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path: root/libavcodec/fmtconvert.h
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* Merge commit 'aebf07075f4244caf591a3af71e5872fe314e87b'Hendrik Leppkes2016-01-02
|\ | | | | | | | | | | | | * commit 'aebf07075f4244caf591a3af71e5872fe314e87b': dca: change the core to work with integer coefficients. Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
| * dca: change the core to work with integer coefficients.Alexandra Hájková2015-12-23
| | | | | | | | | | | | | | | | | | | | | | | | 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.
* | Merge commit 'a0fc780a2093784e8664f88205ee1b215e109cee'Hendrik Leppkes2016-01-02
|\| | | | | | | | | | | | | * commit 'a0fc780a2093784e8664f88205ee1b215e109cee': arm64: int32_to_float_fmul neon asm Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
| * arm64: int32_to_float_fmul neon asmJanne Grunau2015-12-14
| | | | | | | | | | | | | | | | | | | | 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
* | fmtconvert: Remove float_interleave*Timothy Gu2015-08-22
| | | | | | | | They were not public or used anywhere.
* | Merge commit 'd74a8cb7e42f703be5796eeb485f06af710ae8ca'Michael Niedermayer2015-02-28
|\| | | | | | | | | | | | | | | | | | | | | | | * 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>
| * fmtconvert: drop unused functionsAnton Khirnov2015-02-28
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* | Merge commit '31c6f6f65c0ed5a894e26ce44ab0c3e89c82b9a2'Michael Niedermayer2013-07-22
|\| | | | | | | | | | | | | * commit '31c6f6f65c0ed5a894e26ce44ab0c3e89c82b9a2': fmtconvert: Add a new method, int32_to_float_fmul_array8 Merged-by: Michael Niedermayer <michaelni@gmx.at>
| * fmtconvert: Add a new method, int32_to_float_fmul_array8Ben Avison2013-07-22
| | | | | | | | | | | | | | | | | | | | 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>
* | Merge remote-tracking branch 'qatar/master'Michael Niedermayer2013-07-17
|\| | | | | | | | | | | | | | | | | | | | | | | | | * 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>
| * fmtconvert: Explicitly use int32_t instead of intChristophe Gisquet2013-07-17
| | | | | | | | Signed-off-by: Martin Storsjö <martin@martin.st>
* | fmtconvert: int32_t input to int32_to_float_fmul_scalarChristophe Gisquet2013-05-18
| | | | | | | | | | | | | | | | | | | | | | 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>
* | Merge commit '38282149b6ce8f4b8361e3b84542ba9aa8a1f32f'Michael Niedermayer2013-05-01
|\| | | | | | | | | | | | | | | | | | | | | | | * commit '38282149b6ce8f4b8361e3b84542ba9aa8a1f32f': ppc: More consistent arch initialization Conflicts: libavcodec/fft.h libavcodec/mpegaudiodsp.c libavcodec/mpegaudiodsp.h Merged-by: Michael Niedermayer <michaelni@gmx.at>
| * ppc: More consistent arch initializationDiego Biurrun2013-04-30
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* | Merge commit '040c565e51985477a8fa5e42d2ddfb26ebde6608'Michael Niedermayer2013-02-24
|\| | | | | | | | | | | | | | | | | | | | | * commit '040c565e51985477a8fa5e42d2ddfb26ebde6608': doc: developer: Allow tabs in the vim configuration for Automake files Remove pointless av_cold attributes in header files Conflicts: libavcodec/h264.h Merged-by: Michael Niedermayer <michaelni@gmx.at>
| * Remove pointless av_cold attributes in header filesDiego Biurrun2013-02-23
| | | | | | | | | | | | The init functions marked as av_cold have to be executed in any case, so there is no gain from trying to mark paths leading to such functions as unlikely.
* | Optimization of AC3 floating point decoder for MIPSNedeljko Babic2012-09-05
| | | | | | | | | | | | | | | | | | | | 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>
* | Merge remote-tracking branch 'qatar/master'Michael Niedermayer2012-04-04
|\| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * qatar/master: w32threads: Make pthread_cond_wait follow POSIX cosmetics: Consistently place static, inline and av_cold attributes/keywords. sbrdsp: Use standard multiple inclusion guards. pcm: K&R formatting cosmetics rawdec: Support fourccs YV16 and YV24 rtmp: implement bandwidth notification rtmp: update supported audio codecs value Conflicts: libavcodec/pcm.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
| * cosmetics: Consistently place static, inline and av_cold attributes/keywords.Diego Biurrun2012-04-04
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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)
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