEGVideoStreaming
to replace IP of 239.255.42.42 with 10.120.1.254.

And also replace the IP address in testMPEGVideo.sdp of value c with
10.120.1.254. What about IP address (127.0.0.1) of value?

Thanks,

Sandeep

P.S. My regular E-mail address is done, that is why I am using this E-mail 
address.

-----------------------
FROM: Ross FinlaysonDATE: 03/25/2002 10:28:22SUBJECT: RE:  [Live-devel] 
testMPEGVideoStreaming Help -- Important At 04:15 AM 3/25/02, Sandeep Kinra 
wrote:
>Okay, can you tell me the order to execute the programs to view the 
>test.mpg file.
>
>First I execute the testMPEGVideoReciever, then I execute the 
>testMPEGVideoStreaming, then What?
>
>Does it means that next I should open Quicktime player? If yes what rtp:// 
>I should put in?

If you have QuickTime Player (version 5 or later), then you don't need
"testMPEGVideoReceiver".  Instead, you can  just run QuickTime Player, and
have it open the file "testMPEGVideo.sdp" (or "testMPEGAudioVideo" if
you're running "testMPEGAudioVideoStreamer".  (You will find a copy of this
in the "testProgs" directory.)

You can run "testMPEGVideoStreamer" (or "testMPEGAudioVideoStreamer") and
"QuickTime Player" in either order.

         Ross.



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Date: Mon Mar 25 21:50:02 2002
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Status: O

At 09:20 PM 3/25/02, Sandeep Kinra wrote:
>When I ran testMPEGVideoStreamer, it ran quickly and then it was finished 
>pretty quickly. But no video.

This was probably because your input file ("test.mpg") was a MPEG Program 
Stream, instead of a MPEG Video Elementary Stream.

Instead, try streaming using "testMPEGAudioVideoStreamer" (rather than 
"testMPEGVideoStreamer"), and have QuickTime Player open the file 
"testMPEGAudioVideo.sdp" (rather than "testMPEGVideo.sdp").

>If I decide to use unicast

You should first make sure that the code is working OK for multicast, 
before you start to worry about unicast.  (By the way, you only need to use 
unicast if you are streaming over a network that does not support 
multicast.  Note that most local-area networks support multicast.)

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Wed Mar 27 00:49:17 2002
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--0-379438746-1017218957=:6090
Content-Type: text/plain; charset=iso-8859-1
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 Kinra,
I can't explain the difference between the "o" line and the "c". But I think that the "c" line indicates the ip adr where the client machine must look for information coming from the server. It can be either the server ip address or the ip multicast address. Please refer to the RTP rfc for more information about the Session Description Protocol (SDP).
To use unicast solution, in the "testMP3Streamer.cpp"  change multicast address to the address of the client machine and rebuild. In the other side (the client machine) change "c" line to the address of the server in the sdp file (you have to bring the sdp file to the client machine).
first run the testMP3Streamer, then open the SDP file with the appropriate soft such us winamp (you must download the plugin "in_rtp.dll" first from http://www.live.com/multikit/winamp-plugin.html) or quicktime.
Note that RealPlayer is not adviced for unicast stream.
I hope this help
GAZDAR Achraf 



---------------------------------
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--0-379438746-1017218957=:6090
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<P> 
<P>&nbsp;Kinra,
<P>I&nbsp;can't explain the difference between the "o" line and the "c".&nbsp;But I think that the "c" line indicates the&nbsp;ip&nbsp;adr where&nbsp;the client machine&nbsp;must look for information coming from the server. It can be either the server ip address or the ip multicast address. Please refer to the RTP rfc for more information about the Session Description Protocol (SDP).
<P>To&nbsp;use unicast solution,&nbsp;in the&nbsp;"testMP3Streamer.cpp"&nbsp; change multicast address to the address of the client machine and rebuild. In the other side (the client machine)&nbsp;change "c" line to the address of the server in the sdp file (you have to bring the sdp file to the client machine).
<P>first run the testMP3Streamer, then open the SDP file&nbsp;with the appropriate soft such us winamp (you must download the plugin "in_rtp.dll" first from <A href="http://www.live.com/multikit/winamp-plugin.html">http://www.live.com/multikit/winamp-plugin.html</A>)&nbsp;or quicktime.
<P>Note that&nbsp;RealPlayer is not adviced for unicast stream.
<P>I hope this help
<P>GAZDAR Achraf&nbsp;</P><p><br><hr size=1><a href="http://fr.mail.yahoo.com">Yahoo! Mail</a> -- Une adresse (at)yahoo.fr gratuite et en français !<br>
--0-379438746-1017218957=:6090--


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Date: Wed Mar 27 07:02:08 2002
X-Original-Date: Wed, 27 Mar 2002 16:01:25 +0100 (CET)
Status: O

1)how can i measure the size of "frame"and  is it fixe
for MPEG ES,
2) what is the difference between MPEG Video ES and
MPEG 1 video and  MPEG 2 video
3)the number of slices in MPEPG Video dependes on
"test.mpg" ?
4) what is the signification of 'frame';is it picture
or slice or ... and the frame includes the header
(such a picture header) or it is only data

thanks 





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Date: Wed Mar 27 07:07:02 2002
X-Original-Date: Wed, 27 Mar 2002 16:06:07 +0100 (CET)
Status: O

how can i measure the bitrate of testMPEGVideoStreamer
haw can i control this bitrate

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Date: Wed Mar 27 13:53:17 2002
X-Original-Date: Wed, 27 Mar 2002 13:51:43 -0800
Status: O

At 07:06 AM 3/27/02, alouini khalif wrote:
>haw can i control this bitrate

The bitrate of a MPEG video (or audio) file is an attribute of the file 
itself.  That is, the bitrate of the MPEG file was determined when it was 
created (encoded).  "testMPEGVideoStreamer" just streams at whatever 
bitrate the file was created at.

>how can i measure the bitrate of testMPEGVideoStreamer

If you comment out the
         #ifdef DEBUG
         #endif
code that starts at line 374 of "liveMedia/MPEGVideoStreamFramer.cpp", you 
will be able to see the bit (and frame) rate of the input MPEG stream.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Wed Mar 27 14:02:54 2002
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Date: Wed Mar 27 14:03:18 2002
X-Original-Date: Wed, 27 Mar 2002 14:01:44 -0800
Status: O

At 07:01 AM 3/27/02, alouini khalif wrote:
>1)how can i measure the size of "frame"and  is it fixe
>for MPEG ES,
>2) what is the difference between MPEG Video ES and
>MPEG 1 video and  MPEG 2 video

These questions relate to the MPEG video format, so you should refer to the 
ISO MPEG specification.

>3)the number of slices in MPEPG Video dependes on
>"test.mpg" ?

Yes, the number of slices in each MPEG video frame depends on the encoded 
video.

>4) what is the signification of 'frame';is it picture
>or slice or ... and the frame includes the header
>(such a picture header) or it is only data

This is rather complex.  In the "LIVE.COM Streaming Media" code, the term 
'frame' refers to a discrete unit of data that is packed into a RTP 
packet.  So, for streaming MPEG video over RTP (which uses the 
packetization defined in RFC 2250, section 3.1), a 'frame' is a "video 
sequence header", a "GOP header", a "picture header", or a "slice".

However, when people generally refer to MPEG video (outside the context of 
the "LIVE.COM Streaming Media" implementation), a 'frame' means an entire 
'picture' (which consists of a "picture header", plus several "slices").

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Thu Mar 28 02:57:50 2002
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Subject: [Live-devel] clock sync
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Date: Thu Mar 28 02:58:23 2002
X-Original-Date: Thu, 28 Mar 2002 11:46:11 +0100
Status: O

This is a multipart message in MIME format.
--=_alternative 003C0210C1256B8A_=
Content-Type: text/plain; charset="us-ascii"

Hi Ross et al.,

how is clock synchronisation between the sender and receiver done when I 
use the TestMpegVideoStreamer application to stream mpeg-video to a 
receiver (i.e. apple quicktime)? Any idea? 

regards,

luc.
--=_alternative 003C0210C1256B8A_=
Content-Type: text/html; charset="us-ascii"


<br><font size=2 face="sans-serif">Hi Ross et al.,</font>
<br>
<br><font size=2 face="sans-serif">how is clock synchronisation between the sender and receiver done when I use the TestMpegVideoStreamer application to stream mpeg-video to a receiver (i.e. apple quicktime)? Any idea? </font>
<br>
<br><font size=2 face="sans-serif">regards,</font>
<br>
<br><font size=2 face="sans-serif">luc.</font>
--=_alternative 003C0210C1256B8A_=--


From <EMAIL-ADDRESS-HIDDEN> Thu Mar 28 12:14:25 2002
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Date: Thu Mar 28 12:22:50 2002
X-Original-Date: Thu, 28 Mar 2002 12:10:28 -0800
Status: O

At 02:46 AM 3/28/02, you wrote:
>how is clock synchronisation between the sender and receiver done when I 
>use the TestMpegVideoStreamer application to stream mpeg-video to a 
>receiver (i.e. apple quicktime)? Any idea?

The RTP timestamp - present in each streamed RTP packet - is used for time 
synchronization.

Also, if there are separate audio and video streams, then synchronization 
*between* these streams is implemented using RTCP "Sender Report" packets, 
which map each stream's RTP timestamp to NTP (i.e., 'wall clock') time.

For more information about this, see the latest version of the RTP/.RTCP 
specification: 
<http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-new-11.txt>.

The LIVE.COM Streaming Media libraries implement (correctly, I hope) RTP 
timestamps, and RTCP Sender Reports.  Also, QuickTime player should be 
handling these properly.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Sat Mar 30 02:04:34 2002
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Subject: [Live-devel] problem of multicast
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Date: Sat Mar 30 02:05:07 2002
X-Original-Date: Sat, 30 Mar 2002 11:04:32 +0100 (CET)
Status: O

hello

the multicast streaming works perfectly on windoze
envoronment, but when i run the testMPEGVideoStreamer
on linux environment,it works but no packet is
transiting into the network, and there were some meg:
"unable to join ...(IP_MULTICAST..)"
:no such device"

___________________________________________________________
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From <EMAIL-ADDRESS-HIDDEN> Sat Mar 30 02:38:35 2002
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Date: Sat Mar 30 02:39:16 2002
X-Original-Date: Sat, 30 Mar 2002 02:38:30 -0800
Status: O

At 02:04 AM 3/30/02, you wrote:
>the multicast streaming works perfectly on windoze
>envoronment, but when i run the testMPEGVideoStreamer
>on linux environment,it works but no packet is
>transiting into the network, and there were some meg:
>"unable to join ...(IP_MULTICAST..)"
>:no such device"

It may be that your Linux kernel is not configured properly for IP 
multicast.  (Check that "CONFIG_IP_MULTICAST" is set in your kernel 
configuration.)

Another possibility is that your Linux computer's routing table has not 
been set up correctly - in particular, it might not have a default route 
for outgoing multicast packets (224.0.0/4)

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Mon Apr 01 23:23:59 2002
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Subject: Re: [Live-devel] problem:with testMPEGVideoReceiver
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Date: Mon Apr  1 23:34:13 2002
X-Original-Date: Fri, 29 Mar 2002 14:56:28 +0100
Status: O


    rf> Almost every local-area network (e.g., Ethernet) supports
    rf> multicast 'as is'.  However, once you start linking multiple
    rf> local-area networks together (using routers), you will need to
    rf> make sure that these routers also have multicast routing
    rf> turned on.

As a  side  note, if  you need  multicast  aware  routers for  testing
purposes,   two GNU/Linux boxes   with mrouted  on  each  one  is your
friend. Mrouted is very  easy to set up,  and establishes a tunnel (IP
in IP) to forward multicast packets from one subnet to another.

I used it while playing  with vic and rat  (two projects from UCL) and
it worked really smoothly.

I think  iproute2   (the next  generation  routing package  for  Linux
kernels) can be used to established multicast-unicast tunnels, but I
have yet to try.

regards,

andrea glorioso
--
Love is blindness I don't want to see		    andrea glorioso
Won't you wrap the night around me		   
In a parked car in a crowded street		 www.gnutemberg.org
You see your love made complete			sama(at)perchetopi.org


From <EMAIL-ADDRESS-HIDDEN> Tue Apr 02 10:39:44 2002
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Date: Tue Apr  2 10:51:59 2002
X-Original-Date: Tue, 2 Apr 2002 20:40:28 +0200
Status: O

Hello!
 
Do you know if openRTSP is able to set up a stream 
based on rtp in unicast mode ?
 
I'm trying to use openRTSP as a client application that
is able to connect to a  remote video server and then cause
the begin of  streaming.
I need also to receive the sdp file  for the video that is going to
be streamed in order to do some changes, like the substitution
of the addresses of media contents. 
(the parsed sdp file will be then forwarded by email to a group of users)
 
(the remote sever cannot stream using multicast address because
between the server and the client there is a Premium IP network, that is
a network QoS enabled, ad multicasting is not admitted)
 
Can I receive the sdp file using the openRTSP application?
 
Does openRTSP is able to start the remote streamer for a unicast 
connection?
 
Is openRTSP compatible with darwin(quicktime) streaming server ?
 
thank you!

domenico

__________________________________________


Fai i tuoi acquisti su www.kwshopping.it



From <EMAIL-ADDRESS-HIDDEN> Tue Apr 02 11:59:29 2002
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Date: Tue Apr  2 12:15:20 2002
X-Original-Date: Tue, 02 Apr 2002 11:57:47 -0800
Status: O

At 10:40 AM 4/2/02, you wrote:
>Do you know if openRTSP is able to set up a stream
>based on rtp in unicast mode ?

Domenico,

Yes, "openRTSP" can open and play multicast or unicast streams (from a 
"rtsp://..." URL)

>  I'm trying to use openRTSP as a client application that
>is able to connect to a  remote video server and then cause
>the begin of  streaming.
>I need also to receive the sdp file  for the video that is going to
>be streamed in order to do some changes, like the substitution
>of the addresses of media contents.

>Can I receive the sdp file using the openRTSP application?

Yes, openRTSP - as part of its normal operation - will send a "DESCRIBE" 
command to the RTSP server, and, as a result, get back a SDP description.

>  Does openRTSP is able to start the remote streamer for a unicast
>connection?

Yes.

>  Is openRTSP compatible with darwin(quicktime) streaming server ?

Yes.

For more information about "openRTSP", see 
<http://www.live.com/openRTSP/>.  Note that prebuilt binarry versions of 
the application are available, so to learn more about the application, just 
go ahead and try running it.

         Ross.



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This is a multi-part message in MIME format.

--Boundary_(ID_a2MM2eMLsYwxZX0g6kXM0g)
Content-type: text/plain; charset=iso-8859-1
Content-transfer-encoding: 7BIT

Hello !!

Could anyone please suggest me an rtsp url for testing the 
openRTSP programe.

Thanks in advance 
Niranjan

--Boundary_(ID_a2MM2eMLsYwxZX0g6kXM0g)
Content-type: text/html; charset=iso-8859-1
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content="text/html; charset=iso-8859-1" http-equiv=Content-Type>
<META content="MSHTML 5.00.3103.1000" name=GENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=#ffffff>
<DIV><FONT face=Arial size=2>Hello !!</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=Arial size=2>Could anyone please suggest me an rtsp url for 
testing the </FONT></DIV>
<DIV><FONT face=Arial size=2>openRTSP programe.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=Arial size=2>Thanks in advance </FONT></DIV>
<DIV><FONT face=Arial size=2>Niranjan</FONT></DIV></BODY></HTML>

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Date: Wed Apr  3 08:49:41 2002
X-Original-Date: Wed, 3 Apr 2002 18:34:13 +0200 (CEST)
Status: O

preferedPacketSize=1000 byte
maxPacketSize=1450 byte
i don't understand why in 
OutPacketBuffer::OutPacketBuffer(1000,1450)
flimit(20*1450)
what is the meanig of '20' 
it mean the the max size of the buffer is 20*1450
byte?


thanks


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Date: Wed Apr  3 09:00:37 2002
X-Original-Date: Wed, 3 Apr 2002 18:47:38 +0200 (CEST)
Status: O

is there any relation between 
the fbuf and the bank 
i find that the only method that reads from the source
(test.mpg) is fread
(ByteStreamFileSource::DoGetNextframe())
so when we put something into the bank (StreamParser)

what i understand that the buffer is used to packing
as many complete frame from the source,and the parser
parses frame such
as(Video_Sequrnce_Header,Picture_Header,Gop_header)
i don't understand when we put the new frame into the
bank (Streamparser)to be parsed

thanks 


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Date: Wed Apr  3 09:04:57 2002
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Status: O

i'm trying to analyse testMPEGVideoStreamer
FramedSource::doGetNextFrame() (virtual method, called
by getnextFrame)
there is two implementations of this method in
ByteStreamFileSource & MPEGVideoStreamFramer
i don't understand when we call this or that
implementation




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From <EMAIL-ADDRESS-HIDDEN> Wed Apr 03 08:25:21 2002
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Date: Wed Apr  3 09:23:20 2002
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Status: O

rtpHdr=0x80000000
it means
Version=2|P=0| X=0|CC=0 

thanks

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From <EMAIL-ADDRESS-HIDDEN> Wed Apr 03 10:42:58 2002
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Date: Wed Apr  3 10:56:06 2002
X-Original-Date: Wed, 03 Apr 2002 10:42:46 -0800
Status: O

At 04:38 AM 4/3/02, you wrote:
>Hello !!
>
>Could anyone please suggest me an rtsp url for testing the
>openRTSP programe.

Niranjan,

You can try
         rtsp://media.schreiner.edu/realmp3.mp3
This will give you a MP3 stream.

         Ross.



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Date: Wed Apr  3 13:53:03 2002
X-Original-Date: Wed, 03 Apr 2002 13:41:29 -0800
Status: O

At 08:34 AM 4/3/02, you wrote:
>preferedPacketSize=1000 byte
>maxPacketSize=1450 byte
>i don't understand why in
>OutPacketBuffer::OutPacketBuffer(1000,1450)
>flimit(20*1450)
>what is the meanig of '20'
>it mean the the max size of the buffer is 20*1450
>byte?

This buffer is used (by RTP/RTCP sinks) to store a frame of data before it 
gets sent out over the network.  It needs to be larger than the (preferred 
or maximum) network packet size, because some codecs (in particular, MPEG 
video) generate frames that are larger than this, so they need to be split 
into multiple RTP packets for output.  (Any data that's not sent in the 
next outgoing RTP packet gets 'saved' in 'overflow data'.)

The choice of '20' for this buffer size (20*1450) was somewhat arbitrary, 
but it was chosen to take into account the possibility of very large MPEG 
video slices.

(It's unfortunate that a component has to allocate sufficient buffer space 
to allow for the largest possible input frame that it might end up reading; 
it would have been better if the input source could determine this.  This 
is probably a design flaw - one that I will try to fix sometime in the future.)

         Ross.



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Date: Wed Apr  3 14:04:19 2002
X-Original-Date: Wed, 03 Apr 2002 13:55:07 -0800
Status: O

At 08:38 AM 4/3/02, you wrote:
>i'm trying to analyse testMPEGVideoStreamer
>FramedSource::doGetNextFrame() (virtual method, called
>by getnextFrame)
>there is two implementations of this method in
>ByteStreamFileSource & MPEGVideoStreamFramer

That's why "FramedSource::doGetNextFrame()" is a pure virtual member 
function.  Each subclass of "FramedSource" redefines this in its own way 
(to generate its data).

>i don't understand when we call this or that implementation

If you're calling "getNextFrame()" on a "ByteStreamFileSource" object, 
you'll end up calling "ByteStreamFileSource::doGetNextFrame()".  If you're 
calling "getNextFrame()" on a "MPEGVideoStreamFramer" object, you'll end up 
calling "MPEGVideoStreamFramer::doGetNextFrame()".  This is just normal C++ 
object-oriented programming (using virtual functions).

Note that the "testMPEGVideoStreamer" application is built using several 
different 'source' objects (all subclasses of "FramedSource") chained together:
- a "MPEGVideoStreamFramer", which reads from
- a video Elementary Stream, which is implemented as either
         - a "ByteStreamFileSource"
                 (if the input file is a MPEG Elementary Stream), or
         - a stream read from a "MPEGDemux"
                 (if the input file is a MPEG Program Stream)

     Ross.



From <EMAIL-ADDRESS-HIDDEN> Wed Apr 03 14:06:49 2002
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Date: Wed Apr  3 14:16:04 2002
X-Original-Date: Wed, 03 Apr 2002 14:05:09 -0800
Status: O

At 08:47 AM 4/3/02, you wrote:
>is there any relation between
>the fbuf and the bank
>i find that the only method that reads from the source
>(test.mpg) is fread
>(ByteStreamFileSource::DoGetNextframe())
>so when we put something into the bank (StreamParser)

These are different buffers, with different purposes.  In 
"testMPEGVideoStreamer", the "MPEGVideoStreamParser" (a subclass of 
"StreamParser", used to implement "MPEGVideoStreamFramer") reads incoming 
MPEG video data as a byte stream into a 'bank' buffer.  (This data is read 
unstructured - not corresponding to any header, slice etc. 
boundaries.)  The "MPEGVideoStreamParser" then parses this data into 
discrete headers and slices, for delivery to its client - which in this 
case is a "MultiFramedRTPSink".  This client uses its own buffer (a "fBuf") 
to store these frames, while it prepares to send them using RTP.  (Note my 
earlier message about this.)

         Ross.



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Subject: [Live-devel] livemedia errors.....
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Date: Thu Apr  4 00:36:13 2002
X-Original-Date: Thu, 4 Apr 2002 13:52:39 +0530
Status: O

Hi Listers,

I was able to compile the BasicUsageEnvironment/UsageEnvironment/groupsock
libraries successfully but I am unable to compile the livemedia into a
static library in VC++6.0 environment on win2k server due to these compile
errors.

h:\testlive\livemedia\bytestreammultifilesource.cpp(37) : error C2143:
syntax error : missing ';' before '['
h:\testlive\livemedia\bytestreammultifilesource.cpp(46) : error C2143:
syntax error : missing ';' before '['
h:\testlive\livemedia\bytestreammultifilesource.cpp(61) : error C2664:
'delete' : cannot convert parameter 1 from 'const char *' to 'void *'
        Conversion loses qualifiers


any clues how this can be solved ?

I need to port the code to VC++ environment coz then I can do a DEBUG thru'
the code....

PS : I did a ./genWindowsMakefile on a linux machine but after porting the
source files to Windows2000, the VC++6.0 IDE refuses to import the make
file, secondly I also tried clubbing the Head+win32config+tail makefile but
again the VC++6.0 din't like it. Whereas the same make files work well on
the command prompt if I use NMAKE. Can anyone get me started ?


thanks a lot

Deepesh S Gujarathi
Encodex Technologies (India) Pvt. Ltd.
28, Electronic estate,Pune- 411009
Tel- 91 20 4222 401/402
www.encodexindia.com




From <EMAIL-ADDRESS-HIDDEN> Thu Apr 04 00:53:51 2002
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Date: Thu Apr  4 00:54:04 2002
X-Original-Date: Thu, 04 Apr 2002 00:50:31 -0800
Status: O

The file "ByteStreamMultiFileSource.cpp" is included just as an 
example.  It is not meant to be built as part of the "liveMedia" library, 
and, in fact, does not appear in the Makefile for this library.

The fact that you tried compiling that file on Windows suggests that you 
did not use the proper method (as described in 
<http://live.sourceforge.net/#config-windows>) for creating and using 
"Makefile"s for Windows.  Please create a proper Makefile for this library 
(and the others) before you try to build it.

	Ross.



From <EMAIL-ADDRESS-HIDDEN> Thu Apr 04 04:17:52 2002
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Subject: [Live-devel] Multicast
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Date: Thu Apr  4 04:18:17 2002
X-Original-Date: Thu, 4 Apr 2002 13:20:09 +0100
Status: O

	I'm trying to test the multicast capabilities of
testMPEGAudioVideoStreamer. I've set up a private network, connected to a
public network through a Linux router/gateway who I believe is able to
support multicast (v2.2.18, has IP:multicast enabled and IP: multicast
routing with PIM-SM v1 and v2 support enabled). I'm using Cisco's IP/TV as a
multicast client on the public network, but it's not reeciving the stream. I
try using QuickTime in unicast and it works fine.
	I'm not sure how to start debugging the system, since a number of things
can be happening. I started by checking wether the IGMP message (from the
client) was reaching the router (with tcpdump) and they were not. Did I miss
some configuration step? On the client, or the router?


Thank you all very much,
David Pontes



From <EMAIL-ADDRESS-HIDDEN> Thu Apr 04 04:48:30 2002
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Date: Thu Apr  4 04:49:02 2002
X-Original-Date: Thu, 04 Apr 2002 15:47:21 +0300
Status: O

Could anyone give me the knowledge,guideline,links of how to make a LAN 
multicast enabled with the help of a linux box.
I am having problems.

Thank you,
Tamer

At 15:20 04.04.2002, David Pontes wrote:

>         I'm trying to test the multicast capabilities of
>testMPEGAudioVideoStreamer. I've set up a private network, connected to a
>public network through a Linux router/gateway who I believe is able to
>support multicast (v2.2.18, has IP:multicast enabled and IP: multicast
>routing with PIM-SM v1 and v2 support enabled). I'm using Cisco's IP/TV as a
>multicast client on the public network, but it's not reeciving the stream. I
>try using QuickTime in unicast and it works fine.
>         I'm not sure how to start debugging the system, since a number of 
> things
>can be happening. I started by checking wether the IGMP message (from the
>client) was reaching the router (with tcpdump) and they were not. Did I miss
>some configuration step? On the client, or the router?
>
>
>Thank you all very much,
>David Pontes
>
>
>_______________________________________________
>Live-devel mailing list
>Live-devel(at)lists.sourceforge.net
>https://lists.sourceforge.net/lists/listinfo/live-devel



From <EMAIL-ADDRESS-HIDDEN> Thu Apr 04 13:40:04 2002
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Date: Thu Apr  4 13:42:27 2002
X-Original-Date: Thu, 04 Apr 2002 13:31:50 -0800
Status: O

At 04:20 AM 4/4/02, you wrote:
>         I'm trying to test the multicast capabilities of
>testMPEGAudioVideoStreamer. I've set up a private network, connected to a
>public network through a Linux router/gateway who I believe is able to
>support multicast (v2.2.18, has IP:multicast enabled and IP: multicast
>routing with PIM-SM v1 and v2 support enabled). I'm using Cisco's IP/TV as a
>multicast client on the public network, but it's not reeciving the stream. I
>try using QuickTime in unicast and it works fine.
>         I'm not sure how to start debugging the system, since a number of 
> things
>can be happening. I started by checking wether the IGMP message (from the
>client) was reaching the router (with tcpdump) and they were not. Did I miss
>some configuration step? On the client, or the router?

David,

For questions like this (about how to get IP multicast working on a 
network), I suggest a more general IP multicast-related mailing list, such 
as "ipmulticast(at)cloud.lyris.net".

First, though, I suggest that you try running IP/TV on the *same LAN* as 
"testMPEGAudioVideoStreamer", to make sure that it really is working 
properly on a multicast network.  Once you have IP/TV working properly in 
this environment, you can then try moving it to your public network.  That 
way, you will be able to tell which problems are really due to multicast 
routing (or the lack thereof).

         Ross.
   



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Date: Thu Apr  4 13:42:28 2002
X-Original-Date: Thu, 04 Apr 2002 13:35:48 -0800
Status: O

At 04:47 AM 4/4/02, you wrote:
>Could anyone give me the knowledge,guideline,links of how to make a LAN 
>multicast enabled with the help of a linux box.

If your LAN is an Ethernet, then it probably already *is* multicast 
enabled.  If it uses an Ethernet switch, then you may need to reconfigure 
it in some way to support multicast (or else multicast may just work by 
default).

If, however, you are using your Linux box as a router to connect together 
two or more different networks, then you can multicast-enable this larger 
network by running "mrouted" 
<ftp://parcftp.xerox.com/pub/net-research/ipmulti/beta-test/> on the Linux box.

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Thu Apr 04 13:49:37 2002
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Date: Thu Apr  4 14:00:58 2002
X-Original-Date: Thu, 04 Apr 2002 13:45:00 -0800
Status: O

At 10:34 PM 4/3/02, you wrote:
>The RTSP url was very helpfull for me in running the testprogram openRTSP.
>
>1>Could u give me url for a video stream(MPEG 2) type.

I don't know of one.  However, you can set up such a stream yourself by 
running the "testMPEGVideoStreamer" test program (using your own MPEG-2 
video file), and enabling its built-in RTSP server.  (See 
"testMPEGVideoStreamer.cpp" for details on how to do this.)

>2> Is there anyway by which i could store the streaming data locally so as
>to be viewed later.

That's what "openRTSP" does already.  It generates files (named "audio-..." 
and "video-...") containing the data from each incoming stream.  You can 
rename these files with an appropriate filename extension (e.g., ".mp3" for 
MP3 audio, or ".mpg" for MPEG video), and you should be able to view or 
listen to them.

>2>And also could u tell me is there any way to check the in comimg streaming
>packets from the server.I haved used the option -n but does not show the
>streaming pckts.For this should i change anything in the code.

If you're only interested in receiving an audio (or video) stream, you can 
use the "-a" (or "-v") option, and pipe openRTSP's stdout through a program 
of your own that can do whatever it likes with the stream data.

Alternatively, you could modify the "FileSink::afterGettingFrame()" 
function, or - if you wanted to inspect complete RTP packets - the 
"MultiFramedRTPSource::networkReadHandler()" function.  (Both of these can 
be found in the "liveMedia" library.)

         Ross.




From <EMAIL-ADDRESS-HIDDEN> Thu Apr 04 10:37:38 2002
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Subject: [Live-devel] Multicast - problem solved
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Date: Thu Apr  4 16:33:20 2002
X-Original-Date: Thu, 4 Apr 2002 19:40:09 +0100
Status: O

	Ok, I got a tip from someone and they solved the problem. The main link is:

http://jukie.net/~bart/multicast/Linux-Mrouted-MiniHOWTO.html

	(I am in no way connected to the people responsible for the above link).

Thanks anyway, and hope this helps
David Pontes



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Date: Thu Apr  4 18:08:25 2002
X-Original-Date: Thu, 4 Apr 2002 22:11:09 +0200
Status: O

hello!
 
does someone know a rtsp url  that stream video + audio in unicast?
 
thank's
 
domenico

__________________________________________


Fai i tuoi acquisti su www.kwshopping.it



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Date: Thu Apr  4 22:10:02 2002
X-Original-Date: Fri, 5 Apr 2002 11:45:14 +0530
Status: O

Hi Ross,

thanks for the valuable tips, now Im able to build all the live.com
libraries as well as the test progs. I went thru' the 'mailing list
archives' and was wondering if I could get a copy of the archives (as a
tarball) via ftp/offline ?

I observed that most of the compilation and linking related problems have
already been solved on the mailing list before, so it would help a lot for
newbies to read the archives and then post messages. It would also save you
a lot of headache answering the same questions over and over.

I really appreciate your patience in answering all the questions (relevant
or not) on the mailing list with a lot of effort.

PS : Is there any good documentation/links someone can suggest for RTSP/RTP
related work ? I have already read the RFC's and QuickTime documentation
available on the apple.com site.

thanks again

Deepesh S Gujarathi
Encodex Technologies (India) Pvt. Ltd.
28, Electronic estate,Pune- 411009
Tel- 91 20 4222 401/402
www.encodexindia.com




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Date: Fri Apr  5 03:23:02 2002
X-Original-Date: Fri, 5 Apr 2002 13:22:43 +0200 (CEST)
Status: O

thanks ROSS,

i want to setup up some information about the
multimedia traffic in the IP OPTION field (exactly 4
bytes) can i change the SOCK-DRAM en SOCK_RAW to
manipulate the ip hearder ,where can i made a change
in the source code of this librairie

thanks you for your help

___________________________________________________________
Do You Yahoo!? -- Une adresse (at)yahoo.fr gratuite et en français !
Yahoo! Mail : http://fr.mail.yahoo.com


From <EMAIL-ADDRESS-HIDDEN> Fri Apr 05 09:59:49 2002
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Date: Fri Apr  5 10:00:07 2002
X-Original-Date: Fri, 05 Apr 2002 09:55:51 -0800
Status: O

At 10:15 PM 4/4/02, you wrote:
>thanks for the valuable tips, now Im able to build all the live.com
>libraries as well as the test progs. I went thru' the 'mailing list
>archives' and was wondering if I could get a copy of the archives (as a
>tarball) via ftp/offline ?

Unfortunately the mailing list - and the mailing list archive - are not 
maintained by us.  Instead, they're maintained by VA Software 
(sourceforge.net, geocrawler.com).   I have asked them if I can get a 
tarball of the archives, but I don't know whether they can do this.

Because of problems like this, sometime in the future we'll probably be 
moving the mailing list so that it's hosted at LIVE.COM itself, rather than 
at SourceForge.net.  When this happens, I hope to make the (future) 
archives easier to access.


>PS : Is there any good documentation/links someone can suggest for RTSP/RTP
>related work ? I have already read the RFC's and QuickTime documentation
>available on the apple.com site.

You can find more information about RTP at 
<http://www.cs.columbia.edu/~hgs/rtp/>, and about RTSP at 
<http://www.rtsp.org/> and <http://www.cs.columbia.edu/~hgs/rtsp/>

         Ross.



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Date: Fri Apr  5 10:00:08 2002
X-Original-Date: Fri, 05 Apr 2002 09:59:34 -0800
Status: O

At 03:22 AM 4/5/02, you wrote:
>i want to setup up some information about the
>multimedia traffic in the IP OPTION field (exactly 4
>bytes) can i change the SOCK-DRAM en SOCK_RAW to
>manipulate the ip hearder ,where can i made a change
>in the source code of this librairie

You could do this in the "setupDatagramSocket()" function in 
"groupsock/GroupsockHelper.cpp".  Note, though, that if you do this, you 
will also need to set up UDP and IP headers (not just RTP headers) in your 
packets whenever you send them, and you'll also need to allow for these 
other headers when you receive packets.

So I really don't recommend that you do this (but if you do, you're on your 
own).

         Ross.



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From <EMAIL-ADDRESS-HIDDEN> Mon Apr 08 03:48:05 2002
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Date: Mon Apr  8 03:49:02 2002
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Status: O

FYI, I have released a new version (2002.04.08) of the "LIVE.COM Streaming 
Media" source code (at <http://live.sourceforge.net/> or 
<http://www.live.com/liveMedia/>) that fixes a bug that could cause the 
"testMPEGAudioVideoStreamer" test program to crash when it tried streaming 
from its file a second time.

I also made some mostly syntactic changes in several files to eiliminate 
unnecessary compiler warnings.  (Thanks to Vesselin K for suggesting these.)

	Ross.



From <EMAIL-ADDRESS-HIDDEN> Mon Apr 08 05:41:23 2002
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Subject: [Live-devel] SocketNum()
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Date: Mon Apr  8 05:42:02 2002
X-Original-Date: Mon, 8 Apr 2002 15:41:14 +0300
Status: O

Hi!

I'm wondering where function socketNum() is defined? Groupsock  object can
call it and I assume that it return some socket's number if all succeed
else -1. That's right? Ploblem is that my RTSPserver crash when socketNum()
return -1 in RTCP.cpp (Ln 142) . Is there line really something wrong
because Armstrong wrote 12/25/2001:

> It looks like something wrong in this line.
> In RTCP.cpp (Ln 142)
>  // Arrange to handle incoming reports from others:
> env.taskScheduler().turnOnBackgroundReadHandling(RTCPgs->socketNum(),
> (TaskScheduler::BackgroundHandlerProc*)&incomingReportHandler,this);

 >            Armstrong

Any suggestions what's wrong?

Thanks
Tomi




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Date: Mon Apr  8 10:29:15 2002
X-Original-Date: Mon, 08 Apr 2002 10:21:23 -0700
Status: O

At 05:41 AM 4/8/02, you wrote:
>I'm wondering where function socketNum() is defined?

In the class "Socket", which is defined in 
"groupsock/include/NetInterface.hh".  It is just a regular socket number, 
which - at the lowest level of the code - is created as the result of a 
socket() call.

>Groupsock object can
>call it and I assume that it return some socket's number if all succeed
>else -1. That's right? Ploblem is that my RTSPserver crash when socketNum()
>return -1 in RTCP.cpp (Ln 142) .

You can probably avoid this crash by upgrading to the latest version of the 
code (because the code now explicitly checks for a bad socket number (<0) 
in "turnOnBackgroundReadHandling()".)

However, your real problem is that your RTCP 'groupsock' is not being 
created correctly, because a socket is not being created.  You can check 
for this by adding code like the following:

Groupsock rtcpGroupsock(*env, ... ); // this is where you create the 
'groupsock'
if (rtcpGroupsock.socketNum() < 0) {
         fprintf(stderr, "Failed to create RTCP groupsock: %s\n",
                 env->getResultMsg());
}

         Ross.



From <EMAIL-ADDRESS-HIDDEN> Mon Apr 8 12:40:31 2002
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X-Original-Date: Mon, 08 Apr 2002 12:30:54 -0700
Date: Mon, 08 Apr 2002 12:30:54 -0700

FYI,

If you're running the "testMPEGAudioVideoStreamer" test program, and are 
using the SDP file "testMPEGAudioVideo.sdp" to receive and play it, then 
please note that this SDP file has a bug.

The line
	m=audio 6666 RTP/AVP 32
should be
	m=audio 6666 RTP/AVP 14

(Surprisingly, the bug didn't stop QuickTime Player from playing the stream 
OK.)

	Ross.



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From <EMAIL-ADDRESS-HIDDEN> Mon Apr 8 14:16:09 2002
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X-Original-Date: Mon, 08 Apr 2002 14:14:29 -0700
Date: Mon, 08 Apr 2002 14:14:29 -0700

FYI, the archives of this mailing list are now online as a single text file:
	<http://www.live.com/liveMedia/live-devel.mbx>
You may find this useful for searching.  (Note that email addresses have 
been hidden, to protect against spam email harvesters.)

The archives are still online as
	<http://www.geocrawler.com/lists/3/SourceForge/6637/0/>
but this only allows you to look at one message at a time.

	Ross.


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From <EMAIL-ADDRESS-HIDDEN> Tue Apr 9 04:19:24 2002
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X-Original-Date: Tue, 9 Apr 2002 13:12:56 +0200 (CEST)
Date: Tue, 9 Apr 2002 13:12:56 +0200 (CEST)

thanks Ross,
what is the meanig of thesevariables:
NumFrameByteesToUse
FnumframesUsedSoFor

thank you in advance



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X-Original-Date: Tue, 9 Apr 2002 13:16:30 +0200 (CEST)
Date: Tue, 9 Apr 2002 13:16:30 +0200 (CEST)

thanks Ross,
In this librairie the RTP packet at one time (before
sending it into the network) contains only one slice
or a part of slice(frame thats was fragmented )
thank you in advance



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From <EMAIL-ADDRESS-HIDDEN> Tue Apr 9 04:28:07 2002
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X-Original-Date: Tue, 9 Apr 2002 13:26:09 +0200 (CEST)
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thanks Ross,

When we don'tallow frame to appear after last fragment
(the meaning of allowOtherFramesAfterLastFragment and
when it is modified)
when a frame can appear after packet start (the
meaning of FramecanappearAfterPacketStart and when it
is modified)

what is the meaning of
FPreviousFrameEndedFragmentation and when it is
modified


thank you in advance



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From <EMAIL-ADDRESS-HIDDEN> Tue Apr 9 19:37:46 2002
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X-Original-Date: Tue, 09 Apr 2002 19:34:25 -0700
Date: Tue, 09 Apr 2002 19:34:25 -0700

FYI, I have released a patch for the Open Source "mplayer" media player 
(for Linux or FreeBSD), that allows it to receive and play RTP audio/video 
streams, using the "LIVE.COM Streaming Media" libraries.

For more information, please see
         <http://www.live.com/mplayer/>

With this patch, "mplayer" can now receive and play (unicast or multicast) 
MPEG-1 or MPEG-2 RTP streams, using a SDP (".sdp") file as input.  (Future 
versions of this patch will allow "mplayer" to play RTP streams from a 
"rtsp://" URL, and will also support some additional RTP media types (e.g., 
MPEG-4) that "mplayer" has codecs for.)

         Ross.



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From <EMAIL-ADDRESS-HIDDEN> Fri Apr 12 04:09:48 2002
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Subject: [Live-devel] rtp packet recording....
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X-Original-Date: Fri, 12 Apr 2002 16:38:28 +0530
Date: Fri, 12 Apr 2002 16:38:28 +0530
Status: O

Hi Folks,

I am trying to stream a QuickTime presentation .mov file from Darwin
streaming server4.0 using "LIVE.COM Streaming Media" libraries on a Win2000
client machine over RTSP.

I am trying to PAUSE some of the tracks *selectively* within the
presentation. The PAUSE request looks like " PAUSE
rtsp://192.168.0.118:554/hintedmov.mov/trackID=9 RTSP/1.0" and is sent using
sendRequest(writeBuf) function. Now though I receive a "PAUSE response:
RTSP/1.0 200 OK" from the server, how can I determine whether the server has
actually paused the trackID=9 ?

here's what I have tried until now...
-I downloaded an rtpMonitor from
http://www.web3D.org/WorkingGroups/vrtp/rtpMonitor which is excellent RTP
packet recorder but this tool gives me a global statistics of how many
packet (bytes) in all were received, but that doesn't confirm whether the
server did actually pause the requested track.
-Second I tried to put some printfs in the function
FileSink::continuePlaying() and QuickTimeFileSink::continuePlaying() since
they receive the packets but still it didn't work.

Is there any function in any of the LIVE.COM library where I can set a
printf that informs me when an RTP packet arrives and what track it belongs
to, in real time ? or any other rtp packet recorder that can give me the
track to which a packet belongs ?

even the smallest clue would get me going

Deepesh S Gujarathi
Encodex Technologies (India) Pvt. Ltd.
28, Electronic estate,Pune- 411009
Tel- 91 20 4222 401/402
www.encodexindia.com



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Subject: [Live-devel] Help - Regarding openRTSP
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Hello,

I am streaming a quicktime file (of the form sample.mov provided for testing purpose by apple)using the DarwinStreaming Server.
By using the opentRTSP i am able to get the streamed data
and 2 files are created by openRTSP ie video-X-QT and audio-X-QT .But when i try to open these files in a quicktime player , it says unrecognized format .Am i missing something or should i decode the stream which is received by openRTSP.But i assume that openRTSP receives the encoded data from the server , decodes it and then stores it in the video-X-QT files , am i right , Please guide me.


Thanks in advance,
Niranjan.

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<DIV><FONT face=Arial size=2>Hello,</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=Arial size=2>I am streaming a quicktime file (of the form 
sample.mov provided for testing purpose by apple)using the DarwinStreaming 
Server.</FONT></DIV>
<DIV><FONT face=Arial size=2>By using the opentRTSP i am able to get the 
streamed data</FONT></DIV>
<DIV><FONT face=Arial size=2>and 2 files are created by openRTSP ie video-X-QT 
and audio-X-QT .But when i try to open these files in a quicktime player , it 
says unrecognized format .Am i missing something or should i decode the stream 
which is received by openRTSP.But i assume that openRTSP receives the encoded 
data from the server , decodes it and then stores it in the video-X-QT files , 
am i right , Please guide me.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=Arial size=2>Thanks in advance,</FONT></DIV>
<DIV><FONT face=Arial size=2>Niranjan.</FONT></DIV></FONT></DIV></BODY></HTML>

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From <EMAIL-ADDRESS-HIDDEN> Fri Apr 12 09:43:09 2002
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X-Original-Date: Fri, 12 Apr 2002 09:35:13 -0700
Date: Fri, 12 Apr 2002 09:35:13 -0700
Status: O

At 05:07 AM 4/12/02, you wrote:
>Hello,
>
>I am streaming a quicktime file (of the form sample.mov provided for 
>testing purpose by apple)using the DarwinStreaming Server.
>By using the opentRTSP i am able to get the streamed data
>and 2 files are created by openRTSP ie video-X-QT and audio-X-QT .But when 
>i try to open these files in a quicktime player , it says unrecognized format .

By default, "openRTSP" outputs the 'raw' stream data that it receives 
(i.e., without RTP headers).  In the case of  "X-QT" data, this data is in 
the format described in 
<http://developer.apple.com/quicktime/icefloe/dispatch026.html>.  However, 
this is *not* the QuickTime movie format.  Instead, it describes just one 
track of a possible QuickTime movie.  So, it can't be played in QuickTime 
Player, as is.

To generate QuickTime '.mov' file that can be played (perhaps) in QuickTime 
Player, use the "-q" option to "openRTSP", and redirect 'stdout' to a 
"*.mov" file.  See <http://www.live.com/openRTSP/#quicktime> for more details.

         Ross.


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From <EMAIL-ADDRESS-HIDDEN> Fri Apr 12 09:48:10 2002
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X-Original-Date: Fri, 12 Apr 2002 09:46:01 -0700
Date: Fri, 12 Apr 2002 09:46:01 -0700
Status: O

At 04:12 AM 4/9/02, you wrote:
>thanks Ross,
>what is the meanig of thesevariables:
>NumFrameByteesToUse

"numFrameBytesToUse" is a local variable, used in 
"MultiFramedRTPSink::afterGettingFrame1()", that says how much of the 
newly-arrived data (from the source) will get put into the next packet to 
be sent over the network.  Usually, it is the same as "frameSize", which 
means that all of the incoming data will be put into the next 
frame.  However, it can sometimes be less than "frameSize", if:
         - the outgoing packet already has data in it, and the RTP payload 
format
                 does not allow any new data to be appended to it, or
         - the incoming data is too large to fit in a single outgoing packet,
                 in which case "numFrameBytesToUse" will be less than
                 "frameSize"


>FnumframesUsedSoFor

"fNumFramesUsedSoFar" counts the number of (complete, or partial) incoming 
frames that have been packed into the outgoing packet so far.

         Ross.


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From <EMAIL-ADDRESS-HIDDEN> Fri Apr 12 09:48:12 2002
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X-Original-Date: Fri, 12 Apr 2002 09:46:25 -0700
Date: Fri, 12 Apr 2002 09:46:25 -0700
Status: O

At 04:08 AM 4/12/02, you wrote:
>Is there any function in any of the LIVE.COM library where I can set a
>printf that informs me when an RTP packet arrives and what track it belongs
>to, in real time ?

Yes, you can do this in the
         MultiFramedRTPSource::networkReadHandler() function
(See the file "liveMedia/MultiFramedRTPSource.cpp").  This is where 
incoming RTP packets get handled.

To tell what 'track' the incoming packet belongs to, look at the "source" 
parameter.  If this equals "subsession->rtpSource()" (from the 
"RTSPClient.cpp" code), then it will be the one you want.

         Ross.


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From <EMAIL-ADDRESS-HIDDEN> Fri Apr 12 09:53:37 2002
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Subject: [Live-devel] RealOne player
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X-Original-Date: Fri, 12 Apr 2002 12:51:42 -0400
Date: Fri, 12 Apr 2002 12:51:42 -0400
Status: O

Hi,

While browsing through the list archive I saw some references to maybe
possibly sorta kinda maybe being able to use the RealOne player to play MPEG
multi/unicasts. Can anyone confirm or deny being able to stream to a RealOne
player? I haven't had much luck (using mpeg2 program streams controlled by a
local SDP file).

Thanks!

Tom

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Subject: Re: [Live-devel] About RTP packet
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X-Original-Date: Fri, 12 Apr 2002 10:00:18 -0700
Date: Fri, 12 Apr 2002 10:00:18 -0700
Status: O

At 04:16 AM 4/9/02, you wrote:
>thanks Ross,
>In this librairie the RTP packet at one time (before
>sending it into the network) contains only one slice
>or a part of slice(frame thats was fragmented )

You're asking specifically about the RTP payload format for MPEG video 
frames (as described in RFC 2250, and implemented by 
"MPEGVideoRTPSink/Source.cpp").

Although RFC 2250 allows an outgoing packet to contain more than one 
"slice", in this implementation, an outgoing packet can contain only one 
slice (or fragment of a slice) at a time.  See 
"MPEGVideoRTPSink::frameCanAppearAfterPacketStart().

         Ross.


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X-Original-Date: Fri, 12 Apr 2002 10:08:34 -0700 (PDT)
Date: Fri, 12 Apr 2002 10:08:34 -0700 (PDT)
Status: O

Hi -

Is it possible to mix multiple sources with testMP3Streamer/Receiver? Or 
can two different testMP3Streamers share the same multicast session 
address? What I'd like to do (for example) is have two sources 
broadcasting to a single address, and have a listener take the MP3 streams 
from each source, decode them, and mix them together.

Thanks,

Michael



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X-Original-Date: Fri, 12 Apr 2002 10:20:06 -0700
Date: Fri, 12 Apr 2002 10:20:06 -0700
Status: O

At 09:51 AM 4/12/02, you wrote:
>While browsing through the list archive I saw some references to maybe
>possibly sorta kinda maybe being able to use the RealOne player to play MPEG
>multi/unicasts. Can anyone confirm or deny being able to stream to a RealOne
>player? I haven't had much luck (using mpeg2 program streams controlled by a
>local SDP file).

The free RealOne Player (and the QuickTime Player) can play MPEG-1 streams 
(by reading a SDP file), but I don't think they can play MPEG-2 
streams.  (This is probably because to the increased royalties for MPEG-2 
cf. MPEG-1.)  However, it might be possible to purchase a separate plugin 
that will let you play MPEG-2.  (Rob Lanphier from RealNetworks is on this 
mailing list; I'll let him answer this.)

I believe that Cisco's IP/TV viewer can play MPEG-2/RTP streams using a SDP 
file (although they require that it have filename extension ".sdf" rather 
than ".sdp").

Another alternative is to use the Open Source "mplayer" tool, with my 
recently announced patch for RTP: http://www.live.com/mplayer/

         Ross.


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From <EMAIL-ADDRESS-HIDDEN> Fri Apr 12 10:29:58 2002
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X-Original-Date: Fri, 12 Apr 2002 13:28:26 -0400
Date: Fri, 12 Apr 2002 13:28:26 -0400
Status: O

Hi,

Thanks for the info. I looked at mplayer and while it looks very well done,
I'm looking for something to use on Windows desktops (and it didn't look
like that was an option for mplayer, though maybe a possibility for
xine...).

Also, something else I'd like to get your/the list's opinion on: when
transporting mpeg-2 program or transport streams, why go through the effort
of sending the audio and video as two separate RTP streams? Why not just do
the "section 2" part of RFC 2250 and send a simple sequential stream of data
encapsulated in RTP packets without the hassle of the audio and video
specific headers? The streams already have enough information in them for
the client to synchronize a/v playback and to recover from lost/dropped
packets.

Thanks,

Tom

-----Original Message-----
From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
Sent: Friday, April 12, 2002 1:20 PM
To: <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] RealOne player


At 09:51 AM 4/12/02, you wrote:
>While browsing through the list archive I saw some references to maybe
>possibly sorta kinda maybe being able to use the RealOne player to play
MPEG
>multi/unicasts. Can anyone confirm or deny being able to stream to a
RealOne
>player? I haven't had much luck (using mpeg2 program streams controlled by
a
>local SDP file).

The free RealOne Player (and the QuickTime Player) can play MPEG-1 streams 
(by reading a SDP file), but I don't think they can play MPEG-2 
streams.  (This is probably because to the increased royalties for MPEG-2 
cf. MPEG-1.)  However, it might be possible to purchase a separate plugin 
that will let you play MPEG-2.  (Rob Lanphier from RealNetworks is on this 
mailing list; I'll let him answer this.)

I believe that Cisco's IP/TV viewer can play MPEG-2/RTP streams using a SDP 
file (although they require that it have filename extension ".sdf" rather 
than ".sdp").

Another alternative is to use the Open Source "mplayer" tool, with my 
recently announced patch for RTP: http://www.live.com/mplayer/

         Ross.


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From <EMAIL-ADDRESS-HIDDEN> Fri Apr 12 10:43:00 2002
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X-Original-Date: Fri, 12 Apr 2002 10:43:59 -0700 (Pacific Daylight Time)
Date: Fri, 12 Apr 2002 10:43:59 -0700 (Pacific Daylight Time)
Status: O

On Fri, 12 Apr 2002, Pinckney, Thomas wrote:
> While browsing through the list archive I saw some references to maybe
> possibly sorta kinda maybe being able to use the RealOne player to play MPEG
> multi/unicasts. Can anyone confirm or deny being able to stream to a RealOne
> player? I haven't had much luck (using mpeg2 program streams controlled by a
> local SDP file).

RealOne Player can play MPEG-1 via unicast (RTSP/RTP) or multicast (SDP
file/RTP).  MPEG-2 is not supported at this time.

Hope this helps.

Rob



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X-Original-Date: Fri, 12 Apr 2002 10:44:15 -0700
Date: Fri, 12 Apr 2002 10:44:15 -0700
Status: O

At 10:28 AM 4/12/02, you wrote:
>Also, something else I'd like to get your/the list's opinion on: when
>transporting mpeg-2 program or transport streams, why go through the effort
>of sending the audio and video as two separate RTP streams? Why not just do
>the "section 2" part of RFC 2250 and send a simple sequential stream of data
>encapsulated in RTP packets without the hassle of the audio and video
>specific headers? The streams already have enough information in them for
>the client to synchronize a/v playback and to recover from lost/dropped
>packets.

This is a question that would best be posed on the "avt(at)ietf.org" mailing 
list (the IETF AVT working group, which works on RTP).  So, for a more 
detailed discussion of this topic, I suggest using that mailing list.

In fact, many people - expecially in the video broadcast industry - do send 
MPEG streams over RTP using a single RTP stream, as you suggest, and this 
may make sense if you're in an environment where you know that there's only 
going to be a single receiver, and that you will always want to receive and 
process both audio and video.

However, sending audio and video as separate RTP streams gives you more 
flexibility.  For instance:
         - You can choose to receive only audio, or only video, if you 
wish.  (If the audio and video are sent on different multicast groups, then 
a low-bandwidth receiver could choose just to receive audio, without any 
network overhead of receiving video packets.)
         - If you wish, you can have MPEG audio and a non-MPEG video codec, 
or vice versa.

         Ross.


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X-Original-Date: Fri, 12 Apr 2002 14:14:49 -0700
Date: Fri, 12 Apr 2002 14:14:49 -0700
Status: O

At 10:08 AM 4/12/02, you wrote:
>Is it possible to mix multiple sources with testMP3Streamer/Receiver? Or
>can two different testMP3Streamers share the same multicast session
>address? What I'd like to do (for example) is have two sources
>broadcasting to a single address, and have a listener take the MP3 streams
>from each source, decode them, and mix them together.

Michael,

Yes, it should be possible to do this, but with a little bit of work (at 
both your end, and mine).

You can use multiple instances of "testMP3Streamer" for each source, 
provided that
1/ In "testMP3Streamer.cpp", you seed the random number generator with a 
distinct value (e.g., the IP address of the sender), *before* you create a 
'RTPSink'.  You can do this by calling
         our_srandom(<ip-address>);
The reason for doing this is to ensure (or at least, to make it very 
likely) that each instance of "testMP3Streamer.cpp" uses a different RTP 
SSRC.  (The code currently doesn't check for SSRC collisions.)  In a future 
release of the libraries, I will arrange to have this seeding happen 
automatically, so you won't have to worry about it.
2/ You use the RFC 3119 payload format, by uncommenting the line
         #define STREAM_USING_ADUS 1
You need to do this to avoid having a MP3 'bit reservoir' (backpointer) in 
each frame.  However, do *not* uncomment the line
         //#define INTERLEAVE_ADUS 1
(ADU interleaving probably won't work with multiple sources.)
3/ In "testMP3Receiver", you will need to insert a filter - between the 
"RTPSource" and the "MP3FromADUSource" - that reorders each received MP3 
'ADU', based on the 'presentationTime' of each ADU.
4/ I will need to upgrade (fix) the implementation of "RTPSource" to
         - assume a different RTP sequence number space for different SSRCs
         - return correct 'presentation times' for each received frame 
(based on the RTP timestamp and the NTP timestamps in received RTCP SRs)
It's possible that, in doing this, I can also remove the need for you to do 3/

So, the bottom line is that it should be possible, in theory, to do what 
you want to do, but I have some work to do first.

         Ross.


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From <EMAIL-ADDRESS-HIDDEN> Sun Apr 14 05:40:55 2002
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Subject: [Live-devel] RTCP Stats
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X-Original-Date: Sun, 14 Apr 2002 13:36:47 +0100
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Hi -=20
Im new to the Live stack - so this may seem like a very trivial =
question.=20
I was wondering if someone could tell how I can access the RTCP =
statistics i.e.Sender Report and Receiver Report data from the main =
function of the test application testMPEGVideoStreamer and =
testMPEGVideoReceiver.

Tahnking you, rgds,
Nikki

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
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charset=3Diso-8859-1">
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</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Hi - </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Im new to the Live stack - so this may =
seem like a=20
very trivial question. </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>I was wondering if someone could tell =
how I can=20
access the RTCP statistics i.e.Sender Report and Receiver Report data =
from the=20
main function of the test application testMPEGVideoStreamer and=20
testMPEGVideoReceiver.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Tahnking you, rgds,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Nikki</FONT></DIV></BODY></HTML>

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X-Original-Date: Sun, 14 Apr 2002 11:25:11 -0700
Date: Sun, 14 Apr 2002 11:25:11 -0700
Status: O

At 05:36 AM 4/14/02, you wrote:
>Hi -
>Im new to the Live stack - so this may seem like a very trivial question.
>I was wondering if someone could tell how I can access the RTCP statistics 
>i.e.Sender Report and Receiver Report data from the main function of the 
>test application testMPEGVideoStreamer and testMPEGVideoReceiver.

Nikki,

Incoming RTCP Sender Reports and Receiver Reports are received and 
processed in "liveMedia/RTCP.cpp".  This is the best place to look if you 
want to process the RTCP reports directly.

There is also some information that you can get from calling programs 
(i.e., without having to modify "liveMedia/RTCP.cpp":
- You can find out the number of active participants (excluding yourself) 
by calling
         rtcpInstance->numMembers()
         (This is probably useful only for multicast sessions)
- If you're a receiver (e.g., "testMP3Receiver"), you can get some 
information by
         calling rtpSource->receptionStatsDB()
         This gets you a database of reception statistics, which you can 
iterate
                 through, to get a "RTPReceptionStats" structure
         (see "liveMedia/include/RTPSource.hh" for details)
- Unfortunately there's currently no way for sending programs (e.g., 
"testMP3Streamer") to get back information from incoming RTCP Receiver 
Reports, other than "numMembers()".  To do this, you would need to modify 
"liveMedia/RTCP.cpp".

I hope this helps.

         Ross.


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From <EMAIL-ADDRESS-HIDDEN> Mon Apr 15 02:23:42 2002
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Subject: Re: [Live-devel] RTCP Stats - Thankyou! 
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X-Original-Date: Mon, 15 Apr 2002 10:19:07 +0100
Date: Mon, 15 Apr 2002 10:19:07 +0100
Status: O

Hi Ross,
Thats a great help! Actually Im doing research of adaptive QoS for Streamed
multimedia (mp4). So what my aim is to at the server side - evaluate the
RTCP reports and generate a QoS value, then based on this value - adapt the
transmission (and later the encoding) process - so if I want better
relaibility or whatever I can choose a transmission state - and for example,
send certain packets redundantly, for example, if the marker bit is set,
send twice. So thats what I hope to do using this stack.
If you're interested - Ive a few papers on the idea.
Rgds,
Nikki

----- Original Message -----
From: {Ross Finlayson} <EMAIL-ADDRESS-HIDDEN>
To: <EMAIL-ADDRESS-HIDDEN>
Sent: Sunday, April 14, 2002 7:25 PM
Subject: Re: [Live-devel] RTCP Stats


> At 05:36 AM 4/14/02, you wrote:
> >Hi -
> >Im new to the Live stack - so this may seem like a very trivial question.
> >I was wondering if someone could tell how I can access the RTCP
statistics
> >i.e.Sender Report and Receiver Report data from the main function of the
> >test application testMPEGVideoStreamer and testMPEGVideoReceiver.
>
> Nikki,
>
> Incoming RTCP Sender Reports and Receiver Reports are received and
> processed in "liveMedia/RTCP.cpp".  This is the best place to look if you
> want to process the RTCP reports directly.
>
> There is also some information that you can get from calling programs
> (i.e., without having to modify "liveMedia/RTCP.cpp":
> - You can find out the number of active participants (excluding yourself)
> by calling
>          rtcpInstance->numMembers()
>          (This is probably useful only for multicast sessions)
> - If you're a receiver (e.g., "testMP3Receiver"), you can get some
> information by
>          calling rtpSource->receptionStatsDB()
>          This gets you a database of reception statistics, which you can
> iterate
>                  through, to get a "RTPReceptionStats" structure
>          (see "liveMedia/include/RTPSource.hh" for details)
> - Unfortunately there's currently no way for sending programs (e.g.,
> "testMP3Streamer") to get back information from incoming RTCP Receiver
> Reports, other than "numMembers()".  To do this, you would need to modify
> "liveMedia/RTCP.cpp".
>
> I hope this helps.
>
>          Ross.
>
>
> _______________________________________________
> Live-devel mailing list
> Live-devel(at)lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/live-devel


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Subject: [Live-devel] How to get *.mov files ?
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Hi !
I need some *.mov files that openRTSP.exe can play the files =
appropriately (including audio and video).
It means that openRTSP.exe can understand its payload type.
Would you give me some advice how to get *.mov files that I need?
Thanks a lot.


    Armstrong

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</HEAD>
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<DIV><FONT size=3D2>Hi !</FONT></DIV>
<DIV><FONT size=3D2>I need some *.mov files that <FONT =
color=3D#0000ff>openRTSP.exe=20
</FONT>can play the files appropriately (including audio and=20
video).</FONT></DIV>
<DIV><FONT size=3D2>It means that <FONT color=3D#0000ff>openRTSP.exe =
</FONT><FONT=20
color=3D#000000>can understand its payload type.</FONT></FONT></DIV>
<DIV><FONT size=3D2>Would you give me some advice how to get *.mov files =
that I=20
need?</FONT></DIV>
<DIV><FONT size=3D2>Thanks a lot.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT size=3D2>&nbsp;&nbsp;&nbsp; =
Armstrong</FONT></DIV></BODY></HTML>

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Status: O

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Hi All,
I was just wondering if someone could tell me what is going wrong here. =
When I'm checking the SDES items in the RTCP report - it prints out this =
weird stuff - 4 refers to the number of bytes, which is fine. The CNAME =
should be 1 byte long, the length should be 1 byte and the Data should =
be 2 bytes.=20
SDES************
SDES
Size: 4
CNAME: ?
Length 73494115
Data a

Then when Im done with that Im going to use the SDES code as a template =
for APP packets!=20
Thanks a million, rgds,
Nikki


//////////////////////////////////////////////////////////////////////
  case RTCP_PT_SDES: {
   cout<<"SDES****************************"<<endl;
#ifdef DEBUG_PRINT
   fprintf(stderr, "SDES\n");
#endif
   cout<<"SDES"<<endl;
   //only CNAME used by default at the moment!
   //doesnt quite work!! prints out weird numbers=20

   unsigned sdesBlocksSize =3D rc*(4); //i.e. rc =3D sc & only 4 bytes =
for CNAME
   if (length < sdesBlocksSize) break; length -=3D sdesBlocksSize;

   unsigned char CCNAME =3D *pkt;=20
   ADVANCE(1);
   unsigned CNAMELength =3D ntohl(*(unsigned*)pkt);=20
   ADVANCE(1);
   unsigned char CNAMEData =3D *pkt;=20
   ADVANCE(2);

   cout<<"Size: "<<length<<endl;
   cout<<"CNAME: "<<CCNAME<<endl;
   cout<<"Length "<<CNAMELength<<endl;
   cout<<"Data "<<CNAMEData<<endl;
  =20
   subPacketOK =3D True;
   typeOfPacket =3D PACKET_SDES;
   break;
 }


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	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 5.50.4134.600" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Hi All,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>I was just wondering if someone could =
tell me what=20
is going wrong here. When I'm checking the SDES items in the RTCP report =
- it=20
prints out this weird stuff - 4 refers to the number of bytes, which is =
fine.=20
The CNAME should be 1 byte long, the length should be 1 byte and the =
Data should=20
be 2 bytes. </FONT></DIV>
<DIV>
<DIV><FONT face=3DArial size=3D2>SDES************<BR>SDES<BR>Size: =
4<BR>CNAME:=20
&#9786;<BR>Length 73494115<BR>Data a</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Then when Im done with that Im going to =
use the=20
SDES code as a template for APP packets! </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Thanks a million, rgds,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Nikki</FONT></DIV></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial=20
size=3D2>////////////////////////////////////////////////////////////////=
//////<BR>&nbsp;&nbsp;case=20
RTCP_PT_SDES:=20
{<BR>&nbsp;&nbsp;&nbsp;cout&lt;&lt;"SDES****************************"&lt;=
&lt;endl;<BR>#ifdef=20
DEBUG_PRINT<BR>&nbsp;&nbsp; fprintf(stderr, =
"SDES\n");<BR>#endif<BR>&nbsp;&nbsp;=20
cout&lt;&lt;"SDES"&lt;&lt;endl;<BR>&nbsp;&nbsp; //only CNAME used by =
default at=20
the moment!<BR>&nbsp;&nbsp; //doesnt quite work!! prints out weird=20
numbers&nbsp;</FONT></DIV>
<DIV><FONT face=3DArial size=3D2><BR>&nbsp;&nbsp; unsigned =
sdesBlocksSize =3D rc*(4);=20
//i.e. rc =3D sc &amp; only 4 bytes for CNAME<BR>&nbsp;&nbsp; if (length =
&lt;=20
sdesBlocksSize) break; length -=3D sdesBlocksSize;</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp; unsigned char CCNAME =3D =
*pkt;=20
<BR>&nbsp;&nbsp; ADVANCE(1);<BR>&nbsp;&nbsp; unsigned CNAMELength =3D=20
ntohl(*(unsigned*)pkt); <BR>&nbsp;&nbsp; ADVANCE(1);<BR>&nbsp;&nbsp; =
unsigned=20
char CNAMEData =3D *pkt; <BR>&nbsp;&nbsp; ADVANCE(2);</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp; cout&lt;&lt;"Size:=20
"&lt;&lt;length&lt;&lt;endl;<BR>&nbsp;&nbsp; cout&lt;&lt;"CNAME:=20
"&lt;&lt;CCNAME&lt;&lt;endl;<BR>&nbsp;&nbsp; cout&lt;&lt;"Length=20
"&lt;&lt;CNAMELength&lt;&lt;endl;<BR>&nbsp;&nbsp; cout&lt;&lt;"Data=20
"&lt;&lt;CNAMEData&lt;&lt;endl;<BR>&nbsp;&nbsp; <BR>&nbsp;&nbsp; =
subPacketOK =3D=20
True;<BR>&nbsp;&nbsp; typeOfPacket =3D PACKET_SDES;<BR>&nbsp;&nbsp;=20
break;<BR>&nbsp;}</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV></BODY></HTML>

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X-Original-Date: Mon, 15 Apr 2002 11:00:51 -0700
Date: Mon, 15 Apr 2002 11:00:51 -0700
Status: O

Nikki,

What is the source of the RTCP packets that you are (trying to) 
analyze?  Is it another one of the LIVE.COM Streaming Media test programs - 
e.g., "testMP3Streamer" - or is it someone else's application?

	Ross.


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Subject: Re: [Live-devel] How to get *.mov files ?
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X-Original-Date: Mon, 15 Apr 2002 10:59:19 -0700
Date: Mon, 15 Apr 2002 10:59:19 -0700
Status: O

At 07:12 AM 4/15/02, you wrote:
>I need some *.mov files that openRTSP.exe can play the files appropriately 
>(including audio and video).
>It means that openRTSP.exe can understand its payload type.
>Would you give me some advice how to get *.mov files that I need?

You will also need a RTSP *server* that can stream such files.

I suggest that you download and install Apple's QuickTime Streaming Server 
(aka. the "Darwin Streaming Server"): 
<http://www.apple.com/quicktime/products/qtss/>.  This iincludes a file 
"sample.mov" that you can stream (and receive using "openRTSP" using a 
"rtsp://" URL).

         Ross.


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X-Original-Date: Mon, 15 Apr 2002 12:04:16 -0700 (PDT)
Date: Mon, 15 Apr 2002 12:04:16 -0700 (PDT)
Status: O

hi,
 the call to env->taskScheduler().blockMyself() is
giving a segmentation fault.
 what could cause a segmentation fault for the
function call?
Your help would be of great help.
thank you
surya

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X-Original-Date: Mon, 15 Apr 2002 20:16:35 -0700
Date: Mon, 15 Apr 2002 20:16:35 -0700
Status: O

At 12:04 PM 4/15/02, you wrote:
>hi,
>  the call to env->taskScheduler().blockMyself() is
>giving a segmentation fault.
>  what could cause a segmentation fault for the
>function call?

Without more details, I don't know.  Are you *sure* it's crashing in 
blockMyself(), and not in one of the helper routines that it calls?  (You 
could try recompiling with debugging turned on (i.e., with -g if on Unix), 
and run it under a debugger, to see where it's really crashing.)

         Ross.


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Subject: [Live-devel] .mov file sans timing information
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X-Original-Date: Tue, 16 Apr 2002 11:02:39 +0530
Date: Tue, 16 Apr 2002 11:02:39 +0530
Status: O

Date: Fri, 12 Apr 2002 09:35:13 -0700
To: <EMAIL-ADDRESS-HIDDEN>
From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] Help - Regarding openRTSP

>At 05:07 AM 4/12/02, you wrote:
>Hello,
>
>I am streaming a quicktime file (of the form sample.mov provided for
>testing purpose by apple)using the DarwinStreaming Server.
>By using the opentRTSP i am able to get the streamed data
>and 2 files are created by openRTSP ie video-X-QT and audio-X-QT .But when
>i try to open these files in a quicktime player , it says unrecognized
format .

>By default, "openRTSP" outputs the 'raw' stream data that it receives
>(i.e., without RTP headers).  In the case of  "X-QT" data, this data is in
>the format described in
<http://developer.apple.com/quicktime/icefloe/dispatch026.html>.  However,
>this is *not* the QuickTime movie format.  Instead, it describes just one
>track of a possible QuickTime movie.  So, it can't be played in QuickTime
>Player, as is.

>To generate QuickTime '.mov' file that can be played (perhaps) in QuickTime
>Player, use the "-q" option to "openRTSP", and redirect 'stdout' to a
>"*.mov" file.  See <http://www.live.com/openRTSP/#quicktime> for more
details.
>       Ross.

Hi Ross,

I was trying to do the same thing as Niranjan mentioned, and with the above
suggestion that u gave Im now getting the streamed contents in a single
*.mov file. But there's a small glitch, When I play the file generated by
openRTSP in a QuickTIme player, it shows me the final frame only, .i.e. the
movie is not scaled to time. When I did look-up on the documentation
http://www.live.com/openRTSP/#quicktime and I did make another run with the
proper height, width, and frame rate parameters but still when I play the
.mov file in QuickTime player it shows me the final frame.
What could have possibly gone wrong ?

Interestingly when I drag the slider in the QuickTime player manually (with
the help of mouse), I am able to see the entire file, this means that all
the media data is present in the file except the timing information.

PS : I run the program with "openRTSP -p 1900 -V -q -l -w 320 -h 100 -f 12
rtsp://192.168.0.118:554/hintedmov.mov > somefile.mov"

please advise...

thanks
Deepesh S Gujarathi
Encodex Technologies (India) Pvt. Ltd.
28, Electronic estate,Pune- 411009
Tel- 91 20 4222 401/402
www.encodexindia.com



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Subject: Re: [Live-devel] .mov file sans timing information
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X-Original-Date: Tue, 16 Apr 2002 00:01:20 -0700
Date: Tue, 16 Apr 2002 00:01:20 -0700
Status: O


>I was trying to do the same thing as Niranjan mentioned, and with the above
>suggestion that u gave Im now getting the streamed contents in a single
>*.mov file. But there's a small glitch, When I play the file generated by
>openRTSP in a QuickTIme player, it shows me the final frame only, .i.e. the
>movie is not scaled to time. When I did look-up on the documentation
>http://www.live.com/openRTSP/#quicktime and I did make another run with the
>proper height, width, and frame rate parameters but still when I play the
>.mov file in QuickTime player it shows me the final frame.
>What could have possibly gone wrong ?

Unfortunately, to play properly in QuickTime Player, a '.mov' file needs to 
have a lot of codec-specific information.  Unfortunately this information 
can't all be obtained from the incoming data stream, so the implementation 
of the "-q" option works properly only for certain codecs that I know 
about.  It's likely that your file "hintedmov.mov" uses a codec that I 
don't (fully) support.

>PS : I run the program with "openRTSP -p 1900 -V -q -l -w 320 -h 100 -f 12
>rtsp://192.168.0.118:554/hintedmov.mov > somefile.mov"

Could you please put your file "hintedmov.mov" on a public HTTP server (not 
just a RTSP server) so I can take a look at it?  (Note that "192.168.0.118" 
is a private IP address, so I can't access it.)  If I can examing 
"hintedmov.mov", then I can probably figure out how to get the "-q" option 
to work for it.

         Ross.



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Subject: Re: [Live-devel] Print out SDES items
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X-Original-Date: Tue, 16 Apr 2002 09:57:32 +0100
Date: Tue, 16 Apr 2002 09:57:32 +0100
Status: O

Hi Ross,
At the moment Im just using the testMPEGVideoStreamer and Receiver programs
with a test.mpg file - I commented out in RTCP.cpp the line saying to ignore
local msgs - so that I can run the programs on a local host. Also, I
increased the bandwidth to increase the frequency of the packets - just for
testing purposes really as all I am interested in at this stage are the RTP
& RTCP packet header information.
For what I hope to do - it will involve a lot of modification of the RTCP
RTPSink and Source files as the test applications do what I wish them to and
the rest is automated in the stack.
Rgds,
Nikki
----- Original Message -----
From: {Ross Finlayson} <EMAIL-ADDRESS-HIDDEN>
To: <EMAIL-ADDRESS-HIDDEN>
Sent: Monday, April 15, 2002 7:00 PM
Subject: Re: [Live-devel] Print out SDES items


> Nikki,
>
> What is the source of the RTCP packets that you are (trying to)
> analyze?  Is it another one of the LIVE.COM Streaming Media test
programs -
> e.g., "testMP3Streamer" - or is it someone else's application?
>
> Ross.
>
>
> _______________________________________________
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> Live-devel(at)lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/live-devel


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X-Original-Date: Tue, 16 Apr 2002 02:41:06 -0700
Date: Tue, 16 Apr 2002 02:41:06 -0700
Status: O

Nikki,

Your code to handle the reception of SDES RTCP packets has a few bugs:

>   unsigned char CCNAME = *pkt;

That's fine.  However, if you want to print this out in human-readable 
form, don't forget to represent it as an integer.  (If you do this, you 
*should* get the value 1)

>    ADVANCE(1);
>    unsigned CNAMELength = ntohl(*(unsigned*)pkt);

The "length" field is just 1 byte long, so instead, you should do:
         unsigned char CNAMELength = *pkt;

>    ADVANCE(1);
>    unsigned char CNAMEData = *pkt;

Should be
         unsigned char* CNAMEData = pkt;

I hope this helps.

     Ross.


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 hello,
I have done one project, on wich I link with Live.com lib. Thank to Michael I overcome link problems specially LNK 2001 and 2000. Now when I try to execute my project I get an assert error message, when reaching statement:
                                 " Groupsock rtpGroupsock(*env, multicastAddress, rtpPort, ttl);" 
I try to debug the project and I get this Debug message:
                         "HEAP[MP3Streamer.exe]: Invalid Address specified to RtlValidateHeap( 350000, 2f3c40 )"

Any idea? thanks

Achraf




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<P>&nbsp;hello,
<P>I have done one project, on wich I link with Live.com lib. Thank to Michael I overcome link problems specially LNK 2001 and 2000. Now&nbsp;when I try to execute my project I get an assert error message, when&nbsp;reaching statement:
<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; "&nbsp;Groupsock rtpGroupsock(*env, multicastAddress, rtpPort, ttl);"&nbsp;
<P>I try to debug the project and I get this Debug message:
<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; "HEAP[MP3Streamer.exe]: Invalid Address specified to RtlValidateHeap( 350000, 2f3c40 )"</P>
<P>Any idea? thanks</P>
<P>Achraf<BR></P><p><br><hr size=1><a href="http://fr.mail.yahoo.com">Yahoo! Mail</a> -- Une adresse (at)yahoo.fr gratuite et en français !<br>
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X-Original-Date: Tue, 16 Apr 2002 19:14:18 +0530
Date: Tue, 16 Apr 2002 19:14:18 +0530
Status: O

Hi Ross,

I did find a workaround to record incoming packets by logging in the
Boolean QuickTimeFileSink::continuePlaying() function where I log the
parameter

subsession->controlPath()
but the other half of my problem is still unsolved wherein I need to log the
*incoming RTP time* of the packet.
Is there any variable accessible that would give me the time when a packet
arrived ?

thanks and regards

Deepesh S Gujarathi
Encodex Technologies (India) Pvt. Ltd.
28, Electronic estate,Pune- 411009
Tel- 91 20 4222 401/402
www.encodexindia.com



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  hello,
I have done a project with  VC++ V6.0. It's based on live.media.com library. Thank to Michael no compilation error is encountred. The problem is when I try to run the project, I get "address 0x.... can't be readed". When I debug the application I get this Message:
" First-chance exception in VOSserver.exe (MSVCRTD.DLL): 0xC0000005: Access Violation."
I have made an other project making the same thing as "testMP3Streamer". and no problem is encountred when compiling and when running it.
Any idea? thanks
Achraf



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<P> 
<P>&nbsp; hello,
<P>I have done a project with&nbsp; VC++ V6.0. It's based on live.media.com library. Thank to Michael no compilation error is encountred. The problem is when I try to run the project, I get&nbsp;"address 0x.... can't be readed". When I debug the application I get this Message:
<P>"&nbsp;First-chance exception in VOSserver.exe (MSVCRTD.DLL): 0xC0000005: Access Violation."
<P>I have made an other project making the same thing as "testMP3Streamer". and no problem is encountred when compiling and when running it.
<P>Any idea? thanks
<P>Achraf</P><p><br><hr size=1><a href="http://fr.mail.yahoo.com">Yahoo! Mail</a> -- Une adresse (at)yahoo.fr gratuite et en français !<br>
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X-Original-Date: Tue, 16 Apr 2002 08:49:58 -0700
Date: Tue, 16 Apr 2002 08:49:58 -0700
Status: O

At 06:44 AM 4/16/02, you wrote:
>Hi Ross,
>
>I did find a workaround to record incoming packets by logging in the
>Boolean QuickTimeFileSink::continuePlaying() function where I log the
>parameter
>
>subsession->controlPath()
>but the other half of my problem is still unsolved wherein I need to log the
>*incoming RTP time* of the packet.
>Is there any variable accessible that would give me the time when a packet
>arrived ?

Yes, note the function "BufferedPacket::assign()" (in 
"liveMedia/MultiFramedRTPSource.cpp").  This function - which is called 
each time a RTP packet arrives - takes the packet's RTP timestamp as 
parameter, and also the 'wallclock' time at which the packet arrived.

         Ross.


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X-Original-Date: Tue, 16 Apr 2002 08:49:52 -0700
Date: Tue, 16 Apr 2002 08:49:52 -0700
Status: O

At 04:54 AM 4/16/02, you wrote:

>  hello,
>
>I have done one project, on wich I link with Live.com lib. Thank to 
>Michael I overcome link problems specially LNK 2001 and 2000. Now when I 
>try to execute my project I get an assert error message, when reaching 
>statement:
>
>                                  " Groupsock rtpGroupsock(*env, 
> multicastAddress, rtpPort, ttl);"

I don't know what the error is, but you should check that
         rtpGroupsock.socketNum()
is not -1 after returning from the constructor.

>I try to debug the project and I get this Debug message:
>
>                          "HEAP[MP3Streamer.exe]: Invalid Address 
> specified to RtlValidateHeap( 350000, 2f3c40 )"
>
>Any idea?

Sorry, no.  But don't forget that you have the *complete source code* to 
these libraries.  If there's a crash at runtime, you should be able to 
figure out exactly where and why it's happening.

         Ross.


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X-Original-Date: Tue, 16 Apr 2002 11:14:41 +0530
Date: Tue, 16 Apr 2002 11:14:41 +0530
Status: O

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Hello !!


I am streaming a sample.mov file from a Darwin Streaming server.
In the openRTSP i also am redirecting the file instead of stdout 
to "test.mov",

1> i tested openrtsp with -q option , and accordingly the 
file from the server is streamed and stored in test.mov.
But the file size in the server is (0.5MB) and of test.mov
after streaming is (0.23MB).And when i try to open test.mov 
in a quicktime player it says unrecognized format.What
could be the problem.

2> On a similar setup , when i run openRTSP with the option
 -q -v/-a ,(ie for video data/audio data) , i get a 
segmentation fault.

Niranjan.


--Boundary_(ID_wf30+1vXFlj8+lMMDojvvA)
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
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<STYLE></STYLE>
</HEAD>
<BODY bgColor=#ffffff>
<DIV><FONT face=Arial size=2>Hello !!</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=Arial size=2><BR>I am streaming a sample.mov file from a Darwin 
Streaming server.<BR>In the openRTSP i also am redirecting the file instead of 
stdout <BR>to "test.mov",</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=Arial size=2>1&gt; i tested openrtsp with -q option , and 
accordingly the <BR>file from the server is streamed and stored in 
test.mov.<BR>But the file size in the server is (0.5MB) and of test.mov<BR>after 
streaming is (0.23MB).And when i try to open test.mov <BR>in a quicktime player 
it says unrecognized format.What<BR>could be the problem.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=Arial size=2>2&gt; On a similar setup , when i run openRTSP with 
the option<BR>&nbsp;-q -v/-a ,(ie for video data/audio data) , i get a 
<BR>segmentation fault.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=Arial size=2>Niranjan.<BR></FONT></DIV></BODY></HTML>

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--Boundary_(ID_q7025cnLACd9CNaBmiWDZQ)
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Hello !!


I am streaming a sample.mov file from a Darwin Streaming server.
In the openRTSP i also am redirecting the file instead of stdout 
to "test.mov",

1> i tested openrtsp with -q option , and accordingly the 
file from the server is streamed and stored in test.mov.
But the file size in the server is (0.5MB) and of test.mov
after streaming is (0.23MB).And when i try to open test.mov 
in a quicktime player it says unrecognized format.What
could be the problem.

2> On a similar setup , when i run openRTSP with the option
 -q -v/-a ,(ie for video data/audio data) , i get a 
segmentation fault.

Niranjan.


--Boundary_(ID_q7025cnLACd9CNaBmiWDZQ)
Content-type: text/html; charset=iso-8859-1
Content-transfer-encoding: 7BIT

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content="text/html; charset=iso-8859-1" http-equiv=Content-Type>
<META content="MSHTML 5.00.3103.1000" name=GENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=#ffffff>
<DIV><FONT face=Arial size=2>Hello !!</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=Arial size=2><BR>I am streaming a sample.mov file from a Darwin 
Streaming server.<BR>In the openRTSP i also am redirecting the file instead of 
stdout <BR>to "test.mov",</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=Arial size=2>1&gt; i tested openrtsp with -q option , and 
accordingly the <BR>file from the server is streamed and stored in 
test.mov.<BR>But the file size in the server is (0.5MB) and of test.mov<BR>after 
streaming is (0.23MB).And when i try to open test.mov <BR>in a quicktime player 
it says unrecognized format.What<BR>could be the problem.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=Arial size=2>2&gt; On a similar setup , when i run openRTSP with 
the option<BR>&nbsp;-q -v/-a ,(ie for video data/audio data) , i get a 
<BR>segmentation fault.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=Arial size=2>Niranjan.<BR></FONT></DIV></BODY></HTML>

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X-Original-Date: Tue, 16 Apr 2002 17:50:28 -0700
Date: Tue, 16 Apr 2002 17:50:28 -0700
Status: O

At 10:43 PM 4/15/02, you wrote:
>I am streaming a sample.mov file from a Darwin Streaming server.
>In the openRTSP i also am redirecting the file instead of stdout
>to "test.mov",
>
>1> i tested openrtsp with -q option , and accordingly the
>file from the server is streamed and stored in test.mov.
>But the file size in the server is (0.5MB) and of test.mov
>after streaming is (0.23MB).

That, by itself, is not necessarily surprising.  The original .mov file 
contains a 'hint track' (containing RTP headers and other information that 
tells it how to stream).  These will add space to the file, beyond the 
space used by audio and video data.

>And when i try to open test.mov
>in a quicktime player it says unrecognized format.What
>could be the problem.

You're probably encountering the same issue as Deepesh - namely:

Unfortunately, to play properly in QuickTime Player, a '.mov' file needs to 
have a lot of codec-specific information.  Unfortunately this information 
can't all be obtained from the incoming data stream, so the implementation 
of the "-q" option works properly only for certain codecs that I know 
about.  It's likely that your file uses a codec that I don't (fully) support.

Could you please put your original hinted .mov file on a public HTTP server 
(not just a RTSP server) so I can take a look at it?  If I can examine 
this, then I can probably figure out how to get the "-q" option to work for it.

>  2> On a similar setup , when i run openRTSP with the option
>  -q -v/-a ,(ie for video data/audio data) , i get a
>segmentation fault.

I have not seen this error before.  I can probably figure out the problem 
if/when I can experiment with the file that you're streaming.

         Ross.


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From <EMAIL-ADDRESS-HIDDEN> Wed Apr 17 10:37:16 2002
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X-Original-Date: Wed, 17 Apr 2002 19:31:19 +0200 (CEST)
Date: Wed, 17 Apr 2002 19:31:19 +0200 (CEST)
Status: O

thanks Ross,
In the RFC 2250 the field P(picture type contain the
effectif type of the current picture
          1-->I frame ;2-->P frame, 
          3-->B frame ;4 -->D frame)
what i didn't understand that:
MPEGVideoRTPSink::doSpecialFrameHandling this method
is responsible of setting up the Specific Mpeg Video
Header (RFC 2250).
i'm trying the testMPEGVideoStreamer using
test.mpg(MPEG1),I'm printing the type of the
picture,the value of fPictureState.picture_coding_type
is 2 or 3 which means the P frame type or the B frame
type.In the specification of MPEG Video the I frame is
the most important frame in one GOP so where is the I
frame type

thank you in advance!


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X-Original-Date: Wed, 17 Apr 2002 19:39:08 +0200 (CEST)
Date: Wed, 17 Apr 2002 19:39:08 +0200 (CEST)
Status: O

MultiFramedRTPSink::sendPacketIfNecessary()
//#ifdef TEST_LOSS
 if ((our_random()%10) != 0)//simulate 10% packet loss

//#endif
in this way , we have one packet of 10 isn't
transmitted into the network independly if it contain
a completely frame or not and independly of the
picture type????


thanks









thank you in advance!


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X-Original-Date: Wed, 17 Apr 2002 12:59:00 -0700
Date: Wed, 17 Apr 2002 12:59:00 -0700
Status: O

At 10:39 AM 4/17/02, you wrote:
>MultiFramedRTPSink::sendPacketIfNecessary()
>//#ifdef TEST_LOSS
>  if ((our_random()%10) != 0)//simulate 10% packet loss
>
>//#endif
>in this way , we have one packet of 10 isn't
>transmitted into the network independly if it contain
>a completely frame or not and independly of the
>picture type????

This test code - in "MultiFramedRTPSink::sendPacketIfNecessary()" - applies 
to the statement that follows:
         sendPacket(...)
so it simulates random (Poisson) 10% *network packet loss*.

         Ross.


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X-Original-Date: Wed, 17 Apr 2002 23:43:18 -0700
Date: Wed, 17 Apr 2002 23:43:18 -0700
Status: O

At 10:31 AM 4/17/02, you wrote:
>thanks Ross,
>In the RFC 2250 the field P(picture type contain the
>effectif type of the current picture
>           1-->I frame ;2-->P frame,
>           3-->B frame ;4 -->D frame)
>what i didn't understand that:
>MPEGVideoRTPSink::doSpecialFrameHandling this method
>is responsible of setting up the Specific Mpeg Video
>Header (RFC 2250).
>i'm trying the testMPEGVideoStreamer using
>test.mpg(MPEG1),I'm printing the type of the
>picture,the value of fPictureState.picture_coding_type
>is 2 or 3 which means the P frame type or the B frame
>type.In the specification of MPEG Video the I frame is
>the most important frame in one GOP so where is the I
>frame type

Thanks - you found a bug (in "MPEGVideoRTPSink.cpp").  It was not setting 
the RFC 2250 'video-specific header' correctly for type 1 Picture Headers.

I have now released a new version (2002.04.17) of the libraries that fixes 
this bug.

(The new code also seeds the pseudo-random number generator with a seed 
that's based on the machine's IP address and current time of day.  This 
will ensure (with high probability) that each sender streams with a 
different RTP SSRC.)

         Ross.


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From <EMAIL-ADDRESS-HIDDEN> Thu Apr 18 05:21:54 2002
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Subject: [Live-devel] RE : Regarding openRTSP
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X-Original-Date: Thu, 18 Apr 2002 17:55:36 +0530
Date: Thu, 18 Apr 2002 17:55:36 +0530
Status: O

Hi Niranjan,

>I was trying to do the same thing as Niranjan mentioned, and with the above
>suggestion that u gave Im now getting the streamed contents in a single
>*.mov file. But there's a small glitch, When I play the file generated by
>openRTSP in a QuickTIme player, it shows me the final frame only, .i.e. the
>movie is not scaled to time. When I did look-up on the documentation
>http://www.live.com/openRTSP/#quicktime and I did make another run with the
>proper height, width, and frame rate parameters but still when I play the
>.mov file in QuickTime player it shows me the final frame.
>What could have possibly gone wrong ?

Unfortunately, to play properly in QuickTime Player, a '.mov' file needs to
have a lot of codec-specific information.  Unfortunately this information
can't all be obtained from the incoming data stream, so the implementation
of the "-q" option works properly only for certain codecs that I know
about.  It's likely that your file "hintedmov.mov" uses a codec that I
don't (fully) support.


PS : do refer the documentation available on
http://www.live.com/openRTSP/#quicktime  and try to run the openRTSP app
with appropriate height,width and frame rate parameters,  think you will get
a .mov file that can be played in the QUickTime player (atleast it works on
my machine).

Also do send mails in a text (instead of HTML) format on the list, it will
be more readable.

Deepesh S Gujarathi
Encodex Technologies (India) Pvt. Ltd.
28, Electronic estate,Pune- 411009
Tel- 91 20 4222 401/402
www.encodexindia.com



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From <EMAIL-ADDRESS-HIDDEN> Thu Apr 18 05:39:35 2002
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X-Original-Date: Thu, 18 Apr 2002 14:38:34 +0200 (CEST)
Date: Thu, 18 Apr 2002 14:38:34 +0200 (CEST)
Status: O

thank you Ross

I tried the new version of this project,when i'm
printing the picture type i find the I frame.
But what i didn't understand that in the spécification
of the MPEG Video the GOP (Group of Picture) begin
with
I frame,and there are P frames  witch are seperated by
a fixe number of B frame. For example
         
            GOP="IBBPBBPBB"
wht i find in the  first GOP is:
            GOP="IPBBPBBPBBPBB"
normaly we have to find a B frame after a I frame.i
find this only in the first GOP of the first Video
Sequence Header
can you help me, is there any bug??

thanks

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Subject: [Live-devel] openRTSP programming questions
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X-Original-Date: Thu, 18 Apr 2002 19:06:21 +0200
Date: Thu, 18 Apr 2002 19:06:21 +0200
Status: O

Hello!
 
I've some questions about openRTSP.
 
I've observed that RTSP messages are sent to the server rtsp
using a socket TCP. On the server side there is a daemon listening on port 554;
The IP address is obtained form the uri that comes from the command line
 
openRTSP rtsp://....
 
Q: what is the function that converts the address specified to the IP dotted point ?
(I suppose there's a query to the DNS server in some place in the code)
 
Q: On the client side, how is chosen the port number of the socket TCP( the one
that receives answers from server side) ?
 
(I've observed that this port number is always different each time 
a new copy of openRTSP is running)
 
 
The client than sends to the sever text messages.
Q: Which function is responsible of sending these messages?
That is, the function in which these messages are put and then written to the socket .
 
 
The first message that the client RTSP sends to the server is the DESCRIBE method,
including the uri specified on the command line.
 
The server replies sending a file SDP that contains information on how to control the
incoming multimedia flow.
I suppose that the information that are retreived in the sdp file are
 the control (a=control:xxx) and the range (a=range:zzz).
Q: Are there other information that the software looks for in the SDP file?
 
The next command message is SETUP, in which now is specified the "control" label
obtained from sdp file. Another parameter is the desired port number on which
we want to receive the media flow. The client IP address in not specified.
 
Q: On the server side, is the IP address obtained from the header of IP packets?
 
 
I have supposed that the server rtsp forwards this IP address plus the port number to the streamer.
Q: Is it true?
 
Next command message is PLAY, issued after receiving the "200 ok" from the server side.
Q: After receiving the ok from the server, does a timer start, initialized with the time
specified in the "a=range:"  ?
 
I've supposed the use of the timer because after playing 
we must send the TEARDOWN message. So, at the zero the teardown is sent.
 
Q: Is this true?
 
Q: Or is there another method to see the end of the streaming from the server?
 
In the end,
 
  Can I issue the TEARDOWN message during the playing of the stream?
 
It's something like the stop button
 
 
I've done so many questions; I hope these don't cause a headache... ;-)
 
thank you!
 
Domenico
 
 
 

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Subject: Re: [Live-devel] re about picture type
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X-Original-Date: Thu, 18 Apr 2002 10:20:32 -0700
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Status: O

At 05:38 AM 4/18/02, you wrote:
>But what i didn't understand that in the spécification
>of the MPEG Video the GOP (Group of Picture) begin
>with
>I frame,and there are P frames  witch are seperated by
>a fixe number of B frame. For example
>
>             GOP="IBBPBBPBB"
>wht i find in the  first GOP is:
>             GOP="IPBBPBBPBBPBB"
>normaly we have to find a B frame after a I frame.i
>find this only in the first GOP of the first Video
>Sequence Header
>can you help me, is there any bug??

I don't think so.  I've seen the same thing in MPEG files that I've been 
streaming.

By the way, the best place to inspect the "picture_coding_type" is in 
"MPEGVideoStreamParser::parsePictureHeader()".  (Comment out the #ifdef 
DEBUG, #endif pair.)

         Ross.


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Subject: Re: [Live-devel] openRTSP programming questions
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X-Original-Date: Thu, 18 Apr 2002 10:48:13 -0700
Date: Thu, 18 Apr 2002 10:48:13 -0700
Status: O

At 10:06 AM 4/18/02, you wrote:
>The IP address is obtained form the uri that comes from the command line
>
>openRTSP rtsp://....
>
>Q: what is the function that converts the address specified to the IP 
>dotted point ?
>(I suppose there's a query to the DNS server in some place in the code)

This happens in "RTSPClient::parseURL()", line 773, where it calls
         NetAddressList addresses(parseBuffer);
(To see how this works, look in the code for the "groupsock" library, but 
basically, it works by calling "gethostbyname()".)


>  Q: On the client side, how is chosen the port number of the socket TCP( 
> the one
>that receives answers from server side) ?

Note the call to "setupStreamSocket()" in 
"RTSPClient::openConnectionFromURL()".  Because the port number parameter 
is specified as 0, the operating system ends up choosing an 'ephemeral port 
number' for its end of the TCP connection.

If you wanted to specifiy a particular TCP port number here (e.g., because 
you have a firewall), then you could do this in the call to 
"setupStreamSocket()".


>The client than sends to the sever text messages.
>Q: Which function is responsible of sending these messages?

"RTSPClient::sendRequest()"


>The first message that the client RTSP sends to the server is the DESCRIBE 
>method,
>including the uri specified on the command line.
>
>The server replies sending a file SDP that contains information on how to 
>control the
>incoming multimedia flow.
>I suppose that the information that are retreived in the sdp file are
>  the control (a=control:xxx) and the range (a=range:zzz).
>Q: Are there other information that the software looks for in the SDP file?

"openRTSP" looks at many other lines in the SDP description - in 
particular, the "m=", "c=", and "a=rtpmap:" lines.  See the various 
functions named "parseSDP..." in "liveMedia/MediaSession.cpp".


>  The next command message is SETUP, in which now is specified the 
> "control" label
>obtained from sdp file. Another parameter is the desired port number on which
>we want to receive the media flow. The client IP address in not specified.
>
>Q: On the server side, is the IP address obtained from the header of IP 
>packets?

Probably.  For a unicast session, the server will stream to the IP address 
that has been  sending the RTSP requests.


>  Next command message is PLAY, issued after receiving the "200 ok" from 
> the server side.
>Q: After receiving the ok from the server, does a timer start, initialized 
>with the time
>specified in the "a=range:"  ?
>
>I've supposed the use of the timer because after playing
>we must send the TEARDOWN message. So, at the zero the teardown is sent.
>
>Q: Is this true?

Yes, "openRTSP" uses the end time specified in a "a=range: npt" attribute - 
if any - to figure out when the stream is supposed to end.

>  Q: Or is there another method to see the end of the streaming from the 
> server?

Yes, "openRTSP" also looks for RTCP "BYE" packets. If it sees one, it 
assumes that the stream has ended.


>  In the end,
>
>   Can I issue the TEARDOWN message during the playing of the stream?

Yes.  In fact, this is what openRTSP's "-e" option does, if you ask to end 
streaming early.

         Ross.


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X-Original-Date: 18 Apr 2002 18:52:06 -0400
Date: 18 Apr 2002 18:52:06 -0400
Status: O

I've recently been working on the xAudic source code
http://www.xaudic.net/
to add RTP support for mp3 playback, and in the process run into a few
issues.  Some of the background issues are: this is a MacOSX platform,
and xAudic is written in Objective-C.

First, I'd just like to say that the mix of c++ and obj-c called
"obj-c++" is nasty.  One of the primary limitations involves virtual
functions.  Creating the BasicTaskScheduler through it's constructor was
a problem for the compiler.  I was able to work around this by adding a
createNew() static method.  This approach is used elsewhere in the
library so I didn't feel it was too dirty.  I might have another way
around the problem now, but if not, is this an acceptable solution?

Second, there was a naming conflict between the TaskProc typedef and
something used internal to MacOSX (I can dig up more info for this if
necessary).  However, I was able to work around this by renaming it
TaskProcX.  This is a terrible solution, but allowed me to keep going. 
Namespaces are out, since obj-c++ doesn't support them.  Is there anyway
the TaskProc typedef could be made public inside one of the classes to
shield it from the rest of the world without losing any functionality?

Third, I realized a problem with the usage of the
PrioritizedRTPStreamSelector class.  I used the modified version of
freeamp as a reference, and uncovered a bug.  The multi-stream case does
not seem to ever call the destructor for the RTP and RTCP groupsocks.
This has the nasty side-effect of not leaving the multicast groups after
a session has been terminated.

Looking at the code, this is handled externally for single stream
cases.  But doing it externally for the multi-streams would require
storing some vector of GroupSock objects.  Since the
PrioritizedRTPStreamSelector already has a list of
PrioritizedInputStreamDescriptors, which do get destructed.
I modified the destructor for PrioritizedInputStreamDescriptor to add
these two lines.

delete fRTPStream->RTPgs();
delete fRTCPStream->RTCPgs();

Limited testing, but this seems to work.  NOTE: this bug also affects
the use of SLAP streams in the modified version of FreeAmp.

Hopefully we can agree on some workarounds to make this use of the
library work.

Marty Schoch
<mschoch(at)multicasttech.com>






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From <EMAIL-ADDRESS-HIDDEN> Thu Apr 18 18:59:56 2002
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X-Original-Date: Thu, 18 Apr 2002 18:56:46 -0700
Date: Thu, 18 Apr 2002 18:56:46 -0700
Status: O

At 03:52 PM 4/18/02, you wrote:
>I've recently been working on the xAudic source code
>http://www.xaudic.net/
>to add RTP support for mp3 playback

Great stuff!


>First, I'd just like to say that the mix of c++ and obj-c called
>"obj-c++" is nasty.  One of the primary limitations involves virtual
>functions.  Creating the BasicTaskScheduler through it's constructor was
>a problem for the compiler.

Do you mean "creating a BasicTaskScheduler by declaring it as a variable 
(i.e., on the stack, instead of on the heap)"?  Can you give some more 
details as to what the issues are here, and how/why declaring objects on 
the stack is bad for 'obj-c++'?  Is there only a problem when the class 
contains virtual functions??


>   I was able to work around this by adding a
>createNew() static method.  This approach is used elsewhere in the
>library so I didn't feel it was too dirty.  I might have another way
>around the problem now, but if not, is this an acceptable solution?

Yes, and in fact it's probably better than the old approach.  The nice 
thing about the "createNew()" approach (whereby objects are created on the 
heap only, and cannot be created on the stack) is that it makes it easier 
for people to add binary-only extensions, if they wish (because if objects 
are not allocated on the stack, then it doesn't matter if their sizes 
change later).

So, in a future release, I may well change BasicUsageEnvironment so that it 
can only be created on the heap, using a "createNew()" function.  (As this 
will require (minor) changes to people's programs, I'll warn people if/when 
this happens.)


>Second, there was a naming conflict between the TaskProc typedef and
>something used internal to MacOSX (I can dig up more info for this if
>necessary).  However, I was able to work around this by renaming it
>TaskProcX.  This is a terrible solution, but allowed me to keep going.
>Namespaces are out, since obj-c++ doesn't support them.  Is there anyway
>the TaskProc typedef could be made public inside one of the classes to
>shield it from the rest of the world without losing any functionality?

In theory that would be the best solution, but I remember that I had to 
pull some typedefs like this out of classes in order to make someone happy 
who was compiling the code for QNX using some obscure compiler (Watcom).

Probably what I'll do instead is rename "TaskProc" to "TaskFunc" (which is 
a slightly more descriptive name anyway).  I hope that the name "TaskFunc" 
doesn't cause any conflicts?


>Third, I realized a problem with the usage of the
>PrioritizedRTPStreamSelector class.  I used the modified version of
>freeamp as a reference, and uncovered a bug.  The multi-stream case does
>not seem to ever call the destructor for the RTP and RTCP groupsocks.

Yes, you're right - but this is true only if the 'multi streams' were 
created by parsing a 'rtp://' URL of the form 
"rtp://<addr1:port1>|<addr2:port2>|...|<addrn:portn>".  If the multi 
streams were created as a result of reading a SDP description (either from 
a file, or using RTSP), then 'groupsocks' are created within a 
"MediaSession" object, and deleted when this object gets deleted.


>Since the PrioritizedRTPStreamSelector already has a list of
>PrioritizedInputStreamDescriptors, which do get destructed.
>I modified the destructor for PrioritizedInputStreamDescriptor to add
>these two lines.
>
>delete fRTPStream->RTPgs();
>delete fRTCPStream->RTCPgs();

As a general rule, I think it's best for objects to get deleted in the same 
portion in the code where they're created.  So deleting the 'groupsocks' in 
the PrioritizedInputStreamDescriptor destructor is not a good idea, and in 
fact will break if a SDP description was used (because in that case the 
'groupsocks' will get deleted a second time when the "MediaSession" object 
gets destroyed).  (BTW, I hope that your xAudic patch will also support SDP 
files and "rtsp://" URLs, as well as "rtp://" URLs.)


>Looking at the code, this is handled externally for single stream
>cases.  But doing it externally for the multi-streams would require
>storing some vector of GroupSock objects.
[...]
>  NOTE: this bug also affects
>the use of SLAP streams in the modified version of FreeAmp.

One good place to do this, I think, will be in the destructor for 
"RTPEndpointDescriptor", because this is already used in a vector - one 
element for each stream.

What I'll do now is update the FreeAmp code to make this fix.  When I've 
done this, I'll let you know, so you can more easily apply the same fix to 
your code.

         Ross.


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X-Original-Date: Sat, 20 Apr 2002 12:54:17 +0200
Date: Sat, 20 Apr 2002 12:54:17 +0200
Status: O

Ross, thanks for the answers to my questions.


I've red in the documentation of openRTSP that when 

specifying the -r option some application must be able to send RTCP 

packets to ensure that the session doesn't time out.

Without the -r option, does openRTSP sends these packets by himeself,

even if we are receiving in a unicast session?

If so, what is specified in RTCP packet and which procedure build it?

thanks so much

Domenico

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X-Original-Date: Sat, 20 Apr 2002 14:26:03 +0200
Date: Sat, 20 Apr 2002 14:26:03 +0200
Status: O

I've a question about testRelaying.

Can I run the program on a machine, 

and reading the incoming packet from an

IP address belonging to another machine?

That is, can I specify as input for testRelay

an IP address (unicast) other than the one of local machine?

In this case, is the TTL field significant?

Thanks, regards

Domenico


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X-Original-Date: Sat, 20 Apr 2002 21:19:55 -0700
Date: Sat, 20 Apr 2002 21:19:55 -0700
Status: O

At 03:54 AM 4/20/02, you wrote:
>I've red in the documentation of openRTSP that when
>
>specifying the -r option some application must be able to send RTCP
>
>packets to ensure that the session doesn't time out.
>
>Without the -r option, does openRTSP sends these packets by himeself,
>
>even if we are receiving in a unicast session?

Yes.  The RTCP packets will be sent back to the streaming server.

>If so, what is specified in RTCP packet

These are RTCP "Reception Report" packets.  For more information about 
RTCP, see the latest version of the RTP/RTCP specification at 
<http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-new-11.txt>

>and which procedure build it?

RTCPInstance::sendReport()  (See "liveMedia/RTCP.cpp")

         Ross.




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X-Original-Date: Sat, 20 Apr 2002 21:21:07 -0700
Date: Sat, 20 Apr 2002 21:21:07 -0700
Status: O

At 05:26 AM 4/20/02, you wrote:
>That is, can I specify as input for testRelay
>
>an IP address (unicast) other than the one of local machine?

No, because how could the local machine even see these packets, if they're 
not addressed to it?

         Ross.


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X-Original-Date: Sun, 21 Apr 2002 14:20:58 +0200 (CEST)
Date: Sun, 21 Apr 2002 14:20:58 +0200 (CEST)
Status: O

thanks Ross

ok,i tried to modify the GroupSock librairie to
support the RAW_SOCK mode,i succeed to setup the IP
Hearder and the UDP Hearder: i print these packets on
the server using the ethereal tool but nothing arrive
into the client side ,i 'm using testMPEGVideoStreamer
in unicast mode,i have only one question the QuickTime
player when it reads from the network,it setup a
Datagram Socket with the server or not, because it
didn't read anything when i have made this change on
GroupSock lib.
can you help me by telling me where i must make the
change in the GroupSock lib

In advance thank you for your help






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Subject: [Live-devel] 'Picture Type' in parsePictureHeader()
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X-Original-Date: Sun, 21 Apr 2002 14:27:50 +0200 (CEST)
Date: Sun, 21 Apr 2002 14:27:50 +0200 (CEST)
Status: O

thanks Ross

ok,i tried to print the Picture type at the
sourcemodify :
MPEGVideoStreamParser::parsePictureHeader()
i find that only in  the first GOP  --> IPBBP...
i think that it a bug in the test.mpg 


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Subject: [Live-devel] RTPSource - transit calculations?
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X-Original-Date: Mon, 22 Apr 2002 14:34:27 +0100
Date: Mon, 22 Apr 2002 14:34:27 +0100
Status: O

This is a multi-part message in MIME format.

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	charset="iso-8859-1"
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Hi all,
Im curious about these results - not sure what is wrong here - but Im =
working with the testMPEGVideoStreamer and testMPEGVideoReceiver =
programs both running on a localhost - I just wanted to monitor the =
delay and got these values:=20

arrival            rtpTimestamp    transit ??????????
4281287218   1013425969      -1027106047
4281287218   1013425969      -1027106047
4281287218   1013425969      -1027106047
4281287218   1013425969      -1027106047
4281287218   1013425969      -1027106047
4281287218   1013425969      -1027106047

Id expect on a local host for the transit to be (or near) 0. The jitter =
calculations seem ok so Im just not sure what the deal is with the =
delay?
Also, on a related matter (I think!) How can the server monitor the =
average packet delay to the client - could this be found out using the =
RTCP RR or would it have to be relayed some other way (such as the APP =
packet for instance!)

Thanks a million,
Nikki

void RTPReceptionStats
::noteIncomingPacket(unsigned short seqNum, unsigned rtpTimestamp,
       unsigned timestampFrequency,
       Boolean useForJitterCalculation) {
..........................................   =20
struct timeval timeNow;
  gettimeofday(&timeNow, &Idunno);
  unsigned arrival =3D (timestampFrequency*timeNow.tv_sec);
  arrival +=3D (unsigned)
   ((2.0*timestampFrequency*timeNow.tv_usec + 1000000.0)/2000000);
            // note: rounding
  int transit =3D arrival - rtpTimestamp;
  if (fLastTransit =3D=3D (~0)) fLastTransit =3D transit; // hack for =
first time
  int d =3D transit - fLastTransit;
  cout<<arrival<<"\t"<<rtpTimestamp<<"\t"<<transit<<endl;



------=_NextPart_000_0075_01C1EA0A.CE874A70
Content-Type: text/html;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 5.50.4134.600" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Hi all,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Im curious about these results - not =
sure what is=20
wrong here - but Im working with the testMPEGVideoStreamer and=20
testMPEGVideoReceiver programs both running on a localhost -&nbsp;I just =
wanted=20
to monitor the delay and got these values: </FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>arrival&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp; rtpTimestamp&nbsp;&nbsp;&nbsp; transit=20
??????????</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>
<DIV><FONT face=3DArial size=3D2>4281287218&nbsp;&nbsp;=20
1013425969&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
-1027106047<BR>4281287218&nbsp;&nbsp;=20
1013425969&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
-1027106047<BR>4281287218&nbsp;&nbsp;=20
1013425969&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
-1027106047<BR>4281287218&nbsp;&nbsp;=20
1013425969&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
-1027106047<BR>4281287218&nbsp;&nbsp;=20
1013425969&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
-1027106047<BR>4281287218&nbsp;&nbsp;=20
1013425969&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
-1027106047</FONT></DIV></FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Id expect on a local host for the =
transit to be (or=20
near) 0. The jitter calculations seem ok so Im just not sure what the =
deal is=20
with the delay?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Also, on a related matter (I think!) =
How can the=20
server monitor the average packet delay to the client - could this be =
found out=20
using the RTCP RR or would it have to be relayed some other way (such as =
the APP=20
packet for instance!)</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Thanks a million,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Nikki</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>void=20
RTPReceptionStats<BR>::noteIncomingPacket(unsigned short seqNum, =
unsigned=20
rtpTimestamp,<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; unsigned=20
timestampFrequency,<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Boolean=20
useForJitterCalculation) {</FONT></DIV>
<DIV><FONT face=3DArial=20
size=3D2>..........................................&nbsp;&nbsp;&nbsp;=20
</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>struct timeval=20
timeNow;<BR>&nbsp;&nbsp;gettimeofday(&amp;timeNow,=20
&amp;Idunno);<BR>&nbsp;&nbsp;unsigned arrival =3D=20
(timestampFrequency*timeNow.tv_sec);<BR>&nbsp;&nbsp;arrival +=3D=20
(unsigned)<BR>&nbsp;&nbsp;&nbsp;((2.0*timestampFrequency*timeNow.tv_usec =
+=20
1000000.0)/2000000);<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;=20
// note: rounding</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp; int transit =3D arrival -=20
rtpTimestamp;<BR>&nbsp;&nbsp;if (fLastTransit =3D=3D (~0)) fLastTransit =
=3D transit;=20
// hack for first time<BR>&nbsp;&nbsp;int d =3D transit -=20
fLastTransit;<BR>&nbsp;&nbsp;cout&lt;&lt;arrival&lt;&lt;"\t"&lt;&lt;rtpTi=
mestamp&lt;&lt;"\t"&lt;&lt;transit&lt;&lt;endl;</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV></BODY></HTML>

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Subject: [Live-devel] Regarding openRTSP & testMPEGVideoStreamer !!
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X-Original-Date: Mon, 22 Apr 2002 19:57:41 +0530
Date: Mon, 22 Apr 2002 19:57:41 +0530
Status: O

Hello

I am trying to stream a test.mpg file from the testMPEGVideoStreamer.In the
testMPEGVideoStreamer i also setup the RTSP server to run at port number
554.
I then run the opentRTSP using the following command,
./openRTSP -V -n rtsp://localhost:554/test.mpg
The Describe Request is sent and the response for the same is received
successfully and also i receive the sdp description , but soon after i
receive it nothing else happens the program terminates with a segmentation
fault .What might be wrong , have i missed out something , Please guide.
The SDP description from the server (ie testMPEGVideoStreamer) is ,
    v = 0
    o=- 1019484293189188 1 IN IP4 107.108.3.130
    s= SDP description from RTSP server
    t= 0 0
    a = type:broadcast
    m= video 8888 RTP/AVP 32
    a = control:video
    c= IN IP4 239.255.42.42/7


thanks in advance
Niranjan



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From <EMAIL-ADDRESS-HIDDEN> Mon Apr 22 10:34:37 2002
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X-Original-Date: Mon, 22 Apr 2002 10:30:05 -0700
Date: Mon, 22 Apr 2002 10:30:05 -0700
Status: O

At 06:34 AM 4/22/02, Nikki wrote:
>Hi all,
>Im curious about these results - not sure what is wrong here - but Im 
>working with the testMPEGVideoStreamer and testMPEGVideoReceiver programs 
>both running on a localhost - I just wanted to monitor the delay and got 
>these values:
>
>arrival            rtpTimestamp    transit ??????????
>4281287218   1013425969      -1027106047
>4281287218   1013425969      -1027106047
>4281287218   1013425969      -1027106047
>4281287218   1013425969      -1027106047
>4281287218   1013425969      -1027106047
>4281287218   1013425969      -1027106047
>
>Id expect on a local host for the transit to be (or near) 0. The jitter 
>calculations seem ok so Im just not sure what the deal is with the delay?

This is OK.  Because the "transit" number is computed from the RTP 
timestamp (which has a randomly assigned offset from 'real time'), you 
would not expect it to be zero or close ot zero, in general.  What's 
important is the *difference between successive* "transit" values; that's 
what's used to compute the "jitter" value.

>Also, on a related matter (I think!) How can the server monitor the 
>average packet delay to the client - could this be found out using the RTCP RR

Yes.  In fact, this is one of the purposes of RTCP.  The server can compute 
the round-trip delay by looking at the "last SR timestamp" (LSR) and "delay 
since last SR" (DLSR) fields of incoming RTCP RR packets.  See the 
explanation and example at the end of section 6.4.1 of 
<http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-new-11.txt>

         Ross.


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X-Original-Date: Mon, 22 Apr 2002 10:39:30 -0700
Date: Mon, 22 Apr 2002 10:39:30 -0700
Status: O

At 07:27 AM 4/22/02, you wrote:
>I am trying to stream a test.mpg file from the testMPEGVideoStreamer.In the
>testMPEGVideoStreamer i also setup the RTSP server to run at port number
>554.
>I then run the opentRTSP using the following command,
>./openRTSP -V -n rtsp://localhost:554/test.mpg
>The Describe Request is sent and the response for the same is received
>successfully and also i receive the sdp description , but soon after i
>receive it nothing else happens the program terminates with a segmentation
>fault .

This is odd.  Could you please send the complete output from "openRTSP -V ...".

Are you able to receive and play the video OK using QuickTime Player (by 
having it open the file "testMPEGVideo.sdp")?

         Ross.


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Subject: [Live-devel] getting output from testMP3Receiver
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X-Original-Date: Mon, 22 Apr 2002 15:43:58 -0700 (PDT)
Date: Mon, 22 Apr 2002 15:43:58 -0700 (PDT)
Status: O

I'm trying to read the output from testMP3receiver by using popen to pipe 
the output.

    fd1 = popen("testMP3receiver", "r");

I'm trying to read the data into a buffer as follows:

    len = fread((void *)buffer, 1, x, fd1);

The problem is, no matter what x is, len always equals x. So I don't know 
how much data testMP3receiver has put on stdout. Is there a way to 
determine the amount of data that should be read?

Thanks,

Michael Kang


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From <EMAIL-ADDRESS-HIDDEN> Mon Apr 22 16:26:42 2002
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X-Original-Date: Mon, 22 Apr 2002 16:25:46 -0700
Date: Mon, 22 Apr 2002 16:25:46 -0700
Status: O

At 03:43 PM 4/22/02, you wrote:
>I'm trying to read the output from testMP3receiver by using popen to pipe
>the output.
>
>     fd1 = popen("testMP3receiver", "r");
>
>I'm trying to read the data into a buffer as follows:
>
>     len = fread((void *)buffer, 1, x, fd1);
>
>The problem is, no matter what x is, len always equals x. So I don't know
>how much data testMP3receiver has put on stdout.

If you always read the amount of data that you're asking for, then why do 
you care?  Is this because you want to read data on MP3 frame boundaries?

>Is there a way to
>determine the amount of data that should be read?

There's probably some way to figure out how much data is pending in the 
pipe.  That's an OS/runtime-related question, though.)

If you really want to receive and process data on frame boundaries, then a 
better solution would be to modify "FileSink" ("liveMedia/FileSink.cpp") to 
do this.  (Note, in particular, the function 
"FileSink::afterGettingFrame()", which takes a 'frame size' as 
parameter.)  If you do this, then you won't need a separate pipe, and the 
code will also be more efficient.

         Ross.


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X-Original-Date: Mon, 22 Apr 2002 16:56:54 -0700 (PDT)
Date: Mon, 22 Apr 2002 16:56:54 -0700 (PDT)
Status: O

On Mon, 22 Apr 2002, Ross Finlayson wrote:

> >     fd1 = popen("testMP3receiver", "r");
> >     len = fread((void *)buffer, 1, x, fd1);
> 
> If you always read the amount of data that you're asking for, then why do 
> you care?  Is this because you want to read data on MP3 frame boundaries?

Not necessarily... I don't want to wait for the data if it's not there. 
The function that contains the fread statement is called frequently, and 
testMP3receiver may not have put any data in the pipe since the last call.

- Michael


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X-Original-Date: Wed, 24 Apr 2002 01:21:24 -0700
Date: Wed, 24 Apr 2002 01:21:24 -0700
Status: O


>But, though you created an RTPEndpoitDescriptor fTextEndpoint, the
>fRTPSource cannot be set to the BasicUDPSource, so instead no state is kept
>about the GroupSock used for the text stream.  So those groups never leave
>after the player has joined them.
>
>I just added more state to keep a reference to the GroupSock object, since
>there should only be one per session this seems reasonable.  I then clean it
>up myself when it's no longer needed.

That will work.  The "fTextSource" object also need to be closed - i.e.
         Medium::close(fTextSource); fTextSource = NULL
(That's something else that I forgot to do in the "Freeamp" patch.)


>What suggestions do you have about best maintaining the xAudic / live.com
>dependency once they accept my patch?
>
>Perhaps more specifically, would you recommend keeping some version of
>live.com inside their repsitory, for easier developer access (especially
>those not really interested in rtp)?  Then someone, possibly me, would be
>responsible for keeping this code reasonably in sync with main live.com
>tree.
>
>Or, would it be best to keep them entirely external, and just accept that
>anyone trying to build xAudic must independently download the most recent
>version of live.com code?

I like the latter solution the best, because it encourages developers to 
use the latest version of the LIVE.COM libraries.  (The code is not yet 
stable enough to rely on having an old version bundled with the xAudic 
code.)  (Of course, the drawback here is that sometimes the *API* changes 
as well, although API changes are becoming less frequent.)

This (the latter solution) is what I do with both the "Freeamp" and 
"mplayer" patches.  You might also look at making the LIVE.COM patch 
optional, so that people who don't want RTP/RTSP support won't have to deal 
with it.

         Ross.


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From <EMAIL-ADDRESS-HIDDEN> Wed Apr 24 07:02:34 2002
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Subject: Re: [Live-devel] RTPSource - transit calculations?
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X-Original-Date: Wed, 24 Apr 2002 14:56:35 +0100
Date: Wed, 24 Apr 2002 14:56:35 +0100
Status: O

Hi Ross,
From what I can see in the RTCP DLSR and LSR fields is that only the transit
delay for that particular RTCP packet can be inferred.
i.e. TimeNow - DLSR - LSR = transit delay of this RTCP packet?
But for arguments sake - what if this particular packet was anomaly
exceptionally delayed - it would not reflect the true average RTP transit
delay.
What I would like to do is somehow keep track of the average network delay
But this cant be done (or can it) unless the receiver can somehow infer the
RTP timestamp offset?
Anyway - does anyone know why an RTP timestamp offset is used?

Thanks,
Nikki

----- Original Message -----
From: {Ross Finlayson} <EMAIL-ADDRESS-HIDDEN>
To: <EMAIL-ADDRESS-HIDDEN>
Sent: Monday, April 22, 2002 6:30 PM
Subject: Re: [Live-devel] RTPSource - transit calculations?


> At 06:34 AM 4/22/02, Nikki wrote:
> >Hi all,
> >Im curious about these results - not sure what is wrong here - but Im
> >working with the testMPEGVideoStreamer and testMPEGVideoReceiver programs
> >both running on a localhost - I just wanted to monitor the delay and got
> >these values:
> >
> >arrival            rtpTimestamp    transit ??????????
> >4281287218   1013425969      -1027106047
> >4281287218   1013425969      -1027106047
> >4281287218   1013425969      -1027106047
> >4281287218   1013425969      -1027106047
> >4281287218   1013425969      -1027106047
> >4281287218   1013425969      -1027106047
> >
> >Id expect on a local host for the transit to be (or near) 0. The jitter
> >calculations seem ok so Im just not sure what the deal is with the delay?
>
> This is OK.  Because the "transit" number is computed from the RTP
> timestamp (which has a randomly assigned offset from 'real time'), you
> would not expect it to be zero or close ot zero, in general.  What's
> important is the *difference between successive* "transit" values; that's
> what's used to compute the "jitter" value.
>
> >Also, on a related matter (I think!) How can the server monitor the
> >average packet delay to the client - could this be found out using the
RTCP RR
>
> Yes.  In fact, this is one of the purposes of RTCP.  The server can
compute
> the round-trip delay by looking at the "last SR timestamp" (LSR) and
"delay
> since last SR" (DLSR) fields of incoming RTCP RR packets.  See the
> explanation and example at the end of section 6.4.1 of
> <http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-new-11.txt>
>
>          Ross.
>
>
> _______________________________________________
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> Live-devel(at)lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/live-devel


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X-Original-Date: Wed, 24 Apr 2002 09:30:44 -0700
Date: Wed, 24 Apr 2002 09:30:44 -0700
Status: O

At 06:56 AM 4/24/02, you wrote:
>Hi Ross,
> From what I can see in the RTCP DLSR and LSR fields is that only the transit
>delay for that particular RTCP packet can be inferred.
>i.e. TimeNow - DLSR - LSR = transit delay of this RTCP packet?

No, this measures the *round-trip* delay of a *pair* of RTCP packets: a SR 
packet (from the server to the receiving client), and a RR packet (from the 
receiving client back to the server).

>But for arguments sake - what if this particular packet was anomaly
>exceptionally delayed - it would not reflect the true average RTP transit
>delay.

Again, what's being measured here is the round-trip delay, which includes 
the time taken for a RR packet to get back to the server.

>What I would like to do is somehow keep track of the average network delay

Do you mean the *one way* delay from the server to the receiving client?

The way to measure this is to look at incoming SR packets (from the 
server), and looking at the "NTP timestamp" and "RTP timestamp" fields in 
these packets.  This tells you the mapping between 'wall clock' time, and 
the sender's RTP timestamp.  I.e., it tells you the "RTP timestamp 
offset".  You can use this to compute the 'wall clock' time that 
corresponds to the RTP timestamp in each RTP packet, and thus you can 
compute how much it was delayed.

>Anyway - does anyone know why an RTP timestamp offset is used?

The RTP timestamp was never intended to represent 'wall clock' time - it's 
just not large enough.  Instead, the "NTP timestamp" in SR packets is used 
for this purpose.  (The RTP timestamp begins with a random offset so as to 
make known plaintext analysis more difficult when RTP packets are encrypted.)

         Ross.


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Subject: [Live-devel] Regarding testing openRTSP with testMPEGVideoStreamer
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X-Original-Date: Wed, 24 Apr 2002 22:07:02 +0530
Date: Wed, 24 Apr 2002 22:07:02 +0530
Status: O

Hi

I am trying to test the openRTSP with the testMPEGVideoStreamer .I have a
sample test.mpg file .I enable the RTSP server in the
testMPEGVideoStreamer.I also set the "multicastAddressStr" to that of my
machine IP and also in the testMPEGVideo.sdp file i set the field c= to my
machines IP.
Then i start the openRTSP with the command
./openRTSP -V rtsp://localhost:554/test.mpg (as both run in the same
machine)
1>But once i get the response for the describe from testMPEGVideoStreamer
the openRTSP terminates with a segmentation fault .There is no play request
sent from the openRTSP , what could be the problem .

2> Since i am using my own test.mpg file for streaming should i change
something in the testMPEGVideo.sdp file or create a seperate sdp for this
file

thanks in advance,
Niranjan


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From <EMAIL-ADDRESS-HIDDEN> Wed Apr 24 09:53:39 2002
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X-Original-Date: Wed, 24 Apr 2002 09:51:30 -0700
Date: Wed, 24 Apr 2002 09:51:30 -0700
Status: O

At 09:37 AM 4/24/02, you wrote:
>Hi
>
>I am trying to test the openRTSP with the testMPEGVideoStreamer .I have a
>sample test.mpg file .I enable the RTSP server in the
>testMPEGVideoStreamer.I also set the "multicastAddressStr" to that of my
>machine IP and also in the testMPEGVideo.sdp file i set the field c= to my
>machines IP.

If you're using RTSP, then the "testMPEGVideo.sdp" file is not used at all, 
so there's no point in changing it.

The problem here is that the RTSP server that's built in to 
"testMPEGVideoStreamer" (and "testMP3Streamer") currently supports 
*multicast only* - not unicast.  If you want to use "openRTSP" to receive 
the stream sent by "testMPEGVideoStreamer", then - for now - you will need 
to continue streaming using multicast.

Fixing the RTSP server to support unicast streams (and audio+video streams, 
e.g., with "testMPEGAudioVideoStreamer") is on my "to do" list.

If you insist on replacing multicast streaming with unicast (and your only 
reason for wanting to do this should be that the source and destination are 
on different networks without multicast routing between them), then you 
should change "testMPEGVideoStreamer" and "testMPEGVideo.sdp" as follows:

- In "testMPEGVideoStreamer", change "multicastAddressStr" to be the 
(unicast) IP address of the intended destination.
- In "testMPEGVideo.sdp" change the string "239.255.42.42" to the IP 
address of the sender.  (You can also change it to "0.0.0.0", but if you do 
this, no RTCP Reception Reports will get sent back to the sender.)

If you do this, you will be able to watch the stream using "QuickTime 
Player" (by opening the (modified) "testMPEGVideo.sdp" file).  (As I noted 
above, you won't be able to use a "rtsp://" URL.)  Note, however, that you 
will *not* be able to view the stream using the "RealOne Player", as this 
seems to be able to handle multicast streams only.

         Ross.


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X-Original-Date: Thu, 25 Apr 2002 17:20:03 +0530
Date: Thu, 25 Apr 2002 17:20:03 +0530
Status: O

Hi ,


>If you're using RTSP, then the "testMPEGVideo.sdp" file is not used at all,
so there's no point in changing it.

>The problem here is that the RTSP server that's built in to
>"testMPEGVideoStreamer" (and "testMP3Streamer") currently supports
>*multicast only* - not unicast.  If you want to use "openRTSP" to receive
>the stream sent by "testMPEGVideoStreamer", then - for now - you will need
>to continue streaming using multicast.

I agree to you Ross ,and i just want to test the openRTSP with
testMPEGVideoStreamer over Multicast (atleast for the moment).
I tested the testMPEGVideStreamer with the QuickTime player - by making it
open the testMPEGVideo.sdp , everything works fine .
1> But when i tested openRTSP , with the following command --
./openRTSP -V rtsp://239.255.42.42/test.mpg , its says "connection failed()
Network not reachable".But both the programs are running on the same machine
and also Multicasting is enabled in my machine  -- still what could be wrong
. Is my way correct of running the openRTSP for testing
testMPEGVideoStreamer on Multicast .

2> Alternatively i changed the "multicastAddressStr " to my machines IP and
also in the testMPEGVideo.sdp file.
When i run the openRTSP with the command -- ./openRTSP -V
rtsp://localhost/test.mpg .
The connection is established.
But the program terminates with a segmentation fault after receiving the sdp
file , here is the output of the openRTSP programe which i ran on Linux,
[root(at)user]# ./openRTSP -V rtsp://localhost/test.mpg

Sending request: DESCRIBE rtsp://107.108.3.130/test.mpg RTSP/1.0

CSeq: 1

Accept: application/sdp



Received DESCRIBE response: RTSP/1.0 200 OK

CSeq: 1

Content-Type: application/sdp

Content-Length: 178

v=0

o=- 1019733968030495 1 IN IP4 107.108.3.130

s=SDP description from RTSPServer

t=0 0

a=type:broadcast

m=video 8888 RTP/AVP 32

a=control:video

c=IN IP4 107.108.3.130/7

Opened URL "rtsp://107.108.3.130/test.mpg", returning a SDP description:

v=0

o=- 1019733968030495 1 IN IP4 107.108.3.130

s=SDP description from RTSPServer

t=0 0

a=type:broadcast

m=video 8888 RTP/AVP 32

a=control:video

c=IN IP4 107.108.3.130/7

Segmentation fault (core dumped).

Please let me know what could be the problem ,

thanks Ross,
Niranjan



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X-Original-Date: Thu, 25 Apr 2002 09:15:27 -0700 (PDT)
Date: Thu, 25 Apr 2002 09:15:27 -0700 (PDT)
Status: O

hi,
 I am trying to combine Live code with mpeglib code to
give rtsp support to teh mpeglib code.
 in one part of the code,the call to blockMyself()
routine is in a never ending while loop.. when i call
blockMyself() it is retrieving 836 bytes of data and
coming out..reentering fetching again 836 bytes coming
out and reentering..so on 
as i understand, the blockMyself() routine is a never
ending loop.. but i couldnt figure out the reason for
it to come out of the while(1) loop..
it would be of great help if u could help

thank you
surya

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From <EMAIL-ADDRESS-HIDDEN> Thu Apr 25 10:45:29 2002
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X-Original-Date: Thu, 25 Apr 2002 10:41:12 -0700
Date: Thu, 25 Apr 2002 10:41:12 -0700
Status: O

At 04:50 AM 4/25/02, you wrote:
>I tested the testMPEGVideStreamer with the QuickTime player - by making it
>open the testMPEGVideo.sdp , everything works fine .
>1> But when i tested openRTSP , with the following command --
>./openRTSP -V rtsp://239.255.42.42/test.mpg , its says "connection failed()
>Network not reachable".

Instead of "239.255.42.42", the RTSP URL should contain the name (or 
address) of the computer that's running "testMPEGVideoStreamer".  Then it 
should work.  (You can use this URL with "openRTSP", or also with QuickTime 
Player (instead of opening the .sdp file)

         Ross.



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From <EMAIL-ADDRESS-HIDDEN> Thu Apr 25 11:05:57 2002
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X-Original-Date: Thu, 25 Apr 2002 10:57:15 -0700
Date: Thu, 25 Apr 2002 10:57:15 -0700
Status: O

At 09:15 AM 4/25/02, you wrote:
>as i understand, the blockMyself() routine is a never
>ending loop.. but i couldnt figure out the reason for
>it to come out of the while(1) loop..

"blockMyself()" enters an event-handling loop, repeatedly handling pending 
events (arriving network packets, or delay timer expirations).

"blockMyself()" takes an optional "watchVariable" parameter.  If this 
parameter is the address of a 'char', you can use this to signal when 
"blockMyself()" should return.

For example,

         char flag; // global
         ...
         flag = 0;
         blockMyself(&flag);

and then, within an event handler:

         extern char flag;
         flag = ~0;

     Ross.


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X-Original-Date: Fri, 26 Apr 2002 02:05:19 +0530
Date: Fri, 26 Apr 2002 02:05:19 +0530
Status: O

Hi ,

>Instead of "239.255.42.42", the RTSP URL should ?>contain the name (or
>address) of the computer that's running >"testMPEGVideoStreamer".  Then it
>should work.  (You can use this URL with "openRTSP", >or also with
QuickTime
>Player (instead of opening the .sdp file)

I have tried the following command for testing openRTSP with
testMPEGVideoStreamer
./openRTSP -V rtsp://107.108.3.130/test.mpg (Both run in the same machine).
But after the DESCRIBE response the openrtsp program terminates with
segmentation fault.
I still dont understand what could be the problem.Please Guide
The output is as follows,
sending request: DESCRIBE rtsp://----------------------------
Received DESCRIBE response: RTSP/1.0 200 OK -----------------------
Opened URL "rtsp://107.108.3.130/testMPEGVideo.sdp",--------------
v=0,o=-101......,c= IN IP4 239.255.42.42/7
segementation fault(core dumped)

I have tested testMPEGVideoStreamer with the quicktimeplayer it works fine.

thanks
niranjan.



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X-Original-Date: Thu, 25 Apr 2002 13:55:43 -0700
Date: Thu, 25 Apr 2002 13:55:43 -0700
Status: O

At 01:35 PM 4/25/02, you wrote:
>I have tried the following command for testing openRTSP with
>testMPEGVideoStreamer
>./openRTSP -V rtsp://107.108.3.130/test.mpg (Both run in the same machine).
>But after the DESCRIBE response the openrtsp program terminates with
>segmentation fault.
>I still dont understand what could be the problem.Please Guide
>The output is as follows,
>sending request: DESCRIBE rtsp://----------------------------
>Received DESCRIBE response: RTSP/1.0 200 OK -----------------------
>Opened URL "rtsp://107.108.3.130/testMPEGVideo.sdp",--------------
>v=0,o=-101......,c= IN IP4 239.255.42.42/7
>segementation fault(core dumped)

This is a mystery to me also.  What operating system are you running 
"openRTSP" (and "testMPEGVideoStreamer") on?

Do you get the same result if you use the pre-built binary version of 
"openRTSP", available at <http://www.live.com/openRTSP/binaries/>

         Ross.


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X-Original-Date: Thu, 25 Apr 2002 16:22:34 +0200
Date: Thu, 25 Apr 2002 16:22:34 +0200
Status: O

>>>>> "n" == 94426082  <Nikki> writes:

    n> Hi Ross, Thats a great help! Actually Im doing research of
    n> adaptive QoS for Streamed multimedia (mp4). 

[SNIP]

    n> So thats what I hope to do using this stack.  If you're
    n> interested - Ive a few papers on the idea.

Are there papers publicly available?

bye,

andrea glorioso
--
Ma tu rifiuti di ascoltare			    andrea glorioso
Ogni segnale che ti fa cambiare			   
Perche` ti fa paura quello che succedera`	       www.cpsr.org
Se poi ti senti uguale				sama(at)perchetopi.org

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Subject: [Live-devel] RTCP SR Offset calculations!
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Hi all,
I have been struggling trying to calculate the offset from the Sender =
report RTP Timestamp to transit calculations.=20
I have tried many many different solutions but none worked. Basically =
the problem I seem to be having is that the when the time is being =
converted to a RTP timestamp in RTPSink.cpp there are 2 unknowns that =
need to be inferred - the TimestampBase and TimestampFrequency. I =
thought the timestamp frequency could be taken/guessed from the =
ReceptionStatsDB. Then replicate the RTPTimestamp decoding in a similar =
way as in the RTPSource where the jitter calculations are done. But for =
some strange reason the offset calculated changes value - thus =
monitoring any values is near impossible!=20
Unless I could modify the method=20
unsigned RTPSink::convertToRTPTimestamp(......)
so thats it easier to infer the offset on the client side.
Any help with this would be greatly appreciated,
Rgds,
Nikki=20

RTCP.cpp   =20

   struct timeval timeNow;
  gettimeofday(&timeNow, &Idunno);

  unsigned freq =3D instance->fSource->timestampFrequency();

  unsigned arrival =3D (freq*timeNow.tv_sec);
  arrival +=3D (unsigned)
   ((2.0*freq*timeNow.tv_usec + 1000000.0)/2000000);

  unsigned ntptime =3D (freq*NTPmsw);
  ntptime +=3D (unsigned)
   ((2.0*freq*NTPlsw + 1000000.0)/2000000);

  unsigned offset =3D ntptime - RTPts;
            // note: rounding
  int delay =3D arrival - RTPts;

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 5.50.4134.600" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Hi all,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>I have been struggling trying to =
calculate the=20
offset from the Sender report RTP Timestamp to transit calculations.=20
</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>I have tried many many different =
solutions but none=20
worked. Basically the problem I seem to be having is that the when the =
time is=20
being converted to a RTP timestamp in RTPSink.cpp there are 2 unknowns =
that need=20
to be inferred - the TimestampBase and TimestampFrequency. I thought the =

timestamp frequency could be taken/guessed from the =
ReceptionStatsDB.&nbsp;Then=20
replicate the&nbsp;RTPTimestamp decoding&nbsp;in a similar way as in the =

RTPSource&nbsp;where the jitter calculations are done. But for some =
strange=20
reason the offset calculated changes value - thus monitoring any values =
is near=20
impossible! </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Unless I could modify the method =
</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>unsigned=20
RTPSink::convertToRTPTimestamp(......)</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>so thats it easier to infer the offset =
on the=20
client side.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Any help with this would be greatly=20
appreciated,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Rgds,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Nikki</FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>RTCP.cpp&nbsp;&nbsp;&nbsp; =
</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp; struct timeval=20
timeNow;<BR>&nbsp;&nbsp;gettimeofday(&amp;timeNow,=20
&amp;Idunno);</FONT></DIV><FONT face=3DArial size=3D2>
<DIV><BR>&nbsp;&nbsp;unsigned freq =3D=20
instance-&gt;fSource-&gt;timestampFrequency();<BR></DIV>
<DIV>&nbsp;&nbsp;unsigned arrival =3D=20
(freq*timeNow.tv_sec);<BR>&nbsp;&nbsp;arrival +=3D=20
(unsigned)<BR>&nbsp;&nbsp;&nbsp;((2.0*freq*timeNow.tv_usec +=20
1000000.0)/2000000);</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp;unsigned ntptime =3D=20
(freq*NTPmsw);<BR>&nbsp;&nbsp;ntptime +=3D=20
(unsigned)<BR>&nbsp;&nbsp;&nbsp;((2.0*freq*NTPlsw +=20
1000000.0)/2000000);</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp;unsigned offset =3D ntptime =
-=20
RTPts;<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp; //=20
note: rounding</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp;int delay =3D arrival -=20
RTPts;</FONT></DIV></BODY></HTML>

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From <EMAIL-ADDRESS-HIDDEN> Fri Apr 26 10:08:34 2002
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Subject: Re: [Live-devel] RTCP SR Offset calculations!
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X-Original-Date: Fri, 26 Apr 2002 10:06:36 -0700
Date: Fri, 26 Apr 2002 10:06:36 -0700
Status: O

At 08:05 AM 4/26/02, you wrote:
>I have been struggling trying to calculate the offset from the Sender 
>report RTP Timestamp to transit calculations.

Again, the way to measure this is to look at incoming SR packets (from the 
server), and look at the "NTP timestamp" and "RTP timestamp" fields in 
these packets.  This tells you the mapping between 'wall clock' time, and 
the sender's RTP timestamp.

I.e., let N be the "NTP timestamp" in the last-received SR packet, and let 
R be the "RTP timestamp" in the last-received SR packet.  Then, whenever 
you receive a subsequent RTP packet, with timestamp T, you know that this 
corresponds to 'wall clock' time
         N + (T-R)/F
where F is the RTP timestamp frequency, which you can determine by calling
         RTPSource::timestampFrequency()

I hope this helps.

         Ross.


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From <EMAIL-ADDRESS-HIDDEN> Fri Apr 26 10:21:06 2002
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X-Original-Date: Fri, 26 Apr 2002 10:19:22 -0700
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One more thing.  In the formular that I gave in my last message
	N + (T-R)/F
the subtraction (T-R) is done modulo 65536.

	Ross.


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From <EMAIL-ADDRESS-HIDDEN> Sat Apr 27 01:03:10 2002
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Subject: [Live-devel] Can any player be used to play from an rtsp url?
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 hello,
I want to Know if there is a player that can play video/audio, directly from an rtsp url? and what is the aim of the openRTSP?
thanks?



---------------------------------
Yahoo! Mail -- Une adresse (at)yahoo.fr gratuite et en français !

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<P>&nbsp;hello,
<P>I want to Know if there is a player that can play video/audio, directly from an rtsp url? and what is the aim of the openRTSP?
<P>thanks?</P><p><br><hr size=1><a href="http://fr.mail.yahoo.com">Yahoo! Mail</a> -- Une adresse (at)yahoo.fr gratuite et en français !<br>
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From <EMAIL-ADDRESS-HIDDEN> Sat Apr 27 02:24:00 2002
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X-Original-Date: Sat, 27 Apr 2002 02:21:57 -0700
Date: Sat, 27 Apr 2002 02:21:57 -0700
Status: O

At 01:01 AM 4/27/02, you wrote:
>I want to Know if there is a player that can play video/audio, directly 
>from an rtsp url?

Yes, QuickTime Player and RealOne Player can both do this.  Also, for MP3 
streams, Winamp can do this, if you use the "in_rtp.dll" plugin: 
<http://www.live.com/multikit/winamp-plugin.html>


>  and what is the aim of the openRTSP?

The "openRTSP" program has two main purposes:
1/ to allow people to record incoming audio and/or video streams (rather 
than only playing them)
2/ to let people visualize the various RTSP commands as they are being sent

         Ross.


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Subject: [Live-devel] opening an RTSP url??
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  The "openRTSP" program has two main purposes:
1/ to allow people to record incoming audio and/or video streams (rather 
than only playing them)

OK but how can I play video directly from an rtsp url using Realplayer with testMPEGvideostreamer for example ? 

thanks 

achraf  


.




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<P> 
<P>&nbsp; The "openRTSP" program has two main purposes:<BR>1/ to allow people to record incoming audio and/or video streams (rather <BR>than only playing them)</P>
<P>OK but&nbsp;how can I&nbsp;play video directly from an rtsp url using Realplayer with testMPEGvideostreamer for example ?&nbsp;</P>
<P>thanks&nbsp;</P>
<P>achraf&nbsp;&nbsp;<BR><BR><BR>.<BR><BR><BR><BR><BR>--__--__--<BR><BR>_______________________________________________<BR>Live-devel mailing list<BR>Live-devel(at)lists.sourceforge.net<BR>https://lists.sourceforge.net/lists/listinfo/live-devel<BR><BR><BR>End of Live-devel Digest</P><p><br><hr size=1><a href="http://fr.mail.yahoo.com">Yahoo! Mail</a> -- Une adresse (at)yahoo.fr gratuite et en français !<br>
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From <EMAIL-ADDRESS-HIDDEN> Tue Apr 30 03:39:49 2002
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X-Original-Date: Tue, 30 Apr 2002 03:37:09 -0700
Date: Tue, 30 Apr 2002 03:37:09 -0700
Status: O

At 01:46 AM 4/29/02, Achraf wrote:
>OK but how can I play video directly from an rtsp url using Realplayer 
>with testMPEGvideostreamer for example ?

Unfortunately RealPlayer (including the new "RealOne Player") has a bug 
that prevents it from playing multicast streams via RTSP.

However, QuickTime Player (version 5 or later) will work, provided that the 
MPEG video stream is MPEG-1, not MPEG-2.

First, of course, the RTSP server that's built into "testMPEGVideoStreamer" 
needs to be enabled - by uncommenting the line
         #define IMPLEMENT_RTSP_SERVER 1

Then, in QuickTime Player, do
- File
- Open URL in New Player
- enter
         rtsp://<server-name>
where <server-name> is the domain name of the computer on which you're 
running "testMPEGVideoStreamer".

         Ross.

ps. *Don't* try to change "testMPEGVideoStreamer" to stream using unicast 
rather than multicast.  The built-in RTSP server currently doesn't work for 
unicast streams.



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From <EMAIL-ADDRESS-HIDDEN> Tue Apr 30 13:07:35 2002
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Subject: [Live-devel] does realplayer support the rtp potocol?
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X-Original-Date: Tue, 30 Apr 2002 22:01:26 +0200
Date: Tue, 30 Apr 2002 22:01:26 +0200
Status: O

hello!

I red in the documentation that openRTSP that realplayer do not
support RTP protocol, since it implements the RDT technology.

I tried to open the suggested url rtsp://media.schreiner.edu/realmp3.mp3
using realplayer and the media is correctly played.
However the SDP file specifies RTP/AVP in the media field.

Does realplayer now support the rtp protocol?


about the mplayer, does it support the rtsp protocol?

thanks, domenico

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X-Original-Date: Tue, 30 Apr 2002 13:17:24 -0700 (Pacific Daylight Time)
Date: Tue, 30 Apr 2002 13:17:24 -0700 (Pacific Daylight Time)
Status: O

On Tue, 30 Apr 2002, miros2001(at)katamail.com wrote:
> I red in the documentation that openRTSP that realplayer do not
> support RTP protocol, since it implements the RDT technology.

Not true.  RealPlayer supports both RTP and RDT.  When a RealPlayer
connects to a RealServer, it will use RDT by default.  However, when a
RealPlayer connects to a standard RTSP/RTP server, it will use RTP.

The RealOne Player supports RFC 2250 playback of MP3 data.  The RealSystem
Server 8.0 does not support RTP playback from .mp3 files.  However, a
properly-hinted .mov file with MP3 data on a the server should play back
to any RTSP/RTP/MP3 client using either RFC 2250 or RFC 3119.

Rob



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From <EMAIL-ADDRESS-HIDDEN> Tue Apr 30 13:21:10 2002
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X-Original-Date: Tue, 30 Apr 2002 13:18:13 -0700
Date: Tue, 30 Apr 2002 13:18:13 -0700
Status: RO

At 01:01 PM 4/30/02, you wrote:
>I red in the documentation that openRTSP that realplayer do not
>support RTP protocol, since it implements the RDT technology.

Not quite.  RealPlayer does not use RTP when playing *RealAudio* or 
*RealVideo* sessions - i.e., sessions that use RealNetworks' proprietary 
codecs.  However, when playing other codecs (such as MP3, or MPEG video), 
it uses RTP.  That's why <rtsp://media.schreiner.edu/realmp3.mp3> works.


>about the mplayer, does it support the rtsp protocol?

No, not to my knowledge.  It supports some proprietary Microsoft control 
protocol only.

         Ross.


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From <EMAIL-ADDRESS-HIDDEN> Wed May 1 06:52:24 2002
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X-Original-Date: Wed, 1 May 2002 15:49:23 +0200
Date: Wed, 1 May 2002 15:49:23 +0200

Ross, Rob, 

thanks for your clarifications ;-)

Here is the address for MPEG1 streaming player for Linux:

http://opensource.eprocess.fr/popcorn/index_html

It is an open source code, and supports the RTSP protocol;

However I don’t know  if the RTP is also supported.

I hope it is interesting.

Domenico


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From <EMAIL-ADDRESS-HIDDEN> Wed May 1 12:25:05 2002
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X-Original-Date: Wed, 01 May 2002 11:55:39 -0700
Date: Wed, 01 May 2002 11:55:39 -0700

At 06:49 AM 5/1/02, you wrote:
>Here is the address for MPEG1 streaming player for Linux:
>
>http://opensource.eprocess.fr/popcorn/index_html
>
>It is an open source code, and supports the RTSP protocol;
>
>However I don't know  if the RTP is also supported.

Domenico,

This is interesting - I hadn't seen this before.  If it supports RTSP, then 
it must also support RTP, although I didn't see anything in the source code 
that mentioned RTP.  The code seems to be built using XMMS plugins, so 
perhaps it relies upon some  XMMS MPEG plugin that also supports RTP?

In any case, there is also the RTP streaming patch for "mplayer" that I 
recently developed.  This uses the LIVE.COM Streaming Media library.  For 
more information about this, see <http://www.live.com/mplayer/>

         Ross.


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X-Original-Date: Thu, 2 May 2002 20:04:15 +0200
Date: Thu, 2 May 2002 20:04:15 +0200

>The code seems to be built using XMMS plugins, so 
perhaps it relies upon some  XMMS MPEG plugin that also supports RTP?

I think so..


I'd like to see if it works  as soon as possible.

Regards :-)

Domenico




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Subject: [Live-devel] Changing testMPEGVideoStreaming to streaming more than one video!!!??
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X-Original-Date: Mon, 6 May 2002 10:34:20 +0200 (CEST)
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  Hello,
In order to streaming more than one video (two videos for instance) throught testMPEGAudioVideoStreamer, I have made those changes:
- Initialisate two groupsockets for the the two video as follow:
        *

  const unsigned short rtpPortNumAudio2 = 6666;
  const unsigned short rtcpPortNumAudio2 = rtpPortNumAudio2+1;
  const unsigned short rtpPortNumVideo2 = 8888;
  const unsigned short rtcpPortNumVideo2 = rtpPortNumVideo2+1;
  //
  const unsigned short rtpPortNumAudio1 = 7777;
  const unsigned short rtcpPortNumAudio1 = rtpPortNumAudio1+1;
  const unsigned short rtpPortNumVideo1 = 9999;
  const unsigned short rtcpPortNumVideo1 = rtpPortNumVideo1+1;
  const unsigned char ttl = 7; // low, in case routers don't admin scope 

const Port rtpPortAudio1(rtpPortNumAudio1);
  const Port rtcpPortAudio1(rtcpPortNumAudio1);
  const Port rtpPortVideo1(rtpPortNumVideo1);
  const Port rtcpPortVideo1(rtcpPortNumVideo1);

  const Port rtpPortAudio2(rtpPortNumAudio2);
  const Port rtcpPortAudio2(rtcpPortNumAudio2);
  const Port rtpPortVideo2(rtpPortNumVideo2);
  const Port rtcpPortVideo2(rtcpPortNumVideo2);

........

  Groupsock rtpGroupsockAudio2(*env, multicastAddress, rtpPortAudio2, ttl);
  Groupsock rtcpGroupsockAudio2(*env, multicastAddress2, rtcpPortAudio2, ttl);
  
  Groupsock rtpGroupsockAudio1(*env, multicastAddress, rtpPortAudio1, ttl);
  Groupsock rtcpGroupsockAudio1(*env, multicastAddress, rtcpPortAudio1, ttl);
  
  Groupsock rtpGroupsockVideo2(*env, multicastAddress, rtpPortVideo2, ttl);
  Groupsock rtcpGroupsockVideo2(*env, multicastAddress, rtcpPortVideo2, ttl);
  
  Groupsock rtpGroupsockVideo1(*env, multicastAddress, rtpPortVideo1, ttl);
  Groupsock rtcpGroupsockVideo1(*env, multicastAddress, rtcpPortVideo1, ttl);

........

  videoSink1->startPlaying(*videoSource1, afterPlaying, videoSink1);
  audioSink1->startPlaying(*audioSource1, afterPlaying, audioSink1);
  
   videoSink2->startPlaying(*videoSource2, afterPlaying, videoSink2);
   audioSink2->startPlaying(*audioSource2, afterPlaying, audioSink2);


I have wrote two sdp files for the two videos. But always the player play only from the first groupsockets (the first appearing in the code source (i.e, rtpGroupsock...2) ).

Any Idea thanks.

Achraf  

 



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<P> 
<P>&nbsp; Hello,
<P>In order to streaming more than one video (two videos for instance) throught testMPEGAudioVideoStreamer, I have made those changes:
<P>- Initialisate two groupsockets for the&nbsp;the&nbsp;two video as follow:
<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; *</P>
<P>&nbsp; const unsigned short rtpPortNumAudio2 = 6666;<BR>&nbsp; const unsigned short rtcpPortNumAudio2 = rtpPortNumAudio2+1;<BR>&nbsp; const unsigned short rtpPortNumVideo2 = 8888;<BR>&nbsp; const unsigned short rtcpPortNumVideo2 = rtpPortNumVideo2+1;<BR>&nbsp; //<BR>&nbsp; const unsigned short rtpPortNumAudio1 = 7777;<BR>&nbsp; const unsigned short rtcpPortNumAudio1 = rtpPortNumAudio1+1;<BR>&nbsp; const unsigned short rtpPortNumVideo1 = 9999;<BR>&nbsp; const unsigned short rtcpPortNumVideo1 = rtpPortNumVideo1+1;<BR>&nbsp; const unsigned char ttl = 7; // low, in case routers don't admin scope </P>
<P>const Port rtpPortAudio1(rtpPortNumAudio1);<BR>&nbsp; const Port rtcpPortAudio1(rtcpPortNumAudio1);<BR>&nbsp; const Port rtpPortVideo1(rtpPortNumVideo1);<BR>&nbsp; const Port rtcpPortVideo1(rtcpPortNumVideo1);</P>
<P>&nbsp; const Port rtpPortAudio2(rtpPortNumAudio2);<BR>&nbsp; const Port rtcpPortAudio2(rtcpPortNumAudio2);<BR>&nbsp; const Port rtpPortVideo2(rtpPortNumVideo2);<BR>&nbsp; const Port rtcpPortVideo2(rtcpPortNumVideo2);</P>
<P>........</P>
<P>&nbsp; Groupsock rtpGroupsockAudio2(*env, multicastAddress, rtpPortAudio2, ttl);<BR>&nbsp; Groupsock rtcpGroupsockAudio2(*env, multicastAddress2, rtcpPortAudio2, ttl);<BR>&nbsp; <BR>&nbsp; Groupsock rtpGroupsockAudio1(*env, multicastAddress, rtpPortAudio1, ttl);<BR>&nbsp; Groupsock rtcpGroupsockAudio1(*env, multicastAddress, rtcpPortAudio1, ttl);<BR>&nbsp; <BR>&nbsp; Groupsock rtpGroupsockVideo2(*env, multicastAddress, rtpPortVideo2, ttl);<BR>&nbsp; Groupsock rtcpGroupsockVideo2(*env, multicastAddress, rtcpPortVideo2, ttl);<BR>&nbsp; <BR>&nbsp; Groupsock rtpGroupsockVideo1(*env, multicastAddress, rtpPortVideo1, ttl);<BR>&nbsp; Groupsock rtcpGroupsockVideo1(*env, multicastAddress, rtcpPortVideo1, ttl);</P>
<P>........</P>
<P>&nbsp; videoSink1-&gt;startPlaying(*videoSource1, afterPlaying, videoSink1);<BR>&nbsp; audioSink1-&gt;startPlaying(*audioSource1, afterPlaying, audioSink1);<BR>&nbsp; <BR>&nbsp;&nbsp; videoSink2-&gt;startPlaying(*videoSource2, afterPlaying, videoSink2);<BR>&nbsp;&nbsp; audioSink2-&gt;startPlaying(*audioSource2, afterPlaying, audioSink2);<BR></P>
<P>I have wrote two sdp files for the two videos.&nbsp;But&nbsp;always the&nbsp;player play only from the first groupsockets&nbsp;(the first appearing in the code source (i.e, rtpGroupsock...2) ).</P>
<P>Any Idea thanks.</P>
<P>Achraf&nbsp; </P>
<P>&nbsp;</P><p><br><hr size=1><a href="http://fr.mail.yahoo.com">Yahoo! Mail</a> -- Une adresse (at)yahoo.fr gratuite et en français !<br>
--0-921052892-1020674060=:48939--

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X-Original-Date: Mon, 06 May 2002 01:42:56 -0700
Date: Mon, 06 May 2002 01:42:56 -0700

At 01:34 AM 5/6/02, you wrote:
>   const unsigned short rtpPortNumAudio1 = 7777;
>   const unsigned short rtpPortNumVideo1 = 9999;

This is the problem.  The RTP port number must always be even (and the RTCP 
port - the next port - is odd).  So, use something like 7776 and 9998; not 
7777 and 9999.

         Ross.


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Subject: [Live-devel] RTSP server - DESCRIBE response
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X-Original-Date: Mon, 6 May 2002 18:14:11 +0100
Date: Mon, 6 May 2002 18:14:11 +0100

	Greetings
	While using the RTSP server in testMPEGVideo, I've noticed the following
behavior. I'd like to know if this is correct, and also make some questions:

	- The client opens the URL rtsp://server.add , sending the DESCRIBE command
	- The server replies with an RTSP/SDP message, which has a=control:video
(being 'video' the sesion identifier on the server).
	- The client responds with a SETUP message with content:
"RTP/AVP;unicast;client_port=8888-8889" (etc.)
	- The server replies with
"RTP/AVP;multicast;destination=239.255.42.42;port=8888-8889;ttl=7" (etc.)

	Questions:
	1) Is it really necessary to include the "a=control:video" line? Have you
tried, while using QT player sending an RTSP/SDP reply without this
information?
	2) From RFC 2326 I gather that if the client is able to receive multicast
it will indicate so on the transport field of the SETUP message. QT does not
do that, but still you respond to the SETUP with a multicast address. Is
this correct?

Thank you,
David Pontes


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X-Original-Date: Mon, 06 May 2002 11:57:07 -0700
Date: Mon, 06 May 2002 11:57:07 -0700

At 10:14 AM 5/6/02, you wrote:
>         Greetings
>         While using the RTSP server in testMPEGVideo, I've noticed the 
> following
>behavior. I'd like to know if this is correct, and also make some questions:
>
>         - The client opens the URL rtsp://server.add , sending the 
> DESCRIBE command
>         - The server replies with an RTSP/SDP message, which has 
> a=control:video
>(being 'video' the sesion identifier on the server).
>         - The client responds with a SETUP message with content:
>"RTP/AVP;unicast;client_port=8888-8889" (etc.)
>         - The server replies with
>"RTP/AVP;multicast;destination=239.255.42.42;port=8888-8889;ttl=7" (etc.)
>
>         Questions:
>         1) Is it really necessary to include the "a=control:video" line?

In this case, probably not, because there's only one medium (video) in this 
session.  However, if the RTSP server were to support a combined 
audio+video session (something that it doesn't yet do, by the way), then 
the "a=control:..." lines could be useful for clients who wished to play 
only one of the two media.

>Have you
>tried, while using QT player sending an RTSP/SDP reply without this
>information?

No, but it's something that you could try yourself.  Remember that you have 
full source code to "testMPEGVideoStreamer".  (The code for the RTSP server 
is "liveMedia/RTSPServer.cpp")

But I'm curious - why do you care?  QT Player can play the multicast 
session from "testMPEGVideoStreamer" using RTSP just fine.  Are you seeing 
a problem?


>         2) From RFC 2326 I gather that if the client is able to receive 
> multicast
>it will indicate so on the transport field of the SETUP message. QT does not
>do that, but still you respond to the SETUP with a multicast address. Is
>this correct?

I'm not sure, but in this case it's a moot point, because multicast is the 
only transport that this server currently supports.

         Ross.


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X-Original-Date: Tue, 7 May 2002 15:54:12 +0200 (CEST)
Date: Tue, 7 May 2002 15:54:12 +0200 (CEST)

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 Thanks Ross,
The problem of streaming many video is resolved! Now I want to know if I can change the streaming bit rate when pumping video. and This is, in order to accomodate the maximum of video streams.
Thanks
Achraf   



---------------------------------
Yahoo! Mail -- Une adresse (at)yahoo.fr gratuite et en français !

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<P>&nbsp;Thanks Ross,
<P>The problem of streaming many video is resolved! Now I want to know if I can change the streaming bit rate when pumping video.&nbsp;and This is, in&nbsp;order to accomodate the maximum of&nbsp;video streams.
<P>Thanks
<P>Achraf&nbsp;&nbsp; </P><p><br><hr size=1><a href="http://fr.mail.yahoo.com">Yahoo! Mail</a> -- Une adresse (at)yahoo.fr gratuite et en français !<br>
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From: {David Pontes} <EMAIL-ADDRESS-HIDDEN>
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Subject: RE: [Live-devel] RTSP server - DESCRIBE response
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X-Original-Date: Tue, 7 May 2002 18:42:29 +0100
Date: Tue, 7 May 2002 18:42:29 +0100


> At 10:14 AM 5/6/02, you wrote:
> >         Questions:
> >         1) Is it really necessary to include the "a=control:video" line?
>
> >Have you
> >tried, while using QT player sending an RTSP/SDP reply without this
> >information?
>
> No, but it's something that you could try yourself.  Remember
> that you have
> full source code to "testMPEGVideoStreamer".  (The code for the
> RTSP server
> is "liveMedia/RTSPServer.cpp")
>
> But I'm curious - why do you care?  QT Player can play the multicast
> session from "testMPEGVideoStreamer" using RTSP just fine.  Are
> you seeing
> a problem?
>

	I've tried to change other RTSP servers to reply with an SDP without an
'a=control' line and QT didn't do well. I was trying to ascertain if the
problem was with QT or with my hack.
	I've altered the DESCRIBE response the RTSP Server gives (implemented in
RTSPServer.cpp) as to reply with "m=video RTP/AVP 8888 32" and "c=IN IP4
239.255.42.42", among the other parameters (no "a=control:video", though),
and QT reported '5402 - Bad Data'. This is the same result I got in my other
experiment with the other RTSP server (Quick Time Streaming Server, FYI).
Added to the opinions I got from other people, I'm concluding that QT has
some limitations with joining an RTP session based only on the SDP file
returned with RTSP's DESCRIBE response (no SETUP sequence).
	If this were not true (if there was no limitation with QT), it should be
easy to have a client receive both audio and video with the RTSP server you
now implement - the client saw the SDP file and simply joined the session,
as if you opened the SDP file with QTPlayer. The limitation doesn't make
this possible, but there's always HTTP.

Thanks,
David Pontes


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Subject: [Live-devel] SDP Response.
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Hello:

I'm getting the following information ONLY from my RTSP Server:

RTSP 1.0
Session
Range: ntp=3D

My RTSP server doesn't send the information like 'a=3Dcontro...' and =
'c=3D...' etc.  Is there any solution for that.

Regards
Kannan

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2600.0" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Hello:</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>I'm getting the following information =
ONLY from my=20
RTSP Server:</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>RTSP 1.0</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Session</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Range: ntp=3D</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>My RTSP server doesn't send the =
information like=20
'a=3Dcontro...' and 'c=3D...' etc.&nbsp; Is there any solution for=20
that.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Regards</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Kannan</FONT></DIV></BODY></HTML>

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X-Original-Date: Tue, 07 May 2002 15:29:10 -0700
Date: Tue, 07 May 2002 15:29:10 -0700

At 03:00 PM 5/7/02, you wrote:
>Hello:
>
>I'm getting the following information ONLY from my RTSP Server:
>
>RTSP 1.0
>Session
>Range: ntp=
>
>My RTSP server doesn't send the information like 'a=contro...' and 'c=...' 
>etc.

If this is the case, then your RTSP server is badly broken.

Are you sure that this is all that the server is sending back (in response 
to a "DESCRIBE" request)?  Could you mail the output of running
         openRTSP -V
with a rtsp:// URL for your server?

>Is there any solution for that.

Yes, get your RTSP server fixed :-)  Where is your servre running?  Who 
wrote it?

         Ross.


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X-Original-Date: Tue, 07 May 2002 15:46:25 -0700
Date: Tue, 07 May 2002 15:46:25 -0700

At 10:42 AM 5/7/02, you wrote:
>         I've tried to change other RTSP servers to reply with an SDP 
> without an
>'a=control' line and QT didn't do well. I was trying to ascertain if the
>problem was with QT or with my hack.
>         I've altered the DESCRIBE response the RTSP Server gives 
> (implemented in
>RTSPServer.cpp) as to reply with "m=video RTP/AVP 8888 32" and "c=IN IP4
>239.255.42.42", among the other parameters (no "a=control:video", though),
>and QT reported '5402 - Bad Data'. This is the same result I got in my other
>experiment with the other RTSP server (Quick Time Streaming Server, FYI).

OK, so QT Player requires that the SDP returned by a "DESCRIBE" request 
contain "a=control:..." lines.  This is not surprising (and probably not a 
bug).


>Added to the opinions I got from other people, I'm concluding that QT has
>some limitations with joining an RTP session based only on the SDP file
>returned with RTSP's DESCRIBE response (no SETUP sequence).

I'm not sure I understand what these 'limitations' are.  The RTSP protocol 
*requires* that "SETUP" be used (following a "DESCRIBE"), and the QT Player 
does, indeed, do a "SETUP" (following the "DESCRIBE"), so I don't see that 
the QT Player is doing anything wrong.


>         If this were not true (if there was no limitation with QT), it 
> should be
>easy to have a client receive both audio and video with the RTSP server you
>now implement - the client saw the SDP file and simply joined the session,
>as if you opened the SDP file with QTPlayer.

As far as I can tell, the only reason why QT Player can't play a RTSP 
audio+video session using my RTSP server (the one in "liveMedia") is that 
my RTSP server implementation currently doesn't support this.

The QuickTime Streaming Server - another implementation of a RTSP server - 
apparently has a different bug: It cannot serve up multicast sessions to 
RTSP clients.

         Ross.


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X-Original-Date: Tue, 07 May 2002 15:53:54 -0700
Date: Tue, 07 May 2002 15:53:54 -0700

At 06:54 AM 5/7/02, you wrote:
>The problem of streaming many video is resolved! Now I want to know if I 
>can change the streaming bit rate when pumping video. and This is, in 
>order to accomodate the maximum of video streams.

The MPEG streaming code should already be streaming each stream at its 
natural frame rate - which it gets from the "frame_rate_code" parameter in 
each MPEG Video Sequence Header.  See "MPEGVideoStreamFramer.cpp", lines 
372-373.  If you want to change this frame rate (although I'm not sure why 
you'd want to do this), then this would be the place to do it.

         Ross.


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Subject: Re: [Live-devel] SDP Response.
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X-Original-Date: Tue, 07 May 2002 16:02:25 -0700
Date: Tue, 07 May 2002 16:02:25 -0700

At 03:56 PM 5/7/02, you wrote:
>Thanks Ross.  Below is the output of  -V option.
>
>---------------------
>Opened URL "rtsp://24.126.219.38/YimaIntro.mov", returning a SDP
>description:
>RTSP/1.0
>  OK Range: npt=0.000000-604.000000
>session: 2171515680890927956
>
>Failed to create a MediaSession object from the SDP description: Invalid SDP
>line: RTSP/1.0
>  OK Range: npt=0.000000-604.000000
>session: 2171515680890927956
>---------------------

OK, so this confirms that your 'RTSP server', by not returning a proper SDP 
description in response to a "DESCRIBE", is horribly 
non-compliant.  Whatever this thing is, it shouldn't be calling itself a 
RTSP server.

         Ross.


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Subject: [Live-devel] running more than RTSP server ?
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X-Original-Date: Thu, 9 May 2002 14:20:13 +0200 (CEST)
Date: Thu, 9 May 2002 14:20:13 +0200 (CEST)

 
Hello,
 In order to send more than DESCRIBE message (i.e,I
stream more than one video, on different RTP session)I
have added an other RTSP Server to
testMPEGVideoStreamer working on port 552 (the first
is working on prot 554).
 Open I try to connect to the two RTSP server throught
the openRTSP (openRTSP -p 552 ...) I get the DESCRIBE
of the first video session.
 when I let only one server working on port other than
554 I can't receive the DESCRIBE message from the
server. When I set the port number to 554 the problem
is fixed.
 
 Note that I stream two videos on different RTP
session(
the first on port 7778 and the secon is on port 8888)
Any Idea thanks
Achraf

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X-Original-Date: Thu, 09 May 2002 08:26:09 -0700
Date: Thu, 09 May 2002 08:26:09 -0700

At 05:20 AM 5/9/02, you wrote:
>Hello,
>  In order to send more than DESCRIBE message (i.e,I
>stream more than one video, on different RTP session)I
>have added an other RTSP Server to
>testMPEGVideoStreamer working on port 552 (the first
>is working on prot 554).
>  Open I try to connect to the two RTSP server throught
>the openRTSP (openRTSP -p 552 ...)

No, that's not the right way to run "openRTSP".  The "-p" argument is used 
to (optionally) specify the port number that's used by the client to 
receive *RTP* packets.  (In your case, you won't need to use this argument, 
because - for multicast sessions - these client port numbers are specified 
in the SDP description.)

The port number used for the *RTSP* (i.e., TCP) session is specified in the 
"rtsp://" URL.  (If you don't specify a port number in the "rtsp://" URL, 
then 554 is assumed; i.e., 554 is the default port number for RTSP 
(analogous to port 80 for HTTP).)

So, instead, you should be running
         openRTSP rtsp://<server-name>:554/
or just
         openRTSP rtsp://<server-name>/
for your "554" session, and
         openRTSP rtsp://<server-name>:552/
for the "552" session.

         Ross.


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Subject: [Live-devel] Regarding smil
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X-Original-Date: Mon, 13 May 2002 17:52:37 +0530
Date: Mon, 13 May 2002 17:52:37 +0530

Hi ,

1>Can the openRTSP be made to support smil(Synchronized multimedia
Integration Language).

2>Can i use the live media library , modify it and reuse it for my own
embedded application, for this should i take
any permission or is this library  free.

regards,
niranjan


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X-Original-Date: Mon, 13 May 2002 10:35:40 -0700
Date: Mon, 13 May 2002 10:35:40 -0700

At 05:22 AM 5/13/02, you wrote:
>1>Can the openRTSP be made to support smil(Synchronized multimedia
>Integration Language).

I'm not sure.  How does SMIL relate to RTSP?  Are you interested in using 
RTSP's "DESCRIBE" command to retrieve a SMIL description, rather than a SDP 
description?  (Is this possible?)  Or are you interested in using HTTP to 
read a web page that contains SMIL, and then use this SMIL to start/record 
media streams?

Can you give some more details about what you're looking for

>2>Can i use the live media library , modify it and reuse it for my own
>embedded application

Yes, of course.  Several people are currently doing this, and this was the 
main reason why I made this code available. Note, however, that the code is 
licensed under the LGPL, which means that if you make modifications to the 
library source code, and later distribute a product that uses these 
modifications, then you should also make the source code modifications 
available.  (The easiest way to do this is to get your modifications 
included in the library code itself; then you need only publish a link to 
<http://live.sourceforge.net/>)

However, because the license is the LGPL (rather than the GPL), you are 
also free to *link* your own proprietary code (for example, proprietary 
codecs, or a proprietary GUI) against the library code, without having to 
make your proprietary code available.

         Ross.


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From <EMAIL-ADDRESS-HIDDEN> Tue May 14 00:23:45 2002
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X-Original-Date: Tue, 14 May 2002 00:20:48 -0700
Date: Tue, 14 May 2002 00:20:48 -0700

FYI, I have released a new version (2002.5.13) of the library code.  This 
new version has just minor changes:

- The "testMPEGVideoReceiver" test program now outputs to "stdout" (just 
like MP3Receiver), rather than using a built-in HTTP server (which was hard 
to use, and was confusing people anyway).
- For Windows Makefiles, the default Visual Studio tool path is now
	c:\Program Files\DevStudio\Vc
which should match most people's existing configuration.
- In each of the "test*" programs, I explain better (in comments) how to 
(optionally) use unicast instead of multicast.
- I fixed some bugs in the "StreamParser" class that *might* have affected 
MPEG Video streaming (but probably didn't).  Nonetheless, if you're using 
the library for MPEG video streaming, then you should probably upgrade.

	Ross.




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Subject: [Live-devel] MP3 ADUs and SideInfo backpointer
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X-Original-Date: Thu, 16 May 2002 15:16:50 -0400
Date: Thu, 16 May 2002 15:16:50 -0400

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Hi Ross,

I am studying RFC-3119 (A More Loss-Tolerant RTP Payload Format for MP3 =
Audio), and I don't understand the way MP3 ADUs are created from MP3 =
Frames.
In section 3.1, it is said that:

  An "ADU frame" is defined as:

      -  The 4-byte MPEG header
         (the same as the original MP3 frame, except that the first 11
         bits are (optionally) replaced by an "Interleaving Sequence
         Number", as described in section 6 below)
      -  The optional 2-byte CRC field
         (the same as the original MP3 frame)
      -  The "side info" structure
         (the same as the original MP3 frame)
      -  The complete sequence of encoded audio data (and any ancillary
         data) for the ADU (i.e., running from the start of this MP3
         frame's "main_data_begin" back-pointer, up to the start of the
         next MP3 frame's back-pointer)


What I don't understand is for the side-info structure which is the same =
as the original MP3 frame. First of all, which MP3 frame if we consider =
that an ADU frame is build from several parts of different MP3 frames ? =
(3 frames for example) Then, if the freshly built ADU frame is complete, =
the side-info's main_data_begin should be set to 0, right ?

Please help me to understand this, I am stuck here in my work.

Thanks in advance

Nicolas.

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
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<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Hi Ross,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>I am studying RFC-3119 (A More =
Loss-Tolerant RTP=20
Payload Format for MP3 Audio), and I don't understand the way MP3 ADUs =
are=20
created from MP3 Frames.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>In section 3.1, it is said =
that:</FONT></DIV>
<DIV><FONT face=3DArial size=3D2><FONT =
size=3D3></FONT></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><FONT size=3D3>&nbsp; An "ADU frame" is =
defined=20
as:<BR><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; -&nbsp; The 4-byte MPEG=20
header<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; (the same as =
the=20
original MP3 frame, except that the first=20
11<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; bits are =
(optionally)=20
replaced by an "Interleaving=20
Sequence<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Number", as =

described in section 6 below)<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; -&nbsp; =
The=20
optional 2-byte CRC =
field<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
(the same as the original MP3 frame)<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
-&nbsp;=20
The "side info" =
structure<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
(the same as the original MP3 frame)<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
-&nbsp;=20
The complete sequence of encoded audio data (and any=20
ancillary<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; data) for =
the ADU=20
(i.e., running from the start of this=20
MP3<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; frame's=20
"main_data_begin" back-pointer, up to the start of=20
the<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; next MP3 frame's =

back-pointer)<BR></FONT></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><FONT size=3D3><FONT=20
size=3D2></FONT></FONT></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><FONT size=3D3><FONT size=3D2>What I =
don't understand=20
is for the side-info structure which is the same as the original MP3 =
frame.=20
First of all, which MP3 frame if we consider that an ADU frame =
is&nbsp;build=20
from several parts of different MP3 frames ?&nbsp;(3&nbsp;frames for =
example)=20
Then, if the freshly built ADU frame is complete, the side-info's=20
main_data_begin should be set to 0, right ?</FONT></FONT></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><FONT size=3D3><FONT=20
size=3D2></FONT></FONT></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><FONT size=3D3><FONT size=3D2>Please =
help me to=20
understand this, I am stuck here in my work.</FONT></FONT></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><FONT size=3D3><FONT=20
size=3D2></FONT></FONT></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><FONT size=3D3><FONT size=3D2>Thanks in =

advance</FONT></FONT></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><FONT size=3D3><FONT=20
size=3D2></FONT></FONT></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><FONT size=3D3><FONT=20
size=3D2>Nicolas.</FONT></DIV></FONT></FONT></BODY></HTML>

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From <EMAIL-ADDRESS-HIDDEN> Thu May 16 13:06:42 2002
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Subject: Re: [Live-devel] MP3 ADUs and SideInfo backpointer
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X-Original-Date: Thu, 16 May 2002 12:51:31 -0700
Date: Thu, 16 May 2002 12:51:31 -0700

At 12:16 PM 5/16/02, Nicolas Drapeau wrote:
>What I don't understand is for the side-info structure which is the same 
>as the original MP3 frame. First of all, which MP3 frame if we consider 
>that an ADU frame is build from several parts of different MP3 frames ? (3 
>frames for example)

When constructing an ADU, you should think of the sequence of bytes
         <4-byte header><optional CRC><side-info structure>
as being an indivisible structure.  Let's call this S.  So, a sequence of 
input MP3 frames has the following form:
         [S1,D1] [S2,D2] [S3,D3] [S4,D4]
where Di is the encoded audio data that appears in the remaining of the ith 
MP3 frame (i.e., following the 4-byte header, optional CRC, and side-info 
structure).

Suppose, for example, that each frame is from a 128 kbps MPEG-1 44.1 kHz 
stereo encoding, with a padding bit - so, is 418 bytes in size.  Then, each 
Si will be 36 bytes (=4+32) in size, and each Di will be 382 bytes 
(=418-36) in size.  Suppose also, for example, that the "main_data_begin" 
(aka back-pointer) field in S3 has value 400, and that the 
"main_data_begin" field in S4 has value 100.

Then, the 3rd ADU will consist of
         - S3, followed by
         - the last 18 bytes (=400-382) from D2, followed by
         - all of D3 (=382 bytes), followed by
         - the first 282 bytes (=382-100) from D4
I.e., the 3rd ADU will consist of S3 (36 bytes), plus 682 bytes of encoded 
audio data.

>Then, if the freshly built ADU frame is complete, the side-info's 
>main_data_begin should be set to 0, right ?

No!  You must leave the side-info structure alone.  IN particular, a 
receiver will need to know what the "main_data_begin" values were, so that 
it can construct a sequence of valid MP3 frames from the input ADU 
frames.  (Note that a receiver cannot play ADUs without first converting 
them back to MP3 frames.)

I hope this helps explain things...

         Ross.


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From: {David Pontes} <EMAIL-ADDRESS-HIDDEN>
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Subject: [Live-devel] RTSP Implementation
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X-Original-Date: Tue, 21 May 2002 12:18:04 +0100
Date: Tue, 21 May 2002 12:18:04 +0100

	Hello
	I'm considering adapting the Live.Com RTSP server to enable the following
features:

	- Run stand-alone (separate from the media streamer).
	- Serve both audio and video.
	- Accept multiple clients in both unicast and multicast.
	- Issue the RTSP REDIRECT message to instruct the client to connect to a
different stream.

	The packets shall not be processed by the application. In the case of
unicast, the application will simply direct the packets to the client by
changing their destination address. This is achieved with a script which
configures iptables on a linux machine.

	Questions:
	- Is the Live.com server the best starting point? Has this been made
already, either in whole or in part? If so, is that work available as open
source?
	- Do you see any major flaws to this approach, namely going against RFC
2326, or neglecting some security aspect?
	- Does the following list contain all the necessary files?

	RTSPServer.cpp/hh
		GroupsockHelper.cpp/hh
			UsageEnvironment.cpp/hh
		NetAddress.cpp/hh
			Hashtable.cpp/hh
			Boolean.hh
		RTPSink.cpp/hh
			MediaSink.cpp/hh
			Groupsock.cpp/hh
				NetInterface.cpp/hh
					TunnelEncaps.hh
					IOHandler.cpp/hh
				GroupEId.cpp/hh


Thanks,
David Pontes


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X-Original-Date: Tue, 21 May 2002 19:41:44 -0700
Date: Tue, 21 May 2002 19:41:44 -0700

At 04:18 AM 5/21/02, David Pontes wrote:
>         Hello
>         I'm considering adapting the Live.Com RTSP server to enable the 
> following
>features:
>
>         - Run stand-alone (separate from the media streamer).

Can you explain what you mean by this?  To be able to "DESCRIBE" a media 
session (and later, "SETUP" and "PLAY" it), a RTSP server needs to 
communicate with the media streamer.  So, if you were to separate the RTSP 
server from the media streamer, you'd still have to have some 
communications protocol for talking between them.

At present, the "RTSPServer" class takes a "RTPSink" object as 
parameter.  (In the next revision, to appear soon, it'll be able to take 
multiple of these objects - e.g., audio + video.)  It needs this in order 
to be able to form a SDP description that describes the session.

>         - Serve both audio and video.

As I noted above, the next version of the code - to appear soon, will be 
able to support both audio and video.  So, if you plan on working with this 
code, you might want to hold off on this for a few days.

>         - Accept multiple clients in both unicast and multicast.

Multicast works now.  It is easy, because it doesn't require setting up 
separate streams for each client.

Supporting unicast would be quite a bit more work.

>         The packets shall not be processed by the application. In the case of
>unicast, the application will simply direct the packets to the client by
>changing their destination address.  This is achieved with a script which
>configures iptables on a linux machine.

I'm not sure I follow this.  For multicast sessions, the streamer can send 
directly to the destination multicast address.  For unicast sessions, you 
will, in general, need to send out separate (duplicate) streams 
anyway.  Are you somehow proposing to have this packet duplication 
performed inside the Linux kernel somehow, instead of in the streamer?

>         Questions:
>         - Is the Live.com server the best starting point?

Maybe not, because the RTSPServer implementation is currently quite limited 
as to what part of the RTSP spec it implements.  You may be better 
off  using Apple's 'Darwin Streaming Server' as a starting point.

         Ross.


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X-Original-Date: Wed, 22 May 2002 08:11:07 +0000
Date: Wed, 22 May 2002 08:11:07 +0000

Hi there,

I am back to trying to get an MPEG transport stream off of a hard disk and 
sent using RTP.  I was wondering if anyone else has done this before I write 
an MPEGTransportStreamFramer.cpp to cope with the format.  If anyone else 
has done this I would really appreciate some help if not I will hopefully 
start it soon and will feed the results back.

Cheers
Dan


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X-Original-Date: Wed, 22 May 2002 01:42:42 -0700
Date: Wed, 22 May 2002 01:42:42 -0700

At 01:11 AM 5/22/02, Daniel Laird wrote:
>I am back to trying to get an MPEG transport stream off of a hard disk and 
>sent using RTP.  I was wondering if anyone else has done this before I 
>write an MPEGTransportStreamFramer.cpp to cope with the format.  If anyone 
>else has done this I would really appreciate some help

Dan,

You're in luck.  Because the RTP payload format for MPEG transport streams 
is so simple (see RFC 2250, section 2), the existing "liveMedia" library 
classes "ByteStreamFileSource" and "SimpleRTPSink" can be used to stream this.

You can create your media source as follows:
         ByteStreamFileSource::createNew(*env, fileName, 188)
Note that this will cause data to be read in 188-byte chunks: the size of a 
MTS 'packet'.  Thus, you won't need a separate 'Framer' object.

You can create your media sink as follows:
         SimpleRTPSink::createNew(*env, &rtpGroupsock, 33, 90000, "video")
Note that "33" is the static RTP payload format number defined for MPEG-2 
Transport Streams over RTP.   (See 
<http://www.ietf.org/internet-drafts/draft-ietf-avt-profile-new-12.txt>, 
section 8.)

I hope this helps.

         Ross.


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X-Original-Date: Wed, 22 May 2002 01:50:30 -0700
Date: Wed, 22 May 2002 01:50:30 -0700

A follow-up comment: In the call to SimpleRTPSink::createNew(), you 
probably want to use the string "application" rather than "video" (as I 
said in my earlier message).  (This string would be used by a colocated 
RTSPServer, if any, to construct a SDP description for the session.)

Also, you can program a receiver for this stream using a "SimpleRTPSource" 
- e.g.,
SimpleRTPSource::createNew(*env, &rtpGroupsock, 33, 90000, "application/mp2t")

	Ross.


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X-Original-Date: Wed, 22 May 2002 21:51:32 +0530
Date: Wed, 22 May 2002 21:51:32 +0530

Hi ,

I wanted to know the following things,

1>I want to directly stream MPEG Video data from the net on to a HDTV(a
special DTV- digital TV) which has a (MPEG)Decoder(basically a hardware
decoder) , now my need is that i want to use openRTSP : get the stream from
the net and instead of storing the stream onto a file i want to send the
streams on to this decoder , can i do this using openRTSP without any
changes.

2> OR Do i need to add to a buffering mechanism to openRTSP to  acheive this
.

Please guide me ,

thanks in advance,

Niranjan



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X-Original-Date: Wed, 22 May 2002 11:53:29 -0700
Date: Wed, 22 May 2002 11:53:29 -0700

At 09:21 AM 5/22/02, you wrote:
>1>I want to directly stream MPEG Video data from the net on to a HDTV(a
>special DTV- digital TV) which has a (MPEG)Decoder(basically a hardware
>decoder) , now my need is that i want to use openRTSP : get the stream from
>the net and instead of storing the stream onto a file i want to send the
>streams on to this decoder , can i do this using openRTSP without any
>changes.

Niranjan,

Because you're asking about "openRTSP", I presume that your MPEG Video data 
(on the net) is served by a RTSP server.

What you can do is:
1/ Run "openRTSP" with the "-r" option, to start playing the video stream 
(from the RTSP server), but not actually receive it.  (See 
<http://www.live.com/openRTSP/#no-receive>.)  Then also:
2/ Run a separate program that receives the MPEG data from the net, 
decapsulates the RTP headers, and outputs the resulting MPEG data to your 
TV's MPEG decoder.

If your input network data is a pair of MPEG Elementary Streams (audio and 
video) using RTP, then you should be able to receive this using slightly 
modified versions of "testMP3Receiver" and "testMPEGVideoReceiver".  (Note 
that "testMP3Receiver" should work OK even on layer I or layer II audio.)

If your input network data is MPEG Transport or Program Stream data (also 
using RTP), then you should be able to receive it using the 
"SimpleRTPSource" class.  (Note the message that I sent yesterday about this.)

I hope this helps.  If you have any questions, just let me know.

         Ross.


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X-Original-Date: Wed, 22 May 2002 12:03:41 -0700
Date: Wed, 22 May 2002 12:03:41 -0700

Niranjan,

One more thing: If your input data consists just of a *single* video RTP 
stream (i.e., not separate video + audio RTP streams), then you can also 
run "openRTSP" with the "-v" (play video only).  This will cause the video 
data to be output to 'stdout', so you can just pipe it to your MPEG 
decoder.  If you do this, you won't need to run a separate receiver program 
at all.

	Ross.


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X-Original-Date: Thu, 23 May 2002 12:30:50 +0530
Date: Thu, 23 May 2002 12:30:50 +0530

Hi ,

thanks Ross for the valuable reply :)

>What you can do is:
>1/ Run "openRTSP" with the "-r" option, to start playing the video stream
>(from the RTSP server), but not actually receive it.  (See
><http://www.live.com/openRTSP/#no-receive>.)  Then also:

I have understood this , just let me breif up : the main purpose of openRTSP
here would be to firstly obtain the SDP from the server and start playing
the Stream but not receive the stream data and then go into infinite
blockmyself ()function.

>2/ Run a separate program that receives the MPEG data from the net,
>decapsulates the RTP headers, and outputs the resulting MPEG data to your
>TV's MPEG decoder

But this part seems a bit confusing to me .By a separate program do u mean
that i have to write a separate receiver code for receiving the stream and
directing it to the MPEG decoder.
1>Just decapsulating the RTP headers and sending the packet(one by one) to
the MPEG decoder , is that all sufficent (ie appart from RTCP stuff) , but
who will take care of the packet reordering , packetBuffering etc should
this program do this or the decoder will take care of that.
2>For implementing this program could i make use of the live library , if
yes please direct me with what classes i need to use from the library or has
anyone already implemented such a thing or do u have a sample test program
for this functionality let me know.

Please guide me on this

Niranjan



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Subject: [Live-devel] 1 ADU frame fragmented in more than 2 MP3 frames ?
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X-Original-Date: Thu, 23 May 2002 11:45:51 -0400
Date: Thu, 23 May 2002 11:45:51 -0400

This is a multi-part message in MIME format.

------=_NextPart_000_0097_01C2024F.63456780
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	charset="iso-8859-1"
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Hi Ross,

I would like to know if an ADU can be fragmented in more than 2 MP3 =
frames.

If this can happen then I have another problem to submit to you, else =
forget the following.

Lets take for example an ADU frame fragmented over 3 MP3 frames.
When I will try to build this ADU I have to use the same side-info =
structure as the original MP3 frame says RFC 3119.
But which one ?

Lets consider these 3 MP3 frames :
+-------+-------+---------------------------------------------+
|  H1   |  SI1  |             D1                     |xxxxxxxx|
+-------+-------+---------------------------------------------+
                                                     |<- 20 ->|

+-------+-------+--------------------------------------------+
|  H2   |  SI2  |xxxxxxxxxxxx D2 xxxxxxxxxxxxxxxxxxxxxxxxxxxx|
+-------+-------+--------------------------------------------+


+-------+-------+--------------------------------------------+
|  H3   |  SI3  |xxxxxxxx|    D3                             |
+-------+-------+--------------------------------------------+
                |<- 30 ->|

The ADU data will consist of : 20 last bytes of D1 + D2 + 30 first bytes =
of D3

So the backpointers for each side-info will be :
BP2 for SI2 : 20
BP3 for SI3 : 20+length_of_D2

To build the ADU now, which SideInfo should I use, SI2 or SI3 ? (same =
question of course for the MPEG header and the optional CRC)

Thanks in advance=20

Nicolas

------=_NextPart_000_0097_01C2024F.63456780
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	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2715.400" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3D"Courier New" size=3D2>Hi Ross,</FONT></DIV>
<DIV><FONT face=3D"Courier New" size=3D2></FONT>&nbsp;</DIV>
<DIV>
<DIV><FONT face=3D"Courier New" size=3D2>I would like to know if an ADU =
can be=20
fragmented in more than 2 MP3 frames.</FONT></DIV></DIV>
<DIV><FONT face=3D"Courier New" size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3D"Courier New" size=3D2>If this can happen then&nbsp;I =
have another=20
problem to submit to you, else&nbsp;forget the following.</FONT></DIV>
<DIV><FONT face=3D"Courier New" size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3D"Courier New" size=3D2>Lets take for example&nbsp;an =
ADU=20
frame&nbsp;fragmented&nbsp;over 3 MP3 frames.</FONT><FONT=20
face=3D"Courier New"></FONT></DIV>
<DIV><FONT face=3D"Courier New" size=3D2>When I will try to build this =
ADU I have to=20
use the same side-info structure as the original MP3 frame says RFC=20
3119.</FONT></DIV>
<DIV><FONT face=3D"Courier New" size=3D2>But which one ?</FONT></DIV>
<DIV><FONT face=3D"Courier New" size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3D"Courier New" size=3D2>Lets consider these 3 MP3 =
frames=20
:</FONT></DIV>
<DIV><FONT face=3D"Courier New"=20
size=3D2>+-------+-------+---------------------------------------------+<=
/FONT></DIV>
<DIV><FONT face=3D"Courier New" size=3D2>|&nbsp; H1&nbsp;&nbsp; |&nbsp; =
SI1&nbsp;=20
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
=20
D1&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;|xxxxxxxx|</FONT><=
/DIV>
<DIV><FONT face=3D"Courier New"=20
size=3D2>+-------+-------+---------------------------------------------+<=
/FONT></DIV>
<DIV><FONT face=3D"Courier New"=20
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;|&lt;-=20
20 -&gt;|</FONT></DIV>
<DIV><FONT face=3D"Courier New" size=3D2></FONT>&nbsp;</DIV>
<DIV>
<DIV><FONT face=3D"Courier New"=20
size=3D2>+-------+-------+--------------------------------------------+</=
FONT></DIV>
<DIV><FONT face=3D"Courier New" size=3D2>|&nbsp; H2&nbsp;&nbsp; |&nbsp; =
SI2&nbsp;=20
|xxxxxxxxxxxx D2 xxxxxxxxxxxxxxxxxxxxxxxxxxxx|</FONT></DIV>
<DIV><FONT face=3D"Courier New"=20
size=3D2>+-------+-------+--------------------------------------------+</=
FONT>
<DIV><FONT face=3DArial></FONT>&nbsp;</DIV>
<DIV><FONT face=3D"Courier New" size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3D"Courier New"=20
size=3D2>+-------+-------+--------------------------------------------+</=
FONT></DIV>
<DIV><FONT face=3D"Courier New" size=3D2>|&nbsp; H3&nbsp;&nbsp; |&nbsp; =
SI3&nbsp;=20
|xxxxxxxx|&nbsp;&nbsp;&nbsp;=20
D3&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;=20
|</FONT></DIV>
<DIV><FONT face=3D"Courier New"=20
size=3D2>+-------+-------+--------------------------------------------+</=
FONT></DIV>
<DIV><FONT face=3D"Courier New"=20
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;|&lt;-&nbsp;30=20
-&gt;|</FONT></DIV>
<DIV><FONT face=3D"Courier New" size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3D"Courier New" size=3D2>The ADU data will consist of =
:&nbsp;20 last=20
bytes&nbsp;of D1 + D2 +&nbsp;30 first bytes&nbsp;of D3</FONT></DIV>
<DIV><FONT face=3D"Courier New" size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3D"Courier New" size=3D2>So the backpointers for each =
side-info will=20
be :</FONT></DIV>
<DIV><FONT face=3D"Courier New" size=3D2>BP2 for SI2 : 20</FONT></DIV>
<DIV><FONT face=3D"Courier New" size=3D2>BP3 for SI3 : =
20+length_of_D2</FONT></DIV>
<DIV><FONT face=3D"Courier New" size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3D"Courier New" size=3D2>To build the ADU now, which =
SideInfo should=20
I use, SI2 or SI3 ? (same question of course for the MPEG header and the =

optional CRC)</FONT></DIV>
<DIV><FONT face=3D"Courier New" size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3D"Courier New" size=3D2>Thanks in advance </FONT></DIV>
<DIV><FONT face=3D"Courier New" size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3D"Courier New"=20
size=3D2>Nicolas</FONT></DIV></DIV></DIV></BODY></HTML>

------=_NextPart_000_0097_01C2024F.63456780--


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>So the backpointers for each side-info will be :
>BP2 for SI2 : 20

No.  The backpointer for SI2 should be something >= 20, because the encoded 
audio data (if any) for ADU 2 will have to come from somewhere in D1 
*before* the last 20 bytes (because those last 20 bytes are for ADU 3).

>BP3 for SI3 : 20+length_of_D2

That's correct.

>To build the ADU now, which SideInfo should I use, SI2 or SI3 ? (same 
>question of course for the MPEG header and the optional CRC)

To build the ADU frame for MP3 frame 1, use SI1.  To build the ADU frame 
for MP3 frame 2, use SI2.  To build the ADU frame for MP3 frame 3, use SI3, 
etc.  There is a one-to-one correspondence between MP3 frames and ADU 
frames.  I.e., each MP3 frame has a corresponding ADU frame, with the same 
header, side info and optional CRC.  It's only the encoded audio data that 
can come from different MP3 frames.

         Ross.


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X-Original-Date: Thu, 23 May 2002 10:35:53 -0700
Date: Thu, 23 May 2002 10:35:53 -0700

At 12:00 AM 5/23/02, you wrote:
> >What you can do is:
> >1/ Run "openRTSP" with the "-r" option, to start playing the video stream
> >(from the RTSP server), but not actually receive it.  (See
> ><http://www.live.com/openRTSP/#no-receive>.)  Then also:
>
>I have understood this , just let me breif up : the main purpose of openRTSP
>here would be to firstly obtain the SDP from the server and start playing
>the Stream but not receive the stream data

That's right.

>  and then go into infinite blockmyself ()function.

Or, it could be told to "TEARDOWN" the stream after a period of time 
(either specified in the SDP description, or by the user).

> >2/ Run a separate program that receives the MPEG data from the net,
> >decapsulates the RTP headers, and outputs the resulting MPEG data to your
> >TV's MPEG decoder
>
>But this part seems a bit confusing to me .By a separate program do u mean
>that i have to write a separate receiver code for receiving the stream and
>directing it to the MPEG decoder.

Not necessarily.  As I noted in my second message yesterday, an alternative 
would be to use openRTSP's "-v" option.  This will cause openRTSP to 
receive the video stream (only), decapsulate the RTP, and output the 
resulting video to 'stdout'.  So, you can redirect (or pipe) this to your 
MPEG decoder.

To help you figure out what best to do, I need to know more about your 
input data.  So, could you please run "openRTSP" - without any options - on 
your "rtsp://" URL, and let me know what SDP description you get.

         Ross.


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X-Original-Date: Thu, 23 May 2002 15:08:00 -0400
Date: Thu, 23 May 2002 15:08:00 -0400

This is a multi-part message in MIME format.

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	charset="iso-8859-1"
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>>So the backpointers for each side-info will be :
>>BP2 for SI2 : 20

>No.  The backpointer for SI2 should be something >=3D 20, because the =
encoded=20
>audio data (if any) for ADU 2 will have to come from somewhere in D1=20
>*before* the last 20 bytes (because those last 20 bytes are for ADU 3).


Why those last 20 bytes from D1 are for ADU 3 ? I mean how can I =
determine that. I think I missed a lot.



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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2715.400" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>&gt;&gt;So the backpointers for each side-info =
will be=20
:<BR>&gt;&gt;BP2 for SI2 : 20<BR><BR>&gt;No.&nbsp; The backpointer for =
SI2=20
should be something &gt;=3D 20, because the encoded <BR>&gt;audio data =
(if any)=20
for ADU 2 will have to come from somewhere in D1 <BR>&gt;*before* the =
last 20=20
bytes (because those last 20 bytes are for ADU 3).<BR>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Why those last 20 bytes from D1 are for =
ADU 3 ? I=20
mean how can I determine that. I think I missed a lot.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV>&nbsp;</DIV></BODY></HTML>

------=_NextPart_000_000B_01C2026B.A0BAFB40--


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X-Original-Date: Thu, 23 May 2002 12:58:57 -0700
Date: Thu, 23 May 2002 12:58:57 -0700

At 12:08 PM 5/23/02, you wrote:
> >>So the backpointers for each side-info will be :
> >>BP2 for SI2 : 20
>
> >No.  The backpointer for SI2 should be something >= 20, because the encoded
> >audio data (if any) for ADU 2 will have to come from somewhere in D1
> >*before* the last 20 bytes (because those last 20 bytes are for ADU 3).
>
>Why those last 20 bytes from D1 are for ADU 3 ?

Because, in the example you constructed, the backpointer ("main_data_begin" 
field) SI3 is 20+length_of_D2.  So ADU 3 must have that much audio data, 
which means that - in this example - the first 20 bytes will come from the 
end of frame 1.

         Ross.


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X-Original-Date: Fri, 24 May 2002 08:53:20 -0700 (PDT)
Date: Fri, 24 May 2002 08:53:20 -0700 (PDT)

hi Ross,
 I was trying to understand scheduling delay for
tasks.
i printed out the timeToDelay.seconds() in the
block
Myself() function and saw that initially it was
1 sec and then changed to 4 sec and started decreasing
by 1.

can u tell me what is the relevance of 4 sec and in
which part of the code are u setting it?

Your help would be greatly appriciated.

thank you
surya

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From: {David Pontes} <EMAIL-ADDRESS-HIDDEN>
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Subject: RE: [Live-devel] RTSP Implementation
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X-Original-Date: Fri, 24 May 2002 17:18:11 +0100
Date: Fri, 24 May 2002 17:18:11 +0100

> -----Original Message-----
>
> At 04:18 AM 5/21/02, David Pontes wrote:
> >         Hello
> >         I'm considering adapting the Live.Com RTSP server to enable the
> > following
> >features:
> >
> >         - Run stand-alone (separate from the media streamer).
>
> Can you explain what you mean by this?  To be able to "DESCRIBE" a media
> session (and later, "SETUP" and "PLAY" it), a RTSP server needs to
> communicate with the media streamer.  So, if you were to separate
> the RTSP
> server from the media streamer, you'd still have to have some
> communications protocol for talking between them.
>

	I'm using the same technique as DSS's reflector mode, where the server has
a local copy of the sessions's SDP file and uses it as a base to reply to
the client. Again, I assume this is all it takes, but would be thankful for
some confirmation or a hint on where can this go wrong.

> At present, the "RTSPServer" class takes a "RTPSink" object as
> parameter.  (In the next revision, to appear soon, it'll be able to take
> multiple of these objects - e.g., audio + video.)  It needs this in order
> to be able to form a SDP description that describes the session.
>

	In this form, the server would not follow this procedure. Adding sound
support would simply consist on adding control information regarding the
sound media stream on the RTSP/SDP DESCRIBE response. The client will then
SETUP both video and audio (this is what happens with DSS, AFAIK).

> >         - Accept multiple clients in both unicast and multicast.
>
> Multicast works now.  It is easy, because it doesn't require setting up
> separate streams for each client.
>
> Supporting unicast would be quite a bit more work.
>

	In this approach, in unicast the server would be able to listen to the
multicast stream (thus be a recipient of the stream) and resend the packets
back to the clients in unicast (acting as a reflector as needed). The
RTSP/SDP DESCRIBE response would change accordingly.

> >         The packets shall not be processed by the application.
> In the case of
> >unicast, the application will simply direct the packets to the client by
> >changing their destination address.  This is achieved with a script which
> >configures iptables on a linux machine.
>
> I'm not sure I follow this.  For multicast sessions, the streamer
> can send
> directly to the destination multicast address.  For unicast sessions, you
> will, in general, need to send out separate (duplicate) streams
> anyway.  Are you somehow proposing to have this packet duplication
> performed inside the Linux kernel somehow, instead of in the streamer?
>

	Sorry, my bad. I thought it would be easy to duplicate the packets in the
kernel. This is not the case. You're right - the server application must be
responsible for that.

> >         Questions:
> >         - Is the Live.com server the best starting point?
>
> Maybe not, because the RTSPServer implementation is currently
> quite limited
> as to what part of the RTSP spec it implements.  You may be better
> off  using Apple's 'Darwin Streaming Server' as a starting point.
>

	DSS has almost all I need to act as a reflector module, but I still cannot
use it when serving clients in multicast (it automatically reflects the
stream in unicast). I tried altering this and found it unfeasible. Plus, I
need to add the REDIRECT command (upon application criteria), but I guess
that would be a bit simpler and am considering it.


Thanks,
David Pontes


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X-Original-Date: Fri, 24 May 2002 09:16:31 -0700
Date: Fri, 24 May 2002 09:16:31 -0700

"DelayQueue" is used to implement 
"BasicTaskScheduler::scheduleDelayedTask()" (and 
"BasicTaskScheduler::unscheduleDelayedTask()"), so note the places in the 
code where "scheduleDelayedTask()" is called.

	Ross.


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From <EMAIL-ADDRESS-HIDDEN> Mon May 27 21:57:38 2002
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X-Original-Date: Mon, 27 May 2002 21:51:28 -0700
Date: Mon, 27 May 2002 21:51:28 -0700

I have installed a new version (2002.05.27) of the "LIVE.COM Streaming 
Media" libraries.  This new version has some (minor) changes to the API, 
which will probably require minor changes to programs that use it.

1/ "BasicTaskScheduler" and "BasicUsageEnvironment" objects must now be 
heap allocated, using "createNew()" static member functions.

2/ The "TaskProc" typedef (in "UsageEnvironment/UsageEnvironment.hh") has 
been renamed to "TaskFunc".

Changes 1/ and 2/ were done to overcome minor problems that Marty Schoch 
had in porting the code to work with Objective C on MacOS X.

3/ The API for creating "RTSPServer"s has changed.  You now create a 
"RTSPServer" using a new "ServerMediaSession" object, which describes the 
media session(s) that the RTSP server will be serving.

4/ The change in 3/ makes it possible to have a "RTSPServer" for an 
audio+video session.  I have updated the "testMPEGAudioVideoStreamer" test 
program to include an optional RTSP server - just like "testMP3Streamer" 
and "testMPEGVideoStreamer".

Note that, as before, "RTSPServer" currently works only for multicast 
sessions - not unicast.

	Ross.


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From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
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X-Original-Date: Mon, 27 May 2002 21:58:13 -0700
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At 09:18 AM 5/24/02, David Pontes wrote:
> > At present, the "RTSPServer" class takes a "RTPSink" object as
> > parameter.  (In the next revision, to appear soon, it'll be able to take
> > multiple of these objects - e.g., audio + video.)  It needs this in order
> > to be able to form a SDP description that describes the session.
> >
>
>         In this form, the server would not follow this procedure. Adding 
> sound
>support would simply consist on adding control information regarding the
>sound media stream on the RTSP/SDP DESCRIBE response. The client will then
>SETUP both video and audio (this is what happens with DSS, AFAIK).

David,

Note that this is now possible with the new version of the library.  You 
can now create a "ServerMediaSession" object to be used by the RTSP server, 
and describe each subsession (audio, video) by calling 
"addSubsessionByComponents()" for each.  See 
"liveMedia/ServerMediaSession.hh" for more information.

You then create the RTSP server using this "ServerMediaSession" object as 
parameter.

This will let you create a RTSP server that does not actually receive the 
media data that its serving.

         Ross.



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From: <EMAIL-ADDRESS-HIDDEN> (S. Kannan)
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Subject: [Live-devel] Playing avi buffer.
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This is a multi-part message in MIME format.

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Hello All:

I have a different RTSP server (written by southern california =
university). From my RTSP client, I'm able to receive the buffer of the =
movie (.MOV and .AVI), I have no idea how to play that buffer.  The =
linux version of this application (RTSP Client) uses libaviplay.so =
library for playing. Does anyone know how to play this avi (or .mov) =
buffer or any third pary library or source code.  Please let me know, =
I'm really struggling.....

I know how to play an AVI file using Microsoft Multimedia APIs, but not =
AVI Stream buffer.

Regards
Senthu

------=_NextPart_000_0068_01C2094C.35B2F400
Content-Type: text/html;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2600.0" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Hello All:</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>I have a different RTSP server (written =
by southern=20
california university). From my RTSP client, I'm able to receive the =
buffer of=20
the movie (.MOV and .AVI), I have no idea how to play that buffer.&nbsp; =
The=20
linux version of this application (RTSP Client)&nbsp;uses=20
<STRONG>libaviplay.so</STRONG> library for playing. Does anyone know how =
to play=20
this avi (or .mov)&nbsp;buffer or any third pary library or source =
code.&nbsp;=20
Please let me know, I'm really struggling.....</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>I know how to play an AVI file using =
Microsoft=20
Multimedia APIs, but not AVI Stream buffer.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Regards</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Senthu</FONT></DIV></BODY></HTML>

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Hi Ross,

Thank you very much for your help! I still have several questions.

1. In the "testMPEGVideoStreamer" and "testMPEGVideoReceiver",
"blockMyself()" gets called without a parameter but in the class
BasicTaskScheduler, "blockMyself(char* watchVariable)" does have a
parameter "watchvariable". What's the difference?

2. After the "getNextFrame()" call, suppose a frame is avaliable
immediately, the "afterReading()" will be called immediately. The last
line in "afterReading()" is this:
        dp->flags = ~0;
I feel this is the the same "dp->flags" area that is used in this call

rtpSource->envir().taskScheduler().blockMyself((char*)&dp->flags);
am I right?

When the "blockMyself()" is called, in the infinite while loop, the
"watchVariable" is checked first in the statement
    if (watchVariable != NULL && *watchVariable != 0) break;
If the above assumption is correct (a frame is avaliable immediately
when "getNextFrame()" is called), this "if" statement will be true and
the while loop (the loop to call select()) is break. And thus the
"SingleStep()" will not be called. Am I right? If "SingleStep()" is not
called, what about a packet arrives at this time??

I added some buffer tracking code in the "MultiFramedRTPSource.cpp"
file. I found sometimes when a RTP packet is received out of order, the
"fReorderingBuffer" will store it and afterwards packets in the buffer
"fReorderingBuffer" untill the packet with correct seq# is received or a
threshold time is passed. At this time, the "fReorderingBuffer" may
store some numbers of packets. When the threshold passed,
"getNextFrame()" call will always get a packet from the "rtpSource"
immediately. That means, the "afterReading()" will be called immediately
either. When the "blockMyself()" is called afterwards, the "if"
statement is always true and thus the "SingleStep()" call is always
skipped. If this is the case, how can we deal with packets arrived at
this period of time if we dont call the "select()"??

The "SingleStep()" will only be called when all packets in the buffer
"fReorderingBuffer" are consumed. At this time, "afterReading()" can not
be called immediately, and thus the "if" statement in the
"blockMyself()" is false and "SingleStep()" will be called. After the
"select()" reading a packet from the socket, a 'read handler' (is
MultiFramedRTPSource::networkReadHandler(), right?) will process the
packet and at the end, will call "doGetNextFrame1()", the same function
call as from the "getNextFrame()". Is my thought right?

My question is still the same, if "SingleStep()" is skipped, how can we
deal with packet arrived at this time??


Thanks and Best Regards
Michael



Ross Finlayson wrote:

> >I'm PhD student in the School of Computer Scince and Engineering,
> >University of New South Wales, Australia. I'm currently working on a
> >multimedia project that is using live's streaming library. I pached
> >live.com's library to mplayer and now mplayer can play an sdp file.
>
> G'day Michael,
>
> To get a good feel as to how the "LIVE.COM Streaming Media" library
> works,
> you may find it more instructive to look at the test programs that
> come
> with the library - in particular, "testMPEGAudioVideoStreamer" and
> "testMPEGVideoReceiver".  These have less MPlayer-specific 'baggage'
> associated with them, and so should be easier to understand.
>
> >rtpSource->getNextFrame(dp->buffer, bufferSize, afterReading, ds,
> >onSourceClosure, ds);
> >rtpSource->envir().taskScheduler().blockMyself((char*)&dp->flags);
> >
> >To my feeling, the first line is trying to get an RTP frame from the
> >fReorderingBuffer, which is a variable defined in the class
> >MultiFramedRTPSource. If there is no packet in the
> >buffer(fReorderingBuffer), the second line
> >will call the blockMyself() function defined in the
> BasicTaskScheduler
> >class.
>
> No, not quite. The call to "getNextFrame()" will arrange for the next
> frame
> to be read from "rtpSource".  When this frame is available,
> "afterReading()" will be called.  (If the frame is not immediately
> available, then a 'read handler' will be registered, so that
> "afterReading()" will be called when the frame becomes available.)
>
> Then, *regardless of whether or not a frame was immediately
> delivered*,
> "blockMyself()" gets called.  "blockMyself()" works by sitting in a
> 'select()' loop, handling the arrival of network packets (or other
> events).  When a network packet arrives, a read handler (if
> registered)
> will be called to handle it.
>
> >During the interval of two frame times (decoding and
> >rendering frame, sleeping untill next frame playing time), I feel the
>
> >process doesn't read socket paralell for other RTP packet may
> >already arrive in the lower level buffer (UDP/IP).
>
> I'm not sure what you mean by this.  Whenever a packet arrives, the
> 'select()' loop in "blockMyself()" will notice this, and call the
> appropriate read handler to deliver the data.  The receiving code will
>
> always be 'busy' - it will never be blocked doing nothing, as long as
> a new
> incoming packet is available.
>
> Once again, I think you may find the "testMPEGVideoReceiver" code to
> be a
> bit more instructive here.
>
> >That means, two parallel process
>
> No!  The library code is delibererately designed to be
> single-threaded, and
> will *not* work with multiple threads.  As I noted above, the code
> should
> not be blocking unnecesarily, so multiple threads (in a uniprocessor
> system, at least) would not buy you anything anyway.
>
> >I also found that for every mpeg frame(I, P, B), the RTP stream
> server
> >slices it into 16 pieces and creates the RTP packets, no matter how
> larger
> >or small the mpeg frame is. Is that true?
>
> This must be a feature of your RTP streaming server.  Which RTP
> streaming
> server implementation are you using?  The "LIVE.COM Streaming Media"
> library code (e.g., when used in "testMPEGVideoStreamer") doesn't do
> this.
>
> (In the future, could you please use the developers' mailing list
> <live-devel(at)lists.sourceforge.net> for questions like this.  See
> <http://lists.sourceforge.net/lists/listinfo/live-devel> for
> information on
> how to subscribe.  For one-on-one email, I generally give priority to
> paying consulting clients.)
>
>          Ross Finlayson
>          LIVE.COM
>          <http://www.live.com/>



--------------79D8FBBA37BE340ECD460B14
Content-Type: text/html; charset=us-ascii
Content-Transfer-Encoding: 7bit

<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
Hi Ross,
<p>Thank you very much for your help! I still have several questions.
<p>1. In the "testMPEGVideoStreamer" and "testMPEGVideoReceiver", "blockMyself()"
gets called without a parameter but in the class BasicTaskScheduler, "blockMyself(char*
watchVariable)" does have a parameter "watchvariable". What's the difference?
<p>2. After the "getNextFrame()" call, suppose a frame is avaliable immediately,
the "afterReading()" will be called immediately. The last line in "afterReading()"
is this:
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <font face="Courier New,Courier"><font size=-1>dp->flags
= ~0;</font></font>
<br>I feel this is the the same "dp->flags" area that is used in this call
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <font face="Courier New,Courier"><font size=-1>rtpSource->envir().taskScheduler().blockMyself((char*)&amp;dp->flags);</font></font>
<br>am I right?
<p>When the "blockMyself()" is called, in the infinite while loop, the
"watchVariable" is checked first in the statement
<br><font face="Courier New,Courier"><font size=-1>&nbsp;&nbsp;&nbsp; if
(watchVariable != NULL &amp;&amp; *watchVariable != 0) break;</font></font>
<br>If the above assumption is correct (a frame is avaliable immediately
when "getNextFrame()" is called), this "if" statement will be true and
the while loop (the loop to call select()) is break. And thus the "SingleStep()"
will not be called. Am I right? If "SingleStep()" is not called, what about
a packet arrives at this time??
<p>I added some buffer tracking code in the "MultiFramedRTPSource.cpp"
file. I found sometimes when a RTP packet is received out of order, the
"fReorderingBuffer" will store it and afterwards packets in the buffer
"fReorderingBuffer" untill the packet with correct seq# is received or
a threshold time is passed. At this time, the "fReorderingBuffer" may store
some numbers of packets. When the threshold passed, "getNextFrame()" call
will always get a packet from the "rtpSource" immediately. That means,
the "afterReading()" will be called immediately either. When the "blockMyself()"
is called afterwards, the "if" statement is always true and thus the "SingleStep()"
call is always skipped. If this is the case, how can we deal with packets
arrived at this period of time if we dont call the "select()"??
<p>The "SingleStep()" will only be called when all packets in the buffer
"fReorderingBuffer" are consumed. At this time, "afterReading()" can not
be called immediately, and thus the "if" statement in the "blockMyself()"
is false and "SingleStep()" will be called. After the "select()" reading
a packet from the socket, a 'read handler' (is MultiFramedRTPSource::networkReadHandler(),
right?) will process the packet and at the end, will call "doGetNextFrame1()",
the same function call as from the "getNextFrame()". Is my thought right?
<p>My question is still the same, if "SingleStep()" is skipped, how can
we deal with packet arrived at this time??
<br>&nbsp;
<p>Thanks and Best Regards
<br>Michael
<br>&nbsp;
<br>&nbsp;
<p>Ross Finlayson wrote:
<blockquote TYPE=CITE>>I'm PhD student in the School of Computer Scince
and Engineering,
<br>>University of New South Wales, Australia. I'm currently working on
a
<br>>multimedia project that is using live's streaming library. I pached
<br>>live.com's library to mplayer and now mplayer can play an sdp file.
<p>G'day Michael,
<p>To get a good feel as to how the "LIVE.COM Streaming Media" library
works,
<br>you may find it more instructive to look at the test programs that
come
<br>with the library - in particular, "testMPEGAudioVideoStreamer" and
<br>"testMPEGVideoReceiver".&nbsp; These have less MPlayer-specific 'baggage'
<br>associated with them, and so should be easier to understand.
<p>>rtpSource->getNextFrame(dp->buffer, bufferSize, afterReading, ds,
<br>>onSourceClosure, ds);
<br>>rtpSource->envir().taskScheduler().blockMyself((char*)&amp;dp->flags);
<br>>
<br>>To my feeling, the first line is trying to get an RTP frame from the
<br>>fReorderingBuffer, which is a variable defined in the class
<br>>MultiFramedRTPSource. If there is no packet in the
<br>>buffer(fReorderingBuffer), the second line
<br>>will call the blockMyself() function defined in the BasicTaskScheduler
<br>>class.
<p>No, not quite. The call to "getNextFrame()" will arrange for the next
frame
<br>to be read from "rtpSource".&nbsp; When this frame is available,
<br>"afterReading()" will be called.&nbsp; (If the frame is not immediately
<br>available, then a 'read handler' will be registered, so that
<br>"afterReading()" will be called when the frame becomes available.)
<p>Then, *regardless of whether or not a frame was immediately delivered*,
<br>"blockMyself()" gets called.&nbsp; "blockMyself()" works by sitting
in a
<br>'select()' loop, handling the arrival of network packets (or other
<br>events).&nbsp; When a network packet arrives, a read handler (if registered)
<br>will be called to handle it.
<p>>During the interval of two frame times (decoding and
<br>>rendering frame, sleeping untill next frame playing time), I feel
the
<br>>process doesn't read socket paralell for other RTP packet may
<br>>already arrive in the lower level buffer (UDP/IP).
<p>I'm not sure what you mean by this.&nbsp; Whenever a packet arrives,
the
<br>'select()' loop in "blockMyself()" will notice this, and call the
<br>appropriate read handler to deliver the data.&nbsp; The receiving code
will
<br>always be 'busy' - it will never be blocked doing nothing, as long
as a new
<br>incoming packet is available.
<p>Once again, I think you may find the "testMPEGVideoReceiver" code to
be a
<br>bit more instructive here.
<p>>That means, two parallel process
<p>No!&nbsp; The library code is delibererately designed to be single-threaded,
and
<br>will *not* work with multiple threads.&nbsp; As I noted above, the
code should
<br>not be blocking unnecesarily, so multiple threads (in a uniprocessor
<br>system, at least) would not buy you anything anyway.
<p>>I also found that for every mpeg frame(I, P, B), the RTP stream server
<br>>slices it into 16 pieces and creates the RTP packets, no matter how
larger
<br>>or small the mpeg frame is. Is that true?
<p>This must be a feature of your RTP streaming server.&nbsp; Which RTP
streaming
<br>server implementation are you using?&nbsp; The "LIVE.COM Streaming
Media"
<br>library code (e.g., when used in "testMPEGVideoStreamer") doesn't do
this.
<p>(In the future, could you please use the developers' mailing list
<br>&lt;live-devel(at)lists.sourceforge.net> for questions like this.&nbsp;
See
<br>&lt;<a href="http://lists.sourceforge.net/lists/listinfo/live-devel">http://lists.sourceforge.net/lists/listinfo/live-devel</a>>
for information on
<br>how to subscribe.&nbsp; For one-on-one email, I generally give priority
to
<br>paying consulting clients.)
<p>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Ross Finlayson
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; LIVE.COM
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;<a href="http://www.live.com/">http://www.live.com/</a>></blockquote>

<br>&nbsp;</html>

--------------79D8FBBA37BE340ECD460B14--


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From <EMAIL-ADDRESS-HIDDEN> Fri May 31 09:24:23 2002
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X-Original-Date: Fri, 31 May 2002 09:21:59 -0700
Date: Fri, 31 May 2002 09:21:59 -0700

At 12:03 AM 5/31/02, Michael Liu wrote:
>1. In the "testMPEGVideoStreamer" and "testMPEGVideoReceiver", 
>"blockMyself()" gets called without a parameter but in the class 
>BasicTaskScheduler, "blockMyself(char* watchVariable)" does have a 
>parameter "watchvariable". What's the difference?

"watchVariable" is an optional parameter (with default value NULL).  See 
the comment in "UsageEnvironment/include/UsageEnvironment.hh".


>2. After the "getNextFrame()" call, suppose a frame is avaliable 
>immediately, the "afterReading()" will be called immediately. The last 
>line in "afterReading()" is this:
>         dp->flags = ~0;

Note that the code you're referring to occurs only in the MPlayer 
patch.  As I noted earlier, this code is *not* a good example of how to use 
the "LIVE.COM Streaming Media" libary.  It contains some hacks that are 
necessary to get the code to work with MPlayer, but which I would not 
recommend that people use when using the library code 'from scratch'.

In particular, the code that you're referring to in the MPlayer patch  (the 
call to "getNextFrame()" followed by a call to "blockMyself()", with the 
flag being set in "afterReading") is a hack that's used to implement a 
*single, blocking* read - because that's what the enclosing MPlayer code 
expects.  That's why only one packet is received each time this code is 
executed.

For a much better use of the library code, see the code for 
"testMPEGVideoReceiver" or "testMP3Receiver" (or, for transmitting, 
"testMPEGAudioVideoStreamer" or "testMP3Streamer").  These programs use 
"getNetFrame()" to do asychronous (non-blocking) reads, the way it's 
intended, and then use a single "blockMyself()" call to act as an event 
handler.

So, please don't concern yourself to much with the MPlayer code.  It's a 
hack (and will likely undergo significant changes soon anyway).

         Ross.


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From <EMAIL-ADDRESS-HIDDEN> Mon Jun 3 00:35:21 2002
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From: Michael Liu <EMAIL-ADDRESS-HIDDEN>
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X-Original-Date: Mon, 03 Jun 2002 17:33:53 +1000
Date: Mon, 03 Jun 2002 17:33:53 +1000

Hi Ross,

Thank you very much! But unfortunately, I have to use mplayer because my project need a MPEG media player with RTP enabled. Live patch for MPlayer is the one I found with open
source. Is there a way to get the buffer size on the receiver side with current code? Are you going to revise the patch for MPlayer with  "asychronous (non-blocking) reads" in the
future? When?

Thanks
Michael


>
>
> Message: 2
> Date: Fri, 31 May 2002 09:21:59 -0700
> To: live-devel(at)lists.sourceforge.net
> Subject: Re: [Live-devel] Re: packet buffer
>
> At 12:03 AM 5/31/02, Michael Liu wrote:
> >1. In the "testMPEGVideoStreamer" and "testMPEGVideoReceiver",
> >"blockMyself()" gets called without a parameter but in the class
> >BasicTaskScheduler, "blockMyself(char* watchVariable)" does have a
> >parameter "watchvariable". What's the difference?
>
> "watchVariable" is an optional parameter (with default value NULL).  See
> the comment in "UsageEnvironment/include/UsageEnvironment.hh".
>
> >2. After the "getNextFrame()" call, suppose a frame is avaliable
> >immediately, the "afterReading()" will be called immediately. The last
> >line in "afterReading()" is this:
> >         dp->flags = ~0;
>
> Note that the code you're referring to occurs only in the MPlayer
> patch.  As I noted earlier, this code is *not* a good example of how to use
> the "LIVE.COM Streaming Media" libary.  It contains some hacks that are
> necessary to get the code to work with MPlayer, but which I would not
> recommend that people use when using the library code 'from scratch'.
>
> In particular, the code that you're referring to in the MPlayer patch  (the
> call to "getNextFrame()" followed by a call to "blockMyself()", with the
> flag being set in "afterReading") is a hack that's used to implement a
> *single, blocking* read - because that's what the enclosing MPlayer code
> expects.  That's why only one packet is received each time this code is
> executed.
>
> For a much better use of the library code, see the code for
> "testMPEGVideoReceiver" or "testMP3Receiver" (or, for transmitting,
> "testMPEGAudioVideoStreamer" or "testMP3Streamer").  These programs use
> "getNetFrame()" to do asychronous (non-blocking) reads, the way it's
> intended, and then use a single "blockMyself()" call to act as an event
> handler.
>
> So, please don't concern yourself to much with the MPlayer code.  It's a
> hack (and will likely undergo significant changes soon anyway).
>
>          Ross.
>
> --__--__--
>
> _______________________________________________
> Live-devel mailing list
> Live-devel(at)lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/live-devel
>
> End of Live-devel Digest


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Subject: [Live-devel] regarding openRTSP
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X-Original-Date: Mon, 03 Jun 2002 22:11:39 +0530
Date: Mon, 03 Jun 2002 22:11:39 +0530


Hi ,

I wanted to know the following things ,

1> Can the openRTSP be made to receive the RTP streams by giving a http url
like - http://www.starwars.com/starwars.mpeg  for a particular mpeg stream.
Coz one of the things i fell is many of the streaming stations (servers)
initiate through HTTP rather than RTSP.
So i feel the above task could be acheived by the concept of http tunneling
the rtp packets ... but i am not sure , if any one has thought of this or
already have implemented this , kindly help or suggest me how to proceed
using the existing library for openRTSP.

thanks in advance ,
Niranjan.



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From <EMAIL-ADDRESS-HIDDEN> Mon Jun 3 13:34:12 2002
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X-Original-Date: Mon, 03 Jun 2002 13:32:10 -0700
Date: Mon, 03 Jun 2002 13:32:10 -0700

At 09:41 AM 6/3/02, you wrote:
>1> Can the openRTSP be made to receive the RTP streams by giving a http url
>like - http://www.starwars.com/starwars.mpeg  for a particular mpeg stream.

That depends: Is the server a RTSP+RTP server (that just happens to be 
accessible via HTTP)?  If the server just streams using HTTP, without using 
RTSP and RTP at all, then you wouldn't use "openRTSP" to receive its 
data.  (Instead, you'd just use something like "lynx -source".)

>Coz one of the things i fell is many of the streaming stations (servers)
>initiate through HTTP rather than RTSP.

Can you give an example of such a server (that's publically accessible over 
the Internet) - and a working HTTP URL on it?

         Ross.


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X-Original-Date: Mon, 03 Jun 2002 21:05:33 -0700
Date: Mon, 03 Jun 2002 21:05:33 -0700

At 12:33 AM 6/3/02, you wrote:
>Is there a way to get the buffer size on the receiver side with current code?

What "buffer size" are you referring to?  But anyway, remember that you 
have complete source code, so the answer is probably "yes" :-)

>  Are you going to revise the patch for MPlayer with  "asychronous 
> (non-blocking) reads" in the future?

Perhaps, but I'm not 100% sure yet.  Unfortunately, MPlayer was designed to 
do synchronous reads (because it was intended initially to read from a file 
only), but this is potentially a problem when reading from a network 
stream, because it's possible - while blocked waiting to receive an audio 
packet - to miss incoming video packets (or vice versa), unless they happen 
to get buffered in the OS kernel.  So I may end up having to implement my 
own packet buffers (one for audio; one for video) that the asynchronous 
LIVE.COM network reading code would read into.  Then, the 
synchronous-reading MPlayer code would just read from these packet buffers 
(and hopefully rarely, if ever, block).  (I would probably need to do this 
anyway, to implement audio/video sync (something that the code doesn't 
currently do).)

>When?

I may take a look at this within a few weeks, but, as always, tasks 
requested by paying clients take priority...

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>


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Subject: [Live-devel] Winamp RTP plugin
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Hi Ross,

Is there a way to force the Winamp' input plugin in_rtp.dll to generate =
some logs/outputs about its status and actions (for example : =
received/queued/processed/discarded frames) ?

