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.
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>
Fixes out of array read
Fixes: 04442da73d935b776d2236282588d4f9/signal_sigsegv_2625a69_8790_ae85ffc889070663319b3417ede777b0.mov
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
All MDCT outputs must be checked in case of 128point MDCTs
Fixes: out of array read
Fixes: 04442da73d935b776d2236282588d4f9/signal_sigsegv_2625a69_351_52ca6226eb83547a2d26e322ce84ed84.mov
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Change the condition for application of the M/S transform to match
that of the decoder. Namely, that no special coding books must be
in use in either channel. While the condition ought to be
equivalent to the current one when the invariant of is_mask is
kept, matching the decoder's condition is safer and easier to
maintain.
The type of last_frame_pb_count was chosen to be an int since overflow
is impossible (the spec says the maximum bits per frame is 6144 per
channel and the encoder checks for that).
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
PSNR doesn't change as expected. The AAC spec doesn't really say
anything about how exactly to generate noise.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This fixes out-of-bounds reads in avoid_clipping.
Reviewed-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
These variables are coming from mpegvideoenc where are supposedly used
as bit counters on various frame properties. However their use is
unclear as they lack documentation, are available only from a very small
subset of encoders, and they are hardly used in the wild. Also frame_bits
in aacenc is employed in a similar way.
Remove this functionality from AVCodecContex, these variable are mostly
frame properties, and too few encoders support setting them with anything
useful.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
The original plan was to have TNS use data from the PNS search to better
tune itself to noise but this was never used nor necessary. This should
slightly boost the PNS accuracy if TNS was used.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Thiss commit removes the experimental flag from the native AAC Encoder
and thus makes it the default.
After a lot of work, done by myself and Claudio Freire, the quality of
this encoder rivals and surpasses libfdk_aac in some situations. The
encoder had instability issues earlier which prevented it from having
its experimental flag removed, however the last commits done by Claudio
removed the last known source of instability and solved a lot of
problems which were previously observed. The issues were caused by the
various coding tools interfering with the scalefactor indices. Thus,
with these problems solved, it should now be possible to declare this
encoder as the default and recommend that the users should use it
instead of others provided by external libraries, as it is both faster
and has a subjectively higher quality with selected tracks.
The encoder has still yet to be fine tuned for every possible audio file
type like music or voice, so it is hoped that with the experimental flag
removed the users should be able to provide feedback and make the
encoder better than the alternatives for every type of audio and at
every bitrate.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
ANMR has some interesting things coming up but is currently not in a
shape fit for non-experimental usage. Same with "FAST".
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Andreas Rheinhardt
Michael Niedermayer
Lynne
Andreas Rheinhardt
Anton Khirnov
Limin Wang
Marton Balint
Moritz Barsnick
Rostislav Pehlivanov
Diego Biurrun
James Almer
Claudio Freire
Reimar Döffinger
Ganesh Ajjanagadde
Andreas Cadhalpun
Vittorio Giovara