Commit Graph

98 Commits

Author SHA1 Message Date
Michael Niedermayer
a9b1536a01 Merge remote-tracking branch 'qatar/master'
* qatar/master:
  bgmc: Fix av_malloc checks in ff_bgmc_init()
  rtp: set the payload type as stream id

Conflicts:
	libavformat/rtpenc_chain.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-11-15 11:31:13 +01:00
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
Michael Niedermayer
7a72695c05 Merge commit '36ef5369ee9b336febc2c270f8718cec4476cb85'
* commit '36ef5369ee9b336febc2c270f8718cec4476cb85':
  Replace all CODEC_ID_* with AV_CODEC_ID_*
  lavc: add AV prefix to codec ids.

Conflicts:
	doc/APIchanges
	doc/examples/decoding_encoding.c
	doc/examples/muxing.c
	ffmpeg.c
	ffprobe.c
	ffserver.c
	libavcodec/8svx.c
	libavcodec/avcodec.h
	libavcodec/dnxhd_parser.c
	libavcodec/dvdsubdec.c
	libavcodec/error_resilience.c
	libavcodec/h263dec.c
	libavcodec/libvorbisenc.c
	libavcodec/mjpeg_parser.c
	libavcodec/mjpegenc.c
	libavcodec/mpeg12.c
	libavcodec/mpeg4videodec.c
	libavcodec/mpegvideo.c
	libavcodec/mpegvideo_enc.c
	libavcodec/pcm.c
	libavcodec/r210dec.c
	libavcodec/utils.c
	libavcodec/v210dec.c
	libavcodec/version.h
	libavdevice/alsa-audio-dec.c
	libavdevice/bktr.c
	libavdevice/v4l2.c
	libavformat/asfdec.c
	libavformat/asfenc.c
	libavformat/avformat.h
	libavformat/avidec.c
	libavformat/caf.c
	libavformat/electronicarts.c
	libavformat/flacdec.c
	libavformat/flvdec.c
	libavformat/flvenc.c
	libavformat/framecrcenc.c
	libavformat/img2.c
	libavformat/img2dec.c
	libavformat/img2enc.c
	libavformat/ipmovie.c
	libavformat/isom.c
	libavformat/matroska.c
	libavformat/matroskadec.c
	libavformat/matroskaenc.c
	libavformat/mov.c
	libavformat/movenc.c
	libavformat/mp3dec.c
	libavformat/mpeg.c
	libavformat/mpegts.c
	libavformat/mxf.c
	libavformat/mxfdec.c
	libavformat/mxfenc.c
	libavformat/nsvdec.c
	libavformat/nut.c
	libavformat/oggenc.c
	libavformat/pmpdec.c
	libavformat/rawdec.c
	libavformat/rawenc.c
	libavformat/riff.c
	libavformat/sdp.c
	libavformat/utils.c
	libavformat/vocenc.c
	libavformat/wtv.c
	libavformat/xmv.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-08-07 22:45:46 +02:00
Anton Khirnov
36ef5369ee Replace all CODEC_ID_* with AV_CODEC_ID_* 2012-08-07 16:00:24 +02:00
Michael Niedermayer
93342de1d8 Merge remote-tracking branch 'qatar/master'
* qatar/master:
  rtmp: Add credit/copyright to librtmp authors for parts of the RTMPE code
  rtmp: Move the CONFIG_ condition into the if conditions
  aac: Mention abbreviation as well in long_name
  build: Skip compiling rtmpdh.h if ffrtmpcrypt protocol is not enabled
  doc: Add Git configuration section
  configure: Add a dependency on https for rtmpts
  rtp: Only choose static payload types if the sample rate and channels are right

