check AVCodecContext->sample_fmt against AVCodec->sample_fmts[] to ensure
that the encoder supports the specified sample format. Error out if it doesn't.
Previously, it would continue and output garbage. Fixes issue 2587.
(cherry picked from commit 2cfa2d9258)
Decode times for big_buck_bunny_720p_stereo:
1 thread:
real 1m14.227s
user 1m13.104s
sys 0m1.108s
2 threads: (33% faster)
real 0m49.329s
user 1m33.735s
sys 0m1.834s
3 threads: (44% faster)
real 0m41.593s
user 1m44.884s
sys 0m1.967s
(cherry picked from commit d23845f311)
As a side effect of the last commit, avcodec_open() now calls it automatically,
so there is no longer any need for clients to call it.
Instead they should set AVCodecContext.thread_count.
avcodec_thread_free() is deprecated, and will be removed from avcodec.h at the
next MAJOR libavcodec bump.
Rename the functions to ff_thread_init/free, since they are now internal.
Wrappers are provided to maintain API compatibility.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit c0b102ca03)
s->windowed_samples will always have a range of [-32767,32767] due to the
window function, so the return value from log2_tab() will always be in the
range [0,14].
Signed-off-by: Mans Rullgard <mans@mansr.com>
Instead of returning an error when bytes are left over, just return
the number of actually used bytes as other decoders do.
Instead add a special case so an error will be returned when none
of the data looks valid to avoid making debugging a pain.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
The avcodec_thread_free() compatibility wrapper calls ff_thread_free(),
which is not defined when threading is disabled. Make this call
conditional.
Signed-off-by: Mans Rullgard <mans@mansr.com>
check AVCodecContext->sample_fmt against AVCodec->sample_fmts[] to ensure
that the encoder supports the specified sample format. Error out if it doesn't.
Previously, it would continue and output garbage. Fixes issue 2587.
Decode times for big_buck_bunny_720p_stereo:
1 thread:
real 1m14.227s
user 1m13.104s
sys 0m1.108s
2 threads: (33% faster)
real 0m49.329s
user 1m33.735s
sys 0m1.834s
3 threads: (44% faster)
real 0m41.593s
user 1m44.884s
sys 0m1.967s
As a side effect of the last commit, avcodec_open() now calls it automatically,
so there is no longer any need for clients to call it.
Instead they should set AVCodecContext.thread_count.
avcodec_thread_free() is deprecated, and will be removed from avcodec.h at the
next MAJOR libavcodec bump.
Rename the functions to ff_thread_init/free, since they are now internal.
Wrappers are provided to maintain API compatibility.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Also allow qmin/qmax to go up to 69 (the current max value for libx264). This
will have to increase when we add 9/10-bit support.
(cherry picked from commit c7ac200d15)
Due to being pants-on-head retarded, libavcodec defaults this to zero, which
results in broken output. This didn't affect ffmpeg.c, which sets it itself,
but caused problems for other calling apps using VBV.
(cherry picked from commit f7f8120fb9)
Fix emu_edge_v_extend_15 to be <128 bytes on Win64, by being more strict
on the size of registers and which registers are being used for operations
where multiple are available. This fixes segfaults in emulated_edge()
function calls on Win64.
(cherry picked from commit 17cf7c68ed)
In all 3 cases, the decoding continues and thus a warning would be sufficient.
Helps application that catch them with own log handers to handle them
accordingly.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit ae2104791f)
This adds NEON optimised versions of all functions in VP8DSPContext.
Based on initial work by Rob Clark.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit a1c1d3c003)
Makes playing QDMC files in MPlayer work when using the libavformat demuxer.
Problem was that the extradata was not passed from demuxer to decoder.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit ed19fafd48)
Due to being pants-on-head retarded, libavcodec defaults this to zero, which
results in broken output. This didn't affect ffmpeg.c, which sets it itself,
but caused problems for other calling apps using VBV.
Fix emu_edge_v_extend_15 to be <128 bytes on Win64, by being more strict
on the size of registers and which registers are being used for operations
where multiple are available. This fixes segfaults in emulated_edge()
function calls on Win64.
In all 3 cases, the decoding continues and thus a warning would be sufficient.
Helps application that catch them with own log handers to handle them
accordingly.
Signed-off-by: Mans Rullgard <mans@mansr.com>
In some places, dvbsubdec passes improper input buffer size to
bitstream reading functions, not accounting for reading pointer
updates.
Fixed by using buffer_end - buffer pointer instead of fixed buffer length.
Signed-off-by: Jindrich Makovicka <makovick@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
In some places, dvbsubdec passes improper input buffer size to
bitstream reading functions, not accounting for reading pointer
updates.
Fixed by using buffer_end - buffer pointer instead of fixed buffer length.
Signed-off-by: Jindrich Makovicka <makovick@gmail.com>
Signed-off-by: Mans Rullgard <mans@mansr.com>
Makes playing QDMC files in MPlayer work when using the libavformat demuxer.
Problem was that the extradata was not passed from demuxer to decoder.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
This was missed when pkt_pts was first added.
Signed-off-by: Nicolas George <nicolas.george@normalesup.org>
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 62ecd3635a)
This patch changes the exponent difference threshold in the exponent
strategy decision function of the AC-3 encoder. I tested lowering in
increments of 100. From 1000 down to 500 generally increased in quality
with each step, but 400 was generally much worse.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit c3beafa0f1)
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)
Gcc 4.6 only preserves the first value when using an array with an "m"
constraint.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 770c410fbb)
This was missed when pkt_pts was first added.
