This cap is currently used to mark multithreading-capable codecs that
wrap external libraries with their own multithreading code. The name is
highly confusing for our API users, since libavcodec ALWAYS handles
thread_count=0 (see commit message in previous commit). Therefore rename
the cap and update its documentation to make its meaning clear.
The old name is kept deprecated until next+1 major bump.
This callback is functionally the same as get_buffer2() is for decoders, and
implements for the new encode API the functionality of the old encode API had
where the user could provide their own buffers.
Reviewed-by: Lynne <dev@lynne.ee>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Reviewed-by: Mark Thompson <sw@jkqxz.net>
Signed-off-by: James Almer <jamrial@gmail.com>
av_packet_add_side_data() already defines size as a size_t, so this makes it
consistent across all side data functions
Signed-off-by: James Almer <jamrial@gmail.com>
Originally deprecated in 748c2fca7e,
publically deprecated in 9a07c1332c
(merged into FFmpeg in 1885ffb03d).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Enables the usage of such values as AV_EF_EXPLODE in encoders, which
can be useful in cases such as subtitle encoders where they have the
responsibility to validate the correctness of an incoming ASS dialog line.
Signed-off-by: Jan Ekström <jan.ekstrom@24i.com>
AVFrame hasn't been a struct defined in libavcodec for a decade now, when
it was moved to libavutil.
Found-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
avcodec_find_best_pix_fmt2 has been deprecated and replaced by
avcodec_find_best_pix_fmt_of_2 in 2a54ae9df8.
avcodec_find_best_pix_fmt_of_2 and avcodec_get_pix_fmt_loss meanwhile
were deprecated in 617e866e25 when these
functions were de facto moved to libavutil; this has been mentioned in
APIchanges in f7a1c5e4d2. Yet the
attribute_deprecated was never set for the latter two functions and they
were not wrapped in an FF_API define. This commit does this.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Reviewed-by: James Almer <jamrial@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
AVID streams - currently handled by the AVRN decoder - can be (depending
on extradata contents) either MJPEG or raw video. To decode the MJPEG
variant, the AVRN decoder currently instantiates a MJPEG decoder
internally and forwards decoded frames to the caller (possibly after
cropping them).
This is suboptimal, because the AVRN decoder does not forward all the
features of the internal MJPEG decoder, such as direct rendering.
Handling such forwarding in a full and generic manner would be quite
hard, so it is simpler to just handle those streams in the MJPEG decoder
directly.
The AVRN decoder, which now handles only the raw streams, can now be
marked as supporting direct rendering.
This also removes the last remaining internal use of the obsolete
decoding API.
Neither the feature, public fields, or AVOptions were ever truly deprecated,
nor will have been removed if this FF_API_ define was left in place, so
get rid of it as it's misleading.
Signed-off-by: James Almer <jamrial@gmail.com>
It has been deprecated for 4 years and certain new codecs do not work
with it.
Also include AVCodecContext.refcounted_frames, as it has no effect with
the new API.
In order to fine-control referencing schemes in VP9 encoding, there
is a need to use VP9E_SET_SVC_REF_FRAME_CONFIG method. This commit
provides a way to use the API through frame metadata.
The AC3 encoder used to be a separate library called "Aften", which
got merged into libavcodec (literally, SVN commits and all).
The merge preserved as much features from the library as possible.
The code had two versions - a fixed point version and a floating
point version. FFmpeg had floating point DSP code used by other
codecs, the AC3 decoder including, so the floating-point DSP was
simply replaced with FFmpeg's own functions.
However, FFmpeg had no fixed-point audio code at that point. So
the encoder brought along its own fixed-point DSP functions,
including a fixed-point MDCT.
The fixed-point MDCT itself is trivially just a float MDCT with a
different type and each multiply being a fixed-point multiply.
So over time, it got refactored, and the FFT used for all other codecs
was templated.
Due to design decisions at the time, the fixed-point version of the
encoder operates at 16-bits of precision. Although convenient, this,
even at the time, was inadequate and inefficient. The encoder is noisy,
does not produce output comparable to the float encoder, and even
rings at higher frequencies due to the badly approximated winow function.
Enter MIPS (owned by Imagination Technologies at the time). They wanted
quick fixed-point decoding on their FPUless cores. So they contributed
patches to template the AC3 decoder so it had both a fixed-point
and a floating-point version. They also did the same for the AAC decoder.
They however, used 32-bit samples. Not 16-bits. And we did not have
32-bit fixed-point DSP functions, including an MDCT. But instead of
templating our MDCT to output 3 versions (float, 32-bit fixed and 16-bit fixed),
they simply copy-pasted their own MDCT into ours, and completely
ifdeffed our own MDCT code out if a 32-bit fixed point MDCT was selected.
This is also the status quo nowadays - 2 separate MDCTs, one which
produces floating point and 16-bit fixed point versions, and one
sort-of integrated which produces 32-bit MDCT.
MIPS weren't all that interested in encoding, so they left the encoder
as-is, and they didn't care much about the ifdeffery, mess or quality - it's
not their problem.
So the MDCT/FFT code has always been a thorn in anyone looking to clean up
code's eye.
Backstory over. Internally AC3 operates on 25-bit fixed-point coefficients.