Conflicts:
	doc/git-howto.texi
	libavformat/rtmpproto.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-07-24 21:15:57 +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
Michael Niedermayer
e2cc39b609 Merge remote-tracking branch 'qatar/master'
* qatar/master: (40 commits)
  swf: check return values for av_get/new_packet().
  wavpack: Don't shift minclip/maxclip
  rtpenc: Expose the max packet size via an avoption
  rtpenc: Move max_packet_size to a context variable
  rtpenc: Add an option for not sending RTCP packets
  lavc: drop encode() support for video.
  snowenc: switch to encode2().
  snowenc: don't abuse input picture for storing information.
  a64multienc: switch to encode2().
  a64multienc: don't write into output buffer when there's no output.
  libxvid: switch to encode2().
  tiffenc: switch to encode2().
  tiffenc: properly forward error codes in encode_frame().
  lavc: drop libdirac encoder.
  gifenc: switch to encode2().
  libvpxenc: switch to encode2().
  flashsvenc: switch to encode2().
  Remove libpostproc.
  lcl: don't overwrite input memory.
  swscale: take first/lastline over/underflows into account for MMX.
  ...

Conflicts:
	.gitignore
	Makefile
	cmdutils.c
	configure
	doc/APIchanges
	libavcodec/Makefile
	libavcodec/allcodecs.c
	libavcodec/libdiracenc.c
	libavcodec/libxvidff.c
	libavcodec/qtrleenc.c
	libavcodec/tiffenc.c
	libavcodec/utils.c
	libavformat/mov.c
	libavformat/movenc.c
	libpostproc/Makefile
	libpostproc/postprocess.c
	libpostproc/postprocess.h
	libpostproc/postprocess_altivec_template.c
	libpostproc/postprocess_internal.h
	libpostproc/postprocess_template.c
	libswscale/swscale.c
	libswscale/utils.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-02-24 02:57:18 +01:00
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
Michael Niedermayer
fae714a9fb Merge remote-tracking branch 'qatar/master'
* qatar/master:
  avconv: add presets
  rtsp: Expose the flag options via private AVOptions for sdp and rtp, too
  rtsp: Make the rtsp flags avoptions set via a define
  rtpenc: Set a default video codec
  avoptions: Fix av_opt_flag_is_set
  rtp: Fix ff_rtp_get_payload_type
  doc: Update the documentation on setting options for RTSP
  rtsp: Remove the separate filter_source variable
  rtsp: Accept options via private avoptions instead of URL options
  rtsp: Simplify AVOption definitions
  rtsp: Merge the AVOption lists
  lavfi: port libmpcodecs delogo filter
  lavfi: port boxblur filter from libmpcodecs
  lavfi: add negate filter
  lavfi: add LUT (LookUp Table) generic filters
  AVOptions: don't segfault on NULL parameter in av_set_options_string()
  avio: Check for invalid buffer length.
  mpegenc/mpegtsenc: add muxrate private options.
  lavf: deprecate AVFormatContext.file_size
  mov: add support for TV metadata atoms tves, tvsn and stik

Conflicts:
	Changelog
	doc/filters.texi
	doc/protocols.texi
	libavfilter/Makefile
	libavfilter/allfilters.c
	libavfilter/avfilter.h
	libavfilter/formats.c
	libavfilter/internal.h
	libavfilter/vf_boxblur.c
	libavfilter/vf_delogo.c
	libavfilter/vf_lut.c
	libavformat/mpegtsenc.c
	libavformat/utils.c
	libavformat/version.h
	libavutil/opt.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-10-18 01:54:40 +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
Michael Niedermayer
f884ef00de Merge remote-tracking branch 'qatar/master'
* qatar/master: (31 commits)
  tiffenc: initialize forgotten avctx.
  avplay: free the active audio packet at exit.
  avplay: free rdft data used for spectrogram analysis.
  log.h: make AVClass a named struct
  fix ac3 encoder documentation
  vc1: more prettyprinting cosmetics
  vc1: prettyprint some tables
  vc1: K&R formatting cosmetics
  AVOptions: bump minor and add APIchanges entry.
  cmdutils/avtools: simplify show_help() by using av_opt_child_class_next()
  AVOptions: rename FF_OPT_TYPE_* => AV_OPT_TYPE_*
  Remove all uses of deprecated AVOptions API.
  AVOptions: add av_opt_next, deprecate av_next_option.
  AVOptions: add functions for evaluating option strings.
  AVOptions: split get_number().
  AVOptions: add av_opt_get*, deprecate av_get*.
  AVOptions: add av_opt_set*().
  AVOptions: add new API for enumerating children.
  rv34: move inverse transform functions to DSP context
  flvenc: Write the right metadata entry count
  ...

