Age | Commit message (Collapse) | Author |
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Also print port numbers for this protocol.
Signed-off-by: Martin Storsjö <martin@martin.st>
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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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Signed-off-by: Martin Storsjö <martin@martin.st>
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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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Signed-off-by: Martin Storsjö <martin@martin.st>
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This is required for playback with the Stagefright RTSP framework
on Android.
Signed-off-by: Martin Storsjö <martin@martin.st>
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Signed-off-by: Martin Storsjö <martin@martin.st>
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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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Also remove one pointless zero initialization in rangecoder.c.
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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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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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Signed-off-by: Martin Storsjö <martin@martin.st>
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It's used in lavf.
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Specifically, ff_mpeg4audio_sample_rates, ff_mpeg4audio_get_config and
ff_copy_pce_data
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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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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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Signed-off-by: Martin Storsjö <martin@martin.st>
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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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Options from the AVFormatContext can be read for modifying
the generated SDP.
Signed-off-by: Martin Storsjö <martin@martin.st>
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This is more consistent with the rest of the API.
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The new name is more consistent with the rest of the API.
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Signed-off-by: Mans Rullgard <mans@mansr.com>
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Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
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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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Should fix compilation in environments unaware of IPv6.
Originally committed as revision 25528 to svn://svn.ffmpeg.org/ffmpeg/trunk
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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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Originally committed as revision 25390 to svn://svn.ffmpeg.org/ffmpeg/trunk
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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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Originally committed as revision 25125 to svn://svn.ffmpeg.org/ffmpeg/trunk
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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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Originally committed as revision 24919 to svn://svn.ffmpeg.org/ffmpeg/trunk
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IPv4 any longer
Originally committed as revision 24918 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 24917 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 24916 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 24915 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Patch by Josh Allmann, joshua dot allmann at gmail
Originally committed as revision 24797 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 24749 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Patch by Josh Allmann, joshua dot allmann at gmail
Originally committed as revision 24735 to svn://svn.ffmpeg.org/ffmpeg/trunk
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ff_url_split() is retained as an alias, as it was used by ffserver,
to avoid breaking ABI compatibility with it.
Originally committed as revision 23822 to svn://svn.ffmpeg.org/ffmpeg/trunk
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an SDP.
This allows to generate correct SDPs for H.264 video in "MP4 syntax".
Originally committed as revision 23572 to svn://svn.ffmpeg.org/ffmpeg/trunk
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This is in preparation for RTP hinting in the MOV muxer, where
it needs to be able to create SDP fragments for each media stream.
Originally committed as revision 23160 to svn://svn.ffmpeg.org/ffmpeg/trunk
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is deprecated and will be dropped at the next major bump.
Originally committed as revision 22735 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 22665 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Since this function isn't in the public API, it should have an ff_ prefix.
Originally committed as revision 22321 to svn://svn.ffmpeg.org/ffmpeg/trunk
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