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path: root/libavcodec/fmtconvert.c
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2016-01-02Merge commit 'aebf07075f4244caf591a3af71e5872fe314e87b'Hendrik Leppkes
* commit 'aebf07075f4244caf591a3af71e5872fe314e87b': dca: change the core to work with integer coefficients. Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
2016-01-02Merge commit 'a0fc780a2093784e8664f88205ee1b215e109cee'Hendrik Leppkes
* commit 'a0fc780a2093784e8664f88205ee1b215e109cee': arm64: int32_to_float_fmul neon asm Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
2015-12-23dca: change the core to work with integer coefficients.Alexandra Hájková
The DCA core decoder converts integer coefficients read from the bitstream to floats just after reading them (along with dequantization). All the other steps of the audio reconstruction are done with floats which makes the output for the DTS lossless extension (XLL) actually lossy. This patch changes the DCA core to work with integer coefficients until QMF. At this point the integer coefficients are converted to floats. The coefficients for the LFE channel (lfe_data) are not touched. This is the first step for the really lossless XLL decoding.
2015-12-14arm64: int32_to_float_fmul neon asmJanne Grunau
3% faster dts decoding on a cortex-a57. cortex-a57 cortex-a53 int32_to_float_fmul_array8_c: 1270.9 4475.6 int32_to_float_fmul_array8_neon: 328.6 569.2 int32_to_float_fmul_scalar_c: 928.5 4119.6 int32_to_float_fmul_scalar_neon: 309.1 524.1
2015-08-22fmtconvert: Remove float_interleave*Timothy Gu
They were not public or used anywhere.
2015-03-01Merge commit 'd74a8cb7e42f703be5796eeb485f06af710ae8ca'Michael Niedermayer
* commit 'd74a8cb7e42f703be5796eeb485f06af710ae8ca': fmtconvert: drop unused functions Conflicts: libavcodec/arm/fmtconvert_vfp_armv6.S libavcodec/x86/fmtconvert.asm libavcodec/x86/fmtconvert_init.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
2015-02-28fmtconvert: drop unused functionsAnton Khirnov
2013-07-22Merge commit '31c6f6f65c0ed5a894e26ce44ab0c3e89c82b9a2'Michael Niedermayer
* commit '31c6f6f65c0ed5a894e26ce44ab0c3e89c82b9a2': fmtconvert: Add a new method, int32_to_float_fmul_array8 Merged-by: Michael Niedermayer <michaelni@gmx.at>
2013-07-22fmtconvert: Add a new method, int32_to_float_fmul_array8Ben Avison
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>
2013-07-17Merge remote-tracking branch 'qatar/master'Michael Niedermayer
* 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>
2013-07-17fmtconvert: Explicitly use int32_t instead of intChristophe Gisquet
Signed-off-by: Martin Storsjö <martin@martin.st>
2013-05-18fmtconvert: int32_t input to int32_to_float_fmul_scalarChristophe Gisquet
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>
2013-05-01Merge commit '38282149b6ce8f4b8361e3b84542ba9aa8a1f32f'Michael Niedermayer
* commit '38282149b6ce8f4b8361e3b84542ba9aa8a1f32f': ppc: More consistent arch initialization Conflicts: libavcodec/fft.h libavcodec/mpegaudiodsp.c libavcodec/mpegaudiodsp.h Merged-by: Michael Niedermayer <michaelni@gmx.at>
2013-04-30ppc: More consistent arch initializationDiego Biurrun
2012-10-08Merge remote-tracking branch 'qatar/master'Michael Niedermayer
* 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>
2012-10-08x86: call most of the x86 dsp init functions under if (ARCH_X86)Janne Grunau
Rename the called dsp init functions to *_init_x86.
2012-09-05Optimization of AC3 floating point decoder for MIPSNedeljko Babic
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>
2012-08-16Merge remote-tracking branch 'qatar/master'Michael Niedermayer
* 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>
2012-08-15Don't include common.h from avutil.hMartin Storsjö
Signed-off-by: Martin Storsjö <martin@martin.st>
2011-05-19Merge remote-tracking branch 'qatar/master'Michael Niedermayer
* 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>
2011-05-19Add float_interleave() to FmtConvertContext with x86-optimized versions.Justin Ruggles
Partially based on patches by clsid2 in ffdshow-tryout. ff_float_interleave6() x86 improvements by Loren Merrit.
2011-04-04Libavcodec AC3/E-AC3/DTS decoders now output floating point data.clsid2
git-svn-id: https://ffdshow-tryout.svn.sourceforge.net/svnroot/ffdshow-tryout@3769 3b938f2f-1a1a-0410-8054-a526ea5ff92c
2011-03-19Replace FFmpeg with Libav in licence headersMans Rullgard
Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-03-08cosmetics: rename ff_fmt_convert_init_ppc() to ff_fmt_convert_init_altivec().Justin Ruggles
It only has Altivec functions and is not compiled if Altivec is disabled. (cherry picked from commit d21be5f15bec15933cb6360aa0159961d987f449)
2011-03-07cosmetics: rename ff_fmt_convert_init_ppc() to ff_fmt_convert_init_altivec().Justin Ruggles
It only has Altivec functions and is not compiled if Altivec is disabled.
2011-03-07Fix compilation on powerpc with --disable-altivec.Carl Eugen Hoyos
2011-03-04Fix compilation on powerpc with --disable-altivec.Carl Eugen Hoyos
2011-02-04Separate format conversion DSP functions from DSPContext.Justin Ruggles
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)
2011-02-02Separate format conversion DSP functions from DSPContext.Justin Ruggles
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>
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