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This commit reworks the TNS implementation to a hybrid between what
the specifications say, what the decoder does and what's the best
thing to do.
The filter application function was copied from the decoder and
modified such that it applies the inverse AR filter to the
coefficients. The LPC coefficients themselves are fed into the
same quantization expression that the specifications say should
be used however further processing is not done, instead they're
converted to the form that the decoder expects them to be in
and are sent off to the compute_lpc_coeffs function exactly the
way the decoder does. This function does all conversions and will
return the exact coefficients that the decoder will generate, which
are then applied to the coefficients.
Having the exact same coefficients on both the encoder and decoder
is a must since otherwise the entire sfb's over which the filter
is applied will be attenuated.
Despite this major rework, TNS might not work fine on some audio
types at very low bitrates (e.g. sub 90kbps) as it can attenuate
some coefficients too much. Users are advised to experiment with
TNS at higher bitrates if they wish to use this tool or simply
wait for the implementation to be improved.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
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This commit abandons the way the specifications state to
quantize the coefficients, makes use of the new LPC float
functions and is much better.
The original way of converting non-normalized float samples
to int32_t which out LPC system expects was wrong and it was
wrong to assume the coefficients that are generated are also
valid. It was essentially a full garbage-in, garbage-out
system and it definitely shows when looking at spectrals
and listening. The high frequencies were very overattenuated.
The new LPC function performs the analysis directly.
The specifications state to quantize the coefficients into
four bit index values using an asin() function which of course
had to have ugly ternary operators because the function turns
negative if the coefficients are negative which when encoding
causes invalid bitstream to get generated.
This deviates from this by using the direct TNS tables, which
are fairly small since you only have 4 bits at most for index
values. The LPC values are directly quantized against the tables
and are then used to perform filtering after the requantization,
which simply fetches the array values.
The end result is that TNS works much better now and doesn't
attenuate anything but the actual signal, e.g. TNS removes
quantization errors and does it's job correctly now.
It might be enabled by default soon since it doesn't hurt and
helps reduce nastyness at low bitrates.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
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This commit completely alters the algorithm of prediction.
The original commit which introduced prediction was completely
incorrect to even remotely care about what the actual coefficients
contain or whether any options were enabled. Not my actual fault.
This commit treats prediction the way the decoder does and expects
to do: like lossy encryption. Everything related to prediction now
happens at the very end but just before quantization and encoding
of coefficients. On the decoder side, prediction happens before
anything has had a chance to even access the coefficients.
Also the original implementation had problems because it actually
touched the band_type of special bands which already had their
scalefactor indices marked and it's a wonder the asserion wasn't
triggered when transmitting those.
Overall, this now drastically increases audio quality and you should
think about enabling it if you don't plan on playing anything encoded
on really old low power ultra-embedded devices since they might not
support decoding of prediction or AAC-Main. Though the specifications
were written ages ago and as times change so do the FLOPS.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
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This commit finalizes AAC-Main profile encoding support
by implementing all mandatory and optional tools available
in the specifications and current decoders.
The AAC-Main profile reqires that prediction support be
present (although decoders don't require it to be enabled)
for an encoder to be deemed capable of AAC-Main encoding,
as well as TNS, PNS and IS, all of which were implemented
with previous commits or earlier of this year.
Users are encouraged to test the new functionality using either
-profile:a aac_main or -aac_pred 1, the former of which will enable
the prediction option by default and the latter will change the
profile to AAC-Main. No other options shall be changed by enabling
either, it's currently up to the users to decide what's best.
The current implementation works best using M/S and/or IS,
so users are also welcome to enable both options and any
other options (TNS, PNS) for maximum quality.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
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This commit implements temporal noise shaping support in the
encoder, along with an -aac_tns option to toggle it on or off
(off by default for now). TNS will increase audio quality
and reduce quantization noise by applying a multitap FIR filter
across allowed coefficients and transmit side information to the
decoder so it could create an inverse filter.
Users are encouraged to test the new functionality by enabling
-aac_tns 1 during encoding.