Signed-off-by: Nicolas George <nicolas.george@normalesup.org>
Signed-off-by: Mans Rullgard <mans@mansr.com>
This patch changes the exponent difference threshold in the exponent
strategy decision function of the AC-3 encoder. I tested lowering in
increments of 100. From 1000 down to 500 generally increased in quality
with each step, but 400 was generally much worse.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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>
Adds some duplicated code, but avoids duplicate edge checks and similar.
~0.5% faster overall on Parkjoy test sample.
(cherry picked from commit 64233e702a)
This moves the fields needed by asm near the top, before any
structs or other members which complicate the offset calculation.
Modifying other structs will no longer require updating the offsets,
and the asm code is slightly simpler due to the smaller offsets.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit d461a47317)
From ~780 cycles to 551 cycles, mostly just by using libc memcpy()
instead of manually shuffling individual bytes around.
(cherry picked from commit e5262ec44a)
This significantly reduces the size of the symbol table in the generated ELF
shared object (as well as the other linked tables).
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit ac28ce5fac)
This moves the fields needed by asm near the top, before any
structs or other members which complicate the offset calculation.
Modifying other structs will no longer require updating the offsets,
and the asm code is slightly simpler due to the smaller offsets.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Fix C VP8 H+V MC functions which do two-dimensional 4/6-tap filters to
not overread beyond their edges if the second filter is 4-tap, since
the outer pixels aren't there anymore since
44002d8323.
(cherry picked from commit 22893e10ae)
None of these symbols should be accessed directly, so declare them as
hidden.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit d36beb3f69)
The iff.h header only declared one function that is now static, the
libavformat/iff.c source file wasn't using it before. Drop the file
entirely.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 2d162e3825)
It is only used to generate band_start_tab, which about the same size, at
runtime, so it's simpler just to always hardcode band_start_tab.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 24e3ad3031)
This significantly reduces the size of the symbol table in the generated ELF
shared object (as well as the other linked tables).
Signed-off-by: Mans Rullgard <mans@mansr.com>
Fix C VP8 H+V MC functions which do two-dimensional 4/6-tap filters to
not overread beyond their edges if the second filter is 4-tap, since
the outer pixels aren't there anymore since
44002d8323.
The iff.h header only declared one function that is now static, the
libavformat/iff.c source file wasn't using it before. Drop the file
entirely.
Signed-off-by: Mans Rullgard <mans@mansr.com>
It is only used to generate band_start_tab, which about the same size, at
runtime, so it's simpler just to always hardcode band_start_tab.
Signed-off-by: Mans Rullgard <mans@mansr.com>
The header is empty after making the function static, so delete it and
drop its usage.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
(cherry picked from commit 13eb6b9097)
Both functions seem to be commanded by the ff_spatial_idwt function
instead.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
(cherry picked from commit ebb06d96ed)
Do not emulate larger edges than we will actually use for this round of
MC. Decoding goes from avg+SE 29.972+/-0.023sec to 29.856+/-0.023, i.e.
0.12sec or ~0.4% faster.
(cherry picked from commit 44002d8323)
This symbol is only ever used to calculate the non-hardcoded tables, so
only enable it in that case, and static to the source unit that uses it.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 6ed3b504f9)
1d4da6a460 added static to the
prototypes for these fuctions. Adding it to the definitions
as well.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit aa61e39eac)
The PCM_DVD encoder would be left unused, as allcodecs.c properly declared
it as being decoder-only, but it would still be built into the object file.
Since there is no block of code to properly encode this PCM format, it's
not a full codec.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 5b5083b5fe)
The dprintf macro is no-op when DEBUG is unset, so there is no need to
put it conditional to DEBUG.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 73a0b19ba3)
This ensures a locally-unique name as well as marks the symbol as
FFmpeg-private at least by declaration.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 82e1f217f2)
Do not emulate larger edges than we will actually use for this round of
MC. Decoding goes from avg+SE 29.972+/-0.023sec to 29.856+/-0.023, i.e.
0.12sec or ~0.4% faster.
This symbol is only ever used to calculate the non-hardcoded tables, so
only enable it in that case, and static to the source unit that uses it.
Signed-off-by: Mans Rullgard <mans@mansr.com>
The PCM_DVD encoder would be left unused, as allcodecs.c properly declared
it as being decoder-only, but it would still be built into the object file.
Since there is no block of code to properly encode this PCM format, it's
not a full codec.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This ensures a locally-unique name as well as marks the symbol as
FFmpeg-private at least by declaration.
Signed-off-by: Mans Rullgard <mans@mansr.com>
With the removal of the libmpeg2 bitstream reader, MIN_CACHE_BITS
is always >= 25, so tests against smaller values can be removed.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit f162e988aa)
These whitespace changes improve the readability of the get_bits
macros.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit fb5c841d5f)
Some of the macros in get_bits.h include a final semicolon,
some do not. This removes these or adds do {} while(0) around
the macros as appropriate and adds semicolons where needed in
calling code.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit bf5f9b528b)
Using the libmpeg2 reader causes errors in a multitude of places,
including MPEG and H264 codecs. As the advantage of this reader
is questionable, removing it seems the sensible course of action,
especially considering the simplifications this allows elsewhere
with the bit cache size increasing from 17 to 25 bits as minimum.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 938f72e199)
Don't free RBSP tables (containing decoded NAL units) on resolution
change, because we actually need this data to decode the frame after
reiniting (with new resolution). Fixed issue 2393.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
(cherry picked from commit 9107892624)
C99 variadic macros require more arguments than there are named
parameters in the definition. This means we must use an extra
indirection to avoid having two different macros for arrays with
one resp more than one dimension.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 96aad41e81)
It's incomplete, no one is working on it, and when someone asks about
working on it we advise them not to.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit ff3d43104f)
Use backwards compatible explicit signalling to denote the absence of
SBR.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
(cherry picked from commit 8ae0fa243e)
I did not notice that the filter implementation uses a reversed history state.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
(cherry picked from commit 98cfadd648)
DTS-HD HRA streams do not always have an XBR extension in the extension
substream. Instead they can have only XXCh and X96 extensions in
there and still be considered DTS-HD HRA.
