RTCP_* conflict with live.com and they seem not to be used anywhere so commenting them out
Originally committed as revision 4406 to svn://svn.ffmpeg.org/ffmpeg/trunk
RTP/RTSP and MPEG4-AAC audio - preliminary support for mpeg4-aac rtp payload (no interleaving support) - use udp transport as default (makes more sense with rtp, doesn't it ?) - some code factorization, so adding support for new rtp payload will be easier...
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
Originally committed as revision 2448 to svn://svn.ffmpeg.org/ffmpeg/trunk
renamed libav to libavformat
Originally committed as revision 1276 to svn://svn.ffmpeg.org/ffmpeg/trunk