Thanks in advance

Nicolas


------=_NextPart_000_003E_01C20BCB.3E6C5970
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	charset="iso-8859-1"
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2715.400" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>
<DIV><FONT face=3DArial size=3D2>Hi Ross,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Is there a way to force the Winamp' =
input plugin=20
in_rtp.dll to generate some logs/outputs about its status and actions =
(for=20
example : received/queued/processed/discarded frames) ?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Thanks in advance</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Nicolas</FONT></DIV>
<DIV><FONT face=3DArial =
size=3D2></FONT>&nbsp;</DIV></FONT></DIV></BODY></HTML>

------=_NextPart_000_003E_01C20BCB.3E6C5970--


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From <EMAIL-ADDRESS-HIDDEN> Tue Jun 4 11:30:21 2002
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X-Original-Date: Tue, 04 Jun 2002 11:24:16 -0700
Date: Tue, 04 Jun 2002 11:24:16 -0700

At 10:25 AM 6/4/02, Nicolas Drapeau wrote:
>Is there a way to force the Winamp' input plugin in_rtp.dll to generate 
>some logs/outputs about its status and actions (for example : 
>received/queued/processed/discarded frames) ?

No, but the plugin (like all LIVE.COM RTP receiving software) periodically 
sends RTCP "Reception Report" packets (to the session's multicast address, 
and to the session port number+1).  These RTCP packets include packet loss 
statistics.  You could try listening for these packets.

         Ross.


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From <EMAIL-ADDRESS-HIDDEN> Wed Jun 5 05:56:27 2002
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X-Original-Date: Wed,  5 Jun 2002 05:50:54 -0700
Date: Wed,  5 Jun 2002 05:50:54 -0700

Hi Ross,

its been quite a long since I've been off list.

Heres some food for thought...
I am streaming audio/video content embeded in a proprietary file format from
a streaming server. I am using the openRTSP application to recieve rtp
packets. my question is

1/ Which could be the best place to register a callback function to get a
new frame as it is constructed ?
2/ Also since I havent yet integrated the rendering code with  openRTSP, how
can I test to be sure whether I am getting valid data ?

thanks for your patience
Deepesh

Absoluteum obsoletum  (If it works, its obsolete)
    -wanda the fish




Deepesh S Gujarathi
Encodex Technologies (India) Pvt. Ltd.
28, Electronic estate,Pune- 411009
Tel- 91 20 4222 401/402
www.encodexindia.com




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X-Original-Date: Wed, 05 Jun 2002 11:28:02 -0700
Date: Wed, 05 Jun 2002 11:28:02 -0700

At 05:50 AM 6/5/02, you wrote:
>I am streaming audio/video content embeded in a proprietary file format from
>a streaming server. I am using the openRTSP application to recieve rtp
>packets. my question is
>
>1/ Which could be the best place to register a callback function to get a
>new frame as it is constructed ?

Do you mean "get a new frame as it's *received*"?  This callback function 
is the "afterGettingFunc" parameter to "FramedSource::getNextFrame()"..

In "openRTSP", this is done in the call to "fSource->getNextFrame()" in 
"FileSink::continuePlaying()" (file: "liveMedia/FileSink.cpp").

>2/ Also since I havent yet integrated the rendering code with  openRTSP, how
>can I test to be sure whether I am getting valid data ?

The simplest thing you could do is modify the function 
"FileSink::afterGettingFrame()" so that it analyzes each received frame 
instead of (or as well as) writing it to a file.  Each frame's data will be 
in "sink->fBuffer", and its size will be "frameSize" bytes.

         Ross.


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X-Original-Date: Wed, 05 Jun 2002 11:33:29 -0700
Date: Wed, 05 Jun 2002 11:33:29 -0700

I have updated "openRTSP" (both the prebuilt binaries, and the source code) 
to take an optional argument
	-u <username> <password>
This can be used to stream from a access-controlled RTSP server  - i.e., 
one that requires a username and password.

(The code uses "digest authentication", so the user's password will not be 
sent over the net in the clear.)

	Ross.


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X-Original-Date: Thu, 6 Jun 2002 14:51:43 +0200 (CEST)
Date: Thu, 6 Jun 2002 14:51:43 +0200 (CEST)

hello Ross,
ok, i'm trying to stream an MPEG1 video, but what i
remark that when there is a new GOP( Group Of
Picture)there automaticly one parse of Video Sequence
Header, i'm wondering if there is only one GOP in a
Video Sequence if not how can i count the number of
GOPs in a Video Sequence.

thanks in advance


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From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] Question  about GOP
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X-Original-Date: Thu, 06 Jun 2002 09:09:53 -0700
Date: Thu, 06 Jun 2002 09:09:53 -0700

At 05:51 AM 6/6/02, you wrote:
>hello Ross,
>ok, i'm trying to stream an MPEG1 video, but what i
>remark that when there is a new GOP( Group Of
>Picture)there automaticly one parse of Video Sequence
>Header, i'm wondering if there is only one GOP in a
>Video Sequence if not how can i count the number of
>GOPs in a Video Sequence.

I'm not quite sure what you mean, but if you look at the code for 
"MPEGVideoStreamParser", you'll see that Video Sequence Headers are parsed 
in "parseVideoSequenceHeader()", and GOP Headers are parsed in 
"parseGOPHeader()".

You can add print statements to these routines if you want to see exactly 
what your input stream contains.

         Ross.


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Subject: RE: [Live-devel] Winamp RTP plugin
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X-Original-Date: Thu, 6 Jun 2002 14:19:14 -0400
Date: Thu, 6 Jun 2002 14:19:14 -0400

This is a multi-part message in MIME format.

------=_NextPart_000_0037_01C20D65.226C3510
Content-Type: text/plain;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

Hi Ross,

I am not sure if this is a bug or something I misconfigured or did =
wrong, but the Winamp RTP plugin behaves very strangely:

I sniffed RTCP packets going out from the Winamp RTP plugin using =
Ethereal, and I had this surprise: RTCP packets are coming from the good =
source port on the client side (the one given in RTSP transactions, 4701 =
in the example below), but the destination is port 1 (one) on the server =
side (instead of 48101 in the example below). That's why I can't receive =
RTCP packets from the Winamp client.

By the way, my RTSP response to the following RTSP request :

SETUP rtsp://uuu.uuu.uuu.uuu:554/xxx/format=3Dmpa-robust:96 RTSP/1.0\r\n
CSeq: 2\r\n
Transport: RTP/AVP;unicast;client_port=3D4700-4701\r\n
\r\n

is :

RTSP/1.0 200 OK\r\n
CSeq: 2\r\n
Transport: =
RTP/AVP;unicast;client_port=3D4700-4701;server_port=3D48100-48101\r\n
Session: 1855052543\r\n
\r\n

What do you think about that ?

Nicolas

------=_NextPart_000_0037_01C20D65.226C3510
Content-Type: text/html;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2715.400" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DCourier size=3D2>Hi Ross,</FONT></DIV>
<DIV><FONT face=3DCourier size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DCourier size=3D2>I am not sure if this is a bug or =
something I=20
misconfigured or did wrong, but the Winamp RTP plugin behaves very=20
strangely:</FONT></DIV>
<DIV><FONT face=3DCourier size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DCourier size=3D2>I sniffed RTCP packets going out from =
the Winamp=20
RTP plugin using Ethereal, and I had this surprise: RTCP packets are =
coming from=20
the good source port on the client side (the one given in RTSP =
transactions,=20
4701 in the example below), but the destination is port 1 (one) on the =
server=20
side (instead of 48101 in the example below). That's why I can't receive =
RTCP=20
packets from the Winamp client.</FONT></DIV>
<DIV><FONT face=3DCourier size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DCourier size=3D2>By the way, my RTSP response to the =
following=20
RTSP request :</FONT></DIV>
<DIV><FONT face=3DCourier size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DCourier size=3D2>SETUP=20
rtsp://uuu.uuu.uuu.uuu:554/xxx/format=3Dmpa-robust:96 =
RTSP/1.0\r\n</FONT></DIV>
<DIV><FONT face=3DCourier size=3D2>CSeq: 2\r\n</FONT></DIV>
<DIV><FONT face=3DCourier size=3D2>Transport:=20
RTP/AVP;unicast;client_port=3D4700-4701\r\n</FONT></DIV>
<DIV><FONT face=3DCourier size=3D2>\r\n</FONT></DIV>
<DIV><FONT face=3DCourier size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DCourier size=3D2>is :</FONT></DIV>
<DIV><FONT face=3DCourier size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DCourier size=3D2>RTSP/1.0 200 OK\r\n</FONT></DIV>
<DIV><FONT face=3DCourier size=3D2>CSeq: 2\r\n</FONT></DIV>
<DIV><FONT face=3DCourier size=3D2>Transport:=20
RTP/AVP;unicast;client_port=3D4700-4701;server_port=3D48100-48101\r\n</FO=
NT></DIV>
<DIV><FONT face=3DCourier size=3D2>Session: 1855052543\r\n</FONT></DIV>
<DIV><FONT face=3DCourier size=3D2>\r\n</FONT></DIV>
<DIV><FONT face=3DCourier size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DCourier size=3D2>What do you think about that =
?</FONT></DIV>
<DIV><FONT face=3DCourier size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DCourier size=3D2>Nicolas</FONT></DIV></BODY></HTML>

------=_NextPart_000_0037_01C20D65.226C3510--


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Date: Thu, 06 Jun 2002 11:38:28 -0700


>What do you think about that ?

Thanks for the report; it looks like you may, indeed, have found a 
bug.  I'll look into this.  Stay tuned...

         Ross.


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X-Original-Date: Fri, 7 Jun 2002 10:39:06 +0200 (CEST)
Date: Fri, 7 Jun 2002 10:39:06 +0200 (CEST)

hi Ross

 ok, i look at the code of  "MPEGVideoStreamParser"
and when i tryed to print the parse stat i find before
parsing each GOPHeader there is a perse of
VideoSequenceHeader, the problem here that in the
standard MPEG a Sequence video is composed by one or
more GOP, by in the live each Sequence is maked by
only one GOP???

thanks in advance

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X-Original-Date: Fri, 07 Jun 2002 02:17:55 -0700
Date: Fri, 07 Jun 2002 02:17:55 -0700

At 01:39 AM 6/7/02, you wrote:
>  ok, i look at the code of  "MPEGVideoStreamParser"
>and when i tryed to print the parse stat i find before
>parsing each GOPHeader there is a perse of
>VideoSequenceHeader, the problem here that in the
>standard MPEG a Sequence video is composed by one or
>more GOP, by in the live each Sequence is maked by
>only one GOP???

The code is parsing whatever it gets as input.  If the code is parsing only 
one GOP following each Video Sequence Header, then that means that *your 
input data* has only one GOP following each Video Sequence Header!

As far as I can tell, there's nothing wrong with the 
"MPEGVideoStreamParser" code.  But if you know for *sure* that 
the"MPEGVideoStreamParser" is not parsing MPEG video input data correctly, 
then let me know (preferably pointing to an example of MPEG data that is 
not being parsed correctly).

         Ross.



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X-Original-Date: Fri, 07 Jun 2002 10:01:24 -0700
Date: Fri, 07 Jun 2002 10:01:24 -0700

Thanks to Nicolas Drapeau's bug report, I have now fixed the problem that 
would sometimes cause the Winamp RTP plugin 
<http://www.live.com/multikit/windows/in_rtp.dll> to not send RTCP 
Reception reports correctly when streaming from certain RTSP servers.

I have also installed new binary versions of "openRTSP" (which had the same 
problem), and a new release of the "LIVE.COM Streaming Media" libary code 
which includes the fix.

	Ross.


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X-Original-Date: Mon, 10 Jun 2002 11:27:29 -0700 (PDT)
Date: Mon, 10 Jun 2002 11:27:29 -0700 (PDT)

hi Ross,
 in the shutdown routine , there is a call to 
closeMediaSinks().. is it necessary??
are more than one mediaSinks created?
isnt it sufficient if we just close the rtspClient. ??

the call to closeMediaSinks is giving a segmentation
fault at " delete toDelete; " in Media.cpp

Your help would be greatly appreciated.

thankyou
surya




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X-Original-Date: Mon, 10 Jun 2002 12:25:10 -0700
Date: Mon, 10 Jun 2002 12:25:10 -0700

At 11:27 AM 6/10/02, you wrote:
>  in the shutdown routine , there is a call to
>closeMediaSinks()

OK, I presume you're referring to "openRTSP".

>.. is it necessary??

In this case, not really, because the output files will get flushed & 
closed automatically anyway once the program exits.

>are more than one mediaSinks created?

There's one 'mediaSink' for each output file.

>the call to closeMediaSinks is giving a segmentation
>fault at " delete toDelete; " in Media.cpp

This should not be happening.  Have you made any alterations to the 
code?  If not, please tell me how you are running "openRTSP" - i.e., what 
arguments you are giving it.  Also, please let me know what it's 
outputting.  If this crash is happening with an unaltered "openRTSP", I'd 
like to track it down.

         Ross.




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X-Original-Date: Mon, 10 Jun 2002 21:27:02 +0000
Date: Mon, 10 Jun 2002 21:27:02 +0000

Hello

Currently, live.2002.05.20 streams only MPEG-1 and MPEG-2 video. I could 
build testmpegvideostreamer and testmpegvideoreceiver on Windows-ME platform 
and stream and receive a test.mpg MPEG-1 video.  I am very much impressed by 
LIVE software.

Can you please let me know how to achieve the following:
1) How to dump RTCP communication packets, so that I can study and 
understand the RTCP protocol.
2) I would like modify the software to stream MPEG-4 simple-profile video.  
Its header structure is very similar to that of MPEG-1 video.
Can you please tell me which files in LIVE I need to modify for this purpose 
or any other tips how to implement it?

Regards
Putti



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X-Original-Date: Mon, 10 Jun 2002 14:46:29 -0700
Date: Mon, 10 Jun 2002 14:46:29 -0700


>Can you please let me know how to achieve the following:
>1) How to dump RTCP communication packets, so that I can study and 
>understand the RTCP protocol.

RTCP packets are sent/received in "liveMedia/RTCP.cpp".  You could modify 
this code.

>2) I would like modify the software to stream MPEG-4 simple-profile video.

FYI, supporting MPEG-4 video (sending & receiving) is on my "to do" 
list.  But, of course, others are welcome to start working on this also.

>Its header structure is very similar to that of MPEG-1 video.
>Can you please tell me which files in LIVE I need to modify for this 
>purpose or any other tips how to implement it?

To send MPEG-4 video, you would need a new "MPEG4VideoStreamFramer" class, 
and a new "MPEG4VideoRTPSink" class - analogous to the existing 
"MPEGVideoStreamFramer" and "MPEGVideoRTPSink" classes (for MPEG-1,2).

To receive MPEG-4 video, you would need a new "MPEG4VideoRTPSource" class - 
analogous the existing "MPEGVideoRTPSource" class (for MPEG-1,2).

         Ross.


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X-Original-Date: Mon, 10 Jun 2002 18:10:46 -0400
Date: Mon, 10 Jun 2002 18:10:46 -0400

On Mon, Jun 10, 2002 at 09:27:02PM +0000, putti gutti wrote:
> Hello
> 
> Currently, live.2002.05.20 streams only MPEG-1 and MPEG-2 video. I could 
> build testmpegvideostreamer and testmpegvideoreceiver on Windows-ME platform 
> 
> and stream and receive a test.mpg MPEG-1 video.  I am very much impressed by 
> 
> LIVE software.
> 
> Can you please let me know how to achieve the following:
> 1) How to dump RTCP communication packets, so that I can study and 
> understand the RTCP protocol.
> 2) I would like modify the software to stream MPEG-4 simple-profile video.  
> Its header structure is very similar to that of MPEG-1 video.
> Can you please tell me which files in LIVE I need to modify for this purpose 
> 
> or any other tips how to implement it?

You'd probably need to start with a MPEG4 codec implementation.  In the
OSS world, your best bets are: 

Xvid (http://www.xvid.org/), good quality/speed, ISO compliant, used by
	the mpeg4ip project

ffmpeg (http://ffmpeg.sourceforge.net/), simple to use and understand
	code (in libavcodec), I don't know about it's compliance yet.

There's also the ISO reference codec developed by MS, which has it's own
	strange copyright (and is not optimized very much (slow)).

-SteveK

-- 
  Steve Kann - Chief Engineer - 520 8th Ave #2300 NY 10018 -  (212) 533-1775
                HorizonLive.com - collaborate . interact . learn
   "The box said 'Requires Windows 95, NT, or better,' so I installed Linux."

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X-Original-Date: Mon, 10 Jun 2002 15:27:34 -0700
Date: Mon, 10 Jun 2002 15:27:34 -0700


>You'd probably need to start with a MPEG4 codec implementation.

Of course, that's true only if he doesn't already have a MPEG-4 encoder.

The existing MPEG-1,2 support also requires an encoder, of course - or else 
the use of -pre-encoded data.  MPEG-4 will be no different.

         Ross.


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From: {putti gutti} <EMAIL-ADDRESS-HIDDEN>
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Subject: Re: [Live-devel] on RTCP and MPEG-4 video streaming
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X-Original-Date: Tue, 11 Jun 2002 14:57:19 +0000
Date: Tue, 11 Jun 2002 14:57:19 +0000


Dear Ross,

Thank you very much for your reply.

I observe that LIVE form an RTP packet for each slice in the Picture.  And 
sends an RTCP packet/report before each RTP packet.  This leads to lot of 
overheads in the form of RTP and RTCP header bits.

Why is it not advisible to send
1) a RTP packet for every picture ( i.e. every 15 slices in a picture of 
height = 240) instead of every slice and
2) an RTCP packet periodically after every few RTP packets?

Thank you and Steve very much for the info on MPEG-4.  As you said, I was 
thinking of simply streaming an MPEG-4 video stream using LIVE.  May I know 
roughly when is it likely to have MPEG-4 video streaming capability in LIVE?

regards
Putti


>From: Ross Finlayson <finlayson(at)live.com>
>To: live-devel(at)lists.sourceforge.net
>Subject: Re: [Live-devel] on RTCP and MPEG-4 video streaming
>Date: Mon, 10 Jun 2002 14:46:29 -0700
>
>
>>Can you please let me know how to achieve the following:
>>1) How to dump RTCP communication packets, so that I can study and
>>understand the RTCP protocol.
>
>RTCP packets are sent/received in "liveMedia/RTCP.cpp".  You could modify
>this code.
>
>>2) I would like modify the software to stream MPEG-4 simple-profile video.
>
>FYI, supporting MPEG-4 video (sending & receiving) is on my "to do"
>list.  But, of course, others are welcome to start working on this also.
>
>>Its header structure is very similar to that of MPEG-1 video.
>>Can you please tell me which files in LIVE I need to modify for this
>>purpose or any other tips how to implement it?
>
>To send MPEG-4 video, you would need a new "MPEG4VideoStreamFramer" class,
>and a new "MPEG4VideoRTPSink" class - analogous to the existing
>"MPEGVideoStreamFramer" and "MPEGVideoRTPSink" classes (for MPEG-1,2).
>
>To receive MPEG-4 video, you would need a new "MPEG4VideoRTPSource" class -
>analogous the existing "MPEGVideoRTPSource" class (for MPEG-1,2).
>
>         Ross.
>
>
>_______________________________________________________________
>
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>August 25-28 in Las Vegas - 
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X-Original-Date: Tue, 11 Jun 2002 12:37:58 -0700
Date: Tue, 11 Jun 2002 12:37:58 -0700


>Why is it not advisible to send
>1) a RTP packet for every picture ( i.e. every 15 slices in a picture of 
>height = 240) instead of every slice

Because the RTP payload format for MPEG-1,2 video (as described in RFC 
2250) specifies that each slice is sent in a separate RTP packet.  Note 
that slices are typically quite large, so the overhead of a RTP header is 
negligible.  Also, having each slice in a separate RTP packet allows 
receivers to display the rest of the slices for each frame, even if one RTP 
packet gets lost.

>2) an RTCP packet periodically after every few RTP packets?

The RTCP implementation already does this.  Note that it does *not* "send 
an RTCP packet/report before each RTP packet".

>  May I know roughly when is it likely to have MPEG-4 video streaming 
> capability in LIVE?

It's uncertain - as always, tasks requested by paying clients take 
priority.  Many of the features of the library have been added as a result 
of  funding from outside companies.  So, if your organization is interested 
in providing funding for MPEG-4 support, then please let me know (via 
private email).)

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>


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X-Original-Date: Tue, 11 Jun 2002 15:43:53 -0700 (PDT)
Date: Tue, 11 Jun 2002 15:43:53 -0700 (PDT)

hi Ross,
 i am trying to compile the livemedia library for
compaq ipaq using arm-linux-c++.

the inet.c in groupsock subdirectory which is using
old style of C programming which is not being
recognized by g++ or arm-linux-c++.
so i tried to change the function prtotypes for
example

 void fun(x)
  int x;
 { ...}

to

  void fun(int x)
  {... }
and i added the header file #include <netdb.h>

The inet.c compiled but the functions in the inet.c
which are being used by other cpp files like
Groupsock.cpp, etc are giving undefined reference
errors.

 what should i do to correct these errors??

Your help would be great.
thank you
surya
  


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X-Original-Date: Tue, 11 Jun 2002 15:55:52 -0700
Date: Tue, 11 Jun 2002 15:55:52 -0700

At 03:43 PM 6/11/02, you wrote:
>  i am trying to compile the livemedia library for
>compaq ipaq using arm-linux-c++.
>
>the inet.c in groupsock subdirectory which is using
>old style of C programming which is not being
>recognized by g++ or arm-linux-c++.
>so i tried to change the function prtotypes for
>example
>
>  void fun(x)
>   int x;
>  { ...}
>
>to
>
>   void fun(int x)
>   {... }

Could you please mail me your modified version of "inet.c", so I can look 
at including it in the next source code release.

>and i added the header file #include <netdb.h>

Out of curiosity, why did you need to add this #include?  I.e., what was 
undefined by leaving this out?

>The inet.c compiled but the functions in the inet.c
>which are being used by other cpp files like
>Groupsock.cpp, etc are giving undefined reference
>errors.

Please let me know what these error messages are.

         Ross.


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X-Original-Date: Wed, 12 Jun 2002 08:01:30 -0700 (PDT)
Date: Wed, 12 Jun 2002 08:01:30 -0700 (PDT)

--0-1774403452-1023894090=:81051
Content-Type: text/plain; charset=us-ascii
Content-Disposition: inline

hi Ross,
 i included #include<netdb.h> because i got an error
saying 
 implicit declaration of gethostbyname(..)

i am sending the changed inet.c and the errors of the
build process as attachments.

please note that i tried to compile all teh files
using c++.. i changed the 
C_COMPILER= c++ 
in    config.linux

i am also sending the config.linux file.

please do let me know if i am doing wrong.
thank you
surya

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--0-1774403452-1023894090=:81051
Content-Type: text/plain; name="inet.c"
Content-Description: inet.c
Content-Disposition: inline; filename="inet.c"

/* Some systems (e.g., SunOS) have header files that erroneously declare
 * inet_addr(), inet_ntoa() and gethostbyname() as taking no arguments.
 * This confuses C++.  To overcome this, we use our own routines,
 * implemented in C.
 */

#if defined(__WIN32__) || defined(_WIN32)
#include <windows.h>
#else
#include <sys/types.h>
#include <netinet/in.h>
#include <arpa/inet.h>
#include <stdlib.h>
#include <netdb.h>
#endif
#include <stdio.h>

u_long our_inet_addr(char *cp)
//	register /*#####const*/ char *cp;
{
	return inet_addr(cp);
}

char* our_inet_ntoa( struct in_addr in)
//        struct in_addr in;
{
  return inet_ntoa(in);
}

#if defined(__WIN32__) || defined(_WIN32)
#define WS_VERSION_CHOICE1 MAKEWORD(2,2)
#define WS_VERSION_CHOICE2 MAKEWORD(1,1)
int initializeWinsockIfNecessary() {
	// We need to call an initialization routine before
	// we can do anything with winsock.  (How fucking lame!):
	static int _haveInitializedWinsock = 0;
	WSADATA	wsadata;

	if (!_haveInitializedWinsock) {
		if (WSAStartup(WS_VERSION_CHOICE1, &wsadata) != 0
		    && WSAStartup(WS_VERSION_CHOICE2, &wsadata) != 0) {
			return 0; /* error in initialization */
		}
	    	if (wsadata.wVersion != WS_VERSION_CHOICE1
	    	    && wsadata.wVersion != WS_VERSION_CHOICE2) {
	        	WSACleanup();
				return 0; /* desired Winsock version was not available */
		}
		_haveInitializedWinsock = 1;
	}

	return 1;
}
#else
#define initializeWinsockIfNecessary() 1
#endif

#ifndef NULL
#define NULL 0
#endif

struct hostent* our_gethostbyname(char* name)
//     char* name;
{
	if (!initializeWinsockIfNecessary()) return NULL;

	return (struct hostent*) gethostbyname((char*)name);
}


/* Some systems don't seem to know about bcopy(); provide it ourselves
 *
 * sizeof(word) MUST BE A POWER OF TWO
 * SO THAT wmask BELOW IS ALL ONES
 */
typedef	int word;		/* "word" used for optimal copy speed */

#define	wsize	sizeof(word)
#define	wmask	(wsize - 1)

/*
 * Copy a block of memory, handling overlap.
 */
void our_bcopy(void *src0, void *dst0, size_t length)
//	void *dst0;
//	/*#####const*/ void *src0;
//	register size_t length;
{
	register char *dst =(char *)dst0;
	register /*#####const*/ char *src =(char *)src0;
	register size_t t;

	if (length == 0 || dst == src)		/* nothing to do */
		goto done;

	/*
	 * Macros: loop-t-times; and loop-t-times, t>0
	 */
#define	TLOOP(s) if (t) TLOOP1(s)
#define	TLOOP1(s) do { s; } while (--t)

	if ((unsigned long)dst < (unsigned long)src) {
		/*
		 * Copy forward.
		 */
		t = (int)src;	/* only need low bits */
		if ((t | (int)dst) & wmask) {
			/*
			 * Try to align operands.  This cannot be done
			 * unless the low bits match.
			 */
			if ((t ^ (int)dst) & wmask || length < wsize)
				t = length;
			else
				t = wsize - (t & wmask);
			length -= t;
			TLOOP1(*dst++ = *src++);
		}
		/*
		 * Copy whole words, then mop up any trailing bytes.
		 */
		t = length / wsize;
		TLOOP(*(word *)dst = *(word *)src; src += wsize; dst += wsize);
		t = length & wmask;
		TLOOP(*dst++ = *src++);
	} else {
		/*
		 * Copy backwards.  Otherwise essentially the same.
		 * Alignment works as before, except that it takes
		 * (t&wmask) bytes to align, not wsize-(t&wmask).
		 */
		src += length;
		dst += length;
		t = (int)src;
		if ((t | (int)dst) & wmask) {
			if ((t ^ (int)dst) & wmask || length <= wsize)
				t = length;
			else
				t &= wmask;
			length -= t;
			TLOOP1(*--dst = *--src);
		}
		t = length / wsize;
		TLOOP(src -= wsize; dst -= wsize; *(word *)dst = *(word *)src);
		t = length & wmask;
		TLOOP(*--dst = *--src);
	}
done:
	return;
}

/*
 * random.c:
 *
 * An improved random number generation package.  In addition to the standard
 * rand()/srand() like interface, this package also has a special state info
 * interface.  The our_initstate() routine is called with a seed, an array of
 * bytes, and a count of how many bytes are being passed in; this array is
 * then initialized to contain information for random number generation with
 * that much state information.  Good sizes for the amount of state
 * information are 32, 64, 128, and 256 bytes.  The state can be switched by
 * calling the our_setstate() routine with the same array as was initiallized
 * with our_initstate().  By default, the package runs with 128 bytes of state
 * information and generates far better random numbers than a linear
 * congruential generator.  If the amount of state information is less than
 * 32 bytes, a simple linear congruential R.N.G. is used.
 *
 * Internally, the state information is treated as an array of longs; the
 * zeroeth element of the array is the type of R.N.G. being used (small
 * integer); the remainder of the array is the state information for the
 * R.N.G.  Thus, 32 bytes of state information will give 7 longs worth of
 * state information, which will allow a degree seven polynomial.  (Note:
 * the zeroeth word of state information also has some other information
 * stored in it -- see our_setstate() for details).
 *
 * The random number generation technique is a linear feedback shift register
 * approach, employing trinomials (since there are fewer terms to sum up that
 * way).  In this approach, the least significant bit of all the numbers in
 * the state table will act as a linear feedback shift register, and will
 * have period 2^deg - 1 (where deg is the degree of the polynomial being
 * used, assuming that the polynomial is irreducible and primitive).  The
 * higher order bits will have longer periods, since their values are also
 * influenced by pseudo-random carries out of the lower bits.  The total
 * period of the generator is approximately deg*(2**deg - 1); thus doubling
 * the amount of state information has a vast influence on the period of the
 * generator.  Note: the deg*(2**deg - 1) is an approximation only good for
 * large deg, when the period of the shift register is the dominant factor.
 * With deg equal to seven, the period is actually much longer than the
 * 7*(2**7 - 1) predicted by this formula.
 */

/*
 * For each of the currently supported random number generators, we have a
 * break value on the amount of state information (you need at least this
 * many bytes of state info to support this random number generator), a degree
 * for the polynomial (actually a trinomial) that the R.N.G. is based on, and
 * the separation between the two lower order coefficients of the trinomial.
 */
#define	TYPE_0		0		/* linear congruential */
#define	BREAK_0		8
#define	DEG_0		0
#define	SEP_0		0

#define	TYPE_1		1		/* x**7 + x**3 + 1 */
#define	BREAK_1		32
#define	DEG_1		7
#define	SEP_1		3

#define	TYPE_2		2		/* x**15 + x + 1 */
#define	BREAK_2		64
#define	DEG_2		15
#define	SEP_2		1

#define	TYPE_3		3		/* x**31 + x**3 + 1 */
#define	BREAK_3		128
#define	DEG_3		31
#define	SEP_3		3

#define	TYPE_4		4		/* x**63 + x + 1 */
#define	BREAK_4		256
#define	DEG_4		63
#define	SEP_4		1

/*
 * Array versions of the above information to make code run faster --
 * relies on fact that TYPE_i == i.
 */
#define	MAX_TYPES	5		/* max number of types above */

static int degrees[MAX_TYPES] =	{ DEG_0, DEG_1, DEG_2, DEG_3, DEG_4 };
static int seps [MAX_TYPES] =	{ SEP_0, SEP_1, SEP_2, SEP_3, SEP_4 };

/*
 * Initially, everything is set up as if from:
 *
 *	our_initstate(1, &randtbl, 128);
 *
 * Note that this initialization takes advantage of the fact that srandom()
 * advances the front and rear pointers 10*rand_deg times, and hence the
 * rear pointer which starts at 0 will also end up at zero; thus the zeroeth
 * element of the state information, which contains info about the current
 * position of the rear pointer is just
 *
 *	MAX_TYPES * (rptr - state) + TYPE_3 == TYPE_3.
 */

static long randtbl[DEG_3 + 1] = {
	TYPE_3,
	0x9a319039, 0x32d9c024, 0x9b663182, 0x5da1f342, 0xde3b81e0, 0xdf0a6fb5,
	0xf103bc02, 0x48f340fb, 0x7449e56b, 0xbeb1dbb0, 0xab5c5918, 0x946554fd,
	0x8c2e680f, 0xeb3d799f, 0xb11ee0b7, 0x2d436b86, 0xda672e2a, 0x1588ca88,
	0xe369735d, 0x904f35f7, 0xd7158fd6, 0x6fa6f051, 0x616e6b96, 0xac94efdc,
	0x36413f93, 0xc622c298, 0xf5a42ab8, 0x8a88d77b, 0xf5ad9d0e, 0x8999220b,
	0x27fb47b9,
};

/*
 * fptr and rptr are two pointers into the state info, a front and a rear
 * pointer.  These two pointers are always rand_sep places aparts, as they
 * cycle cyclically through the state information.  (Yes, this does mean we
 * could get away with just one pointer, but the code for random() is more
 * efficient this way).  The pointers are left positioned as they would be
 * from the call
 *
 *	our_initstate(1, randtbl, 128);
 *
 * (The position of the rear pointer, rptr, is really 0 (as explained above
 * in the initialization of randtbl) because the state table pointer is set
 * to point to randtbl[1] (as explained below).
 */
static long *fptr = &randtbl[SEP_3 + 1];
static long *rptr = &randtbl[1];

/*
 * The following things are the pointer to the state information table, the
 * type of the current generator, the degree of the current polynomial being
 * used, and the separation between the two pointers.  Note that for efficiency
 * of random(), we remember the first location of the state information, not
 * the zeroeth.  Hence it is valid to access state[-1], which is used to
 * store the type of the R.N.G.  Also, we remember the last location, since
 * this is more efficient than indexing every time to find the address of
 * the last element to see if the front and rear pointers have wrapped.
 */
static long *state = &randtbl[1];
static int rand_type = TYPE_3;
static int rand_deg = DEG_3;
static int rand_sep = SEP_3;
static long *end_ptr = &randtbl[DEG_3 + 1];

/*
 * srandom:
 *
 * Initialize the random number generator based on the given seed.  If the
 * type is the trivial no-state-information type, just remember the seed.
 * Otherwise, initializes state[] based on the given "seed" via a linear
 * congruential generator.  Then, the pointers are set to known locations
 * that are exactly rand_sep places apart.  Lastly, it cycles the state
 * information a given number of times to get rid of any initial dependencies
 * introduced by the L.C.R.N.G.  Note that the initialization of randtbl[]
 * for default usage relies on values produced by this routine.
 */
long our_random(); /*forward*/

void our_srandom(unsigned int x)
//	unsigned int x;
{
	register int i, j;

	if (rand_type == TYPE_0)
		state[0] = x;
	else {
		j = 1;
		state[0] = x;
		for (i = 1; i < rand_deg; i++)
			state[i] = 1103515245 * state[i - 1] + 12345;
		fptr = &state[rand_sep];
		rptr = &state[0];
		for (i = 0; i < 10 * rand_deg; i++)
			(void)our_random();
	}
}

/*
 * our_initstate:
 *
 * Initialize the state information in the given array of n bytes for future
 * random number generation.  Based on the number of bytes we are given, and
 * the break values for the different R.N.G.'s, we choose the best (largest)
 * one we can and set things up for it.  srandom() is then called to
 * initialize the state information.
 *
 * Note that on return from srandom(), we set state[-1] to be the type
 * multiplexed with the current value of the rear pointer; this is so
 * successive calls to our_initstate() won't lose this information and will be
 * able to restart with our_setstate().
 *
 * Note: the first thing we do is save the current state, if any, just like
 * our_setstate() so that it doesn't matter when our_initstate is called.
 *
 * Returns a pointer to the old state.
 */
char* our_initstate(unsigned int seed,char *arg_state,int n)
//	unsigned int seed;		/* seed for R.N.G. */
//	char *arg_state;		/* pointer to state array */
//	int n;				/* # bytes of state info */
{
	register char *ostate = (char *)(&state[-1]);

	if (rand_type == TYPE_0)
		state[-1] = rand_type;
	else
		state[-1] = MAX_TYPES * (rptr - state) + rand_type;
	if (n < BREAK_0) {
		(void)fprintf(stderr,
		    "random: not enough state (%d bytes); ignored.\n", n);
		return(0);
	}
	if (n < BREAK_1) {
		rand_type = TYPE_0;
		rand_deg = DEG_0;
		rand_sep = SEP_0;
	} else if (n < BREAK_2) {
		rand_type = TYPE_1;
		rand_deg = DEG_1;
		rand_sep = SEP_1;
	} else if (n < BREAK_3) {
		rand_type = TYPE_2;
		rand_deg = DEG_2;
		rand_sep = SEP_2;
	} else if (n < BREAK_4) {
		rand_type = TYPE_3;
		rand_deg = DEG_3;
		rand_sep = SEP_3;
	} else {
		rand_type = TYPE_4;
		rand_deg = DEG_4;
		rand_sep = SEP_4;
	}
	state = &(((long *)arg_state)[1]);	/* first location */
	end_ptr = &state[rand_deg];	/* must set end_ptr before srandom */
	our_srandom(seed);
	if (rand_type == TYPE_0)
		state[-1] = rand_type;
	else
		state[-1] = MAX_TYPES*(rptr - state) + rand_type;
	return(ostate);
}

/*
 * our_setstate:
 *
 * Restore the state from the given state array.
 *
 * Note: it is important that we also remember the locations of the pointers
 * in the current state information, and restore the locations of the pointers
 * from the old state information.  This is done by multiplexing the pointer
 * location into the zeroeth word of the state information.
 *
 * Note that due to the order in which things are done, it is OK to call
 * our_setstate() with the same state as the current state.
 *
 * Returns a pointer to the old state information.
 */
char* our_setstate(char *arg_state)
//	char *arg_state;
{
	register long *new_state = (long *)arg_state;
	register int type = new_state[0] % MAX_TYPES;
	register int rear = new_state[0] / MAX_TYPES;
	char *ostate = (char *)(&state[-1]);

	if (rand_type == TYPE_0)
		state[-1] = rand_type;
	else
		state[-1] = MAX_TYPES * (rptr - state) + rand_type;
	switch(type) {
	case TYPE_0:
	case TYPE_1:
	case TYPE_2:
	case TYPE_3:
	case TYPE_4:
		rand_type = type;
		rand_deg = degrees[type];
		rand_sep = seps[type];
		break;
	default:
		(void)fprintf(stderr,
		    "random: state info corrupted; not changed.\n");
	}
	state = &new_state[1];
	if (rand_type != TYPE_0) {
		rptr = &state[rear];
		fptr = &state[(rear + rand_sep) % rand_deg];
	}
	end_ptr = &state[rand_deg];		/* set end_ptr too */
	return(ostate);
}

/*
 * random:
 *
 * If we are using the trivial TYPE_0 R.N.G., just do the old linear
 * congruential bit.  Otherwise, we do our fancy trinomial stuff, which is
 * the same in all the other cases due to all the global variables that have
 * been set up.  The basic operation is to add the number at the rear pointer
 * into the one at the front pointer.  Then both pointers are advanced to
 * the next location cyclically in the table.  The value returned is the sum
 * generated, reduced to 31 bits by throwing away the "least random" low bit.
 *
 * Note: the code takes advantage of the fact that both the front and
 * rear pointers can't wrap on the same call by not testing the rear
 * pointer if the front one has wrapped.
 *
 * Returns a 31-bit random number.
 */
long our_random()
{
	long i;

	if (rand_type == TYPE_0)
		i = state[0] = (state[0] * 1103515245 + 12345) & 0x7fffffff;
	else {
		*fptr += *rptr;
		i = (*fptr >> 1) & 0x7fffffff;	/* chucking least random bit */
		if (++fptr >= end_ptr) {
			fptr = state;
			++rptr;
		} else if (++rptr >= end_ptr)
			rptr = state;
	}
	return(i);
}


#ifdef USE_OUR_BZERO
void __bzero (char *to,int count)
//  char *to;
//  int count;
{
  while (count-- > 0)
    {
      *to++ = 0;
    }
}             
#endif

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--0-1774403452-1023894090=:81051
Content-Type: text/plain; name="config.linux"
Content-Description: config.linux
Content-Disposition: inline; filename="config.linux"

COMPILE_OPTS =		$(INCLUDES) -I. -g -O -DSOCKLEN_T=socklen_t
C =			c
C_COMPILER =		c++	
C_FLAGS =		$(COMPILE_OPTS) -DUSE_OUR_BZERO=1
CPP =			cpp
CPLUSPLUS_COMPILER =	c++
CPLUSPLUS_FLAGS =	$(COMPILE_OPTS) -Wall -DBSD=1
OBJ =			o
LINK =			c++ -o
LINK_OPTS =		-L.
INCLUDE =		/usr/local/rtp2/mpeglib/mpeglib/lib/input
CONSOLE_LINK_OPTS =	$(LINK_OPTS)
LIBRARY_LINK =		ld -o
LIBRARY_LINK_OPTS =	$(LINK_OPTS) -r -Bstatic
LIB_SUFFIX =			a
LIBS_FOR_CONSOLE_APPLICATION =
LIBS_FOR_GUI_APPLICATION =
EXE =

--0-1774403452-1023894090=:81051--

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From <EMAIL-ADDRESS-HIDDEN> Wed Jun 12 08:43:30 2002
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X-Original-Date: Wed, 12 Jun 2002 08:40:54 -0700
Date: Wed, 12 Jun 2002 08:40:54 -0700

At 08:01 AM 6/12/02, you wrote:
>please note that i tried to compile all teh files
>using c++

Bad idea!  This explains the problems that you had.  The "*.c" files must 
be compiled with a C compiler, not a C++ compiler.  Because they're C 
files, the functions in these files get compiled with C linkage - that's 
why the rest of the code declares these functions as 'extern "C"'

>.. i changed the
>C_COMPILER= c++
>in    config.linux

No, please leave this as "cc", or whatever your C compiler is called.  If 
you do this, you should not have to change the rest of the code much (if at 
all), and you should have little problem compiling it.

         Ross.


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From <EMAIL-ADDRESS-HIDDEN> Thu Jun 13 09:54:52 2002
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X-Original-Date: Thu, 13 Jun 2002 09:52:54 -0700 (PDT)
Date: Thu, 13 Jun 2002 09:52:54 -0700 (PDT)

hi Ross,
 i have another question..
 can the livemedia RTSPServer.cpp be used as a server
instead of real server to test the streaming?
what should i do to use openRTSP as the client and
RTSPServer.cpp as the server?

thank you
surya

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X-Original-Date: Thu, 13 Jun 2002 10:15:09 -0700
Date: Thu, 13 Jun 2002 10:15:09 -0700

At 09:52 AM 6/13/02, you wrote:
>hi Ross,
>  i have another question..
>  can the livemedia RTSPServer.cpp be used as a server
>instead of real server to test the streaming?

Yes, this will work (although note that the implementation in 
RTSPServer.cpp currently works only for multicast sessions).

>what should i do to use openRTSP as the client and
>RTSPServer.cpp as the server?

See the "testMP3Streamer", "testMPEGVideoStreamer" and 
"testMPEGAudioVideoStreamer" test programs for examples of how to use the 
RTSP server.  (Note that in these test programs, you'll need to uncomment 
the line
#define IMPLEMENT_RTSP_SERVER 1

"openRTSP" can be run the same way as for any other RTSP server.

         Ross.


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Subject: [Live-devel] integrating GUI player with live
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X-Original-Date: Fri, 14 Jun 2002 18:40:18 +0530
Date: Fri, 14 Jun 2002 18:40:18 +0530

Hi Ross,

I need to integrate my player GUI and codec with the 'live' library.
Actually I'm dealing with a custom file format, rtp payload(#101) and
codec(proprietary implementation of MPEG). I have gone through the 'live'
code and implementing the rtp payload seems achievable.
The most dificult part to me is the Usage Environment, Task Scheduling. I
admit I don't have a enough understanding of these classes. Anywayz, my
question is:

a/ how difficult would t be to write similar implementation without using
UsageEnvironment, BasicUsageEnvironment and TaskScheduler ?
b/ how can I integrate the library within the GUI thread ?
c/ what files would require changes ?

PS : My player can take care of codec/buffering/rendering related issues. I
would make use of the RTP/RTSP functionality of the 'live' library.

thanks for your time and patience

Deepesh

Absoluteum obsoletum  (If it works, its obsolete)
    -wanda the fish




Deepesh S Gujarathi
Encodex Technologies (India) Pvt. Ltd.
28, Electronic estate,Pune- 411009
Tel- 91 20 4222 401/402
www.encodexindia.com



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X-Original-Date: Fri, 14 Jun 2002 10:33:48 -0700
Date: Fri, 14 Jun 2002 10:33:48 -0700


>a/ how difficult would t be to write similar implementation without using
>UsageEnvironment, BasicUsageEnvironment and TaskScheduler ?

Deepesh,

The library code requires the use of instances of the "UsageEnvironment" 
and "TaskScheduler" classes.  (These are abstract base classes.)  However, 
the  objects that implement these classes do not *have to* be of type 
"BasicUsageEnvironment" or "BasicTaskScheduler"; instead, you *can* 
implement your own subclasses of "UsageEnvironment" and/or "TaskScheduler", 
and use these instead.

The biggest decision you have to make is how to subclass 
"TaskScheduler".  If the rest of your application has its own 
scheduling/event handling mechanism, then you may need to write your own 
subclass of "TaskScheduler" that fits into this.  Alternatively, you may be 
able to use the scheduling/event handling mechanism that's built into 
"BasicTaskScheduler", and build the rest of your application around 
this.  It depends on your application...

>b/ how can I integrate the library within the GUI thread ?

It's impossible for me to answer questions like this without knowing more 
the architecture of your player application.  If your company is 
interested, I would be happy to provide personalized consulting to help you 
integrate the "LIVE.COM Streaming Media" library into your 
application.  (If you or your management are interested in this, please let 
me know (via private email).

         Ross.


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X-Original-Date: Sat, 15 Jun 2002 18:00:53 +0530
Date: Sat, 15 Jun 2002 18:00:53 +0530


Hi Ross,

thanks for your prompt response. I went through all the files that you
mentioned and now I do have little understanding of the BasicTaskScheduler
and BasicUsageEnvironment and DelayQueue, except for the function

env->taskScheduler().blockMyself(); // does not return

is it necessary to make a call to this function ? if yes in which situation
?

PS : I think it would be wonderful if there was some basic documentation on
the LIVE.com library available for reference to the developer community. For
a start, I'm working on some HTML docs (class diagrams etc.). Do let me know
how I can make it available publicly (I don't own a public web space).
Either I upload it to some site or send them to you via mail offlist (you
may want to edit it before making them public).

Absoluteum obsoletum  (If it works, its obsolete)
    -wanda the fish

Deepesh S Gujarathi
Encodex Technologies (India) Pvt. Ltd.
28, Electronic estate,Pune- 411009
Tel- 91 20 4222 401/402
www.encodexindia.com



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X-Original-Date: Sat, 15 Jun 2002 09:24:28 -0700
Date: Sat, 15 Jun 2002 09:24:28 -0700

At 05:30 AM 6/15/02, you wrote:
>env->taskScheduler().blockMyself(); // does not return
>
>is it necessary to make a call to this function ? if yes in which situation

"blockMyself()" is used to implement an event loop.  When this function is 
called, the code will enter an event-handling loop, repeatedly handling 
pending events (arriving network packets, or delay timer expirations).  So, 
you call "blockMyself()" when you're ready to handle incoming network 
packets, or delayed tasks.  (Conversely, if you *don't* call 
"blockMyself()", then incoming network packets and delayed tasks won't get 
processed.)


>PS : I think it would be wonderful if there was some basic documentation on
>the LIVE.com library available for reference to the developer community. For
>a start, I'm working on some HTML docs (class diagrams etc.). Do let me know
>how I can make it available publicly (I don't own a public web space).
>Either I upload it to some site or send them to you via mail offlist (you
>may want to edit it before making them public).

Yes, if you like, please send this to me as an archive file (".tar" or 
".zip").  I'll take a look at it, and put it on the web if appropriate.

         Ross.


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X-Original-Date: Mon, 17 Jun 2002 21:58:02 -0700
Date: Mon, 17 Jun 2002 21:58:02 -0700

FYI, Deepesh Gujarathi ran "Doxygen" on the "LIVE.COM Streaming Media" 
library code, and sent me the result.  I have now placed this online as
	<http://www.live.com/liveMedia/doc/>

Thanks, Deepesh!

Questions to Deepesh: Did you have to make many changes to the existing 
source code to get this to work?  What Doxygen configuration file did you use?

	Ross.


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X-Original-Date: Tue, 18 Jun 2002 14:07:06 -0300 (EST)
Date: Tue, 18 Jun 2002 14:07:06 -0300 (EST)

Hi,

I'd like to know how much bytes are sent to the openRTSP. Which function 
can I use to take the size of the packets sent by the server?

Thank you very much
  Marisa Vasconcelos



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X-Original-Date: Tue, 18 Jun 2002 10:14:38 -0700
Date: Tue, 18 Jun 2002 10:14:38 -0700

At 10:07 AM 6/18/02, you wrote:
>I'd like to know how much bytes are sent to the openRTSP.

This depends on the RTSP server, and on the codec(s) that are being streamed.

>Which function
>can I use to take the size of the packets sent by the server?

Look in the function "MultiFramedRTPSource::networkReadHandler()" (in 
"liveMedia/MultiFramedRTPSource.cpp").  After the call to
         source->RTPgs()->handleRead(...)
you can inspect the "packetSize" variable.  This will be the size of the 
received RTP packet (including the RTP header, but excluding UDP and IP 
headers).

         Ross.


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X-Original-Date: Wed, 19 Jun 2002 13:45:16 -0500
Date: Wed, 19 Jun 2002 13:45:16 -0500

Hi Ross,
I was going through the library developed by Live and in the RTCP 
implementation I found two files of concern to us.
1. RTCP.cpp along with the header
2. rtcp_from_spec.c along with the header.
Now, I would like to know the folowing things:
1. The relevance of both the files for a developer of RTCP implementation.
2. Is rtcp_from_spec.c implements the code given in the ietf draft of 1889?
3. Please give me a broad picture on the interfacing / interactions 
between the functions / classes in the two files.

Waiting anxiously for your reply,
Prashanth



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Subject: Re: [Live-devel] ************** RTCP *************
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X-Original-Date: Wed, 19 Jun 2002 01:51:39 -0700
Date: Wed, 19 Jun 2002 01:51:39 -0700

At 11:45 AM 6/19/02, B Prashanth wrote:
>Hi Ross,
>I was going through the library developed by Live and in the RTCP 
>implementation I found two files of concern to us.
>1. RTCP.cpp along with the header
>2. rtcp_from_spec.c along with the header.
>Now, I would like to know the folowing things:
>1. The relevance of both the files for a developer of RTCP implementation.

These files, together, implement RTCP.  (Note, by the way, that RTCP is 
*not* optional functionality; implementations of RTP are *required* to also 
implement RTCP.)

These files should both build properly for your environment.  (If they 
don't, please let me know.)

>2. Is rtcp_from_spec.c implements the code given in the ietf draft of 1889?

Close.  It is actually the (exact!) code that's given in Appendix A.7 of 
the most recent draft of the RTP/RTCP specification 
<http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-new-11.txt>.  (Note 
that this will eventually replace RFC 1889.)

This code implements RTCP's retransmission (etc.) algorithm.  By using the 
exact code that's given in the RTP/RTCP specification, this becomes easier 
to implement correctly.  Note, though, that several routines used by this 
code are left unimplemented.  (See below.)

>3. Please give me a broad picture on the interfacing / interactions 
>between the functions / classes in the two files.

Note the various "extern "C"" routines defined in RTCP.cpp.  These 
functions implement the various unimplemented functions in "rtcp_from_spec.c".

Note also, in "RTCP.cpp", the calls from "RTCPInstance" member functions to 
"OnExpire()" and "OnReceive()".  This is how the code in "rtcp_from_spec.c" 
gets called from the rest of the code.

I hope this helps.

         Ross.


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X-Original-Date: Wed, 19 Jun 2002 01:50:13 -0700
Date: Wed, 19 Jun 2002 01:50:13 -0700

At 11:45 AM 6/19/02, B Prashanth wrote:
>Hi Ross,
>I was going through the library developed by Live and in the RTCP 
>implementation I found two files of concern to us.
>1. RTCP.cpp along with the header
>2. rtcp_from_spec.c along with the header.
>Now, I would like to know the folowing things:
>1. The relevance of both the files for a developer of RTCP implementation.

These files, together, implement RTCP.  (Note, by the way, that RTCP is 
*not* optional functionality; implementations of RTP are *required* to also 
implement RTCP.)

These files should both build properly for your environment.  (If they 
don't, please let me know.)

>2. Is rtcp_from_spec.c implements the code given in the ietf draft of 1889?

Close.  It is actually the (exact!) code that's given in Appendix A.7 of 
the most recent draft of the RTP/RTCP specification 
<http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-new-11.txt>.  (Note 
that this will eventually replace RFC 1889.)

This code implements RTCP's retransmission (etc.) algorithm.  By using the 
exact code that's given in the RTP/RTCP specification, this becomes easier 
to implement correctly.  Note, though, that several routines used by this 
code are left unimplemented.  (See below.)

>3. Please give me a broad picture on the interfacing / interactions 
>between the functions / classes in the two files.

Note the various "extern "C"" routines defined in RTCP.cpp.  These 
functions implement the various unimplemented functions in "rtcp_from_spec.c".

Note also, in "RTCP.cpp", the calls from "RTCPInstance" member functions to 
"OnExpire()" and "OnReceive()".  This is how the code in "rtcp_from_spec.c" 
gets called from the rest of the code.

I hope this helps.

         Ross.


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X-Original-Date: Wed, 19 Jun 2002 18:31:43 -0500
Date: Wed, 19 Jun 2002 18:31:43 -0500

Dear All,
1. Is the current implementation of OpenRTSP adhereing to ISMA standards 
in RTP/RTCP/RTSP ?
2. If I have to do an implementation of only RTP then what are the 
relevant files for me i.e. I want to know what are the core files of RTP 
implementation independent of the payload types or payload formats.
Please do mail back and thanks in advance.
Prashanth
Sr. Design Engr.


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From <EMAIL-ADDRESS-HIDDEN> Wed Jun 19 14:32:03 2002
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X-Original-Date: Wed, 19 Jun 2002 12:47:55 -0700
Date: Wed, 19 Jun 2002 12:47:55 -0700

At 04:31 PM 6/19/02, B Prashanth wrote:
>Dear All,
>1. Is the current implementation of OpenRTSP adhereing to ISMA standards 
>in RTP/RTCP/RTSP ?

"openRTSP" (and the "LIVE.COM Streaming Media" libraries that it's built 
from) adhere to *IETF* standards.  (Note that "ISMA" isn't really a 
standards organization.  Instead, they're an industry consortium that 
rubber-stamps a group of selected IETF standards, and promotes 
interoperability.)

The LIVE.COM software supports many of the IETF standards that have been 
adopted  by ISMA.  The most notable exception is the RTP payload format for 
MPEG-4 Elementary Streams.  We do not currently support this (but will in 
the not too distant future).  However, we do currently support the RTP 
payload formats for MPEG-1,2 audio and video, H.263+ video, GSM and QCELP 
audio, and several others.

>2. If I have to do an implementation of only RTP then what are the 
>relevant files for me i.e. I want to know what are the core files of RTP 
>implementation independent of the payload types or payload formats.

For sending RTP, look at "RTPSink" and "MultiFramedRTPSink".  For receiving 
RTP, look at "RTPSource" and "MultiFramedRTPSource".

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>


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Subject: [Live-devel] cannot use testprogs with media player
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X-Original-Date: 20 Jun 2002 08:44:20 -0000
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Hi,

I have recently downloaded the RTP library and using it with VC++ on windows 2000 adv. server.

When I run the testMPEGAudioVideoStreamer program and provide it with a .mpg file that has both audio and video content, then I receive the output using the testMP3Receiver and testMPEGVdeoReceiver in two separate files (.mpg and .mp3). Why does the testMPEGAudioVideoStreamer program breaks (separates) the stream into audio and video elementary streams. What I intend to do is to send both audio and video together and receive them together without separating. How do I do this.

I also want to use the windows media player to receive output from the  testMPEGVideoStreamer program. I have uncommented the top three #defines for this program. How do I get to see the output with media player.

Basically I want to develop an application that can stream/broadcast (multicast) live-real time audio and video. For now I wish to stream data from a file and receive it using media player, later on I would like to capture live data from a web-cam.

Please help me….

Thanks in advance

Manoj

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Subject: [Live-devel] Play Mp3 Stream
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X-Original-Date: Thu, 20 Jun 2002 09:54:33 -0400
Date: Thu, 20 Jun 2002 09:54:33 -0400

I am running test MP3 audio test programs on windows 2000. It would be great
if someone could help find out, what changes I would have to make to
testMP3Receiver, so that I could play the Mp3 Stream.

Thanks, 
Mahua 
 



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From: Marisa Affonso Vasconcelos <EMAIL-ADDRESS-HIDDEN>
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X-Original-Date: Thu, 20 Jun 2002 14:29:57 -0300 (EST)
Date: Thu, 20 Jun 2002 14:29:57 -0300 (EST)

Hi,

I'd like to know the sequence number of the received packets RTP in the
openRTSP. Does anyone know which part of the code, there is this
information? 

Thanks in advance 
  Marisa



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X-Original-Date: Thu, 20 Jun 2002 12:12:21 -0700
Date: Thu, 20 Jun 2002 12:12:21 -0700

At 10:29 AM 6/20/02, you wrote:
>I'd like to know the sequence number of the received packets RTP in the
>openRTSP. Does anyone know which part of the code, there is this
>information?

Marisa,

The member function "RTPSource::curPacketRTPSeqNum()" will return the 
sequence number of the most recently received RTP packet (for the given RTP 
source).

         Ross.



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Subject: Re: [Live-devel] Play Mp3 Stream
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X-Original-Date: Thu, 20 Jun 2002 12:50:09 -0700
Date: Thu, 20 Jun 2002 12:50:09 -0700

At 06:54 AM 6/20/02, you wrote:
>I am running test MP3 audio test programs on windows 2000. It would be great
>if someone could help find out, what changes I would have to make to
>testMP3Receiver, so that I could play the Mp3 Stream.

Mahua,

"testMP3Receiver" should already be able to receive a MP3/RTP stream that 
is streamed by "testMP3Streamer".  Note, though, that "testMP3Receiver" 
outputs the received MP3 data to 'stdout', so you can run, for example
         testMP3Receiver > test.mp3
and this will produce a MP3 file that you can play.

Alternatively, if you have a MP3 player tool that reads from 'stdin', you run
         testMP3Receiver | your-mp3-player-tool

(For some strange reason, though, few MP3 players support reading from stdin.)

Alternatively, you can run Winamp <http://www.winamp.com/>, with the 
"in_rtp.dll" plugin <http://www.live.com/multikit/winamp-plugin.html> to 
play MP3/RTP streams.

The "RealOne Player" will also work.  (To use it, have it read the file 
"testMP3.sdp".)

         Ross.



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X-Original-Date: Thu, 20 Jun 2002 12:42:30 -0700
Date: Thu, 20 Jun 2002 12:42:30 -0700


>When I run the testMPEGAudioVideoStreamer program and provide it with a 
>.mpg file that has both audio and video content, then I receive the output 
>using the testMP3Receiver and testMPEGVdeoReceiver in two separate files 
>(.mpg and .mp3). Why does the testMPEGAudioVideoStreamer program breaks 
>(separates) the stream into audio and video elementary streams.

Because that's the preferred way that RTP works.  By sending audio and 
video as separate RTP streams, you have the flexibility to
         - have different processes (or even different computers) handling 
incoming audio and video
         - have different processes (or even different computers) streaming 
the outgoing audio and video

>  What I intend to do is to send both audio and video together and receive 
> them together without separating. How do I do this.

It *is* possible to send MPEG Program Stream data in a single RTP session 
(see RFC 2250, section 2), and the LIVE.COM Streaming Media  Library 
supports this (using "SimpleRTPSink", with media type name "video" and MIME 
type "MP2P"), but I don't recommend doing this, because probably few 
existing media players will recognize and play this RTP payload type.

Instead, just continue to send the audio and video streams separately, and 
rely upon the receiving tool (e.g., QuickTime Player or RealOne Player) to 
receive and play them.

>I also want to use the windows media player to receive output from 
>the  testMPEGVideoStreamer program.

Sorry - to my knowledge Windows Media Player can't be used to receive these 
streams; they support Microsoft's proprietary streaming protocols, rather 
than the industry standard RTP/RTCP and RTSP protocols.  Instead, you can 
use QuickTime Player or RealOne Player.  To receive/play the stream from 
"testMPEGAudioVideoStreamer", have the player open the file 
"testMPEGAudioVideo.sdp".  (This file is included in the "testProgs" 
directory.)

>  I have uncommented the top three #defines for this program.

There are only two commented-out #defines in this program.  The only one of 
these that you should consider uncommenting is
         #define IMPLEMENT_RTSP_SERVER 1
if you want to access the stream using RTSP.  You should leave the other 
#define commented out (unless you understand what the term "SSM" means :-)

Note, though, that due to a bug in RealOne Player, it can receive/play the 
stream only by reading the ".sdp" file.  It will not be able to play the 
stream using a "rtsp://" URL.  (That will work for QuickTime Player, though.)

>  How do I get to see the output with media player.

As noted above, Windows Media Player won't work.  I recommend QuickTime 
Player instead.

         Ross.



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From <EMAIL-ADDRESS-HIDDEN> Fri Jun 21 02:29:44 2002
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Subject: [Live-devel] RE: rtp sequence number
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X-Original-Date: Fri, 21 Jun 2002 14:59:28 +0530
Date: Fri, 21 Jun 2002 14:59:28 +0530

Hi Marisa,

the RTP sequence number is a public member of *BufferedPacket
(QCELPBufferedPacket,)class
You will find in the *RTPSource.cpp files.
For e.g. in the MultiFramedRTPSource.hh file it is defined as

unsigned short fRTPSeqNo;


PS: please do include an appropriate 'subject' while posting mails to the
list

hope this helps
Deepesh

--To err is human
to screw things up requires the 'root' password

Deepesh S Gujarathi
Encodex Technologies (India) Pvt. Ltd.
28, Electronic estate,Pune- 411009
Tel- 91 20 4222 401/402
www.encodexindia.com




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X-Original-Date: Sat, 22 Jun 2002 04:29:26 -0700
Date: Sat, 22 Jun 2002 04:29:26 -0700


>As you had directed in the last mail, I have tried the following options:
>
>1) started the testMP3Streamer, feeded it with test.mp3 file, then opened 
>the testMP3.sdp file with QuickTime 4.0 Player. Result-> player status 
>shows live broadcasting but i cant hear any sound.

Unfortunately QuickTime Player currently does not support MP3/RTP 
streams.  (However, MPEG layer I or layer II audio RTP streams will work.)

To play MP3/RTP streams, use try using Winamp with the "in_rtp.dll" plugin 
<http://www.live.com/multikit/winamp-plugin.html>, or the "RealOne" player.


>2) started the testMPEGVideoStreamer, feeded it with test.mpg file, then 
>opened the testMPEGVideo.sdp with the same player. Result-> player status 
>shows live broadcasting but i cant see any picture.
>
>3) did the same with testMPEGAudioVideoStreamer, same Result...

Try using QuickTime version *5* (or 6) Player, rather than version 4.  If 
you do this, then exactly one of 2) or 3) should work.  (If "test.mpg" is a 
video Elementary Stream file, then 2) should work; if "test.mpg" is an 
audio/video Program Stream file, then 3) should work.


>4)uncommented the line #define IMPLEMENT_RTSP_SERVER 1, started 
>testMPEGAudioVideo, feeded it with test.mpg, then specified the following 
>url with the player.
>rtsp://239.255.42.42:6666 and rtsp://239.255.42.42:8888
>The status for the says connecting... and then 10049: maybe this is some 
>sort of an error number.

The RTSP server implemented in "testMPEGAudioVideoStreamer" uses the 
default RTSP port: 554.  Also, in your RTSP URL, you should specify the 
address of the *server*.  So, instead, you should try:
         rtsp://<server-address>
or, equivalently:
         rtsp://<server-address>:554

I hope this helps.

         Ross.



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X-Original-Date: Mon, 24 Jun 2002 14:12:45 -0300 (EST)
Date: Mon, 24 Jun 2002 14:12:45 -0300 (EST)


Hi,

Thanks for the help. I've printed the sequence number using the two ways (
fRTPSeqNo and curPacketRTPSeqNum()). So, I've observed that there were two
number sequence of packets. Are these sequence, one for audio and one for
video? How can I distinguish these two types?    

Thanks in advance
 Marisa

> Hi Marisa,
> 
> the RTP sequence number is a public member of *BufferedPacket
> (QCELPBufferedPacket,)class
> You will find in the *RTPSource.cpp files.
> For e.g. in the MultiFramedRTPSource.hh file it is defined as
> 
> unsigned short fRTPSeqNo;
> 
> 
> PS: please do include an appropriate 'subject' while posting mails to the
> list
> 
> hope this helps
> Deepesh
> 
> --To err is human
> to screw things up requires the 'root' password
> 
> Deepesh S Gujarathi
> Encodex Technologies (India) Pvt. Ltd.
> 28, Electronic estate,Pune- 411009
> Tel- 91 20 4222 401/402
> www.encodexindia.com
> 
> 
> 
> 
> 
> --__--__--
> 
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> 
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X-Original-Date: Mon, 24 Jun 2002 12:26:29 -0700
Date: Mon, 24 Jun 2002 12:26:29 -0700

At 10:12 AM 6/24/02, Marisa Affonso Vasconcelos wrote:
>Thanks for the help. I've printed the sequence number using the two ways (
>fRTPSeqNo and curPacketRTPSeqNum()). So, I've observed that there were two
>number sequence of packets. Are these sequence, one for audio and one for
>video?

Yes.  Each RTP stream uses its own sequence number space.

>How can I distinguish these two types?

Different "RTPSource" objects are used to receive each RTP stream.  So, you 
can distinguish them by printing out the address of the "RTPSource" object 
(or "this" if you're doing this from a member function), as well as the 
sequence number.

         Ross.



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X-Original-Date: Mon, 24 Jun 2002 12:39:16 -0700
Date: Mon, 24 Jun 2002 12:39:16 -0700

(In the future, could you please use the developers' mailing list 
<live-devel(at)lists.sourceforge.net>.  For one-on-one email, I generally give 
priority to paying consulting clients.)

At 04:11 AM 6/24/02, manoj someone singh wrote:
>When I play the testMP3.sdp file with the Winamp player I can hear the 
>audio. The play list shows the entry for the stream as: 
>http://localhost:10006. What does that mean.

The "in_rtp.dll" plugin is implemented by creating an internal HTTP server 
(on port 10006), and telling the rest of Winamp to stream from that HTTP 
server (i.e., using the URL http://localhost:10006").  The reason it does 
this is that this seems to be the only way to get the rest of Winamp to 
read MP3 data.  Unfortunately, like many Windows weenies, the authors of 
Winamp have apparently never heard of 'stdio'; it would have been much 
simpler if the plugin could just pipe MP3 data to Winamp proper (which 
would just read from 'stdin').


>But when I specify the url to the player like, rtp://239.255.42.42:8888 or 
>rtcp://239.255.42.42:8889 or rtp//239.255.42.42:554, the player gives an 
>illegal operation and closes. What should I do.

First, note that the "rtcp://..." URL is bogus - that will never 
work.  (Note that you get RTCP automatically, so you never need to specify 
it directly.)

The following URLs *should* work:
         rtp://239.255.42.42:8888
and, *if* the built-in RTSP server has been turned on in "testMP3Streamer"
         rtsp://<name-of-the-computer-running-testMP3Streamer>:554/

But note that (due to a bug in Winamp proper; not the plugin) Winamp 
sometimes has problems playing more than one RTP or RTSP session in a 
row.  So, I suggest quitting and restarting Winamp from scratch each time 
you attempt to play one of these streams.

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>



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X-Original-Date: 25 Jun 2002 07:27:33 -0000
Date: 25 Jun 2002 07:27:33 -0000

Dear Sir,

As directed in the last mail, I could play the audio with 
testMP3Streamer using rtsp://<computer name>:554, but not using 
rtp://239.255.42.42:8888.

When I try to use the testMPEGVideoStreamer with RealOne Player 
and play the testMPEGVideo.sdp file, only the first frame (in two 
sections) of the movie is shown. Why cant I see the entire 
movie.

When I try to use the testMP3Streamer with RealOne Player and play 
the testMP3.sdp, it does not play the audio. Same happens with 
testMPEGAudioVideoStreamer. It seems that the RealOne Player has 
some problem playing the file if it contains audio.

When I try using the URL in the form rtp://239.255.42.42:8888 or 
rtp://239.255.42.42:6666 the RealOne player says it needs some 
add-ins and jumps to the RealOne site, but later says it cannot 
find any information about rtp://.

When I try using the URL in the form rtsp://<computer name>:554, 
it says connecting.... but will not connect for a long time.

Thanks in advance...

Regards.

Manoj.



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Subject: [Live-devel] Capture & Stream Live Audio/Video
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X-Original-Date: 25 Jun 2002 09:23:37 -0000
Date: 25 Jun 2002 09:23:37 -0000

Dear Sir,

The test programs available with the RTP Library, accept data for 
.mp3 or .mpg file. I want to capure live feed form a web-cam and 
give it to the streamer. I can use the DirectShow API to capture 
video from the web-cam and display it on the rendering surface 
maybe the screen or a file. But how do I combine it with the test 
programs.

The DirectX API is COM based, while the test programs are written 
in plain C++. How do I go about using this feature. What I need is 
to develop an application using Microsoft VC++ and the RTP 
Library, which can Stream (Broadcast/multicast) live audio/video 
over the internet.

Please provide me with the necessary guidelines.

Thanks in advance.

Regards.

Manoj.
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X-Original-Date: Tue, 25 Jun 2002 03:11:19 -0700
Date: Tue, 25 Jun 2002 03:11:19 -0700

At 12:27 AM 6/25/02, manoj someone singh wrote:
>As directed in the last mail, I could play the audio with testMP3Streamer 
>using rtsp://<computer name>:554, but not using rtp://239.255.42.42:8888.

That's strange.  This should be working.  What version of Winamp, and what 
version of Windows are you using?


>When I try to use the testMPEGVideoStreamer with RealOne Player and play 
>the testMPEGVideo.sdp file, only the first frame (in two sections) of the 
>movie is shown. Why cant I see the entire movie.

I'm not sure, but are you sure that your file test.mpg is a Video 
Elementary Stream file - *not* a Program Stream file?  (On the other hand, 
"testMPEGAudioVideoStreamer" requires that test.mpg be a Program Stream file.)


>When I try to use the testMP3Streamer with RealOne Player and play the 
>testMP3.sdp, it does not play the audio.

Again, that's strange.  RealOne player is supposed to be able to play 
MP3/RTP streams (from a .sdp file).


>Same happens with testMPEGAudioVideoStreamer.

I suggest that you try using QuickTime Player instead.


>When I try using the URL in the form rtp://239.255.42.42:8888 or 
>rtp://239.255.42.42:6666 the RealOne player says it needs some add-ins and 
>jumps to the RealOne site, but later says it cannot find any information 
>about rtp://.

"rtp://" URLs are a hack, and neither RealOne Player nor QuickTime Player 
supports them.  They work only for MP3/RTP streams, using Winamp or Freeamp.


>When I try using the URL in the form rtsp://<computer name>:554, it says 
>connecting.... but will not connect for a long time.

RealOne Player has a bug that prevents it from playing multicast MPEG/RTP 
streams from a "rtsp://" URL.

Bottom line: At this time I don't recommend the use of RealOne Player for 
playing RTP streams - it's just too flaky.  Instead, I recommend QuickTime 
Player (version 5 or later).  This can play MPEG/RTP streams - either using 
a "rtsp://" URL, or from a ".sdp" file.  Note, however, that QuickTime 
Player cannot currently play *MP3*/RTP streams, only MPEG audio-video/RTP 
streams (which use layer I or layer II audio, rather than layer III audio 
(MP3)).

Summary: To play RTP streams on Windows, I recommend:
- For MP3 streams, use Winamp, with the "in_rtp.dll" plugin
- For MPEG streams (other than MP3 audio), use QuickTime Player

         Ross.



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X-Original-Date: Tue, 25 Jun 2002 12:41:56 -0700
Date: Tue, 25 Jun 2002 12:41:56 -0700

At 02:23 AM 6/25/02, manoj someone singh wrote:
>The test programs available with the RTP Library, accept data for .mp3 or 
>.mpg file. I want to capure live feed form a web-cam and give it to the 
>streamer. I can use the DirectShow API to capture video from the web-cam 
>and display it on the rendering surface maybe the screen or a file. But 
>how do I combine it with the test programs.

First, note that the "test programs" are just that - programs that 
illustrate how to make use of the libraries.  With the exception of 
"openRTSP", they are not intended to be complete, end-user applications.

To use the libraries to stream live video from a web cam, you would need to 
write the following software components:
1/ an object (subclass of "FramedSource") that encapsulates the video frame 
grabber.  (This will have to be operating system-specific code.)
2/ an object (subclass of "FramedFilter") that encodes the video frame into 
MPEG (possibly by encapsulating an existing, binary-only encoder).

You would then assemble these components into a chain, and play them using 
a "MPEGVideoRTPSink".

If you already have MPEG encoder hardware, then instead of writing new 
software components 1/ and 2/, you can just take the output of the encoder, 
pass it through a "MPEGVideoStreamFramer", and then stream it using a 
"MPEGVideoRTPSink".  In fact, I have one client who has done this by making 
a simple modification to "testMPEGVideoStreamer.cpp": changing "test.mpg" 
to "stdin".  He then just pipes the output of his MPEG video encoder into 
the (modified) "testMPEGVideoStreamer" application.

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>




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X-Original-Date: Tue, 25 Jun 2002 13:01:08 -0700 (PDT)
Date: Tue, 25 Jun 2002 13:01:08 -0700 (PDT)

hi Ross,
 i have written an application using livemedia and
compiled for ipaq which has familiar linux. the
application runs fine on PC with normal linux
distribution but on the ipaq it doesnt stream. i tried
using both real server and also Darwin streaming
server.
 do i have any changes to be made??

ur help would be great.
thank you
surya

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X-Original-Date: Tue, 25 Jun 2002 13:08:15 -0700 (PDT)
Date: Tue, 25 Jun 2002 13:08:15 -0700 (PDT)

hi,
 i have a basic question,
 if i want to use multicast instead of unicast with
openrtsp, is it sufficient if i just give a multicast
address as the IP address?

what changes should i make to listen for multicast??

thank you'
surya

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X-Original-Date: Tue, 25 Jun 2002 13:13:38 -0700
Date: Tue, 25 Jun 2002 13:13:38 -0700

At 01:01 PM 6/25/02, suryakumari sagiraju wrote:
>  i have written an application using livemedia and
>compiled for ipaq which has familiar linux. the
>application runs fine on PC with normal linux
>distribution but on the ipaq it doesnt stream.

 From your (very) limited description, it's not possible to tell what might 
be going wrong.

Remember, though, that you have *complete source code* to the 
libraries.  So, please try debugging your code (e.g., using printf()s) to 
figure out why your Linux/x86 version is working, but your Linux/iPaq 
version is not.

         Ross.



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X-Original-Date: Tue, 25 Jun 2002 13:18:40 -0700
Date: Tue, 25 Jun 2002 13:18:40 -0700

At 01:08 PM 6/25/02, you wrote:
>  i have a basic question,
>  if i want to use multicast instead of unicast with
>openrtsp, is it sufficient if i just give a multicast
>address as the IP address?

No.  The RTSP protocol uses TCP, so the IP address (or machine name) given 
in the "rtsp//" URL must always be unicast.

However, if you want the actual *RTP streaming* to be done using multicast, 
then that's for the RTSP server to decide.  I.e., if the server returns - 
in response to the RTSP "DESCRIBE" request - a SDP description that 
contains a multicast address, then the RTP streaming will be done using 
multicast.

>what changes should i make to listen for multicast??

The changes will need to be made at the server end.  (This depends on which 
brand of RTSP server you are using.)

         Ross.



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Subject: Re: [Live-devel] Capture & Stream Live Audio/Video
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X-Original-Date: Wed, 26 Jun 2002 01:33:10 -0700
Date: Wed, 26 Jun 2002 01:33:10 -0700

A quick follow-up to my earlier message.

If you wish to write a source object that encapsulates a video (or audio) 
grabber device, then you should take a look at the files
	liveMedia/include/DeviceSource.hh
	liveMedia/DeviceSource.cpp

These provide a template for you could implement such an object.  You would 
fill in the various member function implementations to complete this.

	Ross.



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Subject: [Live-devel] Using Capture device with test programs
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X-Original-Date: 26 Jun 2002 11:13:49 -0000
Date: 26 Jun 2002 11:13:49 -0000

Dear Sir,

This is a follow up to the latest message for you regarding how to 
use a capture device with the test programs. You have mentioned 
the devicesouce.hh and devicesource.cpp file in the 
livemedia\(include)directory, but i cant find these files there.

Thanks in advance

Regards.

Manoj.
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Subject: [Live-devel] MultiFramedRTPSink class + MTU size
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X-Original-Date: Wed, 26 Jun 2002 18:14:41 -0500
Date: Wed, 26 Jun 2002 18:14:41 -0500

Hi Ross,

I am going through your library and I wanted to clarify a few issues...

1. Firstly, do all the profiles (Quicktime, H.263, MPEG 2 A/V etc) that 
have been implemented in this library use MultiFramedRTPSink class for 
sending data? If yes, then does RFC recommends this? Please tell me 
which of the implementations would not use MultiFramedRTPSink class.

2. How have you decided on the size of the MTU packet that is finally to 
be sent? ( I think you taken it to be 1450 bytes). I am asking this 
because the size of the MTU changes for wireless to LAN networks.

Thank you in advance for your patience,
Prashanth




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X-Original-Date: Wed, 26 Jun 2002 18:32:43 -0500
Date: Wed, 26 Jun 2002 18:32:43 -0500

Sorry Ross i forgot to put thisin my earlier mail....
What does a statement like this mean....?
MPEGVideoRTPSink::MPEGVideoRTPSink(UsageEnvironment& env, Groupsock* 
RTPgs) 
: MultiFramedRTPSink(env, RTPgs, 32, 90000)

Thank you for your patience...
Prashanth



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X-Original-Date: Wed, 26 Jun 2002 08:52:26 -0700
Date: Wed, 26 Jun 2002 08:52:26 -0700

At 04:13 AM 6/26/02, you wrote:
>This is a follow up to the latest message for you regarding how to use a 
>capture device with the test programs. You have mentioned the 
>devicesouce.hh and devicesource.cpp file in the 
>livemedia\(include)directory, but i cant find these files there.

Sorry - they are present only in the newest version of the library code - 
released yesterday.

         Ross.



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X-Original-Date: Wed, 26 Jun 2002 12:42:54 -0700
Date: Wed, 26 Jun 2002 12:42:54 -0700


>What does a statement like this mean....?
>MPEGVideoRTPSink::MPEGVideoRTPSink(UsageEnvironment& env, Groupsock* 
>RTPgs) : MultiFramedRTPSink(env, RTPgs, 32, 90000)

This is just regular C++.  In a subclass constructor, you need to first 
call the constructor of the parent class (unless the parent class has a 
default constructor).

         Ross.



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Subject: Re: [Live-devel] MultiFramedRTPSink class + MTU size
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X-Original-Date: Wed, 26 Jun 2002 12:58:58 -0700
Date: Wed, 26 Jun 2002 12:58:58 -0700


>1. Firstly, do all the profiles (Quicktime, H.263, MPEG 2 A/V etc) that 
>have been implemented in this library use MultiFramedRTPSink class for 
>sending data? If yes, then does RFC recommends this? Please tell me which 
>of the implementations would not use MultiFramedRTPSink class.

I originally intended there to be a distinction between "RTPSink" and its 
subclass "MultiFramedRTPSink", but, as the code evolved, the distinction 
became less important.  As you noted, it turns out that every RTP payload 
format implementation uses "MultiFramedRTPSink" (rather than "RTPSink"), so 
at some time in the future, I'll probably merge the two classes into one.

Note that the virtual function "frameCanAppearAfterPacketStart()" defines 
whether or not a payload format will allow more than one frame to be packed 
into a single outgoing RTP packet.  The default implementation of this 
function just returns "True", meaning that, by default, multiple frames can 
be packed into an outgoing RTP packet (until it reaches its preferred size 
- see below).  However, some payload format implementations redefine 
this.  E.g., the payload format for H.263+ video allows only one frame per 
RTP packet, and the payload format for MPEG video allows more than one 
'frame' per RTP packet in certain circumstances only.


>2. How have you decided on the size of the MTU packet that is finally to 
>be sent? ( I think you taken it to be 1450 bytes). I am asking this 
>because the size of the MTU changes for wireless to LAN networks.

I chose 1450 bytes as the maximum RTP packet size, because most networks 
have a MTU >= 1500 bytes.  (The extra 50 bytes allow for the IP and UDP 
headers.)  If you know *for sure* that you will be streaming *only* over a 
network with a larger MTU, then you can increase "maxPacketSize" (and 
"preferredPacketSize") in "MultiFramedRTPSink.cpp".  I don't recommend 
this, though...

         Ross.




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From <EMAIL-ADDRESS-HIDDEN> Wed Jun 26 23:37:21 2002
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X-Original-Date: 27 Jun 2002 06:25:15 -0000
Date: 27 Jun 2002 06:25:15 -0000

Dear Sir,

In the last mail, you have mentioned that the 
DeviceSource(.hh,.cpp) files are available in the latest release 
of the RTP Library. Is it available for download, and from where 
can I get it.

Thanks.

Manoj.
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X-Original-Date: Thu, 27 Jun 2002 00:10:06 -0700
Date: Thu, 27 Jun 2002 00:10:06 -0700

At 11:25 PM 6/26/02, you wrote:
>In the last mail, you have mentioned that the DeviceSource(.hh,.cpp) files 
>are available in the latest release of the RTP Library. Is it available 
>for download, and from where can I get it.

You can find these files in the "liveMedia/" (and "liveMedia/include/") 
directory in the latest source code release (2002.06.25).

         Ross.



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X-Original-Date: Thu, 27 Jun 2002 14:23:30 -0700 (PDT)
Date: Thu, 27 Jun 2002 14:23:30 -0700 (PDT)

Hi,
   I want to develop a streaming application server
which streams video to IPAQ devices thro' 802.11
network. Since the bandwidth is limited, i want to use
H.263 for compression. Has anyone implemented the
framer program "H263VideoStreamframer" for this
streaming application? If so, can you please help me?
               Thank you in advance,
Arvind 


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X-Original-Date: Thu, 27 Jun 2002 14:50:22 -0700
Date: Thu, 27 Jun 2002 14:50:22 -0700

At 02:23 PM 6/27/02, Arvind Karunanidhi wrote:
>   I want to develop a streaming application server
>which streams video to IPAQ devices thro' 802.11
>network. Since the bandwidth is limited, i want to use
>H.263 for compression. Has anyone implemented the
>framer program "H263VideoStreamframer" for this
>streaming application? If so, can you please help me?

Arvind,

Streaming H.263 (actually, H.263+, aka. H.263-1998) video using the 
libraries is straightforward - using the "H263plusVideoRTPSink" object.  Of 
course, to stream live video, you will also need to write a 'source' object 
that encapsulates your video frame grabber, and a 'filter' object that 
performs the actual H.263+ encoding.

On the other hand, if you are planning to stream already-encoded H.263+ 
data (e.g., from a file), then you almost have enough already to be able to 
build a H.263+ streaming application, similar to 
"testMPEGVideoStreamer".  The one thing that's still missing is - as you 
note - a "H263plusVideoStreamFramer" class (note, a C++ class, not a 
program), that implements the "doGetNextFrame()" virtual function by 
returning a frame at a time.

A "H263plusVideoStreamFramer" class would be fairly simple to write 
(probably a lot simpler than "MPEGVideoStreamFramer".  (However, if you 
would prefer to hire LIVE.COM to write this for you, please let me know by 
email, off the list.)

         Ross.



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X-Original-Date: Thu, 27 Jun 2002 21:17:10 -0700
Date: Thu, 27 Jun 2002 21:17:10 -0700

At 04:07 PM 6/27/02, S. Kannan wrote:
>You remember me, I was struggling with an RTSP server which is not
>compatible with others.  Now I have a plan to use your RTSP Server. I built
>the source codes and have OpenRTSP.exe, testMPEGAudioVideoStreamer.exe,
>testMPEGSplitter.exe and a test.mpeg file too.  How can I play around these
>applications to stream this mpeg file and see the video stream using the
>openRTSP client.

If you want to "see" the video stream, you don't want to use 
"openRTSP".  "openRTSP" is just a command-line RTSP client that saves the 
incoming stream(s) to files.

To see the video stream, I suggest using QuickTime Player (version 5.0 or 
later).  If you have enabled the RTSP server in 
"testMPEGAudioVideoStreamer", you can then view the video in QuickTime 
Player by opening the URL
         rtsp://<server>/
(Replace "<server>" with the name (or IP address) of the computer on which 
you're running "testMPEGAudioVideoStreamer".)

>   Do I need any SDP file?

No, if you've enabled the RTSP server, then you won't need the SDP 
file.  However, instead of opening a "rtsp://..." URL, you can have 
QuickTime Player open the file "testMPEGAudioVideo.sdp" instead.

         Ross.

ps. Your file "test.mpg" must be MPEG-1, not MPEG-2.  (QuickTime Player 
doesn't support MPEG-2.)

pps. In the future, could you please use the developers' mailing list 
<live-devel(at)lists.sourceforge.net>.  For one-on-one email, I generally give 
priority to paying consulting clients.




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Subject: [Live-devel] RTSP - live vs qt
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X-Original-Date: Fri, 28 Jun 2002 12:06:43 +0100
Date: Fri, 28 Jun 2002 12:06:43 +0100


	Greetings
	While experimenting with an RTSP server, I noticed a small but nevertheless
important detail: when the RTSP server replies with an SDP description like:

v=0
o=- 49451 3 IN IP4 10.0.0.2
s=Test MPEG Audio+Video session
i=Parameters for the session streamed by "testMPEGAudioVideoStreamer"
t=0 0
a=tool:testMPEGAudioVideoStreamer
a=type:broadcast
m=audio 0 RTP/AVP 14
a=control:trackID=0
c=IN IP4 0.0.0.0/127
m=video 0 RTP/AVP 32
a=control:trackID=1
c=IN IP4 0.0.0.0/127

	, openRTSP sends the SETUP message:

SETUP rtsp://my.rtsp.server/myfile.sdp/trackID=0 RTSP/1.0
[etc]

	while QTPlayer replies with:

SETUP rtsp://my.rtsp.server/trackID=0 RTSP/1.0
[etc]

	Basically, the name of the file is missing in the QuickTime Player SETUP
request.
	This might not sound like much, but it makes a huge difference when you're
writing the parser for the reply, and that parser is supposed to be
universal. What I ask is, why the difference? Which one of them is right? Or
is there something missing in the RTSP/SDP DESCRIBE response that accounts
for the different behavior?

Thanks in advance,
David Pontes



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From: <EMAIL-ADDRESS-HIDDEN> (S. Kannan)
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Subject: Re: [Live-devel] Re: How to play around!
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X-Original-Date: Fri, 28 Jun 2002 09:10:43 -0400
Date: Fri, 28 Jun 2002 09:10:43 -0400

I modified the testMPEGAudioVideoStreamer project to replace the hard-coded
IP Address with my machine's IP address and built it.  When I started this
testMPEGAudioVideoStreamer.exe file in the command line, I saw

beginning streaming.....
reading the file......
reading the file.....
reading the file... etc..

When I run the command line openRTSP it says failed to connect().

What do you mean by enabling RTSP server.

--Senthamarai.

----- Original Message -----
From: {Ross Finlayson} <EMAIL-ADDRESS-HIDDEN>
To: <EMAIL-ADDRESS-HIDDEN>
Sent: Friday, June 28, 2002 12:17 AM
Subject: [Live-devel] Re: How to play around!


> At 04:07 PM 6/27/02, S. Kannan wrote:
> >You remember me, I was struggling with an RTSP server which is not
> >compatible with others.  Now I have a plan to use your RTSP Server. I
built
> >the source codes and have OpenRTSP.exe, testMPEGAudioVideoStreamer.exe,
> >testMPEGSplitter.exe and a test.mpeg file too.  How can I play around
these
> >applications to stream this mpeg file and see the video stream using the
> >openRTSP client.
>
> If you want to "see" the video stream, you don't want to use
> "openRTSP".  "openRTSP" is just a command-line RTSP client that saves the
> incoming stream(s) to files.
>
> To see the video stream, I suggest using QuickTime Player (version 5.0 or
> later).  If you have enabled the RTSP server in
> "testMPEGAudioVideoStreamer", you can then view the video in QuickTime
> Player by opening the URL
>          rtsp://<server>/
> (Replace "<server>" with the name (or IP address) of the computer on which
> you're running "testMPEGAudioVideoStreamer".)
>
> >   Do I need any SDP file?
>
> No, if you've enabled the RTSP server, then you won't need the SDP
> file.  However, instead of opening a "rtsp://..." URL, you can have
> QuickTime Player open the file "testMPEGAudioVideo.sdp" instead.
>
>          Ross.
>
> ps. Your file "test.mpg" must be MPEG-1, not MPEG-2.  (QuickTime Player
> doesn't support MPEG-2.)
>
> pps. In the future, could you please use the developers' mailing list
> <live-devel(at)lists.sourceforge.net>.  For one-on-one email, I generally
give
> priority to paying consulting clients.
>
>
>
>
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X-Original-Date: Fri, 28 Jun 2002 09:23:29 -0700
Date: Fri, 28 Jun 2002 09:23:29 -0700


>What do you mean by enabling RTSP server.

Uncomment the line
         #define IMPLEMENT_RTSP_SERVER 1
in "testMPEGAudioVideoStreamer".

         Ross.



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X-Original-Date: Fri, 28 Jun 2002 09:39:45 -0700
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>         , openRTSP sends the SETUP message:
>
>SETUP rtsp://my.rtsp.server/myfile.sdp/trackID=0 RTSP/1.0
>[etc]
>
>         while QTPlayer replies with:
>
>SETUP rtsp://my.rtsp.server/trackID=0 RTSP/1.0
>[etc]
>
>         Basically, the name of the file is missing in the QuickTime 
> Player SETUP
>request.
>         This might not sound like much, but it makes a huge difference 
> when you're
>writing the parser for the reply, and that parser is supposed to be
>universal. What I ask is, why the difference? Which one of them is right?

"openRTSP" is doing the right thing by using the entire URL in the "SETUP" 
request.  If QT Player is leaving out the file name, then that's probably a 
bug.  (It may happen to work OK with Apple's QuickTime (aka. "Darwin") 
Streaming Server, but may not with other RTSP servers.)

         Ross.



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Subject: [Live-devel] RTCP Question
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X-Original-Date: Mon, 1 Jul 2002 13:32:20 +0400
Date: Mon, 1 Jul 2002 13:32:20 +0400

I created 2 files similar to testMpegVideoSender and testMpegVideoReceiver,
but my sender program uses a SimpleRTPSink and ByteStreamFileSource and
receiver program uses a FileSink and a SimpleRTPSource and these both
communicate in a unicast session. Both programs work fine and I am able to
receive the streamed data (text file) which the receiver saves in a file.
However, I would like to view RTCP receiver and sender reports. I recompiled
the file RTCP.cpp on my client and server such that it contains :
#define DEBUG_PRINT 1, to be informed when RRs or SRs are received/sent, but
on my server, I do not get any output other than one line when it is started
which is:
schedule(2.279881->1025513488.646882)

and in my client I only get the following lines printed at regular
intervals:
sending REPORT
sending RTCP packet
81c90007 225.....(cont)
schedule...(cont)

I was expecting to see printouts of the lines between #ifdef DEBUG_PRINT and
#endif such as the ones that print RR and SR in the method:
incomingReportHandler(RTCPInstance *instance, int mask) in RTCP.cpp

and when I terminate the server program the client program prints at regular
intervals: "saw incoming RTCP packet" which I find strange since the server
program has terminated.

I intend to use this library to stream compressed audio and video data
(from&to directshow), but I also need to see the full details of the RRs and
SRs for info and to determine if the server (or client) program has
terminated unexpectedly. How can I get the details of the RRs and SRs that
are received&sent?

Ahmed





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X-Original-Date: Mon, 01 Jul 2002 12:56:13 -0700
Date: Mon, 01 Jul 2002 12:56:13 -0700

Ahmed,

In your receiver program (your copy of "testMPEGVideoReceiver.cpp"), you 
should change "sessionAddressStr" to be the unicast address of the 
sender.  (Plus, of course, in your copy of "testMPEGVideoStreamer", you 
should change "destinationAddressStr" to be the unicast address of your 
receiver.)

If you do this, your receiver should receive RTCP Sender Report (SR) 
packets from the sender, and the sender should receive RTCP Reception 
Report (RR) packets from the receiver.

	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Mon, 1 Jul 2002 15:56:05 -0400
Date: Mon, 1 Jul 2002 15:56:05 -0400


 I started the server as well as the client in the same machine.  the client
 (openRTSP) sends request and received the SDP description also.  It created
 an output file but nothing was there in the fie, it is a 0 KB file.  The
 client says "Receiving Streamed data..." but I don't see the file size
 increasing...could you please guide me what is wrong...

 --Kannan

> ----- Original Message -----
> From: "S. Kannan" <S.Kannan(at)emessages.com>
> To: <live-devel(at)lists.sourceforge.net>
> Sent: Friday, June 28, 2002 9:10 AM
> Subject: Re: [Live-devel] Re: How to play around!
>
>
> > I modified the testMPEGAudioVideoStreamer project to replace the
> hard-coded
> > IP Address with my machine's IP address and built it.  When I started
this
> > testMPEGAudioVideoStreamer.exe file in the command line, I saw
> >
> > beginning streaming.....
> > reading the file......
> > reading the file.....
> > reading the file... etc..
> >
> > When I run the command line openRTSP it says failed to connect().
> >
> > What do you mean by enabling RTSP server.
> >
> > --Senthamarai.
> >
> > ----- Original Message -----
> > From: "Ross Finlayson" <finlayson(at)live.com>
> > To: <live-devel(at)lists.sourceforge.net>
> > Sent: Friday, June 28, 2002 12:17 AM
> > Subject: [Live-devel] Re: How to play around!
> >
> >
> > > At 04:07 PM 6/27/02, S. Kannan wrote:
> > > >You remember me, I was struggling with an RTSP server which is not
> > > >compatible with others.  Now I have a plan to use your RTSP Server. I
> > built
> > > >the source codes and have OpenRTSP.exe,
testMPEGAudioVideoStreamer.exe,
> > > >testMPEGSplitter.exe and a test.mpeg file too.  How can I play around
> > these
> > > >applications to stream this mpeg file and see the video stream using
> the
> > > >openRTSP client.
> > >
> > > If you want to "see" the video stream, you don't want to use
> > > "openRTSP".  "openRTSP" is just a command-line RTSP client that saves
> the
> > > incoming stream(s) to files.
> > >
> > > To see the video stream, I suggest using QuickTime Player (version 5.0
> or
> > > later).  If you have enabled the RTSP server in
> > > "testMPEGAudioVideoStreamer", you can then view the video in QuickTime
> > > Player by opening the URL
> > >          rtsp://<server>/
> > > (Replace "<server>" with the name (or IP address) of the computer on
> which
> > > you're running "testMPEGAudioVideoStreamer".)
> > >
> > > >   Do I need any SDP file?
> > >
> > > No, if you've enabled the RTSP server, then you won't need the SDP
> > > file.  However, instead of opening a "rtsp://..." URL, you can have
> > > QuickTime Player open the file "testMPEGAudioVideo.sdp" instead.
> > >
> > >          Ross.
> > >
> > > ps. Your file "test.mpg" must be MPEG-1, not MPEG-2.  (QuickTime
Player
> > > doesn't support MPEG-2.)
> > >
> > > pps. In the future, could you please use the developers' mailing list
> > > <live-devel(at)lists.sourceforge.net>.  For one-on-one email, I generally
> > give
> > > priority to paying consulting clients.
> > >
> > >
> > >
> > >
> > > -------------------------------------------------------
> > > This sf.net email is sponsored by:ThinkGeek
> > > Bringing you mounds of caffeinated joy.
> > > http://thinkgeek.com/sf
> > > _______________________________________________
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> > > https://lists.sourceforge.net/lists/listinfo/live-devel
> >
> >
> >
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>



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X-Original-Date: Mon, 01 Jul 2002 13:21:02 -0700
Date: Mon, 01 Jul 2002 13:21:02 -0700

At 12:56 PM 7/1/02, you wrote:
>  I started the server as well as the client in the same machine.  the client
>  (openRTSP) sends request and received the SDP description also.  It created
>  an output file but nothing was there in the fie, it is a 0 KB file.  The
>  client says "Receiving Streamed data..." but I don't see the file size
>  increasing...could you please guide me what is wrong...

The message "Receiving streamed data" indicates only that "openRTSP" is 
*listening* for incoming data.  To check whether RTP packets are actually 
arriving, use the "-n" flag to "openRTSP".  With this flag, once the first 
RTP packet arrives, the program will print out:
         "Data packets have begun arriving"

If you're not receiving any data packets, then it maybe because you have a 
firewall somewhere that is blocking UDP packets.  You will need to 
reconfigure it to enable UDP packets - at least to the ports that 
"openRTSP" uses - in order to receive RTP-based streams like this.  (Note 
that you can specify openRTSP's port numbers using the "-p <startPortNum>" 
command.)

         Ross.



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X-Original-Date: Sat, 1 Jun 2002 19:43:47 -0400
Date: Sat, 1 Jun 2002 19:43:47 -0400

Any idea about '405: Method Not Allowed' message?


----- Original Message -----
From: {Ross Finlayson} <EMAIL-ADDRESS-HIDDEN>
To: <EMAIL-ADDRESS-HIDDEN>
Sent: Monday, July 01, 2002 4:21 PM
Subject: Re: [Live-devel] Re: How to play around!


> At 12:56 PM 7/1/02, you wrote:
> >  I started the server as well as the client in the same machine.  the
client
> >  (openRTSP) sends request and received the SDP description also.  It
created
> >  an output file but nothing was there in the fie, it is a 0 KB file.
The
> >  client says "Receiving Streamed data..." but I don't see the file size
> >  increasing...could you please guide me what is wrong...
>
> The message "Receiving streamed data" indicates only that "openRTSP" is
> *listening* for incoming data.  To check whether RTP packets are actually
> arriving, use the "-n" flag to "openRTSP".  With this flag, once the first
> RTP packet arrives, the program will print out:
>          "Data packets have begun arriving"
>
> If you're not receiving any data packets, then it maybe because you have a
> firewall somewhere that is blocking UDP packets.  You will need to
> reconfigure it to enable UDP packets - at least to the ports that
> "openRTSP" uses - in order to receive RTP-based streams like this.  (Note
> that you can specify openRTSP's port numbers using the "-p <startPortNum>"
> command.)
>
>          Ross.
>
>
>
> -------------------------------------------------------
> This sf.net email is sponsored by:ThinkGeek
> Welcome to geek heaven.
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X-Original-Date: Mon, 01 Jul 2002 17:02:16 -0700
Date: Mon, 01 Jul 2002 17:02:16 -0700

At 04:43 PM 6/1/02, you wrote:
>Any idea about '405: Method Not Allowed' message?


The RTSP server is unhappy with one of the RTSP client requests, for some 
reason.

Please re-run "openRTSP" with the "-V" flag ('verbose'), and include the 
output in your reply.

         Ross.





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From: <EMAIL-ADDRESS-HIDDEN> (S. Kannan)
To: <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] Re: How to play around!
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X-Original-Date: Sat, 1 Jun 2002 22:08:57 -0400
Date: Sat, 1 Jun 2002 22:08:57 -0400

This is a multi-part message in MIME format.

------=_NextPart_000_018F_01C209B8.ED0B2C00
Content-Type: text/plain;
	charset="iso-8859-1"
Content-Transfer-Encoding: 7bit

Ross:

I fixed the problem because, the server expects "describe" (in lowercase)
but the client sends "DESCRIBE".  Now my client saying

-----------------------------------------
Now I have another problem: the server keep says
Beginning streaming...
Beginning to read from file...
-----------------------------------------

But the client creates a file with zero KB, and no more addition to that.

Please take a look at the attached output file.

--Kannan

Ross:

I fixed the problem because, the server expects "describe" (in lowercase)
but the client sends "DESCRIBE".  Now my client saying

-----------------------------------------
Now I have another problem: the server keep says
Beginning streaming...
Beginning to read from file...
-----------------------------------------

But the client creates a file with zero KB, and no more addition to that.

Please take a look at the attached output file.

--Kannan


----- Original Message -----
From: {Senthamarai Kannan} <EMAIL-ADDRESS-HIDDEN>
To: <EMAIL-ADDRESS-HIDDEN>
Sent: Saturday, June 01, 2002 7:43 PM
Subject: Re: [Live-devel] Re: How to play around!


> Any idea about '405: Method Not Allowed' message?
>
>
> ----- Original Message -----
> From: "Ross Finlayson" <finlayson(at)live.com>
> To: <live-devel(at)lists.sourceforge.net>
> Sent: Monday, July 01, 2002 4:21 PM
> Subject: Re: [Live-devel] Re: How to play around!
>
>
> > At 12:56 PM 7/1/02, you wrote:
> > >  I started the server as well as the client in the same machine.  the
> client
> > >  (openRTSP) sends request and received the SDP description also.  It
> created
> > >  an output file but nothing was there in the fie, it is a 0 KB file.
> The
> > >  client says "Receiving Streamed data..." but I don't see the file
size
> > >  increasing...could you please guide me what is wrong...
> >
> > The message "Receiving streamed data" indicates only that "openRTSP" is
> > *listening* for incoming data.  To check whether RTP packets are actuall
y
> > arriving, use the "-n" flag to "openRTSP".  With this flag, once the
first
> > RTP packet arrives, the program will print out:
> >          "Data packets have begun arriving"
> >
> > If you're not receiving any data packets, then it maybe because you have
a
> > firewall somewhere that is blocking UDP packets.  You will need to
> > reconfigure it to enable UDP packets - at least to the ports that
> > "openRTSP" uses - in order to receive RTP-based streams like this.
(Note
> > that you can specify openRTSP's port numbers using the "-p
<startPortNum>"
> > command.)
> >
> >          Ross.
> >
> >
> >
> > -------------------------------------------------------
> > This sf.net email is sponsored by:ThinkGeek
> > Welcome to geek heaven.
> > http://thinkgeek.com/sf
> > _______________________________________________
> > Live-devel mailing list
> > Live-devel(at)lists.sourceforge.net
> > https://lists.sourceforge.net/lists/listinfo/live-devel
>

------=_NextPart_000_018F_01C209B8.ED0B2C00
Content-Type: text/plain;
	name="rtsp_output.txt"
Content-Transfer-Encoding: quoted-printable
Content-Disposition: attachment;
	filename="rtsp_output.txt"

/* SERVER OUTPUT */
/*-----------------------------------------------------------------------=
-----------------------------------------*/
Beginning streaming...
Beginning to read from file...
...done reading from file
Beginning to read from file...
...done reading from file
Beginning to read from file...
...done reading from file
Beginning to read from file...
...done reading from file
Beginning to read from file...
...done reading from file

/*-----------------------------------------------------------------------=
-----------------------------------------*/



/* CLIENT OUTPUT */
/*-----------------------------------------------------------------------=
-----------------------------------------*/
Opened URL "rtsp://64.220.235.40/test.mpg", returning a SDP description:
v=3D0
o=3D- 1025564405602000 1 IN IP4 64.220.235.40
s=3DSDP description from RTSPServer
t=3D0 0
a=3Dtype:broadcast
m=3Dvideo 8888 RTP/AVP 32
a=3Dcontrol:video
c=3DIN IP4 64.220.235.40/7

Created receiver for "video/MPV" subsession (client ports 8888-8889)
Issued RTSP "SETUP" on "video/MPV" subsession (client ports 8888-8889)
Created output file: "video-MPV-1"
Issued RTSP "PLAY" command on "video/MPV" subsession
Receiving streamed data...
^C
D:\live\testProgs\OpenRTSP\Debug>openrtsp -n -p 8888 =
rtsp://64.220.235.40/test.mpg
Opened URL "rtsp://64.220.235.40/test.mpg", returning a SDP description:
v=3D0
o=3D- 1025564405602000 1 IN IP4 64.220.235.40
s=3DSDP description from RTSPServer
t=3D0 0
a=3Dtype:broadcast
m=3Dvideo 8888 RTP/AVP 32
a=3Dcontrol:video
c=3DIN IP4 64.220.235.40/7

Created receiver for "video/MPV" subsession (client ports 8888-8889)
Issued RTSP "SETUP" on "video/MPV" subsession (client ports 8888-8889)
Created output file: "video-MPV-1"
Issued RTSP "PLAY" command on "video/MPV" subsession
Receiving streamed data...
/*-----------------------------------------------------------------------=
-----------------------------------------*/
------=_NextPart_000_018F_01C209B8.ED0B2C00--



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From <EMAIL-ADDRESS-HIDDEN> Mon Jul 1 20:01:28 2002
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X-Original-Date: Mon, 01 Jul 2002 19:57:59 -0700
Date: Mon, 01 Jul 2002 19:57:59 -0700

OK, the problem is that you have apparently modified 
"testMPEGVideoStreamer.cpp" to send to a unicast address.  However, the 
built-in RTSP server currently will not work with unicast streams.

If you change "testMPEGVideoStreamer.cpp" back to using multicast, then you 
will be able to use the built-in RTSP server.

(Also, I think you're using an old version of the code.  I recommend 
upgrading to the newest version of the code.)

	Ross.



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X-Original-Date: Tue, 02 Jul 2002 13:37:06 -0500
Date: Tue, 02 Jul 2002 13:37:06 -0500

hi
i m trying to build the code for on windows. but i m not able to do so 
using the instructions given on the site (using genWindowsmakefile etc.) 
. If anyone has done this then please reply, as to how to build it. I am 
using MS visual studio 6.0, enterprise edition.


mukul



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X-Original-Date: Tue, 02 Jul 2002 02:21:44 -0700
Date: Tue, 02 Jul 2002 02:21:44 -0700

At 11:37 AM 7/2/02, you wrote:
>hi
>i m trying to build the code for on windows. but i m not able to do so 
>using the instructions given on the site (using genWindowsmakefile etc.)

What problem are you having with this?

         Ross.



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X-Original-Date: Tue, 02 Jul 2002 10:22:22 -0700
Date: Tue, 02 Jul 2002 10:22:22 -0700

At 03:27 PM 7/2/02, you wrote:
>1. all the makefiles generated (on linux machine) using the script 
>genWindowsMakefiles, are stored in main directory and not respective 
>subdirectories.

That shouldn't be happening.  After you unpack  (using "tar -xzf") the 
"live...tar.gz" file, you should end up with a directory named "live", with 
*sub*directories named "liveMedia", "groupsock", "UsageEnvironment", 
"BasicUsageEnvironment", and "testProgs".

If you then
         cd live
and then
         ./genWindowsMakefiles
the various ".mak" files should be created *inside* each of the 
subdirectories.  If they're not, then you have a problem.

>2. now in VC, as written, a project with files of type "makefile" is 
>chosen, but nmake fails due to problem of path mismatch.

What type of "path mismatch"?  Note that if your copy of VC is installed 
somewhere other than "c:\Program Files\DevStudio\Vc", you will need to 
change the definition of "TOOLS32" in the file "win32config", *before* you 
run "./genWindowsMakefiles".


>Also i wanted to ask , cant we directly compile in VC without going 
>through the route of makefiles, by just adding source files in empty 
>project, and building it.

I'm sure this is possible, but this would be *a lot* more 
complicated/difficult than just using the Makefiles.

         Ross.

ps. In the future, could you please use the developers' mailing list 
<live-devel(at)lists.sourceforge.net>.  For one-on-one email, I generally give 
priority to paying consulting clients.




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X-Original-Date: Thu, 04 Jul 2002 20:14:58 -0700
Date: Thu, 04 Jul 2002 20:14:58 -0700

At 03:43 PM 6/11/02, you wrote:
>  i am trying to compile the livemedia library for
>compaq ipaq using arm-linux-c++.
>
>the inet.c in groupsock subdirectory which is using
>old style of C programming which is not being
>recognized by g++ or arm-linux-c++.

Someone (perhaps it was you) asked about this back on June 11th & 12th; 
here was the answer that I gave then:

-------------------
At 08:01 AM 6/12/02, you wrote:
>please note that i tried to compile all teh files
>using c++

Bad idea!  This explains the problems that you had.  The "*.c" files must 
be compiled with a C compiler, not a C++ compiler.  Because they're C 
files, the functions in these files get compiled with C linkage - that's 
why the rest of the code declares these functions as 'extern "C"'

>.. i changed the
>C_COMPILER= c++
>in    config.linux

No, please leave this as "cc", or whatever your C compiler is called.  If 
you do this, you should not have to change the rest of the code much (if at 
all), and you should have little problem compiling it.

         Ross.



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X-Original-Date: Thu, 04 Jul 2002 20:16:53 -0700
Date: Thu, 04 Jul 2002 20:16:53 -0700

I ran across an old message in my mail file, and didn't realize that it was 
old.

	Ross.



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X-Original-Date: Tue, 09 Jul 2002 20:58:09 +0530
Date: Tue, 09 Jul 2002 20:58:09 +0530

Dear Ross
Could you please help me in understanding how the doGetNextFrame( ) and 
doGetNextFrame1( ) are bring used? where is doGetNextFrame( ) eventually 
called. I am trying to understand the complete procedure for receiving 
RTP packets and I find that these functions play an important role in 
your design.
Thanks in advance
Prashanth



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X-Original-Date: Tue, 09 Jul 2002 10:12:53 -0700
Date: Tue, 09 Jul 2002 10:12:53 -0700


>Could you please help me in understanding how the doGetNextFrame( ) and 
>doGetNextFrame1( ) are bring used? where is doGetNextFrame( ) eventually 
>called.

"doGetNextFrame()" is called by "FramedSource::getNextFrame()" (see 
"liveMedia/FramedSource.cpp").  It is a pure virtual function that is 
implemented differently for each type of source object.  It is the function 
that actually implements the generation of data from the source object.

"doGetNextFrame1()" is just the name of a helper member function that is 
used, by some source objects, to implement "doGetNextFrame()".

         Ross.



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X-Original-Date: Tue, 9 Jul 2002 13:36:17 -0700 (PDT)
Date: Tue, 9 Jul 2002 13:36:17 -0700 (PDT)

Hi,
Is there any full-fledged movie player on linux(like
quicktime) with full support of rtsp protocol for your
 "live" streaming server?
In the client, I have to give rtsp
url...eg.rtsp://www.foo.com/streamingvideoaudio.sdp
Is this supported by Mplayer(with your patch)?

Arvind


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X-Original-Date: Tue, 09 Jul 2002 13:52:10 -0700
Date: Tue, 09 Jul 2002 13:52:10 -0700


>Is there any full-fledged movie player on linux(like
>quicktime) with full support of rtsp protocol for your
>  "live" streaming server?
>In the client, I have to give rtsp
>url...eg.rtsp://www.foo.com/streamingvideoaudio.sdp
>Is this supported by Mplayer(with your patch)?

Not yet, but I hope to update the patch soon, to bring it up-to-date with 
the latest version of MPlayer, and (among other things) add RTSP client 
support.

(I may also try doing the same with the "videoLAN" player.)

I'll be announcing this on this mailing list when it's available.

         Ross.



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Subject: [Live-devel] Regarding openRTSP !!!
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X-Original-Date: Sat, 13 Jul 2002 18:38:25 +0530
Date: Sat, 13 Jul 2002 18:38:25 +0530


Hi Ross,
 
I am planning to integrate the openRTSP to a MPEG2 Hardware Decoder, in 
a settopBOX of a Digital TV.
My question is ,

1> To acheive the above task i think just pumping the data from 
FileSink.cpp to the decoder input would be sufficient.Am i right ..

2>One more problem is , if the decoder is having only one input which 
takes both audio and video input and then demultiplux it , how can i 
acheive this through openRTSP .Because openRTSP receives audio and video 
separately , am i right .What more changes i need to do .

3> i also heard that you have integrated the openRTSP with a linux MPEG 
player s/w .Did u have to make many changes on the library side.Please let
me know , Because i guess this would be very helpful for my integration.

4> What is the maximum bitrate speed the openRTSP programe can receive 
the stream data.

Please guide me on this,

thanks in advance,
niranjan






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X-Original-Date: Sat, 13 Jul 2002 19:17:56 -0700
Date: Sat, 13 Jul 2002 19:17:56 -0700


>I am planning to integrate the openRTSP to a MPEG2 Hardware Decoder, in
>a settopBOX of a Digital TV.
>My question is ,
>
>1> To acheive the above task i think just pumping the data from
>FileSink.cpp to the decoder input would be sufficient.Am i right ..

Yes.  If your settop box's OS supports pipes, then you could just use 
FileSink, as is, with an output 'file name' of "stdout", and then just pipe 
it to your decoder.

Alternatively, you could write your own 'sink' object (replacing the use of 
"FileSink") that outputs directly to your decoder.


>2>One more problem is , if the decoder is having only one input which
>takes both audio and video input and then demultiplux it , how can i
>acheive this through openRTSP .Because openRTSP receives audio and video
>separately , am i right .What more changes i need to do .

You should first check to make sure that your decoder really can't take 
separate Elementary Streams as input, because if they can, this would make 
things a lot simpler.

If your decoder *really* can take only a MPEG Program or Transport Stream 
(i.e., combining audio and video) as input, then you have two choices.

1/ If you have control over your RTSP *servers* - i.e., if you know that 
your client will be getting data from only certain RTSP server(s), and you 
have control over these server(s), then you can modify the server(s) to 
stream a Program or Transport Stream in a single RTP stream.  It will then 
be easy for your client to send that data to your decoder.

2/ If, however, you have no control over the RTSP servers, then you will 
need to write a mixer object that combines the input Audio and Video 
Elementary Streams into a single output Program or Transport Stream.


>3> i also heard that you have integrated the openRTSP with a linux MPEG
>player s/w .Did u have to make many changes on the library side.

No, not many.  All of the changes that I made are now included in the 
latest release of the library code.


>4> What is the maximum bitrate speed the openRTSP programe can receive
>the stream data.

There's no way to answer a question like this without having the hardware 
and OS in front of you...

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>





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X-Original-Date: Tue, 16 Jul 2002 21:35:25 -0700 (PDT)
Date: Tue, 16 Jul 2002 21:35:25 -0700 (PDT)

I can access the webpage at
http://sourceforge.net/projects/live. From it I downed
the live package. I wanna compile it and use it .
but I cannnot access the http://live.sourceforge.net/ 
so I cannot get the documents about it .
so I don't know how to compile this package and use
the testprogs . 
Where Can i get these documents ?
Can you give me a hand ?

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X-Original-Date: Tue, 16 Jul 2002 22:00:21 -0700
Date: Tue, 16 Jul 2002 22:00:21 -0700

At 09:35 PM 7/16/02, you wrote:
>I can access the webpage at
>http://sourceforge.net/projects/live. From it I downed
>the live package. I wanna compile it and use it .
>but I cannnot access the http://live.sourceforge.net/

Do you know why not?  I can get to it right now just fine.  Sounds like you 
should contact your ISP about this...

>so I cannot get the documents about it .
>so I don't know how to compile this package and use
>the testprogs .
>Where Can i get these documents ?

They're at <http://live.sourceforge.net/> - once you can get to it.

         Ross.




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X-Original-Date: Wed, 17 Jul 2002 21:31:14 -0700 (PDT)
Date: Wed, 17 Jul 2002 21:31:14 -0700 (PDT)

I access Internet through education net . And I dont
know why  I cant get it . But I do be able to go any
else site home and abroad.
Does this website has any mirror site?
Or any of you could give me a hand by mailing me one
copy ?



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X-Original-Date: Wed, 17 Jul 2002 21:53:49 -0700
Date: Wed, 17 Jul 2002 21:53:49 -0700

At 09:31 PM 7/17/02, you wrote:
>I access Internet through education net . And I dont
>know why  I cant get it . But I do be able to go any
>else site home and abroad.
>Does this website has any mirror site?

I've made a*temporary* mirror at
         <http://www.live.com/liveMedia/sourceforge-index.html>

However, I'd really like to find out why you can't access the real web 
page: <http://live.sourceforge.net/>; I wonder if perhaps your ISP is 
censoring your web access, and has mistakenly added this page to a filter.

Could you tell me the URL of your ISP?

         Ross.



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From <EMAIL-ADDRESS-HIDDEN> Wed Jul 17 23:06:26 2002
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X-Original-Date: Thu, 18 Jul 2002 11:35:38 +0530
Date: Thu, 18 Jul 2002 11:35:38 +0530

Hi,
I want to learn about streaming, particulary about the part where socket
programming is used to transfer data from the server to the client. Which
part of the code should I start from, seeing that there are a lot of things
given. Any other resources to facilitate the learning?
thanks n regards
Anurag
---------------------------------------------------------------

> -----Original Message-----
> From:	Ross Finlayson [SMTP:finlayson(at)live.com]
> Sent:	Thursday, July 18, 2002 10:24 AM
> To:	live-devel(at)lists.sourceforge.net
> Cc:	tony Tong
> Subject:	Re: [Live-devel] Re:  How to get Documents of live?
> 
> At 09:31 PM 7/17/02, you wrote:
> >I access Internet through education net . And I dont
> >know why  I cant get it . But I do be able to go any
> >else site home and abroad.
> >Does this website has any mirror site?
> 
> I've made a*temporary* mirror at
>          <http://www.live.com/liveMedia/sourceforge-index.html>
> 
> However, I'd really like to find out why you can't access the real web 
> page: <http://live.sourceforge.net/>; I wonder if perhaps your ISP is 
> censoring your web access, and has mistakenly added this page to a filter.
> 
> Could you tell me the URL of your ISP?
> 
>          Ross.
> 
> 
> 
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X-Original-Date: Thu, 18 Jul 2002 11:54:13 +0530
Date: Thu, 18 Jul 2002 11:54:13 +0530

While opening the file MPEG4IP_Guide.pdf, it says that there is an error
opening it. Where can I get a different copy? 
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From <EMAIL-ADDRESS-HIDDEN> Thu Jul 18 10:41:09 2002
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Subject:  [Live-devel] Start Up Trouble.
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X-Original-Date: Thu, 18 Jul 2002 23:10:54 +0530
Date: Thu, 18 Jul 2002 23:10:54 +0530

Hi again,
I realized I had started exploring the wrong Project. I've got the Live.Com
Streaming Media one now.
I have been able to build using nmake the BasicUsageEnvironment, groupsock,
liveMedia, UsageEnvironment straight from the comand promt. The testProgs I
had to do from the VC++ environment, as from the command promt it was giving
the error "cannot Open input filemsvcirt.lib". I dont know why!! 

Now, whats the best way to proceed? I want to be able to Debug and study the
programs. I have  a WinNT machine with MS VC++.
When I open the worksapce testProgs1.dsw, it shows only the makefile!! How
do I include the correct source files into the workspace?

regards
Anurag

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X-Original-Date: Thu, 18 Jul 2002 11:28:10 -0700
Date: Thu, 18 Jul 2002 11:28:10 -0700


>I have been able to build using nmake the BasicUsageEnvironment, groupsock,
>liveMedia, UsageEnvironment straight from the comand promt. The testProgs I
>had to do from the VC++ environment, as from the command promt it was giving
>the error "cannot Open input filemsvcirt.lib". I dont know why!!

It sounds like you have not been following the instructions in 
<http://live.sourceforge.net/#config-windows>.  Before building any of the 
project directories using VC++, you must first create each of the ".mak" 
files (one in each subdirectory).  Once you've done this, you can build 
each subdirectory from VC++ by having it open (and create a project from) 
the ".mak" file.  The "testProgs" are built the same way.

         Ross.



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X-Original-Date: Thu, 18 Jul 2002 22:48:10 -0700
Date: Thu, 18 Jul 2002 22:48:10 -0700

I finally figured out why Tony Tong was having problems accessing the 
project web page at "sourceforge.net".  It turns out that he's in China, 
and apparently his government's firewall ("The Great Firewall of China") is 
blocking access to "live.sourceforge.net"!

Although it's flattering to think that the Chinese government finds this 
Open Source streaming media project subversive, it's more likely that 
they've just made a mistake by blocking it.  In any case, I don't want to 
prevent more than one billion people from being able to access the web page.

So, I've moved the main project web page to
	<http://www.live.com/liveMedia/>

Please update any links that you may have.

(The old page on sourceforge.net still exists, but points to the new page.)

	Ross.



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X-Original-Date: Thu, 18 Jul 2002 22:52:30 -0700
Date: Thu, 18 Jul 2002 22:52:30 -0700

Following up on my previous message: This mailing list has not moved, it 
remains
	live-devel(at)lists.sourceforge.net

( Apparently "lists.sourceforge.net", unlike "live.sourceforge.net", is not 
blocked from within China.)

However, I can't rule out the possibility of the mailing list moving some 
time in the future.

	Ross.



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X-Original-Date: Fri, 19 Jul 2002 16:21:52 +0530
Date: Fri, 19 Jul 2002 16:21:52 +0530

Well, I redid the whole thing, and just to make things clear, let me state
step wise what I did.

1. Unpacked it on a Unix machine
2. Made path changes in the TOOLS32 line in the file win32config.
3. ./genWindowsMakefiles.
4. I got the make files for each directory.
5. Copied the folders on to the Windows Machine.
6. Did "Open Workspace", using this new make file. Visual Studio warns that
Continuuing will create a new Develepor Studio project to wrap this
makefile. Do you want to continue?" I say "Yes". It again prompts to select
the platform in the external makefile. I choose Win32. (Thats the only one
displayed.) It again prompts a "Save As" so I save it as
<directory-name>.dsp
7. I do "Rebuild All". 0 Errors. I do this for all the 4 projects first and
then for the testProgs. No Errors.

Now what? I want to be able to debug the programs and understand them.
Please advice. When I open the workspace testProgs.dsw, only the file
testProgs.mak is included in the project. How about the other files? Which
files are  for which Project? 
Thanks for taking the trouble Ross.
Anurag
> -----Original Message-----
> From:	Ross Finlayson [SMTP:finlayson(at)live.com]
> Sent:	Thursday, July 18, 2002 11:58 PM
> To:	live-devel(at)lists.sourceforge.net
> Subject:	Re: [Live-devel] Start Up Trouble.
> 
> 
> >I have been able to build using nmake the BasicUsageEnvironment,
> groupsock,
> >liveMedia, UsageEnvironment straight from the comand promt. The testProgs
> I
> >had to do from the VC++ environment, as from the command promt it was
> giving
> >the error "cannot Open input filemsvcirt.lib". I dont know why!!
> 
> It sounds like you have not been following the instructions in 
> <http://live.sourceforge.net/#config-windows>.  Before building any of the
> 
> project directories using VC++, you must first create each of the ".mak" 
> files (one in each subdirectory).  Once you've done this, you can build 
> each subdirectory from VC++ by having it open (and create a project from) 
> the ".mak" file.  The "testProgs" are built the same way.
> 
>          Ross.
> 
> 
> 
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>7. I do "Rebuild All". 0 Errors. I do this for all the 4 projects first and
>then for the testProgs. No Errors.

That's good.  You should now be able to run each of the test programs (from 
a command shell).


>Now what? I want to be able to debug the programs and understand them.
>Please advice. When I open the workspace testProgs.dsw, only the file
>testProgs.mak is included in the project. How about the other files?

The other files are mentioned in the ".mak" file (i.e., Makefile) only.

There are two possible ways for you to continue working on the code in 
Windows using VC++:
1/ Make any 'project' modifications to the ".mak" files (i.e., Makefiles) 
*only*.
         Don't use the VC++ GUI at all for this.
         In particular, to get debugging, you can comment out the line
                 NODEBUG=1
         that's near the top of each ".mak" file.
         To add files to the project, do so by adding their names to the 
Makefile.
         This is the way I do things (because I'm used to doing development
         in a Unix environment, using Makefiles, and I rarely do any
         Windows-specific development anyway).
2/ Use the VC++ GUI to make 'project' modifications.
         If you want to do things this way, then I can't help you much, because
         I'm not much of an expert with VC++.  As I noted above, I usually 
just work
         from Makefiles.  However, maybe someone on the mailing list can
         help you out here??

     Ross.




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> -----Original Message-----
> From:	achraf [SMTP:achraf.gazdar(at)cristal.rnu.tn]
> Sent:	Saturday, July 20, 2002 1:56 PM
> To:	live-devel(at)lists.sourceforge.net; anurag.s(at)sonata-software.com
> Subject:	Re: using VC++
> 
		|......         Windows-specific development anyway).
> 	|2/ Use the VC++ GUI to make 'project' modifications.
> 	|         If you want to do things this way, then I can't help you
> much, because
> 	|         I'm not much of an expert with VC++.  As I no
> 	|ted above, I usually 
> 	|just work
> 	|         from Makefiles.  However, maybe someone on the mailing
> list can
> 	|         help you out here??
> 	|
> 	|     Ross.
		|
		|
> 	|That's right, to use VC++ GUI, you just need to make a new project.
> 
> 
		When I make a new Project, what kind of Project should I
choose to make? 
			1. Win32 Application
			2. Win32 Console Application
		      3. Other.
		And why do we need the GUI stuff? Can't we just use a
console based Project? Maybe if you send me the project you have using VC++
I can understand things better.
		In case the size of your project exceeds the size limit for
me, can u send a cc to anushuks(at)yahoo.com?
		Thanks a lot
		Anurag




		| than youe create your Graphic interfaces (probably with
wizard). But, before building  
		|   you must  make those changes:

	|1. add ..\include directory of each live directory to the includes
directory of the developpment envirenment |you find this under the menu
"Tools"/"Options"/"Directories"/"Include Files")
> |2. change your project settings as fallow:
>  |    -menu "Project"/"Settings..."
>    |   -under "General": if you want to use MFC library you must choose
> "use MFC in a shared DLL"   
>  |    -under "C/C++" "category code génération" choose "Multithread DLL"
> as a run time library
>  |    -under "link" you must link with "Wsock32.lib
> libUsageEnvironment.lib libBasicUsageEnvironment.lib |libgroupsock.lib
> libliveMedia.lib "
>      
> |3.finally you must add to your project (under menu "Project"/"add to
> project"/"new") live.com library such as|(libLiveMedia.lib...)
> |4. finally don't forget to make your include in your testPrograms as they
> are maked in testProgs of |ivemedia"
> |(I will send to you one project witch use live.com Maked with VC++)
> I| hope this help   
> 
> 
> 
> 	 
> 	
> 	
> 	
> 	
> 	--__--__--
> 	
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> 	
> 	
> 
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X-Original-Date: Mon, 22 Jul 2002 12:51:52 +0530
Date: Mon, 22 Jul 2002 12:51:52 +0530

When I try to tun the test programs, I get the message "this exe is not a
valid NT application". Why so? How do I run it then?
thanks
Anurag

> -----Original Message-----
> From:	Ross Finlayson [SMTP:finlayson(at)live.com]
> Sent:	Friday, July 19, 2002 4:46 PM
> To:	live-devel(at)lists.sourceforge.net
> Subject:	RE: [Live-devel] Start Up Trouble.
> 
> 
> >7. I do "Rebuild All". 0 Errors. I do this for all the 4 projects first
> and
> >then for the testProgs. No Errors.
> 
> That's good.  You should now be able to run each of the test programs
> (from 
> a command shell).
> 
> 
> >Now what? I want to be able to debug the programs and understand them.
> >Please advice. When I open the workspace testProgs.dsw, only the file
> >testProgs.mak is included in the project. How about the other files?
> 
> The other files are mentioned in the ".mak" file (i.e., Makefile) only.
> 
> There are two possible ways for you to continue working on the code in 
> Windows using VC++:
> 1/ Make any 'project' modifications to the ".mak" files (i.e., Makefiles) 
> *only*.
>          Don't use the VC++ GUI at all for this.
>          In particular, to get debugging, you can comment out the line
>                  NODEBUG=1
>          that's near the top of each ".mak" file.
>          To add files to the project, do so by adding their names to the 
> Makefile.
>          This is the way I do things (because I'm used to doing
> development
>          in a Unix environment, using Makefiles, and I rarely do any
>          Windows-specific development anyway).
> 2/ Use the VC++ GUI to make 'project' modifications.
>          If you want to do things this way, then I can't help you much,
> because
>          I'm not much of an expert with VC++.  As I noted above, I usually
> 
> just work
>          from Makefiles.  However, maybe someone on the mailing list can
>          help you out here??
> 
>      Ross.
> 
> 
> 
> 
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X-Original-Date: Mon, 22 Jul 2002 00:43:53 -0700
Date: Mon, 22 Jul 2002 00:43:53 -0700

At 12:21 AM 7/22/02, you wrote:
>When I try to tun the test programs, I get the message "this exe is not a
>valid NT application". Why so?

I don't know.  If you're building the test programs (and, before that, the 
libraries) from the ".mak" files, exactly as explained in the instructions 
on <http://www.live.com/liveMedia/>, then you should get valid ".exe" 
executables that you can run.  (Of course, if you *don't* use the ".mak" 
files, then all bets are off)

Has anyone else seen this problem?

         Ross.




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X-Original-Date: Tue, 23 Jul 2002 11:34:17 +0000 (GMT)
Date: Tue, 23 Jul 2002 11:34:17 +0000 (GMT)

 =

=3Chtml=3E=3Cnobr=3E=3Cimg src=3Dhttp=3A//maildb=2Esamsung=2Ecom=3A8001/o=
penconfirm=3Fparm=3Dniranjan=5F=5Fniranjan20020723203932751 width=3D=221=22=
 height=3D=221=22=3E=3C/nobr=3E
=3Cfont size=3D3 face=3D=22=B1=BC=B8=B2=C3=BC=22=3E =3Cpre=3EHi =2C

I have a question =2C =

openRTSP can stream only through a rtsp url from a rtsp server =2C but wh=
at more =

changes need to be done to make it support streaming from a http url =2C =
lets
say suppose a media server is streaming using http and rtp=2E
For eg =3A =

http=3A//www=2Ethc=2Eac=2Ekr/=7Echu5505/0107yen/video/yust=2Empg

thanks in advance
niranjan=3C/font=3E=3C/pre=3E=3C/html=3E



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X-Original-Date: Tue, 23 Jul 2002 21:11:00 +0530
Date: Tue, 23 Jul 2002 21:11:00 +0530

Hi,
I get the following error when I try to build openRTSP. Any clues? Im using
the VC++ environment. The libraries have been built successfully.

Hey Achraf, I couldnt receive the project sample you sent. Maybe theres some
problem. Could you send it again?

Thanks
**********************************************************
Deleting intermediate files and output files for project 'LiveTry2 - Win32
Debug'.
--------------------Configuration: LiveTry2 - Win32
Debug--------------------
Compiling...
openRTSP.cpp
Linking...
LINK : fatal error LNK1209: program database
"C:\Live\LiveTry2\Debug\LiveTry2.pdb" differs from previous link; relink or
rebuild
Error executing link.exe.

LiveTry2.exe - 1 error(s), 0 warning(s)
*********************************************************
Anurag
---------------------------------------------------------------

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X-Original-Date: Tue, 23 Jul 2002 10:03:37 -0700
Date: Tue, 23 Jul 2002 10:03:37 -0700


>I have a question ,
>openRTSP can stream only through a rtsp url from a rtsp server , but what 
>more
>changes need to be done to make it support streaming from a http url , lets
>say suppose a media server is streaming using http and rtp.
>For eg :
>http://www.thc.ac.kr/~chu5505/0107yen/video/yust.mpg

No, this is *not* being streamed using RTP.  Instead, it's just regular 
HTTP streaming - i.e., using TCP.
So, to stream (and receive a copy of) this movie, just use a HTTP client, 
for example,
         lynx -source <http-url> > yust.mpg

     Ross.



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X-Original-Date: Wed, 24 Jul 2002 10:53:43 +0530
Date: Wed, 24 Jul 2002 10:53:43 +0530

Hi,
To run openRTSP program, ( openRTSP <http://www.live.com/openRTSP/>
programis a command-line program that can be used to open, stream, receive,
and (optionally) record media streams that are specified by a RTSP URL -
i.e., an URL that begins with rtsp:// ) what are the pre-requiesites I must
have on my comp? Is it necessary to have my machine on the Internet? Mine is
not. Can I run the program witha normal mp3 file on my machine?

Thanks
Anurag
---------------------------------------------------------------

> -----Original Message-----
> From:	Anurag Shukla [SMTP:anurag.s(at)sonata-software.com]
> Sent:	Tuesday, July 23, 2002 9:11 PM
> To:	live-devel(at)lists.sourceforge.net
> Cc:	Ross Finlayson; achraf
> Subject:	[Live-devel] Linking Error
> 
> Hi,
> I get the following error when I try to build openRTSP. Any clues? Im
> using
> the VC++ environment. The libraries have been built successfully.
> 
> Hey Achraf, I couldnt receive the project sample you sent. Maybe theres
> some
> problem. Could you send it again?
> 
> Thanks
> **********************************************************
> Deleting intermediate files and output files for project 'LiveTry2 - Win32
> Debug'.
> --------------------Configuration: LiveTry2 - Win32
> Debug--------------------
> Compiling...
> openRTSP.cpp
> Linking...
> LINK : fatal error LNK1209: program database
> "C:\Live\LiveTry2\Debug\LiveTry2.pdb" differs from previous link; relink
> or
> rebuild
> Error executing link.exe.
> 
> LiveTry2.exe - 1 error(s), 0 warning(s)
> *********************************************************
> Anurag
> ---------------------------------------------------------------
> 
> *********************************************************************
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X-Original-Date: Tue, 23 Jul 2002 22:44:57 -0700
Date: Tue, 23 Jul 2002 22:44:57 -0700

At 10:23 PM 7/23/02, you wrote:
>Hi,
>To run openRTSP program, ( openRTSP <http://www.live.com/openRTSP/>
>programis a command-line program that can be used to open, stream, receive,
>and (optionally) record media streams that are specified by a RTSP URL -
>i.e., an URL that begins with rtsp:// ) what are the pre-requiesites I must
>have on my comp? Is it necessary to have my machine on the Internet? Mine is
>not. Can I run the program witha normal mp3 file on my machine?

"openRTSP" is a RTSP client, which means that it works by talking to a RTSP 
server (the server that's named in the "rtsp://" URL).  So, if you have a 
RTSP server running on your computer, you can use "openRTSP" to stream it 
(using "rtsp://localhost/<filename>"), although I'm not sure why you would 
want to do this.  (If the file that you're interested in is already on your 
computer you can access it directly; you won't need RTSP.)

But anyway, your question suggests that perhaps you don't quite understand 
what RTSP is, and how it is used.  Could you explain exactly what it is you 
are trying to do?

         Ross.



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X-Original-Date: Wed, 24 Jul 2002 00:21:22 -0700
Date: Wed, 24 Jul 2002 00:21:22 -0700

At 12:06 AM 7/24/02, you wrote:
>Actually your observation is correct. All I know about RTSP is that its a
>protoco used for streaming.

I recommend reading <http://www.rtsp.org/>, and, in particular, 
<http://www.rtsp.org/2001/faq.html>

         Ross.



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X-Original-Date: Wed, 24 Jul 2002 12:06:29 +0100
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After several hours with a packetlogger and a hex editor, I have made
some progress with extracting playable media from RDT streams.  However,
I notice that so far, the LIVE project only deals with open protocols.
Would my RDT implementation be welcome here, or could it create legal
problems?

Glyn

-- 
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On Wed, 24 Jul 2002, Glyn wrote:
> 
> After several hours with a packetlogger and a hex editor, I have
> made some progress with extracting playable media from RDT streams.
> However, I notice that so far, the LIVE project only deals with
> open protocols. Would my RDT implementation be welcome here, or
> could it create legal problems?

Sounds fascinating, but what's RDT?


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X-Original-Date: Wed, 24 Jul 2002 15:02:02 +0100
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Eric Jacobs wrote:
> Sounds fascinating, but what's RDT?

The proprietary protocol used for a lot of RealMedia streams.  Quite
similar to RTP, from what I can make out.

Glyn

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X-Original-Date: Wed, 24 Jul 2002 16:32:13 -0700
Date: Wed, 24 Jul 2002 16:32:13 -0700

At 04:06 AM 7/24/02, you wrote:
>After several hours with a packetlogger and a hex editor, I have made
>some progress with extracting playable media from RDT streams.  However,
>I notice that so far, the LIVE project only deals with open protocols.
>Would my RDT implementation be welcome here, or could it create legal
>problems?

Re. "legal problems?" - I'm not sure.  The safest thing might be to host 
the document on your own web site, and just post a link here.

However, I'm not sure how useful a RDT implementation would be by 
itself.  RealNetworks' RDT streams are initiated using RTSP; however, 
RealNetworks' RTSP server includes a cryptographic 'challenge-response', 
test to ensure that only valid RealNetworks players can start the 
stream.  This is why "openRTSP" (for example) probably couldn't start 
streaming RealAudio and/or RealVideo from a RealNetworks server, even if we 
implemented RDT.

However, RealNetworks is said to be releasing some player client source 
code within 90 days (as part of their new "Helix" project).  Maybe this 
will include sufficient information to be able to receive 
RealAudio/RealVideo streams??

(Rob Lanphier of RealNetworks is on this mailing list; maybe he'll clarify 
this?)

         Ross.



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From: Glyn <EMAIL-ADDRESS-HIDDEN>
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Subject: Re: [Live-devel] RDT
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X-Original-Date: Thu, 25 Jul 2002 12:01:57 +0100
Date: Thu, 25 Jul 2002 12:01:57 +0100

Ross Finlayson wrote:
> >After several hours with a packetlogger and a hex editor, I have made
> >some progress with extracting playable media from RDT streams.  However,
> >I notice that so far, the LIVE project only deals with open protocols.
> >Would my RDT implementation be welcome here, or could it create legal
> >problems?
> 
> Re. "legal problems?" - I'm not sure.  The safest thing might be to host 
> the document on your own web site, and just post a link here.

Well, my ISP (ox.ac.uk) came down like a ton of bricks on someone
hosting DeCSS on their website, (and didn't apologise when it turned out
that it was the Cascading Style Sheet perlscript), but as this isn't so
high profile, and I won't be uploading any actual source for a while,
I'll risk it.  I will post the URL at the weekend, when I actually
manage to get the machine (the one with the documents on it) near to a
network point.

My main concern in asking whether my code would be welcome here is
that if this project is intended to be based on open protocols only,
then code for reverse-engineered, undocumented protocols would be
considered too much of an unreliable hack to be included.  Though I'd
probably have to wait for you to all see how bad my code is before
anyone can comment on that.

> However, I'm not sure how useful a RDT implementation would be by 
> itself.  RealNetworks' RDT streams are initiated using RTSP; however, 
> RealNetworks' RTSP server includes a cryptographic 'challenge-response', 
> test to ensure that only valid RealNetworks players can start the 
> stream.

I had noticed the `RealChallenge' lines in the rtsp logs, but hadn't
attached much significance to them.  I may have another look.

> However, RealNetworks is said to be releasing some player client source 
> code within 90 days (as part of their new "Helix" project).  Maybe this 
> will include sufficient information to be able to receive 
> RealAudio/RealVideo streams??

The way I understand it, the Helix server will be used for
RealAudio/video streaming, but won't work in the same way as the
existing RealNetworks servers, so the source won't help with the
challenge/response bits at all.  That was the impression I got from the
press-release.

Glyn

-- 
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From: Glyn <EMAIL-ADDRESS-HIDDEN>
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X-Original-Date: Sat, 27 Jul 2002 18:11:52 +0100
Date: Sat, 27 Jul 2002 18:11:52 +0100

I wrote:
> I will post the URL at the weekend, when I actually manage to get the
> machine (the one with the documents on it) near to a network point.

OK, first draft at http://users.ox.ac.uk/~hert1023/rdt/ .  Comments
always welcome.

Glyn

-- 
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From: Niranjan <EMAIL-ADDRESS-HIDDEN>
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Subject: [Live-devel] Regarding streaming MPEG transport stream
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X-Original-Date: Thu, 01 Aug 2002 12:01:44 +0000 (GMT)
Date: Thu, 01 Aug 2002 12:01:44 +0000 (GMT)

 =

=3Chtml=3E=3Cnobr=3E=3Cimg src=3Dhttp=3A//maildb=2Esamsung=2Ecom=3A8001/o=
penconfirm=3Fparm=3Dniranjan=5F=5Fniranjan20020801210543865 width=3D=221=22=
 height=3D=221=22=3E=3C/nobr=3E
=3Cfont size=3D3 face=3D=22=B1=BC=B8=B2=C3=BC=22=3E =3Cpre=3EHi =2C =


I have following questions=2C

1=3E Is it possible for the livemedia library to stream an MPEG Transport=
 Stream=2E
If so then =2C exactly what changes shud be made =2E
At the streaming side(for eg testMPEvideoStreamer)
I have to change the video sink as SimpleRTPSink=3Acreatnew(*env=2Crtpgs=2C=
33=2C90000=2E=22application)=22
Since the data is to be read as 188 - byte chunks so where shud i modify =
the =

ByteStreamFileSource=2E
On the Streaming side are only these two class modification to be made=2E=

And =2C then on the receiving side (for eg openRTSP) i need to just creat=
e a source
SimpleRTPSource=3AcreateNew(*env=2Crtpgs=2C33=2C90000=2C=22application/mp=
2ts=22)=3B
Is this all i need to do
=2E
Any thing else Please let me =2EAnd has anyone worked on this type of str=
eam with =

live library=2E

regards
niranjan=2E



 =





=3C/font=3E=3C/pre=3E=3C/html=3E



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X-Original-Date: Thu, 01 Aug 2002 09:33:36 -0700
Date: Thu, 01 Aug 2002 09:33:36 -0700


>1> Is it possible for the livemedia library to stream an MPEG Transport 
>Stream.
>If so then , exactly what changes shud be made .
>At the streaming side(for eg testMPEvideoStreamer)
>I have to change the video sink as 
>SimpleRTPSink:creatnew(*env,rtpgs,33,90000."application)"

Yes, that should work.

>Since the data is to be read as 188 - byte chunks so where shud i modify the
>ByteStreamFileSource.

You won't need to modify the "ByteStreamFileSource" class.  Instead you can 
create your "ByteStreamFileSource" object as 
follows:        ByteStreamFileSource::createNew(*env, fileName, 188)
it will cause data to be read in 188-byte chunks

>On the Streaming side are only these two class modification to be made.

Yes, I think so.

>And , then on the receiving side (for eg openRTSP) i need to just create a 
>source
>SimpleRTPSource:createNew(*env,rtpgs,33,90000,"application/mp2ts");
>Is this all i need to do
Yes, I think so.

         Ross.



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X-Original-Date: Fri, 02 Aug 2002 02:39:27 +0000 (GMT)
Date: Fri, 02 Aug 2002 02:39:27 +0000 (GMT)

 =

=3Chtml=3E=3Cnobr=3E=3Cimg src=3Dhttp=3A//maildb=2Esamsung=2Ecom=3A8001/o=
penconfirm=3Fparm=3Dniranjan=5F=5Fniranjan20020802114327997 width=3D=221=22=
 height=3D=221=22=3E=3C/nobr=3E
=3Cfont size=3D3 face=3D=22=B1=BC=B8=B2=C3=BC=22=3E =3Cpre=3EHi Ross=2C

Thanks for your response=2E

I did the changes you mentioned in the previous mail =2E
But when i ran the testMPEGVideoStreamer with a sample MPEG2transportstre=
am
 file it says =22didnt find pack=22=2E
I guess the problem is with the demultiplexer =2C the MPEGDemux=2Ecpp dem=
ultiplexes =

only mpeg 1/2 program stream=2E
Am i right =2ESo what changes shud be made to support transport stream de=
muxing=2E

Please let me know your opinion about me=2E

regards
niranjan=2E



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X-Original-Date: Thu, 01 Aug 2002 20:00:04 -0700
Date: Thu, 01 Aug 2002 20:00:04 -0700


>I did the changes you mentioned in the previous mail .
>But when i ran the testMPEGVideoStreamer with a sample MPEG2transportstream
>  file it says "didnt find pack".
>I guess the problem is with the demultiplexer , the MPEGDemux.cpp 
>demultiplexes
>only mpeg 1/2 program stream.
>Am i right .So what changes shud be made to support transport stream demuxing.


Your chain of objects should just be
         ByteStreamFileSource(188)->SimpleRTPSink
That's all.  You don't need a "MPEGVideoStreamFramer" object, and you don't 
need a "MPEGDemux".

So, if you're using "testMPEGVideoStreamer.cpp" as a model, you should make 
sure that SOURCE_IS_PROGRAM_STREAM is *not* defined.  You should also 
remove the call to
         MPEGVideoStreamFramer::createNew()

   Ross.



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From <EMAIL-ADDRESS-HIDDEN> Fri Aug 2 10:04:51 2002
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X-Original-Date: Fri, 2 Aug 2002 10:03:45 -0700 (PDT)
Date: Fri, 2 Aug 2002 10:03:45 -0700 (PDT)

Hi Ross

Thanks for your cool live streaming libraries, it
makes life much easier.

I used your patched version of mplayer to play a
mpeg-2 program stream file sent by
testMPEGAudioVideoStreamer i.e

mplayer -vo xv testMPEGAudioVideo.sdp 

It hangs on audio decoder. After some debugging, found
that the RTP audio session always initiated with a
codecName 'MPA', which trigger mplayer to use mp3lib. 

When mplayer play that program stream file directly,
it use AC3.

In libmpdemux/demux_sdp_rtp.cpp, you have the code for
AC3, however, in MediaSession::lookupPayloadFormat,I
saw a long list of formats,  but  AC3 is not there. 

Am I using the correct version or something is missing
there?

Thanks

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Subject: [Live-devel] QuickTime Protocol
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X-Original-Date: Fri, 2 Aug 2002 15:51:37 -0300 (EST)
Date: Fri, 2 Aug 2002 15:51:37 -0300 (EST)

Hi,

I'm having troubles with quicktime protocol. I've built a streaming proxy
which receives the stream from the server and repasses it to quicktime. It
receive and send the first packet, but from the second on, the connection
is refused. Anyone knows what could be?

Thanks in advance
  Marisa




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X-Original-Date: Fri, 02 Aug 2002 12:53:32 -0700
Date: Fri, 02 Aug 2002 12:53:32 -0700

Defang,

Within a few days, I'll be releasing a new, much improved version of the 
mplayer RTP streaming patch.  This will work with the newest CVS version of 
mplayer (instead of just the April 10th version that the current patch 
works with).

However, as you discovered, the LIVE.COM Streaming Media libraries do not 
yet implement the RTP payload format for AC3 audio (which, by the way, is 
still in the early stages of IETF standardization).  So, the libraries 
cannot yet be used to stream or receive AC3 audio, even though mplayer has 
the ability to play AC3 streams.

(If your company is interested in funding the addition of AC3 support to 
the libraries, please let me know (via private email).)

	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Fri, 02 Aug 2002 12:55:04 -0700
Date: Fri, 02 Aug 2002 12:55:04 -0700


>I'm having troubles with quicktime protocol. I've built a streaming proxy
>which receives the stream from the server and repasses it to quicktime. It
>receive and send the first packet, but from the second on, the connection
>is refused. Anyone knows what could be?

Marisa,

I'm sorry, but from your description I can't tell what you are trying to 
do, and what problem(s) you might be having.

Can you be more specific about what you are trying to do?  What "LIVE.COM 
Streaming Media" software are you trying to run?

         Ross.



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X-Original-Date: Fri, 2 Aug 2002 16:03:13 -0400
Date: Fri, 2 Aug 2002 16:03:13 -0400

Hello,

I'm going through the LiveMedia Code for the openrtsp application and I've a
question regarding Quick Time files served by Darwin Streaming Server. When
I send a DESCRIBE command to the server for a simple .mov file, I get a
response indicating that the audio track is sampled at 22050 Hz.When I play
the media, the RTP timestamp is incremented by 2048 for each
audio packet. The RTP encoding for QuickTime Media spec
http://developer.apple.com/quicktime/icefloe/dispatch026.html) states that
RTP Timestamp is incremented according to the current time scale and the
first RTP packet states that the current timescale is 0x5622(=22050). An
increment of 2048 would correspond to about 92 ms(2048 * 1000 /22050). But I
see packets are generated roughly at the rate of one every 9-10 ms. Can
somebody tell me where I'm going wrong? Does openrtsp care about this at all
while saving the media to a file?

Thanks,
Nagarjuna



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X-Original-Date: Fri, 02 Aug 2002 13:33:46 -0700
Date: Fri, 02 Aug 2002 13:33:46 -0700


>and the
>first RTP packet states that the current timescale is 0x5622(=22050).

Are you getting this from the "Timescale" field in the "QuickTime Payload 
Description Data" (inside the "QuickTime Header" at the start of each RTP 
packet)?

>  An
>increment of 2048 would correspond to about 92 ms(2048 * 1000 /22050). But I
>see packets are generated roughly at the rate of one every 9-10 ms. Can
>somebody tell me where I'm going wrong?

Could you email the result of running "openRTSP -V" on your "rtsp://" 
URL?  It's possible that the RTP timestamp frequency specified in the SDP 
"a=rtpmap" line differs from that specified in the QuickTime "timescale"...

>  Does openrtsp care about this at all
>while saving the media to a file?

Probably not.  In general, the receiving code assumes that the RTP 
timestamp frequency is that which was specified in the SDP "a=rtpmap" 
line.  However, when recording 'Generic QuickTime RTP streams' to a 
QuickTime file, "openRTSP" uses only the frequency that's specified in the 
QuickTime "timescale".  (See "QuickTimeFileSink.cpp", line 514)

         Ross.



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[Added missing subject line]

My answers are inline


-----Original Message-----
From: Ross Finlayson {[} <EMAIL-ADDRESS-HIDDEN> <EMAIL-ADDRESS-HIDDEN>
]
Sent: Friday, August 02, 2002 4:34 PM
To: <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] (no subject)



>and the
>first RTP packet states that the current timescale is 0x5622(=22050).

Are you getting this from the "Timescale" field in the "QuickTime Payload
Description Data" (inside the "QuickTime Header" at the start of each RTP
packet)?

That's right. This value is from the QuickTime Header in the beginning of
the packet

>  An
>increment of 2048 would correspond to about 92 ms(2048 * 1000 /22050). But
I
>see packets are generated roughly at the rate of one every 9-10 ms. Can
>somebody tell me where I'm going wrong?

Could you email the result of running "openRTSP -V" on your "rtsp://"
URL?  It's possible that the RTP timestamp frequency specified in the SDP
"a=rtpmap" line differs from that specified in the QuickTime "timescale"...

Actually they are the same values. It is 22050. The message is here:

Sending request: DESCRIBE rtsp://localhost/firstever.sdp RTSP/1.0
CSeq: 1
Accept: application/sdp


Received DESCRIBE response: RTSP/1.0 200 OK
Server: DSS/4.0 [v410]-Win32
Cseq: 1
Cache-Control: no-cache
Content-length: 359
Date: Fri, 02 Aug 2002 17:55:21 GMT
Expires: Fri, 02 Aug 2002 17:55:21 GMT
Content-Type: application/sdp
x-Accept-Retransmit: our-retransmit
Content-Base: rtsp://localhost/firstever.sdp/

v=0
o=QTSS_Play_List 1744863317 1744881743 IN IP4 10.10.20.113
s=C:\Program Files\Darwin Streaming Server\Playlists\TheFirstOne\T
a=x-broadcastcontrol:RTSP
t=0 0
b=AS:94
c=IN IP4 0.0.0.0
a=control:*
m=audio 0 RTP/AVP 96
b=AS:18
a=rtpmap:96 X-QT/22050/1
a=control:trackID=1
m=video 0 RTP/AVP 97
b=AS:76
a=rtpmap:97 X-QT/600
a=control:trackID=2

>  Does openrtsp care about this at all
>while saving the media to a file?

Probably not.  In general, the receiving code assumes that the RTP
timestamp frequency is that which was specified in the SDP "a=rtpmap"
line.  However, when recording 'Generic QuickTime RTP streams' to a
QuickTime file, "openRTSP" uses only the frequency that's specified in the
QuickTime "timescale".  (See "QuickTimeFileSink.cpp", line 514)

Actually, I found the answer to my question on DSS mailing list. DSS sends
packets initially at a higher rate so that the client can start playing
media quickly. After about a minute or so, the packets are sent at around
the 90 ms gap as I computed. 

As a side note, I have noticed that openRTSP does not do aggregate control
of streams even when it can. If you see the above SDP, it indicates that the
streams can be controlled as one unit. However, openRTSP sends SETUP for the
streams in a way that results in 2 sessions, so it issues 2 PLAY requests
instead of one that should be enough here.

 
Thanks,
nagarjuna



         Ross.



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<P><FONT size=2>[Added missing subject line]</FONT></P>
<P><FONT size=2>My answers are <FONT 
color=#0000ff>inline<BR></FONT><BR><BR>-----Original Message-----<BR>From: Ross 
Finlayson [<A 
href="mailto:finlayson(at)live.com">mailto:finlayson(at)live.com</A>]<BR>Sent: Friday, 
August 02, 2002 4:34 PM<BR>To: live-devel(at)lists.sourceforge.net<BR>Subject: Re: 
[Live-devel] (no subject)<BR><BR><BR><BR>&gt;and the<BR>&gt;first RTP packet 
states that the current timescale is 0x5622(=22050).<BR><BR>Are you getting this 
from the "Timescale" field in the "QuickTime Payload<BR>Description Data" 
(inside the "QuickTime Header" at the start of each RTP<BR>packet)?</FONT></P>
<DIV><FONT size=2><FONT face=Arial color=#0000ff>That's right. This value is 
from the QuickTime Header in the beginning of the packet</FONT></DIV><FONT 
face=Arial color=#0000ff></FONT>
<P><BR>&gt;&nbsp; An<BR>&gt;increment of 2048 would correspond to about 92 
ms(2048 * 1000 /22050). But I<BR>&gt;see packets are generated roughly at the 
rate of one every 9-10 ms. Can<BR>&gt;somebody tell me where I'm going 
wrong?<BR><BR>Could you email the result of running "openRTSP -V" on your 
"rtsp://"<BR>URL?&nbsp; It's possible that the RTP timestamp frequency specified 
in the SDP<BR>"a=rtpmap" line differs from that specified in the QuickTime 
"timescale"...<BR><BR><FONT face=Arial color=#0000ff>Actually they are the same 
values. It is 22050. The&nbsp;message is here:</FONT></P>
<P><FONT face=Arial color=#0000ff>Sending request: DESCRIBE 
rtsp://localhost/firstever.sdp RTSP/1.0<BR>CSeq: 1<BR>Accept: 
application/sdp</FONT></P><FONT face=Arial color=#0000ff>
<P><BR>Received DESCRIBE response: RTSP/1.0 200 OK<BR>Server: DSS/4.0 
[v410]-Win32<BR>Cseq: 1<BR>Cache-Control: no-cache<BR>Content-length: 
359<BR>Date: Fri, 02 Aug 2002 17:55:21 GMT<BR>Expires: Fri, 02 Aug 2002 17:55:21 
GMT<BR>Content-Type: application/sdp<BR>x-Accept-Retransmit: 
our-retransmit<BR>Content-Base: rtsp://localhost/firstever.sdp/</P>
<P>v=0<BR>o=QTSS_Play_List 1744863317 1744881743 IN IP4 
10.10.20.113<BR>s=C:\Program Files\Darwin Streaming 
Server\Playlists\TheFirstOne\T<BR>a=x-broadcastcontrol:RTSP<BR>t=0 
0<BR>b=AS:94<BR>c=IN IP4 0.0.0.0<BR>a=control:*<BR>m=audio 0 RTP/AVP 
96<BR>b=AS:18<BR>a=rtpmap:96 X-QT/22050/1<BR>a=control:trackID=1<BR>m=video 0 
RTP/AVP 97<BR>b=AS:76<BR>a=rtpmap:97 X-QT/600<BR>a=control:trackID=2</FONT></P>
<P>&gt;&nbsp; Does openrtsp care about this at all<BR>&gt;while saving the media 
to a file?<BR><BR>Probably not.&nbsp; In general, the receiving code assumes 
that the RTP<BR>timestamp frequency is that which was specified in the SDP 
"a=rtpmap"<BR>line.&nbsp; However, when recording 'Generic QuickTime RTP 
streams' to a<BR>QuickTime file, "openRTSP" uses only the frequency that's 
specified in the<BR>QuickTime "timescale".&nbsp; (See "QuickTimeFileSink.cpp", 
line 514)</P>
<P><FONT face=Arial color=#0000ff>Actually, I found the answer to&nbsp;my 
question on DSS mailing list. DSS sends packets initially at a higher rate so 
that the client can start playing media quickly. After about a minute or so, the 
packets are sent at&nbsp;around the 90 ms&nbsp;gap as I computed. </FONT></P>
<P><FONT face=Arial color=#0000ff>As a side note, I have noticed that openRTSP 
does not do aggregate control of streams even when it can. If you see the above 
SDP, it indicates that the streams can be controlled as one unit. However, 
openRTSP sends SETUP for the streams in a way that results in 2 sessions, so it 
issues 2 PLAY requests instead of one that should be enough here.</FONT></P>
<DIV><FONT face=Arial color=#0000ff></FONT>&nbsp;</DIV>
<DIV><FONT face=Arial color=#0000ff>Thanks,</FONT></DIV>
<DIV><FONT face=Arial color=#0000ff>nagarjuna</FONT></DIV>
<P><BR><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
Ross.<BR><BR><BR><BR>-------------------------------------------------------<BR>This 
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Date: Fri, 02 Aug 2002 14:03:36 -0700


>As a side note, I have noticed that openRTSP does not do aggregate control 
>of streams even when it can. If you see the above SDP, it indicates that 
>the streams can be controlled as one unit. However, openRTSP sends SETUP 
>for the streams in a way that results in 2 sessions, so it issues 2 PLAY 
>requests instead of one that should be enough here.

Yes, you're probably right.  However, I tried to make openRTSP use the RTSP 
protocol  as conservatively as possible, to avoid possible problems with 
non-compliant servers.

         Ross.



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From: Niranjan <EMAIL-ADDRESS-HIDDEN>
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Subject: [Live-devel] Regarding streaming MPEG transport stream !!
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X-Original-Date: Sun, 04 Aug 2002 10:00:54 +0000 (GMT)
Date: Sun, 04 Aug 2002 10:00:54 +0000 (GMT)

 =

=3Chtml=3E=3Cnobr=3E=3Cimg src=3Dhttp=3A//maildb=2Esamsung=2Ecom=3A8001/o=
penconfirm=3Fparm=3Dniranjan=5F=5Fniranjan20010804190619194 width=3D=221=22=
 height=3D=221=22=3E=3C/nobr=3E
=3Cfont size=3D3 face=3D=22=B1=BC=B8=B2=C3=BC=22=3E =3Cpre=3EHi Ross=2C

I have done the modifications you had suggested in the previous reply=2E
At the streaming end ie =2E using the testMPEGVideoStreamer as a model=2C=


I created a ByteStreamFileSource object to read 188 chunk bytes of data =2E=

ByteStreamFileSource* fileSource =3D ByteStreamFileSource=3A=3AcreateNew(=
*env=2Cfilename=2C188)=3B

and also a video sink object as=2C
videoSink =3D SimpleRTPSink(*env=2C=26rtpGroupSock=2C33=2C90000=2C=22appl=
ication/mp2ts=22)=3B

In the play function of testMPEGVideoStreamer=2Ecpp in the last line =2C =
ie
videoSink-=3EstartPlaying(*videoSource=2CafterPlaying=2CvideoSink)=3B is =
used for a program stream
I modified the first parameter for the transport stream and i am passing =
the fileSource itself =2C
videoSink-=3EstartPlaying(*fileSource=2CafterPlaying=2CvideoSink)=3B is t=
his right =3F=3F

And at the receiving end ie (using openRTSP) =2C I made the following add=
itions=2C
In the MediaSession=2Ecpp =2C in the initiate method i am creating a new =
SimpleRTPSource =

object for the codec MP2T as below=2C
fReadSource =3D fRTPSource =3D SimpleRTPSource=3A=3AcreateNew(env()=2CfRT=
PSocket=2C33=2C90000=2C=22application/mp2ts=22) =

=22)=3B

I guess these are the changes you had mentioned =2E
But When i run the testMPEGVideoStreamer file with a sample test=2Empg wh=
ich is a transport stream file=2C
it just says =2C
=22Beginning to read the file =22
=22=2E=2E=2E=2E=2E done reading the file=22
and when i run openRTSP no streams are received at all =2E When i debugge=
d =2C
the DESCRIBE request is sent =2C but no response is received=2E
The RTSP server is enabled in the testMPEGVideoStreamer=2E

What might be wrong with the way i am doing for the transport stream=2E
Please Guide me in this =2E

A sample transport stream url =2C http=3A//www=2Einterra=2Etv/products/mp=
dnld=2Ehtm
here u can download sample transport stream files=2E

Please do try this if you have time=2E
thanks in advance=2C
niranjan=2E


=3C/font=3E=3C/pre=3E=3C/html=3E



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Subject: [Live-devel] FYI: Updated RTSP/ RTP streaming patch for "mplayer"
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X-Original-Date: Sun, 04 Aug 2002 16:40:23 -0700
Date: Sun, 04 Aug 2002 16:40:23 -0700

I have recently updated my patch for the Open Source "mplayer" media player 
(for Linux or FreeBSD), that allows it to receive and play RTP audio/video 
streams, using the "LIVE.COM Streaming Media" libraries.

This newest version now includes RTSP client support (using a "rtsp://" 
URL), as well as being able to read SDP files.  I have also made several 
improvements to the mplayer-specific RTP buffering code, so that incoming 
RTP packets are less likely to be dropped.

For more information, see
         <http://www.live.com/mplayer/>

There's a good chance that these changes will eventually be merged into the 
official "mplayer" sources, but for now, you'll need to obtain the patch 
from the URL above.

         Ross.




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X-Original-Date: Sun, 04 Aug 2002 23:54:08 -0700
Date: Sun, 04 Aug 2002 23:54:08 -0700

An update to my earlier message:

The maintainers of the "mplayer" code have now added my patch to their 
official source distribution, so you will no longer need to patch the code 
yourself.  (You will need to use their "Latest CVS snapshot".)

Once again, instructions for configuring this new version of "mplayer" can 
be found at:
	<http://www.live.com/mplayer/>

	Ross.



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X-Original-Date: Mon, 05 Aug 2002 03:23:17 -0700
Date: Mon, 05 Aug 2002 03:23:17 -0700

In my previous messages, I'd made a couple of mistakes about this.  Here, 
now, are what (I hope) are correct instructions for streaming a MPEG 
Transport Stream.

- First, make sure you're using the most up-to-date version of the 
libraries (2002.06.25).

- Create your file source as follows:
	ByteStreamFileSource::createNew(*env, inputFileName, 188, 0.001);

The "0.001" parameter is the time taken, in seconds, by each 188-byte MPEG 
Transport Stream frame.  Replace this with the correct value for your 
stream.  This parameter tells the streaming code how much time to delay 
after sending each RTP packet.  (If you are streaming live input - i.e., if 
your input file is 'stdin' - then you can omit this parameter.)

- Create your RTP sink as follows:
	SimpleRTPSink::createNew(*env, &rtpGroupsock, 33, 90000,"application", 
"mp2t");

Note that "application" and "mp2t" are in separate parameters, and that 
it's "mp2t", *not* "mp2ts".

- If you do this, you should be able to stream a MPEG Transport Stream 
using RTP.

- Note that you *should not* need to make any changes to "openRTSP".  It is 
already set up to be able to handle MPEG Transport Stream RTP streams.

I hope this helps...

	Ross.



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X-Original-Date: Wed, 07 Aug 2002 02:55:32 -0400
Date: Wed, 07 Aug 2002 02:55:32 -0400

Hello,

I ran the testMPEGAudioVideoStreamer test program and then using Quicktime on 2 different PCs, I opened the testMPEGAudioVideo.sdp file and viola, I could see the Mpg file(video and audio) being broadcast live on both the PCs. 

Then, I paused the video/audio playback on 2nd PC. Then after 20-30 secs, I un-paused and resumed playback, however, I realised that there was a significant delay between the video/audio playback on the 1st PC and that displayed on the 2nd PC... The 2nd PC was lagging the 1st PC by about 10-20 secs.

My question is, for multicast, shouldn't the Quicktime player on the 2nd PC be getting the latest Audio and video data from the RTP address e.g. 239.255.42.42, ports 8888 and 6666? i.e. when I un-paused the video/audio playback on the 2nd PC after 20-30 secs, why does it still lag behind the 1st PC by around 10-20 secs? 

Is it due to buffering of the data in Quicktime? if so, is there any way to remedy this problem?

Thanks,

Joseph






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X-Original-Date: Wed, 07 Aug 2002 00:12:24 -0700
Date: Wed, 07 Aug 2002 00:12:24 -0700


>My question is, for multicast, shouldn't the Quicktime player on the 2nd 
>PC be getting the latest Audio and video data from the RTP address e.g. 
>239.255.42.42, ports 8888 and 6666?

Yes, because the stream is multicast, each RTP packet will be arriving at 
each PC at the same time.

>  i.e. when I un-paused the video/audio playback on the 2nd PC after 20-30 
> secs, why does it still lag behind the 1st PC by around 10-20 secs?
>
>Is it due to buffering of the data in Quicktime?

Yes, and/or buffering inside the PCs' operating system.

>  if so, is there any way to remedy this problem?

I don't know of any, but you might look to see if there's any way to reduce 
buffering in QT Player or your OS.  (For the former, you might want to ask 
on one of Apple's Quick Time mailing lists.)

         Ross.



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X-Original-Date: Wed, 07 Aug 2002 03:31:30 -0400
Date: Wed, 07 Aug 2002 03:31:30 -0400

Hello,

I am a newcomer to this mailing list, and I was just looking through the old archive and saw something that I am also trying to accomplish.

I am also trying to Capture & Stream Live Audio/Video...however, I'm trying to incorporate the audio from the mic connected to the sound card together with the video extracted from the web-cam... has anyone succeeded in doing this before? 

Thanks.

Joseph

P.S: Mr manoj someone singh, have u succeeded in capturing and streaming live audio/video successful? could you email me? Thanks!



---------------------------------------------------------------------
Message: 9022363 
FROM: manoj someone singh
DATE: 06/25/2002 02:23:37
SUBJECT:  [Live-devel] Capture & Stream Live Audio/Video

Dear Sir,

The test programs available with the RTP Library, accept data for 
.mp3 or .mpg file. I want to capure live feed form a web-cam and 
give it to the streamer. I can use the DirectShow API to capture 
video from the web-cam and display it on the rendering surface 
maybe the screen or a file. But how do I combine it with the test 
programs.

The DirectX API is COM based, while the test programs are written 
in plain C++. How do I go about using this feature. What I need is 
to develop an application using Microsoft VC++ and the RTP 
Library, which can Stream (Broadcast/multicast) live audio/video 
over the internet.

Please provide me with the necessary guidelines.

Thanks in advance.

Regards.

Manoj.

 



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X-Original-Date: Wed, 07 Aug 2002 13:42:20 -0700
Date: Wed, 07 Aug 2002 13:42:20 -0700


>I am a newcomer to this mailing list, and I was just looking through the 
>old archive and saw something that I am also trying to accomplish.
>
>I am also trying to Capture & Stream Live Audio/Video...however, I'm 
>trying to incorporate the audio from the mic connected to the sound card 
>together with the video extracted from the web-cam... has anyone succeeded 
>in doing this before?

Joseph,

Here, FYI, is the response that I sent to "manoj someone singh"s earlier 
message.  The short answer is that the streaming media framework was 
designed to allow adding grabbing/encoding components to it, but quite a 
bit of work is involved.  (There aren't yet any grabber/encoder components 
that are publically available.)

--------- my earlier response -----
First, note that the "test programs" are just that - programs that 
illustrate how to make use of the libraries.  With the exception of 
"openRTSP", they are not intended to be complete, end-user applications.

To use the libraries to stream live video from a web cam, you would need to 
write the following software components:
1/ an object (subclass of "FramedSource") that encapsulates the video frame 
grabber.  (This will have to be operating system-specific code.)
2/ an object (subclass of "FramedFilter") that encodes the video frame into 
MPEG (possibly by encapsulating an existing, binary-only encoder).

You would then assemble these components into a chain, and play them using 
a "MPEGVideoRTPSink".

If you already have MPEG encoder hardware, then instead of writing new 
software components 1/ and 2/, you can just take the output of the encoder, 
pass it through a "MPEGVideoStreamFramer", and then stream it using a 
"MPEGVideoRTPSink".  In fact, I have one client who has done this by making 
a simple modification to "testMPEGVideoStreamer.cpp": changing "test.mpg" 
to "stdin".  He then just pipes the output of his MPEG video encoder into 
the (modified) "testMPEGVideoStreamer" application.

----------
A quick follow-up to my earlier message.

If you wish to write a source object that encapsulates a video (or audio) 
grabber device, then you should take a look at the files
         liveMedia/include/DeviceSource.hh
         liveMedia/DeviceSource.cpp

These provide a template for you could implement such an object.  You would 
fill in the various member function implementations to complete this.

         Ross.



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Subject: [Live-devel] problem destructing Medium
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X-Original-Date: Fri, 9 Aug 2002 08:01:44 -0700 (PDT)
Date: Fri, 9 Aug 2002 08:01:44 -0700 (PDT)

hi Ross,
 i tried streaming mp3 files using live media on the
ipaq and it worked fine except that while destructing,

 when i call shutdown(0)
and trying to close the Medium::close(Medium *)

i am getting a segmentation fault.
t tried to trace using gdb and the result is the
following..

Program received signal SIGSEGV, Segmentation fault.
[Switching to Thread 1024 (LWP 4404)]
0x080a39f8 in ?? ()
(gdb) where
#0  0x080a39f8 in ?? ()
#1  0x400d14ef in
__static_initialization_and_destruction_0
(__initialize_p=0, 
    __priority=65535) at Media.cpp:44
#2  0x400d1526 in global destructors keyed to
Medium::Medium () at Media.cpp:79
#3  0x4005f12b in __do_global_dtors_aux () from
./libmpeg.so
#4  0x400f8389 in _fini () from ./libmpeg.so
#5  0x4000d4ea in _dl_fini () at dl-fini.c:169
#6  0x4032650b in exit (status=0) at exit.c:54
#7  0x40311630 in __libc_start_main (main=0x805c790
<main>, argc=1, 
    ubp_av=0xbffffa14, init=0x805c070 <_init>,
fini=0x807816c <_fini>, 
    rtld_fini=0x4000d2cc <_dl_fini>,
stack_end=0xbffffa0c)
    at ../sysdeps/generic/libc-start.c:129



can u please tell me what is wrong??
thank you
surya

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X-Original-Date: Fri, 09 Aug 2002 13:24:46 -0700
Date: Fri, 09 Aug 2002 13:24:46 -0700


>  when i call shutdown(0)

Surya,

Are you referring to the "shutdown()" function in "openRTSP.cpp"?

>and trying to close the Medium::close(Medium *)
>
>i am getting a segmentation fault.

Which call to "Medium::close()" (in the "shutdown()" function) is failing?

Please make sure that you're using a recent version of the "LIVE.COM 
Streaming Media" code (from <http://www.live.com/liveMedia/>).  Versions 
more than a couple of months old had a bug that might have caused the fault 
that you saw.

If you're still seeing problems with the newest version of the code, then 
please email the result of running "openRTSP" with the "-V" (verbose) flag.

         Ross.



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X-Original-Date: Tue, 13 Aug 2002 06:41:13 +0100 (BST)
Date: Tue, 13 Aug 2002 06:41:13 +0100 (BST)

Hello 

Can someone tell why jitter is being calculated only
in the case of "I" frames.
Function packetIsUsableInJitterCalculation(..) in
class
MPEGVideoRTPSource ,checks wether the frame is I or
not .

Manik

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From <EMAIL-ADDRESS-HIDDEN> Mon Aug 12 23:06:18 2002
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X-Original-Date: Mon, 12 Aug 2002 23:01:15 -0700
Date: Mon, 12 Aug 2002 23:01:15 -0700

At 10:41 PM 8/12/02, you wrote:
>Can someone tell why jitter is being calculated only
>in the case of "I" frames.
>Function packetIsUsableInJitterCalculation(..) in
>class
>MPEGVideoRTPSource ,checks wether the frame is I or
>not .

This came about as a result of a comment/suggestion that was made just over 
a year ago on the IETF's AVT (Audio Video Transport) mailing list:

----------- quoted message:
4. Subject: Jitter computation for several packets containing data from the
same frame which carry the same timestamp and packets that belong to P- and
B-frames when B-frames are used
     Background: An MPEG encoded video frame may need to be carried using
more than one RTP packet. All of such packets carry the same timestamp
although their actual sent-times have to be different. When B-frames are
used, sent-times of both P- and B-frames are not related to their timestamps
because a future P-frame must be sent prior to all B-frames that refers to
it. As a result, jitter computations done based on such packets can be
severely wrong. The first part of this problem is common to other payload
formats that carry large data samples. B-frame situation is MPEG specific.
     Proposal: Add the following note:
Since several packets containing data from the same frame carry the same
timestamp, the jitter computation for RTCP reports SHOULD be based on the
timestamps of the first packets of each frame (I- or P-) when B-frames are
not used. Otherwise, only the timestamps of the first packets of I- frames
SHOULD be used because of the specific transport order required by B-frames.
     Modifies: RTCP sender and receiver reports for MPEG1 & MPEG2 ES video
payload formats.
----------- end quoted message

To simplify the implementation, I decided to use only I-frames' timestamps 
in the jitter calculation.

         Ross.




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X-Original-Date: Tue, 13 Aug 2002 12:01:58 +0200
Date: Tue, 13 Aug 2002 12:01:58 +0200

elo,
i've read the manual and all instructions how to compile it on win32. but
when try to read .mak files borland just says "Project rule not found". What
should i do? anyone compiled it with borland ? anyone can pass binaries ?
a.



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X-Original-Date: Tue, 13 Aug 2002 10:48:06 -0700
Date: Tue, 13 Aug 2002 10:48:06 -0700

At 03:01 AM 8/13/02, you wrote:
>elo,
>i've read the manual and all instructions how to compile it on win32. but
>when try to read .mak files borland just says "Project rule not found". What
>should i do? anyone compiled it with borland ?

I assume you followed the instructions in the section "If you're using the 
Borland C++ compiler" in <http://www.live.com/liveMedia/#config-windows>.

Has anyone else had this problem when using Borland's C++ compiler?

         Ross.




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X-Original-Date: Wed, 14 Aug 2002 17:06:09 +0530
Date: Wed, 14 Aug 2002 17:06:09 +0530

hi,
I have a query regarding your RTSP code (RTSPClient.cpp etc).
Does this allow playing of multiple medias located at different servers 
in a single presentation. If yes , then do you require to make multiple 
instances of RTSPClient class or is it (can it be) handled under a 
single instance.
This is with reference to the example given in RTSP document (RFC - 
2326), section 14 "Examples", subsection 1 (ie 14.1) "Media on demand 
(Unicast)", page number 63.

thanks and regards
mukul



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Subject: Re: [Live-devel] regarding RTSP
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X-Original-Date: Wed, 14 Aug 2002 10:20:23 -0700
Date: Wed, 14 Aug 2002 10:20:23 -0700

At 04:36 AM 8/14/02, you wrote:
>hi,
>I have a query regarding your RTSP code (RTSPClient.cpp etc).
>Does this allow playing of multiple medias located at different servers in 
>a single presentation. If yes , then do you require to make multiple 
>instances of RTSPClient class

Yes.  The "RTSPClient" class encapsulates a single TCP socket number (see 
"liveMedia/include/RTSPClient.hh"), so to access more than one RTSP server 
simultaneously, you will need separate instances of "RTSPClient".  (So, the 
name "RTSPClient" is somewhat of a misnomer; something like 
"RTSPClientSession" would perhaps have been more accurate.)


>This is with reference to the example given in RTSP document (RFC - 2326), 
>section 14 "Examples", subsection 1 (ie 14.1) "Media on demand (Unicast)", 
>page number 63.

Unfortunately, the example that's given there isn't supported by the 
current code.  (In that example, the client is first getting a SDP 
description from one server, and then interacting with two different 
servers (one for audio; the other for video), based on that SDP description.

Note also, by the way, that the RTSP specification is currently being 
revised by the IETF.  If you want to ensure that you have the most 
up-to-date version of the RTSP specification, see the Internet Drafts in 
<http://www.rtsp.org/2002/drafts/>.)

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>



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Subject: [Live-devel] Regarding PAUSE usage in openRTSP
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X-Original-Date: Thu, 15 Aug 2002 10:54:47 +0000 (GMT)
Date: Thu, 15 Aug 2002 10:54:47 +0000 (GMT)

 =

=3Chtml=3E=3Cnobr=3E=3Cimg src=3Dhttp=3A//maildb=2Esamsung=2Ecom=3A8001/o=
penconfirm=3Fparm=3Dniranjan=5F=5Fniranjan20020815200040582 width=3D=221=22=
 height=3D=221=22=3E=3C/nobr=3E
=3Cfont size=3D3 face=3D=22=B1=BC=B8=B2=C3=BC=22=3E =3Cpre=3EHi =2C

I wanted to know regarding the following things=2C

1=3E Is it possible to make openRTSP support the pause option=2EIf so the=
n how can i
do it =2EAre there any major changes to be done=2EIf possible please spec=
ify the additions to be made=2E

2=3E One thing is confusing to me from the RTSP spec =2C relating to the =
pause option=2E
If once a pause is triggered from the client on to the server =2C the ser=
ver may stop sending =

the RTP packets for that session =2C but how can the client restart playi=
ng (ie release the pause)=2E
Should the client issue a play request again on that particular session t=
o which he had
paused=2E

thanks in advance=2E
Niranjan=2E=3C/font=3E=3C/pre=3E=3C/html=3E



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From <EMAIL-ADDRESS-HIDDEN> Thu Aug 15 07:48:09 2002
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From: suryakumari sagiraju <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] Regarding PAUSE usage in openRTSP
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X-Original-Date: Thu, 15 Aug 2002 07:46:50 -0700 (PDT)
Date: Thu, 15 Aug 2002 07:46:50 -0700 (PDT)

hi,
 yes, pause can be implemented. I have tried it and it
works.
The changes should be done in RTSPClient.cpp
After the pause command, play command should be issued
without specifying the range which would start the
play from the point where it paused.

surya

--- Niranjan <niranjan(at)samsung.com> wrote:

<HR>
 
<html><nobr><img
src=http://maildb.samsung.com:8001/openconfirm?parm=niranjan__niranjan20020815200040582
width="1" height="1"></nobr>
<font size=3 face="±¼¸²Ã¼"> <pre>Hi ,

I wanted to know regarding the following things,

1> Is it possible to make openRTSP support the pause
option.If so then how can i
do it .Are there any major changes to be done.If
possible please specify the additions to be made.

2> One thing is confusing to me from the RTSP spec ,
relating to the pause option.
If once a pause is triggered from the client on to the
server , the server may stop sending 
the RTP packets for that session , but how can the
client restart playing (ie release the pause).
Should the client issue a play request again on that
particular session to which he had
paused.

thanks in advance.
Niranjan.</font></pre></html>



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From <EMAIL-ADDRESS-HIDDEN> Thu Aug 15 20:54:28 2002
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X-Original-Date: 16 Aug 2002 13:21:16 +0930
Date: 16 Aug 2002 13:21:16 +0930

OK - I managed to get mplayer compiled with the live stuff.

I did not work "out of the box".  In particular the mplayer build failed
link any executables.  It could find symbols like __builtin_delete. 
There were lots of undefined references.

After a bit of playing I noticed that all the symbols were c++ related. 
So I ran the link command by hand replacing gcc3 with g++ and everything
worked.

I don't know if this is a problem with the mplayer build or my version
of gcc.






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X-Original-Date: Thu, 15 Aug 2002 21:01:48 -0700
Date: Thu, 15 Aug 2002 21:01:48 -0700


>OK - I managed to get mplayer compiled with the live stuff.
>
>I did not work "out of the box".  In particular the mplayer build failed
>link any executables.  It could find symbols like __builtin_delete.
>There were lots of undefined references.
>
>After a bit of playing I noticed that all the symbols were c++ related.
>So I ran the link command by hand replacing gcc3 with g++ and everything
>worked.
>
>I don't know if this is a problem with the mplayer build or my version
>of gcc.

Malcolm,

I'm not sure, but note that the "Makefile" in the "libmpdemux" subdirectory 
compiles "demux_rtp.cpp" (the only C++ file in the "mplayer" code) using a 
program named "c++".  So, you should make sure that you have "c++" aliased 
correctly.

         Ross.



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X-Original-Date: Fri, 16 Aug 2002 12:15:53 +0530
Date: Fri, 16 Aug 2002 12:15:53 +0530

Hello Ross,
I wanted some clarification on the RTP buffer handling being done in 
your stack implemantation for testMPEGVideoReceiver.
Firstly, all the RTP packets are being stored in the buffer according to 
the sequence number of the RTP packets (in the storepacket function). 
And when asked for presenting a stored packet, the topmost packet of the 
buffer, the head packet, is given for presentation in the getnextframe 
function. My question is how does the software decide the time of 
presentation of B frames as they are stored after the P frames in the 
RTP buffer, when they need to be presented "before" the P frames? i.e. 
the P frames will be retrieved before the B frames according to your 
buffer implementation but the presentation of P frames needs to be 
deferred till the intermediate B frames are presented. How is this being 
managed?
I can see that in your aftergettingframe function you are passing the 
presentation time in timeval structure. But I do not see how you would 
be using it as not further processing is being done with it? So please 
tell me when do you set this variable and when is this information being 
used....?
thanks in advance
Prashanth



Ross Finlayson wrote:

>
>> OK - I managed to get mplayer compiled with the live stuff.
>>
>> I did not work "out of the box".  In particular the mplayer build failed
>> link any executables.  It could find symbols like __builtin_delete.
>> There were lots of undefined references.
>>
>> After a bit of playing I noticed that all the symbols were c++ related.
>> So I ran the link command by hand replacing gcc3 with g++ and everything
>> worked.
>>
>> I don't know if this is a problem with the mplayer build or my version
>> of gcc.
>
>
> Malcolm,
>
> I'm not sure, but note that the "Makefile" in the "libmpdemux" 
> subdirectory compiles "demux_rtp.cpp" (the only C++ file in the 
> "mplayer" code) using a program named "c++".  So, you should make sure 
> that you have "c++" aliased correctly.
>
>         Ross.
>
>
>
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X-Original-Date: Fri, 16 Aug 2002 00:04:40 -0700
Date: Fri, 16 Aug 2002 00:04:40 -0700


>I wanted some clarification on the RTP buffer handling being done in your 
>stack implemantation for testMPEGVideoReceiver.
>Firstly, all the RTP packets are being stored in the buffer according to 
>the sequence number of the RTP packets (in the storepacket function). And 
>when asked for presenting a stored packet, the topmost packet of the 
>buffer, the head packet, is given for presentation in the getnextframe 
>function. My question is how does the software decide the time of 
>presentation of B frames as they are stored after the P frames in the RTP 
>buffer, when they need to be presented "before" the P frames? i.e. the P 
>frames will be retrieved before the B frames according to your buffer 
>implementation but the presentation of P frames needs to be deferred till 
>the intermediate B frames are presented. How is this being managed?

The RTP receiving software presents frames to callers in the exact order 
(based on RTP sequence numbers) in which they were sent in data 
packets.  It is up to the software that uses this - e.g., MPEG video 
playing software - to change the order of these frames, if necessary.

>I can see that in your aftergettingframe function you are passing the 
>presentation time in timeval structure. But I do not see how you would be 
>using it as not further processing is being done with it? So please tell 
>me when do you set this variable and when is this information being used....?

As you observed, the RTP receiver implementation (i.e., RTPSource) is 
currently not setting the "fPresentationTime" variable at all.  This is 
something that I plan to fix soon, because I'll need this to implement 
improved audio/video synchronization in the "mplayer" RTP receiving 
code.  (Doing this right is going to be a bit tricky, because it requires 
listening to RTP->NTP timestamp mappings in incoming RTCP Sender Reports.)

         Ross.



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From <EMAIL-ADDRESS-HIDDEN> Sun Aug 18 06:07:11 2002
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Subject: [Live-devel] continuePlaying
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X-Original-Date: Sun, 18 Aug 2002 15:02:39 +0200
Date: Sun, 18 Aug 2002 15:02:39 +0200

Hello,

I just started working with the source code (2002.08.06) and there are
several questions I have.
First, in many classes there's a recursion that - in my opinion - consumes
Stack space proportional to the number of frames received:

Boolean FileSink::continuePlaying()
	{
	if (!fSource)
		return False;
	
	fSource->getNextFrame(fBuffer, sizeof fBuffer,
			afterGettingFrame, this,
			onSourceClosure, this);
	
	return True;
	}

void FileSink::afterGettingFrame(void* clientData, unsigned
frameSize, struct timeval /*presentationTime*/)
	{
	FileSink* sink = (FileSink*) clientData;
	
	//Do something
	
	// Then try getting the next frame:
	sink->continuePlaying();
	}

So, for each frame, the functions afterGettingFrame and continuePlaying
are called but the stack space used is not freed because continuePlaying
calls afterGettingFrame indirectly.
This means that - not considering stack space used by getNextFrame() and
subsequent function calls - you're wasting at least 40 stack bytes per
frame.

The same effect, but without wasting stack space, could be achieved
through:

Boolean FileSink::continuePlaying()
        {
        while (fSource) {
        fSource->getNextFrame(fBuffer, sizeof fBuffer,
                        afterGettingFrame, this,
                        onSourceClosure, this);
		}
        return True;
        }

void FileSink::afterGettingFrame(void* clientData, unsigned
frameSize, struct timeval /*presentationTime*/)
        {
        FileSink* sink = (FileSink*) clientData;            

        //Do something
        }


Another question I asked myself was:
Why do I have to pass a function pointer to various functions instead
of a class pointer? the clientData argument could be used for better
things than passing the this pointer and casting clientData back to an
object whose method you just called is kind of ugly.
You could, for example, define an interface like

class Playhandler {
	public:
		void afterPlaying(void* clientData);
	}

and make startPlaying etc. accept a pointer to it...


Cheers,
Robert



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X-Original-Date: Sun, 18 Aug 2002 21:52:12 -0400
Date: Sun, 18 Aug 2002 21:52:12 -0400

Hi there,

Just like to check...can RTP be used on PDA devices? I know that TCP and UDP is supported for PDA devices, but I'm not too sure if RTP is support?

Does anyone know? thanks!

Joseph


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X-Original-Date: Sun, 18 Aug 2002 22:05:52 -0700
Date: Sun, 18 Aug 2002 22:05:52 -0700

At 06:52 PM 8/18/02, you wrote:
>Hi there,
>
>Just like to check...can RTP be used on PDA devices?

Absolutely.  In fact, I know of people who have used the LIVE.COM RTP 
library on a Compaq IPaq.

         Ross.



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X-Original-Date: Mon, 19 Aug 2002 11:23:40 +0000 (GMT)
Date: Mon, 19 Aug 2002 11:23:40 +0000 (GMT)

 =

=3Chtml=3E=3Cnobr=3E=3Cimg src=3Dhttp=3A//maildb=2Esamsung=2Ecom=3A8001/o=
penconfirm=3Fparm=3Dniranjan=5F=5Fniranjan20020819202933286 width=3D=221=22=
 height=3D=221=22=3E=3C/nobr=3E
=3Cfont size=3D3 face=3D=22=B1=BC=B8=B2=C3=BC=22=3E =3Cpre=3EHi surya=2C

Firstly thanks for your response=2C

Regarding the pause additions =2C i have made the following changes=2C

I have made changes to the RTSPClient file=2EI have added a new method=2C=

pauseMediaSession()=2E
Then in the BasicUsageEnvironment file in the blockMyself() method i =

wait upon an external request ie for pause request on an ipc (message que=
ue)=2E
So when ever a pause is triggered through an external application(or rath=
er a process) =2C =

the message queue in the blockMyself() method will receive it and then =

issue the pauseMediaSession() commands=2EThe pause commands are successfu=
lly =

sent to the server and accordingly the server halts the streaming =2E
Until this its fine =2C
But when i play again(ie issuse a play request )=2EI dont understand what=
 exactly =

happens =2C the message queue is not able to receive the play request=2E


How was your approach towards the pause implementation=2E
Could you please share it with me=2E =


Regards
Niranjan=3C/font=3E=3C/pre=3E=3C/html=3E



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From <EMAIL-ADDRESS-HIDDEN> Mon Aug 19 19:04:38 2002
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X-Original-Date: Tue, 20 Aug 2002 02:02:35 +0000
Date: Tue, 20 Aug 2002 02:02:35 +0000

Hi,
I am new to Live Media Library.
Can any one tell me if this library handles issues like 
reliability and flow/congestion control?

Thanks
Faruq

  


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X-Original-Date: Mon, 19 Aug 2002 22:17:47 -0700
Date: Mon, 19 Aug 2002 22:17:47 -0700

At 07:02 PM 8/19/02, you wrote:
>Hi,
>I am new to Live Media Library.
>Can any one tell me if this library handles issues like
>reliability and flow/congestion control?

Faruq,

The library implements streaming using RTP (see 
<http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-new-11.txt>), which 
is a protocol layered on top of UDP, which is an unreliable protocol.  This 
unreliability is actually a *benefit* for streaming time-sensitive media 
(like audio and video), because for such media it is usually better to 
occasionally lose packets, rather than delay their arrival (by 
retransmissions) by delivering them reliably.

Nonetheless, there is work currently underway in the IETF to define an 
optional retransmission mechanism for RTP.  This would allow RTP sessions 
to recover some packet loss, provided that it can do so within a short 
period of time.  It is possible that the library will support this 
retransmission/loss receovery in the future (especially if someone is 
interested in providing funding for this task).

As for flow/congestion control: This could be implemented on top of the 
existing library, although for multicast sessions, doing this properly is 
still basically a research problem.  For unicast sessions, however, you can 
implement flow control by having the sender look at the packet loss rates 
noted in incoming RTCP Reception Reports (coming back from the 
recipient).  The sender can adjust its streaming bitrate based on these 
reported packet loss rates.

         Ross.



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X-Original-Date: Tue, 20 Aug 2002 14:08:50 +0530
Date: Tue, 20 Aug 2002 14:08:50 +0530

Dear Ross,
Has live.com's implementation taken into consideration the issue of SSRC 
collision? As far as I can see it this issue still remains to be taken 
up in your implementation. Can you please give some clarity on this issue?
Thanks in advance
Prashanth



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X-Original-Date: Tue, 20 Aug 2002 07:52:46 -0700
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>Has live.com's implementation taken into consideration the issue of SSRC 
>collision?

No.  That part of the RTP specification has not yet been 
implemented.  (It's on the 'to do' list, though.)

         Ross.



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X-Original-Date: Tue, 20 Aug 2002 15:39:36 -0700
Date: Tue, 20 Aug 2002 15:39:36 -0700

Hi

I found some SSM suport code while poking around the test program
sources.  However there doesn't appear to be any corresponding support
code for parsing SDP files.  Is my understanding correct?  Is there a
patch or branch that has this?

thanks
kaifu


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>I found some SSM suport code while poking around the test program
>sources.  However there doesn't appear to be any corresponding support
>code for parsing SDP files.  Is my understanding correct?  Is there a
>patch or branch that has this?

Kaifu,

Not at present.  Currently, only the "multikit" SDP/SAP session browser 
<http://www.live.com/multikit/> supports the proposed "incl:" attribute for 
specifying a SSM session.  The RTSP client code doesn't yet support 
this.  (Part of the reason for this is that the "incl:" proposal is just 
that - a proposal.  The final specification that ends up going through the 
IETF standards process may end up being different.)

However, if you want to modify the existing test programs to receive SSM 
sessions, you can do so by modifying the creation of each 'Groupsock' 
object so that it takes the IP address of the SSM source as parameter 
(after the IP multicast address).  (See "groupsock/include/Groupsock.hh")

         Ross.



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From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] continuePlaying
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X-Original-Date: Tue, 20 Aug 2002 21:12:04 -0700
Date: Tue, 20 Aug 2002 21:12:04 -0700

Robert,

Thanks for the feedback.

>Boolean FileSink::continuePlaying()
>         {
>         if (!fSource)
>                 return False;
>
>         fSource->getNextFrame(fBuffer, sizeof fBuffer,
>                         afterGettingFrame, this,
>                         onSourceClosure, this);
>
>         return True;
>         }
>
>void FileSink::afterGettingFrame(void* clientData, unsigned
>frameSize, struct timeval /*presentationTime*/)
>         {
>         FileSink* sink = (FileSink*) clientData;
>
>         //Do something
>
>         // Then try getting the next frame:
>         sink->continuePlaying();
>         }

[...]

>The same effect, but without wasting stack space, could be achieved
>through:
>
>Boolean FileSink::continuePlaying()
>         {
>         while (fSource) {
>         fSource->getNextFrame(fBuffer, sizeof fBuffer,
>                         afterGettingFrame, this,
>                         onSourceClosure, this);
>                 }
>         return True;
>         }
>
>void FileSink::afterGettingFrame(void* clientData, unsigned
>frameSize, struct timeval /*presentationTime*/)
>         {
>         FileSink* sink = (FileSink*) clientData;
>
>         //Do something
>         }

Actually, this is not equivalent to the original code, because 
"getNextFrame()" doesn't *always* cause "afterGettingFrame()" to be 
called.  That happens only if a new frame is immediately available.  If, 
however, a new frame is not immediately available (e.g., because a new 
incoming RTP packet needs to arrive on the network), then the code returns 
to the loop inside "blockMyself()".  At that point either some other event 
handler is called, or else the code blocks (at the "select()") until a new 
network packet arrives.

>First, in many classes there's a recursion that - in my opinion - consumes
>Stack space proportional to the number of frames received

Yes, there is a recursion here, but the length of the recursion is limited 
by the number of outstanding frames in a network packet.  E.g., if an 
incoming RTP packets contains at most 3 frames, then there is at most 3 
levels of recursion.  After that, the call stack gets popped, and  the code 
returns to the loop inside "blockMyself()" (as noted above).


>Another question I asked myself was:
>Why do I have to pass a function pointer to various functions instead
>of a class pointer? the clientData argument could be used for better
>things than passing the this pointer and casting clientData back to an
>object whose method you just called is kind of ugly.
>You could, for example, define an interface like
>
>class Playhandler {
>         public:
>                 void afterPlaying(void* clientData);
>         }
>
>and make startPlaying etc. accept a pointer to it...

Unfortunately I don't quite follow what you're proposing here.  Can you 
give an example of how code would change if a pointer to a "Playhandler" 
class were used?

         Ross.



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From <EMAIL-ADDRESS-HIDDEN> Wed Aug 21 13:57:52 2002
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Subject: Re: [Live-devel] SSM support?
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X-Original-Date: Wed, 21 Aug 2002 13:55:59 -0700
Date: Wed, 21 Aug 2002 13:55:59 -0700

hi Ross,

Actually I was looking to integrate SSM support into mplayer, which
uses Live's libs.  However it appears that playRTPMPEG does support
SSM, so that will do for now.

Yeah I did notice that the draft for source-filter SDP is rather
dated, so i can understand your reasoning.  I'm glad the SSM support
code is there though!

thanks
kaifu


On Tue, Aug 20, 2002 at 04:31:24PM -0700, Ross Finlayson wrote:
> 
> >I found some SSM suport code while poking around the test program
> >sources.  However there doesn't appear to be any corresponding support
> >code for parsing SDP files.  Is my understanding correct?  Is there a
> >patch or branch that has this?
> 
> Kaifu,
> 
> Not at present.  Currently, only the "multikit" SDP/SAP session browser 
> <http://www.live.com/multikit/> supports the proposed "incl:" attribute for 
> specifying a SSM session.  The RTSP client code doesn't yet support 
> this.  (Part of the reason for this is that the "incl:" proposal is just 
> that - a proposal.  The final specification that ends up going through the 
> IETF standards process may end up being different.)
> 
> However, if you want to modify the existing test programs to receive SSM 
> sessions, you can do so by modifying the creation of each 'Groupsock' 
> object so that it takes the IP address of the SSM source as parameter 
> (after the IP multicast address).  (See "groupsock/include/Groupsock.hh")
> 
>          Ross.
> 
> 
> 
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X-Original-Date: Thu, 22 Aug 2002 21:32:10 -0700
Date: Thu, 22 Aug 2002 21:32:10 -0700

At 11:07 PM 8/21/02, you wrote:
>   This is regarding MPEG1 (ES-Video) streaming.
>    I would like to know about time calculation at transmitter
>    end before sending the RTP packet.
>    Do we need to calculate wait_time per RTP packet Or per frame,
>    and send the packet after waiting this much amount of time?

Rakesh,

This calculation (the wait time between successive frames) is already done 
by the "MultiFramedRTPSink" class, as each RTP packet is sent to the 
network (see "liveMedia/MultiFramedRTPSink.cpp", line 289).  This works by 
calling "getPlayTime()" on the media source.

"getPlayTime()" is a pure virtual function that is implemented by each 
source class.  In particular, it is implemented by "MPEGVideoStreamFramer".

So, if you are streaming MPEG video elementary streams over RTP (e.g., 
using the library as illustrated by the "testMPEGVideoStreamer" test 
program), then the code already spaces outgoing RTP packets correctly.

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>

ps. In the future, please send email to the mailing list address: 
"live-devel(at)lists.sourceforge.net".  (The email address that you used - 
"live-devel-admin(at)lists.sourceforge.net" - goes only to me.)



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X-Original-Date: Mon, 26 Aug 2002 13:34:04 -0700
Date: Mon, 26 Aug 2002 13:34:04 -0700

At 06:51 PM 8/25/02, Joseph Teo wrote:
>Sorry for the delayed reply, just a question, you said you know people who 
>have used the LIVE.COM RTP library on the Compaq IPaq, so, does that mean 
>that I could just compile the source codes provided on your website using 
>Embedded Visual C++ and I'll have the RTP library for PDAs?

I know of know reason why this would not work (although you may need to 
tweak the "win32config" and/or ".mak" files).

The people that I had heard of who had used the code on the iPaq had done 
so when it was running Linux, but I know of no reason why the code could 
not also be compiled to run on Wince.

         Ross.

ps. In the future, could you please use the developers' mailing list 
<live-devel(at)lists.sourceforge.net>.  For one-on-one email, I generally give 
priority to paying consulting clients.



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--0-510066413-1030414083=:30411
Content-Type: text/plain; charset=us-ascii


Hello Ross,

In one of my project requirement,i need to implement a RTSP compliant
streaming client protocol.I came across your library and i found it very 
useful for my development purpose.But before i go ahead with it , i had 
a few queries regarding the usage of the source codes.

1> Instead of me developing the same from the scratch , can i use 
some of the source code available in the library which is appropriate 
to my requirements.Can i Modify them and use them in my project or 
rather a embedded product.Is this software totally free.

2>But the source codes have LIVE NETWORKS copyrights , should i need
to maintain these copyrights even after i modify the sources.


Regards,
Arun Kumar.



---------------------------------
Do You Yahoo!?
Yahoo! Finance - Get real-time stock quotes
--0-510066413-1030414083=:30411
Content-Type: text/html; charset=us-ascii

<P>Hello Ross,</P>
<P>In one of my project requirement,i need to implement a RTSP compliant<BR>streaming client protocol.I came across your library and i found it very <BR>useful for my development purpose.But before i go ahead with it , i had <BR>a few queries regarding the usage of the source codes.</P>
<P>1&gt; Instead of me developing the same from the scratch , can i use <BR>some of the source code available in the library which is appropriate <BR>to my requirements.Can i Modify them and use them in my project or <BR>rather a embedded product.Is this software totally free.</P>
<P>2&gt;But the source codes have LIVE NETWORKS copyrights , should i need<BR>to maintain these copyrights even after i modify the sources.</P>
<P><BR>Regards,<BR>Arun Kumar.</P><p><br><hr size=1><b>Do You Yahoo!?</b><br>
<a href="http://rd.yahoo.com/finance/mailsig/new/*http://finance.yahoo.com">Yahoo! Finance</a> - Get real-time stock quotes
--0-510066413-1030414083=:30411--


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Subject: Re: [Live-devel] Hi !!
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X-Original-Date: Mon, 26 Aug 2002 20:11:33 -0700
Date: Mon, 26 Aug 2002 20:11:33 -0700


>In one of my project requirement,i need to implement a RTSP compliant
>streaming client protocol.I came across your library and i found it very
>useful for my development purpose.But before i go ahead with it , i had
>a few queries regarding the usage of the source codes.
>
>1> Instead of me developing the same from the scratch , can i use
>some of the source code available in the library which is appropriate
>to my requirements.Can i Modify them and use them in my project or
>rather a embedded product.Is this software totally free.

Arun,

This code is "free software", but it's not "public domain".  I.e., the 
software is covered by copyright, and is licensed under the GNU LGPL (see 
<http://www.gnu.org/copyleft/lesser.html>).

To summarize:
- You must not remove the copyright/licensing comment(s) that appear at the 
top of each source file.
- It is OK to use the code in a library that is linked with other, 
proprietary code - e.g., in a released product.  For example, it would be 
OK for you to release a product that has its own, proprietary GUI, but 
which is implemented by linking to one or more libraries containing the 
LIVE.COM code.  It would also be OK if your product has proprietary codecs 
(e.g., implemented as liveMedia 'source' or 'sink' objects) that are linked 
with the LIVE.COM libraries.
- However, if you make any modifications to the *LIVE.COM code itself* 
(rather than just to your own code that happens to link with it), and then 
release this (in a product), then you must make these code modifications 
available.  The simplest way to do this is just to post your modifications 
(or a URL) to this mailing list.
- Of course, if you're just playing around with the code for fun, without 
using it in a product, then you're not *required* to make your 
modifications available.  I encourage you to do so, however, especially if 
the modifications improve upon, and/or fix bugs in the original.

>2>But the source codes have LIVE NETWORKS copyrights , should i need
>to maintain these copyrights even after i modify the sources.

Yes, you must leave the copyright (and licensing) notices intact.

I hope this helps...

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>



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From <EMAIL-ADDRESS-HIDDEN> Tue Aug 27 10:06:39 2002
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X-Original-Date: Tue, 27 Aug 2002 12:55:19 -0400
Date: Tue, 27 Aug 2002 12:55:19 -0400

Hi Ross,

I'm sorry that I emailed to you privately, I pressed the reply button, and I guess it went directly to your mailbox...

Sorry for any inconvenience caused.

Well...I currently have a VC++ program that uses a MPEG4 codec to encode a video file and transmit it via TCP socket to a decoder/player which is able to decode the transmited video and display it on the PDA.

As I'm trying to do multicast, that's why I'm trying to change the TCP to RTP using your marvellous LIVE.COM RTP Library. Could you give me some pointers as to which files I shld be look at?

Can I just use the RTP library just replace the TCP socket currently in use in my program? 

Thanks

Best Regards,
Joseph




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X-Original-Date: Tue, 27 Aug 2002 12:13:05 -0700
Date: Tue, 27 Aug 2002 12:13:05 -0700


>Well...I currently have a VC++ program that uses a MPEG4 codec to encode a 
>video file and transmit it via TCP socket to a decoder/player which is 
>able to decode the transmited video and display it on the PDA.
>
>As I'm trying to do multicast, that's why I'm trying to change the TCP to 
>RTP using your marvellous LIVE.COM RTP Library. Could you give me some 
>pointers as to which files I shld be look at?
>
>Can I just use the RTP library just replace the TCP socket currently in 
>use in my program?

No, because RTP is a packet-based protocol, so you can't simply use it to 
replace TCP (which is a byte-stream protocol).

In particular, to stream data using RTP, the data first needs to be parsed 
into discrete frames.  Also, the RTP packet format is (in general) 
different for each codec that's being streamed.  E.g., MPEG-4 video RTP 
packets have a different format than MPEG-1 (or 2) video RTP packets, which 
have a different format from MPEG-1 (or 2) audio RTP packets, which have a 
different format from H.263 video RTP packets, etc. etc.  The LIVE.COM 
library currently implements the RTP payload format(s) for MPEG-1 or 2, but 
not MPEG-4 (either MPEG-4 video, or AAC audio), although that's on my 'to 
do' list.  (If your company is interested in providing funding for MPEG-4 
support in the LIVE.COM software, then please let me know.)

What I suggest you do for now is see if you can get the existing 
"testMPEGVideoStreamer" and/or "testMPEGAudioVideoStreamer" demo 
applications to run, as is, on your WinCE iPaq.  (These applications stream 
MPEG-1 or 2, but not (yet) MPEG-4.)

         Ross.



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Subject: [Live-devel] Play continuously
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X-Original-Date: Tue, 27 Aug 2002 19:08:39 -0300 (EST)
Date: Tue, 27 Aug 2002 19:08:39 -0300 (EST)

Hi,

I can't play continuously,  even when I using the option -c. Does anyone
has a idea? 

Thanks in advance
  Marisa




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X-Original-Date: Tue, 27 Aug 2002 22:21:23 -0700
Date: Tue, 27 Aug 2002 22:21:23 -0700


>I can't play continuously,  even when I using the option -c. Does anyone
>has a idea?

(I presume you're referring to "openRTSP" here.)

The "-c" option works OK for me.  Please run openRTSP with the "-V" option 
(in addition to "-c"), and mail me the output, so I can look at what might 
be going wrong.

         Ross.



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Subject: [Live-devel] New option for MPEGVideoStreamFramer: "I" frames only
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X-Original-Date: Tue, 27 Aug 2002 22:34:49 -0700
Date: Tue, 27 Aug 2002 22:34:49 -0700

I have added a new optional Boolean parameter "iFramesOnly" to 
"MPEGVideoStreamFramer::createNew()".  If this parameter is True, then the 
framer object will return only data from "I" frames.  (By default, this 
parameter is False, meaning that all frames will be returned.)  This 
parameter can be used to reduce the bandwidth of streamed MPEG (1 or 2) video.

I have also updated "testMPEGVideoStreamer.cpp" and 
"testMPEGAudioVideoStreamer.cpp" to note that this parameter can be used.

	Ross.



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From: umar janjua <EMAIL-ADDRESS-HIDDEN>
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Subject: [Live-devel] H.263
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X-Original-Date: Wed, 28 Aug 2002 14:05:49 +0500
Date: Wed, 28 Aug 2002 14:05:49 +0500

well dears, how are you.
I am looking for the support of h.263 codec with the current library.
I dont see it in the hierarchy except the RTPpayload format (RFC 2429).

IS there an implementation of H.263 
Or I need to implement it from scratch

A general Question:
If I am to send both video and audio ( that are related , like in movies ) ,
can RTP module keep them synchronised.
If RTP cannot , then who else?

regards,
umar


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From <EMAIL-ADDRESS-HIDDEN> Wed Aug 28 01:46:35 2002
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X-Original-Date: Wed, 28 Aug 2002 01:32:18 -0700
Date: Wed, 28 Aug 2002 01:32:18 -0700

At 02:05 AM 8/28/02, umar janjua wrote:
>I am looking for the support of h.263 codec with the current library.
>I dont see it in the hierarchy except the RTPpayload format (RFC 2429).

The library supports RTP streaming of the 1998 version of H.263 video 
(commonly called H.263+).  Note the files
         liveMedia/H263plusVideoRTPSource.cpp
         liveMedia/include/H263plusVideoRTPSource.hh
         liveMedia/H263plusVideoRTPSink.cpp
         liveMedia/include/H263plusVideoRTPSink.hh

Of course, the library does not contain a H.263 *encoder* or *decoder* (or 
encoders/decoders for any other codec).  But if you have the means to 
encoder or decode H.263, then the library code can stream it using RTP.

>A general Question:
>If I am to send both video and audio ( that are related , like in movies ) ,
>can RTP module keep them synchronised.

Yes, audio/video synchronization is performed using the RTP timestamps in 
each stream, and, in particular, the NTP->RTP timestamp mapping that's 
included in periodic RTCP Sender Report ("SR") packets.  See the current 
RTP/RTCP specification
         http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-new-11.txt
for details.

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>




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From <EMAIL-ADDRESS-HIDDEN> Wed Aug 28 02:14:01 2002
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From: umar janjua <EMAIL-ADDRESS-HIDDEN>
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Subject: RE: [Live-devel] H.263
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X-Original-Date: Wed, 28 Aug 2002 15:07:19 +0500
Date: Wed, 28 Aug 2002 15:07:19 +0500

Thank you for your message.
I have some foolish:) confusions.
Who is going to be responsible for making H.263+header for h.263 data.
I think an encoder is in the best position to do so.
What about the VRC code?

regards,
umar


-----Original Message-----
From: <EMAIL-ADDRESS-HIDDEN>
[mailto:live-devel-admin(at)lists.sourceforge.net]On Behalf Of Ross
Finlayson
Sent: Wednesday, August 28, 2002 1:32 PM
To: <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] H.263


At 02:05 AM 8/28/02, umar janjua wrote:
>I am looking for the support of h.263 codec with the current library.
>I dont see it in the hierarchy except the RTPpayload format (RFC 2429).

The library supports RTP streaming of the 1998 version of H.263 video 
(commonly called H.263+).  Note the files
         liveMedia/H263plusVideoRTPSource.cpp
         liveMedia/include/H263plusVideoRTPSource.hh
         liveMedia/H263plusVideoRTPSink.cpp
         liveMedia/include/H263plusVideoRTPSink.hh

Of course, the library does not contain a H.263 *encoder* or *decoder* (or 
encoders/decoders for any other codec).  But if you have the means to 
encoder or decode H.263, then the library code can stream it using RTP.

>A general Question:
>If I am to send both video and audio ( that are related , like in movies )
,
>can RTP module keep them synchronised.

Yes, audio/video synchronization is performed using the RTP timestamps in 
each stream, and, in particular, the NTP->RTP timestamp mapping that's 
included in periodic RTCP Sender Report ("SR") packets.  See the current 
RTP/RTCP specification
         http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-new-11.txt
for details.

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>




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From <EMAIL-ADDRESS-HIDDEN> Wed Aug 28 12:42:42 2002
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From: Marisa Affonso Vasconcelos <EMAIL-ADDRESS-HIDDEN>
To: <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] Play continuously
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X-Original-Date: Wed, 28 Aug 2002 16:35:50 -0300 (EST)
Date: Wed, 28 Aug 2002 16:35:50 -0300 (EST)

Hi,

This is the output. 

Marisa
-----------------------------------------------------------------------------------------------------------


Sending request: DESCRIBE rtsp://bigorna.speed.dcc.ufmg.br/listavideo
RTSP/1.0
CSeq: 1
Accept: application/sdp


SEND REQUEST:   DESCRIBE rtsp://bigorna.speed.dcc.ufmg.br/listavideo
RTSP/1.0
CSeq: 1
Accept: application/sdp


Received DESCRIBE response: RTSP/1.0 200 OK
Server: DSS/4.0 [v410]-Linux
Cseq: 1
Cache-Control: no-cache
Content-length: 306
Date: Wed, 28 Aug 2002 19:14:55 GMT
Expires: Wed, 28 Aug 2002 19:14:55 GMT
Content-Type: application/sdp
x-Accept-Retransmit: our-retransmit
Content-Base: rtsp://bigorna.speed.dcc.ufmg.br/listavideo/

v=0
o=QTSS_Play_List 2884137958 1926779461 IN IP4 150.164.3.150
s=listavideo
a=x-broadcastcontrol:TIME
t=0 0
b=AS:94
c=IN IP4 0.0.0.0
a=control:*
m=audio 0 RTP/AVP 96
b=AS:18
a=rtpmap:96 X-QT/22050/1
a=control:trackID=1
m=video 0 RTP/AVP 97
b=AS:76
a=rtpmap:97 X-QT/600
a=control:trackID=2

Opened URL "rtsp://bigorna.speed.dcc.ufmg.br/listavideo", returning a SDP
description:
v=0
o=QTSS_Play_List 2884137958 1926779461 IN IP4 150.164.3.150
s=listavideo
a=x-broadcastcontrol:TIME
t=0 0
b=AS:94
c=IN IP4 0.0.0.0
a=control:*
m=audio 0 RTP/AVP 96
b=AS:18
a=rtpmap:96 X-QT/22050/1
a=control:trackID=1
m=video 0 RTP/AVP 97
b=AS:76
a=rtpmap:97 X-QT/600
a=control:trackID=2

Created receiver for "audio/X-QT" subsession (client ports 3952-3953)
Created receiver for "video/X-QT" subsession (client ports 3954-3955)
Sending request: SETUP
rtsp://bigorna.speed.dcc.ufmg.br/listavideo/trackID=1 RTSP/1.0
CSeq: 2
Transport: RTP/AVP;unicast;client_port=3952-3953


SEND REQUEST:   SETUP
rtsp://bigorna.speed.dcc.ufmg.br/listavideo/trackID=1 RTSP/1.0
CSeq: 2
Transport: RTP/AVP;unicast;client_port=3952-3953


Received SETUP response: RTSP/1.0 200 OK
Server: DSS/4.0 [v410]-Linux
Cseq: 2
Cache-Control: no-cache
Session: 7337309760789705378
Date: Wed, 28 Aug 2002 19:14:55 GMT
Expires: Wed, 28 Aug 2002 19:14:55 GMT
Transport: RTP/AVP;unicast;client_port=3952-3953;source=150.164.3.150;server_port=6972-6973;ssrc=32A92DA4


Issued RTSP "SETUP" on "audio/X-QT" subsession (client ports 3952-3953)
Sending request: SETUP
rtsp://bigorna.speed.dcc.ufmg.br/listavideo/trackID=2 RTSP/1.0
CSeq: 3
Transport: RTP/AVP;unicast;client_port=3954-3955


SEND REQUEST:   SETUP
rtsp://bigorna.speed.dcc.ufmg.br/listavideo/trackID=2 RTSP/1.0
CSeq: 3
Transport: RTP/AVP;unicast;client_port=3954-3955


Received SETUP response: RTSP/1.0 200 OK
Server: DSS/4.0 [v410]-Linux
Cseq: 3
Cache-Control: no-cache
Session: 7337309760503825625
Date: Wed, 28 Aug 2002 19:14:55 GMT
Expires: Wed, 28 Aug 2002 19:14:55 GMT
Transport: RTP/AVP;unicast;client_port=3954-3955;source=150.164.3.150;server_port=6970-6971;ssrc=68AE14A3


Issued RTSP "SETUP" on "video/X-QT" subsession (client ports 3954-3955)
Created output file: "audio-X-QT"
Created output file: "video-X-QT"
Sending request: PLAY rtsp://bigorna.speed.dcc.ufmg.br/listavideo RTSP/1.0
CSeq: 4
Session: 7337309760789705378
Range: npt=0-


SEND REQUEST:   PLAY rtsp://bigorna.speed.dcc.ufmg.br/listavideo RTSP/1.0
CSeq: 4
Session: 7337309760789705378
Range: npt=0-


Received PLAY response: RTSP/1.0 200 OK
Server: DSS/4.0 [v410]-Linux
Cseq: 4
Session: 7337309760789705378
RTP-Info: url=trackID=1


Issued RTSP "PLAY" command on "audio/X-QT" subsession
Sending request: PLAY rtsp://bigorna.speed.dcc.ufmg.br/listavideo RTSP/1.0
CSeq: 5
Session: 7337309760503825625
Range: npt=0-


SEND REQUEST:   PLAY rtsp://bigorna.speed.dcc.ufmg.br/listavideo RTSP/1.0
CSeq: 5
Session: 7337309760503825625
Range: npt=0-

Received PLAY response: RTSP/1.0 200 OK
Server: DSS/4.0 [v410]-Linux
Cseq: 5
Session: 7337309760503825625
RTP-Info: url=trackID=2


Issued RTSP "PLAY" command on "video/X-QT" subsession
Receiving streamed data...








>I can't play continuously,  even when I using the option -c. Does anyone
>has a idea?

>(I presume you're referring to "openRTSP" here.)

>The "-c" option works OK for me.  Please run openRTSP with the "-V" option
>(in addition to "-c"), and mail me the output, so I can look at what
>might be going wrong.

         Ross.




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X-Original-Date: Wed, 28 Aug 2002 13:01:42 -0700
Date: Wed, 28 Aug 2002 13:01:42 -0700

The problem(?) here is that your RTSP server does not specify an end time 
(in the SDP description that it returns in response to a RTSP "DESCRIBE" 
command).  So, "openRTSP" does not know when to stop - it will receive data 
indefinitely.

So, for this session, if you *really* want to use the "-c" option, then you 
should do so in combination with the "-e <seconds-to-play>" option

E.g.,
		-c -e 30
will tell "openRTSP" to play the stream for 30 seconds, then restart the 
stream and play it for another 30 seconds, etc.

	Ross.



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Subject: RE: [Live-devel] H.263
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X-Original-Date: Wed, 28 Aug 2002 16:12:58 -0700
Date: Wed, 28 Aug 2002 16:12:58 -0700

At 03:07 AM 8/28/02, you wrote:
>Thank you for your message.
>I have some foolish:) confusions.
>Who is going to be responsible for making H.263+header for h.263 data.

The "H263plusVideoRTPSink" class fills in the 16-bit H.263+ RTP payload 
header (as described in RFC 2429, section 4.1).  However, it doesn't 
(currently) include the optional 8-bit VRC field (that would follow this 
payload header).  (So, the "V" flag in the 16-bit header is set to 0.)

>What about the VRC code?

If you wanted to include this field, you would need to modify the 
implementation of 
"H263plusVideoRTPSink::doSpecialFrameHandling()".  Presumably, it would 
call back its source (encoder or framer) object in some H.263+-specific 
way, to get whatever information that it needed in order to set this field.

         Ross.



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X-Original-Date: Thu, 29 Aug 2002 09:49:38 +0500
Date: Thu, 29 Aug 2002 09:49:38 +0500

Its encouraging to hear replies so precise and well in time.

I suggest making a TODO list for adding a new codec type and respective 
streaming RTP class with the description of interaction between the actual
encoder and the RTP streamer.

Your testprograms do give some indication on that , but a formal how to do
would 
certainly matter a lot.



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X-Original-Date: Thu, 29 Aug 2002 15:39:34 +0500
Date: Thu, 29 Aug 2002 15:39:34 +0500


I have run a testMPEG streamer  programm and specified the local IP
"0.0.0.0" and 
enabled the RTSP server.

With the Apple's Quick Time player,
i  open the url
"rtsp://localhost:554"
but it returns the bad request error.


I just want to stream a .mpg file using RTP to my own PC and want to view it
-----Original Message-----
From: <EMAIL-ADDRESS-HIDDEN>
[mailto:live-devel-admin(at)lists.sourceforge.net]On Behalf Of umar janjua
Sent: Thursday, August 29, 2002 9:50 AM
To: <EMAIL-ADDRESS-HIDDEN>
Subject: [Live-devel] an idea


Its encouraging to hear replies so precise and well in time.

I suggest making a TODO list for adding a new codec type and respective 
streaming RTP class with the description of interaction between the actual
encoder and the RTP streamer.

Your testprograms do give some indication on that , but a formal how to do
would 
certainly matter a lot.



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X-Original-Date: Thu, 29 Aug 2002 04:20:22 -0700
Date: Thu, 29 Aug 2002 04:20:22 -0700

At 03:39 AM 8/29/02, you wrote:
>I have run a testMPEG streamer  programm and specified the local IP
>"0.0.0.0" and
>enabled the RTSP server.

No, that won't work (for several reasons, including the fact that the RTSP 
server implementation currently doesn't work for unicast streams).  You 
should just leave "destinationAddressStr" the way it is now - i.e., as a 
multicast address.  Remember, multicast is your friend :-)

>I just want to stream a .mpg file using RTP to my own PC and want to view it

Just continue to stream to a multicast IP address.  Then, from your local 
PC run QuickTime Player, and give it the URL
         rtsp://localhost/

This should work.

         Ross.




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X-Original-Date: Thu, 29 Aug 2002 04:51:02 -0700
Date: Thu, 29 Aug 2002 04:51:02 -0700

At 09:49 PM 8/28/02, you wrote:
>Its encouraging to hear replies so precise and well in time.
>
>I suggest making a TODO list for adding a new codec type and respective
>streaming RTP class with the description of interaction between the actual
>encoder and the RTP streamer.
>
>Your testprograms do give some indication on that , but a formal how to do
>would
>certainly matter a lot.

Here's something that may help for now.  It's a repeat of a message that I 
sent out earlier this month.

         Ross.

----------------------
>I am a newcomer to this mailing list, and I was just looking through the 
>old archive and saw something that I am also trying to accomplish.
>
>I am also trying to Capture & Stream Live Audio/Video...however, I'm 
>trying to incorporate the audio from the mic connected to the sound card 
>together with the video extracted from the web-cam... has anyone succeeded 
>in doing this before?

Joseph,

Here, FYI, is the response that I sent to "manoj someone singh"s earlier 
message.  The short answer is that the streaming media framework was 
designed to allow adding grabbing/encoding components to it, but quite a 
bit of work is involved.  (There aren't yet any grabber/encoder components 
that are publically available.)

--------- my earlier response -----
First, note that the "test programs" are just that - programs that 
illustrate how to make use of the libraries.  With the exception of 
"openRTSP", they are not intended to be complete, end-user applications.

To use the libraries to stream live video from a web cam, you would need to 
write the following software components:
1/ an object (subclass of "FramedSource") that encapsulates the video frame 
grabber.  (This will have to be operating system-specific code.)
2/ an object (subclass of "FramedFilter") that encodes the video frame into 
MPEG (possibly by encapsulating an existing, binary-only encoder).

You would then assemble these components into a chain, and play them using 
a "MPEGVideoRTPSink".

If you already have MPEG encoder hardware, then instead of writing new 
software components 1/ and 2/, you can just take the output of the encoder, 
pass it through a "MPEGVideoStreamFramer", and then stream it using a 
"MPEGVideoRTPSink".  In fact, I have one client who has done this by making 
a simple modification to "testMPEGVideoStreamer.cpp": changing "test.mpg" 
to "stdin".  He then just pipes the output of his MPEG video encoder into 
the (modified) "testMPEGVideoStreamer" application.

----------
A quick follow-up to my earlier message.

If you wish to write a source object that encapsulates a video (or audio) 
grabber device, then you should take a look at the files
         liveMedia/include/DeviceSource.hh
         liveMedia/DeviceSource.cpp

These provide a template for you could implement such an object.  You would 
fill in the various member function implementations to complete this.

         Ross.



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X-Original-Date: Thu, 29 Aug 2002 09:51:45 -0700
Date: Thu, 29 Aug 2002 09:51:45 -0700

FYI, I have installed a new version of the code that updates the 
"liveMedia" library.  There is a minor API change, that most of you will 
not notice.  However, if you have developed your own subclass(es) of 
"MultiFramedRTPSource", then note that the signature of the virtual 
function "processSpecialHeader()" has changed - it now includes a Boolean 
parameter indicating whether or not the packet's RTP marker ("M") bit has 
been set.

In another change, I have added an optional "applicationName" parameter to 
"RTSPClient::createNew()".  This allows RTSP clients to specify the name of 
their application - to be used in RTSP "User-Agent:" headers that are 
included in RTSP commands.  I have also updated "openRTSP" and the 
"in_rtp.dll" Winamp plugin to pass this parameter.

	Ross.



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From: Bertrand Baudet <EMAIL-ADDRESS-HIDDEN>
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Subject: Re: [Live-devel] New "liveMedia" library, with minor API changes
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X-Original-Date: Thu, 29 Aug 2002 14:31:38 -0700
Date: Thu, 29 Aug 2002 14:31:38 -0700

> FROM: Ross Finlayson
> DATE: 08/29/2002 09:51:45
> SUBJECT:  [Live-devel] New "liveMedia" library, with minor API changes
> 
>  
> 
> 
> FYI, I have installed a new version of the code that updates the 
> "liveMedia" library.  There is a minor API change, that most of you will 
> not notice.  However, if you have developed your own subclass(es) of 
> "MultiFramedRTPSource", then note that the signature of the virtual 
> function "processSpecialHeader()" has changed - it now includes a Boolean 
> parameter indicating whether or not the packet's RTP marker ("M") bit has 
> been set.
> 
> In another change, I have added an optional "applicationName" parameter to 
> "RTSPClient::createNew()".  This allows RTSP clients to specify the name of 
> their application - to be used in RTSP "User-Agent:" headers that are 
> included in RTSP commands.  I have also updated "openRTSP" and the 
> "in_rtp.dll" Winamp plugin to pass this parameter.

As I wrote to Ross on the mplayer-dev mailing list which can interest people
on this list also:


VERSION CHECKING:
================
Could you provide a LIVE_VERSION macro that
can be used at compilation time to use the proper version of the LIVE API.
That help a lot to avoid bug report on broken compilation because of not
up-to-date extern lib dependencies.
Can you add this in your liveMedia.hh file for example:

#define  LIVEMEDIA_VERSION(maj,min,micro)  ((maj)*10000 + (min)*100 + (micro))
#define   LIVEMEDIA_VERSION      DVDREAD_VERSION(0,9,0)
#if  LIVEMEDIA_VERSION >=  LIVEMEDIA_VERSION(0,9,0)


ADDING EXTRA FIELDS IN THE RTSP REQUESTS:
======================================
> [Bertrand]:
> >I also need to be able to set any kind of field in all RTSP requests
> >(DESCRIBE, SETUP, PLAY...) on mplayer code side.
> 
> [Ross]:
> Why do you need to do this?  The RTSP client implementation (in the 
> LIVE.COM libraries) should be setting the RTSP protocol fields 
> correctly.  Apart from fields like "User-Agent" that are specific to the 
> particular type of client, what RTSP field(s) do you want to be able to set 
> or modify from the client, and why?
> 
[Bertrand]
To put proprietary fields like in the MS stuff. I guess QT and Real has there
own field too. So if you want to add stuff that was not meant to be done in
the RFC you can do it. I didn't say it was it was a good idea to do so, but
it can be usefull to connect on RTSP server that require those extra fields.

> This is a serious question.  Note that the RTSP client functionality 
> (implemented by the LIVE.COM libraries) are being used in many applications 
> (not just "mplayer"), so I'm interested in learning if there's any need for 
> a richer API to this code.

I think it would be nice to have a generic method to add/del field in any RTSP
requests, so the people can add the "User-Agent" field for example, without
to have to change the API.
Note that you are allow to have several Pragma fields so to remove a 
specific one that can be tricky.


Bertrand


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X-Original-Date: Fri, 30 Aug 2002 12:31:58 +0530
Date: Fri, 30 Aug 2002 12:31:58 +0530

Dear Ross,
I wanted to know how different is the implementation for a MPEG-4 
profile than MPEG-2, at a conceptual level. I am asking you this because 
you have worked with the former profile and the later is in your to-do 
list. Could you please give an insight into this?
Thanks in advance
Prashanth



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X-Original-Date: Fri, 30 Aug 2002 00:25:54 -0700
Date: Fri, 30 Aug 2002 00:25:54 -0700

At 12:01 AM 8/30/02, you wrote:
>Dear Ross,
>I wanted to know how different is the implementation for a MPEG-4 profile 
>than MPEG-2, at a conceptual level. I am asking you this because you have 
>worked with the former profile and the later is in your to-do list. Could 
>you please give an insight into this?

