Hi I’m new here, as I’m trying to learn some of the intricacies around media streaming.
I’ve been building a little client using libav to experiment as I’m learning, and I’ve hit a wall that I’m hoping someone can help me resolve. Basically, I’m trying to build a specialized client that displays h.264 and MPEG4 streams via rtsp with audio and video muxed over a TCP/IP channel. I’ve got the video coming through and displaying ok (well, it’s a bit choppy my client, but it looks the same in VLC). The problem is, I’m having a dickens of a time figuring out how to get the audio codecs configured and able to play the audio channel. Specifically, I’m trying to figure out how to deal with the various parameters on the a=fmtp line of the SDP message that’s delivered from the DESCRIBE command via rtsp. The RFCs all say this is codec-specific data and is “out-of-band” as far as the specs go, so they don’t explain any of it. Obviously, this code library is able to get audio and video demuxed properly based on the data in the SDP message, since it has no problem playing h.264 and MPEG4 files. Is there anything that explains this stuff anywhere? Also, can anybody point me to the specific libav code units that deal with parsing the a=fmtp lines and initializing the audio codecs? Thanks heaps! -David _______________________________________________ libav-tools mailing list [email protected] https://lists.libav.org/mailman/listinfo/libav-tools