No major bugs are observable at this time so after a while if no
new problems appear and if the current implementation is deemed
of high enough quality and stability it will be enabled by default,
possibly at the same time the encoder has its experimental flag
removed and becomes the standard aac encoder in ffmpeg.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
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This commit permits for the use of the Main profile
in encoding. The functionality of that profile will
be added in the commits following. By itself, this
commit does not alter anything.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
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This commit moves the quantizer to a separate header file.
This allows the quantizer to be used from a separate files outside
of aaccoder without having to put another function pointer and will
result in a slight speedup as the compiler can do more optimizations.
This is required for commits following.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
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This commit only creates and initializes an LTP
context which is needed for upcoming commits (TNS).
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
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This commit removes a redundant argument from the functions in aaccoder.
The argument lambda was redundant as it was just a copy of s->lambda,
to which all functions have access to anyway. This cleans up the function
pointers a bit which is helpful as there are a lot of other search_for_*
functions under development and with them populated it gets messy.
Reviewed-by: Claudio Freire <klaussfreire@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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Avoid clipping due to quantization noise to produce audible
artifacts, by detecting near-clipping signals and both attenuating
them a little and encoding escape-encoded bands (usually the
loudest) rounding towards zero instead of nearest, which tends to
decrease overall energy and thus clipping.
Currently fate tests measure numerical error so this change makes
tests using asynth (which are near clipping) report higher error
not less, because of window attenuation. Yet, they sound better,
not worse (albeit subtle, other samples aren't subtle at all).
Only measuring psychoacoustically weighted error would make for
a representative test, so that will be left for a future patch.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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This commit moves the generation of ff_aac_pow34sf_tab[] out of the
encoder and into the table generator. The original commit log for
this table in 2011 actually mentions that it should be moved outside
but this never happened.
This is the first commit which cleans up the encoder a little.
Reviewed-by: Claudio Freire <klaussfreire@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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This commit implements intensity stereo coding support
to the native aac encoder. This is a way to increase the efficiency
of the encoder by zeroing the right channel's spectral coefficients
(in a channel pair) and rederiving them in the decoder using information
from the scalefactor indices of special band types. This commit
confomrs to the official ISO 13818-7 specifications, although due to
their ambiguity certain deviations have been taken to ensure maximum
sound quality. This commit has been extensively tested and has shown
to not result in audiable audio artifacts unless in extreme cases.
This commit also adds an option, aac_is, which has the value of
0 by default. Intensity Stereo is part of the scalable aac profile
and is thus non-default.
The way IS coding works is that it rederives the right channel's
spectral coefficients from the left channel via the scalefactor
index values left in the right channel. Since an entire band's
spectral coefficients do not need to be coded, the encoder's
efficiency jumps up and it unzeroes some high frequency values
which it previously did not have enough bits to encode. That way
less information is lost than the information lost by rederiving
the spectral coefficients with some error. This is why the
filesize of files encoded with IS do not decrease significantly.
Users wishing that IS coding should reduce filesize are expected
to reduce their encoding bitrates appropriately.
This is V2 of the commit. The old version did not mark ms_mask as
0 since M/S and IS coding are incompactible, which resulted in
distortions with M/S coding enabled. This version also improves
phase detection by measuring it for every spectral coefficient in
the band and using a simple majority rule to determine whether the
coefficients are in or out of phase. Also, the energy values per
spectral coefficient were changed as to reflect the
official specifications.
Reviewed-by: Claudio Freire <klaussfreire@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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This commit finalizes the PNS implementation previously added to the encoder
by moving it to a seperate function search_for_pns() and thus making it
coder-generic. This new implementation makes use of the spread field of
the psy bands and the lambda quality feedback paremeter. The spread of the
spectrum in a band prevents PNS from being used excessively and thus preserve
more phase information in high frequencies. The lambda parameter allows
the number of PNS-marked bands to vary based on the lambda parameter and the
amount of bits available, making better choices on which bands are to be marked
as noise. Comparisons with the previous PNS implementation can be found
here: https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/
This is V2 of the patch, the changes from the previous version being that this
version uses the new band->spread metric from aacpsy and normalizes the
energy using the group size. These changes were suggested by Claudio Freire
on the mailing list. Another change is the use of lambda to alter the
frequency threshold. This change makes the actual threshold frequencies
vary between +-2Khz of what's specified, depending on frame encoding performance.
