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Commit Graph

10 Commits

Author SHA1 Message Date
Michael Niedermayer
aedc908601 Merge remote-tracking branch 'qatar/master'
* qatar/master: (35 commits)
  flvdec: Do not call parse_keyframes_index with a NULL stream
  libspeexdec: include system headers before local headers
  libspeexdec: return meaningful error codes
  libspeexdec: cosmetics: reindent
  libspeexdec: decode one frame at a time.
  swscale: fix signed shift overflows in ff_yuv2rgb_c_init_tables()
  Move timefilter code from lavf to lavd.
  mov: add support for hdvd and pgapmetadata atoms
  mov: rename function _stik, some indentation cosmetics
  mov: rename function _int8 to remove ambiguity, some indentation cosmetics
  mov: parse the gnre atom
  mp3on4: check for allocation failures in decode_init_mp3on4()
  mp3on4: create a separate flush function for MP3onMP4.
  mp3on4: ensure that the frame channel count does not exceed the codec channel count.
  mp3on4: set channel layout
  mp3on4: fix the output channel order
  mp3on4: allocate temp buffer with av_malloc() instead of on the stack.
  mp3on4: copy MPADSPContext from first context to all contexts.
  fmtconvert: port float_to_int16_interleave() 2-channel x86 inline asm to yasm
  fmtconvert: port int32_to_float_fmul_scalar() x86 inline asm to yasm
  ...

Conflicts:
	libavcodec/arm/h264dsp_init_arm.c
	libavcodec/h264.c
	libavcodec/h264.h
	libavcodec/h264_cabac.c
	libavcodec/h264_cavlc.c
	libavcodec/h264_ps.c
	libavcodec/h264dsp_template.c
	libavcodec/h264idct_template.c
	libavcodec/h264pred.c
	libavcodec/h264pred_template.c
	libavcodec/x86/h264dsp_mmx.c
	libavdevice/Makefile
	libavdevice/jack_audio.c
	libavformat/Makefile
	libavformat/flvdec.c
	libavformat/flvenc.c
	libavutil/pixfmt.h
	libswscale/utils.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-10-22 01:16:41 +02:00
Justin Ruggles
45add995de fmtconvert: fix and extend documentation for float_interleave() 2011-10-21 10:13:05 -04:00
Michael Niedermayer
75a37b57a5 Merge remote-tracking branch 'qatar/master'
* qatar/master:
  APIchanges: fill in date and commit for request_sample_fmt
  Add floating-point sample format support to the ac3, eac3, dca, aac, and vorbis decoders.
  Add support for request_sample_format in ffmpeg and ffplay.
  Add APIchanges entry for request_sample_fmt.
  Add request_sample_fmt field to AVCodecContext.
  Add float_interleave() to FmtConvertContext with x86-optimized versions.
  Remove unused make variable SEEK_REFFILE
  fate: remove redundant aref and vref references
  fate: remove do_ffmpeg_nocheck function
  fate: do not collect -benchmark output
  mpegaudiodec: remove decode_end() function
  fate: run aref and vref as regular tests
  mpegaudio: sanitise compute_antialias_* names
  mpeg12: add slice-threading checks to slice-threading initializers.
  h264: copy pixel_shift between slice threading contexts.
  mdec: enable frame-level multithreading.
  mdec.c: fix overread.

Conflicts:
	libavcodec/aacdec.c
	libavcodec/ac3dec.c
	libavcodec/avcodec.h
	libavcodec/dca.c
	libavcodec/h264.c
	libavcodec/mdec.c
	libavcodec/mpeg12.c
	libavcodec/options.c
	libavcodec/version.h
	libavcodec/vorbisdec.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-05-19 06:00:31 +02:00
Justin Ruggles
32f8fb8ecf Add float_interleave() to FmtConvertContext with x86-optimized versions.
Partially based on patches by clsid2 in ffdshow-tryout.
ff_float_interleave6() x86 improvements by Loren Merrit.
2011-05-18 17:27:05 -04:00
clsid2
0e09997fa4 Libavcodec AC3/E-AC3/DTS decoders now output floating point data.
git-svn-id: https://ffdshow-tryout.svn.sourceforge.net/svnroot/ffdshow-tryout@3769 3b938f2f-1a1a-0410-8054-a526ea5ff92c
2011-04-03 22:52:58 +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
Justin Ruggles
539244eeb6 cosmetics: rename ff_fmt_convert_init_ppc() to ff_fmt_convert_init_altivec().
It only has Altivec functions and is not compiled if Altivec is disabled.
(cherry picked from commit d21be5f15b)
2011-03-08 02:09:31 +01:00
Justin Ruggles
d21be5f15b cosmetics: rename ff_fmt_convert_init_ppc() to ff_fmt_convert_init_altivec().
It only has Altivec functions and is not compiled if Altivec is disabled.
2011-03-07 11:15:29 -05:00
Justin Ruggles
fe2ff6d247 Separate format conversion DSP functions from DSPContext.
This will be beneficial for use with the audio conversion API without
requiring it to depend on all of dsputil.

Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit c73d99e672)
2011-02-04 03:08:09 +01:00
Justin Ruggles
c73d99e672 Separate format conversion DSP functions from DSPContext.
This will be beneficial for use with the audio conversion API without
requiring it to depend on all of dsputil.

Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-02-02 02:44:53 +00:00