Diego Biurrun
6c1a7d07eb
Use proper "" quotes for local header #includes
12 years ago
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>
12 years ago
Martin Storsjö
e90820d4f8
rtp: Make sure priv_data is set before reading it
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This fixes crashes with muxing H263 into RTSP.
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
12 years ago
Martin Storsjö
c44784c9bb
rtp: Rename a static variable to normal naming conventions
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Signed-off-by: Martin Storsjö <martin@martin.st>
12 years ago
Martin Storsjö
58b5971881
rtp: Cosmetic cleanup
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Remove leftover debug comments, fix brace placement and
add whitespace, remove unnecessary and weirdly placed braces.
Signed-off-by: Martin Storsjö <martin@martin.st>
12 years ago
Martin Storsjö
a925f723a9
rtp: Don't read priv_data unless it is allocated
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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>
12 years ago
Luca Barbato
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>
12 years ago
Anton Khirnov
36ef5369ee
Replace all CODEC_ID_* with AV_CODEC_ID_*
13 years ago
Adriano Pallavicino
999c63e4ca
rtp: Only choose static payload types if the sample rate and channels are right
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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>
13 years ago
Mohamed Naufal Basheer
55c3a4f617
G.723.1 demuxer and decoder
...
Signed-off-by: Kostya Shishkov <kostya.shishkov@gmail.com>
13 years ago
Martin Storsjö
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>
13 years ago
Martin Storsjö
2e69dd66b6
rtp: Fix ff_rtp_get_payload_type
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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>
13 years ago
Anton Khirnov
3b3ea34655
Remove all uses of deprecated AVOptions API.
13 years ago
Mohamed Naufal Basheer
f990dc374e
Add the G723.1 demuxer and decoder
13 years ago
Rafaël Carré
1430ae44e8
rtp: Simplify ff_rtp_get_payload_type
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Signed-off-by: Martin Storsjö <martin@martin.st>
13 years ago
Rafaël Carré
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>
13 years ago
Rafaël Carré
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>
13 years ago
Mans Rullgard
2912e87a6c
Replace FFmpeg with Libav in licence headers
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Signed-off-by: Mans Rullgard <mans@mansr.com>
14 years ago
Martin Storsjö
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
14 years ago
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
15 years ago
Luca Abeni
4bf0faaafe
Remove the inclusion of unneeded headers
...
Originally committed as revision 21152 to svn://svn.ffmpeg.org/ffmpeg/trunk
15 years ago
Stefano Sabatini
9106a698e7
Rename bitstream.h to get_bits.h.
...
Originally committed as revision 18494 to svn://svn.ffmpeg.org/ffmpeg/trunk
16 years ago
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
16 years ago
Luca Abeni
bf6d981806
Remame rtp_get_codec_info() to ff_rtp_get_codec_info(), as it is not
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a static function
Originally committed as revision 17390 to svn://svn.ffmpeg.org/ffmpeg/trunk
16 years ago
Luca Abeni
0550b58f4e
Rename rtp_get_payload_type() to ff_rtp_get_payload_type(), as it is not
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a static function
Originally committed as revision 17364 to svn://svn.ffmpeg.org/ffmpeg/trunk
16 years ago
Luca Abeni
20631a9c15
Merge rtp_internal.h in rtp.h, and remove rtp_internal.h
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Originally committed as revision 16817 to svn://svn.ffmpeg.org/ffmpeg/trunk
16 years ago
Diego Biurrun
406792e7b0
cosmetics: Remove pointless period after copyright statement non-sentences.
...
Originally committed as revision 16684 to svn://svn.ffmpeg.org/ffmpeg/trunk
16 years ago
Luca Abeni
309d32b0db
Do not set sample_rate = 90000 for mp2 and mp3 audio over RTP
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Originally committed as revision 13943 to svn://svn.ffmpeg.org/ffmpeg/trunk
17 years ago
Diego Biurrun
245976da2a
Use full path for #includes from another directory.
