Commit Graph

90 Commits

Author SHA1 Message Date
Michael Niedermayer
6101e5322f Merge remote-tracking branch 'qatar/master'
* qatar/master:
  rtpdec_asf: Set the no_resync_search option for the chained asf demuxer
  asfdec: Add an option for not searching for the packet markers
  cosmetics: Clean up the tiffenc pix_fmts declaration to match the style of others
  cosmetics: Align codec declarations
  cosmetics: Convert mimic.c to utf-8
  avconv: remove an unused function parameter.
  avconv: remove now pointless variables.
  avconv: drop support for building without libavfilter.
  nellymoserenc: fix crash due to memsetting the wrong area.
  libavformat: Only require first packet to be known for audio/video streams
  avplay: Don't try to scale timestamps if the tb isn't set

Conflicts:
	Changelog
	configure
	ffmpeg.c
	libavcodec/aacenc.c
	libavcodec/bmpenc.c
	libavcodec/dnxhddec.c
	libavcodec/dnxhdenc.c
	libavcodec/ffv1.c
	libavcodec/flacenc.c
	libavcodec/fraps.c
	libavcodec/huffyuv.c
	libavcodec/libopenjpegdec.c
	libavcodec/mpeg12enc.c
	libavcodec/mpeg4videodec.c
	libavcodec/pamenc.c
	libavcodec/pgssubdec.c
	libavcodec/pngenc.c
	libavcodec/qtrleenc.c
	libavcodec/rawdec.c
	libavcodec/sgienc.c
	libavcodec/tiffenc.c
	libavcodec/v210dec.c
	libavcodec/wmv2dec.c
	libavformat/utils.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-04-07 22:41:37 +02:00
Martin Storsjö
00c3b67b8a cosmetics: Align codec declarations
Also break some long lines, remove codec function placeholder comments
and add spaces in sample/pixel format lists.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-04-06 22:37:38 +03:00
Paul B Mahol
ae2c33b0c2 cosmetics: remove superfluous curly brackets
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-03-23 03:09:07 +01:00
Michael Niedermayer
c5ea6a5c75 g726enc: switch to ff_alloc_packet2()
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-03-22 19:03:19 +01:00
Michael Niedermayer
967facb695 Merge remote-tracking branch 'qatar/master'
* qatar/master: (26 commits)
  adxenc: use AVCodec.encode2()
  adxenc: Use the AVFrame in ADXContext for coded_frame
  indeo4: fix out-of-bounds function call.
  configure: Restructure help output.
  configure: Internal-only components should not be command-line selectable.
  vorbisenc: use AVCodec.encode2()
  libvorbis: use AVCodec.encode2()
  libopencore-amrnbenc: use AVCodec.encode2()
  ra144enc: use AVCodec.encode2()
  nellymoserenc: use AVCodec.encode2()
  roqaudioenc: use AVCodec.encode2()
  libspeex: use AVCodec.encode2()
  libvo_amrwbenc: use AVCodec.encode2()
  libvo_aacenc: use AVCodec.encode2()
  wmaenc: use AVCodec.encode2()
  mpegaudioenc: use AVCodec.encode2()
  libmp3lame: use AVCodec.encode2()
  libgsmenc: use AVCodec.encode2()
  libfaac: use AVCodec.encode2()
  g726enc: use AVCodec.encode2()
  ...

Conflicts:
	configure
	libavcodec/Makefile
	libavcodec/ac3enc.c
	libavcodec/adxenc.c
	libavcodec/libgsm.c
	libavcodec/libvorbis.c
	libavcodec/vorbisenc.c
	libavcodec/wmaenc.c
	tests/ref/acodec/g722
	tests/ref/lavf/asf
	tests/ref/lavf/ffm
	tests/ref/lavf/mkv
	tests/ref/lavf/mpg
	tests/ref/lavf/rm
	tests/ref/lavf/ts
	tests/ref/seek/lavf_asf
	tests/ref/seek/lavf_ffm
	tests/ref/seek/lavf_mkv
	tests/ref/seek/lavf_mpg
	tests/ref/seek/lavf_rm
	tests/ref/seek/lavf_ts