Reviewed-by: Claudio Freire <klaussfreire@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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This commit enables the function added with commit 7c10b87 and uses that
new function for setting any special scalefactor indices. This commit does
not change the behaviour of the encoder since no bands are being marked as
either NOISE_BT(due to the previous PNS implementation removed in the
previous commit) or INTENSITY_BT2/INTENSITY_BT.
Reviewed-by: Claudio Freire <klaussfreire@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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This commit implements the perceptual noise substitution AAC extension. This is a proof of concept
implementation, and as such, is not enabled by default. This is the fourth revision of this patch,
made after some problems were noted out. Any changes made since the previous revisions have been indicated.
In order to extend the encoder to use an additional codebook, the array holding each codebook has been
modified with two additional entries - 13 for the NOISE_BT codebook and 12 which has a placeholder function.
The cost system was modified to skip the 12th entry using an array to map the input and outputs it has. It
also does not accept using the 13th codebook for any band which is not marked as containing noise, thereby
restricting its ability to arbitrarily choose it for bands. The use of arrays allows the system to be easily
extended to allow for intensity stereo encoding, which uses additional codebooks.
The 12th entry in the codebook function array points to a function which stops the execution of the program
by calling an assert with an always 'false' argument. It was pointed out in an email discussion with
Claudio Freire that having a 'NULL' entry can result in unexpected behaviour and could be used as
a security hole. There is no danger of this function being called during encoding due to the codebook maps introduced.
Another change from version 1 of the patch is the addition of an argument to the encoder, '-aac_pns' to
enable and disable the PNS. This currently defaults to disable the PNS, as it is experimental.
The switch will be removed in the future, when the algorithm to select noise bands has been improved.
The current algorithm simply compares the energy to the threshold (multiplied by a constant) to determine
noise, however the FFPsyBand structure contains other useful figures to determine which bands carry noise more accurately.
Some of the sample files provided triggered an assertion when the parameter to tune the threshold was set to
a value of '2.2'. Claudio Freire reported the problem's source could be in the range of the scalefactor
indices for noise and advised to measure the minimal index and clip anything above the maximum allowed
value. This has been implemented and all the files which used to trigger the asserion now encode without error.
The third revision of the problem also removes unneded variabes and comparisons. All of them were
redundant and were of little use for when the PNS implementation would be extended.
The fourth revision moved the clipping of the noise scalefactors outside the second loop of the two-loop
algorithm in order to prevent their redundant calculations. Also, freq_mult has been changed to a float
variable due to the fact that rounding errors can prove to be a problem at low frequencies.
Considerations were taken whether the entire expression could be evaluated inside the expression
, but in the end it was decided that it would be for the best if just the type of the variable were
to change. Claudio Freire reported the two problems. There is no change of functionality
(except for low sampling frequencies) so the spectral demonstrations at the end of this commit's message were not updated.
Finally, the way energy values are converted to scalefactor indices has changed since the first commit,
as per the suggestion of Claudio Freire. This may still have some drawbacks, but unlike the first commit
it works without having redundant offsets and outputs what the decoder expects to have, in terms of the
ranges of the scalefactor indices.
Some spectral comparisons: https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/Original.png (original),
https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/PNS_NO.png (encoded without PNS),
https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/PNS1.2.png (encoded with PNS, const = 1.2),
https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/Difference1.png (spectral difference).