So for the floating point version, the encoder simply runs the float MDCT,
and converts the resulting coefficients to 25-bit fixed-point, as AC3 is inherently
a fixed-point codec. For the fixed-point version, the input is 16-bit samples,
so to maximize precision the frame samples are analyzed and the highest set
bit is detected via ac3_max_msb_abs_int16(), and the coefficients are then
scaled up via ac3_lshift_int16(), so the input for the FFT is always at least 14 bits,
computed in normalize_samples(). After FFT, the coefficients are scaled up to 25 bits.
This patch simply changes the encoder to accept 32-bit samples, reusing
the already well-optimized 32-bit MDCT code, allowing us to clean up and drop
a large part of a very messy code of ours, as well as prepare for the future lavu/tx
conversion. The coefficients are simply scaled down to 25 bits during windowing,
skipping 2 separate scalings, as the hacks to extend precision are simply no longer
necessary. There's no point in running the MDCT always at 32 bits when you're
going to drop 6 bits off anyway, the headroom is plenty, and the MDCT rounds
properly.
This also makes the encoder even slightly more accurate over the float version,
as there's no coefficient conversion step necessary.
SIZE SAVINGS:
ARM32:
HARDCODED TABLES:
BASE - 10709590
DROP DSP - 10702872 - diff: -6.56KiB
DROP MDCT - 10667932 - diff: -34.12KiB - both: -40.68KiB
DROP FFT - 10336652 - diff: -323.52KiB - all: -364.20KiB
SOFTCODED TABLES:
BASE - 9685096
DROP DSP - 9678378 - diff: -6.56KiB
DROP MDCT - 9643466 - diff: -34.09KiB - both: -40.65KiB
DROP FFT - 9573918 - diff: -67.92KiB - all: -108.57KiB
ARM64:
HARDCODED TABLES:
BASE - 14641112
DROP DSP - 14633806 - diff: -7.13KiB
DROP MDCT - 14604812 - diff: -28.31KiB - both: -35.45KiB
DROP FFT - 14286826 - diff: -310.53KiB - all: -345.98KiB
SOFTCODED TABLES:
BASE - 13636238
DROP DSP - 13628932 - diff: -7.13KiB
DROP MDCT - 13599866 - diff: -28.38KiB - both: -35.52KiB
DROP FFT - 13542080 - diff: -56.43KiB - all: -91.95KiB
x86:
HARDCODED TABLES:
BASE - 12367336
DROP DSP - 12354698 - diff: -12.34KiB
DROP MDCT - 12331024 - diff: -23.12KiB - both: -35.46KiB
DROP FFT - 12029788 - diff: -294.18KiB - all: -329.64KiB
SOFTCODED TABLES:
BASE - 11358094
DROP DSP - 11345456 - diff: -12.34KiB
DROP MDCT - 11321742 - diff: -23.16KiB - both: -35.50KiB
DROP FFT - 11276946 - diff: -43.75KiB - all: -79.25KiB
PERFORMANCE (10min random s32le):
ARM32 - before - 39.9x - 0m15.046s
ARM32 - after - 28.2x - 0m21.525s
Speed: -30%
ARM64 - before - 36.1x - 0m16.637s
ARM64 - after - 36.0x - 0m16.727s
Speed: -0.5%
x86 - before - 184x - 0m3.277s
x86 - after - 190x - 0m3.187s
Speed: +3%
Monochrome encoding with libaom was buggy for a long time, but this was
finally sorted out in libaom 2.0.1 (2.0.0 is almost there but was still
buggy in realtime mode).
We'll keep support for libaom 1.x around until the LTS distros that
include it are EOL (which is still a long time from now).
Fixes: https://trac.ffmpeg.org/ticket/7599
The nvidia hardware explicitly supports decoding monochrome content,
presumably for the AVIF alpha channel. Supporting this requires an
adjustment in av1dec and explicit monochrome detection in nvdec.
I'm not sure why the monochrome path in av1dec did what it did - it
seems non-functional - YUV440P doesn't seem a logical pix_fmt for
monochrome and conditioning on chroma sub-sampling doesn't make sense.
So I changed it.
I've tested 8bit content, but I haven't found a way to create a 10bit
sample, so that path is untested for now.
They add considerable complexity to frame-threading implementation,
which includes an unavoidably leaking error path, while the advantages
of this option to the users are highly dubious.
It should be always possible and desirable for the callers to make their
get_buffer2() implementation thread-safe, so deprecate this option.
This function is so extremely simple that it is preferable to make it
inline rather than deal with all the complications arising from it being
an exported symbol.
Keep avpriv_align_put_bits() around until the next major bump to
preserve ABI compatibility.
Added VDPAU to list of supported formats for VP9 420 10 and 12 bit
formats. Add VP9 10/12 Bit support for VDPAU
Signed-off-by: Philip Langdale <philipl@overt.org>
This AV1 decoder is currently only used for hardware accelerated decoding.
It can be extended into a native decoder in the future, so set its name to
"av1" and temporarily give it the lowest priority in the codec list.
Signed-off-by: Fei Wang <fei.w.wang@intel.com>
Signed-off-by: James Almer <jamrial@gmail.com>
This patch allows setting a compression ratio and to
set multiple layers. The user has to input a compression
ratio for each layer.
The per layer compression ration can be set as follows:
-layer_rates "r1,r2,...rn"
for to create 'n' layers.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