This is also confirmed with Onkyo TX-SR607 receiver which recognizes
such a stream as HiRes Audio.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 8f4a5d225c)
With the removal of the libmpeg2 bitstream reader, MIN_CACHE_BITS
is always >= 25, so tests against smaller values can be removed.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Some of the macros in get_bits.h include a final semicolon,
some do not. This removes these or adds do {} while(0) around
the macros as appropriate and adds semicolons where needed in
calling code.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Using the libmpeg2 reader causes errors in a multitude of places,
including MPEG and H264 codecs. As the advantage of this reader
is questionable, removing it seems the sensible course of action,
especially considering the simplifications this allows elsewhere
with the bit cache size increasing from 17 to 25 bits as minimum.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Don't free RBSP tables (containing decoded NAL units) on resolution
change, because we actually need this data to decode the frame after
reiniting (with new resolution). Fixed issue 2393.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
C99 variadic macros require more arguments than there are named
parameters in the definition. This means we must use an extra
indirection to avoid having two different macros for arrays with
one resp more than one dimension.
Signed-off-by: Mans Rullgard <mans@mansr.com>
It's incomplete, no one is working on it, and when someone asks about
working on it we advise them not to.
Signed-off-by: Mans Rullgard <mans@mansr.com>
DTS-HD HRA streams do not always have an XBR extension in the extension
substream. Instead they can have only XXCh and X96 extensions in
there and still be considered DTS-HD HRA.
This is also confirmed with Onkyo TX-SR607 receiver which recognizes
such a stream as HiRes Audio.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Simplifies error handling and makes it easier to add additional filter types.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 0361d13cf3)
The original functions did not work correctly for edge pixels, e.g.
when CODEC_FLAG_EMU_EDGE is set, leading to corrupt output in e.g. VLC.
Based on a patch by Daniel Kang <daniel d kang gmail com>.
Signed-off-by: Ronald S. Bultje <rsbultje gmail com>
(cherry picked from commit b9c7f66e6d)
The original functions did not work correctly for edge pixels, e.g.
when CODEC_FLAG_EMU_EDGE is set, leading to corrupt output in e.g. VLC.
Based on a patch by Daniel Kang <daniel d kang gmail com>.
Signed-off-by: Ronald S. Bultje <rsbultje gmail com>
Improves CABAC performance about ~1.2%.
Trick originates from x264 and has also been used in ffvp8. It's useful because
coded block flags are usually zero, so it helps to have the early termination
inlined into the main function.
Originally committed as revision 26375 to svn://svn.ffmpeg.org/ffmpeg/trunk
The hunk is not fully understood but it just makes a check tighter so its
safer for us to apply until it is fully understood.
Might fix issue 2550 (and Chrome issue 68115 and unknown CERT issues).
Our bugtracker issue though should stay open until this has been fully
investiagted
Patch by Frank Barchard, fbarchard at google
Originally committed as revision 26368 to svn://svn.ffmpeg.org/ffmpeg/trunk
Fixes issue 2548 (and Chrome issue 68115 and unknown CERT issues).
Patch by Frank Barchard, fbarchard at google
Originally committed as revision 26365 to svn://svn.ffmpeg.org/ffmpeg/trunk
instead of real width, this fixes decoding of some Bink files with odd width.
Originally committed as revision 26364 to svn://svn.ffmpeg.org/ffmpeg/trunk
color value instead of always taking 0 (resulting in green frames).
Fixes issue issue2531.
Originally committed as revision 26363 to svn://svn.ffmpeg.org/ffmpeg/trunk
exponent strategies for a single channel to compute_exp_strategy_ch().
This allows for removal of the temporary pointer arrays.
Originally committed as revision 26356 to svn://svn.ffmpeg.org/ffmpeg/trunk
No speed improvement, but necessary for some future stuff.
Also opens up the possibility of asm chroma dc idct/dequant.
Originally committed as revision 26349 to svn://svn.ffmpeg.org/ffmpeg/trunk
Doesn't help speed as there isn't an asm implementation yet, but consistency
is a good thing.
Originally committed as revision 26348 to svn://svn.ffmpeg.org/ffmpeg/trunk
Since we no longer have non-transposed scantables, the problem it warns about
no longer exists.
Originally committed as revision 26339 to svn://svn.ffmpeg.org/ffmpeg/trunk
Useful so that we don't have to run the hierarchical DC iDCT if there aren't
any coefficients. Opens up some future opportunities for optimization as well.
Originally committed as revision 26337 to svn://svn.ffmpeg.org/ffmpeg/trunk
About 2.5x the speed.
NOTE: the way that the asm code handles large qmuls is a bit suboptimal.
If x264-style dequant was used (separate shift and qmul values), it might
be possible to get some extra speed.
