FFT in MIPS implementation is working iteratively instead
of "recursively" calling functions for smaller FFT sizes.
Some of DSP and format convert utils functions are also optimized.
Signed-off-by: Nedeljko Babic <nbabic@mips.com>
Reviewed-by: Vitor Sessak <vitor1001@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* 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>
* 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>
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)
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>