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-03-22 00:40:11 +01:00
Justin Ruggles
59041fd053 g726enc: use AVCodec.encode2() 2012-03-20 18:47:23 -04:00
Michael Niedermayer
e4de71677f Merge remote-tracking branch 'qatar/master'
* qatar/master:
  aac_latm: reconfigure decoder on audio specific config changes
  latmdec: fix audio specific config parsing
  Add avcodec_decode_audio4().
  avcodec: change number of plane pointers from 4 to 8 at next major bump.
  Update developers documentation with coding conventions.
  svq1dec: avoid undefined get_bits(0) call
  ARM: h264dsp_neon cosmetics
  ARM: make some NEON macros reusable
  Do not memcpy raw video frames when using null muxer
  fate: update asf seektest
  vp8: flush buffers on size changes.
  doc: improve general documentation for MacOSX
  asf: use packet dts as approximation of pts
  asf: do not call av_read_frame
  rtsp: Initialize the media_type_mask in the rtp guessing demuxer
  Cleaned up alacenc.c

Conflicts:
	doc/APIchanges
	doc/developer.texi
	libavcodec/8svx.c
	libavcodec/aacdec.c
	libavcodec/ac3dec.c
	libavcodec/avcodec.h
	libavcodec/nellymoserdec.c
	libavcodec/tta.c
	libavcodec/utils.c
	libavcodec/version.h
	libavcodec/wmadec.c
	libavformat/asfdec.c
	tests/ref/seek/lavf_asf

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-12-03 03:00:30 +01:00
Justin Ruggles
0eea212943 Add avcodec_decode_audio4().
Deprecate avcodec_decode_audio3().
Implement audio support in avcodec_default_get_buffer().
Implement the new audio decoder API in all audio decoders.
2011-12-02 17:40:40 -05:00
Michael Niedermayer
988f585fcb Merge remote-tracking branch 'qatar/master'
* qatar/master: (44 commits)
  replacement Indeo 3 decoder
  gsm demuxer: do not allocate packet twice.
  flvenc: use first packet delay as global delay.
  ac3enc: doxygen update.
  imc: return error codes instead of 0 for error conditions.
  imc: return meaningful error codes instead of -1
  imc: do not set channel layout for stereo
  imc: validate channel count
  imc: check for ff_fft_init() failure
  imc: check output buffer size before decoding
  imc: use DSPContext.bswap16_buf() to byte-swap packet data
  rtsp: add allowed_media_types option
  libgsm: add flush function to reset the decoder state when seeking
  libgsm: simplify decoding by using a loop
  gsm: log error message when packet is too small
  libgsmdec: do not needlessly set *data_size to 0
  gsmdec: do not needlessly set *data_size to 0
  gsmdec: add flush function to reset the decoder state when seeking
  libgsmdec: check output buffer size before decoding
  gsmdec: log error message when output buffer is too small.
  ...

Conflicts:
	Changelog
	ffplay.c
	libavcodec/indeo3.c
	libavcodec/mjpeg_parser.c
	libavcodec/vp3.c
	libavformat/cutils.c
	libavformat/id3v2.c
	libavutil/parseutils.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-03 02:16:26 +01:00
Justin Ruggles
da24963725 g726dec: add flush() function to reset state when seeking 2011-11-01 21:23:04 -04:00
Justin Ruggles
97f5dd1d84 g726: don't pass index to g726_reset()
calculate it from c->code_size instead.
2011-11-01 21:23:04 -04:00
Justin Ruggles
615b2a2cf5 g726enc: add private option for setting code size directly.
This is an easy alternative to setting bit_rate. This patch also selects the
closest bit_rate to the requested one rather than requiring an exact value.
2011-11-01 21:23:04 -04:00
Justin Ruggles
7abb73d4ba g726: wrap the decoder functions with a CONFIG_ADPCM_G726_DECODER check 2011-11-01 21:23:04 -04:00
Justin Ruggles
437c11ca16 g726: group the g726_encoder AVCodec with the other encoding functions 2011-11-01 21:23:04 -04:00
Justin Ruggles
50969c0f46 g726: return AVERROR(EINVAL) instead of -1 for invalid channel count 2011-11-01 21:23:03 -04:00
Justin Ruggles
50c466d609 g726enc: use av_assert0() for sample_rate validation
This should never happen, but the check avoids a divide-by-zero.
2011-11-01 21:23:03 -04:00
Justin Ruggles
9e78d8cfdf g726: treat sample rates other than 8kHz as unofficial. 2011-11-01 21:23:03 -04:00
Justin Ruggles
6e8d4a7afb g726dec: remove the sample_rate validation 2011-11-01 21:23:03 -04:00
Justin Ruggles
6ac34eed54 g726: use bits_per_coded_sample instead of bitrate to determine mode
This requires some workarounds in the WAV muxer and demuxer. We need to write
the correct bits_per_coded_sample and block_align in the muxer. In the
demuxer, we cannot rely on the bits_per_coded_sample value, so we use the bit
rate and sample rate to determine the value.

