- 15 Aug, 2014 1 commit
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Gabriel Dume authored
Signed-off-by: Diego Biurrun <diego@biurrun.de>
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- 04 Jan, 2014 1 commit
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Martin Storsjö authored
This avoids crashes when avserver tries to create an SDP, since d77f4afa. CC: libav-stable@libav.org Signed-off-by: Martin Storsjö <martin@martin.st>
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- 14 Dec, 2013 1 commit
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Martin Storsjö authored
This avoids a memory leak (or having to worry about freeing the config string) if the colorspace isn't accepted. Signed-off-by: Martin Storsjö <martin@martin.st>
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- 15 Jan, 2013 1 commit
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Martin Storsjö authored
Also print port numbers for this protocol. Signed-off-by: Martin Storsjö <martin@martin.st>
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- 14 Nov, 2012 1 commit
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Luca Barbato authored
Support multiple video/audio streams with different format in the same session. Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
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- 09 Oct, 2012 1 commit
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Martin Storsjö authored
Signed-off-by: Martin Storsjö <martin@martin.st>
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- 08 Oct, 2012 1 commit
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Anton Khirnov authored
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- 26 Sep, 2012 1 commit
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Dmitry Samonenko authored
This packetization scheme simply places the full packets into the RTP packet without any extra header bytes. Signed-off-by: Martin Storsjö <martin@martin.st>
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- 23 Sep, 2012 1 commit
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Samuel Pitoiset authored
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- 28 Aug, 2012 2 commits
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Martin Storsjö authored
Signed-off-by: Martin Storsjö <martin@martin.st>
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Martin Storsjö authored
This is required for playback with the Stagefright RTSP framework on Android. Signed-off-by: Martin Storsjö <martin@martin.st>
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- 07 Aug, 2012 1 commit
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Anton Khirnov authored
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- 18 Jun, 2012 2 commits
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Martin Storsjö authored
Signed-off-by: Martin Storsjö <martin@martin.st>
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Martin Storsjö authored
This requires all NAL units to fit within single RTP packets. It doesn't change the actual packetization for packets that fit, but errors out and gives a helpful hint if the NAL units would have to be split, and signals the right packetization mode in the SDP. Signed-off-by: Martin Storsjö <martin@martin.st>
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- 28 Mar, 2012 1 commit
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Diego Biurrun authored
Also remove one pointless zero initialization in rangecoder.c.
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- 23 Feb, 2012 1 commit
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Martin Storsjö authored
According to newer RFCs, this packetization scheme should only be used for interfacing with legacy systems. Implementing this packetization mode properly requires parsing the full H263 bitstream to find macroblock boundaries (and knowing their macroblock and gob numbers and motion vector predictors). This implementation tries to look for GOB headers (which can be inserted by using -ps <small number>), but if the GOBs aren't small enough to fit into the MTU, the packetizer blindly splits packets at any offset and claims it to be a GOB boundary (by using Mode A from the RFC). While not correct, this seems to work with some receivers. Signed-off-by: Martin Storsjö <martin@martin.st>
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- 28 Jan, 2012 1 commit
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Alex Converse authored
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- 27 Jan, 2012 1 commit
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Anton Khirnov authored
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- 10 Dec, 2011 1 commit
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Martin Storsjö authored
If the sdp is generated before the rtp muxer is initialized (e.g. as when called from the rtsp muxer), this has to be done, otherwise the rtp muxer doesn't know that the input really is in mp4 format. Signed-off-by: Martin Storsjö <martin@martin.st>
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- 01 Dec, 2011 1 commit
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Martin Storsjö authored
Signed-off-by: Martin Storsjö <martin@martin.st>
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- 20 Oct, 2011 2 commits
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Anton Khirnov authored
It's used in lavf.
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Anton Khirnov authored
Specifically, ff_mpeg4audio_sample_rates, ff_mpeg4audio_get_config and ff_copy_pce_data
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- 26 Sep, 2011 1 commit
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Rafaël Carré authored
Specifying the payload type is useful when the type number has already been negotiated before creating the stream, for example in SIP protocol. Signed-off-by: Martin Storsjö <martin@martin.st>
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- 23 Sep, 2011 1 commit
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Rafaël Carré authored
Move the identical code in rtp_write_header() and ff_sdp_write_media() inside ff_rtp_get_payload_type() Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
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- 10 Jun, 2011 3 commits
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Martin Storsjö authored
Signed-off-by: Martin Storsjö <martin@martin.st>
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Juan Carlos Rodriguez authored
This is enabled with an AVOption on the RTP muxer. The SDP generator looks for a latm flag in the rtpflags field. Signed-off-by: Martin Storsjö <martin@martin.st>
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Martin Storsjö authored
Options from the AVFormatContext can be read for modifying the generated SDP. Signed-off-by: Martin Storsjö <martin@martin.st>
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- 08 Jun, 2011 1 commit
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Anton Khirnov authored
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- 08 Apr, 2011 2 commits
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Anton Khirnov authored
This is more consistent with the rest of the API.
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Anton Khirnov authored
The new name is more consistent with the rest of the API.
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- 19 Mar, 2011 1 commit
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Mans Rullgard authored
Signed-off-by: Mans Rullgard <mans@mansr.com>
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- 17 Feb, 2011 1 commit
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Anton Khirnov authored
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
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- 20 Dec, 2010 1 commit
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Martin Storsjö authored
While not mentioned in RFC 4629, this is required for H.263 in 3GPP TS 26.234. It is in practice required for playback with Android stagefright and on Samsung bada phones. Originally committed as revision 26062 to svn://svn.ffmpeg.org/ffmpeg/trunk
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- 19 Oct, 2010 1 commit
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Martin Storsjö authored
Should fix compilation in environments unaware of IPv6. Originally committed as revision 25528 to svn://svn.ffmpeg.org/ffmpeg/trunk
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- 08 Oct, 2010 1 commit
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Martin Storsjö authored
According to RFC 4566, a TTL value must not be present for IPv6 multicast. Originally committed as revision 25412 to svn://svn.ffmpeg.org/ffmpeg/trunk
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- 07 Oct, 2010 2 commits
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Martin Storsjö authored
Originally committed as revision 25390 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Martin Storsjö authored
No such option is used anywhere else. Instead, detect the address type. Originally committed as revision 25389 to svn://svn.ffmpeg.org/ffmpeg/trunk
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- 15 Sep, 2010 1 commit
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Martin Storsjö authored
Originally committed as revision 25125 to svn://svn.ffmpeg.org/ffmpeg/trunk
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- 25 Aug, 2010 2 commits
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Andreas Öman authored
Used when compiled without CONFIG_RTP_MUXER Fallout from r24915 Originally committed as revision 24935 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Martin Storsjö authored
Originally committed as revision 24919 to svn://svn.ffmpeg.org/ffmpeg/trunk
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