Originally committed as revision 26336 to svn://svn.ffmpeg.org/ffmpeg/trunk
It was an ugly hack to begin with and didn't give any performance.
NOTE: this patch opens up some future simplifications to be made (such as
removing some of the scantables from H264Context) but doesn't take advantage
of them yet.
Originally committed as revision 26329 to svn://svn.ffmpeg.org/ffmpeg/trunk
This improves the audio quality significantly for stereo source with both the
fixed-point and floating-point AC-3 encoders.
Update acodec-ac3_fixed and seek-ac3_rm test references.
Originally committed as revision 26271 to svn://svn.ffmpeg.org/ffmpeg/trunk
for invalid header up before reading data.
Fixes issue 2500.
Patch by Daniel Kang, daniel.d.kang at gmail
Originally committed as revision 26248 to svn://svn.ffmpeg.org/ffmpeg/trunk
instead of doing it separately in 2 different functions.
This makes float AC-3 encoding approx. 3-7% faster overall.
Also, the coefficient conversion can now be easily SIMD-optimized.
Originally committed as revision 26232 to svn://svn.ffmpeg.org/ffmpeg/trunk
accessing of structs and arrays inside the loop.
Approx. 30% faster in function extract_exponents().
Originally committed as revision 26226 to svn://svn.ffmpeg.org/ffmpeg/trunk
Fixed-point AC-3 encoder renamed to ac3_fixed.
Regression test acodec-ac3 renamed to acodec-ac3_fixed.
Regression test lavf-rm changed to use ac3_fixed encoder.
Originally committed as revision 26209 to svn://svn.ffmpeg.org/ffmpeg/trunk
maximum value of 1023.
This speeds up overall encoding depending on the content and bitrate.
The most improvement is with high bitrates and/or low complexity content.
Originally committed as revision 26181 to svn://svn.ffmpeg.org/ffmpeg/trunk
instead of 64. This will change output in some cases, but it happens to not
affect the AC-3 regression tests.
Originally committed as revision 26180 to svn://svn.ffmpeg.org/ffmpeg/trunk
Jason, Loren, Holger) to FFmpeg. Patch by Daniel Kang <daniel dot d dot kang
at gmail com>, as part of Google's GCI 2010.
Originally committed as revision 26162 to svn://svn.ffmpeg.org/ffmpeg/trunk
Jason, Loren, Holger) to FFmpeg. Patch by Daniel Kang <daniel dot d dot kang
at gmail com>, as part of Google's GCI 2010.
Originally committed as revision 26159 to svn://svn.ffmpeg.org/ffmpeg/trunk
Jason, Loren, Holger) to FFmpeg. Patch by Daniel Kang <daniel dot d dot kang
at gmail com>, as part of Google's GCI 2010.
Originally committed as revision 26158 to svn://svn.ffmpeg.org/ffmpeg/trunk
(authors:Jason, Loren, Holger) to FFmpeg. Patch by Daniel Kang <daniel dot
d dot kang at gmail com>, as part of Google's GCI 2010.
Originally committed as revision 26157 to svn://svn.ffmpeg.org/ffmpeg/trunk
Jason, Loren, Holger) to FFmpeg. Patch by Daniel Kang <daniel dot d dot kang
at gmail com>, as part of Google's GCI 2010.
Originally committed as revision 26156 to svn://svn.ffmpeg.org/ffmpeg/trunk
Jason, Loren, Holger) to FFmpeg. Patch by Daniel Kang <daniel dot d dot kang
at gmail com>, as part of Google's GCI 2010.
Originally committed as revision 26155 to svn://svn.ffmpeg.org/ffmpeg/trunk
(authors: Jason, Loren, Holger) to FFmpeg. Patch by Daniel Kang <daniel dot
d dot kang at gmail com>, as part of Google's GCI 2010.
Originally committed as revision 26151 to svn://svn.ffmpeg.org/ffmpeg/trunk
(authors: Jason, Loren, Holger) to FFmpeg. Patch by Daniel Kang <daniel dot
d dot kang at gmail com>, as part of Google's GCI 2010.
Originally committed as revision 26150 to svn://svn.ffmpeg.org/ffmpeg/trunk
(authors: Jason, Loren, Holger) to FFmpeg. Patch by Daniel Kang <daniel dot
d dot kang at gmail com>, as part of Google's GCI 2010.
Originally committed as revision 26149 to svn://svn.ffmpeg.org/ffmpeg/trunk
(authors: Jason, Loren, Holger) to FFmpeg. Patch by Daniel Kang <daniel dot
d dot kang at gmail com>, as part of Google's GCI 2010.
Originally committed as revision 26148 to svn://svn.ffmpeg.org/ffmpeg/trunk
(authors: Jason, Loren, Holger) to FFmpeg. Patch by Daniel Kang <daniel dot
d dot kang at gmail com>, as part of Google's GCI 2010.
Originally committed as revision 26147 to svn://svn.ffmpeg.org/ffmpeg/trunk
(authors: Jason, Loren, Holger) to FFmpeg. Patch by Daniel Kang <daniel dot
d dot kang at gmail com>, as part of Google's GCI 2010.
Originally committed as revision 26146 to svn://svn.ffmpeg.org/ffmpeg/trunk
(authors: Jason, Loren, Holger) to FFmpeg. Patch by Daniel Kang <daniel dot
d dot kang at gmail com>, as part of Google's GCI 2010.
Originally committed as revision 26145 to svn://svn.ffmpeg.org/ffmpeg/trunk
This gives slightly better quality in PEAQ tests.