This avoids having the decoder rely on AVCodecContext.bit_rate, which is not
required to be set by the user for decoding according to our API.
2011-11-01 21:23:03 -04:00
Justin Ruggles
d405237bae g726: split the init function for the encoder and decoder
This also allows for not having a decoder close function.
2011-11-01 21:23:03 -04:00
Justin Ruggles
c8d36d254e g726: pre-calculate the number of output samples.
Allows for checking output buffer size and simplification of decoding loop.
2011-11-01 21:23:03 -04:00
Justin Ruggles
e61a670b53 g726: use int16_t instead of short 2011-11-01 21:23:02 -04:00
Michael Niedermayer
faba79e080 Merge remote-tracking branch 'qatar/master'
* qatar/master:
  mxfdec: Include FF_INPUT_BUFFER_PADDING_SIZE when allocating extradata.
  H.264: tweak some other x86 asm for Atom
  probe: Fix insane flow control.
  mpegts: remove invalid error check
  s302m: use nondeprecated audio sample format API
  lavc: use designated initialisers for all codecs.
  x86: cabac: add operand size suffixes missing from 6c32576

Conflicts:
	libavcodec/ac3enc_float.c
	libavcodec/flacenc.c
	libavcodec/frwu.c
	libavcodec/pictordec.c
	libavcodec/qtrleenc.c
	libavcodec/v210enc.c
	libavcodec/wmv2dec.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-07-30 06:46:08 +02:00
Anton Khirnov
ec6402b7c5 lavc: use designated initialisers for all codecs.
It's more readable and less prone to breakage.
2011-07-29 08:42:34 +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
Diego Elio Pettenò
e7e2df27f8 Add ff_ prefix to data symbols of encoders, decoders, hwaccel, parsers, bsf.
None of these symbols should be accessed directly, so declare them as
hidden.

Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit d36beb3f69)
2011-01-28 03:15:34 +01:00
Diego Elio Pettenò
d36beb3f69 Add ff_ prefix to data symbols of encoders, decoders, hwaccel, parsers, bsf.
None of these symbols should be accessed directly, so declare them as
hidden.

Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-01-26 16:08:45 +00:00
Stefano Sabatini
5d6e4c160a Replace deprecated symbols SAMPLE_FMT_* with AV_SAMPLE_FMT_*, and enum
SampleFormat with AVSampleFormat.

Originally committed as revision 25730 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-11-12 11:04:40 +00:00
Justin Ruggles
c7d89948a3 Set a constant frame size for encoding G.726 audio.
Originally committed as revision 25107 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-09-11 19:52:09 +00:00
Reimar Döffinger
edac49daf5 Use "const" qualifier for pointers that point to input data of
audio encoders.
This is purely for clarity/documentation purposes.

Originally committed as revision 24481 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-07-24 13:59:49 +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
Reimar Döffinger
b5f09d31c2 Make sample_fmts and channel_layouts compound literals const to reduce size of
.data section.

Originally committed as revision 19787 to svn://svn.ffmpeg.org/ffmpeg/trunk
2009-09-06 09:15:07 +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
Stefano Sabatini
b275500706 Split bitstream.h, put the bitstream writer stuff in the new file
put_bits.h.

