<div>Hi there,<br></div><div><br></div><div>I'm currently using mediaServer to stream various media on a local network.</div><div>Next, I implemented a small RTSP client to receive & play that stream.</div><div><br>
</div><div>It works really great with WAV/PCM files, but I experience some issues with MP3 files.</div><div>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)... </div>
<div>Incorrect value of course, and it messes up all my playback routines afterwards...</div><div><br></div><div>Is it possible to obtain an accurate sample rate value through liveMedia classes, or do I need to do some computing ?</div>
<div><br></div><div>Thanks in advance,</div><div><br></div><div>Kind regards,</div><div>Guillaume. </div>