Code 3 gives a dBpb value of 2816 = -132dB (128 psd units = -6dB), which
corresponds to 22 bits. Since the exponents have an offset applied, the
16-bit source looks like 24-bit source to the bit allocation routine.
So using dBpb code=3 is a closer match to the exponent range.
Regression test refs updated for acodec-ac3, lavf-rm, and seek-ac3_rm.
Originally committed as revision 26144 to svn://svn.ffmpeg.org/ffmpeg/trunk
Jason, Loren, Holger) to FFmpeg. Patch by Daniel Kang <daniel dot d dot kang
at gmail com>, as part of Google's GCI 2010.
Originally committed as revision 26143 to svn://svn.ffmpeg.org/ffmpeg/trunk
Jason, Loren, Holger) to FFmpeg. Patch by Daniel Kang <daniel dot d dot kang at
gmail com>, as part of Google's GCI 2010.
Originally committed as revision 26142 to svn://svn.ffmpeg.org/ffmpeg/trunk
FFmpeg. Original authors: Holger Lubitz <holger lubitz org>, Jason Garrett-
Glaser <darkshikari gmail com> (approves LGPL relicensing for this code) and
Loren Merritt <lorenm at u dot washington dot edu> (approves LGPL relicensing
for this code). Patch by Daniel Kang <daniel dot d dot kang at gmail com>, as
part of Google's GCI 2010.
Originally committed as revision 26140 to svn://svn.ffmpeg.org/ffmpeg/trunk
FFmpeg. Original authors: Holger Lubitz <holger lubitz org>, Jason Garrett-
Glaser <darkshikari gmail com> (approves LGPL relicensing for this code) and
Loren Merritt <lorenm at u dot washington dot edu> (approves LGPL relicensing
for this code). Patch by Daniel Kang <daniel dot d dot kang at gmail com>, as
part of Google's GCI 2010.
Originally committed as revision 26139 to svn://svn.ffmpeg.org/ffmpeg/trunk
Original authors: Holger Lubitz <holger lubitz org>, Jason Garrett-Glaser
<darkshikari gmail com> (approves LGPL relicensing for this code) and Loren
Merritt <lorenm at u dot washington dot edu> (approves LGPL relicensing for
this code). Patch by Daniel Kang <daniel dot d dot kang at gmail com>, as
part of Google's GCI 2010.
Originally committed as revision 26138 to svn://svn.ffmpeg.org/ffmpeg/trunk
Original authors: Holger Lubitz <holger lubitz org>, Jason Garrett-Glaser
<darkshikari gmail com> (approves LGPL relicensing for this code) and Loren
Merritt <lorenm at u dot washington dot edu> (approves LGPL relicensing for
this code). Patch by Daniel Kang <daniel dot d dot kang at gmail com>, as
part of Google's GCI 2010.
Originally committed as revision 26137 to svn://svn.ffmpeg.org/ffmpeg/trunk
authors: Holger Lubitz <holger lubitz org>, Jason Garrett-Glaser <darkshikari
gmail com> (approves LGPL relicensing for this code) and Loren Merritt <lorenm
at u dot washington dot edu> (approves LGPL relicensing for this code). Patch
by Daniel Kang <daniel dot d dot kang at gmail com>, as part of Google's GCI
2010.
Originally committed as revision 26135 to svn://svn.ffmpeg.org/ffmpeg/trunk
Original authors: Holger Lubitz <holger lubitz org>, Jason Garrett-Glaser
<darkshikari gmail com> (approves LGPL relicensing for this code) and Loren
Merritt <lorenm at u dot washington dot edu> (approves LGPL relicensing for
this code). Patch by Daniel Kang <daniel dot d dot kang at gmail com>, as
part of Google's GCI 2010.
Originally committed as revision 26132 to svn://svn.ffmpeg.org/ffmpeg/trunk
initially said he'd be OK with relicensing, he also said he wanted to have
another look at the patch, and then he went on vacation, so let's play it
safe for now. We can consider removing this again later.
Originally committed as revision 26131 to svn://svn.ffmpeg.org/ffmpeg/trunk
LGPL relicensing approved by original authors: Holger Lubitz <holger lubitz
org>, Jason Garrett-Glaser <darkshikari gmail com> and Loren Merritt <lorenm
at u dot washington dot edu>. Patch by Daniel Kang <daniel dot d dot kang at
gmail com>, as part of Google's GCI 2010.
Originally committed as revision 26087 to svn://svn.ffmpeg.org/ffmpeg/trunk
This is optional for encoders, but it's a good idea and has minimal impact
on performance.
This will change the output for some files, but it happens not to affect the
regression tests.
Originally committed as revision 26083 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes compilation with --disable-everything --enable-<component>,
for all encoders, decoders, muxers, demuxers, parsers, protocols, bsfs,
indevs, outdevs and filters at the moment. (All those that work without
any external dependencies at least.)
Originally committed as revision 26076 to svn://svn.ffmpeg.org/ffmpeg/trunk
This avoids a 16-bit overflow in mdct512() due to a -32768 value in costab.
References updated for acodec-ac3, lavf-rm, and seek-ac3_rm tests.
Thanks to Måns Rullgård for finding the bug.
Originally committed as revision 26071 to svn://svn.ffmpeg.org/ffmpeg/trunk
Fixes AC-3 encoding on OpenBSD 4.8 x86_32 and hopefully other similar
configurations.