The constant is the value which multiplies the threshold when it gets compared to the energy, larger
values means more noise will be substituded by PNS values. Example when const = 2.2:
https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/PNS_2.2.png
Reviewed-by: Claudio Freire <klaussfreire@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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Signed-off-by: Timothy Gu <timothygu99@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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Signed-off-by: Bojan Zivkovic <bojan@mips.com>
Reviewed-by: Nedeljko Babic <Nedeljko.Babic@imgtec.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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* commit 'c242bbd8b6939507a1a6fb64101b0553d92d303f':
Remove unnecessary dsputil.h #includes
Conflicts:
libavcodec/ffv1.c
libavcodec/h261dec.c
libavcodec/h261enc.c
libavcodec/h264pred.c
libavcodec/lpc.h
libavcodec/mjpegdec.c
libavcodec/rectangle.h
libavcodec/x86/idct_sse2_xvid.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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* commit '42d324694883cdf1fff1612ac70fa403692a1ad4':
floatdsp: move vector_fmul_reverse from dsputil to avfloatdsp.
Conflicts:
libavcodec/arm/dsputil_init_vfp.c
libavcodec/arm/dsputil_vfp.S
libavcodec/dsputil.c
libavcodec/ppc/float_altivec.c
libavcodec/x86/dsputil.asm
libavutil/x86/float_dsp.asm
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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Now, nellymoserenc and aacenc no longer depends on dsputil. Independent
of this patch, wmaprodec also does not depend on dsputil, so I removed
it from there also.
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* qatar/master:
float_dsp: ppc: add a separate header for Altivec function prototypes
ARM: fix float_dsp breakage from d5a7229
Add a float DSP framework to libavutil
PPC: Move types_altivec.h and util_altivec.h from libavcodec to libavutil
ARM: Move asm.S from libavcodec to libavutil
vc1dsp: mark put/avg_vc1_mspel_mc() always_inline
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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Move vector_fmul() from DSPContext to AVFloatDSPContext.
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* qatar/master: (26 commits)
adxenc: use AVCodec.encode2()
adxenc: Use the AVFrame in ADXContext for coded_frame
indeo4: fix out-of-bounds function call.
configure: Restructure help output.
configure: Internal-only components should not be command-line selectable.
vorbisenc: use AVCodec.encode2()
libvorbis: use AVCodec.encode2()
libopencore-amrnbenc: use AVCodec.encode2()
ra144enc: use AVCodec.encode2()
nellymoserenc: use AVCodec.encode2()
roqaudioenc: use AVCodec.encode2()
libspeex: use AVCodec.encode2()
libvo_amrwbenc: use AVCodec.encode2()
libvo_aacenc: use AVCodec.encode2()
wmaenc: use AVCodec.encode2()
mpegaudioenc: use AVCodec.encode2()
libmp3lame: use AVCodec.encode2()
libgsmenc: use AVCodec.encode2()
libfaac: use AVCodec.encode2()
g726enc: use AVCodec.encode2()
...
Conflicts:
configure
libavcodec/Makefile
libavcodec/ac3enc.c
libavcodec/adxenc.c
libavcodec/libgsm.c
libavcodec/libvorbis.c
libavcodec/vorbisenc.c
libavcodec/wmaenc.c
tests/ref/acodec/g722
tests/ref/lavf/asf
tests/ref/lavf/ffm
tests/ref/lavf/mkv
tests/ref/lavf/mpg
tests/ref/lavf/rm
tests/ref/lavf/ts
tests/ref/seek/lavf_asf
tests/ref/seek/lavf_ffm
tests/ref/seek/lavf_mkv
tests/ref/seek/lavf_mpg
tests/ref/seek/lavf_rm
tests/ref/seek/lavf_ts
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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* qatar/master:
Remove ffmpeg.
aacenc: Simplify windowing
aacenc: Move saved overlap samples to the beginning of the same buffer as incoming samples.
aacenc: Deinterleave input samples before processing.
aacenc: Store channel count in AACEncContext.
aacenc: Move Q^3/4 calculation to it's own table
aacenc: Request normalized float samples instead of converting s16 samples to float.
aacpsy: Replace an if with FFMAX in LAME windowing.
aacenc: cosmetics, replace 'rd' with 'bits' in codebook_trellis_rate to make it more clear what is being calculated.
aacpsy: cosmetics, change a FIXME to a NOTE about subshort comparisons
aacenc: cosmetics: move init() and end() to the bottom of the file.
aacenc: aac_encode_init() cleanup
XWD encoder and decoder
vc1: don't read the interpfrm and bfraction elements for interlaced frames
mxfdec: fix memleak on mxf_read_close()
westwood: split the AUD and VQA demuxers into separate files.
