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95 Commits

Author SHA1 Message Date
Diego Biurrun
6c1a7d07eb Use proper "" quotes for local header #includes 2013-02-01 12:51:15 +01:00
Martin Storsjö
932117171f rtp: Make sure the output format pointer is set 
Not sure if this actually happens, but we do the same check when
checking payload_type further above in the function, so it might
be needed.

Signed-off-by: Martin Storsjö <martin@martin.st>
 2013-01-24 11:31:35 +02:00
Martin Storsjö
e90820d4f8 rtp: Make sure priv_data is set before reading it 
This fixes crashes with muxing H263 into RTSP.

CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
 2013-01-23 23:30:58 +02:00
Martin Storsjö
c44784c9bb rtp: Rename a static variable to normal naming conventions 
Signed-off-by: Martin Storsjö <martin@martin.st>
 2013-01-12 00:01:51 +02:00
Martin Storsjö
58b5971881 rtp: Cosmetic cleanup 
Remove leftover debug comments, fix brace placement and
add whitespace, remove unnecessary and weirdly placed braces.

Signed-off-by: Martin Storsjö <martin@martin.st>
 2013-01-12 00:01:28 +02:00
Martin Storsjö
a925f723a9 rtp: Don't read priv_data unless it is allocated 
This makes all users of rtpenc_chain (rtsp muxer, sapenc, mov
rtp hinting) work again, broken since 8034130e0.

Signed-off-by: Martin Storsjö <martin@martin.st>
 2012-12-20 14:25:49 +02:00
Luca Barbato
8034130e06 rtp: set the payload type as stream id 
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
Anton Khirnov
36ef5369ee Replace all CODEC_ID_* with AV_CODEC_ID_* 2012-08-07 16:00:24 +02:00
Adriano Pallavicino
999c63e4ca rtp: Only choose static payload types if the sample rate and channels are right 
If using a different sample rate or number of channels, use a dynamic
payload type instead, where the parameters are passed in the SDP.

G722 is a special case where the normal rules don't apply.

Signed-off-by: Martin Storsjö <martin@martin.st>
 2012-07-24 00:42:58 +03:00
Mohamed Naufal Basheer
55c3a4f617 G.723.1 demuxer and decoder 
Signed-off-by: Kostya Shishkov <kostya.shishkov@gmail.com>
 2012-07-22 07:58:54 +02:00
Martin Storsjö
c4584f3c1f rtpenc: Allow packetizing H263 according to the old RFC 2190 
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
Martin Storsjö
2e69dd66b6 rtp: Fix ff_rtp_get_payload_type 
It was broken in 3b3ea34655
"Remove all uses of deprecated AVOptions API", where any
presence of a payload_type AVOption caused its value to
be returned, even if it wasn't set (and thus had the default
-1 value).

This caused the RTP muxer to be broken.

Signed-off-by: Martin Storsjö <martin@martin.st>
 2011-10-17 20:40:20 +03:00
Anton Khirnov
3b3ea34655 Remove all uses of deprecated AVOptions API. 2011-10-12 16:51:16 +02:00
Rafaël Carré
1430ae44e8 rtp: Simplify ff_rtp_get_payload_type 
Signed-off-by: Martin Storsjö <martin@martin.st>
 2011-09-26 21:55:27 +03:00
Rafaël Carré
9152880e95 rtpenc: Add a payload type private option 
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
Rafaël Carré
0c378ea1f7 rtp: factorize dynamic payload type fallback 
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
Mans Rullgard
2912e87a6c Replace FFmpeg with Libav in licence headers 
Signed-off-by: Mans Rullgard <mans@mansr.com>
 2011-03-19 13:33:20 +00:00
Martin Storsjö
0048a2a8d3 Handle G.722 in RTP, and all the exceptions mandated in RFC 3551 
Originally committed as revision 25125 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2010-09-15 17:35:39 +00:00
Stefano Sabatini
72415b2adb Define AVMediaType enum, and use it instead of enum CodecType, which 
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
Luca Abeni
4bf0faaafe Remove the inclusion of unneeded headers 
Originally committed as revision 21152 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2010-01-11 19:55:14 +00:00
Stefano Sabatini
9106a698e7 Rename bitstream.h to get_bits.h. 
Originally committed as revision 18494 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2009-04-13 16:20:26 +00:00
Luca Abeni
215037887d Do not return payload type 34 for H.263 (it is deprecated) 
Originally committed as revision 18346 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2009-04-07 06:37:12 +00:00
Luca Abeni
bf6d981806 Remame rtp_get_codec_info() to ff_rtp_get_codec_info(), as it is not 
a static function

Originally committed as revision 17390 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2009-02-17 08:12:51 +00:00
Luca Abeni
0550b58f4e Rename rtp_get_payload_type() to ff_rtp_get_payload_type(), as it is not 
a static function