Originally committed as revision 26070 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes building with --disable-everything --enable-muxer=matroska and/or
--enable-muxer=webm
Originally committed as revision 26067 to svn://svn.ffmpeg.org/ffmpeg/trunk
Since SVN rev 25866, this table is used by the trellis encoder, too,
not only by the decoder.
Originally committed as revision 26065 to svn://svn.ffmpeg.org/ffmpeg/trunk
Galvão Póvoa <marspeoplester gmail com>, mentored by Robert Swain <robert
dot swain gmail com>.
Originally committed as revision 26051 to svn://svn.ffmpeg.org/ffmpeg/trunk
This allows encoding with lower bitrates by decreasing exponent bits first,
then decreasing bandwidth if the user did not specify a specific cutoff
frequency.
Originally committed as revision 26050 to svn://svn.ffmpeg.org/ffmpeg/trunk
We can do this because exponents are the only bit allocation parameters which
change from block-to-block currently.
Approx. 57% faster in function bit_alloc().
Approx. 25% faster overall encoding.
Originally committed as revision 26040 to svn://svn.ffmpeg.org/ffmpeg/trunk
allocation for each block.
24% faster in function bit_alloc(). Approx. 10% faster overall encoding.
Originally committed as revision 26039 to svn://svn.ffmpeg.org/ffmpeg/trunk
in encode_exponents_blk_ch() by removing the inner loops. This is about 30-40%
faster for the modified sections.
Originally committed as revision 26036 to svn://svn.ffmpeg.org/ffmpeg/trunk
longer required. This gets rid of the temp buffer as well as encoded_exp in
AC3EncodeContext. It also allows for skipping the exponent grouping for
EXP_D15. 56% faster in encode_exponents_blk_ch().
Originally committed as revision 26034 to svn://svn.ffmpeg.org/ffmpeg/trunk
This reduces the memory footprint when using less than 6 channels.
Modify bit allocation to swap the 2 buffers instead of using memcpy() and use
per-block pointers for bap. This is slightly faster (0.3%) in function
cbr_bit_allocation().
Originally committed as revision 26023 to svn://svn.ffmpeg.org/ffmpeg/trunk
Avoids memcpy that was used to store last samples for next frame.
Approx. 3% faster in function deinterleave_input_samples() and reduces memory
usage by 3kB.
Originally committed as revision 26021 to svn://svn.ffmpeg.org/ffmpeg/trunk
svq3 still doesn't support multithreading, but it's simpler for clients if
they can enable threading for all codecs by default.
Originally committed as revision 26015 to svn://svn.ffmpeg.org/ffmpeg/trunk
Th new function only needs to be called at initialization because bit
allocation parameters currently do not change during encoding.
Originally committed as revision 26003 to svn://svn.ffmpeg.org/ffmpeg/trunk
per-channel exponent strategy decision. This will also make it easier to
plug-in other exponent strategy algorithms.
Originally committed as revision 25995 to svn://svn.ffmpeg.org/ffmpeg/trunk
This reduces the amount of memcpy() by using pointers to overlap samples for
successive blocks rather than copying.
Originally committed as revision 25986 to svn://svn.ffmpeg.org/ffmpeg/trunk
Return AVERROR(EINVAL) instead of -1. Propogate errors from called functions.
Add some error log printouts.
Originally committed as revision 25982 to svn://svn.ffmpeg.org/ffmpeg/trunk
This is an av_cold function, and we don't need to duplicate variables just to
save a few characters.
Originally committed as revision 25979 to svn://svn.ffmpeg.org/ffmpeg/trunk
Turn it into 2 macros, and use av_clip_int16() and lrintf().
This matches the int16 to float sample conversion in audioconvert.c.
The regression test output is different due to lrintf() rounding.
Originally committed as revision 25956 to svn://svn.ffmpeg.org/ffmpeg/trunk
(without buffering extra input).
Patch by Michael Chinen [mchinen at gmail]
Originally committed as revision 25932 to svn://svn.ffmpeg.org/ffmpeg/trunk
libavcodec to libavcore.
Remove another compile-time dependancy of libavfilter on libavcodec.
Originally committed as revision 25923 to svn://svn.ffmpeg.org/ffmpeg/trunk
data thanks to the recently added FLAC parser.
Patch by Michael Chinen [mchinen at gmail]
Originally committed as revision 25915 to svn://svn.ffmpeg.org/ffmpeg/trunk
Seek test reference updated because FLAC seeking now works properly.
Fixes roundup issue 1150.
Patch by Michael Chinen [mchinen at gmail]
Originally committed as revision 25914 to svn://svn.ffmpeg.org/ffmpeg/trunk
to optionally silence the error messages.
Patch by Michael Chinen [mchinen at gmail]
Originally committed as revision 25912 to svn://svn.ffmpeg.org/ffmpeg/trunk
than just per-block. Patch by Sprezz [sprezzatura gmx com]. Fixes Issue 2387.
Originally committed as revision 25898 to svn://svn.ffmpeg.org/ffmpeg/trunk
I dont know if this is the best way to handle it. But it fixes http://kuwatan.jp/temp/n-02b.3gp
Fixes issue 2373.
Originally committed as revision 25875 to svn://svn.ffmpeg.org/ffmpeg/trunk
Wraparound in ssd is mainly avoided by subtracting the ssd of the
best node from all the others once it has grown large enough.
If using very large trellis sizes (e.g. -trellis 15), the frontier
is so large that the difference between the best and the worst is
large enough to cause wraparound, even if the ssd of the best one
is subtracted regularly.