Conflicts:
.gitignore
Changelog
Makefile
configure
doc/ffmpeg.texi
ffmpeg.c
libavcodec/Makefile
libavcodec/aacenc.c
libavcodec/allcodecs.c
libavcodec/avcodec.h
libavcodec/version.h
libavformat/Makefile
libavformat/img2.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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Signed-off-by: Alex Converse <alex.converse@gmail.com>
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Signed-off-by: Alex Converse <alex.converse@gmail.com>
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This should be moved to tablegen at some point.
Signed-off-by: Alex Converse <alex.converse@gmail.com>
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to float.
Signed-off-by: Alex Converse <alex.converse@gmail.com>
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Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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* qatar/master:
cosmetics: fix some then/than typos
doxygen: Include libavcodec and libavformat examples into the documentation
avutil: elaborate documentation for av_get_random_seed
Add support for aac streams in mp4/mov without extradata.
aes: whitespace cosmetics
adler32: whitespace cosmetics
swscale: fix another yuv range conversion overflow in 16bit scaling.
Fix cpu flags test program
opt-test: Add missing braces to silence compiler warnings.
build: Eliminate obsolete test targets.
udp: Fix a compilation warning
swscale: Unbreak build with --enable-small
base64: add fate test
aes: improve test program and add fate test
adler32: make test program more useful and add fate test
swscale: fix yuv range correction when using 16-bit scaling.
aacenc: Make chan_map const correct
Conflicts:
Makefile
doc/examples/muxing-example.c
libavformat/udp.c
libavutil/random_seed.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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* qatar/master: (21 commits)
swscale: Add Doxygen for hyscale_fast/hScale.
fate: enable lavfi-pixmt tests on big endian systems
PPC: swscale: disable altivec functions for unsupported formats
fate: merge identical pixdesc_be/le tests
swscale: Add Doxygen for yuv2planar*/yuv2packed* functions.
build: call texi2pod.pl with full path instead of symlink
build: include sub-makefiles using full path instead of symlinks
swscale: update big endian reference values after dff5a835.
wavpack: skip blocks with no samples
cosmetics: remove outdated comment that is no longer true
build: replace some addprefix/addsuffix with substitution refs
avutil: Remove unused arbitrary precision integer code.
configure: Drop check for availability of ten assembler operands.
aacenc: Save channel configuration for later use.
aacenc: Fix codebook trellising for zeroed bands.
swscale: change prototypes of scaled YUV output functions.
swscale: re-add support for non-native endianness.
swscale: disentangle yuv2rgbX_c_full() into small functions.
swscale: split yuv2packed[12X]_c() remainders into small functions.
swscale: split yuv2packedX_altivec in smaller functions.
...
Conflicts:
Makefile
configure
libavcodec/x86/dsputil_mmx.c
libavfilter/Makefile
libavformat/Makefile
libavutil/integer.c
libavutil/integer.h
libswscale/swscale.c
libswscale/swscale_internal.h
libswscale/x86/swscale_template.c
tests/ref/lavfi/pixdesc_le
tests/ref/lavfi/pixfmts_scale
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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* qatar/master: (25 commits)
Replace custom DEBUG preprocessor trickery by the standard one.
vorbis: Remove non-compiling debug statement.
vorbis: Remove pointless DEBUG #ifdef around debug output macros.
cook: Remove non-compiling debug output.
Remove pointless #ifdefs around function declarations in a header.
Replace #ifdef + av_log() combinations by av_dlog().
Replace custom debug output functions by av_dlog().
cook: Remove unused debug functions.