The RTP payload format for MPEG-4 Elementary Streams is currently defined 
in 
<http://www.ietf.org/internet-drafts/draft-ietf-avt-mpeg4-simple-04.txt>. 
I haven't yet looked at this in detail, but it's more complicated than the 
payload format for  MPEG-1 and 2 (defined in RFC 2250).

         Ross.



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X-Original-Date: Fri, 30 Aug 2002 02:38:13 -0700
Date: Fri, 30 Aug 2002 02:38:13 -0700


>VERSION CHECKING:
>================
>Could you provide a LIVE_VERSION macro that
>can be used at compilation time to use the proper version of the LIVE API.

Good idea!  OK, I've now done this (in the newly installed code).  There is 
now a 'version header' file for each library:
         liveMedia/include/liveMedia_version.hh
         UsageEnvironment/include/UsageEnvironment_version.hh
         BasicUsageEnvironment/include/BasicUsageEnvironment_version.hh
         groupsock/include/groupsock_version.hh

Each of these define a version string, and a corresponding integer (that 
can be used for version comparisons).

The RTSP client code now includes the version string (for the "liveMedia" 
library) in the RTSP "User-Agent:" header.


>ADDING EXTRA FIELDS IN THE RTSP REQUESTS:
>======================================
[...]
>I think it would be nice to have a generic method to add/del field in any RTSP
>requests

I want to keep the RTSPClient API as simple as possible, and also ensure 
that clients, in using the API, will always be using the RTSP protocol in a 
legal, conformant way.  Because of this, I don't want to allow clients to 
add (or remove) arbitrary RTSP headers.  Instead, when there are specific 
variations that we want to support (such as the "User-Agent" header, which 
is now supported), we can add support for them to the API on a case-by-case 
basis.

(If anyone really wants to do arbitrary hacking on the RTSP client code, 
they can always do so, as the source is available.  But I'd prefer to keep 
the API simple.)

         Ross.



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From <EMAIL-ADDRESS-HIDDEN> Sun Sep 1 10:25:56 2002
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X-Original-Date: Sun, 01 Sep 2002 13:22:23 -0400
Date: Sun, 01 Sep 2002 13:22:23 -0400

 
Hi Ross,

I'm a university student and I'm trying to use your library in my final year project.

Btw, for the PDA side, I'm just trying to play the received Mpeg file and not to stream video, the streaming side is to be done via a PC.

I'll try to implement the receiver part in PocketPC using embedded Visual C++ 3.0.

btw...I don't suppose you have the instructions to compile your LIVE.COM library using embedded Visual C++, do you?

It's kinda similar to Visual C++ 6.0, but I don't think it supports mak file...I might be wrong though...

Best Regards,
Joseph

--

On Tue, 27 Aug 2002 12:13:05  
 Ross Finlayson wrote:
>
>>Well...I currently have a VC++ program that uses a MPEG4 codec to encode a 
>>video file and transmit it via TCP socket to a decoder/player which is 
>>able to decode the transmited video and display it on the PDA.
>>
>>As I'm trying to do multicast, that's why I'm trying to change the TCP to 
>>RTP using your marvellous LIVE.COM RTP Library. Could you give me some 
>>pointers as to which files I shld be look at?
>>
>>Can I just use the RTP library just replace the TCP socket currently in 
>>use in my program?
>
>No, because RTP is a packet-based protocol, so you can't simply use it to 
>replace TCP (which is a byte-stream protocol).
>
>In particular, to stream data using RTP, the data first needs to be parsed 
>into discrete frames.  Also, the RTP packet format is (in general) 
>different for each codec that's being streamed.  E.g., MPEG-4 video RTP 
>packets have a different format than MPEG-1 (or 2) video RTP packets, which 
>have a different format from MPEG-1 (or 2) audio RTP packets, which have a 
>different format from H.263 video RTP packets, etc. etc.  The LIVE.COM 
>library currently implements the RTP payload format(s) for MPEG-1 or 2, but 
>not MPEG-4 (either MPEG-4 video, or AAC audio), although that's on my 'to 
>do' list.  (If your company is interested in providing funding for MPEG-4 
>support in the LIVE.COM software, then please let me know.)
>
>What I suggest you do for now is see if you can get the existing 
>"testMPEGVideoStreamer" and/or "testMPEGAudioVideoStreamer" demo 
>applications to run, as is, on your WinCE iPaq.  (These applications stream 
>MPEG-1 or 2, but not (yet) MPEG-4.)
>
>         Ross.
>
>
>
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>btw...I don't suppose you have the instructions to compile your LIVE.COM 
>library using embedded Visual C++, do you?
>
>It's kinda similar to Visual C++ 6.0, but I don't think it supports mak 
>file...I might be wrong though...

Dunno, but if it's anything like Visual C++ 6.0, you should be able to open 
a 'workspace' from a ".mak" file.  Do an "Open Workspace", then look at the 
"Files of type" selection menu.  If there's an option for "Makefiles 
(.mak)", then that is what you would use.

         Ross.



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From: Anurag Shukla <EMAIL-ADDRESS-HIDDEN>
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Subject: RE: [Live-devel] RTP in PDAs?
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X-Original-Date: Mon, 2 Sep 2002 11:39:47 +0530
Date: Mon, 2 Sep 2002 11:39:47 +0530

I had compiled the libraries using VC++ some time back. I think I could help
you with that. Just give me some time though so that I can give yu a
comprehensive procedure to do that.

" I'm  the Master of My Fate, the Captain of my Soul, because I have the
Power to Control My Thoughts."
Regards
Anurag
-------------------------------------------------

> -----Original Message-----
> From:	Ross Finlayson [SMTP:finlayson(at)live.com]
> Sent:	Monday, September 02, 2002 1:39 AM
> To:	live-devel(at)lists.sourceforge.net
> Subject:	Re: [Live-devel] RTP in PDAs?
> 
> 
> >btw...I don't suppose you have the instructions to compile your LIVE.COM 
> >library using embedded Visual C++, do you?
> >
> >It's kinda similar to Visual C++ 6.0, but I don't think it supports mak 
> >file...I might be wrong though...
> 
> Dunno, but if it's anything like Visual C++ 6.0, you should be able to
> open 
> a 'workspace' from a ".mak" file.  Do an "Open Workspace", then look at
> the 
> "Files of type" selection menu.  If there's an option for "Makefiles 
> (.mak)", then that is what you would use.
> 
>          Ross.
> 
> 
> 
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X-Original-Date: Mon, 02 Sep 2002 13:14:38 -0400
Date: Mon, 02 Sep 2002 13:14:38 -0400

Hi Anurag,

You have successfully compiled the libraries using Embedded VC++ before? Great!!!

I can't wait to hear from you soon... 

Thanks in advance! 

:) Joseph


--

On Mon, 2 Sep 2002 11:39:47   
 Anurag Shukla wrote:
>I had compiled the libraries using VC++ some time back. I think I could help
>you with that. Just give me some time though so that I can give yu a
>comprehensive procedure to do that.
>
>" I'm  the Master of My Fate, the Captain of my Soul, because I have the
>Power to Control My Thoughts."
>Regards
>Anurag
>-------------------------------------------------
>
>> -----Original Message-----
>> From:	Ross Finlayson [SMTP:finlayson(at)live.com]
>> Sent:	Monday, September 02, 2002 1:39 AM
>> To:	live-devel(at)lists.sourceforge.net
>> Subject:	Re: [Live-devel] RTP in PDAs?
>> 
>> 
>> >btw...I don't suppose you have the instructions to compile your LIVE.COM 
>> >library using embedded Visual C++, do you?
>> >
>> >It's kinda similar to Visual C++ 6.0, but I don't think it supports mak 
>> >file...I might be wrong though...
>> 
>> Dunno, but if it's anything like Visual C++ 6.0, you should be able to
>> open 
>> a 'workspace' from a ".mak" file.  Do an "Open Workspace", then look at
>> the 
>> "Files of type" selection menu.  If there's an option for "Makefiles 
>> (.mak)", then that is what you would use.
>> 
>>          Ross.
>> 
>> 
>> 
>> -------------------------------------------------------
>> This sf.net email is sponsored by: OSDN - Tired of that same old
>> cell phone?  Get a new here for FREE!
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X-Original-Date: Mon, 02 Sep 2002 13:14:54 -0400
Date: Mon, 02 Sep 2002 13:14:54 -0400

Hi Anurag,

You have successfully compiled the libraries using Embedded VC++ before? Great!!!

I can't wait to hear from you soon... 

Thanks in advance! 

:) Joseph


--

On Mon, 2 Sep 2002 11:39:47   
 Anurag Shukla wrote:
>I had compiled the libraries using VC++ some time back. I think I could help
>you with that. Just give me some time though so that I can give yu a
>comprehensive procedure to do that.
>
>" I'm  the Master of My Fate, the Captain of my Soul, because I have the
>Power to Control My Thoughts."
>Regards
>Anurag
>-------------------------------------------------
>
>> -----Original Message-----
>> From:	Ross Finlayson [SMTP:finlayson(at)live.com]
>> Sent:	Monday, September 02, 2002 1:39 AM
>> To:	live-devel(at)lists.sourceforge.net
>> Subject:	Re: [Live-devel] RTP in PDAs?
>> 
>> 
>> >btw...I don't suppose you have the instructions to compile your LIVE.COM 
>> >library using embedded Visual C++, do you?
>> >
>> >It's kinda similar to Visual C++ 6.0, but I don't think it supports mak 
>> >file...I might be wrong though...
>> 
>> Dunno, but if it's anything like Visual C++ 6.0, you should be able to
>> open 
>> a 'workspace' from a ".mak" file.  Do an "Open Workspace", then look at
>> the 
>> "Files of type" selection menu.  If there's an option for "Makefiles 
>> (.mak)", then that is what you would use.
>> 
>>          Ross.
>> 
>> 
>> 
>> -------------------------------------------------------
>> This sf.net email is sponsored by: OSDN - Tired of that same old
>> cell phone?  Get a new here for FREE!
>> https://www.inphonic.com/r.asp?r=sourceforge1&refcode1=vs3390
>> _______________________________________________
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>> Live-devel(at)lists.sourceforge.net
>> https://lists.sourceforge.net/lists/listinfo/live-devel
>*********************************************************************
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>confidential / privileged. It is intended solely for the addressee or
>addressees. If you are not the addressee indicated in this message, you may
>not copy or deliver this message to anyone. In such case, you should destroy
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>


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Subject: [Live-devel] audio-video streaming
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X-Original-Date: Tue, 03 Sep 2002 02:58:46 +0000
Date: Tue, 03 Sep 2002 02:58:46 +0000

Hi Live Developers,

I find in the play() method of testMPEGAudioVideoStreamer.cpp the following 
code:

audioSink->startPlaying(*audioSource, afterPlaying, audioSink);
videoSink->startPlaying(*videoSource, afterPlaying, videoSink);

How does the audio and video streaming work together?  Is it that entire 
audio is streamed and then entire video is stremed?  I guess it can not be 
so and I belive I am missing something here.

I have hard time to look into the entire code in detail. Can anyone let me 
know how startPlaying is implemented such that a chunk of audio and a chunk 
of video are streamed alternatively?  Your explaination will help me to 
quickly understand the code.  Is there any flowgraph of the software 
available, so that we can understand how the software works?

Thanks a lot!!

Putti





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X-Original-Date: Mon, 02 Sep 2002 20:28:58 -0700
Date: Mon, 02 Sep 2002 20:28:58 -0700


Has anyone tried live.2002.08.30.tar on OSX 10.1.5?

FYI:

When stuffit automatically unzipped and untarred the file,
 two file names were truncated leaving off a 'h' on .hh and
 a 'p' on .cpp

The library built just fine.

testMP3Streamer though seems to have a problem sending the data stream
 (or RealOne has a problem decoding the stream)  When listening to a
 MP3 sourced from XP system running livecaseter, RealOne does fine.
 When listening to a stream sourced from MacOSX, the stream is interrupted
 after 30 to 90 seconds.  (It is still being sent, its that RealOne
 drops off).

Neither testMPEGAudioVideoStreamer and testMPEGVideostreamer can read
 an MPEG1 file.  (Which plays fine in Quicktime).


Finally, is there any way to convert a file to MPEG1 in OSX?  My
 Quicktime Pro 6 does not provide that as an option.

Thanks




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Subject: Re: [Live-devel] audio-video streaming
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X-Original-Date: Tue, 03 Sep 2002 03:13:07 -0700
Date: Tue, 03 Sep 2002 03:13:07 -0700


>I find in the play() method of testMPEGAudioVideoStreamer.cpp the 
>following code:
>
>audioSink->startPlaying(*audioSource, afterPlaying, audioSink);
>videoSink->startPlaying(*videoSource, afterPlaying, videoSink);
>
>How does the audio and video streaming work together?  Is it that entire 
>audio is streamed and then entire video is stremed?

No, those statements only *begin* the streaming of audio and video.  Most 
of the actual streaming takes place within the subsequent call to
         env->taskScheduler().blockMyself();
at which time the audio and video gets streamed concurrently.

"BasicTaskScheduler::blockMyself()" (see 
"BasicUsageEnvironment/BasicUsageEnvironment.cpp") is an event loop - it 
repeatedly handles delayed tasks, and incoming network packets.  That's 
where most of the work gets done.

"MediaSink::startPlaying()" (see "liveMedia/MediaSink.cpp") works by 
calling "continuePlaying()", which is a pure virtual member function, 
implemented by each subclass of "MediaSink".

In this case, the subclass is "MultiFramedRTPSink" (see 
"liveMedia/MultiFramedRTPSink.cpp"). 
"MultiFramedRTPSink::continuePlaying()" works by (i) reading enough data to 
fill the first RTP packet, (ii) sending this packet, then (iii) callng
         scheduleDelayedTask()   ["MultiFramedRTPSink.cpp", line 311]
to delay an appropriate amount of time before trying to send the next 
packet.  (If this delay did not happen, the program would stream out the 
data as fast as it could read it from the file, which is something we would 
not want to happen.)

Once this period of time has elapsed, the event loop 
("BasicTaskScheduler::blockMyself()") calls the event handler function - in 
this case, "MultiFramedRTPSink::sendNext()".  This function arranges to 
read enough data to fill the next RTP packet, which then gets sent, and 
then "scheduleDelayedTask()" is called once more to delay an appropriate 
amount of time, etc. etc.

In summary, the key to understanding this code is to understand how
1/ The call to "startPlaying()" sets up initial event handler tasks, and
2/ "blockMyself()" is the event handler loop, where most work gets 
done.  It repeatedly calls event handler functions, which in turn set up 
more event handler tasks, and so on.
(It helps to be familiar with event-driven programming.)

         Ross.



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X-Original-Date: Tue, 3 Sep 2002 09:09:28 -0400
Date: Tue, 3 Sep 2002 09:09:28 -0400


On Monday, September 2, 2002, at 11:28 PM, John Zwiebel wrote:

> Has anyone tried live.2002.08.30.tar on OSX 10.1.5?

I helped with the initial makefile cleanup for Mac OS X, but I have to 
admit I have not been keeping up with recent versions of the live.com 
library.  I suppose now that I have 10.2 box running that's the next 
major thing to test out.

> FYI:
>
> When stuffit automatically unzipped and untarred the file,
>  two file names were truncated leaving off a 'h' on .hh and
>  a 'p' on .cpp

I noticed this also, but the command-line tar seems to work fine.

> The library built just fine.
>
> testMP3Streamer though seems to have a problem sending the data stream
>  (or RealOne has a problem decoding the stream)  When listening to a
>  MP3 sourced from XP system running livecaseter, RealOne does fine.
>  When listening to a stream sourced from MacOSX, the stream is 
> interrupted
>  after 30 to 90 seconds.  (It is still being sent, its that RealOne
>  drops off).

What OS were you running RealOne from?  I know the Real One Beta for 
Mac OS X can be quite flaky.  Marshall Eubanks has reported several 
bugs related to it's handling of multicast.

> Neither testMPEGAudioVideoStreamer and testMPEGVideostreamer can read
>  an MPEG1 file.  (Which plays fine in Quicktime).

Hmmn, this should work.  I'll test this later today.

> Finally, is there any way to convert a file to MPEG1 in OSX?  My
>  Quicktime Pro 6 does not provide that as an option.

What is the source file format?  I know clean can export MPEG1 files, 
but it generally handles the same file formats as QuickTime.

Marty Schoch
<mschoch(at)multicasttech.com>



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X-Original-Date: Tue, 3 Sep 2002 09:21:15 -0400
Date: Tue, 3 Sep 2002 09:21:15 -0400

> I know clean can export MPEG1 files, but it generally handles the same 
> file

What I meant to say here is that the product "Cleaner" can export MPEG1.

http://www.discreet.com/products/cleaner/

Marty Schoch
<mschoch(at)multicasttech.com>



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Subject: Re: [Live-devel] Live Library on OSX
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X-Original-Date: Tue, 03 Sep 2002 06:35:54 -0700
Date: Tue, 03 Sep 2002 06:35:54 -0700

On 9/3/02 6:09 AM, "Marty Schoch" <mschoch(at)multicasttech.com> wrote:

> 
> On Monday, September 2, 2002, at 11:28 PM, John Zwiebel wrote:

>> 
>> testMP3Streamer though seems to have a problem sending the data stream
>>  (or RealOne has a problem decoding the stream)  When listening to a
>>  MP3 sourced from XP system running livecaseter, RealOne does fine.
>>  When listening to a stream sourced from MacOSX, the stream is
>> interrupted
>>  after 30 to 90 seconds.  (It is still being sent, its that RealOne
>>  drops off).
> 
> What OS were you running RealOne from?  I know the Real One Beta for
> Mac OS X can be quite flaky.  Marshall Eubanks has reported several
> bugs related to it's handling of multicast.

Yes, it was RealOne for OSX and so the Beta.  However, that same RealOne
 on the same iBook doesn't (or hasn't) drop off the multicast stream
 sourced from an XP via livecaster.  That may or may not mean anything.

> 
>> Finally, is there any way to convert a file to MPEG1 in OSX?  My
>>  Quicktime Pro 6 does not provide that as an option.
> 
> What is the source file format?  I know clean can export MPEG1 files,
> but it generally handles the same file formats as QuickTime.
>

I tried several.  AVI primarily.  But there were also .mov.

Please elaborate on "clean".  I fail to grok the context unless "clean"
 is some application I don't know about.

Thanks



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X-Original-Date: Tue, 3 Sep 2002 09:51:42 -0400
Date: Tue, 3 Sep 2002 09:51:42 -0400


On Tuesday, September 3, 2002, at 09:35 AM, John Zwiebel wrote:

> I tried several.  AVI primarily.  But there were also .mov.

Be careful with AVI files.  AVI is a wrapper format, and many different 
audio/video codecs are used inside them.  QuickTime can handle some of 
them, but there are others that I've never been able to find a Mac 
plugin for.

In general Cleaner will only be able to handle the same types of AVI's 
as QuickTime.

Marty Schoch
<mschoch(at)multicasttech.com>



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X-Original-Date: Tue, 3 Sep 2002 18:35:19 +0100
Date: Tue, 3 Sep 2002 18:35:19 +0100

Hi all,
I downloaded the latest library (30 Aug).

Built it under redhat 7 using gcc 2.96. with -DIMPLEMENT_RTSP_SERVER
I then run testMPEGAudioVisualStreamer

Sometimes (like 50% of the time) when I access the RTSP server 
using QuickTime 6 Player from an XP machine I get a SIGSEGV. 

If I do a backtrace in gdb 
DelayQueueEntry::handleTimeout() was excuting

handleTimeout does 
delete this;

Which is kinda an odd thing to do,
but it is in a linked list and all that jazz.

I removed the 
delete this;
and got a memory leak, as you'd expect, I guess.

The openRTSP and playRTPMPEG tools don't seem to cause
as much of a problem though

And the old library worked fine.

Anyone else had the same problems?
I'll keep looking though.

J.P.




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X-Original-Date: Tue, 03 Sep 2002 11:08:03 -0700
Date: Tue, 03 Sep 2002 11:08:03 -0700

I suspect I've found the problem (I had added a new "a=tool:" line to the 
SDP description returned by the RTSP server, but had forgotten to allocate 
enough space for it).

I'll be uploading a new library shortly.  In the meantime, you should be 
able to fix the problem by changing the
	20
at line 181 of "liveMedia/ServerMediaSession.cpp" to
	120

     Ross.



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X-Original-Date: Wed, 04 Sep 2002 23:22:15 +0800
Date: Wed, 04 Sep 2002 23:22:15 +0800

Hi,everybody.Hi,Ross :P

I complied the live.com library and test programs in VC6 successfully. But when I
use live.com library in a MFC application (I copied TestMPEGAudioVideo.cpp's code
into my application), it showed "Debug assertion fail".It seems that	class
GroupSock's Constructor cause it. Has anybody used live.com lib in custom MFC
application??

I used Borland C++ Builder before. I am not familiar with VC6. And when I opened
.mak in my BCB6,it showed "Project rule not found". Some suggestion?

Thanks
		 
               Capton
               capton(at)263.net
					2002-09-03 




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X-Original-Date: Wed, 04 Sep 2002 09:19:25 -0700
Date: Wed, 04 Sep 2002 09:19:25 -0700


>I complied the live.com library and test programs in VC6 successfully. But 
>when I
>use live.com library in a MFC application (I copied 
>TestMPEGAudioVideo.cpp's code
>into my application), it showed "Debug assertion fail".It seems that    class
>GroupSock's Constructor cause it.

Can you be more specific?  Can you find out exactly why that code is failing?

Everyone, please remember that you have *complete source code* to the 
LIVE.COM libraries.  It may not always be easy, but if you encounter a 
problem, it should be possible to find out exactly where the code is failing.

>I used Borland C++ Builder before. I am not familiar with VC6. And when I 
>opened
>.mak in my BCB6,it showed "Project rule not found". Some suggestion?

Did you follow the special instructions for the Borland compiler, listed at
         http://www.live.com/liveMedia/#borland
?

         Ross.



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> >I used Borland C++ Builder before. I am not familiar with VC6. And when I
> >opened
> >.mak in my BCB6,it showed "Project rule not found". Some suggestion?
>
> Did you follow the special instructions for the Borland compiler, listed
at
>          http://www.live.com/liveMedia/#borland
> ?

i followed exactly with instructions for bcb given on the website with the
same result as one above.
maybe it's time to build .dll libraries for win32 users ?

>          Ross.

a.



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Zaufaj swojej intuicji... >>> http://link.interia.pl/f1646 




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X-Original-Date: Wed, 04 Sep 2002 11:54:14 -0700
Date: Wed, 04 Sep 2002 11:54:14 -0700

At 11:16 AM 9/4/02, altair wrote:
> > >I used Borland C++ Builder before. I am not familiar with VC6. And when I
> > >opened
> > >.mak in my BCB6,it showed "Project rule not found". Some suggestion?
> >
> > Did you follow the special instructions for the Borland compiler, listed
>at
> >          http://www.live.com/liveMedia/#borland
> > ?
>
>i followed exactly with instructions for bcb given on the website with the
>same result as one above.
>maybe it's time to build .dll libraries for win32 users ?

No, maybe it's time for someone who's familiar with the Borland compiler 
(not I) to figure out what needs to be changed in the instructions, and, in 
particular, to figure out if anything needs to be changed in the 
"win32config.Borland" file.

I will not be releasing the libraries in binary form.  This code is 
intended primarily for use by developers who are experienced in developing 
systems software.  Programmers who use this software are expected to be 
able to compile it themselves.  If - as is apparently the case here - the 
instructions are incomplete or inaccurate, then please try to figure out 
what's wrong, so I can update the instructions, as appropriate.

         Ross.



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Subject: Re: [Live-devel] Live Library on OSX
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X-Original-Date: Thu, 05 Sep 2002 20:23:48 -0700
Date: Thu, 05 Sep 2002 20:23:48 -0700

On 9/3/02 6:21 AM, "Marty Schoch" <mschoch(at)multicasttech.com> wrote:


> 
> What I meant to say here is that the product "Cleaner" can export MPEG1.
> 
> http://www.discreet.com/products/cleaner/
>

Marty, the Cleaner demo took in an avi file and said it output a
 MPEG1 file.  I can open the file using QuickTime player and it
 Hhs a different "look" to it when compared to the original file
 plus, MacOSX thinks its an MPEG file.

However, when I tried sending that file with live.com testMPEGVideoStreamer,
 Quicktime wouldn't play it back.

I guess I'm saying that I'm not so sure that Cleaner works, at least
 it isn't obvious that I'm doing something wrong.

I was able to obtain a "real" MPEG file, and things work fine.

Which brings me back to my original problem, where does one find "real"
 MPEG1 files.  :-) 



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Subject: [Live-devel] How to compile the LIVE.COM library code using Borland
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X-Original-Date: Thu, 05 Sep 2002 20:20:33 -0700
Date: Thu, 05 Sep 2002 20:20:33 -0700

Many thanks to Vesselin Kostadinov for sending me instructions on how to 
build the code using the Borland C++/C compiler.  For updated instructions, see
	http://www.live.com/liveMedia/#borland

Note also that if you want to try this, you will need the most up-to-date 
version (2002.09.06) of the source code package, because it contains an 
updated (corrected) version of the "win32config.Borland" file.

	Ross.



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X-Original-Date: Fri, 6 Sep 2002 13:31:53 +0930
Date: Fri, 6 Sep 2002 13:31:53 +0930

Ross,

There is a slight problem with
http://www.live.com/liveMedia/borland-additional-instructions.txt.
Line 143 should be joined to line 142 to form a single line:
----------------------
$(LINK) $(CONSOLE_LINK_OPTS) $(...SAME..WHATEVER..._OBJS),$(at),,$(LIBS),,
----------------------

I do not now if my mail client decided to reformat it. If this is the case -
sorry.

The same thing applies to some of the lines thereafter but they are somewhat
easier to spot.



We could make the process of building with Borland tools even easier if you
agree on the following changes:

1. Add this line in win32config.Borland:

USE_BORLAND_TOOLS = Yes


2. Change each of the Makefile.tail files:

!if $d(USE_BORLAND_TOOLS)
        $(LIBRARY_LINK) $(at) (at)&&!
+-$(**: = &^
+-)
!
!else  #Non-Borland tools
       $(LIBRARY_LINK)$(at) $(LIBRARY_LINK_OPTS) $(OBJS)
!endif

(the example is for UsageEnvironment/Makefile.tail but you get the point)

The forward/backslash issue for the test programs could be addressed with a
similar approach.

However I am not sure whether all non-Borland make programs support !if
$d(...) ... !else ... !endif statements, so the decision is entirely yours.

Once this is done the whole process could be as simple as starting a .BAT
file, wich I wouldn't mind writing.


Regards

Vesselin.


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X-Original-Date: Thu, 05 Sep 2002 21:24:08 -0700
Date: Thu, 05 Sep 2002 21:24:08 -0700


>There is a slight problem with
>http://www.live.com/liveMedia/borland-additional-instructions.txt.
>Line 143 should be joined to line 142 to form a single line:
>----------------------
>$(LINK) $(CONSOLE_LINK_OPTS) $(...SAME..WHATEVER..._OBJS),$(at),,$(LIBS),,
>----------------------
>
>I do not now if my mail client decided to reformat it. If this is the case -
>sorry.
>
>The same thing applies to some of the lines thereafter but they are somewhat
>easier to spot.

OK, thanks.  I've fixed these now.


>We could make the process of building with Borland tools even easier if you
>agree on the following changes:
>
>1. Add this line in win32config.Borland:
>
>USE_BORLAND_TOOLS = Yes
>
>
>2. Change each of the Makefile.tail files:
>
>!if $d(USE_BORLAND_TOOLS)
>         $(LIBRARY_LINK) $(at) (at)&&!
>+-$(**: = &^
>+-)
>!
>!else  #Non-Borland tools
>        $(LIBRARY_LINK)$(at) $(LIBRARY_LINK_OPTS) $(OBJS)
>!endif
>
>(the example is for UsageEnvironment/Makefile.tail but you get the point)
>
>The forward/backslash issue for the test programs could be addressed with a
>similar approach.
>
>However I am not sure whether all non-Borland make programs support !if
>$d(...) ... !else ... !endif statements, so the decision is entirely yours.

No, unfortunately these additions to the Makefiles would break on Unix 
systems.  Probably the best way to do this would be to update the 
"genWindowsMakefiles" script to automatically add these "!if ... !else ... 
!endif" lines when generating the Windoze ".mak" files.  If anyone wants to 
give this a try, be my guest :-)

         Ross.



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--0-2068275556-1031315706=:65621
Content-Type: text/plain; charset=iso-8859-1
Content-Transfer-Encoding: 8bit


 
Hi to all,
I have modified Berckeley MPEG1 player in order to read from an UDP socket. I have planed to read from an RTSP session. So I have some questions about that:
1--When receiving data with OpenRTSP, are they umpacketised and then ordred when they are pumped to 'stdout'? if this is true, can I pump to 'stdin' and then reading fom 'stdin' to the buffer of the decoder ?
thanks a lot
Achraf,



---------------------------------
Yahoo! Mail -- Une adresse (at)yahoo.fr gratuite et en français !

--0-2068275556-1031315706=:65621
Content-Type: text/html; charset=iso-8859-1
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<P> 
<P>Hi to all,
<P>I have modified Berckeley MPEG1 player in order to read from an UDP socket. I have planed to read from an RTSP session. So I have some questions about that:
<P>1--When receiving data&nbsp;with OpenRTSP, are they&nbsp;umpacketised and then ordred when they are pumped to 'stdout'? if this is true, can&nbsp;I pump to 'stdin' and then reading fom 'stdin' to the buffer of the decoder ?
<P>thanks a lot
<P>Achraf,</P><p><br><hr size=1><a href="http://fr.mail.yahoo.com">Yahoo! Mail</a> -- Une adresse (at)yahoo.fr gratuite et en français !<br>
--0-2068275556-1031315706=:65621--


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hi all,

I have an MPEG1-player witch read from a video-file (elemantary stream). My question is:

Can I, write data with OpenRTSP to 'stdin' And then giving 'stdin' as a parameter to my player?

thanks



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<P>hi all,</P>
<P>I have an MPEG1-player witch read from a video-file (elemantary stream). My question is:</P>
<P>Can I, write&nbsp;data with OpenRTSP&nbsp;to 'stdin' And then giving 'stdin' as a parameter to my player?</P>
<P>thanks</P><p><br><hr size=1><a href="http://fr.mail.yahoo.com">Yahoo! Mail</a> -- Une adresse (at)yahoo.fr gratuite et en français !<br>
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Subject: Re: [Live-devel] About OpenRTSP,
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X-Original-Date: Fri, 06 Sep 2002 10:27:29 -0700
Date: Fri, 06 Sep 2002 10:27:29 -0700

At 05:35 AM 9/6/02, Achraf wrote:
>I have modified Berckeley MPEG1 player in order to read from an UDP 
>socket. I have planed to read from an RTSP session. So I have some 
>questions about that:
>
>1--When receiving data with OpenRTSP, are they umpacketised and then 
>ordred when they are pumped to 'stdout'? if this is true, can I pump to 
>'stdin' and then reading fom 'stdin' to the buffer of the decoder ?

Yes, you can do this by giving openRTSP the "-v" option (meaning, output 
video to 'stdout'), and then pipe this to your MPEG-1 player.  I.e.,

         openRTSP -v rtsp://...  |  player

(Replace "rtsp://..." with the RTSP url of your session, and replace 
"player" with the name of your MPEG-1 player tool.)

This assumes that your MPEG-1 player tool can read from 'stdin'.

I'm also assuming that you're doing this on a Unix system.  (I don't know 
whether the "|" pipe mechanism will work on Windows.)

         Ross.



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 Hi all,
I fail to pipe the output of OpenRTSP to the player on Windose OS!. But when I check the player's code I find that It can read Data from socket (Using TCP). So I want to ask :
1--Can Iuse the test Relay with open RTSP write data to one socket (on "localhost") instead of outputing them to "stdout". I thank that I can give "127.0.0.1:port" to my player as a parameter.
2--If yes, what functions are to use.
Thanks
Achraf 



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<P>&nbsp;Hi all,
<P>I fail to pipe the output of OpenRTSP to the player on Windose OS!. But when I check the player's code I find that It can read Data from socket (Using TCP). So I want to ask :
<P>1--Can Iuse the test Relay with open RTSP write data to one socket (on "localhost") instead of outputing them to "stdout". I thank that I can give "127.0.0.1:port" to my player as a parameter.
<P>2--If yes, what functions are to use.
<P>Thanks
<P>Achraf&nbsp;</P><p><br><hr size=1><a href="http://fr.mail.yahoo.com">Yahoo! Mail</a> -- Une adresse (at)yahoo.fr gratuite et en français !<br>
--0-1063352799-1031387252=:71205--


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X-Original-Date: Sat, 07 Sep 2002 01:41:42 -0700
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>I fail to pipe the output of OpenRTSP to the player on Windose OS!. But 
>when I check the player's code I find that It can read Data from socket 
>(Using TCP). So I want to ask :
>
>1--Can Iuse the test Relay with open RTSP write data to one socket (on 
>"localhost") instead of outputing them to "stdout".

No, because "testRelay" is an application for relaying UDP packets, not TCP.

If your player application really can read from 'stdin', then the best 
solution is to figure out how to set up a pipe between "openRTSP" and your 
player.  I don't know how to set up pipes in Windoze, but in Unix, this 
kind of thing is trivial.  (Why not just switch to Unix? :-)

         Ross.



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X-Original-Date: Sun, 08 Sep 2002 10:20:22 +0000
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Hi Ross,
I am trying to stream live capture using Live media 
libraby. I am using H263 on Solaris. Could you please 
give me some hints on how to "register a callback 
function with the video device/card" to make sure that a 
frame is ready for processing? I couldn't find anything 
is XIL guides.

Thanks in advance.
Faruq   


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>I am trying to stream live capture using Live media
>libraby. I am using H263 on Solaris. Could you please
>give me some hints on how to "register a callback
>function with the video device/card" to make sure that a
>frame is ready for processing? I couldn't find anything
>is XIL guides.

Unfortunately I can't help you here, as I'm not familiar with XIL.

However, once you've figured out how to do this, you can use 
"liveMedia/DeviceSource.cpp" as a model for a source object that 
encapsulates your video capture device.  (Of course, you will need to do 
H.263 encoding also.)

         Ross.



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From: Jean-Francois Cadiere <EMAIL-ADDRESS-HIDDEN>
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Subject: [Live-devel] How to extract RTSP client side code from Streaming Media source
 code ?
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X-Original-Date: Mon, 09 Sep 2002 14:08:43 +0200
Date: Mon, 09 Sep 2002 14:08:43 +0200

I want to extract the part of code necessary to get the RTSP client 
functionnality only (i.e., openRTSP).

I just tried removing all files whose names contain the words "Server" 
or "Sink" (and their "#include"s from the file "include/liveMedia.hh".)
but this is not enough. I adapted the Makefile to not compile theses 
files anymore, but some objects defined in these files
are still be used in other classes implementation.

Has anyone already done this successfully ?

Cheers,
JF CADIERE





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Subject: Re: [Live-devel] How to extract RTSP client side code from
  Streaming Media source code ?
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X-Original-Date: Mon, 09 Sep 2002 10:57:26 -0700
Date: Mon, 09 Sep 2002 10:57:26 -0700

At 05:08 AM 9/9/02, you wrote:
>I want to extract the part of code necessary to get the RTSP client 
>functionnality only (i.e., openRTSP).
>
>I just tried removing all files whose names contain the words "Server" or 
>"Sink" (and their "#include"s from the file "include/liveMedia.hh".)
>but this is not enough. I adapted the Makefile to not compile theses files 
>anymore, but some objects defined in these files
>are still be used in other classes implementation.

Jean-Francois,

Sorry, the instructions that I gave you earlier were inaccurate.  You will 
still need the following files:
         FileSink.cpp                    include/FileSink.hh
         QuickTimeFileSink.cpp   include/QuickTimeFileSink.hh
         MediaSink.cpp           include/MediaSink.hh

because these are needed so that "openRTSP" can record the stored 
audio/video data that it receives.

However, you should definitely be able to remove any file whose name 
contains "RTPSink", "HTTPSink", or "Server".

         Ross.



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Subject: [Live-devel] parallel rtsp client !!!
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  Hello all,
In order to implement Staggered Broadcasting protocol used on VOD, I have changed testMPEGVideostreamer so it can stream more than one video at once. I have used as video session number as RTSP server
my question is about openRTSP, is it possible to receive every session description througth one RTSP client, and than save them at the same time??
Achraf 



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<P> 
<P>&nbsp; Hello all,
<P>In order to implement Staggered Broadcasting protocol used on VOD, I have changed testMPEGVideostreamer so it can stream more than one video at once. I have used as video session number as RTSP server
<P>my question is about openRTSP, is it possible to receive every session description througth one RTSP client, and than save them at the same time??
<P>Achraf&nbsp;</P><p><br><hr size=1><a href="http://fr.mail.yahoo.com">Yahoo! Mail</a> -- Une adresse (at)yahoo.fr gratuite et en français !<br>
--0-408065118-1031645845=:86174--


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X-Original-Date: Tue, 10 Sep 2002 01:40:30 -0700
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>In order to implement Staggered Broadcasting protocol used on VOD, I have 
>changed testMPEGVideostreamer so it can stream more than one video at 
>once. I have used as video session number as RTSP server
>
>my question is about openRTSP, is it possible to receive every session 
>description througth one RTSP client, and than save them at the same time??

If I understand you correctly, I think you can do this - but it would 
require significant modification to the "openRTSP" source code.

I.e., you can use the RTSP "DESCRIBE" command on several different 
"rtsp://" URLs to retrieve several different SDP descriptions, *provided 
that* the "rtsp://" URLs all point to the same server.  (If that's not the 
case, then you will need to use different "RTSPClient" objects.)

Then, once you have each SDP descriptions, you can create a separate 
"MediaSession" object from each.

However, you may just find it simplest to just run "openRTSP" multiple 
types, once for each "rtsp://" URL!

         Ross.



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X-Original-Date: Tue, 10 Sep 2002 14:23:50 +0200
Date: Tue, 10 Sep 2002 14:23:50 +0200

Hi Ross,

Thank you very much.
It works well now!

Just a last precision: The file RTPSink.hh needs also to be kept.

Regards,
JF



>
>
>Today's Topics:
>
>   1. How to extract RTSP client side code from Streaming Media source
>       code ? (Jean-Francois Cadiere)
>   2. Re: How to extract RTSP client side code from
>       Streaming Media source code ? (Ross Finlayson)
>
>--__--__--
>
>Message: 1
>Date: Mon, 09 Sep 2002 14:08:43 +0200
>From: Jean-Francois Cadiere <jfcadiere(at)eprocess.fr>
>To: live-devel(at)lists.sourceforge.net, jfcadiere(at)eprocess.fr
>Subject: [Live-devel] How to extract RTSP client side code from Streaming Media source
> code ?
>
>I want to extract the part of code necessary to get the RTSP client 
>functionnality only (i.e., openRTSP).
>
>I just tried removing all files whose names contain the words "Server" 
>or "Sink" (and their "#include"s from the file "include/liveMedia.hh".)
>but this is not enough. I adapted the Makefile to not compile theses 
>files anymore, but some objects defined in these files
>are still be used in other classes implementation.
>
>Has anyone already done this successfully ?
>
>Cheers,
>JF CADIERE
>
>
>
>
>
>--__--__--
>
>Message: 2
>Date: Mon, 09 Sep 2002 10:57:26 -0700
>To: live-devel(at)lists.sourceforge.net
>From: Ross Finlayson <finlayson(at)live.com>
>Subject: Re: [Live-devel] How to extract RTSP client side code from
>  Streaming Media source code ?
>
>At 05:08 AM 9/9/02, you wrote:
>  
>
>>I want to extract the part of code necessary to get the RTSP client 
>>functionnality only (i.e., openRTSP).
>>
>>I just tried removing all files whose names contain the words "Server" or 
>>"Sink" (and their "#include"s from the file "include/liveMedia.hh".)
>>but this is not enough. I adapted the Makefile to not compile theses files 
>>anymore, but some objects defined in these files
>>are still be used in other classes implementation.
>>    
>>
>
>Jean-Francois,
>
>Sorry, the instructions that I gave you earlier were inaccurate.  You will 
>still need the following files:
>         FileSink.cpp                    include/FileSink.hh
>         QuickTimeFileSink.cpp   include/QuickTimeFileSink.hh
>         MediaSink.cpp           include/MediaSink.hh
>
>because these are needed so that "openRTSP" can record the stored 
>audio/video data that it receives.
>
>However, you should definitely be able to remove any file whose name 
>contains "RTPSink", "HTTPSink", or "Server".
>
>         Ross.
>
>
>
>
>--__--__--
>
>_______________________________________________
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>https://lists.sourceforge.net/lists/listinfo/live-devel
>
>
>End of Live-devel Digest
>
>  
>





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Hi,
I have this query about the way rtp timestamp is calculated. Especially =
when ByteStreamFileSource is used as the source.
The ByteStreamFileSource::doGetNextFrame() function assigns system time =
to timestamp field. This is what is used by the ' =
RTPSink::convertToRTPTimestamp' to calculate the timestamp that goes =
with the packet.=20
Here the rtp timestamp does not increase linearly as it is is dependent =
on the system time. If the packets are being read from a file, and each =
packet contains same number of frames, the rtp timestamp must increase =
linearly.=20
Because of this I am unable to work with the live library as the source =
and open rtp stack and applications that use it, like linphone as the =
destination. Open rtp assumes that if the packet has n frames then the =
next packet's rtp time stamp must be 160*n greater than the current one.
I have gone through the entire archive, but haven't come across a single =
query in this regard. Any pointers in this regard would be a great help.
Regards,
Anant Akinchan

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<DIV><FONT face=3DArial size=3D2>
<DIV><FONT face=3DArial size=3D2>Hi,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>I have this query about the way rtp =
timestamp is=20
calculated. Especially&nbsp;when ByteStreamFileSource is used as the=20
source.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>The =
ByteStreamFileSource::doGetNextFrame()=20
function&nbsp;assigns system time to timestamp field. This is what is =
used by=20
the ' RTPSink::convertToRTPTimestamp' to calculate the timestamp that =
goes with=20
the packet. </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Here the&nbsp;rtp timestamp does not =
increase=20
linearly as it is is dependent on the system time.&nbsp;If the packets =
are being=20
read from a file, and each packet contains same number of frames, the =
rtp=20
timestamp must increase linearly. </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Because of this I am unable to work =
with the live=20
library as the source and open rtp stack and applications that use it, =
like=20
linphone as the destination. Open rtp assumes that if the packet has n =
frames=20
then the next packet's rtp time stamp must be 160*n greater than the =
current=20
one.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>I have gone through the entire archive, =
but haven't=20
come across a single query in this regard. Any pointers in this regard =
would be=20
a great help.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Regards,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Anant=20
Akinchan</FONT></DIV></FONT></DIV></BODY></HTML>

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Date: Tue, 10 Sep 2002 18:26:13 -0700


>I have this query about the way rtp timestamp is calculated. Especially 
>when ByteStreamFileSource is used as the source.
>The ByteStreamFileSource::doGetNextFrame() function assigns system time to 
>timestamp field. This is what is used by the ' 
>RTPSink::convertToRTPTimestamp' to calculate the timestamp that goes with 
>the packet.
>Here the rtp timestamp does not increase linearly as it is is dependent on 
>the system time. If the packets are being read from a file, and each 
>packet contains same number of frames, the rtp timestamp must increase 
>linearly.

Yes, good point.  I've gone ahead now and modified "ByteStreamFileSource", 
and installed a new version of the library code (2002.09.11).  If a 
ByteStreamFileSource is created using both the "preferredFrameSize" and 
"playTimePerFrame" parameters, then the 'presentation time' (used for 
computing the RTP timestamp) will now be computed based on these 
parameters, rather than on 'wall clock' time.

Note that, to use this feature, you *must* give the "preferredFrameSize" 
and "playTimePerFrame" parameters to 
"ByteStreamFileSource::createNew()".  For example, to stream a file that 
contains GSM data:
         ByteStreamFileSource::createNew(*env, inputFileName, 33, 0.02);
(because each GSM frames is 33 bytes, and represents a duration of 20 ms).

>Because of this I am unable to work with the live library as the source 
>and open rtp stack and applications that use it, like linphone as the 
>destination. Open rtp assumes that if the packet has n frames then the 
>next packet's rtp time stamp must be 160*n greater than the current one.

OK, this should now work (assuming, of course, that you have a "RTPSink" 
that implements the particular RTP payload format (i.e., codec) that you're 
using).  Out of curiosity, which codec(s) do you plan to use?

         Ross.



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Subject: [Live-devel] Re: openRTSP question
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X-Original-Date: Tue, 10 Sep 2002 21:50:06 -0700
Date: Tue, 10 Sep 2002 21:50:06 -0700

At 06:06 AM 9/10/02, Romulus Grigoras wrote:
>First of all, thanks for the great software you put online.
>
>Secondly, I would like to use openRTSP for some kind of caching server
>for Darwin Streaming Server.

First, it's important to understand that "openRTSP" is merely an 
application - and, in particular, a RTSP *client* application.  So it 
probably does not make sense to try to turn it into a caching server.

>My aplication (research paper at
>http://www.enseeiht.fr/lima/sigma/docs/acm2002.pdf) needs to request
>each stream at a speed that varies in time in order to buffer it
>locally. At some point I may need to buffer non-contiguous pieces of the
>stream. I suppose I can use openRTSP for this, what do you think ?

"openRTSP" is an *example* of an application that can use the "LIVE.COM 
Streaming Media" libraries to do RTSP.  When designing your own 
application, you may find it useful to use the "openRTSP" code as a 
guideline, but it would probably not be a good idea to merely try to morph 
the existing code into doing what you want.

>I think I need to use the SPEED header, but I don't know where to hook
>it in testProgs/openRTSP.cpp . It has to be a header of the SETUP or the
>PLAY request ?

RTSP requests are implemented in the file "liveMedia/RTSPClient.cpp" in the 
"LIVE.COM Streaming Media" libraries.  This is the file to modify if you 
wish to add headers to RTSP requests.

>I am thinking of using JMF to play the stored streams. Do you have any
>experience with these things (architecture, codecs that WORK etc.)

No, not I, although others on the "live-devel" mailing list may have 
experience with JMF.  (If, instead, you were willing to use C/C++, you 
could use "mplayer" to play the streams - see 
<http://www.live.com/mplayer/> for details.)

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>



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Hello All!

I'm using "Live" tools to test the RTSP protocol...
so I wish to use the new version of "mplayer" supporting
RTSP/RTP streaming too.
I follow all the directions include in your mplayer web page
in order to install the software, but when I try to do "make"
during the installation some errors occur. This is what I can
read from my monitor:

"...
demux_rtp.cpp: In function `void demux_open_rtp(demuxer_t*)':
demux_rtp.cpp:114: no matching function for call to
   `RTSPClient::createNew(UsageEnvironment&, int&, const char[8])'
/usr/local/lib/live/liveMedia/include/RTSPClient.hh:43: candidates are:
static
   RTSPClient* RTSPClient::createNew(UsageEnvironment&, int = 0)
make[1]: *** [demux_rtp.o] Error 1
make[1]: Leaving directory 
`/home/tomosan/prove/mplayer-rtsp/cvs/MPlayer-2002091
1/libmpdemux'
make: *** [libmpdemux/libmpdemux.a] Error 2
"

Could someone help me to solve this problem?

Thank you, Santino Tomo




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Subject: [Live-devel] why different files for RTCP?
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X-Original-Date: Wed, 11 Sep 2002 16:37:35 +0530
Date: Wed, 11 Sep 2002 16:37:35 +0530

Hi Ross,
I wanted to know as to why have you kept the RFC specific implementation 
of RTCP in a different C (rtcp_from_spec.c and .h) files when your 
entire application is implemented in C++? Does the RFC restricts in 
using its code in a modified form? Please give an insight in this.
thanking you in advance
Prashanth



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Subject: [Live-devel] parallel rtsp client !!!
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 If I understand you correctly, I think you can do this - but it would 
require significant modification to the "openRTSP" source code.

I.e., you can use the RTSP "DESCRIBE" command on several different 
"rtsp://" URLs to retrieve several different SDP descriptions, *provided 
that* the "rtsp://" URLs all point to the same server. (If that's not the 
case, then you will need to use different "RTSPClient" objects.)

Then, once you have each SDP descriptions, you can create a separate 
"MediaSession" object from each.

However, you may just find it simplest to just run "openRTSP" multiple 
types, once for each "rtsp://" URL!

Ross.

Hi Ross

Please find here the code of my new version of "OpenRTSP". I don't know why subsession1->rtpSource() and subsession2->rtpSouurce() return both NULL. please help me to fixe this code I need on my master project. thanks in advance.

The code


#include "liveMedia.hh"
#include "GroupsockHelper.hh"
#include "BasicUsageEnvironment.hh"

 


file://#include "QuickTimeFileSink.hh"

#if defined(__WIN32__) || defined(_WIN32)
#define snprintf _snprintf
#else
#include <stdlib.h>
#include <signal.h>
#include <string.h>
#include <sys/time.h>
#define USE_SIGNALS 1 
#endif

static int Idunno;
char const* progName;
UsageEnvironment* env;
struct timeval startTime;
TaskToken currentTimerTask = NULL;
float endTime = 0;
float endTimeSlop = 5.0;

RTSPClient* rtspClient = NULL;
RTSPClient* rtspClient1 = NULL;
RTSPClient* rtspClient2 = NULL;

MediaSession* session = NULL;
MediaSession* session1 = NULL;
MediaSession* session2 = NULL;

int simpleRTPoffsetArg = -1;

void shutdown(int exitCode = 1);
void setupRTSPStreams();
void subsessionAfterPlaying(void* clientData=NULL);
void subsessionByeHandler(void* clientData); 
void startPlayingRTSPStreams();
void sessionAfterPlaying(void* clientData);
file://void subsessionByeHandler(void* clientData);
void sessionTimerHandler(void* clientData);

void main()
{
  
  gettimeofday(&startTime, &Idunno);
 
  char* url = "rtsp://serveur3:554/";
  char* url1= "rtsp://serveur3:552/";
  char* url2 = "rtsp://serveur3:550/";
  
  BasicTaskScheduler scheduler;
  env = new BasicUsageEnvironment(scheduler);

  rtspClient = RTSPClient::createNew(*env, 1);
  rtspClient1 = RTSPClient::createNew(*env,1);
  rtspClient2 = RTSPClient::createNew(*env,1);

  char*  sdpDescription = rtspClient->describeURL(url);
  char*  sdpDescription1 = rtspClient1->describeURL(url1);
  char*  sdpDescription2 = rtspClient2->describeURL(url2);

  Boolean madeProgress = False;
  

  if ((sdpDescription == NULL)||(sdpDescription1 == NULL)||
      (sdpDescription2 == NULL))
  {
    fprintf(stderr, "Failed to get a SDP description from URL \"%s\": %s\n",
     url, env->getResultMsg());
    shutdown();
  }

  session = MediaSession::createNew(*env, sdpDescription);
  delete sdpDescription;
  
  session1 = MediaSession::createNew(*env, sdpDescription1);
  delete sdpDescription1;
  
  session2 = MediaSession::createNew(*env, sdpDescription2);
  delete sdpDescription2;
  
   
  

  if ((session == NULL)||(session1 == NULL)||
   (session2 == NULL))
  {
    fprintf(stderr, "Failed to create a MediaSession object from the SDP description: %s\n", env->getResultMsg());
    shutdown();
  } else 
 if (!session->hasSubsessions()||!session1->hasSubsessions()||!session2->hasSubsessions()) {
    fprintf(stderr, "This session has no media subsessions (i.e., \"m=\" lines)\n");
    shutdown();
  }

  
  
  MediaSubsession *subsession;
  MediaSubsession *subsession1;
  MediaSubsession *subsession2;
  file://MediaSubsession *subsession;
  
  MediaSubsessionIterator iter(*session);
  MediaSubsessionIterator iter1(*session1);
  MediaSubsessionIterator iter2(*session2);
  
  
  while (((subsession =iter.next()) != NULL)&&
      ((subsession1 =iter1.next()) != NULL)&&
   ((subsession2 =iter2.next()) != NULL))
  {
  

 if (!subsession->initiate(simpleRTPoffsetArg)&&!subsession1->initiate(simpleRTPoffsetArg)
    &&!subsession2->initiate(simpleRTPoffsetArg))
 {
  fprintf(stderr, "Unable to create receiver for \"%s/%s\" subsession: %s\n",
  subsession->mediumName(), subsession->codecName(),
  env->getResultMsg());
 }
 else
 {
   fprintf(stderr, "Created receiver for \"%s/%s\" subsession (client ports %d-%d)\n",
   subsession1->mediumName(), subsession1->codecName(),
   subsession1->clientPortNum(), subsession1->clientPortNum()+1);
   fprintf(stderr, "Created receiver for \"%s/%s\" subsession (client ports %d-%d)\n",
   subsession->mediumName(), subsession->codecName(),
   subsession->clientPortNum(), subsession->clientPortNum()+1);
       
        fprintf(stderr, "Created receiver for \"%s/%s\" subsession (client ports %d-%d)\n",
   subsession2->mediumName(), subsession2->codecName(),
   subsession2->clientPortNum(), subsession2->clientPortNum()+1);   
     madeProgress = True;
 }
    
 if(subsession1->rtpSource()==NULL) printf("session1 \n");
 if(subsession->rtpSource()==NULL) printf("session \n");
    
    if(subsession2->rtpSource()==NULL) printf("session2 \n");
 
 
 if ( (subsession->rtpSource()!= NULL)&&
   (subsession1->rtpSource()!= NULL)&&
      (subsession2->rtpSource()!= NULL))
 {
   unsigned const thresh = 1000000; // 1 second 
   subsession->rtpSource()->setPacketReorderingThresholdTime(thresh);
      subsession1->rtpSource()->setPacketReorderingThresholdTime(thresh);
      subsession2->rtpSource()->setPacketReorderingThresholdTime(thresh);
 }
    else
   madeProgress = False;

    if (!madeProgress) shutdown(0);

    setupRTSPStreams();

    madeProgress = False;
      
   iter.reset();
   iter1.reset();
   iter2.reset();
     
   while (((subsession = iter.next()) != NULL)&&
       ((subsession1 = iter1.next()) != NULL)&&
    ((subsession2 = iter.next()) != NULL)) 
   {
   
    if ((subsession->readSource() == NULL)||
   (subsession1->readSource() == NULL)||
   (subsession2->readSource() == NULL)) continue; // was not initiated
 
      // Create an output file for each desired stream:
       char outFileName[1000];
          char outFileName1[1000];
          char outFileName2[1000];
    
    static unsigned streamCounter = 0;
    static unsigned streamCounter1 = 0;
    static unsigned streamCounter2 = 0;

       snprintf(outFileName, sizeof outFileName, "%s-%s-%d",
     subsession->mediumName(), subsession->codecName(),
     ++streamCounter);
          snprintf(outFileName1, sizeof outFileName1, "%s-%s-%d",
     subsession1->mediumName(), subsession1->codecName(),
     ++streamCounter1);
          snprintf(outFileName2, sizeof outFileName2, "%s-%s-%d",
     subsession2->mediumName(), subsession2->codecName(),
     ++streamCounter2);
   

    file://sprintf(outFileName, "stdout");
    
    subsession->sink = FileSink::createNew(*env, outFileName);
    subsession1->sink = FileSink::createNew(*env, outFileName1);
          subsession2->sink = FileSink::createNew(*env, outFileName2);
    
    if ((subsession->sink == NULL)||(subsession1->sink == NULL)||
     (subsession2->sink == NULL))
    {
     fprintf(stderr, "Failed to create FileSink for \"%s\": %s\n",
      outFileName, env->getResultMsg());
    }
    else 
    {
     if (1/*singleMedium == NULL*/) 
     {
      fprintf(stderr, "Created output file: \"%s\"\n", outFileName);
     }
     else 
     {
      fprintf(stderr, "Outputting data from the \"%s/%s\" subsession to 'stdout'\n",
       subsession->mediumName(), subsession->codecName());
     }

     subsession->sink->startPlaying(*(subsession->readSource()),
      subsessionAfterPlaying,
      subsession);
     subsession1->sink->startPlaying(*(subsession1->readSource()),
      subsessionAfterPlaying,
      subsession1);
     subsession2->sink->startPlaying(*(subsession2->readSource()),
      subsessionAfterPlaying,
      subsession2);
   // Also set a handler to be called if a RTCP "BYE" arrives
   // for this subsession:
     if ((subsession->rtcpInstance() != NULL)&&
      (subsession1->rtcpInstance() != NULL)&&
      (subsession2->rtcpInstance() != NULL))
     {
      subsession->rtcpInstance()->setByeHandler(subsessionByeHandler,
       subsession);
      subsession1->rtcpInstance()->setByeHandler(subsessionByeHandler,
       subsession1);
      subsession2->rtcpInstance()->setByeHandler(subsessionByeHandler,
       subsession2);
     }
     madeProgress = True;

    }
    if (!madeProgress) shutdown(0);

   }

  }
  
  startPlayingRTSPStreams();

  env->taskScheduler().blockMyself(); // does not return

  return ; // only to prevent compiler warning


}
  

void shutdown(int exitCode) 
{
  if (env != NULL) {
    env->taskScheduler().unscheduleDelayedTask(currentTimerTask);
  }

  if ((rtspClient != NULL)&&
   (rtspClient1 != NULL)&&
   (rtspClient2 != NULL))
  {
    // Teardown any outstanding RTSP sessions, and close media sinks:
    if ((session != NULL)&&
  (session1 != NULL)&&
  (session2 != NULL))
 {
      file://tearDownRTSPStreams();
      file://closeMediaSinks();
    }

    // Then shut down the RTSP client:
    Medium::close(rtspClient);
    Medium::close(rtspClient1);
    Medium::close(rtspClient2);
  }

  // Shutdown the Media session (and all of the RTP/RTCP subsessions):
  Medium::close(session);
  Medium::close(session1);
  Medium::close(session2);

  // Adios...
  exit(exitCode);
}

void setupRTSPStreams()
{
  MediaSubsessionIterator iter(*session);
  MediaSubsessionIterator iter1(*session1);
  MediaSubsessionIterator iter2(*session2);
  
  MediaSubsession *subsession;
  MediaSubsession *subsession1;
  MediaSubsession *subsession2;
  
  Boolean madeProgress = False;

  while (((subsession = iter.next()) != NULL)&& 
      ((subsession1 = iter1.next()) != NULL)&&
   ((subsession2 = iter2.next()) != NULL))
  {
    if ((subsession->clientPortNum() == 0)||
  (subsession1->clientPortNum() == 0)||
  (subsession2->clientPortNum() == 0)) 
  continue; // port # was not set

    if (!rtspClient->setupMediaSubsession(*subsession)||
  !rtspClient->setupMediaSubsession(*subsession1)||
  !rtspClient->setupMediaSubsession(*subsession2)) 
 {
      fprintf(stderr,
       "Failed to issue RTSP \"SETUP\" on \"%s/%s\" subsession: %s\n",
       subsession->mediumName(), subsession->codecName(),
       env->getResultMsg());
    } else {
      fprintf(stderr, "Issued RTSP \"SETUP\" on \"%s/%s\" subsession (client ports %d-%d)\n",
       subsession->mediumName(), subsession->codecName(),
       subsession->clientPortNum(), subsession->clientPortNum()+1);
      madeProgress = True;
    }
  }
  if (!madeProgress) shutdown();
}

void subsessionAfterPlaying(void* clientData)
 {
  // Begin by closing this media subsession:
  MediaSubsession* subsession = (MediaSubsession*)clientData;
  Medium::close(subsession->sink);
  subsession->sink = NULL;

  // Next, check whether *all* subsessions have now been closed:
  MediaSession& session = subsession->parentSession();
  MediaSubsessionIterator iter(session);
  while ((subsession = iter.next()) != NULL) {
    if (subsession->sink != NULL) return; // this subsession is still active
  }

  // All subsessions have now been closed
  sessionAfterPlaying(NULL);
}

void subsessionByeHandler(void* clientData) 
{
  struct timeval timeNow;
  gettimeofday(&timeNow, &Idunno);
  unsigned secsDiff = timeNow.tv_sec - startTime.tv_sec;

  MediaSubsession* subsession = (MediaSubsession*)clientData;
  fprintf(stderr, "Received RTCP \"BYE\" on \"%s/%s\" subsession (after %d seconds)\n",
   subsession->mediumName(), subsession->codecName(), secsDiff);

  // Act now as if the subsession had closed:
  subsessionAfterPlaying(subsession);
}

void startPlayingRTSPStreams()
{
  MediaSubsessionIterator iter(*session);
  MediaSubsessionIterator iter1(*session1);
  MediaSubsessionIterator iter2(*session2);
 
  MediaSubsession *subsession;
  MediaSubsession *subsession1;
  MediaSubsession *subsession2;
  
  Boolean madeProgress = False;
  while (((subsession = iter.next()) != NULL)&&
      ((subsession1 = iter1.next()) != NULL)&&
   ((subsession2 = iter2.next()) != NULL))  
  {
    if ((subsession->sessionId == NULL)||
        (subsession1->sessionId == NULL)||
  (subsession2->sessionId == NULL)) continue; file://no RTSP sess in progress

    if ((!rtspClient->playMediaSubsession(*subsession))||
  (!rtspClient1->playMediaSubsession(*subsession1))||
  (!rtspClient2->playMediaSubsession(*subsession2))) 
 {
      fprintf(stderr, "RTSP \"PLAY\" command on \"%s/%s\" subsession failed: %s\n",
       subsession->mediumName(), subsession->codecName(),
       env->getResultMsg());
      shutdown();
    } else {
      fprintf(stderr, "Issued RTSP \"PLAY\" command on \"%s/%s\" subsession\n",
       subsession->mediumName(), subsession->codecName());
      madeProgress = True;
    }
  }
  if (!madeProgress) shutdown();

  // Figure out how long to delay (if at all) before shutting down, or
  // repeating the playing
  Boolean timerIsBeingUsed = False;
  float totalEndTime = endTime;
  if (endTime == 0) endTime = session->playEndTime(); // use SDP end time
  if (endTime > 0) {
    float const maxDelayTime = (float)( ((unsigned)0xFFFFFFFF)/1000000.0 );
    if (endTime > maxDelayTime) {
      fprintf(stderr, "Warning: specified end time %g exceeds maximum %g; will not do a delayed shutdown\n", endTime, maxDelayTime);
      endTime = 0.0;
    } else {
      timerIsBeingUsed = True;
      totalEndTime = endTime + endTimeSlop;

      int uSecsToDelay = (int)(totalEndTime*1000000.0);
      currentTimerTask = env->taskScheduler().scheduleDelayedTask(
         uSecsToDelay, (TaskProc*)sessionTimerHandler, (void*)NULL);
    }
  }
  
}

/*void subsessionByeHandler(void* clientData) 
{
  struct timeval timeNow;
  gettimeofday(&timeNow, &Idunno);
  unsigned secsDiff = timeNow.tv_sec - startTime.tv_sec;

  MediaSubsession* subsession = (MediaSubsession*)clientData;
  fprintf(stderr, "Received RTCP \"BYE\" on \"%s/%s\" subsession (after %d seconds)\n",
   subsession->mediumName(), subsession->codecName(), secsDiff);

  // Act now as if the subsession had closed:
  subsessionAfterPlaying(subsession);
}*/

void sessionTimerHandler(void* /*clientData*/)
{
  currentTimerTask = NULL;

  sessionAfterPlaying(NULL);
}

void sessionAfterPlaying(void* /*clientData*/) 
{
  startPlayingRTSPStreams();
  
}

Achraf,

 







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--0-644655950-1031756399=:93469
Content-Type: text/html; charset=iso-8859-1
Content-Transfer-Encoding: 8bit

<P>&nbsp;If I understand you correctly, I think you can do this - but it would <BR>require significant modification to the "openRTSP" source code.<BR><BR>I.e., you can use the RTSP "DESCRIBE" command on several different <BR>"rtsp://" URLs to retrieve several different SDP descriptions, *provided <BR>that* the "rtsp://" URLs all point to the same server. (If that's not the <BR>case, then you will need to use different "RTSPClient" objects.)<BR><BR>Then, once you have each SDP descriptions, you can create a separate <BR>"MediaSession" object from each.<BR><BR>However, you may just find it simplest to just run "openRTSP" multiple <BR>types, once for each "rtsp://" URL!<BR><BR>Ross.</P>
<P>Hi Ross</P>
<P>Please find here the code of my new version of "OpenRTSP".&nbsp;I don't know why subsession1-&gt;rtpSource() and subsession2-&gt;rtpSouurce()&nbsp;return both NULL. please help me to fixe this code I need on my master project. thanks in advance.</P>
<P>The code</P>
<P><BR>#include "liveMedia.hh"<BR>#include "GroupsockHelper.hh"<BR>#include "BasicUsageEnvironment.hh"</P>
<P>&nbsp;</P>
<P><BR><A href="file://#include">file://#include</A> "QuickTimeFileSink.hh"</P>
<P>#if defined(__WIN32__) || defined(_WIN32)<BR>#define snprintf _snprintf<BR>#else<BR>#include &lt;stdlib.h&gt;<BR>#include &lt;signal.h&gt;<BR>#include &lt;string.h&gt;<BR>#include &lt;sys/time.h&gt;<BR>#define USE_SIGNALS 1 <BR>#endif</P>
<P>static int Idunno;<BR>char const* progName;<BR>UsageEnvironment* env;<BR>struct timeval startTime;<BR>TaskToken currentTimerTask = NULL;<BR>float endTime = 0;<BR>float endTimeSlop = 5.0;</P>
<P>RTSPClient* rtspClient = NULL;<BR>RTSPClient* rtspClient1 = NULL;<BR>RTSPClient* rtspClient2 = NULL;</P>
<P>MediaSession* session = NULL;<BR>MediaSession* session1 = NULL;<BR>MediaSession* session2 = NULL;</P>
<P>int simpleRTPoffsetArg = -1;</P>
<P>void shutdown(int exitCode = 1);<BR>void setupRTSPStreams();<BR>void subsessionAfterPlaying(void* clientData=NULL);<BR>void subsessionByeHandler(void* clientData); <BR>void startPlayingRTSPStreams();<BR>void sessionAfterPlaying(void* clientData);<BR><A href="file://void">file://void</A> subsessionByeHandler(void* clientData);<BR>void sessionTimerHandler(void* clientData);</P>
<P>void main()<BR>{<BR>&nbsp; <BR>&nbsp; gettimeofday(&amp;startTime, &amp;Idunno);<BR>&nbsp;<BR>&nbsp; char* url = "rtsp://serveur3:554/";<BR>&nbsp; char* url1= "rtsp://serveur3:552/";<BR>&nbsp; char* url2 = "rtsp://serveur3:550/";<BR>&nbsp; <BR>&nbsp; BasicTaskScheduler scheduler;<BR>&nbsp; env = new BasicUsageEnvironment(scheduler);</P>
<P>&nbsp; rtspClient = RTSPClient::createNew(*env, 1);<BR>&nbsp; rtspClient1 = RTSPClient::createNew(*env,1);<BR>&nbsp; rtspClient2 = RTSPClient::createNew(*env,1);</P>
<P>&nbsp; char*&nbsp; sdpDescription = rtspClient-&gt;describeURL(url);<BR>&nbsp; char*&nbsp; sdpDescription1 = rtspClient1-&gt;describeURL(url1);<BR>&nbsp; char*&nbsp; sdpDescription2 = rtspClient2-&gt;describeURL(url2);</P>
<P>&nbsp; Boolean madeProgress = False;<BR>&nbsp; </P>
<P>&nbsp; if ((sdpDescription == NULL)||(sdpDescription1 == NULL)||<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; (sdpDescription2 == NULL))<BR>&nbsp; {<BR>&nbsp;&nbsp;&nbsp; fprintf(stderr, "Failed to get a SDP description from URL \"%s\": %s\n",<BR>&nbsp;&nbsp;&nbsp;&nbsp; url, env-&gt;getResultMsg());<BR>&nbsp;&nbsp;&nbsp; shutdown();<BR>&nbsp; }</P>
<P>&nbsp; session = MediaSession::createNew(*env, sdpDescription);<BR>&nbsp; delete sdpDescription;<BR>&nbsp; <BR>&nbsp; session1 = MediaSession::createNew(*env, sdpDescription1);<BR>&nbsp; delete sdpDescription1;<BR>&nbsp; <BR>&nbsp; session2 = MediaSession::createNew(*env, sdpDescription2);<BR>&nbsp; delete sdpDescription2;<BR>&nbsp; <BR>&nbsp;&nbsp; <BR>&nbsp; </P>
<P>&nbsp; if ((session == NULL)||(session1 == NULL)||<BR>&nbsp;&nbsp; (session2 == NULL))<BR>&nbsp; {<BR>&nbsp;&nbsp;&nbsp; fprintf(stderr, "Failed to create a MediaSession object from the SDP description: %s\n", env-&gt;getResultMsg());<BR>&nbsp;&nbsp;&nbsp; shutdown();<BR>&nbsp; } else <BR>&nbsp;if (!session-&gt;hasSubsessions()||!session1-&gt;hasSubsessions()||!session2-&gt;hasSubsessions()) {<BR>&nbsp;&nbsp;&nbsp; fprintf(stderr, "This session has no media subsessions (i.e., \"m=\" lines)\n");<BR>&nbsp;&nbsp;&nbsp; shutdown();<BR>&nbsp; }</P>
<P>&nbsp; <BR>&nbsp; <BR>&nbsp; MediaSubsession *subsession;<BR>&nbsp; MediaSubsession *subsession1;<BR>&nbsp; MediaSubsession *subsession2;<BR>&nbsp; <A href="file://MediaSubsession">file://MediaSubsession</A> *subsession;<BR>&nbsp; <BR>&nbsp; MediaSubsessionIterator iter(*session);<BR>&nbsp; MediaSubsessionIterator iter1(*session1);<BR>&nbsp; MediaSubsessionIterator iter2(*session2);<BR>&nbsp; <BR>&nbsp; <BR>&nbsp; while (((subsession =iter.next()) != NULL)&amp;&amp;<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; ((subsession1 =iter1.next()) != NULL)&amp;&amp;<BR>&nbsp;&nbsp; ((subsession2 =iter2.next()) != NULL))<BR>&nbsp; {<BR>&nbsp; </P>
<P>&nbsp;if (!subsession-&gt;initiate(simpleRTPoffsetArg)&amp;&amp;!subsession1-&gt;initiate(simpleRTPoffsetArg)<BR>&nbsp;&nbsp;&nbsp; &amp;&amp;!subsession2-&gt;initiate(simpleRTPoffsetArg))<BR>&nbsp;{<BR>&nbsp;&nbsp;fprintf(stderr, "Unable to create receiver for \"%s/%s\" subsession: %s\n",<BR>&nbsp;&nbsp;subsession-&gt;mediumName(), subsession-&gt;codecName(),<BR>&nbsp;&nbsp;env-&gt;getResultMsg());<BR>&nbsp;}<BR>&nbsp;else<BR>&nbsp;{<BR>&nbsp;&nbsp; fprintf(stderr, "Created receiver for \"%s/%s\" subsession (client ports %d-%d)\n",<BR>&nbsp;&nbsp;&nbsp;subsession1-&gt;mediumName(), subsession1-&gt;codecName(),<BR>&nbsp;&nbsp;&nbsp;subsession1-&gt;clientPortNum(), subsession1-&gt;clientPortNum()+1);<BR>&nbsp;&nbsp; fprintf(stderr, "Created receiver for \"%s/%s\" subsession (client ports %d-%d)\n",<BR>&nbsp;&nbsp;&nbsp;subsession-&gt;mediumName(), subsession-&gt;codecName(),<BR>&nbsp;&nbsp;&nbsp;subsession-&gt;clientPortNum(), subsession-&gt;clientPortNum()+1);<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; fprintf(stderr, "Created receiver for \"%s/%s\" subsession (client ports %d-%d)\n",<BR>&nbsp;&nbsp;&nbsp;subsession2-&gt;mediumName(), subsession2-&gt;codecName(),<BR>&nbsp;&nbsp;&nbsp;subsession2-&gt;clientPortNum(), subsession2-&gt;clientPortNum()+1);&nbsp;&nbsp; <BR>&nbsp;&nbsp;&nbsp;&nbsp; madeProgress = True;<BR>&nbsp;}<BR>&nbsp;&nbsp;&nbsp; <BR>&nbsp;if(subsession1-&gt;rtpSource()==NULL) printf("session1 \n");<BR>&nbsp;if(subsession-&gt;rtpSource()==NULL) printf("session \n");<BR>&nbsp;&nbsp;&nbsp; <BR>&nbsp;&nbsp;&nbsp; if(subsession2-&gt;rtpSource()==NULL) printf("session2 \n");<BR>&nbsp;<BR>&nbsp;<BR>&nbsp;if ( (subsession-&gt;rtpSource()!= NULL)&amp;&amp;<BR>&nbsp;&nbsp; (subsession1-&gt;rtpSource()!= NULL)&amp;&amp;<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; (subsession2-&gt;rtpSource()!= NULL))<BR>&nbsp;{<BR>&nbsp;&nbsp; unsigned const thresh = 1000000; // 1 second <BR>&nbsp;&nbsp; subsession-&gt;rtpSource()-&gt;setPacketReorderingThresholdTime(thresh);<BR>&nbsp;&n
bsp;&nbsp;&nbsp;&nbsp; subsession1-&gt;rtpSource()-&gt;setPacketReorderingThresholdTime(thresh);<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; subsession2-&gt;rtpSource()-&gt;setPacketReorderingThresholdTime(thresh);<BR>&nbsp;}<BR>&nbsp;&nbsp;&nbsp; else<BR>&nbsp;&nbsp; madeProgress = False;</P>
<P>&nbsp;&nbsp;&nbsp; if (!madeProgress) shutdown(0);</P>
<P>&nbsp;&nbsp;&nbsp; setupRTSPStreams();</P>
<P>&nbsp;&nbsp;&nbsp; madeProgress = False;<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <BR>&nbsp;&nbsp; iter.reset();<BR>&nbsp;&nbsp; iter1.reset();<BR>&nbsp;&nbsp; iter2.reset();<BR>&nbsp;&nbsp;&nbsp;&nbsp; <BR>&nbsp;&nbsp; while (((subsession = iter.next()) != NULL)&amp;&amp;<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; ((subsession1 = iter1.next()) != NULL)&amp;&amp;<BR>&nbsp;&nbsp;&nbsp; ((subsession2 = iter.next()) != NULL)) <BR>&nbsp;&nbsp; {<BR>&nbsp;&nbsp; <BR>&nbsp;&nbsp;&nbsp; if ((subsession-&gt;readSource() == NULL)||<BR>&nbsp;&nbsp;&nbsp;(subsession1-&gt;readSource() == NULL)||<BR>&nbsp;&nbsp;&nbsp;(subsession2-&gt;readSource() == NULL)) continue; // was not initiated<BR>&nbsp;<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; // Create an output file for each desired stream:<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; char outFileName[1000];<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; char outFileName1[1000];<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; char outFileName2[1000];<BR>&nbsp;&nbsp;&nbsp; <BR>&nbsp;&nbsp;&nbsp; static unsigned streamCounter = 0;<BR>&nbsp;&nbsp;&nbsp; static unsigned streamCounter1 = 0;<BR>&nbsp;&nbsp;&nbsp; static unsigned streamCounter2 = 0;</P>
<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; snprintf(outFileName, sizeof outFileName, "%s-%s-%d",<BR>&nbsp;&nbsp;&nbsp;&nbsp; subsession-&gt;mediumName(), subsession-&gt;codecName(),<BR>&nbsp;&nbsp;&nbsp;&nbsp; ++streamCounter);<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; snprintf(outFileName1, sizeof outFileName1, "%s-%s-%d",<BR>&nbsp;&nbsp;&nbsp;&nbsp; subsession1-&gt;mediumName(), subsession1-&gt;codecName(),<BR>&nbsp;&nbsp;&nbsp;&nbsp; ++streamCounter1);<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; snprintf(outFileName2, sizeof outFileName2, "%s-%s-%d",<BR>&nbsp;&nbsp;&nbsp;&nbsp; subsession2-&gt;mediumName(), subsession2-&gt;codecName(),<BR>&nbsp;&nbsp;&nbsp;&nbsp; ++streamCounter2);<BR>&nbsp;&nbsp; </P>
<P>&nbsp;&nbsp;&nbsp; <A href="file://sprintf">file://sprintf</A>(outFileName, "stdout");<BR>&nbsp;&nbsp;&nbsp; <BR>&nbsp;&nbsp;&nbsp; subsession-&gt;sink = FileSink::createNew(*env, outFileName);<BR>&nbsp;&nbsp;&nbsp; subsession1-&gt;sink = FileSink::createNew(*env, outFileName1);<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; subsession2-&gt;sink = FileSink::createNew(*env, outFileName2);<BR>&nbsp;&nbsp;&nbsp; <BR>&nbsp;&nbsp;&nbsp; if ((subsession-&gt;sink == NULL)||(subsession1-&gt;sink == NULL)||<BR>&nbsp;&nbsp;&nbsp;&nbsp; (subsession2-&gt;sink == NULL))<BR>&nbsp;&nbsp;&nbsp; {<BR>&nbsp;&nbsp;&nbsp;&nbsp; fprintf(stderr, "Failed to create FileSink for \"%s\": %s\n",<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; outFileName, env-&gt;getResultMsg());<BR>&nbsp;&nbsp;&nbsp; }<BR>&nbsp;&nbsp;&nbsp; else <BR>&nbsp;&nbsp;&nbsp; {<BR>&nbsp;&nbsp;&nbsp;&nbsp; if (1/*singleMedium == NULL*/) <BR>&nbsp;&nbsp;&nbsp;&nbsp; {<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; fprintf(stderr, "Created output file: \"%s\"\n", outFileName);<BR>&nbsp;&nbsp;&nbsp;&nbsp; }<BR>&nbsp;&nbsp;&nbsp;&nbsp; else <BR>&nbsp;&nbsp;&nbsp;&nbsp; {<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; fprintf(stderr, "Outputting data from the \"%s/%s\" subsession to 'stdout'\n",<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; subsession-&gt;mediumName(), subsession-&gt;codecName());<BR>&nbsp;&nbsp;&nbsp;&nbsp; }</P>
<P>&nbsp;&nbsp;&nbsp;&nbsp; subsession-&gt;sink-&gt;startPlaying(*(subsession-&gt;readSource()),<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; subsessionAfterPlaying,<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; subsession);<BR>&nbsp;&nbsp;&nbsp;&nbsp; subsession1-&gt;sink-&gt;startPlaying(*(subsession1-&gt;readSource()),<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; subsessionAfterPlaying,<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; subsession1);<BR>&nbsp;&nbsp;&nbsp;&nbsp; subsession2-&gt;sink-&gt;startPlaying(*(subsession2-&gt;readSource()),<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; subsessionAfterPlaying,<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; subsession2);<BR>&nbsp;&nbsp; // Also set a handler to be called if a RTCP "BYE" arrives<BR>&nbsp;&nbsp; // for this subsession:<BR>&nbsp;&nbsp;&nbsp;&nbsp; if ((subsession-&gt;rtcpInstance() != NULL)&amp;&amp;<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; (subsession1-&gt;rtcpInstance() != NULL)&amp;&amp;<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; (subsession2-&gt;rtcpInstance() != NULL))<BR>&nbsp;&nbsp;&nbsp;&nbsp; {<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; subsession-&gt;rtcpInstance()-&gt;setByeHandler(subsessionByeHandler,<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; subsession);<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; subsession1-&gt;rtcpInstance()-&gt;setByeHandler(subsessionByeHandler,<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; subsession1);<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; subsession2-&gt;rtcpInstance()-&gt;setByeHandler(subsessionByeHandler,<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; subsession2);<BR>&nbsp;&nbsp;&nbsp;&nbsp; }<BR>&nbsp;&nbsp;&nbsp;&nbsp; madeProgress = True;</P>
<P>&nbsp;&nbsp;&nbsp; }<BR>&nbsp;&nbsp;&nbsp; if (!madeProgress) shutdown(0);</P>
<P>&nbsp;&nbsp; }</P>
<P>&nbsp; }<BR>&nbsp; <BR>&nbsp; startPlayingRTSPStreams();</P>
<P>&nbsp; env-&gt;taskScheduler().blockMyself(); // does not return</P>
<P>&nbsp; return ; // only to prevent compiler warning</P>
<P><BR>}<BR>&nbsp; </P>
<P>void shutdown(int exitCode) <BR>{<BR>&nbsp; if (env != NULL) {<BR>&nbsp;&nbsp;&nbsp; env-&gt;taskScheduler().unscheduleDelayedTask(currentTimerTask);<BR>&nbsp; }</P>
<P>&nbsp; if ((rtspClient != NULL)&amp;&amp;<BR>&nbsp;&nbsp; (rtspClient1 != NULL)&amp;&amp;<BR>&nbsp;&nbsp; (rtspClient2 != NULL))<BR>&nbsp; {<BR>&nbsp;&nbsp;&nbsp; // Teardown any outstanding RTSP sessions, and close media sinks:<BR>&nbsp;&nbsp;&nbsp; if ((session != NULL)&amp;&amp;<BR>&nbsp;&nbsp;(session1 != NULL)&amp;&amp;<BR>&nbsp;&nbsp;(session2 != NULL))<BR>&nbsp;{<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <A href="file://tearDownRTSPStreams">file://tearDownRTSPStreams</A>();<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <A href="file://closeMediaSinks">file://closeMediaSinks</A>();<BR>&nbsp;&nbsp;&nbsp; }</P>
<P>&nbsp;&nbsp;&nbsp; // Then shut down the RTSP client:<BR>&nbsp;&nbsp;&nbsp; Medium::close(rtspClient);<BR>&nbsp;&nbsp;&nbsp; Medium::close(rtspClient1);<BR>&nbsp;&nbsp;&nbsp; Medium::close(rtspClient2);<BR>&nbsp; }</P>
<P>&nbsp; // Shutdown the Media session (and all of the RTP/RTCP subsessions):<BR>&nbsp; Medium::close(session);<BR>&nbsp; Medium::close(session1);<BR>&nbsp; Medium::close(session2);</P>
<P>&nbsp; // Adios...<BR>&nbsp; exit(exitCode);<BR>}</P>
<P>void setupRTSPStreams()<BR>{<BR>&nbsp; MediaSubsessionIterator iter(*session);<BR>&nbsp; MediaSubsessionIterator iter1(*session1);<BR>&nbsp; MediaSubsessionIterator iter2(*session2);<BR>&nbsp; <BR>&nbsp; MediaSubsession *subsession;<BR>&nbsp; MediaSubsession *subsession1;<BR>&nbsp; MediaSubsession *subsession2;<BR>&nbsp; <BR>&nbsp; Boolean madeProgress = False;</P>
<P>&nbsp; while (((subsession = iter.next()) != NULL)&amp;&amp; <BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; ((subsession1 = iter1.next()) != NULL)&amp;&amp;<BR>&nbsp;&nbsp; ((subsession2 = iter2.next()) != NULL))<BR>&nbsp; {<BR>&nbsp;&nbsp;&nbsp; if ((subsession-&gt;clientPortNum() == 0)||<BR>&nbsp;&nbsp;(subsession1-&gt;clientPortNum() == 0)||<BR>&nbsp;&nbsp;(subsession2-&gt;clientPortNum() == 0)) <BR>&nbsp;&nbsp;continue; // port # was not set</P>
<P>&nbsp;&nbsp;&nbsp; if (!rtspClient-&gt;setupMediaSubsession(*subsession)||<BR>&nbsp;&nbsp;!rtspClient-&gt;setupMediaSubsession(*subsession1)||<BR>&nbsp;&nbsp;!rtspClient-&gt;setupMediaSubsession(*subsession2)) <BR>&nbsp;{<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; fprintf(stderr,<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; "Failed to issue RTSP \"SETUP\" on \"%s/%s\" subsession: %s\n",<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; subsession-&gt;mediumName(), subsession-&gt;codecName(),<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; env-&gt;getResultMsg());<BR>&nbsp;&nbsp;&nbsp; } else {<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; fprintf(stderr, "Issued RTSP \"SETUP\" on \"%s/%s\" subsession (client ports %d-%d)\n",<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; subsession-&gt;mediumName(), subsession-&gt;codecName(),<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; subsession-&gt;clientPortNum(), subsession-&gt;clientPortNum()+1);<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; madeProgress = True;<BR>&nbsp;&nbsp;&nbsp; }<BR>&nbsp; }<BR>&nbsp; if (!madeProgress) shutdown();<BR>}</P>
<P>void subsessionAfterPlaying(void* clientData)<BR>&nbsp;{<BR>&nbsp; // Begin by closing this media subsession:<BR>&nbsp; MediaSubsession* subsession = (MediaSubsession*)clientData;<BR>&nbsp; Medium::close(subsession-&gt;sink);<BR>&nbsp; subsession-&gt;sink = NULL;</P>
<P>&nbsp; // Next, check whether *all* subsessions have now been closed:<BR>&nbsp; MediaSession&amp; session = subsession-&gt;parentSession();<BR>&nbsp; MediaSubsessionIterator iter(session);<BR>&nbsp; while ((subsession = iter.next()) != NULL) {<BR>&nbsp;&nbsp;&nbsp; if (subsession-&gt;sink != NULL) return; // this subsession is still active<BR>&nbsp; }</P>
<P>&nbsp; // All subsessions have now been closed<BR>&nbsp; sessionAfterPlaying(NULL);<BR>}</P>
<P>void subsessionByeHandler(void* clientData) <BR>{<BR>&nbsp; struct timeval timeNow;<BR>&nbsp; gettimeofday(&amp;timeNow, &amp;Idunno);<BR>&nbsp; unsigned secsDiff = timeNow.tv_sec - startTime.tv_sec;</P>
<P>&nbsp; MediaSubsession* subsession = (MediaSubsession*)clientData;<BR>&nbsp; fprintf(stderr, "Received RTCP \"BYE\" on \"%s/%s\" subsession (after %d seconds)\n",<BR>&nbsp;&nbsp; subsession-&gt;mediumName(), subsession-&gt;codecName(), secsDiff);</P>
<P>&nbsp; // Act now as if the subsession had closed:<BR>&nbsp; subsessionAfterPlaying(subsession);<BR>}</P>
<P>void startPlayingRTSPStreams()<BR>{<BR>&nbsp; MediaSubsessionIterator iter(*session);<BR>&nbsp; MediaSubsessionIterator iter1(*session1);<BR>&nbsp; MediaSubsessionIterator iter2(*session2);<BR>&nbsp;<BR>&nbsp; MediaSubsession *subsession;<BR>&nbsp; MediaSubsession *subsession1;<BR>&nbsp; MediaSubsession *subsession2;<BR>&nbsp; <BR>&nbsp; Boolean madeProgress = False;<BR>&nbsp; while (((subsession = iter.next()) != NULL)&amp;&amp;<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; ((subsession1 = iter1.next()) != NULL)&amp;&amp;<BR>&nbsp;&nbsp; ((subsession2 = iter2.next()) != NULL))&nbsp; <BR>&nbsp; {<BR>&nbsp;&nbsp;&nbsp; if ((subsession-&gt;sessionId == NULL)||<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; (subsession1-&gt;sessionId == NULL)||<BR>&nbsp;&nbsp;(subsession2-&gt;sessionId == NULL)) continue; <A href="file://no">file://no</A> RTSP sess in progress</P>
<P>&nbsp;&nbsp;&nbsp; if ((!rtspClient-&gt;playMediaSubsession(*subsession))||<BR>&nbsp;&nbsp;(!rtspClient1-&gt;playMediaSubsession(*subsession1))||<BR>&nbsp;&nbsp;(!rtspClient2-&gt;playMediaSubsession(*subsession2))) <BR>&nbsp;{<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; fprintf(stderr, "RTSP \"PLAY\" command on \"%s/%s\" subsession failed: %s\n",<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; subsession-&gt;mediumName(), subsession-&gt;codecName(),<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; env-&gt;getResultMsg());<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; shutdown();<BR>&nbsp;&nbsp;&nbsp; } else {<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; fprintf(stderr, "Issued RTSP \"PLAY\" command on \"%s/%s\" subsession\n",<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; subsession-&gt;mediumName(), subsession-&gt;codecName());<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; madeProgress = True;<BR>&nbsp;&nbsp;&nbsp; }<BR>&nbsp; }<BR>&nbsp; if (!madeProgress) shutdown();</P>
<P>&nbsp; // Figure out how long to delay (if at all) before shutting down, or<BR>&nbsp; // repeating the playing<BR>&nbsp; Boolean timerIsBeingUsed = False;<BR>&nbsp; float totalEndTime = endTime;<BR>&nbsp; if (endTime == 0) endTime = session-&gt;playEndTime(); // use SDP end time<BR>&nbsp; if (endTime &gt; 0) {<BR>&nbsp;&nbsp;&nbsp; float const maxDelayTime = (float)( ((unsigned)0xFFFFFFFF)/1000000.0 );<BR>&nbsp;&nbsp;&nbsp; if (endTime &gt; maxDelayTime) {<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; fprintf(stderr, "Warning: specified end time %g exceeds maximum %g; will not do a delayed shutdown\n", endTime, maxDelayTime);<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; endTime = 0.0;<BR>&nbsp;&nbsp;&nbsp; } else {<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; timerIsBeingUsed = True;<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; totalEndTime = endTime + endTimeSlop;</P>
<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int uSecsToDelay = (int)(totalEndTime*1000000.0);<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; currentTimerTask = env-&gt;taskScheduler().scheduleDelayedTask(<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; uSecsToDelay, (TaskProc*)sessionTimerHandler, (void*)NULL);<BR>&nbsp;&nbsp;&nbsp; }<BR>&nbsp; }<BR>&nbsp; <BR>}</P>
<P>/*void subsessionByeHandler(void* clientData) <BR>{<BR>&nbsp; struct timeval timeNow;<BR>&nbsp; gettimeofday(&amp;timeNow, &amp;Idunno);<BR>&nbsp; unsigned secsDiff = timeNow.tv_sec - startTime.tv_sec;</P>
<P>&nbsp; MediaSubsession* subsession = (MediaSubsession*)clientData;<BR>&nbsp; fprintf(stderr, "Received RTCP \"BYE\" on \"%s/%s\" subsession (after %d seconds)\n",<BR>&nbsp;&nbsp; subsession-&gt;mediumName(), subsession-&gt;codecName(), secsDiff);</P>
<P>&nbsp; // Act now as if the subsession had closed:<BR>&nbsp; subsessionAfterPlaying(subsession);<BR>}*/</P>
<P>void sessionTimerHandler(void* /*clientData*/)<BR>{<BR>&nbsp; currentTimerTask = NULL;</P>
<P>&nbsp; sessionAfterPlaying(NULL);<BR>}</P>
<P>void sessionAfterPlaying(void* /*clientData*/) <BR>{<BR>&nbsp; startPlayingRTSPStreams();<BR>&nbsp; <BR>}</P>
<P>Achraf,</P>
<P>&nbsp;<BR><BR><BR><BR></P><p><br><hr size=1><a href="http://fr.mail.yahoo.com">Yahoo! Mail</a> -- Une adresse (at)yahoo.fr gratuite et en français !<br>
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Subject: Re: [Live-devel] mplayer-RTSP
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X-Original-Date: Wed, 11 Sep 2002 09:02:53 -0700
Date: Wed, 11 Sep 2002 09:02:53 -0700


>demux_rtp.cpp: In function `void demux_open_rtp(demuxer_t*)':
>demux_rtp.cpp:114: no matching function for call to
>    `RTSPClient::createNew(UsageEnvironment&, int&, const char[8])'
>/usr/local/lib/live/liveMedia/include/RTSPClient.hh:43: candidates are:
>static
>    RTSPClient* RTSPClient::createNew(UsageEnvironment&, int = 0)

You're using an old version of the "liveMedia" library (prior to 
2002.08.29).  Since then, the interface to "RTSPClient::createNew()" has 
changed.

If you download and build the newest version of the library code (from 
<http://www.live.com/liveMedia/public/>), you should then be able to link 
the "mplayer" code.

         Ross.



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X-Original-Date: Wed, 11 Sep 2002 09:10:27 -0700
Date: Wed, 11 Sep 2002 09:10:27 -0700

Check your "if" and "while" statements.  There are several places where you say
	if (case1&&case2&&case3) code
where you should have said
	if (case1||case2||case3) code
or vice versa.

There are also places where you say
	while (case1&&case2&&case3) code
where you should have said
	while (case1) code;
	while (case2) code;
	while (case3) code;

That's all the help you're going to get for now :-)  Remember that you have 
complete source code.

	Ross.



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Subject: [Live-devel] Lip synch
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X-Original-Date: Wed, 11 Sep 2002 17:14:33 +0100
Date: Wed, 11 Sep 2002 17:14:33 +0100

	Greetings
	I've made some experiments with streams generated via the live.com media
server and found some problems with lip synch. How is lip synch achieved
with the RTP/RTCP protocol? Is it just via the RTP timestamps on both
streams, or is the NTP timestamp on RTCP used as well? Could you point to
some documents describing the inter-stream (video+audio) synchronization
process? I have read the description on the RFC but am still not sure. Any
help is much appreciated.

Thanks in advance
David Pontes



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Subject: Re: [Live-devel] why different files for RTCP?
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X-Original-Date: Wed, 11 Sep 2002 09:30:26 -0700
Date: Wed, 11 Sep 2002 09:30:26 -0700


>I wanted to know as to why have you kept the RFC specific implementation 
>of RTCP in a different C (rtcp_from_spec.c and .h) files when your entire 
>application is implemented in C++?

The RTP specification (latest version: 
<http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-new-11.txt>) 
contains pseudo-code - written in C - for computing the RTCP retransmission 
interval (the length of time to delay between successive outgoing RTCP 
messages).  Because this algorithm is quite complex, I chose to use the 
pseudo-code as is, rather than rewriting it in C++ (and risk making a 
mistake in doing so).

         Ross.



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Subject: Re: [Live-devel] Lip synch
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X-Original-Date: Wed, 11 Sep 2002 09:42:13 -0700
Date: Wed, 11 Sep 2002 09:42:13 -0700

At 09:14 AM 9/11/02, you wrote:
>         Greetings
>         I've made some experiments with streams generated via the 
> live.com media
>server and found some problems with lip synch. How is lip synch achieved
>with the RTP/RTCP protocol? Is it just via the RTP timestamps on both
>streams, or is the NTP timestamp on RTCP used as well?

Both of these.  The sequences of RTP timestamps in each stream are 
independent of each other, and of 'wall clock' time (meaning that - if 
you're given a RTP timestamp - you can't, without extra information, figure 
out what 'wall clock' time it corresponds to).

This "extra information" comes from periodic RTCP Sender Report ("SR") 
packets that get sent by each media source.  These packets contain two 
timestamps: (i) a RTP timestamp, and (ii) a 'wall clock' time (in NTP 
format) that corresponds to this RTP timestamp.

Once receivers know the 'wall clock' time that corresponds to each RTP 
timestamp, they can use this to synchronize separate streams (e.g., audio 
and video).

(The bad news: The LIVE.COM code currently does not use the timestamps in 
RTCP Sender Reports to properly set the "fPresentationTime" field (the 
'wall clock' time) in each "RTPSource" object.  This is another thing 
that's on the "to do" list.)

>  Could you point to
>some documents describing the inter-stream (video+audio) synchronization
>process?

Yes, the latest version of the RTP/RTCP specification is online as 
<http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-new-11.txt>

See, in particular, the mentions of "NTP timestamp" in sections 4, 6, and, 
in particular, 6.4.1.

         Ross.



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X-Original-Date: Thu, 12 Sep 2002 10:05:35 +0530
Date: Thu, 12 Sep 2002 10:05:35 +0530

Hi  Ross,

> Yes, good point.  I've gone ahead now and modified "ByteStreamFileSource",
> and installed a new version of the library code (2002.09.11).  If a
> ByteStreamFileSource is created using both the "preferredFrameSize" and
> "playTimePerFrame" parameters, then the 'presentation time' (used for
> computing the RTP timestamp) will now be computed based on these
> parameters, rather than on 'wall clock' time.
>
> Note that, to use this feature, you *must* give the "preferredFrameSize"
> and "playTimePerFrame" parameters to
> "ByteStreamFileSource::createNew()".  For example, to stream a file that
> contains GSM data:
>          ByteStreamFileSource::createNew(*env, inputFileName, 33, 0.02);
> (because each GSM frames is 33 bytes, and represents a duration of 20 ms).
>

Thanks a lot for this quick reply. I tried the new library. The problem I am
facing now is that the rounding off of usec value that happens in the
RTPSink::convertToRTPTimestamp function leads to didfferent values of the
timestamp, depending on the initial value of usec. The difference between
consecutive packets is not constant. At times it is one less than previous
one.
The other problem is that I need to limit the packet size to 512. Is there
an API which would allow me to do this. Right now I working after modifying
the code in MultiFramedRTPSink.cpp:

static unsigned const maxPacketSize = 512;
 // bytes (1500, minus some allowance for IP, UDP, UMTP headers)
static unsigned const preferredPacketSize = 512; // bytes

Any suggestions?

> Out of curiosity, which codec(s) do you plan to use?
Right now we are working on u-law, a-law, gsm and mp3. Shall consider others
in future.

Thanks and regards,
Anant



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X-Original-Date: Thu, 12 Sep 2002 02:34:58 -0700
Date: Thu, 12 Sep 2002 02:34:58 -0700


>Thanks a lot for this quick reply. I tried the new library. The problem I am
>facing now is that the rounding off of usec value that happens in the
>RTPSink::convertToRTPTimestamp function leads to didfferent values of the
>timestamp, depending on the initial value of usec. The difference between
>consecutive packets is not constant. At times it is one less than previous
>one.

What RTP timestamp frequency are you using, and what is the 
"playTimePerFrame" parameter that you're passing to 
"ByteStreamFileSource::createNew()"?  Also, could you please give an 
example of a sequence of RTP timestamps - resulting from this - that are 
not evenly spaced.

Most audio RTP payload formats (except for MPEG audio) use a RTP timestamp 
frequency of 8000 Hz, and the audio frame rate usually divides this evenly.

>The other problem is that I need to limit the packet size to 512. Is there
>an API which would allow me to do this. Right now I working after modifying
>the code in MultiFramedRTPSink.cpp:
>
>static unsigned const maxPacketSize = 512;
>  // bytes (1500, minus some allowance for IP, UDP, UMTP headers)
>static unsigned const preferredPacketSize = 512; // bytes

I believe that will work.

         Ross.



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X-Original-Date: Thu, 12 Sep 2002 19:36:56 +0530
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Hi Ross,
As I had mentioned in my last mail, the rtp timestamp isn't increasing =
linearly in the new library, which containes your fix. I found out that =
there were two reasons. The fPlayTimePerFrame field being a float =
variable would not store 0.02 as 0.02 but as 0.01999999999.... this is =
one source of error. The other source is the initial usec value of the =
fPresentationTime field .

So I have modified the 'ByteStreamFileSource::doGetNextFrame()' function =
thus:

void ByteStreamFileSource::doGetNextFrame() {
  if (feof(fFid) || ferror(fFid)) {
    handleClosure(this);
    return;
  }

  // Try to read as many bytes as will fit in the buffer provided
  // (or "fPreferredFrameSize" if less)
  if (fPreferredFrameSize > 0 && fPreferredFrameSize < fMaxSize) {
    fMaxSize =3D fPreferredFrameSize;
  }
  fFrameSize =3D fread(fTo, 1, fMaxSize, fFid);

  if (fPlayTimePerFrame > 0) {
    unsigned uSeconds =3D fPresentationTime.tv_usec + =
(unsigned)(fLastPlayTime);
    fPresentationTime.tv_sec +=3D uSeconds/1000000;
    fPresentationTime.tv_usec =3D uSeconds%1000000;

    // Remember the play time of this data:
    fLastPlayTime
      =3D (fPlayTimePerFrame*fFrameSize*1000)/fPreferredFrameSize;
  } else {
    // We don't know a specific play time duration for this data,
    // so just record the current time as being the 'presentation time':
    gettimeofday(&fPresentationTime, &Idunno);
  }

  // Switch to another task, and inform the reader that he has data:
  nextTask() =3D envir().taskScheduler().scheduleDelayedTask(0,
    (TaskFunc*)FramedSource::afterGetting, this);
}

Apart from this I have modified the type of fPlayTimePerFrame variable =
to unsigned. Earlier it was float. And all functions which reference it. =
I store it as milli seconds instead of seconds.
 After these changes I am able to work with linphone.

Thanks,
Anant.


------=_NextPart_000_00D7_01C25A93.C11EAAA0
Content-Type: text/html;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 5.50.4134.600" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Hi Ross,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>As I had mentioned in my last mail, the =
rtp=20
timestamp&nbsp;isn't increasing linearly in the new library, which =
containes=20
your fix. I found out that there were two reasons. The fPlayTimePerFrame =
field=20
being a float variable would not store 0.02 as 0.02 but as =
0.01999999999....=20
this is one source of error. The other source is the initial usec value =
of the=20
fPresentationTime field .</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>So I have modified the=20
'ByteStreamFileSource::doGetNextFrame()' function thus:</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>void =
ByteStreamFileSource::doGetNextFrame()=20
{<BR>&nbsp; if (feof(fFid) || ferror(fFid)) {<BR>&nbsp;&nbsp;&nbsp;=20
handleClosure(this);<BR>&nbsp;&nbsp;&nbsp; return;<BR>&nbsp; =
}</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp; // Try to read as many bytes as =
will fit in=20
the buffer provided<BR>&nbsp; // (or "fPreferredFrameSize" if =
less)<BR>&nbsp; if=20
(fPreferredFrameSize &gt; 0 &amp;&amp; fPreferredFrameSize &lt; =
fMaxSize)=20
{<BR>&nbsp;&nbsp;&nbsp; fMaxSize =3D fPreferredFrameSize;<BR>&nbsp; =
}<BR>&nbsp;=20
fFrameSize =3D fread(fTo, 1, fMaxSize, fFid);</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp; if (fPlayTimePerFrame &gt; 0)=20
{<BR>&nbsp;&nbsp;&nbsp; unsigned uSeconds&nbsp;=3D =
fPresentationTime.tv_usec +=20
(unsigned)(fLastPlayTime);<BR>&nbsp;&nbsp;&nbsp; =
fPresentationTime.tv_sec +=3D=20
uSeconds/1000000;<BR>&nbsp;&nbsp;&nbsp; fPresentationTime.tv_usec =3D=20
uSeconds%1000000;<BR></FONT><FONT face=3DArial size=3D2></FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp;&nbsp; // Remember the play =
time of=20
this data:<BR>&nbsp;&nbsp;&nbsp; =
fLastPlayTime<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
=3D (fPlayTimePerFrame*fFrameSize*1000)/fPreferredFrameSize;<BR>&nbsp; } =
else=20
{<BR>&nbsp;&nbsp;&nbsp; // We don't know a specific play time duration =
for this=20
data,<BR>&nbsp;&nbsp;&nbsp; // so just record the current time as being =
the=20
'presentation time':<BR>&nbsp;&nbsp;&nbsp; =
gettimeofday(&amp;fPresentationTime,=20
&amp;Idunno);<BR>&nbsp; }</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp; // Switch to another task, and =
inform the=20
reader that he has data:<BR>&nbsp; nextTask() =3D=20
envir().taskScheduler().scheduleDelayedTask(0,<BR>&nbsp;&nbsp;&nbsp;&nbsp=
;(TaskFunc*)FramedSource::afterGetting,=20
this);<BR>}</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Apart from this I have modified the =
type of=20
fPlayTimePerFrame variable to unsigned. Earlier it was float. And all =
functions=20
which reference it. I store it as milli seconds instead of =
seconds.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;After these changes I am able to =
work with=20
linphone.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Thanks,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Anant.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV></BODY></HTML>

------=_NextPart_000_00D7_01C25A93.C11EAAA0--



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X-Original-Date: Thu, 12 Sep 2002 10:24:08 -0700
Date: Thu, 12 Sep 2002 10:24:08 -0700

Anant,

Thanks for the report.  I'll go ahead and make this change to an upcoming 
release.  (However, I'll probably make the "playTimePerFrame" 
parameter  microseconds rather than milliseconds, so as to match the 
resolution of "struct timeval".)

	Ross.



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X-Original-Date: Fri, 13 Sep 2002 17:08:07 -0700
Date: Fri, 13 Sep 2002 17:08:07 -0700

I have just released a new version (2002.09.13) of the source code.  There 
are two significant changes:

1/ The "playTimePerFrame" parameter to "ByteStreamFileSource::createNew()" 
is now an 'unsigned' rather than a 'float'.  Its value is the time in 
microseconds.  This is to overcome the rounding problem that Anant encountered.

2/ "BasicHashTable" now has a completely new implementation, replacing the 
quick-and-dirty hacked implementation that was there before.  The new 
implementation is more efficient and scalable.

	Ross.



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X-Original-Date: Mon, 16 Sep 2002 11:42:21 +0500
Date: Mon, 16 Sep 2002 11:42:21 +0500






I wish to ask a few important questions!

Lets suppose I have to stream a H263plus enncoded data over the net.

Now,except for the first enocded frame (I frame) or when the I frame is
intentionally inserted, the size of the average frame remains around 150
bytes ( for QCIF format). Now if the RTP packet has an average size of 1000
bytes, the major portion of the RTP packet remains unused leading to greater
overhead.
What if I pack multiple encoded frames with in one RTP packet. If this is
the case, what about  the time stamp .
Also, Can we stream voice and video in the same RTP packet or what is the
other alternative. In the former case , how do you decide the timestamp.

Secondly, if you permit so,  I have some questions with respect to
RTPpayload format for H263plus rfc 2429. 

1) Who tells me that the frame is fragmented ( i.e whether I have to set P
bit or not )

2) When should I insert the extra picture header.


How does RFC 2429 help error resilience and improve efficiency. ( you add
header on one side, and strip it off on the other side before you give it to
the decoder , then whats the use of it )

regards,
umar


-----Original Message-----
From: <EMAIL-ADDRESS-HIDDEN>
[mailto:live-devel-admin(at)lists.sourceforge.net]On Behalf Of Ross
Finlayson
Sent: Saturday, September 14, 2002 5:08 AM
To: <EMAIL-ADDRESS-HIDDEN>
Subject: [Live-devel] New source code release


I have just released a new version (2002.09.13) of the source code.  There 
are two significant changes:

1/ The "playTimePerFrame" parameter to "ByteStreamFileSource::createNew()" 
is now an 'unsigned' rather than a 'float'.  Its value is the time in 
microseconds.  This is to overcome the rounding problem that Anant
encountered.

2/ "BasicHashTable" now has a completely new implementation, replacing the 
quick-and-dirty hacked implementation that was there before.  The new 
implementation is more efficient and scalable.

	Ross.



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X-Original-Date: Mon, 16 Sep 2002 04:08:12 -0700
Date: Mon, 16 Sep 2002 04:08:12 -0700


>Lets suppose I have to stream a H263plus enncoded data over the net.
>
>Now,except for the first enocded frame (I frame) or when the I frame is
>intentionally inserted, the size of the average frame remains around 150
>bytes ( for QCIF format). Now if the RTP packet has an average size of 1000
>bytes, the major portion of the RTP packet remains unused leading to greater
>overhead.

No, if you are sending a 150-byte H.263+ frame in a RTP packet, then the 
outgoing RTP packet will be only as large as it needs to be to fit this 
packet.  I.e., it will be 150 bytes, plus a few more bytes for the H.263+ 
payload header (RFC 2429, section 4.1), RTP header, UDP header, and IP header.

>What if I pack multiple encoded frames with in one RTP packet.

No, you can't do this (at least, according to my interpretation of RFC 2429),

>Also, Can we stream voice and video in the same RTP packet

No, audio and video make up separate RTP streams (with separate RTP payload 
format codes, timestamps, etc.), and so are sent in separate RTP packets.

>Secondly, if you permit so,  I have some questions with respect to
>RTPpayload format for H263plus rfc 2429.
>
>1) Who tells me that the frame is fragmented ( i.e whether I have to set P
>bit or not )

The "H263plusVideoRTPSink" class - included in the "liveMedia" library - 
does this for you.  I.e., if you feed a "H263plusVideoRTPSink" object 
complete H.263+ frames, then the implementation (in particular, 
"H263plusVideoRTPSink::doSpecialFrameHandling()") will take care of setting 
up the "H.263+ payload header" for you.  (And, the parent class - 
"MultiFramedRTPSink" - will take care of fragmentation into multiple RTP 
packets, if necessary.)

>2) When should I insert the extra picture header.

"H263plusVideoRTPSink" currently doesn't implement this (i.e., it currently 
always sets the PLEN field to 0).  You would need to modify this 
implementation to add an extra picture header.

>How does RFC 2429 help error resilience and improve efficiency. ( you add
>header on one side, and strip it off on the other side before you give it to
>the decoder , then whats the use of it )

I think that the idea is to add redundancy to the data, by including extra 
picture headers.  If a packet (containing a picture header) gets lost, then 
fewer of the following packets will be useless to the decoder.

Also, I think the VRC extension mechanism (section 4.2) requires that the 
input data already be in a special form to begin with.

But I'm not an expert in H.263+...

         Ross.



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Subject: [Live-devel] RTP over radio ?
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X-Original-Date: Mon, 16 Sep 2002 13:14:03 +0000
Date: Mon, 16 Sep 2002 13:14:03 +0000

Hello everyone,

I am implementing a RTP based audio/video server for streaming media to 
handheld devices like the iPaq over radio. I want to know if RTP+UDP (I 
cannot use TCP, due to backoff problems that ensue with dropped packets) 
would be a good vehicle for this kind of application. Packets can get dropped 
during wireless broadcasts, could some kind of of carousel scheme be to added 
to the server to transmit every packet with a certain multiple so that the 
end points get the packets in a more reliably. ? Can RTP help with this ? 

Thanks in advance for your help. 

- asit
   


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X-Original-Date: Tue, 17 Sep 2002 01:27:38 -0700
Date: Tue, 17 Sep 2002 01:27:38 -0700


>I am implementing a RTP based audio/video server for streaming media to
>handheld devices like the iPaq over radio. I want to know if RTP+UDP (I
>cannot use TCP, due to backoff problems that ensue with dropped packets)
>would be a good vehicle for this kind of application.

Absolutely.  (Another advantage of using UDP rather than TCP for streaming 
media is that with UDP you can use multicast - which is a good fit for 
wireless.)

>Packets can get dropped
>during wireless broadcasts, could some kind of of carousel scheme be to added
>to the server to transmit every packet with a certain multiple so that the
>end points get the packets in a more reliably. ? Can RTP help with this ?

Unfortunately, the very features of RTP/UDP (low delay, support for 
multicast) that make it suitable for streaming media also make it difficult 
to recover from packet loss.  But there are some things that you can do:

1/ First, one of the best approaches is to use a codec and/or RTP payload 
format that is resilient to packets loss, so that *if* packet loss occurs, 
the receiver can deal with it gracefully.  For example, the RTP payload 
format for H.263+ video has some features to make packet loss more 
resilient, and RFC 3119 describes an alternative RTP payload format for MP3 
audio that tolerates packet loss better than RFC 2250.  (This payload 
format is supported by the LIVE.COM code, BTW.)

2/ You can add forward error correction (FEC) to RTP.  This increases 
bandwidth (by adding extra error-correcting code packets), but if done well 
allows receivers to recover from much packet loss.  (Also, this is much 
better than simply repeating packets.)  One possible approach to using FEC 
in RTP is described in RFC 2733.

3/ Finally, for unicast streams with a back-channel, you can also recover 
from some packet loss by having the receiver ask the sender to retransmit 
lost packets - provided that this can be done without causing excessive 
delay.  A standard RTP mechanism for retransmission is currently going 
through the IETF standards process.

I hope this helps...

         Ross.





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X-Original-Date: Tue, 17 Sep 2002 18:54:41 +0530
Date: Tue, 17 Sep 2002 18:54:41 +0530

Thanks Ross for the latest update.
Anant.

----- Original Message -----
From: <EMAIL-ADDRESS-HIDDEN>
To: <EMAIL-ADDRESS-HIDDEN>
Sent: Tuesday, September 17, 2002 12:32 AM
Subject: Live-devel digest, Vol 1 #243 - 2 msgs


> Send Live-devel mailing list submissions to
> live-devel(at)lists.sourceforge.net
>
> To subscribe or unsubscribe via the World Wide Web, visit
> https://lists.sourceforge.net/lists/listinfo/live-devel
> or, via email, send a message with subject or body 'help' to
> live-devel-request(at)lists.sourceforge.net
>
> You can reach the person managing the list at
> live-devel-admin(at)lists.sourceforge.net
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of Live-devel digest..."
>
>
> Today's Topics:
>
>    1. FW: some questions (umar janjua)
>    2. Re: some questions (Ross Finlayson)
>
> --__--__--
>
> Message: 1
> From: umar janjua <umarj(at)enabtech.com>
> To: live-devel(at)lists.sourceforge.net
> Date: Mon, 16 Sep 2002 11:42:21 +0500
> Subject: [Live-devel] FW: some questions
>
>
>
>
>
>
> I wish to ask a few important questions!
>
> Lets suppose I have to stream a H263plus enncoded data over the net.
>
> Now,except for the first enocded frame (I frame) or when the I frame is
> intentionally inserted, the size of the average frame remains around 150
> bytes ( for QCIF format). Now if the RTP packet has an average size of
1000
> bytes, the major portion of the RTP packet remains unused leading to
greater
> overhead.
> What if I pack multiple encoded frames with in one RTP packet. If this is
> the case, what about  the time stamp .
> Also, Can we stream voice and video in the same RTP packet or what is the
> other alternative. In the former case , how do you decide the timestamp.
>
> Secondly, if you permit so,  I have some questions with respect to
> RTPpayload format for H263plus rfc 2429.
>
> 1) Who tells me that the frame is fragmented ( i.e whether I have to set P
> bit or not )
>
> 2) When should I insert the extra picture header.
>
>
> How does RFC 2429 help error resilience and improve efficiency. ( you add
> header on one side, and strip it off on the other side before you give it
to
> the decoder , then whats the use of it )
>
> regards,
> umar
>
>
> -----Original Message-----
> From: live-devel-admin(at)lists.sourceforge.net
> [mailto:live-devel-admin(at)lists.sourceforge.net]On Behalf Of Ross
> Finlayson
> Sent: Saturday, September 14, 2002 5:08 AM
> To: live-devel(at)lists.sourceforge.net
> Subject: [Live-devel] New source code release
>
>
> I have just released a new version (2002.09.13) of the source code.  There
> are two significant changes:
>
> 1/ The "playTimePerFrame" parameter to "ByteStreamFileSource::createNew()"
> is now an 'unsigned' rather than a 'float'.  Its value is the time in
> microseconds.  This is to overcome the rounding problem that Anant
> encountered.
>
> 2/ "BasicHashTable" now has a completely new implementation, replacing the
> quick-and-dirty hacked implementation that was there before.  The new
> implementation is more efficient and scalable.
>
> Ross.
>
>
>
> -------------------------------------------------------
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>
>
> --__--__--
>
> Message: 2
> Date: Mon, 16 Sep 2002 04:08:12 -0700
> To: live-devel(at)lists.sourceforge.net
> From: Ross Finlayson <finlayson(at)live.com>
> Subject: [Live-devel] Re: some questions
>
>
> >Lets suppose I have to stream a H263plus enncoded data over the net.
> >
> >Now,except for the first enocded frame (I frame) or when the I frame is
> >intentionally inserted, the size of the average frame remains around 150
> >bytes ( for QCIF format). Now if the RTP packet has an average size of
1000
> >bytes, the major portion of the RTP packet remains unused leading to
greater
> >overhead.
>
> No, if you are sending a 150-byte H.263+ frame in a RTP packet, then the
> outgoing RTP packet will be only as large as it needs to be to fit this
> packet.  I.e., it will be 150 bytes, plus a few more bytes for the H.263+
> payload header (RFC 2429, section 4.1), RTP header, UDP header, and IP
header.
>
> >What if I pack multiple encoded frames with in one RTP packet.
>
> No, you can't do this (at least, according to my interpretation of RFC
2429),
>
> >Also, Can we stream voice and video in the same RTP packet
>
> No, audio and video make up separate RTP streams (with separate RTP
payload
> format codes, timestamps, etc.), and so are sent in separate RTP packets.
>
> >Secondly, if you permit so,  I have some questions with respect to
> >RTPpayload format for H263plus rfc 2429.
> >
> >1) Who tells me that the frame is fragmented ( i.e whether I have to set
P
> >bit or not )
>
> The "H263plusVideoRTPSink" class - included in the "liveMedia" library -
> does this for you.  I.e., if you feed a "H263plusVideoRTPSink" object
> complete H.263+ frames, then the implementation (in particular,
> "H263plusVideoRTPSink::doSpecialFrameHandling()") will take care of
setting
> up the "H.263+ payload header" for you.  (And, the parent class -
> "MultiFramedRTPSink" - will take care of fragmentation into multiple RTP
> packets, if necessary.)
>
> >2) When should I insert the extra picture header.
>
> "H263plusVideoRTPSink" currently doesn't implement this (i.e., it
currently
> always sets the PLEN field to 0).  You would need to modify this
> implementation to add an extra picture header.
>
> >How does RFC 2429 help error resilience and improve efficiency. ( you add
> >header on one side, and strip it off on the other side before you give it
to
> >the decoder , then whats the use of it )
>
> I think that the idea is to add redundancy to the data, by including extra
> picture headers.  If a packet (containing a picture header) gets lost,
then
> fewer of the following packets will be useless to the decoder.
>
> Also, I think the VRC extension mechanism (section 4.2) requires that the
> input data already be in a special form to begin with.
>
> But I'm not an expert in H.263+...
>
>          Ross.
>
>
>
>
> --__--__--
>
> _______________________________________________
> Live-devel mailing list
> Live-devel(at)lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/live-devel
>
>
> End of Live-devel Digest
>



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Subject: [Live-devel] Any good books on RTP/UDP programming and concepts?
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X-Original-Date: Tue, 17 Sep 2002 22:09:57 -0400
Date: Tue, 17 Sep 2002 22:09:57 -0400

Hi there,

Just like if you have any recommendations on good books for RTP/UDP concepts and programming?

Thanks a lot!
Regards,
Joseph


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X-Original-Date: Wed, 18 Sep 2002 18:32:41 +0500
Date: Wed, 18 Sep 2002 18:32:41 +0500

If I am to send both Video and Audio through RTP, I have following probable
choices.

1) Create separte channels with in RTP for video and audio, and make the
same time stamp.
  (suggestion)
2) How to synchronise the audio and video if I dont have the same time
stamp.

			


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Subject: Re: [Live-devel] Any good books on RTP/UDP programming and concepts?
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X-Original-Date: Wed, 18 Sep 2002 08:06:59 -0700
Date: Wed, 18 Sep 2002 08:06:59 -0700

On Tue, Sep 17, 2002 at 10:09:57PM -0400, Joseph Teo wrote:
> Hi there,
> 
> Just like if you have any recommendations on good books for RTP/UDP
> concepts and programming?

i haven't found anything that deals with rtp specifically that was
worth reading, but for udp, there's one book that stands out above the
rest: the late w. richard stevens' "unix network programming vol. 1."

-john

-- 
"nothing brings people together like doom."
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  concepts?
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X-Original-Date: Wed, 18 Sep 2002 11:48:02 -0700
Date: Wed, 18 Sep 2002 11:48:02 -0700

At 07:09 PM 9/17/02, you wrote:
>Hi there,
>
>Just like if you have any recommendations on good books for RTP/UDP 
>concepts and programming?

I don't know of any RTP books yet (so far, RTP has stayed below the radar 
of people who like to exploit hot new topics by writing books about them :-)

Apart from the RTP/RTCP spec itself - 
<http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-new-11.txt> - you 
can also find useful information about RTP at 
<http://www.cs.columbia.edu/~hgs/rtp/>

         Ross.



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X-Original-Date: Wed, 18 Sep 2002 11:58:29 -0700
Date: Wed, 18 Sep 2002 11:58:29 -0700

At 06:32 AM 9/18/02, you wrote:
>If I am to send both Video and Audio through RTP, I have following probable
>choices.
>
>1) Create separte channels with in RTP for video and audio, and make the
>same time stamp.
>   (suggestion)

No, "making the same time stamp" does not work - and, in fact, doesn't 
really make sense, because RTP timestamps for separate RTP streams (e.g., 
audio & video) are independent, and (usually) have different frequencies.

>2) How to synchronise the audio and video if I dont have the same time
>stamp.

Synchronization of separate RTP streams is done using the contents of RTCP 
"Sender Report" messages - which map each stream's RTP timestamps to 'real 
time' (expressed as a NTP timestamp).

Please wait a day or so until I've installed a new version of the LIVE.COM 
library code.  This new version will correctly set the "presentationTime" 
parameter when a "RTPSource" object is being read.  You can use this to do 
time synchronization.

         Ross.



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X-Original-Date: Thu, 19 Sep 2002 02:01:39 -0700
Date: Thu, 19 Sep 2002 02:01:39 -0700

I have installed a new version (2002.09.19) of the source code.  This new 
version improves the implementation of "RTPSource" objects: The 
"presentationTime" parameter (to the 'afterGettingFunc' finish-up function) 
will now be set correctly whenever a "RTPSource" object is read.  Clients 
that receive multiple RTP streams (e.g., audio & video) that make up a 
single session can now use this information to do media synchronization.

To get best results, each "RTPSource" should also have an associated 
"RTCPInstance", so that the implementation can use RTCP  "Sender Report" 
messages to properly map each stream's RTP timestamps to 'real time'.

	Ross.



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Subject: [Live-devel] quicktime over RTP: what am I seeing?
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X-Original-Date: Thu, 19 Sep 2002 16:46:28 -0700
Date: Thu, 19 Sep 2002 16:46:28 -0700

Hello,

First of all this RTP library looks really awesome. Looks like very good
work!

We have a Quicktime component that encodes video with our own codec.
Because our target client platform is a PDA, we have to build our own
client. What I need to do is get the streamed data from Darwin and strip
off all the QT so that our decoder can do its thing.

I've looked through the code and did some compiles and some tests. I
tried the following:

openRTSP -q rtsp://192.168.1.100/tv.mov >test1.mov

tv.mov is a quicktime with our own video codec. test1.mov opens in
quicktime, but doesn't play properly. It plays the whole thing in less
than a second (5 second clip). I was encouraged to see actual video,
though. ;-)

Now, if we omit the -q option, as I understand it what is being stored
into the file "video-X-QT-1" is the video track (track meaning quicktime
track) of the movie. Is this correct?

Assuming it is, how would I extract our coded video data for passing to
our decoder? I browsed through the quicktime file format PDF, a 300-page
monstrosity, and the format looks pretty complicated. Any preexisting
methods?

Secondly, I saw a post in the archives from Ross Finlayson dated 8/18
indicating that the library will compile on CE. Are there any pitfalls I
should be aware of before I attempt this next week?

Thanks, I look forward to any advice.

Gabe



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Hi all,

As I have talk about on my last mail, I want to stream video throught one protocol named Staggered Broadcasting. the idea is to cut movies to many equal segments and the streaming them on different RTSP session (different channels). after playing the first segment from the first channel, the clien switch to the second one to get the second segment. This process continu to the end of the movie. 

I have implemented the server part wich stream different segment through diffrent RTSP/RTP session ( based on Live.com). Now I want to complete the client part, for now I' am able to get data from diffrent channels (on client part) ond saving them on different files (this is based on openRTSP). if I suppose that every segment length  is 12 minutes, how can I schedul the "startplaying function" of the second session as it will be started after 12 min after the first one??

thanks in advance.

Achraf .      



---------------------------------
Yahoo! Mail -- Une adresse (at)yahoo.fr gratuite et en français !

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<P>Hi all,</P>
<P>As I have talk about on my last mail, I want&nbsp;to&nbsp;stream video throught one protocol named Staggered Broadcasting.&nbsp;the idea is to cut movies to many&nbsp;equal segments and the streaming them on different RTSP session (different channels).&nbsp;after playing the first segment from the first channel, the clien switch to the second one to get the second segment. This process continu to the end of the movie. </P>
<P>I have implemented the server part&nbsp;wich stream different segment through diffrent RTSP/RTP session ( based on Live.com). Now I want to complete the client part, for now&nbsp;I' am able to get data from diffrent channels (on client part) ond saving them on different files (this is based on openRTSP). if&nbsp;I suppose that every segment length&nbsp;&nbsp;is 12 minutes,&nbsp;how&nbsp;can I&nbsp;schedul the "startplaying function"&nbsp;of the second session as it will be started after 12 min after the first one??</P>
<P>thanks in advance.</P>
<P>Achraf&nbsp;.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </P><p><br><hr size=1><a href="http://fr.mail.yahoo.com">Yahoo! Mail</a> -- Une adresse (at)yahoo.fr gratuite et en français !<br>
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X-Original-Date: Fri, 20 Sep 2002 10:52:40 -0700
Date: Fri, 20 Sep 2002 10:52:40 -0700


>First of all this RTP library looks really awesome. Looks like very good
>work!

Thanks!

>We have a Quicktime component that encodes video with our own codec.
>Because our target client platform is a PDA, we have to build our own
>client. What I need to do is get the streamed data from Darwin and strip
>off all the QT so that our decoder can do its thing.

OK, it looks like you should be able to do this, by writing your own 
RTSP/RTP client application, using the existing "openRTSP" code as a 
model.  In the meantime, you should be able to use the existing "openRTSP" 
application for testing, as you are doing.

If you want to feed incoming video data directly to your decoder, then you 
probably do *not* want to use the "-q" option to "openRTSP", as that 
constructs a new QuickTime movie file from the incoming RTP data.  (I'll 
say more about this later.)  Omit the "-q", and instead use the "-v" option 
(meaning: output video to stdout), and just pipe the video to your decoder, 
e.g.
         openRTSP -v rtsp://... | your-decoder-application

>Now, if we omit the -q option, as I understand it what is being stored
>into the file "video-X-QT-1" is the video track (track meaning quicktime
>track) of the movie. Is this correct?

Sort of, but it's important to note that "track" is a QuickTime 
movie-specific notion.  What gets sent across the network are RTP packets, 
using a particular "RTP payload format".  It just so happens that, in your 
case, the RTP payload format that's being used is Apple's "QuickTime 
Generic RTP Payload Format", as described in 
<http://developer.apple.com/quicktime/icefloe/dispatch026.html>. This is 
the RTP payload format that Apple uses to stream data from a QuickTime 
file, if there's not already a RTP payload format defined for that codec.

I note that you're streaming from a "Darwin Streaming Server", which 
requires that ".mov" files be 'hinted' before they can be 
streamed.  "Hinting", in this case, means running through the track(s) of 
your source ".mov" file, and using a particular RTP payload format 
definition to construct a sequence of RTP packet headers to be used when 
streaming.  So presumably, before you streamed "tv.mov", you ran some sort 
of 'hinting' tool on it, at which time you specified - at least implicitly 
- which RTP payload format should be used for streaming.  (If you didn't 
specify this explicitly, then your hinting tool probably chose the 
"QuickTime Generic RTP Payload Format" instead.)

Ultimately, because you have your own codec, you should probably define 
your own RTP payload format for it, write this up as an Internet Draft, and 
submit it to the IETF's "AVT" (audio/video transport) working group.  In 
the meantime, you may be able to continue using the "QuickTime Generic RTP 
Payload Format", because the LIVE.COM code includes a "RTPSource" subclass 
("QuickTimeGenericRTPSource") that implements this.


>I've looked through the code and did some compiles and some tests. I
>tried the following:
>
>openRTSP -q rtsp://192.168.1.100/tv.mov >test1.mov
>
>tv.mov is a quicktime with our own video codec. test1.mov opens in
>quicktime, but doesn't play properly. It plays the whole thing in less
>than a second (5 second clip). I was encouraged to see actual video,
>though. ;-)

The "-q" option to "openRTSP" is basically a hack, because it turns out 
that the ".mov" format is not well-suited for storing captured incoming RTP 
streams.  To be able to play back such a ".mov" file properly, you need to 
give "openRTSP" some extra, out-of-band information for it to store in the 
file.  In particular, you need to tell "openRTSP" the frame rate (in 
frames-per-second) of the video stream, and also the video's width and 
height (in pixels).  I.e., you should add the "-w", "-h", and "-f" options: 
See <http://www.live.com/openRTSP/#quicktime>.  If you do this, you may end 
up with a ".mov" file that plays at the correct rate.

However, even if you do succeed in producing a playable ".mov" file using 
"openRTSP -q", then you won't be able to use this to feed video data to 
your decoder in real time. The reason for this is that the QuickTime movie 
track 'meta data' (a 'moov atom, in QuickTime parlance) is not written to 
the file until the very end.


>Secondly, I saw a post in the archives from Ross Finlayson dated 8/18
>indicating that the library will compile on CE. Are there any pitfalls I
>should be aware of before I attempt this next week?

I haven't tried this myself, but it should be possible, by following the 
instructions given in <http://www.live.com/liveMedia/#config-windows>.  You 
may also need to make modifications to the "win32config" file, and/or the 
resulting ".mak" makefiles; I'm not sure...

I hope this helps.

         Ross.



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> (meaning: output video to stdout), and just pipe the video to your 
> decoder, 
> e.g.
>         openRTSP -v rtsp://... | your-decoder-application

OK, so I did some investigation and this is almost what I need. The 
data that gets piped into the decoder with this command is frame data 
in the format our decoder expects, with one big exception: it is NOT 
prefaced by a sequence header.

Our decoder needs the sequence header to set up basic video parameters. 
I assume this is related to needing to specify w/h/framerate info when 
using -q.

Because streaming from Darwin to Apple's quicktime player with our 
codec works fine, the sequence header has to somehow move from the 
server to the client.

The $64k question is where/when/how is this information transferred, 
and how can I grab it to send to the decoder?

> >Secondly, I saw a post in the archives from Ross Finlayson dated 8/18
> >indicating that the library will compile on CE. Are there any 
> pitfalls I
> >should be aware of before I attempt this next week?
> 
> I haven't tried this myself, but it should be possible, by 
> following the 
> instructions given in <." 
> target="l">http://www.live.com/liveMedia/#config-windows>.  You 
> may also need to make modifications to the "win32config" file, 
> and/or the 
> resulting ".mak" makefiles; I'm not sure...

MS Embedded VC++ doesn't seem to be able to open .mak workspaces, and 
I'm skeptical that it will open a DSW generated by regular VC++. I'll 
look into this; at the limit I guess we could compile this manually at 
a command prompt.

Thanks Ross,

Gabe



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X-Original-Date: Fri, 20 Sep 2002 13:16:24 -0700
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>I have implemented the server part wich stream different segment through 
>diffrent RTSP/RTP session ( based on Live.com). Now I want to complete the 
>client part, for now I' am able to get data from diffrent channels (on 
>client part) ond saving them on different files (this is based on 
>openRTSP). if I suppose that every segment length  is 12 minutes, how can 
>I schedul the "startplaying function" of the second session as it will be 
>started after 12 min after the first one??

Achraf,

You can do this using "TaskScheduler::scheduleDelayedTask()".  E.g., define 
a function - to be called after 12 minutes - of the form
         void delayedTask(void* clientData)

and call
         your_scheduler->scheduleDelayedTask(12*60*1000000,
                                                         (TaskFunc*)delayedTask,
                                                         some_data);

Then, 12 minutes later, "delayedTask(some_data)" will get called.

You may find it useful to look at the various places in the code where 
"scheduleDelayedTask()" is called.  (There are a couple of good examples in 
"openRTSP.cpp".)

         Ross.



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X-Original-Date: Fri, 20 Sep 2002 13:31:50 -0700
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At 01:14 PM 9/20/02, you wrote:
> > (meaning: output video to stdout), and just pipe the video to your
> > decoder,
> > e.g.
> >         openRTSP -v rtsp://... | your-decoder-application
>
>OK, so I did some investigation and this is almost what I need. The
>data that gets piped into the decoder with this command is frame data
>in the format our decoder expects, with one big exception: it is NOT
>prefaced by a sequence header.

Gabe,

What is a "sequence header"?  Is it one of the fields described in 
<http://developer.apple.com/quicktime/icefloe/dispatch026.html>?  Or is it 
something else?  If so, where is this defined?

         Ross.



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X-Original-Date: Sat, 21 Sep 2002 11:22:48 -0700
Date: Sat, 21 Sep 2002 11:22:48 -0700

> What is a "sequence header"?  Is it one of the fields described in
> <http://developer.apple.com/quicktime/icefloe/dispatch026.html>?  Or
is it
> something else?  If so, where is this defined?

Our codec requires a 12-byte header to initialize. It contains version,
height, width, rate, etc. In the quicktime encoding process, we call
AddImageDescriptionExtension <
http://developer.apple.com/techpubs/quicktime/qtdevdocs/APIREF/SOURCESI/
addimdescext.htm > to stick it in.

Thanks to the debug mode in QuickTimeGenericRTPSource, and confirming my
hunch, I see that our header is contained in an 'sd' TLV atom in the
first packet:

Receiving streamed data (for up to 10.0 seconds)...
PCK: 3, S: 1, Q: 1, L: 0, D: 0, payloadId: 1
        K: 1, F: 0, A: 1, Z: 1, payloadDescriptionLength: 150
        mediaType: 'vide', timescale: 600, 138 bytes of TLVs left
                TLV 'sd', length 110, leaving 24 remaining bytes
00:00:00:6e:54:52:55:56:00:00:00:00:00:00:00:01:00:01:00:01:61:70:70:6c:
00:00:04
:00:00:00:04:00:00:a0:00:78:00:48:00:00:00:48:00:00:00:00:00:00:00:01:08
:54:72:7
5:76:69:64:65:6f:00:00:00:00:00:00:00:00:00:00:00:00:00:00:00:00:00:00:0
0:00:00:
00:00:00:18:ff:ff:00:00:00:14:54:52:55:56:01:04:10:01:0a:01:e0:28:00:00:
00:21:00
:00:00:00:

Of the last 16 bytes, the first 12 are our header.

I think I'll be able to add some code to QuickTimeGenericRTPSource after
"case ('s'<<8|'d'):" to find our header and put it in the output buffer
before we start reading frames. I tested manually by cutting-and-pasting
the header in front of the received video-X-QT file and our decoder
reads it perfectly!

Thanks for your tips,

Gabe



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Subject: RE: [Live-devel] quicktime over RTP: what am I seeing?
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X-Original-Date: Sat, 21 Sep 2002 20:17:50 -0700
Date: Sat, 21 Sep 2002 20:17:50 -0700


>I think I'll be able to add some code to QuickTimeGenericRTPSource after
>"case ('s'<<8|'d'):" to find our header and put it in the output buffer
>before we start reading frames.

Another possibility - perhaps simpler - is to take advantage of the fact 
that the "QuickTimeGenericRTPSource" implementation also saves this header 
- in a field called "qtState".

You can add code like the following to the start of 
"FileSink::afterGettingFrame()":

         #include <QuickTimeGenericRTPSource.hh>
...
         QuickTimeGenericRTPSource* qtSource
             = (QuickTimeGenericRTPSource*)fSource;
             // Hack - relies on the source being a "QuickTimeGenericRTPSource"
         // A copy of your 'sd' atom will be in "qtSource->qtState.sdAtom"
         // and its length will be "qtSource->qtState.sdAtomSize"

         // Write the "sdAtom" to the output file:
         fwrite(qtSource->qtState.sdAtom, qtSource->qtState.sdAtomSize, 1, 
sink->fOutFid);

         // Then write the RTP data to the file, as usual...


I hope this helps.

         Ross.



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Subject: [Live-devel] use AVI(MS Video or Divx) with LiveMedia Library
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X-Original-Date: Mon, 23 Sep 2002 00:10:54 +0800
Date: Mon, 23 Sep 2002 00:10:54 +0800

Hi,Ross

I want to use liveMedia library to stream my .avi file (use MS Video(now maybe
DirectShow) format or Divx format)to client. How can I implement from your
liveMedia class?? From FramedSource,MultiFramedRTPSink???

Thanks




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  Library
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X-Original-Date: Sun, 22 Sep 2002 15:23:18 -0700
Date: Sun, 22 Sep 2002 15:23:18 -0700


>I want to use liveMedia library to stream my .avi file (use MS Video(now maybe
>DirectShow) format or Divx format)to client. How can I implement from your
>liveMedia class?? From FramedSource,MultiFramedRTPSink???

First, note that "AVI" is a file format - in particular, a container for 
one or more media tracks (audio, video).  So, AVI is not something that you 
can stream (at least, not using RTP).  Instead, you would stream one or 
more of the individual media tracks that are contained within an AVI 
file.  So, what you could do - at least in theory - is implement a new 
liveMedia class that took an AVI file as input, and output 'source' objects 
that correspond to each media track that you wish to stream.  Such an 
implementation might be similar to the existing "MPEGDemux" and 
"MPEGDemuxedElementaryStream" classes (which are used to extract one or 
more MPEG Elementary Streams from a MPEG Program Stream).

Second, to be able to stream a media track (that comes from an AVI file), 
there would need to be a "RTP Payload Format" defined for its codec.  A 
"RTP Payload Format" is a set of rules that define how the codec's media 
frames are packed within RTP packets.  It is usually defined by an IETF RFC 
(or, for newer payload formats, an IETF Internet-Draft).  For example, 
there are currently RTP Payload Formats defined for MPEG-1,2, or 4 audio 
and video, PCM, GSM and QCELP audio, and H.263+ video, and many others.

However, there is currently no RTP Payload Format defined for "MS Video", 
because Microsoft has - in the past - not used RTP for streaming.  Instead, 
they have used their own, proprietary streaming protocol.  (Apparently 
Microsoft's new "Windows Media 9" system will use RTP, in which case they 
*may* publish a RTP Payload Format for their Windows Media codecs, but I 
have yet to see such a document.)

As for Divx: Because Divx is related to MPEG-4 video, it *may* be possible 
to stream it using the RTP Payload Format for MPEG-4.  (The liveMedia 
library does not yet implement that payload format, although it's on the 
"to do" list.)

         Ross.



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Subject: Re: RE: [Live-devel] quicktime over RTP: what am I seeing?
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X-Original-Date: Mon, 23 Sep 2002 14:47:16 -0700
Date: Mon, 23 Sep 2002 14:47:16 -0700

VC++ won't compile at

QuickTimeGenericRTPSource* qtSource
            = (QuickTimeGenericRTPSource*)fSource;

reporting error C2440: 'type cast' : cannot convert from 'class 
FramedSource *MediaSink::* ' to 'class QuickTimeGenericRTPSource *'
        There is no context in which this conversion is possible

I'm still trying to understand the structure of the stack, so I'm not 
sure what the problem is.

Gabe


> Another possibility - perhaps simpler - is to take advantage of 
> the fact 
> that the "QuickTimeGenericRTPSource" implementation also saves 
> this header 
> - in a field called "qtState".
> 
> You can add code like the following to the start of 
> "FileSink::afterGettingFrame()":
> 
>         #include <QuickTimeGenericRTPSource.hh>
> ...
>         QuickTimeGenericRTPSource* qtSource
>             = (QuickTimeGenericRTPSource*)fSource;
>             // Hack - relies on the source being a 
> "QuickTimeGenericRTPSource"         // A copy of your 'sd' atom 
> will be in "qtSource->qtState.sdAtom"
>         // and its length will be "qtSource->qtState.sdAtomSize"
> 
>         // Write the "sdAtom" to the output file:
>         fwrite(qtSource->qtState.sdAtom, qtSource-
> >qtState.sdAtomSize, 1, 
> sink->fOutFid);
> 
>         // Then write the RTP data to the file, as usual...
> 
> 
> I hope this helps.
> 
>         Ross.
> 
> 



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X-Original-Date: Mon, 23 Sep 2002 15:01:47 -0700
Date: Mon, 23 Sep 2002 15:01:47 -0700

At 02:47 PM 9/23/02, gnell(at)EECS.Berkeley.EDU wrote:
>VC++ won't compile at
>
>QuickTimeGenericRTPSource* qtSource
>             = (QuickTimeGenericRTPSource*)fSource;
>
>reporting error C2440: 'type cast' : cannot convert from 'class
>FramedSource *MediaSink::* ' to 'class QuickTimeGenericRTPSource *'

Gabe,

Oops, my bad.

Change "fSource" in the above code to
         sink->fSource

and make sure that this code occurs just after the (existing) statement:
         FileSink* sink = (FileSink*)clientData;

If you still get an error (about "fSource" being out of scope), then also 
change the definition of "fSource" in "include/MediaSink.hh" from 
"protected:" to "public:".

         Ross.



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X-Original-Date: Tue, 24 Sep 2002 09:15:55 +0200
Date: Tue, 24 Sep 2002 09:15:55 +0200

Hi everybody, 
first of all I would like to gratulate the Live-Team, to the developement of
such a useful and efficient working library ! I am working on an
Live-Streaming Application and i am using VC++6.0 on windoze 2000. I am
using a "Shinet- MPEGPIA "- Encoder card and I modified the
testMPEGAudioVideoStreamer to send the MPEG-Live Stream that the card
delivers. To achieve that, I  had to pipe the data buffer that the
encoder-card delivers to the stdin of the of the
testMPEGAudioVideoStreamer.That worked well, (unicast and multicast). Then I
used the Quick-Time 5.0 to see the live stream and realized that the
QTPlayer always buffers 3 sec. of data. That means that the stream is not
"live". I think that a  delay of 3.5 sec (0.5 sec. = Card- encoding time),
between the recording and the seeing, is too much. I checked the settings of
the QT-Player but I couldn't find any settings that could help out. Then I
tried it wirh QT6.0. There is a setting called "instand-on". I tried all
possible configurations, but the delay was the same.   Does anyone have an
idea how to solve this problem? . Are there any other tools, that i could
use, as client-decoder?

Thx a lot, 
Andreas!   


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Hi all,
If I suppose that a delayed task function, containing a call to a "startplaying" function was called by the scheduler and I want to stop playing stream, after x minutes : 
is it sufficient to unschedule the delayed task function (after x minutes) to stop the play
Thanks in advance
Achraf



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<P>Hi all,
<P>If I suppose that&nbsp;a delayed task function, containing a call to a "startplaying" function was called by the scheduler and I want to stop playing stream, after x minutes : 
<P>is it sufficient to unschedule the delayed task function (after x minutes) to stop the play
<P>Thanks in advance
<P>Achraf</P><p><br><hr size=1><a href="http://fr.mail.yahoo.com">Yahoo! Mail</a> -- Une adresse (at)yahoo.fr gratuite et en français !<br>
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X-Original-Date: Tue, 24 Sep 2002 14:29:35 -0700
Date: Tue, 24 Sep 2002 14:29:35 -0700

At 02:37 AM 9/24/02, you wrote:

>Hi all,
>
>If I suppose that a delayed task function, containing a call to a 
>"startplaying" function was called by the scheduler and I want to stop 
>playing stream, after x minutes :
>
>is it sufficient to unschedule the delayed task function (after x minutes) 
>to stop the play

If the delayed task has not yet started, then you can cancel it by calling
         turnOffBackgroundReadHandling()

If the delayed task has already started, then it will have to stop itself 
(by just returning from its handler function).

         Ross.



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X-Original-Date: Tue, 24 Sep 2002 14:34:50 -0700
Date: Tue, 24 Sep 2002 14:34:50 -0700


>I used the Quick-Time 5.0 to see the live stream and realized that the
>QTPlayer always buffers 3 sec. of data. That means that the stream is not
>"live". I think that a  delay of 3.5 sec (0.5 sec. = Card- encoding time),
>between the recording and the seeing, is too much. I checked the settings of
>the QT-Player but I couldn't find any settings that could help out. Then I
>tried it wirh QT6.0. There is a setting called "instand-on". I tried all
>possible configurations, but the delay was the same.   Does anyone have an
>idea how to solve this problem?

I don't think it's possible to overcome this problem using QuickTime 
Player.  It was designed fror 'video-on-demand' style streaming, rather 
than live streaming, its buffering delay is inherent, and can't be 
overcome.  (At least, that's what I've read on various QuickTime-related 
mailing lists.)

>Are there any other tools, that i could use, as client-decoder?

You could use Cisco's "IP/TV" viewer tool.

On Linux, you can also run "mplayer" (with LIVE.COM Streaming Media 
support) - see <http://www.live.com/mplayer/>.  This probably has less 
delay (although audio/video sync is currently not working properly - but 
will get fixed soon).

         Ross.



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X-Original-Date: Tue, 24 Sep 2002 16:15:59 -0700
Date: Tue, 24 Sep 2002 16:15:59 -0700

Ross,

Thanks, it works now. Had to add another hack to get it to only write
once since the decoder expects it only at the beginning, but Darwin
seems to put it in each header.

Continuing in the spirit of quick-fixes, because our decoder isn't fully
ready to handle dropped frames, I want to tunnel our data through TCP
for the time being. I tried the (to me) obvious way of changing
RTSPClient.hh to not automatically set the streamOutgoing and
streamUsingTCP arguments of setupMediaSubsession to False, then in
openRTSP changing the line in setupRTSPStreams to
if (!rtspClient->setupMediaSubsession(*subsession, False, True))

I can see that openRTSP seems to handshake alright
Received SETUP response: RTSP/1.0 200 OK
Server: DSS/4.1.1 (Build/412.36; Platform/Win32)
Cseq: 2
Last-Modified: Wed, 11 Sep 2002 20:05:46 GMT
Cache-Control: must-revalidate
Session: 37456409820384
Date: Tue, 24 Sep 2002 22:56:52 GMT
Expires: Tue, 24 Sep 2002 22:56:52 GMT
Transport: RTP/AVP/TCP;unicast;interleaved=0-1;ssrc=00003C32

It says TCP ... it starts everything fine, and terminates fine, but
isn't outputting the stream. Is it capable of streaming through TCP at
this point, or am I setting things up incorrectly ...

Thanks as always

Gabe

> -----Original Message-----
> From: Ross Finlayson [mailto:finlayson(at)live.com]
> Sent: Monday, September 23, 2002 3:02 PM
> To: live-devel(at)lists.sourceforge.net
> Subject: Re: RE: [Live-devel] quicktime over RTP: what am I seeing?
> 
> At 02:47 PM 9/23/02, gnell(at)EECS.Berkeley.EDU wrote:
> >VC++ won't compile at
> >
> >QuickTimeGenericRTPSource* qtSource
> >             = (QuickTimeGenericRTPSource*)fSource;
> >
> >reporting error C2440: 'type cast' : cannot convert from 'class
> >FramedSource *MediaSink::* ' to 'class QuickTimeGenericRTPSource *'
> 
> Gabe,
> 
> Oops, my bad.
> 
> Change "fSource" in the above code to
>          sink->fSource
> 
> and make sure that this code occurs just after the (existing)
statement:
>          FileSink* sink = (FileSink*)clientData;
> 
> If you still get an error (about "fSource" being out of scope), then
also
> change the definition of "fSource" in "include/MediaSink.hh" from
> "protected:" to "public:".
> 
>          Ross.




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X-Original-Date: Tue, 24 Sep 2002 19:44:27 -0700
Date: Tue, 24 Sep 2002 19:44:27 -0700


>Continuing in the spirit of quick-fixes, because our decoder isn't fully
>ready to handle dropped frames, I want to tunnel our data through TCP
>for the time being.

Gabe,

RTP-over-TCP support is still only half implemented (it currently works in 
the outgoing direction, but not in the incoming direction - which is what 
you want).  I hope to fix this within the next few days, and when I do, 
I'll be adding an option to "openRTSP" to make it stream RTP-over-TCP.

So, your best bet is probably to wait until I've completed this.  (I'll be 
announcing it on the mailing list when it's done.)

         Ross.



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Subject: [Live-devel] reflect MPEGVideoRTPSource to another network
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X-Original-Date: Wed, 25 Sep 2002 11:03:30 +0800
Date: Wed, 25 Sep 2002 11:03:30 +0800

Hi,Ross

I am writting a program using LiveMedia lib to reflect a incoming
MPEGVideoRTPSource to another Multicast address.I used testMPEGAudioVideoStreamer
to generate the incoming data.

I modified testMPEGVideoReceiver.cpp to fit my need. I changed the "sink" from
'stdout' to MPEGVideoRTPSink. I changed the multicast address, i.e. ,
'239.255.42.43'. I also modified the SDP file to fit the address.

I used QT to play the movie from my SDP file. It work well first. But then it was
paused. after starting, my 'refletor' report "Warning:
MPEGVideoRTPSink::doSpecialFrameHandling saw strange first 4 bytes 0xa4f0dfb1, but
we're not a fragment" again and again.

Can I connect MPEGVideoRTPSource to MPEGVideoRTPSink directly? And how about audio
stream?

Thank you for your great work.




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X-Original-Date: Tue, 24 Sep 2002 20:19:55 -0700
Date: Tue, 24 Sep 2002 20:19:55 -0700


>I am writting a program using LiveMedia lib to reflect a incoming
>MPEGVideoRTPSource to another Multicast address.I used 
>testMPEGAudioVideoStreamer
>to generate the incoming data.

Note the "testRelay" test program (see "testRelay.cpp").  You should be 
able to modify this to do what you want.


>I modified testMPEGVideoReceiver.cpp to fit my need. I changed the "sink" from
>'stdout' to MPEGVideoRTPSink.

No, this won't work, because "MPEGVideoRTPSink" expects its input to be 
discrete MPEG video frames.  "MPEGVideoRTPSource" does not always output 
complete MPEG video frames (if frames are fragmented over more than one RTP 
packet).  So, instead, you would need to insert a "MPEGVideoStreamFramer" 
object between the "MPEGVideoRTPSource" and the "MPEGVideoRTPSink".

However, this approach is unnecessarily inefficient.  A much simpler 
approach is to simply reflect each RTP packet to another address/port, 
without worrying about the contents of each packet.  I.e., you should use 
something like "testRelay".

         Ross.



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Subject: [Live-devel] Compiling with MS eMbedded tools (attn: A Shukla, J Teo)
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X-Original-Date: Wed, 25 Sep 2002 13:49:50 -0700
Date: Wed, 25 Sep 2002 13:49:50 -0700

In early September there was some discussion about compiling for a PDA. 
I'm now trying to do that. I'm hoping someone who sucessfully compiled 
the libraries on CE can help me.

I am able to compile UsageEnvironment and create a library for windows 
CE ARM, using this command line:
"C:\Program Files\Microsoft eMbedded Tools\EVC\WCE300\BIN\clarm.exe" -
c -Iinclude -I../groupsock/include -Ox -c -W3 -DCRTAPI1=_cdecl -
DCRTAPI2=_cdecl -nologo -D_WIN95 -D_WIN32_WINDOWS=0x0400 -
D_WIN32_IE=0x0300 -DWINVER=0x0400 -DWIN32 -D_WIN32 -D_MT -
D_DLL /D "ARM" /D "_ARM_" -I. UsageEnvironment.cpp

The arguments were derived from how VC++ for the PC compiles it, plus 
some definitions for the ARM environment.

However, the other libraries won't compile -- the most obvious problem 
being that many files need iostream.h, which AFAIK doesn't exist on 
windows CE!

Minor things like <sys/types.h> being just <types.h> on CE are easily 
fixable ... but how to deal with no iostream?

Thanks,

Gabe



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From <EMAIL-ADDRESS-HIDDEN> Wed Sep 25 17:15:57 2002
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X-Original-Date: Wed, 25 Sep 2002 17:11:09 -0700
Date: Wed, 25 Sep 2002 17:11:09 -0700

At 01:49 PM 9/25/02, you wrote:
>In early September there was some discussion about compiling for a PDA.
>I'm now trying to do that. I'm hoping someone who sucessfully compiled
>the libraries on CE can help me.
>
>I am able to compile UsageEnvironment and create a library for windows
>CE ARM, using this command line:
>"C:\Program Files\Microsoft eMbedded Tools\EVC\WCE300\BIN\clarm.exe" -
>c -Iinclude -I../groupsock/include -Ox -c -W3 -DCRTAPI1=_cdecl -
>DCRTAPI2=_cdecl -nologo -D_WIN95 -D_WIN32_WINDOWS=0x0400 -
>D_WIN32_IE=0x0300 -DWINVER=0x0400 -DWIN32 -D_WIN32 -D_MT -
>D_DLL /D "ARM" /D "_ARM_" -I. UsageEnvironment.cpp
>
>The arguments were derived from how VC++ for the PC compiles it, plus
>some definitions for the ARM environment.

If you manage to get this all working, it might be useful to come up with a 
variant of "win32config" specifically for ARM Wince PDAs.

>However, the other libraries won't compile -- the most obvious problem
>being that many files need iostream.h, which AFAIK doesn't exist on
>windows CE!

Which file(s) need "iostream.h", and why?  It is not being explicitly 
"#include"d anywhere in the LIVE.COM code?

         Ross.



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Subject: [Live-devel] about h.263 codec
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X-Original-Date: 26 Sep 2002 13:40:44 -0000
Date: 26 Sep 2002 13:40:44 -0000

hi,
myself is nidhish.
i am using livemedia stacks for rtp transmission of video.
i have seen the test prog. for mpeg streaming and receiving.

but i have to rtp with h.263
question : can any one tell me how to use those test pro. for rtp 
of h.263 bitstream?
h.263videosource and sink are there in stacks but in mpeg pro. it 
uses mpeg framer to frame bitstreams. how to deal with this issue 
in h.263 bitstream?
waiting fot the reply.
nidhish



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Subject: [Live-devel] streaming by mplayer-rtsp
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Hi All!

I have finally installed the new mplayer on my computer...
by the "--disable-gcc-checking" option for the configure file!
I used some libraries to add more functionality to the player like
win32, Xanim and Vivo codecs and to be able to view a lot of different
media format...so I can now open *.mp3, *.rm, *.mov, *.mp4, *.mpg, *.avi
local files.
Then I tried to do streaming using mplayer like rtsp client and 
RealNetworks "Helix" Server like server (I have installed it before and
it 
works correctly).
The problem is the following one:
I'm able to open only *.mp3/*.mpg files via rtsp because when I try to 
stream other files (*.avi *.mov *.mp4 *.rm) mplayer says:
"Stream not seekable!
 Failed to initiate "XXXXXXXX" RTP subsession: RTP payload format 
unknown or not supported
 No stream found"

I think it is because Live libraries can support only MPEG-1 or 2 media,
is 
it so?
How can I do to stream other files?
Thank you, Santino Tomo.




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X-Original-Date: Thu, 26 Sep 2002 13:24:28 -0700
Date: Thu, 26 Sep 2002 13:24:28 -0700


>but i have to rtp with h.263
>question : can any one tell me how to use those test pro. for rtp of h.263 
>bitstream?
>h.263videosource and sink are there in stacks but in mpeg pro. it uses 
>mpeg framer to frame bitstreams. how to deal with this issue in h.263 
>bitstream?

You can use a "H263plusVideoRTPSink" to stream H.263+ video over RTP, but 
its source needs to provide it complete H.263+ frames.  So, you can't use a 
"ByteStreamFileSource" as the input to a "H263plusVideoRTPSink".  Instead, 
you will need a source object that produces complete frames.

If your H.263+ source is a (hardware or software) encoder, then it would be 
easy to write a "FramedSource" subclass that encapsulates it.  (See 
"DeviceSource.cpp" for a model of how you might do this.)

If, on the other hand, you are reading your H.263+ data from a file, then 
you can write a 'FramedFilter" subclass that parses the data into discrete 
frames.  In this case, the I/O chain would be
     ByteStreamFileSource -> your-filter-class -> H263plusVideoRTPSink

         Ross.



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From <EMAIL-ADDRESS-HIDDEN> Thu Sep 26 13:42:00 2002
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X-Original-Date: Thu, 26 Sep 2002 13:39:52 -0700
Date: Thu, 26 Sep 2002 13:39:52 -0700

At 07:15 AM 9/26/02, you wrote:
>by the "--disable-gcc-checking" option for the configure file!

Why did you need to do this?  What errors occurred if you omitted the 
"--disable-gcc-checking" flag?  (Were any of them fatal?)

>I used some libraries to add more functionality to the player like
>win32, Xanim and Vivo codecs and to be able to view a lot of different
>media format...so I can now open *.mp3, *.rm, *.mov, *.mp4, *.mpg, *.avi
>local files.
>Then I tried to do streaming using mplayer like rtsp client and
>RealNetworks "Helix" Server like server (I have installed it before and
>it
>works correctly).
>The problem is the following one:
>I'm able to open only *.mp3/*.mpg files via rtsp because when I try to
>stream other files (*.avi *.mov *.mp4 *.rm) mplayer says:
>"Stream not seekable!
>  Failed to initiate "XXXXXXXX" RTP subsession: RTP payload format
>unknown or not supported
>  No stream found"
>
>I think it is because Live libraries can support only MPEG-1 or 2 media,
>is it so?

No, the libraries also support several other codecs, including H.263+ 
video, and QCELP, GSM, or PCM audio.  (MPEG-4 video, and AAC and AC-3 audio 
are also on the "to do" list.)

Unfortunately streaming RealAudio and/or RealVideo sessions are not 
supported.  There are two reasons for this.  First, RealNetworks servers - 
even though they use RTSP to initiate streams - typically stream 
RealAudio/RealVideo sessions using their own, proprietary "RDT" protocol, 
rather than the open standard RTP protocol.  Second, when streaming 
RealAudio or RealVideo, RealNetworks servers often do a cryptographic 
'challenge-response' protocol (as part of the RTSP protocol) to ensure that 
only 'authorized' clients can receive the stream.

Unless RealNetworks documents the proprietary protocols that they use to 
stream RealAudio/RealVideo, the LIVE.COM libraries won't be able to support 
them.

(Apart from RealAudio, RealVideo, and MPEG-4, have you seen any other 
codecs that are supported by "mplayer", but for which RTSP/RTP streaming 
gives you a "RTP payload format unknown..." error?)

         Ross.



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X-Original-Date: Thu, 26 Sep 2002 13:50:05 -0700
Date: Thu, 26 Sep 2002 13:50:05 -0700

> If you manage to get this all working, it might be useful to come 
> up with a 
> variant of "win32config" specifically for ARM Wince PDAs.

Certainly.

> Which file(s) need "iostream.h", and why?  It is not being 
> explicitly 
> "#include"d anywhere in the LIVE.COM code?

It's included in NetAddress.hh.

Currently I'm going through compiling them one by one, attemping to fix 
errors as they crop up (slow work). Silly things like "strdup" not 
being defined, so having to malloc and strcpy on my own ... sigh.

Gabe



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X-Original-Date: Thu, 26 Sep 2002 14:05:28 -0700
Date: Thu, 26 Sep 2002 14:05:28 -0700


> > Which file(s) need "iostream.h", and why?  It is not being
> > explicitly
> > "#include"d anywhere in the LIVE.COM code?
>
>It's included in NetAddress.hh.

Oops, you're right.  I stand corrected.

>Currently I'm going through compiling them one by one, attemping to fix
>errors as they crop up (slow work). Silly things like "strdup" not
>being defined, so having to malloc and strcpy on my own ... sigh.

VC++ for Wince sounds like a total piece of crap.  IMHO, there's no excuse 
for not providing "iostream".  (The fact that PDA's don't really have a 
'console' is not an excuse.)  Maybe it's possible to port the Gnu 
"iostream" implementation to Wince (although that may be a lot of 
work).  It might be worthwhile looking around to see if anyone's done this.

Perhaps you should switch to an iPAQ or a Sharp Zaurus running Linux (a 
real operating system)?

         Ross.



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Hey everybody,
i am new to live development and i have a little question.
I am trying to build a new project using vc++ and the 
different live libraries. 
Following the advices of Anurag Skula, I did all the 
includes, the linking and the settings but i need those 
libraries : libUsageEnvironment.lib 
libBasicUsageEnvironment.lib libgroupsock.lib
libliveMedia.lib .
Where could i get those ? Thanks in advance
renato
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X-Original-Date: Mon, 30 Sep 2002 01:52:39 -0700
Date: Mon, 30 Sep 2002 01:52:39 -0700

At 01:54 AM 9/30/02, you wrote:
>Hey everybody,
>i am new to live development and i have a little question.
>I am trying to build a new project using vc++ and the
>different live libraries.
>Following the advices of Anurag Skula, I did all the
>includes, the linking and the settings but i need those
>libraries : libUsageEnvironment.lib
>libBasicUsageEnvironment.lib libgroupsock.lib
>libliveMedia.lib .
>Where could i get those ?

Renato,

Please follow the instructions at 
<http://www.live.com/liveMedia/#config-windows>.  You will need to build 
each of the "UsageEnvironment", "groupsock", "liveMedia", and 
"BasicUsageEnvironment" projects first, before building "testProgs".

         Ross.



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X-Original-Date: Mon, 30 Sep 2002 02:05:56 -0700
Date: Mon, 30 Sep 2002 02:05:56 -0700

The newly installed version (2002.09.30) of the "liveMedia" library 
contains an API change that may affect anyone who has subclassed (or 
modified) one of the existing "Medium" subclasses.

The "Medium::addNew()" member function has now been removed.  Because it 
was always called each time a new media object was created, I moved its 
functionality to the "Medium::Medium()" constructor.  This makes the code 
simpler.

If you currently have code that calls "Medium::addNew()", then you can just 
remove it.

	Ross.



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Hello!
I needed to use the "--disable-gcc-checking" options during the mplayer

configuration because without this instruction the configuration process

couldn't be started since mplayer doesn't support GCC version 2.96.
The problem was that I upgraded my GCC from 2.96 to 3.2 version but 
the new one is not detected during the configuration so I can't go on:
"...Checking for cc version ... 2.96, bad
 ...Error:Bad gcc version..."

Moreover, apart from *.rm and *.mov/*.mp4, also with *.avi and *.wmv files

I can't use the RTSP/RTP functionality,although they are supported by 
mplayer.

Thank you for the great availability, Santino Tomo.




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X-Original-Date: Fri, 04 Oct 2002 13:59:18 -0700
Date: Fri, 04 Oct 2002 13:59:18 -0700

A new version (2002.10.04) of the "LIVE.COM Streaming Media" libraries has 
now been installed.  This new version contains full support (in the 
"liveMedia") library for streaming RTP sessions (initiated using RTSP) over 
a TCP connection, instead of using UDP packets (which is what is usually done).

I have also added a new option "-t" to the "openRTSP" test program.  With 
this option, the RTP (and RTCP) packets will be streamed over the RTSP TCP 
connection, rather than as UDP packets.  For more information, see
	http://www.live.com/openRTSP/#other-options

(Note to Gabe Nell: This is what you were asking about a few days ago.)

	Ross.



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X-Original-Date: Sat, 5 Oct 2002 13:44:41 -0700
Date: Sat, 5 Oct 2002 13:44:41 -0700

Hey Ross,

Streaming over TCP seems to work great! Thanks.

I've created my own sink and got our decoder integrated.

I was looking through the DSS source and I notice they've got some sort
of "reliable" RTP scheme involving sending ACKs through RTCP APP
packets. I wonder if the quicktime player actually uses this ....

Gabe

> I have also added a new option "-t" to the "openRTSP" test program.
With
> this option, the RTP (and RTCP) packets will be streamed over the RTSP
TCP
> connection, rather than as UDP packets.  For more information, see
> 	http://www.live.com/openRTSP/#other-options
> 
> (Note to Gabe Nell: This is what you were asking about a few days
ago.)



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X-Original-Date: Sat, 05 Oct 2002 13:56:07 -0700
Date: Sat, 05 Oct 2002 13:56:07 -0700


>I was looking through the DSS source and I notice they've got some sort
>of "reliable" RTP scheme involving sending ACKs through RTCP APP
>packets. I wonder if the quicktime player actually uses this ....

It might, but this is something that's non-standard.  A (differnt) RTP 
retransmission mechanism is currently going through the IETF, so that's 
something I might end up supporting sometime in the future...

I'm glad to hear that you have your decoder working with the LIVE.COM 
libraries.  BTW, I'm interested in learning more about your video codec; 
are there any documents online that describe it?

         Ross.



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From: {Sunny Lai} <EMAIL-ADDRESS-HIDDEN>
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Subject: [Live-devel] mpeg1 video and QuickTime Client
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X-Original-Date: Mon, 7 Oct 2002 10:10:10 +0800
Date: Mon, 7 Oct 2002 10:10:10 +0800

Hello,

I install the lib and use the 'testMPEGAudioVideoStreamer' to stream to
QuickTime 6 client. It is very good, I test both rtsp and sdp file.

But I find some files I encoded by TMPGEnc on window has some problem. Such
as the video looks paused for one or two sec then catch up with the audio.
The Audio is ok. I encode those file using VCD quality.

Other then that, other file is ok. I try use VCDgear to covert 'Dat' file to
'mpeg' file then stream by your software, it is great.

Would you advice me what is wrong with my encoding??? Thanks!

Best Regards,

Sunny Lai




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Subject: Re: [Live-devel] mpeg1 video and QuickTime Client
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Date: Sun, 06 Oct 2002 19:28:08 -0700


>I install the lib and use the 'testMPEGAudioVideoStreamer' to stream to
>QuickTime 6 client. It is very good, I test both rtsp and sdp file.
>
>But I find some files I encoded by TMPGEnc on window has some problem. Such
>as the video looks paused for one or two sec then catch up with the audio.
>The Audio is ok. I encode those file using VCD quality.

Sunny,

Do these files play OK when you play them (directly, without streaming) 
using the QuickTime 6 Player?  I.e., does the problem that you're seeing 
occur only when you're streaming the file?

If so, could you put one of these MPEG files online (i.e., on a WWW or FTP 
site), and send me the URL, so I can take a look at it?  (Please don't send 
it via email; email's not meant for files that large.)

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>



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FYI

-----Original Message-----
From: Sunny Lai <EMAIL-ADDRESS-HIDDEN>
Sent:	Mon 10/7/2002 7:40 AM
To: <EMAIL-ADDRESS-HIDDEN>
Cc:
Subject:	[Live-devel] mpeg1 video and QuickTime Client

Hello,

I install the lib and use the 'testMPEGAudioVideoStreamer' to stream to
QuickTime 6 client. It is very good, I test both rtsp and sdp file.

But I find some files I encoded by TMPGEnc on window has some problem. Such
as the video looks paused for one or two sec then catch up with the audio.
The Audio is ok. I encode those file using VCD quality.

Other then that, other file is ok. I try use VCDgear to covert 'Dat' file to
'mpeg' file then stream by your software, it is great.

Would you advice me what is wrong with my encoding??? Thanks!

Best Regards,

Sunny Lai




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Hi all,
   please ignore my mail, it was a big mistake.
   Sorry for the trouble.

Best Regards
Rakesh 

-----Original Message-----
From: Rakesh Jain
Sent: Monday, October 07, 2002 10:20 AM
To: Sunny {Lai;} <EMAIL-ADDRESS-HIDDEN> Raj {Rajan;} Prakash Shrikrishna Bharankar
Subject: RE: [Live-devel] mpeg1 video and QuickTime Client


FYI

-----Original Message-----
From: Sunny Lai <EMAIL-ADDRESS-HIDDEN>
Sent:	Mon 10/7/2002 7:40 AM
To: <EMAIL-ADDRESS-HIDDEN>
Cc:
Subject:	[Live-devel] mpeg1 video and QuickTime Client

Hello,

I install the lib and use the 'testMPEGAudioVideoStreamer' to stream to QuickTime 6 client. It is very good, I test both rtsp and sdp file.

But I find some files I encoded by TMPGEnc on window has some problem. Such as the video looks paused for one or two sec then catch up with the audio. The Audio is ok. I encode those file using VCD quality.

Other then that, other file is ok. I try use VCDgear to covert 'Dat' file to 'mpeg' file then stream by your software, it is great.

Would you advice me what is wrong with my encoding??? Thanks!

Best Regards,

Sunny Lai




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X-Original-Date: Mon, 07 Oct 2002 02:09:54 -0700
Date: Mon, 07 Oct 2002 02:09:54 -0700

Sunny,

Thanks for sending me the file to look at.  Unfortunately I wasn't able to 
reproduce the problem that you were seeing.  (I ran 
"testMPEGAudioVideoStreamer" on one computer on my LAN, and viewed the 
video using QuickTime Player 6.0 running on another computer.)

I notice that the file's bitrate was about 1.3 Mbps.  Did the other files 
that you encoded - the ones where you saw no problems - have the same 
bitrate, or less?  Also, were you running QuickTime Player on the same 
computer as "testMPEGAudioVideoStreamer", or on a different 
computer.  QuickTime Player is a CPU hog, and I have sometime seen it pause 
its video output when it's overloaded.

	Ross.



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From: {Sunny Lai} <EMAIL-ADDRESS-HIDDEN>
To: {Ross Finlayson} <EMAIL-ADDRESS-HIDDEN> <EMAIL-ADDRESS-HIDDEN>
Subject: RE: [Live-devel] mpeg1 video and QuickTime Client
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X-Original-Date: Tue, 8 Oct 2002 09:13:17 +0800
Date: Tue, 8 Oct 2002 09:13:17 +0800

Hi Ross,

Thanks! I also guess it is due to data rate. But my PC running QuickTime
Player is 2G P4 and I find the CPU usage is only 58%. The data is on the
internal network so it is not bandwidth.

I also do another experiment. I use ffmpeg to encode again find if I use bit
rate at 1152kb/s (-b option), the problem occur. But if I use bit rate at
1000kb/s, it is ok. So what is the mpeg encoder you use???

Is it rtp or rtcp related??? Thanks!! By the way, I use mulitcast for above
test.

Regards,
Sunny

> -----Original Message-----
> From: Ross Finlayson [mailto:finlayson(at)live.com]
> Sent: Monday, October 07, 2002 5:10 PM
> To: live-devel(at)lists.sourceforge.net
> Cc: Sunny Lai
> Subject: RE: [Live-devel] mpeg1 video and QuickTime Client
>
>
> Sunny,
>
> Thanks for sending me the file to look at.  Unfortunately I
> wasn't able to
> reproduce the problem that you were seeing.  (I ran
> "testMPEGAudioVideoStreamer" on one computer on my LAN, and viewed the
> video using QuickTime Player 6.0 running on another computer.)
>
> I notice that the file's bitrate was about 1.3 Mbps.  Did the other files
> that you encoded - the ones where you saw no problems - have the same
> bitrate, or less?  Also, were you running QuickTime Player on the same
> computer as "testMPEGAudioVideoStreamer", or on a different
> computer.  QuickTime Player is a CPU hog, and I have sometime
> seen it pause
> its video output when it's overloaded.
>
> 	Ross.
>
>



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X-Original-Date: Mon, 07 Oct 2002 19:18:37 -0700
Date: Mon, 07 Oct 2002 19:18:37 -0700


>Thanks! I also guess it is due to data rate. But my PC running QuickTime
>Player is 2G P4 and I find the CPU usage is only 58%. The data is on the
>internal network so it is not bandwidth.
>
>I also do another experiment. I use ffmpeg to encode again find if I use bit
>rate at 1152kb/s (-b option), the problem occur. But if I use bit rate at
>1000kb/s, it is ok. So what is the mpeg encoder you use???

??? The LIVE.COM code doesn't do any MPEG encoding.  *You* do the MPEG 
encoding, when you produce the MPEG file to be streamed.

Or are you asking what MPEG encoder I use to produce the files that I use 
to test the code?  I don't - I just use MPEG files that other people have 
already produced.

I don't know what's causing the problem that you're seeing, but I suspect 
that QuickTime Player is to blame (plus, possibly, brain-damaged task 
scheduling in your OS.)  I suspect that if you try streaming from a 
different computer (on the same local-area network), QuickTime Player will 
be happier.

         Ross.



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X-Original-Date: 09 Oct 2002 13:01:41 -0400
Date: 09 Oct 2002 13:01:41 -0400



I have been playing with the PrioritizedRTPStreamSelector code again to
try and see why it doesn't quite work like our own implementation.

The PacketWarehouse is constructed with a seqNumStagger parameter.
Internally two other variables are initialized as follows.

    fMinSpanForDelivery(1*seqNumStagger)
    fMaxSpanForDelivery(2*seqNumStagger)

I have found that once in operation, the code will drain the warehouse
of packets to keep it at fMinSpanForDelivery.  Unfortunately I found
that the value 1*seqNumStagger arranges things such that packets from
the last substream often arrive just after the corresponding seq-number
has been shipped out of the warehouse.

I am able to work around this by artificially inflating the
seqNumStagger value.  2*seqNumStagger is definitely sufficient, but
probably overbuffers unnecessarily.  

Marty Schoch
<mschoch(at)multicasttech.com>



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X-Original-Date: Wed, 09 Oct 2002 13:59:22 -0700
Date: Wed, 09 Oct 2002 13:59:22 -0700

FYI, I have updated the RTSP/RTP streaming support for "mplayer" so that it 
can now (optionally) receive RTP (and RTCP) data over TCP, rather than 
UDP.  So, if you have a broken Internet connection that doesn't allow 
incoming UDP packets, you can now use "mplayer" to play RTSP-initiated 
media streams.

For details, see: http://www.live.com/mplayer/

	Ross.



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X-Original-Date: Wed, 09 Oct 2002 14:28:25 -0700
Date: Wed, 09 Oct 2002 14:28:25 -0700


>The PacketWarehouse is constructed with a seqNumStagger parameter.
>Internally two other variables are initialized as follows.
>
>     fMinSpanForDelivery(1*seqNumStagger)
>     fMaxSpanForDelivery(2*seqNumStagger)
>
>I have found that once in operation, the code will drain the warehouse
>of packets to keep it at fMinSpanForDelivery.  Unfortunately I found
>that the value 1*seqNumStagger arranges things such that packets from
>the last substream often arrive just after the corresponding seq-number
>has been shipped out of the warehouse.
>
>I am able to work around this by artificially inflating the
>seqNumStagger value.  2*seqNumStagger is definitely sufficient, but
>probably overbuffers unnecessarily.

Marty,

Thanks for taking a look at this.  We can probably get by with changing 
fMinSpanForDelivery and fMaxSpanForDelivery from 1 and 2 times 
seqNumStagger, respectively, to 1.5 and 3 times.  I'll make that change in 
the next source code release.

         Ross.




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X-Original-Date: Fri, 11 Oct 2002 09:17:18 -0400
Date: Fri, 11 Oct 2002 09:17:18 -0400

Has anyone tried to use the libraries with Windows CE and specifically
Pocket PC 2002?  Sorta new to this... maybe there is a webpage explaining
some stuff?

Thanks.

-josh 


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X-Original-Date: Fri, 11 Oct 2002 16:36:44 -0700
Date: Fri, 11 Oct 2002 16:36:44 -0700

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Hi All,
I=92m a newbie here and really eager to learn the art of streaming. =
I=92ve
interest in RTSP and would like to see myself implementing it. Still not
finding it easy. So please does anyone have a sample RTSP server, some
not too difficult to grasp code. I shall be grateful to have it or if
you can give me a pointer to where I can lay my hands on one. Please
source should be in c/c++.
=20
Thanks.
=20
-Yau

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<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Hi All,<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>I&#8217;m a newbie here and really eager to learn the =
art of
streaming. I&#8217;ve interest in RTSP and would like to see myself
implementing it. <span class=3DGramE>Still not finding it easy.</span> =
So please
does anyone have a sample RTSP server, some not too difficult to grasp =
code. I
shall be grateful to have it or if you can give me a pointer to where I =
can lay
my hands on one. Please source should be in =
c/c++.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Thanks.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>-Yau<o:p></o:p></span></font></p>

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X-Original-Date: Fri, 11 Oct 2002 14:10:25 -0700
Date: Fri, 11 Oct 2002 14:10:25 -0700


>I m a newbie here and really eager to learn the art of streaming. I ve 
>interest in RTSP and would like to see myself implementing it. Still not 
>finding it easy. So please does anyone have a sample RTSP server, some not 
>too difficult to grasp code. I shall be grateful to have it or if you can 
>give me a pointer to where I can lay my hands on one. Please source should 
>be in c/c++.

The "RTSPServer" class implements a simple RTSP server.  See 
"liveMedia/include/RTSPServer.hh" and "liveMedia/RTSPServer.cpp".  (Note 
that, at present, this implementation works only for multicast sessions.)

For examples of how this class is used to implement an embedded RTSP 
server, see the source code for the "testMP3Streamer", 
"testMPEGVideoStreamer", and "testMPEGAudioVideoStreamer" test programs.

         Ross.





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X-Original-Date: Fri, 11 Oct 2002 14:16:34 -0700
Date: Fri, 11 Oct 2002 14:16:34 -0700

At 06:17 AM 10/11/02, you wrote:
>Has anyone tried to use the libraries with Windows CE and specifically
>Pocket PC 2002?

A few people have been trying to do this, but Microsoft's "Embedded Visual 
C++" development environment is apparently brain damaged in a way that 
makes this difficult.  In particular:
         - it, unlike the regular VC++, apparently does not read make 
(".mak") files.
         - it does not provide several standard C++ interfaces, such as 
"iostream".

If, instead, you have a PDA that runs Linux, then there should be no 
problem getting the "LIVE.COM Streaming Media" code to work on 
it.  (Several people have already done this.)

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>



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X-Original-Date: Sun, 13 Oct 2002 13:23:02 -0700
Date: Sun, 13 Oct 2002 13:23:02 -0700

Hi Ross,
Thanks for the info. But I could not access the following pages:

http://www.live.com/liveMedia/include/RTSPServer.hh

http://www.live.com/liveMedia/RTSPServer.cpp

Also, I couldn't figure out where to pick up the source for
testMP3Streamer", "testMPEGVideoStreamer", and
"testMPEGAudioVideoStreamer"

Thanks.
-Yau


The "RTSPServer" class implements a simple RTSP server.  See 
"liveMedia/include/RTSPServer.hh" and "liveMedia/RTSPServer.cpp".  (Note

that, at present, this implementation works only for multicast
sessions.)

For examples of how this class is used to implement an embedded RTSP 
server, see the source code for the "testMP3Streamer", 
"testMPEGVideoStreamer", and "testMPEGAudioVideoStreamer" test programs.

         Ross.







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Subject: [Live-devel] How to get the source code
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X-Original-Date: Sun, 13 Oct 2002 09:18:23 -0700
Date: Sun, 13 Oct 2002 09:18:23 -0700

At 01:23 PM 10/13/02, you wrote:
>Hi Ross,
>Thanks for the info. But I could not access the following pages:
>
>http://www.live.com/liveMedia/include/RTSPServer.hh
>
>http://www.live.com/liveMedia/RTSPServer.cpp

No, that's not how you get fiese files.  They're part of the source code 
release, which is available - as a ".tar.gz" file - from 
http://www.live.com/liveMedia/public/

Please read the instructions at http://www.live.com/liveMedia/

         Ross.



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X-Original-Date: Sun, 13 Oct 2002 20:33:17 -0700
Date: Sun, 13 Oct 2002 20:33:17 -0700

eMbedded VC++ ALSO doesn't seem to like preprocessor directives, making
it really hard to write in a portable way, especially as many of the
#includes do not exist or are named differently.

We've made some progress compiling some skeleton functionality but
aren't there. Groupsock is the most difficult.

BTW, the newest embedded VC++ 4.0 with the service pack implements a lot
of what was missing from 3, seemingly with iostream.

Hopefully we'll be able to pull something together. 

Gabe

> A few people have been trying to do this, but Microsoft's "Embedded
Visual
> C++" development environment is apparently brain damaged in a way that
> makes this difficult.  In particular:
>          - it, unlike the regular VC++, apparently does not read make
> (".mak") files.
>          - it does not provide several standard C++ interfaces, such
as
> "iostream".



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Thread-Topic: Problem during MPEG-2 Streaming.
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From: {Prakash Shrikrishna  Bharankar} <EMAIL-ADDRESS-HIDDEN>
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Subject: [Live-devel] Problem during MPEG-2 Streaming.
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Hello
=20
I have downloaded the latest source code of live streaming media =
utility.
Using this code there is no problem while streaming MPEG-1 video only =
(video elementary stream)  to MPlayer client.
But when I try the same thing using MPEG-2 video only (video elementary =
stream) file, the MPlayer client is not able to play it properly.The =
picture quality is poor.
=20
Can anyone kindly let me know what could be the problem??
=20
Regards,
Prakash

------_=_NextPart_001_01C27684.90AD11A0
Content-Type: text/html;
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<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">
<TITLE>Message</TITLE>

<META content=3D"MSHTML 5.00.3315.2870" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D810075208-18102002>Hello</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D810075208-18102002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D810075208-18102002>I have =
downloaded=20
the latest source code of live streaming media =
utility.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D810075208-18102002>Using =
this=20
code&nbsp;there is no problem while streaming MPEG-1 video only (video=20
elementary stream)&nbsp; to MPlayer client.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D810075208-18102002>But =
when I try the=20
same thing using MPEG-2 video only (video elementary stream) file, the =
MPlayer=20
client is not able to play it properly.The picture quality is=20
poor.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D810075208-18102002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D810075208-18102002>Can=20
anyone&nbsp;kindly let me know what could be the =
problem??</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D810075208-18102002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D810075208-18102002>Regards,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D810075208-18102002>Prakash</SPAN></FONT></DIV></BODY></HTML>

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X-Original-Date: Fri, 18 Oct 2002 02:30:15 -0700
Date: Fri, 18 Oct 2002 02:30:15 -0700

At 01:58 AM 10/18/02, you wrote:
>Hello
>
>I have downloaded the latest source code of live streaming media utility.
>Using this code there is no problem while streaming MPEG-1 video only 
>(video elementary stream)  to MPlayer client.
>But when I try the same thing using MPEG-2 video only (video elementary 
>stream) file, the MPlayer client is not able to play it properly.The 
>picture quality is poor.
>
>Can anyone kindly let me know what could be the problem??

Prakash,

I think the problem here is in the mplayer code (rather than the LIVE.COM 
library code that it uses).  Its buffering of incoming data could be 
improved; at present, incoming packets in high-bitrate streams (such as 
MPEG-2) get dropped more often than they should.  Fixing this is on my "to 
do" list.

One alternative thing that you can do is run the "testMPEGVideoReceiver" 
test program to receive the video stream (and redirect stdout to a 
file).  You should then be able to play this file using mplayer (or any 
other MPEG player tool).

         Ross.



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From: {David Pontes} <EMAIL-ADDRESS-HIDDEN>
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Subject: RE: [Live-devel] mpeg1 video and QuickTime Client
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X-Original-Date: Tue, 22 Oct 2002 15:06:37 +0100
Date: Tue, 22 Oct 2002 15:06:37 +0100


	I'm experiencing the same effect on my trials with MPEG-1 audiovisual
bitstreams both at 1200 and 300 kbit/s. I attributed this to the lip-sync,
ie. when the player receives and interprets the first (?) RTCP packets for
each bitstream, it has to synchronize audio and video in some way. In this
case, it stops the video for a few frames. I have experienced the opposite
as well (audio pause) though not as often.
	Can anyone confirm this?

Thanks,
David Pontes


> -----Original Message-----
> From: live-devel-admin(at)lists.sourceforge.net
> [mailto:live-devel-admin(at)lists.sourceforge.net]On Behalf Of Sunny Lai
> Sent: terca-feira, 8 de Outubro de 2002 2:13
> To: Ross Finlayson; live-devel(at)lists.sourceforge.net
> Subject: RE: [Live-devel] mpeg1 video and QuickTime Client
>
>
> Hi Ross,
>
> Thanks! I also guess it is due to data rate. But my PC running QuickTime
> Player is 2G P4 and I find the CPU usage is only 58%. The data is on the
> internal network so it is not bandwidth.
>
> I also do another experiment. I use ffmpeg to encode again find
> if I use bit
> rate at 1152kb/s (-b option), the problem occur. But if I use bit rate at
> 1000kb/s, it is ok. So what is the mpeg encoder you use???
>
> Is it rtp or rtcp related??? Thanks!! By the way, I use mulitcast
> for above
> test.
>
> Regards,
> Sunny
>
> > -----Original Message-----
> > From: Ross Finlayson [mailto:finlayson(at)live.com]
> > Sent: Monday, October 07, 2002 5:10 PM
> > To: live-devel(at)lists.sourceforge.net
> > Cc: Sunny Lai
> > Subject: RE: [Live-devel] mpeg1 video and QuickTime Client
> >
> >
> > Sunny,
> >
> > Thanks for sending me the file to look at.  Unfortunately I
> > wasn't able to
> > reproduce the problem that you were seeing.  (I ran
> > "testMPEGAudioVideoStreamer" on one computer on my LAN, and viewed the
> > video using QuickTime Player 6.0 running on another computer.)
> >
> > I notice that the file's bitrate was about 1.3 Mbps.  Did the
> other files
> > that you encoded - the ones where you saw no problems - have the same
> > bitrate, or less?  Also, were you running QuickTime Player on the same
> > computer as "testMPEGAudioVideoStreamer", or on a different
> > computer.  QuickTime Player is a CPU hog, and I have sometime
> > seen it pause
> > its video output when it's overloaded.
> >
> > 	Ross.
> >
> >
>
>
>
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X-Original-Date: Tue, 22 Oct 2002 14:15:25 -0700
Date: Tue, 22 Oct 2002 14:15:25 -0700

At 07:06 AM 10/22/02, you wrote:

>         I'm experiencing the same effect on my trials with MPEG-1 audiovisual
>bitstreams both at 1200 and 300 kbit/s. I attributed this to the lip-sync,
>ie. when the player receives and interprets the first (?) RTCP packets for
>each bitstream, it has to synchronize audio and video in some way. In this
>case, it stops the video for a few frames. I have experienced the opposite
>as well (audio pause) though not as often.

This sounds plausible.  One test: Does the video (or audio) pausing happen 
just once, near the beginning of the stream, or does it happen often, 
throughout the stream?  If it happens just once, then a/v sync (based on 
RTCP) is probably the reason.  However, if this is happening several times, 
throughout the stream, then there might be a problem (e.g., with the 
timestamps generated by "testMPEGAudioVideoStreamer").

         Ross.



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 Streaming Media libraries
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X-Original-Date: Wed, 23 Oct 2002 10:23:26 -0700
Date: Wed, 23 Oct 2002 10:23:26 -0700

FYI.

>From: Marco Lohse
>Subject: NMM uses live
>
>Hi,
>I would like to inform you that our Open Source project called
>Network-Integrated Multimedia Middleware (NMM) and the Multimedia-Box
>application use the live library.
>
>If you would like more information about NMM, visit
>http://freshmeat.net/projects/nmm/
>
>Regards,
>Marco.
>--
>Marco Lohse, Computer Graphics Lab, Saarland University,
>Saarbruecken, Germany, http://graphics.cs.uni-sb.de/~mlohse




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X-Original-Date: Wed, 06 Nov 2002 07:21:21 GMT+1
Date: Wed, 06 Nov 2002 07:21:21 GMT+1

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hi everybody,
when i run the make file testprogs i don't have any sdp 
files created anymore. (i thought that when i first tried 
it last month 3 files were created and that was how i could 
launch quicktime).
Are sdp files only created for multicast session?
how could i manage to retrieve with quicktime a video+audio 
unicast streaming session
Thank you in advance
renato
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X-Original-Date: Wed, 06 Nov 2002 04:00:44 -0800
Date: Wed, 06 Nov 2002 04:00:44 -0800

At 11:21 PM 11/5/02, you wrote:
>hi everybody,
>when i run the make file testprogs i don't have any sdp
>files created anymore. (i thought that when i first tried
>it last month 3 files were created and that was how i could
>launch quicktime).

You can find the ".sdp" files in the "testProgs" directory.  (They aren't 
created by 'make'; instead, I  generated them 'by hand', and included them 
in the source file distribution.)

>how could i manage to retrieve with quicktime a video+audio
>unicast streaming session

First, modify the appropriate "test*Streamer" program for unicast, as 
described in the comments in the "test*Streamer.cpp" file.  (grep for 
"unicast")

Then, modify the corresponding "test*.sdp" file, replacing the IP address 
that's in each "c=" line with the IP address of the sender.

If you do this, then you may then be able to play the stream using 
QuickTime Player.

(I'm not 100% sure if this works; I haven't tried it lately...)

         Ross.



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X-Original-Date: Thu, 7 Nov 2002 16:52:11 +0100
Date: Thu, 7 Nov 2002 16:52:11 +0100

Hi everybody..
I would like to send a ISOCD 11172 MPEG-Systems Stream that contains two
video and one video Stream using the live-Library. Any hints how to achieve
this ? 
Thx for any help .
Andreas   


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X-Original-Date: Thu, 07 Nov 2002 09:27:13 -0800
Date: Thu, 07 Nov 2002 09:27:13 -0800

At 07:52 AM 11/7/02, you wrote:
>Hi everybody..
>I would like to send a ISOCD 11172 MPEG-Systems Stream that contains two
>video and one video Stream using the live-Library. Any hints how to achieve
>this ?

Andreas,

This depends on whether you want to stream this as a single System Stream 
(i.e., in a single RTP session), or as separate Elementary Streams (i.e., 
in multiple RTP sessions).

If you want to stream this as separate Elementary Streams, then I think you 
can do this using the existing "MPEGDemux" object.  See the code for the 
"testMPEGAudioVideoStreamer" test program for an example of how to do 
this.  (This test program currently demuxes and streams one audio and one 
video Elementary Stream from the input source, but you can easily adapt 
this to handle additional streams.)

If, on the other hand, you want to stream this as a single System Stream, 
then this is easy, because the RTP payload format for this is very simple 
(see RFC2250, section 2):

1/ Create your file source as follows:
ByteStreamFileSource::createNew(*env, inputFileName, <block-size>, <time>);

where <block-size> is the size (in bytes) of a discrete block of data that 
you wish to include in each outgoing RTP packet, and <time> is the time, in 
seconds, taken by this sized block of data.  These parameters tell the 
streaming code how much time to delay after sending each RTP packet.  (If 
you are streaming live input - i.e., if your input file is 'stdin' - then 
you can omit these two parameters.)

2/ Create your RTP sink as follows:
         SimpleRTPSink::createNew(*env, &rtpGroupsock, 96,  90000,"video", 
"mp1s");

(The "96" is a dyncmic RTP payload format code.  It can actually be any 
value in the range [96,127].)

I hope this helps...

         Ross.



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X-Original-Date: Fri, 8 Nov 2002 12:20:49 +0100
Date: Fri, 8 Nov 2002 12:20:49 +0100

Hello everybody.
    I'm playing with 'Live' library for a while and have a suggestion about
time consumption on windows. Since this actually happens because 'socket' call
fails if it was called with no entries set in "readSet", why not putting Sleep()
call after that has been detected?
    Or this issue is solved in new versions of library?

basicusageenvironment.cpp - line #254:
#if defined(__WIN32__) || defined(_WIN32)
    // Windows sucks - ignore bogus 'errors' that aren't real errors at all:

    if (errno == 0) {
      errno = WSAGetLastError();
    }
    // For some unknown reason, select() in Windoze sometimes fails with WSAEINVAL
if
    // it was called with no entries set in "readSet".  If this happens, ignore
it:
    if (errno == WSAEINVAL && readSet.fd_count == 0) {
      Sleep(100);  // Take a break.
      errno = 0;
    }
    if (errno != 0)
#endif

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X-Original-Date: Fri, 8 Nov 2002 13:25:28 -0200 (EDT)
Date: Fri, 8 Nov 2002 13:25:28 -0200 (EDT)

Hi,

I am doing a rtp proxy for exchange packets between Darwin Streaming
Server and openRTSP. I don't know if it is necessary exchange the RTCP
packets too. Does anyone know how this protocol works? I am just repassing
the openRTSP packets to the server. Am I right? 

Thanks in advance 
  Marisa




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  entries in windows
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X-Original-Date: Fri, 08 Nov 2002 13:12:45 -0800
Date: Fri, 08 Nov 2002 13:12:45 -0800

At 03:20 AM 11/8/02, you wrote:
>Hello everybody.
>     I'm playing with 'Live' library for a while and have a suggestion about
>time consumption on windows. Since this actually happens because 'socket' call
>fails if it was called with no entries set in "readSet", why not putting 
>Sleep()
>call after that has been detected?

Silvije,

The real problem here is the fact that select() is failing at all.  This 
isn't supposed to happen - it's apparently a bug in Windows.

Inserting a "Sleep()" call may reduce CPU consumption, but it will slow 
down the application unnecessarily, making it less responsive.  (Note that 
"BasicTaskScheduler::SingleStep()" checks not only for available data on 
read sockets, but also for delayed events that have come due.)

Are you seeing this Windows error (select() failing unexpectedly) 
often?  If so, what version of Windows are you running?

         Ross.



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X-Original-Date: Fri, 08 Nov 2002 13:15:10 -0800
Date: Fri, 08 Nov 2002 13:15:10 -0800


>I am doing a rtp proxy for exchange packets between Darwin Streaming
>Server and openRTSP.

Can you explain some more about what you're trying to do?  Note that 
"openRTSP" can already receive data that is streamed by a Darwin Streaming 
Server, so it should not be necessary to insert any 'proxy'  between the 
server  and openRTSP.

         Ross.



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Subject: Re: [Live-devel] BasicTaskScheduler - 'socket' call
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X-Original-Date: Mon, 11 Nov 2002 09:36:17 +0100
Date: Mon, 11 Nov 2002 09:36:17 +0100

>>Hello everybody.
>>     I'm playing with 'Live' library for a while and have a suggestion about

>>time consumption on windows. Since this actually happens because 'socket'
call
>>fails if it was called with no entries set in "readSet", why not putting
>> 
>>Sleep()
>>call after that has been detected?
>
>Silvije,
>
>The real problem here is the fact that select() is failing at all.  This 
>isn't supposed to happen - it's apparently a bug in Windows.
>
>Inserting a "Sleep()" call may reduce CPU consumption, but it will slow 
>down the application unnecessarily, making it less responsive.  (Note that

>"BasicTaskScheduler::SingleStep()" checks not only for available data on 
>read sockets, but also for delayed events that have come due.)
>
>Are you seeing this Windows error (select() failing unexpectedly) 
>often?  If so, what version of Windows are you running?
>
>         Ross.

select() fails every time when it is called with no entries set in "readSet",
and it is like that until at least one stream is created. After that, processor
time consumption is lower. I'm using Win2k professional.

- Silvije

--
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X-Original-Date: Mon, 11 Nov 2002 02:11:25 -0800
Date: Mon, 11 Nov 2002 02:11:25 -0800


>select() fails every time when it is called with no entries set in "readSet",
>and it is like that until at least one stream is created. After that, 
>processor
>time consumption is lower. I'm using Win2k professional.

Silvije,

The ideal solution would be to fix whatever is causing select() to fail on 
Windoze.  Could you please try the following:

In "BasicUsageEnvironment.cpp", line 186, change
         : fMaxNumSockets(0) {
to
         : fMaxNumSockets(1) {

(This will prevent select() from ever being called with a "nfds" parameter 
of 0.  Unix systems don't have a problem with this, but perhaps Windows does.)

Let me know if whether or not you still see select() failing after you make 
this change.

         Ross.



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X-Original-Date: Mon, 11 Nov 2002 10:00:53 -0200 (EDT)
Date: Mon, 11 Nov 2002 10:00:53 -0200 (EDT)


I am creating a proxy to distribute a video stream to many
clients. I don't want to create the RTCP packets, I just want to repass
them.  


	Marisa.

>>I am doing a rtp proxy for exchange packets between Darwin Streaming
>>Server and openRTSP.

>Can you explain some more about what you're trying to do?  Note that 
>"openRTSP" can already receive data that is streamed by a Darwin Streaming 
>Server, so it should not be necessary to insert any 'proxy'  between the 
>server  and openRTSP.

>         Ross.




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X-Original-Date: Mon, 11 Nov 2002 13:28:15 -0800
Date: Mon, 11 Nov 2002 13:28:15 -0800

At 04:00 AM 11/11/02, you wrote:

>I am creating a proxy to distribute a video stream to many
>clients.

OK, thanks for the clarification.

To answer your question: Yes, you should proxy RTCP packets as well as RTP 
packets.  This proxying should be done in both directions.  (Note that the 
DSS uses incoming RTCP packets from clients to check for 'liveness'.)

To learn more about RTP and RTCP, see the latest version of the RTP 
specification: 
<http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-new-11.txt>

         Ross.



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X-Original-Date: Tue, 12 Nov 2002 18:01:29 +0100
Date: Tue, 12 Nov 2002 18:01:29 +0100

>Silvije,

>The ideal solution would be to fix whatever is causing select() to fail on

>Windoze.  Could you please try the following:

>In "BasicUsageEnvironment.cpp", line 186, change
>         : fMaxNumSockets(0) {
>to
>         : fMaxNumSockets(1) {

>(This will prevent select() from ever being called with a "nfds" parameter

>of 0.  Unix systems don't have a problem with this, but perhaps Windows does.)


>Let me know if whether or not you still see select() failing after you make

>this change.

Thank you for your advice, but I&#8217;ve just downloaded new &#8216;Live&#8217;
library release and will first try to compile it and link with my applications.
Not everything is the same, because I&#8217;ve been using version from before
summer.
I did some changes and created some new classes that support sub session aggregation
the way Darwin Streaming Server accepts it. This way, they control all session
states using single session url and Darwin doesn&#8217;t hang after teardown.
I&#8217;m not sure if I was clear, but you get the point, right?
That has reminded me to ask, if I make just some minor changes to old classes
and use them create new (lots of code copy-pasting, if you know what I mean,
:) ), what are my obligations considering license of that code? New classes
also have to be open source and made available somewhere? 
Support considering session aggregation is really just few lines of code, and
I could post here part of it or something? Are you interested in that?

--
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X-Original-Date: Tue, 12 Nov 2002 10:42:37 -0800
Date: Tue, 12 Nov 2002 10:42:37 -0800


>I did some changes and created some new classes that support sub session 
>aggregation
>the way Darwin Streaming Server accepts it. This way, they control all session
>states using single session url and Darwin doesn&#8217;t hang after teardown.
>I&#8217;m not sure if I was clear, but you get the point, right?
>That has reminded me to ask, if I make just some minor changes to old classes
>and use them create new (lots of code copy-pasting, if you know what I mean,
>:) ), what are my obligations considering license of that code?

Under the LGPL license, if you make changes to any of the *existing* 
classes, and then distribute the changed version (e.g., in a product), then 
you are required to make the modified source code available.

Of course, if you make these changes only for your own personal use (e.g., 
for experimentation), rather than in a released product, then you're not 
required to distribute your changes.  However, if your changes improve upon 
the original code, then it would still make sense to post the changes to 
this list, so they can be included in a future source code release.

>  New classes
>also have to be open source and made available somewhere?

No, if these are completely new classes that are linked with the existing 
code, then you are *not* required to distribute the code for your new classes.

>Support considering session aggregation is really just few lines of code, and
>I could post here part of it or something? Are you interested in that?

Yes, please.  Once you've updated your code to match the latest version, 
please post your changes to this list.

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>




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From <EMAIL-ADDRESS-HIDDEN> Wed Nov 13 07:05:16 2002
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Subject: [Live-devel] Aggregate operation support - suggestion
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X-Original-Date: Wed, 13 Nov 2002 15:59:03 +0100
Date: Wed, 13 Nov 2002 15:59:03 +0100

What new RTSPClient class has to do to support session aggregation is to save
SessionID from first SETUP command response and use it to aggregate required
additional streams. I am researching movie streaming that requires audio and
video stream to be in sync, so this is very handy feature. First SETUP command
format is unchanged:
char* const cmdFmt = "SETUP %s%s%s RTSP/1.0\r\nCSeq: %d\r\nTransport: RTP/AVP%s%s%s=%d-%d\r\n\r\n";

Next calls to &#8216;setupMediaSubsession&#8217; should format SETUP command
using:
char* const cmdFmt = "SETUP %s%s%s RTSP/1.0\r\nCSeq: %d\r\nTransport: RTP/AVP%s%s%s=%d-%d\r\nSession:
%s\r\n\r\n"; [notice &#8216;Session: %s&#8217; line at the end]
After that, single &#8216;PLAY&#8217; and &#8216;TEARDOWN&#8217; commands with
&#8216;Session: <session id>&#8217; line are required to have effect on all
streams.

I&#8217;ve added this to RTSPClient:

Boolean RTSPClient::setupMediaSession(MediaSession& session, Boolean streamOutgoing,
Boolean streamUsingTCP) 
{
	MediaSubsessionIterator iter(session);
	MediaSubsessionEx *subsession;
	Boolean madeProgress = False;

	while ((subsession = iter.next()) != NULL) {
		if (subsession->clientPortNum() == 0) continue; // port # was not set
		if (setupMediaSubsession3(*subsession,streamOutgoing,streamUsingTCP)) {
			madeProgress = True;
		}
	}
	if (!madeProgress)
		return -1;

	return 0;
}

and setupMediaSubsession3, playMediaSession, teardownMediaSession methods that
use session id attribute to control state of all streams. Session id is saved
in MediaSession class.

Comments?

- Silvije

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X-Original-Date: Wed, 13 Nov 2002 17:37:11 +0100
Date: Wed, 13 Nov 2002 17:37:11 +0100



Dear Sir,

I want to develop a RTP diffusion from an embedded board, supporting an
real-time OS compatible
with 8086.
I have a daughetr card encodding in real-time in MP3 and diffusing me the
stream.
I want just to encapsulate it by RTP to diffuse via IP (Ethernet, Internet).

Is there an open source of RTP encpsulation compatible with 8086 develpement
?


Thnaks for your answer


Didier Dufournet



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X-Original-Date: Wed, 13 Nov 2002 12:03:40 -0800
Date: Wed, 13 Nov 2002 12:03:40 -0800


>I want to develop a RTP diffusion from an embedded board, supporting an
>real-time OS compatible
>with 8086.
>I have a daughetr card encodding in real-time in MP3 and diffusing me the
>stream.
>I want just to encapsulate it by RTP to diffuse via IP (Ethernet, Internet).
>
>Is there an open source of RTP encpsulation compatible with 8086 develpement?

Yes, the LIVE.COM Streaming Media software - http://www.live.com/liveMedia 
- was designed with environments like this in mind.  Note, for example, the 
"testMP3Streamer" test program, which streams MP3 over RTP.  You should be 
able to adapt this for your application.

Of course, you will need a C/C++ compiler.  Also, your operating system 
should support Unix-style network sockets, and "select()".  (If it doesn't, 
then you would need to modify the "groupsock" and "BasicUsageEnvironment" 
libraries to fit your operating system.)

I hope this helps.

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>



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X-Original-Date: Wed, 13 Nov 2002 12:10:32 -0800
Date: Wed, 13 Nov 2002 12:10:32 -0800

Silvije,

Could you please send your changes (to "MediaSession.cpp" and/or 
"RTSPClient.cpp") as a patch, apllied to the latest source code 
release.  (I.e., run "diff -u <oldfile> <newfile>")

	Ross.



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Subject: [Live-devel] RTP MPEG player for WIN CE (eVC++) ?
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X-Original-Date: Thu, 14 Nov 2002 15:08:16 -0500
Date: Thu, 14 Nov 2002 15:08:16 -0500

Hi all,
I am newbie to this field and and this list specifically. I briefly went 
thru the mailing list archives and I have a couple of questions.

  We  are working on a project to test the QoS guarantees of the  IPAQ 
Win CE, and in this regard we are looking for a RTP,  MPEG/MP 3 ( or any 
audio or video format  ) (client)  Player  to run over  Win CE,  
 Can we use the libraries / player provided here  on WIN CE ?
 If not is it  difficult /  possible to port it to eVC++ ?  

As  for our project is concerned , we are only concerned with the 
 statistics obtained  due to  streaming and not really on playing the 
MPEG we received( though it will be good to have it) and we plan to vary 
the background traffic load and other parameters and perform the study .

Any input / pointers or help in this regard will be greatly appreciated.
Thanks in advance.
Guru




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X-Original-Date: Thu, 14 Nov 2002 12:57:41 -0800
Date: Thu, 14 Nov 2002 12:57:41 -0800


>  We are working on a project to test the QoS guarantees of the  IPAQ Win 
> CE, and in this regard we are looking for a RTP,  MPEG/MP 3 ( or any 
> audio or video format  ) (client)  Player  to run over  Win CE,
>Can we use the libraries / player provided here  on WIN CE ?
>If not is it  difficult /  possible to port it to eVC++ ?

It is possible to build the code using eVC++, but it's non-trivial, because 
eVC++ is such a brain-damaged development environment.  (E.g., it 
apparently doesn't support "errno", and some versions (< 4.0) don't support 
"iostream".)  In other words, you can do this, but it will take some work.

         Ross.



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X-Original-Date: Thu, 14 Nov 2002 19:15:28 -0500
Date: Thu, 14 Nov 2002 19:15:28 -0500

Thanks Ross,
I was wondering  if there  is a way we can reconstruct the  mpeg or mp3 
files from the  RTP payload  we receive?
do we need to do any special processing for that or just dump it to a 
file  and that  produces a mpeg or .mp3 file ?

also is there any simple  open source RTP player in  eVC++ ?
so we that we can add our tabs in that for our Qos Study !

thanks
Guru


Ross Finlayson wrote:

>
>>  We are working on a project to test the QoS guarantees of the  IPAQ 
>> Win CE, and in this regard we are looking for a RTP,  MPEG/MP 3 ( or 
>> any audio or video format  ) (client)  Player  to run over  Win CE,
>> Can we use the libraries / player provided here  on WIN CE ?
>> If not is it  difficult /  possible to port it to eVC++ ?
>
>
> It is possible to build the code using eVC++, but it's non-trivial, 
> because eVC++ is such a brain-damaged development environment.  (E.g., 
> it apparently doesn't support "errno", and some versions (< 4.0) don't 
> support "iostream".)  In other words, you can do this, but it will 
> take some work.
>
>         Ross.
>
>
>
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-- 
Guru Ranganathan, Masters in Computer Networking, NCSU
Ph:919-412-6112  Res:1713,Crest Rd, #4 Raleigh, NC 27606






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From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] RTP MPEG player for WIN CE (eVC++)?
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X-Original-Date: Fri, 15 Nov 2002 02:12:15 -0800
Date: Fri, 15 Nov 2002 02:12:15 -0800


>I was wondering  if there  is a way we can reconstruct the  mpeg or mp3 
>files from the  RTP payload  we receive?

Yes, for MP3 audio or MPEG (1 or 2) video RTP streams you can do this very 
easily.  In each packet, skip the initial, 12-byte RTP header, and the 
following, 4 byte 'media-specific header'.  I.e., ignore the first 16 bytes 
of each MP3/RTP or MPEG/RTP packet.  The remaining data will be data (that 
you can play in a regular media player).

For more information, see RFC 2250, or its implementation in 
"MPEGAudioRTPSource.cpp" and "MPEGVideoRTPSource.cpp".

         Ross Finlayson
         LIVE.COM



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Date: Fri, 15 Nov 2002 11:49:30 +0100

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>Silvije,

>Could you please send your changes (to "MediaSession.cpp" and/or 
>"RTSPClient.cpp") as a patch, apllied to the latest source code 
>release.  (I.e., run "diff -u <oldfile> <newfile>")

>	Ross.

Ok, diff files are attached to this mail. They are created using cygwin shell
in windows environment, so I guess they have DOS formatted line ends, so be
careful.
New methods aren&#8217;t using authentication feature found in new releases
of &#8216;Live&#8217; library. MediaSession::initialize also has minimal support
for MPEG4 video RTP packets using SimpleRTPSource class.

- Silvije

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From <EMAIL-ADDRESS-HIDDEN> Fri Nov 15 07:40:35 2002
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X-Original-Date: Fri, 15 Nov 2002 16:33:18 +0100
Date: Fri, 15 Nov 2002 16:33:18 +0100

Thanks Ross.

I have compiled succesfully MP3Stremer project but I have the following
message when executing :

16:31:42 Groupsock(104: 239.255.42.42, 6666, 1): failed to join group:
setsockop
t(IP_ADD_MEMBERSHIP) error: Unknown error
Unable to determine our source address: setsockopt(IP_ADD_MEMBERSHIP) error:
Unk
nown error
16:31:42 Groupsock(112: 239.255.42.42, 6667, 1): failed to join group:
setsockop
t(IP_ADD_MEMBERSHIP) error: Unknown error
Unable to determine our source address: setsockopt(IP_ADD_MEMBERSHIP) error:
Unk
nown error
Beginning streaming...

Any idea ?



-----Message d'origine-----
De : live-devel-admin(at)lists.sourceforge.net
[mailto:live-devel-admin(at)lists.sourceforge.net]De la part de Ross
Finlayson
Envoye : mercredi 13 novembre 2002 21:04
A : live-devel(at)lists.sourceforge.net
Objet : RE: [Live-devel] BasicTaskScheduler - 'socket' call



>I want to develop a RTP diffusion from an embedded board, supporting an
>real-time OS compatible
>with 8086.
>I have a daughetr card encodding in real-time in MP3 and diffusing me the
>stream.
>I want just to encapsulate it by RTP to diffuse via IP (Ethernet,
Internet).
>
>Is there an open source of RTP encpsulation compatible with 8086
develpement?

Yes, the LIVE.COM Streaming Media software - http://www.live.com/liveMedia
- was designed with environments like this in mind.  Note, for example, the
"testMP3Streamer" test program, which streams MP3 over RTP.  You should be
able to adapt this for your application.

Of course, you will need a C/C++ compiler.  Also, your operating system
should support Unix-style network sockets, and "select()".  (If it doesn't,
then you would need to modify the "groupsock" and "BasicUsageEnvironment"
libraries to fit your operating system.)

I hope this helps.

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>



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X-Original-Date: Fri, 15 Nov 2002 08:48:42 -0800
Date: Fri, 15 Nov 2002 08:48:42 -0800


>I have compiled succesfully MP3Stremer project but I have the following
>message when executing :
>
>16:31:42 Groupsock(104: 239.255.42.42, 6666, 1): failed to join group:
>setsockopt(IP_ADD_MEMBERSHIP) error: Unknown error

These errors suggest that your operating system does not support IP 
multicast.  If possible, you should reconfigure your OS kernel to support 
IP multicast.  (It's not necessary to support IP multicast *routing*, just 
IP multicast.)

         Ross.



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From: <EMAIL-ADDRESS-HIDDEN> (S. Kannan)
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Subject: RE: [Live-devel] BasicTaskScheduler - 'socket' call
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X-Original-Date: Fri, 15 Nov 2002 12:30:34 -0500
Date: Fri, 15 Nov 2002 12:30:34 -0500

Hello Ross:

Could you please take of my email (s.kannan(at)emessages.com) from the group.
Thanks.

:Kannan

-----Original Message-----
From: <EMAIL-ADDRESS-HIDDEN>
[mailto:live-devel-admin(at)lists.sourceforge.net]On Behalf Of Ross
Finlayson
Sent: Friday, November 15, 2002 11:49 AM
To: <EMAIL-ADDRESS-HIDDEN>
Subject: RE: [Live-devel] BasicTaskScheduler - 'socket' call



>I have compiled succesfully MP3Stremer project but I have the following
>message when executing :
>
>16:31:42 Groupsock(104: 239.255.42.42, 6666, 1): failed to join group:
>setsockopt(IP_ADD_MEMBERSHIP) error: Unknown error

These errors suggest that your operating system does not support IP
multicast.  If possible, you should reconfigure your OS kernel to support
IP multicast.  (It's not necessary to support IP multicast *routing*, just
IP multicast.)

         Ross.



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X-Original-Date: Fri, 15 Nov 2002 11:30:27 -0800
Date: Fri, 15 Nov 2002 11:30:27 -0800


>Could you please take of my email (s.kannan(at)emessages.com) from the group.

You should do this yourself, by sending an email message, with subject
         unsubscribe
to
         live-devel-request(at)lists.sourceforge.net

Please don't send unsubscribe requests directly to this (or any other) 
mailing list.

         Ross.



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X-Original-Date: Fri, 15 Nov 2002 23:31:31 +0100
Date: Fri, 15 Nov 2002 23:31:31 +0100

Hello,

I've been trying to compile Live library on AIX at university and I had
to make
some small changes, I don't know very well why,

1.- I had to change in Makefile 'gmake' instead of 'make', and 'gcc'
instead of
'cc'
2.- I had to add <strings.h> in RTSPClient.hh to make work strncasecmp()
2.- I had to add <strings.h> in GroupsockHelper.hh to make work bzero()

After this, test examples worked fine, but I think I still had some
warnings...
Anybody else had this problem? That's all, maybe this could help to
someone on Aix, or maybe not!
Thanks Ross for this great library :)

        Ruben.


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X-Original-Date: Fri, 15 Nov 2002 17:02:02 -0800
Date: Fri, 15 Nov 2002 17:02:02 -0800

Ruben,

Thanks for the tip.  FYI, thanks to your feedback, the latest source code 
release (2002.11.15) now does "#include <strings.h>" (as well as 
<string.h>) in two of the base include files: 
"groupsock/GroupsockHelper.hh" and "liveMedia/Media.hh".  This should make 
the code more portable.

	Ross.



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X-Original-Date: Sun, 17 Nov 2002 10:53:06 -0500
Date: Sun, 17 Nov 2002 10:53:06 -0500

Hi,

Just like to ask if I were to use UDP in Windows Socket (CSocket class in VC++) for transmisson of audio and video signals across a LAN, what is the difference between using UDP and using Live.com RTP?

I'm just a little confused about RTP and UDP..are they the same? 

Thanks..

Joe


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From <EMAIL-ADDRESS-HIDDEN> Sun Nov 17 20:32:21 2002
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X-Original-Date: Sun, 17 Nov 2002 20:15:28 -0800
Date: Sun, 17 Nov 2002 20:15:28 -0800


>Just like to ask if I were to use UDP in Windows Socket (CSocket class in 
>VC++) for transmisson of audio and video signals across a LAN, what is the 
>difference between using UDP and using Live.com RTP?

RTP is the industry standard Internet protocol for transporting streaming 
media (including videoconferencing and voice-over-ip).  FYI, the most 
up-to-date version of the RTP specification can be found at
         http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-new-11.txt

UDP is also a standard Internet protocol - for transporting data in 
(unreliable) datagrams.  RTP is usually implemented using UDP.  I.e., a RTP 
packet is just a UDP packet, but containing data fields that are specific 
to RTP - for example, a timestamp (for media synchronization), and a 
sequence number (for detecting data loss).

In other words, RTP is one specific application of UDP, but UDP can also be 
used for applications other than RTP.

It is certainly possible to use UDP, without RTP, to send audio/video data, 
but if you did this, you wouldn't get the benefits of RTP, and you wouldn't 
be able to interoperate with other, existing tools (such as RealOne Player 
or QuickTime Player).  So, you shouldn't do this.  Instead, use RTP>

I hope this helps.

         Ross Finlayson
         LIVE.COM



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From <EMAIL-ADDRESS-HIDDEN> Mon Nov 18 05:35:32 2002
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Subject: RE: [Live-devel] RTP MPEG player for WIN CE (eVC++)?
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X-Original-Date: Mon, 18 Nov 2002 08:26:45 -0500
Date: Mon, 18 Nov 2002 08:26:45 -0500

Guru,

I would check out this open source project called Pocket DivX...

http://home.adelphia.net/~mdukette/

or

http://www.projectmayo.com/

It's a MPEG1,MP3, and DivX player with RTP written in eVC++.

Hope this helps.


-josh

-----Original Message-----
From: Guru Ranganathan <EMAIL-ADDRESS-HIDDEN>
Sent: Thu, November 14, 2002 7:15 PM
To: <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] RTP MPEG player for WIN CE (eVC++)?


Thanks Ross,
I was wondering  if there  is a way we can reconstruct the  mpeg or mp3 
files from the  RTP payload  we receive?
do we need to do any special processing for that or just dump it to a 
file  and that  produces a mpeg or .mp3 file ?

also is there any simple  open source RTP player in  eVC++ ?
so we that we can add our tabs in that for our Qos Study !

thanks
Guru


Ross Finlayson wrote:

>
>>  We are working on a project to test the QoS guarantees of the  IPAQ 
>> Win CE, and in this regard we are looking for a RTP,  MPEG/MP 3 ( or 
>> any audio or video format  ) (client)  Player  to run over  Win CE,
>> Can we use the libraries / player provided here  on WIN CE ?
>> If not is it  difficult /  possible to port it to eVC++ ?
>
>
> It is possible to build the code using eVC++, but it's non-trivial, 
> because eVC++ is such a brain-damaged development environment.  (E.g., 
> it apparently doesn't support "errno", and some versions (< 4.0) don't 
> support "iostream".)  In other words, you can do this, but it will 
> take some work.
>
>         Ross.
>
>
>
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-- 
Guru Ranganathan, Masters in Computer Networking, NCSU
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Subject: [Live-devel] MPEG Systems (ISO 11172-1) Multiplexer
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X-Original-Date: Mon, 18 Nov 2002 22:29:40 +0100
Date: Mon, 18 Nov 2002 22:29:40 +0100

Hi everybody, well I would like to ask, if it is possible to build a iso
11172-1 multiplexer based upon the Live-Libs.I need to multiplex: 
2 Audio (11172-3) and
1 Video 11172-2) Stream
and then stream it by using rtp.

Is it possible that i could do  this by building three new Classes
"MPEGMux", "MPEGVideoReceiver", "MPEGAudioReceiver" ?



The streaming App would do something like:

//Begin code
// Declarations
MPEGVideoReceiver *mpegVideoReceiverOne; (new Class from
testMPEGVideoReceiver)
MPEGVideoReceiver *mpegVideoReceiverTwo;
MPEGAudioReceiver *mpegAudioReceiverOne; (new Class)
FramedSource* audioSourceOne=NULL;
FramedSource* videoSourceTwo=NULL;
FramedSource* videoSourceTwo=NULL;
MPEGMux mpegSystemMux;

mpegVideoReceiverOne= createNew(*env, multicastAddress, rtpPort); 
mpegVideoReceiverTwo= createNew(*env, multicastAddress, rtpPort);
mpegAudioReceiverOne= createNew(*env, multicastAddress, rtpPort);

FramedSource * videoOne  = mpegVideoReceiverOne->newVideoStream();
videoSourceOne= MPEGVideoStreamFramer::createNew(*env, videoSourceOne,
iFramesOnly);

FramedSource*  = mpegVideoReceiverTwo->newVideoStream();
videoSourceTwo= MPEGVideoStreamFramer::createNew(*env, , iFramesOnly);

FramedSource* audioOne = mpegAudioReceiverOne->newAudioStream();
audioSourceOne= MPEGAudioStreamFramer::createNew(*env, );

ByteStreamFileSource* systemStream;
mpegSystemMux= MPEGMux::createNew(*env,
videoSourceOne,videoSourceTwo,audioSourceOne);
systemStream=mpegSystemMux->newSystemStream();

/*** stream systemStream **/
}
//End code

Would something like that work?
Thanks for your help. 
Best regards 




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X-Original-Date: Mon, 18 Nov 2002 15:14:32 -0800
Date: Mon, 18 Nov 2002 15:14:32 -0800

At 01:29 PM 11/18/02, you wrote:
>Hi everybody, well I would like to ask, if it is possible to build a iso
>11172-1 multiplexer based upon the Live-Libs.I need to multiplex:
>2 Audio (11172-3) and
>1 Video 11172-2) Stream
>and then stream it by using rtp.

Andreas,

Just to clarify, do you want to build a multiplexor, or a *de*multiplexor?

1/ A multiplexor would take the three Elementary Streams (2 audio, 1 video) 
as input, combine them into a single MPEG Program or Transport Stream, and 
then transmit this (one) stream using RTP.

2/ A demultiplexor would take a MPEG Program (or Transport) Stream, split 
this into three separate Elementary Streams (2 audio, 1 video), and 
transmit these (three) streams using RTP.

Which of these do you want?

(2/ could be done very easily, using existing code, if your input is a 
Program Stream.  It would just be a slight modification of the existing 
"testMPEGAudioVideoStreamer" test program.)

         Ross.



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Subject: [Live-devel] TCP tunneling bug
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X-Original-Date: Mon, 18 Nov 2002 17:08:20 -0800
Date: Mon, 18 Nov 2002 17:08:20 -0800

There appears to be a small bug when communicating with Darwin over TCP.

For example, with Darwin there are some included sample movies, like 
sample_100kbit.mov

comparing the audio-X-QT from

openrtsp rtsp://server/sample_100kbit.mov

to

openrtsp -t rtsp://server/sample_100kbit.mov

yields a byte size of 118,312 for the UDP version and 118,128 for the TCP 
tunneled version. I think it's dropping the first audio frame/frames or 
something. I've tried it with different .mov's and it seems to happen with 
audio but not video.

Joe

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X-Original-Date: Mon, 18 Nov 2002 20:47:31 -0500
Date: Mon, 18 Nov 2002 20:47:31 -0500

How about for 1) i.e. multiplex the 2 audio and 1 video? 

Will it be tough to do that? Do you have any suggestion or sample on doing that?

Thanks.

Joseph

 
--

On Mon, 18 Nov 2002 15:14:32  
 Ross Finlayson wrote:
>At 01:29 PM 11/18/02, you wrote:
>>Hi everybody, well I would like to ask, if it is possible to build a iso
>>11172-1 multiplexer based upon the Live-Libs.I need to multiplex:
>>2 Audio (11172-3) and
>>1 Video 11172-2) Stream
>>and then stream it by using rtp.
>
>Andreas,
>
>Just to clarify, do you want to build a multiplexor, or a *de*multiplexor?
>
>1/ A multiplexor would take the three Elementary Streams (2 audio, 1 video) 
>as input, combine them into a single MPEG Program or Transport Stream, and 
>then transmit this (one) stream using RTP.
>
>2/ A demultiplexor would take a MPEG Program (or Transport) Stream, split 
>this into three separate Elementary Streams (2 audio, 1 video), and 
>transmit these (three) streams using RTP.
>
>Which of these do you want?
>
>(2/ could be done very easily, using existing code, if your input is a 
>Program Stream.  It would just be a slight modification of the existing 
>"testMPEGAudioVideoStreamer" test program.)
>
>         Ross.
>
>
>
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Subject: Re: [Live-devel] TCP tunneling bug
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X-Original-Date: Mon, 18 Nov 2002 17:49:05 -0800
Date: Mon, 18 Nov 2002 17:49:05 -0800

... and I think it has to do with the data sometimes discarded in 
RTSPClient::getResponse

Joe


>From: "Joe Shipp" <sgw1178(at)hotmail.com>
>Reply-To: live-devel(at)lists.sourceforge.net
>To: live-devel(at)lists.sourceforge.net
>Subject: [Live-devel] TCP tunneling bug
>Date: Mon, 18 Nov 2002 17:08:20 -0800
>
>There appears to be a small bug when communicating with Darwin over TCP.
>
>For example, with Darwin there are some included sample movies, like 
>sample_100kbit.mov
>
>comparing the audio-X-QT from
>
>openrtsp rtsp://server/sample_100kbit.mov
>
>to
>
>openrtsp -t rtsp://server/sample_100kbit.mov
>
>yields a byte size of 118,312 for the UDP version and 118,128 for the TCP 
>tunneled version. I think it's dropping the first audio frame/frames or 
>something. I've tried it with different .mov's and it seems to happen with 
>audio but not video.
>
>Joe
>
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From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
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X-Original-Date: Mon, 18 Nov 2002 20:26:22 -0800
Date: Mon, 18 Nov 2002 20:26:22 -0800

At 05:47 PM 11/18/02, you wrote:
>How about for 1) i.e. multiplex the 2 audio and 1 video?
>
>Will it be tough to do that? Do you have any suggestion or sample on doing 
>that?

I've already replied to Andreas (who originally asked about this); here's 
what I told him:

-----
>I want to make "1/". I want to stream a ISO-11172-1 Stream that is build of
>1Audio Streams and 2 VideoStreams.

Thanks for the clarification.

Yes it should be possible to use the LIVE.COM Streaming Media libraries to 
do this, but it would require that a new "System Stream Multiplexor" class 
be written.  An object of this class would take an arbitrary number of 
'FramedSources' (each for an Elementary Stream) as input, and output a 
single System Stream.  (This System Stream could then be easily transmitted 
using RTP, using the existing "SimpleRTPSink" class.)

For help in figuring out how to write the new "System Stream Multiplexor" 
class, you may find it useful to examine the code for
"PrioritizedRTPStreamSelector".  This also takes N input streams, and 
produces 1 output stream.
-----

         Ross.



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X-Original-Date: Mon, 18 Nov 2002 20:48:41 -0800
Date: Mon, 18 Nov 2002 20:48:41 -0800

At 05:08 PM 11/18/02, you wrote:
>There appears to be a small bug when communicating with Darwin over TCP.
>
>For example, with Darwin there are some included sample movies, like 
>sample_100kbit.mov
>
>comparing the audio-X-QT from
>
>openrtsp rtsp://server/sample_100kbit.mov
>
>to
>
>openrtsp -t rtsp://server/sample_100kbit.mov
>
>yields a byte size of 118,312 for the UDP version and 118,128 for the TCP 
>tunneled version. I think it's dropping the first audio frame/frames or 
>something.

>... and I think it has to do with the data sometimes discarded in 
>RTSPClient::getResponse

Yes, you're correct.  The problem can occur when there are two (or more) 
media tracks in the RTSP session.  After the RTSP client issues the RTSP 
"PLAY" command for the first media track, and gets back the "OK" response, 
it then sends the RTSP "PLAY" command for the second media track.  At this 
point, it is still reading (synchronously) from the TCP socket, in order to 
read the response (hopefully "OK") from the second "PLAY".  But at this 
point it's possible for the server to already be sending back RTP data for 
the first track, embedded within the TCP stream.  When the client sees this 
RTP data, there's no easy way to tell the RTP client to read it (in part 
because the RTP client(s), unlike the RTSP client, reads from the TCP 
socket asynchronously rather than synchronously), so the RTSP client just 
discards this embedded RTP data, until it gets the RTSP response that it's 
looking for.

You can see this by running openRTSP with the "-V" (verbose) flag.

Fixing this bug would be non-trivial and messy.  How important is this 
initial data to you? :-)

         Ross.



