Thanks Ross. By the way, I assume the 'numChannels' parameter is unrelated too ? I mean, a stereo mp3 file wouldn't have 'numChannels=2' set.
I'll dive further into libavformat to probe those values (if you have an idea...you're welcome ;-) ) Thanks again, Best regards, Guillaume. 2009/12/1 Ross Finlayson <finlay...@live555.com> > I'm currently using mediaServer to stream various media on a local network. >> Next, I implemented a small RTSP client to receive & play that stream. >> >> It works really great with WAV/PCM files, but I experience some issues >> with MP3 files. >> After some investigation, I found out what is going on : I use >> systematically provided samplerate (rtpTimeStampFrequency), which is set to >> 90khz for mP3 files (RTP payload :14)... >> Incorrect value of course, and it messes up all my playback routines >> afterwards... >> >> Is it possible to obtain an accurate sample rate value through liveMedia >> classes, or do I need to do some computing ? >> > > The latter. The sampling frequency for MPEG-1 or 2 audio (which includes > 'MP3') is encoded in the data's MPEG audio header. It has nothing to do > with the RTP timestamp frequency. > -- > > Ross Finlayson > Live Networks, Inc. > http://www.live555.com/ > _______________________________________________ > live-devel mailing list > live-devel@lists.live555.com > http://lists.live555.com/mailman/listinfo/live-devel >
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