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>
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>
* commit 'b805482b1fba1d82fbe47023a24c9261f18979b6':
aac: Provide more information on the failure message
Merged-by: Hendrik Leppkes <h.leppkes@gmail.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>
This coder produces a much lower quality audio than the rest, is much
slower and is unstable. Hasn't been updated for a very long time as
well, hence it is more appropriate to remove it since it also depends on
a big burden of a code (the encode_window_bands_info function which is
just as old, just as unstable and bad and in no way modifiable or
fixable).
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This patch does 4 things, all of which interact and thus it
woudln't be possible to commit them separately without causing
either quality regressions or assertion failures.
Fate comparison targets don't all reflect improvements in
quality, yet listening tests show substantially improved quality
and stability.
1. Increase SF range utilization.
The spec requires SF delta values to be constrained within the
range -60..60. The previous code was applying that range to
the whole SF array and not only the deltas of consecutive values,
because doing so requires smarter code: zeroing or otherwise
skipping a band may invalidate lots of SF choices.
This patch implements that logic to allow the coders to utilize
the full dynamic range of scalefactors, increasing quality quite
considerably, and fixing delta-SF-related assertion failures,
since now the limitation is enforced rather than asserted.
2. PNS tweaks
The previous modification makes big improvements in twoloop's
efficiency, and every time that happens PNS logic needs to be
tweaked accordingly to avoid it from stepping all over twoloop's
decisions. This patch includes modifications of the sort.
3. Account for lowpass cutoff during PSY analysis
The closer PSY's allocation is to final allocation the better
the quality is, and given these modifications, twoloop is now
very efficient at avoiding holes. Thus, to compute accurate
thresholds, PSY needs to account for the lowpass applied
implicitly during twoloop (by zeroing high bands).
This patch makes twoloop set the cutoff in psymodel's context
the first time it runs, and makes PSY account for it during
threshold computation, making PE and threshold computations
closer to the final allocation and thus achieving better
subjective quality.
4. Tweaks to RC lambda tracking loop in relation to PNS
Without this tweak some corner cases cause quality regressions.
Basically, lambda needs to react faster to overall bitrate
efficiency changes since now PNS can be quite successful in
enforcing maximum bitrates, when PSY allocates too many bits
to the lower bands, suppressing the signals RC logic uses to
lower lambda in those cases and causing aggressive PNS.
This tweak makes PNS much less aggressive, though it can still
use some further tweaks.
Also update MIPS specializations and adjust fuzz
Also in lavc/mips/aacpsy_mips.h: remove trailing whitespace
ff_aac_tableinit is a macro in the case of hardcoded tables, so wrap
that up in a function (similar to how the decoder template does it) and
use that as the argument for ff_thread_once().
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
AAC-Fixed decoder segfaulted. This commit makes the aac encoder
and decoder init the table twice in case of transcoding again.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Since the ff_aac_tableinit() can be called by both the encoder and
the decoder (in case of transcoding) this commit shares the AVOnce
variable to prevent this.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
When both M/S coding and PNS are enabled, scalefactors
and coding books would be mistakenly clobbered when setting
the M/S flag on PNS'd bands. The flag needs to be set to
signal the generation of correlated noise, but the scalefactors,
coefficients and the coding books need to be kept intact.
It didn't work out because of the exceptions that needed to be made
for the "-1" cases and was overall more confusing that just manually
checking and setting options for each profile.
Long Term Prediction allows for prediction of spectral coefficients
via the previously decoded time-dependent samples. This feature
works well with harmonic content 2 or more frames long, like speech,
human or non-human, piano music or any constant tones at very low
bitrates.
It should be noted that the current coder is highly efficient and
the rate control system is unable to encode files at extremely
low bitrates (less than 14kbps seems to be impossible) so this
extension isn't capable of optimum operation. Dramatic difference
is observable with some types of audio and speech but for the most
part the audiable differences are subtle. The spectrum looks better
however so the encoder is able to harvest the additional bits that
this feature provies, should the user choose to enable it. So
it's best to enable this feature only if encoding at the absolutely
lowest bitrate that the encoder is capable of.
Apparently it was set to be enabled by default but after the
profile commits it was reverted to be off by default because
I didn't notice.
Works well so (re)enable it.