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X-Original-Date: Tue, 19 Nov 2002 00:02:59 +0000
Date: Tue, 19 Nov 2002 00:02:59 +0000

On Tuesday 19 November 2002 04:48 am, Ross Finlayson wrote:
> At 05:08 PM 11/18/02, you wrote:
> >There appears to be a small bug when communicating with Darwin over TCP.
> >
> >For example, with Darwin there are some included sample movies, like
> >sample_100kbit.mov
> >
> >comparing the audio-X-QT from
> >
> >openrtsp rtsp://server/sample_100kbit.mov
> >
> >to
> >
> >openrtsp -t rtsp://server/sample_100kbit.mov
> >
> >yields a byte size of 118,312 for the UDP version and 118,128 for the TCP
> >tunneled version. I think it's dropping the first audio frame/frames or
> >something.
> >
> >... and I think it has to do with the data sometimes discarded in
> >RTSPClient::getResponse
>
> Yes, you're correct.  The problem can occur when there are two (or more)
> media tracks in the RTSP session.  After the RTSP client issues the RTSP
> "PLAY" command for the first media track, and gets back the "OK" response,
> it then sends the RTSP "PLAY" command for the second media track.  At this
> point, it is still reading (synchronously) from the TCP socket, in order to
> read the response (hopefully "OK") from the second "PLAY".  But at this
> point it's possible for the server to already be sending back RTP data for
> the first track, embedded within the TCP stream.  When the client sees this
> RTP data, there's no easy way to tell the RTP client to read it (in part
> because the RTP client(s), unlike the RTSP client, reads from the TCP
> socket asynchronously rather than synchronously), so the RTSP client just
> discards this embedded RTP data, until it gets the RTSP response that it's
> looking for.
>
> You can see this by running openRTSP with the "-V" (verbose) flag.
>
> Fixing this bug would be non-trivial and messy.  How important is this
> initial data to you? :-)
>
>          Ross.
>

Wouldn't the aggregate specification in the RTSP RFC solve this?


C.3 Aggregate Control Available

   In this scenario, the server has multiple streams that can be
   controlled as a whole. In this case, there are both media-level
   "a=control:" attributes, which are used to specify the stream URLs,
   and a session-level "a=control:" attribute which is used as the
   request URL for aggregate control. If the media-level URL is
   relative, it is resolved to absolute URLs according to Section C.1.1
   above.

   If the presentation comprises only a single stream, the media-level
   "a=control:" attribute may be omitted altogether. However, if the
   presentation contains more than one stream, each media stream section
   MUST contain its own "a=control" attribute.

   Example:
     v=0
     o=- 2890844256 2890842807 IN IP4 204.34.34.32
     s=I contain
     i=<more info>
     t=0 0
     c=IN IP4 0.0.0.0
     a=control:rtsp://example.com/movie/
     m=video 8002 RTP/AVP 31
     a=control:trackID=1
     m=audio 8004 RTP/AVP 3
     a=control:trackID=2

   In this example, the client is required to establish a single RTSP
   session to the server, and uses the URLs
   rtsp://example.com/movie/trackID=1 and
   rtsp://example.com/movie/trackID=2 to set up the video and audio
   streams, respectively. The URL rtsp://example.com/movie/ controls the
   whole movie.


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From <EMAIL-ADDRESS-HIDDEN> Tue Nov 19 01:00:07 2002
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>Wouldn't the aggregate specification in the RTSP RFC solve this?

Yes, probably, although that can be used only if we're setting up and 
playing *every* track in the RTSP session.

         Ross.



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From <EMAIL-ADDRESS-HIDDEN> Tue Nov 19 07:02:10 2002
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X-Original-Date: Tue, 19 Nov 2002 09:56:09 -0500
Date: Tue, 19 Nov 2002 09:56:09 -0500


I believe it is not very difficult to determine if all tracks in the session
are being played (after parsing the SDP) and if yes send a single PLAY
command. However, the problem will still exist if there are more than 2
tracks and all of them are not being played.

Thanks


-----Original Message-----
From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
Sent: Tuesday, November 19, 2002 3:55 AM
To: <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] TCP tunneling bug



>Wouldn't the aggregate specification in the RTSP RFC solve this?

Yes, probably, although that can be used only if we're setting up and 
playing *every* track in the RTSP session.

         Ross.



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From <EMAIL-ADDRESS-HIDDEN> Wed Nov 20 12:39:34 2002
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X-Original-Date: Wed, 20 Nov 2002 12:35:46 -0800
Date: Wed, 20 Nov 2002 12:35:46 -0800

>Fixing this bug would be non-trivial and messy.  How important is this 
>initial data to you? :-)

I take it from the smiley you know that it is quite important. :) Synching 
A/V is difficult if you're missing some (especially an unknown amount), and 
if it happens to be the video track that loses initial data you could be in 
BIG trouble.

I guess for now I'll toss in a hack that buffers rather than discards the 
data, and when my own Sink starts receiving, look in that buffer first.

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X-Original-Date: Tue, 26 Nov 2002 16:26:17 -0500
Date: Tue, 26 Nov 2002 16:26:17 -0500

hi all,
RTP  when sending the  mpeg file splits the audio and video streams as 
two distint /sepearate connections and sends.  How is it reconstructed 
at the client side ?
Where can I find more info. on  this ( that is adding the sepearated 
audio and video streams)  .
Also, where in the live.com code is this done ?

Your help would be greatly appreciated.
Thanks in advance
Guru


-- 
Guru Ranganathan, Masters in Computer Networking, NCSU







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From <EMAIL-ADDRESS-HIDDEN> Tue Nov 26 13:55:10 2002
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X-Original-Date: Tue, 26 Nov 2002 13:52:08 -0800
Date: Tue, 26 Nov 2002 13:52:08 -0800

At 01:26 PM 11/26/02, you wrote:
>hi all,
>RTP  when sending the  mpeg file splits the audio and video streams as two 
>distint /sepearate connections and sends.  How is it reconstructed at the 
>client side ?

By "reconstructed", do you mean "joining the audio and video streams back 
together" (e.g., into a new MPEG Program Stream)?  If so, then RTP clients 
typically don't work this way.  Instead, they keep the audio and video 
streams separate, and play the audio stream through a (real or virtual) 
audio device, and play the video stream through a (real or virtual) video 
device.

For example, in the RTSP/RTP extensions for the "MPlayer" tool (documented 
in <http://www.live.com/mplayer/>), each stream is read by a separate 
"demux" object - one for audio, the other for video.

The reason for sending audio and video in separate RTP streams is that it 
allows for a lot of flexibility.  For example, it makes it possible for a 
player to receive only video, but not audio (or vice-versa).  It would even 
be possible to have one computer receive and play audio, and a separate 
computer receive and play video.

The only way in which the audio and video streams are 'tied together' is to 
synchronize them.  This is done using RTCP "Sender Report" packets (which 
map each stream's RTP timestamp to 'wall clock' (NTP) time.  Receivers can 
then use this for synchronization.  The LIVE.COM code does this 
automatically: For subclasses of "RTPSource", the "presentationTime" 
parameter that's passed to the 'afterGettingFunc' of "getNextFrame()" (see 
"liveMedia/include/FramedSource.hh") will be an accurate, time-synchronized 
time.  (For this to work, you need to have also created a "RTCPInstance" 
for each RTP source.)

For example, if you use "openRTSP" to receive RTSP/RTP streams, then the 
contents of each RTP stream (audio and video) are written into separate 
files.  This is done using the "FileSink" class.  If you look at the 
"FileSink::afterGettingFrame()" member function, you'll notice that there's 
a "presentationTime" parameter (that happens to be unused).  If you wanted 
to, though, you could use that "presentationTime" parameter to synchronize 
audio and video.

I hope this helps.

         Ross Finlayson
         LIVE.COM




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From <EMAIL-ADDRESS-HIDDEN> Tue Nov 26 19:00:07 2002
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Subject: [Live-devel] Using RTP with my audio and video codec
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X-Original-Date: Tue, 26 Nov 2002 21:53:19 -0500
Date: Tue, 26 Nov 2002 21:53:19 -0500

Hi Ross and all,

My fellow classmate and I have done up a Visual C++ 6 program which will encode audio signals to MP3 and video signals to H.263 format and send it via 2 seperate Ports (TCP/IP) to the PDA...

Now, my final mission is to incorporate the RTP to it so that I could 

1)synchronise the audio and video (currently, the audio comes before the video and sometimes the other way round)

2)Multicast to more than 1 PDA...

so, if I want to incorporate the Live.com RTP library into my Visual C++ program..which files shld I add into my workspace in order to use just the RTP library for transmission and control of my encoded video/audio data?

Thanks in advance!

Best Regards,
Joseph




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From <EMAIL-ADDRESS-HIDDEN> Tue Nov 26 19:52:14 2002
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X-Original-Date: Tue, 26 Nov 2002 19:47:55 -0800
Date: Tue, 26 Nov 2002 19:47:55 -0800

Joseph,

First, you should be familiar with building the code (including the various 
'test programs') on Windows - see http://www.live.com/liveMedia/#config-windows

You can use the "testMP3Streamer", "testMPEGVideoStreamer" and 
"testMPEGAudioVideoStreamer" test programs as a model for your application.

For streaming MP3, you will need an instance of the "MPEGAudioRTPSink" 
class.  To feed this, you will also need a 'source' object.  If your MP3 
input source generates discrete frames, then you can easily write a source 
object that encapsulates this; see "liveMedia/DeviceSource.cpp" for an 
example of how to do this.  If, on the other hand, your MP3 input source is 
a byte stream, then you will need to chain together a 
"ByteStreamFileSource" and a "MPEGAudioStreamFramer", and use this as your 
input source.

For streaming H.263+, you will need an instance of the 
"H263plusVideoRTPSink" class.  To feed this, you will also need a 'source' 
object.  As before, if your H.263+ input source generates discrete frames, 
then you can easily write a source object that encapsulates 
this.  Otherwise, you will need to write your own 'framer' class that 
parses the input stream into discrete H.263+ frames.

Also, for audio/video synchronization to work, you'll need a "RTCPInstance" 
object for each of your two RTP sinks.

	Ross Finlayson
	LIVE.COM



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X-Original-Date: Wed, 27 Nov 2002 22:43:45 -0500
Date: Wed, 27 Nov 2002 22:43:45 -0500

 
Hi Ross,

On Tue, 26 Nov 2002 19:47:55  
 Ross Finlayson wrote:

>For streaming H.263+, you will need an instance of the 
>"H263plusVideoRTPSink" class.  To feed this, you will also need >a 'source' object.  As before, if your H.263+ input source generates >discrete frames, then you can easily write a source object that >encapsulates this.  Otherwise, you will need to write your >own 'framer' class that parses the input stream into discrete H.263+ >frames.
>

Thanks for the useful advice, however, the codec in my software is H.263 format, and not H.263+, so, can I still use the "H263plusVideoRTPSink" class? I don't think so, right? If I can't, then do you know if I could modify the "H263plusVideoRTPSink" class to suit H.263?

Thanks in advance.

Regards,
Joseph





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X-Original-Date: Wed, 27 Nov 2002 19:57:10 -0800
Date: Wed, 27 Nov 2002 19:57:10 -0800


>Thanks for the useful advice, however, the codec in my software is H.263 
>format, and not H.263+

Note that "H.263+" is merely a common abbreviation for "the 1998 version of 
ITU-T Recommendation H.263" (see RFC 2429).  You may well find that this is 
the version that you're using.

But, in any case, RFC 2429 also says:
    "Because the 1998 version of H.263
    is a superset of the 1996 syntax, this format can also be used with
    the 1996 version of H.263 [3], and is recommended for this use by new
    implementations."

>so, can I still use the "H263plusVideoRTPSink" class?

Yes, and in fact, you *should* use this.

         Ross.



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From <EMAIL-ADDRESS-HIDDEN> Wed Nov 27 20:22:48 2002
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X-Original-Date: Wed, 27 Nov 2002 23:16:52 -0500
Date: Wed, 27 Nov 2002 23:16:52 -0500

 
Alright, Thanks again...:)

One more thing is, someone was telling me that H.263 is pretty much similar to MPEG4 codec, how true is that? 

Joe



--

On Wed, 27 Nov 2002 19:57:10  
 Ross Finlayson wrote:
>
>>Thanks for the useful advice, however, the codec in my software is H.263 
>>format, and not H.263+
>
>Note that "H.263+" is merely a common abbreviation for "the 1998 version of 
>ITU-T Recommendation H.263" (see RFC 2429).  You may well find that this is 
>the version that you're using.
>
>But, in any case, RFC 2429 also says:
>    "Because the 1998 version of H.263
>    is a superset of the 1996 syntax, this format can also be used with
>    the 1996 version of H.263 [3], and is recommended for this use by new
>    implementations."
>
>>so, can I still use the "H263plusVideoRTPSink" class?
>
>Yes, and in fact, you *should* use this.
>
>         Ross.
>
>
>
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X-Original-Date: Thu, 28 Nov 2002 04:11:24 -0500
Date: Thu, 28 Nov 2002 04:11:24 -0500

Hi Ross,

I was trying out your testMPEGAudioVideoStreamer.cpp and there's something I'm confused about...

the line after you call the play() function to start streaming, i.e.

  env->taskScheduler().blockMyself(); // does not return

What is this line used for? I noted that if I remarked this line, the stream not work.

Also, It says in the remark that it does not return, so do you mean it keeps looping in the taskScheduler()? 

If I want to terminate the RTP transmission halfway, how can I do so? 
if I used 

env->taskScheduler().blockMyself(); // does not return

I can't seem to terminate the transmission unless I close the program.

Thanks,

Best Regards,
Joseph



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X-Original-Date: Thu, 28 Nov 2002 10:34:05 -0800
Date: Thu, 28 Nov 2002 10:34:05 -0800


>I was trying out your testMPEGAudioVideoStreamer.cpp and there's something 
>I'm confused about...
>
>the line after you call the play() function to start streaming, i.e.
>
>   env->taskScheduler().blockMyself(); // does not return
>
>What is this line used for? I noted that if I remarked this line, the 
>stream not work.

Joseph,

The "liveMedia" libray works using an event-driven execution model - tasks 
are executed from an event loop.  This event loop is executed by the 
"TaskScheduler::blockMyself()" function.  In other words, when the program 
calls "blockMyself()", it enters an event loop, which dispatches pending 
events (delaying if necessary).  (To see a concrete implementation of this, 
see "BasicUsageEnvironment/BasicUsageEnvironent.cpp".)

>Also, It says in the remark that it does not return, so do you mean it 
>keeps looping in the taskScheduler()?

That's right.

(On reflection,"blockMyself()" was not the best choice of name.  I probably 
should have called it "enterEventLoop()".)

>If I want to terminate the RTP transmission halfway, how can I do so?
>if I used
>
>env->taskScheduler().blockMyself(); // does not return
>
>I can't seem to terminate the transmission unless I close the program.

You're in luck.  "blockMyself()" takes an optional parameter
         char* watchVariable = NULL
(See "UsageEnvironment/include/UsageEnvironment.hh")  If "watchVariable" is 
not NULL, then the event loop will exit when "*watchVariable" is non-zero.

E.g.,
         char flag = 0;
         ...
         env->taskScheduler().blockMyself(&flag);

and then later, when you want the event loop to end:
         flag = 1;
and then you'll return from the "blockMyself()" call the next time the 
event loop executes.

         Ross.



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From <EMAIL-ADDRESS-HIDDEN> Thu Nov 28 18:44:33 2002
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X-Original-Date: Thu, 28 Nov 2002 21:41:29 -0500
Date: Thu, 28 Nov 2002 21:41:29 -0500

Hi Ross...

Thanks for the great advice again. ;)

I tried implementing your testMPEGAudioVideoStreamer.cpp on a Visual C++ environment, i.e. I created a simple project and used 2 buttons, one for starting the audio/video streaming 

following your advice, I used 

         char flag = 0;
         ...
         env->taskScheduler().blockMyself(&flag);

and as for the other button (Stop button), I tried to set flag = 1, i.e. 

void CVideoDlg::OnButtonStop() 
{
         flag = 1;
}

It was able to stream the audio and video perfectly alright, however, it seems once the 

      env->taskScheduler().blockMyself(&flag);

is executed, my program will no longer be responding... i.e. I can't press the Stop button to set the "flag = 1" to terminate the event loop... in fact after a while, there is a message saying my program is not responding anymore! But the streaming is taking place all the while...

Do you know how I can remedy this problem? i.e. how can i make the GUI still be active during the streaming so that I could press the Stop button to stop the streaming?

Thanks in advance!

Best Regards,
Joseph







--

On Thu, 28 Nov 2002 10:34:05  
 Ross Finlayson wrote:
>
>>I was trying out your testMPEGAudioVideoStreamer.cpp and there's something 
>>I'm confused about...
>>
>>the line after you call the play() function to start streaming, i.e.
>>
>>   env->taskScheduler().blockMyself(); // does not return
>>
>>What is this line used for? I noted that if I remarked this line, the 
>>stream not work.
>
>Joseph,
>
>The "liveMedia" libray works using an event-driven execution model - tasks 
>are executed from an event loop.  This event loop is executed by the 
>"TaskScheduler::blockMyself()" function.  In other words, when the program 
>calls "blockMyself()", it enters an event loop, which dispatches pending 
>events (delaying if necessary).  (To see a concrete implementation of this, 
>see "BasicUsageEnvironment/BasicUsageEnvironent.cpp".)
>
>>Also, It says in the remark that it does not return, so do you mean it 
>>keeps looping in the taskScheduler()?
>
>That's right.
>
>(On reflection,"blockMyself()" was not the best choice of name.  I probably 
>should have called it "enterEventLoop()".)
>
>>If I want to terminate the RTP transmission halfway, how can I do so?
>>if I used
>>
>>env->taskScheduler().blockMyself(); // does not return
>>
>>I can't seem to terminate the transmission unless I close the program.
>
>You're in luck.  "blockMyself()" takes an optional parameter
>         char* watchVariable = NULL
>(See "UsageEnvironment/include/UsageEnvironment.hh")  If "watchVariable" is 
>not NULL, then the event loop will exit when "*watchVariable" is non-zero.
>
>E.g.,
>         char flag = 0;
>         ...
>         env->taskScheduler().blockMyself(&flag);
>
>and then later, when you want the event loop to end:
>         flag = 1;
>and then you'll return from the "blockMyself()" call the next time the 
>event loop executes.
>
>         Ross.
>
>
>
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X-Original-Date: Thu, 28 Nov 2002 22:07:15 -0500
Date: Thu, 28 Nov 2002 22:07:15 -0500

Hi again...

just like to confirm with you...u wrote the other day:

>If, on the other hand, your MP3 input source is a byte stream, then >you will need to chain together a 
>"ByteStreamFileSource" and a "MPEGAudioStreamFramer", and use this >as your input source.
>

I was looking at the testMP3streamer.cpp, so am I right to say that the "MP3FileSource" is the one that chain together a 
"ByteStreamFileSource" and "MPEGAudioStreamFramer" for the testMP3streamer?

Joseph



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X-Original-Date: Thu, 28 Nov 2002 19:37:18 -0800
Date: Thu, 28 Nov 2002 19:37:18 -0800


>It was able to stream the audio and video perfectly alright, however, it 
>seems once the
>
>       env->taskScheduler().blockMyself(&flag);
>
>is executed, my program will no longer be responding... i.e. I can't press 
>the Stop button to set the "flag = 1" to terminate the event loop... in 
>fact after a while, there is a message saying my program is not responding 
>anymore! But the streaming is taking place all the while...

Joseph,

The problem here is that your program has two separate event loops:
         i) the event loop used by the "liveMedia" library, and
         ii) the event loop that's used for your GUI.
Once you call "blockMyself()", you enter the first of these event loops, 
but don't get to execute the second (GUI) event loop.

The solution is to combine these two event loops into one.  There are three 
possible ways to do this:
1/ From within your GUI's event loop, call 
"BasicTaskScheduler::SingleStep()".  (This is the operation that's called 
inside the "blockMyself()" event loop - see 
"BasicUsageEnvironment/BasicUsageEnvironment.cpp".)  Note that if you 
choose this solution, then you won't call "blockMyself()" at all, because 
you'll be using the GUI's event loop as the (sole) event loop.
2/ From within liveMedia's event loop (i.e., "blockMyself()", call some 
operation that performs a single GUI event loop operation.  (I.e., you 
would make a small modification to "BasicTaskScheduler::blockMyself()".)
3/ Write a new subclass of "TaskScheduler" to replace 
"BasicTaskScheduler".  This new task scheduler would implement a single 
event loop for both 'liveMedia' and your GUI.

I suggest trying solution 1/ if you can.  That would seem to be the 
simplest solution (although I don't know much about how Windows' GUI event 
loop is typically programmed...)

         Ross.



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X-Original-Date: Thu, 28 Nov 2002 19:38:54 -0800
Date: Thu, 28 Nov 2002 19:38:54 -0800

At 07:07 PM 11/28/02, you wrote:
>just like to confirm with you...u wrote the other day:
>
> >If, on the other hand, your MP3 input source is a byte stream, then >you 
> will need to chain together a
> >"ByteStreamFileSource" and a "MPEGAudioStreamFramer", and use this >as 
> your input source.
> >
>
>I was looking at the testMP3streamer.cpp, so am I right to say that the 
>"MP3FileSource" is the one that chain together a
>"ByteStreamFileSource" and "MPEGAudioStreamFramer" for the testMP3streamer?

Yes, you can just use "MP3FileSource" - it basically combines the 
functionality of "ByteStreamFileSource" and 
"MPEGAudioStreamFramer".  (Note, though, that "MP3FileSource" is old code 
that might end up getting removed at some point in the future.)

         Ross.



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X-Original-Date: Fri, 29 Nov 2002 00:34:46 -0500
Date: Fri, 29 Nov 2002 00:34:46 -0500

Hi Ross and the rest of the Live-developers,

Thanks... I tried creating a "ByteStreamFileSource" class for reading from my MP3 file and then chaining it to the MPEGAudioStreamFramer, and it works fine.

However, I just got a slight problem...when I used WinAmp to play the MP3 stream on some other PC...It played for a couple of seconds, and then starts to buffer...then there's no sound...then it continues playing again...

I think it's got something to do with the winamp buffering...has anyone managed to get to get continous play? Any setting that I need to take a look at?

Also, I can't seem to get the RealOnePlayer (the latest I downloaded) to run the the stream that I sent using testMPEGAudioVideoStreamer.cpp... 

I just open the testMPEGAudioVideo.sdp in RealOne Player..then it starts buffering...but when it reached 100%, nothing happens...

Are there any setting that I should set? 

Thanks in advance...

Regards,
Joseph

P.S: This RTP library is really cool..;)





--

On Thu, 28 Nov 2002 19:38:54  
 Ross Finlayson wrote:
>At 07:07 PM 11/28/02, you wrote:
>>just like to confirm with you...u wrote the other day:
>>
>> >If, on the other hand, your MP3 input source is a byte stream, then >you 
>> will need to chain together a
>> >"ByteStreamFileSource" and a "MPEGAudioStreamFramer", and use this >as 
>> your input source.
>> >
>>
>>I was looking at the testMP3streamer.cpp, so am I right to say that the 
>>"MP3FileSource" is the one that chain together a
>>"ByteStreamFileSource" and "MPEGAudioStreamFramer" for the testMP3streamer?
>
>Yes, you can just use "MP3FileSource" - it basically combines the 
>functionality of "ByteStreamFileSource" and 
>"MPEGAudioStreamFramer".  (Note, though, that "MP3FileSource" is old code 
>that might end up getting removed at some point in the future.)
>
>         Ross.
>
>
>
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X-Original-Date: Fri, 29 Nov 2002 01:11:20 -0500
Date: Fri, 29 Nov 2002 01:11:20 -0500

Hi again...

I tried the 1st method you suggested... so instead of calling

      env->taskScheduler().blockMyself();

I tried calling 

      BasicTaskScheduler::SingleStep(); 

directly... but, I got an error saying 
 
error C2352: 'BasicTaskScheduler::SingleStep' : illegal call of non-static member function

I thought SingleStep is a public function (I looked at BasicUsageEnvironment.hh)

Do you know what changes I should make in order to solve this error?

Thanks ...

Joseph



--

On Thu, 28 Nov 2002 19:37:18  
 Ross Finlayson wrote:
>
>>It was able to stream the audio and video perfectly alright, however, it 
>>seems once the
>>
>>       env->taskScheduler().blockMyself(&flag);
>>
>>is executed, my program will no longer be responding... i.e. I can't press 
>>the Stop button to set the "flag = 1" to terminate the event loop... in 
>>fact after a while, there is a message saying my program is not responding 
>>anymore! But the streaming is taking place all the while...
>
>Joseph,
>
>The problem here is that your program has two separate event loops:
>         i) the event loop used by the "liveMedia" library, and
>         ii) the event loop that's used for your GUI.
>Once you call "blockMyself()", you enter the first of these event loops, 
>but don't get to execute the second (GUI) event loop.
>
>The solution is to combine these two event loops into one.  There are three 
>possible ways to do this:
>1/ From within your GUI's event loop, call 
>"BasicTaskScheduler::SingleStep()".  (This is the operation that's called 
>inside the "blockMyself()" event loop - see 
>"BasicUsageEnvironment/BasicUsageEnvironment.cpp".)  Note that if you 
>choose this solution, then you won't call "blockMyself()" at all, because 
>you'll be using the GUI's event loop as the (sole) event loop.
>2/ From within liveMedia's event loop (i.e., "blockMyself()", call some 
>operation that performs a single GUI event loop operation.  (I.e., you 
>would make a small modification to "BasicTaskScheduler::blockMyself()".)
>3/ Write a new subclass of "TaskScheduler" to replace 
>"BasicTaskScheduler".  This new task scheduler would implement a single 
>event loop for both 'liveMedia' and your GUI.
>
>I suggest trying solution 1/ if you can.  That would seem to be the 
>simplest solution (although I don't know much about how Windows' GUI event 
>loop is typically programmed...)
>
>         Ross.
>
>
>
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X-Original-Date: Fri, 29 Nov 2002 01:39:54 -0800
Date: Fri, 29 Nov 2002 01:39:54 -0800

At 10:11 PM 11/28/02, you wrote:
>Hi again...
>
>I tried the 1st method you suggested... so instead of calling
>
>       env->taskScheduler().blockMyself();
>
>I tried calling
>
>       BasicTaskScheduler::SingleStep();
>
>directly... but, I got an error saying
>
>error C2352: 'BasicTaskScheduler::SingleStep' : illegal call of non-static 
>member function


Sorry, when I said "call BasicTaskScheduler::SingleStep()", I didn't mean 
this literally.

Because "SingleStep()" is a (non-static) member function, you need to do
         env->taskScheduler().SingleStep();

     Ross.




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X-Original-Date: Fri, 29 Nov 2002 01:45:23 -0800
Date: Fri, 29 Nov 2002 01:45:23 -0800

At 09:34 PM 11/28/02, you wrote:
>Hi Ross and the rest of the Live-developers,
>
>Thanks... I tried creating a "ByteStreamFileSource" class for reading from 
>my MP3 file and then chaining it to the MPEGAudioStreamFramer, and it 
>works fine.
>
>However, I just got a slight problem...when I used WinAmp to play the MP3 
>stream on some other PC...It played for a couple of seconds, and then 
>starts to buffer...then there's no sound...then it continues playing again...

There is a bug in Winamp that causes this to happen (with the "in_rtp.dll" 
plugin) if you click the "stop" button, and then the "play" button again.

Unfortunately the only way I know to get around it is to quit Winamp, and 
restart it.


>Also, I can't seem to get the RealOnePlayer (the latest I downloaded) to 
>run the the stream that I sent using testMPEGAudioVideoStreamer.cpp...
>
>I just open the testMPEGAudioVideo.sdp in RealOne Player..then it starts 
>buffering...but when it reached 100%, nothing happens...

This *should* work.

One thing you can try instead is have it read "testMPEGVideo.sdp", to play 
just the video stream, or "testMP3.sdp", to play just the audio 
stream.  But "testMPEGAudioVideo.sdp" *should* work.

         Ross.



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X-Original-Date: Sat, 30 Nov 2002 02:26:52 -0800
Date: Sat, 30 Nov 2002 02:26:52 -0800

As noted in a comment earlier this week, I have now renamed the 
"TaskScheduler::blockMyself()" member function as "doEventLoop()", to more 
accurately describe what the function does.

If you call this routine from your programs (which you probably do, if you 
use the "liveMedia" library), then don't forget to make this change when 
you upgrade to the new version of the library.

	Ross.



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Subject: [Live-devel] MPEG  non interpolated ?
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X-Original-Date: Sun, 01 Dec 2002 01:59:32 -0500
Date: Sun, 01 Dec 2002 01:59:32 -0500

HI all,
 I am not very much familiar with the inside details of MPEG standard 
and so....
I have one basic question, CAN  we find a non interpolated MPEG video 
stream anywhere ?
or else is there anyway we can create one ?  
thanks for your input on this .
Guru




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Subject: Re: [Live-devel] MPEG  non interpolated ?
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X-Original-Date: Sat, 30 Nov 2002 23:13:02 -0800
Date: Sat, 30 Nov 2002 23:13:02 -0800

At 10:59 PM 11/30/02, you wrote:
>HI all,
>I am not very much familiar with the inside details of MPEG standard and 
>so....
>I have one basic question, CAN  we find a non interpolated MPEG video 
>stream anywhere ?

I'm not sure exactly what you mean by "non-interpolated MPEG video 
stream".  Do you mean a file that consists just of MPEG video frames, 
rather than a MPEG file that contains both audio and video?

If so, then you can create such a file by running the "testMPEGSplitter" 
program (which is included in "testProgs").  This program takes as input a 
(audio+video) MPEG file called "in.mpg", and outputs two files: 
"out_audio.mpg" and "out_video.mpg".  The latter file is a MPEG video-only 
(i.e., Elementary Stream) file.

         Ross.



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X-Original-Date: Sun, 01 Dec 2002 02:59:03 -0500
Date: Sun, 01 Dec 2002 02:59:03 -0500

Just to clarify :
i.e   I need a MPEG file with a no interpolated B frames.  
btw.. is there anything such as this ?
Thanks




Ross Finlayson wrote:

> At 10:59 PM 11/30/02, you wrote:
>
>> HI all,
>> I am not very much familiar with the inside details of MPEG standard 
>> and so....
>> I have one basic question, CAN  we find a non interpolated MPEG video 
>> stream anywhere ?
>
>
> I'm not sure exactly what you mean by "non-interpolated MPEG video 
> stream".  Do you mean a file that consists just of MPEG video frames, 
> rather than a MPEG file that contains both audio and video?
>
> If so, then you can create such a file by running the 
> "testMPEGSplitter" program (which is included in "testProgs").  This 
> program takes as input a (audio+video) MPEG file called "in.mpg", and 
> outputs two files: "out_audio.mpg" and "out_video.mpg".  The latter 
> file is a MPEG video-only (i.e., Elementary Stream) file.
>
>         Ross.
>
>
>
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X-Original-Date: Sun, 01 Dec 2002 21:28:10 -0500
Date: Sun, 01 Dec 2002 21:28:10 -0500

Hi Ross,

Hope you had a nice weekend...

I tried what you have suggested, i.e. using 

          env->taskScheduler().SingleStep();

instead, however, I got an error:

error C2039: 'SingleStep' : is not a member of 'TaskScheduler'

I found another way of getting it done..but like to check with you if this is okay...what I did was I changed

   // Begin by setting up our usage environment:
   TaskScheduler* scheduler = BasicTaskScheduler::createNew();

to

   // Begin by setting up our usage environment:
   BasicTaskScheduler* scheduler = BasicTaskScheduler::createNew();  

i.e. from TaskScheduler* to BasicTaskScheduler*, then instead of

   env->taskScheduler().SingleStep();

I used 

   scheduler->SingleStep();

The streaming works fine...but is this okay, Ross?

Also, I'm not sure myself about the event loop of windows GUI..what I did was to create a thread that runs the a "testMPEGAudioVideoStreamer" program when the "start streaming" button is pressed...then if I want to end the streaming, I just pressed the "stop" button to terminate the thread.

This seems to work fine. So, if I do it this way..do you think it's better if i called directly the SingleStep() as in 
         scheduler->SingleStep();

or stick to the usual

   env->taskScheduler().blockMyself();

?

Thanks...

Best Regards,
Joseph

--

On Fri, 29 Nov 2002 01:39:54  
 Ross Finlayson wrote:
>At 10:11 PM 11/28/02, you wrote:
>>Hi again...
>>
>>I tried the 1st method you suggested... so instead of calling
>>
>>       env->taskScheduler().blockMyself();
>>
>>I tried calling
>>
>>       BasicTaskScheduler::SingleStep();
>>
>>directly... but, I got an error saying
>>
>>error C2352: 'BasicTaskScheduler::SingleStep' : illegal call of non-static 
>>member function
>
>
>Sorry, when I said "call BasicTaskScheduler::SingleStep()", I didn't mean 
>this literally.
>
>Because "SingleStep()" is a (non-static) member function, you need to do
>         env->taskScheduler().SingleStep();
>
>     Ross.
>
>
>
>
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X-Original-Date: Mon, 02 Dec 2002 01:02:12 -0500
Date: Mon, 02 Dec 2002 01:02:12 -0500

Hi there...

Just like to ask... currently, my program records the wave/mic/line-in that's connected to the sound card, does a MP3 conversion and stores the data into a char buffer 

char	buffer[AUDIO_TXBUFFER_SIZE];

Note that if I write this buffer to a file, e.g. test.mp3, then I could play this test.mp3 using winamp etc...

So, I'm now trying to use the "ByteStreamFileSource" class to read from this buffer instead of reading from a file, how can I go about doing it?

Do you have any recommendations? I've thought of one way, and that is to create a new "ByteStreamBufferSource" class that's similar in a way to the "ByteStreamFileSource" class, do you have any pointers as to what I should change in the "ByteStreamFileSource" class?

Thanks!

Joseph


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X-Original-Date: Mon, 2 Dec 2002 07:52:36 -0500
Date: Mon, 2 Dec 2002 07:52:36 -0500

I haven't followed this thread too closely, so I may be way off;

but most MP3 encoders output 1 frame at a time.  If that's the case, 
you should keep this framing intact and be able to go straight to a 
class that extends FramedSource.  So you'd write something like 
MyFramedSource which extends FramedSource, implementing the method 
doGetNextFrame() to move frames from your encoder to the streaming 
pipeline.  You could still buffer frames if you wanted, but try to keep 
the framing intact, otherwise you have to scan the stream for frames 
again, wasting time.

Marty Schoch
<mschoch(at)multicasttech.com>

On Monday, December 2, 2002, at 01:02 AM, Joseph Teo wrote:

> Hi there...
>
> Just like to ask... currently, my program records the wave/mic/line-in 
> that's connected to the sound card, does a MP3 conversion and stores 
> the data into a char buffer
>
> char	buffer[AUDIO_TXBUFFER_SIZE];
>
> Note that if I write this buffer to a file, e.g. test.mp3, then I 
> could play this test.mp3 using winamp etc...
>
> So, I'm now trying to use the "ByteStreamFileSource" class to read 
> from this buffer instead of reading from a file, how can I go about 
> doing it?
>
> Do you have any recommendations? I've thought of one way, and that is 
> to create a new "ByteStreamBufferSource" class that's similar in a way 
> to the "ByteStreamFileSource" class, do you have any pointers as to 
> what I should change in the "ByteStreamFileSource" class?
>
> Thanks!
>
> Joseph
>
>
> _____________________________________________________________
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X-Original-Date: Mon, 02 Dec 2002 13:10:52 -0800
Date: Mon, 02 Dec 2002 13:10:52 -0800

At 04:52 AM 12/2/02, you wrote:
>I haven't followed this thread too closely, so I may be way off;
>
>but most MP3 encoders output 1 frame at a time.  If that's the case, you 
>should keep this framing intact and be able to go straight to a class that 
>extends FramedSource.

Yes, Marty is correct.  The best solution would be to encapsulate your MP3 
encoder in a "FramedSource" (subclass) that delivers discrete frames.

For a model of how to do this, see "liveMedia/DeviceSource.cpp" (and 
"liveMedia/include/DeviceSource.hh").  You will need to fill in parts of 
this code to do the actual read from your encoder.

If you do this, then the rest of your code becomesvery simple: You can just 
feed this FramedSource (subclass) directly into "MPEGAudioRTPSink".  You 
will no longer need a "MPEGAudioStreamFramer".

However, if you *really* don't want to write any 'liveMedia' code, then 
another possible solution would be to implement your program as two 
processes, joined by a pipe.  The first process would write the encoded MP3 
frames to 'stdout'; the second would use a ByteStreamFileSource (reading 
from 'stdin').  However, this solution is less efficient, both because 
you're using a pipe, and because you will need to parse (i.e., frame) the 
MP3 frames all over again (using a "MPEGAudioStreamFramer").

         Ross.



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X-Original-Date: Mon, 02 Dec 2002 13:37:28 -0800
Date: Mon, 02 Dec 2002 13:37:28 -0800


>I tried what you have suggested, i.e. using
>
>           env->taskScheduler().SingleStep();
>
>instead, however, I got an error:
>
>error C2039: 'SingleStep' : is not a member of 'TaskScheduler'

Yes, SingleStep() is actually a member function in the subclass: BasicTask 
Scheduler.  Sorry about that.


>Also, I'm not sure myself about the event loop of windows GUI..what I did 
>was to create a thread that runs the a "testMPEGAudioVideoStreamer" 
>program when the "start streaming" button is pressed...then if I want to 
>end the streaming, I just pressed the "stop" button to terminate the thread.
>
>This seems to work fine. So, if I do it this way..do you think it's better 
>if i called directly the SingleStep() as in
>          scheduler->SingleStep();
>
>or stick to the usual
>
>    env->taskScheduler().blockMyself();

"SingleStep()" *must* be called from within a loop.  So, if you don't call 
it from within your GUI's event loop, then you'll need to call 
"blockMyself()" instead.

         Ross.



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X-Original-Date: Mon, 02 Dec 2002 20:43:30 -0500
Date: Mon, 02 Dec 2002 20:43:30 -0500

Hi Ross,

You're right, if I call the SingleStep() directly, I had to use a while(1){ } loop, i.e. keep calling it repeatedly...

currently, I'm sticking to your original implementation of just calling

env->taskScheduler().blockMyself();

but there's one thing...

for the setting up the usage environment:

what's the main difference if I used *BasicTaskScheduler* :

BasicTaskScheduler* scheduler = BasicTaskScheduler::createNew();  

as compared to *TaskScheduler*:

TaskScheduler* scheduler = BasicTaskScheduler::createNew();

?

Joe




--

On Mon, 02 Dec 2002 13:37:28  
 Ross Finlayson wrote:
>
>>I tried what you have suggested, i.e. using
>>
>>           env->taskScheduler().SingleStep();
>>
>>instead, however, I got an error:
>>
>>error C2039: 'SingleStep' : is not a member of 'TaskScheduler'
>
>Yes, SingleStep() is actually a member function in the subclass: BasicTask 
>Scheduler.  Sorry about that.
>
>
>>Also, I'm not sure myself about the event loop of windows GUI..what I did 
>>was to create a thread that runs the a "testMPEGAudioVideoStreamer" 
>>program when the "start streaming" button is pressed...then if I want to 
>>end the streaming, I just pressed the "stop" button to terminate the thread.
>>
>>This seems to work fine. So, if I do it this way..do you think it's better 
>>if i called directly the SingleStep() as in
>>          scheduler->SingleStep();
>>
>>or stick to the usual
>>
>>    env->taskScheduler().blockMyself();
>
>"SingleStep()" *must* be called from within a loop.  So, if you don't call 
>it from within your GUI's event loop, then you'll need to call 
>"blockMyself()" instead.
>
>         Ross.
>
>
>
>-------------------------------------------------------
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X-Original-Date: Mon, 02 Dec 2002 18:06:54 -0800
Date: Mon, 02 Dec 2002 18:06:54 -0800


>for the setting up the usage environment:
>
>what's the main difference if I used *BasicTaskScheduler* :
>
>BasicTaskScheduler* scheduler = BasicTaskScheduler::createNew();
>
>as compared to *TaskScheduler*:
>
>TaskScheduler* scheduler = BasicTaskScheduler::createNew();

Basic C++ programming questions are generally off-topic for this mailing list.

But anyway, "BasicTaskScheduler::createNew()" returns a pointer to an 
object of type "BasicTaskScheduler" - which means that you could assign it 
to a "BasicTaskScheduler*", or to a 
"UsageEnvironment*".  ("UsageEnvironment" is the parent class of 
"BasicUsageEnvironment".)  By assigning it to a "BasicTaskScheduler*", you 
can use this variable to refer to member functions - such as singleStep() - 
that are defined only for "BasicUsageEnvironment".

         Ross.



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X-Original-Date: Mon, 02 Dec 2002 21:16:02 -0500
Date: Mon, 02 Dec 2002 21:16:02 -0500

Hi again,

Thanks for the tips...it was really very helpful ;)

Yeap, what I did was created another class called "ByteStreamBufferSource" which extends FramedSource... I looked at DeviceSource.cpp and worked it out from there.

It's rather similar to ByteStreamFileSource but this time, I'm not reading from a file...instead, I'm reading from a buffer

 Then inside my ByteStreamBufferSource::doGetNextFrame(), 

I call my *homemade* function 

ringbuf_read(&myAudioRingBuf1,gAudioCriticalSection,fTo,fMaxSize);

where 

myAudioRingBuf1 = buffer containing the encoded MP3 data
fTo = buffer I am suppose to write to
fMaxSize = the amount of bytes to be read

This function will look at the fMaxSize parameter passed in..and if the buffer less than fMaxSize, it will return -1, else if the read was successful, it returns 0.

there is just one problem... my ringbuf_read() function does not return the number of bytes read...i.e. the fFrameSize will not be showing the frame size...will this cause a PROBLEM? 

Could I just fix the frame size to a fixed MP3 Frame size and just read one frame at a time? If so, can anyone tell me what is the FrameSize for 1 MP3 frame? i.e. what is the fMaxSize I should set to read 1 Frame at a time?

Once again, Thanks in advance! ;) 

Regards,
Joseph

--

On Mon, 02 Dec 2002 13:10:52  
 Ross Finlayson wrote:
>At 04:52 AM 12/2/02, you wrote:
>>I haven't followed this thread too closely, so I may be way off;
>>
>>but most MP3 encoders output 1 frame at a time.  If that's the case, you 
>>should keep this framing intact and be able to go straight to a class that 
>>extends FramedSource.
>
>Yes, Marty is correct.  The best solution would be to encapsulate your MP3 
>encoder in a "FramedSource" (subclass) that delivers discrete frames.
>
>For a model of how to do this, see "liveMedia/DeviceSource.cpp" (and 
>"liveMedia/include/DeviceSource.hh").  You will need to fill in parts of 
>this code to do the actual read from your encoder.
>
>If you do this, then the rest of your code becomesvery simple: You can just 
>feed this FramedSource (subclass) directly into "MPEGAudioRTPSink".  You 
>will no longer need a "MPEGAudioStreamFramer".
>
>However, if you *really* don't want to write any 'liveMedia' code, then 
>another possible solution would be to implement your program as two 
>processes, joined by a pipe.  The first process would write the encoded MP3 
>frames to 'stdout'; the second would use a ByteStreamFileSource (reading 
>from 'stdin').  However, this solution is less efficient, both because 
>you're using a pipe, and because you will need to parse (i.e., frame) the 
>MP3 frames all over again (using a "MPEGAudioStreamFramer").
>
>         Ross.
>
>
>
>-------------------------------------------------------
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X-Original-Date: Mon, 02 Dec 2002 18:28:47 -0800
Date: Mon, 02 Dec 2002 18:28:47 -0800


>there is just one problem... my ringbuf_read() function does not return 
>the number of bytes read...

Why not?  Does your MP3 encoder not tell you the size of each frame that it 
produces?

>i.e. the fFrameSize will not be showing the frame size...will this cause a 
>PROBLEM?

Yes.  If you want to read arbitrary amounts of data (rather than a frame at 
a time) from your encoder, then you'll need to pass it through a 
"MPEGAudioStreamFramer" before it goed to the "MPEGAudioRTPSink".

         Ross.



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X-Original-Date: Mon, 02 Dec 2002 21:50:00 -0500
Date: Mon, 02 Dec 2002 21:50:00 -0500

Hi Ross,

yes... if I'm not wrong, my MP3 encoder buffer is 2048 bytes... by the way...do you know if the MP3 frame size is always fixed?

What my MP3 encoder does is that it encodes the data, and then it will throw it into a *AudioBuffer*...and this ringbuf_read() will be used to read from that *AudioBuffer*...

in my program...after calling this ByteStreamBufferSource object to read from the *AudioBuffer*, I will pass the data to a MPEGAudioStreamFramer to frame it up...

so when I read from this AudioBuffer...I do not know what is my frame size... 

So if I pass the data through a "MPEGAudioStreamFramer" before it goes to the "MPEGAudioRTPSink"., do I still have to give my fFrameSize
at the ByteStreamBufferSource class?

Thanks...

Joe




--

On Mon, 02 Dec 2002 18:28:47  
 Ross Finlayson wrote:
>
>>there is just one problem... my ringbuf_read() function does not return 
>>the number of bytes read...
>
>Why not?  Does your MP3 encoder not tell you the size of each frame that it 
>produces?
>
>>i.e. the fFrameSize will not be showing the frame size...will this cause a 
>>PROBLEM?
>
>Yes.  If you want to read arbitrary amounts of data (rather than a frame at 
>a time) from your encoder, then you'll need to pass it through a 
>"MPEGAudioStreamFramer" before it goed to the "MPEGAudioRTPSink".
>
>         Ross.
>
>
>
>-------------------------------------------------------
>This SF.net email is sponsored by: Get the new Palm Tungsten T 
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X-Original-Date: Mon, 02 Dec 2002 22:00:46 -0500
Date: Mon, 02 Dec 2002 22:00:46 -0500

Hi again...

sorry, I made a mistake..my MP3 encoder buffer is 2048 bytes..but it seems that my MP3 encoder outputs 400+ bytes (frame size not fixed) for each frame...

so if I want to do buffering, i.e. keep writing to an AudioBuffer and call the *MPEGAudioStreamFramer*, do I still need to indicate the frame size in fFrameSize? if so, I guess I have to change my ringbuf_read()...right?

Thanks!

--

On Mon, 02 Dec 2002 21:50:00  
 Joseph Teo wrote:
>Hi Ross,
>
>yes... if I'm not wrong, my MP3 encoder buffer is 2048 bytes... by the way...do you know if the MP3 frame size is always fixed?
>
>What my MP3 encoder does is that it encodes the data, and then it will throw it into a *AudioBuffer*...and this ringbuf_read() will be used to read from that *AudioBuffer*...
>
>in my program...after calling this ByteStreamBufferSource object to read from the *AudioBuffer*, I will pass the data to a MPEGAudioStreamFramer to frame it up...
>
>so when I read from this AudioBuffer...I do not know what is my frame size... 
>
>So if I pass the data through a "MPEGAudioStreamFramer" before it goes to the "MPEGAudioRTPSink"., do I still have to give my fFrameSize
>at the ByteStreamBufferSource class?
>
>Thanks...
>
>Joe
>
>
>
>
>--
>
>On Mon, 02 Dec 2002 18:28:47  
> Ross Finlayson wrote:
>>
>>>there is just one problem... my ringbuf_read() function does not return 
>>>the number of bytes read...
>>
>>Why not?  Does your MP3 encoder not tell you the size of each frame that it 
>>produces?
>>
>>>i.e. the fFrameSize will not be showing the frame size...will this cause a 
>>>PROBLEM?
>>
>>Yes.  If you want to read arbitrary amounts of data (rather than a frame at 
>>a time) from your encoder, then you'll need to pass it through a 
>>"MPEGAudioStreamFramer" before it goed to the "MPEGAudioRTPSink".
>>
>>         Ross.
>>
>>
>>
>>-------------------------------------------------------
>>This SF.net email is sponsored by: Get the new Palm Tungsten T 
>>handheld. Power & Color in a compact size! 
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>
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X-Original-Date: Mon, 02 Dec 2002 19:02:11 -0800
Date: Mon, 02 Dec 2002 19:02:11 -0800


>yes... if I'm not wrong, my MP3 encoder buffer is 2048 bytes... by the 
>way...do you know if the MP3 frame size is always fixed?

No, it depends on the stream's bitrate and sampling frequency.  For VBR 
(variable-bit rate) streams, it can even vary within a single frame.


>in my program...after calling this ByteStreamBufferSource object to read 
>from the *AudioBuffer*, I will pass the data to a MPEGAudioStreamFramer to 
>frame it up...

Yes, that should work.

>So if I pass the data through a "MPEGAudioStreamFramer" before it goes to 
>the "MPEGAudioRTPSink"., do I still have to give my fFrameSize
>at the ByteStreamBufferSource class?

Yes.  As for all subclasses of "FramedSource", you need to set "fFrameSize" 
when delivering data to the next stage in the pipeline (the 
MPEGAudioStreamFramer).  In this case, "fFrameSize" will be the actual 
amount of data that you're delivering (even though it doesn't make up a 
complete audio 'frame').  See "ByteStreamFileSource" for an example of this.

         Ross.




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X-Original-Date: Mon, 02 Dec 2002 23:07:28 -0500
Date: Mon, 02 Dec 2002 23:07:28 -0500

Hi Ross,

Success!!! ;) 

I did some modifications to to my ringbuf_read to return the number of bytes read...and now it's working fine... it now return the number of bytes it read from the AudioBuffer...

Thanks for all the support once again!

However, when I tried my program with a microphone...It takes quite some time (ard 20secs) before my voice traveled to the WinAmp that's receiving the RTP... could it be coz of my Audiobuffer size?

Do you or anyone out there have any recommendation on wat size would be ideal for my audio buffer?

Thanks again...

Regards,
Joseph


--

On Mon, 02 Dec 2002 19:02:11  
 Ross Finlayson wrote:
>
>>yes... if I'm not wrong, my MP3 encoder buffer is 2048 bytes... by the 
>>way...do you know if the MP3 frame size is always fixed?
>
>No, it depends on the stream's bitrate and sampling frequency.  For VBR 
>(variable-bit rate) streams, it can even vary within a single frame.
>
>
>>in my program...after calling this ByteStreamBufferSource object to read 
>>from the *AudioBuffer*, I will pass the data to a MPEGAudioStreamFramer to 
>>frame it up...
>
>Yes, that should work.
>
>>So if I pass the data through a "MPEGAudioStreamFramer" before it goes to 
>>the "MPEGAudioRTPSink"., do I still have to give my fFrameSize
>>at the ByteStreamBufferSource class?
>
>Yes.  As for all subclasses of "FramedSource", you need to set "fFrameSize" 
>when delivering data to the next stage in the pipeline (the 
>MPEGAudioStreamFramer).  In this case, "fFrameSize" will be the actual 
>amount of data that you're delivering (even though it doesn't make up a 
>complete audio 'frame').  See "ByteStreamFileSource" for an example of this.
>
>         Ross.
>
>
>
>
>-------------------------------------------------------
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Subject: [Live-devel] how to writing a H.263 Framer class?
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X-Original-Date: Tue, 03 Dec 2002 20:45:32 -0500
Date: Tue, 03 Dec 2002 20:45:32 -0500

Hi all,

Now that my audio part is kinda working, I am now focusing on the video part, which I mentioned earlier that it's a H.263 encoder.

The implementation for my video is pretty much similar to the audio portion I mentioned earlier...i.e. I have an encoder that encodes my video into H.263 format and place this encoded data into a buffer...

Then I would call a "ByteStreamBufferSource", similar to the one I created for the audio part to fetch the encoded data from this buffer...

As mentioned earlier, I have to write a H.263+ Stream Framer to frame up my data before calling the "H263plusVideoRTPSink" class...

Has anyone written this H.263+ framer class before? or can anyone give me some advice or tips as to how it should be done?

I'm currently reading RFC 2429 (H.263+)... and also looking at the "MPEGVideoStreamFramer" class... 

Thanks!

Best Regards,
Joseph


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From <EMAIL-ADDRESS-HIDDEN> Tue Dec 3 19:25:33 2002
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X-Original-Date: Tue, 03 Dec 2002 22:20:20 -0500
Date: Tue, 03 Dec 2002 22:20:20 -0500

hi there...

I see a rtpPayloadFormat in the "H263plusVideoRTPSink" class...

What values should I set for this rtpPayloadFormat? can I set it to 96?

Also, does anyone know if there are any H.263 RTP decoders out there for me to playback my transmitted H.263 video?

Thanks!

Joseph


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Thread-Topic: Problems while streaming MPEG1/2 file
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From: {Prakash Shrikrishna Bharankar} <EMAIL-ADDRESS-HIDDEN>
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Subject: [Live-devel] Problems while streaming MPEG1/2 file
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X-Original-Date: Wed, 4 Dec 2002 15:03:50 +0530
Date: Wed, 4 Dec 2002 15:03:50 +0530

This is a multi-part message in MIME format.

------=_NextPartTM-000-7a760cb2-fe0c-11d6-ba7f-006067005148
Content-Type: multipart/alternative;
	boundary="----_=_NextPart_001_01C29B78.40B59044"

------_=_NextPart_001_01C29B78.40B59044
Content-Type: text/plain;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

Hello Ross,
=20
    I am facing problems while streaming MPEG-1 video-only file from =
Livemedia server to RealOne Player client.
    I am using testMPEGVideoStreamer at server end for this purpose.
    On issuing rtsp://...... at client side after some time I receive =
the following message
   =20
    Message from RealOne Player=20

    "This server is not using a recognized protocol"

    There is no problem on playing this MPEG file locally using MPlayer.

    What could be the reason?

Thanks and Regards
Prakash
   =20

Encl: RTSP signalling messages between client and server.

RTSP Packets

OPTIONS rtsp://10.111.10.65:554 RTSP/1.0\r\n
CSeq: 1\r\n
User-Agent: RealMedia Player Version 6.0.9.1753 (win32)\r\n
Pragma: initiate-session\r\n
\r\n

RTSP/1.0 200 OK\r\n
CSeq: 1\r\n
Public: OPTIONS, DESCRIBE, SETUP, TEARDOWN, PLAY\r\n
\r\n

DESCRIBE rtsp://10.111.10.65:554/test.mpg RTSP/1.0\r\n
CSeq: 2\r\n
Accept: application/sdp\r\n
Bandwidth: 10485800\r\n
Language: en\r\n
Require: com.real.retain-entity-for-setup\r\n
SupportsMaximumASMBandwidth: 1\r\n
\r\n

RTSP/1.0 200 OK\r\n
CSeq: 2\r\n
Content-Type: application/sdp\r\n
Content-Length: 250\r\n
\r\n

SETUP rtsp://10.111.10.65:554/test.mpg/track1 RTSP/1.0\r\n
CSeq: 3\r\n
Transport: =
rtp/avp;unicast;client_port=3D8888-8889;mode=3Dplay,rtp/avp/tcp;unicast;m=
ode=3Dplay\r\n
\r\n

RTSP/1.0 200 OK\r\n
CSeq: 3\r\n
Transport: =
RTP/AVP;unicast;client_port=3D8888-8889;server_port=3D8888-8889\r\n
\r\n

PLAY rtsp://10.111.10.65:554/test.mpg RTSP/1.0\r\n
CSeq: 4\r\n
Session: 1\r\n
Range: npt=3D0-\r\n
\r\n


RTSP/1.0 200 OK\r\n
CSeq: 4\r\n
Session: 1\r\n
\r\n


------_=_NextPart_001_01C29B78.40B59044
Content-Type: text/html;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">
<TITLE>Message</TITLE>

<META content=3D"MSHTML 5.00.3315.2870" name=3DGENERATOR></HEAD>
<BODY>
<DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D375032808-04122002>Hello=20
Ross,</SPAN></FONT></DIV>
<DIV><SPAN class=3D375032808-04122002></SPAN><FONT face=3DArial=20
size=3D2>&nbsp;</FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN =
class=3D375032808-04122002>&nbsp;&nbsp;&nbsp; I=20
am facing problems while streaming MPEG-1 video-only file from Livemedia =
server=20
to RealOne Player client.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN =
class=3D375032808-04122002>&nbsp;&nbsp;&nbsp; I=20
am using testMPEGVideoStreamer at server end for this=20
purpose.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D375032808-04122002>&nbsp;&nbsp;&nbsp;&nbsp;On&nbsp;issuing&nbsp;r=
tsp://......=20
at client side&nbsp;after some time I receive the following=20
message</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D375032808-04122002>&nbsp;&nbsp;&nbsp;&nbsp;</SPAN></FONT></DIV>
<DIV><SPAN class=3D375032808-04122002><FONT face=3DArial=20
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;<B style=3D"mso-bidi-font-weight: =
normal">Message=20
from RealOne Player<?xml:namespace prefix =3D o ns =3D=20
"urn:schemas-microsoft-com:office:office" /><o:p></o:p></B> </FONT>
<P class=3DMsoNormal><FONT size=3D2><FONT face=3DArial><SPAN=20
class=3D375032808-04122002>&nbsp;&nbsp;&nbsp;&nbsp;</SPAN><SPAN=20
class=3D375032808-04122002>"</SPAN>This server is not using a recognized =

protocol<SPAN class=3D375032808-04122002>"</SPAN></FONT></FONT></P><FONT =

size=3D2><FONT face=3DArial>
<P class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;</SPAN><SPAN=20
class=3D375032808-04122002>There is no problem&nbsp;on playing this MPEG =
file=20
locally using MPlayer.</SPAN></FONT></FONT></P><FONT face=3DArial><FONT =
size=3D2>
<P class=3DMsoNormal><SPAN class=3D375032808-04122002>&nbsp;&nbsp;&nbsp; =
<SPAN=20
class=3D375032808-04122002>What coul</SPAN><SPAN =
class=3D375032808-04122002>d be the=20
reason?</SPAN></SPAN></FONT></FONT></P></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D375032808-04122002>Thanks =
and=20
Regards</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D375032808-04122002>Prakash</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN =
class=3D375032808-04122002>&nbsp;&nbsp;&nbsp;=20
</SPAN></FONT></DIV>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN =
class=3D375032808-04122002>Encl:=20
RTSP&nbsp;signalling messages between client and =
server.</SPAN></FONT></P>
<P class=3DMsoNormal><FONT size=3D2><FONT face=3DArial><STRONG>RTSP=20
Packets</STRONG></FONT></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><o:p><SPAN=20
class=3D889351909-04122002>OPTIONS rtsp://10.111.10.65:554 =
RTSP/1.0\r\n<BR>CSeq:=20
1\r\n<BR>User-Agent: RealMedia Player Version 6.0.9.1753 =
(win32)\r\n<BR>Pragma:=20
initiate-session\r\n<BR>\r\n</SPAN></o:p></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><o:p><SPAN=20
class=3D889351909-04122002>RTSP/1.0 200 OK\r\n<BR>CSeq: 1\r\n<BR>Public: =
OPTIONS,=20
DESCRIBE, SETUP, TEARDOWN, PLAY\r\n<BR>\r\n</SPAN></o:p></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><o:p><SPAN=20
class=3D889351909-04122002>DESCRIBE rtsp://10.111.10.65:554/test.mpg=20
RTSP/1.0\r\n<BR>CSeq: 2\r\n<BR>Accept: application/sdp\r\n<BR>Bandwidth: =

10485800\r\n<BR>Language: en\r\n<BR>Require:=20
com.real.retain-entity-for-setup\r\n<BR>SupportsMaximumASMBandwidth:=20
1\r\n<BR>\r\n</SPAN></o:p></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><o:p><SPAN=20
class=3D889351909-04122002>RTSP/1.0 200 OK\r\n<BR>CSeq: =
2\r\n<BR>Content-Type:=20
application/sdp\r\n<BR>Content-Length: =
250\r\n<BR>\r\n</SPAN></o:p></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><o:p><SPAN=20
class=3D889351909-04122002>SETUP rtsp://10.111.10.65:554/test.mpg/track1 =

RTSP/1.0\r\n<BR>CSeq: 3\r\n<BR>Transport:=20
rtp/avp;unicast;client_port=3D8888-8889;mode=3Dplay,rtp/avp/tcp;unicast;m=
ode=3Dplay\r\n<BR>\r\n</SPAN></o:p></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><o:p><SPAN=20
class=3D889351909-04122002>RTSP/1.0 200 OK\r\n<BR>CSeq: =
3\r\n<BR>Transport:=20
RTP/AVP;unicast;client_port=3D8888-8889;server_port=3D8888-8889\r\n<BR>\r=
\n</SPAN></o:p></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><o:p><SPAN=20
class=3D889351909-04122002>PLAY rtsp://10.111.10.65:554/test.mpg=20
RTSP/1.0\r\n<BR>CSeq: 4\r\n<BR>Session: 1\r\n<BR>Range:=20
npt=3D0-\r\n<BR>\r\n</SPAN></o:p></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><o:p><SPAN=20
class=3D889351909-04122002><BR>RTSP/1.0 200 OK\r\n<BR>CSeq: =
4\r\n<BR>Session:=20
1\r\n<BR>\r\n</SPAN></o:p></FONT></P></DIV></BODY></HTML>

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X-Original-Date: Wed, 04 Dec 2002 02:27:43 -0800
Date: Wed, 04 Dec 2002 02:27:43 -0800

At 01:33 AM 12/4/02, you wrote:
>Hello Ross,
>
>     I am facing problems while streaming MPEG-1 video-only file from 
> Livemedia server to RealOne Player client.
>     I am using testMPEGVideoStreamer at server end for this purpose.

RealOne Player has a bug that prevents it from playing multicast streams 
using a "rtsp://" URL.  Instead, have it open the file "testMPEGVideo.sdp" 
(which is included in the "testProgs/" directory).

Alternatively, you can run QuickTime Player.  It should be able to play the 
"rtsp://" URL (or, alternatively, the ".sdp" file).

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>



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X-Original-Date: Wed, 04 Dec 2002 02:32:18 -0800
Date: Wed, 04 Dec 2002 02:32:18 -0800


>The implementation for my video is pretty much similar to the audio 
>portion I mentioned earlier...i.e. I have an encoder that encodes my video 
>into H.263 format and place this encoded data into a buffer...
>
>Then I would call a "ByteStreamBufferSource", similar to the one I created 
>for the audio part to fetch the encoded data from this buffer...
>
>As mentioned earlier, I have to write a H.263+ Stream Framer to frame up 
>my data before calling the "H263plusVideoRTPSink" class...

I still don't think you need to do this.  You should just be getting 
discrete frames from your H.263 encoder.  I don't understand why you would 
not know the sizes of each frame that your encoder produces.  Surely you 
must have this information?

If you get discrete frames from your encoder, then you can feed them 
directly into a "H263plusRTPSink".  You shouldn't need a separate 'framer' 
object.

         Ross.



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X-Original-Date: Wed, 04 Dec 2002 02:37:25 -0800
Date: Wed, 04 Dec 2002 02:37:25 -0800

At 07:20 PM 12/3/02, you wrote:
>hi there...
>
>I see a rtpPayloadFormat in the "H263plusVideoRTPSink" class...
>
>What values should I set for this rtpPayloadFormat? can I set it to 96?

Yes.  This payload format does not have a static payload type defined, so 
you need to choose one from the dynamic range [96-127].  So, 96 is OK.

>Also, does anyone know if there are any H.263 RTP decoders out there for 
>me to playback my transmitted H.263 video?

Yes, QuickTime Player (for MacOS or Windows) can probably play it.  To play 
the stream, have QuickTime Player open a ".sdp" file that looks like the 
existing "testMPEGVideo.sdp" (or "testMPEGAudioVideo.sdp") file, except 
with the line:
         m=video 8888 RTP/AVP 32
replaced by:
         m=video 8888 RTP/AVP 96
         a=rtpmap:96 H263-1998/90000

     Ross.



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Hello again,

	Thanks Ross.
	Its true in the case of multicast but I am using unicast address for streaming. Still it doesn't work?
	What could be the problem?
	I think for QT to work with liveMedia for MPEG-1/2 we require some plugin.
	Also I was not able to find the testMPEGVideo.sdp file in testProgs/ directory

Regards
Prakash
	


-----Original Message-----
From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
Sent: Wednesday, December 04, 2002 3:58 PM
To: <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] Problems while streaming MPEG1/2 file


At 01:33 AM 12/4/02, you wrote:
>Hello Ross,
>
>     I am facing problems while streaming MPEG-1 video-only file from
> Livemedia server to RealOne Player client.
>     I am using testMPEGVideoStreamer at server end for this purpose.

RealOne Player has a bug that prevents it from playing multicast streams 
using a "rtsp://" URL.  Instead, have it open the file "testMPEGVideo.sdp" 
(which is included in the "testProgs/" directory).

Alternatively, you can run QuickTime Player.  It should be able to play the 
"rtsp://" URL (or, alternatively, the ".sdp" file).

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>



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X-Original-Date: Wed, 04 Dec 2002 10:12:23 -0800
Date: Wed, 04 Dec 2002 10:12:23 -0800

At 04:14 AM 12/4/02, you wrote:
>Hello again,
>
>         Thanks Ross.
>         Its true in the case of multicast but I am using unicast address 
> for streaming. Still it doesn't work?
>         What could be the problem?

The RTSP server that's built in to the "testMP*Streamer" test programs 
currently does not work for unicast (regardless of which player you're 
using to try to play it).  You should change the destination address back 
to multicast (e.g., 239.255.42.42) if you want to use RTSP.  But in any 
case, RealOne Player won't work (with a "rtsp://" URL); you'll need to use 
QuickTime Player

>         I think for QT to work with liveMedia for MPEG-1/2 we require 
> some plugin.

No, MPEG-1 video is supported. (Also, a plugin for MPEG-2 video can be 
purchased from Apple.)

>         Also I was not able to find the testMPEGVideo.sdp file in 
> testProgs/ directory

It's there.  Please look again.

         Ross.



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X-Original-Date: Wed, 04 Dec 2002 21:10:43 -0500
Date: Wed, 04 Dec 2002 21:10:43 -0500

Hi Ross,

The H.263 encoder that I'm currently using was originally built for TCP/IP streaming (unicast) using the "CSocket" class. 

So basically, how it initially works is that starts 2 threads, one is for encoding the raw video into H.263 format and putting this into a buffer, and the other thread will read from this buffer and write it to the CSocket to transmit out to the receiver.

Okay...correct me if I'm wrong, so what I should do is :

inside the doGetNextFrame() function, I would use the H.263 encoder to encode a frame and place this encoded frame into fTo, then I would use the "H263plusRTPSink" class directly after that, is that right?

But what I'm WORRIED about is that if I don't buffer up the encoded H.263 frames, and just sink every frame I encoded, then won't I lose some of the frames if the "263plusRTPSink" is slow? but then if I buffer up, the problem would be I won't know how large is each frame!

Thanks! Your advices are greatly appreciated!

Joseph

--

On Wed, 04 Dec 2002 02:32:18  
 Ross Finlayson wrote:
>
>>The implementation for my video is pretty much similar to the audio 
>>portion I mentioned earlier...i.e. I have an encoder that encodes my video 
>>into H.263 format and place this encoded data into a buffer...
>>
>>Then I would call a "ByteStreamBufferSource", similar to the one I created 
>>for the audio part to fetch the encoded data from this buffer...
>>
>>As mentioned earlier, I have to write a H.263+ Stream Framer to frame up 
>>my data before calling the "H263plusVideoRTPSink" class...
>
>I still don't think you need to do this.  You should just be getting 
>discrete frames from your H.263 encoder.  I don't understand why you would 
>not know the sizes of each frame that your encoder produces.  Surely you 
>must have this information?
>
>If you get discrete frames from your encoder, then you can feed them 
>directly into a "H263plusRTPSink".  You shouldn't need a separate 'framer' 
>object.
>
>         Ross.
>
>
>
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Hi Ross,

Sorry...I'm a little confused now...

Am I right to say that after reading the data from a file (e.g test.mpg) and if I call MPEGVideoStreamFramer, the MPEGVideoStreamFramer class will attach MPEG Video Headers to the data I read from the file to make it into one MPEG frame?

Joe




--

On Wed, 04 Dec 2002 02:32:18  
 Ross Finlayson wrote:
>
>>The implementation for my video is pretty much similar to the audio 
>>portion I mentioned earlier...i.e. I have an encoder that encodes my video 
>>into H.263 format and place this encoded data into a buffer...
>>
>>Then I would call a "ByteStreamBufferSource", similar to the one I created 
>>for the audio part to fetch the encoded data from this buffer...
>>
>>As mentioned earlier, I have to write a H.263+ Stream Framer to frame up 
>>my data before calling the "H263plusVideoRTPSink" class...
>
>I still don't think you need to do this.  You should just be getting 
>discrete frames from your H.263 encoder.  I don't understand why you would 
>not know the sizes of each frame that your encoder produces.  Surely you 
>must have this information?
>
>If you get discrete frames from your encoder, then you can feed them 
>directly into a "H263plusRTPSink".  You shouldn't need a separate 'framer' 
>object.
>
>         Ross.
>
>
>
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X-Original-Date: Wed, 04 Dec 2002 19:48:08 -0800
Date: Wed, 04 Dec 2002 19:48:08 -0800


>So basically, how it initially works is that starts 2 threads, one is for 
>encoding the raw video into H.263 format and putting this into a buffer

*This* is the point at which you know the size of the frame.  So, at this 
point, your buffer data structure should store not only the frame data, but 
also where the frame begins and ends (and thus, its size).  (I'll let you 
work out what kind of data structure you need to do this.)

>inside the doGetNextFrame() function, I would use the H.263 encoder to 
>encode a frame and place this encoded frame into fTo

Yes, and also set "fFrameSize" to be the size of this frame.

>, then I would use the "H263plusRTPSink" class directly after that, is 
>that right?

Sort of.  You would call "scheduleDelayedTask(0, ...)", which will end up 
causing the "H263PlusRTPSink" code to be executed.  See 
"DeviceSource::deliverFrame()" for an example of how to do this.

***Please read and understand "DeviceSource.cpp"***.  I deliberately wrote 
this to be an example that people could use when writing a source that 
encapsulates an encoder.

>But what I'm WORRIED about is that if I don't buffer up the encoded H.263 
>frames, and just sink every frame I encoded, then won't I lose some of the 
>frames if the "263plusRTPSink" is slow? but then if I buffer up, the 
>problem would be I won't know how large is each frame!

As I noted above, you need to remember the size of each frame when it gets 
written into the buffer.

         Ross.



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Date: Wed, 04 Dec 2002 19:53:10 -0800


>Am I right to say that after reading the data from a file (e.g test.mpg) 
>and if I call MPEGVideoStreamFramer, the MPEGVideoStreamFramer class will 
>attach MPEG Video Headers to the data I read from the file to make it into 
>one MPEG frame?

No!  "MPEGVideoStreamFramer" merely parses a data stream into discrete 
frames - i.e., figures out where the frame boundaries are.  It doesn't add 
any MPEG headers.  (The MPEG headers will have already been included by 
your encoder, when it generated the frame.)

(The same is true for "MPEGAudioStreamFramer".)

Questions about the code are always welcome, but please try to spend a 
little more time understanding what the code does beforehand.  One of the 
benefits of having complete source code is that it gives you the 
opportunity to figure out exactly how it works.

         Ross.




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X-Original-Date: Wed, 04 Dec 2002 22:14:41 -0500
Date: Wed, 04 Dec 2002 22:14:41 -0500

Hi Ross,

Sorry...I'm a little confused now...

Am I right to say that after reading the data from a file (e.g test.mpg) and if I call MPEGVideoStreamFramer, the MPEGVideoStreamFramer class will attach MPEG Video Headers to the data I read from the file to make it into one MPEG frame?

Joe




--

On Wed, 04 Dec 2002 02:32:18  
 Ross Finlayson wrote:
>
>>The implementation for my video is pretty much similar to the audio 
>>portion I mentioned earlier...i.e. I have an encoder that encodes my video 
>>into H.263 format and place this encoded data into a buffer...
>>
>>Then I would call a "ByteStreamBufferSource", similar to the one I created 
>>for the audio part to fetch the encoded data from this buffer...
>>
>>As mentioned earlier, I have to write a H.263+ Stream Framer to frame up 
>>my data before calling the "H263plusVideoRTPSink" class...
>
>I still don't think you need to do this.  You should just be getting 
>discrete frames from your H.263 encoder.  I don't understand why you would 
>not know the sizes of each frame that your encoder produces.  Surely you 
>must have this information?
>
>If you get discrete frames from your encoder, then you can feed them 
>directly into a "H263plusRTPSink".  You shouldn't need a separate 'framer' 
>object.
>
>         Ross.
>
>
>
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X-Original-Date: Thu, 05 Dec 2002 00:52:53 -0500
Date: Thu, 05 Dec 2002 00:52:53 -0500

Hi Ross,

Thanks for the advices, tips and everything...

Okay, I kinda get the picture now... I'm just a little confused on one part, could you please enlighten me?

if I am able to supply a discrete H.263 frame, but however if this frame is too large to fit into 1 RTP packet, then how am I able to transmit this frame in parts (i.e. multiple RTP packets)? 

Regards,
Joseph

P.S: I've just spoken to the developer of the H.263 encoder that I'm currently using...basically what my encoder does is, as it is generating 1 H.263 frame, it will put what it has encoded into the buffer (even if the frame hasn't been completely generated) and then the other thread will send what's in this buffer to the receiver...

So, perhaps the first block of data from the buffer sent may contain the the first 1/3 of the full H.263 frame, followed by the 2nd block containing the 2/3 of the frame and the last block contains the final portion of the H.263 frame.

I think I should be able to modify the encoder part so that it puts the entire frame into the buffer...


 
--

On Wed, 04 Dec 2002 19:48:08  
 Ross Finlayson wrote:
>
>>So basically, how it initially works is that starts 2 threads, one is for 
>>encoding the raw video into H.263 format and putting this into a buffer
>
>*This* is the point at which you know the size of the frame.  So, at this 
>point, your buffer data structure should store not only the frame data, but 
>also where the frame begins and ends (and thus, its size).  (I'll let you 
>work out what kind of data structure you need to do this.)
>
>>inside the doGetNextFrame() function, I would use the H.263 encoder to 
>>encode a frame and place this encoded frame into fTo
>
>Yes, and also set "fFrameSize" to be the size of this frame.
>
>>, then I would use the "H263plusRTPSink" class directly after that, is 
>>that right?
>
>Sort of.  You would call "scheduleDelayedTask(0, ...)", which will end up 
>causing the "H263PlusRTPSink" code to be executed.  See 
>"DeviceSource::deliverFrame()" for an example of how to do this.
>
>***Please read and understand "DeviceSource.cpp"***.  I deliberately wrote 
>this to be an example that people could use when writing a source that 
>encapsulates an encoder.
>
>>But what I'm WORRIED about is that if I don't buffer up the encoded H.263 
>>frames, and just sink every frame I encoded, then won't I lose some of the 
>>frames if the "263plusRTPSink" is slow? but then if I buffer up, the 
>>problem would be I won't know how large is each frame!
>
>As I noted above, you need to remember the size of each frame when it gets 
>written into the buffer.
>
>         Ross.
>
>
>
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>if I am able to supply a discrete H.263 frame, but however if this frame 
>is too large to fit into 1 RTP packet, then how am I able to transmit this 
>frame in parts (i.e. multiple RTP packets)?

You don't need to worry about this - the "H263RTPSink" class does this 
automatically.  Just be sure to feed it complete H.263 frames, and it will 
take care of fragmenting/packetizing them into RTP packets (as defined in 
RFC 2429).

         Ross.



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X-Original-Date: 07 Dec 2002 00:33:36 +0100
Date: 07 Dec 2002 00:33:36 +0100

I'm trying to play files like
rtsp://real01.dr.dk/uland/bertelsen/2_bertelsen_03.rm
in mplayer with liveMedia support, but I get the following errors:

Failed to initiate "audio/X-PN-REALAUDIO" RTP subsession: RTP payload
format unknown or not supported
Failed to initiate "video/X-PN-REALVIDEO" RTP subsession: RTP payload
format unknown or not supported

Any hints/ideas?

-- 
Jonas Jensen <jbj(at)knef.dk>



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X-Original-Date: Fri, 06 Dec 2002 21:27:28 -0800
Date: Fri, 06 Dec 2002 21:27:28 -0800

At 03:33 PM 12/6/02, you wrote:
>I'm trying to play files like
>rtsp://real01.dr.dk/uland/bertelsen/2_bertelsen_03.rm
>in mplayer with liveMedia support, but I get the following errors:
>
>Failed to initiate "audio/X-PN-REALAUDIO" RTP subsession: RTP payload
>format unknown or not supported
>Failed to initiate "video/X-PN-REALVIDEO" RTP subsession: RTP payload
>format unknown or not supported
>
>Any hints/ideas?

Yes, the RTP payload format is unknown or not supported :-)

Seriously: the "liveMedia" library does not support RealAudio or RealVideo 
streams, because:
1/ These streams typically do not use RTP for transport.  Instead, they use 
RealNetworks' proprietary (and undocumented) "RDT" protocol.
2/ RealNetworks servers often use a cryptographic 'challenge-response' 
mechanism to authenticate clients.  This prevents non-RealNetworks clients 
(such as "openRTSP" or "MPlayer") from playing these streams.

However, you *can* use MPlayer to play many "rtsp://" URLs that are 
streamed from other kinds of RTSP servers (e.g., Apple's QuickTime/Darwin 
Streaming Server).  You can also play some types of streams from 
RealNetworks' servers, as long as the media are not RealAudio or RealVideo.

         Ross.



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From: Bertrand Baudet <EMAIL-ADDRESS-HIDDEN>
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Subject: Re: [Live-devel] audio/X-PN-REALAUDIO and video/X-PN-REALVIDEO
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X-Original-Date: Sat, 7 Dec 2002 01:06:36 +0000
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BTW, Ross, did you get a chance to downloal the helix source code and 
figure out what they provide on their rstp protocol network?

I didn't get a chance to look at it for now, so any feedback on that is 
welcome.

Bertrand

On Saturday 07 December 2002 05:27 am, Ross Finlayson wrote:
> At 03:33 PM 12/6/02, you wrote:
> >I'm trying to play files like
> >rtsp://real01.dr.dk/uland/bertelsen/2_bertelsen_03.rm
> >in mplayer with liveMedia support, but I get the following errors:
> >
> >Failed to initiate "audio/X-PN-REALAUDIO" RTP subsession: RTP payload
> >format unknown or not supported
> >Failed to initiate "video/X-PN-REALVIDEO" RTP subsession: RTP payload
> >format unknown or not supported
> >
> >Any hints/ideas?
>
> Yes, the RTP payload format is unknown or not supported :-)
>
> Seriously: the "liveMedia" library does not support RealAudio or RealVideo
> streams, because:
> 1/ These streams typically do not use RTP for transport.  Instead, they use
> RealNetworks' proprietary (and undocumented) "RDT" protocol.
> 2/ RealNetworks servers often use a cryptographic 'challenge-response'
> mechanism to authenticate clients.  This prevents non-RealNetworks clients
> (such as "openRTSP" or "MPlayer") from playing these streams.
>
> However, you *can* use MPlayer to play many "rtsp://" URLs that are
> streamed from other kinds of RTSP servers (e.g., Apple's QuickTime/Darwin
> Streaming Server).  You can also play some types of streams from
> RealNetworks' servers, as long as the media are not RealAudio or RealVideo.
>
>          Ross.
>
>
>
> -------------------------------------------------------
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X-Original-Date: Sat, 07 Dec 2002 01:35:42 -0800
Date: Sat, 07 Dec 2002 01:35:42 -0800

At 05:06 PM 12/6/02, you wrote:
>BTW, Ross, did you get a chance to downloal the helix source code and
>figure out what they provide on their rstp protocol network?

No, not yet.  However, Rob Lanphier - the manager of RealNetworks' Helix 
Project - is on this mailing list, so he may end up commenting on whether 
any of their proprietary protocols have been opened up by Helix.

         Ross.



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X-Original-Date: Sun, 08 Dec 2002 00:02:49 -0500
Date: Sun, 08 Dec 2002 00:02:49 -0500

Hi Ross,

Alright...Thanks again for the information... I'm getting a clearer picture now...was confused about the "Framer" class and the "sink" class previously.

Were you trying to develop a MPEG4sink class previously? How is the development? 


Regards,
Joseph



--

On Thu, 05 Dec 2002 01:45:33  
 Ross Finlayson wrote:
>
>>if I am able to supply a discrete H.263 frame, but however if this frame 
>>is too large to fit into 1 RTP packet, then how am I able to transmit this 
>>frame in parts (i.e. multiple RTP packets)?
>
>You don't need to worry about this - the "H263RTPSink" class does this 
>automatically.  Just be sure to feed it complete H.263 frames, and it will 
>take care of fragmenting/packetizing them into RTP packets (as defined in 
>RFC 2429).
>
>         Ross.
>
>
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X-Original-Date: Sat, 07 Dec 2002 21:20:26 -0800
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>Were you trying to develop a MPEG4sink class previously? How is the 
>development?

This remains on our "to do" list, but, as always, tasks requested by paying 
clients take priority.  (So, if anyone is interested in providing funding 
for MPEG-4 support in our software, then please let me know.)

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>



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X-Original-Date: Mon, 09 Dec 2002 22:22:48 -0500
Date: Mon, 09 Dec 2002 22:22:48 -0500

Hi Dr Ross and fellow Live developers,

Like I said earlier...in my program, one of the thread is constantly encoding the data from the camera and I will then store the discrete frames together with its frame size into a buffer.

Currently my buffer size is 100, i.e. I can store up to 100 frames before the "H263PlusVideoSink" sends the frames...

Do you think my buffer size is too big or too small? Any comments?

Thanks...

Regards,
Joseph


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X-Original-Date: Mon, 09 Dec 2002 22:25:10 -0500
Date: Mon, 09 Dec 2002 22:25:10 -0500

Hi...

just wondering if anyone know of any programs that can receive MPEG4 RTP packets?

How about Quicktime or RealOnePlayer ?

Joseph


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>just wondering if anyone know of any programs that can receive MPEG4 RTP 
>packets?
>
>How about Quicktime or RealOnePlayer ?

QuickTime Player definitely can.  I think RealOne Player can also (perhaps 
with a special plugin).

         Ross.



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X-Original-Date: Tue, 10 Dec 2002 02:01:19 -0500
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Hi all,

Has anyone tried the "H263PlusVideoSink" class with a H.263 encoder before?

I'm thinking my encoder might not be an actual H.263 encoder... does anyone know where I could find a open source H.263 encoder?

Thanks...

Joseph




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X-Original-Date: Tue, 10 Dec 2002 03:59:15 -0500
Date: Tue, 10 Dec 2002 03:59:15 -0500

Hi...

I've read on the internet, 

"At low-level MPEG-4 is a Motion Picture Experts Group (MPEG) standard that can be thought of as H.263 enhanced with features such as error resilience. MPEG-4 also includes higher level support for content-based interactivity and application specific flexibility. "

If that is the case, Why can't I simply carry MPEG4 on the 
"H263plusVideoSink" if they are pretty similar?

Why then is there another standard to carry MPEG4 (i.e. the RFC 3016) and then there is another RFC 2429 for H.263+?


Regards,
Joseph




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Subject: Re: [Live-devel] Question on MPEG4 and H.263?
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X-Original-Date: Tue, 10 Dec 2002 02:20:59 -0800
Date: Tue, 10 Dec 2002 02:20:59 -0800

At 12:59 AM 12/10/02, you wrote:
>Hi...
>
>I've read on the internet,
>
>"At low-level MPEG-4 is a Motion Picture Experts Group (MPEG) standard 
>that can be thought of as H.263 enhanced with features such as error 
>resilience. MPEG-4 also includes higher level support for content-based 
>interactivity and application specific flexibility. "
>
>If that is the case, Why can't I simply carry MPEG4 on the
>"H263plusVideoSink" if they are pretty similar?

Because the packet formats (i.e., headers) are different.  A receiver 
that's expecting MPEG-4 video content won't understand it if it appears in 
a RFC 2429 RTP packet.

>Why then is there another standard to carry MPEG4 (i.e. the RFC 3016) and 
>then there is another RFC 2429 for H.263+?

Apparently the two codecs were sufficiently different to make this necessary.

         Ross.



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Subject: [Live-devel] RTP DirectShow Filter
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X-Original-Date: Thu, 12 Dec 2002 00:37:27 +0100
Date: Thu, 12 Dec 2002 00:37:27 +0100

This is a multi-part message in MIME format.

------=_NextPart_000_000B_01C2A176.A59F5250
Content-Type: text/plain;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

I need a filter for direct show that send a RTP Stream. Anybody knows =
where can I find it?


Thanks

------=_NextPart_000_000B_01C2A176.A59F5250
Content-Type: text/html;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2600.0" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>I need a filter for direct show that =
send a RTP=20
Stream. Anybody knows where can I find it?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Thanks</FONT></DIV></BODY></HTML>

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X-Original-Date: Thu, 12 Dec 2002 01:51:42 +0100
Date: Thu, 12 Dec 2002 01:51:42 +0100

C'est un message de format MIME en plusieurs parties.

------=_NextPart_000_0016_01C2A181.0505E240
Content-Type: text/plain;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

We're working on (based on liveMedia) still early beta and undocumented =
:

http://www.morgan-multimedia.com/RTP/

Giom.
  ----- Original Message -----=20
  From: Francisco J. Dominguez=20
  To: live-devel(at)lists.sourceforge.net=20
  Sent: Thursday, December 12, 2002 12:37 AM
  Subject: [Live-devel] RTP DirectShow Filter


  I need a filter for direct show that send a RTP Stream. Anybody knows =
where can I find it?


  Thanks

------=_NextPart_000_0016_01C2A181.0505E240
Content-Type: text/html;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 5.50.4807.2300" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>We're working on (based on liveMedia) =
still early=20
beta and undocumented :</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><A=20
href=3D"http://www.morgan-multimedia.com/RTP/">http://www.morgan-multimed=
ia.com/RTP/</A></FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Giom.</FONT></DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
  <DIV style=3D"FONT: 10pt arial">----- Original Message ----- </DIV>
  <DIV=20
  style=3D"BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: =
black"><B>From:</B>=20
  <A title=3Dgimmacks(at)telepolis.com =
href=3D"mailto:gimmacks(at)telepolis.com">Francisco=20
  J. Dominguez</A> </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>To:</B> <A=20
  title=3Dlive-devel(at)lists.sourceforge.net=20
  =
href=3D"mailto:live-devel(at)lists.sourceforge.net">live-devel(at)lists.sourcef=
orge.net</A>=20
  </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Sent:</B> Thursday, December 12, =
2002 12:37=20
  AM</DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Subject:</B> [Live-devel] RTP =
DirectShow=20
  Filter</DIV>
  <DIV><BR></DIV>
  <DIV><FONT face=3DArial size=3D2>I need a filter for direct show that =
send a RTP=20
  Stream. Anybody knows where can I find it?</FONT></DIV>
  <DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial =
size=3D2>Thanks</FONT></DIV></BLOCKQUOTE></BODY></HTML>

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X-Original-Date: Wed, 11 Dec 2002 17:41:10 -0800
Date: Wed, 11 Dec 2002 17:41:10 -0800

At 04:51 PM 12/11/02, you wrote:
>We're working on (based on liveMedia) still early beta and undocumented :
>
><http://www.morgan-multimedia.com/RTP/>http://www.morgan-multimedia.com/RTP/

Wow - this looks really good.  Great stuff!

         Ross.



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X-Original-Date: Thu, 12 Dec 2002 03:57:44 +0100
Date: Thu, 12 Dec 2002 03:57:44 +0100

Thanks, liveMedia is great stuff too =)

Giom.

----- Original Message -----
From: {Ross Finlayson} <EMAIL-ADDRESS-HIDDEN>
To: <EMAIL-ADDRESS-HIDDEN>
Sent: Thursday, December 12, 2002 2:41 AM
Subject: Re: [Live-devel] RTP DirectShow Filter


> At 04:51 PM 12/11/02, you wrote:
> >We're working on (based on liveMedia) still early beta and undocumented :
> >
>
><http://www.morgan-multimedia.com/RTP/>http://www.morgan-multimedia.com/RTP
/
>
> Wow - this looks really good.  Great stuff!
>
>          Ross.
>
>
>
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>




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Hello again.

I have been looking your web, and it looks very very well!!!.
I'm testing the destination filter with graph edit, but I don=B4t know =
how to specify the ip and the port.
Could you tell me how to do it?

Thanks.

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
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<DIV><FONT face=3DArial size=3D2>Hello again.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>I have been looking your web, and it =
looks very=20
very well!!!.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>I'm testing the destination filter with =
graph edit,=20
but I don=B4t know how to specify the ip and the port.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Could you tell me how to do =
it?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
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At 09:27 AM 12/12/02, you wrote:
>Hello again.
>
>I have been looking your web, and it looks very very well!!!.
>I'm testing the destination filter with graph edit, but I don´t know how 
>to specify the ip and the port.
>Could you tell me how to do it?

Questions specifically about the Morgan Multimedia DirectShow software 
should go to Guillaume de Bailliencourt: bailliencourt(at)morgan-multimedia.com

(This mailing list is for questions specifically about the LIVE.COM 
Streaming Media software libraries.)

         Ross.



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Hi Ross,

I have a problem in running the program that was taken from the testMP3Streamer.cpp. All the tests from live.com are working fine. But when I take source from Streamer.cpp and place in my application it compiles and links well. but WIll not run.

I'm using WIN98 os. i have network interfaces. i get problem n GroupSocks. It is not able to create a new sock( probably not able to addto the IP_ADD member ship) I assume that, an error comes access violation  and the application aborts 

Please help me. I have gone thru the archives  but i could ot find any solution  for this. 

thanks 

Satya



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<P>Hi Ross,</P>
<P>I have a problem in running the program that was taken from the testMP3Streamer.cpp. All the tests from live.com are working fine. But when I take source from Streamer.cpp and place in my application it compiles and links well. but WIll not run.</P>
<P>I'm using WIN98 os. i have network interfaces. i get problem n GroupSocks. It is not able to create a new sock( probably not able to addto the IP_ADD member ship) I assume that, an error comes access violation&nbsp; and the application aborts </P>
<P>Please help me.&nbsp;I have gone thru the archives&nbsp; but i could ot find any solution&nbsp; for this. </P>
<P>thanks </P>
<P>Satya</P><p><br><hr size=1>Do you Yahoo!?<br>
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>I have a problem in running the program that was taken from the 
>testMP3Streamer.cpp. All the tests from live.com are working fine. But 
>when I take source from Streamer.cpp and place in my application it 
>compiles and links well. but WIll not run.

The fact that the test programs worked OK for you is a good sign.  This 
suggests that the problem is not your system, but the way in which you're 
using the library code in your application.

>I'm using WIN98 os. i have network interfaces. i get problem n GroupSocks. 
>It is not able to create a new sock( probably not able to addto the IP_ADD 
>member ship) I assume that, an error comes access violation  and the 
>application aborts

As you have complete source code, you should be able to figure out exactly 
where (and why) the memory access violation error occurs.  Once you've done 
that, you should be able to figure out why "testMP3Streamer" is working OK, 
but your application is not.  If not, then please let us know the details 
of the error.

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>



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HI Ross,
 Thanks Ross for your quick reply.
 My application is an MFC  one. I have created a dialog box  and taking teh file nam eas teh input and passing it as a parameter. Rest is junk MFC code and code from TestMP3Streamer.cpp code. 
 The application is able to create a session. it fails when "new" call is called for the groupsocket. i have tried it in WIN32  console application  and MFC application . the result is same fails while allocating teh new Groupsock for teh pointer  in the Session Structure(same as in testStreamer.cpp).
I have even looked at the compile options  ahev included the same. i get error when  i GIVE /NODEFAULTLIB option(bcos its given in the makefile) but it compiles without that option.
Thanks once again
Satya
 



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<P>HI Ross,
<P>&nbsp;Thanks Ross for your quick reply.
<P>&nbsp;My application is an MFC&nbsp; one. I have created a dialog box&nbsp; and taking teh file nam eas teh input and passing it as a parameter. Rest is junk MFC code and code from TestMP3Streamer.cpp code. 
<P>&nbsp;The application is able to create a session. it fails when "new" call is called for the groupsocket. i have tried it in WIN32&nbsp; console application&nbsp; and MFC application . the result is same fails while allocating teh new Groupsock for teh pointer&nbsp; in the Session Structure(same as in testStreamer.cpp).
<P>I have even looked at the compile options&nbsp; ahev included the same. i get error when&nbsp; i&nbsp;GIVE /NODEFAULTLIB option(bcos its given in the makefile)&nbsp;but it compiles without that option.
<P>Thanks once again
<P>Satya
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>  The application is able to create a session. it fails when "new" call is 
> called for the groupsocket.

First, a dumb question: Are you remembering to create a "UsageEnvironment" 
- i.e., the variable "env" - beforehand, and checking that it's 
non-NULL?  The reason I ask is that the call
         new Groupsock(*env, destinationAddress, rtpPort, ttl)
will, of course, fail if "env" is NULL.

If this is OK, then does any part of the Groupsock::Groupsock() constructor 
code actually get executed?

         Ross.



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HI Ross,

 I looked whiel debugiing  initially it was null but later teh pointer is getting allocated in env variable.

i could not get into code of group sock. here the CallStack of the function failure

_free_dbg_lk(void * 0x00791010, int 1) line 1017 + 82 bytes
_free_dbg(void * 0x00791010, int 1) line 970 + 13 bytes
free(void * 0x00791010) line 926 + 11 bytes
MP3TESTER! Scope::clean(void) + 10 bytes
00791010() ---------------------------->>>>>>>>>>>>>>>>>>> play fucntion comes upto                                                                                new groupsock
CMp3testerDlg::OnButtonPlay() line 116
_AfxDispatchCmdMsg(CCmdTarget * 0x0066fcec {CMp3testerDlg}, unsigned int 1000, int 0, void (void)* 0x0040104b CMp3testerDlg::OnButtonPlay(void), void * 0x00000000, unsigned int 12, AFX_CMDHANDLERINFO * 0x00000000) line 88
CCmdTarget::OnCmdMsg(unsigned int 1000, in

 

I need help to get into the code of groupsock while debugging.

Thanks

Live-devel(at)lists.sourceforge.net




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<P>HI Ross,</P>
<P>&nbsp;I looked whiel debugiing&nbsp; initially it was null but later teh pointer is getting allocated in env variable.</P>
<P>i could not get into code of group sock. here the CallStack of the function failure</P>
<P>_free_dbg_lk(void * 0x00791010, int 1) line 1017 + 82 bytes<BR>_free_dbg(void * 0x00791010, int 1) line 970 + 13 bytes<BR>free(void * 0x00791010) line 926 + 11 bytes<BR>MP3TESTER! Scope::clean(void) + 10 bytes<BR>00791010() ----------------------------&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; play fucntion comes upto&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; new groupsock<BR>CMp3testerDlg::OnButtonPlay() line 116<BR>_AfxDispatchCmdMsg(CCmdTarget * 0x0066fcec {CMp3testerDlg}, unsigned int 1000, int 0, void (void)* 0x0040104b CMp3testerDlg::OnButtonPlay(void), void * 0x00000000, unsigned int 12, AFX_CMDHANDLERINFO * 0x00000000) line 88<BR>CCmdTarget::OnCmdMsg(unsigned int 1000, in</P>
<P>&nbsp;</P>
<P>I need help to get into the code of groupsock while debugging.</P>
<P>Thanks</P>
<P>Live-devel(at)lists.sourceforge.net<BR></P><p><br><hr size=1>Do you Yahoo!?<br>
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HI Ross,

 I looked whiel debugiing  initially it was null but later teh pointer is getting allocated in env variable.

i could not get into code of group sock. here the CallStack of the function failure

_free_dbg_lk(void * 0x00791010, int 1) line 1017 + 82 bytes
_free_dbg(void * 0x00791010, int 1) line 970 + 13 bytes
free(void * 0x00791010) line 926 + 11 bytes
MP3TESTER! Scope::clean(void) + 10 bytes
00791010() ---------------------------->>>>>>>>>>>>>>>>>>> play fucntion comes upto                                                                                new groupsock
CMp3testerDlg::OnButtonPlay() line 116
_AfxDispatchCmdMsg(CCmdTarget * 0x0066fcec {CMp3testerDlg}, unsigned int 1000, int 0, void (void)* 0x0040104b CMp3testerDlg::OnButtonPlay(void), void * 0x00000000, unsigned int 12, AFX_CMDHANDLERINFO * 0x00000000) line 88
CCmdTarget::OnCmdMsg(unsigned int 1000, in

 

I need help to get into the code of groupsock while debugging.

Thanks

Live-devel(at)lists.sourceforge.net




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<P>HI Ross,</P>
<P>&nbsp;I looked whiel debugiing&nbsp; initially it was null but later teh pointer is getting allocated in env variable.</P>
<P>i could not get into code of group sock. here the CallStack of the function failure</P>
<P>_free_dbg_lk(void * 0x00791010, int 1) line 1017 + 82 bytes<BR>_free_dbg(void * 0x00791010, int 1) line 970 + 13 bytes<BR>free(void * 0x00791010) line 926 + 11 bytes<BR>MP3TESTER! Scope::clean(void) + 10 bytes<BR>00791010() ----------------------------&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; play fucntion comes upto&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; new groupsock<BR>CMp3testerDlg::OnButtonPlay() line 116<BR>_AfxDispatchCmdMsg(CCmdTarget * 0x0066fcec {CMp3testerDlg}, unsigned int 1000, int 0, void (void)* 0x0040104b CMp3testerDlg::OnButtonPlay(void), void * 0x00000000, unsigned int 12, AFX_CMDHANDLERINFO * 0x00000000) line 88<BR>CCmdTarget::OnCmdMsg(unsigned int 1000, in</P>
<P>&nbsp;</P>
<P>I need help to get into the code of groupsock while debugging.</P>
<P>Thanks</P>
<P>Live-devel(at)lists.sourceforge.net<BR></P><p><br><hr size=1>Do you Yahoo!?<br>
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Hi Ross,

 Thanks Ross once again for the RTP library. 

I have compiled my Vc application. I found that i need to ignore the libc.lib and include teh msvcrt.lib  and link the files.  other thing is add teh winsock2.h in the stdafx.h. rest is all common . no warnings and no errors.  runs smoothly. 

While debugging i could get into the code  of groupsock and other files after compiling them with debug option .  

I want ot know  U have said that for PCM streams we can us e teh simpleRTPSink 

 hwo does this work. i want to implement the both Audio streams and live from mike and camera  how to integrate all. 

i understand "testAudioVideoStreamer.cpp" woudl transmit both Audio and Video streams is this correct?

Thanks 

Satya



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<P>Hi Ross,</P>
<P>&nbsp;Thanks Ross once again for the RTP library. </P>
<P>I have compiled my Vc application. I found that i need to ignore the libc.lib and include teh msvcrt.lib&nbsp; and link the files.&nbsp; other thing is add teh winsock2.h in the stdafx.h. rest is all common . no warnings and no errors.&nbsp; runs smoothly. </P>
<P>While debugging i could get into the code&nbsp; of groupsock and other files after compiling them with debug option .&nbsp; </P>
<P>I want ot know&nbsp; U have said that for PCM streams we can us e teh simpleRTPSink </P>
<P>&nbsp;hwo does this work. i want to implement the both Audio streams and live from mike and camera&nbsp; how to integrate all. </P>
<P>i understand "testAudioVideoStreamer.cpp" woudl transmit both Audio and Video streams is this correct?</P>
<P>Thanks </P>
<P>Satya</P><p><br><hr size=1>Do you Yahoo!?<br>
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>I want ot know  U have said that for PCM streams we can us e teh 
>simpleRTPSink
>
>  hwo does this work.

Because the RTP payload for PCM audio is very simple, you can stream it 
using a "RTPSink" object, created as follows:
         SimpleRTPSink::createNew(*env, &rtpGroupsock, 0,  8000,"audio", 
"pcmu");
This is for u-law PCM.  For a-law PCM, change "0" (the RTP payload format 
number) to "8", and "pcmu" to "pcma".

         Ross.



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X-Original-Date: Thu, 19 Dec 2002 11:49:21 +1300
Date: Thu, 19 Dec 2002 11:49:21 +1300

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Hi,
 
I am hoping someone can help me, I have been searching the site for
documentation on streaming from a windows machine to a Unix server. I
have installed the openRTSP via ports on a freebsd server and would like
to stream some video to mplayer. I have downloaded and tried the
openRTSP on a windows machine but it doesnt seem to have a stream to
server function. Perhaps someone has done this before I knows where I
can get a little more help.
 
My freebsd server runs an IRCD on port 6667-7000 so I was hoping that I
could use an alternate port.
 
This is what I want to do...
 
|HOME PC| ------ stream -----> |FREEBSD SERVER| ------ stream ------>
|INTERNET USERS VIA MPLAYER|
 
Thanks
Kindest Regards
Barry Murphy

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	charset="iso-8859-1"
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">
<TITLE>Message</TITLE>

<META content=3D"MSHTML 5.00.3315.2870" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D390154122-18122002>Hi,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D390154122-18122002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D390154122-18122002>I am =
hoping someone=20
can help me, I have been searching the site for documentation on =
streaming from=20
a windows machine to a Unix server. I have installed the openRTSP via =
ports on a=20
freebsd server and would like to stream some video to mplayer. I have =
downloaded=20
and tried the openRTSP on a windows machine but it doesnt seem to have a =
stream=20
to server function. Perhaps someone has done this before I knows where I =
can get=20
a little more help.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D390154122-18122002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D390154122-18122002>My =
freebsd server=20
runs an IRCD on port 6667-7000 so I was hoping that I could use an =
alternate=20
port.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D390154122-18122002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D390154122-18122002>This =
is what I want=20
to do...</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D390154122-18122002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D390154122-18122002>|HOME =
PC| ------=20
stream -----&gt; |FREEBSD SERVER| ------ stream ------&gt; |INTERNET =
USERS VIA=20
MPLAYER|</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D390154122-18122002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D390154122-18122002>Thanks</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN =
class=3D390154122-18122002>Kindest=20
Regards</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D390154122-18122002>Barry=20
Murphy</SPAN></FONT></DIV></BODY></HTML>

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Subject: Re: [Live-devel] Streaming
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X-Original-Date: Wed, 18 Dec 2002 17:34:07 -0800
Date: Wed, 18 Dec 2002 17:34:07 -0800


>This is what I want to do...
>
>|HOME PC| ------ stream -----> |FREEBSD SERVER| ------ stream ------> 
>|INTERNET USERS VIA MPLAYER|

G'day Barry,

Can you tell us some more about the video that you wish to stream?  In 
particular, are these pre-encoded video files, and if so, what format/codec 
do they use?  Or are you planning to stream live video, and if so, do you 
have any existing (hardware or sofware) video encoder?

Also, can you possibly substitute "Home Macintosh" for "Home PC"?

         Ross.



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Subject: [Live-devel] Differences  between MPEG-1 and H261/263
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X-Original-Date: Thu, 19 Dec 2002 04:21:56 +0100
Date: Thu, 19 Dec 2002 04:21:56 +0100

Hallo everyone,
I have a little problem that might not be a topic on this list. I have to
write something about the differences between MPEG-1 and H261 for my diploma
thesis. I found a lot of Books upon these standards but the question I have
could be answered. The question is: Is there any reason, why someone should
use MPEG and not H263 to develop a conferencing system ? Does anybody has a
idea? 

Thx a lot
Andreas      
  


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Subject: [Live-devel] Capture Audio from Mike
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--0-406815720-1040275457=:3694
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Hi Ross,

 I want to capture the Audio using Mike like and transmit. how do i do it?Is ther a sample application similar using the SimpleRTPSource and SimpleRTPSink()

thanks

 



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<P>Hi Ross,</P>
<P>&nbsp;I want to capture the Audio using Mike like and transmit. how do i do it?Is ther a sample application similar using the SimpleRTPSource and SimpleRTPSink()</P>
<P>thanks</P>
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From <EMAIL-ADDRESS-HIDDEN> Thu Dec 19 00:50:47 2002
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Subject: Re: [Live-devel] Capture Audio from Mike
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X-Original-Date: Thu, 19 Dec 2002 00:44:05 -0800
Date: Thu, 19 Dec 2002 00:44:05 -0800

At 09:24 PM 12/18/02, you wrote:

>Hi Ross,
>
>  I want to capture the Audio using Mike like and transmit. how do i do 
> it?Is ther a sample application similar using the SimpleRTPSource and 
> SimpleRTPSink()

No, but you can use the "testMP3Streamer" application as a model.  Instead 
of using a "MPEGAudioRTPSink", use a "SimpleRTPSink" (as I outlined in my 
earlier message).  And instead of using a "MP3FileSource" for input, use a 
new source class - that you would write - that encapsulates your audio 
input device.  For a model for how to write such a class, see 
"liveMedia/DeviceSource.cpp" (and "liveMedia/include/DeviceSource.hh").

Also, if you want the receiving client (such as QuickTimePlayer) to read 
from a ".sdp" file, you can use "testMP3.sdp" as an example, but you will 
need to change the "RTP payload format" number that's there.  I.e., change 
"14" to 0 (for PCM u-law), or 8 (for PCM a-law).

         Ross.



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X-Original-Date: Thu, 19 Dec 2002 08:31:08 -0800
Date: Thu, 19 Dec 2002 08:31:08 -0800

On Thu, Dec 19, 2002 at 04:21:56AM +0100,  Koustoumpardis, Andreas wrote:
> Hallo everyone,
> I have a little problem that might not be a topic on this list. I have to
> write something about the differences between MPEG-1 and H261 for my diploma
> thesis. I found a lot of Books upon these standards but the question I have
> could be answered. The question is: Is there any reason, why someone should
> use MPEG and not H263 to develop a conferencing system ? Does anybody has a
> idea? 

keeping in mind that i'm not an expert in the technical minutiae of
either codec, the primary difference seems to be political: mpeg is an
iso standard, whereas h261 is an itu standard. on the technical front,
h261 tends to be computationally cheaper, which makes it more
attractive for sticking into embedded devices.
                                       
-john
                                      

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Subject: [Live-devel] How does RTCP synchronize the video and audio streams?
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X-Original-Date: Sun, 22 Dec 2002 06:36:14 -0500
Date: Sun, 22 Dec 2002 06:36:14 -0500

Hi Ross,

You mentioned previously that if I had a H.263VideoSink class (for video source streaming) and MP3AudioSink class (for audio source streaming) and if I wanted to synchronize them, all I have to do is just create an instance of RTCP for each of these class...and it will automatically be synchronized?

I would like to know, how does this RTCP instance take care of the synchronization part at the receivers' end? Is it possible for you to give a brief explaination here while I try to understand your codes on this portion?

Also, I am currently reading RFC 1889, but I am still confused on the RTCP part...is there any other reference drafts I can read?

Thanks and Merry Christmas in advance!

Regards,
Joseph


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X-Original-Date: Sun, 22 Dec 2002 17:53:50 -0800
Date: Sun, 22 Dec 2002 17:53:50 -0800


>Also, I am currently reading RFC 1889, but I am still confused on the RTCP 
>part...is there any other reference drafts I can read?

The latest version of the RTP specification is online as
         http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-new-11.txt
(This will eventually turn into a RFC that will update RFC 1889.)

>I would like to know, how does this RTCP instance take care of the 
>synchronization part at the receivers' end? Is it possible for you to give 
>a brief explaination here while I try to understand your codes on this portion?

Receivers use incoming RTCP "SR" (Sender Report) messages - from the sender 
- to synchronize the RTP stream with 'wall clock time'.  See the code for
         case RTCP_PT_SR:
in "RTCP.cpp".  See also
         noteIncomingSR
in "RTPSource.cpp".  This code sets (for each "RTPSource") the 'wall clock 
time' ("fSyncTime") that corresponds to the RTP timestamp 
("fSyncTimestamp") that was in the most recently received RTCP "SR" 
packet.  This is used to compute an accurate "presentationTime" for each 
incoming RTP packet.  (See
         noteIncomingPacket
in "RTPSource.cpp".)

The key thing to note is that as long as your receiver code creates a 
"RTCPInstance" for each "RTPSource", all of this will get taken care of 
automatically, and the "presentationTime" parameter set for each incoming 
packet will be an accurate timestamp that can be used for synchronization.

         Ross.




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X-Original-Date: Mon, 23 Dec 2002 01:32:59 -0500
Date: Mon, 23 Dec 2002 01:32:59 -0500

Hi there...

I'd just like to know...if I do not use the MPEGAudioStreamFramer to frame up my MP3 source before sending it out via the AudioSink... Why is it that the receiver could not playback? 

What does the MPEGAudioStreamFramer do? What information does it add to the RTP packet? I'm currently looking at MPEGAudioStreamFramer.cpp but I still cannot figure that out, it would be nice if you could give me some hints...

How is it that if I don't have the Framer class, I won't be able to playback MP3?

Also, what size of buffer do you recommend for the receiver size for receiving the RTP packets? Currently I am using a 1MB buffer, and I believe it is too large and it is causing large delays in my audio stream!

Thanks..

Best Regards,
Joseph


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Hi Ross,

 As u have mentioned i have implementede the AudioDeviceSource class. I'm struck. Teh audio device is open and started also but the the callback structure the input even thouhg the device is open. 

Can i send my source if ulet me knwo

Thanks and MErry christmas

 

Satya

 

 



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<P>Hi Ross,</P>
<P>&nbsp;As u have mentioned&nbsp;i have implementede the AudioDeviceSource class. I'm struck. Teh audio device is open and started also but the the callback structure the input even thouhg the device is open. </P>
<P>Can i send my source if ulet me knwo</P>
<P>Thanks and MErry christmas</P>
<P>&nbsp;</P>
<P>Satya</P>
<P>&nbsp;</P>
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>  As u have mentioned i have implementede the AudioDeviceSource class. I'm 
> struck. Teh audio device is open and started also but the the callback 
> structure the input even thouhg the device is open.
>
>Can i send my source if ulet me knwo

Yes, feel free to post your source code to this mailing list.

         Ross.



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X-Original-Date: Mon, 23 Dec 2002 14:47:47 -0800
Date: Mon, 23 Dec 2002 14:47:47 -0800


>What does the MPEGAudioStreamFramer do?

It is a filter that parses its input into discrete MPEG audio frames - 
i.e., each time a reader calls "getNextFrame()" on a 
"MPEGAudioStreamFramer", it will get back a complete MPEG audio frame 
(i.e., beginning with a MPEG audio 'syncword').

>I'd just like to know...if I do not use the MPEGAudioStreamFramer to frame 
>up my MP3 source before sending it out via the AudioSink... Why is it that 
>the receiver could not playback?

The RTP payload format for MPEG audio - as described in RFC 2250 - 
specifies that  each RTP packet shall contain discrete MPEG audio frames. - 
i.e., the MPEG audio data that's packed into RTP packets must be aligned on 
audio frame boundaries (unless a single audio frame is too large for a 
packet, in which case it can get fragmented over multiple RTP packets).

If you didn't use a "MPEGAudioStreamFramer", the resulting output RTP 
packets would not conform to RFC 2250, and some receivers might not be able 
to handle them properly.

         Ross.



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Hi Ross,

 I have attached source  for the audio device. It is windows specific code. I could start the session and  deliver the frames but i 'm not able to listen. i could see that Quick time is playing  and real player is buffering. bt no sound.

 Tha sample program is similar to testMp3Streamer but changed teh source to "audiodevicesource" and sink to "simplertpsink". 

Please let me know where the fault is....if any

Thanks 

Satya



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--0-1231157336-1040696981=:68555
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<P>Hi Ross,</P>
<P>&nbsp;I have attached source&nbsp; for the audio device. It is windows specific code. I could start the session and&nbsp; deliver the frames but i 'm not able to listen. i could see that Quick time is playing&nbsp; and real player is buffering. bt no sound.</P>
<P>&nbsp;Tha sample program is similar to testMp3Streamer but changed teh source to "audiodevicesource" and sink to "simplertpsink". </P>
<P>Please let me know where the fault is....if any</P>
<P>Thanks </P>
<P>Satya</P><p><br><hr size=1>Do you Yahoo!?<br>
<a href="http://rd.yahoo.com/mail/mailsig/*http://mailplus.yahoo.com">Yahoo! Mail Plus</a> - Powerful. Affordable. <a href="http://rd.yahoo.com/mail/mailsig/*http://mailplus.yahoo.com">Sign up now</a>
--0-1231157336-1040696981=:68555--
--0-295618899-1040696981=:68555
Content-Type: text/plain; name="AudioDeviceSource.cpp"
Content-Description: AudioDeviceSource.cpp
Content-Disposition: inline; filename="AudioDeviceSource.cpp"

/**********
This library is free software; you can redistribute it and/or modify it under
the terms of the GNU Lesser General Public License as published by the
Free Software Foundation; either version 2.1 of the License, or (at your
option) any later version. (See <http://www.gnu.org/copyleft/lesser.html>.)

This library is distributed in the hope that it will be useful, but WITHOUT
ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS
FOR A PARTICULAR PURPOSE.  See the GNU Lesser General Public License for
more details.

You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
**********/
// "liveMedia"
// Copyright (c) 1996-2002 Live Networks, Inc.  All rights reserved.
// A template for a MediaSource encapsulating an audio/video input device
// Implementation


#include "AudioDeviceSource.hh"
#include "include/util.h"
#include "include/memory.h"
#include "include/debug.h"


#ifdef BSD
static struct timezone Idunno;
#else
static int Idunno;
#endif

#ifndef UNUSED
#define UNUSED(x) x=x
#endif
static HWAVEIN	shWaveIn;               /* Handle for wave input                                */
static WAVEHDR	*whReadHdrs;            /* Pointer to block of wavehdr's allocated for reading  */
static u_char	*lpReadData;            /* Pointer to raw audio data buffer                     */
static WAVEHDR  *whReadList;            /* List of wave headers that have been read but not     */
                                        /* given to the application.                            */ 
static DWORD     dwBytesUsedAtReadHead; /* Number of bytes that have already been read at head  */
static HANDLE    hAudioReady;           

static int blksz;
static int nblks;
static int smplsz;

 

AudioDeviceSource*
AudioDeviceSource::createNew(UsageEnvironment& env,
			unsigned int preferredFrameSize,
			unsigned int  playTimePerFrame) {
  return new AudioDeviceSource(env,  preferredFrameSize,
			 playTimePerFrame);
}

AudioDeviceSource::AudioDeviceSource(UsageEnvironment& env,
			   unsigned int preferredFrameSize,
					   unsigned int playTimePerFrame)
  : FramedSource(env), fPreferredFrameSize(preferredFrameSize),
    fPlayTimePerFrame(playTimePerFrame), fLastPlayTime(0)
 {
	nMixersWithFullDuplex = 0;

  // Any initialization of the device would be done here
	//Chek for all the devices
	audio_dev_init();
	//Fill the Formatex Structure for opening the device and Communicate

	WAVEFORMATEX wfx;
	

		wfx.wFormatTag      = WAVE_FORMAT_PCM;
        wfx.nChannels       = 1;
        wfx.nSamplesPerSec  = 64;
        wfx.wBitsPerSample  = 8000;
       int  smplsz          = wfx.wBitsPerSample / 8;
        wfx.nAvgBytesPerSec = wfx.nChannels * wfx.nSamplesPerSec * smplsz;
        wfx.nBlockAlign     = (WORD)(wfx.nChannels * smplsz);
        wfx.cbSize          = 0;
		audio_format af;
		af.bits_per_sample = 16;
        af.bytes_per_block = 320;
        af.channels        = 1;
        af.encoding        = DEV_S16;
        af.sample_rate     = 8000;
		for ( int i =0;i<nMixersWithFullDuplex ;i++)
		{
			w32sdk_audio_open(mixerIdMap[i],&af,&af);
		}
	//w32sdk_audio_open_in_probe(

	/*UINT  nDev = waveInGetNumDevs(); 
	//Take teh first device to get data 

	UINT_PTR     uDeviceID;
	LPWAVEINCAPS pwic;
	UINT         cbwic;
	//Get the Details of the Auidio Input device
	waveInGetDevCaps(WAVE_MAPPER, 
					 &pwic,
					 cbwic);
	

		wfx.wFormatTag      = WAVE_FORMAT_PCM;
        wfx.nChannels       = 2;
        wfx.nSamplesPerSec  = fParams.Samplerate;
        wfx.wBitsPerSample  = 8000;
       int  smplsz              = wfx.wBitsPerSample / 8;
        wfx.nAvgBytesPerSec = wfx.nChannels * wfx.nSamplesPerSec * smplsz;
        wfx.nBlockAlign     = (WORD)(wfx.nChannels * smplsz);
        wfx.cbSize          = 0;
		int nRes  = waveInOpen (ID,WAVE_MAPPER,&wfx, 0, 0,CALLBACK_NULL);
		*/
}

AudioDeviceSource::~AudioDeviceSource() {
}

float AudioDeviceSource::getPlayTime(unsigned numFrames) const {
  return 0.0; // Replace this with the actual playing time, in seconds
}

void AudioDeviceSource::doGetNextFrame() {

  // Arrange here for our "deliverFrame" member function to be called
  // when the next frame of data becomes available from the device.
  // This must be done in a non-blocking fashion - i.e., so that we
  // return immediately from this function even if no data is
  // currently available.
  //
  // If the device can be implemented as a readable socket, then one easy
  // way to do this is using a call to
  //     envir().taskScheduler().turnOnBackgroundReadHandling( ... )
  // (See examples of this call in the "liveMedia" directory.)
	

  // If, for some reason, the source device stops being readable
  // (e.g., it gets closed), then you do the following:
	if (hAudioReady)
	{

		deliverFrame();	
	}
	else
	{

	}
  if (0 /* the source stops being readable */) {
    handleClosure(this);
    return;
  }
}

void AudioDeviceSource::deliverFrame() {
  // This would be called when new frame data is available from the device.
  // This function should deliver the next frame of data from the device,
  // using the following parameters (class members):
  // fTo: The frame data is copied here
  // fMaxSize: This is the maximum number of bytes that can be copied
  //     (If the actual frame is larger than this, then it should
  //      be truncated.)
  // fFrameSize: The resulting frame size is copied here 
  // fPresentationTime: The frame's relative presentation time
  //     (seconds, microseconds) is copied here
	for ( int i =0;i<nMixersWithFullDuplex ;i++)
	{
		if ( w32sdk_audio_is_ready(mixerIdMap[i]))
			fFrameSize = w32sdk_audio_read(mixerIdMap[i],  fTo, fMaxSize);

	}
	if (fPlayTimePerFrame > 0 )
	{
		if (fPresentationTime.tv_sec == 0 && fPresentationTime.tv_usec == 0) 
		{     
			// This is the first frame, so use the current time:
			gettimeofday(&fPresentationTime, &Idunno);
		}
		else
		{
			  // Increment by the play time of the previous data:
			  unsigned uSeconds	= fPresentationTime.tv_usec + fLastPlayTime;
			  fPresentationTime.tv_sec += uSeconds/1000000;
			  fPresentationTime.tv_usec = uSeconds%1000000;
		}
	}
	else
	{
		gettimeofday(&fPresentationTime, &Idunno);
	}
	  // Deliver the data here:

  // After delivering the data, switch to another task, and inform
  // the reader that he has data:
  nextTask()
    = envir().taskScheduler().scheduleDelayedTask(0, (TaskFunc*)afterGetting,
						  this);
}


UINT 
AudioDeviceSource::mapAudioDescToMixerID(audio_desc_t ad)
{
        return mixerIdMap[ad];
}

 int AudioDeviceSource::audio_dev_init(void)
{
        audio_format af;
        unsigned int i;

        mixerIdMap = (UINT*)xmalloc(sizeof(UINT) * mixerGetNumDevs());

        af.bits_per_sample = 16;
        af.bytes_per_block = 320;
        af.channels        = 1;
        af.encoding        = DEV_S16;
        af.sample_rate     = 8000;
        
        for(i = 0; i < mixerGetNumDevs(); i++) {
                if (w32sdk_audio_open_mixer_probe(i, &af, &af, 1)) {
                        w32sdk_audio_close_mixer(i);
                        mixerIdMap[nMixersWithFullDuplex] = i;
                        nMixersWithFullDuplex++;
                }
        }

        return nMixersWithFullDuplex;
}





 int
AudioDeviceSource::w32sdk_audio_is_ready(audio_desc_t ad)
{
        UNUSED(ad);
        return (whReadList != NULL);
}

 void CALLBACK
AudioDeviceSource::waveInProc(HWAVEIN hwi,
           UINT    uMsg,
           DWORD   dwInstance,
           DWORD   dwParam1,
           DWORD   dwParam2)
{
        WAVEHDR *whRead, **whInsert;

        switch(uMsg) {
        case WIM_DATA:
                whRead = (WAVEHDR*)dwParam1;       
		/* Insert block at the available list */
                whRead->lpNext   = NULL;  
		whInsert = &whReadList;
                while(*whInsert != NULL) {
                        whInsert = &((*whInsert)->lpNext);
                }
                *whInsert = whRead;
                SetEvent(hAudioReady);
		break;
        default:
                ;  /* nothing to do currently */
        }
        UNUSED(dwInstance);
        UNUSED(dwParam2);
        UNUSED(hwi);
        
        return;
}

 int
AudioDeviceSource::w32sdk_audio_open_in_probe(UINT uId, WAVEFORMATEX *pwfx, int probe)
{
        MMRESULT mmr;
        int      i;
        
        if (shWaveIn) {
		return (TRUE);
	}
        
        if (lpReadData != NULL) {
		xfree(lpReadData);
	}
        lpReadData = (u_char*)xmalloc(nblks * blksz);
        memset(lpReadData, 0, nblks * blksz);

        if (whReadHdrs != NULL) {
		xfree(whReadHdrs);
	}
        whReadHdrs = (WAVEHDR*)xmalloc(sizeof(WAVEHDR)*nblks); 
        memset(whReadHdrs, 0, sizeof(WAVEHDR)*nblks); 

        mmr = waveInOpen(&shWaveIn, 
                         uId, 
                         pwfx,
                         (DWORD)waveInProc,
                         0,
                         (probe ? CALLBACK_FUNCTION/*CALLBACK_NULL*/: CALLBACK_FUNCTION));
        
	if (mmr != MMSYSERR_NOERROR) {
		char errorText[20]; 
                waveInGetErrorText(mmr, errorText, sizeof(errorText));
                debug_msg("waveInOpen: (%d) %s\n", mmr, errorText);
                return (FALSE);
        }

        if (!probe) {
          /* Initialize wave headers */
	  for (i = 0; i < nblks; i++) {
                whReadHdrs[i].lpData         = (char*)(lpReadData + i * blksz);
                whReadHdrs[i].dwBufferLength = blksz;
                whReadHdrs[i].dwFlags        = 0;
                mmr = waveInPrepareHeader(shWaveIn, &whReadHdrs[i], sizeof(WAVEHDR));
				assert(mmr == MMSYSERR_NOERROR);               
                mmr = waveInAddBuffer(shWaveIn, &whReadHdrs[i], sizeof(WAVEHDR));
                assert(mmr == MMSYSERR_NOERROR);
	  }

	  whReadList           = NULL;
	  dwBytesUsedAtReadHead = 0;
	  
	 int  error = waveInStart(shWaveIn);
	  if (error) {
		  	char errorText[20]; 
                waveInGetErrorText(error, errorText, sizeof(errorText));
                debug_msg("Win32Audio: waveInStart: (%d) %s\n", error, errorText);
                exit(1);
	  }
	  hAudioReady = CreateEvent(NULL, TRUE, FALSE, "Audio Ready");
        }

	return (TRUE);
}
int	 AudioDeviceSource::w32sdk_audio_open_in(UINT uId, WAVEFORMATEX *pwfx)
{
  return w32sdk_audio_open_in_probe(uId, pwfx, 0);
}

 void
AudioDeviceSource::w32sdk_audio_close_in()
{
        int		i;
        
        if (shWaveIn == 0)
                return;
        
        waveInStop(shWaveIn);
        waveInReset(shWaveIn);
        
        for (i = 0; i < nblks; i++) {
                if (whReadHdrs[i].dwFlags & WHDR_PREPARED) {
                        waveInUnprepareHeader(shWaveIn, &whReadHdrs[i], sizeof(WAVEHDR));
                }
        }
        whReadList = NULL;

        waveInClose(shWaveIn);           
	shWaveIn = 0;
        
	xfree(whReadHdrs);               
	whReadHdrs = NULL;
        
	xfree(lpReadData);               
	lpReadData  = NULL;
        
        xmemchk();
}

int
AudioDeviceSource::w32sdk_audio_read(audio_desc_t ad, u_char *buf, int buf_bytes)
{
        WAVEHDR *whCur;
	MMRESULT mmr;
        int done = 0, this_read;
        static int added;
        
	/* This is slightly ugle because we want to be able to operate when     */
        /* buf_bytes has any value, not just a multiple of blksz.  In principle */
        /* we do this so the device blksz does not have to match application    */
        /* blksz.  I.e. can reduce process usage by using larger blocks at      */
        /* device whilst the app operates on smaller blocks.                    */

        while(whReadList != NULL && done < buf_bytes) {
		whCur = whReadList;
		this_read = min((int)(whCur->dwBytesRecorded - dwBytesUsedAtReadHead), buf_bytes - done);
                if (buf) {
			memcpy(buf + done, 
			       whCur->lpData + dwBytesUsedAtReadHead,
			       this_read);
		}
                done                  += this_read;
                dwBytesUsedAtReadHead += this_read;
                if (dwBytesUsedAtReadHead == whCur->dwBytesRecorded) {
                        whReadList = whReadList->lpNext;
			/* Finished with the block give it device */
			assert(whCur->dwFlags & WHDR_DONE);
			assert(whCur->dwFlags & ~WHDR_INQUEUE);
                   	whCur->lpNext          = NULL; 
			whCur->dwBytesRecorded = 0; 
			whCur->dwFlags        &= ~WHDR_DONE;
			mmr = waveInAddBuffer(shWaveIn, whCur, sizeof(WAVEHDR)); 
            assert(mmr == MMSYSERR_NOERROR);
			assert(whCur->dwFlags & WHDR_INQUEUE);
                        dwBytesUsedAtReadHead = 0;
			added++;
                }
		assert((int)dwBytesUsedAtReadHead < blksz);
        }
	
        assert(done <= buf_bytes);
	UNUSED(ad);
        return done;
}

void
AudioDeviceSource::w32sdk_audio_drain(audio_desc_t ad)
{
	waveInStop(shWaveIn);
        w32sdk_audio_read(ad, NULL, 10000000);
	waveInStart(shWaveIn);
}

 void AudioDeviceSource::dumpReadHdrStats()
{
	WAVEHDR *whp;
	int i, done, inqueue, prepared, ready;
	
	done = inqueue = prepared = 0;
	for(i = 0; i < nblks; i++) {
		if (whReadHdrs[i].dwFlags & WHDR_DONE) {
			done++;
		}
		if (whReadHdrs[i].dwFlags & WHDR_INQUEUE) {
			inqueue++;
		}
		if (whReadHdrs[i].dwFlags & WHDR_PREPARED) {
			prepared++;
		}
	}

	ready = 0;
	whp = whReadList;
	while (whp != NULL) {
		ready++;
		whp = whp->lpNext;
	}
	debug_msg("done %d inqueue %d prepared %d ready %d\n",
		done, inqueue, prepared, ready);
}

void
AudioDeviceSource::w32sdk_audio_wait_for(audio_desc_t ad, int delay_ms)
{        
        if (whReadList == NULL) {
                DWORD dwRes;
		dwRes = WaitForSingleObject(hAudioReady, delay_ms);
		switch(dwRes) {
		case WAIT_TIMEOUT:
			debug_msg("No audio (%d ms wait timeout)\n", delay_ms);
			dumpReadHdrStats();
			break;
		case WAIT_FAILED:
			debug_msg("Wait failed (error %u)\n", GetLastError());
			break;
		}
                ResetEvent(hAudioReady);
        }
	waveInStart(shWaveIn);
        UNUSED(ad);
}




static int audio_dev_open = 0;

int
AudioDeviceSource::audio_format_match(audio_format *fmt1, audio_format *fmt2)
{
        return !memcmp(fmt1, fmt2, sizeof(audio_format));
}
int
AudioDeviceSource::w32sdk_audio_open_mixer_probe(audio_desc_t ad, audio_format *fmt, audio_format *ofmt, int probe)
{
        static int virgin;
        WAVEFORMATEX owfx, wfx;
        UINT uWavIn;

        if (audio_dev_open) {
                debug_msg("Device not closed! Fix immediately");
                w32sdk_audio_close(ad);
        }
        
        assert(audio_format_match(fmt, ofmt));
        if (fmt->encoding != DEV_S16) {
                return FALSE; /* Only support L16 for time being */
        }
        
        if (mixGetInputInfo(ad, &uWavIn, &dwRecLineID) != TRUE) {
                debug_msg("Could not get wave in or mixer destination for mix %u\n", ad);
                return FALSE;
        }

      /*  if (mixGetOutputInfo(ad, &uWavOut, &dwVolLineID) != TRUE) {
                debug_msg("Could not get wave out or mixer destination for mix %u\n", ad);
                return FALSE;
        }
       */
        if (mixSetup(ad) == FALSE) {
                return FALSE; /* Could not secure mixer */
        }

        mixSaveControls(ad, &control_list);

        wfx.wFormatTag      = WAVE_FORMAT_PCM;
        wfx.nChannels       = (WORD)fmt->channels;
        wfx.nSamplesPerSec  = fmt->sample_rate;
        wfx.wBitsPerSample  = (WORD)fmt->bits_per_sample;
        smplsz              = wfx.wBitsPerSample / 8;
        wfx.nAvgBytesPerSec = wfx.nChannels * wfx.nSamplesPerSec * smplsz;
        wfx.nBlockAlign     = (WORD)(wfx.nChannels * smplsz);
        wfx.cbSize          = 0;
        
        memcpy(&owfx, &wfx, sizeof(wfx));
        
        /* Use 1/8 sec device buffer */	
        blksz  = fmt->bytes_per_block;
        nblks  = (wfx.nAvgBytesPerSec / blksz) / 8;
        
        if (w32sdk_audio_open_in_probe(uWavIn, &wfx, probe) == FALSE){
                debug_msg("Open input failed\n");
                return FALSE;
        }
        
        assert(memcmp(&owfx, &wfx, sizeof(WAVEFORMATEX)) == 0);
        
        /*if (w32sdk_audio_open_out_probe(uWavOut, &wfx, probe) == FALSE) {
                debug_msg("Open output failed\n");
                w32sdk_audio_close_in();
                return FALSE;
        }*/

        /* because these get can corrupted... */
        assert(memcmp(&owfx, &wfx, sizeof(WAVEFORMATEX)) == 0);
    
        /* Set process priority as high as we can go without special permissions on */
        /* on NT.  Although this priority may seem anti-social, it's not that bad   */
        /* since we block whilst waiting for audio events.                          */
        SetPriorityClass(GetCurrentProcess(), HIGH_PRIORITY_CLASS);
        
        /* SetThreadPriority(GetCurrentThread(), THREAD_PRIORITY_HIGHEST);  */

        if (!have_probed[ad]) {
                have_probed[ad] = w32sdk_probe_formats(ad);
        }
 
        audio_dev_open = TRUE;
        return TRUE;
}
 int
AudioDeviceSource::w32sdk_audio_open_mixer(audio_desc_t ad, audio_format *fmt, audio_format *ofmt)
{
  return w32sdk_audio_open_mixer_probe(ad, fmt, ofmt, 0);
}

int 
AudioDeviceSource::w32sdk_audio_open(audio_desc_t ad, audio_format *ifmt, audio_format *ofmt)
{
        ad = mapAudioDescToMixerID(ad);
        return w32sdk_audio_open_mixer(ad, ifmt, ofmt);
}

void
AudioDeviceSource::w32sdk_audio_close_mixer(audio_desc_t ad)
{
        MMRESULT mmr;

        debug_msg("Closing input device.\n");
        w32sdk_audio_close_in();
        
        debug_msg("Closing output device.\n");
        //w32sdk_audio_close_out();
        
        if (input_ports != NULL) {
                xfree(input_ports);
                input_ports = NULL;
        }
        
        if (loop_ports != NULL) {
                xfree(loop_ports);
                loop_ports = NULL;
        }
        
        mixRestoreControls(ad, &control_list);
        mmr = mixerClose(hMixer); hMixer = 0;
        if (mmr != MMSYSERR_NOERROR) {
                debug_msg("mixerClose failed: %s\n", mixGetErrorText(mmr));
        }
        
        audio_dev_open = FALSE;
        UNUSED(ad);
}

void
AudioDeviceSource::w32sdk_audio_close(audio_desc_t ad)
{
        w32sdk_audio_close_mixer(ad);
}

int
AudioDeviceSource::w32sdk_audio_duplex(audio_desc_t ad)
{
        UNUSED(ad);
        return (TRUE);
}



/* mixGetInputInfo - attempt to find corresponding wavein index
* for mixer uMix and corresponding destination line of mixer.  
* Returns TRUE if successful.
*/

int AudioDeviceSource::mixGetInputInfo(UINT uMix, UINT *puWavIn, DWORD *pdwLineID)
{
        UINT i, nWavIn;
        MIXERLINE  ml;
        MMRESULT   mmr;
        WAVEINCAPS wic;
        MIXERCAPS  mc;
        
        mmr = mixerGetDevCaps(uMix, &mc, sizeof(mc));
        if (mmr != MMSYSERR_NOERROR) {
                debug_msg("mixerGetDevCaps: %s\n", mixGetErrorText(mmr));
                return FALSE;
        }
        
        nWavIn = waveInGetNumDevs();
        for(i = 0; i < nWavIn; i++) {
                mmr = waveInGetDevCaps(i, &wic, sizeof(wic));
                if (mmr != MMSYSERR_NOERROR) {
                        debug_msg("waveInGetDevCaps: %s\n", mixGetErrorText(mmr));
                        continue;
                }
                
                ml.cbStruct       = sizeof(ml);
                ml.Target.dwType  = MIXERLINE_TARGETTYPE_WAVEIN;
                strncpy(ml.Target.szPname, wic.szPname, MAXPNAMELEN);
                ml.Target.vDriverVersion = wic.vDriverVersion;
                ml.Target.wMid    = wic.wMid;
                ml.Target.wPid    = wic.wPid;
                
                mmr = mixerGetLineInfo((HMIXEROBJ)uMix, &ml, MIXER_OBJECTF_MIXER | MIXER_GETLINEINFOF_TARGETTYPE);
                if (mmr == MMSYSERR_NOERROR) {
                        *puWavIn          = i;
                        *pdwLineID = ml.dwLineID;
                        debug_msg("Input: %s(%d - %d)\n", ml.szName, ml.dwDestination, ml.dwLineID);
                        return TRUE;
                } else {
                        debug_msg("mixerGetLineInfo (ignore this error): %s\n", mixGetErrorText(mmr));
                }
        }
        return FALSE;
}

/* mixGetOutputInfo - attempt to find corresponding waveout index
 * and corresponding destination line of mixer.  Returns TRUE if
 * successful.
 *
int 
AudioDeviceSource::mixGetOutputInfo(UINT uMix, UINT *puWavOut, DWORD *pdwLineID)
{
        UINT i, nWavOut;
        MIXERLINE  ml;
        MMRESULT   mmr;
        WAVEOUTCAPS woc;
        MIXERCAPS  mc;
        
        mmr = mixerGetDevCaps(uMix, &mc, sizeof(mc));
        if (mmr != MMSYSERR_NOERROR) {
                debug_msg("mixerGetDevCaps: %s\n", mixGetErrorText(mmr));
                return FALSE;
        }
        
        nWavOut = waveOutGetNumDevs();
        for(i = 0; i < nWavOut; i++) {
                mmr = waveOutGetDevCaps(i, &woc, sizeof(woc));
                if (mmr != MMSYSERR_NOERROR) {
                        debug_msg("waveOutGetDevCaps: %s\n", mixGetErrorText(mmr));
                        continue;
                }
                ml.cbStruct       = sizeof(ml);
                ml.Target.dwType  = MIXERLINE_TARGETTYPE_WAVEOUT;
                strncpy(ml.Target.szPname, woc.szPname, MAXPNAMELEN);
                ml.Target.vDriverVersion = woc.vDriverVersion;
                ml.Target.wMid    = woc.wMid;
                ml.Target.wPid    = woc.wPid;
                
                mmr = mixerGetLineInfo((HMIXEROBJ)uMix, &ml, MIXER_OBJECTF_MIXER | MIXER_GETLINEINFOF_TARGETTYPE);
                if (mmr == MMSYSERR_NOERROR) {
                        *puWavOut  = i;
                        *pdwLineID = ml.dwLineID;
                        debug_msg("Output: %s(%d - %d)\n", ml.szName, ml.dwDestination, ml.dwLineID);
                        return TRUE;
                } 
        }
        return FALSE;
}*/
int 
AudioDeviceSource::mixSetup(UINT uMixer)
{
        MIXERCAPS mc;
        MMRESULT  res;
        
        if (hMixer)  {mixerClose(hMixer);  hMixer  = 0;}
        
        res = mixerOpen(&hMixer, uMixer, (unsigned long)NULL, (unsigned long)NULL, MIXER_OBJECTF_MIXER);
        if (res != MMSYSERR_NOERROR) {
                debug_msg("mixerOpen failed: %s\n", mixGetErrorText(res));
                return FALSE;
        }
        
        res = mixerGetDevCaps((UINT)hMixer, &mc, sizeof(mc));
        if (res != MMSYSERR_NOERROR) {
                debug_msg("mixerGetDevCaps failed: %s\n", mixGetErrorText(res));
                return FALSE;
        }
        
        if (mc.cDestinations < 2) {
                debug_msg("mixer does not have 2 destinations?\n");
                return FALSE;
        }
        
        if (input_ports != NULL) {
                xfree(input_ports);
                input_ports   = NULL;
                n_input_ports = 0;
        }
        
        n_input_ports = mixQueryControls((HMIXEROBJ)hMixer, dwRecLineID, &input_ports);
        debug_msg("Input ports %d\n", n_input_ports);
        if (n_input_ports == 0) {
                return FALSE;
        }
        
        make_microphone_first_port(input_ports, n_input_ports);

        if (loop_ports != NULL) {
                xfree(loop_ports);
                loop_ports   = NULL;
                n_loop_ports = 0;
        }
        
        n_loop_ports = mixQueryControls((HMIXEROBJ)hMixer, dwVolLineID, &loop_ports);
        debug_msg("Loop ports %d\n", n_loop_ports);
        if (n_loop_ports == 0) {
                return 0;
        }
        return TRUE;
}



void
AudioDeviceSource::mixSaveControls(UINT uMix, mcd_elem_t **pplist)
{
        MIXERLINE ml, sml;
        MIXERCAPS mc;
        MMRESULT  mmr;
        UINT i,j;
        
        mmr = mixerGetDevCaps(uMix, &mc, sizeof(mc));
        if (mmr != MMSYSERR_NOERROR) {
                debug_msg("mixerGetDevCaps: %s\n", mixGetErrorText(mmr));
                return;
        }
        
        for(i = 0; i < mc.cDestinations; i++) {
                memset(&ml, 0, sizeof(ml));
                ml.cbStruct      = sizeof(ml);
                ml.dwDestination = i;
                mmr = mixerGetLineInfo((HMIXEROBJ)uMix, &ml, MIXER_OBJECTF_MIXER | MIXER_GETLINEINFOF_DESTINATION); 
                if (mmr != MMSYSERR_NOERROR) {
                        debug_msg("mixerGetLineInfo: %s\n", mixGetErrorText(mmr));
                        continue;
                }
                mixSaveLine(uMix, &ml, pplist);
                for (j = 0; j < ml.cConnections; j++) {
                        memset(&sml, 0, sizeof(sml));
                        sml.cbStruct = sizeof(sml);
                        sml.dwSource = j;
                        mmr = mixerGetLineInfo((HMIXEROBJ)uMix, &sml, MIXER_OBJECTF_MIXER | MIXER_GETLINEINFOF_SOURCE); 
                        if (mmr != MMSYSERR_NOERROR) {
                                debug_msg("mixerGetLineInfo: %s\n", mixGetErrorText(mmr));
                                continue;
                        }
                        mixSaveLine(uMix, &sml, pplist);
                }
        }
}





/* mixGetOutputInfo - attempt to find corresponding waveout index
 * and corresponding destination line of mixer.  Returns TRUE if
 * successful.
 *
int 
AudioDeviceSource::mixGetOutputInfo(UINT uMix, UINT *puWavOut, DWORD *pdwLineID)
{
        UINT i, nWavOut;
        MIXERLINE  ml;
        MMRESULT   mmr;
        WAVEOUTCAPS woc;
        MIXERCAPS  mc;
        
        mmr = mixerGetDevCaps(uMix, &mc, sizeof(mc));
        if (mmr != MMSYSERR_NOERROR) {
                debug_msg("mixerGetDevCaps: %s\n", mixGetErrorText(mmr));
                return FALSE;
        }
        
        nWavOut = waveOutGetNumDevs();
        for(i = 0; i < nWavOut; i++) {
                mmr = waveOutGetDevCaps(i, &woc, sizeof(woc));
                if (mmr != MMSYSERR_NOERROR) {
                        debug_msg("waveOutGetDevCaps: %s\n", mixGetErrorText(mmr));
                        continue;
                }
                ml.cbStruct       = sizeof(ml);
                ml.Target.dwType  = MIXERLINE_TARGETTYPE_WAVEOUT;
                strncpy(ml.Target.szPname, woc.szPname, MAXPNAMELEN);
                ml.Target.vDriverVersion = woc.vDriverVersion;
                ml.Target.wMid    = woc.wMid;
                ml.Target.wPid    = woc.wPid;
                
                mmr = mixerGetLineInfo((HMIXEROBJ)uMix, &ml, MIXER_OBJECTF_MIXER | MIXER_GETLINEINFOF_TARGETTYPE);
                if (mmr == MMSYSERR_NOERROR) {
                        *puWavOut  = i;
                        *pdwLineID = ml.dwLineID;
                        debug_msg("Output: %s(%d - %d)\n", ml.szName, ml.dwDestination, ml.dwLineID);
                        return TRUE;
                } 
        }
        return FALSE;
}*/

/* mixerEnableInputLine - enables the input line whose name starts with beginning of portname.  
 * We cannot just use the port index like we do for volume because the mute controls are
 * not necessarily in the same order as the volume controls (grrr!).  The only card
 * that we have seen where this is necessary is the Winnov Videum AV, but there are
 * bound to be others.
 * Muting for input lines on the toplevel control (Rec, or whatever driver happens to call it).
 * It usually has a single control a MUX/Mixer that has "multiple items", one mute for
 * each input line.  Depending on the control type it may be legal to have multiple input
 * lines enabled, or just one.  So mixerEnableInputLine disables all lines other than
 * one selected.
 */
/*
 int
AudioDeviceSource::mixerEnableInputLine(HMIXEROBJ hMix, char *portname)
{
        MIXERCONTROLDETAILS_BOOLEAN *mcdbState;
        MIXERCONTROLDETAILS_LISTTEXT *mcdlText;
        MIXERCONTROLDETAILS mcd;
        MIXERLINECONTROLS mlc;
        MIXERCONTROL mc;
        MIXERLINE ml;
        MMRESULT  mmr;
        UINT      i, matchLine;
        
        ml.cbStruct = sizeof(ml);
        ml.dwLineID = dwRecLineID;
        
        mmr = mixerGetLineInfo(hMix, &ml, MIXER_GETLINEINFOF_LINEID|MIXER_OBJECTF_HMIXER);
        if (mmr != MMSYSERR_NOERROR) {
                debug_msg("mixerGetLineInfo: %s\n", mixGetErrorText(mmr));
        }
        
        /* Get Mixer/MUX control information (need control id to set and get control details) *******
        mlc.cbStruct      = sizeof(mlc);
        mlc.dwLineID      = ml.dwLineID;
        mlc.pamxctrl      = &mc;
        mlc.cbmxctrl      = sizeof(mc);
        
        mlc.dwControlType = MIXERCONTROL_CONTROLTYPE_MUX; /* Single Select *
        mmr = mixerGetLineControls(hMix, &mlc, MIXER_GETLINECONTROLSF_ONEBYTYPE|MIXER_OBJECTF_HMIXER);
        if (mmr != MMSYSERR_NOERROR) {
                mlc.dwControlType = MIXERCONTROL_CONTROLTYPE_MIXER; /* Multiple Select *
                mmr = mixerGetLineControls(hMix, &mlc, MIXER_GETLINECONTROLSF_ONEBYTYPE|MIXER_OBJECTF_HMIXER);
                if (mmr != MMSYSERR_NOERROR) {
                        debug_msg("mixerGetLineControls: %s\n", mixGetErrorText(mmr));
                        return FALSE;
                }
        }
        
        mcd.cbStruct    = sizeof(mcd);
        mcd.dwControlID = mc.dwControlID;
        mcd.cChannels   = 1;
        mcd.cMultipleItems = mc.cMultipleItems;
        mcdlText = (MIXERCONTROLDETAILS_LISTTEXT*)xmalloc(sizeof(MIXERCONTROLDETAILS_LISTTEXT)*mc.cMultipleItems);        
        mcd.paDetails = mcdlText;
        mcd.cbDetails = sizeof(MIXERCONTROLDETAILS_LISTTEXT);
        mmr = mixerGetControlDetails(hMix, &mcd, MIXER_GETCONTROLDETAILSF_LISTTEXT | MIXER_OBJECTF_MIXER);
        
        matchLine = 0;
        for(i = 0; i < mcd.cMultipleItems; i++) {
                if (!strcmp(mcdlText[i].szName, portname)) {
                        matchLine = i;
                        break;
                }
        }
        xfree(mcdlText);

        /* Now get control itself *
        mcd.cbStruct    = sizeof(mcd);
        mcd.dwControlID = mc.dwControlID;
        mcd.cChannels   = 1;
        mcd.cMultipleItems = mc.cMultipleItems;
        mcdbState = (MIXERCONTROLDETAILS_BOOLEAN*)xmalloc(sizeof(MIXERCONTROLDETAILS_BOOLEAN)*mc.cMultipleItems);        
        mcd.paDetails = mcdbState;
        mcd.cbDetails = sizeof(MIXERCONTROLDETAILS_BOOLEAN);
        
        mmr = mixerGetControlDetails(hMix, &mcd, MIXER_GETCONTROLDETAILSF_VALUE|MIXER_OBJECTF_MIXER);
        if (mmr != MMSYSERR_NOERROR) {
                debug_msg("mixerGetControlDetails: %s\n", mixGetErrorText(mmr));
                xfree(mcdbState);
                return FALSE;
        }
        
        for(i = 0; i < mcd.cMultipleItems; i++) {
                if (i == matchLine) {
                        mcdbState[i].fValue = TRUE;
                } else {
                        mcdbState[i].fValue = FALSE;
                }
        }
        
        mmr = mixerSetControlDetails(hMix, &mcd, MIXER_OBJECTF_MIXER);
        if (mmr != MMSYSERR_NOERROR) {
                debug_msg("mixerSetControlDetails: %s\n", mixGetErrorText(mmr));
                xfree(mcdbState);
                return FALSE;
        }
        
        xfree(mcdbState);
        return TRUE;
}

int
AudioDeviceSource::mixerEnableOutputLine(HMIXEROBJ hMix, DWORD dwLineID, int state)
{
        MIXERCONTROLDETAILS_BOOLEAN mcdbState;
        MIXERCONTROLDETAILS mcd;
        MIXERLINECONTROLS mlc;
        MIXERCONTROL      mc;
        MMRESULT          mmr;
        
        mlc.cbStruct      = sizeof(mlc);
        mlc.pamxctrl      = &mc;
        mlc.cbmxctrl      = sizeof(MIXERCONTROL);
        mlc.dwLineID      = dwLineID;
        mlc.dwControlType = MIXERCONTROL_CONTROLTYPE_MUTE;
        
        mmr = mixerGetLineControls(hMix, &mlc, MIXER_GETLINECONTROLSF_ONEBYTYPE | MIXER_OBJECTF_HMIXER);
        if (mmr != MMSYSERR_NOERROR) {
                mlc.cbStruct      = sizeof(mlc);
                mlc.pamxctrl      = &mc;
                mlc.cbmxctrl      = sizeof(MIXERCONTROL);
                mlc.dwLineID      = dwLineID;
                mlc.dwControlType = MIXERCONTROL_CONTROLTYPE_ONOFF;
                mmr = mixerGetLineControls(hMix, &mlc, MIXER_GETLINECONTROLSF_ONEBYTYPE | MIXER_OBJECTF_HMIXER);
                if (mmr != MMSYSERR_NOERROR) {
                        debug_msg("Could not get mute control for line 0x%08x: %s\n", 
                                dwLineID,
                                mixGetErrorText(mmr));
                        mixerDumpLineInfo(hMix, dwLineID);
                        return FALSE;
                }
        }
        
        mcd.cbStruct       = sizeof(mcd);
        mcd.dwControlID    = mc.dwControlID;
        mcd.cChannels      = 1;
        mcd.cMultipleItems = mc.cMultipleItems;
        mcd.cbDetails      = sizeof(MIXERCONTROLDETAILS_BOOLEAN);
        mcd.paDetails      = &mcdbState;
        mcdbState.fValue   = !((UINT)state);
        
        mmr = mixerSetControlDetails((HMIXEROBJ)hMix, &mcd, MIXER_OBJECTF_HMIXER);
        if (mmr != MMSYSERR_NOERROR) {
                debug_msg("Could not set mute state for line 0x%08x\n", dwLineID);
                return FALSE;
        }
        return TRUE;
}
*/
/* MixQueryControls: Get all line names and id's, fill into ppapd, and return number of lines 
static*/ int
AudioDeviceSource::mixQueryControls(HMIXEROBJ hMix, DWORD dwLineID, audio_port_details_t** ppapd)
{
        MIXERCAPS mc;
        MIXERLINE mlt, mlc;
        audio_port_details_t *papd;
        MMRESULT mmr;
        UINT     i;
        
        /* Videum bug work around - Videum does not fill in number of connections if
         * called using MIXER_GETLINEINFOF_LINEID.  This is the only driver with this problem, 
         * but this seems to work in all cases.
         */        
        mixerGetDevCaps((UINT)hMix, &mc, sizeof(mc));
        for(i = 0; i < mc.cDestinations; i++) {
                memset(&mlt, 0, sizeof(mlt));
                mlt.cbStruct = sizeof(mlt);
                mlt.dwDestination = i;
                mmr = mixerGetLineInfo(hMix, &mlt, MIXER_GETLINEINFOF_DESTINATION);
                if (mmr != MMSYSERR_NOERROR) {
                        debug_msg("mixerGetLineInfo: %s\n", mixGetErrorText(mmr));
                        continue;
                }
                if (mlt.dwLineID == dwLineID) {
                        break;
                }
        }
        
        papd = (audio_port_details_t*)xmalloc(sizeof(audio_port_details_t)*mlt.cConnections);
        if (papd == NULL) {
                return 0;
        }
        
        mixerDumpLineInfo((HMIXEROBJ)hMix, mlt.dwLineID);
        
        for(i = 0; i < mlt.cConnections; i++) {
                memcpy(&mlc, &mlt, sizeof(mlc));
                mlc.dwSource = i;
                mmr = mixerGetLineInfo((HMIXEROBJ)hMixer, &mlc, MIXER_GETLINEINFOF_SOURCE|MIXER_OBJECTF_HMIXER);
                if (mmr != MMSYSERR_NOERROR) {
                        xfree(papd);
                        return 0;
                }
                strncpy(papd[i].name, mlc.szName, AUDIO_PORT_NAME_LENGTH);
                papd[i].port = mlc.dwLineID;
        }
        
        *ppapd = papd;
        return (int)mlt.cConnections;
}

/* XXX make_microphone_first_port is a hack to make microphone 
 * the first (default) port.  Of course this only works for
 * english language drivers...
 */

/*static */int
AudioDeviceSource::make_microphone_first_port(audio_port_details_t *ports, int n_ports)
{
        audio_port_details_t tmp;
        int i;
        
        for(i = 1; i < n_ports; i++) {
                if (!strncasecmp("mic", ports[i].name, 3) ||
                    !strncasecmp("mik", ports[i].name, 3) /*||
					!strncasecmp("Stereo", ports[i].name, 6)*/) 
				{
                        memcpy(&tmp, ports + i, sizeof(tmp));
                        memcpy(ports + i , ports, sizeof(ports[0]));
                        memcpy(ports, &tmp, sizeof(ports[0]));
                        return TRUE;
                }
        }

        return FALSE;
}
void
AudioDeviceSource::mixRestoreControls(UINT uMix, mcd_elem_t **pplist)
{
        MIXERCONTROLDETAILS *pmcd;
        MMRESULT mmr;

        return;

        while((pmcd = mcd_elem_get_control(pplist)) != NULL) {
                mmr = mixerSetControlDetails((HMIXEROBJ)uMix, pmcd, MIXER_OBJECTF_MIXER);
                xfree(pmcd->paDetails);
                xfree(pmcd);
                if (mmr != MMSYSERR_NOERROR) {
                        debug_msg("mixerSetControlDetails: %s\n", mixGetErrorText(mmr));
                        continue;
                }
        }
        assert(*pplist == NULL);
}


void
AudioDeviceSource::mixSaveLine(UINT uMix, MIXERLINE *pml, mcd_elem_t **pplist)
{
        MIXERCONTROLDETAILS *pmcd;
        MIXERLINECONTROLS mlc;
        MIXERCONTROL     *pmc;
        MMRESULT          mmr;
        UINT              i;
        
        /* Retrieve control types */
        pmc = (MIXERCONTROL*)xmalloc(sizeof(MIXERCONTROL)*pml->cControls);
        
        mlc.cbStruct  = sizeof(mlc);
        mlc.dwLineID  = pml->dwLineID;
        mlc.cControls = pml->cControls;
        mlc.pamxctrl  = pmc;
        mlc.cbmxctrl  = sizeof(MIXERCONTROL);
        
        debug_msg("Saving %s\n", pml->szName);

        mmr = mixerGetLineControls((HMIXEROBJ)uMix, &mlc, MIXER_GETLINECONTROLSF_ALL | MIXER_OBJECTF_MIXER);
        if (mmr != MMSYSERR_NOERROR) {
                debug_msg("mixerGetLineControls: %s\n", mixGetErrorText(mmr));
                xfree(pmc);
                return;
        }
        
        for(i = 0; i < pml->cControls; i++) {
                DWORD itemCnt, itemSz;
                if (pmc[i].cMultipleItems == 0) {
                        itemCnt = 1;	
                } else {
                        itemCnt = pmc[i].cMultipleItems;
                } 
                
                switch(pmc[i].dwControlType & MIXERCONTROL_CT_UNITS_MASK) {
                        /* Our application on affects boolean types (mute, on/off) and unsigned (vol) */
                case MIXERCONTROL_CT_UNITS_BOOLEAN:
                        itemSz = sizeof(MIXERCONTROLDETAILS_BOOLEAN);
                        break;
                case MIXERCONTROL_CT_UNITS_UNSIGNED:
                        itemSz = sizeof(MIXERCONTROLDETAILS_UNSIGNED);
                        break;
                default:
                        debug_msg("not done %s\n", pmc[i].szName);
                        continue;
                }
                pmcd = (MIXERCONTROLDETAILS*)xmalloc(sizeof(MIXERCONTROLDETAILS));
                pmcd->cbStruct       = sizeof(MIXERCONTROLDETAILS);
                pmcd->cMultipleItems = pmc[i].cMultipleItems;
                pmcd->dwControlID    = pmc[i].dwControlID;
                pmcd->cChannels      = 1;
                pmcd->paDetails      = (void*)xmalloc(itemSz * itemCnt);
                pmcd->cbDetails      = itemSz;
                
                mmr = mixerGetControlDetails((HMIXEROBJ)uMix, pmcd, MIXER_GETCONTROLDETAILSF_VALUE | MIXER_OBJECTF_MIXER);
                if (mmr != MMSYSERR_NOERROR) {
                        debug_msg("mixerGetControlDetails: %s\n", mixGetErrorText(mmr));
                        continue;
                }
                mcd_elem_add_control(pplist, pmcd);
        }
        xfree(pmc);
}

const char *AudioDeviceSource::mixGetErrorText(MMRESULT mmr)
{
#ifndef NDEBUG
        switch (mmr) {
        CASE_STRING(MMSYSERR_NOERROR);     
        CASE_STRING(MIXERR_INVALLINE);     
        CASE_STRING(MIXERR_INVALCONTROL);  
        CASE_STRING(MIXERR_INVALVALUE);    
        CASE_STRING(WAVERR_BADFORMAT);     
        CASE_STRING(MMSYSERR_BADDEVICEID); 
        CASE_STRING(MMSYSERR_INVALFLAG);   
        CASE_STRING(MMSYSERR_INVALHANDLE); 
        CASE_STRING(MMSYSERR_INVALPARAM);  
        CASE_STRING(MMSYSERR_NODRIVER);
        default:
                return "Undefined Error";
        }
#endif /* NDEBUG */
        return "Mixer Error.";
}




void
AudioDeviceSource::mixerDumpLineInfo(HMIXEROBJ hMix, DWORD dwLineID)
{
        MIXERLINECONTROLS mlc;
        LPMIXERCONTROL    pmc;
        MIXERLINE ml;
        MMRESULT  mmr;
        UINT      i;
        
        /* Determine number of controls */
        ml.cbStruct = sizeof(ml);
        ml.dwLineID = dwLineID;
        
        mmr = mixerGetLineInfo((HMIXEROBJ)hMix, &ml, MIXER_GETLINEINFOF_LINEID | MIXER_OBJECTF_HMIXER);
        if (mmr != MMSYSERR_NOERROR) {
                debug_msg(mixGetErrorText(mmr));
                return;
        }
        
        pmc = (LPMIXERCONTROL)xmalloc(sizeof(MIXERCONTROL)*ml.cControls);
        mlc.cbStruct  = sizeof(MIXERLINECONTROLS);
        mlc.cbmxctrl  = sizeof(MIXERCONTROL);
        mlc.pamxctrl  = pmc;
        mlc.cControls = ml.cControls;
        mlc.dwLineID  = dwLineID;
        
        mmr = mixerGetLineControls((HMIXEROBJ)hMix, &mlc, MIXER_GETLINECONTROLSF_ALL | MIXER_OBJECTF_HMIXER);
        if (mmr != MMSYSERR_NOERROR) {
                debug_msg(mixGetErrorText(mmr));
                xfree(pmc);
                return;
        }
        
        for(i = 0; i < ml.cControls; i++) {
                debug_msg("- %u %s\t\t %s\n", i, pmc[i].szName, mixGetControlType(pmc[i].dwControlType));
        }
        xfree(pmc);
}

/* Code for saving control states when claiming device, so we can restore the 
 * config when we release the device.  Lots of request for this 
 */

int
AudioDeviceSource::mcd_elem_add_control(mcd_elem_t **pplist, MIXERCONTROLDETAILS *pmcd)
{
        mcd_elem_t *elem;
        
        elem = (mcd_elem_t*)xmalloc(sizeof(mcd_elem_t));
        if (elem) {
                elem->pmcd = pmcd;
                elem->next = *pplist;
                *pplist    = elem;
                return TRUE;
        }
        return FALSE;
}

MIXERCONTROLDETAILS*
AudioDeviceSource::mcd_elem_get_control(mcd_elem_t **pplist)
{
        MIXERCONTROLDETAILS *pmcd;
        mcd_elem_t *elem;
        
        elem = *pplist;
        if (elem) {
                pmcd    = elem->pmcd;
                *pplist = elem->next;
                xfree(elem);
                return pmcd;
        }
        return NULL;
}

const char *
AudioDeviceSource::mixGetControlType(DWORD dwCtlType)
{
#ifndef NDEBUG
        switch(dwCtlType) {
        CASE_STRING(MIXERCONTROL_CONTROLTYPE_CUSTOM);        
        CASE_STRING(MIXERCONTROL_CONTROLTYPE_BOOLEANMETER);  
        CASE_STRING(MIXERCONTROL_CONTROLTYPE_SIGNEDMETER);   
        CASE_STRING(MIXERCONTROL_CONTROLTYPE_PEAKMETER);     
        CASE_STRING(MIXERCONTROL_CONTROLTYPE_UNSIGNEDMETER); 
        CASE_STRING(MIXERCONTROL_CONTROLTYPE_BOOLEAN);       
        CASE_STRING(MIXERCONTROL_CONTROLTYPE_ONOFF);         
        CASE_STRING(MIXERCONTROL_CONTROLTYPE_MUTE);          
        CASE_STRING(MIXERCONTROL_CONTROLTYPE_MONO);          
        CASE_STRING(MIXERCONTROL_CONTROLTYPE_LOUDNESS);      
        CASE_STRING(MIXERCONTROL_CONTROLTYPE_STEREOENH);     
        CASE_STRING(MIXERCONTROL_CONTROLTYPE_BUTTON);        
        CASE_STRING(MIXERCONTROL_CONTROLTYPE_DECIBELS);      
        CASE_STRING(MIXERCONTROL_CONTROLTYPE_SIGNED);        
        CASE_STRING(MIXERCONTROL_CONTROLTYPE_UNSIGNED);      
        CASE_STRING(MIXERCONTROL_CONTROLTYPE_PERCENT);       
        CASE_STRING(MIXERCONTROL_CONTROLTYPE_SLIDER);        
        CASE_STRING(MIXERCONTROL_CONTROLTYPE_PAN);           
        CASE_STRING(MIXERCONTROL_CONTROLTYPE_QSOUNDPAN);     
        CASE_STRING(MIXERCONTROL_CONTROLTYPE_FADER);         
        CASE_STRING(MIXERCONTROL_CONTROLTYPE_VOLUME);        
        CASE_STRING(MIXERCONTROL_CONTROLTYPE_BASS);          
        CASE_STRING(MIXERCONTROL_CONTROLTYPE_TREBLE);        
        CASE_STRING(MIXERCONTROL_CONTROLTYPE_EQUALIZER);     
        CASE_STRING(MIXERCONTROL_CONTROLTYPE_SINGLESELECT);  
        CASE_STRING(MIXERCONTROL_CONTROLTYPE_MUX);           
        CASE_STRING(MIXERCONTROL_CONTROLTYPE_MULTIPLESELECT);
        CASE_STRING(MIXERCONTROL_CONTROLTYPE_MIXER);         
        CASE_STRING(MIXERCONTROL_CONTROLTYPE_MICROTIME);     
        CASE_STRING(MIXERCONTROL_CONTROLTYPE_MILLITIME);    
        }
#endif /* NDEBUG */
        return "Unknown";
}






int
AudioDeviceSource::w32sdk_probe_format(int rate, int channels)
{
        WAVEFORMATEX wfx;
        
        wfx.cbSize = 0; /* PCM format */
        wfx.wFormatTag      = WAVE_FORMAT_PCM;
        wfx.wBitsPerSample  = 16; /* 16 bit linear */
        wfx.nChannels       = channels;
        wfx.nSamplesPerSec  = rate;
        wfx.nBlockAlign     = wfx.wBitsPerSample / 8 * wfx.nChannels;
        wfx.nAvgBytesPerSec = wfx.nSamplesPerSec * wfx.nBlockAlign;
        
        if (waveInOpen(NULL, (UINT)shWaveIn, &wfx, (UINT)NULL, (UINT)NULL, WAVE_FORMAT_QUERY)) {
                debug_msg("%d %d supported\n", rate, channels);
                return TRUE;
        }
        
        debug_msg("%d %d not supported\n", rate, channels);
        return FALSE;      
}

int 
AudioDeviceSource::w32sdk_probe_formats(audio_desc_t ad) 
{
        int rate, channels;

        for (rate = 8000; rate <= 48000; rate+=8000) {
                if (rate == 24000 || rate == 40000) continue;
                for(channels = 1; channels <= 2; channels++) {
                        if (w32sdk_probe_format(rate, channels)) {
                                af_sup[ad][n_af_sup[ad]].sample_rate = rate;
                                af_sup[ad][n_af_sup[ad]].channels    = channels;
                                n_af_sup[ad]++;
                        }
                }
        }
        return (n_af_sup[ad] ? TRUE : FALSE); /* Managed to find at least 1 we support    */
                                              /* We have this test since if we cannot get */
                                              /* device now (because in use elsewhere)    */
                                              /* we will want to test it later            */
}

--0-295618899-1040696981=:68555
Content-Type: text/plain; name="AudioDeviceSource.hh"
Content-Description: AudioDeviceSource.hh
Content-Disposition: inline; filename="AudioDeviceSource.hh"

/**********
This library is free software; you can redistribute it and/or modify it under
the terms of the GNU Lesser General Public License as published by the
Free Software Foundation; either version 2.1 of the License, or (at your
option) any later version. (See <http://www.gnu.org/copyleft/lesser.html>.)

This library is distributed in the hope that it will be useful, but WITHOUT
ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS
FOR A PARTICULAR PURPOSE.  See the GNU Lesser General Public License for
more details.

You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
**********/
// "liveMedia"
// Copyright (c) 1996-2002 Live Networks, Inc.  All rights reserved.
// A template for a MediaSource encapsulating an audio/video input device
// C++ header


#include <windows.h>
#include <Mmsystem.h>

#include "GroupsockHelper.hh"



#ifndef _FRAMED_SOURCE_HH
#include "FramedSource.hh"
#endif

#ifdef NDEBUG
#define assert(x) if ((x) == 0) fprintf(stderr, "%s:%u: failed assertion\n", __FILE__, __LINE__)
#else
#include <assert.h>
#endif
typedef unsigned __int32	uint32_t;
typedef uint32_t audio_port_t;

#define AUDIO_PORT_NAME_LENGTH 20
#define CASE_STRING(x) case x: return #x
//for the String comparisons
#define strncasecmp strnicmp
#define strcasecmp stricmp


#define AUDIO_PORT_PHONE      "Phone"
#define AUDIO_PORT_SPEAKER    "Speaker"
#define AUDIO_PORT_HEADPHONE  "Headphone"
#define AUDIO_PORT_LINE_OUT   "Line-Out"
#define AUDIO_PORT_MICROPHONE "Microphone"
#define AUDIO_PORT_LINE_IN    "Line-In"
#define AUDIO_PORT_CD         "CD"

typedef struct {
        audio_port_t port;
        char         name[AUDIO_PORT_NAME_LENGTH + 1];
} audio_port_details_t;

#define W32SDK_MAX_DEVICES 20

typedef int audio_desc_t;

// The following class can be used to define specific encoder parameters
class AudioDeviceParameters {
public:
int Samplerate;
protected:
 UINT h_InDev;
 WAVEINCAPS wic;
 
};

typedef struct s_mcd_elem {
        MIXERCONTROLDETAILS *pmcd;
        struct s_mcd_elem   *next;
} mcd_elem_t;

//static mcd_elem_t *control_list;

typedef enum {
	DEV_PCMU, /* mu-law (8 bits) */
        DEV_PCMA, /*  a-law (8 bits) */
	DEV_S8,   /* signed 8 bits   */
        DEV_U8,   /* unsigned 8 bits */
	DEV_S16   /* signed 16 bits  */
} deve_e;

typedef struct s_audio_format {
  deve_e encoding;
  unsigned int     sample_rate; 	/* Should be one of 8000, 16000, 24000, 32000, 48000 	*/
  int    bits_per_sample;	/* Should be 8 or 16 					*/
  int    channels;  		/* Should be 1 or 2  					*/
  int    bytes_per_block;       /* size of unit we will read/write in 			*/
} audio_format;

class AudioDeviceSource: public FramedSource {
public:
  static AudioDeviceSource* createNew(UsageEnvironment& env,
				 unsigned int  preferredFrameSize=0,
			unsigned int playTimePerFrame=0);
  static void CALLBACK waveInProc(HWAVEIN hwi, UINT    uMsg,
								DWORD   dwInstance,
								DWORD   dwParam1,
								DWORD   dwParam2);
//Mixer funxons
	int w32sdk_audio_open_mixer_probe(audio_desc_t ad, audio_format *fmt, audio_format *ofmt, int probe);
	int w32sdk_audio_open_mixer(audio_desc_t ad, audio_format *fmt, audio_format *ofmt);
	
	void w32sdk_audio_close_mixer(audio_desc_t ad);
	void w32sdk_audio_close(audio_desc_t ad);
	int w32sdk_audio_duplex(audio_desc_t ad);
	int w32sdk_audio_open(audio_desc_t ad, audio_format *ifmt, audio_format *ofmt);
	int audio_dev_init(void);
	void w32sdk_audio_wait_for(audio_desc_t ad, int delay_ms);
	void w32sdk_audio_drain(audio_desc_t ad);
	int  w32sdk_audio_is_ready(audio_desc_t ad);
	void dumpReadHdrStats();
	/*global functions*/
	int  w32sdk_audio_read(audio_desc_t ad, unsigned char *buf, int buf_bytes);
	void w32sdk_audio_close_in();
	int  w32sdk_audio_open_in(UINT uId, WAVEFORMATEX *pwfx);
	int  w32sdk_audio_open_in_probe(UINT uId, WAVEFORMATEX *pwfx, int probe);
	int audio_format_match(audio_format *fmt1, audio_format *fmt2);
	UINT mapAudioDescToMixerID(audio_desc_t ad);
	int mixGetInputInfo(UINT uMix, UINT *puWavIn, DWORD *pdwLineID);
	int mixSetup(UINT uMixer);
	int mixQueryControls(HMIXEROBJ hMix, DWORD dwLineID, audio_port_details_t ** ppapd);
	void mixSaveControls(UINT uMix, mcd_elem_t **pplist);
	int make_microphone_first_port(audio_port_details_t *ports, int n_ports);
	void mixRestoreControls(UINT uMix, mcd_elem_t **pplist);
	void mixSaveLine(UINT uMix, MIXERLINE *pml, mcd_elem_t **pplist);
	void mixerDumpLineInfo(HMIXEROBJ hMix, DWORD dwLineID);
	MIXERCONTROLDETAILS* mcd_elem_get_control(mcd_elem_t **pplist);
	int mcd_elem_add_control(mcd_elem_t **pplist, MIXERCONTROLDETAILS *pmcd);
	int w32sdk_probe_formats(audio_desc_t ad) ;
	int w32sdk_probe_format(int rate, int channels);
	
	//Error mesages
	const char * mixGetErrorText(MMRESULT mmr);
	const char * mixGetControlType(DWORD dwCtlType);

	
protected:
  AudioDeviceSource(UsageEnvironment& env, unsigned int  preferredFrameSize,
		       unsigned int playTimePerFrame);
  // called only by createNew(), or by subclass constructors
  virtual ~AudioDeviceSource();
 /*static*/ int   nMixersWithFullDuplex ;
 /*static*/ UINT *mixerIdMap;
 /*static*/ DWORD    dwRecLineID, dwVolLineID; 
	/*static*/ mcd_elem_t *control_list;
	
	/*static*/ int have_probed[W32SDK_MAX_DEVICES];


	audio_port_details_t *input_ports, *loop_ports;
	int                   n_input_ports, n_loop_ports;
	struct timeval fPresentationTime; // of the start of the data
	unsigned fFrameSize;
	HMIXER   hMixer;
////////////////////////////////////////////
//Staically delcared n Rat funtions so 
//added to teh class
	
	audio_format af_sup[W32SDK_MAX_DEVICES][10];
	int          n_af_sup[W32SDK_MAX_DEVICES];



private:
  // redefined virtual functions:
  virtual void doGetNextFrame();
  virtual float getPlayTime(unsigned numFrames) const;

private:
  void deliverFrame();

private:
  AudioDeviceParameters fParams;
  unsigned fPreferredFrameSize;
  unsigned fPlayTimePerFrame;
  unsigned fLastPlayTime;
  audio_desc_t  openAd;
};



--0-295618899-1040696981=:68555--


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From <EMAIL-ADDRESS-HIDDEN> Tue Dec 24 01:13:15 2002
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Subject: Re: [Live-devel] Audio Input from mic
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X-Original-Date: Tue, 24 Dec 2002 01:04:41 -0800
Date: Tue, 24 Dec 2002 01:04:41 -0800

At 06:29 PM 12/23/02, you wrote:
>Please let me know where the fault is....if any


>void AudioDeviceSource::doGetNextFrame() {
>
>   // Arrange here for our "deliverFrame" member function to be called
>   // when the next frame of data becomes available from the device.
>   // This must be done in a non-blocking fashion - i.e., so that we
>   // return immediately from this function even if no data is
>   // currently available.
[...]
>         if (hAudioReady)
>         {
>
>                 deliverFrame();
>         }
>         else
>         {
>
>         }

This is wrong.  If no audio is immediately available when doGetNextFrame() 
is called, then it needs to arrange for "deliverFrame()" to be called 
later, when audio does become available.

I.e., you could call "deliverFrame()" from the callback function.  Make 
sure, though, that "deliverFrame()" gets called exactly as many times as 
"doGetFrame()" got called.

         Ross.



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--0-1484939309-1040723132=:29645
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Hi Ross,
 I only send data when the there  is  in the Header list. But if need to call from the Call back function the it need to make the teh function asnon static and that is supported in the waveInProc(procedure) . Is there a better way than that
COming to other part of the source is that OK
Thanks for the help
satya



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<P>Hi Ross,
<P>&nbsp;I only send data when the there&nbsp; is&nbsp; in the Header list. But if need to call from the Call back function the it need to make the teh function asnon static and that is supported in the waveInProc(procedure) . Is there a better way than that
<P>COming to other part of the source is that OK
<P>Thanks for the help
<P>satya</P><p><br><hr size=1>Do you Yahoo!?<br>
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X-Original-Date: Tue, 24 Dec 2002 02:12:34 -0800
Date: Tue, 24 Dec 2002 02:12:34 -0800

One other thing you can do is poll (within a delay loop) in 
"doGetNextFrame().  If you do this, you won't need to have the callback 
function invoke any of the LIVE.COM C++ code.  For example

class AudioInputDevice ... {
	...
	unsigned fTotalPollingDelay;
	static void audioReadyPoller(void* clientData);
	void audioReadyPoller1();
	...
};

void AudioInputDevice::doGetNextFrame() {
	fTotalPollingDelay = 0;
	audioReadyPoller1();
}

void AudioInputDevice::audioReadyPoller(void* clientData) {
	AudioInputDevice* inputDevice = (AudioInputDevice*)clientData;
	inputDevice->audioReadyPoller1();
}

void AudioInputDevice::audioReadyPoller1() {
	if (/*audio is available*/) {
		deliverFrame();
	} else {
		if (fTotalPollingDelay > 100000 /* 0.1 seconds */) {
			// We've waited too long for the audio device
			// - assume it's down
			handleClosure(this);
			return;
		}
		// Try again after a short delay:
		unsigned const uSecondsToDelay = 10000;
		fTotalPollingDelay += uSecondsToDelay;
		nextTask() = envir().taskScheduler()
			.scheduleDelayedTask(uSecondsToDelay,
					(TaskFunc*)audioReadyPoller, this);
	}
}

Note that the 'delay' *must* be done using the LIVE.COM scheduler's 
"scheduleDelayedTask()" operation.  Don't use the Windoze Sleep() function 
(or anything like that), otherwise other parts of the program won't get 
executed during the delay period.

	Ross.



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Hi Ross,
Thanx Ross for the Code snippet.
  1.i have changed the code  as u have suggested.   The code here is playing football here. IT just gives access voilation closing the handle. even efore the data is written.
2. I'm able to send the data. i could see that . and also receing teh data  at the other end ie client end  but after some delay more than a minute. there i sno network congestion  as i was teh only user at that time (lan Speed i s 10mbps).
3.Even after receivng the data teh Quick player is able to play the data. how to write sample similar to testmp3receiver.
Thanks
Satya
 
 
.
 
void AudioInputDevice::audioReadyPoller1() {
if (/*audio is available*/) {
deliverFrame();
} else {
if (fTotalPollingDelay > 100000 /* 0.1 seconds */) {
// We've waited too long for the audio device
// - assume it's down
handleClosure(this);
return;
}




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--0-926292858-1040908239=:10978
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<P>Hi Ross,
<P>Thanx Ross for the Code snippet.
<P>&nbsp; 1.i have changed the code&nbsp; as u have suggested.&nbsp;&nbsp; The code here is playing football here. IT just gives access voilation&nbsp;closing the handle. even efore the data is written.
<P>2. I'm able to send the data. i could see that&nbsp;. and also receing teh data&nbsp; at the other end ie client end&nbsp; but after some delay more than a minute. there i sno network congestion&nbsp; as i was teh only user at that time (lan Speed i s 10mbps).
<P>3.Even after receivng the data teh Quick player is able to play the data. how to write sample similar to testmp3receiver.
<P>Thanks
<P>Satya
<P>&nbsp;
<P>&nbsp;
<P>.
<P>&nbsp;
<P>void AudioInputDevice::audioReadyPoller1() {<BR>if (/*audio is available*/) {<BR>deliverFrame();<BR>} else {<BR>if (fTotalPollingDelay &gt; 100000 /* 0.1 seconds */) {<BR>// We've waited too long for the audio device<BR>// - assume it's down<BR>handleClosure(this);<BR>return;<BR>}<BR></P><p><br><hr size=1>Do you Yahoo!?<br>
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Hi Ross 
This is the error i get when I try to handle the Close using th HandleClosure() function is called. 
any Insight  Please
Satya
_free_dbg_lk(void * 0x00794240, int 1) line 1017 + 82 bytes
_free_dbg(void * 0x00794240, int 1) line 970 + 13 bytes
operator delete(void * 0x00794240) line 351 + 12 bytes
MICTEST! BasicHashTable::deleteKey(class BasicHashTable::TableEntry *) + 48 bytes
MICTEST! BasicHashTable::deleteEntry(unsigned int,class BasicHashTable::TableEntry *) + 220 bytes
MICTEST! BasicHashTable::Remove(char const *) + 54 bytes
MediaLookupTable::remove(const char * 0x00eb2d68) line 132
Medium::close(Medium * 0x00eb2d60) line 80 + 19 bytes
CMictestDlg::afterPlaying(void * 0x00000000) line 263 + 12 bytes
MediaSink::onSourceClosure(void * 0x00eb2d60) line 91 + 13 bytes
MultiFramedRTPSink::sendPacketIfNecessary() line 287 + 9 bytes
MultiFramedRTPSink::ourHandleClosure(void * 0x00eb2d60) line 336
FramedSource::handleClosure(void * 0x00da4684) line 96 + 13 bytes
AudioDeviceSource::audioReadyPoller1() line 171 + 9 bytes
AudioDeviceSource::doGetNextFrame() line 152

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Today's Topics:

1. Re: Live-devel digest, Vol 1 #303 - 3 msgs (satya vangara)

--__--__--

Message: 1
Date: Thu, 26 Dec 2002 05:10:39 -0800 (PST)
From: satya vangara
To: <EMAIL-ADDRESS-HIDDEN>
Subject: [Live-devel] Re: Live-devel digest, Vol 1 #303 - 3 msgs
Reply-To: <EMAIL-ADDRESS-HIDDEN>

--0-926292858-1040908239=:10978
Content-Type: text/plain; charset=us-ascii


Hi Ross,
Thanx Ross for the Code snippet.
1.i have changed the code as u have suggested. The code here is playing football here. IT just gives access voilation closing the handle. even efore the data is written.
2. I'm able to send the data. i could see that . and also receing teh data at the other end ie client end but after some delay more than a minute. there i sno network congestion as i was teh only user at that time (lan Speed i s 10mbps).
3.Even after receivng the data teh Quick player is able to play the data. how to write sample similar to testmp3receiver.
Thanks
Satya


.

void AudioInputDevice::audioReadyPoller1() {
if (/*audio is available*/) {
deliverFrame();
} else {
if (fTotalPollingDelay > 100000 /* 0.1 seconds */) {
// We've waited too long for the audio device
// - assume it's down
handleClosure(this);
return;
}




---------------------------------
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--0-926292858-1040908239=:10978
Content-Type: text/html; charset=us-ascii


Hi Ross,

Thanx Ross for the Code snippet.

  1.i have changed the code  as u have suggested.   The code here is playing football here. IT just gives access voilation closing the handle. even efore the data is written.

2. I'm able to send the data. i could see that . and also receing teh data  at the other end ie client end  but after some delay more than a minute. there i sno network congestion  as i was teh only user at that time (lan Speed i s 10mbps).

3.Even after receivng the data teh Quick player is able to play the data. how to write sample similar to testmp3receiver.

Thanks

Satya


 

 
.


 
void AudioInputDevice::audioReadyPoller1() {
if (/*audio is available*/) {
deliverFrame();
} else {
if (fTotalPollingDelay > 100000 /* 0.1 seconds */) {
// We've waited too long for the audio device
// - assume it's down
handleClosure(this);
return;
}




---------------------------------
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--0-1962078051-1040984128=:14582
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<P>Hi Ross 
<P>This is the error i get when I try to handle the Close using th HandleClosure() function is called. 
<P>any Insight&nbsp; Please
<P>Satya
<P>_free_dbg_lk(void * 0x00794240, int 1) line 1017 + 82 bytes<BR>_free_dbg(void * 0x00794240, int 1) line 970 + 13 bytes<BR>operator delete(void * 0x00794240) line 351 + 12 bytes<BR>MICTEST! BasicHashTable::deleteKey(class BasicHashTable::TableEntry *) + 48 bytes<BR>MICTEST! BasicHashTable::deleteEntry(unsigned int,class BasicHashTable::TableEntry *) + 220 bytes<BR>MICTEST! BasicHashTable::Remove(char const *) + 54 bytes<BR>MediaLookupTable::remove(const char * 0x00eb2d68) line 132<BR>Medium::close(Medium * 0x00eb2d60) line 80 + 19 bytes<BR>CMictestDlg::afterPlaying(void * 0x00000000) line 263 + 12 bytes<BR>MediaSink::onSourceClosure(void * 0x00eb2d60) line 91 + 13 bytes<BR>MultiFramedRTPSink::sendPacketIfNecessary() line 287 + 9 bytes<BR>MultiFramedRTPSink::ourHandleClosure(void * 0x00eb2d60) line 336<BR>FramedSource::handleClosure(void * 0x00da4684) line 96 + 13 bytes<BR>AudioDeviceSource::audioReadyPoller1() line 171 + 9 bytes<BR>AudioDeviceSource::doGetNextFrame() line 152<BR>
<P>&nbsp;<B><I>live-devel-request(at)lists.sourceforge.net</I></B> wrote:
<BLOCKQUOTE style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #1010ff 2px solid">Send Live-devel mailing list submissions to<BR>live-devel(at)lists.sourceforge.net<BR><BR>To subscribe or unsubscribe via the World Wide Web, visit<BR>https://lists.sourceforge.net/lists/listinfo/live-devel<BR>or, via email, send a message with subject or body 'help' to<BR>live-devel-request(at)lists.sourceforge.net<BR><BR>You can reach the person managing the list at<BR>live-devel-admin(at)lists.sourceforge.net<BR><BR>When replying, please edit your Subject line so it is more specific<BR>than "Re: Contents of Live-devel digest..."<BR><BR><BR>Today's Topics:<BR><BR>1. Re: Live-devel digest, Vol 1 #303 - 3 msgs (satya vangara)<BR><BR>--__--__--<BR><BR>Message: 1<BR>Date: Thu, 26 Dec 2002 05:10:39 -0800 (PST)<BR>From: satya vangara <SATPAD2002(at)YAHOO.COM><BR>To: live-devel(at)lists.sourceforge.net<BR>Subject: [Live-devel] Re: Live-devel digest, Vol 1 #303 - 3 msgs<BR>Reply-To: live-devel(at)lists.sourceforge.net<BR><BR>--0-926292858-1040908239=:10978<BR>Content-Type: text/plain; charset=us-ascii<BR><BR><BR>Hi Ross,<BR>Thanx Ross for the Code snippet.<BR>1.i have changed the code as u have suggested. The code here is playing football here. IT just gives access voilation closing the handle. even efore the data is written.<BR>2. I'm able to send the data. i could see that . and also receing teh data at the other end ie client end but after some delay more than a minute. there i sno network congestion as i was teh only user at that time (lan Speed i s 10mbps).<BR>3.Even after receivng the data teh Quick player is able to play the data. how to write sample similar to testmp3receiver.<BR>Thanks<BR>Satya<BR><BR><BR>.<BR><BR>void AudioInputDevice::audioReadyPoller1() {<BR>if (/*audio is available*/) {<BR>deliverFrame();<BR>} else {<BR>if (fTotalPollingDelay &gt; 100000 /* 0.1 seconds */) {<BR>// We've waited too long for the audio device<BR>// - assume it's down<BR>handleClosure(this);<BR>return;<BR>}<BR><BR><BR><BR><BR>-------------------------------
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<P>Hi Ross,<BR>
<P>Thanx Ross for the Code snippet.<BR>
<P>&nbsp; 1.i have changed the code&nbsp; as u have suggested.&nbsp;&nbsp; The code here is playing football here. IT just gives access voilation&nbsp;closing the handle. even efore the data is written.<BR>
<P>2. I'm able to send the data. i could see that&nbsp;. and also receing teh data&nbsp; at the other end ie client end&nbsp; but after some delay more than a minute. there i sno network congestion&nbsp; as i was teh only user at that time (lan Speed i s 10mbps).<BR>
<P>3.Even after receivng the data teh Quick player is able to play the data. how to write sample similar to testmp3receiver.<BR>
<P>Thanks<BR>
<P>Satya<BR>
<P><BR>&nbsp;
<P><BR>&nbsp;
<P>.<BR>
<P><BR>&nbsp;
<P>void AudioInputDevice::audioReadyPoller1() {<BR>if (/*audio is available*/) {<BR>deliverFrame();<BR>} else {<BR>if (fTotalPollingDelay &gt; 100000 /* 0.1 seconds */) {<BR>// We've waited too long for the audio device<BR>// - assume it's down<BR>handleClosure(this);<BR>return;<BR>}<BR></P>
<P><BR>
<HR SIZE=1>
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X-Original-Date: Fri, 27 Dec 2002 16:59:49 -0800
Date: Fri, 27 Dec 2002 16:59:49 -0800

At 02:15 AM 12/27/02, you wrote:

>Hi Ross
>
>This is the error i get when I try to handle the Close using th 
>HandleClosure() function is called.
>
>any Insight  Please

Satya,

I suspect that Medium::close() is being called on the same object more than 
once (in which case the crash would be happening the second time it's called).

So, I suggest that you check whether you're calling handleClosure() more 
than once on your AudioDeviceSource object.

         Ross.




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Thanks Ross,
  As I was discussing with regarding the playback of  the audio that was transmitted . How do I do this?  In Live.Com integration of mplayer is given but the  instructions to build are only for LInux system. So How do i play back teh audio input from microphone. Quicktime-6 is not playing the transmitted data.  But I can see it buffering..
1. How to capture the data in to  a file . i have used File Sink  and playbacked audio using another program hwich takes buffer and puts in WAVEFORMATEX header and calls out waveOutOpen. But this cannot be done at runtime write into file and read from file. Which I want to avoid. I want to playback teh audio realtime using program. i cannot use it as command line. The Buffer details i sthat I Get Teh buffer in WAVEFORMATEX struct na dcopy teh data from teh structture t to fTO and the no of bytes filled from  the struct  to fMaximum and transmitted using th RTP and to play do the reverse copy the buffer to struct and play. i cannot send the dat in teh struct format becos of teh output device.  For doing realtime  i can use the a buffer back up and write out  to  playback. But I couldnot find how to capture the buffer.
2. I could not get access to the documentation.
 
Thnx
Satya
 live-devel-request(at)lists.sourceforge.net wrote:Send Live-devel mailing list submissions to
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Today's Topics:

1. Re: Live-devel digest, Vol 1 #303 - 3 msgs (satya vangara)

--__--__--

Message: 1
Date: Thu, 26 Dec 2002 05:10:39 -0800 (PST)
From: satya vangara
To: <EMAIL-ADDRESS-HIDDEN>
Subject: [Live-devel] Re: Live-devel digest, Vol 1 #303 - 3 msgs
Reply-To: <EMAIL-ADDRESS-HIDDEN>

--0-926292858-1040908239=:10978
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Hi Ross,
Thanx Ross for the Code snippet.
1.i have changed the code as u have suggested. The code here is playing football here. IT just gives access voilation closing the handle. even efore the data is written.
2. I'm able to send the data. i could see that . and also receing teh data at the other end ie client end but after some delay more than a minute. there i sno network congestion as i was teh only user at that time (lan Speed i s 10mbps).
3.Even after receivng the data teh Quick player is able to play the data. how to write sample similar to testmp3receiver.
Thanks
Satya


.

void AudioInputDevice::audioReadyPoller1() {
if (/*audio is available*/) {
deliverFrame();
} else {
if (fTotalPollingDelay > 100000 /* 0.1 seconds */) {
// We've waited too long for the audio device
// - assume it's down
handleClosure(this);
return;
}




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Hi Ross,

Thanx Ross for the Code snippet.

  1.i have changed the code  as u have suggested.   The code here is playing football here. IT just gives access voilation closing the handle. even efore the data is written.

2. I'm able to send the data. i could see that . and also receing teh data  at the other end ie client end  but after some delay more than a minute. there i sno network congestion  as i was teh only user at that time (lan Speed i s 10mbps).

3.Even after receivng the data teh Quick player is able to play the data. how to write sample similar to testmp3receiver.

Thanks

Satya


 

 
.


 
void AudioInputDevice::audioReadyPoller1() {
if (/*audio is available*/) {
deliverFrame();
} else {
if (fTotalPollingDelay > 100000 /* 0.1 seconds */) {
// We've waited too long for the audio device
// - assume it's down
handleClosure(this);
return;
}




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<P>Thanks Ross,
<P>&nbsp; As I was discussing with regarding the playback of&nbsp;&nbsp;the audio that was transmitted . How do I do this?&nbsp; In Live.Com integration of mplayer is&nbsp;given but the&nbsp; instructions to build are only for LInux system. So How do i play back teh audio input from microphone. Quicktime-6 is not playing the transmitted data.&nbsp; But I can see it buffering..
<P>1. How to capture the data in to&nbsp; a file . i have used File Sink&nbsp; and playbacked audio&nbsp;using another program hwich takes buffer and puts in WAVEFORMATEX header and calls out waveOutOpen. But this cannot be done at runtime write into file and read from file. Which I want to avoid. I want to playback teh audio realtime using program. i cannot use it as command line. The Buffer details i sthat I Get Teh buffer in WAVEFORMATEX struct na dcopy teh data from teh structture t to fTO and the no of bytes filled from&nbsp;&nbsp;the struct&nbsp; to fMaximum and transmitted using th RTP and to play do the reverse copy the buffer to struct and play. i cannot send the dat in teh struct format becos of teh output device.&nbsp; For doing realtime&nbsp; i can use the a buffer back up and&nbsp;write out&nbsp; to &nbsp;playback. But I couldnot find how to capture the buffer.
<P>2.&nbsp;I could not get access to the documentation.
<P>&nbsp;
<P>Thnx
<P>Satya
<P>&nbsp;<B><I>live-devel-request(at)lists.sourceforge.net</I></B> wrote:
<BLOCKQUOTE style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #1010ff 2px solid">Send Live-devel mailing list submissions to<BR>live-devel(at)lists.sourceforge.net<BR><BR>To subscribe or unsubscribe via the World Wide Web, visit<BR>https://lists.sourceforge.net/lists/listinfo/live-devel<BR>or, via email, send a message with subject or body 'help' to<BR>live-devel-request(at)lists.sourceforge.net<BR><BR>You can reach the person managing the list at<BR>live-devel-admin(at)lists.sourceforge.net<BR><BR>When replying, please edit your Subject line so it is more specific<BR>than "Re: Contents of Live-devel digest..."<BR><BR><BR>Today's Topics:<BR><BR>1. Re: Live-devel digest, Vol 1 #303 - 3 msgs (satya vangara)<BR><BR>--__--__--<BR><BR>Message: 1<BR>Date: Thu, 26 Dec 2002 05:10:39 -0800 (PST)<BR>From: satya vangara <SATPAD2002(at)YAHOO.COM><BR>To: live-devel(at)lists.sourceforge.net<BR>Subject: [Live-devel] Re: Live-devel digest, Vol 1 #303 - 3 msgs<BR>Reply-To: live-devel(at)lists.sourceforge.net<BR><BR>--0-926292858-1040908239=:10978<BR>Content-Type: text/plain; charset=us-ascii<BR><BR><BR>Hi Ross,<BR>Thanx Ross for the Code snippet.<BR>1.i have changed the code as u have suggested. The code here is playing football here. IT just gives access voilation closing the handle. even efore the data is written.<BR>2. I'm able to send the data. i could see that . and also receing teh data at the other end ie client end but after some delay more than a minute. there i sno network congestion as i was teh only user at that time (lan Speed i s 10mbps).<BR>3.Even after receivng the data teh Quick player is able to play the data. how to write sample similar to testmp3receiver.<BR>Thanks<BR>Satya<BR><BR><BR>.<BR><BR>void AudioInputDevice::audioReadyPoller1() {<BR>if (/*audio is available*/) {<BR>deliverFrame();<BR>} else {<BR>if (fTotalPollingDelay &gt; 100000 /* 0.1 seconds */) {<BR>// We've waited too long for the audio device<BR>// - assume it's down<BR>handleClosure(this);<BR>return;<BR>}<BR><BR><BR><BR><BR>-------------------------------
--<BR>Do you Yahoo!?<BR>Yahoo! Mail Plus - Powerful. Affordable. Sign up now<BR>--0-926292858-1040908239=:10978<BR>Content-Type: text/html; charset=us-ascii<BR><BR>
<P>Hi Ross,<BR>
<P>Thanx Ross for the Code snippet.<BR>
<P>&nbsp; 1.i have changed the code&nbsp; as u have suggested.&nbsp;&nbsp; The code here is playing football here. IT just gives access voilation&nbsp;closing the handle. even efore the data is written.<BR>
<P>2. I'm able to send the data. i could see that&nbsp;. and also receing teh data&nbsp; at the other end ie client end&nbsp; but after some delay more than a minute. there i sno network congestion&nbsp; as i was teh only user at that time (lan Speed i s 10mbps).<BR>
<P>3.Even after receivng the data teh Quick player is able to play the data. how to write sample similar to testmp3receiver.<BR>
<P>Thanks<BR>
<P>Satya<BR>
<P><BR>&nbsp;
<P><BR>&nbsp;
<P>.<BR>
<P><BR>&nbsp;
<P>void AudioInputDevice::audioReadyPoller1() {<BR>if (/*audio is available*/) {<BR>deliverFrame();<BR>} else {<BR>if (fTotalPollingDelay &gt; 100000 /* 0.1 seconds */) {<BR>// We've waited too long for the audio device<BR>// - assume it's down<BR>handleClosure(this);<BR>return;<BR>}<BR></P>
<P><BR>
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>   As I was discussing with regarding the playback of  the audio that was 
> transmitted . How do I do this?  In Live.Com integration of mplayer is 
> given but the  instructions to build are only for LInux system. So How do 
> i play back teh audio input from microphone. Quicktime-6 is not playing 
> the transmitted data.  But I can see it buffering..

Are you trying to play the stream by having QuickTime Player read a ".sdp" 
file?  If so, make sure that the payload format number is correct, and 
matches the payload format number that you used when you created your 
"SimpleRTPSink" (to stream the audio).

In particular, the payload format number (which follows "RTP/AVP" on the 
"m=" line) must be 0 (for PCM u-law), or 8 (for PCM a-law).

         Ross.



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stop funking bugging me ass holes take me off your fucken list


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At 02:43 PM 12/30/02, you wrote:
>stop funking bugging me ass holes take me off your fucken list

Stop being a moron.  You're on this mailing list by choice.  *You* were the 
person who put yourself on the list.

If you wish to unsubscribe, you can do so from this web page
http://lists.sourceforge.net/lists/options/live-devel/emostarkd--at--netscape.net

         Ross.



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X-Original-Date: Wed, 01 Jan 2003 22:23:54 -0500
Date: Wed, 01 Jan 2003 22:23:54 -0500

Hi Ross,

Happy New Year!

Got a couple of question on the testMPEGAudioVideoStreamer.cpp...

Am I right to say that synchronisation for Audio and Video for the testMPEGAudioVideoStreamer program DEPENDS on the getPlayTime() function inside the MPEGAudioStreamFramer and MPEGVideoStreamFramer class? if my calculations of the audio and video play time is wrong, will the audio and video become out of sync?

Right now, the project I'm working on involves capturing from 2 DIFFERENT video(webcam) and audio(mic) source, encoding the captured data (using "MPEG4-simple profile" for the video source and "MP3" for the audio source) and transmiting using RTP... 

Hence, if my calculation of the getPlayTime() is wrong, will the audio video become out of sync?

When I used the original MPEGAudioStreamFramer::getPlayTime(), it seems that the delay at the receiver side becomes longer and longer... What I did to solve this problem was just to return a fraction of the original getPlayTime, i.e. instead of 

return numFrames*fpt;

I returned numFrames*fpt*0.5;

in order to push the schedule time for the next RTP transmission faster, i.e. speed up the RTP transmission...I suspect that the problem is because the encoding rate for my audio is faster than the RTP transmission rate which is based on the returned value of the getPlayTime(), right? 

Thanks in advance and have a good year ahead...

Best Regards,
Joseph








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Hi Ross,

Happy New Year

>Are you trying to play the stream by having QuickTime Player read a ".sdp" 
>file?  If so, make sure that the payload format number is correct, and 
>matches the payload format number that you used when you created your 
>"SimpleRTPSink" (to stream the audio).

>In particular, the payload format number (which follows "RTP/AVP" on the 
>"m=" line) must be 0 (for PCM u-law), or 8 (for PCM a-law).


 I am using the Quick time to play my received data from RTP for Audio Input 

Device. The format i set for the input source is 0 and same i'm using in teh SDP file 

. i change d the format to 0 as i'm using U-law. Sample rate is 8000 wich is also

 specified in the SDP. But the  out put from QuickTime is  Empty.

 

 I have added a class to which takes the input from SimplRTPSource and Writes in 

WaveHdr  structure . With this i have aslo added the sample program. Th eprogram 

works fine but the problem here is at the receving end there is lot flickering  some

 times the sound is jarring. how do I remove that?

Thanks

 

 



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<P>Hi Ross,</P>
<P>Happy New Year</P>
<P>&gt;Are you trying to play the stream by having QuickTime Player read a ".sdp" <BR>&gt;file?&nbsp; If so, make sure that the payload format number is correct, and <BR>&gt;matches the payload format number that you used when you created your <BR>&gt;"SimpleRTPSink" (to stream the audio).<BR><BR>&gt;In particular, the payload format number (which follows "RTP/AVP" on the <BR>&gt;"m=" line) must be 0 (for PCM u-law), or 8 (for PCM a-law).<BR></P>
<P>&nbsp;I am using the Quick time to play my received data from RTP for Audio Input </P>
<P>Device. The format i set for the input source is 0 and same i'm using in teh SDP file </P>
<P>. i change d the format to 0 as i'm using U-law. Sample rate is 8000 wich is also</P>
<P>&nbsp;specified in the SDP. But the&nbsp; out put from QuickTime is&nbsp;&nbsp;Empty.</P>
<P>&nbsp;</P>
<P>&nbsp;I have added a class to which takes the input from SimplRTPSource and Writes in </P>
<P>WaveHdr&nbsp; structure . With this i have aslo added the sample program. Th eprogram </P>
<P>works fine but the problem here is at the receving end there is lot flickering&nbsp; some</P>
<P>&nbsp;times the sound is jarring. how do I remove that?</P>
<P>Thanks</P>
<P>&nbsp;</P>
<P>&nbsp;</P><p><br><hr size=1>Do you Yahoo!?<br>
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>Am I right to say that synchronisation for Audio and Video for the 
>testMPEGAudioVideoStreamer program DEPENDS on the getPlayTime() function 
>inside the MPEGAudioStreamFramer and MPEGVideoStreamFramer class?

No, not really.  Synchronization depends on the RTP timestamps in outgoing 
packets, which in turn depends on the "presentation time" parameter that's 
set for each input frame.

"RTPSink"s use "getPlayTime()" to figure out how long to delay between 
sending each output RTP packet.

But in any case, both "MPEGAudioStreamFramer" and "MPEGVideoStreamFramer" 
should set presentation times (and implement "getPlayTime()") correctly, 
provided that their input is actual MPEG (1 or 2) audio or video data.  If 
you've modified this code, then you're on your own (but, as always, 
remember that you have complete source code available :-)

         Ross.



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Hi Ross
 I want to Add  teh support for the .DAT files and .AVI  files .  i dont know the formats for these files. I could play a file in real Media when it was renamed to Dat from Mp3. but not in Windows Media player and WinAmp.
Please Help Me.
Satya          



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<P>Hi Ross
<P>&nbsp;I want to Add&nbsp; teh support for the .DAT files and .AVI&nbsp; files .&nbsp; i dont know the formats for these files. I could play a file in real Media when it&nbsp;was renamed to&nbsp;Dat from Mp3. but not in Windows Media player and WinAmp.
<P>Please Help Me.
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>  I want to Add  teh support for the .DAT files and .AVI  files .

I don't know about ".DAT", but ".AVI" is a *container* format, not a 
codec.  So, you wouldn't stream AVI files directly.  Instead, you would 
extract the specific media stream(s) from it, and stream those 
directly.  (This assumes, of course, that a RTP payload format exists for 
those media types.)

>   i dont know the formats for these files.

Google is your friend.  Here's a description of AVI:

http://www.jmcgowan.com/avi.html

         Ross.



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Hi Ross,
Once again it is me. I have implemented the Audio input device but Teh at the receiving end I get noisy  eating up the soung from mike. hwo do i remove/reduce the noise at the receiving end. 
FYI
Just after writing  into the buffer of DeliverFrame from temp buffer i'm also writng into a file. The file content is not noisy. where as teh transmitted content is noisy. The noise level is very high. 
PLease help ME.
Thnx in Advance
Satya 
 
 



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<P>Hi Ross,
<P>Once again it is me. I have implemented the Audio input device but Teh at the receiving end I get noisy&nbsp; eating up the soung from mike. hwo do i remove/reduce the noise at the receiving end. 
<P>FYI
<P>Just after writing&nbsp; into the buffer of DeliverFrame&nbsp;from temp buffer&nbsp;i'm also writng into a file. The file content is not noisy. where as teh transmitted content is noisy. The noise level is very high. 
<P>PLease help ME.
<P>Thnx in Advance
<P>Satya 
<P>&nbsp;
<P>&nbsp;</P><p><br><hr size=1>Do you Yahoo!?<br>
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Subject: Re: [Live-devel] Audio Device Input PRoblem
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X-Original-Date: Mon, 06 Jan 2003 18:19:15 -0800
Date: Mon, 06 Jan 2003 18:19:15 -0800

At 01:17 AM 1/6/03, you wrote:

>Hi Ross,
>
>Once again it is me. I have implemented the Audio input device but Teh at 
>the receiving end I get noisy  eating up the soung from mike. hwo do i 
>remove/reduce the noise at the receiving end.

I dunno, but the first thing I'd check is whether the sound is bad at both 
the sending and receiving ends, or whether it's bad only at the receiving end.

I.e., I'd save another copy of the audio at the sending end, after you've 
captured it (but before you pack it into RTP packets and send it).  If this 
audio also sounds bad, then the problem is with your audio capture code 
(perhaps, dropping audio samples).

         Ross.



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>I dunno, but the first thing I'd check is whether the sound is bad at both 
>the sending and receiving ends, or whether it's bad only at the receiving end.

1. The audio is  bad only at teh receveing end. I have made a copy of the buffer that is being sent  and played it. the auodio is clear. 

2. I aslo used teh FileSink to write into  a file at the receiving end  also aand also played it bak the result it is induced with noise.

PRoblem It ould be either the way i'm writing to into teh sending the buffer or playing it. 

problem  1. former part i'm writing the buffer what i have copied into the fTo buff pointer.

So the latter part should be problem.... i'm using the same program to play it .

Please.

 thx in Advance. 

Satya.

I.e., I'd save another copy of the audio at the sending end, after you've 
captured it (but before you pack it into RTP packets and send it). If this 
audio also sounds bad, then the problem is with your audio capture code 
(perhaps, dropping audio samples).









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<P>&gt;I dunno, but the first thing I'd check is whether the sound is bad at both <BR>&gt;the sending and receiving ends, or whether it's bad only at the receiving end.</P>
<P>1. The audio is&nbsp; bad only at teh receveing end. I have made a copy of the buffer that is being sent&nbsp; and played it. the auodio is clear. </P>
<P>2. I aslo used teh FileSink to write into&nbsp; a file at the receiving end&nbsp; also aand also played it bak the result it is induced with noise.</P>
<P>PRoblem It ould be either the way i'm writing to into teh sending the buffer or playing it. </P>
<P>problem&nbsp; 1. former part i'm writing the buffer what i have copied into the fTo buff pointer.</P>
<P>So the latter part should be problem.... i'm using the same program to play it .</P>
<P>Please.</P>
<P>&nbsp;thx in Advance. </P>
<P>Satya.<BR><BR>I.e., I'd save another copy of the audio at the sending end, after you've <BR>captured it (but before you pack it into RTP packets and send it). If this <BR>audio also sounds bad, then the problem is with your audio capture code <BR>(perhaps, dropping audio samples).<BR><BR><BR><BR><BR><BR></P><p><br><hr size=1>Do you Yahoo!?<br>
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X-Original-Date: Tue, 07 Jan 2003 20:11:38 -0800
Date: Tue, 07 Jan 2003 20:11:38 -0800

At 06:23 PM 1/7/03, you wrote:

> >I dunno, but the first thing I'd check is whether the sound is bad at both
> >the sending and receiving ends, or whether it's bad only at the 
> receiving end.
>
>1. The audio is  bad only at teh receveing end. I have made a copy of the 
>buffer that is being sent  and played it. the auodio is clear.

You should now have enough information to figure out, by yourself, what's 
going wrong.  I.e., you have a comunication chain of the form:

         (a) audio capture -> (b) RTP sink -> (c) network transmission
                 -> (d) RTP source -> (e) audio reception

You know that there's at least one point in this chain (a) where the PCM 
audio data is good, and that there's at least one point in this chain (e) 
where the PCM audio data is bad.  So, look at the data at the other points 
(b),(c),(d) in the chain, to figure out where and why you're ending up with 
bad data.

Good luck.

         Ross.



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Hi!

I'm making and application than streams video compressed with DivX,using =
DirectShow, now I'm using UDP to send the packets, but it`s doesn=B4t =
work proprely because it lose a lot of packets.
I have been testing live.com libraries and it works fine sending mpeg.=20
I have thought to use the live.com rtp protocol to send my divx packets =
over the network.=20
When DirectShow has compressed a frame it calls a routine I have created =
and then I have to send the packet over the network.
Is posible to use your libraries to do this? In than case, How can i do =
it?

Thanks for all

------=_NextPart_000_016A_01C2BA6A.73062F00
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
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<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Hi!</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>I'm making and application than streams =
video=20
compressed with DivX,using DirectShow, now I'm using UDP to send the =
packets,=20
but it`s doesn=B4t work proprely because it lose a lot of =
packets.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>I have been testing live.com libraries =
and it works=20
fine sending mpeg. </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>I have thought to use the live.com rtp =
protocol to=20
send my divx packets over the network. </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>When DirectShow has compressed a frame =
it calls a=20
routine I have created and then I have to send the packet over the=20
network.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Is posible to use your libraries to do =
this? In=20
than case, How can i do it?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Thanks for =
all</FONT></DIV></BODY></HTML>

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At 09:43 AM 1/12/03, you wrote:
>Hi!
>
>I'm making and application than streams video compressed with DivX,using 
>DirectShow, now I'm using UDP to send the packets, but it`s doesn´t work 
>proprely because it lose a lot of packets.

Note that RTP also uses UDP, so if you don't eliminate packet loss by using 
RTP.  (You will, however, be able to *detect* packet loss, because RTP 
packets include a sequence number.)

>I have been testing live.com libraries and it works fine sending mpeg.
>I have thought to use the live.com rtp protocol to send my divx packets 
>over the network.
>When DirectShow has compressed a frame it calls a routine I have created 
>and then I have to send the packet over the network.
>Is posible to use your libraries to do this? In than case, How can i do it?

You would need to do two things:

1/ You would need to write a source class that encapsulates your 
'DirectShow' video interface (i.e., to deliver encoded frames).  For a 
model for how to write such a class, see "liveMedia/DeviceSource.cpp" (and 
"liveMedia/include/DeviceSource.hh").

2/ You would need to figure out the rules for how DivX frames are carried 
in RTP packets.  (These rules are called a "RTP payload format".)  As DivX 
is based on MPEG-4 (I think), you may be able to use the IETF-standard RTP 
payload format that was defined for MPEG-4 video.  (This is not yet 
implemented as part of the released LIVE.COM libraries, however.)

         Ross.



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X-Original-Date: Tue, 14 Jan 2003 13:36:07 +0100
Date: Tue, 14 Jan 2003 13:36:07 +0100

Hi,

I'm having a look at the sources in order to see how to add proxy
support. I'm planning to add two more options to openRTSP (proxy host
and proxy port) and also modify RTSPClient accordingly. If found useful,
I'll post the changes.

romulus


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From <EMAIL-ADDRESS-HIDDEN> Thu Jan 16 13:52:58 2003
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X-Original-Date: Thu, 16 Jan 2003 13:50:02 -0800
Date: Thu, 16 Jan 2003 13:50:02 -0800

FYI, the latest version (2003.01.16) of the LIVE.COM Streaming Media 
libraries includes a new source class - MPEG4LATMAudioRTPSource - that 
allows you to receive MPEG-4 audio RTP streams (as defined in RFC 3016).

	Ross.



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--0-452219799-1043119987=:95765
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hi,

Currently i am developing streaming support media player using live RTSP/RTP library. When i make the range request( for example movie from 0-5 seconds or 5 to 10 seconds) , this works in windows as well with out any problem. When i try same thing in linux, it gets in to loop in the select() system call in SingleStep( ) function. I think it behaves as if I placed request for full content. What my worry is that, it is working in Windows w/o any problem, why does it not work in linux.

Please suggest me to come out from the looping.

execting your reply 

rama

 



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<P>hi,</P>
<P>Currently i am developing streaming support media player using live RTSP/RTP library. When i make the range request( for example movie from 0-5 seconds or 5 to 10 seconds) , this works in windows as well with out any problem. When i try same thing in linux, it gets in to loop in the select() system call in SingleStep( ) function. I think it behaves as if I placed request for full content. What my worry is that, it is working in Windows w/o any problem, why does it not work in linux.</P>
<P>Please suggest me to come out from the looping.</P>
<P>execting your reply </P>
<P>rama</P>
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X-Original-Date: Mon, 20 Jan 2003 22:11:35 -0800
Date: Mon, 20 Jan 2003 22:11:35 -0800


>Currently i am developing streaming support media player using live 
>RTSP/RTP library. When i make the range request( for example movie from 
>0-5 seconds or 5 to 10 seconds) , this works in windows as well with out 
>any problem. When i try same thing in linux, it gets in to loop in the 
>select() system call in SingleStep( ) function.

 From your description, it's hard to know what might be going wrong.  Note 
that BasicUsageEnvironment::SingleStep() is *supposed* to be be executed 
within a loop - i.e., the library's event loop.

Note that - within the event loop - events get handled in two different 
ways: (i) by a timer going off (so that a 'delayed task' gets executed), or 
(ii) by the arrival of a network packet (so that a 'network read handler') 
gets called.  If one of these two things should be happening, but isn't, 
then it should be possible for you to figure out how & why - perhaps using 
your (working) Windows application for comparison.  Remember, YHCSC ("you 
have complete source code" :-)

Note, by the way, that the "MPlayer" media player already supports 
RTSP/RTP, so you might find it useful to use this application as a model: 
see <http://www.live.com/mplayer/>.  (This, however, is only for Linux - 
not Windows.)

         Ross.



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From <EMAIL-ADDRESS-HIDDEN> Fri Jan 24 01:34:35 2003
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X-Original-Date: Fri, 24 Jan 2003 10:31:11 +0100
Date: Fri, 24 Jan 2003 10:31:11 +0100

Hello All,

Is there any significant documentation on the API? Besides the examples
there is not so much to work on.

Best Regards,

Ruud Schramp


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X-Original-Date: Fri, 24 Jan 2003 03:22:18 -0800
Date: Fri, 24 Jan 2003 03:22:18 -0800

At 01:31 AM 1/24/03, you wrote:
>Hello All,
>
>Is there any significant documentation on the API? Besides the examples
>there is not so much to work on.

Ruud,

For now, the best way to understand how to use the libraries is to (i) 
study the example programs in the "testProgs" directory, (ii) study the 
library code itself, and (especially)  (iii) ask lots of questions on this 
mailing list

Unfortunately there's no additional documentation right now (more 
documentation is on the "to do" list, but features requested by paying 
consulting clients currently take priority).

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>



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X-Original-Date: Fri, 24 Jan 2003 22:30:44 +0000
Date: Fri, 24 Jan 2003 22:30:44 +0000


Ruud,

<< Is there any significant documentation on the API? Besides the examples
there is not so much to work on. >>

There are the RFCs -- 1889 and 2326 for RTP / RTCP / RTSP.  If you're interested in MPEG, there is a ton of info on the web, and if you want the real MPEG info from the source, you can visit ansi.org and pay for ISO-IEC 13818-x (MPEG2, I forget the number for MPEG1 or 4).  These really go a long way to understanding how the stuff works, and everything is documented.  Try getting Microsoft to  make that sort of documentation available with Windows Media :-)

If you're interested in other reference implementations of RTP / RTSP / RTCP aside from Live, you can also check out Apple's open-source Darwin Streaming Server.  The code is pretty easy to follow, though there is a lot of it.  All of this assumes that what you're interested in is under the Live API.  If you're only interested in the Live API, then Ross has the best answer -- look at the test code.

Regards,

John

--
Falling You - exploring the beauty 
of voice and sound
http://www.mp3.com/fallingyou








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Subject: [Live-devel] RTSP with realaudio
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I'm trying to use Live library with mplayer to listen radio which stream in 
realaudio (.ra), the mime returned is "audio/X-PN-REALAUDIO".
Is it supported?

Thanx,
Filippo


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X-Original-Date: Tue, 28 Jan 2003 12:48:36 -0500
Date: Tue, 28 Jan 2003 12:48:36 -0500


Nope, there's no support for the "payload format" used by RealAudio/Video.
I want to use it aswell so I'm looking into getting it added myself.

-ross

Quoting Gioachin Filippo <gioachin(at)ieee.org>:

> I'm trying to use Live library with mplayer to listen radio which stream =
> in=20
> realaudio (.ra), the mime returned is "audio/X-PN-REALAUDIO".
> Is it supported?
> 
> Thanx,
> Filippo
> 
> 



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X-Original-Date: Mon, 27 Jan 2003 09:54:00 -0800
Date: Mon, 27 Jan 2003 09:54:00 -0800

At 09:40 AM 1/27/03, you wrote:
>I'm trying to use Live library with mplayer to listen radio which stream in
>realaudio (.ra), the mime returned is "audio/X-PN-REALAUDIO".
>Is it supported?

No, not yet, because RealAudio and RealVideo streams don't use standard RTP 
for transport.  Instead, they use RealNetworks' own "RDT" protocol.  Also, 
RealNetworks' servers use a cryptographic 'challenge-response' mechanism to 
authenticate clients.

The good news, though, is that the developers of "xine" have apparently 
reverse engineered these protocols, so they may eventually end up getting 
reimplemented in the LIVE.COM Streaming Media libraries also.  (Or, in the 
meantime, you could just try building/running "xine".)

Note also that you *can* use MPlayer to play many "rtsp://" URLs that are 
streamed from other kinds of RTSP servers (e.g., Apple's QuickTime/Darwin 
Streaming Server).  You can also play some types of streams from 
RealNetworks' servers, as long as the media are not RealAudio or RealVideo.

         Ross.



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X-Original-Date: Tue, 28 Jan 2003 13:06:33 -0500
Date: Tue, 28 Jan 2003 13:06:33 -0500


Do they really still use RDT? I thought they were all excited about using all
open standards...
Anyway, I wouldn't have thought reverse engineering would be necessary because
they have released the source to a player that can stream realvideo from Real
servers...  helixcommunity.org... I downloaded the source and built the player
and it seems to work.  I'm pretty sure the only part that's only available in
binary form is the codecs themselves.
I've asked someone to look into it and see if he can figure out the payload
format from the helix source and add that support to the live library.(i'm not a
coder myself)  I'll let you know what my coder finds out.

-ross currie
(no relation)

Quoting Ross Finlayson <finlayson(at)live.com>:

> At 09:40 AM 1/27/03, you wrote:
> >I'm trying to use Live library with mplayer to listen radio which stream
> in
> >realaudio (.ra), the mime returned is "audio/X-PN-REALAUDIO".
> >Is it supported?
> 
> No, not yet, because RealAudio and RealVideo streams don't use standard RTP
> 
> for transport.  Instead, they use RealNetworks' own "RDT" protocol.  Also, 
> RealNetworks' servers use a cryptographic 'challenge-response' mechanism to
> 
> authenticate clients.
> 
> The good news, though, is that the developers of "xine" have apparently 
> reverse engineered these protocols, so they may eventually end up getting 
> reimplemented in the LIVE.COM Streaming Media libraries also.  (Or, in the 
> meantime, you could just try building/running "xine".)
> 
> Note also that you *can* use MPlayer to play many "rtsp://" URLs that are 
> streamed from other kinds of RTSP servers (e.g., Apple's QuickTime/Darwin 
> Streaming Server).  You can also play some types of streams from 
> RealNetworks' servers, as long as the media are not RealAudio or RealVideo.
> 
>          Ross.
> 
> 
> 
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> 





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X-Original-Date: Thu, 30 Jan 2003 17:06:06 +1000
Date: Thu, 30 Jan 2003 17:06:06 +1000

Hi,

I was getting this error with testMPEGAudioVideoStreamer.exe:

--
11:55:30 Groupsock(1992: 239.255.42.42, 6666, 7):
failed to join group: setsockpt(IP_ADD_MEMBERSHIP) error: Unknown error
...
--

The workaround to link with "Wsock32.lib" (instead of "Ws2_32.lib")
by setting "TARGETOS = WIN95" in "win32config" no longer works with
VisualC++.Net (i.e. VC++ 7). "win32config" includes "ntwin32.mak",
which in turn just includes "win32.mak", and here is an extract
from "win32.mak" :

----
# if building for basic Windows 95, use WinSock1, else use WinSock2
!IF "$(TARGETOS)" == "WIN95"
!IF "$(APPVER)" == "4.0"
winsocklibs = wsock32.lib
!ELSE
winsocklibs = ws2_32.lib mswsock.lib
!ENDIF
!ELSE
winsocklibs = ws2_32.lib mswsock.lib
!ENDIF
----

Adding "APPVER = 4.0" to "win32config" seems to get rid of the error
  message with testMPEGAudioVideoStreamer.exe, but
testMPEGVideoReceiver.exe doesn't output anything to stdout.

Anyway, I've started looking into why multicasting might be failing
with WinSock2, I think Microsoft Knowledge Base Article - Q257460,
might give an explanation, see:

  http://support.microsoft.com/default.aspx?scid=kb%3ben-us%3b257460

So I was thinking of including "Winsock2.h" and "Ws2tcpip.h" before
"Windows.h" in a few stratigic places (this stops "winsock.h" being
included by "windows.h"). I haven't gotten around to it yet, but
will report back when I do.

Thanks,
Doug.



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From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] WinSock and VisualC++.Net problem
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X-Original-Date: Thu, 30 Jan 2003 14:36:29 -0800
Date: Thu, 30 Jan 2003 14:36:29 -0800


>Anyway, I've started looking into why multicasting might be failing
>with WinSock2, I think Microsoft Knowledge Base Article - Q257460,
>might give an explanation, see:
>
>  http://support.microsoft.com/default.aspx?scid=kb%3ben-us%3b257460
>
>So I was thinking of including "Winsock2.h" and "Ws2tcpip.h" before
>"Windows.h" in a few stratigic places (this stops "winsock.h" being
>included by "windows.h"). I haven't gotten around to it yet, but
>will report back when I do.

Doug,

Thanks for looking into this.  (Un(?)fortunately I don't have Visual 
C++.Not, so I can't look into this myself.)

If you find a way to make things work (that doesn't also break with earlier 
versions of VisualC++), please let us know.

         Ross.



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From <EMAIL-ADDRESS-HIDDEN> Sun Feb 2 18:17:23 2003
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From: Douglas Kosovic <EMAIL-ADDRESS-HIDDEN>
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X-Original-Date: Mon, 03 Feb 2003 12:06:32 +1000
Date: Mon, 03 Feb 2003 12:06:32 +1000

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Hi Ross,

> If you find a way to make things work (that doesn't also break with 
> earlier versions of VisualC++), please let us know.

Please find attached a Winsock2 patch (for live.2003.01.28) that works
with both Visual C++ 6 & .NET, I don't have Visual C++ 5, so couldn't
test that combination.

It is still possible to build with Winsock1, I've tested it with both
VC++.NET & 6 and it should still work with VC++5. Extract from modified
"win32config" explains how to pick between either Winsock1 or Winsock2:

--
# The following definition ensures that we are properly matching
# the WinSock2 library file with the correct header files.
# (will link with "ws2_32.lib" and include "winsock2.h" & "Ws2tcpip.h")
TARGETOS = WINNT

# If for some reason you wish to use WinSock1 instead, uncomment the
# following two definitions.
# (will link with "wsock32.lib" and include "winsock.h")
#TARGETOS = WIN95
#APPVER = 4.0
--


The WinSock2 modifications to the source code basically consist of
adding the following preprocessor directives:

#if defined(_WINNT)
#define _MSWSOCK_
#include <winsock2.h>
#include <ws2tcpip.h>
#endif

The _MSWSOCK_ definition is for VC++6, it stops the inclusion of
"mswsock.h" which has a dependency conflict when "winsock2.h" is
explicitly included.

Here is the invocation of the patch showing which files are patched:

   bash$ cd live
   bash$ patch -p1 < ~/ws2patch.txt
   patching file BasicUsageEnvironment/DelayQueue.cpp
   patching file BasicUsageEnvironment/include/DelayQueue.hh
   patching file groupsock/GroupsockHelper.cpp
   patching file groupsock/include/GroupEId.hh
   patching file groupsock/include/NetAddress.hh
   patching file groupsock/inet.c
   patching file liveMedia/HTTPSink.cpp
   patching file liveMedia/MP3HTTPSource.cpp
   patching file liveMedia/MP3StreamState.cpp
   patching file liveMedia/MP3StreamState.hh
   patching file liveMedia/MediaSession.cpp
   patching file liveMedia/RTSPClient.cpp
   patching file liveMedia/RTSPServer.cpp
   patching file liveMedia/SIPClient.cpp
   patching file liveMedia/include/FramedSource.hh
   patching file win32config


Thanks,
Doug.

