These allocations only depend upon sizeof(SampleType)
(and this size is actually the same for both the fixed-point
and the floating-point encoders for most (all supported?)
systems).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This is in preparation for sharing even more stuff
common to the fixed and floating-point encoders.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
There are lots of files that don't need it: The number of object
files that actually need it went down from 2011 to 884 here.
Keep it for external users in order to not cause breakages.
Also improve the other headers a bit while just at it.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This reverts commits fd5aa93a37
and cf00f60bab
("avcodec/kbdwin: Support arbitrary sized windows").
The change in question has only been made for libavradio.
in anticipation of merging it into the main tree. This has
not happened, so this commit reverts the changes to kbdwin
that are not used for anything else. In particular, these
functions are no longer exported (as avpriv functions);
notice that the fixed-point function has been exported
despite having never been used outside of lavc.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Also remove some internal.h inclusions which have been
unnecessarily added recently.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
AVCodec.channel_layouts is deprecated and Clang (unlike GCC)
warns when setting this field in a codec definition.
Fortunately, Clang (unlike GCC) allows to use
FF_DISABLE_DEPRECATION_WARNINGS inside a definition (of an FFCodec),
so that one can create simple macros to set AVCodec.channel_layouts
that also suppress deprecation warnings for Clang.
(Notice that some of the codec definitions were already
inside FF_DISABLE/ENABLE_DEPRECATION_WARNINGS (that were not
guarded by FF_API_OLD_CHANNEL_LAYOUT); these have been removed.
Also notice that setting AVCodec.channel_layouts was not guarded
by FF_API_OLD_CHANNEL_LAYOUT either, so testing disabling it
it without removing all the codeblocks would not have worked.)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
It reduces typing: Before this patch, there were 105 codecs
whose long_name-definition exceeded the 80 char line length
limit. Now there are only nine of them.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
and remove FF_CODEC_CAP_INIT_THREADSAFE
All our native codecs are already init-threadsafe
(only wrappers for external libraries and hwaccels
are typically not marked as init-threadsafe yet),
so it is only natural for this to also be the default state.
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This is possible, because every given FFCodec has to implement
exactly one of these. Doing so decreases sizeof(FFCodec) and
therefore decreases the size of the binary.
Notice that in case of position-independent code the decrease
is in .data.rel.ro, so that this translates to decreased
memory consumption.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Up until now, codec.h contains both public and private parts
of AVCodec. This exposes the internals of AVCodec to users
and leads them into the temptation of actually using them
and forces us to forward-declare structures and types that
users can't use at all.
This commit changes this by adding a new structure FFCodec to
codec_internal.h that extends AVCodec, i.e. contains the public
AVCodec as first member; the private fields of AVCodec are moved
to this structure, leaving codec.h clean.
Reviewed-by: Anton Khirnov <anton@khirnov.net>
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>
Since the removal of the 16-bit FFT said define is unnecessary as
FFT_FIXED_32 is always !FFT_FLOAT. But one wouldn't believe it when
looking at the code.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
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>
Given that the AVCodec.next pointer has now been removed, most of the
AVCodecs are not modified at all any more and can therefore be made
const (as this patch does); the only exceptions are the very few codecs
for external libraries that have a init_static_data callback.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.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>
* commit 'e22c63ac74b2968075be8bf0d2deb1ee63b28976':
ac3enc: Reshuffle some float/fixed-mode ifdefs to avoid a dummy function
Merged-by: James Almer <jamrial@gmail.com>
* 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>
* qatar/master:
wmaenc: use float planar sample format
(e)ac3enc: use planar sample format
aacenc: use planar sample format
adpcmenc: use planar sample format for adpcm_ima_wav and adpcm_ima_qt
adpcmenc: move 'ch' variable to higher scope
adpcmenc: fix 3 instances of variable shadowing
adpcm_ima_wav: simplify encoding
libvorbis: use planar sample format
libmp3lame: use planar sample formats
vorbisenc: use float planar sample format
ffm: do not write or read the audio sample format
parseutils: fix parsing of invalid alpha values
doc/RELEASE_NOTES: update for the 9 release.
smoothstreamingenc: Add a more verbose error message
smoothstreamingenc: Ignore the return value from mkdir
smoothstreamingenc: Try writing a manifest when opening the muxer
smoothstreamingenc: Move the output_chunk_list and write_manifest functions up
smoothstreamingenc: Properly return errors from ism_flush to the caller
smoothstreamingenc: Check the output UrlContext before accessing it
Conflicts:
doc/RELEASE_NOTES
libavcodec/aacenc.c
libavcodec/ac3enc_template.c
libavcodec/wmaenc.c
tests/ref/lavf/ffm
Merged-by: Michael Niedermayer <michaelni@gmx.at>
