* newdev/master:
avio: make udp_set_remote_url/get_local_port internal.
asfdec: also subtract preroll when reading simple index object
matroskaenc: remove a variable that's unused after bc17bd9.
avio: cosmetics - nicer vertical alignment.
Remove unnecessary icc version checks
Disable 'attribute "foo" ignored' warnings from icc
rtsp: Don't use a locale dependent format string
Add xd55 codec tag for XDCAM HD422 720p25 CBR files.
configure: get libavcodec version from new version.h header
lavc: move the version macros to a new installed header.
matroskaenc: simplify get_aac_sample_rates by using ff_mpeg4audio_get_config
Do not use format string "%0.3f" for RTSP Range field.
Add apply_window_int16() to DSPContext with x86-optimized versions and use it in the ac3_fixed encoder.
Document usage of import libraries created by dlltool
configure: Set the correct lib target for arm/wince dlltool
fate: simplify regression-funcs.sh
fate: add support for multithread testing
Conflicts:
libavformat/rtspdec.c
libavutil/attributes.h
libavutil/internal.h
libavutil/mem.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* newdev/master: (33 commits)
Fix an infinite loop when RoQ encoded generated a frame with a size greater than the maximum valid size.
Add kbdwin.o to AC3 decoder
Detect byte-swapped AC-3 and support decoding it directly.
cosmetics: indentation
Always copy input data for AC3 decoder.
ac3enc: make sym_quant() branch-free
cosmetics: indentation
Add a CPU flag for the Atom processor.
id3v2: skip broken tags with invalid size
id3v2: don't explicitly skip padding
Make sure kbhit() is in conio.h
fate: update wmv8-drm reference
vc1: make P-frame deblock filter bit-exact.
configure: Add the -D parameter to the dlltool command
amr: Set the AVFMT_GENERIC_INDEX flag
amr: Set the pkt->pos field properly to the start of the packet
amr: Set the codec->bit_rate field based on the last packet
rtsp: Specify unicast for TCP interleaved streams, too
Set the correct target for mingw64 dlltool
applehttp: Change the variable for stream position in seconds into int64_t
...
Conflicts:
ffmpeg.c
ffplay.c
libavcodec/ac3dec.c
libavformat/avio.h
libavformat/id3v2.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This allows the AC-3 decoder to be used directly with RealMedia
decoders that unlike the libavformat one do not byte-swap automatically.
Since the new code is only used in case we would fail directly otherwise
there should be no risk for regressions.
Depending on error_recognition is not correct, low values do
certainly not mean it is ok to crash.
Based on a patch by Reimar Döffinger <Reimar.Doeffinger@gmx.de>
These windows do not really belong in fft/mdct files and were
easily confused with the similarly named tables used by rdft.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This function is not tightly coupled to mdct, and it's in the way
of making a fixed-point mdct implementation.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This removes the rather pointless wrappers (one not even inline)
for calling the fft_calc and related function pointers.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Since initially committed in 2004, this codec has only been touched
for maintenanance. Functionally, it contains no novel ideas and
its intended audience is better served by existing mature codecs.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This allows the AC-3 decoder to be used directly with RealMedia
decoders that unlike the libavformat one do not byte-swap automatically.
Since the new code is only used in case we would fail directly otherwise
there should be no risk for regressions.
The "buf" pointer needs to be overwritten since otherwise the CRC check fails.
3GPP:
Remove ffac from and move min_snr out of AacPsyBand.
Rearrange AacPsyCoeffs to make it easier to implement energy spreading.
Rename the band[] array to bands[]
Copy energies and thresholds at the end of analysis.
LAME:
Use a loop instead of an if chain in LAME windowing.
There are several places where a buffer is byte-swapped in 16-bit units.
This allows them to share code which can be optimised for various
architectures.
Signed-off-by: Mans Rullgard <mans@mansr.com>
If the function is not inlined, an immmediate cannot be used for the
shift parameter, so the %cl register must be used instead in that case.
This fixes compilation for x86-32 using gcc with --disable-optimizations.
This fixes unexpected name collisions that were occurring with variables
declared within the macros.
It also fixes the fate-acodec-ac3_fixed regression test on x86-32.
This reverts commit cc4d3dd3e2.
revert at authors request due to better impementation being available
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This increases the accuracy of coefficients, leading to improved quality.
Rescaling of the coefficients to full 25-bit accuracy is done rather than
offsetting the exponent values. This requires coefficient scaling to be done
before determining the rematrixing strategy. Also, the rematrixing strategy
calculation must use 64-bit math to prevent overflow due to the higher
precision coefficients.
The Broadcom CrystalHD decoder chips provide hardware video
decoding for a number of video formats. It does so using a
memory:memory interface where a compressed bitstream is fed
in and decompressed pictures are copied out. As such, it works
independent of any graphics hardware in the system.
