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ffmpeg / libavformat / rtp.h @ ed90d370

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# Date Author Comment
5509bffa 01/12/2006 10:43 PM Diego Biurrun

Update licensing information: The FSF changed postal address.

Originally committed as revision 4842 to svn://svn.ffmpeg.org/ffmpeg/trunk

115329f1 12/17/2005 06:14 PM Diego Biurrun

COSMETICS: Remove all trailing whitespace.

Originally committed as revision 4749 to svn://svn.ffmpeg.org/ffmpeg/trunk

33785a3a 06/28/2005 02:51 PM Michael Niedermayer

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

d1ccf0e0 05/26/2005 07:47 AM Romain Degez

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...

8b1ab7bf 10/29/2003 02:25 PM Fabrice Bellard

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

abac6175 11/25/2002 07:07 PM Fabrice Bellard

renamed libav to libavformat

Originally committed as revision 1276 to svn://svn.ffmpeg.org/ffmpeg/trunk