Hi there, 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 ? Thanks in advance, Kind regards, Guillaume.
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