When using -trellis 10 on a 30 second sample, this causes only a slight
slowdown, from 61 to 64 seconds.
Originally committed as revision 25858 to svn://svn.ffmpeg.org/ffmpeg/trunk
Patch by Mark Goodman [mark goodman gmail com] with some modifications by me.
Originally committed as revision 25796 to svn://svn.ffmpeg.org/ffmpeg/trunk
The new implementation is more compact, more correct and doesn't hurt
the eyes.
Originally committed as revision 25792 to svn://svn.ffmpeg.org/ffmpeg/trunk
so extend decoder to output only one channel for it.
This fixes issue 2368.
Originally committed as revision 25790 to svn://svn.ffmpeg.org/ffmpeg/trunk
This also allows to remove a linking dependency of libavfilter on
libavcodec.
Originally committed as revision 25789 to svn://svn.ffmpeg.org/ffmpeg/trunk
const char *avcodec_get_channel_name(int channel_id)
which was never implemented.
Originally committed as revision 25788 to svn://svn.ffmpeg.org/ffmpeg/trunk
PaletteControl.
This also fixes playback of some files with ffplay (images were
corrupted for a short time after a palette change).
Originally committed as revision 25778 to svn://svn.ffmpeg.org/ffmpeg/trunk
This makes the wording consistent with how people usually talk about heaps.
Originally committed as revision 25775 to svn://svn.ffmpeg.org/ffmpeg/trunk
This increases the PSNR slightly (about 0.1 dB) for trellis sizes
below 8, and gives equal PSNR for sizes above that.
Originally committed as revision 25769 to svn://svn.ffmpeg.org/ffmpeg/trunk
This lowers the run time from 158 to 21 seconds, for -trellis 8
with a 30 second sample on my machine.
This requires 64 KB additional memory.
Originally committed as revision 25768 to svn://svn.ffmpeg.org/ffmpeg/trunk
beginning of the frame, so make it use skip_bits_long() instead of
skip_bits() for that.
Originally committed as revision 25754 to svn://svn.ffmpeg.org/ffmpeg/trunk
By not looking for the exactly largest node, we avoid an O(n) seek through
the leaf nodes. Just pick one (not the same one every time) and try replacing
that node with the new one.
For -trellis 8, this lowers the run time from 190 to 158 seconds,
for a 30 second 44 kHz mono sample, on my machine.
Originally committed as revision 25733 to svn://svn.ffmpeg.org/ffmpeg/trunk
This avoids having to memmove the large parts of the array when inserting into
it.
For -trellis 8, this lowers the run time from 245 seconds to 190 seconds,
for a 30 second 44 kHz mono sample, on my machine.
Originally committed as revision 25731 to svn://svn.ffmpeg.org/ffmpeg/trunk
An intermediate value in the floor 1 linear interpolation was
overflowing
resulting in obvious artifacts on some files.
e.g.
http://upload.wikimedia.org/wikipedia/commons/7/79/Big_Buck_Bunny_small.ogv
Prior to this fix 87 out of 128 64kbit/s mono files decoded with ffmpeg
have
lower PEAQ ODG values than the same files decoded with libvorbis. With
this
fix none of that set have significantly worse ODG values than libvorbis.
Fixes issue 2352
Patch by Gregory Maxwell <greg@xiph.org>
Originally committed as revision 25724 to svn://svn.ffmpeg.org/ffmpeg/trunk
Nobody ever uses it correctly, and ffmpeg sets it incorrectly, so we'll just
leave it out.
Originally committed as revision 25720 to svn://svn.ffmpeg.org/ffmpeg/trunk
descriptors for printing the number of channels/components.
Also replace the term "nb_channels" with "nb_components" which is more
consistent with the FFmpeg internal terminology, and is somehow
different with respect to the current definition of nb_channels in
PixFmtInfo.
See thread:
Subject: [FFmpeg-devel] [PATCH 6/8] Make avcodec_pix_fmt_string() use the
information in the pixel format descriptors for printing the
number of planes. Also replace the term "nb_channels" with
"nb_planes" which is more correct.
Date: Fri, 5 Nov 2010 12:01:38 +0100
Originally committed as revision 25717 to svn://svn.ffmpeg.org/ffmpeg/trunk
eval API.
More grep-friendly and more consistent with the rest of the FFmpeg
API.
Originally committed as revision 25708 to svn://svn.ffmpeg.org/ffmpeg/trunk
Muxlength does not include the 3 bytes of AudioSyncStream() before the
AudioMuxElement(). If these three bytes are not accounted for the last three
bytes of the LATM packet are sent back to the decoder again.
Fixes issue244/mux2.share.ts
Originally committed as revision 25685 to svn://svn.ffmpeg.org/ffmpeg/trunk
the object number is, it determines whether we should continue
parsing the presentation description and whether we should
clear the subtitles on the next display command.
Based on patch by Mark Goodman [mark goodman gmail com]
Originally committed as revision 25682 to svn://svn.ffmpeg.org/ffmpeg/trunk
Previously it was releasing the buffer which was returned to the user,
which was resulting in a crash in case of direct rendering.
Originally committed as revision 25678 to svn://svn.ffmpeg.org/ffmpeg/trunk
Spotted by Alex after Carl Eugen found errors some samples. There no errors or
noticeable artifacts in the samples I used during development.
Originally committed as revision 25676 to svn://svn.ffmpeg.org/ffmpeg/trunk
Contrary to progressive, just being able to crop up to 14/15 pixels
is not enough to encode all supported resolutions, and the new
behaviour is also consistent with e.g. MPEG-2 etc.
