| Commit message (Collapse) | Author | Age | Files | Lines |
|\
| |
| |
| |
| |
| |
| | |
* commit '31c6f6f65c0ed5a894e26ce44ab0c3e89c82b9a2':
fmtconvert: Add a new method, int32_to_float_fmul_array8
Merged-by: Michael Niedermayer <michaelni@gmx.at>
|
| |
| |
| |
| |
| |
| |
| |
| |
| |
| | |
This is similar to int32_to_float_fmul_scalar, but
loads a new scalar multiplier every 8 input samples.
This enables the use of much larger input arrays, which
is important for pipelining on some CPUs (such as
ARMv6).
Signed-off-by: Martin Storsjö <martin@martin.st>
|
|\ \
| |/
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| | |
* qatar/master:
fmtconvert: Explicitly use int32_t instead of int
Conflicts:
libavcodec/ac3dec.c
libavcodec/fmtconvert.c
libavcodec/fmtconvert.h
See: f49564c6075935443323abf4571a62205e7b3c59
Merged-by: Michael Niedermayer <michaelni@gmx.at>
|
| |
| |
| |
| | |
Signed-off-by: Martin Storsjö <martin@martin.st>
|
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| | |
It was previously declared as int.
Does not change fate results for x86.
Conflicts:
libavcodec/ppc/fmtconvert_altivec.c
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
|
|\ \
| |/
| |
| |
| |
| |
| |
| |
| |
| |
| |
| | |
* commit '38282149b6ce8f4b8361e3b84542ba9aa8a1f32f':
ppc: More consistent arch initialization
Conflicts:
libavcodec/fft.h
libavcodec/mpegaudiodsp.c
libavcodec/mpegaudiodsp.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
|
| | |
|
|\ \
| |/
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| | |
* qatar/master:
x86: vc1: call ff_vc1dsp_init_x86() under if (ARCH_X86)
x86: cavs: call ff_cavsdsp_init_x86() under if (ARCH_X86)
x86: call most of the x86 dsp init functions under if (ARCH_X86)
doc: support the new website layout
doc: remove a warning from filters.texi
doc: initial nut documentation
segment: drop global headers setting
lavu: fix typo in Makefile
Conflicts:
doc/Makefile
doc/filters.texi
doc/t2h.init
libavcodec/fmtconvert.c
libavcodec/proresdsp.c
libavcodec/x86/Makefile
libavcodec/x86/vc1dsp_mmx.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
|
| |
| |
| |
| | |
Rename the called dsp init functions to *_init_x86.
|
| |
| |
| |
| |
| |
| |
| |
| |
| |
| | |
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:
Fix even more missing includes after the common.h removal
build: Factor out rangecoder dependencies to CONFIG_RANGECODER
build: Factor out error resilience dependencies to CONFIG_ERROR_RESILIENCE
x86: avcodec: Consistently name all init files
Add more missing includes after removing the implicit common.h
Add some more missing includes after removing the implicit common.h
Don't include common.h from avutil.h
rtmp: Automatically compute the hash for SWFVerification
Conflicts:
configure
doc/APIchanges
doc/examples/decoding_encoding.c
libavcodec/Makefile
libavcodec/assdec.c
libavcodec/audio_frame_queue.c
libavcodec/avpacket.c
libavcodec/dv_profile.c
libavcodec/dwt.c
libavcodec/libtheoraenc.c
libavcodec/rawdec.c
libavcodec/rv40dsp.c
libavcodec/tiff.c
libavcodec/tiffenc.c
libavcodec/v210dec.h
libavcodec/vc1dsp.c
libavcodec/x86/Makefile
libavfilter/asrc_anullsrc.c
libavfilter/avfilter.c
libavfilter/buffer.c
libavfilter/formats.c
libavfilter/vf_ass.c
libavfilter/vf_drawtext.c
libavfilter/vf_fade.c
libavfilter/vf_select.c
libavfilter/video.c
libavfilter/vsrc_testsrc.c
libavformat/version.h
libavutil/audioconvert.c
libavutil/error.h
libavutil/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
|
| |
| |
| |
| | |
Signed-off-by: Martin Storsjö <martin@martin.st>
|
|\ \
| |/
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| | |
* 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>
|
| |
| |
| |
| |
| | |
Partially based on patches by clsid2 in ffdshow-tryout.
ff_float_interleave6() x86 improvements by Loren Merrit.
|
| |
| |
| |
| | |
Signed-off-by: Mans Rullgard <mans@mansr.com>
|
| |
| |
| |
| | |
It only has Altivec functions and is not compiled if Altivec is disabled.
|
| | |
|
|
|
|
|
|
|
| |
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>
|
|
|
|
| |
git-svn-id: https://ffdshow-tryout.svn.sourceforge.net/svnroot/ffdshow-tryout@3769 3b938f2f-1a1a-0410-8054-a526ea5ff92c
|
|
|
|
|
| |
It only has Altivec functions and is not compiled if Altivec is disabled.
(cherry picked from commit d21be5f15bec15933cb6360aa0159961d987f449)
|
| |
|
|
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)
|