<div>Hi Ross,<br></div><div><br></div><div>I'm still stuck into this mp3 streaming case : I must be missing something, but here are some details.</div><div><br></div><div>1) A standard mp3 is being streamed by liveMedia server, works just fine (checked with VLC)</div>
<div>2) I have a sink client (a really simple MediaSink subclass), in which I grab & store frames, through callback 'afterGettingFrame', with a specific framesize.</div><div>3) I'm using FFmpeg API (libavformat / libavcodec) to decode this frame, using provided framesize, and I inject it into a sample player, for instance PortAudio.</div>
<div><br></div><div>FFmpeg decoding seems ok (no errors), but when I feed PortAudio with those samples, I can hear nothing but noise, and after a few seconds, an app crash.</div><div><br></div><div>This client / audio player works really fine with remote PCM files, but MP3 files won't play.</div>
<div><br></div><div>Could you give me some insights about received RTP frames ? Do I need some specific processing before decoding ?</div><div><br></div><div>Thanks in advance,</div><div>Guillaume.</div><div><br></div><br>
<div class="gmail_quote">2009/12/1 Guillaume Ferry <span dir="ltr"><<a href="mailto:guiluge@gmail.com">guiluge@gmail.com</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<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" target="_blank">finlayson@live555.com</a>></span><div>
<div class="h5"><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div><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>
_______________________________________________<br>
live-devel mailing list<br>
<a href="mailto:live-devel@lists.live555.com" target="_blank">live-devel@lists.live555.com</a><br>
<a href="http://lists.live555.com/mailman/listinfo/live-devel" target="_blank">http://lists.live555.com/mailman/listinfo/live-devel</a><br>
</font></blockquote></div></div></div><br>
</blockquote></div><br>