Improve RTSP/RTP layer: Difference between revisions
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** untangle the AAC and mpeg4 format specific code from rtsp.c, make them proper dynamic payload handlers | ** untangle the AAC and mpeg4 format specific code from rtsp.c, make them proper dynamic payload handlers | ||
*** <font color="#FFFF00">Patch submitted, awaiting review</font> | *** <font color="#FFFF00">Patch submitted, awaiting review</font> | ||
** factorize out common code for parsing SDP (fmtp) lines, that is duplicated in parse_h264_sdp_line, amr_parse_sdp_line and xiph_parse_sdp_line | |||
** add RTP packetizers for codecs that we already have depacketizers for (Theora, Vorbis, any other?) | ** add RTP packetizers for codecs that we already have depacketizers for (Theora, Vorbis, any other?) | ||
** support RTCP/Bye as end-of-file (see [[Small_FFmpeg_Tasks#Make_the_rtp_demuxer_support_rtcp_BYE_packets|small task]]) | ** support RTCP/Bye as end-of-file (see [[Small_FFmpeg_Tasks#Make_the_rtp_demuxer_support_rtcp_BYE_packets|small task]]) |
Revision as of 01:33, 21 June 2010
Roadmap/checklist for Josh Allmann's Summer of Code Project:
- Primary goal: improve the receiver compatibility
- Add support for more widespread formats ([list will follow check gst live555 and feng])
- X-Qt/quicktime depacketizer (see X-QT patch)
- SVQ3 [1] and QDM2 [2] depacketizers
- VP8 (see https://groups.google.com/a/webmproject.org/group/webm-discuss/browse_thread/thread/550f946b0e22ead2# )
- support Quicktime http tunnel mode
- Initial implementation done and committed, not supported in the RTSP muxer yet
- untangle the AAC and mpeg4 format specific code from rtsp.c, make them proper dynamic payload handlers
- Patch submitted, awaiting review
- factorize out common code for parsing SDP (fmtp) lines, that is duplicated in parse_h264_sdp_line, amr_parse_sdp_line and xiph_parse_sdp_line
- add RTP packetizers for codecs that we already have depacketizers for (Theora, Vorbis, any other?)
- support RTCP/Bye as end-of-file (see small task)
- Add support for more widespread formats ([list will follow check gst live555 and feng])
- Secondary goal: improve application integration
- Provide an API to expose the rtcp layer (and the equivalent in RDT dialect)
- Try to support subtitle streams (either as rtcp-xr or application/text stream)
- Make VideoLanClient, MPlayer or Xine use FFmpeg RTSP
- Patch submitted to VLC, awaiting review
- Implement protocol variations (e.g. DCCP or improve SCTP)
Mentor: Luca Barbato, Martin Storsjö, Ronald S. Bultje