<div>Thanks Ross.<br></div><div>By the way, I assume the 'numChannels' parameter is unrelated too ?</div><div>I mean, a stereo mp3 file wouldn't have 'numChannels=2' set.</div><div><br></div><div>I'll dive further into libavformat to probe those values (if you have an idea...you're welcome ;-) )</div>
<div><br></div><div>Thanks again,</div><div><br></div><div>Best regards,</div><div>Guillaume.</div><div><br></div><div class="gmail_quote">2009/12/1 Ross Finlayson <span dir="ltr"><<a href="mailto:finlayson@live555.com">finlayson@live555.com</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><div><div class="h5"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
I'm currently using mediaServer to stream various media on a local network.<br>
Next, I implemented a small RTSP client to receive & play that stream.<br>
<br>
It works really great with WAV/PCM files, but I experience some issues with MP3 files.<br>
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)...<br>
Incorrect value of course, and it messes up all my playback routines afterwards...<br>
<br>
Is it possible to obtain an accurate sample rate value through liveMedia classes, or do I need to do some computing ?<br>
</blockquote>
<br></div></div>
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.<br>
-- <br><font color="#888888">
<br>
Ross Finlayson<br>
Live Networks, Inc.<br>
<a href="http://www.live555.com/" target="_blank">http://www.live555.com/</a><br>
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