Features supported in this initial version:
* Support for Linux (using current drivers/library from git.wilsonet.com)
* Support for 70015 hardware
* Formats: MPEG2, MPEG4 Part 2, H.264, VC1 and DivX 3.11 (untested)
* Progressive content
* Non-H.264 Interlaced content
* H.264 MBAFF content
Features missing in this initial version:
* Support for OSX (might work - untested)
* Support for Windows
* Support for 70012 hardware
* H.264 PAFF content
Signed-off-by: Philip Langdale <philipl@overt.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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>
This makes channel coupling more accurate, increasing quality for stereo
content. It also simplifies exponent extraction and mantissa quantization
by no longer needing to apply an offset to the exponents.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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>
When decoding latm config, use the corresponding aac context and its
m4ac instead of using NULL and a local variable. This fixes decoding of
audio in MPEG TS from SBTVD (the Brazillian Digital TV Sytem), when
there is no extradata. This is the case when using the decoder with
gst-ffmpeg and a GStreamer mpegts demuxer.
Signed-off-by: Thadeu Lima de Souza Cascardo <cascardo@holoscopio.com>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 36864ac354)
Should an AVC-1 in MP4 stream not contain SPS or PPS NAL units,
this BSF is then unable to allocate an output buffer for the
modified stream. Warn that the resulting stream may be unplayable.
Fix roundup issue #2386.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 02dd3666c2)
When decoding latm config, use the corresponding aac context and its
m4ac instead of using NULL and a local variable. This fixes decoding of
audio in MPEG TS from SBTVD (the Brazillian Digital TV Sytem), when
there is no extradata. This is the case when using the decoder with
gst-ffmpeg and a GStreamer mpegts demuxer.
Signed-off-by: Thadeu Lima de Souza Cascardo <cascardo@holoscopio.com>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Should an AVC-1 in MP4 stream not contain SPS or PPS NAL units,
this BSF is then unable to allocate an output buffer for the
modified stream. Warn that the resulting stream may be unplayable.
Fix roundup issue #2386.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Currently it is always 4, but this change will allow it to be adjusted when
bandwidth-related features are added such as channel coupling, enhanced
channel coupling, and spectral extension.
(cherry picked from commit 53e35fd340)
Currently it is always 4, but this change will allow it to be adjusted when
bandwidth-related features are added such as channel coupling, enhanced
channel coupling, and spectral extension.
This moves setting the thread count to a minimum of 1 to
frame_thread_init(), allowing a value of zero to propagate
through to the codec if frame threading is not used. This
makes auto-threads work in libx264.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit ff1efc524c)
For intra codecs, ff_thread_finish_setup() is called before decoding starts
automatically. However, get_buffer can only be used before it's called, so
adding this requirement broke frame threading for them. Fixed by moving the
call until after get_buffer is finished.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit ad9791e12b)
This moves setting the thread count to a minimum of 1 to
frame_thread_init(), allowing a value of zero to propagate
through to the codec if frame threading is not used. This
makes auto-threads work in libx264.
Signed-off-by: Mans Rullgard <mans@mansr.com>
For intra codecs, ff_thread_finish_setup() is called before decoding starts
automatically. However, get_buffer can only be used before it's called, so
adding this requirement broke frame threading for them. Fixed by moving the
call until after get_buffer is finished.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
The assembler emits literal pools too far from the load instructions,
so we must do it explicitly at a suitable location.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 8b454c352f)
The assembler emits literal pools too far from the load instructions,
so we must do it explicitly at a suitable location.
Signed-off-by: Mans Rullgard <mans@mansr.com>
decode_init sets bands[0] == 2, so this loop always sets the band table
index (k) to zero.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit a304def1dc)
This avoids the core substream extensions scan when the EXT_AUDIO_ID
field indicates no extensions or only unsupported extensions. The scan
is done only if the value of EXT_AUDIO_ID is unknown or indicates a
present XCh extension which we can decode.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 7e06e0ede3)
There is no need to expand to 16-bits. Just use memcpy() to copy the raw data.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 1108f8998c)
This also adds output buffer size checks for AUDIO and SILENCE block types.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 1574eff3d2)
The size should depend on the output sample size, not the internal bit depth.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit a58bcb40b1)
This avoids the core substream extensions scan when the EXT_AUDIO_ID
field indicates no extensions or only unsupported extensions. The scan
is done only if the value of EXT_AUDIO_ID is unknown or indicates a
present XCh extension which we can decode.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This allows the values to be used without changing C code and is closer to how
the other DEBUG flags work.
If this causes a problem for any user of this flag, please tell me and
ill split the flag in 2.
The 4-tap filters should only access one row/column before the
reference block.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit e0e46cae37)
GCC 4.3 and later are more particular about signedness matching
in vector operations. The operations under if(rangered) were
missing assignments and thus had no effect.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 381efba0ec)
Merging these functions allows merging some loops, which makes the
results (particularly after SIMD optimizations) much faster.
(cherry picked from commit f8bed30d8b)
Advantage is that it allows us to combine several loops into a single
one, and these can eventually be merged into the IDCT itself. Also, it
allows us to remove vc1_put_block(), and makes CODEC_FLAG_GRAY faster.