Conflicts:
	avconv.c
	cmdutils.c
	doc/APIchanges
	ffplay.c
	ffprobe.c
	libavcodec/ac3dec.c
	libavcodec/h264.c
	libavcodec/libvpxenc.c
	libavcodec/libx264.c
	libavcodec/mpeg12enc.c
	libavcodec/options.c
	libavdevice/libdc1394.c
	libavdevice/v4l2.c
	libavfilter/vf_drawtext.c
	libavformat/flvdec.c
	libavformat/mpegtsenc.c
	libavformat/options.c
	libavutil/avutil.h
	libavutil/opt.c
	libswscale/utils.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-10-13 06:00:03 +02:00
Anton Khirnov
3b3ea34655 Remove all uses of deprecated AVOptions API. 2011-10-12 16:51:16 +02:00
Mohamed Naufal Basheer
f990dc374e Add the G723.1 demuxer and decoder 2011-09-29 21:44:03 +02:00
Michael Niedermayer
7c1aba4f01 Merge remote-tracking branch 'qatar/master'
* qatar/master: (21 commits)
  fate: allow testing with libavfilter disabled
  x86: XOP/FMA4 CPU detection support
  ws_snd: misc cosmetic clean-ups
  ws_snd: remove the 2-bit ADPCM table and just subtract 2 instead.
  ws_snd: use memcpy() and memset() instead of loops
  ws_snd: use samples pointer for loop termination instead of a separate iterator variable.
  ws_snd: make sure number of channels is 1
  ws_snd: add some checks to prevent buffer overread or overwrite.
  ws_snd: decode to AV_SAMPLE_FMT_U8 instead of S16.
  flacdec: fix buffer size checking in get_metadata_size()
  rtp: Simplify ff_rtp_get_payload_type
  rtpenc: Add a payload type private option
  rtp: Correct ff_rtp_get_payload_type documentation
  avconv: replace all fprintf() by av_log().
  avconv: change av_log verbosity from ERROR to FATAL for fatal errors.
  cmdutils: replace fprintf() by av_log()
  avtools: parse loglevel before all the other options.
  oggdec: add support for Xiph's CELT codec
  sol: return error if av_get_packet() fails.
  cosmetics: reindent and pretty-print
  ...

Conflicts:
	avconv.c
	cmdutils.c
	libavcodec/avcodec.h
	libavcodec/version.h
	libavformat/oggparsecelt.c
	libavformat/utils.c
	libavutil/avutil.h

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-09-27 02:14:37 +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
Michael Niedermayer
a7758884db Merge remote-tracking branch 'qatar/master'
* qatar/master:
  rtp: factorize  dynamic payload type fallback
  flvdec: Ignore the index if it's from a creator known to be different
  cmdutils: move grow_array out of #if CONFIG_AVFILTER
  avconv: actually set InputFile.rate_emu
  ratecontrol: update last_qscale_for sooner
  Fix unnecessary shift with 9/10bit vertical scaling
  prores: mark prores as intra-only in libavformat/utils.c:is_intra_only()
  prores: return more meaningful error values
  prores: improve error message wording
  prores: cosmetics: prettyprinting, drop useless parentheses
  prores: lowercase AVCodec name entry

Conflicts:
	cmdutils.c
	libavcodec/proresdec_lgpl.c
	tests/ref/lavfi/pixfmts_scale

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-09-24 01:03:07 +02:00
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