This is no longer used since 4608996772.
It also has no implementations other than the plain C one.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
* commit 'aebf07075f4244caf591a3af71e5872fe314e87b':
dca: change the core to work with integer coefficients.
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
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.
* 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: f49564c607
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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 '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>
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.
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>
Rémi Denis-Courmont
Andreas Rheinhardt
Hendrik Leppkes
Alexandra Hájková
Janne Grunau
Timothy Gu
Michael Niedermayer
Anton Khirnov
Ben Avison
Christophe Gisquet
Diego Biurrun
Nedeljko Babic
Justin Ruggles
clsid2
Mans Rullgard