--------------050909080904050001060809
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diff -c -r live.2003.01.28/BasicUsageEnvironment/DelayQueue.cpp live/BasicUsageEnvironment/DelayQueue.cpp
*** live.2003.01.28/BasicUsageEnvironment/DelayQueue.cpp	Wed Jan 29 16:50:25 2003
--- live/BasicUsageEnvironment/DelayQueue.cpp	Fri Jan 31 15:13:20 2003
***************
*** 22,29 ****
--- 22,33 ----
  #include "GroupsockHelper.hh"
  #if defined(__WIN32__) || defined(_WIN32)
  #ifdef __BORLANDC__
+ #if defined(_WINNT)
+ #include <winsock2.h> //struct timeval is here
+ #else
  #include <winsock.h> //struct timeval is here
  #endif
+ #endif
  #else
  #include <signal.h>
  #endif
diff -c -r live.2003.01.28/BasicUsageEnvironment/include/DelayQueue.hh live/BasicUsageEnvironment/include/DelayQueue.hh
*** live.2003.01.28/BasicUsageEnvironment/include/DelayQueue.hh	Wed Jan 29 16:50:25 2003
--- live/BasicUsageEnvironment/include/DelayQueue.hh	Fri Jan 31 15:17:33 2003
***************
*** 25,30 ****
--- 25,35 ----
  #endif
  
  #if defined(__WIN32__) || defined(_WIN32)
+ #if defined(_WINNT)
+ #define _MSWSOCK_
+ #include <winsock2.h>
+ #include <ws2tcpip.h>
+ #endif
  #include <windows.h>
  #else
  #include <sys/time.h>
diff -c -r live.2003.01.28/groupsock/GroupsockHelper.cpp live/groupsock/GroupsockHelper.cpp
*** live.2003.01.28/groupsock/GroupsockHelper.cpp	Wed Jan 29 16:50:26 2003
--- live/groupsock/GroupsockHelper.cpp	Fri Jan 31 15:19:12 2003
***************
*** 24,30 ****
  #include <time.h>
  extern "C" int initializeWinsockIfNecessary();
  #define _close closesocket
! #if defined(_WINNT) || defined(__BORLANDC__)
  #include <ws2tcpip.h>
  #endif
  #else
--- 24,32 ----
  #include <time.h>
  extern "C" int initializeWinsockIfNecessary();
  #define _close closesocket
! #if defined(_WINNT)
! #define _MSWSOCK_
! #include <winsock2.h>
  #include <ws2tcpip.h>
  #endif
  #else
diff -c -r live.2003.01.28/groupsock/include/GroupEId.hh live/groupsock/include/GroupEId.hh
*** live.2003.01.28/groupsock/include/GroupEId.hh	Wed Jan 29 16:50:26 2003
--- live/groupsock/include/GroupEId.hh	Fri Jan 31 15:20:33 2003
***************
*** 26,31 ****
--- 26,36 ----
  #endif
  
  #if defined(__WIN32__) || defined(_WIN32)
+ #if defined(_WINNT)
+ #define _MSWSOCK_
+ #include <winsock2.h>
+ #include <ws2tcpip.h>
+ #endif
  #include <windows.h>
  #else
  #include <stddef.h>
diff -c -r live.2003.01.28/groupsock/include/NetAddress.hh live/groupsock/include/NetAddress.hh
*** live.2003.01.28/groupsock/include/NetAddress.hh	Wed Jan 29 16:50:26 2003
--- live/groupsock/include/NetAddress.hh	Fri Jan 31 15:20:50 2003
***************
*** 26,31 ****
--- 26,36 ----
  #endif
  
  #if defined(__WIN32__) || defined(_WIN32)
+ #if defined(_WINNT)
+ #define _MSWSOCK_
+ #include <winsock2.h>
+ #include <ws2tcpip.h>
+ #endif
  #include <windows.h>
  #else
  #include <sys/types.h>
diff -c -r live.2003.01.28/groupsock/inet.c live/groupsock/inet.c
*** live.2003.01.28/groupsock/inet.c	Wed Jan 29 16:50:26 2003
--- live/groupsock/inet.c	Fri Jan 31 15:23:29 2003
***************
*** 5,10 ****
--- 5,15 ----
   */
  
  #if defined(__WIN32__) || defined(_WIN32)
+ #if defined(_WINNT)
+ #define _MSWSOCK_
+ #include <winsock2.h>
+ #include <ws2tcpip.h>
+ #endif
  #include <windows.h>
  #else
  #include <sys/types.h>
diff -c -r live.2003.01.28/liveMedia/HTTPSink.cpp live/liveMedia/HTTPSink.cpp
*** live.2003.01.28/liveMedia/HTTPSink.cpp	Wed Jan 29 16:50:27 2003
--- live/liveMedia/HTTPSink.cpp	Fri Jan 31 15:25:09 2003
***************
*** 26,31 ****
--- 26,33 ----
  #define _close closesocket
  #define snprintf _snprintf
  #if defined(_WINNT)
+ #define _MSWSOCK_
+ #include <winsock2.h>
  #include <ws2tcpip.h>
  #endif
  #else
diff -c -r live.2003.01.28/liveMedia/MP3HTTPSource.cpp live/liveMedia/MP3HTTPSource.cpp
*** live.2003.01.28/liveMedia/MP3HTTPSource.cpp	Wed Jan 29 16:50:30 2003
--- live/liveMedia/MP3HTTPSource.cpp	Fri Jan 31 15:30:12 2003
***************
*** 24,32 ****
  
  #if defined(__WIN32__) || defined(_WIN32)
  #define _close closesocket
- #if defined(_WINNT)
- #include <ws2tcpip.h>
- #endif
  #else
  #include <unistd.h>
  #include <sys/socket.h>
--- 24,29 ----
diff -c -r live.2003.01.28/liveMedia/MP3StreamState.cpp live/liveMedia/MP3StreamState.cpp
*** live.2003.01.28/liveMedia/MP3StreamState.cpp	Wed Jan 29 16:50:30 2003
--- live/liveMedia/MP3StreamState.cpp	Fri Jan 31 15:30:34 2003
***************
*** 22,33 ****
  #include "GroupsockHelper.hh"
  
  #if defined(__WIN32__) || defined(_WIN32)
- #include <windows.h>
  #define _close closesocket
  #define snprintf _snprintf
- #if defined(_WINNT)
- #include <ws2tcpip.h>
- #endif
  #else
  #include <unistd.h>
  #define _close close
--- 22,29 ----
diff -c -r live.2003.01.28/liveMedia/MP3StreamState.hh live/liveMedia/MP3StreamState.hh
*** live.2003.01.28/liveMedia/MP3StreamState.hh	Wed Jan 29 16:50:30 2003
--- live/liveMedia/MP3StreamState.hh	Fri Jan 31 15:28:13 2003
***************
*** 29,34 ****
--- 29,39 ----
  #endif
  
  #if defined(__WIN32__) || defined(_WIN32)
+ #if defined(_WINNT)
+ #define _MSWSOCK_
+ #include <winsock2.h>
+ #include <ws2tcpip.h>
+ #endif
  #include <windows.h>
  #else
  #include <sys/time.h>
diff -c -r live.2003.01.28/liveMedia/MediaSession.cpp live/liveMedia/MediaSession.cpp
*** live.2003.01.28/liveMedia/MediaSession.cpp	Wed Jan 29 16:50:29 2003
--- live/liveMedia/MediaSession.cpp	Fri Jan 31 15:26:11 2003
***************
*** 23,32 ****
  #include "GroupsockHelper.hh"
  
  #if defined(__WIN32__) || defined(_WIN32)
- #include <windows.h>
  #if defined(_WINNT)
  #include <ws2tcpip.h>
  #endif
  #else
  #include <ctype.h>
  #include <sys/socket.h>
--- 23,34 ----
  #include "GroupsockHelper.hh"
  
  #if defined(__WIN32__) || defined(_WIN32)
  #if defined(_WINNT)
+ #define _MSWSOCK_
+ #include <winsock2.h>
  #include <ws2tcpip.h>
  #endif
+ #include <windows.h>
  #else
  #include <ctype.h>
  #include <sys/socket.h>
diff -c -r live.2003.01.28/liveMedia/RTSPClient.cpp live/liveMedia/RTSPClient.cpp
*** live.2003.01.28/liveMedia/RTSPClient.cpp	Wed Jan 29 16:50:31 2003
--- live/liveMedia/RTSPClient.cpp	Fri Jan 31 15:33:54 2003
***************
*** 23,35 ****
  #include "our_md5.h"
  
  #if defined(__WIN32__) || defined(_WIN32)
  #include <windows.h>
  #define _close closesocket
  #define _strncasecmp strncmp
  #define snprintf _snprintf
- #if defined(_WINNT)
- #include <ws2tcpip.h>
- #endif
  #else
  #include <unistd.h>
  #define _close close
--- 23,37 ----
  #include "our_md5.h"
  
  #if defined(__WIN32__) || defined(_WIN32)
+ #if defined(_WINNT)
+ #define _MSWSOCK_
+ #include <winsock2.h>
+ #include <ws2tcpip.h>
+ #endif
  #include <windows.h>
  #define _close closesocket
  #define _strncasecmp strncmp
  #define snprintf _snprintf
  #else
  #include <unistd.h>
  #define _close close
diff -c -r live.2003.01.28/liveMedia/RTSPServer.cpp live/liveMedia/RTSPServer.cpp
*** live.2003.01.28/liveMedia/RTSPServer.cpp	Wed Jan 29 16:50:31 2003
--- live/liveMedia/RTSPServer.cpp	Fri Jan 31 15:34:48 2003
***************
*** 25,33 ****
  #if defined(__WIN32__) || defined(_WIN32)
  #define _close closesocket
  #define snprintf _snprintf
- #if defined(_WINNT)
- #include <ws2tcpip.h>
- #endif
  #else
  #include <unistd.h>
  #include <sys/socket.h>
--- 25,30 ----
diff -c -r live.2003.01.28/liveMedia/SIPClient.cpp live/liveMedia/SIPClient.cpp
*** live.2003.01.28/liveMedia/SIPClient.cpp	Wed Jan 29 16:50:32 2003
--- live/liveMedia/SIPClient.cpp	Fri Jan 31 15:36:07 2003
***************
*** 23,34 ****
  #include "GroupsockHelper.hh"
  
  #if defined(__WIN32__) || defined(_WIN32)
- #include <windows.h>
- #define _strncasecmp strncmp
- #define snprintf _snprintf
  #if defined(_WINNT)
  #include <ws2tcpip.h>
  #endif
  #else
  #include <unistd.h>
  #if defined(_QNX4)
--- 23,36 ----
  #include "GroupsockHelper.hh"
  
  #if defined(__WIN32__) || defined(_WIN32)
  #if defined(_WINNT)
+ #define _MSWSOCK_
+ #include <winsock2.h>
  #include <ws2tcpip.h>
  #endif
+ #include <windows.h>
+ #define _strncasecmp strncmp
+ #define snprintf _snprintf
  #else
  #include <unistd.h>
  #if defined(_QNX4)
diff -c -r live.2003.01.28/liveMedia/include/FramedSource.hh live/liveMedia/include/FramedSource.hh
*** live.2003.01.28/liveMedia/include/FramedSource.hh	Wed Jan 29 16:50:27 2003
--- live/liveMedia/include/FramedSource.hh	Fri Jan 31 15:25:36 2003
***************
*** 26,31 ****
--- 26,36 ----
  #endif
  
  #if defined(__WIN32__) || defined(_WIN32)
+ #if defined(_WINNT)
+ #define _MSWSOCK_
+ #include <winsock2.h>
+ #include <ws2tcpip.h>
+ #endif
  #include <windows.h>
  #else
  #include <sys/time.h>
diff -c -r live.2003.01.28/win32config live/win32config
*** live.2003.01.28/win32config	Wed Jan 29 16:50:33 2003
--- live/win32config	Fri Jan 31 15:58:19 2003
***************
*** 4,10 ****
  # The following definition ensures that we link with "wsock32.lib"
  # rather than "ws2_32.lib".  For some reason, the standard Berkeley
  # socket calls for multicast don't work properly with "ws2_32.lib".
! TARGETOS = WIN95
  
  !include    <ntwin32.mak>
  
--- 4,21 ----
  # The following definition ensures that we link with "wsock32.lib"
  # rather than "ws2_32.lib".  For some reason, the standard Berkeley
  # socket calls for multicast don't work properly with "ws2_32.lib".
! 
! 
! # The following definition ensures that we are properly matching
! # the WinSock2 library file with the correct header files.
! # (will link with "ws2_32.lib" and include "winsock2.h" & "Ws2tcpip.h")
! TARGETOS = WINNT
! 
! # If for some reason you wish to use WinSock1 instead, uncomment the
! # following two definitions.
! # (will link with "wsock32.lib" and include "winsock.h")
! #TARGETOS = WIN95
! #APPVER = 4.0
  
  !include    <ntwin32.mak>
  

--------------050909080904050001060809--



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X-Original-Date: Mon, 03 Feb 2003 12:30:32 +1000
Date: Mon, 03 Feb 2003 12:30:32 +1000

Douglas Kosovic wrote:

> Adding "APPVER = 4.0" to "win32config" seems to get rid of the error
> message with testMPEGAudioVideoStreamer.exe, but
> testMPEGVideoReceiver.exe doesn't output anything to stdout.

Just a follow up to my original post, I would get something to stdout as 
long as I ran testMPEGAudioVideoStreamer on another host.

The problem was that I have a DVB-S PCI card (i.e DVB/MPEG-2 satellite 
receiver) which has a Class-C private network address (in the range 
192.68.0.0 - 192.168.255.255) and testMPEGAudioVideoStreamer.exe seemed 
to be trying to multicast to it rather than the ethernet card. Disabling 
the DVB-S private virtual network made things work.

Cheers,
Doug.



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X-Original-Date: Sun, 02 Feb 2003 18:52:59 -0800
Date: Sun, 02 Feb 2003 18:52:59 -0800

Doug,

Many thanks for working this out.  I'll go ahead and apply the patch that 
you sent me, simplify the changes, as necessary, to reduce the number of 
files that contain these Windows-specific definitions and #includes, and 
then install a new version of the sources.  I'll send a message to this 
list when a new version is available.

Thanks again for your help.

	Ross.



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X-Original-Date: Mon, 03 Feb 2003 03:43:20 -0800
Date: Mon, 03 Feb 2003 03:43:20 -0800

FYI, I have now installed a new version (2003.02.03a) of the LIVE.COM 
libraries that incorporates the results of Doug Kosovic's discoveries of 
how to get Windows code to properly use the Winsock2 libraries.

The "win32config" script has been updated.  Also, all of the 
Windows-specific networking definitions and #includes are now in a single 
#include file: "groupsock/include/NetCommon.h".  The other source files now 
all #include this (directly or indirectly).

I have tested this with VC++5, and Doug has tested it with VC++6 and VC++ 
.Net.  But if anyone encounters any problems with this change, just let us 
know.

	Ross.




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X-Original-Date: Tue, 4 Feb 2003 02:21:20 +0900 (JST)
Date: Tue, 4 Feb 2003 02:21:20 +0900 (JST)

Hi liveMedia developper.

	I received that reporter, which is Anders Andersson, can't compile
	BasicUsageEnvironment/Lock.cpp in 2002.02.03a.  According to gcc32,
	it say that `NULL' was not declared in this scope.
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
c++ -c -O -pipe -march=pentiumpro -Iinclude -I../UsageEnvironment/include -I../groupsock/include -I. -DBSD=1 -DSOCKLEN_T=socklen_t -Wall Lock.cpp
Lock.cpp:34: `NULL' was not declared in this scope
Lock.cpp: In constructor `RWLock::RWLock()':
Lock.cpp:41: `NULL' undeclared (first use this function)
Lock.cpp:41: (Each undeclared identifier is reported only once for each 
   function it appears in.)
*** Error code 1
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -

	I confirmed and fix this problem like following lines.
	Please check!
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
--- BasicUsageEnvironment/Lock.cpp.orig Mon Feb  3 20:10:22 2003
+++ BasicUsageEnvironment/Lock.cpp      Tue Feb  4 01:55:14 2003
(at)(at) -19,6 +19,7 (at)(at)
 
 #include "Lock.hh"
 #include <signal.h>
+#include "NetCommon.h"
 
 ///// Read-Write Locks /////
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -


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X-Original-Date: Mon, 03 Feb 2003 13:49:21 -0800
Date: Mon, 03 Feb 2003 13:49:21 -0800


>         I confirmed and fix this problem like following lines.
>         Please check!
>- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - 
>- - -
>--- BasicUsageEnvironment/Lock.cpp.orig Mon Feb  3 20:10:22 2003
>+++ BasicUsageEnvironment/Lock.cpp      Tue Feb  4 01:55:14 2003
>(at)(at) -19,6 +19,7 (at)(at)
>
>  #include "Lock.hh"
>  #include <signal.h>
>+#include "NetCommon.h"

Thanks - this fixes the problem.  I've just installed a new version 
(2003.02.03b) of the code with this fix.

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>



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Subject: [Live-devel] Compilation broken on Linux..
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X-Original-Date: Mon, 3 Feb 2003 20:28:28 +0000
Date: Mon, 3 Feb 2003 20:28:28 +0000
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X-MIME-Autoconverted: from quoted-printable to 8bit by ns.live.com id h13NWjvG086114


Today's snapshot of live.com does not compile under Linux 2.4.19-16mdk, 
running gcc 3.2.1

---

[root(at)localhost live]# make
cd groupsock ; make
make[1]: Cambiando a directorio `/usr/lib/live/groupsock'
make[1]: *** Warning: File `GroupsockHelper.cpp' has modification time                                        
in the future (2003-02-03 21:46:04 > 2003-02-03 20:25:47)
c++ -c -Iinclude -I../UsageEnvironment/include -I. -O -DSOCKLEN_T=sockl                                       
en_t -Wall -DBSD=1 GroupsockHelper.cpp
En el fichero incluído de /usr/local/include/c++/3.2.1/backward/iostrea                                       
m.h:31,
                 de include/NetAddress.hh:32,
                 de include/GroupsockHelper.hh:25,
                 de GroupsockHelper.cpp:21:
/usr/local/include/c++/3.2.1/backward/backward_warning.h:32:2: aviso: #                                       
warning This file includes at least one deprecated or antiquated header                                       
. Please consider using one of the 32 headers found in section 17.4.1.2                                        
of the C++ standard. Examples include substituting the <X> header for                                        
the <X.h> header for C++ includes, or <sstream> instead of the deprecat                                       
ed header <strstream.h>. To disable this warning use -Wno-deprecated.
c++ -c -Iinclude -I../UsageEnvironment/include -I. -O -DSOCKLEN_T=socklen_t 
-Wall -DBSD=1 GroupEId.cpp
cc -c -Iinclude -I../UsageEnvironment/include -I. -O -DSOCKLEN_T=socklen_t 
-DUSE_OUR_BZERO=1 inet.c
inet.c:476: error de decodificación antes de '\x0'
inet.c:477: aviso: tipos en conflicto para la función interna 
`__builtin_memset'
inet.c: En la función `__builtin_memset':
inet.c:478: existe la declaración para el parámetro `count' pero no hay tal 
parámetro
inet.c:477: existe la declaración para el parámetro `to' pero no hay tal 
parámetro
make[1]: *** [inet.o] Error 1
make[1]: Saliendo directorio `/usr/lib/live/groupsock'
make: *** [groupsock/libgroupsock.a] Error 2


Damian

-- 
--
I don't want Windows to be only for the 31173. Yes, we've come a long way from
all those security holes, virii, and cryptic commands like "Edit textfile.txt"
(what in the hell is that supposed to mean?)


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X-Original-Date: Tue, 4 Feb 2003 11:31:06 +0900
Date: Tue, 4 Feb 2003 11:31:06 +0900

On Mon, 03 Feb 2003 13:49:21 -0800
Ross Finlayson <finlayson(at)live.com> wrote:
> Thanks - this fixes the problem.  I've just installed a new version 
> (2003.02.03b) of the code with this fix.

	Thank you. I updated new version. It's good!


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>Today's snapshot of live.com does not compile under Linux 2.4.19-16mdk,
>running gcc 3.2.1
[...]
>cc -c -Iinclude -I../UsageEnvironment/include -I. -O -DSOCKLEN_T=socklen_t
>-DUSE_OUR_BZERO=1 inet.c
>inet.c:476: error de decodificación antes de '\x0'
>inet.c:477: aviso: tipos en conflicto para la función interna
>`__builtin_memset'
>inet.c: En la función `__builtin_memset':
>inet.c:478: existe la declaración para el parámetro `count' pero no hay tal
>parámetro
>inet.c:477: existe la declaración para el parámetro `to' pero no hay tal
>parámetro

The problem here appears to be caused by the following code in 
/usr/include/bits/string2.h (assuming that you have the same version that I 
do):

         /* GCC optimizes memset(s, 0, n) but not bzero(s, n).
           The optimization is broken before EGCS 1.1.  */
         # if __GNUC_PREREQ (2, 91)
         #  define __bzero(s, n) __builtin_memset (s, '\0', n)
         # endif

I.e., in your environment, "__bzero()" is defined (by a macro), so it's 
wrong for us to be defining it ourself.

The solution is to remove
         -DUSE_OUR_BZERO=1
from the "C_FLAGS" line in "config.linux", and then rerun
         genMakefiles linux

On the assumption that all Linux environments now define "__bzero()", I'll 
go ahead and make this change (to "config.linux") in future releases.

         Ross.



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Subject: [Live-devel] live isn't 64bit clean
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X-Original-Date: Wed, 5 Feb 2003 22:21:58 +0100
Date: Wed, 5 Feb 2003 22:21:58 +0100

While trying to compile live.2003.01.10 on x86-64 (aka AMD Hammer), I
noticed the warnings included below and found code isn't 64bit clean.

Linux on x86-64, like Linux on AXP, uses LP64, i.e. int is 32 bit and only
'long' (and of cause 'unsigned long') are 64 bit. As pointers on these
platforms are 64bit, only longs/unsigned longs may safely be used for
doing pointer<->integer casts.

I started to check these warnings and some cases could easyly be changed to
use (int)(long)pointer, but then I found code that looked like trying to
store a pointer in an int and that definitely won't work.

As my time to work on such things is currently limited, I can't come up with
patches, but I thought that you should look into these warnings and fix
them. As it is, live sadly just isn't ready for LP64 platforms.

And while I have your eyes, what are the chances of moving to ISO C++
conforming headers, i.e. iostream instead of iostream.h and sstream
instead of strstream.h? That would of cause also require some code changes,
like using std:: where necessary.

The warnings are:
make[1]: Entering directory `/usr/src/packages/BUILD/MPlayer-0.90rc3/live/liveMedia'

c++ -c -Iinclude -I../UsageEnvironment/include -I../groupsock/include -I. -O -DSOCKLEN_T=socklen_t -Wall -DBSD=1 -Wno-deprecated FramedSource.cpp
FramedSource.cpp: In member function `void FramedSource::getNextFrame(unsigned 
   char*, unsigned int, void (*)(void*, unsigned int, timeval), void*, void 
   (*)(void*), void*)':
FramedSource.cpp:64: warning: cast from pointer to integer of different size

c++ -c -Iinclude -I../UsageEnvironment/include -I../groupsock/include -I. -O -DSOCKLEN_T=socklen_t -Wall -DBSD=1 -Wno-deprecated MP3HTTPSource.cpp
MP3HTTPSource.cpp: In static member function `static MP3HTTPSource* 
   MP3HTTPSource::createNew(UsageEnvironment&, const NetAddress&, Port, const 
   char*, const char*)':
MP3HTTPSource.cpp:73: warning: cast to pointer from integer of different size

c++ -c -Iinclude -I../UsageEnvironment/include -I../groupsock/include -I. -O -DSOCKLEN_T=socklen_t -Wall -DBSD=1 -Wno-deprecated MP3StreamState.cpp
MP3StreamState.cpp: In destructor `virtual MP3StreamState::~MP3StreamState()':
MP3StreamState.cpp:50: warning: cast from pointer to integer of different size
MP3StreamState.cpp: In member function `void MP3StreamState::writeGetCmd(const 
   char*, short unsigned int, const char*)':
MP3StreamState.cpp:176: warning: cast from pointer to integer of different size
MP3StreamState.cpp: In member function `unsigned int 
   MP3StreamState::readFromStream(unsigned char*, unsigned int)':
MP3StreamState.cpp:354: warning: cast from pointer to integer of different size

c++ -c -Iinclude -I../UsageEnvironment/include -I../groupsock/include -I. -O -DSOCKLEN_T=socklen_t -Wall -DBSD=1 -Wno-deprecated RTPSource.cpp
RTPSource.cpp: In member function `void 
   RTPReceptionStatsDB::noteIncomingPacket(unsigned int, short unsigned int, 
   unsigned int, unsigned int, unsigned int, timeval&)':
RTPSource.cpp:102: warning: cast to pointer from integer of different size
RTPSource.cpp:108: warning: cast to pointer from integer of different size
RTPSource.cpp: In member function `void 
   RTPReceptionStatsDB::noteIncomingSR(unsigned int, unsigned int, unsigned 
   int, unsigned int)':
RTPSource.cpp:125: warning: cast to pointer from integer of different size
RTPSource.cpp:131: warning: cast to pointer from integer of different size

c++ -c -Iinclude -I../UsageEnvironment/include -I../groupsock/include -I. -O -DSOCKLEN_T=socklen_t -Wall -DBSD=1 -Wno-deprecated MultiFramedRTPSink.cpp
MultiFramedRTPSink.cpp: In static member function `static void 
   MultiFramedRTPSink::sendNext(int)':
MultiFramedRTPSink.cpp:326: warning: cast to pointer from integer of different 
   size

c++ -c -Iinclude -I../UsageEnvironment/include -I../groupsock/include -I. -O -DSOCKLEN_T=socklen_t -Wall -DBSD=1 -Wno-deprecated RTPInterface.cpp
RTPInterface.cpp: In function `SocketDescriptor* lookupSocketDescriptor(int)':
RTPInterface.cpp:67: warning: cast to pointer from integer of different size
RTPInterface.cpp: In member function `void 
   RTPInterface::startNetworkReading(void (*)(void*, int))':
RTPInterface.cpp:114: warning: cast to pointer from integer of different size
RTPInterface.cpp: In member function `void 
   SocketDescriptor::registerRTPInterface(unsigned char, RTPInterface*)':
RTPInterface.cpp:220: warning: cast to pointer from integer of different size
RTPInterface.cpp:228: warning: cast to pointer from integer of different size
RTPInterface.cpp: In member function `RTPInterface* 
   SocketDescriptor::lookupRTPInterface(unsigned char)':
RTPInterface.cpp:235: warning: cast to pointer from integer of different size
RTPInterface.cpp: In member function `void 
   SocketDescriptor::deregisterRTPInterface(unsigned char)':
RTPInterface.cpp:240: warning: cast to pointer from integer of different size
c++ -c -Iinclude -I../UsageEnvironment/include -I../groupsock/include -I. -O -DSOCKLEN_T=socklen_t -Wall -DBSD=1 -Wno-deprecated RTCP.cpp
RTCP.cpp: In member function `Boolean RTCPMemberDatabase::isMember(unsigned 
   int) const':
RTCP.cpp:44: warning: cast to pointer from integer of different size
RTCP.cpp: In member function `Boolean 
   RTCPMemberDatabase::noteMembership(unsigned int, unsigned int)':
RTCP.cpp:55: warning: cast to pointer from integer of different size
RTCP.cpp:55: warning: cast to pointer from integer of different size
RTCP.cpp: In member function `Boolean RTCPMemberDatabase::remove(unsigned 
   int)':
RTCP.cpp:61: warning: cast to pointer from integer of different size
RTCP.cpp: In member function `void RTCPMemberDatabase::reapOldMembers(unsigned 
   int)':
RTCP.cpp:90: warning: cast from pointer to integer of different size
RTCP.cpp:95: warning: cast from pointer to integer of different size

c++ -c -Iinclude -I../UsageEnvironment/include -I../groupsock/include -I. -O -DSOCKLEN_T=socklen_t -Wall -DBSD=1 -Wno-deprecated QuickTimeFileSink.cpp
QuickTimeFileSink.cpp: In member function `unsigned int 
   QuickTimeFileSink::addAtom_elst()':
QuickTimeFileSink.cpp:1340: warning: passing negative value `-1' for argument 1 
   of `unsigned int QuickTimeFileSink::addWord(unsigned int)'

make[1]: Leaving directory `/usr/src/packages/BUILD/MPlayer-0.90rc3/live/liveMedia'

make[1]: Entering directory `/usr/src/packages/BUILD/MPlayer-0.90rc3/live/groupsock'
cc -c -Iinclude -I../UsageEnvironment/include -I. -O -DSOCKLEN_T=socklen_t -DUSE_OUR_BZERO=1 inet.c       
inet.c: In function `our_gethostbyname':
inet.c:68: warning: cast to pointer from integer of different size
inet.c: In function `our_bcopy':
inet.c:108: warning: cast from pointer to integer of different size
inet.c:109: warning: cast from pointer to integer of different size
inet.c:114: warning: cast from pointer to integer of different size
inet.c:136: warning: cast from pointer to integer of different size
inet.c:137: warning: cast from pointer to integer of different size
inet.c:138: warning: cast from pointer to integer of different size

c++ -c -Iinclude -I../UsageEnvironment/include -I. -O -DSOCKLEN_T=socklen_t -Wall -DBSD=1 -Wno-deprecated Groupsock.cpp
In file included from Groupsock.cpp:24:
include/TunnelEncaps.hh: In member function `unsigned int& 
   TunnelEncapsulationTrailer::auxAddress()':
include/TunnelEncaps.hh:58: warning: passing negative value `-4' for argument 1 
   of `char* TunnelEncapsulationTrailer::byteOffset(unsigned int)'
Groupsock.cpp: In member function `int 
   Groupsock::outputToAllMembersExcept(DirectedNetInterface*, unsigned char, 
   unsigned char*, unsigned int, unsigned int)':
Groupsock.cpp:334: warning: cast from pointer to integer of different size
Groupsock.cpp: In function `Boolean unsetGroupsockBySocket(const Groupsock*)':
Groupsock.cpp:416: warning: cast to pointer from integer of different size
Groupsock.cpp:418: warning: cast to pointer from integer of different size
Groupsock.cpp: In function `Boolean setGroupsockBySocket(UsageEnvironment&, 
   int, Groupsock*)':
Groupsock.cpp:443: warning: cast to pointer from integer of different size
Groupsock.cpp:453: warning: cast to pointer from integer of different size
Groupsock.cpp: In function `Groupsock* getGroupsockBySocket(int)':
Groupsock.cpp:468: warning: cast to pointer from integer of different size

c++ -c -Iinclude -I../UsageEnvironment/include -I. -O -DSOCKLEN_T=socklen_t -Wall -DBSD=1 -Wno-deprecated NetInterface.cpp
NetInterface.cpp: In member function `Socket* 
   SocketLookupTable::Fetch(UsageEnvironment&, Port, Boolean&)':
NetInterface.cpp:119: warning: cast to pointer from integer of different size
NetInterface.cpp:124: warning: cast to pointer from integer of different size
NetInterface.cpp: In member function `Boolean SocketLookupTable::Remove(const 
   Socket*)':
NetInterface.cpp:136: warning: cast to pointer from integer of different size

c++ -c -Iinclude -I../UsageEnvironment/include -I. -O -DSOCKLEN_T=socklen_t -Wall -DBSD=1 -Wno-deprecated IOHandlers.cpp
In file included from IOHandlers.cpp:22:
include/TunnelEncaps.hh: In member function `unsigned int& 
   TunnelEncapsulationTrailer::auxAddress()':
include/TunnelEncaps.hh:58: warning: passing negative value `-4' for argument 1 
   of `char* TunnelEncapsulationTrailer::byteOffset(unsigned int)'

make[1]: Leaving directory `/usr/src/packages/BUILD/MPlayer-0.90rc3/live/groupsock'

make[1]: Leaving directory `/usr/src/packages/BUILD/MPlayer-0.90rc3/live/UsageEnvironment'
make[1]: Entering directory `/usr/src/packages/BUILD/MPlayer-0.90rc3/live/BasicUsageEnvironment'
c++ -c -Iinclude -I../UsageEnvironment/include -I../groupsock/include -I. -O -DSOCKLEN_T=socklen_t -Wall -DBSD=1 -Wno-deprecated BasicUsageEnvironment.cpp
BasicUsageEnvironment.cpp: In member function `virtual void* 
   BasicTaskScheduler::scheduleDelayedTask(int, void (*)(void*), void*)':
BasicUsageEnvironment.cpp:207: warning: cast to pointer from integer of 
   different size
BasicUsageEnvironment.cpp: In member function `virtual void 
   BasicTaskScheduler::unscheduleDelayedTask(void*&)':
BasicUsageEnvironment.cpp:211: warning: cast from pointer to integer of 
   different size

c++ -c -Iinclude -I../UsageEnvironment/include -I../groupsock/include -I. -O -DSOCKLEN_T=socklen_t -Wall -DBSD=1 -Wno-deprecated BasicHashTable.cpp
BasicHashTable.cpp: In member function `unsigned int 
   BasicHashTable::hashIndexFromKey(const char*) const':
BasicHashTable.cpp:259: warning: cast from pointer to integer of different size

make[1]: Leaving directory `/usr/src/packages/BUILD/MPlayer-0.90rc3/live/BasicUsageEnvironment'


-- 
Philipp Thomas <pthomas(at)suse.de>
Development,SuSE Linux AG, Deutschherrnstr. 15-19, D-90429 Nuremberg, Germany


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X-Original-Date: Wed, 05 Feb 2003 13:49:49 -0800
Date: Wed, 05 Feb 2003 13:49:49 -0800

At 01:21 PM 2/5/03, you wrote:
>While trying to compile live.2003.01.10 on x86-64 (aka AMD Hammer), I
>noticed the warnings included below and found code isn't 64bit clean.

Guilty as charged.

Thanks for sending these warning messages.  I'll take a look at each of 
them, and see if I can fix them in the next installed version of the 
code.  (Longer term, I'll also be moving through the code, changing types 
like "unsigned" to (e.g.,) "u_int32_t", in places where exact sizes are 
important - e.g., in network protocol implementations.)

>And while I have your eyes, what are the chances of moving to ISO C++
>conforming headers, i.e. iostream instead of iostream.h and sstream
>instead of strstream.h? That would of cause also require some code changes,
>like using std:: where necessary.

To date I've been rather conservative about adopting using some of the 
'newer' features of C++, because I'm scared about breaking older compilers 
that people might be using. E.g., so far the code makes no use of 
templates, namespaces or RTTI, and only limited use of exceptions.

However, it's probably time to start modernizing the code, and in 
particular (as you note), using more ISO-conforming headers.  I'll probably 
start doing this over the next few weeks, provided that it doesn't break 
with MS Visual C++ version 5.0.  (That's the version of VC++ that I use; I 
feel like paying M$ any more money to upgrade to a newer version.)

Is the current header usage breaking for you, or is it just an annoyance 
(e.g., generating warning messages)?

         Ross.



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X-Original-Date: Wed, 05 Feb 2003 17:37:37 -0800
Date: Wed, 05 Feb 2003 17:37:37 -0800

FYI, I have now installed a new version (2003.02.06) of the code that 
should fix most (if not all) of the 64-bit-unsafe warnings that Philipp 
Thomas noticed.

	Ross.



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X-Original-Date: Thu, 06 Feb 2003 11:49:37 +1000
Date: Thu, 06 Feb 2003 11:49:37 +1000

Ross Finlayson wrote:
 >
> However, it's probably time to start modernizing the code, and in 
> particular (as you note), using more ISO-conforming headers.  I'll 
> probably start doing this over the next few weeks, provided that it 
> doesn't break with MS Visual C++ version 5.0.  (That's the version of 
> VC++ that I use; I feel like paying M$ any more money to upgrade to a 
> newer version.)

No need to pay for them, the Visual C++ command-line tools are free, see:
     http://www.winprog.org/tutorial/msvc.html

Cheers,
Doug.



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X-Original-Date: Thu, 6 Feb 2003 10:50:47 +0100
Date: Thu, 6 Feb 2003 10:50:47 +0100

* Ross Finlayson (finlayson(at)live.com) [20030205 22:54]:

> particular (as you note), using more ISO-conforming headers.  I'll probably 
> start doing this over the next few weeks, provided that it doesn't break 
> with MS Visual C++ version 5.0.  (That's the version of VC++ that I use; I 
> feel like paying M$ any more money to upgrade to a newer version.)

And there's at least one problem. iostream.h/strstream.h don't necessarily
put things in namespace std (the libstdc++ headers do, but the backwards
headers contain the necessary code to pull them into the global namespace),
so directly prefixing with std:: is out of the question. The other problem
would be the C compatibility headers (cstdio instead of stdio.h etc.) which
MSVC doesn't offer, at least not in version 5.

Seems like there is no way around some #ifdef guards. For the headers you
could only change the include for the *nix part and in that case use things
like 'using std::ostream' or even (what I dislike) 'using namespace std'.
Another possibility would be to use something STD and then define this to
be either std:: or nothing.

The inclusion of the C compatibility headers would necessitate moving the
inclusion of those C headers affected into parts guarded by __WIN32__, i.e.

#if defined(__WIN32__) || defined (WIN32)
#  include <stdio.h>
#  include <string.h>
#  include <errno.h>
#else
#  include <cstdio>
#  include <cstring>
#  include <cerrno>
#endif

 
> Is the current header usage breaking for you, or is it just an annoyance 
> (e.g., generating warning messages)?

Those headers only generate a warning because the backwards compat headers
include backward_warning.h, which has this:

#ifdef __DEPRECATED
#warning This file includes at least one deprecated or antiquated header. \
Please consider using one of the 32 headers found in section 17.4.1.2 of the \
C++ standard. Examples include substituting the <X> header for the <X.h> \
header for C++ includes, or <sstream> instead of the deprecated header \
<strstream.h>. To disable this warning use -Wno-deprecated.
#endif

These can be suppressed by adding -Wno-deprecated to the compiler flags,
but only gcc 3.X support that.

BTW, would you accept patches that make live use autoconf/automake on
platforms where they are available? That would make the package much more
versatile and would obliviate the need to patch config.linux. If yes, I'll
see what I can come up, though for next 2 months or so I'll be mostly busy
with working on the next release of SuSE Linux.

Philipp

-- 
Philipp Thomas <pthomas(at)suse.de>
SuSE Linux AG, Deutschherrnstr. 15-19, D-90429 Nuremberg, Germany


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X-Original-Date: Thu, 06 Feb 2003 11:21:27 -0800
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> > Is the current header usage breaking for you, or is it just an annoyance
> > (e.g., generating warning messages)?
>
>Those headers only generate a warning because the backwards compat headers
>include backward_warning.h
[...]
>These can be suppressed by adding -Wno-deprecated to the compiler flags,
>but only gcc 3.X support that.

Presumably earlier versions of gcc will just ignore the flag (and won't 
complain about deprecated headers)?  (So maybe I should add this flag to 
the Unix "config.*" files.)

>BTW, would you accept patches that make live use autoconf/automake on
>platforms where they are available?

The problem here, once again, is Windows.  I want to be able to generate 
Makefiles for Windows as well as Unix systems.  The 
"genMakefiles"/"genWindowsMakefiles" scripts do this (using the common 
"Makefile.tail" file for each library).  As far as I know, 
autoconf/automake can't generate Makefiles for Windows (although I'd love 
to be proven wrong here).

         Ross.



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X-Original-Date: Thu, 6 Feb 2003 15:00:45 -0500
Date: Thu, 6 Feb 2003 15:00:45 -0500

On Thu, Feb 06, 2003 at 11:21:27AM -0800, Ross Finlayson wrote:
> 
> >> Is the current header usage breaking for you, or is it just an annoyance
> >> (e.g., generating warning messages)?
> >
> >Those headers only generate a warning because the backwards compat headers
> >include backward_warning.h
> [...]
> >These can be suppressed by adding -Wno-deprecated to the compiler flags,
> >but only gcc 3.X support that.
> 
> Presumably earlier versions of gcc will just ignore the flag (and won't 
> complain about deprecated headers)?  (So maybe I should add this flag to 
> the Unix "config.*" files.)
> 
> >BTW, would you accept patches that make live use autoconf/automake on
> >platforms where they are available?
> 
> The problem here, once again, is Windows.  I want to be able to generate 
> Makefiles for Windows as well as Unix systems.  The 
> "genMakefiles"/"genWindowsMakefiles" scripts do this (using the common 
> "Makefile.tail" file for each library).  As far as I know, 
> autoconf/automake can't generate Makefiles for Windows (although I'd love 
> to be proven wrong here).

Well, it definately can make Makefiles that will work with MinGW gcc or
the cygnus gcc (and gnumake), but I don't know about it making stuff
compatible with the Microsoft toolchain.

And, since your library is C++, you probably need to compile
applications using your library all with the same compiler, so you
couldn't expect people to compile the live libraries with gcc, and the
rest of their app with Visual C++.

-SteveK

-- 
  Steve Kann - Chief Engineer - 520 8th Ave #2300 NY 10018 -  (212) 533-1775
                HorizonLive.com - collaborate . interact . learn
   "The box said 'Requires Windows 95, NT, or better,' so I installed Linux."


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X-Original-Date: Fri, 07 Feb 2003 11:11:02 +1000
Date: Fri, 07 Feb 2003 11:11:02 +1000

Ross Finlayson wrote:
 >
> The problem here, once again, is Windows.  I want to be able to generate 
> Makefiles for Windows as well as Unix systems.  The 
> "genMakefiles"/"genWindowsMakefiles" scripts do this (using the common 
> "Makefile.tail" file for each library).  As far as I know, 
> autoconf/automake can't generate Makefiles for Windows (although I'd 
> love to be proven wrong here).

Visual C++'s "cl.exe" works fine with autoconf, plus cl.exe accepts file 
names with the Unix directory delimiter (i.e '/') and can handle hyphens 
or slashes for the switches (e.g -I is the same as /I) making things 
easy. (Obviously, you need a POSIX like environment such as cygwin to 
build it)

The autoconf AC_OBJEXT & AC_EXEEXT also works with cl.exe. An example 
Makefile.in might contain:

--
# Suffix rule for compiling C++ source.
# Overrides the default cpp rule if any.
.SUFFIXES:
.SUFFIXES: .cpp .(at)OBJEXT(at)

.cpp.(at)OBJEXT(at) :
         $(CXX) -c -I. $(CPPFLAGS) $(CXXFLAGS) $<
--

What I usually do on Windows is make the configure script source another 
script which might contain something like the following:

--
CXX=cl
CC=cl

# C flags
CFLAGS="-nologo"

# C++ flags
CXXFLAGS="-O2 -GX -nologo"

CPPFLAGS="-D__WIN32__ -I../include"

# Linker flags
LDFLAGS="-nologo -link -libpath:\"${PROJ_LIBDIR}\" blah blah...
--

I'm not saying that this is how it should be done, just that it is possible.


Going back to the I/O Streams headers conversation, just thought I would 
mention that they won't be supported with the next release of VC++. 
VC++.NET prints out a lot of pragma deprecated warnings, here is an 
extract from the "useoldio.h" header where the pragma is defined:

--
/*
  * Warning C4995, '_OLD_IOSTREAMS_ARE_DEPRECATED' is a deprecated name,
  * is being issued because the old I/O Streams headers iostreams.h et al
  * will no longer be supported from VC8.  Replace references such as
  * #include <iostreams.h> with #include <iostreams>, using the new, more
  * conformant, I/O Streams headers.
  */

#pragma deprecated(_OLD_IOSTREAMS_ARE_DEPRECATED)
--

Cheers,
Doug.



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X-Original-Date: Fri, 7 Feb 2003 09:00:51 +0100
Date: Fri, 7 Feb 2003 09:00:51 +0100

* Ross Finlayson (finlayson(at)live.com) [20030206 20:26]:

> Presumably earlier versions of gcc will just ignore the flag (and won't 
> complain about deprecated headers)?  (So maybe I should add this flag to 
> the Unix "config.*" files.)

Nope, earlier versions won't just ignore the flag, so there would have to
be a test for gcc and its version somewhere in genMakefiles.
 
> The problem here, once again, is Windows.  I want to be able to generate 
> Makefiles for Windows as well as Unix systems.  The 
> "genMakefiles"/"genWindowsMakefiles" scripts do this (using the common 
> "Makefile.tail" file for each library).  As far as I know, 
> autoconf/automake can't generate Makefiles for Windows (although I'd love 
> to be proven wrong here).

autoconf/automake won't generate Makefiles for Windows. But that shouldn't
be that much of a problem. For Windows you could use a config.h.windows
that's copied to config.h by genWindowsMakefiles and the latter would
generate Makefiles for Windows just as it does now. Only on platforms that
are known to support automake/autoconf would they take over configuration.

Philipp

-- 
Philipp Thomas <pthomas(at)suse.de>
SuSE Linux AG, Deutschherrnstr. 15-19, D-90429 Nuremberg, Germany


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i am new to live.com and i have to design a streaming 
application uppon a CORBA bus. I wanted to know if live.com 
classes could be taken for an implementation (both client 
and server side)regarding compatibility with the 
Audio/video specification for Corba.
_________________________________________________________ 
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X-Original-Date: Sat, 08 Feb 2003 21:16:08 -0800
Date: Sat, 08 Feb 2003 21:16:08 -0800

At 01:40 PM 2/8/03, you wrote:
>i am new to live.com and i have to design a streaming
>application uppon a CORBA bus. I wanted to know if live.com
>classes could be taken for an implementation (both client
>and server side)regarding compatibility with the
>Audio/video specification for Corba.

I'm not familiar with how (if at all) media streaming is defined for CORBA, 
but if it uses the standard RTP protocol, then there should be no reason 
why you couldn't write a CORBA component that encapsulates whichever parts 
of the LIVE.COM code you're interested in.

         Ross.



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X-Original-Date: Sat, 08 Feb 2003 22:18:36 -0800
Date: Sat, 08 Feb 2003 22:18:36 -0800

FYI, the "liveMedia" library now implements a 'RTPSource' for motion-JPEG 
RTP streams.  See "JPEGVideoRTPSource.{hh,cpp}" for details.

(Thanks to Morgan Multimedia for contributing this code to the library.)

	Ross.



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X-Original-Date: Mon, 10 Feb 2003 02:33:08 -0800
Date: Mon, 10 Feb 2003 02:33:08 -0800

FYI, a new application "playSIP" has been added to the "testProgs" directory.

"playSIP" is similar to "openRTSP", except that it opens and plays a SIP 
session (using a "sip:" URL), rather than a RTSP session.  I.e., it can be 
used to record IP telephony sessions.

For more information on "playSIP", see: <http://www.live.com/playSIP/>

	Ross.



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X-Original-Date: Tue, 11 Feb 2003 17:20:55 +0100
Date: Tue, 11 Feb 2003 17:20:55 +0100


Hello, all you gurus,

i'm a bloody boaring beginner and (nearly) don't dare to ask
you this question:

Couldn't there be included a file with some installation hints?
What am i expected to do after a successful "make"?

Is it really that easy to only copy "libliveMedia.a" to /usr/local/lib?
Or do i really need the complete (un-tarred) source-tree "live" there?
Or is it in between? With the groupsocks-library and the testprogs
installed too?

With regards,

Bernt Christandl



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X-Original-Date: Tue, 11 Feb 2003 10:23:48 -0800
Date: Tue, 11 Feb 2003 10:23:48 -0800


>Couldn't there be included a file with some installation hints?
>What am i expected to do after a successful "make"?

Bernt,

There are no real 'installation' instructions.  Instead, you can just leave 
the "live" directory (with all of its subdirectories) as is.

>Is it really that easy to only copy "libliveMedia.a" to /usr/local/lib?

You don't have to copy the libraries ("libliveMedia.lib", 
"libgroupsock.lib", "libUsageEnvironment.lib", 
"libBasicUsageEnvironment.lib") anywhere, but if you do, you will also need 
to modify the Makefile in the "testProgs" directory accordingly.

>Or do i really need the complete (un-tarred) source-tree "live" there?

That is what I'd suggest, if you want the code to be generally 
available.  I.e., copy the entire "live/" directory to 
"/usr/local.lib".  (This makes it easier, BTW, if you want to use this code 
to implement RTSP/RTP streaming in "MPlayer" - see 
<http://www.live.com/mplayer/> for details.)

         Ross  Finlayson
         LIVE.COM
         <http://www.live.com/>



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X-Original-Date: Wed, 12 Feb 2003 07:17:02 +0100
Date: Wed, 12 Feb 2003 07:17:02 +0100


Ross,

thank you very much for your fast answer to my question!

> >Couldn't there be included a file with some installation hints?
> >What am i expected to do after a successful "make"?
> 
> Bernt,
> 
> There are no real 'installation' instructions.  Instead, you can just leave 
> the "live" directory (with all of its subdirectories) as is.
> 
> >Is it really that easy to only copy "libliveMedia.a" to /usr/local/lib?
> 
> You don't have to copy the libraries ("libliveMedia.lib", 
> "libgroupsock.lib", "libUsageEnvironment.lib", 
> "libBasicUsageEnvironment.lib") anywhere, but if you do, you will also need 
> to modify the Makefile in the "testProgs" directory accordingly.
> 
> >Or do i really need the complete (un-tarred) source-tree "live" there?
> 
> That is what I'd suggest, if you want the code to be generally 
> available.  I.e., copy the entire "live/" directory to 
> "/usr/local.lib".  (This makes it easier, BTW, if you want to use this code 
> to implement RTSP/RTP streaming in "MPlayer" - see 
> <http://www.live.com/mplayer/> for details.)
> 
>          Ross  Finlayson
>          LIVE.COM
>          <http://www.live.com/>

With regards,

Bernt Christandl



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X-Original-Date: Fri, 14 Feb 2003 13:15:52 +0100
Date: Fri, 14 Feb 2003 13:15:52 +0100

--=====_10452249523124=_
Content-Type: text/plain; charset="ISO-8859-1"
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Hi!
I'm a collage student and I'm working on a some kind of conferencing=
 software. 
What I'm suposed to do is capture video from camera, encode it to mpeg4 and=
 
then stream over internet, preferably multicast. I'm working under Win xp.=
 So 
for capturing and encoding I'm using directshow filters, but the streaming=
 is 
a problem. I'd like to use rtp protocol. Can I use any of the code from=
 LIVE.COM Streaming Media?
I have seen a rather old post at the mailing list where someone asked a=
 similar thing, and the answer 
was that a mpeg4 payload type for rtp should be implemented instead of=
 mpeg2 payload type in
MPEGAudioRTPSink and MPEGVideoRTPSink classes.

I have read rfc 3016 but it didn't help me much. :)
I really have no idea how to do this, so some advices about that would also=
 be nice.

Any help like links,docs,ideas are welcome because I'm really stuck with=
 this.

If this is not an appropriate message for this forum you can contact me at=
 my email:
sancho(at)fly.cc.fer.hr

Thank you in advance.


--=====_10452249523124=_
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<DIV>Hi!<BR>I'm a collage student and I'm working on a some kind of conferencing 
software. </DIV>
<DIV>What I'm suposed to do is capture video from camera, encode it to mpeg4 and 
</DIV>
<DIV>then stream over internet, preferably multicast. I'm working under Win xp. 
So </DIV>
<DIV>for capturing and encoding I'm using directshow filters, but the streaming 
is </DIV>
<DIV>a problem. I'd like to use rtp protocol. Can I use any of the code from 
LIVE.COM Streaming Media?</DIV>
<DIV>I&nbsp;have seen&nbsp;a rather old post at the mailing&nbsp;list where 
someone asked a similar thing, and the answer </DIV>
<DIV>was that a mpeg4 payload type for rtp should be implemented instead of 
mpeg2 payload type in</DIV>
<DIV>MPEGAudioRTPSink and MPEGVideoRTPSink classes.</DIV>
<DIV>&nbsp;</DIV>
<DIV>I have read rfc 3016 but it didn't help me much. :)</DIV>
<DIV>I really have no idea how to do this, so some advices about that would also 
be nice.</DIV>
<DIV>&nbsp;</DIV>
<DIV>Any help like links,docs,ideas are welcome because I'm really stuck with 
this.</DIV>
<DIV>&nbsp;</DIV>
<DIV>If this is not an appropriate message for this forum you can contact me at 
my email:<BR><A 
href="mailto:sancho(at)fly.cc.fer.hr">sancho(at)fly.cc.fer.hr</A></DIV>
<DIV>&nbsp;</DIV>
<DIV>Thank you in advance.</DIV></BODY></HTML>


--=====_10452249523124=_--



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X-Original-Date: Fri, 14 Feb 2003 13:21:34 +0100
Date: Fri, 14 Feb 2003 13:21:34 +0100

--=====_10452252942703=_
Content-Type: text/plain; charset="ISO-8859-1"
Content-Transfer-Encoding: quoted-printable

Hi!
I'm a collage student and I'm working on a some kind of conferencing=
 software. 
What I'm suposed to do is capture video from camera, encode it to mpeg4 and=
 
then stream over internet, preferably multicast. I'm working under Win xp.=
 So 
for capturing and encoding I'm using directshow filters, but the streaming=
 is 
a problem. I'd like to use rtp protocol. Can I use any of the code from=
 LIVE.COM Streaming Media?
I have seen a rather old post at the mailing list where someone asked a=
 similar thing, and the answer 
was that a mpeg4 payload type for rtp should be implemented instead of=
 mpeg2 payload type in
MPEGAudioRTPSink and MPEGVideoRTPSink classes.

I have read rfc 3016 but it didn't help me much. :)
I really have no idea how to do this, so some advices about that would also=
 be nice.

Any help like links,docs,ideas are welcome because I'm really stuck with=
 this.

If this is not an appropriate message for this forum you can contact me at=
 my email:
sancho(at)fly.cc.fer.hr

Thank you in advance.


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<DIV>Hi!<BR>I'm a collage student and I'm working on a some kind of conferencing 
software. </DIV>
<DIV>What I'm suposed to do is capture video from camera, encode it to mpeg4 and 
</DIV>
<DIV>then stream over internet, preferably multicast. I'm working under Win xp. 
So </DIV>
<DIV>for capturing and encoding I'm using directshow filters, but the streaming 
is </DIV>
<DIV>a problem. I'd like to use rtp protocol. Can I use any of the code from 
LIVE.COM Streaming Media?</DIV>
<DIV>I&nbsp;have seen&nbsp;a rather old post at the mailing&nbsp;list where 
someone asked a similar thing, and the answer </DIV>
<DIV>was that a mpeg4 payload type for rtp should be implemented instead of 
mpeg2 payload type in</DIV>
<DIV>MPEGAudioRTPSink and MPEGVideoRTPSink classes.</DIV>
<DIV>&nbsp;</DIV>
<DIV>I have read rfc 3016 but it didn't help me much. :)</DIV>
<DIV>I really have no idea how to do this, so some advices about that would also 
be nice.</DIV>
<DIV>&nbsp;</DIV>
<DIV>Any help like links,docs,ideas are welcome because I'm really stuck with 
this.</DIV>
<DIV>&nbsp;</DIV>
<DIV>If this is not an appropriate message for this forum you can contact me at 
my email:<BR><A 
href="mailto:sancho(at)fly.cc.fer.hr">sancho(at)fly.cc.fer.hr</A></DIV>
<DIV>&nbsp;</DIV>
<DIV>Thank you in advance.</DIV></BODY></HTML>


--=====_10452252942703=_--



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From <EMAIL-ADDRESS-HIDDEN> Fri Feb 14 13:32:49 2003
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X-Original-Date: Fri, 14 Feb 2003 13:22:16 -0800
Date: Fri, 14 Feb 2003 13:22:16 -0800


>I'm a collage student and I'm working on a some kind of conferencing 
>software.
>What I'm suposed to do is capture video from camera, encode it to mpeg4 and
>then stream over internet, preferably multicast. I'm working under Win xp. So
>for capturing and encoding I'm using directshow filters, but the streaming is
>a problem. I'd like to use rtp protocol. Can I use any of the code from 
>LIVE.COM Streaming Media?

Yes, you could use the "testMPEGAudioVideoStreamer" application as a model, 
but there is some more code that needs to be written (see below).

>I have seen a rather old post at the mailing list where someone asked a 
>similar thing, and the answer
>was that a mpeg4 payload type for rtp should be implemented instead of 
>mpeg2 payload type in
>MPEGAudioRTPSink and MPEGVideoRTPSink classes.

That's correct.   Appropriate MPEG-4 audio and video RTPSink classes would 
need to be written, and added to the library.  (This is on our "to do" 
list, but has not yet been funded.)

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>



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Subject: [Live-devel] cisco ipphone rtp streaming
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X-Original-Date: Mon, 17 Feb 2003 16:32:52 +0900
Date: Mon, 17 Feb 2003 16:32:52 +0900

dear all,

I need to record certain user's voice from cisco ipphone into a file in a server.

Like mentioned in Cisco documentation about  IPPhone Services
Application Development:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_programming_usage_guide09186a00800d6a66.html,

It is possible to invoke RTP streaming via URIs in IPPhone services.
We can instruct the phone to transmit an RTP stream, using
"RTPTx://x.x.x.x:xxxx" syntax to indicate the IPAddress to which the
phone streams, followed by destination port.

But does anyone know how to receive that rtp stream, since there is no
call-setup information..(say, if I want to receive that stream from an
h323 server..)

TIA  

-- 
Best regards,
 handy                          http://handy-tita.da.ru



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X-Original-Date: Mon, 17 Feb 2003 13:49:16 -0800
Date: Mon, 17 Feb 2003 13:49:16 -0800


>I need to record certain user's voice from cisco ipphone into a file in a 
>server.

If you have a "sip:" URL for the call, then you can try using our "playSIP" 
tool <http://www.live.com/playSIP/> to record it.

Alternatively, you could install PBX software like Asterisk 
<http://www.asterisk.org/>, and have your user make a call into this.  You 
can configure Asterisk to make a recording of the call.

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>



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X-Original-Date: Wed, 26 Feb 2003 03:34:14 -0800 (PST)
Date: Wed, 26 Feb 2003 03:34:14 -0800 (PST)

Dear all,

I have build Mplayer using RTSP/RTP libraries from
live.com with --enable-live option. I tried to stram
the data from Streaming21 and Darwin server but could
not succeed. I am using the command 

mplayer rtsp://192.168.0.9/clip1.avi
with  S21 server the response is 

Playing rtsp://192.168.0.9/clip1.avi 
Connected to server 192.168.0.9
stream not seekable!
Failed to get SDP description from URL
"rtsp://192.168.0.9/clip1.avi" :connect() failed:
connection refused.
No stream found.

With Darwin server the error is 

Playing rtsp://192.168.0.9/clip1.avi 
Connected to server 192.168.0.9
stream not seekable!
Failed to get SDP description from URL
"rtsp://192.168.0.9/clip1.avi" :can not handle
DESCRIBE response: RTSP/1.0 404 Not Found.
No stream found.

Pls suggest what could be the reason?.
If the live.com RTSP/RTP protocol is not compatible
with S21 and Darwin server, Which servers I can use
with this player?

Thanks,
Subhash

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X-Original-Date: Wed, 26 Feb 2003 03:35:10 -0800 (PST)
Date: Wed, 26 Feb 2003 03:35:10 -0800 (PST)

Dear all,

I have build Mplayer using RTSP/RTP libraries from
live.com with --enable-live option. I tried to stram
the data from Streaming21 and Darwin server but could
not succeed. I am using the command 

mplayer rtsp://192.168.0.9/clip1.avi
with  S21 server the response is 

Playing rtsp://192.168.0.9/clip1.avi 
Connected to server 192.168.0.9
stream not seekable!
Failed to get SDP description from URL
"rtsp://192.168.0.9/clip1.avi" :connect() failed:
connection refused.
No stream found.

With Darwin server the error is 

Playing rtsp://192.168.0.9/clip1.avi 
Connected to server 192.168.0.9
stream not seekable!
Failed to get SDP description from URL
"rtsp://192.168.0.9/clip1.avi" :can not handle
DESCRIBE response: RTSP/1.0 404 Not Found.
No stream found.

Pls suggest what could be the reason?.
If the live.com RTSP/RTP protocol is not compatible
with S21 and Darwin server, Which servers I can use
with this player?

Thanks,
Subhash

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X-Original-Date: Wed, 26 Feb 2003 03:59:20 -0800
Date: Wed, 26 Feb 2003 03:59:20 -0800

At 03:35 AM 2/26/03, you wrote:
>Dear all,
>
>I have build Mplayer using RTSP/RTP libraries from
>live.com with --enable-live option. I tried to stram
>the data from Streaming21 and Darwin server but could
>not succeed.

I notice that your file name ends with ".avi".  It's unlikely that any RTSP 
server can stream an AVI file (because AVI is a Microsoft-specific format).

I suggest first trying to play your stream using QuickTime Player (on 
Windows or Mac).  If QuickTime Player cannot play a "rtsp://" URL, then 
MPlayer probably won't be able to play it either.

         Ross.



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Subject: [Live-devel] testMPEGAudioVideoStreamer and network flooding
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X-Original-Date: Wed, 26 Feb 2003 12:31:31 -0800
Date: Wed, 26 Feb 2003 12:31:31 -0800


Hello, Ross et al ...

I'm studying the liveMedia stuff (and learning quite a lot, thank you), 
but I run into problems when I run the testMPEGAudioVideoStreamer 
program.  Basically, if I run it at home, it adversely affects my Apple 
Airport Base Station (snow model, not Airport Extreme) i.e. it seems to 
flood it with packets, and no one else can use the base station to 
connect to the outside world.  If I run it on a friend's wired network, 
it does a similar thing to his router / gateway box i.e. other users on 
the network can't get out.

I suspect that there is something I can configure, either on my iBook 
(where I run the test) or my base station, that would stop this from 
happening.  Can someone give me a hint as to what it might be?  I'm no 
stranger to sockets and network programming / streaming, but i'm 
relatively new to multicast and router configuration.  Any assistance 
would be greatly appreciated.

Regards,

John

Falling You - exploring the beauty of voice and sound
New EP, "hope thrown down," available now at
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  flooding
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X-Original-Date: Wed, 26 Feb 2003 14:10:10 -0800
Date: Wed, 26 Feb 2003 14:10:10 -0800


>I'm studying the liveMedia stuff (and learning quite a lot, thank you), 
>but I run into problems when I run the testMPEGAudioVideoStreamer 
>program.  Basically, if I run it at home, it adversely affects my Apple 
>Airport Base Station (snow model, not Airport Extreme) i.e. it seems to 
>flood it with packets, and no one else can use the base station to connect 
>to the outside world.  If I run it on a friend's wired network, it does a 
>similar thing to his router / gateway box i.e. other users on the network 
>can't get out.

It sounds like the bit rate of your MPEG file is simply too high for your 
network.

Either use a lower-bitrate movie, or else set the "iFramesOnly" variable 
(in "testMPEGAudioVideoStreamer.cpp" to "True".  (If you do that, you'll 
stream 'key' frames only.)

         Ross.



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X-Original-Date: Thu, 27 Feb 2003 09:48:13 +1100
Date: Thu, 27 Feb 2003 09:48:13 +1100

Subhash Patil wrote:

>With Darwin server the error is 
>
>Playing rtsp://192.168.0.9/clip1.avi 
>Connected to server 192.168.0.9
>stream not seekable!
>Failed to get SDP description from URL
>"rtsp://192.168.0.9/clip1.avi" :can not handle
>DESCRIBE response: RTSP/1.0 404 Not Found.
>No stream found.
>
>
>  
>
mplayer/darwin server works OK for me.
Darwin server requies "hinted" files though.
Try to connect the server using openRTSP (with -r)
or quicktime movie to verify whether the server accept your files first.

Hope that helps,
Quy.





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X-Original-Date: Wed, 26 Feb 2003 21:43:49 -0800 (PST)
Date: Wed, 26 Feb 2003 21:43:49 -0800 (PST)

> Date: Wed, 26 Feb 2003 03:59:20 -0800
> To: live-devel(at)lists.sourceforge.net
> From: Ross Finlayson <finlayson(at)live.com>
> Subject: Re: [Live-devel] RTSP/RTP supported servers
> Reply-To: live-devel(at)lists.sourceforge.net
> 
> At 03:35 AM 2/26/03, you wrote:
> >Dear all,
> >
> >I have build Mplayer using RTSP/RTP libraries from
> >live.com with --enable-live option. I tried to
> stram
> >the data from Streaming21 and Darwin server but
> could
> >not succeed.
> 
> I notice that your file name ends with ".avi".  It's
> unlikely that any RTSP 
> server can stream an AVI file (because AVI is a
> Microsoft-specific format).
> 
> I suggest first trying to play your stream using
> QuickTime Player (on 
> Windows or Mac).  If QuickTime Player cannot play a
> "rtsp://" URL, then 
> MPlayer probably won't be able to play it either.
 
Sorry, I have tried .mp3, .mpg and .mp4 files but the
results are same. What about the servers like Kassena
and OKI?. Anyway I will check files with QT player.

Thanks,
Subhash

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From <EMAIL-ADDRESS-HIDDEN> Thu Feb 27 05:39:25 2003
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X-Original-Date: Fri, 28 Feb 2003 00:38:44 +1100
Date: Fri, 28 Feb 2003 00:38:44 +1100

Hi all,

This is a patch to MediaSession to enable RTCP packets to be sent to the 
server by the client. Presently the client uses 1 as the server's RTCP port, 
due to the value subsession->serverPortNum not being initialised. The patched 
code causes the client to assume the server will receive RTCP packets on the 
port that the server suggests the client receive RTCP packets on.

--- live/liveMedia/MediaSession.cpp     2003-02-17 20:59:48.000000000 +1100
+++ /tmp/live/liveMedia/MediaSession.cpp        2003-02-27 22:19:11.000000000 
+1100
(at)(at) -131,6 +131,8 (at)(at)
       delete mediumName;
       return False;
     }
+    subsession->serverPortNum = subsession->fClientPortNum;
+
     char const* mStart = sdpLine;
     subsession->fSavedSDPLines = strdup(mStart);


This is a patch to enable the RTP Payload format to be obtained from a 
MediaSession object.

--- live/liveMedia/include/MediaSession.hh      2003-02-17 20:59:48.000000000 
+1100
+++ /tmp/live/liveMedia/include/MediaSession.hh 2003-02-27 22:28:39.000000000 
+1100
(at)(at) -118,6 +118,7 (at)(at)
   unsigned short videoWidth() const { return fVideoWidth; }
   unsigned short videoHeight() const { return fVideoHeight; }
   unsigned videoFPS() const { return fVideoFPS; }
+  unsigned char RTPPayloadFormat() const { return fRTPPayloadFormat; }

   RTPSource* rtpSource() { return fRTPSource; }
   RTCPInstance* rtcpInstance() { return fRTCPInstance; }

I have been working on some software that takes advantage of the LIVE.COM 
libraries. Users of the software are able to use their machines as streaming 
media repeaters (in the absence of multicasting). They can effectively pool 
together unused bandwidth, which can be called upon when needed.

http://www.trevbus.org/trevbus_2.0b.tgz

Alex Pollard




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Subject: Re: [Live-devel] Proposed improvements to MediaSession class
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X-Original-Date: Thu, 27 Feb 2003 14:28:06 -0800
Date: Thu, 27 Feb 2003 14:28:06 -0800


>This is a patch to MediaSession to enable RTCP packets to be sent to the
>server by the client. Presently the client uses 1 as the server's RTCP port,
>due to the value subsession->serverPortNum not being initialised. The patched
>code causes the client to assume the server will receive RTCP packets on the
>port that the server suggests the client receive RTCP packets on.

Yes - good catch.  Note, though, that this applies only for multicast 
sessions and others for which the RTSP "SETUP" response does not include a 
"server_port=" entry in the "Transport:" header.  (If your RTSP server *is* 
returning a "Transport:" header with a "server_port=" entry, then that 
should be used.)

I'll include this fix in the next software release.

>This is a patch to enable the RTP Payload format to be obtained from a
>MediaSession object.
[...]
>+  unsigned char RTPPayloadFormat() const { return fRTPPayloadFormat; }

Actually, another way to get this (without modifying the code) is to do
         subsession->rtpSource()->rtpPayloadFormat()

This assumes that "subsession" has already been initiate()d, to create a 
RTPSource.

(In the future, I may make it possible for a "MediaSubsession" to handle 
more than one RTP payload format code - so I'd like to avoid having too 
many places in the code that assume that there's only one such code.)

         Ross.



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From: B Prashanth <EMAIL-ADDRESS-HIDDEN>
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X-Original-Date: Fri, 28 Feb 2003 15:06:29 +0530
Date: Fri, 28 Feb 2003 15:06:29 +0530

Hi all,
I had a very fundamental question with respect to the multicast RTP 
session. How does RTP handle incoming packets, coming from number of 
sources, on a multicast port?
1. Are the packets (coming from different SSRCs) given to the 
application using the RTP stack in the order of their arrival?
OR,
2. Seperate jitter buffers would be needed to be maintained in the RTP 
stack for each of the SSRCs (senders of RTP data) so that the jitter 
from each of the senders is removed individually and then the payload be 
given to the application that is using a RTP stack?

Thanks in advance,

-- 
Prashanth 
       




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X-Original-Date: Sun, 2 Mar 2003 16:58:06 -0500
Date: Sun, 2 Mar 2003 16:58:06 -0500

This weekend I just began to use your libraries to complete a school
project I'm working on.  I'm trying to stream MPEG1 video out of a linux
box using the liveMedia libraries and I was able to get it to work from
a file using the testMPEGVideoStreamer.  However, what would be the best
way to use the libraries given a file descriptor of a stream?  I tried
it one way, but I ran into some problems.  This is what I did...  I
first opened a stream using the open() method from my MPEG1 encoder.
Then, I used fdopen() to return a file pointer.  Then, I unprotected the
constructor in ByteStreamFileSource.cpp which accepts a file pointer and
created a ByteStreamFileSource.  I was able to stream it this way to my
laptop where I was running quicktime using the testMPEGVideoStreamer.
However, I'm getting major jitter and a 3 second delay.  I'm thinking
there must be a better and much faster way... help.  Is there?  I would
greatly appreciate some help.

Thanks.

-josh unger



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X-Original-Date: Mon, 03 Mar 2003 04:34:33 -0800
Date: Mon, 03 Mar 2003 04:34:33 -0800


>I'm trying to stream MPEG1 video out of a linux
>box using the liveMedia libraries and I was able to get it to work from
>a file using the testMPEGVideoStreamer.  However, what would be the best
>way to use the libraries given a file descriptor of a stream?  I tried
>it one way, but I ran into some problems.  This is what I did...  I
>first opened a stream using the open() method from my MPEG1 encoder.
>Then, I used fdopen() to return a file pointer.  Then, I unprotected the
>constructor in ByteStreamFileSource.cpp which accepts a file pointer and
>created a ByteStreamFileSource.

Josh,

I didn't totally follow your explanation, but is it correct that you are 
modifying the code to simply read from an input MPEG encoder - as a byte 
stream - rather than from a file?  If so, then this should work.

Another, perhaps simpler, way to do this would be to leave 
"ByteStreamFileSource" as is, but change "testMPEGVideoStreamer" to read 
from the file "stdin", rather than "test.mpg".  If you do this, you can 
simply pipe your MPEG encoder into "testMPEGVideoStreamer".

>   I was able to stream it this way to my
>laptop where I was running quicktime using the testMPEGVideoStreamer.
>However, I'm getting major jitter and a 3 second delay.

Most of the 3-second delay is probably happening inside QuickTime 
Player.  That was designed to be a media clip (or stream) player, and not 
necessarily to have low latency.

I'm not sure what would be causing the jitter, but one thing you can try - 
for testing - is run "testMPEGVideoReceiver > foo.mpg" on the receiving 
machine, and then try playing the resulting file "foo.mpg", to make sure 
it's OK.  This will tell you whether the video data is getting streamed OK.

         Ross.



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Subject: RE: [Live-devel] testMPEGVideoStreamer and live streaming
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X-Original-Date: Mon, 3 Mar 2003 09:01:44 -0500
Date: Mon, 3 Mar 2003 09:01:44 -0500

Ross,

Yes.  My code is sorta like this:

int fd;
fd = open("/dev/myencoder/stream");

//set some io ctrl information on the mpeg1 board

FILE* fp;
fp = fdopen(fd, "rb");

// create a new ByteStreamFileSource here

I'm not at my linux box to get the exact coding I'm using, but it's very
similar to the above.  I just learned c and c++ a couple of months ago,
so I apologize if my coding is wrong.

I'll try running the receiver application and see what it produces.  I
appreciate the help.

Thanks!

-josh

-----Original Message-----
From: <EMAIL-ADDRESS-HIDDEN>
[mailto:live-devel-admin(at)lists.sourceforge.net] On Behalf Of Ross
Finlayson
Sent: Monday, March 03, 2003 7:35 AM
To: <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] testMPEGVideoStreamer and live streaming


>I'm trying to stream MPEG1 video out of a linux
>box using the liveMedia libraries and I was able to get it to work from
>a file using the testMPEGVideoStreamer.  However, what would be the
best
>way to use the libraries given a file descriptor of a stream?  I tried
>it one way, but I ran into some problems.  This is what I did...  I
>first opened a stream using the open() method from my MPEG1 encoder.
>Then, I used fdopen() to return a file pointer.  Then, I unprotected
the
>constructor in ByteStreamFileSource.cpp which accepts a file pointer
and
>created a ByteStreamFileSource.

Josh,

I didn't totally follow your explanation, but is it correct that you are

modifying the code to simply read from an input MPEG encoder - as a byte

stream - rather than from a file?  If so, then this should work.

Another, perhaps simpler, way to do this would be to leave 
"ByteStreamFileSource" as is, but change "testMPEGVideoStreamer" to read

from the file "stdin", rather than "test.mpg".  If you do this, you can 
simply pipe your MPEG encoder into "testMPEGVideoStreamer".

>   I was able to stream it this way to my
>laptop where I was running quicktime using the testMPEGVideoStreamer.
>However, I'm getting major jitter and a 3 second delay.

Most of the 3-second delay is probably happening inside QuickTime 
Player.  That was designed to be a media clip (or stream) player, and
not 
necessarily to have low latency.

I'm not sure what would be causing the jitter, but one thing you can try
- 
for testing - is run "testMPEGVideoReceiver > foo.mpg" on the receiving 
machine, and then try playing the resulting file "foo.mpg", to make sure

it's OK.  This will tell you whether the video data is getting streamed
OK.

         Ross.



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X-Original-Date: Tue, 4 Mar 2003 11:46:50 +0200
Date: Tue, 4 Mar 2003 11:46:50 +0200

Hi,
I started using the live library and it is working really well.
I have a question if there is any plans to implement the RTSP command
Pause and the option to seek a stream (via the PLAY command  Range:
attribute ).

I tried to see if the base rtp and sink classes have the option to pause
or seek on a stream and I found that you can only perform start and stop
playing is that true? If not it will be very helpful if you can tell me
how the base class can do that 
Thanks 
eran
   



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Subject: [Live-devel] MPEG4 Video and Handheld Support?
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Hi,
=20
I=92ve recently compiled your rtp libraries for windows.
There appears to be a header =93MPEG4ESVideoRTPSource.hh=94.
Does this encapsulate the mpeg 4 video support?=20
Do the libraries support ISO MPEG4 video currently?
=20
We would like to stream via a push mechanism from an encoder to a server
to a handheld device.
Has anyone used the live.com libraries for pocket pc support?
=20
Many thanks,
Ed Ruffing
=20

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<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Hi,<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>I&#8217;ve recently compiled your <span =
class=3DSpellE>rtp</span>
libraries for windows.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>There appears to be a header
&#8220;MPEG4ESVideoRTPSource.hh&#8221;.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Does this encapsulate the mpeg 4 video support? =
<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Do the libraries support ISO MPEG4 video =
currently?<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>We would like to stream via a push mechanism from an =
encoder
to a server to a handheld device.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Has anyone used the live.com libraries for pocket pc
support?<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Many thanks,<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Ed Ruffing<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

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X-Original-Date: Tue, 04 Mar 2003 13:19:19 -0800
Date: Tue, 04 Mar 2003 13:19:19 -0800


>I started using the live library and it is working really well.
>I have a question if there is any plans to implement the RTSP command
>Pause and the option to seek a stream (via the PLAY command  Range:
>attribute ).

Yes, this is on our "to do" list, but it's not a funded task, so is 
relatively low priority right now...

>I tried to see if the base rtp and sink classes have the option to pause
>or seek on a stream and I found that you can only perform start and stop
>playing is that true?

At the client end, supporting RTSP "PAUSE" and "PLAY" (with ranges) should 
be require only modifications to the "RTSPClient" class.  (No changes to 
the RTP implementation are needed, because it's the server - at the remote 
end - that would actually pause sending the RTP stream.)

OTOH, adding "PAUSE" and range "PLAY" support to our RTSP *server* 
implementation (i.e., "RTSPServer") would be more work (because this RTSP 
server implementation is still quite limited).

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>



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X-Original-Date: Tue, 04 Mar 2003 13:27:28 -0800
Date: Tue, 04 Mar 2003 13:27:28 -0800


>I ve recently compiled your rtp libraries for windows.
>
>There appears to be a header MPEG4ESVideoRTPSource.hh .
>
>Does this encapsulate the mpeg 4 video support?

Yes, this is intended for use by clients that wish to receive/play incoming 
MPEG-4 video RTP streams.

>Do the libraries support ISO MPEG4 video currently?

Yes, the libraries currently have 'RTP source' classes for receiving MPEG-4 
video and audio (generic or LATM).  However, they don't yet have 
corresponding 'RTP sink' classes for *sending* MPEG-4 RTP streams.  (As 
always, should your company be interested in providing funding for this, 
please let me know by separate email.)

>We would like to stream via a push mechanism from an encoder to a server 
>to a handheld device.
>
>Has anyone used the live.com libraries for pocket pc support?

Yes, people have done this, although I've heard reports that the code 
and/or Makefiles need to be tweaked somewhat to work with Microsoft's 
"Embedded Visual C++" tools.

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>



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Subject: [Live-devel] compiling with mplayer fails
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X-Original-Date: Thu, 6 Mar 2003 00:05:46 -0500
Date: Thu, 6 Mar 2003 00:05:46 -0500

Hi.

Trying to compile your stuff into mplayer, and having problems.

Using redhat 7.3.

Downloaded and installed livelibs per instructions on the web page.

Then, grabbed mplayer 20030304 cvs.

./configure --disable-divx4linux --disable-libavcodec --enable-live
--with-livelibdir=/usr/lib/live

and, when we get to the link stage, I get a lot of errors. Here's the tail end:

_tf15DelayQueueEntry+0x1d): undefined reference to `__rtti_user'
/usr/lib/live/BasicUsageEnvironment/libBasicUsageEnvironment.a: In function
`DelayQueue type_info function':
/usr/lib/live/BasicUsageEnvironment/libBasicUsageEnvironment.a(.gnu.linkonce.t.__tf10DelayQueue+0x27):
undefined reference to `__rtti_si'
/usr/lib/live/BasicUsageEnvironment/libBasicUsageEnvironment.a: In function
`BasicHashTable::Iterator::~Iterator(void)':
/usr/lib/live/BasicUsageEnvironment/libBasicUsageEnvironment.a(.gnu.linkonce.t._._Q214BasicHashTable8Iterator+0x2c):
undefined reference to `__builtin_delete'
/usr/lib/live/BasicUsageEnvironment/libBasicUsageEnvironment.a: In function
`BasicHashTable::Iterator type_info function':
/usr/lib/live/BasicUsageEnvironment/libBasicUsageEnvironment.a(.gnu.linkonce.t.__tfQ214BasicHashTable8Iterator+0x27):
undefined reference to `__rtti_si'
/usr/lib/live/BasicUsageEnvironment/libBasicUsageEnvironment.a: In function
`BasicHashTable type_info function':
/usr/lib/live/BasicUsageEnvironment/libBasicUsageEnvironment.a(.gnu.linkonce.t.__tf14BasicHashTable+0x27):
undefined reference to `__rtti_si'
collect2: ld returned 1 exit status
make: *** [mplayer] Error 1

I should add the compile of mplayer appears to be detecting the gcc3.1 install
and using gcc3 for the compile.

Any help is greatly appreciated.

Thanks!

-- 
********************************
David William Botsch
Consultant/Advisor II
CCMR Computing Facility
dwb7(at)ccmr.cornell.edu
********************************


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X-Original-Date: Wed, 05 Mar 2003 21:23:49 -0800
Date: Wed, 05 Mar 2003 21:23:49 -0800

It looks like the final linking of "mplayer" is not including the standard 
C++ libraries.  Is "-lstdc++" included on the link line?  (It should be.)

Were you able to successfully link the LIVE.COM test programs (in 
"live/testProgs/")?  If so, then you should be able to link mplayer also.

>I should add the compile of mplayer appears to be detecting the gcc3.1 install
>and using gcc3 for the compile.

That may (or may not) be significant - I'm not sure.  In any case, make 
sure that you're using the same compilers to compile both the LIVE.COM 
code, and the "mplayer" code.

         Ross.



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X-Original-Date: Thu, 6 Mar 2003 15:19:14 +0800
Date: Thu, 6 Mar 2003 15:19:14 +0800

Hello,

    I've been trying to stream an mpeg 4 file with no luck using Darwin
Streaming Server 4.1.2 on Linux. The scenario is as follow:
1. the mpeg 4 file is converted from avi using QuickTime Pro 6
2. the player is mplayer 0.90rc4 with LIVE.COM Streaming Media v2003.02.17
3. when playing with
    mplayer -v 3 rtsp://server/video.mp4
   the mplayer aborted with following messages (among others)

// error message
==> Found audio stream: 0
Unknown MPlayer format code for MIME type "audio/MPEG4-GENERIC"
[V] filefmt:21  fourcc:0x7634706D  size:0x0  fps: 0.00  ftime:=0.0000
FPS not specified in the header or invalid, use the -fps option.
// error message end

4. The media file can be played locally, that is
   mplayer /path-to-the-file
5. The media file can be played by mpeg4ip player using rtsp.

   It seems to me that the problem might be the audio stream MIME type that
is not recognized by mplayer. If this is the case, how can I specify it in
mplayer? Or did I miss anything?

Rgds,
Loren




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X-Original-Date: Thu, 06 Mar 2003 00:24:05 -0800
Date: Thu, 06 Mar 2003 00:24:05 -0800

MPlayer's RTSP/RTP support doesn't yet handle MPEG-4 video or audio (except 
MPEG-4 LATM audio).  The LIVE.COM libraries implement MPEG-4 audio/video 
reception OK (so, for example, "openRTSP" can be used to download MPEG-4 
RTSP/RTP streams), but MPlayer hasn't yet been updated to handle this.

The good news, though, is that this will be working Real Soon Now (within a 
few days).  I'll send a note to this mailing list when an updated version 
of MPlayer is available.

	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Thu, 6 Mar 2003 16:42:23 +0800
Date: Thu, 6 Mar 2003 16:42:23 +0800

From: {Ross Finlayson} <EMAIL-ADDRESS-HIDDEN>
To: <EMAIL-ADDRESS-HIDDEN>
Sent: Thursday, March 06, 2003 4:24 PM
Subject: Re: [Live-devel] can't play streamed mp4 file


> MPlayer's RTSP/RTP support doesn't yet handle MPEG-4 video or audio
(except
> MPEG-4 LATM audio).  The LIVE.COM libraries implement MPEG-4 audio/video
> reception OK (so, for example, "openRTSP" can be used to download MPEG-4
> RTSP/RTP streams), but MPlayer hasn't yet been updated to handle this.

Exactly, which is what I forgot to add into the scenario. I used openRTSP to
get the stream and save them in two files (video and audio) without problem.

>
> The good news, though, is that this will be working Real Soon Now (within
a
> few days).  I'll send a note to this mailing list when an updated version
> of MPlayer is available.
>

Great! Thanks.


> Ross Finlayson
> LIVE.COM
> <http://www.live.com/>
>





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X-Original-Date: Thu, 6 Mar 2003 20:51:30 +0900
Date: Thu, 6 Mar 2003 20:51:30 +0900

Hello live-devel,

  may be my question is Out of Topic, but if anyone know please help
  me..

  I have built an RTP Receiver that receive stream at certain port for
  -g711 mu-law 20ms-sized- RTP stream sent from IPPhone service using
  IPPhone streaming URI.

  I can sink / render that stream to the speaker and into a wavefile.

  but now, I want to build an RTP Listener that all it does is always
  listen-and-sinkToWavefile. a Wavefile for each RTPTx (start-stop).

  my problem is, how the receiving-side aware that the sending-side stop
  its RTP audio transmission? (ie. by calling RTPTx:Stop). so that we
  can start wrapping received audio stream into a Wavefile ??

-- 
Best regards,
 handy                          http://handy-tita.da.ru



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X-Original-Date: Thu, 06 Mar 2003 13:00:59 -0800
Date: Thu, 06 Mar 2003 13:00:59 -0800


>   may be my question is Out of Topic

This mailing list is for discussion of the "LIVE.COM Streaming Media" 
software, so if your project is using this, then your question is on-topic 
(otherwise, it's off-topic).

>   my problem is, how the receiving-side aware that the sending-side stop
>   its RTP audio transmission?

One way to do this is to look for a RTCP "BYE" packet coming from the 
sender.  (The sender will send a "BYE" when it closes its 
"RTCPInstance".)  See "testProgs/playCommon.cpp" for an illustration of how 
incoming "BYE"s are handled by "openRTSP" and "playSIP".

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>



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Subject: [Live-devel] LiveMedia port for FreeBSD 5.0 ? RTSP+RealMedia still not working?
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X-Original-Date: Thu, 06 Mar 2003 22:12:31 +0100
Date: Thu, 06 Mar 2003 22:12:31 +0100


Hello,

    I just wanted to ask if LiveMedia library's functionality is 
regularly tested on FreeBSD 5.0. In case it was, wouldn't it be nice to 
include it to its Ports collection?


Another question related to the first one: Yes, I did find the e-mail

http://www.geocrawler.com/archives/3/6637/2002/9/0/9711364/

but though my mplayer-gtk-0.90.0.10_1 doesn't work, having exactly the 
same problem, I am not really sure why mplayer, finding out that I am 
trying to play RealMedia via RTSP, displays that I should install 
LIVE.COM, implying that this should solve all of my problems.

Though I realize that this is most likely problem of mplayer people, not 
yours, I am not sure whether the cause of my problems is using FreeBSD - 
a maybe-not-really-supported BSD or still the above-mentioned case. I am 
simply confused that it seems that I am the only one complaining...


Thank you for your patience and the answers!


Ivan Dolezal




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  RTSP+RealMedia still not working?
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X-Original-Date: Thu, 06 Mar 2003 13:28:36 -0800
Date: Thu, 06 Mar 2003 13:28:36 -0800


>   I just wanted to ask if LiveMedia library's functionality is regularly 
> tested on FreeBSD 5.0.

No, but it's tested regularly on FreeBSD 4.x, and I know of no reason why 
it would not also work on 5.x.

>  In case it was, wouldn't it be nice to include it to its Ports collection?

No - the libraries are still changing too much for this.  I want people to 
get the libraries by coming directly to the source - 
<http://www.live.com/liveMedia/> - where they'll get the most up-to-date 
version, rather than getting some pre-packaged older version.

>problem, I am not really sure why mplayer, finding out that I am trying to 
>play RealMedia via RTSP, displays that I should install LIVE.COM, implying 
>that this should solve all of my problems.

No, RealAudio and RealVideo streams are not supported, mainly because 
they're not standards-compliant.  However, other, standards-compliant 
RTSP/RTP streams (including MPEG-1,2, and, very soon, MPEG-4) are supported.

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>



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X-Original-Date: Thu, 6 Mar 2003 17:42:01 -0500
Date: Thu, 6 Mar 2003 17:42:01 -0500

Ok. Got it to compile. The problem was indeed mplayer was compiled with 
gcc3 instead of gcc2.96. Had to throw --disable-gcc-checking on there.

Next problem:

Getting it to actually play the stream. Yes, multicast is turned on on 
my box as I can receive the announcements with sdr. And on my mac right 
next to me (same subnet), I can view the streams. But mplayer just says:

Playing CNN.sdp
No stream found.

if I call it as: mplayer-sdp-pentium -sdp CNN.sdp

The sdp file was saved with the Mac tool MacTV (I cannot seem to find a 
linux tool that will actually let me save a sdp file and have not spent 
enough time with the rfc yet to figure out what is necessary to make 
one).

Here is the sdp file:

> cat CNN.sdp

v=0
o=- 17817 0 IN IP4 IPTV3412.wvulibs.wvu.edu
s=CNN
i=Live Sat. Feed - CNN
e=t <t>
p=t <t>
b=AS:1150
t=0 0
a=type:broadcast
a=tool:IP/TV Content Manager 3.2.24
a=live:capture:1
m=video 54978 RTP/AVP 32 31 96
c=IN IP4 224.2.170.53/63
a=quality:8
a=framerate:30
a=rtpmap:96 WBIH/90000
a=x-iptv-svr:video 157.182.232.16 live
m=audio 27278 RTP/AVP 14 96 0 3 5 97 98 99 100 101 102 10 11 103 104 
105 106
c=IN IP4 224.2.160.5/63
a=rtpmap:96 X-WAVE/8000
a=rtpmap:97 L8/8000/2
a=rtpmap:98 L8/8000
a=rtpmap:99 L8/22050/2
a=rtpmap:100 L8/22050
a=rtpmap:101 L8/11025/2
a=rtpmap:102 L8/11025
a=rtpmap:103 L16/22050/2
a=rtpmap:104 L16/22050
a=rtpmap:105 L16/11025/2
a=rtpmap:106 L16/11025
a=x-iptv-svr:audio 157.182.232.16 live

Throwing -v 3 on the mplayer commandline only shows the additional:

Playing CNN.sdp
Not an URL!
File size is 720 bytes
No stream found.

Thanks for your help!

On 2003.03.06 00:23 Ross Finlayson wrote:
> It looks like the final linking of "mplayer" is not including the 
> standard C++ libraries.  Is "-lstdc++" included on the link line?  
> (It should be.)
> 
> Were you able to successfully link the LIVE.COM test programs (in 
> "live/testProgs/")?  If so, then you should be able to link mplayer 
> also.
> 
>> I should add the compile of mplayer appears to be detecting the 
>> gcc3.1 install
>> and using gcc3 for the compile.
> 
> That may (or may not) be significant - I'm not sure.  In any case, 
> make sure that you're using the same compilers to compile both the 
> LIVE.COM code, and the "mplayer" code.
> 
>         Ross.
> 
> 
> 
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-- 
********************************
David William Botsch
Consultant/Advisor II
CCMR Computing Facility
dwb7(at)ccmr.cornell.edu
********************************


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X-Original-Date: Thu, 06 Mar 2003 15:00:50 -0800
Date: Thu, 06 Mar 2003 15:00:50 -0800


>Playing CNN.sdp
>No stream found.

David,

  I suspect you're using a broken version of the MPlayer code.  (SDP file 
support was accidentally broken in the "0.90rc4" version, but has been 
fixed since.)

This is a reason why I encourage people to use the "latest CVS snapshot" 
version of MPlayer - 
http://www1.mplayerhq.hu/MPlayer/cvs/MPlayer-current.tar.bz2 - ant *not* 
the "rc" version.

>Here is the sdp file:
[snip]
>m=video 54978 RTP/AVP 32 31 96
[snip]
>m=audio 27278 RTP/AVP 14 96 0 3 5 97 98 99 100 101 102 10 11 103 104 105 106

One other thing to be aware of: The LIVE.COM code (and thus MPlayer) 
doesn't yet properly handle multiple RTP payload types specified on SDP 
"m=" lines.  Instead, it just uses the payload type that appears first - in 
this case 32 (==MPEG-1 or 2 video) and 14 (==MPEG-1 or 2 audio).

However, as long as your input streams really are MPEG, then you should be 
OK...

         Ross.



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From: David Botsch <EMAIL-ADDRESS-HIDDEN>
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Subject: Re: [Live-devel] compiling with mplayer fails
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X-Original-Date: Thu, 6 Mar 2003 18:12:47 -0500
Date: Thu, 6 Mar 2003 18:12:47 -0500

Well, I am using cvs, tho, maybe it is broken there as well?

To be exact, I am using cvs from March 4, 2003.

Does cvs work only up to a certain date?

On 2003.03.06 18:00 Ross Finlayson wrote:
> 
>> Playing CNN.sdp
>> No stream found.
> 
> David,
> 
>  I suspect you're using a broken version of the MPlayer code.  (SDP 
> file support was accidentally broken in the "0.90rc4" version, but 
> has been fixed since.)
> 
> This is a reason why I encourage people to use the "latest CVS 
> snapshot" version of MPlayer - 
> http://www1.mplayerhq.hu/MPlayer/cvs/MPlayer-current.tar.bz2 - ant 
> *not* the "rc" version.
> 
>> Here is the sdp file:
> [snip]
>> m=video 54978 RTP/AVP 32 31 96
> [snip]
>> m=audio 27278 RTP/AVP 14 96 0 3 5 97 98 99 100 101 102 10 11 103 104 
>> 105 106
> 
> One other thing to be aware of: The LIVE.COM code (and thus MPlayer) 
> doesn't yet properly handle multiple RTP payload types specified on 
> SDP "m=" lines.  Instead, it just uses the payload type that appears 
> first - in this case 32 (==MPEG-1 or 2 video) and 14 (==MPEG-1 or 2 
> audio).
> 
> However, as long as your input streams really are MPEG, then you 
> should be OK...
> 
>         Ross.
> 
> 
> 
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********************************
David William Botsch
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X-Original-Date: Thu, 06 Mar 2003 15:29:27 -0800
Date: Thu, 06 Mar 2003 15:29:27 -0800

At 03:12 PM 3/6/03, you wrote:
>Well, I am using cvs, tho, maybe it is broken there as well?
>
>To be exact, I am using cvs from March 4, 2003.

That should be OK.  Odd...

I'll take a closer look at this, and get back to you...

         Ross.



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X-Original-Date: Thu, 06 Mar 2003 21:13:10 -0800
Date: Thu, 06 Mar 2003 21:13:10 -0800

David,

	mplayer -sdp <path-to>/CNN.sdp
worked just fine for me.  This was with "MPlayer-20030306" (the latest CVS 
version - 20030307 - should also work), and version 2003.03.03 of the 
LIVE.COM libraries.

	Ross.



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Hello All,

I am using openRTSP to send data to a external rendering program using a pipe.

Currently the FileSink class doesn't verify that the stream it is writing to is still valid (it could fail because of full disks or in my case a closed pipe).

I include some code snippits to describe my solution and I would like to see it included in the next versions.

Code is modified in the function
void FileSink::afterGettingFrame(void* clientData, unsigned frameSize,
				 struct timeval /*presentationTime*/) 

......

  fwrite(sink->fBuffer, frameSize, 1, sink->fOutFid);
  int lEOF=fflush(sink->fOutFid);
  if (lEOF==EOF)
  {
     if (sink->fAfterFunc != NULL) {
    	(*(sink->fAfterFunc))(sink->fAfterClientData);
     }
     sink->stopPlaying();
     return;
  }
  sink->continuePlaying();  




Best regards,

Ruud Schramp


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X-Original-Date: Fri, 07 Mar 2003 01:11:56 -0800
Date: Fri, 07 Mar 2003 01:11:56 -0800

Ruud,

Good idea.  Thanks for the patch.  I've included it in a new version of the 
library - 2003.03.07 - released just now.

	Ross.



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X-Original-Date: Fri, 7 Mar 2003 11:30:10 -0500
Date: Fri, 7 Mar 2003 11:30:10 -0500

Tried with that cvs of mplayer. It's just not working.

live libs are in /usr/lib/live

mplayer was configured with ./configure --disable-divx4linux 
--enable-live
make
then copied the mplayer executable to my local bin dir

Any ideas here on what we can do to troubleshoot?

On 2003.03.07 00:13 Ross Finlayson wrote:
> David,
> 
> 	mplayer -sdp <path-to>/CNN.sdp
> worked just fine for me.  This was with "MPlayer-20030306" (the 
> latest CVS version - 20030307 - should also work), and version 
> 2003.03.03 of the LIVE.COM libraries.
> 
> 	Ross.
> 
> 
> 
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********************************
David William Botsch
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CCMR Computing Facility
dwb7(at)ccmr.cornell.edu
********************************


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Subject: Re: [Live-devel] compiling with mplayer fails
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X-Original-Date: Fri, 07 Mar 2003 09:44:53 -0800
Date: Fri, 07 Mar 2003 09:44:53 -0800

At 08:30 AM 3/7/03, you wrote:
>Tried with that cvs of mplayer. It's just not working.
>
>live libs are in /usr/lib/live
>
>mplayer was configured with ./configure --disable-divx4linux --enable-live
>make
>then copied the mplayer executable to my local bin dir
>
>Any ideas here on what we can do to troubleshoot?

Could you run, from your local bin dir
         ./mplayer -V -sdp <path-to>/CNN.sdp
and send me a copy of mplayer's console output.  That may give me a hint as 
to what's going wrong.

         Ross.



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From: David Botsch <EMAIL-ADDRESS-HIDDEN>
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Subject: Re: [Live-devel] compiling with mplayer fails
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X-Original-Date: Fri, 7 Mar 2003 13:14:19 -0500
Date: Fri, 7 Mar 2003 13:14:19 -0500

Sure... here goes...

domino> ./mplayer-sdp-pentium -V -sdp ~/CNN.sdp Using GNU 
internationalization
Original domain: messages
Original dirname: /usr/share/locale
Current domain: mplayer
Current dirname: /usr/local/share/locale


MPlayer dev-CVS-030306-00:00-2.96 (C) 2000-2003 Arpad Gereoffy (see 
DOCS)

CPU: Intel  (Family: 8, Stepping: 4)
Detected cache-line size is 64 bytes
CPUflags:  MMX: 1 MMX2: 1 3DNow: 0 3DNow2: 0 SSE: 1 SSE2: 1
Compiled for x86 CPU with extensions: MMX MMX2 SSE SSE2

Reading config file /usr/local/etc/mplayer/mplayer.conf: No such file 
or directory
Reading config file 
/afs/msc.cornell.edu/home/mscadm/dwb7/.mplayer/config
Reading /afs/msc.cornell.edu/home/mscadm/dwb7/.mplayer/codecs.conf: 50 
audio & 136 video codecs
CommandLine: '-V' '-sdp' '/afs/msc.cornell.edu/home/mscadm/dwb7/CNN.sdp'
get_path('font/font.desc') -> 
'/afs/msc.cornell.edu/home/mscadm/dwb7/.mplayer/font/font.desc'
font: can't open file: 
/afs/msc.cornell.edu/home/mscadm/dwb7/.mplayer/font/font.desc
font: can't open file: /usr/local/share/mplayer/font/font.desc
init_freetype
Using MMX (with tiny bit MMX2) Optimized OnScreenDisplay
Linux RTC init error in ioctl (rtc_irqp_set 1024): Permission denied
Try adding "echo 1024 > /proc/sys/dev/rtc/max-user-freq" to your system 
startup scripts.
Using usleep() timing
get_path('input.conf') -> 
'/afs/msc.cornell.edu/home/mscadm/dwb7/.mplayer/input.conf'
Can't open input config file 
/afs/msc.cornell.edu/home/mscadm/dwb7/.mplayer/input.conf : No such 
file or directory
Can't open input config file /usr/local/etc/mplayer/input.conf : No 
such file or directory
Falling back on default (hardcoded) input config
get_path('CNN.sdp.conf') -> 
'/afs/msc.cornell.edu/home/mscadm/dwb7/.mplayer/CNN.sdp.conf'

Playing /afs/msc.cornell.edu/home/mscadm/dwb7/CNN.sdp
Not an URL!
File size is 720 bytes
No stream found.

DEMUXER: freeing demuxer at 0x8435558  vo: x11 uninit called but X11 
not inited..

Exiting... (End of file)


On 2003.03.07 12:44 Ross Finlayson wrote:
> At 08:30 AM 3/7/03, you wrote:
>> Tried with that cvs of mplayer. It's just not working.
>> 
>> live libs are in /usr/lib/live
>> 
>> mplayer was configured with ./configure --disable-divx4linux 
>> --enable-live
>> make
>> then copied the mplayer executable to my local bin dir
>> 
>> Any ideas here on what we can do to troubleshoot?
> 
> Could you run, from your local bin dir
>         ./mplayer -V -sdp <path-to>/CNN.sdp
> and send me a copy of mplayer's console output.  That may give me a 
> hint as to what's going wrong.
> 
>         Ross.
> 
> 
> 
> -------------------------------------------------------
> This SF.net email is sponsored by: Etnus, makers of TotalView, The 
> debugger for complex code. Debugging C/C++ programs can leave you 
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> www.etnus.com
> _______________________________________________
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-- 
********************************
David William Botsch
Consultant/Advisor II
CCMR Computing Facility
dwb7(at)ccmr.cornell.edu
********************************


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 <4.3.1.1.20030306144930.00c699c0(at)laptop-localhost>
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X-Original-Date: Sun, 09 Mar 2003 00:05:43 -0800
Date: Sun, 09 Mar 2003 00:05:43 -0800

At 10:14 AM 3/7/03, you wrote:
>Playing /afs/msc.cornell.edu/home/mscadm/dwb7/CNN.sdp
>Not an URL!
>File size is 720 bytes
>No stream found.

David,

It looks like the SDP parsing code (in the LIVE.COM libraries) is - for 
some unknown reason - having trouble parsing your "CNN.sdp" file.

Could you instead download <http://www.live.com/mplayer/CNN.sdp>, and try 
using that instead?  (That file works OK for me.)  If that "CNN.sdp" works 
OK, but yours doesn't, then please let me know how they differ.

If, on the other hand, your "mplayer" works with *neither* version of 
"CNN.sdp", then could you try running the 
<http://www.live.com/mplayer/mplayer> binary instead.  (That is a new 
version of "mplayer" that I built.)

         Ross.



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Subject: Re: [Live-devel] compiling with mplayer fails
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References: <4.3.1.1.20030307094239.00c5aa70(at)laptop-localhost> <4.3.1.1.20030306204524.00b58c80(at)laptop-localhost> <4.3.1.1.20030306144930.00c699c0(at)laptop-localhost> <4.3.1.1.20030305212053.00c5ace0(at)laptop-localhost> <4.3.1.1.20030305212053.00c5ace0(at)laptop-localhost> <4.3.1.1.20030306144930.00c699c0(at)laptop-localhost> <4.3.1.1.20030306204524.00b58c80(at)laptop-localhost> <4.3.1.1.20030307094239.00c5aa70(at)laptop-localhost> <20030307181419.GI5294(at)domino.ccmr.cornell.edu> <4.3.1.1.20030309000100.00ad6f00(at)laptop-localhost>
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X-Original-Date: Sun, 9 Mar 2003 15:05:34 -0500
Date: Sun, 9 Mar 2003 15:05:34 -0500

While I can't 100% confirm it works until I get back to work tomorrow (trying
to run mplayer remotely over a cable modem), I can tell you it does not exit
with the "no stream found". It would appear to parse the stram and iniitiate
the rtp sessions.

domino> diff CNN.sdp CNN-new.sdp 
1d0
< 
33d31
< 


where CNN-new.sdp is your sdp file. The sdp file I have were saved using the
MacTV program.

Am I reading thi scorrectly that the only differences are blank lines in the
file?

On Sun, Mar 09, 2003 at 12:05:43AM -0800, Ross Finlayson wrote:
> At 10:14 AM 3/7/03, you wrote:
> >Playing /afs/msc.cornell.edu/home/mscadm/dwb7/CNN.sdp
> >Not an URL!
> >File size is 720 bytes
> >No stream found.
> 
> David,
> 
> It looks like the SDP parsing code (in the LIVE.COM libraries) is - for 
> some unknown reason - having trouble parsing your "CNN.sdp" file.
> 
> Could you instead download <http://www.live.com/mplayer/CNN.sdp>, and try 
> using that instead?  (That file works OK for me.)  If that "CNN.sdp" works 
> OK, but yours doesn't, then please let me know how they differ.
> 
> If, on the other hand, your "mplayer" works with *neither* version of 
> "CNN.sdp", then could you try running the 
> <http://www.live.com/mplayer/mplayer> binary instead.  (That is a new 
> version of "mplayer" that I built.)
> 
>          Ross.
> 
> 
> 
> -------------------------------------------------------
> This SF.net email is sponsored by: Etnus, makers of TotalView, The debugger 
> for complex code. Debugging C/C++ programs can leave you feeling lost and 
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********************************


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X-Original-Date: Sun, 09 Mar 2003 15:33:24 -0800
Date: Sun, 09 Mar 2003 15:33:24 -0800


>Am I reading thi scorrectly that the only differences are blank lines in the
>file?

Yes.  It turns out that the blank line at the beginning of your file was 
confusing our SDP parsing code.  If you delete this (and the one at the end 
of the file also), then things should work.

(I have now installed a new version (2003.03.09) of the LIVE.COM libraries 
that allows blank lines in SDP descriptions.)

Thanks for helping track this down.  Your patience is appreciated :-)

         Ross.



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From: David Botsch <EMAIL-ADDRESS-HIDDEN>
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Subject: Re: [Live-devel] compiling with mplayer fails
Message-ID: <20030309185533.A31309(at)ccmr.cornell.edu>
References: <4.3.1.1.20030306144930.00c699c0(at)laptop-localhost> <4.3.1.1.20030305212053.00c5ace0(at)laptop-localhost> <4.3.1.1.20030305212053.00c5ace0(at)laptop-localhost> <4.3.1.1.20030306144930.00c699c0(at)laptop-localhost> <4.3.1.1.20030306204524.00b58c80(at)laptop-localhost> <4.3.1.1.20030307094239.00c5aa70(at)laptop-localhost> <20030307181419.GI5294(at)domino.ccmr.cornell.edu> <4.3.1.1.20030309000100.00ad6f00(at)laptop-localhost> <20030309150532.A30862(at)ccmr.cornell.edu> <4.3.1.1.20030309151954.00b7eec0(at)laptop-localhost>
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X-Original-Date: Sun, 9 Mar 2003 18:55:33 -0500
Date: Sun, 9 Mar 2003 18:55:33 -0500

And, thanks for your patience and help.

I'll be able to actually look at mplayer tomorrow at work and see what the
video quality, etc. actually looks like and give some feedback on that.

Thnx again!

On Sun, Mar 09, 2003 at 03:33:24PM -0800, Ross Finlayson wrote:
> 
> >Am I reading thi scorrectly that the only differences are blank lines in the
> >file?
> 
> Yes.  It turns out that the blank line at the beginning of your file was 
> confusing our SDP parsing code.  If you delete this (and the one at the end 
> of the file also), then things should work.
> 
> (I have now installed a new version (2003.03.09) of the LIVE.COM libraries 
> that allows blank lines in SDP descriptions.)
> 
> Thanks for helping track this down.  Your patience is appreciated :-)
> 
>          Ross.
> 
> 
> 
> -------------------------------------------------------
> This SF.net email is sponsored by: Etnus, makers of TotalView, The debugger 
> for complex code. Debugging C/C++ programs can leave you feeling lost and 
> disoriented. TotalView can help you find your way. Available on major UNIX 
> and Linux platforms. Try it free. www.etnus.com
> _______________________________________________
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-- 
********************************
David William Botsch
Consultant/Advisor II
CCMR Computing Facility
dwb7(at)ccmr.cornell.edu
********************************


-------------------------------------------------------
This SF.net email is sponsored by: Etnus, makers of TotalView, The debugger 
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From: David Botsch <EMAIL-ADDRESS-HIDDEN>
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Subject: Re: [Live-devel] compiling with mplayer fails
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X-Original-Date: Mon, 10 Mar 2003 11:53:01 -0500
Date: Mon, 10 Mar 2003 11:53:01 -0500

Well... on to the next problem.

I haven't tried the new cvses of mplayer and the new live.com libs yet, 
but when plyaing the sdp streams (and yes, they do play, now), the 
video/audio has a lot of lost data, it seems, and the console output of 
mplayer shows:

(at)(at)(at) libmpeg2 returned from sig11... (bad file?) (at)(at)(at)
(at)(at)(at) libmpeg2 returned from sig11... (bad file?) (at)(at)(at)
(at)(at)(at) libmpeg2 returned from sig11... (bad file?) (at)(at)(at)
(at)(at)(at) libmpeg2 returned from sig11... (bad file?) (at)(at)(at)
(at)(at)(at) libmpeg2 returned from sig11... (bad file?) (at)(at)(at)

lots of these lines appearing every second. Any general ideas?

On 2003.03.09 18:33 Ross Finlayson wrote:
> 
>> Am I reading thi scorrectly that the only differences are blank 
>> lines in the
>> file?
> 
> Yes.  It turns out that the blank line at the beginning of your file 
> was confusing our SDP parsing code.  If you delete this (and the one 
> at the end of the file also), then things should work.
> 
> (I have now installed a new version (2003.03.09) of the LIVE.COM 
> libraries that allows blank lines in SDP descriptions.)
> 
> Thanks for helping track this down.  Your patience is appreciated :-)
> 
>         Ross.
> 
> 
> 
> -------------------------------------------------------
> This SF.net email is sponsored by: Etnus, makers of TotalView, The 
> debugger for complex code. Debugging C/C++ programs can leave you 
> feeling lost and disoriented. TotalView can help you find your way. 
> Available on major UNIX and Linux platforms. Try it free. 
> www.etnus.com
> _______________________________________________
> Live-devel mailing list
> Live-devel(at)lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/live-devel

-- 
********************************
David William Botsch
Consultant/Advisor II
CCMR Computing Facility
dwb7(at)ccmr.cornell.edu
********************************


-------------------------------------------------------
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From: David Botsch <EMAIL-ADDRESS-HIDDEN>
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Subject: Re: [Live-devel] compiling with mplayer fails
Message-ID: <20030310165509.GC1416(at)domino.ccmr.cornell.edu>
References: <4.3.1.1.20030306204524.00b58c80(at)laptop-localhost> <4.3.1.1.20030306144930.00c699c0(at)laptop-localhost> <4.3.1.1.20030305212053.00c5ace0(at)laptop-localhost> <4.3.1.1.20030305212053.00c5ace0(at)laptop-localhost> <4.3.1.1.20030306144930.00c699c0(at)laptop-localhost> <4.3.1.1.20030306204524.00b58c80(at)laptop-localhost> <4.3.1.1.20030307094239.00c5aa70(at)laptop-localhost> <20030307181419.GI5294(at)domino.ccmr.cornell.edu> <4.3.1.1.20030309000100.00ad6f00(at)laptop-localhost> <4.3.1.1.20030309151954.00b7eec0(at)laptop-localhost>
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X-Original-Date: Mon, 10 Mar 2003 11:55:09 -0500
Date: Mon, 10 Mar 2003 11:55:09 -0500

Just a little more info:

using -vfm ffmpeg gives this output instead:

Error while decoding slice
ac-tex damaged at 17 14-0.460 ct:  0.140   95/ 95   3%  3% 105.6% 85 0 
0%
Error while decoding slice
ac-tex damaged at 16 0 -0.680 ct:  0.133   97/ 97   3%  3% 110.0% 87 0 
0%
Error while decoding slice
ac-tex damaged at 16 0 -0.699 ct:  0.117  102/102   3%  3% 117.6% 92 0 
0%
Error while decoding slice
ac-tex damaged at 16 0 -0.688 ct:  0.093  109/109   3%  3% 119.2% 99 0 
0%
Error while decoding slice
ac-tex damaged at 16 0 -0.680 ct:  0.077  114/114   3%  3% 124.9% 104 0 
0%
Error while decoding slice
slice too long4.9 A-V: -0.523 ct:  0.053  121/121   3%  3% 120.5% 111 0 
0%
Error while decoding slice
invalid cbp at 7 13-V: -0.512 ct:  0.043  124/124   3%  3% 119.7% 114 0 
0%
Error while decoding slice
invalid mb type in B Frame at 1 14 0.040  125/125   3%  3% 120.9% 115 0 
0%
Error while decoding slice
ac-tex damaged at 9 14

On 2003.03.09 18:33 Ross Finlayson wrote:
> 
>> Am I reading thi scorrectly that the only differences are blank 
>> lines in the
>> file?
> 
> Yes.  It turns out that the blank line at the beginning of your file 
> was confusing our SDP parsing code.  If you delete this (and the one 
> at the end of the file also), then things should work.
> 
> (I have now installed a new version (2003.03.09) of the LIVE.COM 
> libraries that allows blank lines in SDP descriptions.)
> 
> Thanks for helping track this down.  Your patience is appreciated :-)
> 
>         Ross.
> 
> 
> 
> -------------------------------------------------------
> This SF.net email is sponsored by: Etnus, makers of TotalView, The 
> debugger for complex code. Debugging C/C++ programs can leave you 
> feeling lost and disoriented. TotalView can help you find your way. 
> Available on major UNIX and Linux platforms. Try it free. 
> www.etnus.com
> _______________________________________________
> Live-devel mailing list
> Live-devel(at)lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/live-devel

-- 
********************************
David William Botsch
Consultant/Advisor II
CCMR Computing Facility
dwb7(at)ccmr.cornell.edu
********************************


-------------------------------------------------------
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 <4.3.1.1.20030306144930.00c699c0(at)laptop-localhost>
 <4.3.1.1.20030306204524.00b58c80(at)laptop-localhost>
 <4.3.1.1.20030307094239.00c5aa70(at)laptop-localhost>
 <20030307181419.GI5294(at)domino.ccmr.cornell.edu>
 <4.3.1.1.20030309000100.00ad6f00(at)laptop-localhost>
 <4.3.1.1.20030309151954.00b7eec0(at)laptop-localhost>
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X-Original-Date: Mon, 10 Mar 2003 10:22:28 -0800
Date: Mon, 10 Mar 2003 10:22:28 -0800


>I haven't tried the new cvses of mplayer and the new live.com libs yet, 
>but when plyaing the sdp streams (and yes, they do play, now), the 
>video/audio has a lot of lost data, it seems, and the console output of 
>mplayer shows:
>
>(at)(at)(at) libmpeg2 returned from sig11... (bad file?) (at)(at)(at)
>(at)(at)(at) libmpeg2 returned from sig11... (bad file?) (at)(at)(at)
>(at)(at)(at) libmpeg2 returned from sig11... (bad file?) (at)(at)(at)
>(at)(at)(at) libmpeg2 returned from sig11... (bad file?) (at)(at)(at)
>(at)(at)(at) libmpeg2 returned from sig11... (bad file?) (at)(at)(at)
>
>lots of these lines appearing every second. Any general ideas?

Yes, the current MPlayer code doesn't deal with high bitrate streams 
particularly well - it drops data far more than it should (which causes the 
MPEG decoder to complain).

Within a few days I'll be checking in a new version of the MPlayer code 
that significantly improves this.  If you're impatient, though, you can try 
running the binary version that I have left online as
         http://www.live.com/mplayer/mplayer

     Ross.



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From: David Botsch <EMAIL-ADDRESS-HIDDEN>
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Subject: Re: [Live-devel] compiling with mplayer fails
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References: <4.3.1.1.20030306144930.00c699c0(at)laptop-localhost> <4.3.1.1.20030305212053.00c5ace0(at)laptop-localhost> <4.3.1.1.20030305212053.00c5ace0(at)laptop-localhost> <4.3.1.1.20030306144930.00c699c0(at)laptop-localhost> <4.3.1.1.20030306204524.00b58c80(at)laptop-localhost> <4.3.1.1.20030307094239.00c5aa70(at)laptop-localhost> <20030307181419.GI5294(at)domino.ccmr.cornell.edu> <4.3.1.1.20030309000100.00ad6f00(at)laptop-localhost> <4.3.1.1.20030309151954.00b7eec0(at)laptop-localhost> <4.3.1.1.20030310101925.00c20200(at)laptop-localhost>
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X-Original-Date: Mon, 10 Mar 2003 16:50:01 -0500
Date: Mon, 10 Mar 2003 16:50:01 -0500

Looks like we will have to wait because our stock redhat 7.3 machines 
don't have libvga and alsa.

On 2003.03.10 13:22 Ross Finlayson wrote:
> 
>> I haven't tried the new cvses of mplayer and the new live.com libs 
>> yet, but when plyaing the sdp streams (and yes, they do play, now), 
>> the video/audio has a lot of lost data, it seems, and the console 
>> output of mplayer shows:
>> 
>> (at)(at)(at) libmpeg2 returned from sig11... (bad file?) (at)(at)(at)
>> (at)(at)(at) libmpeg2 returned from sig11... (bad file?) (at)(at)(at)
>> (at)(at)(at) libmpeg2 returned from sig11... (bad file?) (at)(at)(at)
>> (at)(at)(at) libmpeg2 returned from sig11... (bad file?) (at)(at)(at)
>> (at)(at)(at) libmpeg2 returned from sig11... (bad file?) (at)(at)(at)
>> 
>> lots of these lines appearing every second. Any general ideas?
> 
> Yes, the current MPlayer code doesn't deal with high bitrate streams 
> particularly well - it drops data far more than it should (which 
> causes the MPEG decoder to complain).
> 
> Within a few days I'll be checking in a new version of the MPlayer 
> code that significantly improves this.  If you're impatient, though, 
> you can try running the binary version that I have left online as
>         http://www.live.com/mplayer/mplayer
> 
>     Ross.
> 
> 
> 
> -------------------------------------------------------
> This sf.net email is sponsored by:ThinkGeek
> Welcome to geek heaven.
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-- 
********************************
David William Botsch
Consultant/Advisor II
CCMR Computing Facility
dwb7(at)ccmr.cornell.edu
********************************


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X-Original-Date: Wed, 12 Mar 2003 02:34:07 -0500
Date: Wed, 12 Mar 2003 02:34:07 -0500

Hallo, 
I have a problem, possibly with LIVE.COM library compiled 
on Linux/ppc system. Mplayer craches each time when I try 
to start to receive a stream from Darwin Streaming Server. 
Although the same file (mp4) can be normally played by 
MPlayer locally or through http. And the same stream file 
can be also normally played by MPlayer with LIVE.COM on 
Linux/x86 system. It looks like it is a problem only on 
PowerPC systems. Are there any known problems with 
LIVE.COM library compiling and using and solutions on ppc 
systems? What could I try to do to make my MPlayer 
compiled on ppc stop crashing each time when try to 
receive streamed file? 
-- 
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From <EMAIL-ADDRESS-HIDDEN> Wed Mar 12 01:00:22 2003
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Subject: [Live-devel] Updated MPlayer, with improved RTSP/RTP streaming support
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X-Original-Date: Wed, 12 Mar 2003 00:57:27 -0800
Date: Wed, 12 Mar 2003 00:57:27 -0800

I'm pleased to report that the "MPlayer" media player has now been updated 
with improved RTSP/RTP streaming support.

- MPEG-4 audio and video RTP streams are now supported.
- Buffering of incoming packets has been improved, and there should now be 
significantly less data loss when playing high bitrate streams.
- Audio/video synchronization has been improved (especially after resuming 
from a pause).
  - MPEG-1 or MPEG-2 video streams should now look better when there is 
packet loss.
- In most cases you no longer need to use the "-fps" option to set the 
video frame rate.  Instead, the MPlayer code will try to figure out the 
frame rate automatically, by comparing the RTP presentation times of 
successive packets.

You can get this new version of MPlayer by downloading the "latest CVS 
snapshot":
	http://www1.mplayerhq.hu/MPlayer/cvs/MPlayer-current.tar.bz2

Plus, of course, for RTSP/RTP streaming support, you will also need the 
"LIVE.COM Streaming Media" libraries.  See <http://www.live.com/liveMedia/> 
for more details.

	Ross.



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X-Original-Date: Wed, 12 Mar 2003 01:00:31 -0800
Date: Wed, 12 Mar 2003 01:00:31 -0800


>I have a problem, possibly with LIVE.COM library compiled
>on Linux/ppc system. Mplayer craches each time when I try
>to start to receive a stream from Darwin Streaming Server.
>Although the same file (mp4) can be normally played by
>MPlayer locally or through http.

Elnar,

Previous versions of MPlayer did not support MPEG-4 streams well (if at all).

Please download the newest version of MPlayer 
-  http://www1.mplayerhq.hu/MPlayer/cvs/MPlayer-current.tar.bz2 - and try 
again with that version.

If you still see a crash, please run MPlayer with the "-V" flag, and email 
the console output to this mailing list, so we can take a look at it.

         Ross.



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From: {Rohit Kapoor} <EMAIL-ADDRESS-HIDDEN>
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Subject: [Live-devel] testMPEGAudioVideoStreamer
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Date: Wed, 12 Mar 2003 12:57:50 -0800

This is a multi-part message in MIME format.

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Hi,

I am new to using live streaming. So, this might be a basic question, =
but I would appreciate of somebody could answer it.
I am trying to stream mpg or mp3 files using testMPEGAudioVideoStreamer. =
I have two laptops, one running mplayer and the other has =
testMPEGAudioVideoStreamer.

I am not sure about how mplayer can be made to play the mpg files =
streamed by testMPEGAudioVideoStreamer, I mean, what is the command-line =
argument for mplayer?

Thanks,
Rohit


------=_NextPart_000_0426_01C2E896.FCC6A350
Content-Type: text/html;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content=3D"text/html; charset=3Diso-8859-1" =
http-equiv=3DContent-Type>
<META content=3D"MSHTML 5.00.3315.2870" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Hi,</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>I am new to using live streaming. So, =
this might be=20
a basic question, but I would appreciate of somebody could answer=20
it.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>I am trying to stream mpg or mp3 files =
using=20
testMPEGAudioVideoStreamer. I have two laptops, one running mplayer and =
the=20
other has testMPEGAudioVideoStreamer.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>I am not sure about how mplayer can be =
made to play=20
the mpg files streamed by testMPEGAudioVideoStreamer, I mean, what is =
the=20
command-line argument for mplayer?</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Thanks,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Rohit</FONT></DIV>
<DIV>&nbsp;</DIV></BODY></HTML>

------=_NextPart_000_0426_01C2E896.FCC6A350--



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X-Original-Date: Wed, 12 Mar 2003 14:21:55 -0800
Date: Wed, 12 Mar 2003 14:21:55 -0800


>I am new to using live streaming. So, this might be a basic question, but 
>I would appreciate of somebody could answer it.
>I am trying to stream mpg or mp3 files using testMPEGAudioVideoStreamer

".mpg" (audio+video) files can be streamed using 
"testMPEGAudioVideoStreamer".  To stream ".mp3" files, use "testMP3Streamer".

>I am not sure about how mplayer can be made to play the mpg files streamed 
>by testMPEGAudioVideoStreamer, I mean, what is the command-line argument 
>for mplayer?

mplayer -sdp testMPEGAudioVideo.sdp

(You will find the file "testMPEGAudioVideoStreamer.sdp" in the 
"testProgs/" directory.)

For receiving/playing MP3/RTP streams, use

mplayer -sdp testMP3.sdp

         Ross.



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Subject: [Live-devel] Problems compiling last CVS MPlayer
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X-Original-Date: Thu, 13 Mar 2003 02:58:10 -0500
Date: Thu, 13 Mar 2003 02:58:10 -0500

Here are the errors that I get after compiling MPlayer on 
Linux/x86. The MPlayer with called with following 
parameters: ./mplayer -nosound -fps 25 
rtsp://localhost/video.mp4 
 
Using GNU internationalization  
Original domain: messages  
Original dirname: /usr/share/locale  
Current domain: mplayer  
Current dirname: 
/home/elnar/software/mplayer//share/locale  
  
  
MPlayer dev-CVS-030313-06:00-2.95.3 (C) 2000-2003 Arpad 
Gereoffy (see  
DOCS)  
  
CPU: Intel Celeron 2/Pentium III Coppermine,Geyserville 
(Family: 6,  
Stepping: 1)  
Detected cache-line size is 32 bytes  
CPUflags:  MMX: 1 MMX2: 1 3DNow: 0 3DNow2: 0 SSE: 1 SSE2: 
0  
Compiled for x86 CPU with extensions: MMX MMX2 SSE  
  
Reading config file  
/home/elnar/software/mplayer//etc/mplayer/mplayer.confReading 
config file  
/root/.mplayer/config  
Reading /root/.mplayer/codecs.conf: Reading  
/home/elnar/software/mplayer//etc/mplayer/codecs.conf: 50 
audio & 136  
video codecs  
font: can't open file: /root/.mplayer/font/font.desc  
font: can't open file:  
/home/elnar/software/mplayer//share/mplayer/font/font.desc  
Using Linux hardware RTC timing (1024Hz).  
  
Playing rtsp://suse8-as/miwarze.mp4  
Connected to server: suse8-as  
Stream not seekable!  
==========================================================================  
Opening audio decoder: [faad] AAC (MPEG2/4 Advanced Audio 
Coding)  
AUDIO: 44100 Hz, 2 ch, 16 bit (0x10), ratio: 16000->176400 
(128.0 kbit)  
Selected audio codec: [faad] afm:faad (AAC MPEG2/MPEG4 
Audio)  
==========================================================================  
==========================================================================  
Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec 
family  
Selected video codec: [ffodivx] vfm:ffmpeg (FFmpeg MPEG-4)  
==========================================================================  
Checking audio filter chain for 44100Hz/2ch/16bit -> 
44100Hz/2ch/16bit...  
AF_pre: af format: 2 bps, 2 ch, 44100 hz, little endian 
signed int  
AF_pre: 44100Hz 2ch Signed 16-bit (Little-Endian)  
alsa-init: requested format: 44100 Hz, 2 channels, Signed 
16-bit  
(Little-Endian)  
AO: [null] 44100Hz 2ch Signed 16-bit (Little-Endian) (2 
bps)  
Building audio filter chain for 44100Hz/2ch/16bit -> 
44100Hz/2ch/16bit...  
FPS forced to be 25.000  (ftime: 0.040)  
Starting playback...  
This file was encoded with DivX501 Build450  
This file was encoded with DivX501 Build450  
VDec: vo config request - 320 x 240 (preferred csp: Planar 
YV12)  
VDec: using Planar YV12 as output csp (no 0)  
Movie-Aspect is 1.33:1 - prescaling to correct movie 
aspect.  
VO: [fbdev] 320x240 => 320x240 Planar YV12  
broken, this should use the swscaler  
broken, this should use the swscaler.000    1/  1   0%  0%  
0.0% 0 0 0%  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler.004    2/  2   0%  0%  
0.0% 0 0 0%  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler.008    3/  3   0%  0%  
0.0% 0 0 0%  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler.012    4/  4   0%  0%  
0.0% 0 0 0%  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler.011    5/  5   0%  0%  
0.0% 0 0 0%  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler.011    6/  6   0%  0%  
0.0% 0 0 0%  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler.015    7/  7   0%  0%  
0.0% 0 0 0%  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler.019    8/  8   0%  0%  
0.0% 0 0 0%  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler.023    9/  9   0%  0%  
0.0% 0 0 0%  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler.027   10/ 10   0%  0%  
0.0% 0 0 0%  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler.031   11/ 11   0%  0%  
0.0% 0 0 0%  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler.035   12/ 12   0%  0%  
0.0% 0 0 0%  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler.039   13/ 13   0%  0%  
0.0% 0 0 0%  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler.043   14/ 14  10%  1%  
9.5% 0 0 0%  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler.047   15/ 15  10%  1%  
9.3% 0 0 0%  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler.043   16/ 16  10%  1%  
9.1% 0 0 0%  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler.041   17/ 17  10%  1%  
8.7% 0 0 0%  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler.040   18/ 18  10%  1%  
8.7% 0 0 0%  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler.042   19/ 19  10%  1%  
8.6% 0 0 0%  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler  
broken, this should use the swscaler.046   20/ 20   9%  1%  
8.3% 0 0 0%   
 
and so on thousands of lines.... 
 
 
If I try to start the same video file locally it also does 
not work. The same faults are comming. It seems that this 
CVS version of MPlayer is broken. How could I solve this 
problem and make my program to start working? 
 
Thanks 
 
Elnar 
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Subject: Re: [Live-devel] Problems compiling last CVS MPlayer
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X-Original-Date: Thu, 13 Mar 2003 00:30:21 -0800
Date: Thu, 13 Mar 2003 00:30:21 -0800


>If I try to start the same video file locally it also does
>not work. The same faults are comming. It seems that this
>CVS version of MPlayer is broken.

Elnar,

Because you see this same problem when you (try to) play the file locally, 
I suggest that you ask the "mplayer-users(at)mplayerhq.hu" mailing 
list.  Someone there should be able to help you.

         Ross.



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X-Original-Date: Thu, 13 Mar 2003 10:12:21 -0800
Date: Thu, 13 Mar 2003 10:12:21 -0800

Ross,

Thanks a lot. I got it to work (almost) by enabling RTSP. I am having one
more problem though.
I have the latest mediplayer release (not the CVS). It plays local mp3 and
mpg files correctly. But, when I try to stream, it plays mp3 correctly, but
for mpg, it merely prints out a lot of numbers instead of playing the video.

This is what the output looks like:

Using GNU internationalization
Original domain: messages
Original dirname: /usr/share/locale
Current domain: mplayer
Current dirname: /usr/local/share/locale

MPlayer 0.90rc4-3.2 (C) 2000-2003 Arpad Gereoffy (see DOCS)
CPU: Intel Celeron Covington/Pentium II Deschutes,Tonga/Pentium II Xeon
(Family: 6, Stepping: 2)
Detected cache-line size is 32 bytes
CPUflags: MMX: 1 MMX2: 0 3DNow: 0 3DNow2: 0 SSE: 0 SSE2: 0
Compiled for x86 CPU with extensions: MMX
Reading config file /usr/local/etc/mplayer/mplayer.confReading config file
/root/.mplayer/config
Reading /root/.mplayer/codecs.conf: 50 audio & 135 video codecs
font: can't open file: /root/.mplayer/font/font.desc
font: can't open file: /usr/local/share/mplayer/font/font.desc
Using usleep() timing
Playing rtsp://10.10.10.10/test.mpg
Connected to server: 10.10.10.10
Stream not seekable!
Initiated "audio/MPA" RTP subsession
Initiated "video/MPV" RTP subsession
0x100
0x101
0x100
0x101
0x100
0x101
0x100
0x101
0x100
0x101
0x100
0x101
0x100
0x101
0x100
0x101
0x100
0x101
0x100
0x101
0x100
0x101
0x100
0x101
0x100
0x101
0x100
0x101
0x1B8
0x100
0x101


Any ideas?

Thanks,
Rohit


----- Original Message -----
From: {Ross Finlayson} <EMAIL-ADDRESS-HIDDEN>
To: <EMAIL-ADDRESS-HIDDEN>
Sent: Thursday, March 13, 2003 12:30 AM
Subject: Re: [Live-devel] Problems compiling last CVS MPlayer


>
> >If I try to start the same video file locally it also does
> >not work. The same faults are comming. It seems that this
> >CVS version of MPlayer is broken.
>
> Elnar,
>
> Because you see this same problem when you (try to) play the file locally,
> I suggest that you ask the "mplayer-users(at)mplayerhq.hu" mailing
> list.  Someone there should be able to help you.
>
>          Ross.
>
>
>
> -------------------------------------------------------
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X-Original-Date: Thu, 13 Mar 2003 10:25:05 -0800
Date: Thu, 13 Mar 2003 10:25:05 -0800


>Thanks a lot. I got it to work (almost) by enabling RTSP. I am having one
>more problem though.
>I have the latest mediplayer release (not the CVS).

Bzzt!  Wrong answer.  You should use the "latest CVS" snapshot.  The "rc" 
version is *older* than this, and should *not* be used.

>Any ideas?

You must use the "latest CVS snapshot" version.  (I try to make this clear 
in <http://www.live.com/mplayer/>)

         Ross.



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X-Original-Date: Thu, 13 Mar 03 20:12:09 -0800
Date: Thu, 13 Mar 03 20:12:09 -0800

Ross, et al, I'm using the liveMedia libraries to add multicast support 
to our internet product line and have run into a problem with the Win32 
version of the libraries: pairing strdup() -- a malloc based routine -- 
with the C++ delete causes a heap-check assertion (RtlValidateHeap) when 
delete is called. Specifically, the "delete buffer" call in the 
MediaSession's parseCLine routine after the strdup of sdpLine traps.

Although we intend to bring the libraries up on Linux, Window's is the 
first configuration we need. Of course I can go through and change all 
the delete's to free() calls, but I hate to make that many changes to the 
library code. BTW, we're currently running the 3.11.2003 version of the 
libraries.

For what it's worth, we're using W2K Pro, SP3 with version 6 of Visual 
C++. So, two questions: do you see this when running under Windows and 
any idea when a fix might be released? Regardless, I've been impressed 
with the overall quality of the code. Thanks for making it available 
under the LPGL.

---------------------------------------------------------------	
Bill Kain                             Phone: 650.934.0895
iMNetworks, Inc.                        Fax: 650.967.3924
305 W. Evelyn Ave.                   E-Mail: bk(at)imnetworks.com
Mountain View, CA   94041               Web: www.imnetworks.com




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X-Original-Date: Thu, 13 Mar 2003 20:32:54 -0800
Date: Thu, 13 Mar 2003 20:32:54 -0800


>I'm using the liveMedia libraries to add multicast support
>to our internet product line and have run into a problem with the Win32
>version of the libraries: pairing strdup() -- a malloc based routine --
>with the C++ delete causes a heap-check assertion (RtlValidateHeap) when
>delete is called. Specifically, the "delete buffer" call in the
>MediaSession's parseCLine routine after the strdup of sdpLine traps.
[...]
>For what it's worth, we're using W2K Pro, SP3 with version 6 of Visual
>C++. So, two questions: do you see this when running under Windows

No, I don't, but I use Visual C++ version 5.0, which might not trip over 
the problem.

But I agree that the problem is real.  I'll fix it by defining a C++ 
equivalent to "strdup()" (that uses new instead of malloc), and then doing 
a global replace of "strdup" with this new function.

>  and any idea when a fix might be released?

I'll try to get it done later this evening.

Thanks for pointing out this problem.

         Ross.



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From <EMAIL-ADDRESS-HIDDEN> Thu Mar 13 21:11:07 2003
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X-Original-Date: Thu, 13 Mar 03 21:08:01 -0800
Date: Thu, 13 Mar 03 21:08:01 -0800

Thanks Ross for the amazingly fast turnaround. I thought of the fix you 
suggest, but isn't that just part of it? I believe you also have to use 
the aggregate form of the delete -- delete [] -- along with the new 
strdup...

>Subject:     Re: [Live-devel] malloc/delete clash
>Sent:        3/14/20 4:23 PM
>Received:    3/13/03 9:03 PM
>From:        Ross Finlayson, finlayson(at)live.com
>Reply-To:    live.com mailingList, live-devel(at)lists.sourceforge.net
>To:          live.com mailingList, live-devel(at)lists.sourceforge.net
>
>
>>I'm using the liveMedia libraries to add multicast support
>>to our internet product line and have run into a problem with the Win32
>>version of the libraries: pairing strdup() -- a malloc based routine --
>>with the C++ delete causes a heap-check assertion (RtlValidateHeap) when
>>delete is called. Specifically, the "delete buffer" call in the
>>MediaSession's parseCLine routine after the strdup of sdpLine traps.
>[...]
>>For what it's worth, we're using W2K Pro, SP3 with version 6 of Visual
>>C++. So, two questions: do you see this when running under Windows
>
>No, I don't, but I use Visual C++ version 5.0, which might not trip over 
>the problem.
>
>But I agree that the problem is real.  I'll fix it by defining a C++ 
>equivalent to "strdup()" (that uses new instead of malloc), and then doing 
>a global replace of "strdup" with this new function.
>
>>  and any idea when a fix might be released?
>
>I'll try to get it done later this evening.
>
>Thanks for pointing out this problem.
>
>         Ross.
>
>
>
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X-Original-Date: Thu, 13 Mar 2003 21:19:25 -0800
Date: Thu, 13 Mar 2003 21:19:25 -0800


>I believe you also have to use
>the aggregate form of the delete -- delete [] -- along with the new
>strdup...

Practically speaking, is there any difference between "delete" and "delete 
[]" when you're deleting memory that was allocated using "new 
char[....]"?  I thought that "delete[]" was important only when deleting an 
array of objects that have destructors?

         Ross.



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From <EMAIL-ADDRESS-HIDDEN> Thu Mar 13 21:58:50 2003
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Subject: Re: [Live-devel] malloc/delete clash
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X-Original-Date: Thu, 13 Mar 03 21:50:57 -0800
Date: Thu, 13 Mar 03 21:50:57 -0800

Well, I guess it depends on whether you're feeling lucky ;-) The bible 
(Stroustrup's "The C++ Programming Language, Special Edition") says 
(10.4.7 Arrays, page 250-1) that it's illicit to  pair the "wrong" delete 
with the "right" allocation. Bjarne says "Exactly how arrays and 
individual objects are allocated is implementation-dependent. Therefore, 
different implementaiotns will react differently to incorrect uses of the 
delete and delete [] operators. In simple and uninteresting cases like 
the previous one, a compiler can detect the problem, but generally 
something nasty will happen at run time."

Granted, we're fussier than some since we develop against a canonical 
platform that runs atop several different hardware/OS combinations, both 
pre-emptive and realtime. I suspect that in general it'd just work, but 
it's hard to say what meta-data a given tool chain carries along with 
native data types in the way of degenerate "destructors"...

>Subject:     Re: [Live-devel] malloc/delete clash
>Sent:        3/12/20 5:33 AM
>Received:    3/13/03 9:28 PM
>From:        Ross Finlayson, finlayson(at)live.com
>Reply-To:    live.com mailingList, live-devel(at)lists.sourceforge.net
>To:          live.com mailingList, live-devel(at)lists.sourceforge.net
>
>
>>I believe you also have to use
>>the aggregate form of the delete -- delete [] -- along with the new
>>strdup...
>
>Practically speaking, is there any difference between "delete" and "delete 
>[]" when you're deleting memory that was allocated using "new 
>char[....]"?  I thought that "delete[]" was important only when deleting an 
>array of objects that have destructors?
>
>         Ross.
>
>


---------------------------------------------------------------	
Bill Kain                             Phone: 650.934.0895
iMNetworks, Inc.                        Fax: 650.967.3924
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Mountain View, CA   94041               Web: www.imnetworks.com




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X-Original-Date: Fri, 14 Mar 2003 01:51:16 -0800
Date: Fri, 14 Mar 2003 01:51:16 -0800

FYI, I have now installed a new version - 2003.03.14 - of the code that (i) 
replaces calls to "strdup()" with calls to our own C++ function "strDup()", 
and (ii) calls "delete []" instead of "delete" where appropriate.

Please try out this new version, and let me know if there are any remaining 
problems.

	Ross.



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Subject: [Live-devel] Stream not seekable
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X-Original-Date: Fri, 14 Mar 2003 18:00:57 +0800
Date: Fri, 14 Mar 2003 18:00:57 +0800

Hi,

     When I rtsp'd an mp4 file using Darwin Streaming Server, the 
mplayer  shows "Stream not seekable" on the console and I can't go 
forward nor backward using the hotkeys. How to make a mp4 file seekable? 
Is there any tools for this?

Loren



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Subject: Re: [Live-devel] Stream not seekable
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X-Original-Date: Fri, 14 Mar 2003 02:25:02 -0800
Date: Fri, 14 Mar 2003 02:25:02 -0800


>    When I rtsp'd an mp4 file using Darwin Streaming Server, the 
> mplayer  shows "Stream not seekable" on the console and I can't go 
> forward nor backward using the hotkeys. How to make a mp4 file seekable? 
> Is there any tools for this?

Your mp4 file is not the problem.  MPlayer currently can't seek within 
*any* RTSP/RTP session, because the LIVE.COM RTSP client implementation 
doesn't yet fully implement the parts of the RTSP protocol that are used to 
seek within a stream.  (This is not yet a funded task, so if your company 
is interested in providing funding for this, just let me know.)

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>




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X-Original-Date: Fri, 14 Mar 2003 05:42:01 -0500
Date: Fri, 14 Mar 2003 05:42:01 -0500

The question is if it is possible to make an authentication before   
starting receiving a stream, if it is implemented by LIVE.COM  
library? I meen somethink like access to the MP4 file requires 
authentication. I request a MP4   
video file, then I have to go through authentication process, and   
then I start receiving data. If authentication fails, then I do not   
get any data streamed.  
If not implemented, any tips on how to make it?  
-- 
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X-Original-Date: Fri, 14 Mar 2003 09:16:52 -0800
Date: Fri, 14 Mar 2003 09:16:52 -0800

At 02:42 AM 3/14/03, you wrote:
>The question is if it is possible to make an authentication before
>starting receiving a stream, if it is implemented by LIVE.COM
>library? I meen somethink like access to the MP4 file requires
>authentication. I request a MP4
>video file, then I have to go through authentication process, and
>then I start receiving data. If authentication fails, then I do not
>get any data streamed.
>If not implemented, any tips on how to make it?

First, you are referring to "MPlayer", right?  (The LIVE.COM libraries are 
used in many other things as well.)

I don't think RTSP authentication is currently working in MPlayer.  You 
could *try* using an URL like
         rtsp://<username>:<password>(at)foo.example.com/bar.mp4
(replace <username> and <password> with your user name and password for 
authentication), but I don't think this will work just yet.

I will try to fix this in MPlayer sometime within the next few days, but if 
you're really desperate in the meantime, you could replace the call to
         describeURL(url)
in "libmpdemux/demux_rtp.cpp" (in the MPlayer code) with
         describeURL(url, "<username>", "<password>")
(again, replace <username> and <password> with your user name and 
password).  That should work (but obviously, only for one stream at a time).

         Ross.



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Subject: [Live-devel] Unable to create receiver for "audio/X-PN-REALAUDIO"
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X-Original-Date: 15 Mar 2003 20:35:07 -0800
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I'm having a problem playing realaudio.  I get the same basic error
message when I try to play the stream with mplayer (with live.com and
realplayer 9 codecs enabled).

Does anyone konw how to play this stream?


$ ./openRTSP  rtsp://audio.npr.org/me/20030312_me_rice.rm
Opened URL "rtsp://audio.npr.org/me/20030312_me_rice.rm", returning a
SDP description:
v=0
o=- 1047421221 1047421221 IN IP4 216.35.221.79
s=<No title>
i=NPR 2003
c=IN IP4 0.0.0.0
t=0 0
a=SdpplinVersion:1610643347
a=Flags:integer;2
a=IsRealDataType:integer;1
a=StreamCount:integer;1
a=Copyright:buffer;"MjAwMwA="
a=Author:buffer;"TlBSAA=="
a=range:npt=0-1515.796000
m=audio 0 RTP/AVP 101
b=AS:44
a=control:streamid=0
a=range:npt=0-1515.796000
a=length:npt=1515.796000
a=rtpmap:101 x-pn-realaudio
a=mimetype:string;"audio/x-pn-realaudio"
a=MinimumSwitchOverlap:integer;200
a=StartTime:integer;0
a=AvgBitRate:integer;44137
a=EndOneRuleEndAll:integer;1
a=AvgPacketSize:integer;640
a=EndTime:integer;1513592
a=SeekGreaterOnSwitch:integer;0
a=Preroll:integer;4640
a=MaxPacketSize:integer;640
a=MaxBitRate:integer;44137
a=RMFF 1.0 Flags:buffer;"AAwAAgAAAAIAAAACAAAAAgAAAAIAAAACAAA="
a=OpaqueData:buffer;"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"
a=StreamName:string;"audio/x-pn-multirate-realaudio logical stream"
a=ASMRuleBook:string;"#($OldPNMPlayer),AverageBandwidth=20000,priority=5,PNMKeyframeRule=T;#($OldPNMPlayer),AverageBandwidth=0,priority=5,PNMNonKeyframeRule=T;#($Bandwidth < 16230),AverageBandwidth=11043,Priority=5;#($Bandwidth < 16230),AverageBandwidth=0,Priority=5,OnDepend=\"2\", OffDepend=\"2\";#($Bandwidth >= 16230) && ($Bandwidth < 20000),AverageBandwidth=16230,Priority=5;#($Bandwidth >= 16230) && ($Bandwidth < 20000),AverageBandwidth=0,Priority=5,OnDepend=\"4\", OffDepend=\"4\";#($Bandwidth >= 20000) && ($Bandwidth < 32460),AverageBandwidth=20000,Priority=5;#($Bandwidth >= 20000) && ($Bandwidth < 32460),AverageBandwidth=0,Priority=5,OnDepend=\"6\", OffDepend=\"6\";#($Bandwidth >= 32460) && ($Bandwidth < 44137),AverageBandwidth=32460,Priority=5;#($Bandwidth >= 32460) && ($Bandwidth < 44137),AverageBandwidth=0,Priority=5,OnDepend=\"8\", OffDepend=\"8\";#($Bandwidth >= 44137),AverageBandwidth=44137,Priority=5;#($Bandwidth >= 44137),AverageBandwidth=0,Priority=5,OnDepend=\"10\", OffDepend=\"10\";"
 
Unable to create receiver for "audio/X-PN-REALAUDIO" subsession: RTP
payload format unknown or not supported
$




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X-Original-Date: Sat, 15 Mar 2003 21:20:38 -0800
Date: Sat, 15 Mar 2003 21:20:38 -0800

At 08:35 PM 3/15/03, you wrote:
>I'm having a problem playing realaudio.  I get the same basic error
>message when I try to play the stream with mplayer (with live.com and
>realplayer 9 codecs enabled).
>
>Does anyone konw how to play this stream?

Right now, you can't.  See <http://www.live.com/openRTSP/#real-media>.

         Ross.



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Subject: [Live-devel] Streams MUST be seakable!
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X-Original-Date: Mon, 17 Mar 2003 11:06:34 -0500
Date: Mon, 17 Mar 2003 11:06:34 -0500

Hallo, 
 
I have a question, how can I get the value of position of 
currently being played stream. I meen in C++ code in 
LIVE.COM library. I am currently working on making streams 
seakable and got pretty close to the end. The biggest 
problem is synchronisation problem. So my biggest question 
how can I get where I am now: at the beginning of the 
stream, 10 minutes after abeginning, and so on. 
 
Your answer would be very helpful. 
Thanks in advance, 
Elnar 
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Subject: RE: [Live-devel] Forwarding Stream begin recorded!
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X-Original-Date: Mon, 17 Mar 2003 11:46:48 -0500
Date: Mon, 17 Mar 2003 11:46:48 -0500

Hello:

Anybody knows how to Fast Forward a stream being recorded from an encoder.
Help will be greatly appreciated.  Thanks

:Kannan


-----Original Message-----
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[mailto:live-devel-admin(at)lists.sourceforge.net]On Behalf Of Elnar Hajiev
Sent: Monday, March 17, 2003 11:07 AM
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Subject: [Live-devel] Streams MUST be seakable!


Hallo, 
 
I have a question, how can I get the value of position of 
currently being played stream. I meen in C++ code in 
LIVE.COM library. I am currently working on making streams 
seakable and got pretty close to the end. The biggest 
problem is synchronisation problem. So my biggest question 
how can I get where I am now: at the beginning of the 
stream, 10 minutes after abeginning, and so on. 
 
Your answer would be very helpful. 
Thanks in advance, 
Elnar 
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X-Original-Date: Mon, 17 Mar 2003 09:41:13 -0800
Date: Mon, 17 Mar 2003 09:41:13 -0800


>I have a question, how can I get the value of position of
>currently being played stream.

Assuming you used RTSP to start playing the stream, you could figure this 
out from the "seq" and "rtptime" parameters that were returned in the 
"RTP-Info" header of the RTSP "PLAY" response (see, e.g., 
<http://www.rtsp.org/2003/drafts/draft03/draft-ietf-mmusic-rfc2326bis-03.txt>, 
section 13.33).

(Note, though, that the current RTSP client code does not doing anything 
with the "RTP-Info" header.)

         Ross.



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X-Original-Date: Mon, 17 Mar 2003 09:54:27 -0800
Date: Mon, 17 Mar 2003 09:54:27 -0800

Hi,

How can I get statistics from MPlayer, for example, the number of I frames
received/lost etc?

Thanks,
Rohit



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X-Original-Date: Mon, 17 Mar 2003 10:03:13 -0800
Date: Mon, 17 Mar 2003 10:03:13 -0800


>How can I get statistics from MPlayer, for example, the number of I frames
>received/lost etc?

One simple way to do this would be to 'sniff' for the RTCP "Reception 
Report" packets that the LIVE.COM RTCP implementation sends 
periodically.  These contain packet loss statistics.

Alternatively, you could add your own code to 
"MPEGVideoRTPSource::processSpecialHeader()" (which is called each time a 
MPEG video RTP packet is received) to keep track of whatever statistics you 
wish.

         Ross.



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X-Original-Date: Mon, 17 Mar 2003 10:17:20 -0800
Date: Mon, 17 Mar 2003 10:17:20 -0800

Thanks

----- Original Message -----
From: {Ross Finlayson} <EMAIL-ADDRESS-HIDDEN>
To: <EMAIL-ADDRESS-HIDDEN>
Sent: Monday, March 17, 2003 10:03 AM
Subject: Re: [Live-devel] MPlayer statistics


>
> >How can I get statistics from MPlayer, for example, the number of I
frames
> >received/lost etc?
>
> One simple way to do this would be to 'sniff' for the RTCP "Reception
> Report" packets that the LIVE.COM RTCP implementation sends
> periodically.  These contain packet loss statistics.
>
> Alternatively, you could add your own code to
> "MPEGVideoRTPSource::processSpecialHeader()" (which is called each time a
> MPEG video RTP packet is received) to keep track of whatever statistics
you
> wish.
>
>          Ross.
>
>
>
> -------------------------------------------------------
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X-Original-Date: Mon, 17 Mar 03 11:17:44 -0800
Date: Mon, 17 Mar 03 11:17:44 -0800

Thanks Ross, it worked like a champ under VC++ 6.

>Subject:     Re: [Live-devel] malloc/delete clash
>Sent:        3/16/20 6:33 AM
>Received:    3/14/03 12:10 PM
>From:        Ross Finlayson, finlayson(at)live.com
>Reply-To:    live.com mailingList, live-devel(at)lists.sourceforge.net
>To:          live.com mailingList, live-devel(at)lists.sourceforge.net
>
>FYI, I have now installed a new version - 2003.03.14 - of the code that (i) 
>replaces calls to "strdup()" with calls to our own C++ function "strDup()", 
>and (ii) calls "delete []" instead of "delete" where appropriate.
>
>Please try out this new version, and let me know if there are any remaining 
>problems.
>
>	Ross.
>
>
>
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Subject: [Live-devel] FramedSource/MediaSink buffers
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X-Original-Date: Mon, 17 Mar 03 11:39:31 -0800
Date: Mon, 17 Mar 03 11:39:31 -0800

I'm doing a first time integration of the live.com libraries into an 
internet audio player and have some questions about the buffer's that a 
MediaSink presents via FramedSource::getNextFrame(): are the buffers 
presented required to be the same size each time getNextFrame is called? 
Is there a minimum size required in practice? What happens when a buffer 
is presented to getNextFrame that's smaller than the source's packet 
buffer (i.e. what happens to the residual content)?

The application profile: a subsession iterator is used to walk a session 
selecting out Mpeg streams (i.e. MPA, MPA-ROBUST, X-MP3-DRAFT-00). A 
custom MediaSink delivers the selected source directly to client read 
buffers. Although we're able to connect to the session, select a stream, 
and "play" it, we get a tremendous number of frame sync errors and never 
see more than 2-4 seconds of coherent sound. Since the client reader 
assembles its own frames from the byte stream, the presented read buffer 
size varies from one getNextFrame call to the next.

Although we were trying to avoid extra buffer copies, it occurs that we 
might put a packet fifo between live.com and our reader -- say 2k 
packets. Is this likely to fix the sync errors/data loss? If not, any 
ideas as to what might?

We're running the 2003.03.14 live.com release under Windows2K Pro, SP3, 
building with VC++ version 6...

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iMNetworks, Inc.                        Fax: 650.967.3924
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Subject: Re: [Live-devel] MPlayer statistics
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X-Original-Date: Mon, 17 Mar 2003 12:07:21 -0800
Date: Mon, 17 Mar 2003 12:07:21 -0800

Hi Ross,

I wanted to ask one more thing. Would it be possible to also know frames
which are received, yet discarded by the player (for example, if they
arrived too late)? In a sense, I am interested in knowing only those frames
which are actually played by the player; all others I would like to consider
as lost. Such statistics would be really useful to me.

Thanks,
Rohit


----- Original Message -----
From: {Ross Finlayson} <EMAIL-ADDRESS-HIDDEN>
To: <EMAIL-ADDRESS-HIDDEN>
Sent: Monday, March 17, 2003 10:03 AM
Subject: Re: [Live-devel] MPlayer statistics


>
> >How can I get statistics from MPlayer, for example, the number of I
frames
> >received/lost etc?
>
> One simple way to do this would be to 'sniff' for the RTCP "Reception
> Report" packets that the LIVE.COM RTCP implementation sends
> periodically.  These contain packet loss statistics.
>
> Alternatively, you could add your own code to
> "MPEGVideoRTPSource::processSpecialHeader()" (which is called each time a
> MPEG video RTP packet is received) to keep track of whatever statistics
you
> wish.
>
>          Ross.
>
>
>
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X-Original-Date: Mon, 17 Mar 2003 13:21:39 -0800
Date: Mon, 17 Mar 2003 13:21:39 -0800


>I'm doing a first time integration of the live.com libraries into an
>internet audio player and have some questions about the buffer's that a
>MediaSink presents via FramedSource::getNextFrame(): are the buffers
>presented required to be the same size each time getNextFrame is called?

No, they don't have to be the same size.

>Is there a minimum size required in practice?

Yes, each buffer should be large enough to hold a complete input frame.

>  What happens when a buffer
>is presented to getNextFrame that's smaller than the source's packet
>buffer (i.e. what happens to the residual content)?

It will get truncated.

>The application profile: a subsession iterator is used to walk a session
>selecting out Mpeg streams (i.e. MPA, MPA-ROBUST, X-MP3-DRAFT-00). A
>custom MediaSink delivers the selected source directly to client read
>buffers. Although we're able to connect to the session, select a stream,
>and "play" it, we get a tremendous number of frame sync errors and never
>see more than 2-4 seconds of coherent sound. Since the client reader
>assembles its own frames from the byte stream, the presented read buffer
>size varies from one getNextFrame call to the next.
>
>Although we were trying to avoid extra buffer copies, it occurs that we
>might put a packet fifo between live.com and our reader -- say 2k
>packets. Is this likely to fix the sync errors/data loss?

No, it sounds like the errors you're seeing are caused by a programming 
error on your part.  If you replace your own custom 'sink' object with a 
"FileSink", is the resulting output file formed OK?  If so, then the 
problem is with your sink object.

         Ross.



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X-Original-Date: Mon, 17 Mar 2003 13:25:25 -0800
Date: Mon, 17 Mar 2003 13:25:25 -0800


>I wanted to ask one more thing. Would it be possible to also know frames
>which are received, yet discarded by the player (for example, if they
>arrived too late)?

Yes, you can figure this out by calling "RTPSource::curPacketRTPSeqNum()" 
each time you look at a new incoming packet (e..g., inside 
"MPEGVideoRTPSource::processSpecialHeader()").

         Ross.



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X-Original-Date: Mon, 17 Mar 2003 14:52:43 -0800
Date: Mon, 17 Mar 2003 14:52:43 -0800

Thanks

----- Original Message -----
From: {Ross Finlayson} <EMAIL-ADDRESS-HIDDEN>
To: <EMAIL-ADDRESS-HIDDEN>
Sent: Monday, March 17, 2003 1:25 PM
Subject: Re: [Live-devel] MPlayer statistics


>
> >I wanted to ask one more thing. Would it be possible to also know frames
> >which are received, yet discarded by the player (for example, if they
> >arrived too late)?
>
> Yes, you can figure this out by calling "RTPSource::curPacketRTPSeqNum()"
> each time you look at a new incoming packet (e..g., inside
> "MPEGVideoRTPSource::processSpecialHeader()").
>
>          Ross.
>
>
>
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Subject: Re: [Live-devel] FramedSource/MediaSink buffers
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X-Original-Date: Mon, 17 Mar 03 19:40:22 -0800
Date: Mon, 17 Mar 03 19:40:22 -0800

>Subject:     Re: [Live-devel] FramedSource/MediaSink buffers
>Sent:        3/18/20 7:53 AM
>Received:    3/17/03 2:47 PM
>From:        Ross Finlayson, finlayson(at)live.com
>Reply-To:    live.com mailingList, live-devel(at)lists.sourceforge.net
>To:          live.com mailingList, live-devel(at)lists.sourceforge.net
>
>
>>I'm doing a first time integration of the live.com libraries into an
>>internet audio player and have some questions about the buffer's that a
>>MediaSink presents via FramedSource::getNextFrame(): are the buffers
>>presented required to be the same size each time getNextFrame is called?
>
>No, they don't have to be the same size.
>
>>Is there a minimum size required in practice?
>
>Yes, each buffer should be large enough to hold a complete input frame.
Exactly. By consistently presenting a 2k buffer to getNextFrame, live.com 
now
works perfectly. Is 2k large enough for any FramedSource? If not, what 
is? 

Anyway, thanks Ross for your quick responses and excellent code. It took 
about a man-week to get a baseline understanding of the live.com 
libraries, instantiate them into a complex system, and with your help, 
get it up and running. Outstanding!



---------------------------------------------------------------	
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iMNetworks, Inc.                        Fax: 650.967.3924
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X-Original-Date: Mon, 17 Mar 2003 20:08:21 -0800
Date: Mon, 17 Mar 2003 20:08:21 -0800


> >Yes, each buffer should be large enough to hold a complete input frame.
>Exactly. By consistently presenting a 2k buffer to getNextFrame, live.com
>now
>works perfectly. Is 2k large enough for any FramedSource? If not, what
>is?

It depends on the maximum size of a frame for the particular medium & codec 
that's used.  For regular MP3 audio, 2 kBytes is enough; for mpa-robust MP3 
audio, 5 kBytes is enough; for MPEG-1 or MPEG-2 video, you may want to 
allow something like 50 kBytes, just to be on the safe side.

         Ross.



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Subject: [Live-devel] Streams MUST be seakable!
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X-Original-Date: Tue, 18 Mar 2003 02:51:52 -0500
Date: Tue, 18 Mar 2003 02:51:52 -0500

Hi Ross, 
 
but i get these parameters only once, when I start playing 
a stream. But my question is how can I get these values 
each moment. It is like that: user presses a button to 
seek forward, for example, and it can happen anytime 
during we play the stream. By the moment when user pressed 
the button, I need to find out where I am now (how many 
minutes after the beginning of the film) and then sum this 
value with 10 secs for example and send a new PLAY request 
to server to seek forward for 10 secs. That is how I 
imagine the whole process according to RTSP RFC. Any other 
advices are always wellcome. 
 
Thanks 
Elnar Hajiev 
----- Original Message ----- 
From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
Date: Mon, 17 Mar 2003 09:41:13 -0800  
To: <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] Streams MUST be seakable! 
 
>  
> >I have a question, how can I get the value of position 
of 
> >currently being played stream. 
>  
> Assuming you used RTSP to start playing the stream, you 
could figure this  
> out from the "seq" and "rtptime" parameters that were 
returned in the  
> "RTP-Info" header of the RTSP "PLAY" response (see, 
e.g.,  
> 
<http://www.rtsp.org/2003/drafts/draft03/draft-ietf-mmusic-rfc2326bis-03.txt>,  
> section 13.33). 
>  
> (Note, though, that the current RTSP client code does 
not doing anything  
> with the "RTP-Info" header.) 
>  
>          Ross. 
>  
>  
>  
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X-Original-Date: Tue, 18 Mar 2003 05:01:05 -0800
Date: Tue, 18 Mar 2003 05:01:05 -0800


>but i get these parameters only once, when I start playing
>a stream. But my question is how can I get these values
>each moment.

What I meant was that - given the "seq" and/or "rtptime" parameters (from 
the beginning of the stream), you can later use the current packet's RTP 
sequence number (using "RTPSource::curPacketRTPSeqNum()") and/or RTP 
timestamp (using "RTPSource::curPacketRTPTimestamp()") to tell you where 
you are now.

         Ross.



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X-Original-Date: Tue, 18 Mar 2003 09:09:22 -0500
Date: Tue, 18 Mar 2003 09:09:22 -0500

Absolutaly correct and I can already do so. You can't use seq number  
because it gets always bigger, and never smaller. Even if you start playing  
film from the beginning. But it works very good with rtptime and  
timestamps.  
A couple of another questions:  
How can I reset picture(set it black for example) if I pause just beeing  
played stream or seek it? 
Another question how does pause work when mplayer plays rtsp video stream?  
I ask this question because I could not find any PAUSE RTSP request in  
LIVE.COM library. And it looks like mplayer just stops playing streams, but  
keeps on buffering them in background. I meen the data is still streamed  
even after user has pressed pause, but mplayer does not play it, it just  
gathers it in buffer. And then when user presses play again, it plays data  
from the buffer. But If I press pause for a quite a long time(so that  
buffer is not enogh to keep all received data), some data will get lost. It  
is just my hypothese on how this process works and I am not quite sure if 
it works really this was,  
that is why I ask?  
  
Thanks 
Elnar 
 
----- Original Message -----  
From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
Date: Tue, 18 Mar 2003 05:01:05 -0800   
To: <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] Streams MUST be seakable!  
  
>   
> >but i get these parameters only once, when I start playing  
> >a stream. But my question is how can I get these values  
> >each moment.  
>   
> What I meant was that - given the "seq" and/or "rtptime" parameters (from   
> the beginning of the stream), you can later use the current packet's RTP   
> sequence number (using "RTPSource::curPacketRTPSeqNum()") and/or RTP   
> timestamp (using "RTPSource::curPacketRTPTimestamp()") to tell you where   
> you are now.  
>   
>          Ross.  
>   
>   
>   
> -------------------------------------------------------  
> This SF.net email is sponsored by: Does your code think in ink?   
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From <EMAIL-ADDRESS-HIDDEN> Tue Mar 18 14:58:59 2003
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Subject: Re: [Live-devel] Streams MUST be seekable!
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X-Original-Date: Tue, 18 Mar 2003 14:51:32 -0800
Date: Tue, 18 Mar 2003 14:51:32 -0800


>A couple of another questions:
>How can I reset picture(set it black for example) if I pause just beeing
>played stream or seek it?

I don't know.  That would be a question for the "MPlayer development" 
mailing list: mplayer-dev-eng(at)mplayerhq.hu

>Another question how does pause work when mplayer plays rtsp video stream?
>I ask this question because I could not find any PAUSE RTSP request in
>LIVE.COM library.

That's right - the RTSP client implementation does not yet include the 
ability to send "PAUSE" requests.

>  And it looks like mplayer just stops playing streams, but
>keeps on buffering them in background. I meen the data is still streamed
>even after user has pressed pause, but mplayer does not play it, it just
>gathers it in buffer. And then when user presses play again, it plays data
>from the buffer. But If I press pause for a quite a long time(so that
>buffer is not enogh to keep all received data), some data will get lost. It
>is just my hypothese on how this process works

Yes, this is correct.

         Ross.



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From <EMAIL-ADDRESS-HIDDEN> Wed Mar 19 08:06:43 2003
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Subject: [Live-devel] Slightly OT -- RTP timestamp length only 32 bits?
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X-Original-Date: Wed, 19 Mar 2003 08:03:13 -0800
Date: Wed, 19 Mar 2003 08:03:13 -0800


Hello, all ...

Please forgive the somewhat OT question, but I wanted to ask people who 
were intimately familiar with RTP (i've only worked with it on and off, 
and only in the last couple of months have I been really boning up on 
it).

Anyway, why is the RTP timestamp in the RTP header so small i.e. 32 
bits?  It seems that it will rollover quite often with high-bandwidth, 
highly-seekable media.  Even Windows Media (a format I don't especially 
condone because of it's very closed nature) implements packet 
timestamps that are 64 bits (in 100ns units).  Am I missing something, 
or is 32 bits really enough?

Regards,

John

Falling You - exploring the beauty of voice and sound
New EP, "hope thrown down," available now at
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  bits?
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X-Original-Date: Wed, 19 Mar 2003 09:38:41 -0800
Date: Wed, 19 Mar 2003 09:38:41 -0800


>Please forgive the somewhat OT question, but I wanted to ask people who 
>were intimately familiar with RTP (i've only worked with it on and off, 
>and only in the last couple of months have I been really boning up on it).
>
>Anyway, why is the RTP timestamp in the RTP header so small i.e. 32 
>bits?  It seems that it will rollover quite often with high-bandwidth, 
>highly-seekable media.

John,

Basically, the RTP timestamp was chosen to be 32 bits to save space.  Note 
that the RTP timestamp is not meant to represent 'absolute' time, nor the 
entire time range of a stream.  Instead, they represent the relative timing 
of packets.  RTP timestamps can still be mapped to absolute time using 
periodic RTCP "Sender Report" packets, which include a mapping from RTP 
timestamp to (64-bit) NTP ("network time protocol") 'wall clock' time.

For more information about RTP timestamps, see sections 5.1 and 6.4.1 of 
the latest RTP specification 
<http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-new-12.txt>

         Ross.



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Subject: [Live-devel] Mapping timestamp to ntp time?
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X-Original-Date: Thu, 20 Mar 2003 04:38:34 -0500
Date: Thu, 20 Mar 2003 04:38:34 -0500

Ross, if I send PAUSE request to Server to pause the 
stream, and then later want to resume it sometime again, I 
have to send PLAY again with the position, where the 
stream should be started again(resumed). The problem is I 
currently do not know how to get these position after I've 
sent PAUSE request. I know how to get rtptime parameter 
but I do not know how to map it(convert) to actual time(to 
secs after beginning of the film) and if this timestamp 
value is precise enough so after user presses resume 
button, he will not see any broken frames or any frames 
repeated, caused by not very correct resume time sent with 
PLAY request to Server.You write that it is possible to 
map rtp time to ntp time. If you could just shortly 
describe this process and if it is that what I need at all 
or am I on the wrong way? I've read those parts of RFC but 
could not actually get how the mapping works and what 
values do I actually need? 
Note: according to RTSP RFC Server has to remember pause 
position itself, and after following PLAY request without 
having Range specified, it schould resume from the 
position where it was paused, but that is not the way 
Darwin Streaming Server works. I have dumped RTSP messeges 
sent between QuickTime Player and Darwin and saw that 
QPlayer send the resume position itself after every PAUSE 
request. So I have to make it also this way, but the 
question is how. Hope you can help. 
 
Thanks 
Elnar 
----- Original Message ----- 
From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
Date: Wed, 19 Mar 2003 09:38:41 -0800  
To: <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] Slightly OT -- RTP timestamp 
length only 32 bits? 
 
>  
> >Please forgive the somewhat OT question, but I wanted 
to ask people who  
> >were intimately familiar with RTP (i've only worked 
with it on and off,  
> >and only in the last couple of months have I been 
really boning up on it). 
> > 
> >Anyway, why is the RTP timestamp in the RTP header so 
small i.e. 32  
> >bits?  It seems that it will rollover quite often with 
high-bandwidth,  
> >highly-seekable media. 
>  
> John, 
>  
> Basically, the RTP timestamp was chosen to be 32 bits to 
save space.  Note  
> that the RTP timestamp is not meant to represent 
'absolute' time, nor the  
> entire time range of a stream.  Instead, they represent 
the relative timing  
> of packets.  RTP timestamps can still be mapped to 
absolute time using  
> periodic RTCP "Sender Report" packets, which include a 
mapping from RTP  
> timestamp to (64-bit) NTP ("network time protocol") 
'wall clock' time. 
>  
> For more information about RTP timestamps, see sections 
5.1 and 6.4.1 of  
> the latest RTP specification  
> 
<http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-new-12.txt> 
>  
>          Ross. 
>  
>  
>  
> ------------------------------------------------------- 
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X-Original-Date: Thu, 20 Mar 2003 08:10:37 -0800
Date: Thu, 20 Mar 2003 08:10:37 -0800


>You write that it is possible to
>map rtp time to ntp time. If you could just shortly
>describe this process and if it is that what I need at all
>or am I on the wrong way? I've read those parts of RFC but
>could not actually get how the mapping works and what
>values do I actually need?

You might want to take a look at "RTPSource.cpp".  This is where the 
library maps RTP timestamps (of incoming packets) to its 'presentation time.

         Ross.



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hi!  has the live.com libraries (specifically rtp/rtcp) been compiled and
tested on strongarm? has anyone made a project on this?  i'm new on rtp and
would like to build the stack on arm.  i've also found two other open source
rtp libraries, one from vovida.org and the other ccRTP from GNU.  how does
LIVE.COM compare with them? any ideas on these would be appreciated.
thank you very much in advance.

mildred


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<DIV><SPAN class=3D474471810-21032003><FONT face=3DArial =
size=3D2>hi!&nbsp; has the=20
live.com libraries (specifically rtp/rtcp) been compiled and tested on=20
strongarm? has anyone made a project on this?&nbsp; i'm new on rtp and =
would=20
like to build the stack on arm.&nbsp;&nbsp;i've also found two other =
open source=20
rtp libraries, one from vovida.org and the other ccRTP from GNU.&nbsp; =
how does=20
LIVE.COM compare with them?&nbsp;any ideas on these would be =
appreciated.=20
</FONT></SPAN></DIV>
<DIV><SPAN class=3D474471810-21032003><FONT face=3DArial size=3D2>thank =
you very much=20
in advance.</FONT></SPAN></DIV>
<DIV><SPAN class=3D474471810-21032003><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D474471810-21032003><FONT face=3DArial=20
size=3D2>mildred</FONT></SPAN></DIV>
<DIV><SPAN class=3D474471810-21032003><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV></BODY></HTML>

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hi!  has the live.com libraries (specifically rtp/rtcp) been compiled and
tested on strongarm? has anyone made a project on this?  i'm new on rtp and
would like to build the stack on arm.  i've also found two other open source
rtp libraries, one from vovida.org and the other ccRTP from GNU.  how does
LIVE.COM compare with them? any ideas on these would be appreciated.
thank you very much in advance.

mildred


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<BODY>
<DIV><FONT face=3DArial size=3D2>
<DIV><SPAN class=3D474471810-21032003><FONT face=3DArial =
size=3D2>hi!&nbsp; has the=20
live.com libraries (specifically rtp/rtcp) been compiled and tested on=20
strongarm? has anyone made a project on this?&nbsp; i'm new on rtp and =
would=20
like to build the stack on arm.&nbsp;&nbsp;i've also found two other =
open source=20
rtp libraries, one from vovida.org and the other ccRTP from GNU.&nbsp; =
how does=20
LIVE.COM compare with them?&nbsp;any ideas on these would be =
appreciated.=20
</FONT></SPAN></DIV>
<DIV><SPAN class=3D474471810-21032003><FONT face=3DArial size=3D2>thank =
you very much=20
in advance.</FONT></SPAN></DIV>
<DIV><SPAN class=3D474471810-21032003><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D474471810-21032003><FONT face=3DArial=20
size=3D2>mildred</FONT></SPAN></DIV>
<DIV><SPAN=20
class=3D474471810-21032003></SPAN>&nbsp;</DIV></FONT></DIV></BODY></HTML>=


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Date: Fri, 21 Mar 2003 06:57:48 -0500

Ross, very shortly one question. It is not correctly addressed to this mail 
list, but those guys in MPlayer-users and MPlayer-dev-eng do not respond to 
my questions. 
You know the answer for sure, how can I free the RTSP buffer, so after I 
press pause, and then resume in MPlayer, there will be nothing more played 
from the buffer if the streaming server does not stream anymore. Which 
buffers are those? I suppose demuxer, but anything I do on it does not 
help. 
I would very appreciate your answer. 
Thank you in advance. 
Elnar 
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At 02:21 AM 3/21/03, you wrote:
>hi!  has the live.com libraries (specifically rtp/rtcp) been compiled and 
>tested on strongarm?

Yes it has.

         Ross.



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Date: Fri, 21 Mar 2003 14:16:40 -0800


>how can I free the RTSP buffer, so after I
>press pause, and then resume in MPlayer, there will be nothing more played
>from the buffer if the streaming server does not stream anymore. Which
>buffers are those?

By "RTSP buffer", I presume you really mean "buffering of incoming RTP 
packet data"?

There are two kinds of buffering in MPlayer (for incoming RTP streams):
1/ The buffering - internal to the OS - of incoming packets on each UDP socket.
2/ The "demux_packet_t" structure that MPlayer uses to process (decode) 
incoming data.

See "libmpdemux/demux_rtp.cpp" for the details of how incoming UDP (i.e., 
RTP) data is read into "demux_packet_t" structures.

         Ross.



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X-Original-Date: Mon, 24 Mar 2003 10:04:34 +0800
Date: Mon, 24 Mar 2003 10:04:34 +0800


	great! how much memory did it occupy?  i would like to build it on iPAQ and
install embedded linux on it then put the rtp stack support for its tcp/ip
stack.

-----Original Message-----
From: <EMAIL-ADDRESS-HIDDEN>
[mailto:live-devel-admin(at)lists.sourceforge.net]On Behalf Of Ross
Finlayson
Sent: Saturday, March 22, 2003 6:09 AM
To: <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] rtp on arm


At 02:21 AM 3/21/03, you wrote:
>hi!  has the live.com libraries (specifically rtp/rtcp) been compiled and
>tested on strongarm?

Yes it has.

         Ross.



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>         great! how much memory did it occupy?

I don't know, because I haven't yet done it myself.  (Also, not everyone 
who has done this has compiled it for Linux; some people targeted WinCE 
instead.)

         Ross.



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Subject: [Live-devel] help (MP3 Streaming Media)
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X-Original-Date: Mon, 24 Mar 2003 10:37:21 +0530
Date: Mon, 24 Mar 2003 10:37:21 +0530


Hi There,

     I couldn't compile ur source code for MP3(RTP/RTCP) streaming on
windows NT OS...
     can u pls suggest me the steps "how to do unpack and compile steps on
windows platform?
     I am eagerly waiting for ur reply. Pls mail me the complete
installation procedure for Windows.
     Thanks in advance...

  Thanks & Regards,
    - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    Lakshmi Kumari K
    Telecom-IP
    Tata Consultany Services
    5-9-62, 6th Floor, Khan Lateef Khan Estate
    Fateh Maidan Road, Hyderabad - 500001
    Phone: +91 40 5567 1122
    Fax      : +91 40 5567 1111
    - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -


                                                                                                                   
                    Ross                                                                                           
                    Finlayson            To:     lakshmik(at)hydbad.tcs.co.in                                         
                    <finlayson(at)li        cc:                                                                       
                    ve.com>              Subject:     Re: help                                                     
                                                                                                                   
                    03/22/03                                                                                       
                    09:15 AM                                                                                       
                                                                                                                   
                                                                                                                   





>   I have downloaded ur source code for MP3 Streaming Media. ...
>   unpacket and run on Linux OS. but i am unable to do the sa...
>   If i type as u said "./genWindowsMakefiles" , it is giving...
>   Too many errors encountered; the rest of the message is ignored:
>   "the name specified is not recognized as an internal or external
command,
>   operable program or batch file"

 From your description, it's hard to tell what might be wrong.  Could you
post another message - this time containing the complete error output - to
"live-devel(at)lists.sourceforge.net".  (The email address that you used last
time was incorrect.)

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>






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Date: Mon, 24 Mar 2003 10:49:24 +0530





Hi There,

     I couldn't compile ur source code for MP3(RTP/RTCP) streaming on
windows NT OS...
     can u pls suggest me the steps "how to do unpack and compile steps on
windows platform?
     I am eagerly waiting for ur reply. Pls mail me the complete
installation procedure for Windows.
     Thanks in advance...

    Thanks & Regards,
    - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    Lakshmi Kumari K
    Telecom-IP
    Tata Consultany Services
    5-9-62, 6th Floor, Khan Lateef Khan Estate
    Fateh Maidan Road, Hyderabad - 500001
    Phone: +91 40 5567 1122
    Fax      : +91 40 5567 1111
    - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -



                    Ross
                    Finlayson            To:     lakshmik(at)hydbad.tcs.co.in
                    <finlayson(at)li        cc:
                    ve.com>              Subject:     Re: help

                    03/22/03
                    09:15 AM







>   I have downloaded ur source code for MP3 Streaming Media. ...
>   unpacket and run on Linux OS. but i am unable to do the sa...
>   If i type as u said "./genWindowsMakefiles" , it is giving...
>   Too many errors encountered; the rest of the message is ignored:
>   "the name specified is not recognized as an internal or external
command,
>   operable program or batch file"

 From your description, it's hard to tell what might be wrong.  Could you
post another message - this time containing the complete error output - to
"live-devel(at)lists.sourceforge.net".  (The email address that you used last
time was incorrect.)

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>






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X-Original-Date: Sun, 23 Mar 2003 21:34:36 -0800
Date: Sun, 23 Mar 2003 21:34:36 -0800

Instructions for building the code on Windows are laid out quite clearly at 
<http://www.live.com/liveMedia/#config-windows>.  If there's some part of 
this that you think could be explained more clearly, just let me know.

	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Mon, 24 Mar 2003 11:24:36 +0530
Date: Mon, 24 Mar 2003 11:24:36 +0530


     Thanks for ur  prompt reply..i had gone thru it...still i couldn't
compile it correctly on Windows...
     Pls give me some more details step by step clearly...
     my doubt is "do i need to unpack it on WIndows or Linux.
     what i did actually is " i have done unzipped on Windows..
     and next went to live directory and typed "./genWindowsMakefiles"
....its is giving the error that
     "the name specified is not recognised as an internal or external
command, operable program
     or batch file"

     I am using VC++ 6.0 for compiling and running..i did not change any
settings in VC++...
     u said change 'tools' directory on your Windows machine is something
other than
     "c:\Program Files\DevStudio\Vc", change the "TOOLS32 =" line in the
file "win32config".
     I couldn't get it exactly?
     my VC++ directory is "D:\Program Files\Microsoft Visual Studio\common"
like that...
     do i need to change any settings in this?
     Pls mail me the details immd..

     I am very thankful in advance.....



  Thanks & Regards,
    - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
    Lakshmi Kumari K
    Telecom-IP
    Tata Consultany Services
    5-9-62, 6th Floor, Khan Lateef Khan Estate
    Fateh Maidan Road, Hyderabad - 500001
    Phone: +91 40 5567 1122
    Fax      : +91 40 5567 1111
    - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -


                                                                                                                                  
                    Ross Finlayson                                                                                                
                    <finlayson(at)live.com>                To:     live-devel(at)lists.sourceforge.net                                  
                    Sent by:                            cc:                                                                       
                    live-devel-admin(at)lists.sourc        Subject:     Re: [Live-devel] help (MP3 Streaming Media)                  
                    eforge.net                                                                                                    
                                                                                                                                  
                                                                                                                                  
                    03/24/03 11:04 AM                                                                                             
                    Please respond to live-devel                                                                                  
                                                                                                                                  
                                                                                                                                  




Instructions for building the code on Windows are laid out quite clearly at
<http://www.live.com/liveMedia/#config-windows>.  If there's some part of
this that you think could be explained more clearly, just let me know.

     Ross Finlayson
     LIVE.COM
     <http://www.live.com/>



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X-Original-Date: Mon, 24 Mar 2003 09:16:53 -0500
Date: Mon, 24 Mar 2003 09:16:53 -0500

You were absolutaly right, I ment buffering of incoming RTP packet data. There is a function 
ds_free_packs() which clears all demux_packet_t structures found in a stream. But it does not seem to 
be working. After resume is pressed, MPlayer still plays a very short period of time an old part of 
movie, before it goes to a new position. Where should I look for freeing 1(internal) kind of buffers, 
internal to the OS. 
 
Thanks. 
 
----- Original Message ----- 
From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
Date: Fri, 21 Mar 2003 14:16:40 -0800  
To: <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] RTSP Buffer 
 
>  
> >how can I free the RTSP buffer, so after I 
> >press pause, and then resume in MPlayer, there will be nothing more played 
> >from the buffer if the streaming server does not stream anymore. Which 
> >buffers are those? 
>  
> By "RTSP buffer", I presume you really mean "buffering of incoming RTP  
> packet data"? 
>  
> There are two kinds of buffering in MPlayer (for incoming RTP streams): 
> 1/ The buffering - internal to the OS - of incoming packets on each UDP socket. 
> 2/ The "demux_packet_t" structure that MPlayer uses to process (decode)  
> incoming data. 
>  
> See "libmpdemux/demux_rtp.cpp" for the details of how incoming UDP (i.e.,  
> RTP) data is read into "demux_packet_t" structures. 
>  
>          Ross. 
>  
>  
>  
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X-Original-Date: Mon, 24 Mar 2003 16:56:59 -0800
Date: Mon, 24 Mar 2003 16:56:59 -0800

At 06:16 AM 3/24/03, you wrote:
>You were absolutaly right, I ment buffering of incoming RTP packet data. 
>There is a function
>ds_free_packs() which clears all demux_packet_t structures found in a 
>stream. But it does not seem to
>be working. After resume is pressed, MPlayer still plays a very short 
>period of time an old part of
>movie, before it goes to a new position. Where should I look for freeing 
>1(internal) kind of buffers,
>internal to the OS.

I don't know how (or even if) you can flush the OS's UDP socket 
buffer.  But what you can do, though, is continue to read packets, as 
usual, but discard them (using "free_demux_packet()") if their presentation 
timestamp is less than the value you want.  (See "demux_rtp.cpp" for an 
example of how I already do this (for A/V synchronization) if the audio 
stream gets too far behind the video stream, or vice versa.  Look for 
"ptsBehindThreshold".)

         Ross.





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X-Original-Date: Mon, 24 Mar 2003 20:02:42 -0500
Date: Mon, 24 Mar 2003 20:02:42 -0500

I wanted to go ahead and respond back. While I haven't personally had a chance
to try the new stuff, yet, a coworker next to me did. Worked pretty darn well,
though the video on the CNN stream on the mbone was a tad jerky. 

I do, however, have two additional questions:
1. Will mpalyer/live libs read and respect the value in the sdp file which sets
the packet size? From what I have heard, the videolan client does not respect
this value but does have an option someplace to set it to something else.
Haven't figured out other players yet.
2. Will live.com/mplayer handle raw (not rtp) streams? Not exactly sure what
this means, as I thought some protocol had to be involved in the streaming, but
maybe not (a test project here is streaming raw mpeg2).

Thanks!

On Mon, Mar 10, 2003 at 10:22:28AM -0800, Ross Finlayson wrote:
> 
> >I haven't tried the new cvses of mplayer and the new live.com libs yet, 
> >but when plyaing the sdp streams (and yes, they do play, now), the 
> >video/audio has a lot of lost data, it seems, and the console output of 
> >mplayer shows:
> >
> >(at)(at)(at) libmpeg2 returned from sig11... (bad file?) (at)(at)(at)
> >(at)(at)(at) libmpeg2 returned from sig11... (bad file?) (at)(at)(at)
> >(at)(at)(at) libmpeg2 returned from sig11... (bad file?) (at)(at)(at)
> >(at)(at)(at) libmpeg2 returned from sig11... (bad file?) (at)(at)(at)
> >(at)(at)(at) libmpeg2 returned from sig11... (bad file?) (at)(at)(at)
> >
> >lots of these lines appearing every second. Any general ideas?
> 
> Yes, the current MPlayer code doesn't deal with high bitrate streams 
> particularly well - it drops data far more than it should (which causes the 
> MPEG decoder to complain).
> 
> Within a few days I'll be checking in a new version of the MPlayer code 
> that significantly improves this.  If you're impatient, though, you can try 
> running the binary version that I have left online as
>          http://www.live.com/mplayer/mplayer
> 
>      Ross.
> 
> 
> 
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X-Original-Date: Mon, 24 Mar 2003 17:08:07 -0800
Date: Mon, 24 Mar 2003 17:08:07 -0800


>     Thanks for ur  prompt reply..i had gone thru it...still i couldn't
>compile it correctly on Windows...
>      Pls give me some more details step by step clearly...
>      my doubt is "do i need to unpack it on WIndows or Linux.
>      what i did actually is " i have done unzipped on Windows..
>      and next went to live directory and typed "./genWindowsMakefiles"
>....its is giving the error that
>      "the name specified is not recognised as an internal or external
>command, operable program
>      or batch file"

Sigh...  As the web site explains, "genWindowsMakefiles" is a *Unix* 
command - i.e., one that you run on a Unix machine, to generate ".mak" 
files that you would then copy to a Windows machine.

If, for some reason, Windows computers are all that you have, see the 
following paragraph, titled "If you don't have access to a Unix machine"

         Ross.



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Subject: Re: [Live-devel] compiling with mplayer fails
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X-Original-Date: Mon, 24 Mar 2003 17:18:51 -0800
Date: Mon, 24 Mar 2003 17:18:51 -0800


>I do, however, have two additional questions:
>1. Will mpalyer/live libs read and respect the value in the sdp file which 
>sets
>the packet size?

No.  (What "value in the SDP file" is this??)

>2. Will live.com/mplayer handle raw (not rtp) streams?

No.  Why on Earth would you want to do this?  RTP is the Internet standard 
for A/V streaming.  If you were to try to stream in some other way, it 
would be incompatible with dozens of existing pieces of software (client 
and server).

         Ross.



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X-Original-Date: Mon, 24 Mar 2003 20:58:48 -0500
Date: Mon, 24 Mar 2003 20:58:48 -0500

Good question on why we would want to do RAW. Evidently it is something that
the VBrick streaming servers do. Other than that, I don't know much more as I
am not an expert on multicast. You are right that it seems to break clients.

But, you will see from the included sdp file, the packetformat is set to RAW.

Re the packetsize, I am referring to:
a=packetsize:4136
(which, I am told, is the VBrick server default).

Here is an example file from an internal test stream:

v=0
o=- 0007df0006b4 0 IN IP4 132.236.93.110
s=Cornell - SciFi
i=VBrick Streaming Video
c=IN IP4 239.239.239.236/15
b=AS:5689900
t=0 0
a=packetsize:4136
a=packetformat:RAW
a=mux:m1s
a=mux:m2t
a=keywds:
a=author:_
a=copyright:_
m=video 4444 udp 33



On Mon, Mar 24, 2003 at 05:18:51PM -0800, Ross Finlayson wrote:
> 
> >I do, however, have two additional questions:
> >1. Will mpalyer/live libs read and respect the value in the sdp file which 
> >sets
> >the packet size?
> 
> No.  (What "value in the SDP file" is this??)
> 
> >2. Will live.com/mplayer handle raw (not rtp) streams?
> 
> No.  Why on Earth would you want to do this?  RTP is the Internet standard 
> for A/V streaming.  If you were to try to stream in some other way, it 
> would be incompatible with dozens of existing pieces of software (client 
> and server).
> 
>          Ross.
> 
> 
> 
> -------------------------------------------------------
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-- 
********************************
David William Botsch
Consultant/Advisor II
CCMR Computing Facility
dwb7(at)ccmr.cornell.edu
********************************


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X-Original-Date: Mon, 24 Mar 2003 18:14:13 -0800
Date: Mon, 24 Mar 2003 18:14:13 -0800

At 05:58 PM 3/24/03, you wrote:
>Good question on why we would want to do RAW. Evidently it is something that
>the VBrick streaming servers do.

I'm pretty sure that the VBrick servers can be configured to stream using 
RTP instead.

>Re the packetsize, I am referring to:
>a=packetsize:4136
>(which, I am told, is the VBrick server default).

The LIVE.COM library code should be able to receive and handle RTP packets 
of any size (unless they're ridiculously large - like 50 kBytes).  It 
doesn't recognize any "a=packetsize" line in SDP descriptions, but that's 
just a 'hint' for receivers anyway.  The library code should handle 
incoming RTP packets of this size nonetheless.

         Ross.



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Subject: Re: [Live-devel] MPlayer statistics
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X-Original-Date: Mon, 24 Mar 2003 18:26:09 -0800
Date: Mon, 24 Mar 2003 18:26:09 -0800

Hi Ross,
Just wanted to ask one more thing here. If I understand right,
"RTPSource::curPacketRTPSeqNum()" gives the RTP sequence number of the just
received packet. What do I compare this with to know if this packet is too
late to be played by the MPlayer, i.e., the MPlayer is going to ignore it?

Thanks,
Rohit

----- Original Message -----
From: {Ross Finlayson} <EMAIL-ADDRESS-HIDDEN>
To: <EMAIL-ADDRESS-HIDDEN>
Sent: Monday, March 17, 2003 1:25 PM
Subject: Re: [Live-devel] MPlayer statistics


>
> >I wanted to ask one more thing. Would it be possible to also know frames
> >which are received, yet discarded by the player (for example, if they
> >arrived too late)?
>
> Yes, you can figure this out by calling "RTPSource::curPacketRTPSeqNum()"
> each time you look at a new incoming packet (e..g., inside
> "MPEGVideoRTPSource::processSpecialHeader()").
>
>          Ross.
>
>
>
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X-Original-Date: Mon, 24 Mar 2003 21:28:26 -0500
Date: Mon, 24 Mar 2003 21:28:26 -0500

I'll have to see if I can convince them to do RTP instead. Don't know why they
are doing RAW. Don't know what their reasoning is. Less overhead perhaps?

If that is resolved, maybe I can test the large packet size.

On Mon, Mar 24, 2003 at 06:14:13PM -0800, Ross Finlayson wrote:
> At 05:58 PM 3/24/03, you wrote:
> >Good question on why we would want to do RAW. Evidently it is something that
> >the VBrick streaming servers do.
> 
> I'm pretty sure that the VBrick servers can be configured to stream using 
> RTP instead.
> 
> >Re the packetsize, I am referring to:
> >a=packetsize:4136
> >(which, I am told, is the VBrick server default).
> 
> The LIVE.COM library code should be able to receive and handle RTP packets 
> of any size (unless they're ridiculously large - like 50 kBytes).  It 
> doesn't recognize any "a=packetsize" line in SDP descriptions, but that's 
> just a 'hint' for receivers anyway.  The library code should handle 
> incoming RTP packets of this size nonetheless.
> 
>          Ross.
> 
> 
> 
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X-Original-Date: Mon, 24 Mar 2003 18:42:49 -0800
Date: Mon, 24 Mar 2003 18:42:49 -0800


>Just wanted to ask one more thing here. If I understand right,
>"RTPSource::curPacketRTPSeqNum()" gives the RTP sequence number of the just
>received packet. What do I compare this with to know if this packet is too
>late to be played by the MPlayer, i.e., the MPlayer is going to ignore it?

Rohit,

You don't need to do anything.  The LIVE.COM library automatically makes 
sure that incoming RTP packets are presented to the receiver (such as 
MPlayer) in the correct order.

If there's a gap in the RTP sequence number of received packets, then the 
library code will automatically wait for a certain period of time (by 
default, 100 ms) before deciding that it's not going to wait for the 
missing packet(s).  E.g., if the library receives packet #42, then packet 
#44, it will wait 100 ms for packet #43 to arrive, before giving up on it 
and delivering packet #44 instead.  (This is done in 
"MultiFramedRTPSource.cpp".)

So, again, all this is already done automatically by the library.

         Ross.



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X-Original-Date: Mon, 24 Mar 2003 18:55:29 -0800
Date: Mon, 24 Mar 2003 18:55:29 -0800

Thanks. That answers my question.

Rohit

----- Original Message -----
From: {Ross Finlayson} <EMAIL-ADDRESS-HIDDEN>
To: <EMAIL-ADDRESS-HIDDEN>
Sent: Monday, March 24, 2003 6:42 PM
Subject: Re: [Live-devel] MPlayer statistics


>
> >Just wanted to ask one more thing here. If I understand right,
> >"RTPSource::curPacketRTPSeqNum()" gives the RTP sequence number of the
just
> >received packet. What do I compare this with to know if this packet is
too
> >late to be played by the MPlayer, i.e., the MPlayer is going to ignore
it?
>
> Rohit,
>
> You don't need to do anything.  The LIVE.COM library automatically makes
> sure that incoming RTP packets are presented to the receiver (such as
> MPlayer) in the correct order.
>
> If there's a gap in the RTP sequence number of received packets, then the
> library code will automatically wait for a certain period of time (by
> default, 100 ms) before deciding that it's not going to wait for the
> missing packet(s).  E.g., if the library receives packet #42, then packet
> #44, it will wait 100 ms for packet #43 to arrive, before giving up on it
> and delivering packet #44 instead.  (This is done in
> "MultiFramedRTPSource.cpp".)
>
> So, again, all this is already done automatically by the library.
>
>          Ross.
>
>
>
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X-Original-Date: Mon, 24 Mar 2003 19:01:57 -0800
Date: Mon, 24 Mar 2003 19:01:57 -0800

Ross,
Just one more clarification. Suppose the library waits for 100msec and the
packet does not arrive within that time; the packet then arrives later. Then
the library will drop the packet, right? In that case, will this packet be
displayed by RTPSource::curPacketRTPSeqNum() (when this function is called
from "MPEGVideoRTPSource::processSpecialHeader()" ). In other words, are the
packets which are received in "MPEGVideoRTPSource::processSpecialHeader()"
only those which are actually going to be played by MPlayer?

I am just trying to make sure that my statistics only include those packets
which are actually played by the MPlayer.

Thanks,
Rohit

----- Original Message -----
From: {Ross Finlayson} <EMAIL-ADDRESS-HIDDEN>
To: <EMAIL-ADDRESS-HIDDEN>
Sent: Monday, March 24, 2003 6:42 PM
Subject: Re: [Live-devel] MPlayer statistics


>
> >Just wanted to ask one more thing here. If I understand right,
> >"RTPSource::curPacketRTPSeqNum()" gives the RTP sequence number of the
just
> >received packet. What do I compare this with to know if this packet is
too
> >late to be played by the MPlayer, i.e., the MPlayer is going to ignore
it?
>
> Rohit,
>
> You don't need to do anything.  The LIVE.COM library automatically makes
> sure that incoming RTP packets are presented to the receiver (such as
> MPlayer) in the correct order.
>
> If there's a gap in the RTP sequence number of received packets, then the
> library code will automatically wait for a certain period of time (by
> default, 100 ms) before deciding that it's not going to wait for the
> missing packet(s).  E.g., if the library receives packet #42, then packet
> #44, it will wait 100 ms for packet #43 to arrive, before giving up on it
> and delivering packet #44 instead.  (This is done in
> "MultiFramedRTPSource.cpp".)
>
> So, again, all this is already done automatically by the library.
>
>          Ross.
>
>
>
> -------------------------------------------------------
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X-Original-Date: Mon, 24 Mar 2003 19:07:37 -0800
Date: Mon, 24 Mar 2003 19:07:37 -0800


>Just one more clarification. Suppose the library waits for 100msec and the
>packet does not arrive within that time; the packet then arrives later. Then
>the library will drop the packet, right?

Yes.

>  In that case, will this packet be
>displayed by RTPSource::curPacketRTPSeqNum() (when this function is called
>from "MPEGVideoRTPSource::processSpecialHeader()" ).

No.

>  In other words, are the
>packets which are received in "MPEGVideoRTPSource::processSpecialHeader()"
>only those which are actually going to be played by MPlayer?

Yes.

         Ross.



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X-Original-Date: Tue, 25 Mar 2003 04:06:54 -0500
Date: Tue, 25 Mar 2003 04:06:54 -0500

I found the code, you've mentioned. Here is how I do it using your example: 
 
  float ptsBehind; 
  demux_packet_t *dp; 
  while(1) { 
        dp = getBuffer(demuxer, demuxer->video, True, ptsBehind); 
        if(dp) { 
           free_demux_packet(dp); 
           //printf("ptsBehind %f\n", ptsBehind); 
           //fflush(stdout); 
        } 
        else { 
           //printf("else\n"); 
           break; 
          } 
  } 
I call this code after Darwin Streaming Server has paused its streams. And 
I assume that this code will call getBuffer and then free_demux_packet 
untill getBuffer returns NULL and that means that there are no more 
packets. 
But in reality this loop functiions 2 or 3 times, and then MPlayer halts. 
Why? What do I do wrong? Why blocking function getBuffer() stops 
responding? 
 
 
----- Original Message ----- 
From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
Date: Mon, 24 Mar 2003 16:56:59 -0800  
To: <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] RTP Buffer 
 
> At 06:16 AM 3/24/03, you wrote: 
> >You were absolutaly right, I ment buffering of incoming RTP packet data.  
> >There is a function 
> >ds_free_packs() which clears all demux_packet_t structures found in a  
> >stream. But it does not seem to 
> >be working. After resume is pressed, MPlayer still plays a very short  
> >period of time an old part of 
> >movie, before it goes to a new position. Where should I look for freeing  
> >1(internal) kind of buffers, 
> >internal to the OS. 
>  
> I don't know how (or even if) you can flush the OS's UDP socket  
> buffer.  But what you can do, though, is continue to read packets, as  
> usual, but discard them (using "free_demux_packet()") if their 
presentation  
> timestamp is less than the value you want.  (See "demux_rtp.cpp" for an  
> example of how I already do this (for A/V synchronization) if the audio  
> stream gets too far behind the video stream, or vice versa.  Look for  
> "ptsBehindThreshold".) 
>  
>          Ross. 
>  
>  
>  
>  
>  
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X-Original-Date: Tue, 25 Mar 2003 17:10:21 +0530
Date: Tue, 25 Mar 2003 17:10:21 +0530


Hi ,

     Thank u very much for ur reply. I have done all necessary changes
according to ur "How to configure and build the code on Windows"
documentation.

 while bulding the make file in Windows NT VC++ environment , I am getting
the error
"Microsoft (R) Program Maintenance Utility   Version 6.00.8168.0
Copyright (C) Microsoft Corp 1988-1998. All rights reserved.
NMAKE : fatal error U1073: don't know how to make
'include/UsageEnvironment_version.hh'
Stop.
Error executing NMAKE.

UsageEnvironment1.exe - 1 error(s), 0 warning(s)"

for all files. not only for UsageEnvironment make file...

what exactly i ahve done  is:

1. Unpack it on Linux using "-zxvf"

2. My VC++ is under D folder. so that  modified the win32congi file TOOLS32
= d:\Program Files\Microsoft Visual Studio\Common(which contain the Tools
directory)

"but my bin\cl directory is under "d:\Program Files\Microsoft Visual
Studio\Vc98"  directory..
I changed the TOOLS32  = "d:\Program Files\Microsoft Visual Studio\Vc98"
and tested the application in this case also" getting the same problem "

3. I have copied these sub directories  in a  Windows NT machine.

4. Using VC++ open Workspace, I have opened make file for build.

5. while building the make file , getting the  above error ....

what is the problem exactly? Pls any body can help me in this issuus..... I
am very thankful  to u...




    - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
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                    Ross Finlayson                                                                                                
                    <finlayson(at)live.com>                To:     live-devel(at)lists.sourceforge.net                                  
                    Sent by:                            cc:                                                                       
                    live-devel-admin(at)lists.sourc        Subject:     Re: [Live-devel] help (MP3 Streaming Media)                  
                    eforge.net                                                                                                    
                                                                                                                                  
                                                                                                                                  
                    03/25/03 06:38 AM                                                                                             
                    Please respond to live-devel                                                                                  
                                                                                                                                  
                                                                                                                                  





>     Thanks for ur  prompt reply..i had gone thru it...still i couldn't
>compile it correctly on Windows...
>      Pls give me some more details step by step clearly...
>      my doubt is "do i need to unpack it on WIndows or Linux.
>      what i did actually is " i have done unzipped on Windows..
>      and next went to live directory and typed "./genWindowsMakefiles"
>....its is giving the error that
>      "the name specified is not recognised as an internal or external
>command, operable program
>      or batch file"

Sigh...  As the web site explains, "genWindowsMakefiles" is a *Unix*
command - i.e., one that you run on a Unix machine, to generate ".mak"
files that you would then copy to a Windows machine.

If, for some reason, Windows computers are all that you have, see the
following paragraph, titled "If you don't have access to a Unix machine"

         Ross.



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Subject: Re: [Live-devel] MPlayer statistics
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X-Original-Date: Tue, 25 Mar 2003 08:32:22 -0800
Date: Tue, 25 Mar 2003 08:32:22 -0800

Ross,
Thanks.

----- Original Message -----
From: {Ross Finlayson} <EMAIL-ADDRESS-HIDDEN>
To: <EMAIL-ADDRESS-HIDDEN>
Sent: Monday, March 24, 2003 7:07 PM
Subject: Re: [Live-devel] MPlayer statistics


>
> >Just one more clarification. Suppose the library waits for 100msec and
the
> >packet does not arrive within that time; the packet then arrives later.
Then
> >the library will drop the packet, right?
>
> Yes.
>
> >  In that case, will this packet be
> >displayed by RTPSource::curPacketRTPSeqNum() (when this function is
called
> >from "MPEGVideoRTPSource::processSpecialHeader()" ).
>
> No.
>
> >  In other words, are the
> >packets which are received in
"MPEGVideoRTPSource::processSpecialHeader()"
> >only those which are actually going to be played by MPlayer?
>
> Yes.
>
>          Ross.
>
>
>
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X-Original-Date: Tue, 25 Mar 2003 13:44:10 -0800
Date: Tue, 25 Mar 2003 13:44:10 -0800

Ross,

Thanks for all the help. Can you please answer some questions about the data
buffering in Live/MPlayer. When playing streamed video, does the Live
library buffer data or does the MPlayer do that? Also, how much data (in
kbits or seconds) is buffered? I have noticed that when I stream data from a
server to the Mplayer, the server may have already reached the end of the
stream, but the MPlayer finishes playing around 3-4 seconds after that.

Thanks,
Rohit



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X-Original-Date: Tue, 25 Mar 2003 17:09:06 -0800
Date: Tue, 25 Mar 2003 17:09:06 -0800


>Thanks for all the help. Can you please answer some questions about the data
>buffering in Live/MPlayer. When playing streamed video, does the Live
>library buffer data or does the MPlayer do that? Also, how much data (in
>kbits or seconds) is buffered?

Most of the buffering is done by the underlying OS (it's regular buffering 
of incoming UDP socket data).  For MPlayer, the size of this buffer is set 
to 2000000 bytes for video, and 100000 bytes for audio.  (See 
"demux_rtp.cpp" for details.)

RYHCSC ("Remember, you have complete source code")

         Ross.



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Subject: Re: [Live-devel] RTP Buffer
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X-Original-Date: Tue, 25 Mar 2003 17:16:21 -0800
Date: Tue, 25 Mar 2003 17:16:21 -0800


>Why blocking function getBuffer() stops responding?

Because it's a blocking function :-)

Note that "getBuffer()" never returns NULL (unless an error occurs), and 
will wait, if necessary, until packet data arrives.  In your case, it's 
blocking because (presumably) you've just asked the server to pause, so no 
more RTP packets are arriving.

What you should be doing instead is calling getBuffer() in a loop after you 
have *ended* the pause (i.e., once RTP packets have begin flowing 
again).  Then, discard any packets whose presentation time is too low.

         Ross.



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X-Original-Date: Tue, 25 Mar 2003 22:57:48 -0500
Date: Tue, 25 Mar 2003 22:57:48 -0500

Well, here are some answers.

Except for two mpeg-1 streams, the streams are all mpeg-2. Evidently, mpeg2
cannot be done with rtp under vbrick.

"The mpeg-2 vbricks we're using don't do RTP."

But here are some furhter comments on the raw vs rtp story:

"Though RTP is useful, I do doubt its necessity with mpeg streams like
ours.  Both video and audio are present in the same (native format)
stream, so there's no synchronizing necessary between streams.  Also, an
mpeg stream has its own time clock references buried in it, so the one
that RTP provides is somewhat redundant

"Like I said, a lot of mpeg streaming sources do raw formats (they tend
to be from the broadcasting world vs. IP companies).  I don't know if
one could find "dozens" of pieces of software that can handle mpeg-2
transport streams in any format, let alone RTP ones."

As it turns out, quicktime, w/o a paid plugin from Apple, does not do mpeg 2 at
all.

Anyway, I am not sure I agree with the assesment here that lots of software out
there does RAW. So far, the only piece of software that seems to like these
streams is the VideoLan Client.

As an added note, I finally compild myself, today, the latest mplayer and
live.com cvses. Nice!

On Mon, Mar 24, 2003 at 06:14:13PM -0800, Ross Finlayson wrote:
> At 05:58 PM 3/24/03, you wrote:
> >Good question on why we would want to do RAW. Evidently it is something that
> >the VBrick streaming servers do.
> 
> I'm pretty sure that the VBrick servers can be configured to stream using 
> RTP instead.
> 
> >Re the packetsize, I am referring to:
> >a=packetsize:4136
> >(which, I am told, is the VBrick server default).
> 
> The LIVE.COM library code should be able to receive and handle RTP packets 
> of any size (unless they're ridiculously large - like 50 kBytes).  It 
> doesn't recognize any "a=packetsize" line in SDP descriptions, but that's 
> just a 'hint' for receivers anyway.  The library code should handle 
> incoming RTP packets of this size nonetheless.
> 
>          Ross.
> 
> 
> 
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X-Original-Date: Wed, 26 Mar 2003 03:06:31 -0500
Date: Wed, 26 Mar 2003 03:06:31 -0500

Its a great clue, but it still does not work. I call now my loop now after 
Resume request is sent to Server. But after I call getBuffer, the value I 
get received to ptsBehind is always 0.0 so I cannot compare the packets and 
differe them to discard!? 
 
 
----- Original Message ----- 
From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
Date: Tue, 25 Mar 2003 17:16:21 -0800  
To: <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] RTP Buffer 
 
>  
> >Why blocking function getBuffer() stops responding? 
>  
> Because it's a blocking function :-) 
>  
> Note that "getBuffer()" never returns NULL (unless an error occurs), and  
> will wait, if necessary, until packet data arrives.  In your case, it's  
> blocking because (presumably) you've just asked the server to pause, so 
no  
> more RTP packets are arriving. 
>  
> What you should be doing instead is calling getBuffer() in a loop after 
you  
> have *ended* the pause (i.e., once RTP packets have begin flowing  
> again).  Then, discard any packets whose presentation time is too low. 
>  
>          Ross. 
>  
>  
>  
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Subject: [Live-devel] Size of Movie?
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X-Original-Date: Wed, 26 Mar 2003 04:02:29 -0500
Date: Wed, 26 Mar 2003 04:02:29 -0500

How can I get the size of the Movie? I meen how long is  
the RTSP MPEG4 Movie? Is there any ready function in 
LIVE.COM library or at least any way to get it from RTSP 
source? 
 
Thanks 
Elnar 
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Subject: [Live-devel] H.263 Help
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X-Original-Date: Wed, 26 Mar 2003 08:03:27 -0800 (PST)
Date: Wed, 26 Mar 2003 08:03:27 -0800 (PST)

I have a H.263 encoder which gives me encoded frame
buffer. And I want use your rtp library in order to
make a video conferencing application. I see
H263plusrtpsink/source files, and I understand how to
use them. My problem is I could not find a method to
give the buffer to the rtp library. I read all the
mailing list about this subject and nearly all the
livemedia source code. But I could not find them
enough helpful. I releasize that i must write
something like MPEGVideoStreamFramer class. And I must
concantrate on doGetNextFrame functions. I also assume
that fTo is the buffer that i need. That mpeg class is
very confusing for me. I could not understant how to
start to write the necessary code for my project. If
it is possible i need a step by step explanation how
to write H263VideoStreamFramer class. Thank you for
your reading and help.

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X-Original-Date: Wed, 26 Mar 2003 10:12:21 -0800
Date: Wed, 26 Mar 2003 10:12:21 -0800

At 08:03 AM 3/26/03, you wrote:
>I have a H.263 encoder which gives me encoded frame
>buffer. And I want use your rtp library in order to
>make a video conferencing application. I see
>H263plusrtpsink/source files, and I understand how to
>use them. My problem is I could not find a method to
>give the buffer to the rtp library.

You should be able to do this quite easily, by writing a "FramedSource" 
subclass that encapsulates your encoder, and delivers H.263 frames to 
"H263plusRTPSink".

For a model of how to do this, see "liveMedia/DeviceSource.cpp" (and 
"liveMedia/include/DeviceSource.hh").  You will need to fill in parts of 
this code to do the actual read from your encoder.

If you do this, then the rest of your code becomes very simple: You can 
just feed this FramedSource (subclass) directly into "H263plusRTPSink".

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>



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Please unsubscribe me. Tmoss(at)enerdyne.com

		-----Original Message-----
		From:	live-devel-request(at)lists.sourceforge.net
[mailto:live-devel-request(at)lists.sourceforge.net]
		Sent:	Wednesday, March 26, 2003 12:02 PM
		To:	live-devel(at)lists.sourceforge.net
		Subject:	Live-devel digest, Vol 1 #360 - 8 msgs

		Send Live-devel mailing list submissions to
			live-devel(at)lists.sourceforge.net

		To subscribe or unsubscribe via the World Wide Web, visit
	
https://lists.sourceforge.net/lists/listinfo/live-devel
		or, via email, send a message with subject or body 'help' to
			live-devel-request(at)lists.sourceforge.net

		You can reach the person managing the list at
			live-devel-admin(at)lists.sourceforge.net

		When replying, please edit your Subject line so it is more
specific
		than "Re: Contents of Live-devel digest..."


		Today's Topics:

		   1. MPlayer Buffer (Rohit Kapoor)
		   2. Re: MPlayer Buffer (Ross Finlayson)
		   3. Re: RTP Buffer (Ross Finlayson)
		   4. Re: compiling with mplayer fails (David Botsch)
		   5. Re: RTP Buffer (Elnar Hajiev)
		   6. Size of Movie? (Elnar Hajiev)
		   7. H.263 Help (Çaðatay Tunalý)
		   8. Re: H.263 Help (Ross Finlayson)

		--__--__--

		Message: 1
		From: "Rohit Kapoor" <rohitk(at)CS.UCLA.EDU>
		To: <live-devel(at)lists.sourceforge.net>
		Date: Tue, 25 Mar 2003 13:44:10 -0800
		Subject: [Live-devel] MPlayer Buffer
		Reply-To: live-devel(at)lists.sourceforge.net

		Ross,

		Thanks for all the help. Can you please answer some
questions about the data
		buffering in Live/MPlayer. When playing streamed video, does
the Live
		library buffer data or does the MPlayer do that? Also, how
much data (in
		kbits or seconds) is buffered? I have noticed that when I
stream data from a
		server to the Mplayer, the server may have already reached
the end of the
		stream, but the MPlayer finishes playing around 3-4 seconds
after that.

		Thanks,
		Rohit



		--__--__--

		Message: 2
		Date: Tue, 25 Mar 2003 17:09:06 -0800
		To: live-devel(at)lists.sourceforge.net
		From: Ross Finlayson <finlayson(at)live.com>
		Subject: Re: [Live-devel] MPlayer Buffer
		Reply-To: live-devel(at)lists.sourceforge.net


		>Thanks for all the help. Can you please answer some
questions about the data
		>buffering in Live/MPlayer. When playing streamed video,
does the Live
		>library buffer data or does the MPlayer do that? Also, how
much data (in
		>kbits or seconds) is buffered?

		Most of the buffering is done by the underlying OS (it's
regular buffering 
		of incoming UDP socket data).  For MPlayer, the size of this
buffer is set 
		to 2000000 bytes for video, and 100000 bytes for audio.
(See 
		"demux_rtp.cpp" for details.)

		RYHCSC ("Remember, you have complete source code")

		         Ross.



		--__--__--

		Message: 3
		Date: Tue, 25 Mar 2003 17:16:21 -0800
		To: live-devel(at)lists.sourceforge.net
		From: Ross Finlayson <finlayson(at)live.com>
		Subject: Re: [Live-devel] RTP Buffer
		Reply-To: live-devel(at)lists.sourceforge.net


		>Why blocking function getBuffer() stops responding?

		Because it's a blocking function :-)

		Note that "getBuffer()" never returns NULL (unless an error
occurs), and 
		will wait, if necessary, until packet data arrives.  In your
case, it's 
		blocking because (presumably) you've just asked the server
to pause, so no 
		more RTP packets are arriving.

		What you should be doing instead is calling getBuffer() in a
loop after you 
		have *ended* the pause (i.e., once RTP packets have begin
flowing 
		again).  Then, discard any packets whose presentation time
is too low.

		         Ross.



		--__--__--

		Message: 4
		Date: Tue, 25 Mar 2003 22:57:48 -0500
		From: David Botsch <dwb7(at)ccmr.cornell.edu>
		To: live-devel(at)lists.sourceforge.net
		Subject: Re: [Live-devel] compiling with mplayer fails
		Reply-To: live-devel(at)lists.sourceforge.net

		Well, here are some answers.

		Except for two mpeg-1 streams, the streams are all mpeg-2.
Evidently, mpeg2
		cannot be done with rtp under vbrick.

		"The mpeg-2 vbricks we're using don't do RTP."

		But here are some furhter comments on the raw vs rtp story:

		"Though RTP is useful, I do doubt its necessity with mpeg
streams like
		ours.  Both video and audio are present in the same (native
format)
		stream, so there's no synchronizing necessary between
streams.  Also, an
		mpeg stream has its own time clock references buried in it,
so the one
		that RTP provides is somewhat redundant

		"Like I said, a lot of mpeg streaming sources do raw formats
(they tend
		to be from the broadcasting world vs. IP companies).  I
don't know if
		one could find "dozens" of pieces of software that can
handle mpeg-2
		transport streams in any format, let alone RTP ones."

		As it turns out, quicktime, w/o a paid plugin from Apple,
does not do mpeg 2 at
		all.

		Anyway, I am not sure I agree with the assesment here that
lots of software out
		there does RAW. So far, the only piece of software that
seems to like these
		streams is the VideoLan Client.

		As an added note, I finally compild myself, today, the
latest mplayer and
		live.com cvses. Nice!

		On Mon, Mar 24, 2003 at 06:14:13PM -0800, Ross Finlayson
wrote:
		> At 05:58 PM 3/24/03, you wrote:
		> >Good question on why we would want to do RAW. Evidently
it is something that
		> >the VBrick streaming servers do.
		> 
		> I'm pretty sure that the VBrick servers can be configured
to stream using 
		> RTP instead.
		> 
		> >Re the packetsize, I am referring to:
		> >a=packetsize:4136
		> >(which, I am told, is the VBrick server default).
		> 
		> The LIVE.COM library code should be able to receive and
handle RTP packets 
		> of any size (unless they're ridiculously large - like 50
kBytes).  It 
		> doesn't recognize any "a=packetsize" line in SDP
descriptions, but that's 
		> just a 'hint' for receivers anyway.  The library code
should handle 
		> incoming RTP packets of this size nonetheless.
		> 
		>          Ross.
		> 
		> 
		> 
		> -------------------------------------------------------
		> This SF.net email is sponsored by:
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		-- 
		********************************
		David William Botsch
		Consultant/Advisor II
		CCMR Computing Facility
		dwb7(at)ccmr.cornell.edu
		********************************


		--__--__--

		Message: 5
		From: "Elnar Hajiev" <elik(at)asia.com>
		To: live-devel(at)lists.sourceforge.net
		Date: Wed, 26 Mar 2003 03:06:31 -0500
		Subject: Re: [Live-devel] RTP Buffer
		Reply-To: live-devel(at)lists.sourceforge.net

		Its a great clue, but it still does not work. I call now my
loop now after 
		Resume request is sent to Server. But after I call
getBuffer, the value I 
		get received to ptsBehind is always 0.0 so I cannot compare
the packets and 
		differe them to discard!? 
		 
		 
		----- Original Message ----- 
		From: Ross Finlayson <finlayson(at)live.com> 
		Date: Tue, 25 Mar 2003 17:16:21 -0800  
		To: live-devel(at)lists.sourceforge.net 
		Subject: Re: [Live-devel] RTP Buffer 
		 
		>  
		> >Why blocking function getBuffer() stops responding? 
		>  
		> Because it's a blocking function :-) 
		>  
		> Note that "getBuffer()" never returns NULL (unless an
error occurs), and  
		> will wait, if necessary, until packet data arrives.  In
your case, it's  
		> blocking because (presumably) you've just asked the server
to pause, so 
		no  
		> more RTP packets are arriving. 
		>  
		> What you should be doing instead is calling getBuffer() in
a loop after 
		you  
		> have *ended* the pause (i.e., once RTP packets have begin
flowing  
		> again).  Then, discard any packets whose presentation time
is too low. 
		>  
		>          Ross. 
		>  
		>  
		>  
		> ------------------------------------------------------- 
		> This SF.net email is sponsored by: 
		> The Definitive IT and Networking Event. Be There! 
		> NetWorld+Interop Las Vegas 2003 -- Register today! 
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		--__--__--

		Message: 6
		From: "Elnar Hajiev" <elik(at)asia.com>
		To: live-devel(at)lists.sourceforge.net
		Date: Wed, 26 Mar 2003 04:02:29 -0500
		Subject: [Live-devel] Size of Movie?
		Reply-To: live-devel(at)lists.sourceforge.net

		How can I get the size of the Movie? I meen how long is  
		the RTSP MPEG4 Movie? Is there any ready function in 
		LIVE.COM library or at least any way to get it from RTSP 
		source? 
		 
		Thanks 
		Elnar 
		-- 
		__________________________________________________________
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		--__--__--

		Message: 7
		Date: Wed, 26 Mar 2003 08:03:27 -0800 (PST)
		From: "Çaðatay" "Tunalý" <the_cagatay(at)yahoo.com>
		To: live-devel(at)lists.sourceforge.net
		Subject: [Live-devel] H.263 Help
		Reply-To: live-devel(at)lists.sourceforge.net

		I have a H.263 encoder which gives me encoded frame
		buffer. And I want use your rtp library in order to
		make a video conferencing application. I see
		H263plusrtpsink/source files, and I understand how to
		use them. My problem is I could not find a method to
		give the buffer to the rtp library. I read all the
		mailing list about this subject and nearly all the
		livemedia source code. But I could not find them
		enough helpful. I releasize that i must write
		something like MPEGVideoStreamFramer class. And I must
		concantrate on doGetNextFrame functions. I also assume
		that fTo is the buffer that i need. That mpeg class is
		very confusing for me. I could not understant how to
		start to write the necessary code for my project. If
		it is possible i need a step by step explanation how
		to write H263VideoStreamFramer class. Thank you for
		your reading and help.

		__________________________________________________
		Do you Yahoo!?
		Yahoo! Platinum - Watch CBS' NCAA March Madness, live on
your desktop!
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		--__--__--

		Message: 8
		Date: Wed, 26 Mar 2003 10:12:21 -0800
		To: live-devel(at)lists.sourceforge.net
		From: Ross Finlayson <finlayson(at)live.com>
		Subject: Re: [Live-devel] H.263 Help
		Reply-To: live-devel(at)lists.sourceforge.net

		At 08:03 AM 3/26/03, you wrote:
		>I have a H.263 encoder which gives me encoded frame
		>buffer. And I want use your rtp library in order to
		>make a video conferencing application. I see
		>H263plusrtpsink/source files, and I understand how to
		>use them. My problem is I could not find a method to
		>give the buffer to the rtp library.

		You should be able to do this quite easily, by writing a
"FramedSource" 
		subclass that encapsulates your encoder, and delivers H.263
frames to 
		"H263plusRTPSink".

		For a model of how to do this, see
"liveMedia/DeviceSource.cpp" (and 
		"liveMedia/include/DeviceSource.hh").  You will need to fill
in parts of 
		this code to do the actual read from your encoder.

		If you do this, then the rest of your code becomes very
simple: You can 
		just feed this FramedSource (subclass) directly into
"H263plusRTPSink".

		         Ross Finlayson
		         LIVE.COM
		         <http://www.live.com/>




		--__--__--

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		End of Live-devel Digest


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X-Original-Date: Wed, 26 Mar 2003 16:27:26 -0800
Date: Wed, 26 Mar 2003 16:27:26 -0800

At 04:15 PM 3/26/03, you wrote:
>Please unsubscribe me.

NOTE!!

>                 To subscribe or unsubscribe via the World Wide Web, visit
>
>https://lists.sourceforge.net/lists/listinfo/live-devel
>                 or, via email, send a message with subject or body 'help' to
>                         live-devel-request(at)lists.sourceforge.net

"Unsubscribe" messages should *never* be sent directly to a mailing list.

         Ross.



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X-Original-Date: Thu, 27 Mar 2003 00:06:28 -0800 (PST)
Date: Thu, 27 Mar 2003 00:06:28 -0800 (PST)

Thank you for your reply :)

I implemented device.cpp and h. files according to my
encoder output. Now I want to write a simple program
which streams h.263. When I look testMPEGVideoStream
program i see that play() is the key function. but i
do not understant how to use my clas in this function.
you use 2 key classes. one is ByteStreamFileSource the
other is MPEGVideoStreamFramer. myclass looks like 
ByteStreamFileSource. so i put my class and get off
the ByteStreamFileSource class. Could you tell me what
i am going to use for MPEGVideoStreamFramer.

And as a second question i do not understand the usage
of afterplaying 

videoSink->startPlaying(*videoSource, afterPlaying,
videoSink);

is this type of declaration is possible in cpp. Are
you giving the address of the function
afterPlaying()to the videoSink->startPlaying(....)

Thank you for your help.....:)

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From <EMAIL-ADDRESS-HIDDEN> Thu Mar 27 00:56:44 2003
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X-Original-Date: Thu, 27 Mar 2003 00:49:57 -0800
Date: Thu, 27 Mar 2003 00:49:57 -0800

At 12:06 AM 3/27/03, you wrote:
>Thank you for your reply :)
>
>I implemented device.cpp and h. files according to my
>encoder output. Now I want to write a simple program
>which streams h.263. When I look testMPEGVideoStream
>program i see that play() is the key function. but i
>do not understant how to use my clas in this function.

If you use "testMPEGVideoStreamer" as a model, then the "play()" function 
would look something like this:

void play() {
   // Create the input source:
   DeviceSource* encoder
     = DeviceSource::createNew(*env, <your-encoder-params>);

   // Start playing:
   videoSink->startPlaying(*encoder, afterPlaying, videoSink);
}

Of course, "videoSink" should be a "H263plusVideoRTPSink*", not a 
"MPEGVideoRTPSink*".

>you use 2 key classes. one is ByteStreamFileSource the
>other is MPEGVideoStreamFramer. myclass looks like
>ByteStreamFileSource. so i put my class and get off
>the ByteStreamFileSource class. Could you tell me what
>i am going to use for MPEGVideoStreamFramer.

In your case, you won't need a 'framer' filter, because your "DeviceSource" 
encoder class is already outputting complete H.263 frames - not just an 
unstructured byte stream.

>And as a second question i do not understand the usage
>of afterplaying
>
>videoSink->startPlaying(*videoSource, afterPlaying,
>videoSink);
>
>is this type of declaration is possible in cpp. Are
>you giving the address of the function
>afterPlaying()to the videoSink->startPlaying(....)

I'm passing the function name directly, but that has the same effect as 
passing a pointer to the function.

(FYI, the "afterPlaying" function is called when the input source closes.)

         Ross.



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Subject: [Live-devel] rtp library and vlan
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X-Original-Date: Thu, 27 Mar 2003 17:10:31 +0800
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Hi! Do you know the VLAN project?  It is a video streaming server-client
application supporting different kinds of audio and video formats.  Has the
live rtp library been tested on it?
Thanks

mildred


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<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><SPAN class=3D335210809-27032003><FONT face=3DArial size=3D2>Hi! Do =
you know the=20
VLAN project?&nbsp; It is a video streaming server-client application =
supporting=20
different kinds of audio and video formats.&nbsp; Has the live rtp =
library been=20
tested on it?&nbsp; </FONT></SPAN></DIV>
<DIV><SPAN class=3D335210809-27032003><FONT face=3DArial=20
size=3D2>Thanks</FONT></SPAN></DIV>
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<DIV><SPAN class=3D335210809-27032003><FONT face=3DArial=20
size=3D2>mildred</FONT></SPAN></DIV>
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X-Original-Date: Thu, 27 Mar 2003 01:08:18 -0800
Date: Thu, 27 Mar 2003 01:08:18 -0800

At 01:10 AM 3/27/03, you wrote:
>
>Hi! Do you know the VLAN project?  It is a video streaming server-client 
>application supporting different kinds of audio and video formats.  Has 
>the live rtp library been tested on it?

I don't know anything about "VLAN" (what's the URL?), but the "MPlayer" 
media player now uses the LIVE.COM libraries.  (See 
<http://www.live.com/mplayer/> for information.)

         Ross.



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X-Original-Date: Fri, 28 Mar 2003 09:42:51 +0800
Date: Fri, 28 Mar 2003 09:42:51 +0800


	VLAN - http://www.videolan.org


-----Original Message-----
From: <EMAIL-ADDRESS-HIDDEN>
[mailto:live-devel-admin(at)lists.sourceforge.net]On Behalf Of Ross
Finlayson
Sent: Thursday, March 27, 2003 5:08 PM
To: <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] rtp library and vlan


At 01:10 AM 3/27/03, you wrote:
>
>Hi! Do you know the VLAN project?  It is a video streaming server-client 
>application supporting different kinds of audio and video formats.  Has 
>the live rtp library been tested on it?

I don't know anything about "VLAN" (what's the URL?), but the "MPlayer" 
media player now uses the LIVE.COM libraries.  (See 
<http://www.live.com/mplayer/> for information.)

         Ross.



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X-Original-Date: Thu, 27 Mar 2003 23:53:58 -0800
Date: Thu, 27 Mar 2003 23:53:58 -0800

At 05:42 PM 3/27/03, you wrote:
>         VLAN - http://www.videolan.org

I looked at this a few months ago.  Unfortunately, it's not well-suited for 
playing RTP streams, because its architecture assumes that there is just a 
single, multiplexed input source for both audio and video.  So it couldn't 
easily be modified to play general RTP sessions (in which there are 
separate input packet streams for audio and video).

Instead, I recommend using "MPlayer", which can already play RTP streams 
(using the LIVE.COM libraries).  See <http://www.live.com/mplayer/> for 
details.

         Ross.





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From <EMAIL-ADDRESS-HIDDEN> Fri Mar 28 06:00:10 2003
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Subject: RE: [Live-devel] rtp library and vlan
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X-Original-Date: Fri, 28 Mar 2003 08:57:06 -0500
Date: Fri, 28 Mar 2003 08:57:06 -0500

Ross,

I'm not familiar with VLAN project but it is possible to have multiplexed
media streams over RTP. Take a look at RFC 2343 that describes how to
transmit bundled MPEG over RTP - all synchronization information is now
carried as part of the RTP packet in the MPEG headers, duplicating some of
what is already in RTCP.

Thanks,
nagarjuna

-----Original Message-----
From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
Sent: Friday, March 28, 2003 2:54 AM
To: <EMAIL-ADDRESS-HIDDEN>
Subject: RE: [Live-devel] rtp library and vlan


At 05:42 PM 3/27/03, you wrote:
>         VLAN - http://www.videolan.org

I looked at this a few months ago.  Unfortunately, it's not well-suited for 
playing RTP streams, because its architecture assumes that there is just a 
single, multiplexed input source for both audio and video.  So it couldn't 
easily be modified to play general RTP sessions (in which there are 
separate input packet streams for audio and video).

Instead, I recommend using "MPlayer", which can already play RTP streams 
(using the LIVE.COM libraries).  See <http://www.live.com/mplayer/> for 
details.

         Ross.





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From <EMAIL-ADDRESS-HIDDEN> Fri Mar 28 13:21:37 2003
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X-Original-Date: Fri, 28 Mar 2003 13:13:55 -0800
Date: Fri, 28 Mar 2003 13:13:55 -0800


>I'm not familiar with VLAN project but it is possible to have multiplexed
>media streams over RTP. Take a look at RFC 2343 that describes how to
>transmit bundled MPEG over RTP

Yes, it is certainly 'possible' to have RTP sessions that multiplex 
separate media streams, but most RTP sessions are not like this.  In 
particular, I have yet to see a server that actually implements the payload 
format described in RFC 2343.  However, the LIVE.COM libraries (and, thus, 
MPlayer) could easily be updated to support such a stream, if 
necessary.  (A somewhat more common example of multiplexed media streams is 
the "MPEG transport stream" RTP payload format described in RFC 2250; we 
support this already.)

In contrast, the vast majority of RTP sessions - that use separate packet 
streams for each media - cannot be received/played by "videolan", because 
it makes the fatal assumption that there is just a single input 
stream.  Moral of the story, once again: Forget about "videolan", and use 
"MPlayer" <http://www.live.com/mplayer/>; it's superior software.

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>



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X-Original-Date: Tue, 1 Apr 2003 04:36:58 -0800 (PST)
Date: Tue, 1 Apr 2003 04:36:58 -0800 (PST)

Hi,

I am writing an application that will do
teleconferencing like netmeeting. 
I am grabbing live video and encoding it. Now I want
to send this encoded video 
via rtp. I want use livemedia rtp library (thx a lot
:))

First of all I write a console applicaton which will
read H263 file and send
them like testMPEGVideoStreamer application. It works
fine.

Now I want use the rtp library in mfc application. I
have two problems. 

First is, In my console application, in main function,
I call play() function 
and env->taskScheduler().doEventLoop(). Nothing more.
Now in my mfc program 
I do not want to read H263 data from file instead I
will get this data live. 
I have a callback function which works when it
captures frame from the webcam
(i.e If I capture 30 frames per second, it is called
30 times in one second).

I have a winthread class which acts like a main
function in the console applicaton. 
It initializes variables and calls play () and
env->taskScheduler().doEventLoop(). 
Then I use DeviceSource as a base class and use
doGetNextFrame function like
doGetNextFrame (H263 buffer) (fTo = buffer,
fpresentationTime ....). 
This function is called in callbackfunction. Is this
rigth .
Because I think doGetNextFrame is called from
FramedSource class also. If this is not
right what is the solution. I must find a way to give
the encoded data buffer to rtp
library. 

second problem is, in the class that acts a main
function has two member functions

videoSink = H263plusVideoRTPSink::createNew(*env,
&rtpGroupsock, 96);	

void MyClass::afterPlaying()
{
	play();
}

void MyClass::play()
{
	mySource = DeviceSource::createNew(*env,
deviceParams);
	videoSink->startPlaying(*mySource, afterPlaying,
videoSink);
}

in .h file
	void play();
	void afterPlaying();

When I try to compile this I get

error C2664: 'unsigned int __thiscall
MediaSink::startPlaying(class MediaSource &,void
(__cdecl *)(void *),void *)' : 
cannot convert parameter 2 from 'void (void)' to 'void
(__cdecl *)(void *)'
        None of the functions with this name in scope
match the target type

You have a declaration   typedef void
(afterPlayingFunc)(void* clientData) In MediaSink
file.
But I think this is not suitable for class's member
functions. And what will be the solution of 
this problem.

