611bf39bde
sdp: Include SRTP crypto params if using the srtp protocol
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Also print port numbers for this protocol.
Signed-off-by: Martin Storsjö <martin@martin.st>
2013-01-15 11:55:29 +02:00
8034130e06
rtp: set the payload type as stream id
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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>
2012-11-14 20:38:51 +01:00
c136a813d7
rtp: Support packetization/depacketization of opus
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Signed-off-by: Martin Storsjö <martin@martin.st>
2012-10-09 11:57:11 +03:00
716d413c13
Replace PIX_FMT_* -> AV_PIX_FMT_*, PixelFormat -> AVPixelFormat
2012-10-08 07:13:26 +02:00
490ae95aa8
rtpenc: Add support for packetizing speex
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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>
2012-09-26 19:04:57 +03:00
cee1950bbb
rtp: Packetization of JPEG (RFC 2435)
2012-09-23 21:58:41 +03:00
d4bba93f4d
sdp: Use static const char arrays instead of pointers to strings
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Signed-off-by: Martin Storsjö <martin@martin.st>
2012-08-28 22:35:17 +03:00
06b5246c84
sdp: Include profile-level-id for H264
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This is required for playback with the Stagefright RTSP framework
on Android.
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-08-28 19:32:19 +03:00
36ef5369ee
Replace all CODEC_ID_* with AV_CODEC_ID_*
2012-08-07 16:00:24 +02:00
579fd87b46
rtpenc: Support packetizing iLBC
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Signed-off-by: Martin Storsjö <martin@martin.st>
2012-06-18 22:01:04 +03:00
d77f4afa98
rtpenc: Allow requesting H264 RTP packetization mode 0
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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>
2012-06-18 12:27:56 +03:00
a92be9b856
Replace memset(0) by zero initializations.
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Also remove one pointless zero initialization in rangecoder.c.
2012-03-28 09:38:33 +02:00
c4584f3c1f
rtpenc: Allow packetizing H263 according to the old RFC 2190
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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>
2012-02-23 15:27:52 +02:00
7181c4edee
cosmetics: Remove extra newlines at EOF
2012-01-27 17:19:09 -08:00
1d911bb404
lavf: remove disabled FF_API_SDP_CREATE cruft
2012-01-27 10:52:42 +01:00
0ebd4083e1
sdp: Restore the original mp4 format h264 extradata if converted
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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>
2011-12-11 01:37:26 +02:00
04403ec2e4
rtpenc: Add support for G726 audio
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Signed-off-by: Martin Storsjö <martin@martin.st>
2011-12-01 23:19:25 +02:00
357db4c263
lavc: use avpriv_ prefix for ff_split_xiph_headers.
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It's used in lavf.
2011-10-20 21:06:58 +02:00
59a9a23581
lavc: use avpriv_ prefix for some mpeg4audio symbols used in lavf.
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Specifically, ff_mpeg4audio_sample_rates, ff_mpeg4audio_get_config and
ff_copy_pce_data
2011-10-20 21:06:57 +02:00
9152880e95
rtpenc: Add a payload type private option
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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>
2011-09-26 21:54:57 +03:00
0c378ea1f7
rtp: factorize dynamic payload type fallback
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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>
2011-09-23 22:00:24 +02:00
9c434ce826
sdp: Reindent after the previous commit
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Signed-off-by: Martin Storsjö <martin@martin.st>
2011-06-10 10:49:55 +03:00
0832122880
rtpenc: MP4A-LATM payload support
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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>
2011-06-10 10:49:20 +03:00
0558e266a2
sdp: Allow passing an AVFormatContext to the SDP generation
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Options from the AVFormatContext can be read for modifying
the generated SDP.
Signed-off-by: Martin Storsjö <martin@martin.st>
2011-06-10 10:45:22 +03:00
d2d67e424f
Remove all uses of now deprecated metadata functions.
2011-06-08 07:43:45 +02:00
c234d40bd9
lavf: rename a parameter of av_sdp_create from buff->buf
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This is more consistent with the rest of the API.
2011-04-08 16:45:41 +02:00
c3675dfe1e
lavf: rename avf_sdp_create to av_sdp_create.
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The new name is more consistent with the rest of the API.
2011-04-08 16:45:40 +02:00
2912e87a6c
Replace FFmpeg with Libav in licence headers
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Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-03-19 13:33:20 +00:00
ab0287fcbd
Move find_info_tag to lavu and add av_ prefix to it
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Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
2011-02-17 15:35:18 +01:00
9b14ec5fae
sdp: Add a framesize attribute to H.263 SDP descriptions
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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
2010-12-20 20:46:40 +00:00
095e5c0b6d
sdp: Conditionally compile code using AF_INET6
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Should fix compilation in environments unaware of IPv6.
Originally committed as revision 25528 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-10-19 07:50:40 +00:00
6898677e54
sdp: Only specify the TTL for IPv4 addresses
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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
2010-10-08 11:58:47 +00:00
87b2b40a33
Reindent
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Originally committed as revision 25390 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-10-07 08:00:03 +00:00
10105c9b29
sdp: Don't require the explicit ?multicast option
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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
2010-10-07 07:59:35 +00:00
0048a2a8d3
Handle G.722 in RTP, and all the exceptions mandated in RFC 3551
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Originally committed as revision 25125 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-09-15 17:35:39 +00:00
f06d6c751f
Make stub version of ff_sdp_write_media() match its prototype
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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
2010-08-25 21:07:04 +00:00
5a8693ef59
Reindent
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Originally committed as revision 24919 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-08-25 13:31:33 +00:00
7ad526d288
Simplify resolve_destination in sdp.c further, now that we don't enforce IPv4 any longer
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Originally committed as revision 24918 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-08-25 13:30:06 +00:00
1272ae7e50
10l, add new function parameters to resolve_destination for !config_network too
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Originally committed as revision 24917 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-08-25 12:51:20 +00:00
cc83027c67
Reindent, rewrap lines
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Originally committed as revision 24916 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-08-25 12:34:42 +00:00
7ad1dc5447
Properly handle IPv6 addresses in the SDP generation
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Originally committed as revision 24915 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-08-25 12:29:32 +00:00
7b18d94cb3
Add RTP packetization of VP8
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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
2010-08-16 14:21:17 +00:00
b5c4bb989d
rtpenc_xiph: Set the ident value via a define
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Originally committed as revision 24749 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-08-09 10:31:59 +00:00
91af5601c1
Add RTP packetization of Theora and Vorbis
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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
2010-08-07 11:16:07 +00:00
f3bfe388b5
Make ff_url_split() public
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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
2010-06-27 14:16:46 +00:00
ede13f55f0
Use a bitstream filter for converting the extradata syntax when generating an SDP.
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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
2010-06-11 08:01:45 +00:00
0341b6994a
Make ff_sdp_write_media a lavf-internal function
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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
2010-05-18 19:32:59 +00:00
72415b2adb
Define AVMediaType enum, and use it instead of enum CodecType, which
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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
2010-03-30 23:30:55 +00:00
ddbeb95447
Add a lowercase parameter to ff_data_to_hex
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Originally committed as revision 22665 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-03-25 07:13:20 +00:00
c5c6e67c28
Rename url_split to ff_url_split
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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
2010-03-08 09:03:25 +00:00
Martin Storsjö
Luca Barbato
Anton Khirnov
Dmitry Samonenko
Samuel Pitoiset
Diego Biurrun
Alex Converse
Rafaël Carré
Juan Carlos Rodriguez
Mans Rullgard
Andreas Öman
Josh Allmann
Luca Abeni
Stefano Sabatini