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>
Up until now, these encoders received non-refcounted packets
(whose data was owned by the corresponding AVCodecContext)
from ff_alloc_packet(); these packets were made refcounted lateron
by av_packet_make_refcounted() generically.
This commit makes these encoders accept user-supplied buffers by
replacing av_packet_make_refcounted() with an equivalent function
that is based upon get_encode_buffer().
(I am pretty certain that one can also set the flag for mpegvideo-
based encoders, but I want to double-check this later. What is certain
is that it reallocates the buffer owned by the AVCodecContext
which should maybe be moved to encode.c, so that proresenc_kostya.c
and ttaenc.c can make use of it, too.)
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 more spec-compliant because it does not rely
on dead-code elimination by the compiler. Especially
MSVC has problems with this, as can be seen in
https://ffmpeg.org/pipermail/ffmpeg-devel/2022-May/296373.html
or
https://ffmpeg.org/pipermail/ffmpeg-devel/2022-May/297022.html
This commit does not eliminate every instance where we rely
on dead code elimination: It only tackles branching to
the initialization of arch-specific dsp code, not e.g. all
uses of CONFIG_ and HAVE_ checks. But maybe it is already
enough to compile FFmpeg with MSVC with whole-programm-optimizations
enabled (if one does not disable too many components).
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>
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>
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>
This avoids including version.h in all source files, avoiding
unnecessary rebuilds when the version number is bumped. Only
version_major.h is included by the main header, which defines
availability of e.g. FF_API_* macros, and which is bumped much
less often.
This isn't done for libavutil/version.h, because that header needs
to be included essentially everywhere due to LIBAVUTIL_VERSION_INT
being used wherever an AVClass is constructed.
Signed-off-by: Martin Storsjö <martin@martin.st>
avpriv_mpeg4audio_sample_rates has a size of 64B and it is currently
avpriv; a clone of it exists in aacenctab.h and from there it is inlined
in aacenc.c (which also uses the avpriv version) and in the FLV muxer.
This means that despite it being avpriv both libavformat as well as
libavcodec have copies already.
This situation is clearly suboptimal. Given the overhead of exporting
symbols (for x64 Elf/Linux/GNU: 2x2B version, 2x24B .dynsym, 24B .rela.dyn,
8B .got, 4B hash + twice the size of the name (here 31B)) the object is
unavprived, i.e. duplicated into libavformat when creating a shared
build; but the duplicates in the AAC encoder and FLV muxer are removed.
This involves splitting of the sample rate table into a file of its own;
this allowed to break some spurious dependencies (e.g. both the AAC
encoder as well as the Matroska demuxer actually don't need the
mpeg4audio_get_config stuff).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Up until now, ff_alloc_packet2() has a min_size parameter:
It is supposed to be a lower bound on the final size of the packet
to allocate. If it is not too far from the upper bound (namely,
if it is at least half the upper bound), then ff_alloc_packet2()
already allocates the final, already refcounted packet; if it is
not, then the packet is not refcounted and its data only points to
a buffer owned by the AVCodecContext (in this case, the packet will
be made refcounted in encode_simple_internal() in libavcodec/encode.c).
The goal of this was to avoid data copies and intermediate buffers
if one has a precise lower bound.
Yet those encoders for which precise lower bounds exist have recently
been switched to ff_get_encode_buffer() (which automatically allocates
final buffers), leaving only two encoders to actually set the min_size
to something else than zero (namely aliaspixenc and hapenc). Both of
these encoders use a very low lower bound that is not helpful in any
nontrivial case.
This commit therefore removes the min_size parameter as well as the
codepath in ff_alloc_packet2() for the allocation of final buffers.
Furthermore, the function has been renamed to ff_alloc_packet() and
moved to encode.h alongside ff_get_encode_buffer().
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This used to be the default, but was reverted as it was slower than
the 'fast' coder by around 25%.
Since our encoder is still not very good, change back to the twoloop
coder by default. It has much better rate control management as well,
making it closer to CBR, and it sounds much better.