Thank you for your answers....:)

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Subject: [Live-devel] Using SDP Files with MPlayer
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X-Original-Date: Tue, 1 Apr 2003 08:14:37 -0500 (EST)
Date: Tue, 1 Apr 2003 08:14:37 -0500 (EST)

I've been trying to get mplayer to work the the "-sdp" command line option 
on a Solaris 8 client.  I'm using live.2003.03.25 and MPlayer-0.90rc5 (I 
also tried MPlayer-20030325).  All builds completed successfully with the 
GCC 2.95.3 compiler.  I followed the build instructions on 
http://www.live.com/mplayer/.  I figured that I should be able to do the 
following as a multicast streaming test.

% cp <somempegfile>.mpeg ./test.mpg
% ./testMPEGAudioVideoStreamer

% ./testMPEGVideoReceiver > foo.mpg

That worked.  Also, "./testMPEGVideoReceiver | mplayer -" works fine too.

However, the following mplayer command doesn't work:

% ./mplayer -sdp testMPEGAudioVideo.sdp

It sits forever at the cache fill output line until I interrupt it.  For 
example:

----------------------------------------------------------------------
$ mplayer -sdp testMPEGAudioVideo.sdp
Using GNU internationalization
Original domain: messages
Original dirname: /usr/lib/locale/
Current domain: mplayer
Current dirname: /home/caron/sw28/share/locale


MPlayer 0.90rc5-2.95.3 (C) 2000-2003 Arpad Gereoffy (see DOCS)

Reading config file /home/caron/sw28/etc/mplayer/mplayer.conf
Reading config file /home/caron/.mplayer/config
Reading /home/caron/.mplayer/codecs.conf: 50 audio & 136 video codecs
font: can't open file: /home/caron/.mplayer/font/font.desc
Font /home/caron/sw28/share/mplayer/font/font.desc loaded successfully! 
(206 chars)
Using usleep() timing
Input config file /home/caron/.mplayer/input.conf parsed : 52 binds

Playing testMPEGAudioVideo.sdp
Cache fill:  0.00% (0 bytes)    


MPlayer interrupted by signal 2 in module: enable_cache
----------------------------------------------------------------------

No luck with the "mplayer -nocache -sdp testMPEGAudioVideo.sdp" either.  
Eliminating audio and/or video doesn't help.  Again, it sits there until I 
hit <CTRL>-C.

----------------------------------------------------------------------
$ mplayer -nocache -sdp testMPEGAudioVideo.sdp
Using GNU internationalization
Original domain: messages
Original dirname: /usr/lib/locale/
Current domain: mplayer
Current dirname: /home/caron/sw28/share/locale


MPlayer 0.90rc5-2.95.3 (C) 2000-2003 Arpad Gereoffy (see DOCS)

Reading config file /home/caron/sw28/etc/mplayer/mplayer.conf
Reading config file /home/caron/.mplayer/config
Reading /home/caron/.mplayer/codecs.conf: 50 audio & 136 video codecs
font: can't open file: /home/caron/.mplayer/font/font.desc
Font /home/caron/sw28/share/mplayer/font/font.desc loaded successfully! 
(206 chars)
Using usleep() timing
Input config file /home/caron/.mplayer/input.conf parsed : 52 binds

Playing testMPEGAudioVideo.sdp


MPlayer interrupted by signal 2 in module: demux_open
----------------------------------------------------------------------

I've read through the docs, FAQs, and mailing list but haven't been able to 
find a solution yet.  I ran a debugger on the code and found that the 
hangup is happening at the following section of mplayer.c (Line 1248 for 
MPlayer-0.90rc5).

----------------------------------------------------------------------
if(stream_cache_size>0){
  current_module="enable_cache";
  
if(!stream_enable_cache(stream,stream_cache_size*1024,stream_cache_size*102
4/5,stream_cache_size*1024/20))
    if((eof = libmpdemux_was_interrupted(PT_NEXT_ENTRY))) goto 
goto_next_file;
}
----------------------------------------------------------------------

Any ideas on what I'm doing wrong here?

My biggest point of confusion is that I can't seem to find the section of 
the MPlayer code that initializes the multicast stream.  I used the code in 
testMPEGVideoReceiver.cpp as an example.  Shouldn't there be a call to 
"MPEGVideoRTPSink::createNew" somewhere in the MPlayer code?  I see the 
part where the SDP file is opened and the contents are stuffed into 
stream->priv.  However, I can't find where the SDP file contents are 
actually parsed and used to initiate a multicast session.



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X-Original-Date: Tue, 01 Apr 2003 08:50:42 -0800
Date: Tue, 01 Apr 2003 08:50:42 -0800


>When I try to compile this I get
>
>error C2664: 'unsigned int __thiscall
>MediaSink::startPlaying(class MediaSource &,void
>(__cdecl *)(void *),void *)' :
>cannot convert parameter 2 from 'void (void)' to 'void
>(__cdecl *)(void *)'

The problem here is that your "afterPlaying" function has the wrong 
signature.  It should be:
         void afterPlaying(void* clientData)
If you change this, your code should compile.

As for your other problem, I couldn't quite follow your 
description.  However, because you already have a console application that 
works, you should be able to figure out how to turn that into a MFC 
application that works.

RYHCSC ("Remember, you have complete source code")

         Ross.



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X-Original-Date: Tue, 01 Apr 2003 10:45:14 -0800
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>However, the following mplayer command doesn't work:
>
>% ./mplayer -sdp testMPEGAudioVideo.sdp

That's odd.  That *should* be working.

>I've been trying to get mplayer to work the the "-sdp" command line option
>on a Solaris 8 client.  I'm using live.2003.03.25 and MPlayer-0.90rc5 (I
>also tried MPlayer-20030325).

I tested it just now with MPlayer-20030401 (on Linux), and it works 
OK.  (Note: Do *not* use the "rc" version; it's out of date!)

I didn't know that MPlayer was supposed to be portable to Solaris.  (It was 
targeted originally for Linux.)  So, it wouldn't surprise me if there was 
some part of the MPlayer code that's not quite portable to Solaris.

One other thing you could try is enable the built-in RTSP server in 
"testMPEGAudioVideoStreamer", and then run "mplayer" to read a "rtsp://" 
URL (instead of a SDP file).  Uncomment the line
         #define IMPLEMENT_RTSP_SERVER 1
in "testMPEGAudioVideoStreamer.cpp".  Then, rebuild and rerun 
"testMPEGAudioVideoStreamer" as root (so that it can bind to the RTSP port: 
554).

Then, run mplayer as:
         mplayer rtsp://<name-of-computer-running-testMPEGAudioVideoStreamer>/

     Ross.



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Subject: Re: [Live-devel] Using SDP Files with MPlayer
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X-Original-Date: Tue, 01 Apr 2003 10:47:18 -0800
Date: Tue, 01 Apr 2003 10:47:18 -0800

An update to my previous message:

Even better, run mplayer as:
	mplayer -V rtsp://<name-of-computer-running-testMPEGAudioVideoStreamer>/

(The "-V" option will print out more information that might help track down 
the problem.)

	Ross.



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From: Paul Caron <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] Using SDP Files with MPlayer
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X-Original-Date: Tue, 1 Apr 2003 14:52:03 -0500 (EST)
Date: Tue, 1 Apr 2003 14:52:03 -0500 (EST)

Hmmm.....  The rtsp option worked fine.  Still can't get the "-sdp" option 
to work though.  I've attached the verbose printout of the "-sdp" run after 
the succssful rtsp run shown below.

I'm trying to get this to work since we have some multicast audio/video 
feeds where I work.  Unfortunately, the RTSP option is not available.  I 
only have an SDP file.

Is there any other way to tell mplayer to read from video and audio 
multicast streams if you know the addresses and port numbers?  I couldn't 
find any command line options for this in the docs.

-------------------------------------------------------------------------

$ mplayer -V rtsp://lompoc/
Using GNU internationalization
Original domain: messages
Original dirname: /usr/lib/locale/
Current domain: mplayer
Current dirname: /home/caron/sw28/share/locale


MPlayer 0.90rc5-2.95.3 (C) 2000-2003 Arpad Gereoffy (see DOCS)

Reading config file /home/caron/sw28/etc/mplayer/mplayer.conf
Reading config file /home/caron/.mplayer/config
Reading /home/caron/.mplayer/codecs.conf: 50 audio & 136 video codecs
CommandLine: '-V' 'rtsp://lompoc/'
get_path('font/font.desc') -> '/home/caron/.mplayer/font/font.desc'
font: can't open file: /home/caron/.mplayer/font/font.desc
Font /home/caron/sw28/share/mplayer/font/font.desc loaded successfully! 
(206 chars)
Using Unoptimized OnScreenDisplay
Using usleep() timing
get_path('input.conf') -> '/home/caron/.mplayer/input.conf'
Parsing input config file /home/caron/.mplayer/input.conf
Input config file /home/caron/.mplayer/input.conf parsed : 52 binds
get_path('.conf') -> '/home/caron/.mplayer/.conf'

Playing rtsp://lompoc/
Connected to server: lompoc
This stream is non-cacheable
Stream not seekable!
Sending request: DESCRIBE rtsp://lompoc/ RTSP/1.0
CSeq: 1
Accept: application/sdp
User-Agent: MPlayer (LIVE.COM Streaming Media v2003.03.25)


Received DESCRIBE response: RTSP/1.0 200 OK
CSeq: 1
Content-Type: application/sdp
Content-Length: 326


Need to read 326 extra bytes
Read 326 extra bytes: v=0
o=- 1049225941302234 1 IN IP4 141.162.1.101
s=Session streamed by "testMPEGAudioVideoStreamer"
i=LIVE.COM Streaming Media
t=0 0
a=tool:LIVE.COM Streaming Media
a=type:broadcast
m=audio 6666 RTP/AVP 14
a=control:track1
c=IN IP4 239.255.42.42/7
m=video 8888 RTP/AVP 32
a=control:track2
c=IN IP4 239.255.42.42/7

Initiated "audio/MPA" RTP subsession
Sending request: SETUP rtsp://lompoc//track1 RTSP/1.0
CSeq: 2
Transport: RTP/AVP;multicast;client_port=6666-6667
User-Agent: MPlayer (LIVE.COM Streaming Media v2003.03.25)


Received SETUP response: RTSP/1.0 200 OK
CSeq: 2
Transport: RTP/AVP;multicast;destination=239.255.42.42;port=6666;ttl=7
Session: 1


Sending request: PLAY rtsp://lompoc//track1 RTSP/1.0
CSeq: 3
Session: 1
Range: npt=0-
User-Agent: MPlayer (LIVE.COM Streaming Media v2003.03.25)


Received PLAY response: RTSP/1.0 200 OK
CSeq: 3
Session: 1


==> Found audio stream: 0
Initiated "video/MPV" RTP subsession
Sending request: SETUP rtsp://lompoc//track2 RTSP/1.0
CSeq: 4
Transport: RTP/AVP;multicast;client_port=8888-8889
Session: 1
User-Agent: MPlayer (LIVE.COM Streaming Media v2003.03.25)


Received SETUP response: RTSP/1.0 200 OK
CSeq: 4
Transport: RTP/AVP;multicast;destination=239.255.42.42;port=8888;ttl=7
Session: 1


Sending request: PLAY rtsp://lompoc//track2 RTSP/1.0
CSeq: 5
Session: 1
Range: npt=0-
User-Agent: MPlayer (LIVE.COM Streaming Media v2003.03.25)


Received PLAY response: RTSP/1.0 200 OK
CSeq: 5
Session: 1


==> Found video stream: 0
Searching for sequence header... OK!
VIDEO:  MPEG1  176x132  (aspect 12)  29.97 fps   97.6 kbps (12.2 kbyte/s)
[V] filefmt:21  fourcc:0x10000001  size:176x132  fps:29.97  ftime:=0.0334
get_path('sub/') -> '/home/caron/.mplayer/sub/'
==========================================================================
Opening audio decoder: [mp3lib] MPEG layer-2, layer-3
dec_audio: Allocating 4608 + 65536 = 70144 bytes for output buffer
mp3lib: using generic C decore!
MP3lib: init layer2&3 finished, tables done
MPEG 1.0, Layer II, 44100 Hz 112 kbit Stereo, BPF: 366
Channels: 2, copyright: No, original: Yes, CRC: No, emphasis: 0
AUDIO: 44100 Hz, 2 ch, 16 bit (0x20), ratio: 14000->176400 (112.0 kbit)
Selected audio codec: [mp3] afm:mp3lib (mp3lib MPEG layer-2, layer-3)
==========================================================================
X11 opening display: :0.0
vo: X11 color mask:  FFFFFF  (R:FF G:FF00 B:FF0000)
vo: X11 running at 1280x1024 with depth 24 and 32 bpp (":0.0" => local 
display)
[x11] Unknown wm type...
==========================================================================
Opening video decoder: [mpegpes] MPEG 1/2 Video passthrough
VDec: vo config request - 176 x 132 (preferred csp: Mpeg PES)
Trying filter chain: vo
Could not find matching colorspace - retrying with -vop scale...
Opening video filter: [scale]
SwScale params: -1 x -1 (-1=no scaling)
Trying filter chain: scale vo
The selected video_out device is incompatible with this codec.
VDecoder init failed :(
Opening video decoder: [libmpeg2] MPEG 1/2 Video decoder v2.0
libmpeg2: No accelerated IDCT transform found
libmpeg2: No accelerated motion compensation found
VDec: vo config request - 176 x 132 (preferred csp: Planar YV12)
Trying filter chain: vo
VDec: using Planar YV12 as output csp (no 0)
Movie-Aspect is 1.33:1 - prescaling to correct movie aspect.
VO Config (176x132->176x132,flags=0,'MPlayer',0x32315659)
VO: [x11] 176x132 => 176x132 Planar YV12 
VO: Description: X11 ( XImage/Shm )
VO: Author: Aaron Holtzman <aholtzma(at)ess.engr.uvic.ca>
Sharing memory.
No accelerated colorspace conversion found
SwScaler: using unscaled Planar YV12 -> BGR 32-bit special converter
Selected video codec: [mpeg12] vfm:libmpeg2 (MPEG 1 or 2 (libmpeg2))
==========================================================================
Checking audio filter chain for 44100Hz/2ch/16bit -> 44100Hz/2ch/16bit...
[libaf] Adding filter dummy 
[dummy] Was reinitialized, rate=44100Hz, nch = 2, format = 0x00000000 and 
bps = 2
AF_pre: af format: 2 bps, 2 ch, 44100 hz, big endian signed int 
AF_pre: 44100Hz 2ch Signed 16-bit (Big-Endian)
ao_sun: sample counter increment: 882
ao_sun: minimum sample counter increment per 10msec interval: 882
        using sample counter based timing code
AO: [sun] 44100Hz 2ch Signed 16-bit (Big-Endian) (2 bps)
AO: Description: Sun audio output
AO: Author: jk(at)tools.de
Building audio filter chain for 44100Hz/2ch/16bit -> 44100Hz/2ch/16bit...
[dummy] Was reinitialized, rate=44100Hz, nch = 2, format = 0x00000000 and 
bps = 2
[dummy] Was reinitialized, rate=44100Hz, nch = 2, format = 0x00000000 and 
bps = 2
Starting playback...
*** [vo] Allocating mp_image_t, 176x144x12bpp YUV planar, 38016 bytes
*** [vo] Allocating (slices) mp_image_t, 176x144x12bpp YUV planar, 38016 
bytes
*** [vo] Allocating mp_image_t, 176x144x12bpp YUV planar, 38016 bytes0%
Uninit audio filters...-0.795 ct:  0.035  200/200   1%  0% 25.1% 105 0 0%
[libaf] Removing filter dummy 
uninit audio: mp3lib
uninit video: libmpeg2
DEMUXER: freeing demuxer at 4dee50  
DEMUXER: freeing sh_audio at 533f68  
DEMUXER: freeing sh_video at 53bc10  
vo: uninit ...

Exiting... (Quit)

-------------------------------------------------------------------------

Here's the verbose "-sdp" run:

$ mplayer -V -sdp ./testMPEGAudioVideo.sdp
Using GNU internationalization
Original domain: messages
Original dirname: /usr/lib/locale/
Current domain: mplayer
Current dirname: /home/caron/sw28/share/locale


MPlayer 0.90rc5-2.95.3 (C) 2000-2003 Arpad Gereoffy (see DOCS)

Reading config file /home/caron/sw28/etc/mplayer/mplayer.conf
Reading config file /home/caron/.mplayer/config
Reading /home/caron/.mplayer/codecs.conf: 50 audio & 136 video codecs
CommandLine: '-V' '-sdp' './testMPEGAudioVideo.sdp'
get_path('font/font.desc') -> '/home/caron/.mplayer/font/font.desc'
font: can't open file: /home/caron/.mplayer/font/font.desc
Font /home/caron/sw28/share/mplayer/font/font.desc loaded successfully! 
(206 chars)
Using Unoptimized OnScreenDisplay
Using usleep() timing
get_path('input.conf') -> '/home/caron/.mplayer/input.conf'
Parsing input config file /home/caron/.mplayer/input.conf
Input config file /home/caron/.mplayer/input.conf parsed : 52 binds
get_path('testMPEGAudioVideo.sdp.conf') -> 
'/home/caron/.mplayer/testMPEGAudioVideo.sdp.conf'

Playing ./testMPEGAudioVideo.sdp
Not an URL!
File size is 294 bytes
CACHE_PRE_INIT: 0 [0] 0  pre:0  eof:0  
Cache fill:  0.00% (0 bytes)    

MPlayer interrupted by signal 2 in module: enable_cache
vo: x11 uninit called but X11 not inited..

-------------------------------------------------------------------------

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>Subject: Re: [Live-devel] Using SDP Files with MPlayer
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>X-Spam-Checker-Version: SpamAssassin 2.50 (1.173-2003-02-20-exp)
>
>
>>However, the following mplayer command doesn't work:
>>
>>% ./mplayer -sdp testMPEGAudioVideo.sdp
>
>That's odd.  That *should* be working.
>
>>I've been trying to get mplayer to work the the "-sdp" command line 
option
>>on a Solaris 8 client.  I'm using live.2003.03.25 and MPlayer-0.90rc5 (I
>>also tried MPlayer-20030325).
>
>I tested it just now with MPlayer-20030401 (on Linux), and it works 
>OK.  (Note: Do *not* use the "rc" version; it's out of date!)
>
>I didn't know that MPlayer was supposed to be portable to Solaris.  (It 
was 
>targeted originally for Linux.)  So, it wouldn't surprise me if there was 
>some part of the MPlayer code that's not quite portable to Solaris.
>
>One other thing you could try is enable the built-in RTSP server in 
>"testMPEGAudioVideoStreamer", and then run "mplayer" to read a "rtsp://" 
>URL (instead of a SDP file).  Uncomment the line
>         #define IMPLEMENT_RTSP_SERVER 1
>in "testMPEGAudioVideoStreamer.cpp".  Then, rebuild and rerun 
>"testMPEGAudioVideoStreamer" as root (so that it can bind to the RTSP 
port: 
>554).
>
>Then, run mplayer as:
>         mplayer 
rtsp://<name-of-computer-running-testMPEGAudioVideoStreamer>/
>
>     Ross.
>
>
>
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From <EMAIL-ADDRESS-HIDDEN> Tue Apr 1 13:03:06 2003
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From: Paul Caron <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] Using SDP Files with MPlayer
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X-Original-Date: Tue, 1 Apr 2003 15:44:20 -0500 (EST)
Date: Tue, 1 Apr 2003 15:44:20 -0500 (EST)

I need to revise my last email with results from the CVS versions instead 
of the RC versions to avoid flames, although the results are the same.  The 
RTSP run works fine but the SDP run hangs.

Any ideas?

Versions Used
-------------
MPlayer-20030401
live.2003.03.31
ffmpeg-cvs-2003-04-01 (used libavcodec from this)

------------------------------------------------------------------------

Here's the mplayer run with the RTSP option.  Works fine...

# ./testMPEGAudioVideoStreamer
Beginning streaming...
Beginning to read from file...

$ mplayer -V rtsp://lompoc/
Using GNU internationalization
Original domain: messages
Original dirname: /usr/lib/locale/
Current domain: mplayer
Current dirname: /home/caron/sw28/share/locale


MPlayer dev-CVS-030401-15:17-2.95.3 (C) 2000-2003 Arpad Gereoffy (see DOCS)

Reading config file /home/caron/sw28/etc/mplayer/mplayer.conf
Reading config file /home/caron/.mplayer/config
Reading /home/caron/.mplayer/codecs.conf: 50 audio & 136 video codecs
CommandLine: '-V' 'rtsp://lompoc/'
get_path('font/font.desc') -> '/home/caron/.mplayer/font/font.desc'
font: can't open file: /home/caron/.mplayer/font/font.desc
Font /home/caron/sw28/share/mplayer/font/font.desc loaded successfully! 
(206 chars)
Using Unoptimized OnScreenDisplay
Using usleep() timing
get_path('input.conf') -> '/home/caron/.mplayer/input.conf'
Parsing input config file /home/caron/.mplayer/input.conf
Input config file /home/caron/.mplayer/input.conf parsed : 52 binds
get_path('.conf') -> '/home/caron/.mplayer/.conf'

Playing rtsp://lompoc/
Connected to server: lompoc
This stream is non-cacheable
Stream not seekable!
Sending request: DESCRIBE rtsp://lompoc/ RTSP/1.0
CSeq: 1
Accept: application/sdp
User-Agent: MPlayer (LIVE.COM Streaming Media v2003.03.31)


Received DESCRIBE response: RTSP/1.0 200 OK
CSeq: 1
Content-Type: application/sdp
Content-Length: 326


Need to read 326 extra bytes
Read 326 extra bytes: v=0
o=- 1049229380615971 1 IN IP4 141.162.1.101
s=Session streamed by "testMPEGAudioVideoStreamer"
i=LIVE.COM Streaming Media
t=0 0
a=tool:LIVE.COM Streaming Media
a=type:broadcast
m=audio 6666 RTP/AVP 14
a=control:track1
c=IN IP4 239.255.42.42/7
m=video 8888 RTP/AVP 32
a=control:track2
c=IN IP4 239.255.42.42/7

Initiated "audio/MPA" RTP subsession
Increased audio socket receive buffer to 100000 bytes 
Sending request: SETUP rtsp://lompoc//track1 RTSP/1.0
CSeq: 2
Transport: RTP/AVP;multicast;client_port=6666-6667
User-Agent: MPlayer (LIVE.COM Streaming Media v2003.03.31)


Received SETUP response: RTSP/1.0 200 OK
CSeq: 2
Transport: RTP/AVP;multicast;destination=239.255.42.42;port=6666;ttl=7
Session: 1


Initiated "video/MPV" RTP subsession
Increased video socket receive buffer to 173000 bytes 
Sending request: SETUP rtsp://lompoc//track2 RTSP/1.0
CSeq: 3
Transport: RTP/AVP;multicast;client_port=8888-8889
Session: 1
User-Agent: MPlayer (LIVE.COM Streaming Media v2003.03.31)


Received SETUP response: RTSP/1.0 200 OK
CSeq: 3
Transport: RTP/AVP;multicast;destination=239.255.42.42;port=8888;ttl=7
Session: 1


Sending request: PLAY rtsp://lompoc/ RTSP/1.0
CSeq: 4
Session: 1
Range: npt=0-
User-Agent: MPlayer (LIVE.COM Streaming Media v2003.03.31)


Received PLAY response: RTSP/1.0 200 OK
CSeq: 4
Session: 1


==> Found audio stream: 0
==> Found video stream: 0
Searching for sequence header... OK!
VIDEO:  MPEG1  320x240  (aspect 1)  24.00 fps  480.0 kbps (60.0 kbyte/s)
[V] filefmt:21  fourcc:0x10000001  size:320x240  fps:24.00  ftime:=0.0417
get_path('sub/') -> '/home/caron/.mplayer/sub/'
==========================================================================
Opening audio decoder: [mp3lib] MPEG layer-2, layer-3
dec_audio: Allocating 4608 + 65536 = 70144 bytes for output buffer
mp3lib: using generic C decore!
MP3lib: init layer2&3 finished, tables done
MPEG 1.0, Layer II, 44100 Hz 96 kbit Single-Channel, BPF: 314
Channels: 1, copyright: No, original: Yes, CRC: No, emphasis: 0
AUDIO: 44100 Hz, 2 ch, 16 bit (0x20), ratio: 12000->176400 (96.0 kbit)
Selected audio codec: [mp3] afm:mp3lib (mp3lib MPEG layer-2, layer-3)
==========================================================================
X11 opening display: :0.0
vo: X11 color mask:  FFFFFF  (R:FF G:FF00 B:FF0000)
vo: X11 running at 1280x1024 with depth 24 and 32 bpp (":0.0" => local 
display)
[x11] Unknown wm type...
==========================================================================
Opening video decoder: [mpegpes] MPEG 1/2 Video passthrough
VDec: vo config request - 320 x 240 (preferred csp: Mpeg PES)
Trying filter chain: vo
Could not find matching colorspace - retrying with -vop scale...
Opening video filter: [scale]
SwScale params: -1 x -1 (-1=no scaling)
Trying filter chain: scale vo
The selected video_out device is incompatible with this codec.
VDecoder init failed :(
Opening video decoder: [libmpeg2] MPEG 1/2 Video decoder v2.0
libmpeg2: No accelerated IDCT transform found
libmpeg2: No accelerated motion compensation found
VDec: vo config request - 320 x 240 (preferred csp: Planar YV12)
Trying filter chain: vo
VDec: using Planar YV12 as output csp (no 0)
Movie-Aspect is undefined - no prescaling applied.
VO Config (320x240->320x240,flags=0,'MPlayer',0x32315659)
VO: [x11] 320x240 => 320x240 Planar YV12 
VO: Description: X11 ( XImage/Shm )
VO: Author: Aaron Holtzman <aholtzma(at)ess.engr.uvic.ca>
Sharing memory.
No accelerated colorspace conversion found
SwScaler: using unscaled Planar YV12 -> BGR 32-bit special converter
Selected video codec: [mpeg12] vfm:libmpeg2 (MPEG 1 or 2 (libmpeg2))
==========================================================================
Checking audio filter chain for 44100Hz/2ch/16bit -> 44100Hz/2ch/16bit...
[libaf] Adding filter dummy 
[dummy] Was reinitialized, rate=44100Hz, nch = 2, format = 0x00000000 and 
bps = 2
AF_pre: af format: 2 bps, 2 ch, 44100 hz, big endian signed int 
AF_pre: 44100Hz 2ch Signed 16-bit (Big-Endian)
ao_sun: sample counter increment: 882
ao_sun: minimum sample counter increment per 10msec interval: 882
        using sample counter based timing code
AO: [sun] 44100Hz 2ch Signed 16-bit (Big-Endian) (2 bps)
AO: Description: Sun audio output
AO: Author: jk(at)tools.de
Building audio filter chain for 44100Hz/2ch/16bit -> 44100Hz/2ch/16bit...
[dummy] Was reinitialized, rate=44100Hz, nch = 2, format = 0x00000000 and 
bps = 2
[dummy] Was reinitialized, rate=44100Hz, nch = 2, format = 0x00000000 and 
bps = 2
Starting playback...
*** [vo] Allocating mp_image_t, 320x240x12bpp YUV planar, 115200 bytes
*** [vo] Allocating (slices) mp_image_t, 320x240x12bpp YUV planar, 115200 
bytes
*** [vo] Allocating mp_image_t, 320x240x12bpp YUV planar, 115200 bytes%
video stream has been synchronized using RTCP  42  14%  3%  1.6% 0 0 0%
audio stream has been synchronized using RTCP  43  14%  3%  1.6% 0 0 0%
audio stream is no longer RTCP-synchronized 7/ 47  14%  3%  1.7% 0 0 0%
audio stream has been synchronized using RTCP  62  14%  3%  1.7% 0 0 0%
Uninit audio filters...-0.220 ct: -1.522  685/685  13%  4%  1.7% 0 0 0%
[libaf] Removing filter dummy 
uninit audio: mp3lib
uninit video: libmpeg2
DEMUXER: freeing demuxer at 37ec98  
Sending request: TEARDOWN rtsp://lompoc//track1 RTSP/1.0
CSeq: 5
Session: 1
User-Agent: MPlayer (LIVE.COM Streaming Media v2003.03.31)


Sending request: TEARDOWN rtsp://lompoc//track2 RTSP/1.0
CSeq: 6
Session: 1
User-Agent: MPlayer (LIVE.COM Streaming Media v2003.03.31)


DEMUXER: freeing sh_audio at 540340  
DEMUXER: freeing sh_video at 540428  
vo: uninit ...

Exiting... (Quit)

------------------------------------------------------------------------

Here's the mplayer run with the SDP file.  Still hangs....

$ ./testMPEGAudioVideoStreamer
Beginning streaming...
Beginning to read from file...

$ mplayer -V -sdp ./testMPEGAudioVideo.sdp
Using GNU internationalization
Original domain: messages
Original dirname: /usr/lib/locale/
Current domain: mplayer
Current dirname: /home/caron/sw28/share/locale


MPlayer dev-CVS-030401-15:17-2.95.3 (C) 2000-2003 Arpad Gereoffy (see DOCS)

Reading config file /home/caron/sw28/etc/mplayer/mplayer.conf
Reading config file /home/caron/.mplayer/config
Reading /home/caron/.mplayer/codecs.conf: 50 audio & 136 video codecs
CommandLine: '-V' '-sdp' './testMPEGAudioVideo.sdp'
get_path('font/font.desc') -> '/home/caron/.mplayer/font/font.desc'
font: can't open file: /home/caron/.mplayer/font/font.desc
Font /home/caron/sw28/share/mplayer/font/font.desc loaded successfully! 
(206 chars)
Using Unoptimized OnScreenDisplay
Using usleep() timing
get_path('input.conf') -> '/home/caron/.mplayer/input.conf'
Parsing input config file /home/caron/.mplayer/input.conf
Input config file /home/caron/.mplayer/input.conf parsed : 52 binds
get_path('testMPEGAudioVideo.sdp.conf') -> 
'/home/caron/.mplayer/testMPEGAudioVideo.sdp.conf'

Playing ./testMPEGAudioVideo.sdp
Not an URL!
File size is 294 bytes
CACHE_PRE_INIT: 0 [0] 0  pre:0  eof:0  
Cache fill:  0.00% (0 bytes)    

MPlayer interrupted by signal 2 in module: enable_cache
vo: x11 uninit called but X11 not inited..

------------------------------------------------------------------------



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From <EMAIL-ADDRESS-HIDDEN> Tue Apr 1 13:17:03 2003
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From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] Using SDP Files with MPlayer
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X-Original-Date: Tue, 01 Apr 2003 13:08:35 -0800
Date: Tue, 01 Apr 2003 13:08:35 -0800

--=====================_3953835==_
Content-Type: text/plain; charset="us-ascii"; format=flowed

At 12:44 PM 4/1/03, you wrote:
>I need to revise my last email with results from the CVS versions instead
>of the RC versions to avoid flames, although the results are the same.  The
>RTSP run works fine but the SDP run hangs.

OK, there's definitely a file reading/parsing problem somewhere.  Note that 
the lines

>Initiated "audio/MPA" RTP subsession
>Initiated "video/MPV" RTP subsession

appear in the output of the "rtsp://" test, but *not* the "-sdp" 
test.  This shows that the MPlayer code is failing to read and/or parse the 
".sdp" file properly.

Could you please replace "libmpdemux/demux_rtp.cpp" with the attached 
version (which prints out some debugging info), re-run "mplayer -V -sdp 
testMPEGAudioVideo.sdp", and send me the output?

         Ross.

--=====================_3953835==_
Content-Type: text/plain; charset="us-ascii"

////////// Routines (with C-linkage) that interface between "MPlayer"
////////// and the "LIVE.COM Streaming Media" libraries:

extern "C" {
#include "demux_rtp.h"
#include "stheader.h"
}
#include "demux_rtp_internal.h"

#include "BasicUsageEnvironment.hh"
#include "liveMedia.hh"
#include "GroupsockHelper.hh"
#include <unistd.h>

extern "C" stream_t* stream_open_sdp(int fd, off_t fileSize,
				     int* file_format) {
  *file_format = DEMUXER_TYPE_RTP;
  stream_t* newStream = NULL;
  do {
    fprintf(stderr, "stream_open_sdp()1: fileSize %d\n", fileSize);//#####(at)(at)(at)(at)(at)
    char* sdpDescription = (char*)malloc(fileSize+1);
    if (sdpDescription == NULL) break;

    fprintf(stderr, "stream_open_sdp()2: fileSize %d\n", fileSize);//#####(at)(at)(at)(at)(at)
    ssize_t numBytesRead = read(fd, sdpDescription, fileSize);
    fprintf(stderr, "stream_open_sdp()3: numBytesRead %d\n", numBytesRead);//#####(at)(at)(at)(at)(at)
    if (numBytesRead != fileSize) break;
    sdpDescription[fileSize] = '\0'; // to be safe

    newStream = (stream_t*)calloc(sizeof (stream_t), 1);
    if (newStream == NULL) break;

    // Store the SDP description in the 'priv' field, for later use:
    newStream->priv = sdpDescription; 
    fprintf(stderr, "stream_open_sdp()4: sdpDescription %s\n", sdpDescription);//#####(at)(at)(at)(at)(at)
  } while (0);
  return newStream;
}

extern "C" int _rtsp_streaming_seek(int /*fd*/, off_t /*pos*/,
				    streaming_ctrl_t* /*streaming_ctrl*/) {
  return -1; // For now, we don't handle RTSP stream seeking
}

extern "C" int rtsp_streaming_start(stream_t* stream) {
  stream->streaming_ctrl->streaming_seek = _rtsp_streaming_seek;

  return 0;
}

// A data structure representing input data for each stream:
class ReadBufferQueue {
public:
  ReadBufferQueue(MediaSubsession* subsession, demuxer_t* demuxer,
		  char const* tag);
  virtual ~ReadBufferQueue();

  FramedSource* readSource() const { return fReadSource; }
  RTPSource* rtpSource() const { return fRTPSource; }
  demuxer_t* ourDemuxer() const { return fOurDemuxer; }
  char const* tag() const { return fTag; }

  char blockingFlag; // used to implement synchronous reads

  // For A/V synchronization:
  Boolean prevPacketWasSynchronized;
  float prevPacketPTS;
  ReadBufferQueue** otherQueue;

  // The 'queue' actually consists of just a single "demux_packet_t"
  // (because the underlying OS does the actual queueing/buffering):
  demux_packet_t* dp;

  // However, we sometimes inspect buffers before delivering them.
  // For this, we maintain a queue of pending buffers:
  void savePendingBuffer(demux_packet_t* dp);
  demux_packet_t* getPendingBuffer();

private:
  demux_packet_t* pendingDPHead;
  demux_packet_t* pendingDPTail;

  FramedSource* fReadSource;
  RTPSource* fRTPSource;
  demuxer_t* fOurDemuxer;
  char const* fTag; // used for debugging
};

// A structure of RTP-specific state, kept so that we can cleanly
// reclaim it:
typedef struct RTPState {
  char const* sdpDescription;
  RTSPClient* rtspClient;
  MediaSession* mediaSession;
  ReadBufferQueue* audioBufferQueue;
  ReadBufferQueue* videoBufferQueue;
  unsigned flags;
  struct timeval firstSyncTime;
};

int rtspStreamOverTCP = 0; 

extern "C" void demux_open_rtp(demuxer_t* demuxer) {
  do {
    TaskScheduler* scheduler = BasicTaskScheduler::createNew();
    if (scheduler == NULL) break;
    UsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler);
    if (env == NULL) break;

    RTSPClient* rtspClient = NULL;

    if (demuxer == NULL || demuxer->stream == NULL) break;  // shouldn't happen
    demuxer->stream->eof = 0; // just in case 

    // Look at the stream's 'priv' field to see if we were initiated
    // via a SDP description:
    char* sdpDescription = (char*)(demuxer->stream->priv);
    fprintf(stderr, "demux_open_rtp()1, sdpDescription %s\n", sdpDescription);//#####(at)(at)(at)(at)(at)
    if (sdpDescription == NULL) {
      // We weren't given a SDP description directly, so assume that
      // we were given a RTSP URL:
      char const* url = demuxer->stream->streaming_ctrl->url->url;

      extern int verbose;
      rtspClient = RTSPClient::createNew(*env, verbose, "MPlayer");
      if (rtspClient == NULL) {
	fprintf(stderr, "Failed to create RTSP client: %s\n",
		env->getResultMsg());
	break;
      }

      sdpDescription = rtspClient->describeURL(url);
      if (sdpDescription == NULL) {
	fprintf(stderr, "Failed to get a SDP description from URL \"%s\": %s\n",
		url, env->getResultMsg());
	break;
      }
    }

    // Now that we have a SDP description, create a MediaSession from it:
    MediaSession* mediaSession = MediaSession::createNew(*env, sdpDescription);
    fprintf(stderr, "demux_open_rtp()2, mediaSession %p\n", mediaSession);//#####(at)(at)(at)(at)(at)
    if (mediaSession == NULL) break;


    // Create a 'RTPState' structure containing the state that we just created,
    // and store it in the demuxer's 'priv' field, for future reference:
    RTPState* rtpState = new RTPState;
    rtpState->sdpDescription = sdpDescription;
    rtpState->rtspClient = rtspClient;
    rtpState->mediaSession = mediaSession;
    rtpState->audioBufferQueue = rtpState->videoBufferQueue = NULL;
    rtpState->flags = 0;
    rtpState->firstSyncTime.tv_sec = rtpState->firstSyncTime.tv_usec = 0;
    demuxer->priv = rtpState;

    // Create RTP receivers (sources) for each subsession:
    MediaSubsessionIterator iter(*mediaSession);
    MediaSubsession* subsession;
    unsigned desiredReceiveBufferSize;
    while ((subsession = iter.next()) != NULL) {
      // Ignore any subsession that's not audio or video:
      if (strcmp(subsession->mediumName(), "audio") == 0) {
	desiredReceiveBufferSize = 100000;
      } else if (strcmp(subsession->mediumName(), "video") == 0) {
	desiredReceiveBufferSize = 2000000;
      } else {
	continue;
      }

      if (!subsession->initiate()) {
	fprintf(stderr, "Failed to initiate \"%s/%s\" RTP subsession: %s\n", subsession->mediumName(), subsession->codecName(), env->getResultMsg());
      } else {
	fprintf(stderr, "Initiated \"%s/%s\" RTP subsession\n", subsession->mediumName(), subsession->codecName());

	// Set the OS's socket receive buffer sufficiently large to avoid
	// incoming packets getting dropped between successive reads from this
	// subsession's demuxer.  Depending on the bitrate(s) that you expect,
	// you may wish to tweak the "desiredReceiveBufferSize" values above.
	int rtpSocketNum = subsession->rtpSource()->RTPgs()->socketNum();
	int receiveBufferSize
	  = increaseReceiveBufferTo(*env, rtpSocketNum,
				    desiredReceiveBufferSize);
	if (verbose > 0) {
	  fprintf(stderr, "Increased %s socket receive buffer to %d bytes \n",
		  subsession->mediumName(), receiveBufferSize);
	}

	if (rtspClient != NULL) {
	  // Issue a RTSP "SETUP" command on the chosen subsession:
	  if (!rtspClient->setupMediaSubsession(*subsession, False,
						rtspStreamOverTCP)) break;
	}
      }
    }

    if (rtspClient != NULL) {
      // Issue a RTSP aggregate "PLAY" command on the whole session:
      if (!rtspClient->playMediaSession(*mediaSession)) break;
    }

    // Now that the session is ready to be read, do additional
    // MPlayer codec-specific initialization on each subsession:
    iter.reset();
    while ((subsession = iter.next()) != NULL) {
      if (subsession->readSource() == NULL) continue; // not reading this

      unsigned flags = 0;
      if (strcmp(subsession->mediumName(), "audio") == 0) {
	rtpState->audioBufferQueue
	  = new ReadBufferQueue(subsession, demuxer, "audio");
	rtpState->audioBufferQueue->otherQueue = &(rtpState->videoBufferQueue);
	rtpCodecInitialize_audio(demuxer, subsession, flags);
      } else if (strcmp(subsession->mediumName(), "video") == 0) {
	rtpState->videoBufferQueue
	  = new ReadBufferQueue(subsession, demuxer, "video");
	rtpState->videoBufferQueue->otherQueue = &(rtpState->audioBufferQueue);
	rtpCodecInitialize_video(demuxer, subsession, flags);
      }
      rtpState->flags |= flags;
    }
  } while (0);
}

extern "C" int demux_is_mpeg_rtp_stream(demuxer_t* demuxer) {
  // Get the RTP state that was stored in the demuxer's 'priv' field:
  RTPState* rtpState = (RTPState*)(demuxer->priv);

  return (rtpState->flags&RTPSTATE_IS_MPEG12_VIDEO) != 0;
}

static demux_packet_t* getBuffer(demuxer_t* demuxer, demux_stream_t* ds,
				 Boolean mustGetNewData,
				 float& ptsBehind); // forward

extern "C" int demux_rtp_fill_buffer(demuxer_t* demuxer, demux_stream_t* ds) {
  // Get a filled-in "demux_packet" from the RTP source, and deliver it.
  // Note that this is called as a synchronous read operation, so it needs
  // to block in the (hopefully infrequent) case where no packet is
  // immediately available.

  while (1) {
    float ptsBehind;
    demux_packet_t* dp = getBuffer(demuxer, ds, False, ptsBehind); // blocking
    if (dp == NULL) return 0;

    if (demuxer->stream->eof) return 0; // source stream has closed down
  
    // Before using this packet, check to make sure that its presentation
    // time is not far behind the other stream (if any).  If it is,
    // then we discard this packet, and get another instead.  (The rest of
    // MPlayer doesn't always do a good job of synchronizing when the
    // audio and video streams get this far apart.)
    // (We don't do this when streaming over TCP, because then the audio and
    // video streams are interleaved.)
    const float ptsBehindThreshold = 1.0; // seconds
    if (ptsBehind < ptsBehindThreshold || rtspStreamOverTCP) { // packet's OK
      ds_add_packet(ds, dp);
      break;
    }
    
    free_demux_packet(dp); // give back this packet, and get another one
  }

  return 1;
}

Boolean awaitRTPPacket(demuxer_t* demuxer, demux_stream_t* ds,
		       unsigned char*& packetData, unsigned& packetDataLen,
		       float& pts) {
  // Similar to "demux_rtp_fill_buffer()", except that the "demux_packet"
  // is not delivered to the "demux_stream".
  float ptsBehind;
  demux_packet_t* dp = getBuffer(demuxer, ds, True, ptsBehind); // blocking
  if (dp == NULL) return False;

  packetData = dp->buffer;
  packetDataLen = dp->len;
  pts = dp->pts;

  return True;
}

Boolean insertRTPData(demuxer_t* demuxer, demux_stream_t* ds,
		      unsigned char* data, unsigned dataLen) {
  // Begin by finding the buffer queue that we want to add data to.
  // (Get this from the RTP state, which we stored in
  //  the demuxer's 'priv' field)
  RTPState* rtpState = (RTPState*)(demuxer->priv);
  ReadBufferQueue* bufferQueue = NULL;
  if (ds == demuxer->video) {
    bufferQueue = rtpState->videoBufferQueue;
  } else if (ds == demuxer->audio) {
    bufferQueue = rtpState->audioBufferQueue;
  } else {
    fprintf(stderr, "(demux_rtp)insertRTPData: internal error: unknown stream\n");
    return False;
  }

  if (data == NULL || dataLen == 0) return False;

  demux_packet_t* dp = new_demux_packet(dataLen);
  if (dp == NULL) return False;

  // Copy our data into the buffer, and save it:
  memmove(dp->buffer, data, dataLen);
  dp->len = dataLen;
  dp->pts = 0;
  bufferQueue->savePendingBuffer(dp);
}

static void teardownRTSPSession(RTPState* rtpState); // forward

extern "C" void demux_close_rtp(demuxer_t* demuxer) {
  // Reclaim all RTP-related state:

  // Get the RTP state that was stored in the demuxer's 'priv' field:
  RTPState* rtpState = (RTPState*)(demuxer->priv);
  if (rtpState == NULL) return;

  teardownRTSPSession(rtpState);

  UsageEnvironment* env = NULL;
  TaskScheduler* scheduler = NULL;
  if (rtpState->mediaSession != NULL) {
    env = &(rtpState->mediaSession->envir());
    scheduler = &(env->taskScheduler());
  }
  Medium::close(rtpState->mediaSession);
  Medium::close(rtpState->rtspClient);
  delete rtpState->audioBufferQueue;
  delete rtpState->videoBufferQueue;
  delete rtpState->sdpDescription;
  delete rtpState;

  delete env; delete scheduler;
}

////////// Extra routines that help implement the above interface functions:

#define MAX_RTP_FRAME_SIZE 50000
    // >= the largest conceivable frame composed from one or more RTP packets

static void afterReading(void* clientData, unsigned frameSize,
			 struct timeval presentationTime) {
  if (frameSize >= MAX_RTP_FRAME_SIZE) {
    fprintf(stderr, "Saw an input frame too large (>=%d).  Increase MAX_RTP_FRAME_SIZE in \"demux_rtp.cpp\".\n",
	    MAX_RTP_FRAME_SIZE);
  }
  ReadBufferQueue* bufferQueue = (ReadBufferQueue*)clientData;
  demuxer_t* demuxer = bufferQueue->ourDemuxer();
  RTPState* rtpState = (RTPState*)(demuxer->priv);

  if (frameSize > 0) demuxer->stream->eof = 0;

  demux_packet_t* dp = bufferQueue->dp;
  dp->len = frameSize;

  // Set the packet's presentation time stamp, depending on whether or
  // not our RTP source's timestamps have been synchronized yet: 
  Boolean hasBeenSynchronized
    = bufferQueue->rtpSource()->hasBeenSynchronizedUsingRTCP();
  if (hasBeenSynchronized) {
    if (verbose > 0 && !bufferQueue->prevPacketWasSynchronized) {
      fprintf(stderr, "%s stream has been synchronized using RTCP \n",
	      bufferQueue->tag());
    }

    struct timeval* fst = &(rtpState->firstSyncTime); // abbrev
    if (fst->tv_sec == 0 && fst->tv_usec == 0) {
      *fst = presentationTime;
    }
    
    // For the "pts" field, use the time differential from the first
    // synchronized time, rather than absolute time, in order to avoid
    // round-off errors when converting to a float:
    dp->pts = presentationTime.tv_sec - fst->tv_sec
      + (presentationTime.tv_usec - fst->tv_usec)/1000000.0;
    bufferQueue->prevPacketPTS = dp->pts;
  } else {
    if (verbose > 0 && bufferQueue->prevPacketWasSynchronized) {
      fprintf(stderr, "%s stream is no longer RTCP-synchronized \n",
	      bufferQueue->tag());
    }

    // use the previous packet's "pts" once again:
    dp->pts = bufferQueue->prevPacketPTS;
  }
  bufferQueue->prevPacketWasSynchronized = hasBeenSynchronized;

  dp->pos = demuxer->filepos;
  demuxer->filepos += frameSize;

  // Signal any pending 'doEventLoop()' call on this queue:
  bufferQueue->blockingFlag = ~0;
}

static void onSourceClosure(void* clientData) {
  ReadBufferQueue* bufferQueue = (ReadBufferQueue*)clientData;
  demuxer_t* demuxer = bufferQueue->ourDemuxer();

  demuxer->stream->eof = 1;

  // Signal any pending 'doEventLoop()' call on this queue:
  bufferQueue->blockingFlag = ~0;
}

static demux_packet_t* getBuffer(demuxer_t* demuxer, demux_stream_t* ds,
				 Boolean mustGetNewData,
				 float& ptsBehind) {
  // Begin by finding the buffer queue that we want to read from:
  // (Get this from the RTP state, which we stored in
  //  the demuxer's 'priv' field)
  RTPState* rtpState = (RTPState*)(demuxer->priv);
  ReadBufferQueue* bufferQueue = NULL;
  if (ds == demuxer->video) {
    bufferQueue = rtpState->videoBufferQueue;
  } else if (ds == demuxer->audio) {
    bufferQueue = rtpState->audioBufferQueue;
  } else {
    fprintf(stderr, "(demux_rtp)getBuffer: internal error: unknown stream\n");
    return NULL;
  }

  if (bufferQueue == NULL || bufferQueue->readSource() == NULL) {
    fprintf(stderr, "(demux_rtp)getBuffer failed: no appropriate RTP subsession has been set up\n");
    return NULL;
  }
  
  demux_packet_t* dp;
  if (!mustGetNewData) {
    // Check whether we have a previously-saved buffer that we can use:
    dp = bufferQueue->getPendingBuffer();
    if (dp != NULL) return dp;
  }

  // Allocate a new packet buffer, and arrange to read into it:
  dp = new_demux_packet(MAX_RTP_FRAME_SIZE);
  bufferQueue->dp = dp;
  if (dp == NULL) return NULL;

  // Schedule the read operation:
  bufferQueue->blockingFlag = 0;
  bufferQueue->readSource()->getNextFrame(dp->buffer, MAX_RTP_FRAME_SIZE,
					  afterReading, bufferQueue,
					  onSourceClosure, bufferQueue);
  // Block ourselves until data becomes available:
  TaskScheduler& scheduler
    = bufferQueue->readSource()->envir().taskScheduler();
  scheduler.doEventLoop(&bufferQueue->blockingFlag);

  // Set the "ptsBehind" result parameter:
  if (bufferQueue->prevPacketPTS != 0.0 && *(bufferQueue->otherQueue) != NULL
      && (*(bufferQueue->otherQueue))->prevPacketPTS != 0.0) {
    ptsBehind = (*(bufferQueue->otherQueue))->prevPacketPTS
		 - bufferQueue->prevPacketPTS;
  } else {
    ptsBehind = 0.0;
  }

  if (mustGetNewData) {
    // Save this buffer for future reads:
    bufferQueue->savePendingBuffer(dp);
  }

  return dp;
}

static void teardownRTSPSession(RTPState* rtpState) {
  RTSPClient* rtspClient = rtpState->rtspClient;
  MediaSession* mediaSession = rtpState->mediaSession;
  if (rtspClient == NULL || mediaSession == NULL) return;

  MediaSubsessionIterator iter(*mediaSession);
  MediaSubsession* subsession;

  while ((subsession = iter.next()) != NULL) {
    rtspClient->teardownMediaSubsession(*subsession);
  }
}

////////// "ReadBuffer" and "ReadBufferQueue" implementation:

ReadBufferQueue::ReadBufferQueue(MediaSubsession* subsession,
				 demuxer_t* demuxer, char const* tag)
  : prevPacketWasSynchronized(False), prevPacketPTS(0.0), otherQueue(NULL),
    dp(NULL), pendingDPHead(NULL), pendingDPTail(NULL),
    fReadSource(subsession == NULL ? NULL : subsession->readSource()),
    fRTPSource(subsession == NULL ? NULL : subsession->rtpSource()),
    fOurDemuxer(demuxer), fTag(strdup(tag)) {
} 

ReadBufferQueue::~ReadBufferQueue() {
  delete fTag;

  // Free any pending buffers (that never got delivered):
  demux_packet_t* dp = pendingDPHead;
  while (dp != NULL) {
    demux_packet_t* dpNext = dp->next;
    dp->next = NULL;
    free_demux_packet(dp);
    dp = dpNext;
  }
}

void ReadBufferQueue::savePendingBuffer(demux_packet_t* dp) {
  // Keep this buffer around, until MPlayer asks for it later:
  if (pendingDPTail == NULL) {
    pendingDPHead = pendingDPTail = dp;
  } else {
    pendingDPTail->next = dp;
    pendingDPTail = dp;
  }
  dp->next = NULL;
}

demux_packet_t* ReadBufferQueue::getPendingBuffer() {
  demux_packet_t* dp = pendingDPHead;
  if (dp != NULL) {
    pendingDPHead = dp->next;
    if (pendingDPHead == NULL) pendingDPTail = NULL; 

    dp->next = NULL;
  }

  return dp;
}

--=====================_3953835==_--



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From <EMAIL-ADDRESS-HIDDEN> Tue Apr 1 14:05:17 2003
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From: Paul Caron <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] Using SDP Files with MPlayer
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X-Original-Date: Tue, 1 Apr 2003 16:58:26 -0500 (EST)
Date: Tue, 1 Apr 2003 16:58:26 -0500 (EST)

Here's the results with the new libmpdemux/demux_rtp.cpp....

$ mplayer -V -sdp testMPEGAudioVideo.sdp 
Using GNU internationalization
Original domain: messages
Original dirname: /usr/lib/locale/
Current domain: mplayer
Current dirname: /home/caron/sw28/share/locale


MPlayer dev-CVS-030401-15:17-2.95.3 (C) 2000-2003 Arpad Gereoffy (see DOCS)

Reading config file /home/caron/sw28/etc/mplayer/mplayer.conf
Reading config file /home/caron/.mplayer/config
Reading /home/caron/.mplayer/codecs.conf: 50 audio & 136 video codecs
CommandLine: '-V' '-sdp' 'testMPEGAudioVideo.sdp'
get_path('font/font.desc') -> '/home/caron/.mplayer/font/font.desc'
font: can't open file: /home/caron/.mplayer/font/font.desc
Font /home/caron/sw28/share/mplayer/font/font.desc loaded successfully! 
(206 chars)
Using Unoptimized OnScreenDisplay
Using usleep() timing
get_path('input.conf') -> '/home/caron/.mplayer/input.conf'
Parsing input config file /home/caron/.mplayer/input.conf
Input config file /home/caron/.mplayer/input.conf parsed : 52 binds
get_path('testMPEGAudioVideo.sdp.conf') -> 
'/home/caron/.mplayer/testMPEGAudioVideo.sdp.conf'

Playing testMPEGAudioVideo.sdp
Not an URL!
File size is 294 bytes
stream_open_sdp()1: fileSize 0
stream_open_sdp()2: fileSize 0
stream_open_sdp()3: numBytesRead 294
stream_open_sdp()4: sdpDescription v=0
o=- 49451 3 IN IP4 127.0.0.1
s=Test MPEG Audio+Video session
i=Parameters for the session streamed by "testMPEGAudioVideoStreamer"
t=0 0
a=tool:testMPEGAudioVideoStreamer
a=type:broadcast
m=audio 6666 RTP/AVP 14
c=IN IP4 239.255.42.42/127
m=video 8888 RTP/AVP 32
c=IN IP4 239.255.42.42/127

CACHE_PRE_INIT: 0 [0] 0  pre:0  eof:0  
Cache fill:  0.00% (0 bytes)    

MPlayer interrupted by signal 2 in module: enable_cache
vo: x11 uninit called but X11 not inited..

>X-Sender: rsf(at)laptop-localhost
>To: live-devel(at)lists.sourceforge.net
>From: Ross Finlayson <finlayson(at)live.com>
>Subject: Re: [Live-devel] Using SDP Files with MPlayer
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>X-Original-Date: Tue, 01 Apr 2003 13:08:35 -0800
>Date: Tue, 01 Apr 2003 13:08:35 -0800
>X-Spam-Status: No, hits=-10.1 required=5.0 
tests=AWL,BAYES_20,EMAIL_ATTRIBUTION,IN_REP_TO,KNOWN_MAILING_LIST 
autolearn=ham	version=2.50
>X-Spam-Level: 
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>
>At 12:44 PM 4/1/03, you wrote:
>>I need to revise my last email with results from the CVS versions instead
>>of the RC versions to avoid flames, although the results are the same.  
The
>>RTSP run works fine but the SDP run hangs.
>
>OK, there's definitely a file reading/parsing problem somewhere.  Note 
that 
>the lines
>
>>Initiated "audio/MPA" RTP subsession
>>Initiated "video/MPV" RTP subsession
>
>appear in the output of the "rtsp://" test, but *not* the "-sdp" 
>test.  This shows that the MPlayer code is failing to read and/or parse 
the 
>".sdp" file properly.
>
>Could you please replace "libmpdemux/demux_rtp.cpp" with the attached 
>version (which prints out some debugging info), re-run "mplayer -V -sdp 
>testMPEGAudioVideo.sdp", and send me the output?
>
>         Ross.



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Subject: Re: [Live-devel] Using SDP Files with MPlayer
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X-Original-Date: Tue, 01 Apr 2003 12:46:00 -0800
Date: Tue, 01 Apr 2003 12:46:00 -0800


>Here's the verbose "-sdp" run:
>
>$ mplayer -V -sdp ./testMPEGAudioVideo.sdp
[...]
>MPlayer 0.90rc5-2.95.3 (C) 2000-2003 Arpad Gereoffy (see DOCS)

Please try again using the "latest CVS snapshot" version of MPlayer.

>Is there any other way to tell mplayer to read from video and audio
>multicast streams if you know the addresses and port numbers?

No, because SDP is the IETF standard way to specify multimedia sessions.

For some reason, though, MPlayer is failing to properly read and/or parse 
the SDP file.  I suspect that this is a Solaris-related portability problem 
(because there's no problem on Linux).

One more thing you could try: Replace the contents of 
"testMPEGAudioVideo.sdp" with the SDP description that was returned by the 
RTSP "DESCRIBE" operation:

v=0
o=- 1049225941302234 1 IN IP4 141.162.1.101
s=Session streamed by "testMPEGAudioVideoStreamer"
i=LIVE.COM Streaming Media
t=0 0
a=tool:LIVE.COM Streaming Media
a=type:broadcast
m=audio 6666 RTP/AVP 14
a=control:track1
c=IN IP4 239.255.42.42/7
m=video 8888 RTP/AVP 32
a=control:track2
c=IN IP4 239.255.42.42/7

and try using "-sdp" once again.

         Ross.




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X-Original-Date: Tue, 01 Apr 2003 17:59:25 -0800
Date: Tue, 01 Apr 2003 17:59:25 -0800

The attached patch (to "libmpdemux/stream.c" in the MPlayer source code) 
should fix the problem.

It turns out that the problem showed up only on Solaris, as a consequence 
of the fact that, on Solaris, "lseek(NULL, ...)" returns 0, not -1 as it 
should.  Go figure...

         Ross.



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Subject: Re: [Live-devel] Using SDP Files with MPlayer - Oops! Here's
  the patch
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--=====================_1835609==_
Content-Type: text/plain; charset="us-ascii"; format=flowed

Oops - I forgot to attach the patch to my previous message.  Here it is now.

(In any case, the patch will probably show up in the "latest CVS snapshot" 
tonight or tomorrow.)

-----
The attached patch (to "libmpdemux/stream.c" in the MPlayer source code) 
should fix the problem.

It turns out that the problem showed up only on Solaris, as a consequence 
of the fact that, on Solaris, "lseek(NULL, ...)" returns 0, not -1 as it 
should.  Go figure...

         Ross.

--=====================_1835609==_
Content-Type: text/plain; charset="us-ascii"

diff -Nur -x CVS -x .* main/libmpdemux/stream.c main.dev/libmpdemux/stream.c
--- main/libmpdemux/stream.c	Wed Mar 19 12:56:40 2003
+++ main.dev/libmpdemux/stream.c	Tue Apr  1 17:07:12 2003
(at)(at) -56,7 +56,7 (at)(at)
 
 int stream_fill_buffer(stream_t *s){
   int len;
-  if(s->eof){ s->buf_pos=s->buf_len=0; return 0; }
+  if (s->fd == NULL || s->eof) { s->buf_pos = s->buf_len = 0; return 0; }
   switch(s->type){
 #ifdef LIBSMBCLIENT
   case STREAMTYPE_SMB:

--=====================_1835609==_--



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X-Original-Date: Tue, 1 Apr 2003 21:42:59 -0800 (PST)
Date: Tue, 1 Apr 2003 21:42:59 -0800 (PST)

Hi,

Again I have the same error.
I have class which name is MyClass

in cpp file I have 

void MyClass::afterPlaying(void* clientData)
{
	play();
}

void MyClass::play()
{
mySource = DeviceSource::createNew(*env,
deviceParams);

-error-	videoSink->startPlaying(*mySource,
afterPlaying, videoSink);
}
ýn h file I have 
void play();
void afterPlaying(void* clientData);

I still have the same problem. I think it is about the
member functions.

error C2664: 'unsigned int __thiscall
MediaSink::startPlaying(class MediaSource &,void
(__cdecl *)(void *),void *)' : cannot convert
parameter 2 from 'void' to 'void (__cdecl *)(void *)'
        Expressions of type void cannot be converted
to other types



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Subject: Re: [Live-devel] VC++ compiler error
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X-Original-Date: 02 Apr 2003 17:44:13 +1000
Date: 02 Apr 2003 17:44:13 +1000

Hi,

> I still have the same problem. I think it is about the
> member functions.
> 
> error C2664: 'unsigned int __thiscall
> MediaSink::startPlaying(class MediaSource &,void
> (__cdecl *)(void *),void *)' : cannot convert
> parameter 2 from 'void' to 'void (__cdecl *)(void *)'
>         Expressions of type void cannot be converted
> to other types

I think you are attempting to do a C-style callback with a C++ instance
member function, but callbacks work with static C++ member functions. 

It because C++ member functions use an additional parameter (this) that
can't be created on the stack by the calling code, as the caller is not
aware that it is calling a C++ member function.

Doug.





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X-Original-Date: Wed, 02 Apr 2003 00:24:24 -0800
Date: Wed, 02 Apr 2003 00:24:24 -0800


> > I still have the same problem. I think it is about the
> > member functions.
> >
> > error C2664: 'unsigned int __thiscall
> > MediaSink::startPlaying(class MediaSource &,void
> > (__cdecl *)(void *),void *)' : cannot convert
> > parameter 2 from 'void' to 'void (__cdecl *)(void *)'
> >         Expressions of type void cannot be converted
> > to other types
>
>I think you are attempting to do a C-style callback with a C++ instance
>member function, but callbacks work with static C++ member functions.

That's right.  The callback function cannot be a (non-static) member 
function.  Instead, declare it as static, or else a function defined 
outside of a class.

         Ross.



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X-Original-Date: Wed, 2 Apr 2003 09:49:42 -0500 (EST)
Date: Wed, 2 Apr 2003 09:49:42 -0500 (EST)

Thanks.  Much better!

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>Subject: Re: [Live-devel] Using SDP Files with MPlayer - Oops! Here's the 
patch
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>Oops - I forgot to attach the patch to my previous message.  Here it is 
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>
>(In any case, the patch will probably show up in the "latest CVS snapshot" 
>tonight or tomorrow.)
>
>-----
>The attached patch (to "libmpdemux/stream.c" in the MPlayer source code) 
>should fix the problem.
>
>It turns out that the problem showed up only on Solaris, as a consequence 
>of the fact that, on Solaris, "lseek(NULL, ...)" returns 0, not -1 as it 
>should.  Go figure...
>
>         Ross.



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Subject: [Live-devel] openRTSP crash upon exit (-t option)
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X-Original-Date: Wed, 02 Apr 2003 16:56:13 +0200
Date: Wed, 02 Apr 2003 16:56:13 +0200

Hi,

I've just downloaded the lastest version and it appears that openRTSP
still crashes upon exit (-t option):

==========================================================
strace openRTSP -e 3 -V -t -n rtsp://speedy:8000/count.mov
.............
Session: 5659498657368204914
User-Agent: openRTSP (LIVE.COM Streaming Media v2003.04.01)


) = 178
send(3, "TEARDOWN rtsp://speedy:8000/coun"..., 160, 0) = 160
close(6)                                = 0
munmap(0x40017000, 4096)                = 0
rt_sigprocmask(SIG_BLOCK, [ALRM], NULL, 8) = 0
rt_sigprocmask(SIG_UNBLOCK, [ALRM], NULL, 8) = 0
gettimeofday({1049295259, 901544}, {4294967176, 0}) = 0
send(3, "$", 1, 0)                      = 1
send(3, "\1", 1, 0)                     = -1 EPIPE (Broken pipe)
--- SIGPIPE (Broken pipe) ---
+++ killed by SIGPIPE +++

=========================================================
~/softs/openRTSP/live/testProgs> strace openRTSP -e 3 -V -t -n -q
rtsp://speedy:8000/count.mov
........................
Session: 7958809107880340122
User-Agent: openRTSP (LIVE.COM Streaming Media v2003.04.01)


) = 178
send(3, "TEARDOWN rtsp://speedy:8000/coun"..., 160, 0) = 160
write(1, "\0\0\0\0mdat", 8mdat)             = 8
_llseek(1, 0, 0xbfffeb94, SEEK_SET)     = -1 ESPIPE (Illegal seek)
write(2, "QuickTimeFileSink::setWord(): fs"...,
52QuickTimeFileSink::setWord(): fseek failed (err 29)
) = 52
munmap(0x40017000, 4096)                = 0
_exit(1)                                = ?


Thanks,

romulus


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X-Original-Date: Wed, 02 Apr 2003 12:39:42 -0800
Date: Wed, 02 Apr 2003 12:39:42 -0800


>I've just downloaded the lastest version and it appears that openRTSP
>still crashes upon exit (-t option):

Thanks again for the report.

The first example you showed happens to be an obscure race condition, 
caused by the fact that the remote server has apparently closed down the 
RTCP TCP connection (after handling the RTSP "TEARDOWN" command), before 
"openRTSP" had a chance to stop any further outgoing RTCP packets from 
being sent.

Unfortunately, changing the code to stop RTCP before sending the "TEARDOWN" 
would be messy, so instead I've changed the order of events during shutdown 
to make this race condition less likely.  (In any case, this is only an 
issue when the "-t" option is being used, and occurs only when the program 
is exiting, so it's more of a nuisance than a real problem.)

Your second example is a consequence of the fact that you were (apparently) 
trying to use the "-q" option to output to a non-seekable stream.  Because 
of the way that QuickTime files are created, that won't work.  When using 
the "-q" option, you have to output to a seekable stream - e.g., a file.

         Ross.



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From <EMAIL-ADDRESS-HIDDEN> Thu Apr 3 07:29:58 2003
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X-Original-Date: Thu, 03 Apr 2003 17:14:44 +0200
Date: Thu, 03 Apr 2003 17:14:44 +0200

Hi,

> Your second example is a consequence of the fact that you were
> (apparently) 
> trying to use the "-q" option to output to a non-seekable stream. 
> Because 
> of the way that QuickTime files are created, that won't work.  When
> using 
> the "-q" option, you have to output to a seekable stream - e.g., a file.

I think that I understand what happened. I was trying to stream a movie
that contains a track called "vidéo" (probably created using QT pro
french :) version). As you can see from the strace below, at the end the
file sink tries to _llseek into fd 1(stdout) and the program crashes. It
does work OK when the media type is "video". It is probably a matter of
incorrectly handling a file descriptor table when there is only one
unrecognized track in the movie.

romulus

====================================
 strace openRTSP -e 3 -V  -n -q rtsp://172.16.2.78:8000/ccount.mov

Setup "vidéo/H261" subsession (client ports 32820-32821)
Warning: We don't implement a QuickTime Media Handler for media type
"vidéo", so a track for the "vidéo/H261" subsession will not be included
in the output QuickTime file
Sending request: PLAY rtsp://172.16.2.78:8000/ccount.mov RTSP/1.0
CSeq: 3
.............
write(1, "\0\0\0\0mdat", 8mdat)             = 8
_llseek(1, 0, 0xbfffeb94, SEEK_SET)     = -1 ESPIPE (Illegal seek)
write(2, "QuickTimeFileSink::setWord(): fs"...,
52QuickTimeFileSink::setWord(): fseek failed (err 29)
) = 52
munmap(0x40017000, 4096)                = 0
_exit(1)                                = ?


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>I think that I understand what happened. I was trying to stream a movie
>that contains a track called "vidéo" (probably created using QT pro
>french :) version). As you can see from the strace below, at the end the
>file sink tries to _llseek into fd 1(stdout) and the program crashes.

That's odd.  Do you have a publically-accessible "rtsp://" URL of a stream 
that illustrates this problem?  (I tried accessing the URL that you 
mentioned in your report, but I couldn't reach the host 172.16.2.78)

If that's not possible, could you make available a hinted .mov file (i.e., 
on a web side) that I could try streaming from my own Darwin Streaming 
Server?  (Please do *not* send the file by email, though.)

         Ross.



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From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
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Subject: [Live-devel] A new test program - for streaming DVD content
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X-Original-Date: Fri, 04 Apr 2003 03:07:41 -0800
Date: Fri, 04 Apr 2003 03:07:41 -0800

FYI, a new application - "vobStreamer" - has been added to the "testProgs" 
directory.

"vobStreamer" reads the audio and video tracks from one or more ".vob" 
files (e.g., from a DVD), and streams them - via RTP multicast.

For more information about this application, see: 
<http://www.live.com/vobStreamer/>

	Ross.



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From <EMAIL-ADDRESS-HIDDEN> Fri Apr 4 08:14:49 2003
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From: {Mosher, Joel} <EMAIL-ADDRESS-HIDDEN>
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Subject: [Live-devel] Q: how to convert a quicktime MP4V-ES saved file
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X-Original-Date: Fri, 04 Apr 2003 11:08:51 -0500
Date: Fri, 04 Apr 2003 11:08:51 -0500

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Hi,
 
I have a broadcast that was sent out in Quicktime's implementation of
MPEG-4, and saved it using openRTSP.   However, it says that "A seperate,
codec-specific editing pass will needed before this track can be played."
What tools could be be used for doing that?  I tried home-brewing some using
the quicktime SDK's, but can't see anyway to convert it. 
 
Thanks,
 
Joel Mosher
 
 
 
 

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<DIV><SPAN class=625530716-04042003><FONT face=Arial 
size=2>Hi,</FONT></SPAN></DIV>
<DIV><SPAN class=625530716-04042003><FONT face=Arial 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=625530716-04042003><FONT face=Arial size=2>I&nbsp;have a 
broadcast that was sent out in Quicktime's implementation of MPEG-4, and saved 
it using openRTSP.&nbsp;&nbsp; However, it says that "A seperate, codec-specific 
editing pass will needed before this track can be played."&nbsp;&nbsp; What 
tools could be be used for doing that?&nbsp; I tried home-brewing some using the 
quicktime SDK's, but can't see anyway to convert it. </FONT></SPAN></DIV>
<DIV><SPAN class=625530716-04042003><FONT face=Arial 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=625530716-04042003><FONT face=Arial 
size=2>Thanks,</FONT></SPAN></DIV>
<DIV><SPAN class=625530716-04042003><FONT face=Arial 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=625530716-04042003><FONT face=Arial size=2>Joel 
Mosher</FONT></SPAN></DIV>
<DIV><SPAN class=625530716-04042003><FONT face=Arial 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=625530716-04042003><FONT face=Arial 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><FONT face=Arial size=2></FONT>&nbsp;</DIV>
<DIV>&nbsp;</DIV></BODY></HTML>

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From <EMAIL-ADDRESS-HIDDEN> Fri Apr 4 09:19:02 2003
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From: {Ruth Sadler} <EMAIL-ADDRESS-HIDDEN>
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Subject: [Live-devel] Controlling the streaming rate
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Hi there,
 
I'm using the liveMedia libraries to stream from an encoder producing an
MPEG2 program stream.  My application is based on
testMPEGAudioVideoStreamer - the program stream is read from a
ByteStreamFileSource, the data is then demulitplexed and sent over the
network.  The file is actually a circular buffer written to by the
encoder.
 
Initially everything works great, but over time it seems as though the
data is getting read from the file faster than the encoder is producing
it - eventually the application fails to read from the file because it
has caught up with the data being written by the encoder.  I can't work
out why this is happening - the encoder is producing the program stream
at a constant bitrate, but the RTP side seems to be sending the data
slightly faster.
 
As far as I understand, the rate at which data is read from the file
will depend on how often MultiFramedRTPSink::sendNext is scheduled, and
this depends on calculating when the next frame of data should be sent.
Is this the right place to look to try and solve the problem?  I've
tried using a couple of different ways to calculate this delay, but I
either end up reading from the file still too quickly, or alternatively
too slowly in which case the encoder overwrites some data before I have
a chance to read it.  
 
Any ideas on why this might be happening and what I can do about it
would be appreciated!
 
Thanks,
 
Ruth
 
 

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<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>Hi =
there,<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'><o:p>&nbsp;</o:p></span=
></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>I&#8217;m using the =
<span
class=3DSpellE>liveMedia</span> libraries to stream from an encoder =
producing an
MPEG2 program stream.<span style=3D'mso-spacerun:yes'>&nbsp; </span>My
application is based on <span =
class=3DSpellE>testMPEGAudioVideoStreamer</span> -
the program stream is read from a <span =
class=3DSpellE>ByteStreamFileSource</span>,
the data is then <span class=3DSpellE>demulitplexed</span> and sent over =
the
network. <span style=3D'mso-spacerun:yes'>&nbsp;</span>The file is =
actually a
circular buffer written to by the encoder.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'><o:p>&nbsp;</o:p></span=
></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>Initially everything =
works
great, but over time it seems as though the data is getting read from =
the file faster
than the encoder is producing it &#8211; eventually the application =
fails to
read from the file because it has caught up with the data being written =
by the
encoder.<span style=3D'mso-spacerun:yes'>&nbsp; </span>I can&#8217;t =
work out why
this is happening - the encoder is producing the program stream at a =
constant <span
class=3DSpellE>bitrate</span>, but the RTP side seems to be sending the =
data
slightly faster.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'><o:p>&nbsp;</o:p></span=
></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>As far as I =
understand, the
rate at which data is read from the file will depend on how often <span
class=3DSpellE>MultiFramedRTPSink::sendNext</span> is scheduled, and =
this depends
on calculating when the next frame of data should be sent.<span
style=3D'mso-spacerun:yes'>&nbsp; </span>Is this the right place to look =
to try
and solve the problem?<span style=3D'mso-spacerun:yes'>&nbsp; =
</span>I&#8217;ve
tried using a couple of different ways to calculate this delay, but I =
either
end up reading from the file still too quickly, or alternatively too =
slowly in
which case the encoder overwrites some data before I have a chance to =
read it.<span
style=3D'mso-spacerun:yes'>&nbsp; </span><o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'><o:p>&nbsp;</o:p></span=
></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>Any ideas on why this =
might
be happening and what I can do about it would be =
appreciated!<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'><o:p>&nbsp;</o:p></span=
></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>Thanks,<o:p></o:p></spa=
n></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'><o:p>&nbsp;</o:p></span=
></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>Ruth<o:p></o:p></span><=
/font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'><o:p>&nbsp;</o:p></span=
></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'><o:p>&nbsp;</o:p></span=
></font></p>

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X-Original-Date: Fri, 04 Apr 2003 09:47:04 -0800
Date: Fri, 04 Apr 2003 09:47:04 -0800


>I have a broadcast that was sent out in Quicktime's implementation of 
>MPEG-4, and saved it using openRTSP.   However, it says that "A seperate, 
>codec-specific editing pass will needed before this track can be 
>played."   What tools could be be used for doing that?

One tool that you could try is "Dumpster", which is available (for MacOS 
and Windows) at ftp://ftp.apple.com/developer/Quicktime/

Alternatively, you could try updating "liveMedia/QuickTimeFileSink" to 
handle MPEG-4 codecs.  (Or, if you wish, I could do this for you.  If your 
company is interested in providing funding for this task, please let me 
know by separate email.)

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>



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X-Original-Date: Fri, 04 Apr 2003 10:26:36 -0800
Date: Fri, 04 Apr 2003 10:26:36 -0800


>I m using the liveMedia libraries to stream from an encoder producing an 
>MPEG2 program stream.  My application is based on 
>testMPEGAudioVideoStreamer - the program stream is read from a 
>ByteStreamFileSource, the data is then demulitplexed and sent over the 
>network.  The file is actually a circular buffer written to by the encoder.

Alternatively, you could let your operating system do this for you.  Change 
"testMPEGAudioVideoStreamer.cpp" to read from "stdin" (instead of 
"test.mpg"), and then pipe the output of your encoder to (your modified) 
"testMPEGAudioVideoStreamer".  If you do this, then you don't need to add 
any buffering of your own, and the OS will block your application if it 
runs short of data.

Even better, if your encoder gives you a sequence of frames, rather than a 
sequence of bytes, then you could write your own "FramedSource" subclass 
that encapsulates your encoder, and delivers MPEG frames directly to 
"MPEG*RTPSink" when they are ready (avoiding the need for an intermediate 
'framer' filter that parses the MPEG stream).  This will be a lot more 
efficient.

For a model of how to do that, see "liveMedia/DeviceSource.cpp" (and 
"liveMedia/include/DeviceSource.hh").  You will need to fill in parts of 
this code to do the actual read from your encoder.

>  Initially everything works great, but over time it seems as though the 
> data is getting read from the file faster than the encoder is producing 
> it eventually the application fails to read from the file because it has 
> caught up with the data being written by the encoder.  I can t work out 
> why this is happening - the encoder is producing the program stream at a 
> constant bitrate, but the RTP side seems to be sending the data slightly 
> faster.

How slightly is "slightly"? :-)  The interval between successive outgoing 
frames is calculated using "MPEGVideoStreamFramer::getPlayTime()" and 
"MPEGAudioStreamFramer::getPlayTime()".  It's possible that, over time, 
(because of truncation or roundoff error), this could cause the RTP output 
to overrun your buffer.  But the real problem here, I think, is the way 
you've written your application.  If it finds that no frame is immediately 
available, then it should just block until the next one becomes 
available.  Either of the methods I described above will do this.

         Ross.



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X-Original-Date: Sat, 5 Apr 2003 15:23:49 +0800 (CST)
Date: Sat, 5 Apr 2003 15:23:49 +0800 (CST)

Dear all,
I run 
./openRTSP rtsp://172.16.6.7:19999/sample_100kbit.mov
The movie is streamed by apple darwin streaming
server, but I get the following reponse:

Opened URL
"rtsp://172.16.6.7:19999/sample_100kbit.mov",
returning a SDP description:
v=0
o=StreamingServer 3258516238 1025914806000 IN IP4
172.16.6.7
s=/sample_100kbit.mov
u=http:///
e=admin(at)
c=IN IP4 0.0.0.0
t=0 0
a=control:*
b=AS:94
a=range:npt=0-  66.47000
m=audio 0 RTP/AVP 96
b=AS:10
a=rtpmap:96 X-QDM/22050/2
a=control:trackID=3
a=x-bufferdelay:5.99
m=video 0 RTP/AVP 97
b=AS:84
a=rtpmap:97 X-SV3V-ES/90000
a=control:trackID=4

Unable to create receiver for "audio/X-QDM"
subsession: RTP payload format unknown or not
supported
Unable to create receiver for "video/X-SV3V-ES"
subsession: RTP payload format unknown or not
supported

Why? How could I modify OpenRTSP to paly .mov? 

Thx.
Tak

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X-Original-Date: Sat, 05 Apr 2003 02:05:59 -0800
Date: Sat, 05 Apr 2003 02:05:59 -0800


>Unable to create receiver for "audio/X-QDM"
>subsession: RTP payload format unknown or not
>supported
>Unable to create receiver for "video/X-SV3V-ES"
>subsession: RTP payload format unknown or not
>supported
>
>Why?

A "RTP payload format" is the set of rules that describe how each codec's 
data is packed into RTP packets.

In this case, the stream contains two Apple-specific codecs whose RTP 
payload formats are proprietary, and I don't know what they are.

>How could I modify OpenRTSP to paly .mov?

One thing you could try is use the option
         -s 0

Of course, you probably won't be able to *play* the output files that 
"openRTSP" generates.  To play this "rtsp://" URL, you will need to use 
QuickTime Player.

         Ross.



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X-Original-Date: Sun, 6 Apr 2003 17:25:18 +0800 (CST)
Date: Sun, 6 Apr 2003 17:25:18 +0800 (CST)

Dear all,
I add the live source tree into apple's Project
Builder. I want to create a target based on OpenRTSP
and PlayCommon. In Targets, I add search path to all
include folder, but results to Build failed.
The error is
undefined sysbols: _SECOND _our_inet_our_random
Could anyone help to solve my problem?
How could I compile the whole live source in Project
Builder ?
thx
tak



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X-Original-Date: Sun, 06 Apr 2003 15:56:00 -0700
Date: Sun, 06 Apr 2003 15:56:00 -0700


>I add the live source tree into apple's Project
>Builder.

I presume you are doing this in Mac OS X.  (If you have an earlier version 
of Mac OS, then it's unlikely that the code could be built for it - at 
least, not without a lot of work.)

Please follow the instructions at 
<http://www.live.com/liveMedia/#config-unix>, and build the code on Mac OS 
the correct way (beginning with "genMakefiles macos").  Only then should 
you even think about trying to build the code any other way.

         Ross.



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Date: Sun, 06 Apr 2003 16:02:56 -0700

A small correction to my earlier message:

The config file for Mac OS X is "config.macosx".  So, to build the code on 
Mac OS X, begin with
	genMakefiles macosx

     Ross.



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Dear all,

I found it difficult to learn about the flow of OpenRTSP. Could anyone provide me the class diagram or flow char of the OpenRTSP program? Or, could anyone tell me the flow of events?

Thx

Tak


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<P>Dear all,</P>
<P>I found it&nbsp;difficult&nbsp;to learn about the flow of OpenRTSP. Could anyone provide me the class diagram or flow char of the OpenRTSP program? Or, could anyone tell me the flow of events?</P>
<P>Thx</P>
<P>Tak</P><p><hr size=1><b>Do You Yahoo!?</b><br>µn°O§K¶Oªº (at)yahoo.com ¤¤¤å¹q¤l¶l¥ó (at) <a href="http://chinese.mail.yahoo.com/">Yahoo! Mail</a>.<br>Get your free (at)yahoo.com address at <a href="http://chinese.mail.yahoo.com/">Yahoo! Mail</a>.<br>
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X-Original-Date: Tue, 8 Apr 2003 09:33:37 +0100
Date: Tue, 8 Apr 2003 09:33:37 +0100

Thanks for the suggestion, I have tried piping the output of the encoder
to stdin (it is a sequence of bytes rather than frames) but this causes
a new problem...

I think the MultiFramedRTPSink calculates when to schedule the next read
based on working out the time the current frame will take to play
(getPlayTime), adding that to the time the previous frame should have
been played, and working out the difference between that time and the
current time.  But blocking on the read means the current time increases
a bit, so the next packet is scheduled to be sent sooner.  It ends up in
the situation where the current time is ahead of the time we want to
send the packet so the "sendNext" function is always scheduled with 0
timeout.  

This is fine for the RTP because video and audio are being sent as
quickly as they arrive from the encoder, but the task scheduler is
constantly scheduling the "sendNext", which means that any RTSP requests
to the server don't get serviced because the TaskScheduler doesn't drop
out of the handleAlarm() function.  So no new clients can join!

I think maybe as well as blocking waiting for data from the encoder I
should insert some dummy task in the task scheduler to tell it to drop
out of the handleAlarm() loop if there is no data available to read
right now..  Does that sound like it would break anything, and if so, do
you have any other suggestions about how to handle this situation?

Thanks,

Ruth
 

-----Original Message-----
From: <EMAIL-ADDRESS-HIDDEN>
[mailto:live-devel-admin(at)lists.sourceforge.net] On Behalf Of Ross
Finlayson
Sent: 04 April 2003 19:27
To: <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] Controlling the streaming rate


>I m using the liveMedia libraries to stream from an encoder producing
an 
>MPEG2 program stream.  My application is based on 
>testMPEGAudioVideoStreamer - the program stream is read from a 
>ByteStreamFileSource, the data is then demulitplexed and sent over the 
>network.  The file is actually a circular buffer written to by the
encoder.

Alternatively, you could let your operating system do this for you.
Change 
"testMPEGAudioVideoStreamer.cpp" to read from "stdin" (instead of 
"test.mpg"), and then pipe the output of your encoder to (your modified)

"testMPEGAudioVideoStreamer".  If you do this, then you don't need to
add 
any buffering of your own, and the OS will block your application if it 
runs short of data.

Even better, if your encoder gives you a sequence of frames, rather than
a 
sequence of bytes, then you could write your own "FramedSource" subclass

that encapsulates your encoder, and delivers MPEG frames directly to 
"MPEG*RTPSink" when they are ready (avoiding the need for an
intermediate 
'framer' filter that parses the MPEG stream).  This will be a lot more 
efficient.

For a model of how to do that, see "liveMedia/DeviceSource.cpp" (and 
"liveMedia/include/DeviceSource.hh").  You will need to fill in parts of

this code to do the actual read from your encoder.

>  Initially everything works great, but over time it seems as though
the 
> data is getting read from the file faster than the encoder is
producing 
> it eventually the application fails to read from the file because it
has 
> caught up with the data being written by the encoder.  I can t work
out 
> why this is happening - the encoder is producing the program stream at
a 
> constant bitrate, but the RTP side seems to be sending the data
slightly 
> faster.

How slightly is "slightly"? :-)  The interval between successive
outgoing 
frames is calculated using "MPEGVideoStreamFramer::getPlayTime()" and 
"MPEGAudioStreamFramer::getPlayTime()".  It's possible that, over time, 
(because of truncation or roundoff error), this could cause the RTP
output 
to overrun your buffer.  But the real problem here, I think, is the way 
you've written your application.  If it finds that no frame is
immediately 
available, then it should just block until the next one becomes 
available.  Either of the methods I described above will do this.

         Ross.



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>This is fine for the RTP because video and audio are being sent as
>quickly as they arrive from the encoder, but the task scheduler is
>constantly scheduling the "sendNext", which means that any RTSP requests
>to the server don't get serviced because the TaskScheduler doesn't drop
>out of the handleAlarm() function.  So no new clients can join!

Yes, good point.

>I think maybe as well as blocking waiting for data from the encoder I
>should insert some dummy task in the task scheduler to tell it to drop
>out of the handleAlarm() loop if there is no data available to read
>right now..  Does that sound like it would break anything, and if so, do
>you have any other suggestions about how to handle this situation?

I think a better (and much simpler solution) would be to change the event 
loop 'SingleStep' from
         select
         handle all delayed events that are now due
         call any handlers for sockets with readable data
to
         select
         handle *the first* delayed event (if any) that is now due
         call any handlers for sockets with readable data

You can do this by changing the "while" in "DelayQueue::handleAlarm()" 
(line 206) to an "if".

Please give this a try, and let me know if it makes the RTSP server more 
responsive.

         Ross.



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X-Original-Date: Tue, 8 Apr 2003 18:02:32 +0800 (CST)
Date: Tue, 8 Apr 2003 18:02:32 +0800 (CST)

Dear all,
when I use openRTSP to get video?
Is one port used for receving video and one port used
for receving audio?
if yes, how could two sockets receving data and then
output to one media file? Where is the program code
stating how to do that?
How could I know in program code the ending of getting
packets?
Thx 
Tak

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Subject: RE: [Live-devel] Controlling the streaming rate
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X-Original-Date: Tue, 8 Apr 2003 13:16:37 +0100
Date: Tue, 8 Apr 2003 13:16:37 +0100

Yes, replacing the "while" with an "if" in DelayQueue::handleAlarm()
makes the RTSP server more responsive in the situation I described.

Thanks for the help,

Ruth

-----Original Message-----
From: <EMAIL-ADDRESS-HIDDEN>
[mailto:live-devel-admin(at)lists.sourceforge.net] On Behalf Of Ross
Finlayson
Sent: 08 April 2003 10:01
To: <EMAIL-ADDRESS-HIDDEN>
Subject: RE: [Live-devel] Controlling the streaming rate


>This is fine for the RTP because video and audio are being sent as
>quickly as they arrive from the encoder, but the task scheduler is
>constantly scheduling the "sendNext", which means that any RTSP
requests
>to the server don't get serviced because the TaskScheduler doesn't drop
>out of the handleAlarm() function.  So no new clients can join!

Yes, good point.

>I think maybe as well as blocking waiting for data from the encoder I
>should insert some dummy task in the task scheduler to tell it to drop
>out of the handleAlarm() loop if there is no data available to read
>right now..  Does that sound like it would break anything, and if so,
do
>you have any other suggestions about how to handle this situation?

I think a better (and much simpler solution) would be to change the
event 
loop 'SingleStep' from
         select
         handle all delayed events that are now due
         call any handlers for sockets with readable data
to
         select
         handle *the first* delayed event (if any) that is now due
         call any handlers for sockets with readable data

You can do this by changing the "while" in "DelayQueue::handleAlarm()" 
(line 206) to an "if".

Please give this a try, and let me know if it makes the RTSP server more

responsive.

         Ross.



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X-Original-Date: Tue, 08 Apr 2003 15:43:58 +0200
Date: Tue, 08 Apr 2003 15:43:58 +0200

Hi,

We would like to have an multithreaded application that uses the library
for controlling and receiving the streams. We have at least two streams
(live and background) that need to be managed concurently from two
separate threads. These streams fill up some kind of cache in memory
using the received RTP packets. The streams need to be paused or
restarted many times.

We also need to be able to manage and receive the two separate streams
corresponding to different time ranges of the same media track. For
example, while the user is viewing the seconds 10-30 of a track, we want
to prefetch the seconds 40-50, because we predict it is likely that he
jumps to that segment.

Ideally we would have the main application thread (GUI thread), a
thread  for prediction and another for receiving the packets. The GUI
and prediction threads would handle the RTSP requests by themselves.

As mentioned in the docs, the library follows a single-threaded,
event-driven execution model, rather than a multi-threaded concurrent
execution model. But how can we use the library for our purpose ?

One "quick" way to solve the problem would be to wrap up a separate
environment/scheduler/client in each thread that manages one stream. As
the event loop blocks on the select, we would need to select on an extra
file descriptor in order to process any aditional RTSP commands after
starting receiving RTP packets. The main thread would just write to that
file descriptor to wake up the stream thread.

The problem that remains is that processing an RTSP command and
especially receiving the responses make the current thread block and
therefore RTP packets are not consumed. A solution to this second
problem would be to modify RTSPClient and do select instead of recvfrom.
The state of successive RTSP request needs to be managed separately,
though.

--------------
More generally, what would be the best way to have multiple streams
handled separately by the library, even in a single-threaded model ?

We found it difficult to handle separately multiple media subsessions so
we thought to have separate clients for the streams.

Now, is it possible to use the same environment for having completely
separate clients ? This would allow us to receive packets for various
clients/streams in the same place.

We noticed that openRTSP example uses a lot of global variables, so
everything needs to be packed together.

Thanks for any hints on this,

romulus


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X-Original-Date: Tue, 08 Apr 2003 09:56:45 -0700
Date: Tue, 08 Apr 2003 09:56:45 -0700

At 03:02 AM 4/8/03, you wrote:
>Dear all,
>when I use openRTSP to get video?
>Is one port used for receving video and one port used
>for receving audio?

Yes, because that's the way that the RTP streams are sent (usually).

>if yes, how could two sockets receving data and then
>output to one media file? Where is the program code
>stating how to do that?

It depends on what kind of output file format you want.  If you want a 
QuickTime (i.., '.mov') format file, then you can use the "-q" option, 
which uses a "QuickTimeFileSink" object.

For any other type of file, you will need to write your own 'sink' class to 
implement this.

         Ross.



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Subject: [Live-devel] mplayer and rtp mcast video sources
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X-Original-Date: Tue, 8 Apr 2003 14:05:11 -0500 (CDT)
Date: Tue, 8 Apr 2003 14:05:11 -0500 (CDT)

Hi,

Excuse me if this belongs on the MPlayer mailing list. I'm a little unsure
where to post. I've been experimenting with live.com and mplayer as a
possible soltion for streaming very high quality live video over 
multicast. With the live.com rtsp/rtp streaming support for mplayer I can
watch a single video stream fine, using the sdp option. But when I try to
watch a channel where there are multiple (dozens) of video feeds, it
returns:

Playing ag-lobby.sdp
Initiated "video/H261" RTP subsession
demux_rtp: Guessed the video frame rate as 0 frames-per-second.
        (If this is wrong, use the "-fps <frame-rate>" option instead.)
FPS not specified in the header or invalid, use the -fps option.
No stream found.

and then freezes. 

Am I doing something wrong? If not, is this something that pertains to
the live.com libraries, or just the rtsp/rtp support for MPlayer. Also, if
I add say -fps 1, it yells about the h261xa codec, but thats listed in 
codecs.conf and I've watched h261xa feeds before.

On the todo list at http://www.live.com/liveMedia/ it mentions supporting
more codecs. Is that the problem?

Sorry about this long diverging email :)
Chris



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X-Original-Date: Tue, 08 Apr 2003 12:53:27 -0700
Date: Tue, 08 Apr 2003 12:53:27 -0700

Chris,

Unfortunately multiple video sources are not yet supported.  First, the 
LIVE.COM library currently doesn't demultiplex incoming RTP packets based 
on SSRC.  Although that could probably be fixed without too much trouble, 
the more serious problem is that even if the library *could* demultiplex 
multiple video streams, MPlayer would have no way to display them!  I.e., 
MPlayer currently doesn't have the ability to either (i) pop up separate 
display windows for >1 video stream, or (ii) 'tile' multiple video streams 
in a single display window.

Personally, I doubt that MPlayer could be hacked to do either of these, 
without a *lot* of work.  An easier solution might be to simulate (i) by 
forking off new instances of MPlayer (with the "-nosound") option each time 
the 'base' MPlayer sees a new SSRC.

Also, I haven't personally tested playing H.261 video streams.  Did you 
find that you were able to play a single-source H.261 stream OK?  Or did 
H.261 not work at all for you?

	Ross.



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Subject: Re: [Live-devel] mplayer and rtp mcast video sources
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X-Original-Date: Tue, 8 Apr 2003 16:18:04 -0500 (CDT)
Date: Tue, 8 Apr 2003 16:18:04 -0500 (CDT)

Ross, 

H261 might not work correctly, at least I haven't gotten it working for
me. Correct me if I'm wrong but most of the RTP formats listed in
liveMedia/MediaSession.cpp are just place holders, and are not actually
supported, right? At least thats the error messages I get when I give
MPlayer an .sdp file. For example, m=video 8888 RTP/AVP 34 describes a
H263 session, and MPlayer complains that its not a suppported RTP payload
format. The only format I've been able to successfully stream/receive is
MPV, or case 32. Some, such as JPEG and H261 just wait forever, even
though I'm sending stuff in those formats with VIC to the specified
multicast address. Are these formats not working? Are there any other
formats besides MPV that you know work?

Thanks,
Chris



On Tue, 8 Apr 2003, Ross Finlayson wrote:

> Chris,
> 
> Unfortunately multiple video sources are not yet supported.  First, the 
> LIVE.COM library currently doesn't demultiplex incoming RTP packets based 
> on SSRC.  Although that could probably be fixed without too much trouble, 
> the more serious problem is that even if the library *could* demultiplex 
> multiple video streams, MPlayer would have no way to display them!  I.e., 
> MPlayer currently doesn't have the ability to either (i) pop up separate 
> display windows for >1 video stream, or (ii) 'tile' multiple video streams 
> in a single display window.
> 
> Personally, I doubt that MPlayer could be hacked to do either of these, 
> without a *lot* of work.  An easier solution might be to simulate (i) by 
> forking off new instances of MPlayer (with the "-nosound") option each time 
> the 'base' MPlayer sees a new SSRC.
> 
> Also, I haven't personally tested playing H.261 video streams.  Did you 
> find that you were able to play a single-source H.261 stream OK?  Or did 
> H.261 not work at all for you?
> 
> 	Ross.
> 
> 
> 
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X-Original-Date: Tue, 08 Apr 2003 15:35:20 -0700
Date: Tue, 08 Apr 2003 15:35:20 -0700


>H261 might not work correctly, at least I haven't gotten it working for
>me. Correct me if I'm wrong but most of the RTP formats listed in
>liveMedia/MediaSession.cpp are just place holders, and are not actually
>supported, right?

Every MIME type that's checked in "MediaSubsession::initiate()" (using a 
"strcmp()" call) is supported.  (However, not all of the static RTP payload 
formats listed in "MediaSession::lookupPayloadFormat()" are supported.)

>  At least thats the error messages I get when I give
>MPlayer an .sdp file. For example, m=video 8888 RTP/AVP 34 describes a
>H263 session, and MPlayer complains that its not a suppported RTP payload
>format.

That's correct.  The payload format for H.263 - as described in RFC 2190 - 
is currently not supported.  However, the newer payload format for 
H.263-1998 (aka. H.263+) - as defined by RFC 2429 - *is* supported.

>  The only format I've been able to successfully stream/receive is
>MPV, or case 32. Some, such as JPEG and H261 just wait forever, even
>though I'm sending stuff in those formats with VIC to the specified
>multicast address. Are these formats not working? Are there any other
>formats besides MPV that you know work?

Besides "MPV" (MPEG-1 or 2 Elementary Stream video), the other video RTP 
formats that are known to work are "MP4V-ES" (MPEG-4 Elementary Stream), 
"H263-1998" (i.e., H.263+), and "JPEG" (motion-JPEG).

"H261" is also supported by the LIVE.COM libraries; however, the 
corresponding codec implementation in MPlayer seems to have problems (in 
particular, it seems to need to be spoon-fed the video width and height, 
and likely has other problems also).

I'm surprised to hear you say that motion-JPEG is not working for 
you.  This (and MPEG-4 video, and H.263+) requires the "FFmpeg libavcodec" 
package, so make sure that you have this installed (in the "libavcodec/" 
subdirectory) before you run "configure" on the MPlayer source.

If you've installed the "FFmpeg libavcodec" sources correctly, and MPlayer 
still fails to play a motion-JPEG RTP stream, then please put one up on the 
MBone and send me the SDP file that describes it, and I'll take a look at 
it.  (I'm wondering if perhaps there's some incompatibility between the RTP 
payload format sent by "vic" for motion-JPEG, and my own implementation 
(which is supposed to follow RFC 2435).

Similarly, if you were to put a "H263" stream up on the MBone (& send me a 
SDP file), I could look into supporting the payload format for it (RFC 2190).

         Ross.



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X-Original-Date: Thu, 10 Apr 2003 17:12:28 +0800
Date: Thu, 10 Apr 2003 17:12:28 +0800

Hi!  I have configured and make live source for red hat linux 7.3.  Now, 
I would like to use the test programs starting with 
testMPEGVideoStreamer/Receiver and then testMP3Streamer/Receiver.  How 
do I configure or use them disabling multicast and RTSP?
Thanks in advance.

Mildred








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Subject: Re: [Live-devel] how to configure/use the test programs without
 using multicast and rtsp
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X-Original-Date: Thu, 10 Apr 2003 10:47:15 -0500 (CDT)
Date: Thu, 10 Apr 2003 10:47:15 -0500 (CDT)

Mildred,

Sorry, but I'm not quite sure what you mean. testMPEG*Streamer/Receiver
stream a file across multicast/RTSP and read it. Are you trying 
to test the video/audio sinks? Off hand I do not know of any 
multicast/RTSP address that would specify a local stream. Or do you mean 
that you are trying to use a different protocol than RTP/RTSP for 
streaming?

Sorry I'm not much help,
Chris


On Thu, 10 Apr 2003, Mildred L. Frisco wrote:

> Hi!  I have configured and make live source for red hat linux 7.3.  Now, 
> I would like to use the test programs starting with 
> testMPEGVideoStreamer/Receiver and then testMP3Streamer/Receiver.  How 
> do I configure or use them disabling multicast and RTSP?
> Thanks in advance.
> 
> Mildred
> 
> 
> 
> 
> 
> 
> 
> 
> -------------------------------------------------------
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X-Original-Date: Thu, 10 Apr 2003 10:47:15 -0500 (CDT)
Date: Thu, 10 Apr 2003 10:47:15 -0500 (CDT)

Mildred,

Sorry, but I'm not quite sure what you mean. testMPEG*Streamer/Receiver
stream a file across multicast/RTSP and read it. Are you trying 
to test the video/audio sinks? Off hand I do not know of any 
multicast/RTSP address that would specify a local stream. Or do you mean 
that you are trying to use a different protocol than RTP/RTSP for 
streaming?

Sorry I'm not much help,
Chris


On Thu, 10 Apr 2003, Mildred L. Frisco wrote:

> Hi!  I have configured and make live source for red hat linux 7.3.  Now, 
> I would like to use the test programs starting with 
> testMPEGVideoStreamer/Receiver and then testMP3Streamer/Receiver.  How 
> do I configure or use them disabling multicast and RTSP?
> Thanks in advance.
> 
> Mildred
> 
> 
> 
> 
> 
> 
> 
> 
> -------------------------------------------------------
> This SF.net email is sponsored by: Etnus, makers of TotalView, The debugger 
> for complex code. Debugging C/C++ programs can leave you feeling lost and 
> disoriented. TotalView can help you find your way. Available on major UNIX 
> and Linux platforms. Try it free. www.etnus.com
> _______________________________________________
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> Live-devel(at)lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/live-devel
> 
> 



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X-Original-Date: Thu, 10 Apr 2003 09:34:54 -0700
Date: Thu, 10 Apr 2003 09:34:54 -0700

At 02:12 AM 4/10/03, you wrote:
>Hi!  I have configured and make live source for red hat linux 7.3.  Now, I 
>would like to use the test programs starting with 
>testMPEGVideoStreamer/Receiver and then testMP3Streamer/Receiver.  How do 
>I configure or use them disabling multicast and RTSP?

The RTSP server that's built into "testMP3Streamer" and 
"testMPEGVideoStreamer" is already disabled, by default.

Why would you not want to stream using multicast?

         Ross.



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X-Original-Date: Fri, 11 Apr 2003 10:49:58 +0800
Date: Fri, 11 Apr 2003 10:49:58 +0800


Hi!  Is there a documentation for the live.com libraries/stacks ?
thanks

mildred







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X-Original-Date: Thu, 10 Apr 2003 20:14:39 -0700
Date: Thu, 10 Apr 2003 20:14:39 -0700


>Hi!  Is there a documentation for the live.com libraries/stacks ?

Apart from the web site <http://www.live.com/liveMedia/>, no.

For now, the best way to understand how to use the libraries is to (i) 
study the example programs in the "testProgs" directory, (ii) study the 
library code itself, and (especially)  (iii) ask lots of questions on this 
mailing list

Unfortunately there's no additional documentation right now (more 
documentation is on the "to do" list, but features requested by paying 
consulting clients currently take priority).

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>



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X-Original-Date: Thu, 10 Apr 2003 21:51:19 -0700
Date: Thu, 10 Apr 2003 21:51:19 -0700

A few weeks ago, there was a question on this mailing list about how to 
play access-controlled RTSP sessions using MPlayer.

FYI, the latest version of the LIVE.COM and MPlayer code will now let you 
do this.  See <http://www.live.com/mplayer/#access-control> for more 
information.

	Ross.



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X-Original-Date: Fri, 11 Apr 2003 02:38:16 -0700
Date: Fri, 11 Apr 2003 02:38:16 -0700

As you noted, the library code was designed to be executed as a single 
thread of control, using an event loop (with asynchronous event handlers) 
for concurrency.  Although this execution model has some drawbacks, 
compared to the alternative (multiple threads with blocking I/O), it makes 
the code easier to debug, and more portable.  (It's difficult to write 
threaded code that's portable across both Unix and Windows.)

John Ousterhout has given an excellent presentation (see 
<http://home.pacbell.net/ouster/>, under "Miscellaneous Stuff") in which he 
makes the same argument (although perhaps a bit stronger than I 
would).  Personally, if I had been writing this software only for myself, 
and only for one OS, then I might have used a threading model (as I've been 
programming with threads for 20 years!).  But for this software, the 
event-driven model seems to be a good choice.

>More generally, what would be the best way to have multiple streams
>handled separately by the library, even in a single-threaded model ?

This is actually quite straightforward - and, in fact, "openRTSP" does this 
already (in the general case in which a RTSP stream consists of separate 
incoming RTP streams for audio and video).  It calls "startPlaying()" on 
each sink (to start the stream), and then calls "doEventLoop()".  Each 
event handler identifies the stream that it is handling (indirectly, via 
the "clientData" parameter).

Alternatively, if your multiple streams are processed mostly independent of 
each other, then you could also use separate threads to handle them.  If 
you do this, though, you will have to use separate "UsageEnvironment"s and 
"TaskScheduler"s for each thread.  (This should work, although the library 
code might still have a few lingering shared variables that may cause 
problems.  I'll need to look into this.)

The other point you noted was the fact that "RTSPClient" operations (such 
as "DESCRIBE") are blocking, and potentially long-lasting.  Thus, the 
"RTSPClient" implementation actually does not follow the event-driven 
model, and at some poiint I should probably rewrite it so that it does.

This is not necessarily a problem, though, because any incoming packets 
that arrive during a blocking "RTSPClient" operation will just get buffered 
in the OS, and will be available to be handled once the event loop is 
re-entered later.  Just be sure that you have enough OS buffering for each 
network socket.  By default, each "RTPSource" is given 50 kBytes of OS 
buffer space, but you can increase this if you wish, using the 
"increaseReceiveBufferTo()" operation.

         Ross.



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Subject: Re: [Live-devel] Size of Movie?
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X-Original-Date: Fri, 11 Apr 2003 02:59:41 -0700
Date: Fri, 11 Apr 2003 02:59:41 -0700

At 02:02 AM 3/26/03, you wrote:
>How can I get the size of the Movie? I meen how long is
>the RTSP MPEG4 Movie? Is there any ready function in
>LIVE.COM library or at least any way to get it from RTSP
>source?

I can't remember if anyone answered this already, but one thing you can do, 
once you've opened the RTSP URL (using "DESCRIBE") is call
         session->playEndTime()
to get the session's playing time (in seconds).  (See, for example, 
"playCommon.cpp", which implements "openRTSP".)

Of course, this works only if the session's SDP description (returned by 
the "DESCRIBE") contains an appropriate "a=range ..." attribute.  (So, for 
example, it usually won't work on live streams - only prerecorded streams.)

         Ross.



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Subject: [Live-devel] Slightly off topic: Boost.Threads
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X-Original-Date: Sat, 12 Apr 2003 18:03:37 +1000
Date: Sat, 12 Apr 2003 18:03:37 +1000

On Sat, 12 Apr 2003 01:01 pm, live-devel-request(at)lists.sourceforge.net wrote:

> Message: 3
> Date: Fri, 11 Apr 2003 02:38:16 -0700
> To: live-devel(at)lists.sourceforge.net
> From: Ross Finlayson <finlayson(at)live.com>
> Subject: Re: [Live-devel] multithreading
> Reply-To: live-devel(at)lists.sourceforge.net
>
> As you noted, the library code was designed to be executed as a single
> thread of control, using an event loop (with asynchronous event handlers)
> for concurrency.  Although this execution model has some drawbacks,
> compared to the alternative (multiple threads with blocking I/O), it makes
> the code easier to debug, and more portable.  (It's difficult to write
> threaded code that's portable across both Unix and Windows.)

Hi Ross,

This is a bit broad and off topic, but in terms of making threads portable 
across platforms, the Boost library for C++ looks good.

http://www.boost.org
http://www.boost.org/libs/thread/doc/index.html

Is this what you would use to enable porting of threaded apps or libraries 
across Unix/Linux/Windows?

Alex


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From <EMAIL-ADDRESS-HIDDEN> Sat Apr 12 10:26:18 2003
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Subject: [Live-devel] rtcp question
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X-Original-Date: Sun, 13 Apr 2003 01:20:32 +0800 (CST)
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Dear all,I create one pair rtp and rtcp socket to receive audio and one port pair to receive video data from dawrin server. I could only receive rtcp packets from audio rtcp socket but not video rtcp socket. Is it possible? Is the use of rtp and rtcp port of client is negotiated in rtsp setup method?How could I create different threads to run OpenRTSP at the same time?ThxTak

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<DIV>Dear all,</DIV>
<DIV>I create one pair rtp and rtcp socket to receive audio and one port pair to receive video data from dawrin server. I could only receive rtcp packets from audio rtcp socket but not video rtcp socket. Is it possible? Is the use of rtp and rtcp port of client is negotiated in rtsp setup method?</DIV>
<DIV>How could I create different threads to run OpenRTSP at the same time?</DIV>
<DIV>Thx</DIV>
<DIV>Tak</DIV><p><hr size=1><b>Do You Yahoo!?</b><br>µn°O§K¶Oªº (at)yahoo.com ¤¤¤å¹q¤l¶l¥ó (at) <a href="http://chinese.mail.yahoo.com/">Yahoo! Mail</a>.<br>Get your free (at)yahoo.com address at <a href="http://chinese.mail.yahoo.com/">Yahoo! Mail</a>.<br>
--0-1739205614-1050168032=:12101--


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From <EMAIL-ADDRESS-HIDDEN> Sat Apr 12 19:38:07 2003
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X-Original-Date: Sat, 12 Apr 2003 19:27:48 -0700
Date: Sat, 12 Apr 2003 19:27:48 -0700


>I create one pair rtp and rtcp socket to receive audio and one port pair 
>to receive video data from dawrin server. I could only receive rtcp 
>packets from audio rtcp socket but not video rtcp socket. Is it possible?

Yes, receiving/sending RTCP packets for more than one streams should 
work.  In fact, "openRTSP" does this.

>  Is the use of rtp and rtcp port of client is negotiated in rtsp setup 
> method?

Yes.

>How could I create different threads to run OpenRTSP at the same time?

You can't.  "openRTSP" (and the LIVE.COM libraries in general) are 
single-threaded.  Note the message that I sent about this a couple of days ago.

         Ross.



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From <EMAIL-ADDRESS-HIDDEN> Sun Apr 13 15:45:17 2003
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Subject: [Live-devel] liveMedia questions ...
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X-Original-Date: Sun, 13 Apr 2003 15:39:09 -0700
Date: Sun, 13 Apr 2003 15:39:09 -0700


Hello, Ross et al ...

I've a few questions regarding LiveMedia.  These are at a higher level 
than the source itself, but still development-related.  If someone 
could illuminate me, i'd be quite appreciative.

1. Does LiveMedia include a multiplexer a.k.a. reactor a.k.a. Task 
model to assist in building a highly scalable streaming server?  For 
instance, Darwin Streaming Server implements a Task model (based on 
modwatch() on Mac OSX / Darwin, select() on other *nixes), the ACE 
project implements the Reactor pattern, etc.  These all basically allow 
one to handle hundreds or thousands of simultaneous clients (players) 
on a single server thread.  Does LiveMedia include something like this, 
or does a server application developer need to do this themselves?

2. Can LiveMedia stream MPEG2 -- more specifically, can it stream a DVD 
.VOB file?  What about a .VOB file with, say, a THX / AC3 elementary 
stream (assuming the client player can select that stream).

I've studied the DSS source a lot, i've looked at Live a few times now, 
but haven't really dug into it deeply yet.  I'm currently looking at 
the VideoLAN project at work, but it does not implement an RTSP server, 
so I would need to build one.  Basically, i'm looking for the easiest / 
quickest way to build a highly scalable streaming server / streaming 
client that can stream a ripped DVD .VOB, possibly with THX / ACE 
audio, file over IP multicast using RTP / RTSP.

Regards,

John

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X-Original-Date: Sun, 13 Apr 2003 16:28:44 -0700
Date: Sun, 13 Apr 2003 16:28:44 -0700


>1. Does LiveMedia include a multiplexer a.k.a. reactor a.k.a. Task model 
>to assist in building a highly scalable streaming server?  For instance, 
>Darwin Streaming Server implements a Task model (based on modwatch() on 
>Mac OSX / Darwin, select() on other *nixes), the ACE project implements 
>the Reactor pattern, etc.  These all basically allow one to handle 
>hundreds or thousands of simultaneous clients (players) on a single server 
>thread.  Does LiveMedia include something like this, or does a server 
>application developer need to do this themselves?

Yes, I think liveMedia's single-threaded event-loop model (using 
"UsageEnvironment::doEventLoop()") implements exactly this model.

>2. Can LiveMedia stream MPEG2 -- more specifically, can it stream a DVD 
>.VOB file?  What about a .VOB file with, say, a THX / AC3 elementary stream

Ahh, you must have missed my earlier email announcing the "vobStreamer" 
application - which does just this.  See <http://www.live.com/vobStreamer/>

         Ross.



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X-Original-Date: Wed, 16 Apr 2003 16:03:44 -0700
Date: Wed, 16 Apr 2003 16:03:44 -0700

Hi,

I am trying to stream MP3 using MP3Streamer to Mplayer with the Live
library. I need to find the function in the  LIVE library where the MP3
frame is received? Basically, I need to know the times at which MP3 frames
are received.

Can somebody please tell me this?

Thanks,
Rohit



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Date: Wed, 16 Apr 2003 16:53:03 -0700


>I am trying to stream MP3 using MP3Streamer to Mplayer with the Live
>library.

I also had someone else from UCLA asking about this earlier today.  You 
should first make sure that you're using up-to-date versions of the 
software: the latest version of the "LIVE.COM Streaming Media" code (from 
<http://www.live.com/liveMedia/public/>), and (especially) the "latest CVS 
snapshot" version of MPlayer: 
<http://www1.mplayerhq.hu/MPlayer/cvs/MPlayer-current.tar.bz2>.

>I need to find the function in the  LIVE library where the MP3
>frame is received? Basically, I need to know the times at which MP3 frames
>are received.

See "liveMedia/MultiFramedRTPSource.cpp".  (Search for "gettimeofday".)

         Ross.



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X-Original-Date: Thu, 17 Apr 2003 21:07:18 +0800
Date: Thu, 17 Apr 2003 21:07:18 +0800


i saw that in live.com allows http sink...
how can we implement httpsink ?

does it mean that we can stream videos via the use of http ?
if it does, how can we implement it ?
and we can open up the video streams using any movie player
using http://url

please enlighten me with this problem i would appreciate if someone can 
guide me along
thanks



_________________________________________________________________
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X-Original-Date: Thu, 17 Apr 2003 10:10:12 -0700
Date: Thu, 17 Apr 2003 10:10:12 -0700


>i saw that in live.com allows http sink...
>how can we implement httpsink ?
>
>does it mean that we can stream videos via the use of http ?
>if it does, how can we implement it ?
>and we can open up the video streams using any movie player
>using http://url

"HTTPSink" is a simple, 'bare-bones' HTTP server that allows the input 
stream to be read via HTTP.  I.e., if you were to use this to replace a 
"FileSink" or "*RTPSink" in one of the test programs - e.g., using
         HTTPSink::createNew(env, <some-port-number>)
instead of
         FileSink::createNew(env, ...)
or
         <something>RTPSink::createNew(env, ...)
then the input stream would be readable using HTTP (instead of  being 
written into a file, or streamed using RTP), using the URL
         http://<host-running-HTTPSink>:<some-port-number>

Note, though, that this is only a very limited implementation of a HTTP 
server.  In particular, only one client can read from the server at a time.

HTTPSink is used to implement the "in_rtp.dll" RTP/RTSP plugin for Winamp 
<http://www.live.com/multikit/winamp-plugin.html> (because the only way 
that the rest of Winamp can read a HTTP stream is via HTTP).

         Ross.



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X-Original-Date: Thu, 17 Apr 2003 21:18:28 +0000
Date: Thu, 17 Apr 2003 21:18:28 +0000


Ross et al,

Has anyone managed to build MPlayer with RTSP / RTP support?  I followed the directions and 
snarfed the latest MPlayer CVS snapshot, but it failed to build (on Mac OSX 10.2.5 as well as 
RedHat Linux 9).  I checked out the MPlayer source direct from their CVS repo to see if that would 
build, but it doesn't, either (messages below).  This was with live in /usr/local instead of where I 
normally have it (in ~/multimedia).  Damn.

Has anyone a version of MPlayer (with source) with RTP / RTSP support?  Ross, if I get the 0.90 
version, which I assume is the version you modified to add RTP / RTSP to, could you supply a diff 
or something?  I'm looking at MPlayer for a project i'm involved with (they're using VideoLAN, but, 
while VideoLAN is way cool and everything, it has some issues that might take awhile to solve.  If 
MPlayer / Live is a better (more mature) solution, i'd rather use it.

Regards,

John

--
Falling You - exploring the beauty 
of voice and sound
http://www.mp3.com/fallingyou










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X-Original-Date: Thu, 17 Apr 2003 14:40:04 -0700
Date: Thu, 17 Apr 2003 14:40:04 -0700


>Has anyone managed to build MPlayer with RTSP / RTP support?

I have :-)

>   I followed the directions and
>snarfed the latest MPlayer CVS snapshot, but it failed to build (on Mac 
>OSX 10.2.5 as well as RedHat Linux 9).

I don't know whether or not MPlayer is buildable on Mac OS X, but it 
certainly should be buildable on RedHat Linux.

Because you had problems building the latest MPlayer CVS snapshot on Linux, 
you should report this to the MPlayer developers' mailing list:
         mplayer-dev-eng(at)mplayerhq.hu
That's the right place to report problems like this.  (This mailing list, 
on the other hand, is for any problems that you might have building the 
LIVE.COM code.)

>if I get the 0.90
>version, which I assume is the version you modified to add RTP / RTSP to

No, No, a thousand times: NO!  As I have explained repeatedly, the 0.90 
version is out of date, and RTSP/RTP streaming support will *not* work 
properly with that version.  Instead, use the "latest CVS snapshot" version 
of MPlayer.

I'm hoping that sometime soon the MPlayer people will be releasing a new, 
more up-to-date 'official' version that works - but until that time, you 
need to use the "latest CVS snapshot".

         Ross.



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X-Original-Date: Thu, 17 Apr 2003 14:41:42 -0700
Date: Thu, 17 Apr 2003 14:41:42 -0700

On Thu, Apr 17, 2003 at 09:18:28PM +0000, j.zorko(at)att.net wrote:
> 
> Ross et al,
> 
> Has anyone managed to build MPlayer with RTSP / RTP support?  I followed the directions and 
> snarfed the latest MPlayer CVS snapshot, but it failed to build (on Mac OSX 10.2.5 as well as 
> RedHat Linux 9).  I checked out the MPlayer source direct from their CVS repo to see if that would 
> build, but it doesn't, either (messages below).  This was with live in /usr/local instead of where I 
> normally have it (in ~/multimedia).  Damn.
> 
I've built mplayer 0.90 on debian unstable with live support and am
using it successfully.

I build live in /usr/src/mplayer/live, and then add 
--with-livelibdir=/usr/src/mplayer/live to the mplayer config options,
I have then used testMPEGAudioVideoStreamer to broadcast 2 RTP streams,
an audio and a video stream, and can receive them both with
mplayer rtsp://localhost

I haven't tried it with mplayer cvs, but 0.90 works just fine..

hope this helps,

EG
-- 
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... yes, it does, with --enable-live.  On Mac OSX 10.2.5, anyway (building it on RH9 now).  I wasn't 
too pleased about it loudly proclaiming that my nifty iBook 700 was "too SLOW to play this," but 
hey ... i'll just forget that it said that :-)

Regards,

John

--
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of voice and sound
http://www.mp3.com/fallingyou










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X-Original-Date: Thu, 17 Apr 2003 14:53:55 -0700
Date: Thu, 17 Apr 2003 14:53:55 -0700


>I haven't tried it with mplayer cvs, but 0.90 works just fine..

The 0.90 version of MPlayer works mostly OK, I think, for playing MPEG-1 or 
2 content via RTSP.  However, it won't work for playing streams from a SDP 
file, and also might not play MPEG-4 streams properly.  (It also might not 
handle audio/video sync as well as newer versions.)  For these reasons, you 
should use the "latest CVS snapshot" version if you can.

         Ross.



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X-Original-Date: Thu, 17 Apr 2003 23:38:33 +0000
Date: Thu, 17 Apr 2003 23:38:33 +0000


Ross et al,

The CVS snapshot of MPlayer for 4-16 builds and runs on Mac OSX and RH9 Linux.  It even plays 
streams from DSS running on my iBook.  I've more questions, though ... some of which are 
probably more of an MPlayer than a Live thing, but i'm not sure ...

1. When using vobStreamer, MPlayer (20030416) doesn't work too well -- the display is jumbled.  
Once it played the VOB, but it kept mixing the frames up i.e. a frame from 10-15 seconds later in 
the stream would appear and rapidly switch between that and the real frame.  MPlayer works fine, 
though, with MP4 streams from DSS.

2. Assuming the above is easily fixed (or the MPlayer people fix it soon), what is involved in 
extending RTP / RTSP support in MPlayer to handle PAUSE and PLAY with a start time?  Can I set 
the RTSP scale header to do ff / rewind and expect MPlayer to play it, or does this need to be 
added as well?  I've not looked at the MPlayer source yet ... 

3. Let's say a player adds itself to a multicast group and starts playing what the other player(s) in 
that group are playing.  Let's also say that these players need to be as closely in sync as humanly 
possible.  Assuming CPU load and network bandwidth aren't an issue, how difficult do you think it 
would be to add a client buffer management option (perhaps as an extension header in the RTSP) 
so that the server can say "only buffer up n seconds before starting to play."  Or is there a better 
way to keep multiple players on multiple boxes playing the same content from the same multicast 
address in perfect (or close to perfect) sync?

I _do_ appreciate everyone's input on this ...

Regards,

John

--
Falling You - exploring the beauty 
of voice and sound
http://www.mp3.com/fallingyou










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X-Original-Date: Thu, 17 Apr 2003 18:54:48 -0700
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>1. When using vobStreamer, MPlayer (20030416) doesn't work too well -- the 
>display is jumbled.
>Once it played the VOB, but it kept mixing the frames up i.e. a frame from 
>10-15 seconds later in
>the stream would appear and rapidly switch between that and the real frame.

1/ Does this behavior depend on whether or not you're streaming I-frames 
only (i.e., using the "-i" option)?
2/ Does this behavior still happen if you stream video only (i.e., using 
the "-v" option)?
3/ If the answer to 2/ is "yes", does the behavior still occur if you 
record the video stream into a file (e.g., using "openRTSP"), and then 
later play that file with MPlayer?

>2. Assuming the above is easily fixed (or the MPlayer people fix it soon), 
>what is involved in
>extending RTP / RTSP support in MPlayer to handle PAUSE and PLAY with a 
>start time?

$$$  See the last question in the FAQ: <http://www.live.com/liveMedia/faq.html>

>Or is there a better
>way to keep multiple players on multiple boxes playing the same content 
>from the same multicast address in perfect (or close to perfect) sync?

This would be difficult.  Right now MPlayer has enough trouble keeping 
audio and video in sync on the same box.  Keeping multiple boxes in sync 
would be even harder.

         Ross.



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From <EMAIL-ADDRESS-HIDDEN> Fri Apr 18 05:09:22 2003
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X-Original-Date: Fri, 18 Apr 2003 22:10:31 +1000
Date: Fri, 18 Apr 2003 22:10:31 +1000

Ross,

I would like to be able to implement a change in the way a client handles 
received RTP/RTCP packets. The packets need to be handled in the normal way, 
but I also want to be able to insert a handler that will relay the packets 
without the overhead of extracting the data from the RTP packet (and codec 
frames), only to repacketize it for retransmission.

The ideal place to do this is in RTPInterface, but RTPInterface is pretty 
hard-wired. Would it be possible for some method to be added to RTPSource or 
MultiFramedRTPSource that allows for the behaviour of RTPInterface to be 
changed? I suppose this would involve some virtual methods in, or a callback 
from, RTPInterface, and a setRTPInterface() or setRTPInterfaceFactory() 
method for RTPSource.

Thanks

Alex


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X-Original-Date: Fri, 18 Apr 2003 17:06:29 +0000
Date: Fri, 18 Apr 2003 17:06:29 +0000


Ross et al,

I snarfed the latest MPlayer from their CVS repository this morning, configured it with --enable-
live and built it.  I'm also currently running vobStreamer with -i (I-frames only) and -v.  The 
machines in this test are 2 2GHz RedHat 9 boxes.  One box runs vobStreamer and one MPlayer, the 
other runs another MPlayer.  Both MPlayers are built from the most recent CVS code, and given the 
RTSP URL of the vobStreamer, which is streaming on the default multicast address.  This is what 
i'm seeing:

1. The MPlayer on the same box as vobStreamer behaves better with -i ... the movie is jerky 
(expected) but none of the weird forward / backward frame skipping is happening.

2. The MPlayer on the other box isn't doing anywhere near as well -- slices are all being drawn in 
weird places.

If I run vobStreamer without -i (and with or without -v), the local MPlayer does the weird forward / 
backward frame switching thing, and the remote player is completely jumbled (not just slices, 
macroblocks and blocks as well, the video now completely unrepresentative of the actual movie).

Do these point to issues with MPlayer, live, ffmpeg, our LAN, or something else?  Please cc 
j.zorko(at)att.net -- I turned off digest mode, but it tells me I may get one more.

Regards,

John

--
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of voice and sound
http://www.mp3.com/fallingyou










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Subject: Re: [Live-devel] Altering RTP packet reading
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X-Original-Date: Fri, 18 Apr 2003 10:38:38 -0700
Date: Fri, 18 Apr 2003 10:38:38 -0700


>I would like to be able to implement a change in the way a client handles
>received RTP/RTCP packets. The packets need to be handled in the normal way,
>but I also want to be able to insert a handler that will relay the packets
>without the overhead of extracting the data from the RTP packet (and codec
>frames), only to repacketize it for retransmission.

Can you say some more about what you're trying to do?  Are you trying to 
develop a RTP/RTCP receiver that also relays the incoming packets to 
another destination?

>The ideal place to do this is in RTPInterface, but RTPInterface is pretty
>hard-wired.

Perhaps a better place to do this would be in 
MultiFramedRTPSource::networkReadHandler(), because this is where complete 
incoming RTP packets are actually processed by the RTP layer.  Would it 
work for you if "networkReadHandler()" were split into (i) a static member 
function (as it is now), and (ii) a virtual member function that's called 
by the static member function?  Then, you could - if you wished - redefine 
the virtual member function in your own subclass of "MultiFramedRTPSource".

Or is your change something that you want to to be able to do at runtime, 
for any *existing* subclass of "MultiFramedRTPSource"?  Again, it would be 
helpful if you could say some more about what you're trying to do.

         Ross.



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X-Original-Date: Fri, 18 Apr 2003 10:47:53 -0700
Date: Fri, 18 Apr 2003 10:47:53 -0700


>1. The MPlayer on the same box as vobStreamer behaves better with -i ... 
>the movie is jerky
>(expected) but none of the weird forward / backward frame skipping is 
>happening.
>
>2. The MPlayer on the other box isn't doing anywhere near as well -- 
>slices are all being drawn in
>weird places.

It sounds like your LAN isn't fast enough.  10 Mbps Ethernet won't cut it; 
try upgrading to 100 Mbps.

>If I run vobStreamer without -i (and with or without -v), the local 
>MPlayer does the weird forward /
>backward frame switching thing

Does this happen the first time that "vobStreamer" reads a particular 
'.vob' file, or only after it loops to repeat reading the same file?

Could you put the '.vob' file (or a representative portion of it) online, 
so I can download it and try it for myself?

>and the remote player is completely jumbled (not just slices,
>macroblocks and blocks as well, the video now completely unrepresentative 
>of the actual movie).

Again, it definitely sounds like your LAN is underpowered.

         Ross.




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X-Original-Date: Sat, 19 Apr 2003 09:49:54 +0200
Date: Sat, 19 Apr 2003 09:49:54 +0200


Hi,

Handling individual packets is of great interest to us too. Precisely,
it would be useful to be able to register a packet handler per each
media subsession, so individual RTP streams could be handled separately.
I don't think a virtual method of MultiFramedRTPSource would do the job
...

Generally speaking, I think that a way to handle this would be to
somehow register custom RTP sources.

Thanks,

romulus


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X-Original-Date: Sat, 19 Apr 2003 01:13:45 -0700
Date: Sat, 19 Apr 2003 01:13:45 -0700


>Handling individual packets is of great interest to us too. Precisely,
>it would be useful to be able to register a packet handler per each
>media subsession, so individual RTP streams could be handled separately.
>I don't think a virtual method of MultiFramedRTPSource would do the job
>...
>Generally speaking, I think that a way to handle this would be to
>somehow register custom RTP sources.

Of course, anyone can already do this, by defining a custom subclass of 
"MultiFramedRTPSource".

Note, by the way, the existing subclass "SimpleRTPSource".  This is the 
minimal subclass of "MultiframedRTPSource".  It just delivers the RTP 
payload 'as is' to the caller.

         Ross.



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From <EMAIL-ADDRESS-HIDDEN> Mon Apr 21 13:18:26 2003
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X-Original-Date: Mon, 21 Apr 2003 16:15:22 -0400 (EDT)
Date: Mon, 21 Apr 2003 16:15:22 -0400 (EDT)

I am a student and I use your library for my final
year project

I get a problem when i try to modify the test programs
I tried to modify testmp3Streamer and I get that:

--------------------Configuration: testMP3Streamer -
Win32 Debug--------------------
Linking...
testMP3Streamer.obj : error LNK2001: unresolved
external symbol "public: static class
BasicUsageEnvironment * __cdecl
BasicUsageEnvironment::createNew(class TaskScheduler
&)"
(?createNew(at)BasicUsageEnvironment(at)(at)SAPAV1(at)AAVTaskScheduler(at)(at)(at)Z)
testMP3Streamer.obj : error LNK2001: unresolved
external symbol "public: static class
BasicTaskScheduler * __cdecl
BasicTaskScheduler::createNew(void)"
(?createNew(at)BasicTaskScheduler(at)(at)SAPAV1(at)XZ)
testMP3Streamer.obj : error LNK2001: unresolved
external symbol "public: unsigned int __thiscall
MediaSink::startPlaying(class MediaSource &,void
(__cdecl*)(void *),void *)"
(?startPlaying(at)MediaSink(at)(at)QAEIAAVMediaSource(at)(at)P6AXPAX(at)Z1(at)Z)
testMP3Streamer.obj : error LNK2001: unresolved
external symbol "public: static class RTCPInstance *
__cdecl RTCPInstance::createNew(class UsageEnvironment
&,class Groupsock *,unsigned int,unsigned char const
*,class RTPSink const *,class RTPSource 
const *)"
(?createNew(at)RTCPInstance(at)(at)SAPAV1(at)AAVUsageEnvironment(at)(at)PAVGroupsock(at)(at)IPBEPBVRTPSink(at)(at)PBVRTPSource(at)(at)(at)Z)
testMP3Streamer.obj : error LNK2001: unresolved
external symbol _gethostname(at)8
testMP3Streamer.obj : error LNK2001: unresolved
external symbol "public: static class MPEGAudioRTPSink
* __cdecl MPEGAudioRTPSink::createNew(class
UsageEnvironment &,class Groupsock *)"
(?createNew(at)MPEGAudioRTPSink(at)(at)SAPAV1(at)AAVUsageEnvironment(at)(at)PAVGr
oupsock(at)(at)(at)Z)
testMP3Streamer.obj : error LNK2001: unresolved
external symbol "public: __thiscall
Groupsock::Groupsock(class UsageEnvironment &,struct
in_addr const &,class Port,unsigned char)"
(??0Groupsock(at)(at)QAE(at)AAVUsageEnvironment(at)(at)ABUin_addr(at)(at)VPort(at)(at)E(at)Z)
testMP3Streamer.obj : error LNK2001: unresolved
external symbol "public: __thiscall
Port::Port(unsigned short)" (??0Port(at)(at)QAE(at)G(at)Z)
testMP3Streamer.obj : error LNK2001: unresolved
external symbol _our_inet_addr
testMP3Streamer.obj : error LNK2001: unresolved
external symbol "public: static class MP3FileSource *
__cdecl MP3FileSource::createNew(class
UsageEnvironment &,char const *)"
(?createNew(at)MP3FileSource(at)(at)SAPAV1(at)AAVUsageEnvironment(at)(at)PBD(at)Z)
testMP3Streamer.obj : error LNK2001: unresolved
external symbol "public: static void __cdecl
Medium::close(class Medium *)"
(?close(at)Medium(at)(at)SAXPAV1(at)(at)Z)
testMP3Streamer.obj : error LNK2001: unresolved
external symbol "class DelayInterval  __cdecl
operator*(short,class DelayInterval const &)"
(??D(at)YA?AVDelayInterval(at)(at)FABV0(at)(at)Z)
testMP3Streamer.obj : error LNK2001: unresolved
external symbol "class DelayInterval const  SECOND"
(?SECOND(at)(at)3VDelayInterval(at)(at)B)
Debug/testMP3Streamer.exe : fatal error LNK1120: 13
unresolved externals
Error executing link.exe.

testMP3Streamer.exe - 14 error(s), 0 warning(s)

Regards 

Jocelyn

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X-Original-Date: Mon, 21 Apr 2003 16:36:29 -0400 (EDT)
Date: Mon, 21 Apr 2003 16:36:29 -0400 (EDT)

I followed the instruction on the web site and
everything works well, but when i try to open
testMP3Streamer.cpp and  try to build it, I get these errors

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X-Original-Date: Mon, 21 Apr 2003 16:33:59 -0400 (EDT)
Date: Mon, 21 Apr 2003 16:33:59 -0400 (EDT)

Yes everything works when I compile the testprogs but
when I want to modify the code in one of them I get
the previous errors


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X-Original-Date: Mon, 21 Apr 2003 13:29:58 -0700
Date: Mon, 21 Apr 2003 13:29:58 -0700

At 01:15 PM 4/21/03, you wrote:
>I am a student and I use your library for my final
>year project
>
>I get a problem when i try to modify the test programs

Were you able to successfully build and run the original test programs?  If 
so, then you should also be able to build your modified program.  Remember 
that if you change the names of any of the files (and/or add any new 
files), you will need to make the corresponding change in the 
"testProgs.mak" Makefile.

         Ross.



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X-Original-Date: Mon, 21 Apr 2003 16:41:46 -0400 (EDT)
Date: Mon, 21 Apr 2003 16:41:46 -0400 (EDT)

Do i have to add some .lib file in the project's
settings (link)

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X-Original-Date: Mon, 21 Apr 2003 13:53:14 -0700
Date: Mon, 21 Apr 2003 13:53:14 -0700

At 01:33 PM 4/21/03, you wrote:
>Yes everything works when I compile the testprogs but
>when I want to modify the code in one of them I get
>the previous errors

Unfortunately, since we don't know what you modified, it's hard for us to 
figure out what you did wrong.  So you will have to figure this out for 
yourself.  But since the original test programs work OK for you, you should 
be able to figure this out (by making a series of small changes only).

         Ross.



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X-Original-Date: Mon, 21 Apr 2003 14:00:33 -0700
Date: Mon, 21 Apr 2003 14:00:33 -0700

At 01:41 PM 4/21/03, you wrote:
>Do i have to add some .lib file in the project's
>settings (link)

No.  You already a program that works.  Start by making modifications only 
to the existing source file(s) (e.g., "testMP3Streamer.cpp"), and then 
rebuild the program.

         Ross.



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X-Original-Date: Mon, 21 Apr 2003 17:20:04 -0400 (EDT)
Date: Mon, 21 Apr 2003 17:20:04 -0400 (EDT)

I didn't change anything... just the IP adresses and I
tried to build it...

I really don't know

I am pretty that I have to include some .lib file in
the project's setting


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X-Original-Date: Mon, 21 Apr 2003 17:21:28 -0400 (EDT)
Date: Mon, 21 Apr 2003 17:21:28 -0400 (EDT)

I can't rebuild the original one.. this is exactly
what I did and I got these stupid link errors

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X-Original-Date: Mon, 21 Apr 2003 21:47:40 +0000
Date: Mon, 21 Apr 2003 21:47:40 +0000


Jocelyn,

> I can't rebuild the original one.. this is exactly
> what I did and I got these stupid link errors

I understand your frustration, but Ross is right -- we've no clue what you've done, we're not privvy 
to the changes you've made, and we don't know the state of your Live installation anymore.  If I 
were you, I would re-install the Live stuff ... source, directories, the whole tree, and build Live from 
scratch again.  At this point you should have completely working examples again, just be sure to 
make a copy of any that you want to change before you change them.  When we can compare what 
was there to what you changed, we can better assist you.

Regards,

John

--
Falling You - exploring the beauty 
of voice and sound
http://www.mp3.com/fallingyou









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Subject: [Live-devel] how to build webserver with rtp support ?
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X-Original-Date: Tue, 22 Apr 2003 17:14:51 +0800
Date: Tue, 22 Apr 2003 17:14:51 +0800


Hi!  I have two sets of questions regarding an application i would like 
to make.  
1.  I'm thinking of building a webserver with rtp support on it.  Or if 
I have an existing embedded webserver, do you think it's sensible to put 
rtp support on it to improve streaming of data (e.g. video or audio from 
a camera)?  Somebody told me that you can only use one, http or rtp, for 
the application.  I was thinking http will be used for the connection 
and then rtp (over udp) will be used to send the data.  How do I 
actually implement this using the liveMedia library?  What modules do I 
need?  
2.  Does the other end which would receive the streamed data also need 
to have rtp support?  For example, if I access the webserver through a 
webbrowser, I think it needs a plugin which can display the video (like 
realplayer or quicktime), or just a player which can get input through 
http or url address.  Does the plugin (or webbrowser)  or player need to 
have rtp?

Thank you in advance.

Mildred










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Subject: [Live-devel] Adding a RTP handler
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X-Original-Date: Tue, 22 Apr 2003 20:30:57 +1000
Date: Tue, 22 Apr 2003 20:30:57 +1000

> Message: 3
> Date: Fri, 18 Apr 2003 10:38:38 -0700
> To: live-devel(at)lists.sourceforge.net
> From: Ross Finlayson <finlayson(at)live.com>
> Subject: Re: [Live-devel] Altering RTP packet reading
> Reply-To: live-devel(at)lists.sourceforge.net
>
> >I would like to be able to implement a change in the way a client handles
> >received RTP/RTCP packets. The packets need to be handled in the normal
> > way, but I also want to be able to insert a handler that will relay the
> > packets without the overhead of extracting the data from the RTP packet
> > (and codec frames), only to repacketize it for retransmission.
>
> Can you say some more about what you're trying to do?  Are you trying to
> develop a RTP/RTCP receiver that also relays the incoming packets to
> another destination?

Yes, that's about it. I'm working on an improved version of Trevbus
(http://www.trevbus.org) and I'm looking to include a built-in HTTP server
similar to Icecast. The complete RTP packets need to be relayed out to 
multiple clients as well as sent to the HTTP server so it can do its normal 
RTP client thing.

I think I can accomplish it by using RTP over TCP, and by doing 
setStreamSocket on RTPInterfaces obtained from the static RTPInterface lookup 
table so they read from an internal TCP connection created to send packets 
from the network read handler attached to new RTPInterfaces that actually 
read the RTSP TCP connection. Anyhow, there has to be a better way!

> >The ideal place to do this is in RTPInterface, but RTPInterface is pretty
> >hard-wired.
>
> Perhaps a better place to do this would be in
> MultiFramedRTPSource::networkReadHandler(), because this is where complete
> incoming RTP packets are actually processed by the RTP layer.  Would it
> work for you if "networkReadHandler()" were split into (i) a static member
> function (as it is now), and (ii) a virtual member function that's called
> by the static member function?  Then, you could - if you wished - redefine
> the virtual member function in your own subclass of "MultiFramedRTPSource".
>
> Or is your change something that you want to to be able to do at runtime,
> for any *existing* subclass of "MultiFramedRTPSource"?  Again, it would be
> helpful if you could say some more about what you're trying to do.

Yes, for any codec (RTPSource) I would like to be able to insert a RTP packet 
handler for both RTP and RTCP packets on each subsession, in addition to 
whatever the codec does. Subclassing MultiFramedRTPSource means subclassing 
every possible codec -- or is it possible to do something similar to the way 
MediaSubsession::fReadSource and MediaSubsession::fRTPSource interoperate? 
Though that may preclude intercepting RTCP packets as well, and would require 
writing a new MediaSession class?

Thanks

Alex Pollard


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Subject: Re: [Live-devel] how to build webserver with rtp support ?
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X-Original-Date: Tue, 22 Apr 2003 09:52:30 -0700
Date: Tue, 22 Apr 2003 09:52:30 -0700


>1.  I'm thinking of building a webserver with rtp support on it.  Or if I 
>have an existing embedded webserver, do you think it's sensible to put rtp 
>support on it to improve streaming of data (e.g. video or audio from a 
>camera)?  Somebody told me that you can only use one, http or rtp, for the 
>application.  I was thinking http will be used for the connection and then 
>rtp (over udp) will be used to send the data.

A better (& more standard) solution is to use RTSP (a protocol somewhat 
similar to HTTP) for the control protocol, and then RTP for streaming.

>  How do I actually implement this using the liveMedia library?  What 
> modules do I need?

If you use RTSP, then you won't need to develop your own server.  Instead, 
you can use existing, open source, servers, such as Apple's QuickTime (aka. 
Darwin) Streaming server, or RealNetworks' Helix Server.  (For more 
information about RTSP, see <http://www.rtsp.org/>)

>2.  Does the other end which would receive the streamed data also need to 
>have rtp support?

Yes, of course.  But fortunately there are several clients already 
available, including  QuickTime Player (for Mac OS and Windows), and 
MPlayer (for Linux and *BSD).

>   For example, if I access the webserver through a webbrowser, I think it 
> needs a plugin which can display the video (like realplayer or 
> quicktime), or just a player which can get input through http or url 
> address.  Does the plugin (or webbrowser)  or player need to have rtp?

Yes, Apple's QuickTime plugin does this (using RTSP/RTP).

         Ross.



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X-Original-Date: Tue, 22 Apr 2003 13:06:01 -0400 (EDT)
Date: Tue, 22 Apr 2003 13:06:01 -0400 (EDT)

I finally found a wab to run it.

When I run the test program testMP3Streamer and
tesMP3receiver, the computer that runs the
testMP3receiver displays several symbol continually
and it get stucks.  I can't stop it.

Do you have any idea why?

Regards
Jocelyn

__________________________________________________________
Lèche-vitrine ou lèche-écran ?
magasinage.yahoo.ca


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Jocelyn,

> When I run the test program testMP3Streamer and
> tesMP3receiver, the computer that runs the
> testMP3receiver displays several symbol continually
> and it get stucks.  I can't stop it.

I believe testMP3Receiver outputs everything to stdout, which would explain why it does what you 
describe.  I've had success running testMP3Receiver like this:

testMP3Reciever > stuff.mp3

... then open "stuff.mp3" in whatever MP3 player you use, and it should play what 
testMP3Streamer streamed.

I hope this helps.

Regards,

John

--
Falling You - exploring the beauty 
of voice and sound
http://www.mp3.com/fallingyou









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Jocelyn, 

As the documentation explains ( http://www.live.com/liveMedia ), the test 
programs receive the av stream and output it to stdout. So you  are seeing 
the bytes of the MP3 in stdout. In testMP3Receiver.cpp, if you change the 
"stdout" in sessionState.sink = FileSink::createNew(*env, "stdout"); to a 
file name, then it would output it to a file.

Chris

On Tue, 22 Apr 2003, [iso-8859-1] Jocelyn Lefrancois wrote:

> I finally found a wab to run it.
> 
> When I run the test program testMP3Streamer and
> tesMP3receiver, the computer that runs the
> testMP3receiver displays several symbol continually
> and it get stucks.  I can't stop it.
> 
> Do you have any idea why?
> 
> Regards
> Jocelyn
> 
> __________________________________________________________
> Lèche-vitrine ou lèche-écran ?
> magasinage.yahoo.ca
> 
> 
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Hi,

I wanted to know if there is a means to stream ASF packets using the library?

Thanks & Regards,
Manu



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I can stream MPEG12 video great, but wanted to try out MPEG4. I notice 
that there is no MPEG4*RTPSink class. What will it take to implement one? 
Before I try my hand at writing one, is there anything special going on, 
or should I just take the MPEG*RTPSink classes as examples and dive in?

Thanks,
Chris




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Hi,

Can anyone help me know the source code files that would be required for implementation of RTSP server that does not use RTP? Is documentation on source code available anywhere?

Thanks & Regards,
Manu



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>Can anyone help me know the source code files that would be required for 
>implementation of RTSP server that does not use RTP?

A RTSP server that doesn't use RTP??  I suppose this is possible, in 
principle, but I have never heard of such a thing.

What do you want to use to stream media instead of RTP?  And why?

>  Is documentation on source code available anywhere?

See <http://www.live.com/liveMedia/faq.html#doc>

         Ross.



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X-Original-Date: Tue, 22 Apr 2003 16:04:45 -0700
Date: Tue, 22 Apr 2003 16:04:45 -0700

At 01:38 PM 4/22/03, you wrote:
>I can stream MPEG12 video great, but wanted to try out MPEG4. I notice
>that there is no MPEG4*RTPSink class. What will it take to implement one?
>Before I try my hand at writing one, is there anything special going on,
>or should I just take the MPEG*RTPSink classes as examples and dive in?

Note that there are at least three different RTP payload formats for MPEG-4:
         1/ video/MP4V-ES (defined in RFC 3016)
         2/ audio/MPEG4-GENERIC (defined in 
<http://www.ietf.org/internet-drafts/draft-ietf-avt-mpeg4-simple-07.txt>)
         3/ audio/MP4A-LATM (defined in RFC 3016)
Most MPEG-4 RTP sessions use 1/ and 2/.  (3/ is used mainly for 3GPP audio.)

A RTPSink for 1/ would be very easy.  In fact, you could just use the 
existing "SimpleRTPSink" - e.g.
         SimpleRTPSink::createNew(*env, destGS, 96, 90000, "video", "MP4V-ES");
(assuming that you use the dynamic payload format # 96 for video).  (With 
the current implementation of "SimpleRTPSink", the RTP "M" bit won't be set 
properly, but I'll fix that shortly.)

A general implementation of a RTPSink for 2/ (supporting arbitrary 'modes', 
and interleaving) would be very difficult.  (OTOH, a simpler 
implementation, that supported just the non-interleaved "aac-hbr" mode (the 
most commonly-used mode), would merely be "difficult".)

A RTPSink for 3/ would be fairly easy.

         Ross.




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X-Original-Date: Tue, 22 Apr 2003 16:33:10 -0700
Date: Tue, 22 Apr 2003 16:33:10 -0700


> > Can you say some more about what you're trying to do?  Are you trying to
> > develop a RTP/RTCP receiver that also relays the incoming packets to
> > another destination?
>
>Yes, that's about it. I'm working on an improved version of Trevbus
>(http://www.trevbus.org) and I'm looking to include a built-in HTTP server
>similar to Icecast.

BTW, are you familiar with the "HTTPSink" class?  You may find that useful.

>The complete RTP packets need to be relayed out to
>multiple clients as well as sent to the HTTP server so it can do its normal
>RTP client thing.

Here's a possible solution:
- I could add, to "RTPInterface" (and thus "RTPSource" and "RTCPInstance"), 
the ability to register an optional, auxilliary 'read handler' 
function.  Such a handler function would have the signature
         void auxilliaryReadHandler(unsigned char* packet, unsigned packetSize)
I would modify "RTPInterface::handleRead()" to call this read handler (if 
it exists), just after it reads each new packet.  (Thus, it would include 
the RTP header.)

Would this work for you?

         Ross.



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Subject: RE: [Live-devel] how to build webserver with rtp support ?
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X-Original-Date: Wed, 23 Apr 2003 09:57:35 +0800
Date: Wed, 23 Apr 2003 09:57:35 +0800


Ross,
Thank you so much, this is very insightful.  Is there a similar application
in liveMedia (like Darwin or Helix)?  Is there something which streams MPEG
instead?  Do Darwin and Helix run in Linux?  Can they be made for embedded?
Thanks.

Mildred

-----Original Message-----
From: <EMAIL-ADDRESS-HIDDEN>
[mailto:live-devel-admin(at)lists.sourceforge.net]On Behalf Of Ross
Finlayson
Sent: Wednesday, April 23, 2003 12:53 AM
To: <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] how to build webserver with rtp support ?



>1.  I'm thinking of building a webserver with rtp support on it.  Or if I
>have an existing embedded webserver, do you think it's sensible to put rtp
>support on it to improve streaming of data (e.g. video or audio from a
>camera)?  Somebody told me that you can only use one, http or rtp, for the
>application.  I was thinking http will be used for the connection and then
>rtp (over udp) will be used to send the data.

A better (& more standard) solution is to use RTSP (a protocol somewhat
similar to HTTP) for the control protocol, and then RTP for streaming.

>  How do I actually implement this using the liveMedia library?  What
> modules do I need?

If you use RTSP, then you won't need to develop your own server.  Instead,
you can use existing, open source, servers, such as Apple's QuickTime (aka.
Darwin) Streaming server, or RealNetworks' Helix Server.  (For more
information about RTSP, see <http://www.rtsp.org/>)

>2.  Does the other end which would receive the streamed data also need to
>have rtp support?

Yes, of course.  But fortunately there are several clients already
available, including  QuickTime Player (for Mac OS and Windows), and
MPlayer (for Linux and *BSD).

>   For example, if I access the webserver through a webbrowser, I think it
> needs a plugin which can display the video (like realplayer or
> quicktime), or just a player which can get input through http or url
> address.  Does the plugin (or webbrowser)  or player need to have rtp?

Yes, Apple's QuickTime plugin does this (using RTSP/RTP).

         Ross.



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X-Original-Date: Wed, 23 Apr 2003 10:10:44 +0800
Date: Wed, 23 Apr 2003 10:10:44 +0800


What if I already have a webserver/website? Is it easy to incorporate rtp
with the webserver code for streaming?
Thanks

Mildred

-----Original Message-----
From: <EMAIL-ADDRESS-HIDDEN>
[mailto:live-devel-admin(at)lists.sourceforge.net]On Behalf Of Ross
Finlayson
Sent: Wednesday, April 23, 2003 12:53 AM
To: <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] how to build webserver with rtp support ?



>1.  I'm thinking of building a webserver with rtp support on it.  Or if I
>have an existing embedded webserver, do you think it's sensible to put rtp
>support on it to improve streaming of data (e.g. video or audio from a
>camera)?  Somebody told me that you can only use one, http or rtp, for the
>application.  I was thinking http will be used for the connection and then
>rtp (over udp) will be used to send the data.

A better (& more standard) solution is to use RTSP (a protocol somewhat
similar to HTTP) for the control protocol, and then RTP for streaming.

>  How do I actually implement this using the liveMedia library?  What
> modules do I need?

If you use RTSP, then you won't need to develop your own server.  Instead,
you can use existing, open source, servers, such as Apple's QuickTime (aka.
Darwin) Streaming server, or RealNetworks' Helix Server.  (For more
information about RTSP, see <http://www.rtsp.org/>)

>2.  Does the other end which would receive the streamed data also need to
>have rtp support?

Yes, of course.  But fortunately there are several clients already
available, including  QuickTime Player (for Mac OS and Windows), and
MPlayer (for Linux and *BSD).

>   For example, if I access the webserver through a webbrowser, I think it
> needs a plugin which can display the video (like realplayer or
> quicktime), or just a player which can get input through http or url
> address.  Does the plugin (or webbrowser)  or player need to have rtp?

Yes, Apple's QuickTime plugin does this (using RTSP/RTP).

         Ross.



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X-Original-Date: Tue, 22 Apr 2003 19:12:31 -0700
Date: Tue, 22 Apr 2003 19:12:31 -0700


>Is there a similar application
>in liveMedia (like Darwin or Helix)?  Is there something which streams MPEG
>instead?

No fewer than *four* of the "LIVE.COM Streaming Media" test programs - 
"testMPEGAudioVideoStreamer", "testMPEGVideoStreamer", "testMP3Streamer", 
and "vobStreamer" - stream MPEG (of various forms), and have an optional, 
built-in RTSP server.  (Search the source code for "IMPLEMENT_RTSP_SERVER".)

*Note*, however, that the RTSP server in these applications currently works 
only for multicast streams.

>   Do Darwin and Helix run in Linux?  Can they be made for embedded?

Yes, but they have their own mailing lists.  Please post questions about 
those systems there, not here.

         Ross.



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X-Original-Date: Wed, 23 Apr 2003 10:38:34 +0800
Date: Wed, 23 Apr 2003 10:38:34 +0800


Thanks! If the application's not using rtsp, what does it use for control?
just rtp then?

-----Original Message-----
From: <EMAIL-ADDRESS-HIDDEN>
[mailto:live-devel-admin(at)lists.sourceforge.net]On Behalf Of Ross
Finlayson
Sent: Wednesday, April 23, 2003 10:13 AM
To: <EMAIL-ADDRESS-HIDDEN>
Subject: RE: [Live-devel] how to build webserver with rtp support ?



>Is there a similar application
>in liveMedia (like Darwin or Helix)?  Is there something which streams MPEG
>instead?

No fewer than *four* of the "LIVE.COM Streaming Media" test programs -
"testMPEGAudioVideoStreamer", "testMPEGVideoStreamer", "testMP3Streamer",
and "vobStreamer" - stream MPEG (of various forms), and have an optional,
built-in RTSP server.  (Search the source code for "IMPLEMENT_RTSP_SERVER".)

*Note*, however, that the RTSP server in these applications currently works
only for multicast streams.

>   Do Darwin and Helix run in Linux?  Can they be made for embedded?

Yes, but they have their own mailing lists.  Please post questions about
those systems there, not here.

         Ross.



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From <EMAIL-ADDRESS-HIDDEN> Tue Apr 22 19:39:53 2003
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X-Original-Date: Tue, 22 Apr 2003 19:34:35 -0700
Date: Tue, 22 Apr 2003 19:34:35 -0700


>Thanks! If the application's not using rtsp, what does it use for control?
>just rtp then?

Yes, RTP includes a basic control-protocol, called RTCP, that (among other 
things) provides feedback from the receiver back to the sender.

See the latest version of the RTP/RTCP specification: 
<http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-new-12.txt>

         Ross.



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X-Original-Date: Wed, 23 Apr 2003 23:37:24 +1000
Date: Wed, 23 Apr 2003 23:37:24 +1000

On Wed, 23 Apr 2003 11:59 am, live-devel-request(at)lists.sourceforge.net wrote:
> Date: Tue, 22 Apr 2003 16:33:10 -0700
> To: live-devel(at)lists.sourceforge.net
> From: Ross Finlayson <finlayson(at)live.com>
> Subject: Re: [Live-devel] Adding a RTP handler
> Reply-To: live-devel(at)lists.sourceforge.net
>
....
> Here's a possible solution:
> - I could add, to "RTPInterface" (and thus "RTPSource" and "RTCPInstance"),
> the ability to register an optional, auxilliary 'read handler'
> function.  Such a handler function would have the signature
>          void auxilliaryReadHandler(unsigned char* packet, unsigned
> packetSize) I would modify "RTPInterface::handleRead()" to call this read
> handler (if it exists), just after it reads each new packet.  (Thus, it
> would include the RTP header.)
>
> Would this work for you?

Something like that, but would auxilliaryReadHandler know which RTPInterface 
the packet came from, ie which subsession? Would this be easily determined in 
all cases by examining the RTP or RTCP header for each packet? Would the RTP 
packet's payload type be enough to guarantee this? The UDP port 
number/streamchannelId might be sufficient.

Thanks

Alex


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Hello Ross,

Actually I want to stream ASF data and I don't want to use MMS since it is proprietary to Microsoft. Since, ASF packets can be streamed without wrapping them under RTP, I thought it would be best to use RTSP can medium without RTP to stream the content. Is this possible. Can the library help me do this?

Thanks & Regards,
Manu

>>> finlayson(at)live.com 04/22/03 05:06PM >>>

>Can anyone help me know the source code files that would be required for 
>implementation of RTSP server that does not use RTP?

A RTSP server that doesn't use RTP??  I suppose this is possible, in 
principle, but I have never heard of such a thing.

What do you want to use to stream media instead of RTP?  And why?

>  Is documentation on source code available anywhere?

See <http://www.live.com/liveMedia/faq.html#doc>

         Ross.



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From <EMAIL-ADDRESS-HIDDEN> Wed Apr 23 07:42:46 2003
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X-Original-Date: Wed, 23 Apr 2003 08:39:26 -0400 (EDT)
Date: Wed, 23 Apr 2003 08:39:26 -0400 (EDT)

Hi.  One more quick question about test programs... 
Is it possible to start an instance of window media
player or realOne player to play(MP3) or display
(MPEG).

Can we do it by replacing stdout with a special
command

I can open an instance of WM or RealOne with the
command 
system("folder\\realone.exe or wmplayer.exe") but I
get stuck

Reagrds 
Jocelyn

__________________________________________________________
Lèche-vitrine ou lèche-écran ?
magasinage.yahoo.ca


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X-Original-Date: Wed, 23 Apr 2003 10:03:26 -0700
Date: Wed, 23 Apr 2003 10:03:26 -0700


>Actually I want to stream ASF data and I don't want to use MMS since it is 
>proprietary to Microsoft.

But note that the ASF file format is also 'proprietary to Microsoft'.  See, 
for example,
         http://www.advogato.org/article/101.html

If your goal is to avoid Microsoft's lawyers, then just replacing MMS with 
RTSP won't be enough.  You're better off avoiding all Microsoft proprietary 
bullshit altogether.  I.e., don't just steer clear of MMS, but also ASF, 
and the Windows Media audio and video codecs.

>  Since, ASF packets can be streamed without wrapping them under RTP, I 
> thought it would be best to use RTSP can medium without RTP to stream the 
> content. Is this possible.

Yes, it's certainly 'possible' in principle, but (just as a starting point) 
you would need to define a way for the session's SDP description - returned 
in response to a RTSP "DESCRIBE" request - to denote a transport other than 
"RTP/AVP".  (Ditto for the "Transport:" headers.)  (And if you were to do 
this, then this is something that realy should be documented in an Internet 
Draft.)

>Can the library help me do this?

The short answer is no (unless you're prepared to do a lot of work).

         Ross.



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From <EMAIL-ADDRESS-HIDDEN> Wed Apr 23 11:40:00 2003
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From: Christopher Cope <EMAIL-ADDRESS-HIDDEN>
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Subject: Re: [Live-devel] MPEG4
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X-Original-Date: Wed, 23 Apr 2003 13:33:08 -0500 (CDT)
Date: Wed, 23 Apr 2003 13:33:08 -0500 (CDT)

Thanks Ross. 
Video should be enough for me to play around with for now.

I replaced 
videoSink = MPEGVideoRTPSink::createNew(*env, &rtpGroupsock);
with 
videoSink = SimpleRTPSink::createNew(*env, &rtpGroupsock, 96, 90000, 
"video", "MP4V-ES");
in testMPEGVideoStreamer.cpp.

Two problems:
first, the streamer halts after a short while (~15 seconds) with
the error
StreamParser internal error (100000 + 4 > 100000)
Its from liveMedia/StreamParser.cpp EnsureValidBytes()
This is probably due to streaming data that the parser is not
recognizing, i.e. a differnt codec. Trying to stream mpeg4 with the 
original testMPEGVideoStreamer gives the same error.
Do you know why the parser is having a problem? It might be 
that mencoder's  output is not an elementary video stream. Unfortunately I 
don't know enough about the different codecs to have any idea.

Also, I'm using the stdio hack to send the live stream:
mencoder -tv 
on:driver=v4l:device=/dev/video2:input=3:norm=ntsc:width=640:height=480  
-ovc lavc -lavcopts vcodec=mpeg4 -nosound -o /dev/stdout |
../live/testProgs/testMPEGVideoStreamer
This is a dubious shortcut, but because mpeg1 video works fine this way, 
I'm assuming the problem either lies in the output of mencoder, or my use 
of the RTP sink.

Much appreciated,
Chris



On Tue, 22 Apr 2003, Ross Finlayson wrote:

> At 01:38 PM 4/22/03, you wrote:
> >I can stream MPEG12 video great, but wanted to try out MPEG4. I notice
> >that there is no MPEG4*RTPSink class. What will it take to implement one?
> >Before I try my hand at writing one, is there anything special going on,
> >or should I just take the MPEG*RTPSink classes as examples and dive in?
> 
> Note that there are at least three different RTP payload formats for MPEG-4:
>          1/ video/MP4V-ES (defined in RFC 3016)
>          2/ audio/MPEG4-GENERIC (defined in 
> <http://www.ietf.org/internet-drafts/draft-ietf-avt-mpeg4-simple-07.txt>)
>          3/ audio/MP4A-LATM (defined in RFC 3016)
> Most MPEG-4 RTP sessions use 1/ and 2/.  (3/ is used mainly for 3GPP audio.)
> 
> A RTPSink for 1/ would be very easy.  In fact, you could just use the 
> existing "SimpleRTPSink" - e.g.
>          SimpleRTPSink::createNew(*env, destGS, 96, 90000, "video", "MP4V-ES");
> (assuming that you use the dynamic payload format # 96 for video).  (With 
> the current implementation of "SimpleRTPSink", the RTP "M" bit won't be set 
> properly, but I'll fix that shortly.)

I replaced
videoSink = MPEGVideoRTPSink::createNew(*env, &rtpGroupsock);
with 
videoSink = SimpleRTPSink::createNew(*env, &rtpGroupsock, 96, 90000, 
"video", "MP4V-ES");
in testMPEGVideoStreamer.cpp. 
Two problems im running into:
first, the streamer halts after a short while (~15 seconds) with 
the error
StreamParser internal error (100000 + 4 > 100000)
Its from liveMedia/StreamParser.cpp EnsureValidBytes()
I'm guessing this might be due to streaming data that the parser is not 
recognizing, i.e. parsing thus emptuing the buffer. Though Im just 
guessing. I'm sending the data from mencoder's output using the video 
codec mpeg4 with no sound. Is this 




> 
> A general implementation of a RTPSink for 2/ (supporting arbitrary 'modes', 
> and interleaving) would be very difficult.  (OTOH, a simpler 
> implementation, that supported just the non-interleaved "aac-hbr" mode (the 
> most commonly-used mode), would merely be "difficult".)
> 
> A RTPSink for 3/ would be fairly easy.
> 
>          Ross.
> 
> 
> 
> 
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From: Christopher Cope <EMAIL-ADDRESS-HIDDEN>
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Subject: Re: [Live-devel] MPEG4
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X-Original-Date: Wed, 23 Apr 2003 13:45:31 -0500 (CDT)
Date: Wed, 23 Apr 2003 13:45:31 -0500 (CDT)

That email got away from me before I finsihed. :)
The second problem I ran into, was receiving, or at least trying to,  the 
MP4V-ES stream. I added MP4V-ES to lookupPayloadFormat() in 
MediaSession.cpp. 

case 27: {temp = "MP4V-ES"; rtpTimestampFrequency = 90000; break;}

Saw that it was already in initiate() but not in lookupPayloadFormat(). I 
built live.com again but when I give mplayer an sdp with
m=video 8888 RTP/AVP 27
it cant find the stream and exits, as if there is no codec for case 27.
Is there another piece of code that needs to be modified? I did the same 
for "H263-1998" without this problem.

Chris

On Wed, 23 Apr 2003, Christopher Cope wrote:

> Thanks Ross. 
> Video should be enough for me to play around with for now.
> 
> I replaced 
> videoSink = MPEGVideoRTPSink::createNew(*env, &rtpGroupsock);
> with 
> videoSink = SimpleRTPSink::createNew(*env, &rtpGroupsock, 96, 90000, 
> "video", "MP4V-ES");
> in testMPEGVideoStreamer.cpp.
> 
> Two problems:
> first, the streamer halts after a short while (~15 seconds) with
> the error
> StreamParser internal error (100000 + 4 > 100000)
> Its from liveMedia/StreamParser.cpp EnsureValidBytes()
> This is probably due to streaming data that the parser is not
> recognizing, i.e. a differnt codec. Trying to stream mpeg4 with the 
> original testMPEGVideoStreamer gives the same error.
> Do you know why the parser is having a problem? It might be 
> that mencoder's  output is not an elementary video stream. Unfortunately I 
> don't know enough about the different codecs to have any idea.
> 
> Also, I'm using the stdio hack to send the live stream:
> mencoder -tv 
> on:driver=v4l:device=/dev/video2:input=3:norm=ntsc:width=640:height=480  
> -ovc lavc -lavcopts vcodec=mpeg4 -nosound -o /dev/stdout |
> ../live/testProgs/testMPEGVideoStreamer
> This is a dubious shortcut, but because mpeg1 video works fine this way, 
> I'm assuming the problem either lies in the output of mencoder, or my use 
> of the RTP sink.
> 
> Much appreciated,
> Chris
> 
> 
> 
> On Tue, 22 Apr 2003, Ross Finlayson wrote:
> 
> > At 01:38 PM 4/22/03, you wrote:
> > >I can stream MPEG12 video great, but wanted to try out MPEG4. I notice
> > >that there is no MPEG4*RTPSink class. What will it take to implement one?
> > >Before I try my hand at writing one, is there anything special going on,
> > >or should I just take the MPEG*RTPSink classes as examples and dive in?
> > 
> > Note that there are at least three different RTP payload formats for MPEG-4:
> >          1/ video/MP4V-ES (defined in RFC 3016)
> >          2/ audio/MPEG4-GENERIC (defined in 
> > <http://www.ietf.org/internet-drafts/draft-ietf-avt-mpeg4-simple-07.txt>)
> >          3/ audio/MP4A-LATM (defined in RFC 3016)
> > Most MPEG-4 RTP sessions use 1/ and 2/.  (3/ is used mainly for 3GPP audio.)
> > 
> > A RTPSink for 1/ would be very easy.  In fact, you could just use the 
> > existing "SimpleRTPSink" - e.g.
> >          SimpleRTPSink::createNew(*env, destGS, 96, 90000, "video", "MP4V-ES");
> > (assuming that you use the dynamic payload format # 96 for video).  (With 
> > the current implementation of "SimpleRTPSink", the RTP "M" bit won't be set 
> > properly, but I'll fix that shortly.)
> 
> I replaced
> videoSink = MPEGVideoRTPSink::createNew(*env, &rtpGroupsock);
> with 
> videoSink = SimpleRTPSink::createNew(*env, &rtpGroupsock, 96, 90000, 
> "video", "MP4V-ES");
> in testMPEGVideoStreamer.cpp. 
> Two problems im running into:
> first, the streamer halts after a short while (~15 seconds) with 
> the error
> StreamParser internal error (100000 + 4 > 100000)
> Its from liveMedia/StreamParser.cpp EnsureValidBytes()
> I'm guessing this might be due to streaming data that the parser is not 
> recognizing, i.e. parsing thus emptuing the buffer. Though Im just 
> guessing. I'm sending the data from mencoder's output using the video 
> codec mpeg4 with no sound. Is this 
> 
> 
> 
> 
> > 
> > A general implementation of a RTPSink for 2/ (supporting arbitrary 'modes', 
> > and interleaving) would be very difficult.  (OTOH, a simpler 
> > implementation, that supported just the non-interleaved "aac-hbr" mode (the 
> > most commonly-used mode), would merely be "difficult".)
> > 
> > A RTPSink for 3/ would be fairly easy.
> > 
> >          Ross.
> > 
> > 
> > 
> > 
> > -------------------------------------------------------
> > This sf.net email is sponsored by:ThinkGeek
> > Welcome to geek heaven.
> > http://thinkgeek.com/sf
> > _______________________________________________
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> > Live-devel(at)lists.sourceforge.net
> > https://lists.sourceforge.net/lists/listinfo/live-devel
> > 
> > 
> 
> 
> 
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X-Original-Date: Wed, 23 Apr 2003 13:11:24 -0700
Date: Wed, 23 Apr 2003 13:11:24 -0700


>I replaced
>videoSink = MPEGVideoRTPSink::createNew(*env, &rtpGroupsock);
>with
>videoSink = SimpleRTPSink::createNew(*env, &rtpGroupsock, 96, 90000,
>"video", "MP4V-ES");
>in testMPEGVideoStreamer.cpp.

Yes, that's good.

However, your problem arises *before* data gets to the RTP sink:

>Also, I'm using the stdio hack to send the live stream:
>mencoder -tv
>on:driver=v4l:device=/dev/video2:input=3:norm=ntsc:width=640:height=480
>-ovc lavc -lavcopts vcodec=mpeg4 -nosound -o /dev/stdout |
>../live/testProgs/testMPEGVideoStreamer

That won't work, because the "testMPEGAudioVideoStreamer" code does the 
following: (i) parses the input stream into MPEG-1 or 2 video frames (using 
"MPEGVideoStreamFramer"), and (ii) transmits these video frames using a 
MPEGVideoRTPSink.

Just changing (ii) won't be enough to get what you want.

In addition to changing (ii) (the easy part), you also need to change 
(i).  You will need a new 'framer' (i.e., parser) filter class (e.g., 
called "MPEG4ESVideoStreamFramer") that parses the input stream into MPEG-4 
video frames.  The good news here is that you can probably use the existing 
"MPEGVideoStreamFramer" code as a model, because the structure of MPEG-4 
video frames is somewhat similar to that of MPEG-1 or 2 video frames.

         Ross.



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X-Original-Date: Wed, 23 Apr 2003 13:29:35 -0700
Date: Wed, 23 Apr 2003 13:29:35 -0700


>I added MP4V-ES to lookupPayloadFormat() in
>MediaSession.cpp.
>
>case 27: {temp = "MP4V-ES"; rtpTimestampFrequency = 90000; break;}

No.  Bad!  Very bad!  The static RTP payload format code 27 is unassigned 
(see 
<http://www.ietf.org/internet-drafts/draft-ietf-avt-profile-new-13.txt>, 
Table 5).  You shouldn't use this for MP4V-ES.  Instead, use a *dynamic* 
payload format code (one that's >= 96).  Since you're already using 96 on 
the sender side (when you create a "SimpleRTPSink"), then presumably 96 is 
what you'll use.

Because you're using a dynamic payload type, you should be using (at the 
receiving end) a SDP description with the following lines:
         m=video 8888 RTP/AVP 96
         a=rtpmap:96 MP4V-ES/90000

Note the "a=rtpmap:" line, which describes which codec will be using the 
dynamic payload type 96.

         Ross.



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X-Original-Date: Wed, 23 Apr 2003 18:16:40 -0700
Date: Wed, 23 Apr 2003 18:16:40 -0700

OK, I've now added a new member function - to both "RTPSource" and 
"RTCPInstance" - for registering an optional 'auxillary read handler':

	void setAuxilliaryReadHandler(AuxHandlerFunc* handlerFunc,
					void* handlerClientData);

where "AuxHandlerFunc" is defined as
	typedef void AuxHandlerFunc(void* clientData, unsigned char* packet,
					unsigned packetSize);

If you register such a handler function, then it will be called each time a 
new packet is read.

If you register the same handler function with different "RTPSource"s or 
"RTCPInstance"s, then you can use the "clientData" parameter to distinguish 
between them.

This should give you the functionality that you're looking for.

	Ross.



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X-Original-Date: Thu, 24 Apr 2003 12:56:45 +0530
Date: Thu, 24 Apr 2003 12:56:45 +0530



hey

i've developed my own RTP/RTSP stack for a client end with MP3 playing 
support... and i want to test this with an independent server. can i use the 
testMP3streamer as my server application. If so what files will i need to 
add or do i just build everything and run the testMP3Streamer.

Does the lib respond to RTSP requests.I don't think RTSPServer supports PLAY 
and Pause as of yet.

regards
Vikram




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>i've developed my own RTP/RTSP stack for a client end with MP3 playing 
>support... and i want to test this with an independent server. can i use 
>the testMP3streamer as my server application.

Yes.  Just be sure to uncomment the line
         #define IMPLEMENT_RTSP_SERVER 1
in "testMP3Streamer.cpp", before compiling it.

Other LIVE.COM test programs with a built-in RTSP servers are 
"testMPEGAudioVideoStreamer", "testMPEGVideoStreamer", and "vobStreamer".

>Does the lib respond to RTSP requests.I don't think RTSPServer supports 
>PLAY and Pause as of yet.

OPTIONS, DESCRIBE, SETUP, TEARDOWN, and PLAY is supported, but not PAUSE.

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>



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X-Original-Date: Thu, 24 Apr 2003 18:26:05 +0800
Date: Thu, 24 Apr 2003 18:26:05 +0800


Hi!  I can't seem to find the library files generated when I compiled 
liveMedia in Linux.  I looked in the live directory and /usr/lib 
/usr/local/lib.
Where are they located, or are the executables statically linked then ?  :)
Thanks

Mildred






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X-Original-Date: Thu, 24 Apr 2003 18:42:15 +0800
Date: Thu, 24 Apr 2003 18:42:15 +0800


do you think it's possible to build an application using oRTP without an 
OS?
for example, a voice app using rtp on top of a net186 board?
Thanks  :)

Mildred




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X-Original-Date: Thu, 24 Apr 2003 10:57:01 -0400 (EDT)
Date: Thu, 24 Apr 2003 10:57:01 -0400 (EDT)

Hi... I would like to tahnk you for youe help during
my final year project (RTP).  Now since it is done I
woulk like to remove my name from live's mailing list.

Could you do it please.

Regards
Jocelyn

__________________________________________________________
Lèche-vitrine ou lèche-écran ?
magasinage.yahoo.ca


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X-Original-Date: Thu, 24 Apr 2003 09:06:22 -0700
Date: Thu, 24 Apr 2003 09:06:22 -0700

At 03:42 AM 4/24/03, you wrote:

>do you think it's possible to build an application using oRTP without an OS?

("oRTP" is not our product.)

However, the 'operating system'-level functionality required for porting 
the LIVE.COM Streaming Media libaries are quite modest.  As long as you have:

   1/ Standard POSIX (i.e., 'Berkeley') network sockets (for the 
"groupsock" library)
   2/ select() (for the "BasicUsageEnvironment" library)
   3/ gettimeofday()
   4/ Dynamic memory management (i.e., new/delete (C++) and malloc/free (C))

you should be able to port the code to your system.

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>



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X-Original-Date: Thu, 24 Apr 2003 09:09:14 -0700
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>Hi!  I can't seem to find the library files generated when I compiled 
>liveMedia in Linux.  I looked in the live directory and /usr/lib 
>/usr/local/lib.

Run
         find live -name 'lib*.a' -print

     Ross.



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why is rtsp in the test programs used only for multicast?  how to modify 
it for unicast?
Thanks

mildred



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Hi,

If one wants to stream MPEG-4 data, is it required that the audio and video be multiplexed into systems stream? Can Audio and Video be streamed independently over the network, RFC 3016 does make it seem possible, but what would be the overhead at the decoder end to sync the packets back again.

Is there an RFC that defines the payload format for MPEG-4 system streams?

Thanks & Regards,
Manu



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>If one wants to stream MPEG-4 data, is it required that the audio and 
>video be multiplexed into systems stream? Can Audio and Video be streamed 
>independently over the network

Yes, and in fact, that is usually how it is done.

Most MPEG-4 streams are sent using two separate RTP streams:
         1/ video/MP4V-ES (defined in RFC 3016)
         2/ audio/MPEG4-GENERIC (defined in 
<http://www.ietf.org/internet-drafts/draft-ietf-avt-mpeg4-simple-07.txt>)

         Ross.



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>why is rtsp in the test programs used only for multicast?

Because a RTSP server for a multicast stream is much easier to implement 
(because for multicast, there is just a single stream, sent from the same 
source port to the same destination address & port, regardless of how many 
clients request the stream).

>how to modify it for unicast?

See <http://www.live.com/liveMedia/faq.html#support>

For now, if you want a unicast RTSP server, I suggest using Apple's "Darwin 
Streaming Server", or RealNetworks' "Helix Server".

         Ross.



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Thanks Ross,

But then how is the resynchronization done at decoder end? Does RTP do that while providing ES streams?

Thanks & Regards,
Manu

>>> finlayson(at)live.com 04/25/03 12:58PM >>>

>If one wants to stream MPEG-4 data, is it required that the audio and 
>video be multiplexed into systems stream? Can Audio and Video be streamed 
>independently over the network

Yes, and in fact, that is usually how it is done.

Most MPEG-4 streams are sent using two separate RTP streams:
         1/ video/MP4V-ES (defined in RFC 3016)
         2/ audio/MPEG4-GENERIC (defined in 
<http://www.ietf.org/internet-drafts/draft-ietf-avt-mpeg4-simple-07.txt>)

         Ross.



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>But then how is the resynchronization done at decoder end?

This is done in the usual way: Using RTP timestamps, along with RTP->NTP 
timestamp mappings in periodic RTCP "Sender Report" packets.  See the 
RTP/RTCP specification: 
<http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-new-12.txt>

Note that the "LIVE.COM Streaming Media" RTP reception code 
("*RTPSource.cpp") implements this automatically - so you will get 
accurate, synchronized 'presentation times' whenever you read from a 
"RTPSource" object

         Ross.



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X-Original-Date: Fri, 25 Apr 2003 11:13:58 -0700
Date: Fri, 25 Apr 2003 11:13:58 -0700

Hi,

Maybe this is not the best place to ask this question, but I would
appreciate if somebody could help me with this.

Basically, I am streaming MP3 to MPlayer through live. I wanted to know
where in the MPlayer source code can one check if a received RTP packet is
played or not by the MPlayer.

Thanks,
Rohit



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X-Original-Date: Fri, 25 Apr 2003 13:34:35 -0700
Date: Fri, 25 Apr 2003 13:34:35 -0700


>Basically, I am streaming MP3 to MPlayer through live. I wanted to know
>where in the MPlayer source code can one check if a received RTP packet is
>played or not by the MPlayer.

The question was rather vague... But MPlayer reads incoming RTP packets in 
"libmpdemux/demux_rtp.cpp".  (Note, in particular, the function "getBuffer()".)

This MPlayer code uses the underlying "LIVE.COM Streaming Media" libraries 
in exactly the same as "openRTSP", and other RTSP clients.

         Ross.



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X-Original-Date: Fri, 25 Apr 2003 20:49:10 -0700
Date: Fri, 25 Apr 2003 20:49:10 -0700

Ross,
Thanks. I guess the question was vague. What I really wanted to know was the
sequence numbers of the RTP packets which are (or are not) played by MPlayer
(some packets may be received by Live, but may be too late to be played by
MPlayer, so MPlayer does not play them).

Also, if MPlayer is playing a streaming MP3 file (with a certain bitrate),
how does it decide to not play some packets? I mean, how late does a packet
have to be for MPlayer not to play it? In other words, from MPlayer's point
of view, how is the deadline of a packet (to be played) decided?

Thanks,
Rohit


----- Original Message -----
From: {Ross Finlayson} <EMAIL-ADDRESS-HIDDEN>
To: <EMAIL-ADDRESS-HIDDEN>
Sent: Friday, April 25, 2003 1:34 PM
Subject: Re: [Live-devel] Question on Mplayer


>
> >Basically, I am streaming MP3 to MPlayer through live. I wanted to know
> >where in the MPlayer source code can one check if a received RTP packet
is
> >played or not by the MPlayer.
>
> The question was rather vague... But MPlayer reads incoming RTP packets in
> "libmpdemux/demux_rtp.cpp".  (Note, in particular, the function
"getBuffer()".)
>
> This MPlayer code uses the underlying "LIVE.COM Streaming Media" libraries
> in exactly the same as "openRTSP", and other RTSP clients.
>
>          Ross.
>
>
>
> -------------------------------------------------------
> This sf.net email is sponsored by:ThinkGeek
> Welcome to geek heaven.
> http://thinkgeek.com/sf
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X-Original-Date: Sun, 27 Apr 2003 14:38:04 -0700
Date: Sun, 27 Apr 2003 14:38:04 -0700


>Thanks. I guess the question was vague. What I really wanted to know was the
>sequence numbers of the RTP packets which are (or are not) played by MPlayer
>(some packets may be received by Live, but may be too late to be played by
>MPlayer, so MPlayer does not play them).
>
>Also, if MPlayer is playing a streaming MP3 file (with a certain bitrate),
>how does it decide to not play some packets? I mean, how late does a packet
>have to be for MPlayer not to play it? In other words, from MPlayer's point
>of view, how is the deadline of a packet (to be played) decided?

*Every* incoming RTP packet that the underlying LIVE.COM code receives is 
passed up to the MPlayer code.  No matter how 'late' a packet may be.

(The only exception to this is if packets get reordered in the network, 
with an out-of-order packet getting delayed.  E.g., if the LIVE.COM RTP 
code receives packet "n", and then packet "n+2", it will wait 100ms for the 
missing packet "n+1" to arrive.  If packet "n+1" doesn't arrive within this 
time, then the RTP code will give up on it; if packet "n+1" happens to 
arrive later, it'll get thrown away.  This situation, however, occurs very 
rarely.)

         Ross.



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X-Original-Date: Mon, 28 Apr 2003 22:14:41 -0700
Date: Mon, 28 Apr 2003 22:14:41 -0700

Thanks.

Rohit
----- Original Message -----
From: {Ross Finlayson} <EMAIL-ADDRESS-HIDDEN>
To: <EMAIL-ADDRESS-HIDDEN>
Sent: Sunday, April 27, 2003 2:38 PM
Subject: Re: [Live-devel] Question on Mplayer


>
> >Thanks. I guess the question was vague. What I really wanted to know was
the
> >sequence numbers of the RTP packets which are (or are not) played by
MPlayer
> >(some packets may be received by Live, but may be too late to be played
by
> >MPlayer, so MPlayer does not play them).
> >
> >Also, if MPlayer is playing a streaming MP3 file (with a certain
bitrate),
> >how does it decide to not play some packets? I mean, how late does a
packet
> >have to be for MPlayer not to play it? In other words, from MPlayer's
point
> >of view, how is the deadline of a packet (to be played) decided?
>
> *Every* incoming RTP packet that the underlying LIVE.COM code receives is
> passed up to the MPlayer code.  No matter how 'late' a packet may be.
>
> (The only exception to this is if packets get reordered in the network,
> with an out-of-order packet getting delayed.  E.g., if the LIVE.COM RTP
> code receives packet "n", and then packet "n+2", it will wait 100ms for
the
> missing packet "n+1" to arrive.  If packet "n+1" doesn't arrive within
this
> time, then the RTP code will give up on it; if packet "n+1" happens to
> arrive later, it'll get thrown away.  This situation, however, occurs very
> rarely.)
>
>          Ross.
>
>
>
> -------------------------------------------------------
> This sf.net email is sponsored by:ThinkGeek
> Welcome to geek heaven.
> http://thinkgeek.com/sf
> _______________________________________________
> Live-devel mailing list
> Live-devel(at)lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/live-devel



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X-Original-Date: Tue, 29 Apr 2003 09:44:06 +0200 (CEST)
Date: Tue, 29 Apr 2003 09:44:06 +0200 (CEST)


Hello,

I try modified the live.com libraries to work fine 
with video4linux stream source.

Using TCP/IP stack as buffer between streamer and grabber.

I bit hacked together new class deviceSource. and tryied to stream.
But I get some errors from demuxer. 

First version used gotDATA function (select service)
and worked quite fine when all data arrived. 

Then I rewrote it similar to UDP example.

I know the code is not very nice, its proof of concept only.

Please would you take a look at the rest and tell me what I have done 
wrong ?

I suscpect that some internal buffer is not able to get more data.
But I dont know how to get in my class some clue to handle this situation.

Should I build new ring buffer ? Or is there any other elegant solution?

Btw. there is no fine (free) software to do multicast streaming.
(ffserver is reported to be broken) and maybe this is the chance for
live.com :)

Regards

Rudolf

I run this as one process

 ffmpeg -b 1000 -f mpeg -s 352x288 - | nc ruik -p 60000 -

and this one as another.

 ./testMPEGAudioVideoStreamer
(uses the Devicefile)


Beginning streaming...
Beginning to read from file...
CTU 100000
CTU 95904
CTU 95904
CTU 95904
CTU 95904
CTU 95904
CTU 95904
CTU 95904
CTU 95904
CTU 95904
CTU 95904
CTU 95904
CTU 95904
CTU 95904
CTU 95904
CTU 95904
CTU 95904
CTU 95904
CTU 95904
CTU 95904
CTU 18080
CTU 18080
CTU 18080
CTU 18080
CTU 18080
StreamParser::afterGettingBytes() warning: read 1696 bytes; expected no more
than 1696
CTU 18080
StreamParser::afterGettingBytes() warning: read 0 bytes; expected no more
than 0
CTU 18080
StreamParser::afterGettingBytes() warning: read 0 bytes; expected no more
than 0
CTU 18080
StreamParser::afterGettingBytes() warning: read 0 bytes; expected no more
than 0
CTU 18080
StreamParser::afterGettingBytes() warning: read 0 bytes; expected no more
than 0
CTU 18080
StreamParser::afterGettingBytes() warning: read 0 bytes; expected no more
than 0
CTU 18080
StreamParser::afterGettingBytes() warning: read 0 bytes; expected no more
than 0
CTU 18080
StreamParser::afterGettingBytes() warning: read 0 bytes; expected no more
than 0
CTU 18080
StreamParser::afterGettingBytes() warning: read 0 bytes; expected no more
than 0
CTU 18080
StreamParser::afterGettingBytes() warning: read 0 bytes; expected no more
than 0
CTU 99996
CTU 97948
CTU 97948
CTU 89756
CTU 85660
CTU 81564
CTU 77468
CTU 73372
CTU 69276
CTU 65180
CTU 61084
CTU 56988
CTU 52892
CTU 48796
CTU 44700
CTU 40604
CTU 40604
CTU 32412
CTU 30364
CTU 28316
CTU 24220
CTU 20124
CTU 16028
CTU 11932
CTU 9884
 etc...
/**********
This library is free software; you can redistribute it and/or modify it under
the terms of the GNU Lesser General Public License as published by the
Free Software Foundation; either version 2.1 of the License, or (at your
option) any later version. (See <http://www.gnu.org/copyleft/lesser.html>.)

This library is distributed in the hope that it will be useful, but WITHOUT
ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS
FOR A PARTICULAR PURPOSE.  See the GNU Lesser General Public License for
more details.

You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
**********/
// "liveMedia"
// Copyright (c) 1996-2002 Live Networks, Inc.  All rights reserved.
// A template for a MediaSource encapsulating an audio/video input device
// Implementation

#include "DeviceSource.hh"
#include "GroupsockHelper.hh" 

#include <string.h>
#include <sys/types.h>
#include <sys/socket.h>
#include <netinet/in.h>
#include <sys/poll.h>
 

DeviceSource*
DeviceSource::createNew(UsageEnvironment& env
			) {
  return new DeviceSource(env);
}

DeviceSource::DeviceSource(UsageEnvironment& env)
  : FramedSource(env) {
 
 struct sockaddr_in dest_addr;   // will hold the destination addr
     sockfd = socket(AF_INET, SOCK_STREAM, 0); // do some error checking!
	 
     dest_addr.sin_family = AF_INET;          // host byte order
     dest_addr.sin_port = htons(60000);   // short, network byte order
     dest_addr.sin_addr.s_addr = inet_addr("147.32.126.11");

     memset(&(dest_addr.sin_zero), 0, 8);  // zero the rest of the struct
						 
     // don't forget to error check the connect()!
     connect(sockfd, (struct sockaddr *) &dest_addr, sizeof(struct sockaddr));
  // Any initialization of the device would be done here
cti=0;

}

DeviceSource::~DeviceSource() {
}

float DeviceSource::getPlayTime(unsigned numFrames) const {
// return (float)(numFrames*(fPlayTimePerFrame/1000000.0));
return 0.0;
}

int DeviceSource::gotDATA(int timeout) {
struct pollfd ufds[1];
ufds[0].events=POLLIN;
ufds[0].fd=sockfd;
poll(ufds, 1,timeout);
return (ufds[0].revents&POLLIN);
}


void DeviceSource::incomingConnectionHandler(void* instance, int /*mask*/) {
  DeviceSource* server = (DeviceSource*)instance;  
  
//printf("GOT data!\n");

//server->;
server->deliverFrame();

}
    

void DeviceSource::doGetNextFrame() {
   envir().taskScheduler().turnOnBackgroundReadHandling(sockfd,
         (TaskScheduler::BackgroundHandlerProc*)&incomingConnectionHandler,
	                                                    this);
}

void DeviceSource::deliverFrame() {

printf("CTU %d\n",fMaxSize);
if (fMaxSize<4096) 
{
fFrameSize = read(sockfd, fTo, fMaxSize);
}
else
{ 
fFrameSize = read(sockfd, fTo, 4096);
}

//FramedSource::afterGetting(this);
  // Switch to another task, and inform the reader that he has data:
//  nextTask() = envir().taskScheduler().scheduleDelayedTask(0,
//				(TaskFunc*)FramedSource::afterGetting, this);

afterGetting(this);

cti=0;
}


//  nextTask()
//    = envir().taskScheduler().scheduleDelayedTask(0, (TaskFunc*)afterGetting,
//						  this);
//}



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Subject: [Live-devel] Changing Video Format/Codec during same RTP session
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X-Original-Date: Tue, 29 Apr 2003 11:09:52 +0200
Date: Tue, 29 Apr 2003 11:09:52 +0200

Hello,

Is it possible to change the video Format of a stream during the same 
RTP session ?
In other words, if I'm getting raw video from a whatever source, can I 
start encode the frames in MPEG and send them using RTP for a given 
period than change the encoder (to H263 or else) and continue to use the 
same RTP session (with corresponding timing) to transmit remaining 
frames with the new Format ? Or must I restart a new RTP session for this ?

I'm working on component model that allows the dynamic change of the 
implementation of a component Interface.
For example, I want to have a component  (an object, if you want)   
(AnyFormat)VideoStreamFramer that can change
dynamically the format of video frames it sends through an RTPSink.

After having a glance at the live library and testProgs source code I 
noticed that I need also to have (at least) an
(AnyFormat)VideoRTPSink.

Do you think that it's hard to transorm the Framer internal data (Parser 
.... ) while preserving the video timing to allow this dynamic change ?

Thanks

-- 
Djalel CHEFROUR, PhD student, ARMOR project at IRISA.
IRISA-INRIA, Campus de Beaulieu, 35042 Rennes cedex, France
Tel: +33 02 99 84 72 64, Fax: +33 02 99 84 71 71
http://www.irisa.fr/armor/Armor-Ext/




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Subject: Re: [Live-devel] video live streaming problem
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X-Original-Date: Tue, 29 Apr 2003 10:51:42 -0700
Date: Tue, 29 Apr 2003 10:51:42 -0700

First, note that - in "testMPEGAudioVideoStreamer" - the input to the 
"MPEGVideoStreamerFramer" and "MPEGAudioStreamFramer" has to be encoded 
MPEG-1 or MPEG-2 video data (respectively).  If your input data is MPEG 
data, then you should be OK.  But if it's raw (i.e., uncompressed) 
video/audio data, then your code won't work unless you insert MPEG encoders 
into the chain.

If your "DeviceSource" is reading encoded MPEG data, then your code look 
mostly OK, except that - in "DeviceSource::deliverFrame()" - you should 
schedule "afterGetting" via 
"envir().taskScheduler().scheduleDelayedTask(...)" (as shown in the code), 
rather than calling "afterGetting()" directly.

	Ross.



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X-Original-Date: Tue, 29 Apr 2003 11:00:54 -0700
Date: Tue, 29 Apr 2003 11:00:54 -0700


>Is it possible to change the video Format of a stream during the same RTP 
>session ?

Yes, you can do this, but you will need to close (using "Medum::close()") 
the "RTPSink" and "RTCPInstance" objects, and then create new ones (for the 
new payload format).  (You can leave the existing RTP and RTCP "groupsock" 
objects in place.)

         Ross.



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Subject: Re: [Live-devel] video live streaming problem
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X-Original-Date: Tue, 29 Apr 2003 20:27:49 +0200 (CEST)
Date: Tue, 29 Apr 2003 20:27:49 +0200 (CEST)


Oh yeah. Its MPEG stream.

It works (sometimes :)
But it prints that messages (even with the envir stuff there)

Sometimes It will print smth like saw ... bla bla bad header
or even Internal Parser Error. I played with BANK_SIZE
but I cant get it work.

And thats the point I cant solve.

Should I implement some other cache there?

I know that doNextFrame is scheduled from someone.

If it enters doNextFrame and I have no data to feed 
I cant simply return without the sched. stuff because
it wont return to that function.

I dont know how to reschedule the doNextframe without notifiing the
the consumer about data.

I hope its clearer now.

If I finish this I can give it to you. (GPL or LGPL) Its my private
project and I dont need $ for it. I think this part should be in live.com
library because there is a gap in streaming media with linux.

Regards

Rudolf




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X-Original-Date: Tue, 29 Apr 2003 13:20:02 -0700
Date: Tue, 29 Apr 2003 13:20:02 -0700


>If it enters doNextFrame and I have no data to feed
>I cant simply return without the sched. stuff because
>it wont return to that function.

Yes it will, because when data later arrives, your 
"incomingConnectionHandler()" function will get called, which in turn will 
call "deliverFrame()" - feeding data back to the client.

Your "DeviceSource::doGetNextFrame()" function looks OK.

What I suggest you do is *first* debug your code by having your 
"DeviceSource" feed directly into a "FileSink" (rather than a framer and 
RTP sink).  If you do this, you'll get an output file that you can examine 
(and try playing), to see what's going wrong with your code.

I.e., start by doing:
         DeviceSource -> FileSink
and then later, *only after you get that working*:
         DeviceSource -> MPEGVideoStreamFramer -> MPEGVideoRTPSink
(In fact, if your "DeviceSource" reads discrete MPEG video frames (rather 
than an arbitrary sequence of bytes), then you can omit the 
"MPEGVideoStreamFramer" from the chain.)

RYHCSC ("Remember, You Have Complete Source Code")

         Ross.



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X-Original-Date: Tue, 29 Apr 2003 14:03:40 -0700
Date: Tue, 29 Apr 2003 14:03:40 -0700

(I don't have a Mac, so I can't yet experiment with this for myself.)

Apple has just released a new version of its "iTunes" software 
<http://www.apple.com/itunes/> (currently for Macintosh only) that makes it 
possible to stream AAC or MP3 audio to other computers on a LAN.

If - as is likely - this streaming uses the open standard RTSP/RTP 
protocols, then it may be possible for MPlayer (with the LIVE.COM RTSP/RTP 
streaming support: <http://www.live.com/mplayer/>) to receive and play 
these streams, by giving MPlayer an appropriate "rtsp://" URL.

(Alternatively, it might be possible to use "openRTSP" to record such a 
stream to a file.)

I'd be interested to learn if anyone has tried playing with this, to see if 
if works.

	Ross. 



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X-Original-Date: Wed, 30 Apr 2003 11:47:53 +0800
Date: Wed, 30 Apr 2003 11:47:53 +0800


Hi!  I'm sorry, but I often encounter these terms.
What's the difference between elementary streams, system streams and 
transport streams?
Thanks






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X-Original-Date: Tue, 29 Apr 2003 23:31:52 -0700
Date: Tue, 29 Apr 2003 23:31:52 -0700


Mildred,

> Hi!  I'm sorry, but I often encounter these terms.
> What's the difference between elementary streams, system streams and  
> transport streams?

No worries, and no need to be sorry :-)

In MPEG parlance, elementary streams are the component streams (i.e.  
audio and video) of a system stream.  Transport streams are MPEG  
streams set up to stream over a high error probability system (i.e. a  
network, where packets could be dropped), vs. programs streams, which  
are MPEG streams set up to stream from a low error probability system  
(i.e. a CDROM or hard disk).

There are a lot of details, i've only touched on the basics of course.   
I can personally recommend the following books if you want to really  
know about MPEG:

ATM and MPEG2: Integrating Digital Video into Broadband Networks

http://www.amazon.com/exec/obidos/tg/detail/-/0132437007/ 
qid=1051684006/sr=1-2/ref=sr_1_2/102-7734067-4930555?v=glance&s=books

The MPEG Handbook

http://www.amazon.com/exec/obidos/ASIN/0240516567/qid%3D1051684217/ 
sr%3D11-1/ref%3Dsr%5F11%5F1/102-7734067-4930555

Regards,

John

Falling You - exploring the beauty of voice and sound
New EP, "hope thrown down," available now at
http://www.mp3.com/fallingyou













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X-Original-Date: Tue, 29 Apr 2003 23:31:52 -0700
Date: Tue, 29 Apr 2003 23:31:52 -0700


Mildred,

> Hi!  I'm sorry, but I often encounter these terms.
> What's the difference between elementary streams, system streams and  
> transport streams?

No worries, and no need to be sorry :-)

In MPEG parlance, elementary streams are the component streams (i.e.  
audio and video) of a system stream.  Transport streams are MPEG  
streams set up to stream over a high error probability system (i.e. a  
network, where packets could be dropped), vs. programs streams, which  
are MPEG streams set up to stream from a low error probability system  
(i.e. a CDROM or hard disk).

There are a lot of details, i've only touched on the basics of course.   
I can personally recommend the following books if you want to really  
know about MPEG:

ATM and MPEG2: Integrating Digital Video into Broadband Networks

http://www.amazon.com/exec/obidos/tg/detail/-/0132437007/ 
qid=1051684006/sr=1-2/ref=sr_1_2/102-7734067-4930555?v=glance&s=books

The MPEG Handbook

http://www.amazon.com/exec/obidos/ASIN/0240516567/qid%3D1051684217/ 
sr%3D11-1/ref%3Dsr%5F11%5F1/102-7734067-4930555

Regards,

John

Falling You - exploring the beauty of voice and sound
New EP, "hope thrown down," available now at
http://www.mp3.com/fallingyou













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X-Original-Date: Wed, 30 Apr 2003 09:42:36 +0200
Date: Wed, 30 Apr 2003 09:42:36 +0200

>
>
> Yes, you can do this, but you will need to close (using 
> "Medum::close()") the "RTPSink" and "RTCPInstance" objects, and then 
> create new ones (for the new payload format).  (You can leave the 
> existing RTP and RTCP "groupsock" objects in place.)
>
>         Ross.


OK.

Is there any H26xVideoStreamer (or MPEG4 or JPEG) planned soon or have I 
to make them my self ?
In the latter case, do you think that it's hard or it takes lotof time ?




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--------------080403040208040707080607
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> Is there any H26xVideoStreamer (or MPEG4 or JPEG) planned soon or have 
> I to make them my self ?
> In the latter case, do you think that it's hard or it takes lotof time ?


Sorry, I mean   xxxxVideoStreamFramer

--------------080403040208040707080607
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<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
  <title></title>
</head>
<body>
<blockquote type="cite" cite="mid3EAF7E6C.30708(at)irisa.fr">Is there any H26xVideoStreamer
(or MPEG4 or JPEG) planned soon or have I  to make them my self ? <br>
In the latter case, do you think that it's hard or it takes lotof time ? <br>
 </blockquote>
<br>
Sorry, I mean &nbsp; xxxx<b>VideoStreamFramer</b><br>
</body>
</html>

--------------080403040208040707080607--



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From <EMAIL-ADDRESS-HIDDEN> Wed Apr 30 02:58:47 2003
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X-Original-Date: Wed, 30 Apr 2003 02:52:30 -0700
Date: Wed, 30 Apr 2003 02:52:30 -0700

At 01:11 AM 4/30/03, you wrote:
>>Is there any H26xVideoStreamer (or MPEG4 or JPEG) planned soon or have I 
>>to make them my self ?
>>In the latter case, do you think that it's hard or it takes lotof time ?
>
>Sorry, I mean   xxxxVideoStreamFramer

No, there are no near-term plans to develop any of these framers (as these 
have not been requested by clients).

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>



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From <EMAIL-ADDRESS-HIDDEN> Wed Apr 30 10:17:04 2003
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X-Original-Date: Wed, 30 Apr 2003 18:54:36 +0200
Date: Wed, 30 Apr 2003 18:54:36 +0200

Hi,

We propose to add the possibility of registering a packet handler for
processed packets (that is after the packets are re-ordered and parsed)
to the actual auxilliary handler proposed for raw packets.

In a similar way, the following code snippet could be inserted in
MultiFramedRTPSource. The handler would be called with a handle to the
current BufferedPacket, before the nextPacket->use(...) method is
called.

if(fRTPInterface.fPacketReadHandler!=NULL)
(*(fRTPInterface.fPacketReadHandler))(fRTPInterface.fPacketReadClientData,nextPacket);

The Buffered Packet class could also be extended to provide access to
the timestamp.

unsigned rtpTimestamp() const {return fRTPTimestamp;};

Thanks,

Cezar & Romulus


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X-Original-Date: Wed, 30 Apr 2003 10:33:59 -0700
Date: Wed, 30 Apr 2003 10:33:59 -0700


>We propose to add the possibility of registering a packet handler for
>processed packets (that is after the packets are re-ordered and parsed)
>to the actual auxilliary handler proposed for raw packets.

Romulus,

I'm not inclined to add a hack like this, because the best way to handle 
processed RTP frames is in the Medium (filter or sink) that is already 
reading from the RTPSource.  I.e., the regular data chain will already be 
handling the incoming RTP frames, so any special processing that you need 
to do on these frames should be done the - after you call "getNextFrame()" 
on the RTPSource.

>The Buffered Packet class could also be extended to provide access to
>the timestamp.
>
>unsigned rtpTimestamp() const {return fRTPTimestamp;};

Note that "RTPSource" has a member function
         unsigned curPacketRTPTimestamp() const
that returns the RTP timestamp that was in the most recently received RTP 
packet.  There are similar member functions that return the RTP sequence 
number, the RTP payload format, and whether or not the "M" bit was sent.

See "liveMedia/include/RTPSource.hh".

         Ross.



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From <EMAIL-ADDRESS-HIDDEN> Wed Apr 30 13:34:12 2003
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  receivable/playable by MPlayer?
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X-Original-Date: Wed, 30 Apr 2003 13:29:25 -0700
Date: Wed, 30 Apr 2003 13:29:25 -0700

Some more information I learned about Apple's "iTunes" 
streaming:  Apparently they access online music streams using their own 
(presumably secret) protocol called "DAAP", rather than regular RTSP:
	http://features.slashdot.org/comments.pl?sid=62547&cid=5844931

(The actual audio stream might still be using RTP - I dunno.)

Disappointing, but not too much of a surprise...

	Ross.



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X-Original-Date: Thu, 1 May 2003 10:58:31 -0700
Date: Thu, 1 May 2003 10:58:31 -0700

Ross,

Thanks a lot for the answer. This means that late packets are also played by
the MPlayer as long as reordering doesnt create issues (which it normally
will not).
In a sense, this really implies that audio clients are not really
'hard-realtime'. Is that right? Do you know if other audio clients also
behave in a similar manner or do they just drop 'late' packets?

Thanks,
Rohit

----- Original Message -----
From: {Ross Finlayson} <EMAIL-ADDRESS-HIDDEN>
To: <EMAIL-ADDRESS-HIDDEN>
Sent: Sunday, April 27, 2003 2:38 PM
Subject: Re: [Live-devel] Question on Mplayer


>
> >Thanks. I guess the question was vague. What I really wanted to know was
the
> >sequence numbers of the RTP packets which are (or are not) played by
MPlayer
> >(some packets may be received by Live, but may be too late to be played
by
> >MPlayer, so MPlayer does not play them).
> >
> >Also, if MPlayer is playing a streaming MP3 file (with a certain
bitrate),
> >how does it decide to not play some packets? I mean, how late does a
packet
> >have to be for MPlayer not to play it? In other words, from MPlayer's
point
> >of view, how is the deadline of a packet (to be played) decided?
>
> *Every* incoming RTP packet that the underlying LIVE.COM code receives is
> passed up to the MPlayer code.  No matter how 'late' a packet may be.
>
> (The only exception to this is if packets get reordered in the network,
> with an out-of-order packet getting delayed.  E.g., if the LIVE.COM RTP
> code receives packet "n", and then packet "n+2", it will wait 100ms for
the
> missing packet "n+1" to arrive.  If packet "n+1" doesn't arrive within
this
> time, then the RTP code will give up on it; if packet "n+1" happens to
> arrive later, it'll get thrown away.  This situation, however, occurs very
> rarely.)
>
>          Ross.
>
>
>
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> This sf.net email is sponsored by:ThinkGeek
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From <EMAIL-ADDRESS-HIDDEN> Fri May 2 03:17:14 2003
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X-Original-Date: Fri, 02 May 2003 11:46:12 +0200
Date: Fri, 02 May 2003 11:46:12 +0200

Hi Ross,

The current auxilliary handler is actually useful for, let's say, a
"dumb" proxy, that is one that needs just raw RTP packets. By looking at
the code, packet handling functions as following:
1. call the auxiliary handler
2. parse the raw packet and extract the RTP info
3. create a Buffered Packet
4. store it in the reordering buffer

What we actually need (and we think this could be useful to others) is
to access the already parsed and ordered RTP packets and not frames.
If you provide a way to access parsed/reordered BufferedPackets, the
BufferedPacket already contains the RTP timestamp, so why go elsewhere
to look for it ? Naturally it just needs to have an accessor method like
(getRTPTimestamp).

To conclude, we think that is can be useful to keep the actual
auxilliary handler and eventually make the call a bit later (that is,
after parsing the packet, to avoid double parsing) for anyone wanting to
access RAW RTP packets, but it can also be useful to have access to
parsed/reordered BufferPackets. Additionally, it can be useful to
have a way of setting the seqNoThreshold value from outside the library,
too.

thanks,

cezar & romulus


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X-Original-Date: Fri, 02 May 2003 09:33:20 -0700
Date: Fri, 02 May 2003 09:33:20 -0700


>What we actually need (and we think this could be useful to others) is
>to access the already parsed and ordered RTP packets and not frames.

Fortunately there's already a way to do this - using "SimpleRTPSource".  A 
"SimpleRTPSource" lets the client access the RTP payload as an opaque piece 
of data, without breaking it up into 'frames' at all.

When creating a "SimpleRTPSource", you need to specify the RTP payload 
format number, the RTP timestamp frequency, and the MIME type (e.g., 
"audio/PCMU").

See "MediaSession.cpp" for examples of  how "SimpleRTPSource" is used for 
receiving simple RTP payload formats (such as PCM audio, and MPEG transport 
streams).

>Additionally, it can be useful to
>have a way of setting the seqNoThreshold value from outside the library, too

You can do this using the function
         RTPSource::setPacketReorderingThresholdTime(uSeconds)

     Ross.



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From <EMAIL-ADDRESS-HIDDEN> Sat May 3 02:07:10 2003
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X-Original-Date: Sat, 03 May 2003 02:00:40 -0700
Date: Sat, 03 May 2003 02:00:40 -0700

FYI, the "RTSPClient" class has now been updated with support for the RTSP 
"PAUSE" command, plus the ability to seek to arbitrary times within the 
stream (using the "Range:" header with the "PLAY" command).

Thanks to Romulus Grigoras for contributing this.

	Ross.



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From <EMAIL-ADDRESS-HIDDEN> Sat May 3 21:39:21 2003
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Subject: [Live-devel] MPlayer as a SIP (IP telephony) client
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X-Original-Date: Sat, 03 May 2003 21:33:28 -0700
Date: Sat, 03 May 2003 21:33:28 -0700

FYI, the latest CVS version of MPlayer - with the "LIVE.COM Streaming 
Media" libraries <http://www.live.com/mplayer/> - can now play IP telephony 
sessions - i.e., those specified by a "sip:" URL.  For example, you could 
use MPlayer to listen in on multi-party IP telephony conferences.

Of course, because MPlayer is a receive-only application, you can listen in 
on such sessions - but you won't be able to talk back.   (If you want 
two-way communication on a SIP session, you should instead use a dedicated 
SIP 'softphone' application, such as "linphone".)

For example, here are a couple of SIP 'voicemail announcements' that you 
can now play using MPlayer:
	sip:1234(at)pbx.live.com
	sip:10001(at)fwd.pulver.com

(You might need to use the latest version (2003.05.03) of the LIVE.COM code 
for SIP to work properly.)

	Ross. 



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From <EMAIL-ADDRESS-HIDDEN> Sun May 4 15:02:24 2003
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X-Original-Date: Sun, 04 May 2003 14:57:28 -0700
Date: Sun, 04 May 2003 14:57:28 -0700

I recently ran across the following document
	http://www.mozilla.org/hacking/portable-cpp.html
This is a set of guidelines that the "Mozilla" project uses for their code.

It turns out that I also followed many of these guidelines (although not 
always religiously) when writing the "LIVE.COM Streaming Media" code - to 
try to make it portable across multiple platforms.

	Ross.



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X-Original-Date: Mon, 05 May 2003 14:51:02 +0200
Date: Mon, 05 May 2003 14:51:02 +0200

Hi,

>From: Ross Finlayson <finlayson(at)li...> 
>Re: Re: Adding a RTP handler   
>2003-05-02 09:33 
>>What we actually need (and we think this could be useful to others) is
>>to access the already parsed and ordered RTP packets and not frames.
                              
>Fortunately there's already a way to do this - using "SimpleRTPSource".  A 
>"SimpleRTPSource" lets the client access the RTP payload as an opaque piece 
>of data, without breaking it up into 'frames' at all.                             
>When creating a "SimpleRTPSource", you need to specify the RTP payload 
>format number, the RTP timestamp frequency, and the MIME type (e.g., 
>"audio/PCMU").                              
>See "MediaSession.cpp" for examples of  how "SimpleRTPSource" is used for 
>receiving simple RTP payload formats (such as PCM audio, and MPEG transport 
>streams).

Yes, but we still think that it's much more convenient to have access to
parsed, ordered packets through an auxilliary handler called at the
right moment.
It seems that extending the SimpleRTPSource implies modifications to
MediaSession, which is part of the library ... But this does not solve
our problem (get the packet and it's RTP info), as packets are still
handled by MultiFramedRTPSource that we should not touch...

thanks,

cezar & romulus


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Subject: Re: [Live-devel] Re: Adding a RTP handler
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X-Original-Date: Mon, 05 May 2003 10:15:04 -0700
Date: Mon, 05 May 2003 10:15:04 -0700


> >>What we actually need (and we think this could be useful to others) is
> >>to access the already parsed and ordered RTP packets and not frames.
>
> >Fortunately there's already a way to do this - using "SimpleRTPSource"
[...]

>Yes, but we still think that it's much more convenient to have access to
>parsed, ordered packets through an auxilliary handler called at the
>right moment.

An 'auxillary handler' might be "more convenient", but it would also be 
uglier, because it would be outside the normal data chain 
(Source->Sink).  So, I'm trying to understand what you're really trying to 
do, so I can see if there is any way for you to do what you want within the 
normal data chain.

What are you using as your regular data sink - i.e., the object that you 
call "startPlaying()" on?  This object receives one frame at a time - not 
one packet at a time.  However, it could also figure out packet boundaries 
- by calling "curPacketRTPSeqNum()" on the RTPSource, and checking when 
this changes.

Or is it the case that you really don't want your data sink receiving one 
frame at a time, but instead would prefer it to be receiving one complete 
RTP packet (minus the RTP header) at a time?

Or do you (effectively) want *two* data sinks: One that receives one frame 
at a time, and another that receives one RTP packet (minus the RTP header) 
at a time?

>It seems that extending the SimpleRTPSource implies modifications to
>MediaSession, which is part of the library

Would it help you at all if I were to 'virtualize' the code in 
"MediaSubsession::initiate()" (in "liveMedia/MediaSession.cpp") that 
creates a particular kind of "RTPSource", based on the medium name and 
codec name?  I.e., the default behavior - which creates a different kind of 
"RTPSource" for different codecs - would remain as it is now, but, as an 
alternative, it would be possible for you to change this behavior - e.g., 
to always use a "SimpleRTPSource" - without you having to modify the 
library.  Would this help you?

         Ross.



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X-Original-Date: Mon, 05 May 2003 20:58:50 +0200
Date: Mon, 05 May 2003 20:58:50 +0200

Hi,

We want to finely control the playing of a stream. We therefore use the
Range header of the PLAY request and we are therefore able to seek into
a stream. The Range uses ntp(Normal Play Time) values to specify the
time. The server "normally"(Darwin seems buggy) responds with the actual
Range (according to the RFC 2326) and some other information under the
RTP-Info header.
The actual range can be different from the one requested because the
requested range may not match exactly a frame boundary, for example. If
the server says OK, RTP packets are flowing to the client.

In order to make a precise association between the start ntp of the
Range and the received RTP packets, the RTP-Info contains precious
information: the CSeq and RTP timestamp corresponding to the first
packet sent by the server.

It is useful to parse RTP-Info for the following reasons:

1. To see if there are any missing packets when the first packet is
received.
2. Find out precisely the npt time that corresponds to the first packet
received.
3. Be able to finely synchronize multiple streams (multiple media
subsessions).

We can propose a patch for parsing these infos (in RTSPClient) and are
wondering how to make it available from the RTPSources (there you
receive the packets, so it may be useful to find out to what time they
correspond precisely). How can we get the corresponding subsession from
the source ?

What do you think about it ?

cezar & romulus


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X-Original-Date: Mon, 05 May 2003 12:50:45 -0700
Date: Mon, 05 May 2003 12:50:45 -0700


>We can propose a patch for parsing these infos (in RTSPClient)

Yes, this would be good.  Feel free to mail a patch.

>and are
>wondering how to make it available from the RTPSources (there you
>receive the packets, so it may be useful to find out to what time they
>correspond precisely). How can we get the corresponding subsession from
>the source ?

To map a "RTPSource" object to a "MediaSubsession" object that refers to 
it, you will need to iterate through the "MediaSubsessions", until you find 
one that uses the "RTPSource" that you're looking for.  I.e.,

         MediaSubsessionIterator iter(*session);
         MediaSubsession *subsession;
         while ((subsession = iter.next()) != NULL) {
                 if (subsession->rtpSource() == 'the RTPSource that we're 
looking for') {
                         success!
                 }
         }

Note that "RTPSource" is a 'lower-level' object than "MediaSubsession"; a 
"RTPSource" can be used without also using "MediaSession" or 
"MediaSubsession".  (See, for example, the "testMP3Receiver" and 
"testMPEGVideoReceiver" test programs, that don't use SDP at all, and so 
don't use "MediaSession" or "MediaSubsession".)  That's why there's no way 
to map directly from a "RTPSource" to a "MediaSubession"; you have to do 
this indirectly, by iterating through the "MediaSubsession"s.

         Ross.



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Subject: Re: [Live-devel] Question on Mplayer
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X-Original-Date: Mon, 5 May 2003 13:14:15 -0700
Date: Mon, 5 May 2003 13:14:15 -0700

Ross,

Thanks for replying.

>>Also, if MPlayer is playing a streaming MP3 file (with a certain bitrate),
>>how does it decide to not play some packets? I mean, how late does a
packet
>>have to be for MPlayer not to play it? In other words, from MPlayer's
point
>>of view, how is the deadline of a packet (to be played) decided?

> *Every* incoming RTP packet that the underlying LIVE.COM code receives is
> passed up to the MPlayer code.  No matter how 'late' a packet may be.

Does this mean that under bad channel conditions, an audio client will keep
getting "delayed" with respect to the server? Since in bad channel
conditions, each packet will take a large amount of time to reach the
client, this will keep "delaying" the audio file being played at the client,
i.e., if the server is streaming packet 'a' at some time instant, the client
will be playing "a - x" at the same time, and the value of 'x' keeps
increasing. This would mean that under bad channel conditions, a song would
take a longer time to finish than when the channel is good.

Am I understanding it correctly?


>          Ross.
>
>
>
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X-Original-Date: Mon, 05 May 2003 13:21:22 -0700
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> > *Every* incoming RTP packet that the underlying LIVE.COM code receives is
> > passed up to the MPlayer code.  No matter how 'late' a packet may be.
>
>Does this mean that under bad channel conditions, an audio client will keep
>getting "delayed" with respect to the server?

You'll need to look at the MPlayer playback code ("mplayer.c") to figure 
out exactly what it does in this case.  (I don't fully understand that 
code; it's very complicated.)

But, in any case, the LIVE.COM code will deliver - to MPlayer - correct 
'presentation timestamps' ("pts") for each incoming audio frame, regardless 
of how delayed the RTP packet may have been.  I.e., if you were to take all 
of the incoming audio frames, and play them according to their associated 
'presentation timestamps', then you'd get an accurate playback, with no 
gaps (assuming no packet loss).

But, again, you'll need to look at the MPlayer code to figure out exactly 
what it does.

         Ross.



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X-Original-Date: Tue, 06 May 2003 14:04:19 +0200
Date: Tue, 06 May 2003 14:04:19 +0200

Hi Ross,

And thank you for your reply. Here are some more details about what we
are trying to do:

We would like to receive multiple streams and save them into our own
data structures. We want to finely handle multiple streams (that is,
multiple MediaSubsessions), to retrieve packets or frames from the
sources or the sinks, save them in the memory and compute their media
time (that is, this packet/frame corresponds to the npt range t1-t2 in
the entire movie). 

We had proposed 2 approaches for that : 
1) Per packet. In this case we would like to receive ordered RTP packets
along with the specific RTP informations. At our knowledge, this could
be done by simply being called just before the call nextPacket->use(...)
in MultiFramedRTPSource.cpp. We would have in this case all the
informations that we need and we have them in the BufferedPacket itself,
so we don't have to search for it somewhere else. 

>Would it help you at all if I were to 'virtualize' the code in
> "MediaSubsession::initiate()" (in "liveMedia/MediaSession.cpp") that
> creates a particular kind of "RTPSource", based on the medium name and
> codec name?  I.e., the default behavior - which creates a different kind of
> "RTPSource" for different codecs - would remain as it is now, but, as an
> alternative, it would be possible for you to change this behavior - e.g.,
> to always use a "SimpleRTPSource" - without you having to modify the
> library.  Would this help you?

Yes, you are right, it can be useful to 'virtualize' the code in
"MediaSubsession::initiate()" but I think that it is not enough for our
needs. This is because we'll have to write our own MultiFramedRTPSource
to call our handler within its code. At my knowledge, the right place to
retrieve ordered RTP packets is in method
MultiFramedRTPSource::doGetNextFrame1() and therefore this method should
be 'virtualized', too. 

2) Per frame . We would like to receive each frame individually, but we
are also interested in obtaining the media time. We use here a FrameSink
and we handle the frame in the function afterGettingFrame(void*
clientData, unsigned frameSize,struct timeval presentationTime).
A way to achieve this is to find out the timestamp of this frame (in
fact, the TS of the last RTP packet within the frame) and to compare it
with the initial timestamp transmitted by the server in the PLAY
response (RTP-Info). We can retrieve the TS from the RTP source by
calling  RTPSource::curPacketRTPTimestamp() and the initial TS from the
MediaSubsession as you explained in your previous message. The question
is how to retrieve the MediaSession from a RTP Source ?


Do you find these two approaches correct ? 
Thanks,

cezar & romulus


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>2) Per frame . We would like to receive each frame individually, but we
>are also interested in obtaining the media time.

That's what the "presentationTime" parameter is for.  You can use that 
parameter (to the 'after getting' function) to give you an accurate time 
for each frame that you process.

(Right now, however, the "presentationTime" parameter won't be quite 
correct if there are multiple frames in a single RTP packet - in this case, 
the "presentationTime" will be the same for each such frame.  This is a bug 
that I'll need to fix.)

         Ross.



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X-Original-Date: Thu, 8 May 2003 13:36:04 +0200 (CEST)
Date: Thu, 8 May 2003 13:36:04 +0200 (CEST)

Hi all.

  My name is Juan Carlos Montes and i am from Spain. I
am a student and i have an university project to
multicast live video capture in MPEG2 with RTP. Your
library makes all the work ;D. 

Now i want create a prototype and i am looking for a
MPEG2 encoder board. Witch MPEG2 capturing/encoder
board you advise me? Linux or windows? I need a
cheaper one that gives me at least 15Mb bitrate. 

  Thank you in advance.

    Juan Carlos

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Juan,

First off I'd like to say that I don't know anything about MPEG2 encoder 
cards under windows, but would assume that most, if not all, would 
come with windows drivers, but probably not linux ones. I've been playing 
with Visiontech's Kfir chip simply because there was one laying around 
here. I'd suggest www.linuxtv.org as a good source for information. They 
list two boards: the one I mentioned and a new one using the Phillips 
Empress chip.

Chris



On Thu, 8 May 2003, [iso-8859-1] Juan Carlos Montes wrote:

> Hi all.
> 
>   My name is Juan Carlos Montes and i am from Spain. I
> am a student and i have an university project to
> multicast live video capture in MPEG2 with RTP. Your
> library makes all the work ;D. 
> 
> Now i want create a prototype and i am looking for a
> MPEG2 encoder board. Witch MPEG2 capturing/encoder
> board you advise me? Linux or windows? I need a
> cheaper one that gives me at least 15Mb bitrate. 
> 
>   Thank you in advance.
> 
>     Juan Carlos
> 
> ___________________________________________________
> Yahoo! Messenger - Nueva versión GRATIS
> Super Webcam, voz, caritas animadas, y más...
> http://messenger.yahoo.es
> 
> 
> -------------------------------------------------------
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> The only event dedicated to issues related to Linux enterprise solutions
> www.enterpriselinuxforum.com
> 
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> Live-devel(at)lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/live-devel
> 
> 



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X-Original-Date: Thu, 08 May 2003 10:14:12 -0700
Date: Thu, 08 May 2003 10:14:12 -0700

FYI, Sensoray <http://www.sensoray.com/> makes Linux-compatible MPEG-2 (or 
MPEG-1) video encoder boards for PCI 
<http://www.sensoray.com/html/616data.htm> and PC/104 
<http://www.sensoray.com/html/512data.htm>.

These have already been used with the LIVE.COM Streaming Media code to 
stream MPEG-1/MPEG-2 video over a LAN using multicast RTP.  (The output 
from the encoder board is piped into a (slightly) modified version of 
"testMPEGVideoStreamer" that reads from 'stdin'.)

	Ross.



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X-Original-Date: Thu, 8 May 2003 18:38:50 -0700
Date: Thu, 8 May 2003 18:38:50 -0700

Hi,
I'm trying to test openRTSP, and can't find any streams on
the net to play.  (I tried the two mentioned in the archives
of this list, but those URLs time out without connecting;
I guess they're stale.)
Can anyone help out?

Thanks,
Dan


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X-Original-Date: Fri, 09 May 2003 00:49:36 -0700
Date: Fri, 09 May 2003 00:49:36 -0700


>I'm trying to test openRTSP, and can't find any streams on
>the net to play.  (I tried the two mentioned in the archives
>of this list, but those URLs time out without connecting;
>I guess they're stale.)
>Can anyone help out?

Here are some that you can try:
         rtsp://darwin.horizonlive.com/finlayson/sample-gsm-8000.mov
         rtsp://christianchallenge.org:554/ccistream/sunday.mp4
         rtsp://stream.macwebcaster.com/camp03/murata.mp4

     Ross.



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X-Original-Date: Sat, 10 May 2003 17:59:29 +0200
Date: Sat, 10 May 2003 17:59:29 +0200

Hi,

We are programming in C++ and use the library from various objects. The
callback mechanism that is in place forbids passing an object method as
a handler function, but not a class method (static). It is probably a
problem of having C and C++ code mixed together (only ordinary C
functions and C++ static methods can be used as handlers...).

The workaround seems to be the following:

class A
{
public:
  A() {};
  int c;
  static void mA(void* clientData) {
	A* a=(A*)(clientData);
	a->c=2; // or a.myMethod();
};

and then register the handler as 
      A* a=new A();
      sessionTimerTask = env->taskScheduler().scheduleDelayedTask(
         uSecsToDelay, (TaskFunc*)A::mA, (void*)a);

Is that right ? What a hack ! :o)

romulus & cezar


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X-Original-Date: Sun, 11 May 2003 20:40:50 -0700
Date: Sun, 11 May 2003 20:40:50 -0700


>The workaround seems to be the following:
>
>class A
>{
>public:
>   A() {};
>   int c;
>   static void mA(void* clientData) {
>         A* a=(A*)(clientData);
>         a->c=2; // or a.myMethod();
>};
>
>and then register the handler as
>       A* a=new A();
>       sessionTimerTask = env->taskScheduler().scheduleDelayedTask(
>          uSecsToDelay, (TaskFunc*)A::mA, (void*)a);
>
>Is that right ?

Yes.  In fact, if you look at the "liveMedia" code, you'll see that this is 
the way that most existing handlers (especially 'after getting' handlers) 
in the code work: The handler function is a static member function, and the 
"clientData" parameter is a pointer to the particular object to which the 
handler will apply.

>  What a hack ! :o)

Yes, but it allows for a great deal of flexibility in how handler functions 
are implemented.

         Ross.



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X-Original-Date: Tue, 13 May 2003 16:27:31 +0200
Date: Tue, 13 May 2003 16:27:31 +0200

Hi
I just found a strange gcc bug, happening on 3 different machines:

http://sourceforge.net/tracker/index.php?func=detail&aid=737065&group_id=10577&atid=110577

a segfault on RTCP.cpp:145
fInBuf = new unsigned char[maxPacketSize];
with playSIP.

-- 
Reini Urban - Programmer - http://inode.at



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X-Original-Date: Tue, 13 May 2003 11:26:57 -0700
Date: Tue, 13 May 2003 11:26:57 -0700

Reini,

Many thanks for sending the bug report.  I was able to reproduce the 
problem on Linux (although not on FreeBSD), but only with the SIP URL that 
you used:
	sip:01998240001(at)voip.inode.at

However, I haven't been able to get through to that URL consistently, 
because recently it has always been 'Busy', or 'Ringing' without 
answering.  So, it is hard for me to track down the problem.

Could you tell me another "sip:" URL that will answer more consistently 
(but still demonstrates the bug)?  (E.g., this could be just a prerecorded 
message.)

	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Thu, 15 May 2003 09:14:18 +0200
Date: Thu, 15 May 2003 09:14:18 +0200

Ross,
It was you ringing on my phone all the night!
This is a real cisco ip phone answering, me personally
on the other line.

Okay, I'll install a softphone answering something prerecorded.
My whole deal is a voicemail application, which should answer anyway.
It'll be ready this afternoon (european time).

Ross Finlayson wrote:
> Reini,
> 
> Many thanks for sending the bug report.  I was able to reproduce the 
> problem on Linux (although not on FreeBSD), but only with the SIP URL 
> that you used:
>     sip:01998240001(at)voip.inode.at
> 
> However, I haven't been able to get through to that URL consistently, 
> because recently it has always been 'Busy', or 'Ringing' without 
> answering.  So, it is hard for me to track down the problem.
> 
> Could you tell me another "sip:" URL that will answer more consistently 
> (but still demonstrates the bug)?  (E.g., this could be just a 
> prerecorded message.)

-- 
Reini Urban - Programmer - http://inode.at



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X-Original-Date: Thu, 15 May 2003 00:27:21 -0700
Date: Thu, 15 May 2003 00:27:21 -0700


>It was you ringing on my phone all the night!

Oops!  My apologies if I woke you up :-)

>Okay, I'll install a softphone answering something prerecorded.

Thanks.  Please post the "sip:" URL when it's ready.

         Ross.



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X-Original-Date: Thu, 15 May 2003 13:54:45 +0200
Date: Thu, 15 May 2003 13:54:45 +0200

Reini Urban wrote:
> It was you ringing on my phone all the night!
> This is a real cisco ip phone answering, me personally
> on the other line.

BTW:
the sip server is ser-0.8.10 (from iptel.org),
the phone is a cisco ip phone 7940.