Originally committed as revision 18461 to svn://svn.ffmpeg.org/ffmpeg/trunk
2009-04-12 08:35:26 +00:00
Thilo Borgmann
7a00bbad21 Implement avcodec_decode_video2(), _audio3() and _subtitle2() which takes an
AVPacket argument rather than a const uint8_t *buf + int buf_size. This allows
passing of packet-specific flags from demuxer to decoder, such as the keyframe
flag, which appears necessary to playback corePNG P-frames.

Patch by Thilo Borgmann thilo.borgmann googlemail com, see also the thread
"Google Summer of Code participation" on the mailinglist.

Originally committed as revision 18351 to svn://svn.ffmpeg.org/ffmpeg/trunk
2009-04-07 15:59:50 +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
Aurelien Jacobs
b250f9c66d Change semantic of CONFIG_*, HAVE_* and ARCH_*.
They are now always defined to either 0 or 1.

Originally committed as revision 16590 to svn://svn.ffmpeg.org/ffmpeg/trunk
2009-01-13 23:44:16 +00:00
Diego Biurrun
f544a5fc84 Replace generic CONFIG_ENCODERS preprocessor conditionals by more specific
CONFIG_FOO_ENCODER conditionals where appropriate.

Originally committed as revision 15174 to svn://svn.ffmpeg.org/ffmpeg/trunk
2008-09-03 12:33:21 +00:00
Laurent Aimar
bd10f6e149 Prevent a division by 0 in the g726 decoder when the configured samplerate is 0.
patch by Laurent Aimar, fenrir via.ecp fr

Originally committed as revision 15160 to svn://svn.ffmpeg.org/ffmpeg/trunk
2008-09-02 23:09:14 +00:00
Peter Ross
fd76c37fd9 Modify all codecs to report their supported input and output sample format(s).
Originally committed as revision 14482 to svn://svn.ffmpeg.org/ffmpeg/trunk
2008-07-31 10:47:31 +00:00
Michael Niedermayer
74d9441715 Do not shift F[I] twice, it is also clearer and smaller now.
Originally committed as revision 13818 to svn://svn.ffmpeg.org/ffmpeg/trunk
2008-06-19 11:04:31 +00:00
Michael Niedermayer
50c52d2250 Factorize c->ap += (-c->ap) >> 4 out
Originally committed as revision 13817 to svn://svn.ffmpeg.org/ffmpeg/trunk
2008-06-19 11:00:17 +00:00
Michael Niedermayer
0e0d6cfd48 Get rid of G726Tables.bits.
Originally committed as revision 13816 to svn://svn.ffmpeg.org/ffmpeg/trunk
2008-06-19 10:52:47 +00:00
Michael Niedermayer
05c9f3516c Copy 4 pointers to avid dozends of ptr dereferences.
Originally committed as revision 13815 to svn://svn.ffmpeg.org/ffmpeg/trunk
2008-06-19 10:49:30 +00:00
Michael Niedermayer
fc234250b4 Does not need to be int16.
Originally committed as revision 13814 to svn://svn.ffmpeg.org/ffmpeg/trunk
2008-06-19 10:38:20 +00:00
Michael Niedermayer
cb26c9d664 Factorize I >> (c->tbls->bits - 1) out.
Originally committed as revision 13812 to svn://svn.ffmpeg.org/ffmpeg/trunk
2008-06-19 10:29:36 +00:00
Michael Niedermayer
73ff4f8344 1 abs() less
Originally committed as revision 13810 to svn://svn.ffmpeg.org/ffmpeg/trunk
2008-06-19 10:02:39 +00:00
Michael Niedermayer
4714776b6a simplify
Originally committed as revision 13807 to svn://svn.ffmpeg.org/ffmpeg/trunk
2008-06-18 21:09:36 +00:00
Michael Niedermayer
673a64b89b useless ()
Originally committed as revision 13806 to svn://svn.ffmpeg.org/ffmpeg/trunk
2008-06-18 21:05:07 +00:00
Michael Niedermayer
428c82cbac remove unneeded tr == 0
Originally committed as revision 13805 to svn://svn.ffmpeg.org/ffmpeg/trunk
2008-06-18 21:00:44 +00:00