Also remove some internal.h inclusions which have been
unnecessarily added recently.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
It is only used by encoders; in fact, AVCodecContext.time_base
is only used by encoders, so it is only useful for encoders.
Also constify the AVCodecContext parameter in it.
Also fixup the other headers a bit while removing now unnecessary
internal.h inclusions.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Move AC3HeaderInfo into ac3_parser_internal.h and the rest
into a new header ac3defs.h.
This also breaks an include cycle of ac3.h and ac3tab.h
(the latter now only needs ac3defs.h).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This structure is no longer declared in a public header,
so using an FF-prefix is more appropriate.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Also move FF_CODEC_TAGS_END as well as struct AVCodecDefault.
This reduces the amount of files that have to include internal.h
(which comes with quite a lot of indirect inclusions), as e.g.
most encoders don't need it. It is furthemore in preparation
for moving the private part of AVCodec out of the public codec.h.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This avoids unnecessary rebuilds of most source files if only the
list of enabled components has changed, but not the other properties
of the build, set in config.h.
Signed-off-by: Martin Storsjö <martin@martin.st>
The child_class_next API relied on different AVCodecs to use
different AVClasses; yet this API has been replaced by
child_class_iterate.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
The size of the output buffer is always known in advance and
the code has no alignment requirement (it uses mostly the PutBits API),
so allowing user-supplied buffers is trivial.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
When encoding E-AC-3, whether coupling is on or not determines whether
an additional frame based coupling exponent strategy element frmcplexpstr
(of size five bits) is present in the bitstream. So just add five to the
number of bits when counting them instead of adding 5*s->cpl_on (the
latter field is currently only 0 or 1, so it doesn't make a difference).
Furthermore, move some parts of the bit allocation that doesn't change
per-frame to count_frame_bits_fixed() (which is only run once during
init).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
AC-3 and EAC-3 are codecs whose packet sizes are known in advance,
so one can use the min_size parameter of ff_alloc_packet2() to
allocate exactly this amount. This avoids a memcpy later in
av_packet_make_refcounted() in encode_simple_internal().
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Since the very beginning (since de6d9b6404)
the AC-3 encoder used AC3_MAX_CODED_FRAME_SIZE (namely 3840) for the
size of the output buffer (without any check at all).
This causes problems when encoding EAC-3 for which the maximum is too small,
smaller than the actual size of the buffer: One can run into asserts used
by the PutBits API. Ticket #8513 is about such a case and this commit
fixes it by using the real size of the buffer.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Both AC-3 encoder share the same options, yet they are nevertheless
duplicated in the binary; and the options applying to the EAC-3 encoder
are a proper subset of the options for the AC-3 encoders, so that it can
use the same options as the former by putting the options specific to
AC-3 at the front. This commit implements this.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
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%
ff_eac3_exponent_init() set values twice when initializing a static
table; ergo the initialization code must not run concurrently with
a running EAC-3 encoder. Yet this code is executed every time an EAC-3
encoder is initialized. So use ff_thread_once() for this and also for a
similar initialization performed for all AC-3 encoders to make them all
init-threadsafe.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Due to a typo, it was impossible to write 0.595 / -4.5 dB
of ltrt_cmixlev, ltrt_surmixlev, loro_cmixlev, loro_surmixlev.
Without any error 0.841 / -1.5 dB was written to file.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Log the warning message once per encoder instance instead.
Reviewed-by: Kieran Kunhya <kierank@obe.tv>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
Fixes the FATE-tests unknown_layout-ac3, ac3-fixed-encode, ac3-encode
and eac3-encode. It furthermore fixes the ac3-encoder bugs mentioned in
tickets #7994, #8144 and #8159.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This commit uses smaller types for some static const arrays to reduce
their size in case the entries can be represented in the smaller type.
The biggest savings came from inv_map_table in vp9.c.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
The table in question is a 253 byte one. In fact, it turns out that
dynamic generation of the table results in an increased binary size.
Code compiled with GCC 5.2.0, x86-64 (size in bytes), before and after
patch:
old: 62321064 libavcodec/libavcodec.so.57
new: 62320536 libavcodec/libavcodec.so.57
Thus, it always make sense to statically allocate this.
Tested with FATE with/without --enable-hardcoded-tables.
Reviewed-by: wm4 <nfxjfg@googlemail.com>
Signed-off-by: Ganesh Ajjanagadde <gajjanagadde@gmail.com>
This should fix the undefined behavior reported in:
https://trac.ffmpeg.org/ticket/4727.
I can reproduce this at runtime: simply stick in an abort call in
asym_quant to check if c < 0 and run FATE. I don't know ac3 so I can't
confirm if negative coefficients are intentional, but at the moment they
clearly are according to FATE.
This resolves the undefined behavior. Tested with FATE.
Signed-off-by: Ganesh Ajjanagadde <gajjanagadde@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Commit 7404f3bdb switched bitrate to 64 bits.
This triggers -Wabsolute-value on clang, e.g
http://fate.ffmpeg.org/log.cgi?time=20150917122742&log=compile&slot=x86_64-darwin-clang-3.7-O3.
Therefore, usage of abs is changed to llabs, which is available on all of the platforms.
Unfortunately, LLONG_MAX is not always available, so INT64_MAX is used instead.
Signed-off-by: Ganesh Ajjanagadde <gajjanagadde@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* commit '2df0c32ea12ddfa72ba88309812bfb13b674130f':
lavc: use a separate field for exporting audio encoder padding
Conflicts:
libavcodec/audio_frame_queue.c
libavcodec/avcodec.h
libavcodec/libvorbisenc.c
libavcodec/utils.c
libavcodec/version.h
libavcodec/wmaenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Currently, the amount of padding inserted at the beginning by some audio
encoders, is exported through AVCodecContext.delay. However
- the term 'delay' is heavily overloaded and can have multiple different
meanings even in the case of audio encoding.
- this field has entirely different meanings, depending on whether the
codec context is used for encoding or decoding (and has yet another
different meaning for video), preventing generic handling of the codec
context.
Therefore, add a new field -- AVCodecContext.initial_padding. It could
conceivably be used for decoding as well at a later point.
* commit '9a9e2f1c8aa4539a261625145e5c1f46a8106ac2':
dsputil: Split audio operations off into a separate context
Conflicts:
configure
libavcodec/takdec.c
libavcodec/x86/Makefile
libavcodec/x86/dsputil.asm
libavcodec/x86/dsputil_init.c
libavcodec/x86/dsputil_mmx.c
libavcodec/x86/dsputil_x86.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