Originally committed as revision 25669 to svn://svn.ffmpeg.org/ffmpeg/trunk
av_get_sample_fmt_name()
av_get_sample_fmt()
av_get_sample_fmt_string()
in libavcore, and deprecate the corresponding libavcodec/audioconvert.h functions:
avcodec_get_sample_fmt_name()
avcodec_get_sample_fmt()
avcodec_sample_fmt_string()
Originally committed as revision 25653 to svn://svn.ffmpeg.org/ffmpeg/trunk
The decoder is just a wrapper around the AAC decoder.
based on patch by Paul Kendall { paul <ät> kcbbs gen nz }
Originally committed as revision 25642 to svn://svn.ffmpeg.org/ffmpeg/trunk
Use avctx in all called functions. This allows passing a NULL AACContext
for LATM since the AACContext is only used in output_configure() which
is skipped for LATM parsing.
Originally committed as revision 25641 to svn://svn.ffmpeg.org/ffmpeg/trunk
aac_decode_frame() remains as AVPacket handling a wrapper. The actual
decoding function takes a GetBitContext as input and will be used be the
AAC LATM decoder to avoid copying the unaligned AAC bitstream.
Originally committed as revision 25640 to svn://svn.ffmpeg.org/ffmpeg/trunk
This will be used by the latm decoder to avoid AACContext changes during
audio specific config parsing.
Originally committed as revision 25638 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes compilation with the latest clang trunk version.
Patch by İsmail Dönmez, ismail at namtrac dot org
Originally committed as revision 25628 to svn://svn.ffmpeg.org/ffmpeg/trunk
The 3GPP spec uses the following calculation for high spreading:
thr'_spr = max(thr_scaled, s_h(n) * thr_scaled(n-1))
where, n is defined as the current band, and s_h() is defined as "[...] the
distance of adjacent bands in Bark and a constant slope that is 15 dB/Bark
[...]". This is a little ambiguous as you would assume you want the Bark
width of the previous band for this calculation. However, this assumption
appears to be incorrect, and you really want the Bark width of the current
band. Coincidentally this is exactly what the spec calls for! =P
This noticeably improves Tom's Diner at low bitrates (I tested at 64kbps,
with mid/side disabled).
Patch by: Nathan Caldwell <saintdev@gmail.com>
Originally committed as revision 25622 to svn://svn.ffmpeg.org/ffmpeg/trunk
These blocks depended on the compiler keeping xmm registers untouched between
them.
Originally committed as revision 25619 to svn://svn.ffmpeg.org/ffmpeg/trunk
suncc does not like the leading commas inside the macro, but it has no problem
with trailing commas.
Originally committed as revision 25615 to svn://svn.ffmpeg.org/ffmpeg/trunk
This greatly improves bitrate handling. You will now get within a few
kbps of your requested bitrate instead of 20-40kbps higher.
There is absolutely no analog to this line in the 3GPP spec, that I
can find.
patch by Nathan Caldwell saintdev (at) gmail
Originally committed as revision 25589 to svn://svn.ffmpeg.org/ffmpeg/trunk
Removing the modification vastly improves quality (at a slight bitrate
cost) for some samples. castanets.wav is a good example. The closest
equivalent I see to the modification in the 3GPP spec is a similar
modification (over a specific frequency range) when TNS is used.
This also changes the threshold-in-quiet calculation to match the
3GPP spec.
patch by Nathan Caldwell saintdev (at) gmail
Originally committed as revision 25588 to svn://svn.ffmpeg.org/ffmpeg/trunk
According to the 3GPP spec:
"Thus the pre-echo control is inactive for the first short window (but
not all short windows in a short frame) after a start block and for
all frames with a stop window sequence."
Currently, pre-echo control is only run when the current frame is not
a short frame, and the previous frame is not a short frame.
patch by Nathan Caldwell saintdev (at) gmail
Originally committed as revision 25587 to svn://svn.ffmpeg.org/ffmpeg/trunk
Reading 7 bits as an unsigned int can't result in a value exceeding 127.
Accordingly, remove error message (as it'll never be reached).
Originally committed as revision 25575 to svn://svn.ffmpeg.org/ffmpeg/trunk
Bug caused by the fact that get_bits(gb, 0) is undefined.
Doesn't affect any streams generated by the official Theora encoder, but such
streams are nevertheless valid.
Fixes decoding of CELT-933dd833-nmr-bandt.ogv.
Originally committed as revision 25573 to svn://svn.ffmpeg.org/ffmpeg/trunk
Do decode init in the init function instead of at the first frame.
Fix some possible crash cases.
Originally committed as revision 25572 to svn://svn.ffmpeg.org/ffmpeg/trunk
Some code was initializing some xmm registers in one asm block and using them
in the following block, assuming they wouldn't be changed in between blocks.
Originally committed as revision 25568 to svn://svn.ffmpeg.org/ffmpeg/trunk
thus making forced key frames work.
Patch by Nicolas George, nicolas d george a normalesup d org
Originally committed as revision 25567 to svn://svn.ffmpeg.org/ffmpeg/trunk
I used the same loop counter for the inner and outer initalization loops.
This caused initalization to only run for the first channel. This in turn lead
to any channel other than the first using only short blocks.
Patch by Nathan Caldwell, saintdev at gmail
Originally committed as revision 25566 to svn://svn.ffmpeg.org/ffmpeg/trunk