> Okay, I'll install a softphone answering something prerecorded.
> My whole deal is a voicemail application, which should answer anyway.
> It'll be ready this afternoon (european time).
> 
> Ross Finlayson wrote:
> 
>> Reini,
>>
>> Many thanks for sending the bug report.  I was able to reproduce the 
>> problem on Linux (although not on FreeBSD), but only with the SIP URL 
>> that you used:
>>     sip:01998240001(at)voip.inode.at
>>
>> However, I haven't been able to get through to that URL consistently, 
>> because recently it has always been 'Busy', or 'Ringing' without 
>> answering.  So, it is hard for me to track down the problem.
>>
>> Could you tell me another "sip:" URL that will answer more 
>> consistently (but still demonstrates the bug)?  (E.g., this could be 
>> just a prerecorded message.)
> 
> 


-- 
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Hi all,

first of all, thanks for the LiveMedia library, it's a very good stuff for my work !! I'm a student and I have to work on a video server which has to change the video encoding depending on the brandwith (H263 - MPEG1 - MPEG2 - etc...). The line quality is known via RTCP reports.

I've saw RTCP.cpp source code, but I don't really understand where the Receiver Reports values are stored.. because I've to know some values (like Jitter & Fraction loss) to determine the line quality.
I've also modified the source code of the function "incomingReportHandler1()" to print some RTCP paquet values, but I'm not sure they are OK.
So, can I have some explanations to know how to extract these informations ?

I hope I was as clear as possible,
Sorry for my (so bad) english,

Regards,

DoObS

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X-Original-Date: Thu, 15 May 2003 15:07:28 +0200
Date: Thu, 15 May 2003 15:07:28 +0200

Reini Urban wrote:
> Ross,
> It was you ringing on my phone all the night!
> This is a real cisco ip phone answering, me personally
> on the other line.
> 
> Okay, I'll install a softphone answering something prerecorded.
> My whole deal is a voicemail application, which should answer anyway.
> It'll be ready this afternoon (european time).

I'm sorry, I cannot reproduce the problem with another rtsp server.

I used your testMP3Streamer and a local ser sip server, routed to the 
internal testMP3Streamer, which plays via playSIP just fine.
   ./playSIP -a -V sip:01998240001(at)192.168.1.153
no crash unfortunately.

Our live RTSP server on which the problem occurs is a CISCO 5300.
The problem is that I cannot easily route to my internal RTSP test 
daemon easily, because our SIP server and RTSP router is already in 
production. (we are the first austrian ip tel provider)

I'll check what I can do.

> Ross Finlayson wrote:
>> Many thanks for sending the bug report.  I was able to reproduce the 
>> problem on Linux (although not on FreeBSD), but only with the SIP URL 
>> that you used:
>>     sip:01998240001(at)voip.inode.at
>>
>> However, I haven't been able to get through to that URL consistently, 
>> because recently it has always been 'Busy', or 'Ringing' without 
>> answering.  So, it is hard for me to track down the problem.
>>
>> Could you tell me another "sip:" URL that will answer more 
>> consistently (but still demonstrates the bug)?  (E.g., this could be 
>> just a prerecorded message.)
-- 
Reini Urban - Programmer - http://inode.at



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X-Original-Date: Thu, 15 May 2003 15:12:44 +0200
Date: Thu, 15 May 2003 15:12:44 +0200

Reini Urban wrote:
> I used your testMP3Streamer and a local ser sip server, routed to the 
> internal testMP3Streamer, which plays via playSIP just fine.
>   ./playSIP -a -V sip:01998240001(at)192.168.1.153

oops. the external ip is sip:01998240001(at)81.223.38.34



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Date: Thu, 15 May 2003 09:25:36 -0700


>I've saw RTCP.cpp source code, but I don't really understand where the 
>Receiver Reports values are stored.. because I've to know some values 
>(like Jitter & Fraction loss) to determine the line quality.

Note the function "RTCPInstance::enqueueReportBlock()" (in 
"RTCP.cpp").  This is the function that generates RTCP "Reception Report" 
(or "Sender Report") packets for each 'RTCPInstance'.

This information comes from a "RTPReceptionStats" object, which is stored 
with the "RTPSource" object for this "RTCPInstance".

"RTPReceptionStats" is implemented in the file "RTPSource.cpp".  This is 
where the statistics for jitter, packet loss etc. are stored.  (These 
statistics are updated each time a new RTP packet is received - see the 
functions named  "noteIncomingPacket()".)  Note that there will be one 
"RTPReceptionStats" object for each distinct RTP source ("SSRC") in the 
stream (but most RTP streams have just one "SSRC").

         Ross.
  



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From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] segfault on RTCP.cpp:145
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References: <3EC33E4A.6050901(at)inode.at>
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X-Original-Date: Thu, 15 May 2003 10:37:51 -0700
Date: Thu, 15 May 2003 10:37:51 -0700


>I'm sorry, I cannot reproduce the problem with another rtsp server.

OK, thanks for trying.  If you find another example that demonstrates the 
problem (but which won't keep you awake at nights :-), please let me know.

         Ross.



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unsubscribe

----- Original Message -----
From: {Reini Urban} <EMAIL-ADDRESS-HIDDEN>
To: <EMAIL-ADDRESS-HIDDEN>
Sent: Thursday, May 15, 2003 7:14 PM
Subject: Re: [Live-devel] segfault on RTCP.cpp:145


> Ross,
> It was you ringing on my phone all the night!
> This is a real cisco ip phone answering, me personally
> on the other line.
>
> Okay, I'll install a softphone answering something prerecorded.
> My whole deal is a voicemail application, which should answer anyway.
> It'll be ready this afternoon (european time).
>
> Ross Finlayson wrote:
> > Reini,
> >
> > Many thanks for sending the bug report.  I was able to reproduce the
> > problem on Linux (although not on FreeBSD), but only with the SIP URL
> > that you used:
> >     sip:01998240001(at)voip.inode.at
> >
> > However, I haven't been able to get through to that URL consistently,
> > because recently it has always been 'Busy', or 'Ringing' without
> > answering.  So, it is hard for me to track down the problem.
> >
> > Could you tell me another "sip:" URL that will answer more consistently
> > (but still demonstrates the bug)?  (E.g., this could be just a
> > prerecorded message.)
>
> --
> Reini Urban - Programmer - http://inode.at
>
>
>
> -------------------------------------------------------
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> https://lists.sourceforge.net/lists/listinfo/live-devel
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>




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Date: Thu, 15 May 2003 14:04:01 -0700


>I've saw RTCP.cpp source code, but I don't really understand where the 
>Receiver Reports values are stored.. because I've to know some values 
>(like Jitter & Fraction loss) to determine the line quality.

One more thing:
The "openRTSP" test program now has a new option "-Q" for printing out 
'quality of service' statistics for each stream after it completes.  The 
code for this (in "testProgs/playCommon.cpp") access the "RTPSource" 
statistics (packet loss, inter-packet gap, etc.).  So, you might want to 
look at that code for an example of how to access these statistics.

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>

ps. Thanks to Agilent Technologies for providing funding for the 
development of the new "-Q" option.

  



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thanks for your help !

I'm going to check theses samples...

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From <EMAIL-ADDRESS-HIDDEN> Sat May 17 00:30:52 2003
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X-Original-Date: Sat, 17 May 2003 00:13:36 -0700
Date: Sat, 17 May 2003 00:13:36 -0700

Previously, various parts of the "liveMedia" and "groupsock" libraries 
would output text to the console by writing to "stderr" (or "cerr").

Now, console output is done using "<<" on the 'UsageEnvironment' 
object.  The header file "UsageEnvironment.hh" defines pure virtual 
functions for "operator<<"; subclasses of "UsageEnvironment" redefine these 
with their own implementation.

The "BasicUsageEnvironment" subclass implements "operator<<" by outputting 
to 'cerr' - so, if you're using "BasicUsageEnvironment", console output 
will be the same as before.

However, other "UsageEnvironment" subclasses might "operator<<" this in 
some other way.  This can be useful if your using these libraries in 
embedded systems, for example.

	Ross.



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Subject: [Live-devel] GCC 3.3 compile problems
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X-Original-Date: Sat, 17 May 2003 12:16:37 +0200
Date: Sat, 17 May 2003 12:16:37 +0200

Hi List,

I'm trying to compile the source from 2003.05.16 in my GCC 3.3 environment.

live/liveMedia/liblibeMedia.a
live/groupsock/libgroupsock.a
live/UsageEnvironment/libUsageEnviroment.a

_will_ be build, but the build process stops at
live/BasicUsageEnvironment

with the following error:

<snip>

cd BasicUsageEnvironment ; make
make[1]: Entering directory 
`/usr/local/src/MPlayer-20030515/live/BasicUsageEnvironment'
c++ -c -Iinclude -I../UsageEnvironment/include -I../groupsock/include 
-I. -O -DSOCKLEN_T=socklen_t -Wall -DBSD=1 BasicUsageEnvironment0.cpp
c++ -c -Iinclude -I../UsageEnvironment/include -I../groupsock/include 
-I. -O -DSOCKLEN_T=socklen_t -Wall -DBSD=1 BasicUsageEnvironment.cpp
BasicUsageEnvironment.cpp: In member function `virtual UsageEnvironment&
    BasicUsageEnvironment::operator<<(const char*)':
BasicUsageEnvironment.cpp:38: error: `cerr' undeclared (first use this
    function)
BasicUsageEnvironment.cpp:38: error: (Each undeclared identifier is reported
    only once for each function it appears in.)
BasicUsageEnvironment.cpp: In member function `virtual UsageEnvironment&
    BasicUsageEnvironment::operator<<(void*)':
BasicUsageEnvironment.cpp:58: error: `hex' undeclared (first use this 
function)
make[1]: *** [BasicUsageEnvironment.o] Fehler 1
make[1]: Leaving directory 
`/usr/local/src/MPlayer-20030515/live/BasicUsageEnvironment'
make: *** [BasicUsageEnvironment/BasicUsageEnvironment.a] Fehler 2


Has somebody a patch for the GCC 3.3 environment ?

regards
  Andi
--
So kann es nicht weitergehen! - http://www.feldpolitik.de/



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Subject: Re: [Live-devel] GCC 3.3 compile problems
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X-Original-Date: Sat, 17 May 2003 11:37:38 -0700
Date: Sat, 17 May 2003 11:37:38 -0700

Both "cerr" and "hex" are (or, at least, should be) defined in "iostream.h".

Note that, to compile the code, you need to continue to use the old-style 
C++ headers - e.g., <iostream.h>.  Do *not* change any of the #includes to 
use the new-style C++ headers instead (e.g., do *not* change "#include 
<iostream.h>" to "#include <iostream>").

If you get compiler warning messages about "deprecated or antiquated 
headers", you can turn them off by adding the flag
	-Wno-deprecated
to the "CPLUSPLUS_FLAGS=" line in your 'config.<your-os>' file, before running
	./genMakefiles <your-os>

     Ross.



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X-Original-Date: Sun, 18 May 2003 09:14:56 +0200
Date: Sun, 18 May 2003 09:14:56 +0200

Hi Ross Finlayson,

I've solved my compile Problem. Thank you for the help.
On SuSE 8.2 you have to:
1. patch the source:

diff -urN live.orig/BasicUsageEnvironment/BasicUsageEnvironment.cpp 
live/BasicUsageEnvironment/BasicUsageEnvironment.cpp
--- live.orig/BasicUsageEnvironment/BasicUsageEnvironment.cpp	2003-05-17 
00:08:23.000000000 +0200
+++ live/BasicUsageEnvironment/BasicUsageEnvironment.cpp	2003-05-18 
07:53:37.000000000 +0200
(at)(at) -19,6 +19,7 (at)(at)
 
 #include "BasicUsageEnvironment.hh"
 #include <iostream.h>
+#include <iomanip.h>
 
 ////////// BasicUsageEnvironment //////////
 
diff -urN live.orig/config.linux live/config.linux
--- live.orig/config.linux	2003-05-17 00:08:25.000000000 +0200
+++ live/config.linux	2003-05-18 07:53:11.000000000 +0200
(at)(at) -4,7 +4,7 (at)(at)
 C_FLAGS =		$(COMPILE_OPTS)
 CPP =			cpp
 CPLUSPLUS_COMPILER =	c++
-CPLUSPLUS_FLAGS =	$(COMPILE_OPTS) -Wall -DBSD=1
+CPLUSPLUS_FLAGS =	$(COMPILE_OPTS) -Wall -Wno-deprecated -DBSD=1
 OBJ =			o
 LINK =			c++ -o
 LINK_OPTS =		-L.


2. make a missing link
ln -s strstream /usr/include/g++/backward/strstream.h

I'm using LIVE.COM libs with MPlayer.
And b.t.w. you need some extra for SuSE 8.2 and MPlayer too.

#!/bin/sh
export PATH=$PATH:/opt/kde3/bin
./configure --prefix=/usr --datadir=/usr/share/mplayer --mandir=/usr/share/man 
--confdir=/etc --libdir=/usr/lib \
	--disable-i18n --disable-svga --disable-directfb --disable-aa --disable-dvb 
--disable-lirc --disable-edl \
	--disable-runtime-cpudetection --disable-tv --disable-tv-v4l --disable-vidix 
--disable-vesa --disable-fbdev  \
	--enable-largefiles --enable-mpdvdkit --enable-arts --enable-qtx-codecs 
--enable-faad --enable-live --with-livelibdir=`pwd`/live \
	--disable-gui

This little thing make SuSE rpm kompatible binaries, means you need only to 
overwrite the mplayer binary.
 
The stream that I'm tried to play AND save 
http://dogme.burningman.com/webcast/
is still not working :-(((

Andi
-- 
So kann es nicht weitergehen! - http://www.feldpolitik.de/


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X-Original-Date: Mon, 19 May 2003 18:05:57 +0200
Date: Mon, 19 May 2003 18:05:57 +0200

Reini Urban wrote:
> I just found a strange gcc bug, happening on 3 different machines:
> http://sourceforge.net/tracker/index.php?func=detail&aid=737065&group_id=10577&atid=110577 
> 
> a segfault on RTCP.cpp:145
> fInBuf = new unsigned char[maxPacketSize];
> with playSIP.

I tracked down the cause for this segfault:

One problem is for sure a wrong strDupSize() called by SIPClient in
getting the tag from the sip To: header.

The DMALLOC option is from libdmalloc dmalloc.com, which helped me 
tracking down the still outstanding free problem with
delete[] toTagStr

I tried to exchange the new call in RTCP.cpp with malloc, and this 
helped, when I run the testProgs with the libdmalloc linked.
It doesn't segfault there anymore.
So I guess there is somewhere some mix between delete and free calls.

--- liveMedia/SIPClient.cpp.ori	2003-05-17 00:08:24.000000000 +0200
+++ liveMedia/SIPClient.cpp	2003-05-19 17:31:31.000000000 +0200
(at)(at) -22,6 +22,12 (at)(at)
  #include "our_md5.h"
  #include "GroupsockHelper.hh"

+#ifdef DMALLOC
+# define DEBUG_PRINT
+# define DMALLOC_FUNC_CHECK
+# include "dmalloc.h"
+#endif
+
  #if defined(__WIN32__) || defined(_WIN32)
  #define _strncasecmp strncmp
  #define snprintf _snprintf
(at)(at) -528,7 +534,7 (at)(at)
        nextLineStart = getLine(lineStart);
        if (lineStart[0] == '\0') break; // this is a blank line

-      if (sscanf(lineStart, "To:%*[^;]; tag=%s", toTagStr) == 1) {
+      if (sscanf(lineStart, "To: %*[^;]; tag=%s", toTagStr) == 1) {
  	delete[] (char*)fToTagStr; fToTagStr = strDupSize(toTagStr);
  	fToTagStrSize = strlen(fToTagStr);
        }
(at)(at) -542,7 +548,14 (at)(at)
          }
        }
      }
-    delete[] toTagStr;
+#ifdef DEBUG_PRINT
+    fprintf(stderr,">>> To: tag='%s'\n",toTagStr);
+#endif
+    delete[] toTagStr; // this fails!
+/* dmalloc.log:
+1053354876: 131: free: failed OVER picket-fence magic-number check: 
pointer '0x80d0aa8' from 'ra=0x80703ca' prev access 'unknown'
+1053354876: 131: ERROR: _dmalloc_chunk_free: failed OVER picket-fence 
magic-number check (err 27)
+*/

      // We're now at the end of the response header lines
      if (lineStart == NULL) {

--- UsageEnvironment/strDup.cpp~	2003-05-17 00:08:24.000000000 +0200
+++ UsageEnvironment/strDup.cpp	2003-05-19 17:00:41.000000000 +0200
(at)(at) -34,7 +34,9 (at)(at)
  char* strDupSize(char const* str) {
    size_t len = strlen(str) + 1;
    char* copy = new char[len];
-
+  if (copy != NULL) {
+    memcpy(copy, str, len);
+  }
    return copy;
  }


--- liveMedia/RTCP.cpp.ori	2003-05-17 00:08:24.000000000 +0200
+++ liveMedia/RTCP.cpp	2003-05-19 15:14:57.000000000 +0200
(at)(at) -22,6 +22,12 (at)(at)
  #include "GroupsockHelper.hh"
  #include "rtcp_from_spec.h"

+#ifdef DMALLOC
+# define DEBUG_PRINT
+# define DMALLOC_FUNC_CHECK
+# include "dmalloc.h"
+#endif
+
  ////////// RTCPMemberDatabase //////////

  #ifdef BSD
(at)(at) -133,6 +139,14 (at)(at)
      fTypeOfEvent(EVENT_UNKNOWN), fTypeOfPacket(PACKET_UNKNOWN_TYPE),
      fHaveJustSentPacket(False), fLastPacketSentSize(0),
      fByeHandlerTask(NULL), fByeHandlerClientData(NULL) {
+
+    unsigned char *fInBuf;
+#ifdef DMALLOC
+    char *dFile;
+    unsigned int  dLine;
+    size_t dSize;
+    void *dAddr;
+#endif
  #ifdef DEBUG_PRINT
    fprintf(stderr, "RTCPInstance[%p]::RTCPInstance()\n", this);
  #endif
(at)(at) -142,8 +156,20 (at)(at)
    fPrevReportTime = fNextReportTime = timeNow;

    fKnownMembers = new RTCPMemberDatabase;
+  fInBuf = (unsigned char *) malloc(maxPacketSize);
+  /*
    fInBuf = new unsigned char[maxPacketSize];
+  */
+#ifdef DMALLOC
+  dmalloc_examine(fInBuf,&dSize,&dFile,&dLine,&dAddr);
+  fprintf(stderr, ">>> fInBuf[%p]: %d %s:%d/%p\n", 
fInBuf,dSize,dFile,dLine,dAddr);
+#endif
+
    fOutBuf = new OutPacketBuffer(preferredPacketSize, maxPacketSize);
+#ifdef DMALLOC
+  dmalloc_examine(fOutBuf,&dSize,&dFile,&dLine,&dAddr);
+  fprintf(stderr, ">>> fOutBuf[%p]: %d %s:%d/%p\n", 
fInBuf,dSize,dFile,dLine,dAddr);
+#endif
    if (fKnownMembers == NULL || fOutBuf == NULL) return;

    // Arrange to handle incoming reports from others:
(at)(at) -170,7 +196,8 (at)(at)

    delete fKnownMembers;
    delete fOutBuf;
-  delete[] fInBuf;
+  free (fInBuf);
+  // delete[] fInBuf;
  }

  RTCPInstance* RTCPInstance::createNew(UsageEnvironment& env, 
Groupsock* RTCPgs,



from the dmalloc docs:
fence-post (i.e. bounds) checking
      "Fence-post" memory is the area immediately above or below memory
      allocations.  It is all too easy to write code that accesses above
      or below an allocation - especially when dealing with arrays or
      strings.  The library can write special values in the areas around
      every allocation so it will notice when these areas have been
      overwritten.
      _NOTE_: The library cannot notice when the program _reads_ from
      these areas, only when it writes values.  Also, fence-post checking
      will increase the amount of memory the program allocates.
-- 
Reini Urban - Programmer - http://inode.at



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Subject: Re: [Live-devel] segfault on RTCP.cpp:145
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X-Original-Date: Mon, 19 May 2003 09:31:19 -0700
Date: Mon, 19 May 2003 09:31:19 -0700

Reini,

Thanks for looking into this.

>One problem is for sure a wrong strDupSize() called by SIPClient in
>getting the tag from the sip To: header.

Yes, the call to "strDupSize()" at line 532 was definitely a bug.  It 
should have been "strDup()" instead.

I've now installed a new version of the library that fixes this bug.  With 
that bug fixed, do you still see any crashes?

         Ross.



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Subject: Re: [Live-devel] segfault on RTCP.cpp:145
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X-Original-Date: Tue, 20 May 2003 15:09:49 +0200
Date: Tue, 20 May 2003 15:09:49 +0200

Ross Finlayson wrote:
> Thanks for looking into this.
> 
>> One problem is for sure a wrong strDupSize() called by SIPClient in
>> getting the tag from the sip To: header.
> 
> 
> Yes, the call to "strDupSize()" at line 532 was definitely a bug.  It 
> should have been "strDup()" instead.
> 
> I've now installed a new version of the library that fixes this bug.  
> With that bug fixed, do you still see any crashes?

yes. there's still a heap corruption somewhere:

...
Opened URL "sip:01998240001(at)voip.inode.at", returning a SDP description:
v=0
o=- 1254555217 0 IN IP4 192.168.1.229
s=playSIP session
c=IN IP4 192.168.1.229
t=0 0
m=audio 8000 RTP/AVP 0


Program received signal SIGSEGV, Segmentation fault.
0x400f7459 in chunk_alloc (ar_ptr=0x401a26a0, nb=24) at malloc.c:3075
3075    malloc.c: No such file or directory.
         in malloc.c
(gdb) bt
#0  0x400f7459 in chunk_alloc (ar_ptr=0x401a26a0, nb=24) at malloc.c:3075
#1  0x400f6ae4 in __libc_malloc (bytes=16) at malloc.c:2810
#2  0x4004e649 in __builtin_new (sz=16) at 
../../gcc-2.95.3/gcc/cp/new1.cc:77
#3  0x08060bf7 in RTPSource::RTPSource (this=0x80c8f18, env=(at)0x80c7060, 
RTPgs=0x80c8c48, rtpPayloadFormat=0 '\0',
     rtpTimestampFrequency=8000) at RTPSource.cpp:54
#4  0x080619cb in MultiFramedRTPSource::MultiFramedRTPSource 
(this=0x80c8f18, env=(at)0x80c7060, RTPgs=0x80c8c48,
     rtpPayloadFormat=0 '\0', rtpTimestampFrequency=8000, 
packetFactory=0x0) at MultiFramedRTPSource.cpp:61
#5  0x08062a64 in SimpleRTPSource::SimpleRTPSource (this=0x80c8f18, 
env=(at)0x80c7060, RTPgs=0x80c8c48,
     rtpPayloadFormat=0 '\0', rtpTimestampFrequency=8000, 
mimeTypeString=0x80c8c10 "audio/PCMU", offset=0)
     at SimpleRTPSource.cpp:42
#6  0x080629c8 in SimpleRTPSource::createNew (env=(at)0x80c7060, 
RTPgs=0x80c8c48, rtpPayloadFormat=0 '\0',
     rtpTimestampFrequency=8000, mimeTypeString=0x80c8c10 "audio/PCMU", 
offset=0) at SimpleRTPSource.cpp:34
#7  0x08073879 in MediaSubsession::initiate (this=0x80c8d10, 
useSpecialRTPoffset=0) at MediaSession.cpp:713
#8  0x0804ae3d in main (argc=2, argv=0xbffff8cc) at playCommon.cpp:407
#9  0x400a017d in __libc_start_main (main=0x8049f40 <main>, argc=4, 
ubp_av=0xbffff8c4, init=0x8049394 <_init>,
     fini=0x8089940 <_fini>, rtld_fini=0x4000a534 <_dl_fini>, 
stack_end=0xbffff8bc) at ../sysdeps/generic/libc-start.c:129

dmalloc still reports the same cause of heap corruption:
SipClient.cpp:545
   delete[] tagToStr;


-- 
Reini Urban - Programmer - http://inode.at



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X-Original-Date: Tue, 20 May 2003 15:43:06 +0200
Date: Tue, 20 May 2003 15:43:06 +0200

Ross Finlayson wrote:
> I've now installed a new version of the library that fixes this bug.  
> With that bug fixed, do you still see any crashes?

with a simple malloc it doesn't crash anymore. wonder why.

rurban(at)mail1:/usr/src/live$ diff -bu liveMedia/SIPClient.cpp.ori 
liveMedia/SIPClient.cpp
--- liveMedia/SIPClient.cpp.ori 2003-05-20 14:51:20.000000000 +0200
+++ liveMedia/SIPClient.cpp     2003-05-20 14:49:40.000000000 +0200
(at)(at) -518,7 +518,7 (at)(at)
      // The remaining data is assumed to be the SDP descriptor that we 
want.
      // We should really do some more checking on the headers here - 
e.g., to
      // check for "Content-type: application/sdp", "CSeq", etc. #####
-    char* toTagStr = strDupSize(readBuf);
+    char* toTagStr = (char *) malloc(readBufSize + 1);
      int contentLength = -1;
      char* lineStart;
      while (1) {
(at)(at) -528,7 +528,7 (at)(at)
        nextLineStart = getLine(lineStart);
        if (lineStart[0] == '\0') break; // this is a blank line

-      if (sscanf(lineStart, "To:%*[^;]; tag=%s", toTagStr) == 1) {
+      if (sscanf(lineStart, "To: %*[^;]; tag=%s", toTagStr) == 1) {
         delete[] (char*)fToTagStr; fToTagStr = strDup(toTagStr);
         fToTagStrSize = strlen(fToTagStr);
        }
(at)(at) -542,7 +542,8 (at)(at)
          }
        }
      }
-    delete[] toTagStr;
+    free(toTagStr);

      // We're now at the end of the response header lines
      if (lineStart == NULL) {

-- 
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Hi,

Is there a document that describes the RTSP session used by Windows Media Player Series 9. How can we stream ASF files over RTSP???

Thanks & Regards,
Manu V Batura



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X-Original-Date: Tue, 20 May 2003 18:08:56 +0200
Date: Tue, 20 May 2003 18:08:56 +0200

Hi,

i read the following in the archives:
> Note that, to compile the code, you need to continue to use the
> old-style C++ headers - e.g., <iostream.h>.  Do *not* change any of the
> #includes to use the new-style C++ headers instead (e.g., do *not*
> change "#include <iostream.h>" to "#include <iostream>").

But that's exactly what i had to do to get live to compile with gcc-3.3.
groupsock/Groupsock.cpp and groupsock/NetInterface.cpp include
<strstream.h>, which doesn't exist on my system, changing that to
#include <strstream> makes it compile fine.

-- 
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X-Original-Date: Tue, 20 May 2003 09:53:59 -0700
Date: Tue, 20 May 2003 09:53:59 -0700


>i read the following in the archives:
> > Note that, to compile the code, you need to continue to use the
> > old-style C++ headers - e.g., <iostream.h>.  Do *not* change any of the
> > #includes to use the new-style C++ headers instead (e.g., do *not*
> > change "#include <iostream.h>" to "#include <iostream>").
>
>But that's exactly what i had to do to get live to compile with gcc-3.3.
>groupsock/Groupsock.cpp and groupsock/NetInterface.cpp include
><strstream.h>, which doesn't exist on my system, changing that to
>#include <strstream> makes it compile fine.


But that worked only because you *didn't* have an existing file named 
"strstream.h".  If you had already had a "strstream.h" - as well as a 
"strstream" - then it's possible that those two files were different, and 
incompatible, in which case it would have been wrong to change the code to 
do "#include <strstream>".

         Ross.



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X-Original-Date: Tue, 20 May 2003 10:18:32 -0700
Date: Tue, 20 May 2003 10:18:32 -0700


>yes. there's still a heap corruption somewhere:

Reini,

It would be great if I had a "sip:" URL to use to track this down (that was 
connected to a machine, so that you didn't have to answer the phone in person).

But, failing that, could you try the following experiment:

In "SIPClient.cpp", there are several pairs of lines like the following:
         <some-string> = new char[<some-size>];
         sprintf(<some-string>, ...)

(You can find these by searching for "sprintf")

These *should* all be safe, because, in each case, <some-size> is supposed 
to be have been computed to more than enough size to hold the string.  It's 
conceivable, though, that one of the <some-size>s is incorrect - which 
would cause the crash.

So, could you change each of these "sprintf()" calls to
         snprintf(<some-string>, <some-size>, ...)
and see whether or not the crash goes away.  If the crash *does* go away, 
then change each of the "snprintf()"s back to "sprintf()", until the crash 
reappears.

         Ross.



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is the use of the rtp thru the usage of datagrams ?
if so , when i tried using the commands recvfrom (.......) , i can only receive 12 bytes of data per time...
so may i know what is the packet size and how does live.com sends the rtp packets ?


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<DIV>is the use of the rtp thru the usage of datagrams ?</DIV>
<DIV>if so , when i tried using the commands recvfrom (.......) , i can only receive 12 bytes of data per time...</DIV>
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X-Original-Date: Wed, 21 May 2003 02:31:07 -0700
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At 11:20 PM 5/20/03, you wrote:
>is the use of the rtp thru the usage of datagrams ?
>if so , when i tried using the commands recvfrom (.......) , i can only 
>receive 12 bytes of data per time...
>so may i know what is the packet size and how does live.com sends the rtp 
>packets ?

I'm sorry, but I don't understand your question.

Note that this mailing list is for the discussion of the "LIVE.COM 
Streaming Media" software.  Which specific part of this software do you 
have a question about?

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>



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X-Original-Date: Wed, 21 May 2003 02:34:54 -0700
Date: Wed, 21 May 2003 02:34:54 -0700


>    I want to stream live video and take input from a MPEG-1 encoder 
> instead read from a file. Would you please tell me how to pipe the output 
> of my encoder to  "test*Streamer" application?

Michael,

This question is now answered in the online 'FAQ'.  See 
<http://www.live.com/liveMedia/faq.html#liveInput>

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>



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Dear Ross,
    I have read online 'FAQ' carefully, but I still don't know how to pipe output. Would you please give me some code samples to do that?
    Thanks!
Michael

----- Original Message ----- 
From: {Ross Finlayson} <EMAIL-ADDRESS-HIDDEN>
To: <EMAIL-ADDRESS-HIDDEN>
Sent: Wednesday, May 21, 2003 5:34 PM
Subject: Re: [Live-devel] how to pipe output of a encoder?


> 
> >    I want to stream live video and take input from a MPEG-1 encoder 
> > instead read from a file. Would you please tell me how to pipe the output 
> > of my encoder to  "test*Streamer" application?
> 
> Michael,
> 
> This question is now answered in the online 'FAQ'.  See 
> <http://www.live.com/liveMedia/faq.html#liveInput>
> 
>          Ross Finlayson
>          LIVE.COM
>          <http://www.live.com/>
> 
> 
> 
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From <EMAIL-ADDRESS-HIDDEN> Wed May 21 07:55:22 2003
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X-Original-Date: Wed, 21 May 2003 07:50:27 -0700
Date: Wed, 21 May 2003 07:50:27 -0700

At 02:50 AM 6/5/03, you wrote:
>Dear Ross,
>     I have read online 'FAQ' carefully, but I still don't know how to 
> pipe output. Would you please give me some code samples to do that?

If (i) your MPEG video encoder is an application named "enc", and (ii) you 
have modified the "testMPEGVideoStreamer" application to read from "stdin" 
instead of "test.mpg", then - from a command shell - do

         enc | testMPEGVideoStreamer

(Note the "|" symbol, meaning: create a pipe between the two applications.)

         Ross.



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Dear Ross,
    In fact I want to encapsulate my mpeg encoder and live media in one application, then send the output of encoder to testMPEGVideoStreamer. So I create a pipe as follows:
        int fields[2];
        pipe(fields);
    and then I put the output of mpeg encoder to fields[1], modify testMPEGVideoStreamer to read from fields[0] instead of "test.mpg". Is this right?
    Thank you very much for your kindly replies.
Michael
----- Original Message ----- 
From: {Ross Finlayson} <EMAIL-ADDRESS-HIDDEN>
To: <EMAIL-ADDRESS-HIDDEN>
Sent: Wednesday, May 21, 2003 10:50 PM
Subject: Re: [Live-devel] how to pipe output of a encoder?


> At 02:50 AM 6/5/03, you wrote:
> >Dear Ross,
> >     I have read online 'FAQ' carefully, but I still don't know how to 
> > pipe output. Would you please give me some code samples to do that?
> 
> If (i) your MPEG video encoder is an application named "enc", and (ii) you 
> have modified the "testMPEGVideoStreamer" application to read from "stdin" 
> instead of "test.mpg", then - from a command shell - do
> 
>          enc | testMPEGVideoStreamer
> 
> (Note the "|" symbol, meaning: create a pipe between the two applications.)
> 
>          Ross.
> 
> 
> 
> -------------------------------------------------------
> This SF.net email is sponsored by: ObjectStore.
> If flattening out C++ or Java code to make your application fit in a
> relational database is painful, don't do it! Check out ObjectStore.
> Now part of Progress Software. http://www.objectstore.net/sourceforge
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>    In fact I want to encapsulate my mpeg encoder and live media in one 
> application, then send the output of encoder to testMPEGVideoStreamer. So 
> I create a pipe as follows:
>         int fields[2];
>         pipe(fields);
>     and then I put the output of mpeg encoder to fields[1], modify 
> testMPEGVideoStreamer to read from fields[0] instead of "test.mpg". Is 
> this right?

That should work.  You'll probably want to use separate threads for this: 
one thread running your encoder (and writing to the pipe); the other thread 
runing the modified "testMPEGVideoStreamer" code (and reading from the pipe).

However, I recommend that you *first* write your encoder and streamer as 
separate applications, and pipe them together using the command 
shell.  (Then, once you're sure that this is working OK, you can put them 
into a single application.)

         Ross.



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--0-1523374102-1053573114=:72670
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ok .. i rephrase my question...
 
i am using winsock at the client for receiving the rtp data while using the testvideostreamer program at my server using unicast method
 
While using datagrams at the winsock, i tried to use the recvfrom() command to receive the data
 
however, i could only receive 12 bytes of data per packet. so i would like to know if we can use this method to receive the packets ? if yes, what is the size of the packet sent per time using the testvideostreamer program listed at the testprogs. 


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<DIV>ok .. i rephrase my question...</DIV>
<DIV>&nbsp;</DIV>
<DIV>i am using winsock at the client for receiving the rtp data while using the testvideostreamer program&nbsp;at my server using unicast method</DIV>
<DIV>&nbsp;</DIV>
<DIV>While using datagrams at the winsock, i tried to use the recvfrom() command to receive the data</DIV>
<DIV>&nbsp;</DIV>
<DIV>however, i could only receive 12 bytes of data&nbsp;per&nbsp;packet. so i&nbsp;would like to&nbsp;know&nbsp;if we can use this method to receive the packets ? if yes,&nbsp;what is the size of the packet sent per time using the testvideostreamer program listed at the testprogs.&nbsp;</DIV><p><hr SIZE=1>
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X-Original-Date: Thu, 22 May 2003 02:18:18 -0700
Date: Thu, 22 May 2003 02:18:18 -0700

FYI, I recently ran the (very useful) Linux "valgrind" memory checking tool 
<http://developer.kde.org/~sewardj/> on the code, and found a couple of 
minor bugs.  It's conceivable (although unlikely) that they were 
responsible for the crash that you saw.

But anyway, please let me know if the crash still occurs with the newest 
version of the library (2003.05.22a or later).  If it does still occur, 
then it would be cool if you could run "valgrind openRTSP <args>", and let 
me know what you see.  (Before doing this, rebuild the code for gdb by 
doing "./genMakefiles linux-gdb", so as to get better debugging information.)

	Ross.



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X-Original-Date: Thu, 22 May 2003 02:21:59 -0700
Date: Thu, 22 May 2003 02:21:59 -0700


>i am using winsock at the client for receiving the rtp data while using 
>the testvideostreamer program at my server using unicast method
>
>While using datagrams at the winsock, i tried to use the recvfrom() 
>command to receive the data
>
>however, i could only receive 12 bytes of data per packet. so i would like 
>to know if we can use this method to receive the packets ?

Yes, and in fact "recvfrom()" is the method that the LIVE.COM code uses to 
receive packets.  (See the "readSocket" function in 
"groupsock/GroupsockHelper.cpp".)

>  if yes, what is the size of the packet sent per time using the 
> testvideostreamer program listed at the testprogs.

Typically somewhere in the range 1000 to 1500 bytes.  (If you're only 
getting 12 bytes of data per packet, then that's definitely a bug!)

I suggest that you first run "testMPEGVideoReceiver", to make sure that 
you're receiving the video OK using an application that is known to work.

         Ross.



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Vens,

> however, i could only receive 12 bytes of data per packet. so i would 
> like to know if we can use this method to receive the packets ? if 
> yes, what is the size of the packet sent per time using the 
> testvideostreamer program listed at the testprogs. 

Forgive me if I misunderstand the question, but the RTP header might 
only be 12 bytes long (no CSRCs).  If the extension bit is set, you can 
find out the length of the extension.

Darn -- I need to re-read RFC1889 again.  I just realized that I don't 
know how to determine the length of an RTP packet (nothing in the RTP 
header that I see has this info).

Regards,

John

Falling You - exploring the beauty of voice and sound
New EP, "hope thrown down," available now at
http://www.mp3.com/fallingyou













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X-Original-Date: Fri, 23 May 2003 03:05:01 -0700
Date: Fri, 23 May 2003 03:05:01 -0700

FYI, Reini Urban helped me track down the bug that was causing "playSIP" to 
crash.  The latest source code release contains the bug fix.

	Ross.



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X-Original-Date: Fri, 23 May 2003 07:48:05 -0700 (PDT)
Date: Fri, 23 May 2003 07:48:05 -0700 (PDT)

Hi,

I am using the mp4live as a broadcaster, Darwin
Streaming Server. Then I use openRTSP try to get the
broadcast streaming. I got a video file MP4V-ES-1 and
an audio file. After renaming the MP4V-ES-1 as 1.mp4,
I could use the gmp4player to play it. Also how could
I put the video file and audio file into to one file,
or have the openRTSP to record them as one file, and
got the gmp4player to play it?

Kind regards,

Guang Xu



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X-Original-Date: Fri, 23 May 2003 09:06:42 -0700
Date: Fri, 23 May 2003 09:06:42 -0700


>I am using the mp4live as a broadcaster, Darwin
>Streaming Server. Then I use openRTSP try to get the
>broadcast streaming. I got a video file MP4V-ES-1 and
>an audio file. After renaming the MP4V-ES-1 as 1.mp4,
>I could use the gmp4player to play it. Also how could
>I put the video file and audio file into to one file,
>or have the openRTSP to record them as one file, and
>got the gmp4player to play it?

Because ".mp4" files are similar to ".mov" files, you could - in principle 
- use openRTSP's "-q" option to record the incoming audio and video streams 
into a single (".mov" format) file.

Unfortunately you would need to modify the "QuickTimeFileSink" class in 
order to produce a file that is readable.  (At present, "QuickTimeFileSink" 
works properly only for H.263 video, and PCM, GSM, or QCELP audio.)

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>



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X-Original-Date: Mon, 26 May 2003 10:48:16 +0530
Date: Mon, 26 May 2003 10:48:16 +0530

hai all
i downloaded  tha live.com streaming code.
i am using it ina bluetooth application.
actually i want to play the streamed media directly by using winamp instead
of diaplying it on stdout or
stroing it in a file.

so i am using the Mp3Http source class create new function in
testMp3streamer function
and at the  receiver side http sink class

I am facing problem at this point in the code in the Create new function of
the mp3hhtpsource class

FILE* fid = fdopen(ourSocket, "r+b");

i find an assertion message gets popped up when i try to excute this line

the two assertion messages come because of this
_ASSERTE((unsigned)filedes < (unsigned)_nhandle);
 _ASSERTE(_osfile(filedes) & FOPEN);

regards
mritunjay



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X-Original-Date: Mon, 26 May 2003 02:32:18 -0700
Date: Mon, 26 May 2003 02:32:18 -0700

At 10:18 PM 5/25/03, you wrote:
>hai all
>i downloaded  tha live.com streaming code.
>i am using it ina bluetooth application.
>actually i want to play the streamed media directly by using winamp instead
>of diaplying it on stdout or
>stroing it in a file.

Do you know about the existing "in_rtp.dll" Winamp plugin - 
<http://www.live.com/multikit/winamp-plugin.html> - that uses the "LIVE.COM 
Streaming Media" code to play MP3/RTP streams?  You should be able to use 
this instead of writing your own receiver for Winamp.

>so i am using the Mp3Http source class create new function in
>testMp3streamer function

If your input data comes from a HTTP stream, and you are eventually playing 
the data using Winamp, then why not have Winamp play the HTTP stream (from 
the original "http://" URL) directly?  Is this because you want to 
multicast the input streams to potentially multiple receivers (e.g., on a 
local-area network)?

>I am facing problem at this point in the code in the Create new function of
>the mp3hhtpsource class
>
>FILE* fid = fdopen(ourSocket, "r+b");
>
>i find an assertion message gets popped up when i try to excute this line
>
>the two assertion messages come because of this
>_ASSERTE((unsigned)filedes < (unsigned)_nhandle);
>  _ASSERTE(_osfile(filedes) & FOPEN);

I'm not sure where those assertions are coming from (they're not in our 
code), but it sounds like "fdopen()" isn't working properly (or at all) on 
your system (which, I'm guessing, is Windows).

I suggest commenting out the "fdopen()" call, and just execute the
         if (fid == NULL) {
branch instead.  Maybe that will work for you...

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>



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From: {Mildred L. Frisco} <EMAIL-ADDRESS-HIDDEN>
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 boundary="------------050206010603080805020101"
Subject: [Live-devel] error compiling live using arm-linux tools
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X-Original-Date: Mon, 26 May 2003 17:51:22 +0800
Date: Mon, 26 May 2003 17:51:22 +0800

This is a multi-part message in MIME format.
--------------050206010603080805020101
Content-Type: text/plain; charset=us-ascii; format=flowed
Content-Transfer-Encoding: 7bit


Hi!  i am trying to compile the live library using the arm-linux 
toolchain.  i attached the config.linux which i modified.  i wonder why 
does it generate the following error when i run make. config.linux is 
the only file i changed and the include files mentioned are in the 
library, the include directories are also specified in teh makefile.

here's what i did:

$ ./genMakefiles linux
$ make
    cd liveMedia ; make
make[1]: Entering directory '/home.mildred/armbuild/live/liveMedia'
arm-linux-c++ -c -o Media.o Media.cpp
Media.cpp:21: Media.hh: No such file or directory
Media.cpp:22: HashTable.hh: No such file or directory
make[1]: *** [Media.o] error 1
leaving directory /home.mildred/armbuild/live/liveMedia

it seems that it doesn't include the include directories mentioned in 
INCLUDES in the makefile  

i already built this on linux desktop and it did just fine.  i wonder 
what's wrong with this one.

Mildred





--------------050206010603080805020101
Content-Type: text/plain;
 name="config.linux"
Content-Transfer-Encoding: 7bit
Content-Disposition: inline;
 filename="config.linux"

COMPILE_OPTS =		$(INCLUDES) -I. -O -DSOCKLEN_T=socklen_t
C =			c
C_COMPILER =		arm-linux-gcc
C_FLAGS =		$(COMPILE_OPTS)
CPP =			arm-linux-g++
CPLUSPLUS_COMPILER =	arm-linux-c++
CPLUSPLUS_FLAGS =	$(COMPILE_OPTS) -Wall -DBSD=1
OBJ =			o
LINK =			arm-linux-c++ -o
LINK_OPTS =		-L.
CONSOLE_LINK_OPTS =	$(LINK_OPTS)
LIBRARY_LINK =		arm-linux-ld -o
LIBRARY_LINK_OPTS =	$(LINK_OPTS) -r -Bstatic
LIB_SUFFIX =			a
LIBS_FOR_CONSOLE_APPLICATION =
LIBS_FOR_GUI_APPLICATION =
EXE =

--------------050206010603080805020101--



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Subject: [Live-devel] Newbie: How to PCM streamer & receiver ?
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X-Original-Date: Mon, 26 May 2003 17:13:05 +0200
Date: Mon, 26 May 2003 17:13:05 +0200

Hi everybody,

just started to play with the animal this morning.

So far so good ;o)

But now, I wish to stream some raw PCM files. I've tried to hack a little
bit MP3Streamer & Receiver with no luck.

+++ Streamer's changes:

the file first
------------
void play() {
  // Open the file as a 'MP3 file source':
//  sessionState.source = MP3FileSource::createNew(*env, inputFileName);
  sessionState.source = ByteStreamFileSource::createNew(*env, inputFileName,
0, 0);

  if (sessionState.source == NULL) {

------------

and also the sink
------------
#else
  sessionState.sink
//    = MPEGAudioRTPSink::createNew(*env, sessionState.rtpGroupsock);
    = SimpleRTPSink::createNew(*env, sessionState.rtpGroupsock, 0, 8000,
"audio", "PCMU");

#endif

------------


+++ Receiver's changes:
------------
  // Create the data source: a "MP3 *ADU* RTP source"
  unsigned char rtpPayloadFormat = 0;//96; // a dynamic payload type
  rtpSource
//    = MP3ADURTPSource::createNew(*env, &rtpGroupsock, rtpPayloadFormat);
    = SimpleRTPSource::createNew(*env, &rtpGroupsock, rtpPayloadFormat);
#endif
------------

I've changed the payload format also, but I'm not sure it was necessary....


Results:

The streamer seems to do its job. I've used sapWatch to look from the target
[ yes, btw, unicast is used here. I'm playing between 2 boxes ].

The Receiver seems to see nothing.....

I'm pretty sure I've missed something obvious......

Could you point it out for me please ?

Cheers,
tm



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X-Original-Date: Mon, 26 May 2003 11:21:33 -0700
Date: Mon, 26 May 2003 11:21:33 -0700


>CPP =                   arm-linux-g++

This is the mistake.  "CPP" is the name of the C++ source code file name 
suffix - *not* the name of the compiler (*that* is already defined by 
"CPLUSPLUE_COMPILER").

You should leave "CPP" defined to be "cpp".


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Mon, 26 May 2003 11:44:51 -0700
Date: Mon, 26 May 2003 11:44:51 -0700

Part of the difficulty is that you're trying to make two changes at once: 
(i) streaming using PCM instead of MP3; (ii) streaming unicast instead of 
multicast.  Instead, I suggest doing these one-at-a-time: first, streaming 
PCM (but via multicast), and then using unicast instead of multicast.

1/ To stream using PCM
===============

At the sender ('streamer'), do:

>   sessionState.source = ByteStreamFileSource::createNew(*env, inputFileName,
>0, 0);

That's mostly correct.  However, you probably want to change the last 
parameter ("playTimePerFrame") from 0 to 125 (microseconds), so that 
"getPlayTime()" works correctly, and the RTP packets get streamed out at 
the correct rate.  (If you leave this parameter at 0, the RTP packets will 
be sent out as fast as the program can read data from the file - which is 
not what you want.)

>   sessionState.sink
>//    = MPEGAudioRTPSink::createNew(*env, sessionState.rtpGroupsock);
>     = SimpleRTPSink::createNew(*env, sessionState.rtpGroupsock, 0, 8000,
>"audio", "PCMU");

This is correct.

At the receiver, do:

>   rtpSource
>    = SimpleRTPSource::createNew(*env, &rtpGroupsock, rtpPayloadFormat);

Almost correct (provided that "rtpPayloadFormat" is 0).  You'll also need 
to add the  "mimeTypeString" parameter:
         "audio/PCMU"

2/ To stream using unicast instead of multicast
==========

See <http://www.live.com/liveMedia/faq.html#unicast>

I hope this helps...


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Dear Ross,

    In ttestMPEGAudioVideoStreamer.cpp, You said the sentence "  env->taskScheduler().doEventLoop();" does not return. Now I want it stop when I click the "stop" button in a GUI. How can I do that?

Thanks.

Michael
ÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÓ†+ùÞµéšŠX¬²š'²ŠÞu¼ÿ9¸ÞrÔ­¢·ÿ!÷åjÛ^ž)à¢ëBûê+%«Úr‡^¶‰š‘ì¨º¶©¦XœjØ¨ø­Šv«zV­Š‰Ú•Ö­i¶¬z+)j)ßº_Ý¢ívˆ­ü(^rJ.´æãyËR¶ŠÞüÚ0¥ªí¡óë¢Þ²Ä¨~Ü­ïá¶Úÿÿü0Ãúç-²Ú+{ùÞ·û(º·~Šà{ÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿ.+Þý×¯zYšŠX§‚X¬´¸¯{÷^½éÿ–+-³û(º·~Šà{ùÞ¶m¦ÏÿþX¬¶Ïì¢êÜyú+ïçzßåŠËlþX¬¶)ß£ùb½ïÝz÷¥


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>    In ttestMPEGAudioVideoStreamer.cpp, You said the sentence 
> "  env->taskScheduler().doEventLoop();" does not return. Now I want it 
> stop when I click the "stop" button in a GUI. How can I do that?

Note that "doEventLoop()" takes an optional parameter "watchVariable":
In "UsageEnvironment/include/UsageEnvironment.hh":
         virtual void doEventLoop(char* watchVariable = NULL) = 0;

If the "watchVariable" parameter is not NULL, then "doEventLoop()" will 
return if/when *watchVariable is not 0.  E.g.,

char stopFlag = 0;
env->taskScheduler().doEventLoop(&stopFlag);

and then, in an event handler (e.g., for your GUI, when the user clicks the 
"stop" button:
         extern char stopFlag;
         stopFlag = ~0;

I hope this helps...


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Tue, 27 May 2003 10:31:32 +0530
Date: Tue, 27 May 2003 10:31:32 +0530

hai
actually i hvae included the in_rtp.dll in the specified location as given
in the live.com site.
But is this enough .
shouldn't i make some changes in testmp3receivre.cpp and testmp3
streamer.cpp

the following chnages i made in the two files.
i called the create new function on the MP3http source class insated of
mp3filesource class

and at the receiver side i called httpsink class create new function
instead of Filesink class

don't u feel we need to make those changes.

as u gave the solution to me yesterday since i am working on windows so i
am encountering problem with
fdopen . i even tried the one which u suggested to me before hand

but i find that the thread excution gets blocked on the socket recv call
as given in the code below.

if (fFidIsReallyASocket) {
    long fid_long = (long)fFid;
    int sock = (int)fid_long;
    unsigned totBytesRead = 0;
    do {
      int bytesRead
     = recv(sock, &((char*)buf)[totBytesRead], numChars-totBytesRead, 0);
      if (bytesRead < 0) return 0;

      totBytesRead += (unsigned)bytesRead;
    } while (totBytesRead < numChars);

    return totBytesRead;
  }

i am unable to understand from where the message will be send so that this
socket will receive the message.


regards
mritunjay





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Hi Mr. Ross,
i want to know if is possible to modify the trasmission frequency of =
Receive Report sent by a RTPplayer.
Thanks in advance.
Generoso
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<DIV><FONT face=3DArial size=3D2>i want to know if is possible to modify =
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trasmission frequency of Receive Report sent by&nbsp;a =
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X-Original-Date: Tue, 27 May 2003 02:01:04 -0700
Date: Tue, 27 May 2003 02:01:04 -0700


>i want to know if is possible to modify the trasmission frequency of 
>Receive Report

Yes, of course it's possible.  Remember, You Have Complete Source Code :-)

The RTCP reporting interval is calculated in the files "liveMedia/RTCP.cpp" 
and "liveMedia/rtcp_from_spec.c".  The latter comes directly from the 
RTP/RTCP specification (RFC).  So, you really shouldn't be modifying the 
RTCP reporting interval unless you have a good reason for wanting to 
violate the specification.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Tue, 27 May 2003 01:54:23 -0700
Date: Tue, 27 May 2003 01:54:23 -0700

At 10:01 PM 5/26/03, you wrote:
>hai
>actually i hvae included the in_rtp.dll in the specified location as given
>in the live.com site.
>But is this enough .
>shouldn't i make some changes in testmp3receivre.cpp and testmp3
>streamer.cpp

I'm sorry, but I still don't understand exactly what you're trying to do.

If you want to receive and play a HTTP stream in Winamp, then you don't 
need any LIVE.COM software at all.  Just open the "http://" URL in Winamp.

If you want to receive and play a MP3/RTP stream in Winamp, then you should 
just use the existing "in_rtp.dll" plugin (assuming you're running Winamp 
2.*, not Winamp 3).  It works.

>but i find that the thread excution gets blocked on the socket recv call
>as given in the code below.

If the code is blocking on "recv()", then it's because no packets are 
arriving on the socket!  So, you'll have to track this down yourself (e.g, 
using a network packet analyzer to figure out why you're not receiving packets.

Remember, You Have Complete Source Code.

         Ross.



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Subject: [Live-devel] streaming using winamp
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X-Original-Date: Tue, 27 May 2003 15:38:07 +0530
Date: Tue, 27 May 2003 15:38:07 +0530

hai
actually i have file called test.mp3. let us say i am streaming that file
from one pc(local pc with some ip address)
to another local pc ( with different ip address) at the receiver side
instead of storing that file in the local pc folder  i want to play it
using winamp.

 I am streaming the file let us say present in some folder in one pc i am
using rtp to stream.
I recive the stream at the receiver i want to direct this received stream
to winamp.

In your code u display the file on standard out instead  i directly want to
send that data
to winamp so that i start listening to Mp3 file getting streamed.

hope u could get my problem.

mritunjay



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From <EMAIL-ADDRESS-HIDDEN> Tue May 27 13:20:06 2003
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X-Original-Date: Tue, 27 May 2003 13:11:27 -0700
Date: Tue, 27 May 2003 13:11:27 -0700

At 03:08 AM 5/27/03, you wrote:
>hai
>actually i have file called test.mp3. let us say i am streaming that file
>from one pc(local pc with some ip address)
>to another local pc ( with different ip address) at the receiver side
>instead of storing that file in the local pc folder  i want to play it
>using winamp.
>
>  I am streaming the file let us say present in some folder in one pc i am
>using rtp to stream.
>I recive the stream at the receiver i want to direct this received stream
>to winamp.

OK, you can do this without having to download and build the "LIVE.COM 
Streaming Media" source code.  Instead, you can download *pre-built* binary 
applications that do what you want.

At the receiver (Winamp), you can use the "in_rtp.dll" plugin: 
<http://www.live.com/multikit/winamp-plugin.html>.  This assumes that 
you're running Winamp version 2.<something>.  If you're running Winamp 
version 3, then I suggest you get rid of it, and go back to (the much 
better and more stable) Winamp version 2.91: <http://classic.winamp.com/>

At the sender, you can use "liveCaster": <http://www.live.com/liveCaster/>

So, once again: To stream MP3 files via RTP, and receive/play them using 
Winamp you don't need to build any software.  Instead, you can just use the 
pre-built binaries "in_rtp.dll" and "liveCaster".


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Subject: [Live-devel] Simple Stream Server
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X-Original-Date: Wed, 28 May 2003 11:14:26 +0800
Date: Wed, 28 May 2003 11:14:26 +0800

This is a multi-part message in MIME format.
--------------090409000006030002070008
Content-Type: text/plain; charset=us-ascii; format=flowed
Content-Transfer-Encoding: 7bit

Hi ! Everyone

    Sorry about my poor english . I try to build a simple stream server 
with liveMedia library.
    The attachment code is my first incomplete implementation. I merge 
the RTSPServer.cpp ,
    testMP3Streamer.cpp and testMPEGAudioVideoStream.cpp into my test 
program . The program could
     stream mp3 and mpg file at current directory  . I have test mp3 
file and the program is work fine.
    When I stream mpeg file with audio and video track . And the client 
is linux mplayer . The video usually
    have some decode block like data lose or decode error . And I have 
other problem is stragely .
    I copy the code from testMP3Streamer.cpp that support USE_SSM for 
unicast and multicast .
    The program build ServerMediaSession when client connected and 
request "DESCRIBE" the resource .
    And I try to evaluate multi-client performances. I modify 
openRTSP.cpp and change output file to "/dev/null".
    The performance is fine when the client number is 60 . The loading 
is very low include user and system .
    But When client number is 80 , The system usage is rising to 70%-80% 
. And I profiling the kernel , I found
    loading of sock_poll() is exceptional high. The attach is profiling 
data of my kernel.

    Is anyone have the same experience with the library ?
    Kindly advise how to fix the problems.  Thanks in advance.

             Claud Yu


--------------090409000006030002070008
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X-Original-Date: Wed, 28 May 2003 02:24:24 -0700
Date: Wed, 28 May 2003 02:24:24 -0700


>   When I stream mpeg file with audio and video track . And the client is 
> linux mplayer . The video usually
>    have some decode block like data lose or decode error .

I believe this is a bug somewhere in MPlayer.

>And I have other problem is stragely .
>    I copy the code from testMP3Streamer.cpp that support USE_SSM for 
> unicast and multicast .
>    The program build ServerMediaSession when client connected and request 
> "DESCRIBE" the resource .
>    And I try to evaluate multi-client performances. I modify openRTSP.cpp 
> and change output file to "/dev/null".
>    The performance is fine when the client number is 60 . The loading is 
> very low include user and system .
>    But When client number is 80 , The system usage is rising to 70%-80% . 
> And I profiling the kernel , I found
>    loading of sock_poll() is exceptional high.

I haven't seen this before, and I don't know what might be causing this - 
but I suspect that perhaps something is causing the call to "select()" in 
"BasicTaskScheduler::SingleStep()" to return immediately, even when there 
is no readable socket.  This would cause 
"BasicTaskScheduler0::doEventLoop()" to loop without pausing.

I don't know why "select()" would do this - but this is something that you 
could check yourself.  Remember, You Have Complete Source Code.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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>Let's say I want RTP to stream H.263 bitstream in mode A defiend in RFC 2190.

Note that the payload format defined in RFC 2429 is now preferred over RFC 
2190, so you should be using RFC 2429 for your implementation.

Note that the "LIVE.COM Streaming Media" library implements the RFC 2429 
payload format - using the classes "H263plusVideoRTPSink" and 
"H263plusVideoRTPSource".

So, you can stream H.263 (or H.263+) using a "H263plusVideoRTPSink", 
provided that the input to this sink is a sequence of discrete video frames 
(i.e., not just a sequence of bytes).


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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From <EMAIL-ADDRESS-HIDDEN> Wed May 28 10:02:32 2003
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From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] RTP Payload Format for H.263 Streams
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>But I still don't know how to set those payload header.
>
>My question is that does H.263 codec provide API for RTP to query about or 
>does RTP provide API for codec to set the values or any other means.
>
>So could you tell which method in H263plusVideoRTPSink does this job.

If you feed a "H263plusVideoRTPSink" a sequence of H.263 (or H.263+) 
frames, then it will *automatically* set up the (2-byte) H.263+ payload 
header (as defined in RFC 2429, section 4.1.  This is done by 
"H263plusVideoRTPSink::doSpecialFrameHandling()" (see 
"liveMedia/H263plusVideoRTPSink.cpp").

Note, however, that this code does not include an optional "Video 
Redundancy Coding Header Extension", or an "Extra Picture 
Header".  Therefore, it sets the "V" bit to zero, and the "PLEN" field to 
zero.  If you don't want either of these optional features, then you can 
just use "H263plusVideoRTPSink" as is.  (Otherwise, you would need to 
modifiy the code accordingly.)


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Subject: Re: [Live-devel] how to pipe output of a encoder?
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Dear Ross,

    I made encoder and streamer as separate applications, and run "encoder | streamer". But streamer can't get any data from "stdin"(I have change test.mpg to stdin). I run only encoder to test, it do output data on console. Why?

    Another question ,when I combine streamer and encoder in one application. one thread running encoder (and writing to the pipe), the other thread runing the modified "testMPEGAudioVideoStreamer.cpp" code (and reading from the pipe). My movie player playback the mpeg-1 movie very slow and uncontinuous(when I play the data from testMPEGVideoStreamer using a file, it is ok).  Is that due to that the speed of encoder is too slow to feed the streamer? And the cpu usage is up to 100% and system becomes almost halt. Is there something wrong in "testMPEGAudioVideoStreamer.cpp"?

Regards,

Michael 

----- Original Message ----- 
From: {Ross Finlayson} <EMAIL-ADDRESS-HIDDEN>
To: <EMAIL-ADDRESS-HIDDEN>
Sent: Thursday, May 22, 2003 10:05 AM
Subject: Re: [Live-devel] how to pipe output of a encoder?


> 
> >    In fact I want to encapsulate my mpeg encoder and live media in one 
> > application, then send the output of encoder to testMPEGVideoStreamer. So 
> > I create a pipe as follows:
> >         int fields[2];
> >         pipe(fields);
> >     and then I put the output of mpeg encoder to fields[1], modify 
> > testMPEGVideoStreamer to read from fields[0] instead of "test.mpg". Is 
> > this right?
> 
> That should work.  You'll probably want to use separate threads for this: 
> one thread running your encoder (and writing to the pipe); the other thread 
> runing the modified "testMPEGVideoStreamer" code (and reading from the pipe).
> 
> However, I recommend that you *first* write your encoder and streamer as 
> separate applications, and pipe them together using the command 
> shell.  (Then, once you're sure that this is working OK, you can put them 
> into a single application.)
> 
>          Ross.
> 
> 
> 
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From <EMAIL-ADDRESS-HIDDEN> Thu May 29 00:57:26 2003
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Subject: Re: [Live-devel] Simple Stream Server
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X-Original-Date: Thu, 29 May 2003 09:53:05 +0200 (CEST)
Date: Thu, 29 May 2003 09:53:05 +0200 (CEST)

Hi all,

  I have a similar problem like this. Let me explain.

  I have testMPEGVideoStreamer running with a foo.mpg
file (elementary stream) in a Red Hat 9. I have
compiled MPlayer to play with rtsp format. I can start
several MPlayers and the CPU is low, there is no
problems :D, (i have connected also a QuickTime
instance in a Win System) if i stop a MPlayer instance
with CTRL-c, there is no problem but if i destroy the
MPlayer clicking in the output video form, ... PUMM!!
testMPEGVideoStreamer increases CPU consumtion to
85%/92%. 

  I know that we have source code, but i am new in
Linux systems. How could i debug
testMPEGVideoStreamer? Could i use an IDE like
KDevelop? 

  Thanks.



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From <EMAIL-ADDRESS-HIDDEN> Thu May 29 03:22:38 2003
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X-Original-Date: Thu, 29 May 2003 03:12:22 -0700
Date: Thu, 29 May 2003 03:12:22 -0700


>   I have testMPEGVideoStreamer running with a foo.mpg
>file (elementary stream) in a Red Hat 9. I have
>compiled MPlayer to play with rtsp format. I can start
>several MPlayers and the CPU is low, there is no
>problems :D, (i have connected also a QuickTime
>instance in a Win System) if i stop a MPlayer instance
>with CTRL-c, there is no problem but if i destroy the
>MPlayer clicking in the output video form, ... PUMM!!
>testMPEGVideoStreamer increases CPU consumtion to
>85%/92%.

I presume you're using "testMPEGVideoStreamer"s built-in RTSP server 
(otherwise  there is no way that it could be affected by what a client 
does).  I have just installed a new version (2003.05.29) of the software 
that fixes a bug in "RTSPServer" that could cause it to get into an 
infinite loop if a connected client dies.  I suspect that it will solve 
your problem.

>   I know that we have source code, but i am new in
>Linux systems. How could i debug
>testMPEGVideoStreamer? Could i use an IDE like
>KDevelop?

My main debugging tool is "fprintf()" :-)  But when I want something more 
high-tech, I use "gdb".  To use "gdb", first build the code using the "-g" 
flag.  (You can do this by running "./genMakefiles linux-gdb" before 
building the code.)


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Thu, 29 May 2003 03:05:56 -0700
Date: Thu, 29 May 2003 03:05:56 -0700


>    I made encoder and streamer as separate applications, and run "encoder 
> | streamer". But streamer can't get any data from "stdin"(I have change 
> test.mpg to stdin). I run only encoder to test, it do output data on 
> console. Why?

That's odd.  I don't know why this wouldn't be working.

>    Another question ,when I combine streamer and encoder in one 
> application. one thread running encoder (and writing to the pipe), the 
> other thread runing the modified "testMPEGAudioVideoStreamer.cpp" code 
> (and reading from the pipe). My movie player playback the mpeg-1 movie 
> very slow and uncontinuous(when I play the data from 
> testMPEGVideoStreamer using a file, it is ok).  Is that due to that the 
> speed of encoder is too slow to feed the streamer? And the cpu usage is 
> up to 100% and system becomes almost halt. Is there something wrong in 
> "testMPEGAudioVideoStreamer.cpp"?

No, I suspect that what's happening is that your encoder thread is writing 
to the pipe continuously, and is never blocking, so it's taking up most of 
your CPU time (and thus not giving the 'streamer' thread enough time to 
run).  Instead, you should arrange for the pipe to have a finite (and 
fairly small) maximum buffer size, so that the encoder thread doesn't run 
continuously, but instead only when it's needed to contribute more data.

(Trouble is, I don't know how to set a maximum buffer size on a pipe that 
was created using pipe(2) - perhaps using "ioctl" somehow??)


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Subject: Re: [Live-devel] how to pipe output of a encoder?
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Dear Ross,
    I run only streamer thread(streaming from a file), and the cpu used is also up to 100%. Why?
Regards,
Michael

----- Original Message ----- 
From: {Ross Finlayson} <EMAIL-ADDRESS-HIDDEN>
To: <EMAIL-ADDRESS-HIDDEN>
Sent: Thursday, May 29, 2003 6:05 PM
Subject: Re: [Live-devel] how to pipe output of a encoder?


> 
> >    I made encoder and streamer as separate applications, and run "encoder 
> > | streamer". But streamer can't get any data from "stdin"(I have change 
> > test.mpg to stdin). I run only encoder to test, it do output data on 
> > console. Why?
> 
> That's odd.  I don't know why this wouldn't be working.
> 
> >    Another question ,when I combine streamer and encoder in one 
> > application. one thread running encoder (and writing to the pipe), the 
> > other thread runing the modified "testMPEGAudioVideoStreamer.cpp" code 
> > (and reading from the pipe). My movie player playback the mpeg-1 movie 
> > very slow and uncontinuous(when I play the data from 
> > testMPEGVideoStreamer using a file, it is ok).  Is that due to that the 
> > speed of encoder is too slow to feed the streamer? And the cpu usage is 
> > up to 100% and system becomes almost halt. Is there something wrong in 
> > "testMPEGAudioVideoStreamer.cpp"?
> 
> No, I suspect that what's happening is that your encoder thread is writing 
> to the pipe continuously, and is never blocking, so it's taking up most of 
> your CPU time (and thus not giving the 'streamer' thread enough time to 
> run).  Instead, you should arrange for the pipe to have a finite (and 
> fairly small) maximum buffer size, so that the encoder thread doesn't run 
> continuously, but instead only when it's needed to contribute more data.
> 
> (Trouble is, I don't know how to set a maximum buffer size on a pipe that 
> was created using pipe(2) - perhaps using "ioctl" somehow??)
> 
> 
> Ross Finlayson
> LIVE.COM
> <http://www.live.com/>
> 
> 
> 
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X-Original-Date: Thu, 29 May 2003 07:52:40 -0700
Date: Thu, 29 May 2003 07:52:40 -0700


>    I run only streamer thread(streaming from a file), and the cpu used is 
> also up to 100%. Why?

Does this also happen if you run the original "testMPEGVideoStreamer" 
application - i.e., the version that comes with the source code release - 
unchanged?


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Thu, 29 May 2003 09:51:45 -0700
Date: Thu, 29 May 2003 09:51:45 -0700

Hello,
I need help. I am able to get Mplayer work without enabling live.
After I enable "live", it gives the following error message and stops the 
Mplayer "make" process. How can I fix it? Thanks very much!
Wendy
......more error message here, this is the end of it.
/home/httpd/image/live/BasicUsageEnvironment/libBasicUsageEnvironment.a(.gnu.lin 

konce.d.__vt_22BasicUsageEnvironment0+0x34): undefined reference to 
`__pure_virt
ual'
/home/httpd/image/live/BasicUsageEnvironment/libBasicUsageEnvironment.a(.gnu.lin 

konce.d.__vt_22BasicUsageEnvironment0+0x38): undefined reference to 
`__pure_virt
ual'
/home/httpd/image/live/BasicUsageEnvironment/libBasicUsageEnvironment.a(.gnu.lin 

konce.d.__vt_22BasicUsageEnvironment0+0x3c): more undefined references to 
`__pur
e_virtual' follow
collect2: ld returned 1 exit status
make: *** [mplayer] Error 1



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  "live"
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X-Original-Date: Thu, 29 May 2003 11:12:43 -0700
Date: Thu, 29 May 2003 11:12:43 -0700

At 09:51 AM 5/29/03, you wrote:
>Hello,
>I need help. I am able to get Mplayer work without enabling live.
>After I enable "live", it gives the following error message and stops the 
>Mplayer "make" process. How can I fix it?

Wendy,

Were you able to successfully build the "LIVE.COM Streaming Media" code, 
including the 'test programs' (i.e., those in the "testProgs" directory)?

If so, then I suspect that you were (inadvertently) compiling the LIVE.COM 
code and the rest of MPlayer with two different versions of "gcc".  Please 
check that you're compiling with the same version of "gcc" each time.

         Ross.



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X-Original-Date: Thu, 29 May 2003 14:31:20 -0700
Date: Thu, 29 May 2003 14:31:20 -0700

Ross,
Your suspect is right. Thank you!
I still need help. Since Mplayer needs gcc other than 2.96, I have both 
2.96 and 3.02 installed. Mplayer automatically chooses 3.02 to compile. 
When I run "make" for LIVE, I guess the system might choose 2.96 to run and 
it's successful. I modified config.linux as following. Will it let LIVE 
"make" to run by gcc version 3.02? I tried the whole process again and get 
the same error message for Mplayer "make". I am using RH 7.2 (kernel 
2.4.20) Thanks for the help!
COMPILE_OPTS =		$(INCLUDES) -I. -O -DSOCKLEN_T=socklen_t 
-I/usr/lib/gcc-lib/i386-redhat-linux/3.0.2
CPLUSPLUS_FLAGS =	$(COMPILE_OPTS) -Wall -DBSD=1 -Wno-deprecated



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X-Original-Date: Thu, 29 May 2003 14:47:10 -0700
Date: Thu, 29 May 2003 14:47:10 -0700


>Since Mplayer needs gcc other than 2.96

Are you sure that's true?

>, I have both 2.96 and 3.02 installed. Mplayer automatically chooses 3.02 
>to compile. When I run "make" for LIVE, I guess the system might choose 
>2.96 to run and it's successful. I modified config.linux as following. 
>Will it let LIVE "make" to run by gcc version 3.02?

Yes, you can compile the LIVE.COM code using gcc 3.*.  Because you have 
both compilers installed, you may need to modify the "CPLUSPLUS_COMPILER" 
and "C_COMPILER" definitions in "config.linux" to ensure that it uses the 
correct version.

You might also find that you don't have the include file "strstream.h" 
(although that's unlikely, because you also have gcc 2.* installed).  If 
so, you can just make "strstream.h" a symbolic link to "strstream".


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Subject: Re: [Live-devel] how to pipe output of a encoder?
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X-Original-Date: Fri, 30 May 2003 10:42:04 +0800
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Dear Ross,
 
> >    I run only streamer thread(streaming from a file), and the cpu used is 
> > also up to 100%. Why?
> 
> Does this also happen if you run the original "testMPEGVideoStreamer" 
> application - i.e., the version that comes with the source code release - 
> unchanged?

No, it doesn't. It only happens when I make it run as a thread.
and I also try to less the buffer of pipe(from 5000k to 10k), it seems no much help. 
I wonder what happen when 2 threads run in one application.

Thanks for your help.

Michael
ÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÓ†+ùÞµéšŠX¬²š'²ŠÞu¼ÿx²ëh}øœyê®Š™žž×è®W¬²‰Þ¬ÿ†ÛiÿÿÚuö«›ùžv&©•ìÿr‰¿ißÜ“þõ×ýuëÞÿëÝzÿŸÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿü¸¯{÷^½éfj)bž	b²Òâ½ïÝz÷¥þX¬¶Ïì¢êÜyú+ïçzØm¶›ÿÿùb²Ûÿ²‹«qçè®ÿë–+-³ùb²Ø§~åŠ÷ÿuëÞ


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X-Original-Date: Thu, 29 May 2003 22:02:49 -0700
Date: Thu, 29 May 2003 22:02:49 -0700


> > Does this also happen if you run the original "testMPEGVideoStreamer"
> > application - i.e., the version that comes with the source code release -
> > unchanged?
>
>No, it doesn't. It only happens when I make it run as a thread.

That's good.  You have a program that works properly, and a slightly 
modified version that doesn't.  So, it shouldn't be too hard to debug it, 
to figure out how your modifications have caused a problem.  Please let us 
know what you find.

         Ross.




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Subject: [Live-devel] ADU frame question ...
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X-Original-Date: Fri, 30 May 2003 01:25:46 -0700
Date: Fri, 30 May 2003 01:25:46 -0700


Ross et al,

We're considering adding an RFC3119 implementation to the VideoLAN 
open-source project (http://www.videolan.org).  I have a question, 
though.  Since MP3 frames often need data from others to play back 
correctly, and since the number of other frames could be up to 9, this 
implies that ADU frames are variable length, right?

Regards,

John

Falling You - exploring the beauty of voice and sound
New EP, "hope thrown down," available now at
http://www.mp3.com/fallingyou












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X-Original-Date: Fri, 30 May 2003 03:19:59 -0700
Date: Fri, 30 May 2003 03:19:59 -0700


>ADU frames are variable length, right?

That's correct.

Note, BTW, that there are a handful of typos in RFC 3119 (especially in the 
'pseudo code').  You should refer, instead, to 
<http://www.live.com/rtp-mp3.txt> (which will someday become a RFC, 
replacing RFC 3119.)


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Subject: Re: [Live-devel] how to pipe output of a encoder?
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Dear Ross,

    I change the pipe buffer size to 500k, it seems work. 2 thread work smoothly.
    But when I try to use unicast, I set the receiver ip to 0.0.0.0, but it can't receive media data. why?

Regards,
Michael
----- Original Message ----- 
From: {Ross Finlayson} <EMAIL-ADDRESS-HIDDEN>
To: <EMAIL-ADDRESS-HIDDEN>
Sent: Friday, May 30, 2003 1:02 PM
Subject: Re: [Live-devel] how to pipe output of a encoder?


> 
> > > Does this also happen if you run the original "testMPEGVideoStreamer"
> > > application - i.e., the version that comes with the source code release -
> > > unchanged?
> >
> >No, it doesn't. It only happens when I make it run as a thread.
> 
> That's good.  You have a program that works properly, and a slightly 
> modified version that doesn't.  So, it shouldn't be too hard to debug it, 
> to figure out how your modifications have caused a problem.  Please let us 
> know what you find.
> 
>          Ross.
> 
> 
> 
> 
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X-Original-Date: Fri, 30 May 2003 13:41:17 -0700
Date: Fri, 30 May 2003 13:41:17 -0700


>    But when I try to use unicast, I set the receiver ip to 0.0.0.0, but 
> it can't receive media data. why?

I don't know.  I checked this again just now, to make sure that it still 
works (for either a Windows or Unix receiver).  It worked OK for me.

<http://www.live.com/liveMedia/faq.html#unicast>


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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From <EMAIL-ADDRESS-HIDDEN> Fri May 30 19:04:38 2003
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X-Original-Date: Fri, 30 May 2003 18:54:53 -0700
Date: Fri, 30 May 2003 18:54:53 -0700

If you're using "SimpleRTPSink" - e.g., to stream PCM audio via RTP - then 
you should upgrade to the latest version (2003.05.31) of the LIVE.COM 
Streaming Media code.  I've fixed a bug that prevented RTP timestamps from 
being set correctly on outgoing RTP packets.

	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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From: eran <EMAIL-ADDRESS-HIDDEN>
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Subject: [Live-devel] vobStreamer  on windows
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X-Original-Date: Sun, 1 Jun 2003 09:00:42 +0300
Date: Sun, 1 Jun 2003 09:00:42 +0300

Hello,
I am trying to use the vobStreamer on windows XP, everything compiled
fine but when I run it I am getting the following message.
  
Beginning streaming...
Beginning to read from "VIDEO_TS.VOB"...
Created RTSP server.
Access this stream using the URL:
        rtsp://192.168.1.101/
StreamParser internal error (90307+ 10000 > 100000)

And when I try to play VTS_01_0.VOB it gets into a loop with out doing
any thing 
Beginning streaming...
Beginning to read from "VTS_01_0.VOB"...
...done reading from file
Beginning to read from "VTS_01_0.VOB"...
...done reading from file
Beginning to read from "VTS_01_0.VOB"...
...done reading from file
Beginning to read from "VTS_01_0.VOB"...
...done reading from file

 
........
........

Any help will be appreciated

Thanks 
Eran 






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X-Original-Date: Sat, 31 May 2003 23:17:41 -0700
Date: Sat, 31 May 2003 23:17:41 -0700


>StreamParser internal error (90307+ 10000 > 100000)
>
>And when I try to play VTS_01_0.VOB it gets into a loop with out doing
>any thing
>Beginning streaming...
>Beginning to read from "VTS_01_0.VOB"...
>...done reading from file
>Beginning to read from "VTS_01_0.VOB"...
>...done reading from file
>Beginning to read from "VTS_01_0.VOB"...
>...done reading from file
>Beginning to read from "VTS_01_0.VOB"...
>...done reading from file

It looks like you tried to play an encrypted DVD.  That won't work; 
"vobStreamer" works only on *unencrypted* ".vob" files.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Mon, 2 Jun 2003 14:58:08 -0400
Date: Mon, 2 Jun 2003 14:58:08 -0400

I am interested in adding support for the part of the RTSP protocol 
that allows streams to be paused and seek-able.  I was wondering 
exactly how the interaction between mplayer and the live.com libraries 
actually work.  I am assuming that I will need to modify RTSPClient, 
but will I also have to modify code in mplayer to get this to work 
properly in mplayer.  Do you have any advice for implementing this?  
Thanks.

Patrick



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X-Original-Date: Mon, 02 Jun 2003 12:08:54 -0700
Date: Mon, 02 Jun 2003 12:08:54 -0700

At 11:58 AM 6/2/03, you wrote:
>I am interested in adding support for the part of the RTSP protocol that 
>allows streams to be paused and seek-able.  I was wondering exactly how 
>the interaction between mplayer and the live.com libraries actually 
>work.  I am assuming that I will need to modify RTSPClient

Actually, not.  Recent versions of the "LIVE.COM Streaming Media" code adds 
- to "RTSPClient" - the member function "pauseMediaSubsession()".  Also, 
the "playMediaSubsession()" member function now has optional "start" and 
"end" time parameters.

>, but will I also have to modify code in mplayer to get this to work 
>properly in mplayer.

Yes, you would need to make some changes to the MPlayer code 
("libmpdemux/demux_rtp.cpp") that interfaces between the base MPlayer code, 
and the LIVE.COM libraries.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Tue, 03 Jun 2003 09:14:53 -0700
Date: Tue, 03 Jun 2003 09:14:53 -0700

Ross,
Here is the message from MPlayer - the message I got when I use gcc-2.96.

*** Please downgrade/upgrade C compiler to version gcc-2.95.x or gcc-3.x! ***

You are not using a supported compiler. We do not have the time to make sure
everything works with compilers other than the ones we use.  Use either the
same compiler as we do, or use --disable-gcc-checking but DO *NOT* REPORT BUGS
unless you can reproduce them after recompiling with a 2.95.x or 3.x version!

Note for gcc 2.96 users: Some versions of this compiler are known to miscompile
mplayer and lame (which is used for mencoder).  If you get compile errors,
first upgrade to the latest 2.96 release (minimum 2.96-85) and try again.
If the problem still exists, try with gcc 3.x (or 2.95.x) *BEFORE* reporting
bugs!

         GCC 2.96 IS NOT AND WILL NOT BE SUPPORTED BY US !

      *** For details please read DOCS/users_against_developers.html ***



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X-Original-Date: Tue, 03 Jun 2003 16:42:56 +0000
Date: Tue, 03 Jun 2003 16:42:56 +0000


Ross et al,

I'm adding functionality to the VideoLAN client project (VLC) to make it be able to seek to the start 
of an ADU (not implementing 3119 just yet, though that could happen later -- right now to be 
able to find the first MP3 frame in the sequence without a backpointer would be fine).

So, i'm looking in the liveMedia source, and I find (in MP3Internals.cpp) the following:

void MP3FrameParams::getSideInfo(MP3SideInfo& si) {
  // First skip over the CRC if present:     
  if (hasCRC) getBits(16);

  int single = -1;
  int ms_stereo, i_stereo;
  int sfreq = samplingFreqIndex;

  if (stereo == 1) {
    single = 0;
  }

  ms_stereo = (mode == MPG_MD_JOINT_STEREO) && (mode_ext & 0x2);
  i_stereo = (mode == MPG_MD_JOINT_STEREO) && (mode_ext & 0x1);

  if (isMPEG2) {
    getSideInfo2(*this, si, stereo, ms_stereo, sfreq, single);
  } else {
    getSideInfo1(*this, si, stereo, ms_stereo, sfreq, single);
  }    
}

... now getSideInfo1() does a getBits(8) to fill main_data_begin, while getSideInfo2() does getBits(9).  
Can I infer from this that the size of the backpointer is only 8 bits for MPEG2 layer III, and only 9 
bits for MPEG1 layer III?

Regards,

John

--
Falling You - exploring the beauty 
of voice and sound
http://www.mp3.com/fallingyou










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X-Original-Date: Tue, 03 Jun 2003 09:56:50 -0700
Date: Tue, 03 Jun 2003 09:56:50 -0700

At 09:42 AM 6/3/03, you wrote:
>Can I infer from this that the size of the backpointer is only 8 bits for 
>MPEG2 layer III, and only 9
>bits for MPEG1 layer III?

Yes.

(You could also look in the MPEG (1 and 2) audio specification, noting the 
definition of "main_data_begin".)


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Tue, 03 Jun 2003 10:00:12 -0700
Date: Tue, 03 Jun 2003 10:00:12 -0700


>Here is the message from MPlayer - the message I got when I use gcc-2.96.
>
>*** Please downgrade/upgrade C compiler to version gcc-2.95.x or gcc-3.x! ***

OK, so do what the friendly message says: Downgrade (or upgrade) your "gcc" 
version.

Note that this is an issue with the *MPlayer* code - not the "LIVE.COM 
Streaming Media" code.  As far as I know, the LIVE.COM code doesn't have 
any problems with "gcc" version 2.96.   Of course, though, you should use 
the same version of the compiler to build both.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Tue, 03 Jun 2003 18:48:20 +0000
Date: Tue, 03 Jun 2003 18:48:20 +0000


Ross et al,

Thanks for the confirmation, it is greatly appreciated.  Here are the first 8 bytes of some frames 
from an mp3 that iTunes created (320k, stereo):

FB      E2      0       0       0       6       EF      86
FB      E2      0       0       0       7       65      8C
FB      E2      0       0       0       7       69      8B
FB      E2      0       0       0       7       2A      89
FB      E2      0       0       0       7       D       83
FB      E2      0       0       0       6       9A      87
FB      E0      0       0       0       7       1D      88
FB      E2      0       0       0       7       34      88
FB      E2      0       0       0       7       3E      8B
FB      E2      0       0       0       6       FB      8A
FB      E2      0       0       0       7       68      8A
FB      E2      0       0       0       7       24      87
FB      E2      0       0       0       7       2C      87
FB      E0      0       0       0       6       A3      89
FB      E2      0       0       0       6       E9      87
FB      E2      0       0       0       6       E6      87
FB      E2      0       0       0       7       4D      88
FB      E2      0       0       0       7       27      89
FB      E2      0       0       0       6       F6      89
FB      E2      0       0       0       6       CA      89
FB      E0      0       0       0       7       63      86
FB      E2      0       0       0       6       E5      84
FB      E2      0       0       0       6       96      86

... so the FB are the sync bits, and none of these frames have a CRC (because the second byte is 
always even).  Since there is no CRC, the main_data_begin should begin at the 5th byte, right?  
Since this is MPEG1, Layer III, this seems to imply that all of these are ADU frames (5th byte is 
0x00, 6th byte is less than 0x10).  Is this correct?

Sorry to post bytes at you, i'm just trying to make sure I understand this.  I truly appreciate the 
assistance!

Regards,

John


--
Falling You - exploring the beauty 
of voice and sound
http://www.mp3.com/fallingyou










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Subject: [Live-devel] Interoperability with live.com library (RTP)
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X-Original-Date: Tue, 03 Jun 2003 12:43:56 -0700
Date: Tue, 03 Jun 2003 12:43:56 -0700

Hi,

I've written an RTP/RTSP server (streaming live MPEG-1 Video ES) which I'm
testing with mplayer using the live.com library, and QuickTime 6.1/6.2 (Real
seems to have lots of other problems)

Last week, off list, Ross helped me solve a problem with beginning/end of
slice bits, but asked me to post to the list next time I had a problem, so
here I am. 

The problem I'm seeing in mplayer is that it seems to be losing frames
(video breaks up) every second or so. OpenRTSP test program, also reports
lost packets. In quicktime the stream plays fine continuously (I've got a
stream going for 17.5 hours straight currently, with virtually no packet
loss). 

I'm thinking that it's not that it's actually losing packets, but perhaps
it's discarding them for some reason. Perhaps there is still something wrong
with my RTP headers.

I've analyzed the headers after capturing my stream with ethereal, and all
RTP headers looks fine now (the markers, beginning/end of slice bits,
timestamps based on GOP timecode, all look fine now after fixing some bugs).
I've even compared the captured stream to one captured from
testMPEGVideoStreamer, which is streaming a 60MB file that was output from
my program. The RTP headers from my server look very similar to the ones
from the testMPEGVideoStreamer, or at least I can't tell what the
differences are. Perhaps it's something very subtle, as I've run out of
things to try. The testMPEGVideoStreamer stream is received by mplayer as
well as QuickTime, with no problems.

Quicktime exhibits one problem with my stream, but it's of a different
nature (but perhaps it's related). Essentially once in a while when I open
the stream, QT doesn't know what stream type it is, and displays just the
controller with no Video. After a couple of minutes the video will show up.
If I pause it while no video exists, and unpause it, the video will show up
fine. This problem never occurs when I first open the stream, but usually
after opening/closing it > 10 times.

So I'm thinking that there's still something wrong that I'm doing/not doing.

Is there a debug setting for the live library that I can turn on that would
perhaps complain about the stream I'm sending it, if it encountered
something wrong? Does anyone know of anything that could be causing these
symptoms?

Thanks,

Tym



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X-Original-Date: Tue, 03 Jun 2003 13:13:44 -0700
Date: Tue, 03 Jun 2003 13:13:44 -0700


>The problem I'm seeing in mplayer is that it seems to be losing frames
>(video breaks up) every second or so. OpenRTSP test program, also reports
>lost packets. In quicktime the stream plays fine continuously (I've got a
>stream going for 17.5 hours straight currently, with virtually no packet
>loss).

That's interesting.  This suggests that there might well be a bug in the 
LIVE.COM code that's causing packets to get lost.  Could you try running
         openRTSP -V -e 60 -Q 1 rtsp://<whatever>
and mail me the output, so I can take a look at what it's reporting?

(I assume that you're streaming using UDP in all cases, and that QuickTime 
Player is not inadvertently streaming using TCP.  If your server supports 
RTP-over-TCP streaming, then you could also try running "openRTSP" with the 
-t option, just to verify that you don't get any packet loss in that case.)

>Is there a debug setting for the live library that I can turn on that would
>perhaps complain about the stream I'm sending it, if it encountered
>something wrong?

Not really.  You can find
         #ifdef DEBUG
         #endif
in various places, but there's no systematic debugging output mechanism.

You can check for packet loss in "liveMedia/MultiFramedRTPSource", in the 
function "ReorderingPacketBuffer::getNextCompletedPacket()".  Note the 
statement
         packetLossPreceded = True;
If this statement is reached, the code detected a lost packet.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Subject: Re: [Live-devel] Interoperability with live.com library (RTP)
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X-Original-Date: Tue, 03 Jun 2003 14:14:28 -0700
Date: Tue, 03 Jun 2003 14:14:28 -0700

on 6/3/03 1:13 PM, Ross Finlayson at finlayson(at)live.com wrote:
> (I assume that you're streaming using UDP in all cases, and that QuickTime
> Player is not inadvertently streaming using TCP.  If your server supports

Yes. QT is using UDP as well (I don't support TCP).

Here is the output. Interestingly enough the packet loss isn't that high
this time. I've seen it higher.

Thanks for taking the time to take a look at this. I've also noticed that
the packet loss seems to get lower if I lover my bitrate.

One other thing I noticed while examining packets with ethereal, is that you
seem to be setting the M marker bit in the RTP header at the start of the
last slice in a picture. Ie. If the last slice spans 2 or 3 packets then the
M is set for the first packet instead of the 2nd or 3rd. Could that be
making a difference. I'm going off of the update of RFC 2250 presented at
the 51st IETF, where it says to put it in the last packet containing the
last byte of the last slice in the picture.

Here is the output you asked for:

Sending request: DESCRIBE rtsp://192.168.173.2:3554/1085102592571150095
RTSP/1.0
CSeq: 1
Accept: application/sdp
User-Agent: /usr/lib/live/testProgs/openRTSP (LIVE.COM Streaming Media
v2003.05.19)


Received DESCRIBE response: RTSP/1.0 200 OK
CSeq: 1
Content-base: rtsp://192.168.173.2:3554/1085102592571150095
Content-type: application/sdp
Content-length: 398


Need to read 398 extra bytes
Read 398 extra bytes: v=0
o=- 32233 1 IN IP4 bycast.com
s=Bycast Unicast Channel One (MPEG1)
i=The Universal Header for Quicktime, RealOne, and Mplayer
u=http://www.bycast.com/
e=Bycast <bugs(at)bycast.com>
b=AS:2500
t=0 0
c=IN IP4 192.168.173.2
m=video 0 RTP/AVP 32
a=rtpmap:32 MPV/90000
a=framerate:29.970
a=mimetype:string;"video/MPV/90000"
a=cliprect:0,0,240,320
a=x-bufferdelay:1.1
a=control:trackID=1

Opened URL "rtsp://192.168.173.2:3554/1085102592571150095", returning a SDP
description:
v=0
o=- 32233 1 IN IP4 bycast.com
s=Bycast Unicast Channel One (MPEG1)
i=The Universal Header for Quicktime, RealOne, and Mplayer
u=http://www.bycast.com/
e=Bycast <bugs(at)bycast.com>
b=AS:2500
t=0 0
c=IN IP4 192.168.173.2
m=video 0 RTP/AVP 32
a=rtpmap:32 MPV/90000
a=framerate:29.970
a=mimetype:string;"video/MPV/90000"
a=cliprect:0,0,240,320
a=x-bufferdelay:1.1
a=control:trackID=1

Created receiver for "video/MPV" subsession (client ports 1176-1177)
Sending request: SETUP
rtsp://192.168.173.2:3554/1085102592571150095/trackID=1 RTSP/1.0
CSeq: 2
Transport: RTP/AVP;unicast;client_port=1176-1177
User-Agent: /usr/lib/live/testProgs/openRTSP (LIVE.COM Streaming Media
v2003.05.19)


Received SETUP response: RTSP/1.0 200 OK
CSeq: 2
Session: 547140464
Transport: RTP/AVP;unicast;client_port=1176-1177;server_port=6970-6971


Setup "video/MPV" subsession (client ports 1176-1177)
Created output file: "video-MPV-1"
Sending request: PLAY rtsp://192.168.173.2:3554/1085102592571150095 RTSP/1.0
CSeq: 3
Session: 547140464
Range: npt=0-
User-Agent: /usr/lib/live/testProgs/openRTSP (LIVE.COM Streaming Media
v2003.05.19)


Received PLAY response: RTSP/1.0 200 OK
CSeq: 3
Session: 547140464
Range: npt=0-
RTP-Info: url=trackID=1;seq=0


Started playing session
Receiving streamed data (for up to 60 seconds)...
begin_QOS_statistics
server_availability     100
stream_availability     100
subsession      video/MPV
num_packets_received    13701
num_packets_lost        106
elapsed_measurement_time        60.0014
kBytes_received_total   9344.67
measurement_sampling_interval_ms        100
kbits_per_second_min    0
kbits_per_second_ave    1245.93
kbits_per_second_max    6579.71
packet_loss_percentage_min      0
packet_loss_percentage_ave      0.767727
packet_loss_percentage_max      55.6818
inter_packet_gap_ms_min 0.036
inter_packet_gap_ms_ave 4.35595
inter_packet_gap_ms_max 465.707
end_QOS_statistics
Closing "video/MPV" subsession
Sending request: TEARDOWN
rtsp://192.168.173.2:3554/1085102592571150095/trackID=1 RTSP/1.0
CSeq: 4
Session: 547140464
User-Agent: /usr/lib/live/testProgs/openRTSP (LIVE.COM Streaming Media
v2003.05.19)


Thanks,

Tym Altman
Bycast Inc.



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Subject: [Live-devel] LIVE can not compile with MPlayer
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X-Original-Date: Tue, 3 Jun 2003 15:47:23 -0700
Date: Tue, 3 Jun 2003 15:47:23 -0700

This is a multi-part message in MIME format.

------=_NextPart_000_0346_01C329E7.6C9082A0
Content-Type: text/plain;
	charset="Windows-1252"
Content-Transfer-Encoding: quoted-printable

Ross,=20
It works to complile MPlayer only. I can play movie file and live link =
without any problem. Once I enable LIVE to make MPlayer, it gives me the =
following error. I changed the config.linux as following to have LIVE =
use the same compilers as MPlayer.=20
I ran "genMakefiles linux" first and then ran "make". I need help to =
have LIVE work. Thanks very much!

LIVE make message:=20
                      make
cd BasicUsageEnvironment ; make
make[1]: Entering directory =
`/home/httpd/image/live/BasicUsageEnvironment'
make[1]: Nothing to be done for `all'.
make[1]: Leaving directory =
`/home/httpd/image/live/BasicUsageEnvironment'
cd testProgs ; make
make[1]: Entering directory `/home/httpd/image/live/testProgs'
make[1]: Nothing to be done for `all'.
make[1]: Leaving directory `/home/httpd/image/live/testProgs'


config.linux file
COMPILE_OPTS =3D          $(INCLUDES) -I. -O -DSOCKLEN_T=3Dsocklen_t =
-I/usr/lib/gcc-
lib/i386-redhat-linux/3.0.2
C =3D                     c
C_COMPILER =3D            cc =
-I/home/httpd/image/MPlayer-20030527/TOOLS/c
C_FLAGS =3D               $(COMPILE_OPTS)
CPP =3D                   cpp
CPLUSPLUS_COMPILER =3D    c++ -I/usr/share/doc/ncurses-devel-5.2/c++
CPLUSPLUS_FLAGS =3D       $(COMPILE_OPTS) -Wall -DBSD=3D1
OBJ =3D                   o
LINK =3D                  c++ -o
LINK_OPTS =3D             -L.
CONSOLE_LINK_OPTS =3D     $(LINK_OPTS)
LIBRARY_LINK =3D          ld -o
LIBRARY_LINK_OPTS =3D     $(LINK_OPTS) -r -Bstatic
LIB_SUFFIX =3D                    a
LIBS_FOR_CONSOLE_APPLICATION =3D
LIBS_FOR_GUI_APPLICATION =3D
EXE =3D

LIVE enabled MPlayer make error:=20
....more error - this is the end of it:=20
/home/httpd/image/live/BasicUsageEnvironment/libBasicUsageEnvironment.a(.=
gnu.lin
konce.d.__vt_22BasicUsageEnvironment0+0x38): undefined reference to =
`__pure_virt
ual'
/home/httpd/image/live/BasicUsageEnvironment/libBasicUsageEnvironment.a(.=
gnu.lin
konce.d.__vt_22BasicUsageEnvironment0+0x3c): more undefined references =
to `__pur
e_virtual' follow
collect2: ld returned 1 exit status
make: *** [mplayer] Error 1

------=_NextPart_000_0346_01C329E7.6C9082A0
Content-Type: text/html;
	charset="Windows-1252"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dwindows-1252">
<META content=3D"MSHTML 6.00.2800.1106" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Ross, </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>It works to complile MPlayer only. I =
can play movie=20
file and live link without any problem. Once I&nbsp;enable LIVE to make =
MPlayer,=20
it gives me the following error. I changed the config.linux as following =

to&nbsp;have&nbsp;LIVE use the same compilers as MPlayer. </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>I ran "genMakefiles linux" =
first&nbsp;and then ran=20
"make". I need help to have LIVE work. Thanks very much!</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>LIVE make message: </FONT></DIV>
<DIV><FONT face=3DArial=20
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
make<BR>cd BasicUsageEnvironment ; make<BR>make[1]: Entering directory=20
`/home/httpd/image/live/BasicUsageEnvironment'<BR>make[1]: Nothing to be =
done=20
for `all'.<BR>make[1]: Leaving directory=20
`/home/httpd/image/live/BasicUsageEnvironment'<BR>cd testProgs ;=20
make<BR>make[1]: Entering directory=20
`/home/httpd/image/live/testProgs'<BR>make[1]: Nothing to be done for=20
`all'.<BR>make[1]: Leaving directory=20
`/home/httpd/image/live/testProgs'<BR></FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>config.linux file</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>COMPILE_OPTS=20
=3D&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; $(INCLUDES) =
-I. -O=20
-DSOCKLEN_T=3Dsocklen_t =
-I/usr/lib/gcc-<BR>lib/i386-redhat-linux/3.0.2<BR>C=20
=3D&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
c<BR>C_COMPILER=20
=3D&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; cc =

-I/home/httpd/image/MPlayer-20030527/TOOLS/c<BR>C_FLAGS=20
=3D&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;=20
$(COMPILE_OPTS)<BR>CPP=20
=3D&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
cpp<BR>CPLUSPLUS_COMPILER =3D&nbsp;&nbsp;&nbsp; c++=20
-I/usr/share/doc/ncurses-devel-5.2/c++<BR>CPLUSPLUS_FLAGS=20
=3D&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; $(COMPILE_OPTS) -Wall =
-DBSD=3D1<BR>OBJ=20
=3D&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
o<BR>LINK=20
=3D&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
c++ -o<BR>LINK_OPTS=20
=3D&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;=20
-L.<BR>CONSOLE_LINK_OPTS =3D&nbsp;&nbsp;&nbsp;&nbsp; =
$(LINK_OPTS)<BR>LIBRARY_LINK=20
=3D&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; ld=20
-o<BR>LIBRARY_LINK_OPTS =3D&nbsp;&nbsp;&nbsp;&nbsp; $(LINK_OPTS) -r=20
-Bstatic<BR>LIB_SUFFIX=20
=3D&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
a<BR>LIBS_FOR_CONSOLE_APPLICATION =3D<BR>LIBS_FOR_GUI_APPLICATION =
=3D<BR>EXE=20
=3D<BR></FONT></DIV>
<DIV><FONT face=3DArial size=3D2>LIVE enabled MPlayer make error: =
</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>....more error - this is the end of it: =

</FONT></DIV>
<DIV><FONT face=3DArial=20
size=3D2>/home/httpd/image/live/BasicUsageEnvironment/libBasicUsageEnviro=
nment.a(.gnu.lin<BR>konce.d.__vt_22BasicUsageEnvironment0+0x38):=20
undefined reference to=20
`__pure_virt<BR>ual'<BR>/home/httpd/image/live/BasicUsageEnvironment/libB=
asicUsageEnvironment.a(.gnu.lin<BR>konce.d.__vt_22BasicUsageEnvironment0+=
0x3c):=20
more undefined references to `__pur<BR>e_virtual' follow<BR>collect2: ld =

returned 1 exit status<BR>make: *** [mplayer] Error=20
1<BR></FONT></DIV></BODY></HTML>

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Subject: Re: [Live-devel] LIVE can not compile with MPlayer
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>....more error - this is the end of it:
>/home/httpd/image/live/BasicUsageEnvironment/libBasicUsageEnvironment.a(.gnu.lin
>konce.d.__vt_22BasicUsageEnvironment0+0x38): undefined reference to 
>`__pure_virt
>ual'
>/home/httpd/image/live/BasicUsageEnvironment/libBasicUsageEnvironment.a(.gnu.lin
>konce.d.__vt_22BasicUsageEnvironment0+0x3c): more undefined references to 
>`__pur
>e_virtual' follow
>collect2: ld returned 1 exit status
>make: *** [mplayer] Error 1

This shows that you haven't built the LIVE.COM code and the MPlayer code 
with the same compilers.  You should do a "make clean" in both directories, 
and try again.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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>Here is the output.

Thanks.  Unfortunately it doesn't tell me a whole lot, except that packet 
loss is probably happening in bursts.

Would it be possible for you to make a stream (using your server) available 
on the public Internet so I can take a closer look at this issue?  (I'm 
concerned about the fact that QuickTime Player apparently does not see any 
packet loss.)

>One other thing I noticed while examining packets with ethereal, is that you
>seem to be setting the M marker bit in the RTP header at the start of the
>last slice in a picture. Ie. If the last slice spans 2 or 3 packets then the
>M is set for the first packet instead of the 2nd or 3rd. Could that be
>making a difference. I'm going off of the update of RFC 2250 presented at
>the 51st IETF, where it says to put it in the last packet containing the
>last byte of the last slice in the picture.

Yes, good point.  This will be fixed in the next release.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Tue, 03 Jun 2003 17:20:13 -0700
Date: Tue, 03 Jun 2003 17:20:13 -0700


>Thanks for the confirmation, it is greatly appreciated.  Here are the 
>first 8 bytes of some frames
>from an mp3 that iTunes created (320k, stereo):
>
>FB      E2      0       0       0       6       EF      86

No - this cannot be a MPEG-1 or 2 audio frame.  For such frames, the first 
eleven bits of the 'syncword' must be all-one.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Subject: [Live-devel] Re: LIVE can not compile with MPlayer
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Ross,=20
I used following config.linux and I ran "make clean" as well.=20
No error message for LIVE make and the same error message if I enable =
LIVE with MPlayer. No error message if I run MPlayer only.=20
It seems like MPlayer uses "gcc3" to compile most of time. Should I add =
something different in config.linux for LIVE make? Thanks a bunch!

COMPILE_OPTS =3D          $(INCLUDES) -I. -O -DSOCKLEN_T=3Dsocklen_t =
-I/usr/lib/gcc-
lib/i386-redhat-linux/3.0.2
C =3D                     c
C_COMPILER =3D            cc -I/usr/lib/gcc-lib/i386-redhat-linux/3.0.2
C_FLAGS =3D               $(COMPILE_OPTS)
CPP =3D                   cpp
CPLUSPLUS_COMPILER =3D    c++ -I/usr/lib/gcc-lib/i386-redhat-linux/3.0.2
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OBJ =3D                   o
LINK =3D                  c++ -o
LINK_OPTS =3D             -L.
CONSOLE_LINK_OPTS =3D     $(LINK_OPTS)
LIBRARY_LINK =3D          ld -o
LIBRARY_LINK_OPTS =3D     $(LINK_OPTS) -r -Bstatic
LIB_SUFFIX =3D                    a
LIBS_FOR_CONSOLE_APPLICATION =3D
LIBS_FOR_GUI_APPLICATION =3D
EXE =3D

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<DIV><FONT face=3DArial size=3D2>Ross, </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>I used following config.linux and I ran =
"make=20
clean" as well. </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>No error message for LIVE make and the =
same error=20
message if I enable LIVE with MPlayer. No error message if I run MPlayer =
only.=20
</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>It seems like MPlayer uses "gcc3" to =
compile most=20
of time. Should I add something different in config.linux for LIVE make? =
Thanks=20
a bunch!</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>COMPILE_OPTS=20
=3D&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; $(INCLUDES) =
-I. -O=20
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=3D&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
c<BR>C_COMPILER=20
=3D&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; cc =

-I/usr/lib/gcc-lib/i386-redhat-linux/3.0.2<BR>C_FLAGS=20
=3D&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
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=3D&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; $(COMPILE_OPTS) -Wall =
-DBSD=3D1<BR>OBJ=20
=3D&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
o<BR>LINK=20
=3D&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
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c++ -o<BR>LINK_OPTS=20
=3D&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;=20
-L.<BR>CONSOLE_LINK_OPTS =3D&nbsp;&nbsp;&nbsp;&nbsp; =
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a<BR>LIBS_FOR_CONSOLE_APPLICATION =3D<BR>LIBS_FOR_GUI_APPLICATION =
=3D<BR>EXE=20
=3D<BR></DIV></FONT></BODY></HTML>

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X-Original-Date: Tue, 3 Jun 2003 17:37:14 -0700
Date: Tue, 3 Jun 2003 17:37:14 -0700


Ross,

>> FB      E2      0       0       0       6       EF      86
>
> No - this cannot be a MPEG-1 or 2 audio frame.  For such frames, the 
> first eleven bits of the 'syncword' must be all-one.

Yes -- silly bug in the for loop printing the bytes -- the initial FF 
wasn't printed.  I found out something interesting, though, which I 
guess makes sense, I just never thought of it before.  If I print out 
the frame headers of MP3s that I encode with iTunes at 320K, every 
single one has no backpointer.  Perhaps at 320kbps, the byte reservoir 
just isn't needed, as there are enough bits in every frame to encode 
anything and everything audible?  If I look at the frame headers for, 
say, 128k MP3s, almost every frame has a backpointer (though the 
initial one, predictably, does not).

So this leads me to think of ways I could make my code smarter, which 
makes me want to look at the liveMedia source again (parts of it I 
didn't look much at before, that is).  Is there a limit to how many 
frames another frame can be predicated on?  I mean, if a decoder has to 
cache n previous frames, I would think n would have some upper limit 
(that is manageable on, say, embedded systems with low RAM).

I have to entertain the possibility that this discussion isn't entirely 
appropriate for the live-devel list, though it applies somewhat.  If 
you would prefer I keep this off list, i'll oblige.

Regards,

John

Falling You - exploring the beauty of voice and sound
New EP, "hope thrown down," available now at
http://www.mp3.com/fallingyou













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Subject: Re: [Live-devel] Interoperability with live.com library (RTP)
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X-Original-Date: Tue, 03 Jun 2003 17:48:15 -0700
Date: Tue, 03 Jun 2003 17:48:15 -0700

on 6/3/03 4:26 PM, Ross Finlayson at finlayson(at)live.com wrote:

> Thanks.  Unfortunately it doesn't tell me a whole lot, except that packet
> loss is probably happening in bursts.

Yeah..it's pretty bursty. Perhaps that's the problem.

> Would it be possible for you to make a stream (using your server) available
> on the public Internet so I can take a closer look at this issue?  (I'm
> concerned about the fact that QuickTime Player apparently does not see any
> packet loss.)

QuickTime player has a 3 second buffer, so it plays from this buffer, while
your code doesn't appear to have this buffer (which is actually a good
thing, I wish QT didn't have the minimum 3 second buffer), so maybe that's
why it plays nicely in QT?

I'll have to discuss this with our sys admin. From prior experience I know
he doesn't like making our boxes public, but I'll give it a shot.

Thanks for your help.

Tym Altman



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--- Tym Altman <tym(at)bycast.com> wrote:
> on 6/3/03 4:26 PM, Ross Finlayson at
> finlayson(at)live.com wrote:
> 
> > Thanks.  Unfortunately it doesn't tell me a whole
> lot, except that packet
> > loss is probably happening in bursts.
> 
> Yeah..it's pretty bursty. Perhaps that's the
> problem.
> 
> > Would it be possible for you to make a stream
> (using your server) available
> > on the public Internet so I can take a closer look
> at this issue?  (I'm
> > concerned about the fact that QuickTime Player
> apparently does not see any
> > packet loss.)
> 
> QuickTime player has a 3 second buffer, so it plays
> from this buffer, while
> your code doesn't appear to have this buffer (which
> is actually a good
> thing, I wish QT didn't have the minimum 3 second
> buffer), so maybe that's
> why it plays nicely in QT?
> 
> I'll have to discuss this with our sys admin. From
> prior experience I know
> he doesn't like making our boxes public, but I'll
> give it a shot.
> 
> Thanks for your help.
> 
> Tym Altman
> 
> 
> 
>
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X-Original-Date: Tue, 3 Jun 2003 21:40:03 -0700 (PDT)
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--- vidya sagar yellapu <ysd4477(at)yahoo.com> wrote:
> 
> --- Tym Altman <tym(at)bycast.com> wrote:
> > on 6/3/03 4:26 PM, Ross Finlayson at
> > finlayson(at)live.com wrote:
> > 
> > > Thanks.  Unfortunately it doesn't tell me a
> whole
> > lot, except that packet
> > > loss is probably happening in bursts.
> > 
> > Yeah..it's pretty bursty. Perhaps that's the
> > problem.
> > 
> > > Would it be possible for you to make a stream
> > (using your server) available
> > > on the public Internet so I can take a closer
> look
> > at this issue?  (I'm
> > > concerned about the fact that QuickTime Player
> > apparently does not see any
> > > packet loss.)
> > 
> > QuickTime player has a 3 second buffer, so it
> plays
> > from this buffer, while
> > your code doesn't appear to have this buffer
> (which
> > is actually a good
> > thing, I wish QT didn't have the minimum 3 second
> > buffer), so maybe that's
> > why it plays nicely in QT?
> > 
> > I'll have to discuss this with our sys admin. From
> > prior experience I know
> > he doesn't like making our boxes public, but I'll
> > give it a shot.
> > 
> > Thanks for your help.
> > 
> > Tym Altman
> > 
> > 
> > 
> >
>
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From <EMAIL-ADDRESS-HIDDEN> Wed Jun 4 02:28:21 2003
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X-Original-Date: Wed, 04 Jun 2003 02:18:30 -0700
Date: Wed, 04 Jun 2003 02:18:30 -0700


>Is there a limit to how many frames another frame can be predicated on?  I 
>mean, if a decoder has to cache n previous frames, I would think n would 
>have some upper limit

I don't think there's a limit on the number of previous frames over which a 
backpointer can range - but of course there's a limit on the *range* of the 
backpointer (based on its size in bits).

Anyway, you can probably find out for sure from the MPEG standards...


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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>QuickTime player has a 3 second buffer, so it plays from this buffer, while
>your code doesn't appear to have this buffer

Actually, it does have buffering.  "MultiFramedRTPSource" sets the OS's UDP 
socket buffering to 50 kBytes, by default.  The "MPlayer" code, though, 
increases this to 2 MBytes for video, and 100 kBytes for audio.  (Look for 
"increaseReceiveBufferTo".)


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Wed, 04 Jun 2003 02:27:42 -0700
Date: Wed, 04 Jun 2003 02:27:42 -0700


>Should I add something different in config.linux for LIVE make?

You should make whatever changes are necessary to "config.linux" (in 
particular, the definitions of C_COMPILER and CPLUSPLUS_COMPILER) so that 
the LIVE.COM code builds with the same compiler as the MPlayer code.  You 
have control here!


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Hi all,

I have a simple comprehension question :
I saw in testMPEGVideoStreamer "MediaSource" & "MPEGVideoStreamFramer". This read a MPEG 1 or 2 file and streams via "MPEGVideoRTPSink". 
But how can I stream MPEG4 or H263(+) file ? I've seen "H263plusVideoRTPSink " for example, how can I use it to stream my file ?

regards,

DoObS

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Subject: RE: [Live-devel] Interoperability with live.com library (RTP)
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X-Original-Date: Wed, 4 Jun 2003 18:12:34 +0200
Date: Wed, 4 Jun 2003 18:12:34 +0200


Hi Tim,

   I'm not sure, but this is what I would say, with all the knowledge I have
now (almost nothing in fact).
   Concerning your problem with RTP bad synchronization (might be lost
frames, or simply a resync happening), it's probably due to RTCP timestamp
being wrong. I know, it should be RTCP that fails, but it's what happened
with me.

In RTCP, timestamp are linked (I haven't understood yet how) with network
time, and above all they are used to resync the rtp'ed flux.

I have no idea how MPlayer is supposed to react to (bad) RTCP packet. But if
it reacts correspondingly with the standard, it should resync, and then
seems to loose frames.

I hope it's what it is going wrong with your implementation.
I would feel interested in the real reason linked with this problem, so if
you found it, please tell me.

Sincerly
Cyril Russo









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X-Original-Date: Wed, 04 Jun 2003 10:53:37 -0700
Date: Wed, 04 Jun 2003 10:53:37 -0700


>In RTCP, timestamp are linked (I haven't understood yet how) with network
>time, and above all they are used to resync the rtp'ed flux.

FYI, the following FAQ entry might be helpful:
http://www.live.com/liveMedia/faq.html#separate-rtp-streams


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Wed, 04 Jun 2003 10:52:14 -0700
Date: Wed, 04 Jun 2003 10:52:14 -0700


>I saw in testMPEGVideoStreamer "MediaSource" & "MPEGVideoStreamFramer". 
>This read a MPEG 1 or 2 file and streams via "MPEGVideoRTPSink".
>But how can I stream MPEG4 or H263(+) file ? I've seen 
>"H263plusVideoRTPSink " for example, how can I use it to stream my file ?

To stream H.263+, you would need to write a "H263plusVideoStreamFramer" 
filter that parses the input stream into discrete frames, for packaging in 
outgoing RTP packets (by "H263plusVideoRTPSink ").

To stream MPEG-4 video, you would need (i) a "MPEG4VideoStreamFramer" 
filter (as above), and also (ii) a "MPEG4VideoRTPSink".

(Someday these new classes might be added to the library, but they haven't 
yet been funded.)


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Subject: Re: [Live-devel] Interoperability with live.com library (RTP)
From: Tym Altman <EMAIL-ADDRESS-HIDDEN>
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X-Original-Date: Wed, 04 Jun 2003 11:18:49 -0700
Date: Wed, 04 Jun 2003 11:18:49 -0700

Hi,

Thank you Ross and Vincent for replying to my questions. I'm not using RTCP
at all. I was under the impression that it wasn't needed if I'm just doing a
single MPEG1 Video Elementary stream, and don't need to synchronize it with
any other streams. Am I incorrect in this assumption?

Thanks,

Tym Altman



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Subject: [Live-devel] LIVE Make Error Message
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X-Original-Date: Wed, 4 Jun 2003 12:40:38 -0700
Date: Wed, 4 Jun 2003 12:40:38 -0700

This is a multi-part message in MIME format.

------=_NextPart_000_047B_01C32A96.802B7AE0
Content-Type: text/plain;
	charset="Windows-1252"
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I need to use "gcc3" to compile since MPlayer chooses "gcc3".=20
Does LIVE code compatible with "gcc3"?=20
Thanks!

Error Messag:=20
gcc3 -o testMP3Streamer -L. testMP3Streamer.o =
../liveMedia/libliveMedia.a ../gro
upsock/libgroupsock.a ../UsageEnvironment/libUsageEnvironment.a =
../BasicUsageEnv
ironment/libBasicUsageEnvironment.a
testMP3Streamer.o: In function `play()':
testMP3Streamer.o(.text+0xd3): undefined reference to `operator =
new(unsigned)'
testMP3Streamer.o(.text+0x121): undefined reference to `operator =
delete(void*)'
testMP3Streamer.o(.text+0x13f): undefined reference to `operator =
delete(void*)'
testMP3Streamer.o(.text+0x14c): undefined reference to `operator =
new(unsigned)'
testMP3Streamer.o(.text+0x199): undefined reference to `operator =
delete(void*)'
testMP3Streamer.o(.text+0x1b7): undefined reference to `operator =
delete(void*)'
testMP3Streamer.o: In function `__FRAME_BEGIN__':
testMP3Streamer.o(.eh_frame+0x12): undefined reference to =
`__gxx_personality_v0'
../liveMedia/libliveMedia.a: In function `ourMedia(UsageEnvironment&)':
../liveMedia/libliveMedia.a(.text+0x18): undefined reference to =
`operator new(un
signed)'

config.linux
COMPILE_OPTS =3D          $(INCLUDES) -I. -O -DSOCKLEN_T=3Dsocklen_t =
-I/usr/lib/gcc-
lib/i386-redhat-linux/3.0.2
C =3D                     c
C_COMPILER =3D            gcc3 #cc
C_FLAGS =3D               $(COMPILE_OPTS)
CPP =3D                   cpp
CPLUSPLUS_COMPILER =3D    gcc3 #c++
CPLUSPLUS_FLAGS =3D       $(COMPILE_OPTS) -Wall -DBSD=3D1
OBJ =3D                   o
LINK =3D                  gcc3 -o #c++ -o
LINK_OPTS =3D             -L.
CONSOLE_LINK_OPTS =3D     $(LINK_OPTS)
LIBRARY_LINK =3D          ld -o
LIBRARY_LINK_OPTS =3D     $(LINK_OPTS) -r -Bstatic
LIB_SUFFIX =3D                    a
LIBS_FOR_CONSOLE_APPLICATION =3D
LIBS_FOR_GUI_APPLICATION =3D
EXE =3D


------=_NextPart_000_047B_01C32A96.802B7AE0
Content-Type: text/html;
	charset="Windows-1252"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dwindows-1252">
<META content=3D"MSHTML 6.00.2800.1106" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>I need to use "gcc3" to compile since =
MPlayer=20
chooses "gcc3". </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Does LIVE code compatible with "gcc3"?=20
</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Thanks!</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Error Messag: </FONT></DIV>
<DIV style=3D"FONT: 10pt arial">gcc3 -o testMP3Streamer -L. =
testMP3Streamer.o=20
../liveMedia/libliveMedia.a ../gro<BR>upsock/libgroupsock.a=20
../UsageEnvironment/libUsageEnvironment.a=20
../BasicUsageEnv<BR>ironment/libBasicUsageEnvironment.a<BR>testMP3Streame=
r.o: In=20
function `play()':<BR>testMP3Streamer.o(.text+0xd3): undefined reference =
to=20
`operator new(unsigned)'<BR>testMP3Streamer.o(.text+0x121): undefined =
reference=20
to `operator delete(void*)'<BR>testMP3Streamer.o(.text+0x13f): undefined =

reference to `operator delete(void*)'<BR>testMP3Streamer.o(.text+0x14c): =

undefined reference to `operator=20
new(unsigned)'<BR>testMP3Streamer.o(.text+0x199): undefined reference to =

`operator delete(void*)'<BR>testMP3Streamer.o(.text+0x1b7): undefined =
reference=20
to `operator delete(void*)'<BR>testMP3Streamer.o: In function=20
`__FRAME_BEGIN__':<BR>testMP3Streamer.o(.eh_frame+0x12): undefined =
reference to=20
`__gxx_personality_v0'<BR>../liveMedia/libliveMedia.a: In function=20
`ourMedia(UsageEnvironment&amp;)':<BR>../liveMedia/libliveMedia.a(.text+0=
x18):=20
undefined reference to `operator new(un<BR>signed)'</DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>config.linux</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>COMPILE_OPTS=20
=3D&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; $(INCLUDES) =
-I. -O=20
-DSOCKLEN_T=3Dsocklen_t =
-I/usr/lib/gcc-<BR>lib/i386-redhat-linux/3.0.2<BR>C=20
=3D&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
c<BR>C_COMPILER=20
=3D&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
gcc3=20
#cc<BR>C_FLAGS=20
=3D&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;=20
$(COMPILE_OPTS)<BR>CPP=20
=3D&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
cpp<BR>CPLUSPLUS_COMPILER =3D&nbsp;&nbsp;&nbsp; gcc3 =
#c++<BR>CPLUSPLUS_FLAGS=20
=3D&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; $(COMPILE_OPTS) -Wall =
-DBSD=3D1<BR>OBJ=20
=3D&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
o<BR>LINK=20
=3D&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
gcc3 -o #c++ -o<BR>LINK_OPTS=20
=3D&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;=20
-L.<BR>CONSOLE_LINK_OPTS =3D&nbsp;&nbsp;&nbsp;&nbsp; =
$(LINK_OPTS)<BR>LIBRARY_LINK=20
=3D&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; ld=20
-o<BR>LIBRARY_LINK_OPTS =3D&nbsp;&nbsp;&nbsp;&nbsp; $(LINK_OPTS) -r=20
-Bstatic<BR>LIB_SUFFIX=20
=3D&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
a<BR>LIBS_FOR_CONSOLE_APPLICATION =3D<BR>LIBS_FOR_GUI_APPLICATION =
=3D<BR>EXE=20
=3D<BR></FONT></DIV>
<DIV style=3D"FONT: 10pt arial">&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2></DIV></FONT></BODY></HTML>

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From <EMAIL-ADDRESS-HIDDEN> Wed Jun 4 13:18:22 2003
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X-Original-Date: Wed, 04 Jun 2003 13:08:17 -0700
Date: Wed, 04 Jun 2003 13:08:17 -0700

At 12:40 PM 6/4/03, you wrote:
>I need to use "gcc3" to compile since MPlayer chooses "gcc3".
>Does LIVE code compatible with "gcc3"?

Yes, but - once again - you need to compile *everything* with the same 
compiler.  That means first running "make clean" in both the "live" and 
"MPlayer*" directories, and then rebuilding both from scratch.

You can't just compile some parts of the code with one compiler, and other 
parts of the code with another.  If you do, you'll end up with linkage 
error messages - like you've been seeing.

This is the last message I'll be sending on this subject....

         Ross.




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Subject: Re: [Live-devel] Interoperability with live.com library (RTP)
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X-Original-Date: Wed, 04 Jun 2003 13:18:35 -0700
Date: Wed, 04 Jun 2003 13:18:35 -0700


>I'm not using RTCP
>at all. I was under the impression that it wasn't needed if I'm just doing a
>single MPEG1 Video Elementary stream, and don't need to synchronize it with
>any other streams. Am I incorrect in this assumption?

The RTP spec (see <http://www.live.com/liveMedia/rtp-spec.txt>, section 6) 
says that implementations "SHOULD" include RTCP.  "SHOULD", though, is not 
"MUST", so you can, at least in principle, omit using RTCP if you're 
streaming a single unicast stream, and don't care about packet loss (etc.) 
reporting.

However, it's conceivable that QT Player doesn't work properly if it 
doesn't get RTCP "Sender Report" packets from the source.  To test this, 
you can modify "testMPEGVideoStreamer" to not send RTCP.  To do this, 
comment out the call to "RTCPInstance::createNew" in 
"testMPEGVideoStreamer.cpp".  Then test whether there's a difference in how 
QT Player plays the stream with, and without, RTCP.

But the higher-level question is: Why did you choose to 'roll your own' 
RTSP/RTP server, rather than just using the implementation that comes with 
the "LIVE.COM Streaming Media" code?  Was this because you wanted to stream 
via unicast (the LIVE.COM RTSP server currently works only for multicast 
streams)?




	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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From <EMAIL-ADDRESS-HIDDEN> Wed Jun 4 14:13:39 2003
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Subject: Re: [Live-devel] Interoperability with live.com library (RTP)
From: Tym Altman <EMAIL-ADDRESS-HIDDEN>
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X-Original-Date: Wed, 04 Jun 2003 14:04:09 -0700
Date: Wed, 04 Jun 2003 14:04:09 -0700

on 6/4/03 1:18 PM, Ross Finlayson at finlayson(at)live.com wrote:
> The RTP spec (see <http://www.live.com/liveMedia/rtp-spec.txt>, section 6)
> says that implementations "SHOULD" include RTCP.  "SHOULD", though, is not
> "MUST", so you can, at least in principle, omit using RTCP if you're
> streaming a single unicast stream, and don't care about packet loss (etc.)
> reporting.

Well we care about packet loss, but were under the impression that RTCP
wasn't needed. I'll have to investigate RTCP, which is something I'm not
looking forward to considering I only have another week or so to get this
thing working reliably.

> However, it's conceivable that QT Player doesn't work properly if it
> doesn't get RTCP "Sender Report" packets from the source.  To test this,
> you can modify "testMPEGVideoStreamer" to not send RTCP.  To do this,
> comment out the call to "RTCPInstance::createNew" in
> "testMPEGVideoStreamer.cpp".  Then test whether there's a difference in how
> QT Player plays the stream with, and without, RTCP.

OK. Thanks. I'll try that. I've also posted to the QT API list.
 
> But the higher-level question is: Why did you choose to 'roll your own'
> RTSP/RTP server, rather than just using the implementation that comes with
> the "LIVE.COM Streaming Media" code?  Was this because you wanted to stream
> via unicast (the LIVE.COM RTSP server currently works only for multicast
> streams)?

Because when we started we were unaware of your library. Most of the
RTP/RTSP code we had written about 2 years ago, and then we didn't do much
with it, until recent developments led us to resurrect that code base.
Originally we were streaming MPEG-1 system streams, and recently we wanted
to add ES support, so I dug deeper into the code which was originally
written by someone else. Only when I started having problems, and did
searches on the internet, did I find your library. I was actually looking
for RTP/RTSP client players that I could test with in addition to QuickTime.
Trying out mplayer led me to your website.

The fact that we need unicast support is also a reason, although maybe we
would have worked around it if we were working from scratch.

If I was given more time perhaps I'd look at how I could integrate with the
library.

Thanks for all your help.

Tym



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From <EMAIL-ADDRESS-HIDDEN> Wed Jun 4 14:14:43 2003
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Subject: Re: [Live-devel] Interoperability with live.com library (RTP)
From: Tym Altman <EMAIL-ADDRESS-HIDDEN>
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X-Original-Date: Wed, 04 Jun 2003 14:11:52 -0700
Date: Wed, 04 Jun 2003 14:11:52 -0700

on 6/4/03 1:18 PM, Ross Finlayson at finlayson(at)live.com wrote:
> The RTP spec (see <http://www.live.com/liveMedia/rtp-spec.txt>, section 6)
> says that implementations "SHOULD" include RTCP.  "SHOULD", though, is not
> "MUST", so you can, at least in principle, omit using RTCP if you're
> streaming a single unicast stream, and don't care about packet loss (etc.)
> reporting.

Well we care about packet loss, but were under the impression that RTCP
wasn't needed. I'll have to investigate RTCP, which is something I'm not
looking forward to considering I only have another week or so to get this
thing working reliably.

> However, it's conceivable that QT Player doesn't work properly if it
> doesn't get RTCP "Sender Report" packets from the source.  To test this,
> you can modify "testMPEGVideoStreamer" to not send RTCP.  To do this,
> comment out the call to "RTCPInstance::createNew" in
> "testMPEGVideoStreamer.cpp".  Then test whether there's a difference in how
> QT Player plays the stream with, and without, RTCP.

OK. Thanks. I'll try that. I've also posted to the QT API list.
 
> But the higher-level question is: Why did you choose to 'roll your own'
> RTSP/RTP server, rather than just using the implementation that comes with
> the "LIVE.COM Streaming Media" code?  Was this because you wanted to stream
> via unicast (the LIVE.COM RTSP server currently works only for multicast
> streams)?

Because when we started we were unaware of your library. Most of the
RTP/RTSP code we had written about 2 years ago, and then we didn't do much
with it, until recent developments led us to resurrect that code base.
Originally we were streaming MPEG-1 system streams, and recently we wanted
to add ES support, so I dug deeper into the code which was originally
written by someone else. Only when I started having problems, and did
searches on the internet, did I find your library. I was actually looking
for RTP/RTSP client players that I could test with in addition to QuickTime.
Trying out mplayer led me to your website.

The fact that we need unicast support is also a reason, although maybe we
would have worked around it if we were working from scratch.

If I was given more time perhaps I'd look at how I could integrate with the
library.

Thanks for all your help.

Tym



-------------------------------------------------------
This SF.net email is sponsored by:  Etnus, makers of TotalView, The best
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you've never dreamed of, try TotalView 6 free at www.etnus.com.
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Subject: Re: [Live-devel] Interoperability with live.com library (RTP)
From: Tym Altman <EMAIL-ADDRESS-HIDDEN>
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X-Original-Date: Wed, 04 Jun 2003 14:50:32 -0700
Date: Wed, 04 Jun 2003 14:50:32 -0700

on 6/4/03 1:18 PM, Ross Finlayson at finlayson(at)live.com wrote:
> However, it's conceivable that QT Player doesn't work properly if it
> doesn't get RTCP "Sender Report" packets from the source.  To test this,
> you can modify "testMPEGVideoStreamer" to not send RTCP.  To do this,
> comment out the call to "RTCPInstance::createNew" in
> "testMPEGVideoStreamer.cpp".  Then test whether there's a difference in how
> QT Player plays the stream with, and without, RTCP.

I've done this, as well as enabled the RTSP server, and QT likes it. It
doesn't exhibit any of the problems that exhibits with my stream. So it
looks like it's not the RTCP or lack there of. This is a bit of a relief :)
The only other thing I can think of is perhaps it's a difference between
unicast/multicast, or something I'm not sending in the SDP.

Thanks,

Tym Altman



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Subject: Re: [Live-devel] Interoperability with live.com library (RTP)
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X-Original-Date: Wed, 04 Jun 2003 19:50:11 -0700
Date: Wed, 04 Jun 2003 19:50:11 -0700

I've been playing around with the SDP I'm sending and looks like I found
something that causes the problem with QuickTime Player to go away.

I noticed that the live media library sends a=control:track1, instead of
a=control:trackID=1, like I've seen everywhere else, so I started playing
around with that part.

I discovered that if I send a=control:trackID=1 and then in the PLAY
response 'RTP-Info: url=track1;seq=0 (instead of url=trackID=1) then it
works. If both of them have trackID=1 then it has this strange behavior. So
it looks like it's something to do with the RTP-Info and perhaps the seq=0
part, and the fact that the part after control: and url= were the same. I
noticed that live media doesn't even send the RTP-Info. I'll have to
investigate this tag more. I don't know why it would cause those strange
symptoms though.

Incidentally, these changes have no impact on mplayer & live media dropping
packets, although I got one response on the QuickTime list saying that I
MUST implement RTCP in order for the QT Player to behave properly. So
perhaps that's what I also need to do to get mplayer & live media to work.

Regards,

Tym Altman



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X-Original-Date: Thu, 05 Jun 2003 18:24:35 +0800
Date: Thu, 05 Jun 2003 18:24:35 +0800


has anyone succedded in building live.com applications on ARM?  i've 
cross-compiled the libraries using arm-linux toolchain and downloaded 
the test applications to iPAQ running opie.  I tried to run 
testMPEGVideoReceiver and playSIP but get Segmentation fault

mildred




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Dear Ross,
    I compiled testMPEGAudioVideoStreamer with rtsp support, and use quicktime mplayer rtsp:// and mplayer sdp:// to play mpeg-1 streams sent by testMPEGAudioVideoStreamer , mplayer complains about the slow stream speed and stops. 
    Would you give me some help?

Regards,
Michael
ÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÓ†+ùÞµéšŠX¬²š'²ŠÞu¼ÿÙî³ùš‘êì¡ôèµ©U‰ìÿN›zËm†·šu×›º®‰í…êejw­ü7¬Š	Þw­†Øky§]y» ‚)à}æ­º·¬Ê‹ÿ½éÞ½êÝ­æ¦yÚþÚòN‹Z•XžÃ§ëyæ­ÃÿzÙî³÷(›ÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿü¸¯{÷^½éfj)bž	b²Òâ½ïÝz÷¥þX¬¶Ïì¢êÜyú+ïçzØm¶›ÿÿùb²Ûÿ²‹«qçè®ÿë–+-³ùb²Ø§~åŠ÷ÿuëÞ


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X-Original-Date: Thu, 05 Jun 2003 20:36:44 +0800
Date: Thu, 05 Jun 2003 20:36:44 +0800


how hard is it to add those classes to the liveMedia library?  will it 
take long to program them?  what do i need to know?
thanks

mildred

Ross Finlayson wrote:

>
>> I saw in testMPEGVideoStreamer "MediaSource" & 
>> "MPEGVideoStreamFramer". This read a MPEG 1 or 2 file and streams via 
>> "MPEGVideoRTPSink".
>> But how can I stream MPEG4 or H263(+) file ? I've seen 
>> "H263plusVideoRTPSink " for example, how can I use it to stream my 
>> file ?
>
>
> To stream H.263+, you would need to write a 
> "H263plusVideoStreamFramer" filter that parses the input stream into 
> discrete frames, for packaging in outgoing RTP packets (by 
> "H263plusVideoRTPSink ").
>
> To stream MPEG-4 video, you would need (i) a "MPEG4VideoStreamFramer" 
> filter (as above), and also (ii) a "MPEG4VideoRTPSink".
>
> (Someday these new classes might be added to the library, but they 
> haven't yet been funded.)
>
>
>     Ross Finlayson
>     LIVE.COM
>     <http://www.live.com/>
>
>
>
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X-Original-Date: Thu, 05 Jun 2003 08:30:23 -0700
Date: Thu, 05 Jun 2003 08:30:23 -0700


>has anyone succedded in building live.com applications on ARM?

Yes, several people have done this.

>  i've cross-compiled the libraries using arm-linux toolchain and 
> downloaded the test applications to iPAQ running opie.  I tried to run 
> testMPEGVideoReceiver and playSIP but get Segmentation fault

Remember, You Have Complete Source Code.  Should you find a bug, or some 
other problem that makes portability difficult, please let us know.



	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Thu, 05 Jun 2003 08:45:02 -0700
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>    I compiled testMPEGAudioVideoStreamer with rtsp support, and use 
> quicktime mplayer rtsp:// and mplayer sdp:// to play mpeg-1 streams sent 
> by testMPEGAudioVideoStreamer , mplayer complains about the slow stream 
> speed and stops.

That's odd.  Just to be clear: Did you have problems playing the stream 
using *both* QuickTime Player and MPlayer, or just with MPlayer?

MPlayer's warning message "Your system is too slow to play this" can't 
always be taken seriously, especially when playing RTP streams.  In this 
case, it sometimes really means "Your network is too slow to play this", or 
"I'm too anal retentive".

Are you sure that you're not trying to receive/play your stream over a 
network that's too low-bandwidth for the stream - e.g., that you're not 
trying to stream a high-bandwidth MPEG-2 movie over an 802.11b wireless LAN?

Also, I assume that you used the "latest CVS snapshot" version of the 
MPlayer source code, and not the 0.90 version (which won't work properly).


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Thu, 05 Jun 2003 09:00:18 -0700
Date: Thu, 05 Jun 2003 09:00:18 -0700


>how hard is it to add those classes to the liveMedia library?  will it 
>take long to program them?  what do i need to know?

An "MPEG4ESVideoRTPSink" class would be very easy to do.  In fact, you 
could just use the existing "SimpleRTPSink" class (with 
"rtpTimestampFrequency" = 90000, "sdpMediaTypeString" = "video", 
"rtpPayloadFormatName" = "MP4V-ES"), so you wouldn't need to write any new 
code for this.

"H263plusVideoStreamFramer" and "MPEG4ESVideoStreamFramer" would be more 
difficult to do.  You could use the existing "MPEGVideoStreamFramer" code 
as a guideline, but it would be non-trivial.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Subject: [Live-devel] very nice of you
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X-Original-Date: Thu, 5 Jun 2003 16:56:58 -0700
Date: Thu, 5 Jun 2003 16:56:58 -0700

You might think MPlayer is not friendly that they do not support gcc 2.96.
At least they tell you so and check what version is available to continue
the process. The installation is quite a pleassant process.
It's very nice of you that you wrote this library for MPlayer and answered
my question. Unfortunately I am still in darkness and not able to use it.
frustrating and suffering.
I followed your instruction from your web site and your reply. I asked for
help from my fellows. I instructed config.linux to complile everything with
the same version MPlayer used ( I copied the whole MPlayer compliling
process (without enable LIVE) with "nohup" and found out that everything is
compliled with gcc3) and it just won't work for LIVE compiling. It jumps to
"cc" instead even though the "Makefile" says to use "gcc3". Of cause I run
"make clean" whenever I start a new try. If you specifically wrote the
program for MPlayer, why it's so hard to compile into MPlayer?
You do not have to reply like you declared in the previous reply. just want
to let you know how I feel to use your program.



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> >    I compiled testMPEGAudioVideoStreamer with rtsp support, and use 
> > quicktime mplayer rtsp:// and mplayer sdp:// to play mpeg-1 streams sent 
> > by testMPEGAudioVideoStreamer , mplayer complains about the slow stream 
> > speed and stops.
> 
> That's odd.  Just to be clear: Did you have problems playing the stream 
> using *both* QuickTime Player and MPlayer, or just with MPlayer?

I do all the test in LAN which has 10M bandwidth. QT player sometimes succeed, but most time it failed with "10061: Connection Failed". MPlayer always play the beginning several second length of mpeg-1 movie,  then it stopped and  said something like "slow speed streaming" . I tried mplayer -dropframe, but the result is the same.
 
> MPlayer's warning message "Your system is too slow to play this" can't 
> always be taken seriously, especially when playing RTP streams.  In this 
> case, it sometimes really means "Your network is too slow to play this", or 
> "I'm too anal retentive".
> 
> Are you sure that you're not trying to receive/play your stream over a 
> network that's too low-bandwidth for the stream - e.g., that you're not 
> trying to stream a high-bandwidth MPEG-2 movie over an 802.11b wireless LAN?

No. Though i would likely to do that if i can:-)

> Also, I assume that you used the "latest CVS snapshot" version of the 
> MPlayer source code, and not the 0.90 version (which won't work properly).

I do use the "latest CVS snapshot" version .
ÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÓ†+ùÞµéšŠX¬²š'²ŠÞu¼ÿÙî³ùš‘êì¡ôèµ©U‰ìÿN›zËm†·šu×›º®‰í…êejw­ü7¬Š	Þw­†Øky§]y» ‚)à}æ­º·¬Ê‹ÿ½éÞ½êÝ­æ¦yÚþÚòN‹Z•XžÃ§ëyæ­ÃÿzÙî³÷(›ÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿü¸¯{÷^½éfj)bž	b²Òâ½ïÝz÷¥þX¬¶Ïì¢êÜyú+ïçzØm¶›ÿÿùb²Ûÿ²‹«qçè®ÿë–+-³ùb²Ø§~åŠ÷ÿuëÞ


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X-Original-Date: Fri, 06 Jun 2003 12:44:39 +1000
Date: Fri, 06 Jun 2003 12:44:39 +1000

Wendy Tan wrote:

> I need to use "gcc3" to compile since MPlayer chooses "gcc3".

I think you are getting confused between C compilers and C++ compilers. 
GCC's C++ compiler is g++ (or c++), not gcc (or cc).

 > config.linux
 > ...
 > C_COMPILER =            gcc3 #cc
 > ...
 > CPLUSPLUS_COMPILER =    gcc3 #c++
 > ...
 > LINK =                  gcc3 -o #c++ -o

So in your case you should be using g++3 in CPLUSPLUS_COMPILER and LINK.

Doug.




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X-Original-Date: Thu, 05 Jun 2003 20:46:51 -0700
Date: Thu, 05 Jun 2003 20:46:51 -0700


>QT player sometimes succeed, but most time it failed with "10061: 
>Connection Failed".

Make sure you are using version 2003.05.29 or later of the LIVE.COM 
code.  (That version fixed a bug in the RTSP server that might have caused 
this.)

>  MPlayer always play the beginning several second length of mpeg-1 
> movie,  then it stopped and  said something like "slow speed streaming" . 
> I tried mplayer -dropframe, but the result is the same.

This is strange.  Could you put your "test.mpg" input file online, so I can 
take a look at it, and see if I can reproduce the problem that you're 
seeing?  (Please *don't* send the file by email, though.)


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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> >QT player sometimes succeed, but most time it failed with "10061: 
> >Connection Failed".
> 
> Make sure you are using version 2003.05.29 or later of the LIVE.COM 
> code.  (That version fixed a bug in the RTSP server that might have caused 
> this.)

The version I used is 2003.05.16. I will upgrade it to 2003.05.31 and test it again.

Thanks,

Michael
ÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÓ†+ùÞµéšŠX¬²š'²ŠÞu¼ÿÙî³ùš‘êì¡ôèµ©U‰ìÿN›zËm†·šu×›º®‰í…êejw­ü7¬Š	Þw­†Øky§]y» ‚)à}æ­º·¬Ê‹ÿ½éÞ½êÝ­æ¦yÚþÚòN‹Z•XžÃ§ëyæ­ÃÿzÙî³÷(›ÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿü¸¯{÷^½éfj)bž	b²Òâ½ïÝz÷¥þX¬¶Ïì¢êÜyú+ïçzØm¶›ÿÿùb²Ûÿ²‹«qçè®ÿë–+-³ùb²Ø§~åŠ÷ÿuëÞ


From <EMAIL-ADDRESS-HIDDEN> Tue Jun 10 01:37:04 2003
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Subject: RE: [Live-devel] MPEG4/H263 streaming
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should be very simple:
for Mpeg4:

split content before every 0000
Actually (000001) for Startcodes and (0000) for resync markers




> -----Original Message-----
> From: Mildred L. Frisco [mailto:mildredf(at)asti.dost.gov.ph]
> Sent: donderdag 5 juni 2003 14:37
> To: live-devel(at)lists.sourceforge.net
> Subject: Re: [Live-devel] MPEG4/H263 streaming
> 
> 
> 
> how hard is it to add those classes to the liveMedia library? 
>  will it 
> take long to program them?  what do i need to know?
> thanks
> 
> mildred
> 
> Ross Finlayson wrote:
> 
> >
> >> I saw in testMPEGVideoStreamer "MediaSource" & 
> >> "MPEGVideoStreamFramer". This read a MPEG 1 or 2 file and 
> streams via 
> >> "MPEGVideoRTPSink".
> >> But how can I stream MPEG4 or H263(+) file ? I've seen 
> >> "H263plusVideoRTPSink " for example, how can I use it to stream my 
> >> file ?
> >
> >
> > To stream H.263+, you would need to write a 
> > "H263plusVideoStreamFramer" filter that parses the input 
> stream into 
> > discrete frames, for packaging in outgoing RTP packets (by 
> > "H263plusVideoRTPSink ").
> >
> > To stream MPEG-4 video, you would need (i) a 
> "MPEG4VideoStreamFramer" 
> > filter (as above), and also (ii) a "MPEG4VideoRTPSink".
> >
> > (Someday these new classes might be added to the library, but they 
> > haven't yet been funded.)
> >
> >
> >     Ross Finlayson
> >     LIVE.COM
> >     <http://www.live.com/>
> >
> >
> >
> > -------------------------------------------------------
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> 
> 
> 
> 
> 
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This is a multi-part message in MIME format.
--------------030709070704080808010707
Content-Type: text/plain; charset=us-ascii; format=flowed
Content-Transfer-Encoding: 7bit


Hi!  I compiled the libraries in Red Hat Linux 7.3.  I tried to test the
programs  testMPEGVideoStreamer/Receiver in two different PCs, with
test.mpg a 6 MB mpeg file.  I also tried outputting to file the received
video instead of standard output


pc1 $  ./testMPEGVideoStreamer

MPEGVideoStreamParser::parseSlice(): saw unexpected code : 0x000001e0
MPEGVideoStreamParser::parseSlice(): saw unexpected code : 0x000001e0
MPEGVideoStreamParser::parseSlice(): saw unexpected code : 0x000001e0
....


on the other end
pc2 $ ./testMPEGVideoReceiver > video.mpg


when I tried to run video.mpg in MPlayer, it said that the file format
is not recognized/supported (i attached the log file here)

or am i missing something?  do i correctly use the said test programs?


thanks,
mildred

-- 

Mildred L. Frisco
Science Research Specialist I
Advanced Science and Technology Institute
UP TechnoPark Complex,C.P.Garcia Avenue, 
Diliman, Quezon City 1101
Philippines
Tel. No. 63 2 426 9764
www.asti.dost.gov.ph



--------------030709070704080808010707
Content-Type: text/plain;
 name="video.log"
Content-Transfer-Encoding: 7bit
Content-Disposition: inline;
 filename="video.log"

Using GNU internationalization
Original domain: messages
Original dirname: /usr/share/locale
Current domain: mplayer
Current dirname: /usr/share/locale


MPlayer 0.90-RPM-3.1 (C) 2000-2003 Arpad Gereoffy (see DOCS)

CPU: Intel Pentium III Katmai/Pentium III Xeon Tanner (Family: 6, Stepping: 3)
Detected cache-line size is 32 bytes
CPUflags:  MMX: 1 MMX2: 1 3DNow: 0 3DNow2: 0 SSE: 1 SSE2: 0
Compiled with Runtime CPU Detection - WARNING - this is not optimal!
To get best performance, recompile MPlayer with --disable-runtime-cpudetection
Reading config file /etc/mplayer/mplayer.conf
Reading config file /home/mildred/.mplayer/config
Reading /home/mildred/.mplayer/codecs.conf: can't open '/home/mildred/.mplayer/codecs.conf': No such file or directory
Reading /etc/mplayer/codecs.conf: 50 audio & 136 video codecs
Linux RTC init error in ioctl (rtc_irqp_set 1024): Permission denied
font: can't open file: /home/mildred/.mplayer/font/font.desc
font: can't open file: /usr/share/mplayer/font/font.desc
Try adding "echo 1024 > /proc/sys/dev/rtc/max-user-freq" to your system startup scripts.
Using usleep() timing
Can't open input config file /home/mildred/.mplayer/input.conf : No such file or directory
Input config file /etc/mplayer/input.conf parsed : 52 binds

Playing video.mpg

Cache fill:  0.00% (0 bytes)    
Cache fill: 12.91% (1082749 bytes)    Falling back on trying to parse playlist video.mpg...
============ Sorry, this file format is not recognized/supported =============
=== If this file is an AVI, ASF or MPEG stream, please contact the author! ===


Exiting... (End of file)

--------------030709070704080808010707--



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Subject: Re: [Live-devel] help in running testMPEGVideoStreamer/Receiver
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X-Original-Date: Tue, 10 Jun 2003 08:35:59 -0700
Date: Tue, 10 Jun 2003 08:35:59 -0700

At 02:05 AM 6/10/03, you wrote:

>Hi!  I compiled the libraries in Red Hat Linux 7.3.  I tried to test the
>programs  testMPEGVideoStreamer/Receiver in two different PCs, with
>test.mpg a 6 MB mpeg file.  I also tried outputting to file the received
>video instead of standard output
>
>
>pc1 $  ./testMPEGVideoStreamer
>
>MPEGVideoStreamParser::parseSlice(): saw unexpected code : 0x000001e0
>MPEGVideoStreamParser::parseSlice(): saw unexpected code : 0x000001e0
>MPEGVideoStreamParser::parseSlice(): saw unexpected code : 0x000001e0

I have now added an entry in the FAQ that answers this.  See:
     http://www.live.com/liveMedia/faq.html#mpeg-program-stream


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Date: Wed, 11 Jun 2003 11:54:09 -0400

This is a multi-part message in MIME format.

------=_NextPart_000_0103_01C33010.2AB5E6C0
Content-Type: text/plain;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

Hello, I tried initiating a session from openRTSP from a Linux client to =
a Kasenna server... I am getting a return messege from the server =
"unsupported media type".

Question: can we specify the application/media type (x-rtsp-mh) as an =
example and User Agent parameters?
When I capture the Describe I see that openRTSP is indicating "Accept: =
application/sdp, as well, User-Agent parameters ...=20

When I trace from the Kasenna client-server we see Accept: =
application/x-rtsp-mh and User Agent: QTS/10.....

Thanks in advance !

Steve
------=_NextPart_000_0103_01C33010.2AB5E6C0
Content-Type: text/html;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2800.1141" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>
<P>Hello, I tried initiating a session from openRTSP from a Linux client =
to a=20
Kasenna server... I am getting a return messege from the server =
"unsupported=20
media type".<BR><BR>Question: can we specify the application/media type=20
(x-rtsp-mh) as an example and User Agent parameters?<BR>When I capture =
the=20
Describe I see that openRTSP is indicating "Accept: application/sdp, as =
well,=20
User-Agent parameters ... </P>
<P>When I trace from the Kasenna client-server we see Accept:=20
application/x-rtsp-mh and User Agent: QTS/10.....<BR><BR>Thanks in =
advance=20
!</P></FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Steve</FONT></DIV></BODY></HTML>

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X-Original-Date: Wed, 11 Jun 2003 11:15:37 -0700
Date: Wed, 11 Jun 2003 11:15:37 -0700


>Hello, I tried initiating a session from openRTSP from a Linux client to a 
>Kasenna server... I am getting a return messege from the server 
>"unsupported media type".

Could you re-run "openRTSP" with the "-V" option, and post the output to 
the mailing list, so we can look at what specifically it's complaining about.

>Question: can we specify the application/media type (x-rtsp-mh) as an 
>example and User Agent parameters?
>When I capture the Describe I see that openRTSP is indicating "Accept: 
>application/sdp, as well, User-Agent parameters ...
>
>When I trace from the Kasenna client-server we see Accept: 
>application/x-rtsp-mh and User Agent: QTS/10.....

The "application/x-rtsp-mh" is a red herring.  That string appeared in an 
example in the RTSP specification, but it's meaningless, and not 
needed.  The Kasenna client shouldn't really be including it, but the fact 
that it does is inconsequential.

         Ross.



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X-Original-Date: Wed, 11 Jun 2003 20:05:37 -0400
Date: Wed, 11 Jun 2003 20:05:37 -0400

As requested, please find the included output from the openRTSP client:

# openRTSP -V rtsp://192.168.1.2/TestFile2.3
Sending request: DESCRIBE rtsp://192.168.1.2/TestFile2.3 RTSP/1.0
CSeq: 1
Accept: application/sdp
User-Agent: openRTSP (live.com Streaming Media v2003.05.06)

Received DESCRIBE response: RTSP/1.0 415 Unsupported Media Type
CSeq: 1
Message: Date: Tue, 10 Jun 2003 22:31:52 GMT

Failed to get a SDP description from URL "rtsp://192.168.1.2/TestFile2.3": cannot handle DESCRIBE response: RTSP/1.0 415 Unsupported Media Type


Steve
-----Original Message-----
From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
Date: Wed, 11 Jun 2003 11:15:37 
<EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] openRTSP - MPEG2 streaming


>Hello, I tried initiating a session from openRTSP from a Linux client to a 
>Kasenna server... I am getting a return messege from the server 
>"unsupported media type".

Could you re-run "openRTSP" with the "-V" option, and post the output to 
the mailing list, so we can look at what specifically it's complaining about.

>Question: can we specify the application/media type (x-rtsp-mh) as an 
>example and User Agent parameters?
>When I capture the Describe I see that openRTSP is indicating "Accept: 
>application/sdp, as well, User-Agent parameters ...
>
>When I trace from the Kasenna client-server we see Accept: 
>application/x-rtsp-mh and User Agent: QTS/10.....

The "application/x-rtsp-mh" is a red herring.  That string appeared in an 
example in the RTSP specification, but it's meaningless, and not 
needed.  The Kasenna client shouldn't really be including it, but the fact 
that it does is inconsequential.

         Ross.



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swatt(at)ixiacom.com

Sent from TELUS Blackberry service


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X-Original-Date: Wed, 11 Jun 2003 17:33:43 -0700
Date: Wed, 11 Jun 2003 17:33:43 -0700


># openRTSP -V rtsp://192.168.1.2/TestFile2.3
>Sending request: DESCRIBE rtsp://192.168.1.2/TestFile2.3 RTSP/1.0
>CSeq: 1
>Accept: application/sdp
>User-Agent: openRTSP (live.com Streaming Media v2003.05.06)
>
>Received DESCRIBE response: RTSP/1.0 415 Unsupported Media Type

Is *any* RTSP client able to successfully play that particular URL 
(rtsp://192.168.1.2/TestFile2.3)?  (You noted that you had used a 'Kasenna' 
client, but is it able to successfully play this *same* URL?)

I have seen that error message "Unsupported Media Type" returned by some 
RTSP servers (in particular, the 'Darwin Streaming Server') if the media 
file - stored on the server -  is not in an appropriate form for streaming 
(e.g., it hasn't been 'hinted').  That's why I'm sceptical that *any* RTSP 
client is able to play that URL.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Fri, 13 Jun 2003 15:00:21 +0200
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Hi all,

I'm doing a project in which I need to extend ffmpeg with a
working RTSP protocol. We've chosen to use openRTSP as a basis
for supporting the communication.

Because of this I've started leafing through the code and I
ran into a possible memory leak... (I only started a week ago
don't kill me for this...)

ServerMediaSession maintains a linked list of ServerMediaSubSession
objects. (Each seems to contain a link to the next one through fNext.)
On destruction ServerMediaSession only deletes the fSubsessionsHead,
but the ServerMediaSubSession's destructor DOESN'T  destroy the
next one through 'delete fNext;' for instance.
Doesn't that mean that, when there are more than one subsessions,
only the head of the list is actually deleted and the rest remain
in memory?

Side Question: What does the 'f' convention stand for? A lot of
class variables are called 'fThis' or 'fThat'. I already figured that
it doesn't stand for float since at one time I found 'Boolean
fSomething;'

Thanks for your help,
	Frederik

-- 
- - -- --- ----- -------- ------------- ---------------------
 "Draco Dormiens Nunquam Tittilandus"
 ing. F. Bonte     bonte(at)fel.tno.nl
 TNO e-Business    phone (+31)534802111
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X-Original-Date: Fri, 13 Jun 2003 09:30:52 -0400
Date: Fri, 13 Jun 2003 09:30:52 -0400

Ross, thanks for the replies, my responses embedded.
Steve
----- Original Message -----
From: {Ross Finlayson} <EMAIL-ADDRESS-HIDDEN>
To: <EMAIL-ADDRESS-HIDDEN>
Sent: Wednesday, June 11, 2003 8:33 PM
Subject: Re: [Live-devel] openRTSP - MPEG2 streaming
>
> ># openRTSP -V rtsp://192.168.1.2/TestFile2.3
> >Sending request: DESCRIBE rtsp://192.168.1.2/TestFile2.3 RTSP/1.0
> >CSeq: 1
> >Accept: application/sdp
> >User-Agent: openRTSP (live.com Streaming Media v2003.05.06)
> >
> >Received DESCRIBE response: RTSP/1.0 415 Unsupported Media Type
>
> Is *any* RTSP client able to successfully play that particular URL
> (rtsp://192.168.1.2/TestFile2.3)?  (You noted that you had used a
'Kasenna'
> client, but is it able to successfully play this *same* URL?)
[STEVE]
Yes, we are able to play the asset without a problem using a kassena rtsp
client.  As well, to ensure my end was good, I successfully used the
openRTSP client to request and
receive a few of the sample streams on the net.
Does it matter that the MPEG2 stream is transported via UDP ?

>
> I have seen that error message "Unsupported Media Type" returned by some
> RTSP servers (in particular, the 'Darwin Streaming Server') if the media
> file - stored on the server -  is not in an appropriate form for streaming
> (e.g., it hasn't been 'hinted').  That's why I'm sceptical that *any* RTSP
> client is able to play that URL.
>
>
> Ross Finlayson
> LIVE.COM
> <http://www.live.com/>
>
>
>
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From <EMAIL-ADDRESS-HIDDEN> Fri Jun 13 08:38:09 2003
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X-Original-Date: Fri, 13 Jun 2003 08:30:24 -0700
Date: Fri, 13 Jun 2003 08:30:24 -0700


> > Is *any* RTSP client able to successfully play that particular URL
> > (rtsp://192.168.1.2/TestFile2.3)? (You noted that you had used a
>'Kasenna'
> > client, but is it able to successfully play this *same* URL?)
>[STEVE]
>Yes, we are able to play the asset without a problem using a kassena rtsp
>client.

Interesting.  Are you also able to play the stream using any other RTSP 
clients - in particular QuickTime Player?

Could you please make that server accesible (i.e., using a publically 
accessible "rtsp://" URL), so I can investigate why "openRTSP" is failing 
to play your stream?

>Does it matter that the MPEG2 stream is transported via UDP ?

Not if the UDP packets are RTP.  Are you saying that your server wants to 
use some other protocol (other than RTP/UDP) to send your stream??  I don't 
understand...


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Fri, 13 Jun 2003 08:59:27 -0700
Date: Fri, 13 Jun 2003 08:59:27 -0700


>ServerMediaSession maintains a linked list of ServerMediaSubSession
>objects. (Each seems to contain a link to the next one through fNext.)
>On destruction ServerMediaSession only deletes the fSubsessionsHead,
>but the ServerMediaSubSession's destructor DOESN'T  destroy the
>next one through 'delete fNext;' for instance.
>Doesn't that mean that, when there are more than one subsessions,
>only the head of the list is actually deleted and the rest remain
>in memory?

Yes, you're correct.  Thanks for finding this bug.  (It has now been 
corrected.)

>Side Question: What does the 'f' convention stand for?

It stands for "field".  Giving (most) class member variables names that 
start with "f" is just a convention, used to distinguish the field names 
from member function (or constructor) parameter names.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Hi All,

I've been looking at the code of the RTSPServer and as far as I can
decrypt, it seams only capable of serving one file through multicasting
or unicasting.

Is the RTSPServer able to respond to a request for one of multiple 
content files? Or can it only broadcast one particular file?

Greetings,
	Frederik

-- 
- - -- --- ----- -------- ------------- ---------------------
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>I've been looking at the code of the RTSPServer and as far as I can
>decrypt, it seams only capable of serving one file through multicasting
>or unicasting.

In fact, it's currently even more limited than this - it can serve only a 
single multicast stream.

Note that a RTSP server for a multicast stream is much easier to implement 
(because for multicast, there is just a single stream, sent from the same 
source port to the same destination address & port, regardless of how many 
clients request the stream).

>Is the RTSPServer able to respond to a request for one of multiple
>content files?

This is something that I hope to support someday, but it would be 
non-trivial to do, and has not yet been requested by a paying customer.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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> >I've been looking at the code of the RTSPServer and as far as I can
> >decrypt, it seams only capable of serving one file through multicasting
> >or unicasting.
> In fact, it's currently even more limited than this - it can serve only a
> single multicast stream.
OK, it's the test program that dictates the use of unicast. I see.

> Note that a RTSP server for a multicast stream is much easier to implement
> (because for multicast, there is just a single stream, sent from the same
> source port to the same destination address & port, regardless of how many
> clients request the stream).

> >Is the RTSPServer able to respond to a request for one of multiple
> >content files?
> This is something that I hope to support someday, but it would be
> non-trivial to do, and has not yet been requested by a paying customer.
How much effort do you feel this will take? (General indication)

By the way, I feel a slight problem with this adaptation.
Whatever is streamed shouldn't matter to the RTSPServer, so 
am I correct in assuming that isn't the RTSPServer itself 
that should be adapted to fascilitate this.
In it's purest sense it seems to me that it should only need
to fascilitate the communication, the actions taken on behalf
of that can then be handled by another component that is able
to serve streams. Streams can then contain either MPEG data or
feedback from a measuring device or whatever...

Greetings,
	Frederik

-- 
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> > Note that a RTSP server for a multicast stream is much easier to implement
> > (because for multicast, there is just a single stream, sent from the same
> > source port to the same destination address & port, regardless of how many
> > clients request the stream).
>
> > >Is the RTSPServer able to respond to a request for one of multiple
> > >content files?
> > This is something that I hope to support someday, but it would be
> > non-trivial to do, and has not yet been requested by a paying customer.
>How much effort do you feel this will take? (General indication)

Frederik,

I could probably do this in a week.  If you're interested in providing 
funding for this task, please let me know (by private email).


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Wed, 18 Jun 2003 19:43:20
Date: Wed, 18 Jun 2003 19:43:20


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Hi All,

 My name is Wagner Leonardo , I am a student of Catolica University 
of Peru ...and a try to implement an application with this libraries that i 
download fron \"live.com\" page
Well....I dont have access to a unix machine y only have a Windows Machine 
(win98)...so i follow the instructions that you recommend for this case

-First, unpack and extract the \".tar.gz\" file usin WinZip
....it´s easy :)


-Second,then if necessary , change the \"TOOLS32=\" line in the file \"win32config\"
....the problem stars!
1)when this change is necessary?
2)What kind of change ?

-third, I join together the files to form a new one (the \"*.mak\" file)
...i do this! , but when i tried to compile( openRTSP for example)...so there 
be error messages :
* \"unable to open \'play common.hh\'\" ( but this file exist in this 
directory)....together with this 3 messages of declaration syntax error


well this my first time that i work with this kind of application ..please if 
somenone have a recommendation please tell me! 

*******************************
Wagner Leonardo Sánchez Sánchez
Bach. Ing ELectronica-PUCP
2831517/97190947
sanchez.wl(at)pucp.edu.pe
**********************************
         

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X-Original-Date: Mon, 23 Jun 2003 17:53:59 +0800
Date: Mon, 23 Jun 2003 17:53:59 +0800


Hi!  I have built the liveMedia library and tried to run it on iPAQ. 
 However, I encountered "Segmentation fault" in iPAQ.  I tried to debug 
the program using arm-linux-gdb and gdbserver.  below are the logs for 
arm-linux-gdb and gdbserver(on target)  and also the top-level Makefile 
for liveMedia


------ Makefile ---------------

INCLUDES = -IUsageEnvironment/include -Igroupsock/include 
-ITclUsageEnvironment/include -ImTunnel/include
##### Change the following for your environment:
COMPILE_OPTS =        $(INCLUDES) -I. -O -DSOCKLEN_T=socklen_t -g
C =            c
C_COMPILER =        arm-linux-gcc
C_FLAGS =        $(COMPILE_OPTS)
CPP =            cpp
CPLUSPLUS_COMPILER =    arm-linux-c++
CPLUSPLUS_FLAGS =    $(COMPILE_OPTS) -Wall -DBSD=1
OBJ =            o
LINK =            arm-linux-c++ -o
LINK_OPTS =        -L.
CONSOLE_LINK_OPTS =    $(LINK_OPTS)
LIBRARY_LINK =        arm-linux-ld -o
LIBRARY_LINK_OPTS =    $(LINK_OPTS) -r -Bstatic
LIB_SUFFIX =            a
LIBS_FOR_CONSOLE_APPLICATION =
LIBS_FOR_GUI_APPLICATION =
EXE =
##### End of variables to change

LIVEMEDIA_DIR = liveMedia
LIVEMEDIA_LIB = $(LIVEMEDIA_DIR)/libliveMedia.$(LIB_SUFFIX)
GROUPSOCK_DIR = groupsock
GROUPSOCK_LIB = $(GROUPSOCK_DIR)/libgroupsock.$(LIB_SUFFIX)
USAGE_ENVIRONMENT_DIR = UsageEnvironment
USAGE_ENVIRONMENT_LIB = 
$(USAGE_ENVIRONMENT_DIR)/libUsageEnvironment.$(LIB_SUFFIX)
...
...
...

-----------  on host ----------------
$ arm-linux-gdb testMPEGVideoReceiver
GNU gdb 5.3
Copyright 2002 Free Software Foundation, Inc.
GDB is free software, covered by the GNU General Public License, and you are
welcome to change it and/or distribute copies of it under certain 
conditions.
Type "show copying" to see the conditions.
There is absolutely no warranty for GDB.  Type "show warranty" for details.
This GDB was configured as "--host=i686-pc-linux-gnu --target=arm-linux"...
(gdb) target remote 10.10.4.46:2345
Remote debugging using 10.10.4.46:2345
0x40001c60 in ?? ()
(gdb) load testMOE  PEGVideoReceiver
Loading section .interp, size 0x13 lma 0x20000f4
Loading section .note.ABI-tag, size 0x20 lma 0x2000110
Loading section .hash, size 0x67c lma 0x2000130
Loading section .dynsym, size 0xd80 lma 0x20007ac
Loading section .dynstr, size 0xc13 lma 0x200152c
Loading section .gnu.version, size 0x1b0 lma 0x2002140
Loading section .gnu.version_r, size 0x50 lma 0x20022f0
Loading section .rel.got, size 0x1e8 lma 0x2002340
Loading section .rel.bss, size 0x30 lma 0x2002528
Loading section .rel.plt, size 0x3e0 lma 0x2002558
Loading section .init, size 0x18 lma 0x2002938
Loading section .plt, size 0x7d0 lma 0x2002950
Loading section .text, size 0x44a1c lma 0x2003120
Loading section .fini, size 0x14 lma 0x2047b3c
Loading section .rodata, size 0x4537 lma 0x2047b50
Loading section .data, size 0x7010 lma 0x2054088
Loading section .ctors, size 0x1c lma 0x205b098
Loading section .dtors, size 0xc lma 0x205b0b4
Loading section .got, size 0x3b4 lma 0x205b0c0
Loading section .dynamic, size 0xa8 lma 0x205b474
Start address 0x2003120, load size 341021
Transfer rate: 545633 bits/sec, 158 bytes/write.
(gdb) break main
Breakpoint 1 at 0x20031f0: file testMPEGVideoReceiver.cpp, line 41.
(gdb) run
The program being debugged has been started already.
Start it from the beginning? (y or n) Program not restarted.
(gdb) nexti
0x02003124 in _start ()
(gdb)
0x02003128 in _start ()
(gdb)
0x0200312c in _start ()
(gdb)
0x02003130 in _start ()
(gdb)
0x02003134 in _start ()
(gdb)
0x02003138 in _start ()
(gdb)
0x0200313c in _start ()
(gdb)
0x02003140 in _start ()
(gdb)

Program received signal SIGSEGV, Segmentation fault.
0x00000000 in ?? ()
(gdb)

Program terminated with signal SIGSEGV, Segmentation fault.
The program no longer exists.
(gdb) quit

$


---------------- on iPAQ --------------------

$ gdbserver host:2345 testMPEGVideoReceiver

Process /root/testMPEGVideoReceiver created; pid = 670
Remote debugging from host 10.10.4.52

Child terminated with signal = b

Child terminated with signal = 0xb
GDBserver exiting
$
---------------------------

-- 

Mildred L. Frisco
Science Research Specialist I
Advanced Science and Technology Institute
ASTI Bldg., UP TechnoPark Complex, C. P. Garcia Avenue, 
Diliman, Quezon City 1101
Philippines
Tel. No. 63 2 426 9764
http://www.asti.dost.gov.ph





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X-Original-Date: Mon, 23 Jun 2003 18:09:42 +0200 (CEST)
Date: Mon, 23 Jun 2003 18:09:42 +0200 (CEST)

Hello,

First, I would like to congratulate the developpers of this library for
this great work.

I try to realize a videoconference system for my training. So, I had
decided to use the library FFMPEG/libavcodec for compression/decompression
and LIVE.COM for the transport by RTP/RTCP.

I succeed to implement a client/server on Linux, and it works very well
with H263+ and MPEG4.
But, as the server of the application must be used on Windows, I try to
tranform my client/server for this.

And, unfortunely, I have had a bad surprise : the two libraries work on
Windows but with different compilers,
FFMPEG//libavcodec  works only with MINGW32 while LIVE.COM not.

My first reflex was to try to compile FFMPEG/libavcodec  with MINGW32 and
import the library in Visual C++ 6.0 but it doesn't work : the application
compiles and link well but a mysterious exception appears in the methods
"avcodec_open".

So, I try to use FFMPEG/libavcodec compiled with MINGW32 and LIVE.COM
compiled with CYGWIN.
It doesn't work too : linking incompatibility.

So, I try to compile LIVE.COM with MINGW32/MSYS.

I create the file "config.mingw32" by taking example of the "config.cygwin"

COMPILE_OPTS =  $(INCLUDES) -I. -O -DSOCKLEN_T=int
C =   c
C_COMPILER =  gcc
C_FLAGS =  $(COMPILE_OPTS) -DUSE_OUR_BZERO=1 -D__MINGW32__ -D__WIN32__
CPP =   cpp
CPLUSPLUS_COMPILER = c++
CPLUSPLUS_FLAGS = $(COMPILE_OPTS) -Wall
OBJ =   o
LINK =   c++ -o
LINK_OPTS =  -L. -lws2_32
CONSOLE_LINK_OPTS = $(LINK_OPTS)
LIBRARY_LINK =  ld -o
LIBRARY_LINK_OPTS = $(LINK_OPTS) -r -Bstatic
LIB_SUFFIX =   a
LIBS_FOR_CONSOLE_APPLICATION = -lws2_32
LIBS_FOR_GUI_APPLICATION =
EXE =

And I make little changes :
- In file "GroupSockHelper.cpp", after the line
#if defined(__WIN32__) || defined(_WIN32)
I add
#if defined(__MINGW32__)
#include <winsock2.h>
#include <ws2tcpip.h>
#endif

- In files "GroupSock.cpp" and "NetInterface.cpp", I remplace
#if defined(__WIN32__) || defined(_WIN32)
by
#if not defined(__MINGW32__) && (defined(__WIN32__) || defined(_WIN32))


Then, the library compiles well.
But when I try to execute testMPEGVideoStreamer, it fails with a
mysterious (core dumped).
The problem seems to come of the line :
videoSink->startPlaying(*videoSource, afterPlaying, videoSink);

The problem isn't the file test.mpg I use because it works very well with
a testMPEGVideoStreamer compiled with Visual C++ 6.0.

My application doesn't work too, so I would like know if the problem is
LIVE.COM can't work with MINGW32/MSYS or my changes are bad.

Anybody can help me ?

Thank you to have read this.

The Rolender.

PS : Sorry for my poor english but I am French :-)


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X-Original-Date: Mon, 23 Jun 2003 11:38:05 -0700
Date: Mon, 23 Jun 2003 11:38:05 -0700


>My application doesn't work too, so I would like know if the problem is
>LIVE.COM can't work with MINGW32/MSYS or my changes are bad.

I don't know if anyone else has tried building/running the LIVE.COM code 
with MINGW32, but I don't know of any reason why this would not work.

>Then, the library compiles well.
>But when I try to execute testMPEGVideoStreamer, it fails with a
>mysterious (core dumped).
>The problem seems to come of the line :
>videoSink->startPlaying(*videoSource, afterPlaying, videoSink);

You should check whether "videoSink" or "videoSource" is NULL.  If so, this 
means that the corresponding "createNew()" call failed, for some reason.

In any case, you will need to look into the LIVE.COM code to figure out why 
it's failing.  Remember, you have complete source code.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Tue, 24 Jun 2003 09:43:30 +0200 (CEST)
Date: Tue, 24 Jun 2003 09:43:30 +0200 (CEST)

Hi Ross (and all :D)

  I have compiled live media lib in RH9.0 in a AMD
ATHLON 1.4Ghz with 256Ram DDR without errors (of
course ;)). Then i have plugged a MPEG2 encoder
(sensoray), i have problems trying to open /dev/sxdrvr
(the sensoray board) directly. I have developed a
small program that reads from the board and send all
data to stdout. Afterwards i try to pipe that to
testMPEGAudioVideoStreamer app (modified for stdin
like you tell me!!). When i try to send data with low
bitrates and small picture size to localhost and i
haven't any problems but when i try it with high
bitrates (10Mb) i have several problems the reception
(openRTSP) has frames loss. 

 I thougth that the problem was the small program,
then i record the file (plays with XINE o MPLAYER
perfectly) and send the file, but the result was the
same (with low CPU consumption). The network card is
an Intel 10/100Mb. How could i test the network card
bandwith?

 What could i investigate?

 Best regards.

    Juan Carlos Montes

___________________________________________________
Yahoo! Sorteos  -  http://loteria.yahoo.es
Juega a la Lotería Primitiva sin salir de casa


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X-Original-Date: Tue, 24 Jun 2003 17:16:11 +0800
Date: Tue, 24 Jun 2003 17:16:11 +0800



Hi!  I have built the liveMedia library and tried to run it on iPAQ. 
However, I encountered "Segmentation fault" in iPAQ.  I tried to debug 
the program using arm-linux-gdb and gdbserver.  below are the logs for 
arm-linux-gdb and gdbserver(on target)  and also the top-level Makefile 
for liveMedia


------ Makefile ---------------

INCLUDES = -IUsageEnvironment/include -Igroupsock/include 
-ITclUsageEnvironment/include -ImTunnel/include
##### Change the following for your environment:
COMPILE_OPTS =        $(INCLUDES) -I. -O -DSOCKLEN_T=socklen_t -g
C =            c
C_COMPILER =        arm-linux-gcc
C_FLAGS =        $(COMPILE_OPTS)
CPP =            cpp
CPLUSPLUS_COMPILER =    arm-linux-c++
CPLUSPLUS_FLAGS =    $(COMPILE_OPTS) -Wall -DBSD=1
OBJ =            o
LINK =            arm-linux-c++ -o
LINK_OPTS =        -L.
CONSOLE_LINK_OPTS =    $(LINK_OPTS)
LIBRARY_LINK =        arm-linux-ld -o
LIBRARY_LINK_OPTS =    $(LINK_OPTS) -r -Bstatic
LIB_SUFFIX =            a
LIBS_FOR_CONSOLE_APPLICATION =
LIBS_FOR_GUI_APPLICATION =
EXE =
##### End of variables to change

LIVEMEDIA_DIR = liveMedia
LIVEMEDIA_LIB = $(LIVEMEDIA_DIR)/libliveMedia.$(LIB_SUFFIX)
GROUPSOCK_DIR = groupsock
GROUPSOCK_LIB = $(GROUPSOCK_DIR)/libgroupsock.$(LIB_SUFFIX)
USAGE_ENVIRONMENT_DIR = UsageEnvironment
USAGE_ENVIRONMENT_LIB = 
$(USAGE_ENVIRONMENT_DIR)/libUsageEnvironment.$(LIB_SUFFIX)
...
...
...

-----------  on host ----------------
$ arm-linux-gdb testMPEGVideoReceiver
GNU gdb 5.3
Copyright 2002 Free Software Foundation, Inc.
GDB is free software, covered by the GNU General Public License, and you 
are
welcome to change it and/or distribute copies of it under certain 
conditions.
Type "show copying" to see the conditions.
There is absolutely no warranty for GDB.  Type "show warranty" for details.
This GDB was configured as "--host=i686-pc-linux-gnu --target=arm-linux"...
(gdb) target remote 10.10.4.46:2345
Remote debugging using 10.10.4.46:2345
0x40001c60 in ?? ()
(gdb) load testMOE  PEGVideoReceiver
Loading section .interp, size 0x13 lma 0x20000f4
Loading section .note.ABI-tag, size 0x20 lma 0x2000110
Loading section .hash, size 0x67c lma 0x2000130
Loading section .dynsym, size 0xd80 lma 0x20007ac
Loading section .dynstr, size 0xc13 lma 0x200152c
Loading section .gnu.version, size 0x1b0 lma 0x2002140
Loading section .gnu.version_r, size 0x50 lma 0x20022f0
Loading section .rel.got, size 0x1e8 lma 0x2002340
Loading section .rel.bss, size 0x30 lma 0x2002528
Loading section .rel.plt, size 0x3e0 lma 0x2002558
Loading section .init, size 0x18 lma 0x2002938
Loading section .plt, size 0x7d0 lma 0x2002950
Loading section .text, size 0x44a1c lma 0x2003120
Loading section .fini, size 0x14 lma 0x2047b3c
Loading section .rodata, size 0x4537 lma 0x2047b50
Loading section .data, size 0x7010 lma 0x2054088
Loading section .ctors, size 0x1c lma 0x205b098
Loading section .dtors, size 0xc lma 0x205b0b4
Loading section .got, size 0x3b4 lma 0x205b0c0
Loading section .dynamic, size 0xa8 lma 0x205b474
Start address 0x2003120, load size 341021
Transfer rate: 545633 bits/sec, 158 bytes/write.
(gdb) break main
Breakpoint 1 at 0x20031f0: file testMPEGVideoReceiver.cpp, line 41.
(gdb) run
The program being debugged has been started already.
Start it from the beginning? (y or n) Program not restarted.
(gdb) nexti
0x02003124 in _start ()
(gdb)
0x02003128 in _start ()
(gdb)
0x0200312c in _start ()
(gdb)
0x02003130 in _start ()
(gdb)
0x02003134 in _start ()
(gdb)
0x02003138 in _start ()
(gdb)
0x0200313c in _start ()
(gdb)
0x02003140 in _start ()
(gdb)

Program received signal SIGSEGV, Segmentation fault.
0x00000000 in ?? ()
(gdb)

Program terminated with signal SIGSEGV, Segmentation fault.
The program no longer exists.
(gdb) quit

$


---------------- on iPAQ --------------------

$ gdbserver host:2345 testMPEGVideoReceiver

Process /root/testMPEGVideoReceiver created; pid = 670
Remote debugging from host 10.10.4.52

Child terminated with signal = b

Child terminated with signal = 0xb
GDBserver exiting
$
---------------------------

-- 

Mildred L. Frisco
Science Research Specialist I
Advanced Science and Technology Institute
ASTI Bldg., UP TechnoPark Complex, C. P. Garcia Avenue, Diliman, Quezon 
City 1101
Philippines
Tel. No. 63 2 426 9764
http://www.asti.dost.gov.ph




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X-Original-Date: Tue, 24 Jun 2003 09:33:49 -0700
Date: Tue, 24 Jun 2003 09:33:49 -0700


>  What could i investigate?

I suggest starting with the simplest configuration that demonstrates the 
problem.  In your case, it is streaming from a (pre-recorded) file, and 
receiving using "openRTSP".  Try the following:

1/ Run "openRTSP", with the -Q option, on the same computer that you're 
streaming from.  Do you see any packet loss?

2/ Run "openRTSP", with the -Q option, on a different computer (on the same 
LAN) from the one the streaming from.  Do you see any packet loss in this case?

If you see significant packet loss in case 2/ but not case 1/, then it's 
likely that the data stream is exceeding your LAN's bandwidth.  Make sure 
that your LAN (the interfaces at each end, and the hub(s)) are 100 Mbps, 
rather than 10 Mbps.

Another thing you could try is increasing the UDP read buffer size (look 
for "increaseReadBufferTo" in "liveMedia/MultiFramedRTPSource.cpp"), then 
recompile "liveMedia" and "openRTSP".

If you see significant packet loss even in case 1/ (running sender and 
receiver on the same computer), then you could try increasing the priority 
of both the sender and receiver (openRTSP) applications, using "nice".


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Hi Ross,

    I have use live media to stream mpeg-1/2 and use directShow to play it. I found that the audio and video do not synchronize. The audio seems to be 2 to 3 seconds late. Can you tell me why?

Regards,

Michael
    
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>    I have use live media to stream mpeg-1/2 and use directShow to play 
> it. I found that the audio and video do not synchronize. The audio seems 
> to be 2 to 3 seconds late. Can you tell me why?

Does your client use the "presentationTime" parameter (passed to the 
'afterGettingFunc' of "getNextFrame()") to synchronize audio and video?  If 
not, then it should.  (See 
<http://www.live.com/liveMedia/faq.html#separate-rtp-streams> for more 
information.)


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Hi Ross,

    In fact, I took "testMPEGVideoReceiver.cpp" as an example to receive the audio and video streams. I don't know when "presentationTime" be set and how to pass it to "afterGettingFrame"(right?) and what to do next after passing "presentationTime"?

Many thanks!

Michael
----- Original Message ----- 
From: {Ross Finlayson} <EMAIL-ADDRESS-HIDDEN>
To: <EMAIL-ADDRESS-HIDDEN>
Sent: Wednesday, June 25, 2003 4:00 PM
Subject: Re: [Live-devel] a-v sync


> 
> >    I have use live media to stream mpeg-1/2 and use directShow to play 
> > it. I found that the audio and video do not synchronize. The audio seems 
> > to be 2 to 3 seconds late. Can you tell me why?
> 
> Does your client use the "presentationTime" parameter (passed to the 
> 'afterGettingFunc' of "getNextFrame()") to synchronize audio and video?  If 
> not, then it should.  (See 
> <http://www.live.com/liveMedia/faq.html#separate-rtp-streams> for more 
> information.)
> 
> 
> Ross Finlayson
> LIVE.COM
> <http://www.live.com/>
> 
> 
> 
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From <EMAIL-ADDRESS-HIDDEN> Wed Jun 25 11:04:06 2003
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>    In fact, I took "testMPEGVideoReceiver.cpp" as an example to receive 
> the audio and video streams.

But "testMPEGVideoReceiver" receives video only.  If you added code (e.g., 
from "testMP3Receiver") to also receive a separate audio stream, then that 
will work OK, but you will then need to synchronize the two streams.

>  I don't know when "presentationTime" be set

It is set *automatically* by the "*RTPSource" classes.

>and how to pass it to "afterGettingFrame"(right?)

"presentationTime" is a *parameter* that's passed to the 'afterGetting' 
function.  See "liveMedia/FileSink.cpp" for an example of this.  (In that 
case, the "presentationTime" parameter is not actually used, but, with 
multiple streams, this is what you would use to synchronize them.)


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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From <EMAIL-ADDRESS-HIDDEN> Thu Jun 26 04:44:37 2003
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Subject: [Live-devel] Question about NPT and rtptime of RTP-Info
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X-Original-Date: Thu, 26 Jun 2003 07:39:40 -0400
Date: Thu, 26 Jun 2003 07:39:40 -0400

Greetings,
  I had a quick question about NPT and rtptime mappings. I am currently
using mplayer as the front end to the Live libraries, and I was wanting to
play around with getting seeking working with RTSP streams.  My problem is
that I can not figure out the correlation between NPT and the rtptime of
RTP-Info. E.g. in 12.33 of RFC 2326 it says about rtptime, "..The client
uses this value to calculate the mapping of RTP Time to NPT.."  My overall
question is, if the video has been streaming for some time, and I want to
seek forward or backward a certain number of minutes or seconds, how do I
determine  the NPT value to put in the Range field of the PLAY method ?  I
guess it boils down to, how do you determine your "current" NPT based on the
rtptime so you can add/subtract seconds to go forward or backward ?  Is it
okay just to start a timer at the beginning of the video and use that as the
NPT and add/subtract from that ?  It seems like I would run into lots of
synchronization issues if I did that. Maybe I am just having trouble seeing
the forest because of all the trees, but I am missing something.

Thanks in advance,
Jeff


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X-Original-Date: Thu, 26 Jun 2003 09:26:43 -0700
Date: Thu, 26 Jun 2003 09:26:43 -0700


>   I had a quick question about NPT and rtptime mappings.

Note that it's "NTP" (for Network Time Protocol), not "NPT".

>My overall
>question is, if the video has been streaming for some time, and I want to
>seek forward or backward a certain number of minutes or seconds, how do I
>determine  the NPT value to put in the Range field of the PLAY method ?  I
>guess it boils down to, how do you determine your "current" NPT based on the
>rtptime so you can add/subtract seconds to go forward or backward ?

Your "current " NTP time for a RTP stream is simply the "presentationTime" 
of the most recently-received data - i.e., the parameter that was passed to 
the 'after getting' function named in the call to "getNextFrame()".

(This "presentationTime" will be an accurate NTP time provided that at 
least one RTCP "Sender Report" packet (mapping RTP timestamps to NTP time) 
has already been received. You can check this by calling 
"rtpSource->hasBeenSynchronizedUsingRTCP()".)

Note that, in addition to the NTP "presentationTime", you also know:
- The most recent RTP timestamp: "rtpSource->curPacketRTPTimestamp()"
- The RTP timestamp frequency: "rtpSource->timestampFrequency()"

So, if you wanted to, you could also use these values to map an arbitrary 
NTP time to a RTP timestamp, or vice versa.



	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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From <EMAIL-ADDRESS-HIDDEN> Thu Jun 26 10:13:43 2003
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Subject: RE: [Live-devel] Question about NPT and rtptime of RTP-Info
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X-Original-Date: Thu, 26 Jun 2003 13:08:43 -0400
Date: Thu, 26 Jun 2003 13:08:43 -0400


I think he is talking about the NPT (Normal Play Time) mapping and not NTP.
I believe you need to know the packetization interval to determine the NPT
value you should put in the Range header. This is my understanding of how it
works - 

For codecs that have a constant packetization rate, it should be a
straightforward computation - subtract the rtp timestamp corresponding to
the location you want to seek to from the base rtp timestamp (the timestamp
for which you already know the NPT) and you know how much media is encoded
between those two points. Example G.711 packetization at 20 ms, 8KHz - for
every packet the RTP timestamp goes up by 160. So, for every 160 increment
in RTP the NPT goes up 20ms.

However, for codecs that change the packetization rate you have to keep
track from the beginning of the stream. Quicktime codecs have a way to
change this and it is part of the QT header - you will have to process this
to get the mapping right.

Ross - please correct me if i'm wrong.

Thanks,
nagarjun

-----Original Message-----
From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
Sent: Thursday, June 26, 2003 12:27 PM
To: <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] Question about NPT and rtptime of RTP-Info



>   I had a quick question about NPT and rtptime mappings.

Note that it's "NTP" (for Network Time Protocol), not "NPT".

>My overall
>question is, if the video has been streaming for some time, and I want to
>seek forward or backward a certain number of minutes or seconds, how do I
>determine  the NPT value to put in the Range field of the PLAY method ?  I
>guess it boils down to, how do you determine your "current" NPT based on
the
>rtptime so you can add/subtract seconds to go forward or backward ?

Your "current " NTP time for a RTP stream is simply the "presentationTime" 
of the most recently-received data - i.e., the parameter that was passed to 
the 'after getting' function named in the call to "getNextFrame()".

(This "presentationTime" will be an accurate NTP time provided that at 
least one RTCP "Sender Report" packet (mapping RTP timestamps to NTP time) 
has already been received. You can check this by calling 
"rtpSource->hasBeenSynchronizedUsingRTCP()".)

Note that, in addition to the NTP "presentationTime", you also know:
- The most recent RTP timestamp: "rtpSource->curPacketRTPTimestamp()"
- The RTP timestamp frequency: "rtpSource->timestampFrequency()"

So, if you wanted to, you could also use these values to map an arbitrary 
NTP time to a RTP timestamp, or vice versa.



	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Subject: RE: [Live-devel] Question about NPT and rtptime of RTP-Info
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X-Original-Date: Thu, 26 Jun 2003 13:18:46 -0400
Date: Thu, 26 Jun 2003 13:18:46 -0400

Ross,
  Thanks for your reply.

  But actually, I did mean NPT (Normal Play Time) e.g. taken from RFC 2326 "
the stream absolute position relative to the beginning of the presentation".
It seems I could solve my problem by using the latest bis release of 2326,
but the RTSP server I am using is not compliant with the bis-03 release of
2326.  In the bis release, it says that whenever I send a PAUSE command, the
server SHOULD send back a NPT of where the paused occurred. So in theory, I
could send the PAUSE with no Range parameter, get back a response with a
valid NPT value in the Range parameter and add/subtract from that value and
resend a PLAY using that new NPT.  But, as I mentioned my server does not
send the Range back in the PAUSE response, plus that way of doing it seems
kind of kludgy. So for those reasons I was hoping there was some other way
of getting the current play time.
So given what you said below, it is safe to say that there is no direct
mapping between and RTP timestamp and a NPT (Normal Play Time). If that is
true, and I am probably missing something here, how can a client get the NPT
?

Thanks,
  Jeff

-----Original Message-----
From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
Sent: Thursday, June 26, 2003 12:27 PM
To: <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] Question about NPT and rtptime of RTP-Info



>   I had a quick question about NPT and rtptime mappings.

Note that it's "NTP" (for Network Time Protocol), not "NPT".

>My overall
>question is, if the video has been streaming for some time, and I want to
>seek forward or backward a certain number of minutes or seconds, how do I
>determine  the NPT value to put in the Range field of the PLAY method ?  I
>guess it boils down to, how do you determine your "current" NPT based on
the
>rtptime so you can add/subtract seconds to go forward or backward ?

Your "current " NTP time for a RTP stream is simply the "presentationTime" 
of the most recently-received data - i.e., the parameter that was passed to 
the 'after getting' function named in the call to "getNextFrame()".

(This "presentationTime" will be an accurate NTP time provided that at 
least one RTCP "Sender Report" packet (mapping RTP timestamps to NTP time) 
has already been received. You can check this by calling 
"rtpSource->hasBeenSynchronizedUsingRTCP()".)

Note that, in addition to the NTP "presentationTime", you also know:
- The most recent RTP timestamp: "rtpSource->curPacketRTPTimestamp()"
- The RTP timestamp frequency: "rtpSource->timestampFrequency()"

So, if you wanted to, you could also use these values to map an arbitrary 
NTP time to a RTP timestamp, or vice versa.



	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Thu, 26 Jun 2003 10:30:19 -0700
Date: Thu, 26 Jun 2003 10:30:19 -0700


>   But actually, I did mean NPT (Normal Play Time) e.g. taken from RFC 2326

Sorry, my mistake.

         Ross.



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X-Original-Date: Thu, 26 Jun 2003 13:09:25 -0700
Date: Thu, 26 Jun 2003 13:09:25 -0700


>So given what you said below, it is safe to say that there is no direct
>mapping between and RTP timestamp and a NPT (Normal Play Time). If that is
>true, and I am probably missing something here, how can a client get the NPT?

I would use the "presentationTime".  In general,
         NPT = "presentationTime" of most recently-received packet - 
"presentationTime" of first[*]-received packet

[*] To be precise, this should really be:
         "presentationTime" of the first-received packet for which 
"rtpSource->hasBeenSynchronizedUsingRTCP()" was true,
         minus an adjustment for the delay between the very first-received 
packet and that packet

         Ross.



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X-Original-Date: Fri, 27 Jun 2003 11:38:41 +0800
Date: Fri, 27 Jun 2003 11:38:41 +0800


Hi! I'm using arm-linux-insight to debug the program. I think the error 
lies in the 1st line of main function
BasicTaskScheduler::createNew();

here's additional debug messages:
....
....
Breakpoint 13, main (argc=1, argv=0xbffffe04) at 
testMPEGVideoReceiver.cpp:39

Breakpoint 29, main (argc=34033636, argv=0x1) at 
testMPEGVideoReceiver.cpp:41
BasicTaskScheduler::createNew () at BasicUsageEnvironment.cpp:185
__builtin_new (sz=176) at /usr/src/gcc-2.95.2/gcc/cp/new1.cc:75

Program received signal SIGSEGV, segmentation fault
HandlerDescriptor::HandlerDescriptor (this=0x2075ab0, 
nextHandler=0x2075ab0) at BasicUsageEnvironment.cpp:298

(gdb)

-------
i have another question. why is the program looking for new1.cc of 
gcc-2.95.2? i donot have this directory
is there some setting or configuration of the makefile i needed to do?
please help. i'm really at a loss here.
thanks

mildred


Mildred L. Frisco wrote:

>
>
> Hi! I have built the liveMedia library and tried to run it on iPAQ. 
> However, I encountered "Segmentation fault" in iPAQ. I tried to debug 
> the program using arm-linux-gdb and gdbserver. below are the logs for 
> arm-linux-gdb and gdbserver(on target) and also the top-level Makefile 
> for liveMedia
>
>
> ------ Makefile ---------------
>
> INCLUDES = -IUsageEnvironment/include -Igroupsock/include 
> -ITclUsageEnvironment/include -ImTunnel/include
> ##### Change the following for your environment:
> COMPILE_OPTS = $(INCLUDES) -I. -O -DSOCKLEN_T=socklen_t -g
> C = c
> C_COMPILER = arm-linux-gcc
> C_FLAGS = $(COMPILE_OPTS)
> CPP = cpp
> CPLUSPLUS_COMPILER = arm-linux-c++
> CPLUSPLUS_FLAGS = $(COMPILE_OPTS) -Wall -DBSD=1
> OBJ = o
> LINK = arm-linux-c++ -o
> LINK_OPTS = -L.
> CONSOLE_LINK_OPTS = $(LINK_OPTS)
> LIBRARY_LINK = arm-linux-ld -o
> LIBRARY_LINK_OPTS = $(LINK_OPTS) -r -Bstatic
> LIB_SUFFIX = a
> LIBS_FOR_CONSOLE_APPLICATION =
> LIBS_FOR_GUI_APPLICATION =
> EXE =
> ##### End of variables to change
>
> LIVEMEDIA_DIR = liveMedia
> LIVEMEDIA_LIB = $(LIVEMEDIA_DIR)/libliveMedia.$(LIB_SUFFIX)
> GROUPSOCK_DIR = groupsock
> GROUPSOCK_LIB = $(GROUPSOCK_DIR)/libgroupsock.$(LIB_SUFFIX)
> USAGE_ENVIRONMENT_DIR = UsageEnvironment
> USAGE_ENVIRONMENT_LIB = 
> $(USAGE_ENVIRONMENT_DIR)/libUsageEnvironment.$(LIB_SUFFIX)
> ...
> ...
> ...
>
> ----------- on host ----------------
> $ arm-linux-gdb testMPEGVideoReceiver
> GNU gdb 5.3
> Copyright 2002 Free Software Foundation, Inc.
> GDB is free software, covered by the GNU General Public License, and 
> you are
> welcome to change it and/or distribute copies of it under certain 
> conditions.
> Type "show copying" to see the conditions.
> There is absolutely no warranty for GDB. Type "show warranty" for 
> details.
> This GDB was configured as "--host=i686-pc-linux-gnu 
> --target=arm-linux"...
> (gdb) target remote 10.10.4.46:2345
> Remote debugging using 10.10.4.46:2345
> 0x40001c60 in ?? ()
> (gdb) load testMOE PEGVideoReceiver
> Loading section .interp, size 0x13 lma 0x20000f4
> Loading section .note.ABI-tag, size 0x20 lma 0x2000110
> Loading section .hash, size 0x67c lma 0x2000130
> Loading section .dynsym, size 0xd80 lma 0x20007ac
> Loading section .dynstr, size 0xc13 lma 0x200152c
> Loading section .gnu.version, size 0x1b0 lma 0x2002140
> Loading section .gnu.version_r, size 0x50 lma 0x20022f0
> Loading section .rel.got, size 0x1e8 lma 0x2002340
> Loading section .rel.bss, size 0x30 lma 0x2002528
> Loading section .rel.plt, size 0x3e0 lma 0x2002558
> Loading section .init, size 0x18 lma 0x2002938
> Loading section .plt, size 0x7d0 lma 0x2002950
> Loading section .text, size 0x44a1c lma 0x2003120
> Loading section .fini, size 0x14 lma 0x2047b3c
> Loading section .rodata, size 0x4537 lma 0x2047b50
> Loading section .data, size 0x7010 lma 0x2054088
> Loading section .ctors, size 0x1c lma 0x205b098
> Loading section .dtors, size 0xc lma 0x205b0b4
> Loading section .got, size 0x3b4 lma 0x205b0c0
> Loading section .dynamic, size 0xa8 lma 0x205b474
> Start address 0x2003120, load size 341021
> Transfer rate: 545633 bits/sec, 158 bytes/write.
> (gdb) break main
> Breakpoint 1 at 0x20031f0: file testMPEGVideoReceiver.cpp, line 41.
> (gdb) run
> The program being debugged has been started already.
> Start it from the beginning? (y or n) Program not restarted.
> (gdb) nexti
> 0x02003124 in _start ()
> (gdb)
> 0x02003128 in _start ()
> (gdb)
> 0x0200312c in _start ()
> (gdb)
> 0x02003130 in _start ()
> (gdb)
> 0x02003134 in _start ()
> (gdb)
> 0x02003138 in _start ()
> (gdb)
> 0x0200313c in _start ()
> (gdb)
> 0x02003140 in _start ()
> (gdb)
>
> Program received signal SIGSEGV, Segmentation fault.
> 0x00000000 in ?? ()
> (gdb)
>
> Program terminated with signal SIGSEGV, Segmentation fault.
> The program no longer exists.
> (gdb) quit
>
> $
>
>
> ---------------- on iPAQ --------------------
>
> $ gdbserver host:2345 testMPEGVideoReceiver
>
> Process /root/testMPEGVideoReceiver created; pid = 670
> Remote debugging from host 10.10.4.52
>
> Child terminated with signal = b
>
> Child terminated with signal = 0xb
> GDBserver exiting
> $
> ---------------------------
>


-- 

Mildred L. Frisco
Science Research Specialist I
Advanced Science and Technology Institute
ASTI Bldg., UP TechnoPark Complex, C. P. Garcia Avenue, 
Diliman, Quezon City 1101
Philippines
Tel. No. 63 2 426 9764
http://www.asti.dost.gov.ph






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X-Original-Date: Fri, 27 Jun 2003 13:00:45 -0700
Date: Fri, 27 Jun 2003 13:00:45 -0700

 > nextHandler=0x2075ab0) at BasicUsageEnvironment.cpp:298

"BasicUsageEnvironment.cpp" now has fewer than 298 lines, so it seems that 
you're working with a version of the software that's several weeks 
old.  Since then, the "BasicUsageEnvironment" implementation underwent a 
lot of changes.  Please upgrade to the latest version and try again.

Don't just 'give up' whenever you encounter a "Segmentation fault".  Try to 
find out *why* this is happening.  E.g., it seems that the code is trying 
to dereference a NULL pointer.  Find out what this pointer is, and why it 
is (apparently incorrectly) being set to NULL.  Figure out what code is 
failing to cause this to happen, and why.  Remember, You Have Complete 
Source Code.

	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Sat, 28 Jun 2003 18:35:17 +0200
Date: Sat, 28 Jun 2003 18:35:17 +0200

Hi.

We want to receive a stream which has a payload that is not supported by
liveMedia, so we used "-s" option. 
The problem is that we get called to process a frame every time a packet
is received. 
We had a look at SimpleRTPSource and saw that the marker bit is not used
to set the value of fCurrentPacketCompletesFrame, 
which remains 1 all the time.

fCurrentPacketCompletesFrame could be set for video packets but for
audio...

Thanks,
grig and cezar.


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X-Original-Date: Sat, 28 Jun 2003 14:01:47 -0700
Date: Sat, 28 Jun 2003 14:01:47 -0700


>We had a look at SimpleRTPSource and saw that the marker bit is not used
>to set the value of fCurrentPacketCompletesFrame,
>which remains 1 all the time.
>
>fCurrentPacketCompletesFrame could be set for video packets but for
>audio...

Yes - good idea.  I've made this change now.  (This was already being done 
for "SimpleRTPSink".)


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Subject: [Live-devel] A new library: WindowsAudioInputDevice
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X-Original-Date: Mon, 30 Jun 2003 18:21:36 -0700
Date: Mon, 30 Jun 2003 18:21:36 -0700

FYI, a new library "WindowsAudioInputDevice" has now been added to the 
"LIVE.COM Streaming Media" source code distribution.

This library can optionally be used by Windows applications to read PCM 
audio samples from a input device (such as a microphone).  (These audio 
samples could then be fed to an encoder, for example.)

To use this library, call "AudioInputDevice::createNew()" (see 
"liveMedia/include/AudioInputDevice.hh"), specifying the input port number, 
number of channels (1 or 2) and audio sampling frequency.  To get a list of 
input port names, in order, call "AudioInputDevice::getPortNames()".

(In the future, a similar library for Linux might also be made available.)

	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Subject: [Live-devel] recompose mpeg-1
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Hi All,

    I use liveMedia to streaming mpeg-1 media over the net. The client receives separate audio and video stream. My problem is how to recompose these 2 streams to original mpeg-1 file in client?

Thanks in advance.

Michael
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From <EMAIL-ADDRESS-HIDDEN> Fri Jul 4 03:08:45 2003
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X-Original-Date: Fri, 04 Jul 2003 03:00:39 -0700
Date: Fri, 04 Jul 2003 03:00:39 -0700


>    I use liveMedia to streaming mpeg-1 media over the net. The client 
> receives separate audio and video stream. My problem is how to recompose 
> these 2 streams to original mpeg-1 file in client?

You would need to write a file sink class that reads from the 
"MediaSession"s input 'subsessions' (audio and video), and writes them to a 
single file, in MPEG Program Stream format.

To do this, you will need to be familiar with the MPEG 1 and 2 'Systems' 
specification.  You might also find the "QuickTimeFileSink" class useful as 
a model (because it does something somewhat similar, except that it writes 
out a QuickTime file rather than a MPEG Program Stream file.  (Also, note 
<http://www.live.com/liveMedia/faq.html#separate-rtp-streams>)


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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From <EMAIL-ADDRESS-HIDDEN> Mon Jul 7 14:23:14 2003
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Subject: [Live-devel] Synchronization timestamps
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X-Original-Date: Mon, 7 Jul 2003 14:17:50 -0700
Date: Mon, 7 Jul 2003 14:17:50 -0700

Hi,

I recently inherited the mim/mimd projects at the University of Oregon, and
have been looking at the Live Media library in an attempt to fix some of the
problems that these utilities have. In particular, I am looking at the
synchronization problems.

In the FAQ, it mentions the presentationTime argument that is sent to the
afterGettingFrame() method of FileSink. I took FileSink and added a print
statement to watch the incoming data stream timestamps, and noticed that they
are often out of order, more so that I would expect from just UDP issues. 

What I am trying to figure out is that since FileSink just dumps this to
stdout, why doesn't this mass disorder mess up the decoding? Is there yet
another sequence number embedded in the MPEG data? 

This raises several issues for me. There are almost always multiple calls to
afterGettingFrame with the same time stamp, which must belong to the same
time slice of audio or video. If I want to synchronize, I can easily sort and
buffer by the presentation time stamp, but do I also need to detect sub-frame
order problems, or can I just concatenate the incoming data segments with the
same presentation times into a single buffer in the order they arrive and push
that out to a decoder? 

Thanks!

-- 
Anthony Kay                        Phone: (541) 346-1719
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University of Oregon Computing Center             B4D9 BAF1 C76E AB6E 33E5

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From <EMAIL-ADDRESS-HIDDEN> Tue Jul 8 00:30:00 2003
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X-Original-Date: Tue, 08 Jul 2003 00:16:28 -0700
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>In the FAQ, it mentions the presentationTime argument that is sent to the
>afterGettingFrame() method of FileSink. I took FileSink and added a print
>statement to watch the incoming data stream timestamps, and noticed that they
>are often out of order, more so that I would expect from just UDP issues.
>
>What I am trying to figure out is that since FileSink just dumps this to
>stdout, why doesn't this mass disorder mess up the decoding?

Anthony,

I think you may be getting confused between the RTP sequence number and the 
RTP timestamp  - these are different fields in the RTP packet.  The 
underlying RTP reception code ("MultiFramedRTPSource") always checks the 
RTP sequence number field to make sure that incoming packets arrive in the 
order that they were sent.  (If any packets happen to arrive out-of-order, 
then they will be put back into the correct order before they're delivered 
to the higher-level code.)

So, the frames that are delivered to "afterGettingFrame()" will *always* be 
in the order that they were sent.  However, the RTP timestamp fields (from 
which the "presentationTime" is derived) in each outgoing packet are *not 
necessarily* monotonically non-decreasing.

The classic example of this - which may be what you're seeing - is MPEG-1 
or 2 video frames.  MPEG video frames are often stored (and transmitted) in 
an order different from display order, because of the presence of "B" 
frames (which need information from both the preceding *and* succeeding 
frame in order to be decoded).

Suppose that a sequence of frames will be displayed in this order:
         I1 B2 P3 B4 P5 B6 I7
Then these frames will typically be stored in a file - and streamed over a 
network - in a different order:
         I1 P3 B2 P5 B4 I7 B6
because, for example, frame B2 needs information from both I1 and P3 in 
order to be decoded, so I1 and P3 are usually sent before B2.  That's why 
the RTP timestamps (and thus "presentationTime") can sometimes decrease.

         Ross Finlayson
         LIVE.COM
         <http://www.live.com/>

ps. One other thing to note is that "presentationTime" is not totally 
accurate until the first RTCP "Sender Report" packet has been 
received.  You can test this using the function 
"RTPSource::hasBeenSynchronizedUsingRTCP()".



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From <EMAIL-ADDRESS-HIDDEN> Thu Jul 10 10:07:52 2003
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X-Original-Date: Thu, 10 Jul 2003 18:17:54 +0200 (CEST)
Date: Thu, 10 Jul 2003 18:17:54 +0200 (CEST)

Hi all.

  Thank you for your library, Ross ;D. It´s great!!

  I would like to do an app. that gets MPEG2 frames
(from a file) and send 2 RTP streams at same time. One
of them with all the frames (25 fps-PAL) and the other
only with a number of frames (configurable) per
second. I have some questions for the forum:

  1.- Could i make it with 1 or with 2 RTSP?

  2.- I have seen that you get the frames from
MPEGVideoStreamFramer (parseVideoSequenceHeader
method). Could you explain us how this works? 

  Thanks in advance!!

  Regards.

    Juan Carlos


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X-Original-Date: Fri, 11 Jul 2003 09:58:06 +0200
Date: Fri, 11 Jul 2003 09:58:06 +0200

Hi *,
I'm trying to fix the ffserver rtp code, and I'm currently using 
mplayer+liveMedia as test client. After having some problems with mplayer, I 
decided to dig into the liveMedia code to see what was going wrong. Since I 
don't know mplayer nor liveMedia source code, it was a rather difficult work!
I found that mplayer stops playing in "certain situations". after some debug, 
I observed that when some rtp udp packets get lost, liveMedia begins discard 
all the incoming packets and can't recover from the error situation. So 
mplayer is stuck waiting for a frame that never arrives. With some luck, I 
made things work almost perfectly using this patch:

--- live-org/liveMedia/MultiFramedRTPSource.cpp
+++ live/liveMedia/MultiFramedRTPSource.cpp
(at)(at) -148,7 +148,7 (at)(at)
       }
       if (fPacketLossInFragmentedFrame) {
 	// This packet is unusable; reject it:
-	  fReorderingBuffer->releaseUsedPacket(nextPacket);
+	//  fReorderingBuffer->releaseUsedPacket(nextPacket);
 	  fNeedDelivery = True;
 	  return;
       }

I guess what liveMedia is doing here: if a corrupted rtp frame is detected 
(i.e. some udp packets got lost), then simply discard this frame - right! But 
liveMedia doesn't recover from the error in my test case, even if all the 
next udp packets (so the next rtp frames) are correctly received!
I know this patch is the "wrong way" to fix this behaviour, so I'm asking for 
your help to understand exactly what's going wrong, since I'm not very 
confident with liveMedia libs.
If you need more detailed informations/tests, just ask.
Thanks!

Giancarlo



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From <EMAIL-ADDRESS-HIDDEN> Fri Jul 11 01:25:25 2003
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Subject: RE: [Live-devel] MPEG2 frames
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X-Original-Date: Fri, 11 Jul 2003 09:57:31 +0200
Date: Fri, 11 Jul 2003 09:57:31 +0200
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Hello Juan,

what functionality would you expect of streaming only a certain number of frames per sec: As the frames are dependent of each other you cann't leave out random frames. You could decide to only send the I frames of only once every Nth I frame. this would be possible although you would need to chech which frames are I frames and which are P frames.

Best Regards,

Ruud Schramp

> -----Original Message-----
> From: Juan Carlos Montes [mailto:jcmontes_es(at)yahoo.com]
> Sent: donderdag 10 juli 2003 18:18
> To: live-devel(at)lists.sourceforge.net
> Subject: [Live-devel] MPEG2 frames
> 
> 
> Hi all.
> 
>   Thank you for your library, Ross ;D. It´s great!!
> 
>   I would like to do an app. that gets MPEG2 frames
> (from a file) and send 2 RTP streams at same time. One
> of them with all the frames (25 fps-PAL) and the other
> only with a number of frames (configurable) per
> second. I have some questions for the forum:
> 
>   1.- Could i make it with 1 or with 2 RTSP?
> 
>   2.- I have seen that you get the frames from
> MPEGVideoStreamFramer (parseVideoSequenceHeader
> method). Could you explain us how this works? 
> 
>   Thanks in advance!!
> 
>   Regards.
> 
>     Juan Carlos
> 
> 
> ___________________________________________________
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Subject: Re: [Live-devel] MPEG2 frames
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X-Original-Date: Fri, 11 Jul 2003 02:54:03 -0700
Date: Fri, 11 Jul 2003 02:54:03 -0700


>   I would like to do an app. that gets MPEG2 frames
>(from a file) and send 2 RTP streams at same time. One
>of them with all the frames (25 fps-PAL) and the other
>only with a number of frames (configurable) per
>second. I have some questions for the forum:
>
>   1.- Could i make it with 1 or with 2 RTSP?

The way the RTSP server is set up right now, you will need two instances of 
"RTSPServer" - each one using a different port number for TCP.

In fact, the simplest way to do what you're trying to do would be to have 
two separate applications - both reading from the same file.  Each app 
would have its own RTSP server (on a different port).  One app would send 
all the frames; the other would send only some of them.

>   2.- I have seen that you get the frames from
>MPEGVideoStreamFramer (parseVideoSequenceHeader
>method). Could you explain us how this works?

"MPEGVideoStreamFramer" is a liveMedia filter that parses an input stream 
into discrete MPEG (1 or 2) video frames.  More precisely, it parses the 
stream into a sequence of structures from the following set:
         - a video sequence header
         - a GOP header
         - a picture header
         - a slice
I.e., each time "getNextFrame()" is called on a "MPEGVideoStreamFramer" 
object, the returned 'frame' will be a structure from this set (so, not 
actually an entire 'frame', despite the name).

As I noted in my earlier message, a "MPEGVideoStreamFramer" can also be 
told to skip 'picture headers' and 'slices' that don't belong to 
I-frames.  (To see how this works, note the use of the 
"fSkippingCurrentPicture" field.)


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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>. You could decide to only send the I frames of only once every Nth I 
>frame. this would be possible although you would need to chech which 
>frames are I frames and which are P frames.

Note that the "MPEGVideoStreamFramer" class already does this for you.  It 
takes an optional Boolean parameter telling it whether or to return only 
I-frames.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Fri, 11 Jul 2003 02:34:47 -0700
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>I observed that when some rtp udp packets get lost, liveMedia begins discard
>all the incoming packets and can't recover from the error situation.

It does this only if the lost packet was part of a fragmented frame (i.e., 
a frame that was too large to fit into a single packet, and so is split 
over multiple packets), and it discards only those remaining packets that 
are part of this fragmented frame.

I suspect that your RTP streaming server is not setting the RTP "M" bit 
correctly.  For most video payload formats, the "M" bit is supposed to be 
set on the last packet of a fragmented frame (or on the single packet of a 
non-fragmented frame).

If the input source sets the "M" bit correctly, then the liveMedia code 
should be doing the right thing.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Fri, 11 Jul 2003 15:35:55 +0200
Date: Fri, 11 Jul 2003 15:35:55 +0200

Ross Finlayson wrote:
> >I observed that when some rtp udp packets get lost, liveMedia begins
> > discard all the incoming packets and can't recover from the error
> > situation.
>
> It does this only if the lost packet was part of a fragmented frame (i.e.,
> a frame that was too large to fit into a single packet, and so is split
> over multiple packets), and it discards only those remaining packets that
> are part of this fragmented frame.
>
> I suspect that your RTP streaming server is not setting the RTP "M" bit
> correctly.  For most video payload formats, the "M" bit is supposed to be
> set on the last packet of a fragmented frame (or on the single packet of a
> non-fragmented frame).
>
> If the input source sets the "M" bit correctly, then the liveMedia code
> should be doing the right thing.

Rox, _really_ tanx for your explanation.

Actually, ffserver didn't set M bit. Made it do, didn't work either.
But your suggestion pointed me to the right direction: for payload type 32, 
the fCurrentPacketBeginsFrame flag is set into MPEGVideoRTPSource.cpp, and is 
function of the bit S, B, E specific for this payload type (which ffserver 
didn't set correctly).
Fixed this point too, now ffserver works.

Thank you!

-- Giancarlo



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From <EMAIL-ADDRESS-HIDDEN> Fri Jul 11 13:25:11 2003
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Hi,

I'm currently working on a capability description protocol for my 
diploma thesis which I would like to implement using the live RTSP 
implementation for the signaling part. I did have a look at some other 
possible solutions but openRTSP and the integration into mplayer are 
simply too sweet to resist :-)

So I was wondering if someone could provide me with some insights on 
how openRTSP.cpp, RTSPClient.cpp and playCommon.cpp work together (or 
maybe some state/sequence diagramms) to help me get started. I did run 
doygen over the source code but it's still not completly clear how 
everthing works together.

Any help would be appreciated...

Gruß
Marc


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Hi,

has anyone created a ProjectBuilder Project for working with live? I'm 
pretty much a newbie to PB so I would be eternally gratefull if anyone 
could provide me with a project file that works with live :-)

Gruß
Marc



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>So I was wondering if someone could provide me with some insights on how 
>openRTSP.cpp, RTSPClient.cpp and playCommon.cpp work together

OK, here's a start:

"playCommon.cpp" contains the "main()" routine for both the "openRTSP" and 
"playSIP" test programs.  (These programs are very similar in 
functionality, which is why their main program code has been put into a 
single file - "playCommon.cpp" - with the program-specific functionality 
moved into separate files: "openRTSP.cpp" and "playSIP.cpp" respectively.)

I.e., "openRTSP" is built by linking "playCommon.o" with "openRTSP.o" (and 
the LIVE.COM libraries).  "playSIP" is built by linking "playCommon.o" with 
"playSIP.o" (and the LIVE.COM libraries).

"playCommon.cpp" does the following:
1/ verifies and interprets command-line arguments
2/ creates an appropriate client object ("RTSPClient" or "SIPClient")
3/ uses this client object to 'open' the supplied URL (using RTSP's 
"DESCRIBE" or SIP's "INVITE") to get a SDP description (that describes the 
media session).
4/ creates a "MediaSession" object from this SDP description.  (This 
"MediaSession" object will point to "MediaSubsession" for each of the 
(audio, video, etc.) subsessions that were described in the SDP description.
5/ calls "initiate()" on each media subsession that you're interested in 
playing.  ("initiate()" causes appropriate "RTPSource" and "RTCPInstance" 
objects to be set up for the subsession, so incoming RTP data can be received.)
6/ does any additional protocol operations that are necessary to start the 
data flow.  (For RTSP ("openRTSP"), this means sending RTSP "SETUP" and 
"PLAY" commands.)

IMHO, a better way to learn how RTSP (and SIP) clients can use the LIVE.COM 
code is to study the "MPlayer" RTSP/RTP implementation (see 
<http://www.live.com/mplayer/>).  In particular, note the 
"demux_open_rtp()" function (in "libmpdemux/demux_rtp.cpp").



	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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From <EMAIL-ADDRESS-HIDDEN> Sun Jul 13 13:04:40 2003
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X-Original-Date: Sun, 13 Jul 2003 12:53:50 -0700
Date: Sun, 13 Jul 2003 12:53:50 -0700

A new source class - "WAVAudioFileSource" - has now been added to the 
"liveMedia" library.  This lets you open and stream PCM audio from 
WAV-format audio files.

There are also new filter classes for translating between 16-bit PCM and 
8-bit u-law audio (see "liveMedia/include/uLawAudioFilter.hh" for details).

Finally, there is a new test program: "testWAVAudioStreamer".  This program 
reads from a WAV-format audio file (named "test.wav"), and streams the 
enclosed PCM audio stream via IP multicast, using a built-in RTSP 
server.  (The resulting stream can be played using open standard media 
players - such as QuickTime Player.)  Also, optionally, 16-bit PCM streams 
can be converted to 8-bit u-law format prior to streaming.

	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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> OK, here's a start:
Thanks a lot. This definitively saved me a lot of time. I'm also 
positively surprised by the amount of comments I found in the source 
files. Since the comments are already there, how about using doxygen to 
automatically create an API/code documentation? Someone would have to 
change the comment style in some places though :-)

> IMHO, a better way to learn how RTSP (and SIP) clients can use the 
> LIVE.COM code is to study the "MPlayer" RTSP/RTP implementation (see 
> <http://www.live.com/mplayer/>).  In particular, note the 
> "demux_open_rtp()" function (in "libmpdemux/demux_rtp.cpp").
It actually looks easier to use the LIVE.COM code than to compile 
MPlayer under OS X (it still doesn't compile). Maybe I should write a 
simple player on my own instead of spending more time trying to compile 
MPlayer  :-)

Gruß
Marc



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X-Original-Date: Sun, 13 Jul 2003 15:45:03 -0700
Date: Sun, 13 Jul 2003 15:45:03 -0700


>I'm also positively surprised by the amount of comments I found in the 
>source files. Since the comments are already there, how about using 
>doxygen to automatically create an API/code documentation? Someone would 
>have to change the comment style in some places though :-)

Yes, someday...

         Ross.



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X-Original-Date: Mon, 14 Jul 2003 07:31:17 -0700 (PDT)
Date: Mon, 14 Jul 2003 07:31:17 -0700 (PDT)

Dear All,

I could stream MPEG1 using testMPEGVideoStreamer and
testMPEGAudioVideoStreamer test programs for MPlayer.
I couldn't stream MPEG2 file using any of these
programs. Can I use these test programs to stream
MPEG2 files for MPlayer?. If not what do I need to
modify to get MPEG2 streaming?. Pls suggest.
Thanks in advance. 

Best Regards,
Subhash

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X-Original-Date: Mon, 14 Jul 2003 08:59:17 -0700
Date: Mon, 14 Jul 2003 08:59:17 -0700


>I could stream MPEG1 using testMPEGVideoStreamer and
>testMPEGAudioVideoStreamer test programs for MPlayer.
>I couldn't stream MPEG2 file using any of these
>programs. Can I use these test programs to stream
>MPEG2 files for MPlayer?.

Yes, the "testMPEGVideoStreamer" and "testMPEGAudioVideoStreamer" programs 
work for both MPEG-1 and MPEG-2 Video Elementary Streams, and audio+video 
Program Streams (respectively).

Perhaps what you really mean is that you can't *play* MPEG-2 RTP streams in 
your media player - e.g., in QuickTime Player.  If so, then this is 
expected, because these players do not support MPEG-2 by default (because 
of the cost of licensing).  I believe, however, that you can purchase 
MPEG-2 plugins for these players from the Apple QuickTime web site.

If , however, you're running Linux, you can also "MPlayer" (with LIVE.COM 
Streaming Media support - see <http://www.live.com/mplayer/>) to play 
MPEG-2 RTP streams.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Tue, 15 Jul 2003 09:37:47 +0200
Date: Tue, 15 Jul 2003 09:37:47 +0200

Heya,

I would like to use live.com libs to stream MPEG-4 content to 
QuickTime clients, with the RTP streams tunneled over HTTP to
allow peer-to-peer streaming to clients behind firewalls.

I am aware MPEG-4 streaming support is not available yet in 
live.com libraries, but (for now) I am mostly interested in the
RTP tunneling aspect.

As I understand it, live.com implements a HTTPSink which has
nothing to do with RTP tunneling, but which implements merely 
dumping a file to a client over http. I've had a brief look at 
how RTP tunneling is implemented in Darwin Streaming Server; 
my first impression is that it is not trivial, and seems to be 
closely  related to RTSP...

Anybody with more info or plans related to RTP tunneling over 
HTTP?

Many thanks,

Tom


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X-Original-Date: Tue, 15 Jul 2003 02:47:19 -0700 (PDT)
Date: Tue, 15 Jul 2003 02:47:19 -0700 (PDT)

> Date: Mon, 14 Jul 2003 08:59:17 -0700
> To: live-devel(at)lists.sourceforge.net
> From: Ross Finlayson <finlayson(at)live.com>
> Subject: Re: [Live-devel] Streaming MPEG2 using test
> programs for
>   MPlayer
> Reply-To: live-devel(at)lists.sourceforge.net
> 
> 
> >I could stream MPEG1 using testMPEGVideoStreamer
> and
> >testMPEGAudioVideoStreamer test programs for
> MPlayer.
> >I couldn't stream MPEG2 file using any of these
> >programs. Can I use these test programs to stream
> >MPEG2 files for MPlayer?.
> 
> Yes, the "testMPEGVideoStreamer" and
> "testMPEGAudioVideoStreamer" programs 
> work for both MPEG-1 and MPEG-2 Video Elementary
> Streams, and audio+video 
> Program Streams (respectively).

Yes I could stream MPEG1 ES & PS to Mplayer on linux
machine using RTSP protocol. The test programs are
enabled to Implement RTSP server.
 
> Perhaps what you really mean is that you can't
> *play* MPEG-2 RTP streams in 
> your media player - e.g., in QuickTime Player.  If
> so, then this is 
> expected, because these players do not support
> MPEG-2 by default (because 
> of the cost of licensing).  I believe, however, that
> you can purchase 
> MPEG-2 plugins for these players from the Apple
> QuickTime web site.

Sorry, I am not using any of these players. I am using
Mplayer on linux machine.
 
> If , however, you're running Linux, you can also
> "MPlayer" (with LIVE.COM 
> Streaming Media support - see
> <http://www.live.com/mplayer/>) to play 
> MPEG-2 RTP streams.

Exactly, I am using Mplayer with live.com library to
receive the streams and testMPEGAudioVideoStreamer &
testMPEGVideoStreamer programs as server. I could
stream using these programs MPEG1 ES and PS and play
it using MPlayer with RSTP protocol. 

The problem is with MPEG2 files. The server
(testMPEGxxxStreamer) runs without error and displays
msgs
Begining Streaming..
Begining to read from file...
...done reading from file.

The Mplayer sends the DESCRIBE request. and waits for
the response from server but doesn't receives any
esponse from server. I could locate this in the
RTSPClient::describeURL(xxx  ) method.

What could be the reason?.

> 
> 
> 	Ross Finlayson
> 	LIVE.COM
> 	<http://www.live.com/>
> 
> 
> 
> 
> --__--__--
> 
> _______________________________________________
> Live-devel mailing list
> Live-devel(at)lists.sourceforge.net
>
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> 
> 
> End of Live-devel Digest


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X-Original-Date: Tue, 15 Jul 2003 02:50:57 -0700
Date: Tue, 15 Jul 2003 02:50:57 -0700


>I am aware MPEG-4 streaming support is not available yet in
>live.com libraries

That hasn't been true for several months.  The LIVE.COM libraries support 
MPEG-4 audio and video RTP sources and sinks.  For example, you can play 
incoming MPEG-4 RTSP/RTP streams using "openRTSP" (or "MPlayer").  You can 
also stream outgoing MPEG-4 audio or video RTP streams, provided that the 
data is available as discrete frames.  (What are not yet implemented are 
MPEG-4 'framer' filters that parse MPEG-4 audio or video *byte streams* 
into discrete frames.)

>I would like to use live.com libs to stream MPEG-4 content to
>QuickTime clients, with the RTP streams tunneled over HTTP to
>allow peer-to-peer streaming to clients behind firewalls.

Note that RTSP allows (as an option) for the RTP stream(s) to be sent over 
the RTSP TCP connection.  Our RTSP *client* implementation supports this, 
but the RTSP *server* implementation currently does not.

Apple has also defined a mechanism that allows the RTP stream(s) to be sent 
over a HTTP TCP connection.  This is a non-standard hack, but Apple 
documents it in a file named "RTSP_Over_HTTP.pdf", which it distributes 
with its Darwin Streaming Server code.

>Anybody with more info or plans related to RTP tunneling over HTTP?

Unfortunately there are currently no near-term plans to implement 
RTP-over-HTTP (especially not at the server end).


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] Re: Streaming MPEG2 using RTSP with test
  programs to Mplayer.
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X-Original-Date: Tue, 15 Jul 2003 03:25:52 -0700
Date: Tue, 15 Jul 2003 03:25:52 -0700


>The problem is with MPEG2 files. The server
>(testMPEGxxxStreamer) runs without error and displays
>msgs
>Begining Streaming..
>Begining to read from file...
>...done reading from file.

Does the time gap between "Beginning to read from file..." and "...done 
reading from file" correspond to the actual playing time of the movie 
file?  Or is this gap much shorter than expected?  If it's much shorter, 
then this suggests that the program was not able to parse the file, for 
some reason.  Make sure that it really is a MPEG-2 Elementary Video Stream, 
or a Program Stream file.  (Note that MPEG-2 *Transport* stream files won't 
work.)


>The Mplayer sends the DESCRIBE request. and waits for
>the response from server but doesn't receives any
>esponse from server. I could locate this in the
>RTSPClient::describeURL(xxx  ) method.
>
>What could be the reason?.

If your MPEG-2 files are OK, then I'm not sure.  Are you running a fairly 
recent version of the LIVE.COM code?  (The 2003.04.09 version fixed a bug 
that might have affected this.)

If you disable the built-in RTSP server, are you able to receive and play 
the MPEG-2 RTP stream using the appropriate "testMPEG*Video.sdp" file 
(i.e., using a "sdp://" pseudo-URL in MPlayer)?



	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Subject: Re: [Live-devel] RTP tunneling over HTTP
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X-Original-Date: Tue, 15 Jul 2003 12:56:31 +0200
Date: Tue, 15 Jul 2003 12:56:31 +0200


> >I am aware MPEG-4 streaming support is not available yet in
> >live.com libraries
>
> That hasn't been true for several months.  The LIVE.COM libraries support
> MPEG-4 audio and video RTP sources and sinks.  For example, you can play
> incoming MPEG-4 RTSP/RTP streams using "openRTSP" (or "MPlayer").  You can
> also stream outgoing MPEG-4 audio or video RTP streams, provided that the
> data is available as discrete frames.  (What are not yet implemented are
> MPEG-4 'framer' filters that parse MPEG-4 audio or video *byte streams*
> into discrete frames.)

I know, I was referring to the server (i.e. streaming) side only. The
implementation of such MPEG-4 frame filters should be easy though. Maybe 
a little bit more work is involved for interpretation of the mp4 file format,
but I suspect I can do without for now (and use sdp files instead of RTSP).

> >I would like to use live.com libs to stream MPEG-4 content to
> >QuickTime clients, with the RTP streams tunneled over HTTP to
> >allow peer-to-peer streaming to clients behind firewalls.
>
> Note that RTSP allows (as an option) for the RTP stream(s) to be sent over
> the RTSP TCP connection.  Our RTSP *client* implementation supports this,
> but the RTSP *server* implementation currently does not.

Oh? At least a bit of good news. If RTP over TCP is implemented on the client 
side only, what streaming server that supports RTP over TCP should be used 
for this? How much work is involved in supporting RTP over TCP on the server
side?

> Apple has also defined a mechanism that allows the RTP stream(s) to be sent
> over a HTTP TCP connection.  This is a non-standard hack, but Apple
> documents it in a file named "RTSP_Over_HTTP.pdf", which it distributes
> with its Darwin Streaming Server code.

It might be a non-standard hack, but at least it works for the QT client/DSS
setup. How much does this approach differ from the live.com's RTP over TCP
approach?

Thanks,

Tom



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X-Original-Date: Tue, 15 Jul 2003 05:34:31 -0700
Date: Tue, 15 Jul 2003 05:34:31 -0700


> > Note that RTSP allows (as an option) for the RTP stream(s) to be sent over
> > the RTSP TCP connection.  Our RTSP *client* implementation supports this,
> > but the RTSP *server* implementation currently does not.
>
>Oh? At least a bit of good news. If RTP over TCP is implemented on the client
>side only, what streaming server that supports RTP over TCP should be used
>for this?

Apple's "Darwin" (aka. QuickTime) Streaming Server supports this.  (The 
RealNetworks "Helix" server might also.)

>  How much work is involved in supporting RTP over TCP on the server
>side?

This cannot be done until support for unicast RTP-over-UDP RTSP server 
support is implemented (which, fortunately, may end up happening later this 
summer, pending funding...)