Originally committed as revision 17364 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2009-02-16 09:36:21 +00:00
Luca Abeni
20631a9c15 Merge rtp_internal.h in rtp.h, and remove rtp_internal.h 
Originally committed as revision 16817 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2009-01-27 12:23:16 +00:00
Diego Biurrun
406792e7b0 cosmetics: Remove pointless period after copyright statement non-sentences. 
Originally committed as revision 16684 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2009-01-19 15:46:40 +00:00
Luca Abeni
309d32b0db Do not set sample_rate = 90000 for mp2 and mp3 audio over RTP 
Originally committed as revision 13943 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2008-06-24 11:07:04 +00:00
Diego Biurrun
245976da2a Use full path for #includes from another directory. 
Originally committed as revision 13098 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2008-05-09 11:56:36 +00:00
Luca Abeni
2ccf0a4290 Add a comment about missing entries 
Originally committed as revision 12646 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2008-03-31 09:54:49 +00:00
Luca Abeni
87cb064359 Use the correct size for the enc_name field (removing the arbitrary "50" size) 
Originally committed as revision 12645 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2008-03-31 09:49:58 +00:00
Luca Abeni
19faa9f469 Remove useless entries from AVRtpPayloadTypes 
Originally committed as revision 12644 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2008-03-31 09:48:04 +00:00
Luca Abeni
e4ed1fbf91 Support mp3 audio in the RTP muxer 
Originally committed as revision 12643 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2008-03-31 09:42:45 +00:00
Diego Pettenò
7d51edddd4 Make AVRtpPayloadTypes static and constant 
Patch by Diego 'Flameeyes' Pettenò (flameeyes AT gmail DOT com)

Originally committed as revision 11432 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2008-01-06 10:00:04 +00:00
Luca Abeni
83a0d3878c Split the RTP muxer out of rtp.c, to simplify the RTSP demuxer's dependencies 
Originally committed as revision 11408 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2008-01-04 20:09:48 +00:00
Luca Abeni
8eb793c459 Split the RTP demuxing functions out of rtp.c, to simplify the RTP muxer's dependencies 
Originally committed as revision 11406 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2008-01-04 19:33:50 +00:00
Diego Biurrun
d0b53d05c8 Fix some spelling mistakes. 
Originally committed as revision 11125 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2007-12-02 14:06:28 +00:00
Björn Axelsson
899681cd1d Use dynamically allocated ByteIOContext in AVFormatContext 
patch by: Björn Axelsson, bjorn d axelsson a intinor d se
thread: [PATCH] Remove static ByteIOContexts, 06 nov 2007

Originally committed as revision 11071 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2007-11-21 07:41:00 +00:00
Luca Abeni
db628029c4 Add MPEG2 support to the RTP muxer 
Originally committed as revision 11047 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2007-11-16 13:13:53 +00:00
Luca Abeni
7ed19d7fbf Remove the "AVRtpPayloadTypes[i].pt == i" assumption from RTP and RTSP 
code (this is needed for supporting MPEG2 video in the RTP muxer)

Originally committed as revision 11046 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2007-11-16 07:59:41 +00:00
Luca Abeni
18c05a375b Do not send too many RTCP packets (according to RFC 3550, the minimum 
RTCP interval should be 5 seconds)

Originally committed as revision 10930 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2007-11-05 12:25:10 +00:00
Luca Abeni
0aa7a2e690 Use a symbolic name for the payload size of an RTCP Sender Report packet 
Originally committed as revision 10929 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2007-11-05 10:15:20 +00:00
Luca Abeni
e0d21bfe83 Allow to set the maximum number of frames per RTP packet (and add support for 
this in the AAC packetizer)

Originally committed as revision 10647 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2007-10-02 14:48:08 +00:00
Luca Abeni
d0c3be9568 Fix a warning by removing an useless assignment (buf_ptr should be only 
used in the RTP muxer, and not in the demuxer)

Originally committed as revision 10561 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2007-09-24 10:43:26 +00:00
Luca Abeni
171dce486c Support for AAC streaming over RTP. Fragmentation is not implemented yet 
Originally committed as revision 10491 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2007-09-14 08:17:06 +00:00
Luca Abeni
af74c95a08 Fix timestamps in RTP packets (now, MPEG1 video with B frames works correctly) 
Originally committed as revision 10469 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2007-09-10 07:01:29 +00:00
Luca Abeni
1b31b02ed1 Properly set RTP and NTP timestamps in RTCP SR packets 
Originally committed as revision 10468 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2007-09-10 06:58:19 +00:00
Luca Abeni
98561024ac Move the RTP packetization code for MPEG12 video in its own file (rtp_mpv.c) 
Originally committed as revision 10201 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2007-08-24 07:13:34 +00:00
Panagiotis Issaris
6f3e0b2174 Replace all occurrences of AVERROR_IO with AVERROR(EIO). 
Originally committed as revision 9760 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2007-07-19 15:23:32 +00:00
Ronald S. Bultje
80fb82346e Use AV_RB* macros where appropriate. 
patch by Ronald S. Bultje, rsbultje gmail com
thread: Re: [FFmpeg-devel] remove int readers
date: Sat, 23 Jun 2007 09:32:12 -0400

Originally committed as revision 9499 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2007-07-06 09:32:34 +00:00
Vitor Sessak
7cf0999529 remove empty write trailer functions 
patch by Vitor: [vitor1001 gmail com]

Originally committed as revision 9442 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2007-06-27 06:54:47 +00:00