(cherry picked from commit bbfd2e7ab4)
GCC 4.3 and later are more particular about signedness matching
in vector operations. The operations under if(rangered) were
missing assignments and thus had no effect.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Advantage is that it allows us to combine several loops into a single
one, and these can eventually be merged into the IDCT itself. Also, it
allows us to remove vc1_put_block(), and makes CODEC_FLAG_GRAY faster.
Advanced profile never uses "range reduction", so vc1_put_block() quite
literally just calls put_pixels_clamped() from vc1_decode_i_blocks_adv().
By inlining the function, we can prevent calling IDCT8x8 if
CODEC_FLAG_GRAY is set, and we don't have to scale the coeffs in the
[0,256] range, but can instead use put_signed_pixels_clamped().
(cherry picked from commit 70aa916e46)
With negative stride, the start of the edge_emu buffer should be pointing to
the last line, not the end of the buffer.
With positive stride, pointing to the end of the buffer was completely wrong.
(cherry picked from commit a89f4ca005)
Advanced profile never uses "range reduction", so vc1_put_block() quite
literally just calls put_pixels_clamped() from vc1_decode_i_blocks_adv().
By inlining the function, we can prevent calling IDCT8x8 if
CODEC_FLAG_GRAY is set, and we don't have to scale the coeffs in the
[0,256] range, but can instead use put_signed_pixels_clamped().
With negative stride, the start of the edge_emu buffer should be pointing to
the last line, not the end of the buffer.
With positive stride, pointing to the end of the buffer was completely wrong.
The rematrixing strategy reuse flags are not reset between frames, so they
need to be initialized for all blocks, not just block 0.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 5b54d4b376)
The rematrixing strategy reuse flags are not reset between frames, so they
need to be initialized for all blocks, not just block 0.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This is to match the value in every (E-)AC-3 file from commercial sources.
It has a negligible effect on audio quality.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 50d7140441)
VBV delay is useful for T-STD compliance in some TS muxers. It is
certainly possible to retrieve it by parsing the output of FFmpeg, but
getting it from the context makes it simpler and less error-prone.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This is to match the value in every (E-)AC-3 file from commercial sources.
It has a negligible effect on audio quality.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Perform validity check on AVFormatContext.channels instead of
uninitialised field.
This fixes issue 2001.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 9806fbd535)
It is pretty hopeless that other considerable projects will adopt
libavutil alone in other projects. Projects that need small footprint
are better off with more specialized libraries such as gnulib or rather
just copy the necessary parts that they need. With this in mind, nobody
is helped by having libavutil and libavcore split. In order to ease
maintenance inside and around FFmpeg and to reduce confusion where to
put common code, avcore's functionality is merged (back) to avutil.
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
VBV delay is useful for T-STD compliance in some TS muxers. It is
certainly possible to retrieve it by parsing the output of FFmpeg, but
getting it from the context makes it simpler and less error-prone.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Perform validity check on AVFormatContext.channels instead of
uninitialised field.
This fixes issue 2001.
Signed-off-by: Mans Rullgard <mans@mansr.com>
AC3DSPContext.ac3_max_msb_abs_int16() finds the maximum MSB of the absolute
value of each element in an array of int16_t.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit fbb6b49dab)
This fixes visual glitches in Bink version 'b' files, as the quantization
tables were not being permuted.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 2315392174)
AC3DSPContext.ac3_max_msb_abs_int16() finds the maximum MSB of the absolute
value of each element in an array of int16_t.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Some MPEG4 cameras produce files with empty GOP headers.
This patch makes the decoder ignore such broken headers and proceed
with the following I-frame. Without this change, the following
start code is missed resulting in the entire I-frame being skipped.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This fixes visual glitches in Bink version 'b' files, as the quantization
tables were not being permuted.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Using doubles make the double -> int cast well defined for all the values
used, with the exception of when s[i]==1.0, which is special-cased.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 47d62c965b)
Using doubles make the double -> int cast well defined for all the values
used, with the exception of when s[i]==1.0, which is special-cased.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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>
(cherry picked from commit 626264b11b)
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>
(cherry picked from commit 4a72765a1c)
The function return type is void, so a return statement with an
expression is forbidden (and pointless).
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit b4668274b9)
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>
(cherry picked from commit 9a77a92c2b)
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
This fixes a possibly exploitable buffer overflow and it will likely also be needed for future overreading fixes.
Originally committed as revision 25546 to svn://svn.ffmpeg.org/ffmpeg/trunk
The 8 bits offset (nal unit type) should not be added, as the spec says:
"This bit offset is the offset within the RBSP data for the slice, relative
to the starting position of the slice_header() in the RBSP"
This fixes DXVA2 support for intel GPU.
Patch by Rafaël Carré (funman _AT_ videolan _DOT_ org).
Originally committed as revision 25538 to svn://svn.ffmpeg.org/ffmpeg/trunk
