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8 Commits

Author SHA1 Message Date
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