Hi Ross, I'm still stuck into this mp3 streaming case : I must be missing something, but here are some details.
1) A standard mp3 is being streamed by liveMedia server, works just fine (checked with VLC) 2) I have a sink client (a really simple MediaSink subclass), in which I grab & store frames, through callback 'afterGettingFrame', with a specific framesize. 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. 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. This client / audio player works really fine with remote PCM files, but MP3 files won't play. Could you give me some insights about received RTP frames ? Do I need some specific processing before decoding ? Thanks in advance, Guillaume. 2009/12/1 Guillaume Ferry <guil...@gmail.com> > Thanks Ross. > By the way, I assume the 'numChannels' parameter is unrelated too ? > I mean, a stereo mp3 file wouldn't have 'numChannels=2' set. > > I'll dive further into libavformat to probe those values (if you have an > idea...you're welcome ;-) ) > > Thanks again, > > Best regards, > Guillaume. > > 2009/12/1 Ross Finlayson <finlay...@live555.com> > > 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/ >> _______________________________________________ >> live-devel mailing list >> live-devel@lists.live555.com >> http://lists.live555.com/mailman/listinfo/live-devel >> > >
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