Remove stray extra arguments from av_dlog() invocations.
targa: fix big-endian build
v4l2: remove one forgotten use of AVFormatParameters.pix_fmt.
vfwcap: add a framerate private option.
v4l2: add a framerate private option.
libdc1394: add a framerate private option.
fbdev: add a framerate private option.
bktr: add a framerate private option.
oma: check avio_read() return value
nutdec: remove unused variable
Remove unused variables
swscale: allocate larger buffer to handle altivec overreads.
...
Conflicts:
ffmpeg.c
libavcodec/dca.c
libavcodec/dirac.c
libavcodec/error_resilience.c
libavcodec/h264.c
libavcodec/mpeg12.c
libavcodec/mpeg4videodec.c
libavcodec/mpegvideo.c
libavcodec/mpegvideo_enc.c
libavcodec/pthread.c
libavcodec/rv10.c
libavcodec/s302m.c
libavcodec/shorten.c
libavcodec/truemotion2.c
libavcodec/utils.c
libavdevice/dv1394.c
libavdevice/fbdev.c
libavdevice/libdc1394.c
libavdevice/v4l2.c
libavformat/4xm.c
libavformat/apetag.c
libavformat/asfdec.c
libavformat/avidec.c
libavformat/mmf.c
libavformat/mpeg.c
libavformat/mpegenc.c
libavformat/mpegts.c
libavformat/oggdec.c
libavformat/oggparseogm.c
libavformat/rl2.c
libavformat/rmdec.c
libavformat/rpl.c
libavformat/rtpdec_latm.c
libavformat/sauce.c
libavformat/sol.c
libswscale/utils.c
tests/ref/vsynth1/error
tests/ref/vsynth2/error
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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ms_off is the default, until Mid/Side is no longer buggy.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
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* qatar/master: (23 commits)
ac3enc: correct the flipped sign in the ac3_fixed encoder
Eliminate pointless '#if 1' statements without matching '#else'.
Add AVX FFT implementation.
Increase alignment of av_malloc() as needed by AVX ASM.
Update x86inc.asm from x264 to allow AVX emulation using SSE and MMX.
mjpeg: Detect overreads in mjpeg_decode_scan() and error out.
documentation: extend documentation for ffmpeg -aspect option
APIChanges: update commit hashes for recent additions.
lavc: deprecate FF_*_TYPE macros in favor of AV_PICTURE_TYPE_* enums
aac: add headers needed for log2f()
lavc: remove FF_API_MB_Q cruft
lavc: remove FF_API_RATE_EMU cruft
lavc: remove FF_API_HURRY_UP cruft
pad: make the filter parametric
vsrc_movie: add key_frame and pict_type.
vsrc_movie: fix leak in request_frame()
lavfi: add key_frame and pict_type to AVFilterBufferRefVideo.
vsrc_buffer: add sample_aspect_ratio fields to arguments.
lavfi: add fieldorder filter
scale: make the filter parametric
...
Conflicts:
Changelog
doc/filters.texi
ffmpeg.c
libavcodec/ac3dec.h
libavcodec/dsputil.c
libavfilter/avfilter.h
libavfilter/vf_scale.c
libavfilter/vf_yadif.c
libavfilter/vsrc_buffer.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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Signed-off-by: Reinhard Tartler <siretart@tauware.de>
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Signed-off-by: Mans Rullgard <mans@mansr.com>
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This reverts the removal of scoefs from AACEncContext.
It resulted in scoefs being a NULL pointer when
search_for_quantizers() is called.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
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This reverts the removal of scoefs from AACEncContext.
It resulted in scoefs being a NULL pointer when
search_for_quantizers() is called.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 2790d7a9ffbd51f33e5367a31ace5c44c30401a1)
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Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
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This requires us to code small escapes; we can't avoid it.
Originally committed as revision 23135 to svn://svn.ffmpeg.org/ffmpeg/trunk
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These macros are redundant. All uses are replaced with the generic
DECLARE_ALIGNED macro instead.
Originally committed as revision 22233 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 21377 to svn://svn.ffmpeg.org/ffmpeg/trunk
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