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/
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