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* Merge remote-tracking branch 'qatar/master'Michael Niedermayer2011-10-22
|\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * 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>
| * fmtconvert: fix and extend documentation for float_interleave()Justin Ruggles2011-10-21
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* | Merge remote-tracking branch 'qatar/master'Michael Niedermayer2011-05-19
|\| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * 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>
| * Add float_interleave() to FmtConvertContext with x86-optimized versions.Justin Ruggles2011-05-18
| | | | | | | | | | Partially based on patches by clsid2 in ffdshow-tryout. ff_float_interleave6() x86 improvements by Loren Merrit.
| * Replace FFmpeg with Libav in licence headersMans Rullgard2011-03-19
| | | | | | | | Signed-off-by: Mans Rullgard <mans@mansr.com>
| * cosmetics: rename ff_fmt_convert_init_ppc() to ff_fmt_convert_init_altivec().Justin Ruggles2011-03-07
| | | | | | | | It only has Altivec functions and is not compiled if Altivec is disabled.
| * Separate format conversion DSP functions from DSPContext.Justin Ruggles2011-02-02
| | | | | | | 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>
* Libavcodec AC3/E-AC3/DTS decoders now output floating point data.clsid22011-04-03
| | | | git-svn-id: https://ffdshow-tryout.svn.sourceforge.net/svnroot/ffdshow-tryout@3769 3b938f2f-1a1a-0410-8054-a526ea5ff92c
* cosmetics: rename ff_fmt_convert_init_ppc() to ff_fmt_convert_init_altivec().Justin Ruggles2011-03-08
| | | | | It only has Altivec functions and is not compiled if Altivec is disabled. (cherry picked from commit d21be5f15bec15933cb6360aa0159961d987f449)
* Separate format conversion DSP functions from DSPContext.Justin Ruggles2011-02-04
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 c73d99e672329c8f2df290736ffc474c360ac4ae)