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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From <EMAIL-ADDRESS-HIDDEN> Wed Jul 16 11:02:02 2003
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Subject: [Live-devel] ByteStreamFileSource
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X-Original-Date: Wed, 16 Jul 2003 19:57:10 +0200 (CEST)
Date: Wed, 16 Jul 2003 19:57:10 +0200 (CEST)

Hi all.
  
  I have seen the source code of ByteStreamFileSource,
FramedFileSource and FramedSource, now, i have some
questions for you ;D

  1.-What is the objective of FramedFileSource class?
It seems that it is only to make diferences between
Filters and FileSources...

  ByteStreamFileSource:

  2.- Why do you always get a buffer of 10000 bytes
(fMaxSize) ? 

  3.- How could i use fPreferredFrameSize? (i think
that it depends of bitrate and resolution but i don´t
know how to calculate it :( )

  Regards.

   Juan Carlos

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>   1.-What is the objective of FramedFileSource class?

This is merely a parent class that holds the 'fid' (FILE*) for the 
stream.  (It is used by both "ByteStreamFileSource" and "MP3FileSource".)

>   2.- Why do you always get a buffer of 10000 bytes
>(fMaxSize) ?

"fMaxSize" is set by the client (calling) object, when it calls 
"getNextFrame".  (It is the "maxSize" parameter.)  It isn't 'always' 10000.

>   3.- How could i use fPreferredFrameSize? (i think
>that it depends of bitrate and resolution but i don´t
>know how to calculate it :( )

The "preferredFrameSize" parameter (to "ByteStreamFileSource::createNew()") 
should be used if the file data naturally occurs in fixed-sized frames, or 
if you want to group fine-grain data into larger 'frames' for more 
efficient streaming.

Its use depends on the particular type of data that you're streaming.  If 
you set this parameter, then you should probably set the "playTimePerFrame" 
parameter also.

What kind of data is stored in the file that you're reading?


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Thu, 17 Jul 2003 14:22:47 +0200 (CEST)
Date: Thu, 17 Jul 2003 14:22:47 +0200 (CEST)

> >   1.-What is the objective of FramedFileSource
> class?
> 
> This is merely a parent class that holds the 'fid'
> (FILE*) for the=20
> stream.  (It is used by both "ByteStreamFileSource"
> and "MP3FileSource".)

OK.

> 
> >   2.- Why do you always get a buffer of 10000
> bytes
> >(fMaxSize) ?
> 
> "fMaxSize" is set by the client (calling) object,
> when it calls=20
> "getNextFrame".  (It is the "maxSize" parameter.) 
> It isn't 'always' 10000.
> 
Yes, it´s only the first time.

> >   3.- How could i use fPreferredFrameSize? (i
> think
> >that it depends of bitrate and resolution but i
> don=B4t
> >know how to calculate it :( )
> 
> The "preferredFrameSize" parameter (to
> "ByteStreamFileSource::createNew()")=
> =20
> should be used if the file data naturally occurs in
> fixed-sized frames, or=
> =20
> if you want to group fine-grain data into larger
> 'frames' for more=20
> efficient streaming.
> 
> Its use depends on the particular type of data that
> you're streaming.  If=20
> you set this parameter, then you should probably set
> the "playTimePerFrame"=
> =20
> parameter also.
> 
> What kind of data is stored in the file that you're
> reading?
> 

I try to stream MPEG2 PE over the network polling from
a MPEG2 Encoder card directly. The bitrate is 4Mb and
the resolution is 720x576 (PAL) but could be change.
If i use pipes i have small stops (the video is not
fluid) because the pipes has to switch a lot (the
buffers are 64Kb and pipes block in 4Kb) (all testing
without live).

Now i´m trying to make a DeviceSource implementation
and i have this problems. I take like an example
ByteStreamFileSource and change the fread from
doGetNextFrame() to Card->Read that implements the
read to a card (the card only let me reads a fixed
amount of bytes). I have tested CardManager class with
another program and seem to work perfectly. With live
i have all (debug) messages that i put in printf
function calls but it does not send any data over the
network.

 Below you can see MyDeviceSource implementation and
CardManager::Read implementation


// Implementation

#if defined(__WIN32__) || defined(_WIN32)
#include <io.h>
#include <fcntl.h>
#endif

#include "MyDeviceSource.hh"
#include "CardManager.hh"
#include "GroupsockHelper.hh"

////////// MyDeviceSource //////////

MyDeviceSource::MyDeviceSource(UsageEnvironment& env,
					   CardManager* cardsettings,
					   unsigned preferredFrameSize,
					   unsigned playTimePerFrame)
  : FramedSource(env),
fPreferredFrameSize(preferredFrameSize),
    fPlayTimePerFrame(playTimePerFrame),
fLastPlayTime(0) {
    //fCardManager is CardManager
    fCardManager = cardsettings;
}

MyDeviceSource::~MyDeviceSource() {
}

MyDeviceSource*
MyDeviceSource::createNew(UsageEnvironment& env,
				CardManager* cardsettings,
				unsigned preferredFrameSize,
				unsigned playTimePerFrame) {

  // Open the card and initialize params
  if (cardsettings->init()==-1)
    return NULL;

  do {
    return new MyDeviceSource(env, cardsettings,
preferredFrameSize,
				    playTimePerFrame);
  } while (0);

  return NULL;
}

#ifdef BSD
static struct timezone Idunno;
#else
static int Idunno;
#endif

void MyDeviceSource::doGetNextFrame() {
  if ( fCardManager->haveToClosure() /* tengo algun
problema con la tarjeta feof(fFid) || ferror(fFid)*/)
{
    handleClosure(this);  // la cierro
    return;
  }

  // Try to read as many bytes as will fit in the
buffer provided
  // (or "fPreferredFrameSize" if less)

  if (fPreferredFrameSize > 0 && fPreferredFrameSize <
fMaxSize) {
    fMaxSize = fPreferredFrameSize;
  }


  // Get data from the card...
  fFrameSize = fCardManager->read(fTo, fMaxSize);
  printf ("I have read %i bytes from the card.\n",
fFrameSize);

  // This is all that ByteStreamFileSource do
  //fprintf ( file, "fFrameSize:%i = fread(fTo:%i, 1,
fMaxSize:%i, fFid);\n", fFrameSize,
sizeof(*fTo),fMaxSize);

  // Set the 'presentation time':
  if (fPlayTimePerFrame > 0 && fPreferredFrameSize >
0) {
    //fprintf ( file, "      fPlayTimePerFrame %i > 0
&& fPreferredFrameSize %i\n", fPlayTimePerFrame,
fPreferredFrameSize);
    if (fPresentationTime.tv_sec == 0 &&
fPresentationTime.tv_usec == 0) {
      // This is the first frame, so use the current
time:
      gettimeofday(&fPresentationTime, &Idunno);
    } else {
      // Increment by the play time of the previous
data:
      unsigned uSeconds	= fPresentationTime.tv_usec +
fLastPlayTime;
      fPresentationTime.tv_sec += uSeconds/1000000;
      fPresentationTime.tv_usec = uSeconds%1000000;
    }

    // Remember the play time of this data:
    fLastPlayTime =
(fPlayTimePerFrame*fFrameSize)/fPreferredFrameSize;
    //fprintf ( file, "fLastPlayTime %i\n",
fLastPlayTime);

  } else {
    // We don't know a specific play time duration for
this data,
    // so just record the current time as being the
'presentation time':
    gettimeofday(&fPresentationTime, &Idunno);
  }

  // Switch to another task, and inform the reader
that he has data:
  nextTask() =
envir().taskScheduler().scheduleDelayedTask(0,
				(TaskFunc*)FramedSource::afterGetting, this);
}

float MyDeviceSource::getPlayTime(unsigned numFrames)
const {
  return
(float)(numFrames*(fPlayTimePerFrame/1000000.0));
}


//CardManager::read method gets data from the Card but
to test i 
//have change to read from a file.


unsigned CardManager::read (unsigned char* To,
unsigned MaxSize)
{

  // I have two buffer, one to get the data and the
other to storage the 
  // data from one call to another
  // unsigned char buffer[1024*512];     // maximum
buffer size is 512Kb
  // unsigned char nextbuffer[1024*512];

  unsigned numBytesRead=0;
  unsigned numBytesWanted=0;
  
  printf("	i have nextbuffer %i \n", nnextbuffer);

  if (nnextbuffer>0){
    if (MaxSize<nnextbuffer)
    {
      memmove(&To[numBytesRead], nextbuffer, MaxSize);
// copy to live
      memmove( nextbuffer, &(nextbuffer[MaxSize]),
nnextbuffer-MaxSize); // copy to live	
      nnextbuffer=nnextbuffer-MaxSize;
      numBytesRead += MaxSize;
    }
    else
    {
      memmove(&To[numBytesRead], nextbuffer,
nnextbuffer); // copy to live
      numBytesRead += nnextbuffer;
      nnextbuffer=0;
    }
    printf("	copy from nextbuffer
%i!!\n",nnextbuffer);
  }


  while (numBytesRead < MaxSize) {

      numBytesWanted=MaxSize-numBytesRead; // Bytes
that i need

      printf("	numBytesWanted %i\n", numBytesWanted);

      if ( numBytesWanted > sizeof(buffer) ) // If
it's bigger than the buffer
        numBytesWanted=sizeof(buffer);

   	// read from device this would be change!!!!

	//int thisRead =
read(numTarjeta,(char*)buffer,sizeof(buffer));
      int thisRead =
fread((char*)buffer,1,sizeof(buffer),fid);

	  totBytesRead+=thisRead;

      if (thisRead > 0 )
      {
      	printf("	read from device %i ... trying
copy\n", thisRead);
      	printf("	copy only numBytesWanted %i!!\n",
numBytesWanted);
      	memmove(&To[numBytesRead], buffer,
numBytesWanted); // copy to live
		printf("	copy!!\n");

		if (thisRead>numBytesWanted)
		{
		nnextbuffer = thisRead-numBytesWanted;
		memmove(nextbuffer, &(buffer[numBytesWanted]),
nnextbuffer);
		printf("	for next read i have nextbuffer %i \n",
nnextbuffer);
		}
		numBytesRead += numBytesWanted;
	  }
	  else
      {
		printf("	total read from device %i ...\n",
totBytesRead);
		haveToClose = 1;
		totBytesRead = 0;
		return numBytesRead;
      }
  }
  return numBytesRead;
}


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From <EMAIL-ADDRESS-HIDDEN> Thu Jul 17 07:13:19 2003
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Subject: [Live-devel] Re: Streaming MPEG2 using RTSP with test programs to Mplayer.
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X-Original-Date: Thu, 17 Jul 2003 07:08:06 -0700 (PDT)
Date: Thu, 17 Jul 2003 07:08:06 -0700 (PDT)

Dear Ross,

>The problem is with MPEG2 files. The server
>(testMPEGxxxStreamer) runs without error and displays
>msgs
>Begining Streaming..
>Begining to read from file...
>...done reading from file.

>Does the time gap between "Beginning to read from
>file..." and "...done reading from file" correspond
>to the actual playing time of the movie file?  Or is
>this gap much shorter than expected?  If it's much 
>shorter, then this suggests that the program was not
>able to parse the file, for some reason.

Yes, the gap between two messages corresponds to
actual playing time. The openRTSP client receives and
saves in Video/audio files from both
testMPEG*VideoStreamer programs. After disabling the
IMPLEMENT_SERVER in testMPEGVideoStreamer I can
receive the Mpeg2 files with testVideoReceive. The
problem is with MPlayer.

>Make sure that it really is a MPEG-2 Elementary Video

>Stream, or a Program Stream file.  (Note that MPEG-2
>*Transport* stream files won't work.)

Yes, when played locally in mplayer it shows MPEG-PS
stream.

>The Mplayer sends the DESCRIBE request. and waits for
>the response from server but doesn't receives any
>esponse from server. I could locate this in the
>RTSPClient::describeURL(xxx  ) method.
>
>What could be the reason?.

>If your MPEG-2 files are OK, then I'm not sure.  Are
>you running a fairly recent version of the LIVE.COM
>code?  (The >2003.04.09 version fixed a bug that
>might have affected this.)

I am using 2003.03.16 version of both live.com library
and Mplayer.
Do you think that I may have to use the recent Mplayer
version?. In that case can I use the same live library
or have to use the recent one?

>If you disable the built-in RTSP server, are you able
>to receive and play the MPEG-2 RTP stream using the
>appropriate "testMPEG*Video.sdp" file 
>(i.e., using a "sdp://" pseudo-URL in MPlayer)?

I tried Mplayer with SDP protocol but no luck. The
server is running testMPEGVideoStreamer.
1)Command : mplayer -sdp sdp://testMPEGVideo.sdp

relevant msgs: playing sdp://testMPEGVideo.sdp
Unknown protocol 'sdp'.
Unable to open URL: sdp://testMPEGVideo.sdp.

Exiting...(End of file)

2)Command : mplayer -sdp
sdp://192.168.0.23/testMPEGVideo.sdp
(here 192.168.0.23 is server IP address).

relevant msgs: playing sdp://testMPEGVideo.sdp
Unknown protocol 'sdp'.
Unable to open URL: sdp://testMPEGVideo.sdp.

Exiting...(End of file)



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X-Original-Date: Thu, 17 Jul 2003 09:01:04 -0700
Date: Thu, 17 Jul 2003 09:01:04 -0700


>I am using 2003.03.16 version of both live.com library
>and Mplayer.
>Do you think that I may have to use the recent Mplayer
>version?. In that case can I use the same live library
>or have to use the recent one?

You'll need to use a recent version of the LIVE.COM Streaming Media code 
(from <http://www.live.com/liveMedia/public/>.  You'll also need to use the 
"latest CVS snapshot" version 
<http://www1.mplayerhq.hu/MPlayer/cvs/MPlayer-current.tar.bz2> of the 
MPlayer source code.  See <http://www.live.com/mplayer/> for more information.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Thu, 17 Jul 2003 09:20:26 -0700
Date: Thu, 17 Jul 2003 09:20:26 -0700


>I try to stream MPEG2 PE

What is MPEG "PE"?  Is it a Program Stream (audio+video), or just a video 
elementary stream?  Or something else?

But anyway, I suggest that you start by hooking a "MyDeviceSource" directly 
into a "FileSink".  This will let you test your "MyDeviceSource" 
implementation by recording directly into a file, rather than by streaming 
over the network.  You'll find this easier to debug.

If you're able to successfully record your input data into a file, then you 
should then be able to stream over a network, by replacing 
"ByteStreamFileSource" with "MyDeviceSource" in the 
"testMPEGVideoStreamer.cpp" or "testMPEGAudioVideoStreamer.cpp" - *provided 
that* your data is in MPEG video Elementary Stream or Program Stream format 
(respectively).

If your data is in some *other* format, though, then you may need to do 
more work...


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Subject: [Live-devel] OpenRTSP RTSP client.
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X-Original-Date: Thu, 17 Jul 2003 21:56:45 -0700 (PDT)
Date: Thu, 17 Jul 2003 21:56:45 -0700 (PDT)

Dear All,

Has anybody tried openRTSP with Darwin server?. 
Does OpenRTSP client streams from Darwin server (or
any other server) using RTSP protocol?.
If yes, what formats are supported, MPEG1/2/4 MP3?.

I will be grateful to you for this information.

Thanks,
Subhash.



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Hi Ross,
    I can use testMPEGAudioVideoStreamer to stream MPEG-4 transport streams, but testMPEGAudioVideoStreamer can't stream MPEG-4 program stream. (I noticed your reply in another mail:" You can also stream outgoing MPEG-4 audio or video RTP streams, provided that the data is available as discrete frames. ") What's the difference between these 2 cases concerned with live libraries? Is MPEG-4 transport stream stores data in discrete frames? And what kind of data will testMPEGVideoReceiver.cpp(the client side) receive? Will testMPEGVideoReceiver.cpp can receive video frames of this mpeg-4 transport?

Thanks in advance

Michael
ÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÓ†+ùÞµéšŠX¬²š'²ŠÞu¼ÿTÅš­å¢¶LÁªÞÊ‹œjzîžk¥¶*ezŠ^­«bž2²×¦²‰Ú²)à•éšr§{õˆLs”M8äÈ4oÌ‹âžìZ)Ý£?6‹Þ–[â®Ûš–fœ†)Þ±«m…ë™ëb™ïÅ­ç­®&¥rXœ’«{øm¶Ÿÿÿ0þù°j·ÿr‰¿Â_è}÷«ÿ~9ÿOÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿü¸¯{÷^½éfj)bž	b²Òâ½ïÝz÷¥þX¬¶Ïì¢êÜyú+ïçzØm¶›ÿÿùb²Ûÿ²‹«qçè®ÿë–+-³ùb²Ø§~åŠ÷ÿuëÞ


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X-Original-Date: Fri, 18 Jul 2003 03:56:33 -0700
Date: Fri, 18 Jul 2003 03:56:33 -0700


>Has anybody tried openRTSP with Darwin server?.

Yes.  It works well.

>If yes, what formats are supported, MPEG1/2/4 MP3?.

All of the formats that are checked for in "MediaSubsession::initiate()" 
(see "liveMedia/MediaSession.cpp".


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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From <EMAIL-ADDRESS-HIDDEN> Fri Jul 18 04:18:25 2003
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X-Original-Date: Fri, 18 Jul 2003 13:14:16 +0200 (CEST)
Date: Fri, 18 Jul 2003 13:14:16 +0200 (CEST)

>Now i´m trying to make a DeviceSource implementation
> and i have this problems. I take like an example
> ByteStreamFileSource and change the fread from
> doGetNextFrame() to Card->Read that implements the
> read to a card (the card only let me reads a fixed
> amount of bytes). I have tested CardManager class
with
> another program and seem to work perfectly. With
live
> i have all (debug) messages that i put in printf
> function calls but it does not send any data over
the
> network.
> 
>  Below you can see MyDeviceSource implementation and
> CardManager::Read implementation

Sorry forget this. The problem was that i don´t send a
PE :) 

About "preferredFrameSize" parameter for more
efficient streaming. Could you talk more about this. 

 > 
 > Its use depends on the particular type of data that
 > you're streaming.  If=20
 > you set this parameter, then you should probably
set
 > the "playTimePerFrame"=
 > =20
 > parameter also.


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X-Original-Date: Fri, 18 Jul 2003 04:30:46 -0700
Date: Fri, 18 Jul 2003 04:30:46 -0700


>About "preferredFrameSize" parameter for more
>efficient streaming. Could you talk more about this.

This parameter is used if you want to fix the size of the data chunks that 
get read each time from the "ByteStreamFileSource".

Suppose, for example, that the input file consists of GSM audio 
data.  Because each GSM frame is 33 bytes, you would set 
"preferredFrameSize" to 33 (and "playTimePerFrame" to 20000 (microseconds)) 
when reading GSM audio data.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Fri, 18 Jul 2003 03:57:53 -0700
Date: Fri, 18 Jul 2003 03:57:53 -0700

The "testMPEGVideoStreamer" and "testMPEGAudioVideoStreamer" test programs 
(and "testMPEGVideoReceiver") work only for MPEG-1 or MPEG-2 audio/video - 
not MPEG-4.

	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Subject: Re: [Live-devel] stream mpeg-4 transport streams
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> The "testMPEGVideoStreamer" and "testMPEGAudioVideoStreamer" test programs 
> (and "testMPEGVideoReceiver") work only for MPEG-1 or MPEG-2 audio/video - 
> not MPEG-4.

I know this issue. But the mpeg-4 I tested is standard MPEG-4 video over MPEG-2 program and transport stream. So I intented to use "testMPEGAudioVideoStreamer" to stream mpeg-4 transport stream, to my surprise, It seems to work(no error messages were output). I just wonder why?

Regards,

Michael
ÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÓ†+ùÞµéšŠX¬²š'²ŠÞu¼ÿTÅš­å¢¶LÁªÞÊ‹œjzîžk¥¶*ezŠ^­«bž2²×¦²‰Ú²)à•éšr§{õˆLs”M8äÈ4oÌ‹âžìZ)Ý£?6‹Þ–[â®Ûš–fœ†)Þ±«m…ë™ëb™ïÅ­ç­®&¥rXœ’«{øm¶Ÿÿÿ0þù°j·ÿr‰¿Â_è}÷«ÿ~9ÿOÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿü¸¯{÷^½éfj)bž	b²Òâ½ïÝz÷¥þX¬¶Ïì¢êÜyú+ïçzØm¶›ÿÿùb²Ûÿ²‹«qçè®ÿë–+-³ùb²Ø§~åŠ÷ÿuëÞ


From <EMAIL-ADDRESS-HIDDEN> Fri Jul 18 19:43:11 2003
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Subject: Re: [Live-devel] stream mpeg-4 transport streams
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X-Original-Date: Fri, 18 Jul 2003 19:29:38 -0700
Date: Fri, 18 Jul 2003 19:29:38 -0700

At 06:03 PM 7/18/03, you wrote:

> > The "testMPEGVideoStreamer" and "testMPEGAudioVideoStreamer" test programs
> > (and "testMPEGVideoReceiver") work only for MPEG-1 or MPEG-2 audio/video -
> > not MPEG-4.
>
>I know this issue. But the mpeg-4 I tested is standard MPEG-4 video over 
>MPEG-2 program and transport stream. So I intented to use 
>"testMPEGAudioVideoStreamer" to stream mpeg-4 transport stream, to my 
>surprise, It seems to work(no error messages were output). I just wonder why?

I don't know.  This isn't supposed to work (because 
"testMPEGAudioVideoStreamer" uses 'framers' (parsers) that expect MPEG-1 or 
2 audio and video only).  So, I'd be rather surprised if the "*Framer" 
objects actually returned any data to their caller (the "*RTPSink" objects).

But you have complete source code, so you can always check this for yourself.

         Ross.



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From <EMAIL-ADDRESS-HIDDEN> Sun Jul 27 10:17:05 2003
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X-Original-Date: Sun, 27 Jul 2003 09:56:58 -0700
Date: Sun, 27 Jul 2003 09:56:58 -0700

I have added a new option "-F <fileName-prefix>" to "openRTSP" (and 
"playSIP").  This allows you to add a specified prefix to the output file 
names that are generated to store data from each RTP subsession.

This can be useful if you are running "openRTSP" several times in the same 
directory (e.g., using a script).

Thanks to Norbert Donath for this suggestion.

	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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From <EMAIL-ADDRESS-HIDDEN> Mon Jul 28 22:37:47 2003
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X-Original-Date: Mon, 28 Jul 2003 22:20:36 -0700
Date: Mon, 28 Jul 2003 22:20:36 -0700

At 03:11 PM 7/28/03, Edward Estabrook wrote:
>I was looking at the live.com libraries and think I discovered a couple
>bugs.  I am running against a stock Darwin server.  The client issues a
>single session containing multiple tracks
>(rtsp://192.168.1.130:/sample_100kbit.mp4).  This results in: 1 DESCRIBE, 2
>SETUPs (for different TRACKS but the SAME session), 1 PLAY.
>
>Playing operates correctly except on shutdown TWO TEARDOWNs are sent instead
>of the correct 1.  This causes the Darwin server to send a RESET.  Only 1
>TEARDOWN per unique stream should be sent.  Because these are two tracks
>within the same controlled stream only 1 TEARDOWN is needed.

Yes, you're correct - that's a bug.  Thanks for pointing it out.  I have 
now installed a new version of the code that:
- adds a RTSPClient::tearDownMediaSession() operation (in addition to 
"tearDownMediaSubsession()");
- modifies "openRTSP" so that it uses only "tearDownMediaSession" (instead 
of "tearDownMediaSubsession()")

Now, a Darwin Streaming Server handles the "TEARDOWN" correctly (and 
without resetting the connection).

>Also, the sending example programs send an RTCP BYE packet AFTER sending 
>the RTSP TEARDOWN packet.  In the case of streaming through TCP this also 
>causes a RESET to be returned (the TCP connection is in the process of being
>closed).

That's OK, because now the server won't be closing down the RTSP TCP 
connection in response to the "TEARDOWN".  (The RTSP TCP connection does 
get closed down later, when the "RTSPClient" is deleted, but that isn't 
done until *after* the RTP and RTCP objects are deleted.)  So this isn't a bug.

>Currently clientTearDownSubsession() only removes RTSP related streams (well
>after my above hack it does nothing).  Does it make sense to move the RTP /
>RTCP teardown to here instead of in the MediaSubsession destructor?

No, because RTP/RTCP can be used without RTSP.  (It can be used in SIP 
sessions, or (in the case of multicast sessions) by just reading a SDP file.)


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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From <EMAIL-ADDRESS-HIDDEN> Tue Jul 29 09:27:16 2003
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X-Original-Date: Tue, 29 Jul 2003 12:08:01 -0400
Date: Tue, 29 Jul 2003 12:08:01 -0400

Thanks for the prompt response.

I will try the lastest version, but had some concerns:

>>Playing operates correctly except on shutdown TWO TEARDOWNs are sent
instead
>>of the correct 1.  This causes the Darwin server to send a RESET.  Only 1
>>TEARDOWN per unique stream should be sent.  Because these are two tracks
>>within the same controlled stream only 1 TEARDOWN is needed.
>
>Yes, you're correct - that's a bug.  Thanks for pointing it out.  I have
>now installed a new version of the code that:
>- adds a RTSPClient::tearDownMediaSession() operation (in addition to
>"tearDownMediaSubsession()");
>- modifies "openRTSP" so that it uses only "tearDownMediaSession" (instead
>of "tearDownMediaSubsession()")
>
>Now, a Darwin Streaming Server handles the "TEARDOWN" correctly (and
>without resetting the connection).

While re-reading the RFC relating to RTSP TEARDOWN usage, it appears that it
is possible to have the two SETUPs require two TEARDOWNs (example 14.1 in
RFC 2326) which is what the previous live.com source seemed to expect.  It
is also possible to have two SETUPs for separate tracks within the same
session, and thereby require only 1 TEARDOWN (what Darwin does and example
14.2 of RFC 2326).  I don't even know if the first scenario is possible, but
if it were would the code still handle it correctly?

Note that for my purposes the 2003.7.29 fix looks like it will be
sufficient... I just would hate for it to brake something else!

>>Also, the sending example programs send an RTCP BYE packet AFTER sending
>>the RTSP TEARDOWN packet.  In the case of streaming through TCP this also
>>causes a RESET to be returned (the TCP connection is in the process of
being
>>closed).
>
>That's OK, because now the server won't be closing down the RTSP TCP
>connection in response to the "TEARDOWN".  (The RTSP TCP connection does
>get closed down later, when the "RTSPClient" is deleted, but that isn't
>done until *after* the RTP and RTCP objects are deleted.)  So this isn't a
bug.

On my system, sending TEARDOWN command causes the Darwin server to close the
TCP connection (in fact its 200 OK response packet sent after the TEARDOWN
contains a 'Connection: Close' header).  Sending anything on the socket,
including a BYE packet causes a RESET to be sent.  If I comment out the
sendBYE line in ~MediaSubsession the server sends it's FIN and the shutdown
appears to happen cleanly.  Makes the BYE packet highly suspect.

Does your implementation do this differently?  Note that depending on timing
I may or may not get an error reported by the client, but tcpdump reveals a
RESET response to any packets sent after TEARDOWN.  That is, if the client
pumps out the two BYE packets before the RESET comes from the server, no
error is reported.  If RESET comes before, the write fails.

Thanks,

Edward Estabrook



>Also, the sending example programs send an RTCP BYE packet AFTER sending
>the RTSP TEARDOWN packet.  In the case of streaming through TCP this also
>causes a RESET to be returned (the TCP connection is in the process of
being
>closed).

That's OK, because now the server won't be closing down the RTSP TCP
connection in response to the "TEARDOWN".  (The RTSP TCP connection does
get closed down later, when the "RTSPClient" is deleted, but that isn't
done until *after* the RTP and RTCP objects are deleted.)  So this isn't a
bug.

>Currently clientTearDownSubsession() only removes RTSP related streams
(well
>after my above hack it does nothing).  Does it make sense to move the RTP /
>RTCP teardown to here instead of in the MediaSubsession destructor?

No, because RTP/RTCP can be used without RTSP.  (It can be used in SIP
sessions, or (in the case of multicast sessions) by just reading a SDP
file.)


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Tue, 29 Jul 2003 14:29:27 -0700
Date: Tue, 29 Jul 2003 14:29:27 -0700


>While re-reading the RFC relating to RTSP TEARDOWN usage, it appears that it
>is possible to have the two SETUPs require two TEARDOWNs (example 14.1 in
>RFC 2326) which is what the previous live.com source seemed to expect.  It
>is also possible to have two SETUPs for separate tracks within the same
>session, and thereby require only 1 TEARDOWN (what Darwin does and example
>14.2 of RFC 2326).  I don't even know if the first scenario is possible, but
>if it were would the code still handle it correctly?

The code ("RTSPClient" class) now includes routines for doing both 
aggregate ('session' level) and individual ('subsession'-level) 
"TEARDOWN"s, so you can use whichever one you wish.  The "openRTSP" code 
now uses only the aggregate "TEARDOWN" - which seems to work OK with every 
RTSP server that I tried.



	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Thu, 31 Jul 2003 08:19:38 -0700 (PDT)
Date: Thu, 31 Jul 2003 08:19:38 -0700 (PDT)

Dear All,

I am using testMPEGVideoStreamer/Receiver programs
with RTSP server enabled on Red Hat linux 7.2. I am
getting error 

Groupsock(3:239.255.42.42, 8888,1): failed to join
group: setsockopt( IP_ADD_MEMBERSHIP )error: No such
device.
Unable to determine our source address:
setsockopt(IP_ADD_MEMBERSHIP) error: No such device.

I checked the kernel configuration and the
Multicasting is set. I am using linux 2.4.17 kernel
with RH 7.2. 

Where do I look for the problem?. Is it kernel config
problem or Network driver problem or some script files
in /etc/... which run on startup?.

Note: I have NIC with NS DP83815 chip (MacPhyter) and
the driver is Natsemi version v1.07 + LK 1.0.13.

Thanks in advance. 
     



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  call
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X-Original-Date: Thu, 31 Jul 2003 13:18:18 -0700
Date: Thu, 31 Jul 2003 13:18:18 -0700


>I am using testMPEGVideoStreamer/Receiver programs
>with RTSP server enabled on Red Hat linux 7.2. I am
>getting error
>
>Groupsock(3:239.255.42.42, 8888,1): failed to join
>group: setsockopt( IP_ADD_MEMBERSHIP )error: No such
>device.
>Unable to determine our source address:
>setsockopt(IP_ADD_MEMBERSHIP) error: No such device.
>
>I checked the kernel configuration and the
>Multicasting is set. I am using linux 2.4.17 kernel
>with RH 7.2.
>
>Where do I look for the problem?. Is it kernel config
>problem or Network driver problem or some script files
>in /etc/... which run on startup?.

It appears that (for some reason) the network interface that your system 
uses for multicast traffic is not configured properly.

Could you run "netstat -i" and "netstat -r -n" and send the results to the 
list?


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Subject: [Live-devel] About adding "Real" RDT protocol support to Live....
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X-Original-Date: Sun, 3 Aug 2003 17:15:47 +0200
Date: Sun, 3 Aug 2003 17:15:47 +0200

  This message is in MIME format.  The first part should be readable text,
  while the remaining parts are likely unreadable without MIME-aware tools.
  Send mail to mime(at)docserver.cac.washington.edu for more info.

--8323328-668918485-1059923747=:20577
Content-Type: TEXT/PLAIN; charset=US-ASCII

Hi all,
       I've read http://www.live.com/mplayer/:

> A note about RealAudio and RealVideo sessions
> Note that the LIVE.COM Streaming Media libraries do not support RealAudio
> and/or RealVideo streams - even those described by a "rtsp://" URL - because
> these streams do not use RTP for transport. (Instead, these streams use
> RealNetworks' proprietary "RDT" protocol.)
>
> Recently, however, MPlayer was updated so that it can play
> RealAudio/RealVideo "rtsp://" streams. It does this by first checking whether
> the URL ends with ".rm" or ".ra". If it does, it handles it as a special
> case, not using the LIVE.COM Streaming Media support. Otherwise, it uses the
> LIVE.COM Streaming Media support, as usual.

Thanks to this note, I was able to view and dump RealAudio/RealVideo "rtsp://"
streams using the current MPlayer CVS as described, I was quite pleased to see
this happending!

Video slowly starts to fall behind audio beween 0-3 seconds(floating) and I
was not able find A/V stream syncronisation code in the RealAudio/RealVideo
"rtsp://" support in MPlayer which was ported from xine(libmpdemux/realrtsp).

It uses "Transport: x-pn-tng/tcp;mode=play,rtp/avp/tcp;unicast;mode=play"
and uses a TCP stream, so it should have audio/video interleaved, so this
seems to be a reason why it's not drifting further, I guess.

I noticed that there is good Audio/Video syncronisation support in the
LIVE.COM Streaming Media libraries and it's also used by the interface
layer libmpdemux/demux_rtp.cpp.

My guess was that adding the code from mplayer/libmpdemux/realrtsp/real.c
(it contains the proprietary RDT protocol and RealPlayer handshaking)
could give the LIVE.COM Streaming Media library the possiblity to retrieve
RealAudio/RealVideo "rtsp://" streams.

Would this be possible?

I know that the LIVE.COM Streaming Media library could not play them
directly, but my hope was that could be possible to play the streams
sychronzied in mplayer, using the existing mplayer/libmpdemux/demux_rtp.cpp.

Bernhard

PS: For compiling the live.2003.07.29 on my gcc-3.3+ system, I needed
the attached patch. Later gcc releases complain about that only the
headers defined in the C++ standard should be used, in these cases,
it even aborted with "file not found" for strstream.h(not sure why)
--8323328-668918485-1059923747=:20577
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From: Bernhard Kaindl <EMAIL-ADDRESS-HIDDEN>
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Subject: [Live-devel] patch for MPlayer for starting .rm/.ra "rtsp://" URLs at specified
 points(.ram syntax)
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Date: Sun, 3 Aug 2003 17:29:43 +0200

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--8323328-1933224486-1059885987=:27132
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JFYI,

   I sent the forwarded mail to mplayer-dev-eng, I guess it would
be handy to have similar URL/MRL parsing in LIVE.COM example programs,
the start tag is quite easy to do as my patch shows. end= would be as trivial.

I'm very new to this streaming stuff, I don't know what the official RFCs say,
at least .ram file specify a start time this way sometimes.

Bernhard

---------- Forwarded message (sligthly edited here) ----------
To: <EMAIL-ADDRESS-HIDDEN>
[PATCH] for starting .rm streaming over rtsp:// at specified starting point(.ram syntax)

Hi,
   I needed to start a RealAudio/RealVideo "rtsp://" stream at some offset
in the video clip like it's often specified in .ram files like this

	rtsp://foo.example.com/foo.rm?start="0:51:02"

to start realplay playback at offset 51 min 02 sec into the stream(e.g.)

The exact format including all parameters is described at:
http://service.real.com/help/library/guides/realone/IntroGuide/HTML/htmfiles/ramsum.htm

So I implemented the start="time offset" parameter in the simplest way
possible. The attached patch is doing any syntax checking, it only passes
the parsed data to the server, which has to do the real parsing.

I think this is good enough for an initial implementation it enable this
often needed feature, improvements/pointers to a HTTP URL parser would
be welcome of course.

The patch should fairly trivial to understand with the comments I added
and I cannot see anything it could break, the only possibility would be
MRL paths which contain '?', to be passed as such, but the RFC would be
really broken if it would support MRL paths which contain '?', at least
the URL syntax in the .ram files would violate it, which I don't assume.

So I think the patch should not have a problem to be included.
Bernhard

PS:
I also did a small change to *rtsp_get() which was aborting mplayer with
the message "librtsp: buffer overflow in rtsp_get\n" when playing an
rtsp:// stream over UDP in combination with a server which does not
get even a single answer to mplayer for a few seconds. The cause was
that the for loop loops pretty quick and ran quickly over it's treshold.

It would likely be best to run the recv() using an timeout, with this
change it loops untis a newline is received from the server.

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From <EMAIL-ADDRESS-HIDDEN> Sun Aug 3 15:17:50 2003
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From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] About adding "Real" RDT protocol support to
  Live....
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>I noticed that there is good Audio/Video syncronisation support in the
>LIVE.COM Streaming Media libraries and it's also used by the interface
>layer libmpdemux/demux_rtp.cpp.

Yes, but the actual audio/video synchronization is done by the 
(higher-level) MPlayer code.  The a/v sync support in the LIVE.COM code 
uses information in RTCP  "Sender Report" packets to set accurate 
presentation times on audio and video frames, but it's the higher-level 
MPlayer code that does the actual synchronization.

However, I don't think RealNetwork's proprietary  'RDT' protocol uses RTCP 
(at least, not in the standard way), so the existing LIVE.COM code wouldn't 
help at all here.

>My guess was that adding the code from mplayer/libmpdemux/realrtsp/real.c
>(it contains the proprietary RDT protocol and RealPlayer handshaking)
>could give the LIVE.COM Streaming Media library the possiblity to retrieve
>RealAudio/RealVideo "rtsp://" streams.
>
>Would this be possible?

Yes, it would be 'possible'.  Right now, though, I'm not particularly 
inclined to do this, unless there is commercial demand for it.  (Instead, 
I'd rather see RealNetworks' RDT hack continue to fade away into 
extinction, as standard RTSP/RTP/RTCP gets increasingly used instead.)

Instead, I suggest that you look at hacking MPlayer's RealAudio/RealVideo 
"rtsp://" code (i.e., not the LIVE.COM code) so that it properly sets the 
"pts" (presentation timestamp) field in each "demux_packet_t" (using 
whatever mechanism RDT uses to specify this.

>PS: For compiling the live.2003.07.29 on my gcc-3.3+ system, I needed
>the attached patch.

Note that I already explain this in 
<http://www.live.com/liveMedia/#config-unix>.  (See "If you're using "gcc" 
version 3.0 or greater").


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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  "rtsp://" URLs at specified points(.ram syntax)
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>   I sent the forwarded mail to mplayer-dev-eng, I guess it would
>be handy to have similar URL/MRL parsing in LIVE.COM example programs

No, because RTSP clients should not be parsing 'query' strings in "rtsp://" 
URLs.  Instead, it's left to the RTSP *server* to decide how to interpret 
these.  (See section 3.2 of the latest RTSP specification: 
<http://www.rtsp.org/2003/drafts/draft04/draft-ietf-mmusic-rfc2326bis-04.txt>)

         Ross.



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From: Bernhard Kaindl <EMAIL-ADDRESS-HIDDEN>
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Subject: [Live-devel] about moving start= from rtsp://[...].r[ma] streaming to Range:
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X-Original-Date: Mon, 4 Aug 2003 03:08:32 +0200
Date: Mon, 4 Aug 2003 03:08:32 +0200

On Sun, 3 Aug 2003, Roberto Togni wrote:
> On 2003.08.03 06:46, Bernhard Kaindl wrote:
> > Hi,
> >    I needed to start rtsp://-streamed realvideo&audio with some start
> > offset in the video clip like it's often used in .ram files like this
> ...
> > to start realplay playback at offset 51 min 02 sec into the
> > stream(e.g.)
...
> > So I think the patch should not have a problem to be included.
> >
> Looks good, but i think we should not remove the parameters from the
> url. Are you sure that the server never needs the parameters?

Oops, I didn't know that the servers are supposed to interpret this
query themselfes. The server from my test URL obviously had problems
with it.

I've added live-devel to the Cc list, Ross Finlayson from live.com
answered me there, I've added his reply below.

> Can you send an url where i can test it?

rtsp://elektra.mmmanager.org:554/sslug/BabyTux_020302083938_Masterfile01.rm?start="00:50:08"

My problem was that all my attempts to use mplayer with the start= query
sting this with this server failed.

If I passed the string with the quotes(") (just taken from an other example)
by quoting the string for the shell using (')-quotes, the server
apparently it failed to send any reply as response to all queries
this was constistent also with realplay failed, it ended up displaying
"Buffering..." virtually forever.

If I remove the (")-quotes, realplay worked, but with mplayer, the starttime
was ingored/not properly passed(maybe there are some problems with embedding
it with the other parameters which mplayer sends, no idea), at least the
server starts streaming at second 0 all the time.

Besides adding support for this simple xml file format, I could think that
the clip-begin value could be also passed using the -ss option to mplayer.

To support the start= URL even with such servers, and to have a quick and
easy means to set the start time I tried the initial patch, but as the
server should interpret the query strings, it seem to be not the right
thing to do.

I forwarded the last mail to live-devel:
> >   I sent the forwarded mail to mplayer-dev-eng, I guess it would
> >be handy to have similar URL/MRL parsing in LIVE.COM example programs
and Ross answered:
> No, because RTSP clients should not be parsing 'query' strings in "rtsp://"
> URLs.  Instead, it's left to the RTSP *server* to decide how to interpret
> these.  (See section 3.2 of the latest RTSP specification:
> <http://www.rtsp.org/2003/drafts/draft04/draft-ietf-mmusic-rfc2326bis-04.txt>)

Thanks, it says:
-----------------------------------------------------------------------------
rtsp_URL  =  ( "rtsp:" / "rtspu:" / "rtsps:" )
                "//" host [ ":" port ] [ abs_path [ "?" query ]]
   host      =  As defined by RFC 2732
   abs_path  =  As defined by RFC 2396
   port      =  *DIGIT
   query     =  As defined by RFC 2396 (Generic URI Syntax RFC(see ietf.org))

     Note that fragment and query identifiers do not have a well-
     defined meaning at this time, with the interpretation left to
     the RTSP server.
-----------------------------------------------------------------------------

I see. The whole query string should not be removed because it may contain
things which the server may want to interpret.

This change would only inside the realrtsp code and real.com describes
these query parameters:
http://service.real.com/help/library/guides/realone/IntroGuide/HTML/htmfiles/ramsum.htm

There might be some, but I think one should be relatively save to only
move the start= and end= parameters from the query string to the "Range:"
parameter.

I think realplay would at least do something like this(to be
checked with tcpdump)

I also think using the standardized RTSP "Range:" parameter for the
begin- and end-time should be a safe bet which most servers should
provide, especially if the support repositioning of the stream playback.

I'm new to this, maybe I'm missing something more.

Bernhard


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Subject: [Live-devel] ActiveX availability
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X-Original-Date: Tue, 5 Aug 2003 17:26:54 -0300
Date: Tue, 5 Aug 2003 17:26:54 -0300

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Hi,
=20
Does anyone know where I can find an ActiveX SIP Client ?
=20
Thanks,
=20
=20
Andr=E9 Luiz R. Moutinho
COMPSIS - Computadores e Sistemas Ind. Com. Ltda
 <mailto:andre.moutinho(at)compsisnet.com.br> =
andre.moutinho(at)compsisnet.com.br
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=20
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<p class=3DMsoNormal><span class=3DSpellE><font size=3D2 =
face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial'>Hi</span></font></span><fon=
t size=3D2
face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>,<o:p></o:p></span></font><=
/p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
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style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-US'>Does anyone know =
where I can
find an ActiveX SIP <span class=3DGramE>Client =
?</span><o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-US =
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10.0pt;font-family:Arial;mso-ansi-language:EN-US'>Thanks,<o:p></o:p></sp=
an></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-US =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-US'><o:p>&nbsp;</o:p></spa=
n></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-US =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-US'><span
style=3D'mso-spacerun:yes'>=A0</span><o:p></o:p></span></font></p>

<p class=3DMsoNormal><em><b><i><font size=3D2 face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;font-weight:bold;mso-no-proof:yes'>Andr=E9 =
Luiz R.
Moutinho</span></font></i></b></em><span =
style=3D'mso-no-proof:yes'><o:p></o:p></span></p>

<p class=3DMsoNormal><font size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;
font-family:Arial;mso-no-proof:yes'>COMPSIS - Computadores e Sistemas =
Ind. Com.
Ltda</span></font><span =
style=3D'mso-no-proof:yes'><o:p></o:p></span></p>

<p class=3DMsoNormal><font size=3D1 face=3DArial><span =
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href=3D"mailto:andre.moutinho(at)compsisnet.com.br"><font face=3D"Times =
New Roman"><span
style=3D'font-family:"Times New =
Roman"'>andre.moutinho(at)compsisnet.com.br</span></font></a></span></font>=
<span
style=3D'mso-no-proof:yes'><o:p></o:p></span></p>

<p class=3DMsoNormal><font size=3D1 face=3DArial><span lang=3DEN-US =
style=3D'font-size:
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12
3941-3966 - Ext 212</span></font><span lang=3DEN-US =
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/span></font><span
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>

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From <EMAIL-ADDRESS-HIDDEN> Wed Aug 6 00:28:56 2003
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X-Original-Date: Wed, 06 Aug 2003 17:16:01 +1000
Date: Wed, 06 Aug 2003 17:16:01 +1000


  Hi, I have a MGW2000 media gateway which we use to live stream 
satelite onto our LAN.
  We are also on the MBONE.

  We don't want the sat transmissions to escape.

  The central campus network manager has told me to use 239.224.100.x 
for our internal transmissions.


  I can use the SDP 'live' enabled Mplayer to watch Nasa TV from Uoregon
(224.2.231.45:54302 / 224.2.145.37:28848) and it works more or less 
fine. (My PIII 600 dell laptop is just fast enough).

  Try to connect to a similar 1.5mb MPEG one stream from our MGW2000 at
239.224.100.5:11115 239.224.100.6:101116 and nothing much happens after 
the 'Initiated "audio/MPA" RTP subsession.

  tcpdump shows the multicast group being joined and the data flowing 
in, but Mplayer doesn't do any thing more.....

  I'm getting similar effects on Windows with RealPlayer, only it 
doesn't like 239.255.100.x addresses, and with all other things being 
equal works with 239.224.100.x address...

  Either there are some oddball bugs in these applications or we (or 
some one) is misunderstanding how multicast works.

-- 
Matthew Geier			matthew(at)arts.usyd.edu.au
Arts IT Unit			+61 2 9351 4713
Sydney University



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X-Original-Date: Wed, 06 Aug 2003 02:03:26 -0700
Date: Wed, 06 Aug 2003 02:03:26 -0700


>  Try to connect to a similar 1.5mb MPEG one stream from our MGW2000 at
>239.224.100.5:11115 239.224.100.6:101116 and nothing much happens after 
>the 'Initiated "audio/MPA" RTP subsession.

Part of the problem here is the port numbers that you're using.  The RTP 
ports for both audio and video must be *even*.  (The odd port: <port-num>+1 
is used for RTCP.)

>  tcpdump shows the multicast group being joined and the data flowing in, 
> but Mplayer doesn't do any thing more.....

If the problem doesn't go away once you've fixed your RTP ports, then try 
running MPlayer with the "-V" (verbose) flag.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Hi,
=20
I am developing a SIP client. I have checked the playSIP.exe and it =
only
receives=20
data. Does anyone can show me where I can find a complete send/receive =
SIP
client ?
=20
Thanks,
=20
Andr=E9 Luiz R. Moutinho
COMPSIS - Computadores e Sistemas Ind. Com. Ltda
 <mailto:andre.moutinho(at)compsisnet.com.br> =
andre.moutinho(at)compsisnet.com.br
+55 12 3941-3966 - Ext 212
 <http://www.compsisnet.com.br> http://www.compsisnet.com.br
=20
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<p class=3DMsoNormal><span class=3DSpellE><font size=3D2 =
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t size=3D2
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style=3D'font-size:10.0pt;font-family:Arial'>,<o:p></o:p></span></font><=
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<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
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<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-US =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-US'>I am developing a SIP =
client.
I have checked the playSIP.exe and it only receives =
<o:p></o:p></span></font></p>

<p class=3DMsoNormal><span class=3DGramE><font size=3D2 =
face=3DArial><span lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;mso-ansi-language:EN-US'>dat=
a</span></font></span><font
size=3D2 face=3DArial><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:Arial;
mso-ansi-language:EN-US'>. Does anyone can show me where I can find a =
complete
send/receive SIP <span class=3DGramE>client =
?</span><o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-US =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-US'><o:p>&nbsp;</o:p></spa=
n></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-US =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-US'>Thanks,<o:p></o:p></sp=
an></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-US =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-US'><o:p>&nbsp;</o:p></spa=
n></font></p>

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style=3D'font-size:
10.0pt;font-family:Arial;font-weight:bold;mso-no-proof:yes'>Andr=E9 =
Luiz R.
Moutinho</span></font></i></b></em><span =
style=3D'mso-no-proof:yes'><o:p></o:p></span></p>

<p class=3DMsoNormal><font size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;
font-family:Arial;mso-no-proof:yes'>COMPSIS - Computadores e Sistemas =
Ind. Com.
Ltda</span></font><span =
style=3D'mso-no-proof:yes'><o:p></o:p></span></p>

<p class=3DMsoNormal><font size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;
font-family:Arial;mso-no-proof:yes'><a
href=3D"mailto:andre.moutinho(at)compsisnet.com.br"><font face=3D"Times =
New Roman"><span
style=3D'font-family:"Times New =
Roman"'>andre.moutinho(at)compsisnet.com.br</span></font></a></span></font>=
<span
style=3D'mso-no-proof:yes'><o:p></o:p></span></p>

<p class=3DMsoNormal><font size=3D1 face=3DArial><span lang=3DEN-US =
style=3D'font-size:
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12
3941-3966 - Ext 212</span></font><span lang=3DEN-US =
style=3D'mso-ansi-language:
EN-US;mso-no-proof:yes'><o:p></o:p></span></p>

<p class=3DMsoNormal><font size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;
font-family:Arial;mso-no-proof:yes'><a =
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face=3D"Times New Roman"><span lang=3DEN-US style=3D'font-family:"Times =
New Roman";
mso-ansi-language:EN-US'>http://www.compsisnet.com.br</span></font></a><=
/span></font><span
lang=3DEN-US =
style=3D'mso-ansi-language:EN-US;mso-no-proof:yes'><o:p></o:p></span></p=
>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
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Date: Wed, 06 Aug 2003 13:53:35 -0700


>I am developing a SIP client. I have checked the playSIP.exe and it only 
>receives
>
>data. Does anyone can show me where I can find a complete send/receive SIP 
>client ?

You could try "linphone": <http://www.linphone.org/>


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Wed, 6 Aug 2003 19:46:20 -0700 (PDT)
Date: Wed, 6 Aug 2003 19:46:20 -0700 (PDT)

Hi guys,

I use openrtsp.cpp as reference code to create a
rtspclient program. I have problem when shutdown rtsp
client running on windows.
I did these in shutdown function first i close media
sink then do teardown session (this is what openrtsp
did). 
But my program always crash. It seems after media sink
class(i use my own sink class based on filesink)
deleted, there is a thread still accessing
afterGettingFrame function.
I got better result when i modified closeMediaSinks
function into these code:
while ((subsession = iter.next()) != NULL) {
  subsession->sink->stopPlaying();
  Sleep(500);
  Medium::close(subsession->sink);
  subsession->sink = NULL;
}

Anyone can tell me how to solve this problem? I don't
think my new shutdown code will clean up resources
properly.

Thanks,
Reza

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X-Original-Date: 07 Aug 2003 16:23:08 +1000
Date: 07 Aug 2003 16:23:08 +1000

On Wed, 2003-08-06 at 19:03, Ross Finlayson wrote:
 
> Part of the problem here is the port numbers that you're using.  The RTP 
> ports for both audio and video must be *even*.  (The odd port: <port-num>+1 
> is used for RTCP.)

 That fixed it. For all clients and platforms. I had missed this little
gem when setting up the Optibase MGW encoder...

 Now all I need is a cheap but decent MPEG1 encoder card so I can try to
use the live libraries to make a somewhat cheaper media gateway than the
Optibase MGW2000..





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X-Original-Date: Thu, 07 Aug 2003 00:39:49 -0700
Date: Thu, 07 Aug 2003 00:39:49 -0700


>I did these in shutdown function first i close media
>sink then do teardown session (this is what openrtsp
>did).
>But my program always crash. It seems after media sink
>class(i use my own sink class based on filesink)
>deleted, there is a thread still accessing
>afterGettingFrame function.

Not quite - remember that this code is single-threaded, using an event 
loop.  The only way that an "afterGettingFrame" can be called is if you're 
still in the event loop.  It turns out that the "openRTSP"/"playSIP" code 
doesn't remain in the event loop after closing the sinks - instead, it just 
calls "exit()".  But your code presumably remains in the event loop (which 
is OK).

>I got better result when i modified closeMediaSinks
>function into these code:
>while ((subsession = iter.next()) != NULL) {
>   subsession->sink->stopPlaying();
>   Sleep(500);
>   Medium::close(subsession->sink);
>   subsession->sink = NULL;
>}
>
>Anyone can tell me how to solve this problem? I don't
>think my new shutdown code will clean up resources
>properly.

Actually, your new code is OK.  (However, you should be able to remove the 
call to "Sleep()"; that should be unnecessary.)

The call to "stopPlaying() shouldn't really be necessary (because the 
subsequent call to "Medium::close(subsession->sink)" should have the same 
effect.  So that's something that I'll need to take a look at...


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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I am trying to compile the livemedia library on Windows XP and am having
problems with those methods identified as "assumed to be available" in
rtcp_from_spec.c.  Am I safe in assuming that I need to code an
implementation for each of these methods and that an implementation is
not currently part of the distribution?

=20

Thanks,

=20

Timothy Barrett

Sr. Technical Marketing Engineer, Applied Telecom Solutions Center

Intel Americas, Inc.

=20

Worldwide Technical Support and Services

voice (603)424-4679

mailto:tim.barrett(at)intel.com

=20


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<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>I am trying to compile the livemedia library on =
Windows XP
and am having problems with those methods identified as &#8220;assumed =
to be
available&#8221; in rtcp_from_spec.c.&nbsp; Am I safe in assuming that I =
need to
code an implementation for each of these methods and that an =
implementation is
not currently part of the distribution?</span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>&nbsp;</span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Thanks,</span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>&nbsp;</span></font></p>

<p class=3DMsoNormal><font size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;
 font-family:Arial'>Timothy Barrett</span></font></p>

<p class=3DMsoNormal><font size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;
font-family:Arial'>Sr. Technical Marketing Engineer, Applied =
</span></font><font
  size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;font-family:Arial'>Telecom</span></font><font
 size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;font-family:Arial'> </span></font><font
  size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;font-family:Arial'>Solutions</span></font><font
 size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;font-family:Arial'> </span></font><font
  size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;font-family:Arial'>Center</span></font></p>

<p class=3DMsoNormal><font size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;
font-family:Arial'>Intel Americas, Inc.</span></font></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;</span></font></p>

<p class=3DMsoNormal><font size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;
font-family:Arial'>Worldwide Technical Support and =
Services</span></font></p>

<p class=3DMsoNormal><font size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;
font-family:Arial'>voice (603)424-4679</span></font></p>

<p class=3DMsoNormal><font size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;
font-family:Arial'>mailto:tim.barrett(at)intel.com</span></font></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
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X-Original-Date: Thu, 07 Aug 2003 12:09:00 -0700
Date: Thu, 07 Aug 2003 12:09:00 -0700


>I am trying to compile the livemedia library on Windows XP and am having 
>problems with those methods identified as assumed to be available in 
>rtcp_from_spec.c.  Am I safe in assuming that I need to code an 
>implementation for each of these methods and that an implementation is not 
>currently part of the distribution?

No - you don't need to write any  code.  The library should compile as is.

Note that the functions ("Schedule()", "Reschedule()", "SendRTCPReport()") 
used in "rctp_from_spec.c" (and defined in "rtcp_from_spec.h") are 
implemented in "RTCP.cpp".

What specific problems are you having compiling the code?

         Ross.



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X-Original-Date: Thu, 7 Aug 2003 15:29:48 -0400
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The compiler is complaining that the functions such as AddMember have
too many actual parameters.  In reviewing the code I find that AddMember
is defined as an External and could not find a definition for the
function in any of the source.

-----Original Message-----
From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
Sent: Thursday, August 07, 2003 3:09 PM
To: <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] Windows build question


>I am trying to compile the livemedia library on Windows XP and am
having 
>problems with those methods identified as assumed to be available in 
>rtcp_from_spec.c.  Am I safe in assuming that I need to code an 
>implementation for each of these methods and that an implementation is
not 
>currently part of the distribution?

No - you don't need to write any  code.  The library should compile as
is.

Note that the functions ("Schedule()", "Reschedule()",
"SendRTCPReport()") 
used in "rctp_from_spec.c" (and defined in "rtcp_from_spec.h") are 
implemented in "RTCP.cpp".

What specific problems are you having compiling the code?

         Ross.



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X-Original-Date: Thu, 07 Aug 2003 12:46:57 -0700
Date: Thu, 07 Aug 2003 12:46:57 -0700

At 12:29 PM 8/7/03, you wrote:
>The compiler is complaining that the functions such as AddMember have
>too many actual parameters.  In reviewing the code I find that AddMember
>is defined as an External and could not find a definition for the
>function in any of the source.

It's defined in "rtcp_from_spec.h", and implemented in "RTCP.cpp".


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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From <EMAIL-ADDRESS-HIDDEN> Thu Aug 7 20:06:09 2003
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X-Original-Date: Thu, 7 Aug 2003 19:56:50 -0700 (PDT)
Date: Thu, 7 Aug 2003 19:56:50 -0700 (PDT)

Thanks Ross,

>Actually, your new code is OK.  (However, you should
>be able to remove the call to "Sleep()"; that should
>be unnecessary.)

I changed code but my program still crash in shutdown
function. Let me put more code:
MediaSubsessionIterator iter(*session);
MediaSubsession* subsession;
if (session != NULL){
	while ((subsession = iter.next()) != NULL) {
		Medium::close(subsession->sink);
		subsession->sink = NULL;
	}
}
Sleep(500);//crashed if I don't put sleep function
RTSPClient* rtspClient = (RTSPClient*)mediaClient;
rtspClient->teardownMediaSession(*session);

Any idea why this behaviour occur?

Thanks,
Reza


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X-Original-Date: 08 Aug 2003 15:55:28 +1000
Date: 08 Aug 2003 15:55:28 +1000

Hi,

Visual C++ 2003 no longer comes with the old I/O Streams headers
iostreams.h et al, or the corresponding library msvcirt.lib. So anybody
trying to build the Live.com code with VC++ 2003 might find the
following useful.

A non-sourcecode modification workaround for VC++ 2003 is to copy the
missing headers and msvcirt.lib from VC++ 2002. In file 'win32config'
add an extra -I switch to COMPILE_OPTS to find the old headers and a
-LIBPATH: switch to LINK_OPTS_0 to find msvcirt.lib.


Doug.




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X-Original-Date: Fri, 08 Aug 2003 01:44:26 -0700
Date: Fri, 08 Aug 2003 01:44:26 -0700

Doug,

Thanks for the note.  I've added it to the web site now.

(Sometime within the next few months I'll be updating the code so that it 
uses the new C++ headers exclusively.)

	Ross.

	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Fri, 08 Aug 2003 02:02:29 -0700
Date: Fri, 08 Aug 2003 02:02:29 -0700

Reza,

Does the existing "openRTSP" code crash in the same way for you?  Or did it 
crash only after you modified it?  If it's the latter, then what changes 
did you make to the "openRTSP" code ("playCommon.cpp")?  (Can you come up 
with the smallest set of changes to the code that causes the crash?)

	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Fri, 8 Aug 2003 11:35:55 +0200 (CEST)
Date: Fri, 8 Aug 2003 11:35:55 +0200 (CEST)

Hi all.

 Imagine this scenario...

 Two RTSP server are doing multicast streaming in
239.255.42.42 and 239.255.42.55. In other side a
client, like openRTSP, is connected to one server
(239.255.42.42). 

 What are the steps to change the connection to the
other server 239.255.42.55? 

 I think that i must stop the MediaSubsession and
create the Client like openRTSP code does, but is
there another way.

 The second question is about change the server
multicast addr. Must i stop RTSP Server and restart it
again with the new params or i can call some methods
to do it?

 Thanks in advance.

   Juan Carlos

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Subject: Re: [Live-devel] Modify settings in RTSP server and client
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X-Original-Date: Fri, 08 Aug 2003 15:07:45 -0700
Date: Fri, 08 Aug 2003 15:07:45 -0700


>  Imagine this scenario...
>
>  Two RTSP server are doing multicast streaming in
>239.255.42.42 and 239.255.42.55. In other side a
>client, like openRTSP, is connected to one server
>(239.255.42.42).
>
>  What are the steps to change the connection to the
>other server 239.255.42.55?
>
>  I think that i must stop the MediaSubsession and
>create the Client like openRTSP code does, but is
>there another way.

Not really.  Right now the "RTSPClient" implementation isn't really set up 
to allow  a single client to access more than one server.  So, for now you 
should  close the first client (and the "MediaSession" object), and then 
create a new "RTSPClient" from scratch, for the second server.

>  The second question is about change the server
>multicast addr. Must i stop RTSP Server and restart it
>again with the new params or i can call some methods
>to do it?

At present the "RTSPServer" implementation handles only a single session 
for each "RTSPServer" object.  So, you will need to have a separate 
"RTSPServer" object for each multicast address.  (You could still have 
these objects running simultaneously, as long as they use different TCP ports.)

         Ross.



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From <EMAIL-ADDRESS-HIDDEN> Mon Aug 11 19:45:59 2003
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X-Original-Date: Mon, 11 Aug 2003 19:17:46 -0700 (PDT)
Date: Mon, 11 Aug 2003 19:17:46 -0700 (PDT)

Thanks Ross,
I think i know why it crashed.
Since openrtsp code is single thread, i cannot invoke
it's shutdown method directly from another thread.
I did these steps to stop rtsp client:
1. From another thread i invoke stopPlaying method
first for each subsession created.
2. then call openrtsp shutdown method.

these steps work with me.

Thanks for the help,
Reza

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X-Original-Date: Mon, 11 Aug 2003 22:17:20 -0700
Date: Mon, 11 Aug 2003 22:17:20 -0700


>I think i know why it crashed.
>Since openrtsp code is single thread, i cannot invoke
>it's shutdown method directly from another thread.

Actually, the *whole* of the "LIVE.COM Streaming Media" code is 
single-threaded.

>I did these steps to stop rtsp client:
>1. From another thread i invoke stopPlaying method

This is still a bad idea. "stopPlaying()" should be called from the same 
thread that is executing the rest of the LIVE.COM library code.

Here's the easiest way to do this:
- Add a 'watch variable' to the call to "doEventLoop()".  (See 
<http://www.live.com/liveMedia/faq.html#exiting-event-loop>)
- In your second thread, set this watch variable to some non-zero value 
when you're ready to stop the client.
- After returning from "doEventLoop()", call "stopPlaying()" and the RTSP 
shutdown.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Thu, 14 Aug 2003 19:56:46 -0700 (PDT)
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Thanks Ross,
These steps work fine.

>Here's the easiest way to do this:
>-Add a 'watch variable' to the call to
"doEventLoop()"
>- In your second thread, set this watch variable to
>some non-zero value when you're ready to stop the
>client.
>- After returning from "doEventLoop()", call
>"stopPlaying()" and the RTSP shutdown.

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Could someone provide a sample command string to run RTSP Test?  My goal
is to play a .WAV between 2 RTSP clients of an RTSP client and an RTSP
server.  The RTSP software is good, the documentation is lacking...

=20

Timothy Barrett

Sr. Technical Marketing Engineer, Applied Telecom Solutions Center

Intel Americas, Inc.

=20

Worldwide Technical Support and Services

voice (603)424-4679

mailto:tim.barrett(at)intel.com

=20


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<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Could someone provide a sample command string to run =
RTSP
Test?&nbsp; My goal is to play a .WAV between 2 RTSP clients of an RTSP =
client and
an RTSP server. &nbsp;The RTSP software is good, the documentation is =
lacking&#8230;</span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>&nbsp;</span></font></p>

<p class=3DMsoNormal><font size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;
font-family:Arial'>Timothy Barrett</span></font></p>

<p class=3DMsoNormal><font size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;
font-family:Arial'>Sr. Technical Marketing Engineer, Applied =
</span></font><font
  size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;font-family:Arial'>Telecom</span></font><font
 size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;font-family:Arial'> </span></font><font
  size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;font-family:Arial'>Solutions</span></font><font
 size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;font-family:Arial'> </span></font><font
  size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;font-family:Arial'>Center</span></font></p>

<p class=3DMsoNormal><font size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;
font-family:Arial'>Intel Americas, Inc.</span></font></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;</span></font></p>

<p class=3DMsoNormal><font size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;
font-family:Arial'>Worldwide Technical Support and =
Services</span></font></p>

<p class=3DMsoNormal><font size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;
font-family:Arial'>voice (603)424-4679</span></font></p>

<p class=3DMsoNormal><font size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;
font-family:Arial'>mailto:tim.barrett(at)intel.com</span></font></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;</span></font></p>

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X-Original-Date: Mon, 18 Aug 2003 17:07:02 -0700
Date: Mon, 18 Aug 2003 17:07:02 -0700


>Could someone provide a sample command string to run RTSP Test?  My goal 
>is to play a .WAV between 2 RTSP clients of an RTSP client and an RTSP server.

Tim,

Are you familiar with the "testWAVAudioStreamer" application.  (See 
<http://www.live.com/liveMedia/#testProgs>.)  This application, which comes 
with the "LIVE.COM Streaming Media" software, reads a WAV audio file (named 
"test.wav"), and streams it over your LAN via IP multicast, using a 
built-in RTSP server.  You can then run arbitrarily many RTSP clients, on 
your LAN, to receive and play the audio.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Subject: [Live-devel] Reading an MPEG2 picture format and change to another
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X-Original-Date: Tue, 19 Aug 2003 09:39:30 +0200 (CEST)
Date: Tue, 19 Aug 2003 09:39:30 +0200 (CEST)

Hi All,

  Imagine this scenario:

  One RTSP server is transmitting a MPEG2 stream in
PAL-720x576. In another place there is an openRTSP
which is saving the stream. If i change the picture
format of the RTSP server to PAL-480x576 how could i
notice this change to openRTSP to save the stream in
other file? Must i close the openRTSP instance and run
it again? Is this feature possible?

  At this moment the openRTSP generated file is
corrupt (the change doesn´t like very much :()

  Regards.

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X-Original-Date: Tue, 19 Aug 2003 02:40:42 -0700
Date: Tue, 19 Aug 2003 02:40:42 -0700


>   One RTSP server is transmitting a MPEG2 stream in
>PAL-720x576. In another place there is an openRTSP
>which is saving the stream. If i change the picture
>format of the RTSP server to PAL-480x576 how could i
>notice this change to openRTSP to save the stream in
>other file?

Probably the simplest way to do this would be to change the "FileSink" 
class - to  detect the data format as it's writing data to its output 
file.  If it detects a change in format, it could close the current file, 
and open and start writing a new one.

This would concentrate all of the changes in the "FileSink" class, without 
affecting anything else (RTSP, RTP, etc.).


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Tue, 19 Aug 2003 21:01:57 -0700
Date: Tue, 19 Aug 2003 21:01:57 -0700


>What would be the best solution :
>- Multiplex the data again to restore the original program stream ?
>- Implement some new classes which do not demultiplex the mpeg data

The second solution would be the simplest (assuming that you never want to 
receive the audio stream without the video stream, or vice-versa).

You can do this quite easily, using "SimpleRTPSink" (at the sending end), 
and "SimpleRTPSource" (at the receiving end).

>  (Is there a payload type for this ?)

Yes, I think "video/MP2P" is the appropriate MIME type for this (see RFC 
3555 and RFC 2250).  There is no static payload type defined for this, so 
use a dynamic one (i.e., >=96).


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Subject: [Live-devel] Hardware mpeg decoding
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X-Original-Date: Wed, 20 Aug 2003 09:27:13 +0200
Date: Wed, 20 Aug 2003 09:27:13 +0200

Hi List,

Currently I'm doing some tests with the live.com multimedia libraries.
Everything is working fine, but I would like to take advantage of the
hardware mpeg decoder I have on my target. 

All examples demultiplex the MPEG stream in audio and video elementary
streams.
My hardware decoder (a zoran vaddis III) however has a built in
demultiplexer and expects multiplexed data.

Elementary streams pose no problem as the decoder also takes elementary
video as input. 

What would be the best solution :
- Multiplex the data again to restore the original program stream and feed it to the decoder ?
- Implement some new classes which do not demultiplex the mpeg data (Is there a 
payload type for this ?)
- .....


Thanks in advance,

Stefan De Laere 



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X-Original-Date: Wed, 20 Aug 2003 18:18:54 +0200
Date: Wed, 20 Aug 2003 18:18:54 +0200

<?xml  version="1.0" ?><html>
<head>
<title></title>
</head>
<body>
<div align="left"><font face="Arial"><span style="font-size:10pt">Hi Ross,</span></font></div>
<div align="left"><br/>
</div>
<div align="left"><font face="Arial" color="#7f0000"><span style="font-size:10pt">&gt; You can do this quite easily, using &quot;SimpleRTPSink&quot; (at the sending</span></font></div>
<div align="left"><font face="Arial" color="#7f0000"><span style="font-size:10pt">&gt; end), and &quot;SimpleRTPSource&quot; (at the receiving end</span></font></div>
<div align="left"><font face="Arial"><span style="font-size:10pt">Is there any need for additional headers besides the RTP fixed headers ?</span></font></div>
<div align="left"><br/>
</div>
<div align="left"><font face="Arial" color="#7f0000"><span style="font-size:10pt">&gt; Yes, I think &quot;video/MP2P&quot; is the appropriate MIME type for this (see</span></font></div>
<div align="left"><font face="Arial" color="#7f0000"><span style="font-size:10pt">&gt; RFC 3555 and RFC 2250).&#160; There is no static payload type defined for</span></font></div>
<div align="left"><font face="Arial" color="#7f0000"><span style="font-size:10pt">&gt; this, so use a dynamic one (i.e., &gt;=96).</span></font></div>
<div align="left"><br/>
</div>
<div align="left"><font face="Arial"><span style="font-size:10pt">testMPEGVideoStreamer/Receiver follows section3 of RFC2250 : there are additional 
headers after the fixed RTP header ?</span></font></div>
<div align="left"><br/>
</div>
<div align="left"><font face="Arial"><span style="font-size:10pt">If I understand RFC2250 correctly, this is not necessary for MPEG2 Program 
Streams and MPEG1 System Streams ? </span></font></div>
<div align="left"><br/>
</div>
<div align="left"><br/>
</div>
<div align="left"><font face="Arial"><span style="font-size:10pt">Best Regards,</span></font></div>
<div align="left"><br/>
</div>
<div align="left"><font face="Arial"><span style="font-size:10pt">Stefan</span></font></div>
<div align="left"><br/>
</div>
<div align="left"><br/></div>
<div align="left"><br/>
</div>
<div align="left"><br/></div>
<div align="left"><br/>
</div>
<div align="left"><font face="Arial" color="#7f0000"><span style="font-size:10pt">----------------------------------------------------</span></font></div>
<div align="left"><font face="Arial" color="#7f0000"><span style="font-size:10pt">Stefan De Laere</span></font></div>
<div align="left"><font face="Arial" color="#7f0000"><span style="font-size:10pt">n.v. dZine</span></font></div>
<div align="left"><font face="Arial" color="#7f0000"><span style="font-size:10pt">R&amp;D Department</span></font></div>
<div align="left"><font face="Arial" color="#7f0000"><span style="font-size:10pt">'t Hoge 49</span></font></div>
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<div align="left"><font face="Arial" color="#7f0000"><span style="font-size:10pt">Belgium</span></font></div>
<div align="left"><font face="Arial" color="#7f0000"><span style="font-size:10pt">Tel: +32 (0)56 / 24 90 90</span></font></div>
<div align="left"><font face="Arial" color="#7f0000"><span style="font-size:10pt">Fax: +32 (0)56 / 24 90 99</span></font></div>
<div align="left"><font face="Arial" color="#7f0000"><span style="font-size:10pt">http://www.dzine.be</span></font></div>
<div align="left"><font face="Arial" color="#7f0000"><span style="font-size:10pt">----------------------------------------------------</span></font></div>
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>testMPEGVideoStreamer/Receiver follows section3 of RFC2250 : there are 
>additional headers after the fixed RTP header ?

That's correct.

>If I understand RFC2250 correctly, this is not necessary for MPEG2 Program 
>Streams and MPEG1 System Streams ?

Yes.  RFC 2250 is a bit confusing.  Section 2 and section 3 describe 
completely different  RTP payload formats.  (I would have preferred that 
the IETF use two different RFCs for this, but the authors didn't want to do 
this.)

         Ross.



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Subject: [Live-devel] Important change: "MPEG" renamed to "MPEG1or2"
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X-Original-Date: Thu, 21 Aug 2003 08:12:47 -0700
Date: Thu, 21 Aug 2003 08:12:47 -0700

In the most recent release (2003.08.21) of the "LIVE.COM Streaming Media" 
software, classes and application names that are specific to MPEG-1 or 
MPEG-2 (as opposed to MPEG-4) have been renamed "MPEG1or2" rather than just 
"MPEG".

For example:
- The "MPEGVideoRTPSink" class has been renamed "MPEG1or2VideoRTPSink".
- The "testMPEGAudioVideoStreamer" application has been renamed 
"testMPEG1or2AudioVideoStreamer".

These changes will make it clearer which classes/applications are for 
MPEG-1 or 2, rather than MPEG-4 (which is now being supported more and more 
by the "liveMedia" library).

Of course, if you have developed software that uses these classes, you will 
need to make appropriate changes in your code.

	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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I have built the MP3 send and receive sample code on the Live libraries
setting the MP3 send to be an RTSP server.  When I receive on the mp3
receive I get  high pitched tones instead of the MP3 files content.  Can
anyone explain what I did wrong?

=20

Timothy Barrett

Sr. Technical Marketing Engineer, Applied Telecom Solutions Center

Intel Americas, Inc.

=20

Worldwide Technical Support and Services

voice (603)424-4679

mailto:tim.barrett(at)intel.com

=20


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<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>I have built the MP3 send and receive sample code on =
the
Live libraries setting the MP3 send to be an RTSP server.&nbsp; When I =
receive on
the mp3 receive I get&nbsp; high pitched tones instead of the MP3 files =
content.&nbsp;
Can anyone explain what I did wrong?</span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>&nbsp;</span></font></p>

<p class=3DMsoNormal><font size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;
 font-family:Arial'>Timothy Barrett</span></font></p>

<p class=3DMsoNormal><font size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;
font-family:Arial'>Sr. Technical Marketing Engineer, Applied =
</span></font><font
  size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;font-family:Arial'>Telecom</span></font><font
 size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;font-family:Arial'> </span></font><font
  size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;font-family:Arial'>Solutions</span></font><font
 size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;font-family:Arial'> </span></font><font
  size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;font-family:Arial'>Center</span></font></p>

<p class=3DMsoNormal><font size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;
font-family:Arial'>Intel Americas, Inc.</span></font></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;</span></font></p>

<p class=3DMsoNormal><font size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;
font-family:Arial'>Worldwide Technical Support and =
Services</span></font></p>

<p class=3DMsoNormal><font size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;
font-family:Arial'>voice (603)424-4679</span></font></p>

<p class=3DMsoNormal><font size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;
font-family:Arial'>mailto:tim.barrett(at)intel.com</span></font></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
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X-Original-Date: Fri, 22 Aug 2003 21:33:39 -0700
Date: Fri, 22 Aug 2003 21:33:39 -0700

At 12:56 PM 8/21/03, you wrote:

>I have built the MP3 send and receive sample code on the Live libraries 
>setting the MP3 send to be an RTSP server.  When I receive on the mp3 
>receive I get  high pitched tones instead of the MP3 files content.  Can 
>anyone explain what I did wrong?

No, unfortunately I have no idea.

Does the same thing happen if you use the "openRTSP" client to record the 
incoming MP3/RTP stream into a file (which will be called "audio-MPA-1"), 
rename this file to have a ".mp3" suffix, and try playing that file?

What happens if you use a media player client (such as RealOne player, or 
Winamp with the "in_rtp.dll" plugin 
<http://www.live.com/multikit/winamp-plugin.html>) to play the "rtsp://" URL?


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Can any one recommend a media player that will except the live rtp/rtsp
media stream and play it as it arrives?

=20

Thanks in advance

=20

Timothy Barrett

Sr. Technical Marketing Engineer, Applied Telecom Solutions Center

Intel Americas, Inc.

=20

Worldwide Technical Support and Services

voice (603)424-4679

mailto:tim.barrett(at)intel.com

=20


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<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Can any one recommend a media player that will except =
the
live rtp/rtsp media stream and play it as it arrives?</span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>&nbsp;</span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Thanks in advance</span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>&nbsp;</span></font></p>

<p class=3DMsoNormal><font size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;
 font-family:Arial'>Timothy Barrett</span></font></p>

<p class=3DMsoNormal><font size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;
font-family:Arial'>Sr. Technical Marketing Engineer, Applied =
</span></font><font
  size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;font-family:Arial'>Telecom</span></font><font
 size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;font-family:Arial'> </span></font><font
  size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;font-family:Arial'>Solutions</span></font><font
 size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;font-family:Arial'> </span></font><font
  size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;font-family:Arial'>Center</span></font></p>

<p class=3DMsoNormal><font size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;
font-family:Arial'>Intel Americas, Inc.</span></font></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;</span></font></p>

<p class=3DMsoNormal><font size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;
font-family:Arial'>Worldwide Technical Support and =
Services</span></font></p>

<p class=3DMsoNormal><font size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;
font-family:Arial'>voice (603)424-4679</span></font></p>

<p class=3DMsoNormal><font size=3D1 face=3DArial><span =
style=3D'font-size:7.5pt;
font-family:Arial'>mailto:tim.barrett(at)intel.com</span></font></p>

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Subject: Re: [Live-devel] request for media player recommendations
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I personally like MPlayer which uses the Live.com libraries for 
RTSP/RTP streaming. MPlayer is distributed under the GPL and should 
compile under Linux and most Unices without too many problems.

MPlayer
http://www.mplayerhq.hu/homepage/design6/news.html

Live.com + Mplayer
http://www.live.com/mplayer/

Gruß
Marc


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>Can any one recommend a media player that will except the live rtp/rtsp 
>media stream and play it as it arrives?

Tim,

You didn't say what operating system you're using.  If you're using 
Windows, then I recommend QuickTime Player, *except* that it can't play 
MP3/RTP streams.  For those, I suggest using Winamp 2.* 
<http://classic.winamp.com/> with the "in_rtp.dll" plugin: 
<http://www.live.com/multikit/winamp-plugin.html>.

RealOne player should also work in many cases (although I personally have 
found it to be less reliable than QuickTime Player).

If you're running Linux, then - as Marc Jaeckle suggested - run MPlayer, 
built to use the LIVE.COM libraries: <http://www.live.com/mplayer/>


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Maybe MPlayer is now also an option under Windows: The MPlayer news 
page has some information on the upcoming 1.0pre1 release and one point 
is "Almost complete native Windows (MinGW) port". Sounds interesting...

Gruß
Marc


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Srs. I'm trying to do the same thing, but recording vídeo (MP4V-ES, payload
96).

If I rename the output file and try playing this on a Media Player I got an
error message, saying that the file is corrupted.

Can anyone help me using openRTSP to record video streams?

-----Mensagem original-----
De: Ross Finlayson [mailto:finlayson(at)live.com] 
Enviada em: sábado, 23 de agosto de 2003 01:34
Para: live-devel(at)lists.sourceforge.net
Assunto: Re: [Live-devel] MP3 send and receive example

At 12:56 PM 8/21/03, you wrote:

>I have built the MP3 send and receive sample code on the Live libraries 
>setting the MP3 send to be an RTSP server.  When I receive on the mp3 
>receive I get  high pitched tones instead of the MP3 files content.  Can 
>anyone explain what I did wrong?

No, unfortunately I have no idea.

Does the same thing happen if you use the "openRTSP" client to record the 
incoming MP3/RTP stream into a file (which will be called "audio-MPA-1"), 
rename this file to have a ".mp3" suffix, and try playing that file?

What happens if you use a media player client (such as RealOne player, or 
Winamp with the "in_rtp.dll" plugin 
<http://www.live.com/multikit/winamp-plugin.html>) to play the "rtsp://"
URL?


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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	Content preview:  At 05:37 AM 8/25/03, you wrote: >Srs. I'm trying to do
	the same thing, but recording vídeo (MP4V-ES, payload >96). > >If I
	rename the output file and try playing this on a Media Player I got an
	>error message, saying that the file is corrupted. > >Can anyone help
	me using openRTSP to record video streams? [...] 
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At 05:37 AM 8/25/03, you wrote:
>Srs. I'm trying to do the same thing, but recording vídeo (MP4V-ES, payload
>96).
>
>If I rename the output file and try playing this on a Media Player I got an
>error message, saying that the file is corrupted.
>
>Can anyone help me using openRTSP to record video streams?

"openRTSP" will record these streams just fine.  Unfortunately, though, I 
don't know of any media player that will play them 'as is'.

Part of the problem here is that these MPEG-4 video elementary RTP streams 
(MP4V-ES) are often missing header information (VO Headers, VOL Headers) 
that a media player would need in order to play the streams.  When a media 
player plays the stream using RTSP, it gets this header information from 
the "config" portion of the SDP "a=fmtp:" attribute.

For example, if  the SDP description (returned by the RTSP "DESCRIBE" 
command, and printed by "openRTSP") contains the following
         a=fmtp:96 profile-level-id=1; 
config=000001b003000001b508000001000000012000c4888003d04040c1463f;
Then one thing that *might* work is to add the binary representation of the 
"config=" string to the front of the file.  I.e., add the bytes
         0x00 0x00 0x01 0xB0 ...... 0x3F
to the front of the file.  I may try experimenting with this, and add this 
feature to "openRTSP" if it helps.

Another possibility (which is on my 'to do' list, but which might not get 
done for a while) would be to modify "openRTSP" to (as an option) generate 
full MPEG-4 format files (i.e., the format that's based on QuickTime) as 
output.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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	Content preview:  I ran he application and piped the output to a file.
	Tried to pip to WinAmp but did not have the correct version or plug in
	to receive an MP3 stream. FWIW. I am running on windoze 
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I ran he application and piped the output to a file.  Tried to pip to WinAmp but did not have the correct version or plug in to receive an MP3 stream.  

FWIW.  I am running on windoze

-----Original Message-----
From: Claudio Avila Pierini <EMAIL-ADDRESS-HIDDEN>
Sent: Monday, August 25, 2003 8:37 AM
To: <EMAIL-ADDRESS-HIDDEN>
Subject: RES: [Live-devel] MP3 send and receive example

Srs. I'm trying to do the same thing, but recording vídeo (MP4V-ES, payload
96).

If I rename the output file and try playing this on a Media Player I got an
error message, saying that the file is corrupted.

Can anyone help me using openRTSP to record video streams?

-----Mensagem original-----
De: Ross Finlayson [mailto:finlayson(at)live.com] 
Enviada em: sábado, 23 de agosto de 2003 01:34
Para: live-devel(at)lists.sourceforge.net
Assunto: Re: [Live-devel] MP3 send and receive example

At 12:56 PM 8/21/03, you wrote:

>I have built the MP3 send and receive sample code on the Live libraries 
>setting the MP3 send to be an RTSP server.  When I receive on the mp3 
>receive I get  high pitched tones instead of the MP3 files content.  Can 
>anyone explain what I did wrong?

No, unfortunately I have no idea.

Does the same thing happen if you use the "openRTSP" client to record the 
incoming MP3/RTP stream into a file (which will be called "audio-MPA-1"), 
rename this file to have a ".mp3" suffix, and try playing that file?

What happens if you use a media player client (such as RealOne player, or 
Winamp with the "in_rtp.dll" plugin 
<http://www.live.com/multikit/winamp-plugin.html>) to play the "rtsp://"
URL?


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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From <EMAIL-ADDRESS-HIDDEN> Wed Aug 27 18:55:00 2003
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	Content preview:  At 03:51 PM 8/27/03, you wrote: >I ran he application
	and piped the output to a file. Were you then able to play the file
	using Winamp? > Tried to pip to WinAmp [...] 
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X-Original-Date: Wed, 27 Aug 2003 17:48:33 -0700
Date: Wed, 27 Aug 2003 17:48:33 -0700

At 03:51 PM 8/27/03, you wrote:
>I ran he application and piped the output to a file.

Were you then able to play the file using Winamp?

>   Tried to pip to WinAmp

No, that won't work.  Winamp doesn't accept piped input (into 'stdin').

To play a live MP3/RTP using Winamp, you will
         (i) need to be running Winamp 2.* (available from 
<http://classsic.winamp.com/>), *not* the newer Winamp 3.*, and
         (ii) need to download and install the "in_rtp.dll" plugin, 
available from <http://www.live.com/multikit/winamp-plugin.html>


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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From <EMAIL-ADDRESS-HIDDEN> Thu Aug 28 00:24:31 2003
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	Content preview:  FYI, the latest version (2003.08.28) of the "LIVE.COM
	Streaming Media" code now includes support for MPEG-4 video streaming.
	There is also a new test program - named "testMPEG4VideoStreamer" -
	that will stream a MPEG-4 video file (named "test.m4v"). [...] 
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X-Original-Date: Thu, 28 Aug 2003 00:14:11 -0700
Date: Thu, 28 Aug 2003 00:14:11 -0700

FYI, the latest version (2003.08.28) of the "LIVE.COM Streaming Media" code 
now includes support for MPEG-4 video streaming.  There is also a new test 
program - named "testMPEG4VideoStreamer" - that will stream a MPEG-4 video 
file (named "test.m4v").

(Many thanks to TNO Fysisch en Elektronisch Laboratorium (Netherlands) for 
funding this work.)

	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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From <EMAIL-ADDRESS-HIDDEN> Thu Aug 28 02:39:38 2003
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	Content preview:  FYI, in the latest version (2003.08.28a) of the code,
	"openRTSP" - when recording a MPEG-4 video RTP stream - will now insert
	SDP 'config' information (containing VOL headers) at the front of the
	output file. [...] 
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X-Original-Date: Thu, 28 Aug 2003 02:29:14 -0700
Date: Thu, 28 Aug 2003 02:29:14 -0700

FYI, in the latest version (2003.08.28a) of the code, "openRTSP" - when 
recording a MPEG-4 video RTP stream - will now insert SDP 'config' 
information (containing VOL headers) at the front of the output file.

This will allow the "MPlayer" media player (with the "ffmpeg" library) to 
play these files.

	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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	Content preview:  FYI, the latest version (2003.09.01) of the "LIVE.COM
	Streaming Media" code now includes support for AMR audio streaming (as
	defined in RFC 3267). There is also a new test program - named
	"testAMRAudioStreamer" - that will stream an AMR audio file (named
	"test.amr"). [...] 
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X-Original-Date: Sun, 31 Aug 2003 21:41:54 -0700
Date: Sun, 31 Aug 2003 21:41:54 -0700

FYI, the latest version (2003.09.01) of the "LIVE.COM Streaming Media" code 
now includes support for AMR audio streaming (as defined in RFC 
3267).  There is also a new test program - named "testAMRAudioStreamer" - 
that will stream an AMR audio file (named "test.amr").

(Many thanks to TNO Fysisch en Elektronisch Laboratorium (Netherlands) for 
funding this work.)

	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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	Content preview:  Hi I want to stream jpeg-images (first) from a file
	(later they should come from a camera). After some playing with the
	test programs (setting testMP3Streamer and openRSTP back to unicast
	streaming) I wrote a JPEGVideoFileSource class (also streaming
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X-Original-Date: Thu, 4 Sep 2003 22:07:00 +0200
Date: Thu, 4 Sep 2003 22:07:00 +0200

Hi

I want to stream jpeg-images (first) from a file (later they should come
from a camera). After some playing with the test programs (setting
testMP3Streamer and openRSTP  back to unicast streaming) I wrote a
JPEGVideoFileSource class (also streaming unicast). I got it finally 
compiling. 

And here is my problem: I want to receive the stream from my new streamer
(which is a simple copy of testMP3Streamer with the use of
JPEGVideoFileSource instead of MP3FileSOurce) with openRSTP, too. But there
is no stream on my loopback device (I've set booth on 127.0.0.1, my 
unicast testMP3Server worked also on loopback), only udp packets on a regular
rate (every 5-6 seconds a udp-packet with 48 bytes from the server, and at
a similar rate 28 bytes from the openRSTP-client, which is first sending
that packet and then probably receiving the same packet?).

My assumption (I just got the idea): is that openRSTP still listening on a
multicast socket and my server sending no multicast?

My question is: 
- What can I do to find my mistakes? Is there some kind of general
  debugging switch I haven't found yet (there is somewhere a $define
  DEBUG_PRINT, in the groupsock files there is a "int debug", but no switch
  for all files).
- How are these regular udp-packets triggerd? I haven't found
  the source of them (It must be somewhere deep in the
  basicUsageEnvironment). 

I hope someone can help me out :-) If I can use that library for my work I
will provide some kind of documentation (perhaps a howto).

Joerg

-- 
<istari> he: darf ich euch nicht mal meine glueckwuensche aussprechen?
<HE> istari: Das kannst du auf 3 Arten machen. 1) Anbeten, ruhig sein.
2) Geld geben, ruhig sein. 3) Nur wenn weiblich und gut aussehend: Sex.
   -- gelesen in #debian.de zum 10. Geburtstag von debian


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	Content preview:  >I want to stream jpeg-images (first) from a file
	(later they should come >from a camera). After some playing with the
	test programs (setting >testMP3Streamer and openRSTP back to unicast
	streaming) I wrote a >JPEGVideoFileSource class (also streaming
	unicast). I got it finally >compiling. > >And here is my problem: I
	want to receive the stream from my new streamer >(which is a simple
	copy of testMP3Streamer with the use of >JPEGVideoFileSource instead of
	MP3FileSOurce) [...] 
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X-Original-Date: Thu, 04 Sep 2003 13:47:07 -0700
Date: Thu, 04 Sep 2003 13:47:07 -0700


>I want to stream jpeg-images (first) from a file (later they should come
>from a camera). After some playing with the test programs (setting
>testMP3Streamer and openRSTP  back to unicast streaming) I wrote a
>JPEGVideoFileSource class (also streaming unicast). I got it finally
>compiling.
>
>And here is my problem: I want to receive the stream from my new streamer
>(which is a simple copy of testMP3Streamer with the use of
>JPEGVideoFileSource instead of MP3FileSOurce)

Joerg,

Note that the library *already* has support for JPEG/RTP streaming.  Note 
the classes "JPEGVideoRTPSink" and "JPEGVideoSource".  You should be using 
these, rather than trying to modify the "MP3FileSource" and 
"MPEG1or2AudioRTPSink" that "testMP3Streamer" uses.

Note, though, that "JPEGVideoSource" is an abstract base class, which you 
will need to subclass in order to use.  I suggest that you make your 
"JPEGVideoFileSource" class a subclass of "JPEGVideoSource", and then there 
will just be a handful of virtual member functions:
         virtual void doGetNextFrame();
         virtual float getPlayTime(unsigned numFrames) const;
         virtual u_int8_t type();
         virtual u_int8_t qFactor();
         virtual u_int8_t width();
         virtual u_int8_t height();
(plus a "createNew()" function)
that you will need to implement to get your code working.

I.e., your source can be a "JPEGVideoSource", and your sink can be a 
"JPEGVideoRTPSink".  That's all you need (i.e., you won't need any extra 
'framer' objects inbetween).

As for the rest of your question, where you asked about debugging tips: 
Remember that You Have Complete Source Code.  You can use whatever 
debugging tools/techniques you like.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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From <EMAIL-ADDRESS-HIDDEN> Fri Sep 5 03:24:10 2003
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	Content preview:  Recent changes to the "LIVE.COM Streaming Media" code
	have improved the implementation of MPEG and JPEG video streaming, so
	if you're using the code for this, you should upgrade to the latest
	version. [...] 
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Subject: [Live-devel] Improvements to MPEG and JPEG video streaming
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X-Original-Date: Fri, 05 Sep 2003 03:14:54 -0700
Date: Fri, 05 Sep 2003 03:14:54 -0700

Recent changes to the "LIVE.COM Streaming Media" code have improved the 
implementation of MPEG and JPEG video streaming, so if you're using the 
code for this, you should upgrade to the latest version.

Version "2003.09.03" or later makes the streaming of MPEG or JPEG video 
more efficient (in the common case when slices or frames (respectively) are 
split over multiple RTP packets).  An unnecessary memory copy was eliminated.

Version "2003.09.05" also improves the *reception* of motion-JPEG/RTP 
streams - i.e., the "JPEGVideoRTPSource" class.  In fact, the previous 
implementation of this class didn't work at all in most cases.  But now, 
you can use "openRTSP" (or "MPlayer") to receive (play) JPEG/RTP streams.

	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Hi,
 
Does anyone know where I can find a SIP UA ActiveX ?
 
Thanks,
 
Andre

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<p class=3DMsoNormal><span class=3DSpellE><font size=3D2 =
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<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
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find a SIP UA ActiveX ?<o:p></o:p></span></font></p>

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	Content preview:  Hi, I am recording a H263 and qualcomm purevoice stream
	from with openRTSP to a quicktime file. I then stream this file using
	Darwin SS to a quicktime client. The video is perfect but there is no
	sound in the 'Movie Info' pane no codec is listed for the audio 0Hz and
	0bits while all the correct info for the video is listed H.263 etc..
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X-Original-Date: Sat, 06 Sep 2003 09:29:30 -0400
Date: Sat, 06 Sep 2003 09:29:30 -0400

Hi,
   I am recording a H263 and qualcomm purevoice stream
from with openRTSP to a quicktime file.
I then stream this file using Darwin SS to a quicktime client.
The video is perfect but there is no sound in the 'Movie Info' pane no codec is listed for the audio 0Hz and 0bits while all the correct info for the video is listed
H.263 etc..

The file plays, in QT, correctly locally.

It looks like there is something wrong with openRTSP and its hinting of audio tracks.

If i create a QT file using openRTSP containing only audio i have the same problem.
If i do the same with only video it works correctly.

Please please help i've hit a brick wall with this.
It's been weeks.

thanks
John



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Subject: Re: [Live-devel] openRTSP & Quicktime file
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	Content preview:  At 06:29 AM 9/6/03, you wrote: >Hi, > I am recording a
	H263 and qualcomm purevoice stream >from with openRTSP to a quicktime
	file. >I then stream this file using Darwin SS to a quicktime client.
	>The video is perfect but there is no sound in the 'Movie Info' pane no
	>codec is listed for the audio 0Hz and 0bits while all the correct info
	for >the video is listed >H.263 etc.. > >The file plays, in QT,
	correctly locally. > >It looks like there is something wrong with
	openRTSP and its hinting of >audio tracks. [...] 
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X-Original-Date: Sat, 06 Sep 2003 08:01:59 -0700
Date: Sat, 06 Sep 2003 08:01:59 -0700

At 06:29 AM 9/6/03, you wrote:
>Hi,
>    I am recording a H263 and qualcomm purevoice stream
>from with openRTSP to a quicktime file.
>I then stream this file using Darwin SS to a quicktime client.
>The video is perfect but there is no sound in the 'Movie Info' pane no 
>codec is listed for the audio 0Hz and 0bits while all the correct info for 
>the video is listed
>H.263 etc..
>
>The file plays, in QT, correctly locally.
>
>It looks like there is something wrong with openRTSP and its hinting of 
>audio tracks.

John,

The Bad News: "openRTSP"s support for creating QuickTime format files is 
rather limited.  At present, only PCM (u-law and a-law), GSM and QCELP 
(==PureVoice) audio is supported, and only H.263 video is supported.

The Good News: Fortunately, though, you are using codecs from this set, 
which is why you are able to create a QuickTime file that plays locally.

The Bad News: "openRTSP"s support for creating *hinted* QuickTime format 
files is even more limited: QCELP audio is not currently supported.

If you are able to use PCM or GSM audio instead, then you should be able to 
generate hint tracks that work.

(If QCELP hint track support is important to you, and you're interested in 
providing funding to have it implemented, please let me know (by separate 
email).)



	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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	Content preview:  Ross, thanks for the reply. I'll check it out. thanks
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X-Original-Date: Mon, 08 Sep 2003 09:13:18 -0400
Date: Mon, 08 Sep 2003 09:13:18 -0400

Ross,
       thanks for the reply.  I'll check it out.

thanks

regards
John

--

--------- Original Message ---------

DATE: Sat, 06 Sep 2003 08:01:59
From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
To: <EMAIL-ADDRESS-HIDDEN>
Cc:

>At 06:29 AM 9/6/03, you wrote:
>>Hi,
>>    I am recording a H263 and qualcomm purevoice stream
>>from with openRTSP to a quicktime file.
>>I then stream this file using Darwin SS to a quicktime client.
>>The video is perfect but there is no sound in the 'Movie Info' pane no 
>>codec is listed for the audio 0Hz and 0bits while all the correct info for 
>>the video is listed
>>H.263 etc..
>>
>>The file plays, in QT, correctly locally.
>>
>>It looks like there is something wrong with openRTSP and its hinting of 
>>audio tracks.
>
>John,
>
>The Bad News: "openRTSP"s support for creating QuickTime format files is 
>rather limited.  At present, only PCM (u-law and a-law), GSM and QCELP 
>(==PureVoice) audio is supported, and only H.263 video is supported.
>
>The Good News: Fortunately, though, you are using codecs from this set, 
>which is why you are able to create a QuickTime file that plays locally.
>
>The Bad News: "openRTSP"s support for creating *hinted* QuickTime format 
>files is even more limited: QCELP audio is not currently supported.
>
>If you are able to use PCM or GSM audio instead, then you should be able to 
>generate hint tracks that work.
>
>(If QCELP hint track support is important to you, and you're interested in 
>providing funding to have it implemented, please let me know (by separate 
>email).)
>
>
>
>	Ross Finlayson
>	LIVE.COM
>	<http://www.live.com/>
>
>
>
>-------------------------------------------------------
>This sf.net email is sponsored by:ThinkGeek
>Welcome to geek heaven.
>http://thinkgeek.com/sf
>_______________________________________________
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>



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	does not understand this format, some or all of this message may not be
	legible. Hello, I have started looking for MPEG4 libraries and came
	across this group. I have been looking at the source code of the
	library as well as previous posting on this news group. I have very
	basic questions and will greatly appreciate any input on this. [...] 
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Date: Mon, 15 Sep 2003 10:53:42 -0400

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------_=_NextPart_001_01C37B99.280BF1B0
Content-Type: text/plain;
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Hello,
    I have started looking for MPEG4 libraries and came across this group. I
have been looking at the source code of the library as well as previous
posting on this news group. I have very basic questions and will greatly
appreciate any input on this.
 
- does Live.com library supports MPEG4 ? 
- does the library support RTP payload format for H-263 (RFC 2190) ?
- does the library support RTP payload format for AMR (Adaptive multirate)
codecs (RFC 3267) ?
- does the library provide support for synchronization of audio & video
packets ?
- what are the licensing clauses for a commercial product using this library
including patent fees for MPEG4, AAC and MP3
- finally, is there any documentation or sample programs available on how to
use this library with RTP stacks for recording and playback of MPEG4 files.
Whether support for HTTP streaming for playback is available.
 
Thank you,
Samir Karande


------_=_NextPart_001_01C37B99.280BF1B0
Content-Type: text/html;
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<DIV><SPAN class=625454514-15092003><FONT size=2>Hello,</FONT></SPAN></DIV>
<DIV><SPAN class=625454514-15092003>&nbsp;&nbsp;&nbsp; <FONT size=2>I have 
started looking for MPEG4 libraries and came across this group. I have been 
looking at the source code of the library as well as previous posting on this 
news group. I have very basic questions and will greatly appreciate any input on 
this.</FONT></SPAN></DIV>
<DIV><SPAN class=625454514-15092003><FONT size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=625454514-15092003><FONT size=2>- does Live.com library 
supports MPEG4 ? </FONT></SPAN></DIV>
<DIV><SPAN class=625454514-15092003><FONT size=2>- does the library support RTP 
payload format for H-263 (RFC 2190) ?</FONT></SPAN></DIV>
<DIV><SPAN class=625454514-15092003><FONT size=2>- does the library support RTP 
payload format for AMR (Adaptive multirate) codecs (RFC 3267) 
?</FONT></SPAN></DIV>
<DIV><SPAN class=625454514-15092003><FONT size=2>- does the library provide 
support for synchronization of audio &amp; video packets ?</FONT></SPAN></DIV>
<DIV><SPAN class=625454514-15092003><FONT size=2>- what are the licensing 
clauses for a commercial product using this library including patent fees for 
MPEG4, AAC and MP3</FONT></SPAN></DIV>
<DIV><SPAN class=625454514-15092003><FONT size=2>- finally, is there any 
documentation or sample programs available on how to use this library with RTP 
stacks for recording and playback of MPEG4 files. Whether support for HTTP 
streaming for playback is available.</FONT></SPAN></DIV>
<DIV><SPAN class=625454514-15092003><FONT size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=625454514-15092003><FONT size=2>Thank you,</FONT></SPAN></DIV>
<DIV><SPAN class=625454514-15092003><FONT size=2>Samir 
Karande</FONT></SPAN></DIV>
<P></P></BODY></HTML>

------_=_NextPart_001_01C37B99.280BF1B0--


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	Content preview:  >- does Live.com library supports MPEG4 ? Yes, it
	supports the sending and receiving of MPEG-4 video and audio (generic
	or LATM AAC) RTP streams >- does the library support RTP payload format
	for H-263 (RFC 2190) ? [...] 
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X-Original-Date: Mon, 15 Sep 2003 10:53:43 -0700
Date: Mon, 15 Sep 2003 10:53:43 -0700


>- does Live.com library supports MPEG4 ?

Yes, it supports the sending and receiving of MPEG-4 video and audio 
(generic or LATM AAC) RTP streams

>- does the library support RTP payload format for H-263 (RFC 2190) ?

The RTP payload format described in RFC 2429 is now preferred over the 
format defined in RFC 2190.  The LIVE.COM library supports RFC 2429 
(sending and receiving).

>- does the library support RTP payload format for AMR (Adaptive multirate) 
>codecs (RFC 3267) ?

Yes.

>- does the library provide support for synchronization of audio & video 
>packets ?

Yes, using RTCP.  See also 
<http://www.live.com/liveMedia/faq.html#separate-rtp-streams>

>- what are the licensing clauses for a commercial product using this 
>library including patent fees for MPEG4, AAC and MP3

The LIVE.COM library implements only open IETF standards, and does not 
include any codec (encoder or decoder) implementation that would require 
licensing.  Of course, if you were to combine this library with a 
third-party encoder or decoder (e.g., for MPEG-4), then you would be 
responsible for any license fees required to use that third-party product.

>- finally, is there any documentation or sample programs available on how 
>to use this library with RTP stacks for recording and playback of MPEG4 files.

Note the "testMPEG4VideoStreamer" test program.  See 
<http://www.live.com/liveMedia/#testProgs>

>  Whether support for HTTP streaming for playback is available.

The library supports (as an option) RTP-over-TCP streaming from RTSP 
servers, but not RTP-over-HTTP streaming.

(Many of your questions are already addressed by the online documentation 
at <http://www.live.com/liveMedia/> and 
<http://www.live.com/liveMedia/faq.html>, so you may wish to review these 
web pages.)


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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From <EMAIL-ADDRESS-HIDDEN> Mon Sep 15 17:47:35 2003
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	Content preview:  Hi, i have a "elementary" question :-D I try to use
	livemedia in my streaming application. My application capture video and
	produce an MPEG4 videostream. in a prio mail you wrote: "If you already
	have MPEG encoder hardware, then instead of writing new software
	components 1/ and 2/, you can just take the output of the encoder, pass
	it through a "MPEGVideoStreamFramer", and then stream it using a
	"MPEGVideoRTPSink". [...] 
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X-Original-Date: Tue, 16 Sep 2003 02:41:46 +0200
Date: Tue, 16 Sep 2003 02:41:46 +0200

Hi,

i have a "elementary" question :-D
I try to use livemedia in my streaming application.
My application capture video and produce an MPEG4 videostream.
in a prio mail you wrote:
"If you already have MPEG encoder hardware, then instead of writing new software
components 1/ and 2/, you can just take the output of the encoder, pass it
through a "MPEGVideoStreamFramer", and then stream it using a "MPEGVideoRTPSink".

I have no MPEG hard- but software. I get an videoheader with width and height of
the image,etc..., and the compressed data from the image from my encoder.
What is a "frame" in livemedia ? Only the compressed data or compressed data +
headerinformation ?
Or streams the livemedialib only the data ? But how can the receiver knows which
resultion the video has or its a keyframe or not?

Similar my next question :-D what is a elementary stream ? I know, an none
multiplexed stream. But i produced an mpeg2 stream without sound (with tmpgEnc)
but i got "Saw unexpected code error" like others on the mailinglist.
  

-- 
Mit freundlichen Grüssen
Dominik Acri                         
  



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From <EMAIL-ADDRESS-HIDDEN> Mon Sep 15 20:09:41 2003
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	livemedia in my streaming application. >My application capture video
	and produce an MPEG4 videostream. >in a prio mail you wrote: >"If you
	already have MPEG encoder hardware, then instead of writing new
	>software >components 1/ and 2/, you can just take the output of the
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X-Original-Date: Mon, 15 Sep 2003 20:00:59 -0700
Date: Mon, 15 Sep 2003 20:00:59 -0700


>i have a "elementary" question :-D
>I try to use livemedia in my streaming application.
>My application capture video and produce an MPEG4 videostream.
>in a prio mail you wrote:
>"If you already have MPEG encoder hardware, then instead of writing new 
>software
>components 1/ and 2/, you can just take the output of the encoder, pass it
>through a "MPEGVideoStreamFramer", and then stream it using a 
>"MPEGVideoRTPSink".

Yes, that's true.  However, note that the names of these classes has 
changed.  (Please make sure that you're using an up-to-date version of the 
source code!)

For MPEG-1 or 2 video, use "MPEG1or2VideoRTPSink", and 
"MPEG1or2VideoStreamFramer" (if necessary).

For MPEG-4 video, use "MPEG4ESVideoRTPSink", and "MPEG4VideoStreamFramer" 
(if necessary).

>I have no MPEG hard- but software. I get an videoheader with width and 
>height of
>the image,etc..., and the compressed data from the image from my encoder.
>What is a "frame" in livemedia ? Only the compressed data or compressed data +
>headerinformation ?

As noted in RFC 3016 (which defines the RTP payload format for MPEG-4 
video), the transmitted RTP stream must contain all MPEG-4 video headers - 
not just 'video object plane' data.  I.e., the stream must also contain 
'visual object sequence', 'visual object', 'video object layer', and 'group 
of video object plane' headers.  From the 'liveMedia' library's point of 
view, each of these is treated as a 'frame'.  (See the implementation of 
the "MPEG4VideoStreamFramer" class to see how the library breaks up a 
MPEG-4 video byte stream into these 'frames'.)

I.e., your encoder software must produce all of these headers, in order for 
the data to be properly streamed over RTP.  If your encoder produces 
discrete 'frames' (i.e., MPEG-4 video headers plus 'video object planes'), 
then you can feed it directly into a "MPEG4ESVideoRTPSink"; you won't need 
to use a separate 'framer' filter.  (For this to work, however, your 
encoder will need to implement the "isMPEG4VideoStreamFramer()" virtual 
function, returning "True".)

>Or streams the livemedialib only the data ? But how can the receiver knows 
>which
>resultion the video has or its a keyframe or not?

As noted above, all of the MPEG-4 video header information is transmitted 
in the RTP stream.

>Similar my next question :-D what is a elementary stream ? I know, an none
>multiplexed stream. But i produced an mpeg2 stream without sound (with 
>tmpgEnc)
>but i got "Saw unexpected code error" like others on the mailinglist.

Perhaps your MPEG-2 stream was a *Program* stream, rather than a true 
Elementary Stream.  See 
<http://www.live.com/liveMedia/faq.html#mpeg-program-stream>


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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	Content preview:  Hi, I have been experimenting running more than one
	mplayer on the same host, decoding different RTSP streams. When running
	two mplayers decoding the same RTSP stream everything works fine. If
	decoding two different streams (from two different hosts and multicast
	addresses) mplayer seem to get confused, it looks as if mplayer somehow
	"mix" the two streams. The symptoms are a lot of decoding errors. [...] 
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Hi,

I have been experimenting running more than one mplayer on the same
host, decoding different RTSP streams.
When running two mplayers decoding the same RTSP stream everything works
fine. If decoding two different streams (from two different hosts and
multicast addresses) mplayer seem to get confused, it looks as if
mplayer somehow "mix" the two streams. The symptoms are a lot of
decoding errors.

Any ideas ??

Thanks - Jan


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Subject: Re: [Live-devel] Problems running 2 (or more) mplayer instances
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	Content preview:  >I have been experimenting running more than one
	mplayer on the same >host, decoding different RTSP streams. >When
	running two mplayers decoding the same RTSP stream everything works
	>fine. If decoding two different streams (from two different hosts and
	>multicast addresses) mplayer seem to get confused [...] 
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>I have been experimenting running more than one mplayer on the same
>host, decoding different RTSP streams.
>When running two mplayers decoding the same RTSP stream everything works
>fine. If decoding two different streams (from two different hosts and
>multicast addresses) mplayer seem to get confused

You should make sure that the two multicast streams use not only different 
multicast addresses, but also different *port numbers*.  Many OSs don't do 
a good job of demultiplexing incoming packets that have the same 
destination port number, even though they have different destination 
multicast addresses.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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	Content preview:  It looks as if you are right. We tried using unique
	multicast addresses AND ports, and everything seem to work OK now. This
	is a bug in the implementation of the IP stack, isn't it ?? Thanks !
	-----Oprindelig meddelelse----- Fra: Ross Finlayson
	[mailto:finlayson(at)live.com] Sendt: 16. september 2003 11:58 Til:
	live-devel(at)lists.sourceforge.net Emne: Re: [Live-devel] Problems
	running 2 (or more) mplayer instances [...] 
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It looks as if you are right. We tried using unique multicast addresses
AND ports, and everything seem to work OK now.
This is a bug in the implementation of the IP stack, isn't it ??

Thanks !

-----Oprindelig meddelelse-----
Fra: Ross Finlayson [mailto:finlayson(at)live.com] 
Sendt: 16. september 2003 11:58
Til: live-devel(at)lists.sourceforge.net
Emne: Re: [Live-devel] Problems running 2 (or more) mplayer instances



>I have been experimenting running more than one mplayer on the same 
>host, decoding different RTSP streams. When running two mplayers 
>decoding the same RTSP stream everything works fine. If decoding two 
>different streams (from two different hosts and multicast addresses) 
>mplayer seem to get confused

You should make sure that the two multicast streams use not only
different 
multicast addresses, but also different *port numbers*.  Many OSs don't
do 
a good job of demultiplexing incoming packets that have the same 
destination port number, even though they have different destination 
multicast addresses.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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From <EMAIL-ADDRESS-HIDDEN> Wed Sep 17 04:14:05 2003
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	Content preview:  Hi, A couple of questions concerning RTSP - the Live
	RTSPServer. After reading the RTSP spec. I had a look at the source to
	RTSP client/server code. Please let me know if the following is
	correct: [...] 
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Subject: [Live-devel] RTSP Server
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Hi,

A couple of questions concerning RTSP - the Live RTSPServer.

After reading the RTSP spec. I had a look at the source to RTSP
client/server code. Please let me know if the following is correct:

The live RTSP server starts streaming (RTP/multicasting) immediately on
startup.
The server accepts "SETUP", "DESCRIBE", "PLAY", "PAUSE" and "TEARDOWN",
but only "DESCRIBE" and "TEARDOWN" actually do something (except
returning an appropriate RTSP response to the client). The RTSP client
issues "SETUP", next "DESCRIBE" and then "PLAY" (that doesn't actually
do anything) and starts receiving RTP packets on the DESCRIBE'd
multicast address/port. 
The RTSP connection is persistent, that's why you can avoid checking the
session id of the commands. 

- Jan



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	RTSPServer. > >After reading the RTSP spec. I had a look at the source
	to RTSP >client/server code. Please let me know if the following is
	correct: > >The live RTSP server starts streaming (RTP/multicasting)
	immediately on >startup. [...] 
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>A couple of questions concerning RTSP - the Live RTSPServer.
>
>After reading the RTSP spec. I had a look at the source to RTSP
>client/server code. Please let me know if the following is correct:
>
>The live RTSP server starts streaming (RTP/multicasting) immediately on
>startup.

Not quite.  The current "RTSPServer" implementation works by being passed 
one or more "RTPSink" objects.  These "RTPSink" objects each represent an 
existing, outgoing RTP stream.  These RTP streams could have been working 
for a long time before the "RTSPServet" was created.  (However, the test 
programs that currently use the "RTSPServer" happen to create and start the 
RTP streams ("RTPSink") objects at the same time that they create the 
"RTSPServer".)

>The server accepts "SETUP", "DESCRIBE", "PLAY", "PAUSE" and "TEARDOWN",
>but only "DESCRIBE" and "TEARDOWN" actually do something (except
>returning an appropriate RTSP response to the client).

That's correct, but note that this is true only for the *current* 
"RTSPServer" implementation.  (That's why it can currently be used only for 
multicast streams.)  However, the next version of the "RTSPServer" (which 
is currently being tested, and should be available soon) will be more 
sophisticated.  It will have the ability to serve on-demand unicast 
streams, as well as static multicast streams.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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	the *current* > "RTSPServer" implementation. (That's why it can >
	currently be used only for > multicast streams.) However, the next
	version of > the "RTSPServer" (which > is currently being tested, and
	should be available > soon) will be more > sophisticated. It will have
	the ability to serve > on-demand unicast > streams, as well as static
	multicast streams. > [...] 
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X-Original-Date: Thu, 18 Sep 2003 09:38:22 +0200 (CEST)
Date: Thu, 18 Sep 2003 09:38:22 +0200 (CEST)

> That's correct, but note that this is true only for
> the *current* 
> "RTSPServer" implementation.  (That's why it can
> currently be used only for 
> multicast streams.)  However, the next version of
> the "RTSPServer" (which 
> is currently being tested, and should be available
> soon) will be more 
> sophisticated.  It will have the ability to serve
> on-demand unicast 
> streams, as well as static multicast streams.
> 

Hi Ross,

  When? One month, two? ;D Do you need beta testers?

  Regards.

___________________________________________________
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From: Ross Finlayson <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] New RTSP server
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	Content preview:  > However, the next version of > > the "RTSPServer"
	(which > > is currently being tested, and should be available > > soon)
	will be more > > sophisticated. It will have the ability to serve > >
	on-demand unicast > > streams, as well as static multicast streams.
	[...] > When? One month, two? [...] 
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X-Original-Date: Thu, 18 Sep 2003 09:55:24 -0700
Date: Thu, 18 Sep 2003 09:55:24 -0700


>  However, the next version of
> > the "RTSPServer" (which
> > is currently being tested, and should be available
> > soon) will be more
> > sophisticated.  It will have the ability to serve
> > on-demand unicast
> > streams, as well as static multicast streams.
[...]
>   When? One month, two?

A few days.

(Note that this was done upon request of a client who provided funding for 
it to happen.  That's the way most significant improvements to the code get 
done.)


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Subject: [Live-devel] good start for mp3 RTSP unicast server
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X-Original-Date: Mon, 22 Sep 2003 01:16:35 +0800 (HKT)
Date: Mon, 22 Sep 2003 01:16:35 +0800 (HKT)

Dear all,

I've read some code of the live.com library.  I would like to use it to
write a RTSP mp3 unicast streaming server.  Could anyone tell me how to
make start from group up?!
Thanks!



Chris




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Subject: Re: [Live-devel] good start for mp3 RTSP unicast server
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X-Original-Date: Sun, 21 Sep 2003 15:31:50 -0700
Date: Sun, 21 Sep 2003 15:31:50 -0700


>I've read some code of the live.com library.  I would like to use it to
>write a RTSP mp3 unicast streaming server.  Could anyone tell me how to
>make start from group up?!

Within a few days I'll be releasing a new 'test program' that demonstrates 
how to use the new unicast RTSP/RTP streaming support (e.g., for streaming 
MP3).  Stay tuned...


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Subject: [Live-devel] Unicast RTP streaming from a RTSP server is now supported
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X-Original-Date: Thu, 25 Sep 2003 03:23:37 -0700
Date: Thu, 25 Sep 2003 03:23:37 -0700

I have added a new test program "testOnDemandRTSPServer" that demonstrates 
how a "RTSPServer" object can now stream from a file, on demand, using RTP 
unicast.  Several different kinds of file are supported (including MP3 
audio and MPEG elementary stream video files).

(Thanks to TNO Fysisch en Elektronisch Laboratorium (Netherlands)  for 
funding this work.)

	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Thu, 25 Sep 2003 16:50:15 +0200 (CEST)
Date: Thu, 25 Sep 2003 16:50:15 +0200 (CEST)

Now i´m downloading the new RTSP server...

Congrutalations Ross!!

Regards.

  Juan Carlos

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From: Luca Abeni <EMAIL-ADDRESS-HIDDEN>
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Subject: [Live-devel] about live.com+mplayer
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X-Original-Date: 29 Sep 2003 17:38:14 +0100
Date: 29 Sep 2003 17:38:14 +0100

Hi Ross,

I think I discovered why the fix that you committed in the mplayer cvs
does not work... At least, this is my understanding; please correct me
if I am wrong:
- When a packet arrives, MultiFramedRTPSource::networkReadHandler() is
called, and source->receptionStatsDB().noteIncomingPacket() is used to
compute the presentationTime for the packet, then the packet is queued
with source->fReorderingBuffer->storePacket(bPacket). All packets
arriving before the first SR are queued with a presentationTime computed
based on the local time (I think).
- When the first SR packet arrives, the fSyncTime and fSyncTimestamp are
properly set in source->receptionStatsDB(), and the source is considered
to be syncronized... However, source->fReorderingBuffer still contains
some packets that have the presentationTime based on the wrong (pre SR)
temporal reference!!!
- Hence, when demux_rtp gets the next packets it gets a packet with a
wrong presentation time (because it was assigned before synchronizing
the stream), but it sees that the source is synchronized.
- Hence, it demux_rtp starts to see big ptsBehind values, and starts to
discard packets from the other source :(

What do you think? Is this understanding correct? If yes, I think this
could be a bug in the live.com library... I see two possible solutions:
1) computed the presentation time only when a packet is extracted from
source->fReorderingBuffer, to deliver it to demux_rtp
or
2) consider a source syncronized only when the first synchronized packet
is delivered to demux_rtp.


				Luca
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X-Original-Date: Mon, 29 Sep 2003 13:31:02 -0700
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>- When the first SR packet arrives, the fSyncTime and fSyncTimestamp are
>properly set in source->receptionStatsDB(), and the source is considered
>to be syncronized... However, source->fReorderingBuffer still contains
>some packets that have the presentationTime based on the wrong (pre SR)
>temporal reference!!!

Luca,

That can happen, but only if packets have been arriving out-of-order (based 
on RTP sequence number).  Normally, however, RTP packets arrive in order - 
with no gaps - and in this case no extra packets get kept in the 
'reordering buffer'.

So, what you're describing is, indeed, a bug, but it should be triggered 
only rarely - so I'm surprised that you seem to be seeing it every 
time.  I'm curious as to exactly what's going on in your case - e.g., is 
your input source generating lots of reordered packets, and/or lots of 
packet loss?

Nonetheless, later today I'll check in a new version of the code that 
should fix this bug.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Tue, 30 Sep 2003 00:01:16 -0700
Date: Tue, 30 Sep 2003 00:01:16 -0700

As noted in my last message:

A new version (2003.09.30) of the "LIVE.COM Streaming Media" code has now 
been installed.  It fixes the issue that was noted by Luca Abeni: The 
"RTPSource::hasBeenSynchronizedUsingRTCP()" function now more accurately 
reports whether or not the most recently-delivered packet's presentation 
time was synchronized using RTCP SRs.

	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Subject: Re: [Live-devel] about live.com+mplayer
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X-Original-Date: 30 Sep 2003 11:11:07 +0100
Date: 30 Sep 2003 11:11:07 +0100

Hi Ross,

first of all, thanks for the fix!!!

I checked the streams that I am receiving (I wrote a simple program that
receives a multicast RTP stream and checks the sequence numbers and the
timestamps), and no packets arrive out of order. Also, no packets are
lost.

Hence, I do not know what's going on... I will need to investigate the
problem a little bit more. Is it possible that packets are buffered when
a single video frame is split in more packets? Can this be the cause of
the problem?

Anyway, I tried live.2003.09.30.tar.gz, and it fixes the problem!!!
Mplayer does not block anymore. I am still having A/V synchronization
problems (both with MP2 and MP4 files), but I think they are due to
something else (it seems to me that the code discarding packets in
demux_rtp.cpp interacts very badly with mplayer's A/V synch mechanism...
I am looking at this).

Summing up: the last change that you made to the live.com library fixes
the problem that I was seeing (but am still trying to understand why ;-)

			Thanks,
				Luca
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Subject: [Live-devel] Create mp4 Audio/Video File from rtsp
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This is a multipart message in MIME format.
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Hello,

I've fetched 2 dumps from a rtsp stream with openRTSP.

audio-MPEG4-GENERIC-2  and   video-MP4V-ES-1

The Stream is feeded from mp4live (mpeg4 & aac) and is streamed over a 
Darwin Server.

Now my Problem: 
I want do genereate from the fetched Files a mp4 file which I can provide 
as an on-Demand Stream...

I do not know how to build mp4 files... I've tried to rename the dumps to 
mp4/mpg/avi and want
to encode them with mp4creator. But I think there is someting going 
wrong....


Hope of help from you !



regards,
Tom
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<br><font size=2 face="sans-serif">Hello,</font>
<br>
<br><font size=2 face="sans-serif">I've fetched 2 dumps from a rtsp stream
with openRTSP.</font>
<br>
<br><font size=2 face="sans-serif">audio-MPEG4-GENERIC-2 &nbsp;and &nbsp;
video-MP4V-ES-1</font>
<br>
<br><font size=2 face="sans-serif">The Stream is feeded from mp4live (mpeg4
&amp; aac) and is streamed over a Darwin Server.</font>
<br>
<br><font size=2 face="sans-serif">Now my Problem: </font>
<br><font size=2 face="sans-serif">I want do genereate from the fetched
Files a mp4 file which I can provide as an on-Demand Stream...</font>
<br>
<br><font size=2 face="sans-serif">I do not know how to build mp4 files...
I've tried to rename the dumps to mp4/mpg/avi and want</font>
<br><font size=2 face="sans-serif">to encode them with mp4creator. But
I think there is someting going wrong....</font>
<br>
<br>
<br><font size=2 face="sans-serif">Hope of help from you !</font>
<br>
<br>
<br>
<br><font size=2 face="sans-serif">regards,</font>
<br><font size=2 face="sans-serif">Tom</font>
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Date: Tue, 30 Sep 2003 16:56:07 +0530

Hi ,
         Shall i know what are the media file formats are currently
supports in Live.com media steaming server
Thank you in advance

regards


--
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. 



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>I've fetched 2 dumps from a rtsp stream with openRTSP.
>
>audio-MPEG4-GENERIC-2  and   video-MP4V-ES-1
>
>The Stream is feeded from mp4live (mpeg4 & aac) and is streamed over a 
>Darwin Server.
>
>Now my Problem:
>I want do genereate from the fetched Files a mp4 file which I can provide 
>as an on-Demand Stream...

Tom,

"openRTSP" does not yet output "mp4"-format files; instead, only raw 
elementary stream (audio and/or video) files.

Support for writing "mp4"-format files is on our 'to do' list, but has not 
yet been funded.

If your company is interested in providing funding for this project, please 
me know (by separate email; not on this mailing list).


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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>         Shall i know what are the media file formats are currently
>supports in Live.com media steaming server

First, please note that - unlike some other streaming media projects - 
there isn't a *single* piece of software called *the* LIVE.COM streaming 
server.  Instead, the LIVE.COM software contains a *set* of software 
components (C++ classes) that can be used to build specialized servers for 
particular media types.  See, for example, the various test programs (noted 
in <http://www.live.com/liveMedia/#testProgs>), which show how servers for 
various media types can be constructed using the library.

Among the several media file formats that can currently be streamed by the 
library:
- MPEG-1 or 2 audio (including "MP3")
- MPEG-1,2, or 4 video (elementary stream)
- 'WAV'-format files (containing PCM audio)
- AMR audio files
- VOB-format files (e.g., from an unencrypted DVD), containing AC3 audio 
and MPEG-2 video


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Subject: Re: [Live-devel] List of media file Fomrat Support in Live.com
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X-Original-Date: Thu, 02 Oct 2003 16:42:21 +0930
Date: Thu, 02 Oct 2003 16:42:21 +0930

On Wed, 2003-10-01 at 00:51, Ross Finlayson wrote:
> >         Shall i know what are the media file formats are currently
> >supports in Live.com media steaming server
> 
> First, please note that - unlike some other streaming media projects - 
> there isn't a *single* piece of software called *the* LIVE.COM streaming 
> server.  Instead, the LIVE.COM software contains a *set* of software 
> components (C++ classes) that can be used to build specialized servers for 
> particular media types.  See, for example, the various test programs (noted 
> in <http://www.live.com/liveMedia/#testProgs>), which show how servers for 
> various media types can be constructed using the library.
> 
> Among the several media file formats that can currently be streamed by the 
> library:
> - MPEG-1 or 2 audio (including "MP3")
> - MPEG-1,2, or 4 video (elementary stream)
> - 'WAV'-format files (containing PCM audio)
> - AMR audio files
> - VOB-format files (e.g., from an unencrypted DVD), containing AC3 audio 
> and MPEG-2 video

Does this mean that a PS with MPEG-2 video and MPEG audio is not
supported?

I was thinking of using live.com for MPEG TS - either by streaming the
TS directly or by converting it to a PS.
> 
> 
> 	Ross Finlayson
> 	LIVE.COM
> 	<http://www.live.com/>
> 
> 
> 
> -------------------------------------------------------
> This sf.net email is sponsored by:ThinkGeek
> Welcome to geek heaven.
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> 



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  server?
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> > Among the several media file formats that can currently be streamed by the
> > library:
> > - MPEG-1 or 2 audio (including "MP3")
> > - MPEG-1,2, or 4 video (elementary stream)
> > - 'WAV'-format files (containing PCM audio)
> > - AMR audio files
> > - VOB-format files (e.g., from an unencrypted DVD), containing AC3 audio
> > and MPEG-2 video
>
>Does this mean that a PS with MPEG-2 video and MPEG audio is not
>supported?

No.  The list I gave above was not a complete list.

Streaming MPEG Program Stream files is supported.  That's what the 
"testMPEGAudioVideoStreamer" test program does, for example.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Is it possible for someone to send me a copy of their *NIX Self.txt file =
so that I can manually create/edit it to work on my server?  I am not =
really able to run X on the server I am using.

Darkie
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2800.1226" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Is it possible for someone to send me a =
copy of=20
their *NIX Self.txt file so that I can manually create/edit it to work =
on my=20
server?&nbsp; I am not really able to run X on the server I am=20
using.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Darkie</FONT></DIV></BODY></HTML>

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--=====================_987039==_
Content-Type: text/plain; charset="us-ascii"; format=flowed

At 06:06 PM 10/2/03, you wrote:
>Is it possible for someone to send me a copy of their *NIX Self.txt file 
>so that I can manually create/edit it to work on my server?  I am not 
>really able to run X on the server I am using.

[Note to the rest of the list: This question is about "lc" - the GUI-less 
version of the "liveCaster" application.]

Attached is a "SELF.txt" file that you can use.  (Change the "info" and 
"nickname" fields as appropriate.)

         Ross.

--=====================_987039==_
Content-Type: text/plain; charset="us-ascii"

info	MP3 audio
bwLimit	320
progId	P:66.80.62.35:1065161097
nickname	liveCaster
outputMode	audio
SDPdir	default
GroupEId	236.206.27.53 56028 {255 nokey}
tunnelState	0 {} 10100 10100 1

--=====================_987039==_
Content-Type: text/plain; charset="us-ascii"; format=flowed


--=====================_987039==_--



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From <EMAIL-ADDRESS-HIDDEN> Fri Oct 3 10:53:34 2003
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From: Michel Cunha <EMAIL-ADDRESS-HIDDEN>
To: Mike Melanson <EMAIL-ADDRESS-HIDDEN>
   xine-devel <xine-devel(at)lists.sourceforge.net>
Cc: live-devel <EMAIL-ADDRESS-HIDDEN>
   Yezdi Tamboli <ytamboli(at)callisto.ca>
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Subject: [Live-devel] investigation about rtsp in xine
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X-Original-Date: 03 Oct 2003 13:51:55 -0400
Date: 03 Oct 2003 13:51:55 -0400

I will like your opinion on this :)

After a quick investigation about what would be the easiest and best way
to add more support in your xine rtsp input plugin to support streaming
mp4 from Apple and Sun servers, I discovered that the best solution will
be to include and link xine with the openRTSP libraries.

openRTSP libraries code forms a set of C++ libraries for multimedia
streaming, using open standard protocols (RTP/RTCP, RTSP, SIP). These
libraries can be used to build streaming applications(Unix/Linux) and
can also be used to build basic RTSP or SIP clients and servers, and
have been used to add streaming support to existing media player
applications.

One big advantage to use their libraries, is that we will have two
excellent teams (xine and live.com) working and maintaining each their
project.

Michel




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Subject: [Live-devel] Re: [xine-devel] investigation about rtsp in xine
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Live.com say in their web site at
http://www.live.com/openRTSP/#real-media
:
A note about RealAudio and RealVideo sessions
Note that this program cannot be used to receive RealAudio and/or
RealVideo sessions - even those described by a "rtsp://" URL - because
these sessions do not use RTP for transport. (Instead, these sessions
use RealNetworks' proprietary "RDT" protocol.) 

Can someone in live.com confirm this please ?


Le ven 03/10/2003 Ã  14:00, Guenter Bartsch a Ã©crit :
> hallo michel,
> 
> > I will like your opinion on this :)
> > 
> > After a quick investigation about what would be the easiest and best way
> > to add more support in your xine rtsp input plugin to support streaming
> > mp4 from Apple and Sun servers, I discovered that the best solution will
> > be to include and link xine with the openRTSP libraries.
> > 
> > openRTSP libraries code forms a set of C++ libraries for multimedia
> > streaming, using open standard protocols (RTP/RTCP, RTSP, SIP). These
> > libraries can be used to build streaming applications(Unix/Linux) and
> > can also be used to build basic RTSP or SIP clients and servers, and
> > have been used to add streaming support to existing media player
> > applications.
> > 
> > One big advantage to use their libraries, is that we will have two
> > excellent teams (xine and live.com) working and maintaining each their
> > project.
> 
> does openrtsp support real network's rtsp extensions?
> 
> best regards
> 
>    guenter



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From: Michel Cunha <EMAIL-ADDRESS-HIDDEN>
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Subject: [Live-devel] Re: [xine-devel] investigation about rtsp in xine
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> so the tricky part will be to make sure xine doesn't loos it's ability
> to play back real network streams (so far i haven't seen a single
> non-real network server out there using rtsp btw)

Notice that I try the last version of the Real server with xine and its
does not work.  It seems that the actual xine rtsp input plugin play
only with a old Real server.

However it will be good to support also the standard rtsp used by Sun
and Apple :)

Le ven 03/10/2003 Ã  14:35, Guenter Bartsch a Ã©crit :
> hallo michel,
> 
> > Live.com say in their web site at
> > http://www.live.com/openRTSP/#real-media
> > :
> > A note about RealAudio and RealVideo sessions
> > Note that this program cannot be used to receive RealAudio and/or
> > RealVideo sessions - even those described by a "rtsp://" URL - because
> > these sessions do not use RTP for transport. (Instead, these sessions
> > use RealNetworks' proprietary "RDT" protocol.) 
> 
> so the tricky part will be to make sure xine doesn't loos it's ability
> to play back real network streams (so far i haven't seen a single
> non-real network server out there using rtsp btw)
> 
> guenter
> 
> > 
> > Can someone in live.com confirm this please ?
> > 
> > 
> > Le ven 03/10/2003 Ã  14:00, Guenter Bartsch a Ã©crit :
> > > hallo michel,
> > > 
> > > > I will like your opinion on this :)
> > > > 
> > > > After a quick investigation about what would be the easiest and best way
> > > > to add more support in your xine rtsp input plugin to support streaming
> > > > mp4 from Apple and Sun servers, I discovered that the best solution will
> > > > be to include and link xine with the openRTSP libraries.
> > > > 
> > > > openRTSP libraries code forms a set of C++ libraries for multimedia
> > > > streaming, using open standard protocols (RTP/RTCP, RTSP, SIP). These
> > > > libraries can be used to build streaming applications(Unix/Linux) and
> > > > can also be used to build basic RTSP or SIP clients and servers, and
> > > > have been used to add streaming support to existing media player
> > > > applications.
> > > > 
> > > > One big advantage to use their libraries, is that we will have two
> > > > excellent teams (xine and live.com) working and maintaining each their
> > > > project.
> > > 
> > > does openrtsp support real network's rtsp extensions?
> > > 
> > > best regards
> > > 
> > >    guenter



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Subject: Re: [Live-devel] Re: [xine-devel] investigation about rtsp in
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X-Original-Date: Fri, 03 Oct 2003 13:23:57 -0700
Date: Fri, 03 Oct 2003 13:23:57 -0700

A few things to note (in response to each of these messages):

First, the name of our software libraries is "LIVE.COM Streaming Media" 
(see <http://www.live.com/liveMedia/>, *not* "openRTSP".  ("openRTSP" is 
the name of an example RTSP client application that happens to use the 
"LIVE.COM Streaming Media libraries".)

As previously noted, RealAudio/RealVideo sessions use a bastardized form of 
RTSP - that uses RealNetworks' proprietary 'RDT' protocol for transport, 
rather than the industry-standard RTP protocol that everyone else uses (and 
that the LIVE.COM libraries support).  MPlayer was able to handle both 
types of stream by including special-case code for RealAudio/RealVideo 
streams.  (Actually, this special-case code was ported from Xine!)  The 
special-case hack works by looking at the "rtsp://..." URL.  If it ends 
with a RealAudio/RealVideo suffix - ".ra", ".ram", or ".rv" - then the 
special-case RealAudio/RealVideo-specific code is called; otherwise, the 
regular RTSP/RTP handling code (using the LIVE.COM libraries) is called.

Someone commented that so far they had never seen a RTSP stream that's not 
RealAudio or RealVideo.  If so, then they haven't been looking very 
hard.  All of Apple's QuickTime streams use RTSP.  Ditto for standard 
"3GPP" and "ISMA" streams.

Note that support for the LIVE.COM libraries in MPlayer is optional (using 
"#ifdef"s), and is specified using an option to "configure".  By default, 
it is not enabled.  Thus, people who don't want to have to use the LIVE.COM 
code in MPlayer don't need to use it (although then they won't get support 
for standard RTSP/RTP streaming.

A couple of the MPlayer developers have grizzled about the fact that the 
LIVE.COM code is written in C++, but in practice their have been very few 
problems.  Complaints about C++ might have been credible 15 years ago, but 
not in 2003.  Pretty much every platform on which you could imagine porting 
Xine will have C++ compiler support as well as C.  (With MPlayer, the only 
real portability issue that has arisen has been that you need to use the 
same version of GCC to build both the LIVE.COM libraries and the rest of 
MPlayer.)

If you want to support standard RTSP/RTP streams in Xine, the most 
important technical issue to note is that these streams typically use 
separate, independent RTP packet streams for audio and video.  I.e., audio 
packets usually arrive on one UDP port; video packets arrive on 
another.  This means that if you cannot use a single input source, and 
demultiplex audio and video from this.  Thus, if Xine's code assumes that 
there will always be just a single input source, then you are going to have 
problems implementing support for standard RTSP/RTP streams (whether or not 
you use the LIVE.COM libraries for this).  I don't know enough about the 
Xine code to know whether or not this will be a problem for you - but it 
was a fatal flaw in the "videoLAN player" software when I looked at that a 
year or so ago.  (In contrast, "MPlayer"s software architecture allows for 
separate 'demuxes' for audio and video, which made it possible to add 
support for RTSP/RTP to MPlayer.)

If anyone wants to investigate adding RTSP/RTP support (using the LIVE.COM 
code) to Xine, then I suggest that you look at my implementation of this in 
MPlayer.  (Note, in particular, the files "demux_rtp*" in the "libmpdemux" 
directory.)  Don't bother looking that the code for the "openRTSP" 
application - that code implements a lot of 'bells and whistles' that would 
not be relevant for you.

Finally, please note that the "live-devel(at)lists.sourceforge.net" mailing 
list accepts messages only from subscribers, so you may wish to omit that 
mailing list from any follow-up messages.

	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Subject: [Live-devel] SELF.txt
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Is it possible for someone to send me a copy of their *NIX Self.txt file =
so that I can manually create/edit it to work on my server?  I am not =
really able to run X on the server I am using.

Darkie
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<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Is it possible for someone to send me a =
copy of=20
their *NIX Self.txt file so that I can manually create/edit it to work =
on my=20
server?&nbsp; I am not really able to run X on the server I am=20
using.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Darkie</FONT></DIV></BODY></HTML>

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X-Original-Date: Sun, 05 Oct 2003 13:06:24 -0700
Date: Sun, 05 Oct 2003 13:06:24 -0700


>xine does assume one single input. But this RTSP/RTP streaming would be
>encapsulated in an input plugin, which can open two ports without
>problems. The interesting question is: Will it be possible for the
>input plugin to serialize the two streams to feed one demuxer?

Yes, you could do this - but you would need to define your own, new format 
for representing multiplexed RTP data, and make the rest of xine understand 
this format.  (E.g., you couldn't use an existing multiplexed data format 
like MPEG Program Streams, because RTP data isn't necessarily MPEG.)

Note that incoming data from RTP packets have accurate, time-synchronized 
'presentation times' (see 
<http://www.live.com/liveMedia/faq.html#separate-rtp-streams> for more 
explanation of this).  You also know the data's media type (from SDP 
information that was obtained 'out-of-band' - e.g., using RTSP).  So, you 
could define a multiplexed data format that represents, for each piece of 
RTP data:
1/ its media type (audio or video)
2/ its MIME type or other codec id
3/ its presentation time
4/ the data itself

The rest of xine could then be made to understand - and demultiplex from - 
this format.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Subject: [Live-devel] problem with cross compiling live
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X-Original-Date: Mon, 6 Oct 2003 17:35:28 +0800
Date: Mon, 6 Oct 2003 17:35:28 +0800

hi,all
      now I'm trying to cross compile live library to arm,but met some
problems.
      my live library is live.2003.09.19,toolchain is arm-elf-20030314,from
uclinux.org(http://www.uclinux.org/pub/uClinux/arm-elf-tools/),and I change
Makefile like this:
COMPILE_OPTS =          $(INCLUDES) -I. -O -DSOCKLEN_T=socklen_t
C =                     c
C_COMPILER =            arm-elf-gcc
C_FLAGS =               $(COMPILE_OPTS) -g
CPP =                   cpp
CPLUSPLUS_COMPILER =    arm-elf-c++
CPLUSPLUS_FLAGS =       $(COMPILE_OPTS) -Wall -DBSD=1 -g
OBJ =                   o
LINK =                  arm-elf-c++ -o
LINK_OPTS =             -L.
CONSOLE_LINK_OPTS =     $(LINK_OPTS)
LIBRARY_LINK =          arm-elf-ld -o
      at first,it seems that everything is ok,but soon encounters an error:
arm-elf-c++ -c -Iinclude -I../UsageEnvironment/include
-I../groupsock/include -I. -O -DSOCKLEN_T=socklen_t -Wall -DBSD=1 -g
ByteStreamFileSource.cpp
ByteStreamFileSource.cpp:72: aggregate `struct timezone Idunno' has
incomplete type and cannot be initialized
ByteStreamFileSource.cpp: In method `void
ByteStreamFileSource::doGetNextFrame()':
ByteStreamFileSource.cpp:94: confused by earlier errors, bailing out
make: *** [ByteStreamFileSource.o] Error 1
      i check in the code there,comment "-DBSD=1" in Makefile,then try
again,there is no error,but after a while,another error comes:
arm-elf-c++ -c -Iinclude -I../UsageEnvironment/include
-I../groupsock/include -I. -O -DSOCKLEN_T=socklen_t -Wall  -g
MediaSession.cpp
In file included from include/liveMedia.h:70,
                 from MediaSession.cpp:22:
include/QuickTimeFileSink.h: In method `unsigned int
QuickTimeFileSink::addByte(unsigned char)':
include/QuickTimeFileSink.h:85: implicit declaration of function `int
fputc_unlocked(...)'
MediaSession.cpp: In method `MediaSession::MediaSession(UsageEnvironment
&)':
MediaSession.cpp:67: implicit declaration of function `int gethostname
(...)'
make[1]: *** [MediaSession.o] Error 1
make[1]: Leaving directory `/root/live/liveMedia'
make: *** [liveMedia/libliveMedia.a] Error 2
      it seems that it needs header files,so I tried to add path to
Makefile(path /usr/local/arm-elf/include):
INCLUDES = -Iinclude -I../UsageEnvironment/include -I../groupsock/include
-I/usr
/local/arm-elf/include
      but when i try again,more error encountered:
arm-elf-c++ -c -Iinclude -I../UsageEnvironment/include
-I../groupsock/include -I/usr/local/arm-elf/include -I. -O
-DSOCKLEN_T=socklen_t -Wall -g Media.cpp
In file included from /usr/local/arm-elf/include/netinet/in.h:250,
                 from ../groupsock/include/NetCommon.h:50,
                 from include/Media.h:38,
                 from Media.cpp:21:
/usr/local/arm-elf/include/bits/in.h:69: ANSI C++ forbids data member
`ip_opts' with same name as enclosing class
make: *** [Media.o] Error 1


      i'm sure that these codes can be compiled for x86,can someone helps
me out?when i finish this,i will provide some documents/hows about this.

(btw:i'v change all .hh file to .h,and works well)


di





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X-Original-Date: Mon, 06 Oct 2003 10:33:14 -0700
Date: Mon, 06 Oct 2003 10:33:14 -0700


>      now I'm trying to cross compile live library to arm,but met some
>problems.
>       my live library is live.2003.09.19,toolchain is arm-elf-20030314,from
>uclinux.org(http://www.uclinux.org/pub/uClinux/arm-elf-tools/),and I change
>Makefile

Instead of changing Makefiles, you should modify the appropriate 
"config.<platform>" file, and re-run "genMakefiles <platform>".  You may 
wish to use "config.cris-axis-linux-gnu" as a model.

>arm-elf-c++ -c -Iinclude -I../UsageEnvironment/include
>-I../groupsock/include -I. -O -DSOCKLEN_T=socklen_t -Wall  -g
>MediaSession.cpp
>In file included from include/liveMedia.h:70,
>                  from MediaSession.cpp:22:
>include/QuickTimeFileSink.h: In method `unsigned int
>QuickTimeFileSink::addByte(unsigned char)':
>include/QuickTimeFileSink.h:85: implicit declaration of function `int
>fputc_unlocked(...)'

It appears that your target environment does not implement "putc()" 
properly.  In "liveMedia/include/QuickTimeFileSink.hh", you should try 
replacing "putc" with "fputc".

>MediaSession.cpp: In method `MediaSession::MediaSession(UsageEnvironment
>&)':
>MediaSession.cpp:67: implicit declaration of function `int gethostname

Some embeded Linux environments - for some reason - don't define 
"gethostname()".  You can avoid this problem by adding "-DCRIS" to the 
compilation command line.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Mon, 06 Oct 2003 10:41:57 -0700
Date: Mon, 06 Oct 2003 10:41:57 -0700


>So RTP can contain different formats for audio and video? Say, MPEG
>video and WAV audio?

Yes (although note that "WAV" is a file format, not a codec; you meant 
*PCM* audio).  RTP sessions can contain audio and video streams that use 
completely unrelated codecs (and also other types of stream, such as 
time-synchronized text (e.g., subtitles)).  The audio and video RTP streams 
could even - in principle - come from different sources.


	Ross Finlayson
	LIVE.COM
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X-Original-Date: Mon, 06 Oct 2003 10:38:52 -0700
Date: Mon, 06 Oct 2003 10:38:52 -0700

At 06:35 AM 10/6/03, you wrote:
> > I guess that depends on the actual format being streamed.
>
>mp4, ISO and ISMA, is the format that is (and will be) the most used to
>stream with rtsp, because of is complexity.

Many other codecs can also be streamed using RTSP/RTP - including MPEG-1 or 
2 audio and video, motion JPEG video, H.261, H.263 and H.263+ video, QCELP 
audio, PCM audio, Speex audio, etc. etc.  The "LIVE.COM streaming media 
libraries" support all of these codecs (as does MPlayer, which uses these 
libraries).  So, the use of RTSP/RTP should not be restricted to MPEG-4.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Subject: Re: [Live-devel] Re: [xine-devel] investigation about rtsp in  xine
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yes, I agree with you.

What I wanted to say, is that in my opinion mp4 is a file format that is
going to be more and more used in computers, palms, cellular phones, and
all others devices that stream small and compressed files.

Michel
 

Le lun 06/10/2003 Ã  13:38, Ross Finlayson a Ã©crit :
> At 06:35 AM 10/6/03, you wrote:
> > > I guess that depends on the actual format being streamed.
> >
> >mp4, ISO and ISMA, is the format that is (and will be) the most used to
> >stream with rtsp, because of is complexity.
> 
> Many other codecs can also be streamed using RTSP/RTP - including MPEG-1 or 
> 2 audio and video, motion JPEG video, H.261, H.263 and H.263+ video, QCELP 
> audio, PCM audio, Speex audio, etc. etc.  The "LIVE.COM streaming media 
> libraries" support all of these codecs (as does MPlayer, which uses these 
> libraries).  So, the use of RTSP/RTP should not be restricted to MPEG-4.
> 
> 
> 	Ross Finlayson
> 	LIVE.COM
> 	<http://www.live.com/>



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X-Original-Date: Mon, 06 Oct 2003 14:17:23 -0700
Date: Mon, 06 Oct 2003 14:17:23 -0700


>So do WE WORK ALL TOGETHER on this project ?

Sounds good to me.  FYI, I think the "live-devel(at)lists.sourceforge.net" 
mailing list is the best forum for continued discussion of this project - 
that is the mailing list for discussion of how to make use of the LIVE.COM 
libraries in new applications.

(Note, though, that you need to be s subscriber to that list in order to 
post to it.  To subscribe, visit 
<http://lists.sourceforge.net/lists/listinfo/live-devel>)


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Date: Tue, 7 Oct 2003 15:05:57 +0800

thank you for your hints,i'l try it again
when i finish this work,i'l post what i do in this project,maybe useful i
think




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 each frame
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X-Original-Date: Tue, 07 Oct 2003 02:46:19 -0700
Date: Tue, 07 Oct 2003 02:46:19 -0700

In the latest release (2003.10.07) of the "LIVE.COM Streaming Media" code, 
the "FileSink" class has been updated to take an optional Boolean parameter
"oneFilePerFrame".  If set, a separate file will be output for each
incoming frame.  (The frame's presentation time is used as a
file name suffix, to distinguish the files from each other.)

There is also a new option "-m" to "openRTSP" (and "playSIP").  This option
causes a separate file to be written for each frame (using the
above-mentioned new feature of "FileSink").

(Of course, if you use this feature, you'll typically generate a very large 
number of files.)

Thanks to Kobenhavns Lufthavne (Denmark) for funding this work.



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X-Original-Date: Wed, 8 Oct 2003 11:49:08 +0800
Date: Wed, 8 Oct 2003 11:49:08 +0800

>In the latest release (2003.10.07) of the "LIVE.COM Streaming Media" code,
>the "FileSink" class has been updated to take an optional Boolean
parameter
>"oneFilePerFrame".  If set, a separate file will be output for each
>incoming frame.  (The frame's presentation time is used as a
>file name suffix, to distinguish the files from each other.)
>
>There is also a new option "-m" to "openRTSP" (and "playSIP").  This
option
>causes a separate file to be written for each frame (using the
>above-mentioned new feature of "FileSink").
using the "-m" option, openRTSP will cause a Segmentation fault error when
it wants to write MPEG4 config data to a file,because now it is a null
pointer:
in FineSink.cpp:
FileSink* FileSink::createNew(UsageEnvironment& env, char const* fileName,
                        unsigned bufferSize, Boolean oneFilePerFrame) {
  FileSink* newSink = NULL;

  do {
    FILE* fid;
    char const* perFrameFileNamePrefix;
    if (oneFilePerFrame) {
      // Create the fid for each frame
      fid = NULL;
      perFrameFileNamePrefix = fileName;
    } else {
      // Normal case: create the fid once
      fid = openFileByName(env, fileName);
      if (fid == NULL) break;
      perFrameFileNamePrefix = NULL;
    }

    newSink = new FileSink(env, fid, bufferSize, perFrameFileNamePrefix);
    if (newSink == NULL) break;

    return newSink;
  } while (0);

  delete newSink;
  return NULL;
}
if the oneFilePerFrame paramenter is TRUE,FileSink::createNew will return a
NULL fid,so
the config data writing will cause error.

I'm not very familiar with MPEG,i think that if the "-m" option is used,the
config data writing should also be changed,right?




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X-Original-Date: Tue, 07 Oct 2003 22:50:01 -0700
Date: Tue, 07 Oct 2003 22:50:01 -0700


>using the "-m" option, openRTSP will cause a Segmentation fault error when
>it wants to write MPEG4 config data to a file,because now it is a null
>pointer:

You're right!  Thanks for finding this.

I have now installed a new version (2003.10.08) that fixes this bug.

         Ross.





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Subject: [Live-devel] RTCP APP packet.
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X-Original-Date: Thu, 09 Oct 2003 01:44:52 +0000
Date: Thu, 09 Oct 2003 01:44:52 +0000

Hi, 2 questions.
1) Is every RTCP packet sent in compound format? say, [ SR/RR + SDES + 
Optional single packet..]
2) If so, a usr difined RTCP APP packet should be add in this way:
       i)  Write a "RTCPInstance::addAPP()";  //add a APP report.
       ii)  Insert this "addAPP()" into "RTCPInstance::sendReport()", right 
behind the line of "addSDES()", before the line of "sentBuiltPacket()";  Is 
this right?

Thank you:)

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X-Original-Date: Wed, 08 Oct 2003 20:27:09 -0700
Date: Wed, 08 Oct 2003 20:27:09 -0700

At 06:44 PM 10/8/03, you wrote:
>Hi, 2 questions.
>1) Is every RTCP packet sent in compound format? say, [ SR/RR + SDES + 
>Optional single packet..]

Yes (as required by the RTP/RTCP specification).  This is done in 
"RTCPInstance::sendReport()".

>2) If so, a usr difined RTCP APP packet should be add in this way:
>       i)  Write a "RTCPInstance::addAPP()";  //add a APP report.
>       ii)  Insert this "addAPP()" into "RTCPInstance::sendReport()", 
> right behind the line of "addSDES()", before the line of 
> "sentBuiltPacket()";  Is this right?

Yes, that seems right.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Date: Thu, 9 Oct 2003 10:28:08 +0800


----- Original Message ----- 
From: {Ross Finlayson} <EMAIL-ADDRESS-HIDDEN>
To: <EMAIL-ADDRESS-HIDDEN>
Sent: Wednesday, October 08, 2003 1:50 PM
Subject: Re: [Live-devel] New FileSink (and openRTSP) feature: Output a new
file for each frame


>
> >using the "-m" option, openRTSP will cause a Segmentation fault error
when
> >it wants to write MPEG4 config data to a file,because now it is a null
> >pointer:
>
> You're right!  Thanks for finding this.
>
> I have now installed a new version (2003.10.08) that fixes this bug.
>
>          Ross.
>
>
>
>
>
> -------------------------------------------------------
> This sf.net email is sponsored by:ThinkGeek
> Welcome to geek heaven.
> http://thinkgeek.com/sf
> _______________________________________________
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>
>



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X-Original-Date: Thu, 9 Oct 2003 10:46:41 -0400
Date: Thu, 9 Oct 2003 10:46:41 -0400

Hi,
I'm streaming multiple copies of the same movie on the same client machine.
It appears that some of the client's store packets from the OTHER streams,
resulting in lots of BufferedPacket allocations.  These packets are not
freed until the stream is aborted.  Sometimes there are so many of these
packets that a stack fault occurs during the ~BufferedPacket destructor (due
to the recursive nature in which buffered packets are destroyed).  Or, if
the streams are allowed to run long enough, eventually the system runs out
of memory.

Is there a mechanism, perhaps in MultiFramedRTPSource::networkReadHandler(),
that I can implement to discard packets that are not destined to this
processes stream?  Can I tell from rtpSSRC if the packet is for the current
process or not?

Thanks,
Ed Estabrook



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X-Original-Date: Thu, 09 Oct 2003 08:47:53 -0700
Date: Thu, 09 Oct 2003 08:47:53 -0700


>I'm streaming multiple copies of the same movie on the same client machine.

Ed,

Can you explain this a bit more?  Are you trying to receive multiple 
streams within a single application (i.e., using a single instance of the 
LIVE.COM libraries)?  Or are you running separate copies of a receiving 
application - i.e., in different processes - on the same client?  Also, are 
your receivers trying to receive the same multicast stream (i.e., using the 
same multicast address and port number), or are they each receiving a 
separate unicast stream (i.e., sent to different client port numbers)?  Or 
are the different processes each trying to receive the *same* unicast stream?

>It appears that some of the client's store packets from the OTHER streams,

Again, in order to try to understand this, could you give us some more 
details about exactly what you're trying to do?


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Subject: RE: [Live-devel] Multiple streams on same machine
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X-Original-Date: Thu, 9 Oct 2003 13:26:50 -0400
Date: Thu, 9 Oct 2003 13:26:50 -0400

Ross,
Thanks for getting back so quickly!

There are two processes running separate copies of the recieving
application.  They both, at the same time, request from a Darwin server to
stream a movie, Rally.mov.  They are supposed to be two separate unicast
streams.  The 'RTSP/1.0 200 OK' response that comes back from the Darwin
server to both processes specifies 'client_port=6550-6551'.  Indeed
netstat -nap reveals that both processes have UDP sockets listening on 6550
and 6551 locally and 0.0.0.0:* remotely.  No multicast is involved... the
streams are supposed to be completely separate.  But the Darwin server is
sending both to the same client ports.

Based on this and some printf statements I added in the BufferedPacket
source it looks like both processes hear ALL streams' UDP packets, not just
the packets intended to itself.  The sequence numbers are consulted in
MultiFramedRTPSource::networkReadHandler, but I don't see anything that
ensures the packet is actually part of that process's stream at all.  So the
packet with a sequence number much too large or much too small is simply
stored for future playing.

So it seems the question is how does one tell that a packet is for a
particular stream or not?

In RFC 1889 I read in section 3:
       "All packets from a synchronization source form part of the same
        timing and sequence number space, so a receiver groups packets
        by synchronization source for playback.
        ...
        A participant need not use the same SSRC identifier for all the
        RTP sessions in a multimedia session; the binding of the SSRC
        identifiers is provided through RTCP (see Section 6.4.1).  If a
        participant generates multiple streams in one RTP session, for
        example from separate video cameras, each must be identified as
        a different SSRC."

I see that the SSRC value in the two RTP streams are different.  Is there a
list of 'correct' SSRC values stored in the library?  I've noticed code
elsewhere that observes that a SSRC value is new and adds it to
fKnownMembers (indirectly via NetMember).  Might the wrong SSRC value be put
into this table?  Or is verifying that a each packet's SSRC value is in this
table a viable means of determining if the packet should be queued or
discarded?  How are the SSRC values determined to begin with anyway?

Thanks,
Ed

-----Original Message-----
From: <EMAIL-ADDRESS-HIDDEN>
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Finlayson
Sent: Thursday, October 09, 2003 11:48 AM
To: <EMAIL-ADDRESS-HIDDEN>
Subject: Re: [Live-devel] Multiple streams on same machine



>I'm streaming multiple copies of the same movie on the same client machine.

Ed,

Can you explain this a bit more?  Are you trying to receive multiple
streams within a single application (i.e., using a single instance of the
LIVE.COM libraries)?  Or are you running separate copies of a receiving
application - i.e., in different processes - on the same client?  Also, are
your receivers trying to receive the same multicast stream (i.e., using the
same multicast address and port number), or are they each receiving a
separate unicast stream (i.e., sent to different client port numbers)?  Or
are the different processes each trying to receive the *same* unicast
stream?

>It appears that some of the client's store packets from the OTHER streams,

Again, in order to try to understand this, could you give us some more
details about exactly what you're trying to do?


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Thu, 09 Oct 2003 12:07:06 -0700
Date: Thu, 09 Oct 2003 12:07:06 -0700


>They are supposed to be two separate unicast
>streams.  The 'RTSP/1.0 200 OK' response that comes back from the Darwin
>server to both processes specifies 'client_port=6550-6551'.  Indeed
>netstat -nap reveals that both processes have UDP sockets listening on 6550
>and 6551 locally and 0.0.0.0:* remotely.  No multicast is involved... the
>streams are supposed to be completely separate.  But the Darwin server is
>sending both to the same client ports.

Then this is a bug in the Darwin server.  It should be using different 
client port ranges for each stream, because the destination IP address is 
the same in each case.

>Based on this and some printf statements I added in the BufferedPacket
>source it looks like both processes hear ALL streams' UDP packets, not just
>the packets intended to itself.

That's correct.  Each receiving application is listening to the same UDP 
port (6550), and the incoming packets are being sent to this same port 
(6550).  So each copy of the receiving application will receive a copy of 
each packet.

 From what I can tell, the receiving code is working properly.  The bug is 
in the Darwin server - it should not be sending both streams to the same 
port number.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Thu, 09 Oct 2003 12:22:16 -0700
Date: Thu, 09 Oct 2003 12:22:16 -0700

A correction to my earlier message: The client port number is actually 
specified by the RTSP *client*, when it does the RTSP "SETUP" 
operation.  So, the problem in this case is not with the RTSP *server* - 
it's with the RTSP *client*.

For some reason the RTSP client is choosing the same port number for its 
RTP socket.  It shouldn't be doing this; instead, different instances of 
the same process should be choosing different port numbers.  This may be an 
OS problem - I dunno... (What OS are you running?)

I suggest that you instrument the code for the "getPortNumber()" routine 
(in "liveMedia/MediaSession.cpp") to try to figure out why the same client 
port number is being chosen by both processes.  (I've tested this just now 
on FreeBSD, and it's not happening for me there.)

	Ross.

	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Subject: RE: [Live-devel] Multiple streams on same machine
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X-Original-Date: Thu, 9 Oct 2003 16:35:42 -0400
Date: Thu, 9 Oct 2003 16:35:42 -0400

Ross,
Found the problem.  I am embarrassed to confess that it was an accidental
change in playCommon.cpp that resulted in desiredPortNum being set within
the code.  The result was that all the processes started with port 6650.  So
this situation only appears if two different processes _explicitly_ request
the same local port.  Not nearly such a big problem.

Thanks for your help and patience, and sorry to have spent your time on such
a trivial mistake by me.

Best Regards,
Ed

-----Original Message-----
From: <EMAIL-ADDRESS-HIDDEN>
[mailto:live-devel-admin(at)lists.sourceforge.net]On Behalf Of Ross
Finlayson
Sent: Thursday, October 09, 2003 3:22 PM
To: <EMAIL-ADDRESS-HIDDEN>
Subject: RE: [Live-devel] Multiple streams on same machine


A correction to my earlier message: The client port number is actually
specified by the RTSP *client*, when it does the RTSP "SETUP"
operation.  So, the problem in this case is not with the RTSP *server* -
it's with the RTSP *client*.

For some reason the RTSP client is choosing the same port number for its
RTP socket.  It shouldn't be doing this; instead, different instances of
the same process should be choosing different port numbers.  This may be an
OS problem - I dunno... (What OS are you running?)

I suggest that you instrument the code for the "getPortNumber()" routine
(in "liveMedia/MediaSession.cpp") to try to figure out why the same client
port number is being chosen by both processes.  (I've tested this just now
on FreeBSD, and it's not happening for me there.)

	Ross.

	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Subject: [Live-devel] cross compiling error again
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X-Original-Date: Tue, 14 Oct 2003 17:22:39 +0800
Date: Tue, 14 Oct 2003 17:22:39 +0800

hi.
when I finish my cross compiling,there are many elf2flt error occur:
[root(at)test testProgs]# make
arm-elf-c++ -c -I../UsageEnvironment/include -I../groupsock/include
-I../liveMedia/include -I../BasicUsageEnvironment/include -I.
-I/tbx/uClinux-2.4/include -I../liveMedia -I../liveMedia/include
-I../BasicUsageEnvironment/include -I../groupsock/include
-I../UsageEnvironment/include  -DSOCKLEN_T=socklen_t -DBSD=1 -DCRIS
-fomit-frame-pointer -ffunction-sections -fdata-sections -O2 openRTSP.cpp
arm-elf-c++ -c -I../UsageEnvironment/include -I../groupsock/include
-I../liveMedia/include -I../BasicUsageEnvironment/include -I.
-I/tbx/uClinux-2.4/include -I../liveMedia -I../liveMedia/include
-I../BasicUsageEnvironment/include -I../groupsock/include
-I../UsageEnvironment/include  -DSOCKLEN_T=socklen_t -DBSD=1 -DCRIS
-fomit-frame-pointer -ffunction-sections -fdata-sections -O2 playCommon.cpp
arm-elf-gcc -r openRTSP.o playCommon.o -o rtsp.o -nostdlib
arm-elf-gcc --gc-sections  -Wl,-elf2flt -o rtsp.bin rtsp.o
rtsp.bin.elf2flt:/home/vincent/public/uClinux-sdk/gcc-2.95.3/build/gcc/../../gcc/cp/tinfo.cc:134:

undefined reference to `RTSPClient::createNew(UsageEnvironment &, int, char
const *)'
rtsp.bin.elf2flt:/home/vincent/public/uClinux-sdk/gcc-2.95.3/build/gcc/../../gcc/cp/tinfo.cc:134:

undefined reference to `RTSPClient::sendOptionsCmd(char const *)'
rtsp.bin.elf2flt:/home/vincent/public/uClinux-sdk/gcc-2.95.3/build/gcc/../../gcc/cp/tinfo.h:50:

undefined reference to `RTSPClient::describeWithPassword(char const *, char
const *, char const *)'
......

i have included all needed header files,but ..... these errors seems like
that
linker cannot file these functions,right?
thanks!




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X-Original-Date: Tue, 14 Oct 2003 13:15:46 -0700
Date: Tue, 14 Oct 2003 13:15:46 -0700


>arm-elf-gcc --gc-sections  -Wl,-elf2flt -o rtsp.bin rtsp.o
>rtsp.bin.elf2flt:/home/vincent/public/uClinux-sdk/gcc-2.95.3/build/gcc/../../gcc/cp/tinfo.cc:134:
>
>undefined reference to `RTSPClient::createNew(UsageEnvironment &, int, char
>const *)'

This is easy - you're not including the "LIVE.COM Streaming Media
  libraries (libliveMedia, libgroupsock, libUsageEnvironment, 
libBasicUsageEnvironment) in the linking command line.

Once again, don't modify the "Makefile"s (unless you know what you're doing 
:-).  Instead, modify the appropriate "config.*" file (if necessary), and 
re-run "genMakefiles".


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Wed, 15 Oct 2003 15:06:42 +0800
Date: Wed, 15 Oct 2003 15:06:42 +0800

>This is easy - you're not including the "LIVE.COM Streaming Media
>  libraries (libliveMedia, libgroupsock, libUsageEnvironment,
>libBasicUsageEnvironment) in the linking command line.
>Once again, don't modify the "Makefile"s (unless you know what you're
doing
>:-).  Instead, modify the appropriate "config.*" file (if necessary), and
>re-run "genMakefiles".
thanks for your hints,i have modified the Makefile by myself.
because my toolchain have no C++ library,only C library,and I want to use
liveMedia to create a ISMA client(just like openRTSP) in my uclinux
platform,so I modified the source code,removed/replaced some C++ codes.
following is my config file:
AXIS_DIR = /opt/tools-install/arm-elf
COMPILE_OPTS = $(INCLUDES) -I. -I/tbx/uClinux-2.4/include -I../liveMedia
-I../liveMedia/include -I../BasicUsageEnvironment/include
-I../groupsock/include -I../UsageEnvironment/include  -DSOCKLEN_T=socklen_t
-DBSD=1 -DCRIS
C =               c
C_COMPILER =            arm-elf-gcc
C_FLAGS =         $(COMPILE_OPTS) -O2
CPP =             cpp
CPLUSPLUS_COMPILER =    arm-elf-c++
CPLUSPLUS_FLAGS = $(COMPILE_OPTS) -fomit-frame-pointer -ffunction-sections
-fdata-sections -O2
OBJ =             o
LINK =                  arm-elf-c++ -o
AXIS_LINK_OPTS =  -L$(AXIS_DIR)/lib
LINK_OPTS =       -L.
CONSOLE_LINK_OPTS =     $(LINK_OPTS) -L$(AXIS_DIR)/lib -Wl,-elf2flt
-nostdlib
LIBRARY_LINK =          arm-elf-ld  -o
LIBRARY_LINK_OPTS =     $(LINK_OPTS) -r
LIB_SUFFIX =                  a
LIBS_FOR_CONSOLE_APPLICATION =
LIBS_FOR_GUI_APPLICATION =
EXE =
with this config file,the compile is ok,but when linking,errors occur:
arm-elf-c++ -c -I../UsageEnvironment/include -I../groupsock/include
-I../liveMedia/include -I../BasicUsageEnvironment/include -I.
-I/tbx/uClinux-2.4/include -I../liveMedia -I../liveMedia/include
-I../BasicUsageEnvironment/include -I../groupsock/include
-I../UsageEnvironment/include  -DSOCKLEN_T=socklen_t -DBSD=1 -DCRIS
-fomit-frame-pointer -ffunction-sections -fdata-sections -O2 openRTSP.cpp
arm-elf-c++ -c -I../UsageEnvironment/include -I../groupsock/include
-I../liveMedia/include -I../BasicUsageEnvironment/include -I.
-I/tbx/uClinux-2.4/include -I../liveMedia -I../liveMedia/include
-I../BasicUsageEnvironment/include -I../groupsock/include
-I../UsageEnvironment/include  -DSOCKLEN_T=socklen_t -DBSD=1 -DCRIS
-fomit-frame-pointer -ffunction-sections -fdata-sections -O2 playCommon.cpp
arm-elf-c++ -o openRTSP -L. -L/opt/tools-install/arm-elf/lib -Wl,-elf2flt
-nostdlib openRTSP.o playCommon.o ../liveMedia/libliveMedia.a
../groupsock/libgroupsock.a ../UsageEnvironment/libUsageEnvironment.a
../BasicUsageEnvironment/libBasicUsageEnvironment.a
/opt/tools-install/arm-elf/bin/ld.real: warning: cannot find entry symbol
_start; defaulting to 00000000
openRTSP.elf2flt: In function `rtcp_interval':
openRTSP.elf2flt(.text+0x64): undefined reference to `__floatsidf'
openRTSP.elf2flt(.text+0x74): undefined reference to `__floatsidf'
openRTSP.elf2flt(.text+0x80): undefined reference to `__muldf3'
openRTSP.elf2flt(.text+0x94): undefined reference to `__ltdf2'
openRTSP.elf2flt(.text+0xbc): undefined reference to `__muldf3'
openRTSP.elf2flt(.text+0xfc): undefined reference to `__muldf3'

looks like no C lib,but i have specified the C library directory in config
file




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oh,forget to say,the rtsp.o contains all needed .o objects and .a livemeida
librares,:
arm-elf-c++ -o rtsp.o  -L. -L/opt/tools-install/arm-elf/lib -r  openRTSP.o
playCommon.o ../liveMedia/libliveMedia.a  ../groupsock/libgroupsock.a
../UsageEnvironment/libUsageEnvironment.a
../BasicUsageEnvironment/libBasicUsageEnvironment.a  -nostdlib
arm-elf-c++ --gc-sections -Wl,-elf2flt -o rtsp.bin rtsp.o




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Subject: [Live-devel] Why testOnDemandRTSPServer works as expected
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 Hi,
	I have run the program "testOnDemandRTSPServer",and then run 
"openRTSP -v rtsp://ipaddr:7070/mpeg1or2ESVideoTest",but nothing appeared
on the screen. What's wrong with the programs?
 				

¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡wanghao
¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡wanghao(at)net.pku.edu.cn
¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡2003-10-20




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>         I have run the program "testOnDemandRTSPServer",and then run
>"openRTSP -v rtsp://ipaddr:7070/mpeg1or2ESVideoTest",but nothing appeared
>on the screen. What's wrong with the programs?

The diagnostic output from "openRTSP" appears on 'stderr'.  If you're 
running a brain-damaged operating system (e.g., some older versions of 
Windows), then perhaps 'stderr' doesn't appear on the console.

Try running
         openRTSP -V -n rtsp://ipaddr:7070/mpeg1or2ESVideoTest 2> stderr.txt
then look at the contents of the file "stderr.txt"


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Subject: [Live-devel] frame rate control ?
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X-Original-Date: Fri, 24 Oct 2003 15:56:28 +0200
Date: Fri, 24 Oct 2003 15:56:28 +0200

Hi,

I am using liveMedia libraries to create an outgoing video stream from a 
visualization application.
The question I was unable to find the answer to on my own is : how can I 
specify the minimum/maximum,
desired frame rate at which I want to send out video frames.
More specific : I've implemented a FramedSource class and I'm using a 
MPEG4VideoStreamFramer
as in testMPEG4VideoStreamer to generate the video stream.
What I cannot do is to specify the frequency in which doGetNextFrame() 
in the FramedSource implementation
gets called. So what I get is  a very low cpu usage but only 1-2 fps and 
lots of idle time. My computer could do more
and it does so as soon as I start a vic client that connects to the 
multicast adress that I'm streaming to.
When vic runs, after a time my streaming server increases the frame rate 
to 15-16 fps
and the cpu usage goes to 100% (Because of the mpeg engoder I guess).
Both these modes are good for diferent scenarios but I would like to be 
able to steer
the switch between them somehow on server side. Any ideas ?

If it helps : I'm also starting a RTSP server in parallel ..


Thanks,
Andrei Hutanu






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X-Original-Date: Fri, 24 Oct 2003 09:20:16 -0700
Date: Fri, 24 Oct 2003 09:20:16 -0700


>I am using liveMedia libraries to create an outgoing video stream from a 
>visualization application.
>The question I was unable to find the answer to on my own is : how can I 
>specify the minimum/maximum,
>desired frame rate at which I want to send out video frames.
>More specific : I've implemented a FramedSource class and I'm using a 
>MPEG4VideoStreamFramer
>as in testMPEG4VideoStreamer to generate the video stream.
>What I cannot do is to specify the frequency in which doGetNextFrame() in 
>the FramedSource implementation
>gets called.

Andrei,

The "getPlayTime()" function is used to determine 
this.  "MultiFramedRTPSink" (from which other "*RTPSink" classes are 
subclassed) calls this function to figure out how long to delay before 
asking for the next frame.

For "MPEG4VideoStreamFramer", "getPlayTime()" is implemented in the parent 
class - "MPEGVideoStreamFramer".  It uses the "fFrameRate" variable, which 
- for "MPEG4VideoStreamFramer" - is computed by looking at the 
"vop_time_increment_resolution" from the MPEG-4 VOL header.

So, your problem seems to be that the frame rate that you want from your 
MPEG-4 video stream does not match the rate that's specified by the MPEG-4 
VOL header.  If that's the case, then you should be able to overcome this 
by defining your own subclass of "MPEG4VideoStreamFramer" that redefines 
"getPlayTime()" (but does nothing else new).

By the way, note that the "getPlayTime()" function will probably be going 
away in some future version of the library (probably to be released 
sometime within the next few weeks), because it's an ugly hack, and is 
currently being used only to determine the time duration of individual 
frames.  Instead, a new 'frame duration' result parameter will be added to 
"doGetNextFrame()".  Because this will be a significant change to the API, 
I will, of course, be announcing this change to the mailing list when it 
happens.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Subject: [Live-devel] testOnDemandRTSPServer & openRTSP
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hello!
	I am a undergraduated student from Peking University in China,Our lab is working on the streaming protocol,we are using RTSP of LIVE.COM
	We use the 2003-10-4 version of LIVE.COM.
    There are our problems:
    We use the command "./testOnDemandRTSPServer" ,Then we use "openRTSP rtsp://ipaddr:7070/mpeg1or2ESVideoTest" to connect the server, the result is that
the rtsp C->S,S->C is ok, But wo received nothing.
	When we use the qtplayer to connect the server,it is ok, also we can recieve the data.
    
    I don not Why?
thank you !    
	
 				

¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡Ren Maosheng
¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡rms(at)net.cs.pku.edu.cn
¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡2003-10-23





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>    There are our problems:
>     We use the command "./testOnDemandRTSPServer" ,Then we use "openRTSP 
> rtsp://ipaddr:7070/mpeg1or2ESVideoTest" to connect the server, the result 
> is that
>the rtsp C->S,S->C is ok, But wo received nothing.

Maosheng,

Please add the "-V" and "-n" command-line options to "openRTSP".  This will 
give you more information about what is happening.

In this case, "openRTSP" should be creating a file named "video-MPV-1"; 
that file should contain the MPEG video data received from the server.




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Subject: [Live-devel] frame rate control ?
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X-Original-Date: Fri, 24 Oct 2003 12:58:29 +0200
Date: Fri, 24 Oct 2003 12:58:29 +0200

Hi,

I am using liveMedia libraries to create an outgoing video stream from a 
visualization application.
The question I was unable to find the answer to on my own is : how can I 
specify the minimum/maximum,
desired frame rate at which I want to send out video frames.
More specific : I've implemented a FramedSource class and I'm using a 
MPEG4VideoStreamFramer
as in testMPEG4VideoStreamer to generate the video stream.
What I cannot do is to specify the frequency in which doGetNextFrame() 
in the FramedSource implementation
gets called. So what I get is  a very low cpu usage but only 1-2 fps and 
lots of idle time. My computer could do more
and it does so as soon as I start a vic client that connects to the 
multicast adress that I'm streaming to.
When vic runs, after a time my streaming server increases the frame rate 
to 15-16 fps
and the cpu usage goes to 100% (Because of the mpeg engoder I guess).
Both these modes are good for diferent scenarios but I would like to be 
able to steer
the switch between them somehow on server side. Any ideas ?

If it helps : I'm also starting a RTSP server in parallel ..


Thanks,
Andrei Hutanu




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X-Original-Date: Mon, 27 Oct 2003 20:50:59 +0100
Date: Mon, 27 Oct 2003 20:50:59 +0100

Hi,

Sorry for the double message.
Thanks for your answer.

I looked at the method you pointed out and it's usage in 
MultiFramedRTPSink::sendPacketIfNecesary().
That doesn't seem to be the problem as the calculated uSecondsToGo is 
almost always (just for the first frame is different than) 0,
so that doesn't delay the transmission. Reimplementing the getPlayTime() 
has then no influence.
The sendNext method is called immediately ...

So there must be something else that stops the servers from sending at 
full capacity..
The fact that the transmission rate increases when I connect with vic to 
visualize the stream
means maybe something ?

Andrei

Ross Finlayson wrote:

>
>> I am using liveMedia libraries to create an outgoing video stream 
>> from a visualization application.
>> The question I was unable to find the answer to on my own is : how 
>> can I specify the minimum/maximum,
>> desired frame rate at which I want to send out video frames.
>> More specific : I've implemented a FramedSource class and I'm using a 
>> MPEG4VideoStreamFramer
>> as in testMPEG4VideoStreamer to generate the video stream.
>> What I cannot do is to specify the frequency in which 
>> doGetNextFrame() in the FramedSource implementation
>> gets called.
>
>
> Andrei,
>
> The "getPlayTime()" function is used to determine this.  
> "MultiFramedRTPSink" (from which other "*RTPSink" classes are 
> subclassed) calls this function to figure out how long to delay before 
> asking for the next frame.
>
> For "MPEG4VideoStreamFramer", "getPlayTime()" is implemented in the 
> parent class - "MPEGVideoStreamFramer".  It uses the "fFrameRate" 
> variable, which - for "MPEG4VideoStreamFramer" - is computed by 
> looking at the "vop_time_increment_resolution" from the MPEG-4 VOL 
> header.
>
> So, your problem seems to be that the frame rate that you want from 
> your MPEG-4 video stream does not match the rate that's specified by 
> the MPEG-4 VOL header.  If that's the case, then you should be able to 
> overcome this by defining your own subclass of 
> "MPEG4VideoStreamFramer" that redefines "getPlayTime()" (but does 
> nothing else new).
>
> By the way, note that the "getPlayTime()" function will probably be 
> going away in some future version of the library (probably to be 
> released sometime within the next few weeks), because it's an ugly 
> hack, and is currently being used only to determine the time duration 
> of individual frames.  Instead, a new 'frame duration' result 
> parameter will be added to "doGetNextFrame()".  Because this will be a 
> significant change to the API, I will, of course, be announcing this 
> change to the mailing list when it happens.
>
>
>     Ross Finlayson
>     LIVE.COM
>     <http://www.live.com/>
>
>
>
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Subject: Re: [Live-devel] frame rate control ?
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>I looked at the method you pointed out and it's usage in 
>MultiFramedRTPSink::sendPacketIfNecesary().
>That doesn't seem to be the problem as the calculated uSecondsToGo is 
>almost always (just for the first frame is different than) 0,
>so that doesn't delay the transmission. Reimplementing the getPlayTime() 
>has then no influence.
>The sendNext method is called immediately ...
>
>So there must be something else that stops the servers from sending at 
>full capacity..

No, there's nothing else.  It's the call to "getPlayTime()" (in 
"MultiFramedRTPSink::sendPacketIfNecessary()") that determines when the 
server should ask for the next frame to send.  (Note the computation of 
"fNextSendTime".)

I suggest that you start from the existing "testMPEG4VideoStreamer" 
application - which works - and figure out how your modifications changed 
this.  (See <http://www.live.com/liveMedia/faq.html#my-modification-broke>)


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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From <EMAIL-ADDRESS-HIDDEN> Tue Oct 28 01:32:32 2003
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Hi,

We have been experimenting streaming MPEG2 using
testProgs/testMPEG1or2VideoStreamer.
So far we have been streaming from pre-generated files, and it works
just fine.
Recently we installed a Hauppauge PVR-350 card under Redhat Linux 7.3,
the setup seems to work correctly, mplayer can decode the generated
stream when fed (via stdin) directly from the video device
(/dev/video0), so can mpeg2dec.
However if we modify testProgs/testMPEG1or2VideoStreamer to read from
/dev/video0 (simply by replacing test.mpg with /dev/video0) the stream
seem to be partly invalidated somehow - sympom is lots of 'pixellation'
when decoding.
If we generate a file reading from the video device, eg. Cat /dev/video0
> jantest1.mpg and then stream this file it works just fine as well.

Any help will be appreciated !

- Jan



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From <EMAIL-ADDRESS-HIDDEN> Tue Oct 28 02:03:18 2003
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Hi,

We have been experimenting streaming MPEG2 using
testProgs/testMPEG1or2VideoStreamer.
So far we have been streaming from pre-generated files, and it works
just fine.
Recently we installed a Hauppauge PVR-350 card under Redhat Linux 7.3,
the setup seems to work correctly, mplayer can decode the generated
stream when fed (via stdin) directly from the video device
(/dev/video0), so can mpeg2dec.
However if we modify testProgs/testMPEG1or2VideoStreamer to read from
/dev/video0 (simply by replacing test.mpg with /dev/video0) the stream
seem to be partly invalidated somehow - sympom is lots of 'pixellation'
when decoding.
If we generate a file reading from the video device, eg. Cat /dev/video0
> jantest1.mpg and then stream this file it works just fine as well.

Any help will be appreciated !

- Jan



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>We have been experimenting streaming MPEG2 using
>testProgs/testMPEG1or2VideoStreamer.
>So far we have been streaming from pre-generated files, and it works
>just fine.
>Recently we installed a Hauppauge PVR-350 card under Redhat Linux 7.3,
>the setup seems to work correctly, mplayer can decode the generated
>stream when fed (via stdin) directly from the video device
>(/dev/video0), so can mpeg2dec.
>However if we modify testProgs/testMPEG1or2VideoStreamer to read from
>/dev/video0 (simply by replacing test.mpg with /dev/video0) the stream
>seem to be partly invalidated somehow - sympom is lots of 'pixellation'
>when decoding.

It sounds like there is data loss happening somewhere - most likely within 
MPlayer.  I have noticed the typically high bitrate of MPEG-2 streams can 
sometimes cause MPlayer problems, even when run on a fast CPU.  I'm not 
sure why exactly this is happening, unfortunately.

As a test, you could try modifying "testMPEG1or2VideoStreamer" to send 
I-frames only.  (Do this by setting "iFramesOnly = True" in 
"testMPEG1or2VideoStreamer.cpp".)

Another thing you could try is running "testMPEG1or2VideoReceiver" to 
record the incoming MPEG-2/RTP data into a file, and then try playing this 
file.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Tue, 28 Oct 2003 12:23:12 +0100
Date: Tue, 28 Oct 2003 12:23:12 +0100

OK. Getting closer. Thanks, I found the spot.

As I described before, I have started from the test streamer 
application(s) and just implemented a
new FramedSource as indicated in 
http://www.live.com/liveMedia/faq.html#liveInput

It seems that the delay is caused  by a delay  in the call of 
afterGetting when I do a

nextTask()
    = envir().taskScheduler().scheduleDelayedTask(0, 
(TaskFunc*)afterGetting,
                          this);

after the frame is encoded.
My guess that this is because my FramedSource is implemented as a 
separate thread.

I was hoping this will work fine but I guess I was wrong. Can somebody 
give me a hint
about what I need to do to get rid of the delay ? I guess I need to wake 
up the rtsp thread
somehow, and this is what I think that happens when I connect with vic 
to the stream.
Is there a way to wake up this thread from the "inside" ?


Thanks,
Andrei


>
> No, there's nothing else.  It's the call to "getPlayTime()" (in 
> "MultiFramedRTPSink::sendPacketIfNecessary()") that determines when 
> the server should ask for the next frame to send.  (Note the 
> computation of "fNextSendTime".)
>
> I suggest that you start from the existing "testMPEG4VideoStreamer" 
> application - which works - and figure out how your modifications 
> changed this.  (See 
> <http://www.live.com/liveMedia/faq.html#my-modification-broke>)
>
>
>     Ross Finlayson
>     LIVE.COM
>     <http://www.live.com/>
>
>
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Date: Tue, 28 Oct 2003 03:43:20 -0800


>My guess that this is because my FramedSource is implemented as a separate 
>thread.

Arggh!  No, don't do this.  You can't use multiple threads with this code 
(unless each thread uses its own "UsageEnvironment" and 
"TaskScheduler").  The code is designed to use a single thread of 
execution, using an event loop - rather than multiple threads - for 
concurrency.  (Note the call to "TaskScheduler::doEventLoop()" in each of 
the test programs.)


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Tue, 28 Oct 2003 13:23:24 +0100
Date: Tue, 28 Oct 2003 13:23:24 +0100

Oh, that's very sad ...
Maybe multiple threads will be supported in the future.. ;)
It sems to "almost work" for my case .. so maybe it's not very 
complicated to implement.

Thanks for your help.

Andrei

Ross Finlayson wrote:

>
>> My guess that this is because my FramedSource is implemented as a 
>> separate thread.
>
>
> Arggh!  No, don't do this.  You can't use multiple threads with this 
> code (unless each thread uses its own "UsageEnvironment" and 
> "TaskScheduler").  The code is designed to use a single thread of 
> execution, using an event loop - rather than multiple threads - for 
> concurrency.  (Note the call to "TaskScheduler::doEventLoop()" in each 
> of the test programs.)
>
>
>     Ross Finlayson
>     LIVE.COM
>     <http://www.live.com/>
>
>
>
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Subject: SV: [Live-devel] Streaming MPEG2 directly from video device
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Ross,

I have already tried recording the stream to a file on the receiving
side (using openRTSP) - this doesn't change anything, ie. The recorded
file has the same problem. Somehow it seems that reading directly from
the video device causes the problems.
"iFramesOnly = True" helps a lot, in the sense that the stream is
decoded without errors, but of course, not smooth.
The box on which the test is running has a permanent load on about 2,
and the load seem to be io-bound. 
Since we have the same problem with the recorded file, the dataloss must
happen in the testMPEG1or2VideoStreamer (??)
Still I find it strange since reading directly from file seem to be
working fine. 

- Jan


-----Oprindelig meddelelse-----
Fra: Ross Finlayson [mailto:finlayson(at)live.com] 
Sendt: 28. oktober 2003 12:16
Til: live-devel(at)lists.sourceforge.net
Emne: Re: [Live-devel] Streaming MPEG2 directly from video device



>We have been experimenting streaming MPEG2 using 
>testProgs/testMPEG1or2VideoStreamer.
>So far we have been streaming from pre-generated files, and it works 
>just fine. Recently we installed a Hauppauge PVR-350 card under Redhat 
>Linux 7.3, the setup seems to work correctly, mplayer can decode the 
>generated stream when fed (via stdin) directly from the video device
>(/dev/video0), so can mpeg2dec.
>However if we modify testProgs/testMPEG1or2VideoStreamer to read from
>/dev/video0 (simply by replacing test.mpg with /dev/video0) the stream
>seem to be partly invalidated somehow - sympom is lots of 'pixellation'
>when decoding.

It sounds like there is data loss happening somewhere - most likely
within 
MPlayer.  I have noticed the typically high bitrate of MPEG-2 streams
can 
sometimes cause MPlayer problems, even when run on a fast CPU.  I'm not 
sure why exactly this is happening, unfortunately.

As a test, you could try modifying "testMPEG1or2VideoStreamer" to send 
I-frames only.  (Do this by setting "iFramesOnly = True" in 
"testMPEG1or2VideoStreamer.cpp".)

Another thing you could try is running "testMPEG1or2VideoReceiver" to 
record the incoming MPEG-2/RTP data into a file, and then try playing
this 
file.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Ross,

I have tried looking at your code in BasicTaskScheduler.cpp, trying to
figure out if the problem with our video-device could somehow be
explained by a combination of the way your code read data from the
device, and the way the PVR-350 device driver makes data available,
could the problem be that data becomes available in rather large chunks,
leaving your code "idle" in some periods and overloaded in other ??

- Jan

-----Oprindelig meddelelse-----
Fra: Ross Finlayson [mailto:finlayson(at)live.com] 
Sendt: 28. oktober 2003 12:16
Til: live-devel(at)lists.sourceforge.net
Emne: Re: [Live-devel] Streaming MPEG2 directly from video device



>We have been experimenting streaming MPEG2 using 
>testProgs/testMPEG1or2VideoStreamer.
>So far we have been streaming from pre-generated files, and it works 
>just fine. Recently we installed a Hauppauge PVR-350 card under Redhat 
>Linux 7.3, the setup seems to work correctly, mplayer can decode the 
>generated stream when fed (via stdin) directly from the video device
>(/dev/video0), so can mpeg2dec.
>However if we modify testProgs/testMPEG1or2VideoStreamer to read from
>/dev/video0 (simply by replacing test.mpg with /dev/video0) the stream
>seem to be partly invalidated somehow - sympom is lots of 'pixellation'
>when decoding.

It sounds like there is data loss happening somewhere - most likely
within 
MPlayer.  I have noticed the typically high bitrate of MPEG-2 streams
can 
sometimes cause MPlayer problems, even when run on a fast CPU.  I'm not 
sure why exactly this is happening, unfortunately.

As a test, you could try modifying "testMPEG1or2VideoStreamer" to send 
I-frames only.  (Do this by setting "iFramesOnly = True" in 
"testMPEG1or2VideoStreamer.cpp".)

Another thing you could try is running "testMPEG1or2VideoReceiver" to 
record the incoming MPEG-2/RTP data into a file, and then try playing
this 
file.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Subject: SV: [Live-devel] Streaming MPEG2 directly from video device
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Ross,

I have tried looking at your code in BasicTaskScheduler.cpp, trying to
figure out if the problem with our video-device could somehow be
explained by a combination of the way your code read data from the
device, and the way the PVR-350 device driver makes data available,
could the problem be that data becomes available in rather large chunks,
leaving your code "idle" in some periods and overloaded in other ??

- Jan

-----Oprindelig meddelelse-----
Fra: Ross Finlayson [mailto:finlayson(at)live.com] 
Sendt: 28. oktober 2003 12:16
Til: live-devel(at)lists.sourceforge.net
Emne: Re: [Live-devel] Streaming MPEG2 directly from video device



>We have been experimenting streaming MPEG2 using 
>testProgs/testMPEG1or2VideoStreamer.
>So far we have been streaming from pre-generated files, and it works 
>just fine. Recently we installed a Hauppauge PVR-350 card under Redhat 
>Linux 7.3, the setup seems to work correctly, mplayer can decode the 
>generated stream when fed (via stdin) directly from the video device
>(/dev/video0), so can mpeg2dec.
>However if we modify testProgs/testMPEG1or2VideoStreamer to read from
>/dev/video0 (simply by replacing test.mpg with /dev/video0) the stream
>seem to be partly invalidated somehow - sympom is lots of 'pixellation'
>when decoding.

It sounds like there is data loss happening somewhere - most likely
within 
MPlayer.  I have noticed the typically high bitrate of MPEG-2 streams
can 
sometimes cause MPlayer problems, even when run on a fast CPU.  I'm not 
sure why exactly this is happening, unfortunately.

As a test, you could try modifying "testMPEG1or2VideoStreamer" to send 
I-frames only.  (Do this by setting "iFramesOnly = True" in 
"testMPEG1or2VideoStreamer.cpp".)

Another thing you could try is running "testMPEG1or2VideoReceiver" to 
record the incoming MPEG-2/RTP data into a file, and then try playing
this 
file.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



-------------------------------------------------------
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Ross,

I have tried looking at your code in BasicTaskScheduler.cpp, trying to
figure out if the problem with our video-device could somehow be
explained by a combination of the way your code read data from the
device, and the way the PVR-350 device driver makes data available,
could the problem be that data becomes available in rather large chunks,
leaving your code "idle" in some periods and overloaded in other ??

- Jan

-----Oprindelig meddelelse-----
Fra: Ross Finlayson [mailto:finlayson(at)live.com] 
Sendt: 28. oktober 2003 12:16
Til: live-devel(at)lists.sourceforge.net
Emne: Re: [Live-devel] Streaming MPEG2 directly from video device



>We have been experimenting streaming MPEG2 using 
>testProgs/testMPEG1or2VideoStreamer.
>So far we have been streaming from pre-generated files, and it works 
>just fine. Recently we installed a Hauppauge PVR-350 card under Redhat 
>Linux 7.3, the setup seems to work correctly, mplayer can decode the 
>generated stream when fed (via stdin) directly from the video device
>(/dev/video0), so can mpeg2dec.
>However if we modify testProgs/testMPEG1or2VideoStreamer to read from
>/dev/video0 (simply by replacing test.mpg with /dev/video0) the stream
>seem to be partly invalidated somehow - sympom is lots of 'pixellation'
>when decoding.

It sounds like there is data loss happening somewhere - most likely
within 
MPlayer.  I have noticed the typically high bitrate of MPEG-2 streams
can 
sometimes cause MPlayer problems, even when run on a fast CPU.  I'm not 
sure why exactly this is happening, unfortunately.

As a test, you could try modifying "testMPEG1or2VideoStreamer" to send 
I-frames only.  (Do this by setting "iFramesOnly = True" in 
"testMPEG1or2VideoStreamer.cpp".)

Another thing you could try is running "testMPEG1or2VideoReceiver" to 
record the incoming MPEG-2/RTP data into a file, and then try playing
this 
file.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



-------------------------------------------------------
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X-Original-Date: Tue, 28 Oct 2003 09:24:05 -0800
Date: Tue, 28 Oct 2003 09:24:05 -0800

At 04:23 AM 10/28/03, you wrote:
>Oh, that's very sad ...
>Maybe multiple threads will be supported in the future.. ;)

Well, multiple threads are supported now, *provided that* each thread uses 
its own "UsageEnvironment" and "TaskScheduler" object (or does not access 
the library at all).

That is, a single "TaskScheduler" object must be used within a single 
thread.  The library's execution model is to provide concurrency using a 
single threaded event loop.

I doubt you need more than one thread in your application.  Try 
re-implementing it as just a single thread of control, as in the provided 
test programs.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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>Since we have the same problem with the recorded file, the dataloss must
>happen in the testMPEG1or2VideoStreamer (??)

Not necessarily.  The data loss could also be happening at the receiving 
end (because the same library code is used to implement MPEG/RTP reception 
in "MPlayer", "openRTSP", and "testMPEG1or2VideoReceiver".

(That reminds me: You should make sure that you're using an up-to-date 
version of the LIVE.COM library when building each of these applications.)


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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Subject: Re: [Live-devel] Streaming MPEG2 directly from video device
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>I have tried looking at your code in BasicTaskScheduler.cpp, trying to
>figure out if the problem with our video-device could somehow be
>explained by a combination of the way your code read data from the
>device, and the way the PVR-350 device driver makes data available,
>could the problem be that data becomes available in rather large chunks,
>leaving your code "idle" in some periods and overloaded in other ??

I don't think so.  "testMPEG1or2VideoStreamer" - like all applications that 
use a "RTPSink" object - will read, at a time, only as much data as is 
needed to pack into a single outgoing RTP packet.  Also, after sending each 
RTP packet, it will delay an appropriate amount of time before going back 
and reading more data.  (Note my earlier answers to Andrei's "frame rate 
control?" question.)  I.e., "testMPEG1or2VideoStreamer" will send its RTP 
packets - and read from its input "MPEG1or2VideoStreamFramer" object - at a 
steady pace.  (However, the "MPEG1or2VideoStreamFramer" will, in turn, read 
from the input file (a "ByteStreamFileSource") in a more bursty manner.  It 
has large buffers ('banks') that it tries to keep full.  See 
"StreamParser::ensureValidBytes1()" to understand this.)

Nonetheless, this means that your PVR-350 input device 'file' - as 
implemented within the operating system kernel - needs to buffer data 
properly, because the "ByteStreamFileSource" object that reads from this 
device does not always read, at once, all of the data that's 
available.  E.g., if the "ByteStreamFileSource" reads 150000 bytes from the 
device, and (e.g.) 200000 bytes is currently available, then the operating 
system needs to make sure that the remaining 50000 bytes is buffered, and 
is available for the next time the device is read.

So, you might want to look at your input device driver, to make sure that 
it's buffering data properly.  One thing you try is modify 
"ByteStreamFileSource" to save - into a separate file - a copy of each 
block of data that it reads from the input device.  Then, you could try 
playing this file.  If the file has data loss, then this means that your 
input device driver is not buffering data properly.  If, however, the file 
*does not* have data loss, then this means that your device driver is 
probably OK, and that the data loss is occurring elsewhere.

         Ross.

ps. None of this has anything to do with "BasicTaskScheduler".  That's a 
'red herring'.




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Subject: SV: [Live-devel] Streaming MPEG2 directly from video device
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Thank you for your thorough reply. I'll perform the tests you suggest.
About "The red Herring" in BasicTaskScheduler :-). My search was (as my
understanding of your code is, I must admit) indeed superficial, but I
was searching for the place in your code that controls the frequency at
which data are read from various fd's or sockets, and the timeout
parameters to your select calls - I'll be more thorough next time. 

- Jan

-----Oprindelig meddelelse-----
Fra: Ross Finlayson [mailto:finlayson(at)live.com] 
Sendt: 28. oktober 2003 19:05
Til: live-devel(at)lists.sourceforge.net
Emne: Re: [Live-devel] Streaming MPEG2 directly from video device



>I have tried looking at your code in BasicTaskScheduler.cpp, trying to 
>figure out if the problem with our video-device could somehow be 
>explained by a combination of the way your code read data from the 
>device, and the way the PVR-350 device driver makes data available, 
>could the problem be that data becomes available in rather large 
>chunks, leaving your code "idle" in some periods and overloaded in 
>other ??

I don't think so.  "testMPEG1or2VideoStreamer" - like all applications
that 
use a "RTPSink" object - will read, at a time, only as much data as is 
needed to pack into a single outgoing RTP packet.  Also, after sending
each 
RTP packet, it will delay an appropriate amount of time before going
back 
and reading more data.  (Note my earlier answers to Andrei's "frame rate

control?" question.)  I.e., "testMPEG1or2VideoStreamer" will send its
RTP 
packets - and read from its input "MPEG1or2VideoStreamFramer" object -
at a 
steady pace.  (However, the "MPEG1or2VideoStreamFramer" will, in turn,
read 
from the input file (a "ByteStreamFileSource") in a more bursty manner.
It 
has large buffers ('banks') that it tries to keep full.  See 
"StreamParser::ensureValidBytes1()" to understand this.)

Nonetheless, this means that your PVR-350 input device 'file' - as 
implemented within the operating system kernel - needs to buffer data 
properly, because the "ByteStreamFileSource" object that reads from this

device does not always read, at once, all of the data that's 
available.  E.g., if the "ByteStreamFileSource" reads 150000 bytes from
the 
device, and (e.g.) 200000 bytes is currently available, then the
operating 
system needs to make sure that the remaining 50000 bytes is buffered,
and 
is available for the next time the device is read.

So, you might want to look at your input device driver, to make sure
that 
it's buffering data properly.  One thing you try is modify 
"ByteStreamFileSource" to save - into a separate file - a copy of each 
block of data that it reads from the input device.  Then, you could try 
playing this file.  If the file has data loss, then this means that your

input device driver is not buffering data properly.  If, however, the
file 
*does not* have data loss, then this means that your device driver is 
probably OK, and that the data loss is occurring elsewhere.

         Ross.

ps. None of this has anything to do with "BasicTaskScheduler".  That's a

'red herring'.




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Subject: SV: [Live-devel] Streaming MPEG2 directly from video device
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I modified ByteStreamFileSource as you suggested - so that it writes the
data read from the device to a file. 
Streaming from this file works fine .... No dataloss reading from the
device.

- Jan

-----Oprindelig meddelelse-----
Fra: Ross Finlayson [mailto:finlayson(at)live.com] 
Sendt: 28. oktober 2003 19:05
Til: live-devel(at)lists.sourceforge.net
Emne: Re: [Live-devel] Streaming MPEG2 directly from video device



>I have tried looking at your code in BasicTaskScheduler.cpp, trying to 
>figure out if the problem with our video-device could somehow be 
>explained by a combination of the way your code read data from the 
>device, and the way the PVR-350 device driver makes data available, 
>could the problem be that data becomes available in rather large 
>chunks, leaving your code "idle" in some periods and overloaded in 
>other ??

I don't think so.  "testMPEG1or2VideoStreamer" - like all applications
that 
use a "RTPSink" object - will read, at a time, only as much data as is 
needed to pack into a single outgoing RTP packet.  Also, after sending
each 
RTP packet, it will delay an appropriate amount of time before going
back 
and reading more data.  (Note my earlier answers to Andrei's "frame rate

control?" question.)  I.e., "testMPEG1or2VideoStreamer" will send its
RTP 
packets - and read from its input "MPEG1or2VideoStreamFramer" object -
at a 
steady pace.  (However, the "MPEG1or2VideoStreamFramer" will, in turn,
read 
from the input file (a "ByteStreamFileSource") in a more bursty manner.
It 
has large buffers ('banks') that it tries to keep full.  See 
"StreamParser::ensureValidBytes1()" to understand this.)

Nonetheless, this means that your PVR-350 input device 'file' - as 
implemented within the operating system kernel - needs to buffer data 
properly, because the "ByteStreamFileSource" object that reads from this

device does not always read, at once, all of the data that's 
available.  E.g., if the "ByteStreamFileSource" reads 150000 bytes from
the 
device, and (e.g.) 200000 bytes is currently available, then the
operating 
system needs to make sure that the remaining 50000 bytes is buffered,
and 
is available for the next time the device is read.

So, you might want to look at your input device driver, to make sure
that 
it's buffering data properly.  One thing you try is modify 
"ByteStreamFileSource" to save - into a separate file - a copy of each 
block of data that it reads from the input device.  Then, you could try 
playing this file.  If the file has data loss, then this means that your

input device driver is not buffering data properly.  If, however, the
file 
*does not* have data loss, then this means that your device driver is 
probably OK, and that the data loss is occurring elsewhere.

         Ross.

ps. None of this has anything to do with "BasicTaskScheduler".  That's a

'red herring'.




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X-Original-Date: Wed, 29 Oct 2003 16:39:44 +0100
Date: Wed, 29 Oct 2003 16:39:44 +0100

Hi,

I encounter a somehow strange behavior when streaming live MPEG4 video.
When trying to stream live mpeg4, if the image size is larger than CIF 
(352x288)
(I tried 640x480 and 704x576) then the image on the client side seems to 
have losses.
I guess this is because the encoded mpeg4 frame exceeds a predefined 
value (?)
With CIF it works just fine.

Architecture
I have implemented a FramedSource that generates the MPEG4-encoded 
frames, and
I'm using a MPEG4VideoStreamFramer on top of it + a MPEG4ESVideoRTPSink
to send the images.
On the client side I'm using MPlayer compiled with the same (latest) 
live.com libraries.

A separate question :

When implementing a  FramedSource to send out h263 frames,  in 
doGetNextFrame
I always get a fMaxSize of 28988. This limits the size of the images 
that can
be sent. Is this a h263 limitation ?


I also have a question about the development plan (feel free to igore it 
if you wish) :
Maybe this is also a question for the vic developers, but I'll just put 
it here :
    do you plan to have a live streamer that is compatible with vic 
clients ?
    My current tests, with mpeg1 and h263 have failed. Also 
testMPEG1or2VideoStreamer
    outgoing stream cannot be viewed with vic. Do you know about
    this incopatibility and have any idea what's causing it ?


Thanks,
Andrei



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X-Original-Date: Wed, 29 Oct 2003 13:15:48 -0800
Date: Wed, 29 Oct 2003 13:15:48 -0800

At 04:27 AM 10/29/03, you wrote:
>I modified ByteStreamFileSource as you suggested - so that it writes the
>data read from the device to a file.
>Streaming from this file works fine .... No dataloss reading from the
>device.

OK, the next thing to try is to do the same thing (writing the received 
data into a file, for testing) further along the data chain: When the 
"RTPSink" object has received data from its input source (a 
"MPEG1or2VideoStreamFramer").

Add code to the start of "MultiFramedRTPSink::afterGettingFrame1()" that 
copies the received frame data:
         - pointer: fOutBuf->curPtr()
         - length: frameSize
into a file.  Then, check whether that file has any data loss.  (If so, 
then the problem is occurring in the "MPEG1or2VideoStreamFramer" somehow.)

Are you sure you're not seeing any error messages printed to stderr?


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Wed, 29 Oct 2003 13:32:44 -0800
Date: Wed, 29 Oct 2003 13:32:44 -0800


>I encounter a somehow strange behavior when streaming live MPEG4 video.
>When trying to stream live mpeg4, if the image size is larger than CIF 
>(352x288)
>(I tried 640x480 and 704x576) then the image on the client side seems to 
>have losses.
>I guess this is because the encoded mpeg4 frame exceeds a predefined value (?)
>With CIF it works just fine.

Are you seeing any error messages printed to 'stderr'?

>When implementing a  FramedSource to send out h263 frames,  in doGetNextFrame
>I always get a fMaxSize of 28988. This limits the size of the images that can
>be sent. Is this a h263 limitation ?

I think you may be misunderstanding how the "fMaxSize" field is used.  It 
is not a 'returned' value.  Instead, this field is *set* by the "maxSize" 
parameter to the "FramedSource::getNextFrame()" function.  It is then 
*used* by each source's "doGetNextFrame()" virtual function to tell it the 
maximum amount of data that its sink is requesting.

>I also have a question about the development plan (feel free to igore it 
>if you wish) :
>Maybe this is also a question for the vic developers, but I'll just put it 
>here :
>    do you plan to have a live streamer that is compatible with vic clients ?
>    My current tests, with mpeg1 and h263 have failed. Also 
> testMPEG1or2VideoStreamer
>    outgoing stream cannot be viewed with vic. Do you know about
>    this incopatibility and have any idea what's causing it ?

As far as I know, "vic" does not support MPEG video at all.  It *does* 
support H.263, but I think it does so using the older RFC 2190 payload 
format; not the newer RFC 2429 payload format that the LIVE.COM library 
supports.  In short, "vic" is very old code that hasn't been updated in 
years, so I don't have high hopes that anyone will bring it up-to-date.

Instead of "vic", I suggest using QuickTime Player (on Windows), or 
"MPlayer" (on Linux).


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Thu, 30 Oct 2003 11:03:19 +0100
Date: Thu, 30 Oct 2003 11:03:19 +0100



>> I encounter a somehow strange behavior when streaming live MPEG4 video.
>> When trying to stream live mpeg4, if the image size is larger than 
>> CIF (352x288)
>> (I tried 640x480 and 704x576) then the image on the client side seems 
>> to have losses.
>> I guess this is because the encoded mpeg4 frame exceeds a predefined 
>> value (?)
>> With CIF it works just fine.
>
>
> Are you seeing any error messages printed to 'stderr'?


A lot of error messages printed out by mplayer :

Error at MB: 377
concealing errors
ac-tex damaged at 17 8  0.0% 0 0 0%
Error at MB: 377
concealing errors
ac-tex damaged at 16 8  0.0% 0 0 0%
Error at MB: 376
concealing errors
ac-tex damaged at 13 8  0.0% 0 0 0%
Error at MB: 373
concealing errors
ac-tex damaged at 13 8  0.0% 0 0 0%
Error at MB: 373
concealing errors
ac-tex damaged at 17 8  0.0% 0 0 0%
Error at MB: 377
concealing errors
illegal dc vlc  0%  0%  0.0% 0 0 0%
Error at MB: 376
concealing errors
....
ac-tex damaged at 30 9  0.0% 0 0 0%
Error at MB: 435
concealing errors
illegal dc vlc  0%  0%  0.0% 0 0 0%
Error at MB: 435
concealing errors
2. marker bit missing in 3. esc0 0%
Error at MB: 393
concealing errors

I'm using the MPEG4 stream framer,
and in doGetNextFrame I'm returning just the part of the frame
that "fits" in the current request, the rest being sent in subsequent 
doGetNextFrame calls.
My guess is that the stream framer ignores the last bytes of a too large 
frame, but I cannot
say for sure.

>
>> When implementing a  FramedSource to send out h263 frames,  in 
>> doGetNextFrame
>> I always get a fMaxSize of 28988. This limits the size of the images 
>> that can
>> be sent. Is this a h263 limitation ?
>
>
> I think you may be misunderstanding how the "fMaxSize" field is used.  
> It is not a 'returned' value.  Instead, this field is *set* by the 
> "maxSize" parameter to the "FramedSource::getNextFrame()" function.  
> It is then *used* by each source's "doGetNextFrame()" virtual function 
> to tell it the maximum amount of data that its sink is requesting.

I understood it exactly, my problem is that when using the h263 sink, 
this value is always 28988. Is there a way to modify this ?
Or is this a h263 limitation ?


>
> As far as I know, "vic" does not support MPEG video at all.  It *does* 
> support H.263, but I think it does so using the older RFC 2190 payload 
> format; not the newer RFC 2429 payload format that the LIVE.COM 
> library supports.  In short, "vic" is very old code that hasn't been 
> updated in years, so I don't have high hopes that anyone will bring it 
> up-to-date.
>
> Instead of "vic", I suggest using QuickTime Player (on Windows), or 
> "MPlayer" (on Linux).

OK. Thanks.

Andrei



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Subject: Re: [Live-devel] large image streaming
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X-Original-Date: Thu, 30 Oct 2003 13:31:52 -0800
Date: Thu, 30 Oct 2003 13:31:52 -0800


>>>When implementing a  FramedSource to send out h263 frames,  in 
>>>doGetNextFrame
>>>I always get a fMaxSize of 28988. This limits the size of the images 
>>>that can
>>>be sent. Is this a h263 limitation ?
>>
>>
>>I think you may be misunderstanding how the "fMaxSize" field is used.
>>It is not a 'returned' value.  Instead, this field is *set* by the 
>>"maxSize" parameter to the "FramedSource::getNextFrame()" function.
>>It is then *used* by each source's "doGetNextFrame()" virtual function to 
>>tell it the maximum amount of data that its sink is requesting.
>
>I understood it exactly, my problem is that when using the h263 sink, this 
>value is always 28988. Is there a way to modify this ?

OK, now I understand what's going on.  The maximum buffer size used by your 
"RTPSink" object needs to be increased, because its incoming frames are 
exceeding this value.

You can do this - within your main program - by changing the value of the 
"OutPacketBuffer::numPacketsLimit" variable.  (Note that its default value 
is 20, which corresponds to a max buffer size of 20*1450 = 29000 = 28988 + 
12.  (12 bytes is the size of the RTP header.))

So, for example, if your input frames are a maximum of 50000 bytes in size, 
then - in your main program - you should set:
         OutPacketBuffer::numPacketsLimit = 35;
(because 35 = ceiling(50000/1450).

I have now added a warning message to the most recent "LIVE.COM Streaming 
Media" code release - 2003.10.30 - that will tell you when this limit is 
being exceeded.

NOTE TO JAN BORRIS: I suspect that this is also the cause of the problem 
that you're seeing when you're streaming MPEG-2 video.  Please upgrade to 
the latest version of the LIVE.COM code, rebuild, and try running 
"testMPEG1or2VideoStreamer" again.  If you see warning messages about 
needing to increase "OutPacketBuffer::numPacketsLimit", then please do so 
(by adding such a statement to "testMPEG1or2VideoStreamer.cpp").


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Fri, 31 Oct 2003 15:33:39 +0100
Date: Fri, 31 Oct 2003 15:33:39 +0100


Thanks, that solved the problems, both for MPEG4 and h263.
Changing that variable helps.
Maybe you can have it as a parameter to the createNew method of the sink 
or something similar ?

Andrei



>
> OK, now I understand what's going on.  The maximum buffer size used by 
> your "RTPSink" object needs to be increased, because its incoming 
> frames are exceeding this value.
>
> You can do this - within your main program - by changing the value of 
> the "OutPacketBuffer::numPacketsLimit" variable.  (Note that its 
> default value is 20, which corresponds to a max buffer size of 20*1450 
> = 29000 = 28988 + 12.  (12 bytes is the size of the RTP header.))
>
> So, for example, if your input frames are a maximum of 50000 bytes in 
> size, then - in your main program - you should set:
>         OutPacketBuffer::numPacketsLimit = 35;
> (because 35 = ceiling(50000/1450).
>
> I have now added a warning message to the most recent "LIVE.COM 
> Streaming Media" code release - 2003.10.30 - that will tell you when 
> this limit is being exceeded.
>
> NOTE TO JAN BORRIS: I suspect that this is also the cause of the 
> problem that you're seeing when you're streaming MPEG-2 video.  Please 
> upgrade to the latest version of the LIVE.COM code, rebuild, and try 
> running "testMPEG1or2VideoStreamer" again.  If you see warning 
> messages about needing to increase "OutPacketBuffer::numPacketsLimit", 
> then please do so (by adding such a statement to 
> "testMPEG1or2VideoStreamer.cpp").
>
>
>     Ross Finlayson
>     LIVE.COM
>     <http://www.live.com/>
>
>
>
> -------------------------------------------------------
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From <EMAIL-ADDRESS-HIDDEN> Wed Nov 5 20:09:36 2003
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	0.9 FROM_ENDS_IN_NUMS      From: ends in numbers
Subject: [Live-devel] testOnDemand and openrtsp play problem
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X-Original-Date: Wed, 5 Nov 2003 19:56:04 -0800 (PST)
Date: Wed, 5 Nov 2003 19:56:04 -0800 (PST)

Hi Ross,

I think I have same problem with others.
I run testOnDemand server and open it with openRTSP to
play mp3 stream with -V -n as you suggested. the
result is not as i expected. Sometimes i receive the
data after PLAY command, sometimes i receive nothing
even for 1st session.
I also try with MORGAN rtp client, the problem still
exist.

Any idea how to solve this?

Thanks,
Reza

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	0.9 FROM_ENDS_IN_NUMS      From: ends in numbers
Subject: [Live-devel] RTSPServer and app callback
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X-Original-Date: Wed, 5 Nov 2003 20:33:49 -0800 (PST)
Date: Wed, 5 Nov 2003 20:33:49 -0800 (PST)

Hi Ross,

Are there any method to inform application if
RTSPServer handle_cmd method is invoked?

For example if RTSPServer handleCmd_PLAY method
invoked, i want my app aware of this.

any way to do this?

Thanks,
Reza

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Subject: Re: [Live-devel] testOnDemand and openrtsp play problem
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X-Original-Date: Thu, 06 Nov 2003 01:23:06 -0800
Date: Thu, 06 Nov 2003 01:23:06 -0800

At 07:56 PM 11/5/03, you wrote:
>Hi Ross,
>
>I think I have same problem with others.
>I run testOnDemand server and open it with openRTSP to
>play mp3 stream with -V -n as you suggested. the
>result is not as i expected. Sometimes i receive the
>data after PLAY command, sometimes i receive nothing
>even for 1st session.
>I also try with MORGAN rtp client, the problem still
>exist.
>
>Any idea how to solve this?

Not really - at least not without more information.  How does the output 
from "openRTSP" (using the "-V" option) differ in each case?


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Thu, 06 Nov 2003 01:25:26 -0800
Date: Thu, 06 Nov 2003 01:25:26 -0800

At 08:33 PM 11/5/03, you wrote:
>Hi Ross,
>
>Are there any method to inform application if
>RTSPServer handle_cmd method is invoked?
>
>For example if RTSPServer handleCmd_PLAY method
>invoked, i want my app aware of this.

You could just add code to "RTSPServer.cpp" to set some global flag, and 
have the rest of your application check for this flag being set.

(Of course, if you just want to print debugging information, then you can 
do this directly from within the "RTSPServer.cpp" code - using 
"fprintf(stderr, ...)".)


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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--=====================_856843==_.ALT
Content-Type: text/plain; charset="us-ascii"; format=flowed

Hi -
I'm trying to build Live using Green Hills compiler for ARM7. I have built 
all of the source except for MPEGVideoStreamParser.cpp where I get the 
following compiler error:

error: a pointer to a bound function may only be used to call the 
function
                 usingSource->continueReadProcessing, 
usingSource),
                               ^
Here is the constructor from which the error refers to:

MPEGVideoStreamParser
::MPEGVideoStreamParser(MPEGVideoStreamFramer* usingSource,
                         FramedSource* inputSource)
   : StreamParser(inputSource, FramedSource::handleClosure, usingSource,
                 usingSource->continueReadProcessing, usingSource),
   fUsingSource(usingSource) {
}

I don't understand this problem. Perhaps a compiler switch could solve this 
problem.
Thanks.
--=====================_856843==_.ALT
Content-Type: text/html; charset="us-ascii"

<html>
Hi -<br>
I'm trying to build Live using Green Hills compiler for ARM7. I have
built all of the source except for MPEGVideoStreamParser.cpp where I get
the following compiler error:<br><br>
<b>error: a pointer to a bound function may only be used to call the
function&nbsp;&nbsp; <x-tab>&nbsp;&nbsp;&nbsp;</x-tab><br>
&nbsp;
<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab><x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>usingSource-&gt;continueReadProcessing,
usingSource),&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
<br>
</b><x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab><x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab><x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
<b>^<br>
</b>Here is the constructor from which the error refers to:<br><br>
<b>MPEGVideoStreamParser<br>
::MPEGVideoStreamParser(MPEGVideoStreamFramer* usingSource,<br>
<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab><x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab><x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>FramedSource*
inputSource)<br>
&nbsp; : StreamParser(inputSource, FramedSource::handleClosure,
usingSource,<br>
<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab><x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>usingSource-&gt;continueReadProcessing,
usingSource),<br>
&nbsp; fUsingSource(usingSource) {<br>
}<br><br>
</b>I don't understand this problem. Perhaps a compiler switch could
solve this problem. <br>
Thanks.</html>

--=====================_856843==_.ALT--




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X-Original-Date: Thu, 6 Nov 2003 15:16:41 +0100
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Hi Ross,

To sum up:
We have installed a Hauppauge PVR-350 card on a linux pc; when streaming directly from the device we experience reduced quality in the resulting RTP stream. Strangely enough if we do a "cat /dev/video0 > /var/tmp/devdump.mpg", and stream from the generated file, the stream looks fine.

As you suggested I have made two experiments:
1) dumping data to a file from within ByteStreamFileSource::doGetNextFrame()
Data in this file seem to be OK
2) dumping data to a file from within MultiFramedRTPSink::afterGettingFrame1(), according to your directions.
Data in this file is *NOT* OK, when trying to decode from this file we get a lower quality stream, sometimes with pixellations all over the image, other times just in the bottom. Trying to decode this file using mpeg2dec we get some error messages from mpeg2dec - but I don't know what they mean, they are not very informative; something like

<someaddr> INVALID

Hope this can get us closer to an explanation

- Jan

-----Oprindelig meddelelse-----
Fra: Ross Finlayson [mailto:finlayson(at)live.com] 
Sendt: 29. oktober 2003 22:16
Til: live-devel(at)lists.sourceforge.net
Emne: Re: [Live-devel] Streaming MPEG2 directly from video device


At 04:27 AM 10/29/03, you wrote:
>I modified ByteStreamFileSource as you suggested - so that it writes 
>the data read from the device to a file. Streaming from this file works 
>fine .... No dataloss reading from the device.

OK, the next thing to try is to do the same thing (writing the received 
data into a file, for testing) further along the data chain: When the 
"RTPSink" object has received data from its input source (a 
"MPEG1or2VideoStreamFramer").

Add code to the start of "MultiFramedRTPSink::afterGettingFrame1()" that 
copies the received frame data:
         - pointer: fOutBuf->curPtr()
         - length: frameSize
into a file.  Then, check whether that file has any data loss.  (If so, 
then the problem is occurring in the "MPEG1or2VideoStreamFramer" somehow.)

Are you sure you're not seeing any error messages printed to stderr?


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Thu, 06 Nov 2003 08:17:19 -0800
Date: Thu, 06 Nov 2003 08:17:19 -0800


>As you suggested I have made two experiments:
>1) dumping data to a file from within ByteStreamFileSource::doGetNextFrame()
>Data in this file seem to be OK
>2) dumping data to a file from within 
>MultiFramedRTPSink::afterGettingFrame1(), according to your directions.
>Data in this file is *NOT* OK

Jan,

I suspect that the maximum buffer size used by your "RTPSink" object needs 
to be increased, because its incoming frames are exceeding this size.

To test this, please upgrade to the latest version of the LIVE.COM 
Streaming Media code, and rebuild "testMPEG1or2VideoStreamer".

If you see warning messages about needing to increase 
"OutPacketBuffer::numPacketsLimit", then you can do so - within your main 
program - by changing the value of the "OutPacketBuffer::numPacketsLimit" 
variable.  (Note that its default value is 20, which corresponds to a max 
buffer size of 20*1450 = 29000 = 28988 + 12.  (12 bytes is the size of the 
RTP header.))

So, for example, if your input frames are a maximum of 50000 bytes in size, 
then - in your main program - you should set:
         OutPacketBuffer::numPacketsLimit = 35;
(because 35 = ceiling(50000/1450).


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Thu, 06 Nov 2003 08:05:11 -0800
Date: Thu, 06 Nov 2003 08:05:11 -0800

At 05:52 AM 11/6/03, you wrote:
>Hi -
>I'm trying to build Live using Green Hills compiler for ARM7. I have built 
>all of the source except for MPEGVideoStreamParser.cpp where I get the 
>following compiler error:
>
>error: a pointer to a bound function may only be used to call the 
>function
>                 usingSource->continueReadProcessing, 
> usingSource),
>                               ^
>Here is the constructor from which the error refers to:
>
>MPEGVideoStreamParser
>::MPEGVideoStreamParser(MPEGVideoStreamFramer* usingSource,
>                         FramedSource* inputSource)
>   : StreamParser(inputSource, FramedSource::handleClosure, usingSource,
>                 usingSource->continueReadProcessing, usingSource),
>   fUsingSource(usingSource) {
>}

Bill,

Try replacing
         usingSource->continueReadProcessing
(in the "MPEGVideoStreamParser" constructor) with
         &MPEGVideoStreamFramer::continueReadProcessing

I think that may make your compiler happier.

BTW, when you get all of the source compiled (with the Green Hills compiler 
for ARM), could you please send us the "config.*" file that you used (if 
it's different from one of the original ones), so we can include it in the 
next software distribution.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>



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X-Original-Date: Thu, 06 Nov 2003 15:04:46 -0500
Date: Thu, 06 Nov 2003 15:04:46 -0500

Ross -
see my responses below


>Bill,
>
>Try replacing
>         usingSource->continueReadProcessing
>(in the "MPEGVideoStreamParser" constructor) with
>         &MPEGVideoStreamFramer::continueReadProcessing
>
>I think that may make your compiler happier.
Thanks - it worked.


>BTW, when you get all of the source compiled (with the Green Hills 
>compiler for ARM), could you please send us the "config.*" file that you 
>used (if it's different from one of the original ones), so we can include 
>it in the next software distribution.

Actually, I have compiled the code using the Multi2000 IDE for Windows and 
did not use makefiles. I simply dragged all files into the project 
workspace and built as a library. I added a new define called NET_OS_MULTI 
for a small number of changes that had to be made. I may also build for 
NetOS for GNU.

FYI - the platform I'm using is a custom board based on Netsilicon's 
NetARM-50 (ARM7 core). The NetARM is running NetOS6, for which I built the 
Live library into an existing project. A camera/DSP provide the video input 
to the NetARM. Currently, the system works with JPEG , but I am adding 
support for MPEG4.

Bill




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	0.9 FROM_ENDS_IN_NUMS      From: ends in numbers
Subject: [Live-devel] Re: testOnDemand and openrtsp play problem
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X-Original-Date: Thu, 6 Nov 2003 22:53:59 -0800 (PST)
Date: Thu, 6 Nov 2003 22:53:59 -0800 (PST)

> Not really - at least not without more information. 
>How does the output 
> from "openRTSP" (using the "-V" option) differ in
>each case?
 
Hi Ross,
You can see below the result. The client received
nothing, i checked the dump file and
audio-MPA-ROBUST-1 file
both size were 0 byte. Is it because i was using UDP
(as default) to transfer the data?

Thanks
Reza

c:\>testOpenRtsp.exe -a -V -n
rtsp://192.168.2.224:7070/ >> dump.txt
---- Start
Sending request: DESCRIBE rtsp://192.168.2.224:7070/
RTSP/1.0
CSeq: 1
Accept: application/sdp
User-Agent: testOpenRtsp.exe (LIVE.COM Streaming Media
v2003.10.30)


Received DESCRIBE response: RTSP/1.0 200 OK
CSeq: 1
Content-Type: application/sdp
Content-Length: 268


Need to read 268 extra bytes
Read 268 extra bytes: v=0
o=- 1068167604453000 1 IN IP4 192.168.2.224
s=LIVE.COM Streaming Media v
i=LIVE.COM Streaming Media v
t=0 0
a=tool:LIVE.COM Streaming Media v2003.10.30
a=type:broadcast
m=audio 0 RTP/AVP 96
c=IN IP4 0.0.0.0
a=rtpmap:96 MPA-ROBUST/90000
a=control:track1

Opened URL "rtsp://192.168.2.224:7070/", returning a
SDP description:
v=0
o=- 1068167604453000 1 IN IP4 192.168.2.224
s=LIVE.COM Streaming Media v
i=LIVE.COM Streaming Media v
t=0 0
a=tool:LIVE.COM Streaming Media v2003.10.30
a=type:broadcast
m=audio 0 RTP/AVP 96
c=IN IP4 0.0.0.0
a=rtpmap:96 MPA-ROBUST/90000
a=control:track1

Created receiver for "audio/MPA-ROBUST" subsession
(client ports 4028-4029)
Sending request: SETUP
rtsp://192.168.2.224:7070/track1 RTSP/1.0
CSeq: 2
Transport: RTP/AVP;unicast;client_port=4028-4029
User-Agent: testOpenRtsp.exe (LIVE.COM Streaming Media
v2003.10.30)


Received SETUP response: RTSP/1.0 200 OK
CSeq: 2
Transport: RTP/AVP;unicast
Session: 1


Setup "audio/MPA-ROBUST" subsession (client ports
4028-4029)
Outputting data from the "audio/MPA-ROBUST" subsession
to 'stdout'
Sending request: PLAY rtsp://192.168.2.224:7070/
RTSP/1.0
CSeq: 3
Session: 1
Range: npt=0-
User-Agent: testOpenRtsp.exe (LIVE.COM Streaming Media
v2003.10.30)


Received PLAY response: RTSP/1.0 200 OK
CSeq: 3
Session: 1


Started playing session
Receiving streamed data...

--- End

 

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X-Original-Date: Fri, 07 Nov 2003 11:25:49 -0500
Date: Fri, 07 Nov 2003 11:25:49 -0500

Hi -
This question comes from someone who knows very little about media 
streaming, MPEG, RTP, RTSP, etc.

I have MPEG4 video-only data comming live from a camera/encoder on an 
embedded system. I have an application running based on 
testMPEG4VideoStreamer so that I can multicast the live video. (I haven't 
actually yet connected the video data to Live's stdin but I see there are 
instructions for that in the FAQ.) I have also built this to run on Windows 
for experimenting purposes.

My question is:

Will I be able to stream this live video to Quicktime? I need to embed some 
player in a web page, specifically on Windows running IE.

If so, I understand that you need an .sdp file. How can I create this for 
my live stream?

If not, what are other options for playing this using an embedded player?

Thanks
Bill




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From: David Jones <EMAIL-ADDRESS-HIDDEN>
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Subject: [Live-devel] file input/jpeg streaming
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X-Original-Date: Fri, 7 Nov 2003 17:06:54 -0600 (CST)
Date: Fri, 7 Nov 2003 17:06:54 -0600 (CST)


I'd like to create a program similar to the testMPEG1or2VideoStreamer app 
that reads in .ppm files and then streams as JPEG.  I'm guessing that I'll 
have to write a subclass of JPEGVideoSource which will read in the file, 
store it in YUV format, and ensure that type(), qFactor(), width(), 
height() return the appropriate values.

However, I'm not sure how to implement the doGetNextFrame() or 
getPlayTime() functions (I still haven't wrapped my brain around the entire library 
either).  I'd think there'd have to be some pointer to the YUV data which 
gets packaged by the JPEGVideoRTPSink and sent off to some multicast 
address.  Can anyone fill me in (or tell me if I'm approaching this all 
wrong)?

Thanks much,
Dave J.



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