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Originally committed as revision 13098 to svn://svn.ffmpeg.org/ffmpeg/trunk
17 years ago
Luca Abeni
2ccf0a4290
Add a comment about missing entries
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Originally committed as revision 12646 to svn://svn.ffmpeg.org/ffmpeg/trunk
17 years ago
Luca Abeni
87cb064359
Use the correct size for the enc_name field (removing the arbitrary "50" size)
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Originally committed as revision 12645 to svn://svn.ffmpeg.org/ffmpeg/trunk
17 years ago
Luca Abeni
19faa9f469
Remove useless entries from AVRtpPayloadTypes
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Originally committed as revision 12644 to svn://svn.ffmpeg.org/ffmpeg/trunk
17 years ago
Luca Abeni
e4ed1fbf91
Support mp3 audio in the RTP muxer
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Originally committed as revision 12643 to svn://svn.ffmpeg.org/ffmpeg/trunk
17 years ago
Diego Pettenò
7d51edddd4
Make AVRtpPayloadTypes static and constant
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Patch by Diego 'Flameeyes' Pettenò (flameeyes AT gmail DOT com)
Originally committed as revision 11432 to svn://svn.ffmpeg.org/ffmpeg/trunk
17 years ago
Luca Abeni
83a0d3878c
Split the RTP muxer out of rtp.c, to simplify the RTSP demuxer's dependencies
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Originally committed as revision 11408 to svn://svn.ffmpeg.org/ffmpeg/trunk
17 years ago
Luca Abeni
8eb793c459
Split the RTP demuxing functions out of rtp.c, to simplify the RTP muxer's dependencies
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Originally committed as revision 11406 to svn://svn.ffmpeg.org/ffmpeg/trunk
17 years ago
Diego Biurrun
d0b53d05c8
Fix some spelling mistakes.
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Originally committed as revision 11125 to svn://svn.ffmpeg.org/ffmpeg/trunk
17 years ago
Björn Axelsson
899681cd1d
Use dynamically allocated ByteIOContext in AVFormatContext
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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
17 years ago
Luca Abeni
db628029c4
Add MPEG2 support to the RTP muxer
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Originally committed as revision 11047 to svn://svn.ffmpeg.org/ffmpeg/trunk
17 years ago
Luca Abeni
7ed19d7fbf
Remove the "AVRtpPayloadTypes[i].pt == i" assumption from RTP and RTSP
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code (this is needed for supporting MPEG2 video in the RTP muxer)
Originally committed as revision 11046 to svn://svn.ffmpeg.org/ffmpeg/trunk
17 years ago
Luca Abeni
18c05a375b
Do not send too many RTCP packets (according to RFC 3550, the minimum
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RTCP interval should be 5 seconds)
Originally committed as revision 10930 to svn://svn.ffmpeg.org/ffmpeg/trunk
17 years ago
Luca Abeni
0aa7a2e690
Use a symbolic name for the payload size of an RTCP Sender Report packet
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Originally committed as revision 10929 to svn://svn.ffmpeg.org/ffmpeg/trunk
17 years ago
Luca Abeni
e0d21bfe83
Allow to set the maximum number of frames per RTP packet (and add support for
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this in the AAC packetizer)
Originally committed as revision 10647 to svn://svn.ffmpeg.org/ffmpeg/trunk
17 years ago
Luca Abeni
d0c3be9568
Fix a warning by removing an useless assignment (buf_ptr should be only
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used in the RTP muxer, and not in the demuxer)
Originally committed as revision 10561 to svn://svn.ffmpeg.org/ffmpeg/trunk
18 years ago
Luca Abeni
171dce486c
Support for AAC streaming over RTP. Fragmentation is not implemented yet
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Originally committed as revision 10491 to svn://svn.ffmpeg.org/ffmpeg/trunk
18 years ago
Luca Abeni
af74c95a08
Fix timestamps in RTP packets (now, MPEG1 video with B frames works correctly)
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Originally committed as revision 10469 to svn://svn.ffmpeg.org/ffmpeg/trunk
18 years ago
Luca Abeni
1b31b02ed1
Properly set RTP and NTP timestamps in RTCP SR packets
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Originally committed as revision 10468 to svn://svn.ffmpeg.org/ffmpeg/trunk
18 years ago
Luca Abeni
98561024ac
Move the RTP packetization code for MPEG12 video in its own file (rtp_mpv.c)
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Originally committed as revision 10201 to svn://svn.ffmpeg.org/ffmpeg/trunk
18 years ago
Panagiotis Issaris
6f3e0b2174
Replace all occurrences of AVERROR_IO with AVERROR(EIO).
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Originally committed as revision 9760 to svn://svn.ffmpeg.org/ffmpeg/trunk
18 years ago
Ronald S. Bultje
80fb82346e
Use AV_RB* macros where appropriate.
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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
18 years ago