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>
This table is currently initialized up to three times: Once by the
encoder and twice by the decoders (once by the fixed and once by the
floating-point decoder); each of these initializations is guarded by an
AVOnce, yet the fact that there are three of them implies that there
might be data races (the fact that each entry is only written to once
(to its final value) when initializing means that this is safe in
practice, yet it is still undefined behaviour). Fix this by only
initializing the table from one place that is guarded by a single AVOnce.
This also avoids unnecessary duplications of the init code.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
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.
The twoloop coder sounds decent at low bitrates, however at higher bitrates
it sounds worse than the fast coder (which used to be the old twoloop coder
before October 2015) and needs quite a lot more CPU.
Change the default to fast. It has been well tested and has had little changes
over the years so its been confirmed to be quite stable.
Also change its description (not valid for more than a year) and the
documentation.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This commit implements support for PCE (Program Configuration Elements) in the
AAC encoder, and as such allows for encoding of channel layouts not present
in the presets defined by the spec (which only lists the 8 most common ones).
This has been a highly requested feature and is also the first open source encoder
to support this many layouts.
Many thanks to pkviet <pkv.stream@gmail.com> who implemented support for and
verified all channel layouts.
* commit '97cfe1d8bd1968143e2ba9aa46ebe9504a835e24':
Convert all AVClass struct declarations to designated initializers.
Merged-by: James Almer <jamrial@gmail.com>
The libopus encoder does the same thing and its better than
keeping track of when the empty flush frames appear.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Performance improvements:
quant_bands:
with: 681 decicycles in quant_bands, 8388453 runs, 155 skips
without: 1190 decicycles in quant_bands, 8388386 runs, 222 skips
Around 42% for the function
Twoloop coder:
abs_pow34:
with/without: 7.82s/8.17s
Around 4% for the entire encoder
Both:
with/without: 7.15s/8.17s
Around 12% for the entire encoder
Fast coder:
abs_pow34:
with/without: 3.40s/3.77s
Around 10% for the entire encoder
Both:
with/without: 3.02s/3.77s
Around 20% faster for the entire encoder
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Tested-by: Michael Niedermayer <michael@niedermayer.cc>
Reviewed-by: James Almer <jamrial@gmail.com>
Using lfg was an overkill in this case where the random numbers
were only used for encoder descisions. Should increase result
uniformity between different FPUs and gives a slight speedup.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Fixes occurance of NaN/Inf leading to assertion failures and out of array access
Fixes: d1c38a09acc34845c6be3a127a5aacaf/signal_sigsegv_3982225_6121_d18bd5451d4245ee09408f04badd1b83.wmv
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This version has had much testing so there's little point in keeping it
maked as experimental.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Move wi.clipping computation outside of psy_lame_window, LFE
channels don't even call that, and make the LFE path also
initialize window_type[1] which is needed by analyze_channel
Results in dropping out in channels, usually on EIGHT_SHORT windows.
Will be reenabled once the cause has been investigated and a fix has
been made.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Takes into account whether there's pairing and if there's an LFE channel.
An SCE has more bits than CPE/2 since IS and M/S save quite a lot of bits
when channels are paired. And most of the SCEs we have are in surround
layouts which map it to the center channel, which usually carries all of
the dialogue and compression artifacts there are easily audiable.
Also refactors the init function a little bit and labels some parts of it.
Fixes bug #5233
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This is needed as near infinite values on the input side result in only some
output to be non finite.
Also it may still be insufficient if subsequent computations overflow
Fixes null pointer dereference
Fixes: ae66c0f6c12ac1cd5c2c237031240f57/signal_sigsegv_2618c99_9516_6007026f2185a26d7afea895fbed6e38.ogg
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Reviewed-by: Claudio Freire <klaussfreire@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Has been marked for removal for over a month and has not been improved
or touched at all since it was implemented.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Too many crashes observed. Can't be helped until the autocorrelation
function is massively checked for sanity.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
