90 Commits

Author SHA1 Message Date
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
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
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
c5ea6a5c75 g726enc: switch to ff_alloc_packet2()
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-03-22 19:03:19 +01:00
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
59041fd053 g726enc: use AVCodec.encode2() 2012-03-20 18:47:23 -04:00
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
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
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
da24963725 g726dec: add flush() function to reset state when seeking 2011-11-01 21:23:04 -04:00
97f5dd1d84 g726: don't pass index to g726_reset()
calculate it from c->code_size instead.
2011-11-01 21:23:04 -04:00
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
7abb73d4ba g726: wrap the decoder functions with a CONFIG_ADPCM_G726_DECODER check 2011-11-01 21:23:04 -04:00
437c11ca16 g726: group the g726_encoder AVCodec with the other encoding functions 2011-11-01 21:23:04 -04:00
50969c0f46 g726: return AVERROR(EINVAL) instead of -1 for invalid channel count 2011-11-01 21:23:03 -04:00
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
9e78d8cfdf g726: treat sample rates other than 8kHz as unofficial. 2011-11-01 21:23:03 -04:00
6e8d4a7afb g726dec: remove the sample_rate validation 2011-11-01 21:23:03 -04:00
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
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
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
e61a670b53 g726: use int16_t instead of short 2011-11-01 21:23:02 -04:00
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
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
2912e87a6c Replace FFmpeg with Libav in licence headers
Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-03-19 13:33:20 +00:00
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 d36beb3f6902b1217beda576aa18abf7eb72b03c)
2011-01-28 03:15:34 +01:00
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
73ff4f8344 1 abs() less
Originally committed as revision 13810 to svn://svn.ffmpeg.org/ffmpeg/trunk
2008-06-19 10:02:39 +00:00
4714776b6a simplify
Originally committed as revision 13807 to svn://svn.ffmpeg.org/ffmpeg/trunk
2008-06-18 21:09:36 +00:00
673a64b89b useless ()
Originally committed as revision 13806 to svn://svn.ffmpeg.org/ffmpeg/trunk
2008-06-18 21:05:07 +00:00
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