> I am currently using a file, but eventually it will be live source. > I got the audio streaming, which is good, but it stutters. I looked at the > packet format with wireshark, but found no issues. > However I found out that apparently RTP packets are sent too quickly with my > class. But if you use MP3AudioFileServerMediaSubsession, the audio is fine > and RTP packets are sent at a normal rate.
OK, so you have code that works: "MP3AudioFileServerMediaSubsession", and code that does not work: Your subclass of "OnDemandServerMediaSubsession". By looking at the differences between them, it should be relatively easy, then, for you to figure out what's wrong with your code. > If I use my class with openRTSP for a few seconds, the audio file is 3x > larger that the video file (with your class it's 10x smaller). > Do you have any idea what is happening here? What happens when you try playing this "3x larger" audio file (i.e., after renaming it to have a ".mp3" filename suffix)? I suspect that you are delivering the same MP3 frame into the downstream "MPEG1or2AudioRTPSink" more than once (so you end up with duplicate MP3 frames being sent). Ross Finlayson Live Networks, Inc. http://www.live555.com/
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