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.
This commit adds the ability for a profile to set the default
options, as well as for the user to override such options
by simply stating them in the command line while still keeping
the same profile, as long as those options are still permitted by
the profile.
Example: setting the profile to aac_low (the default) will turn
PNS and IS on. They can be disabled by -aac_pns 0 and -aac_is 0,
respectively. Turning on -aac_pred 1 will cause the profile to be
elevated to aac_main, as long as no options forbidding aac_main
have been entered (like AAC-LTP, which will be pushed soon).
A useful feature is that by setting the profile to mpeg2_aac_low,
all MPEG4 features will be disabled and if the user tries to enable
them then the program will exit with an error. This profile is
signalled with the same bitstream as aac_low (MPEG4) but some devices
and decoders will fail if any MPEG4 features have been enabled.
This commit implements support for 7.1 channel audio. There's no
more predefined bitstream channel mappings so going beyond 8 channels
(and 7 channels exactly) will require programmable channel elements,
which is already underway.
The bulk of calls to quantize_band_cost are replaced
by a call to a version that memoizes, greatly improving
performance, since during coefficient search there is
a great deal of repeat work.
Memoization cannot always be applied, so do this in a
different function, and leave the original as-is.
This finalizes merging of the work in the patches in ticket #2686.
Improvements to twoloop and RC logic are extensive.
The non-exhaustive list of twoloop improvments includes:
- Tweaks to distortion limits on the RD optimization phase of twoloop
- Deeper search in twoloop
- PNS information marking to let twoloop decide when to use it
(turned out having the decision made separately wasn't working)
- Tonal band detection and priorization
- Better band energy conservation rules
- Strict hole avoidance
For rate control:
- Use psymodel's bit allocation to allow proper use of the bit
reservoir. Don't work against the bit reservoir by moving lambda
in the opposite direction when psymodel decides to allocate more/less
bits to a frame.
- Retry the encode if the effective rate lies outside a reasonable
margin of psymodel's allocation or the selected ABR.
- Log average lambda at the end. Useful info for everyone, but especially
for tuning of the various encoder constants that relate to lambda
feedback.
Psy:
- Do not apply lowpass with a FIR filter, instead just let the coder
zero bands above the cutoff. The FIR filter induces group delay,
and while zeroing bands causes ripple, it's lost in the quantization
noise.
- Experimental VBR bit allocation code
- Tweak automatic lowpass filter threshold to maximize audio bandwidth
at all bitrates while still providing acceptable, stable quality.
I/S:
- Phase decision fixes. Unrelated to #2686, but the bugs only surfaced
when the merge was finalized. Measure I/S band energy accounting for
phase, and prevent I/S and M/S from being applied both.
PNS:
- Avoid marking short bands with PNS when they're part of a window
group in which there's a large variation of energy from one window
to the next. PNS can't preserve those and the effect is extremely
noticeable.
M/S:
- Implement BMLD protection similar to the specified in
ISO-IEC/13818:7-2003, Appendix C Section 6.1. Since M/S decision
doesn't conform to section 6.1, a different method had to be
implemented, but should provide equivalent protection.
- Move the decision logic closer to the method specified in
ISO-IEC/13818:7-2003, Appendix C Section 6.1. Specifically,
make sure M/S needs less bits than dual stereo.
- Don't apply M/S in bands that are using I/S
Now, this of course needed adjustments in the compare targets and
fuzz factors of the AAC encoder's fate tests, but if wondering why
the targets go up (more distortion), consider the previous coder
was using too many bits on LF content (far more than required by
psy), and thus those signals will now be more distorted, not less.
The extra distortion isn't audible though, I carried extensive
ABX testing to make sure.
A very similar patch was also extensively tested by Kamendo2 in
the context of #2686.
This patch modifies the encode frame function to
retry encoding the frame when the resulting bit count
is too far off target, but only adjusting lambda
in small, incremental step. It also makes the logic
more conservative - otherwise it will contend with
bit reservoir-related variations in bit allocation,
and result in artifacts when frame have to be truncated
(usually at high bit rates transitioning from low
complexity to high complexity).
This commit reorders the coding tools such that they're doing what
the decoder does in reverse order. The very first thing the decoder
does is to decode M/S stereo if that's signalled, then prediction,
IS, and finally TNS and PNS in another function.
adjust_frame_information()'s application of IS and M/S was taken
out into two separate functions since prediction doesn't expect
to get the raw coefficients but rathe the coefficients at that
part of the encoding process.
The results show a much better PSNR when any combination of
Intensity Stereo, Mid/Side stereo and Prediction is used, which
is a sign of an increased encoder efficiency as well as the fact
that the decoder gets what it expects.
Otherwise, with only IS, PNS or prediction there are neither
regressions nor improvements except in the case of IS, which
now by itself (or with PNS) is less prone to artifacts. Enabling
M/S (using stereo_mode) as well will also reduce stereo artifacts
induced by IS, so in the very near future M/S may be enabled
by default.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
If the selected coder isn't twoloop, this commit temporarily
disables IS and PNS.
The problem is in the encode_window_bands_info() being confused
and setting invalid band_types for non-marked (normal) bands.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Since the changes made a few week ago (which were done more than a
month ago) the quality and stability of intensity stereo has been
notably good. There were some requests and wishes to have in on by
default and therefore it has been enabled. Should any regressions
arise changes will be made to preferably keep it operating rather
than just disabling it by default again.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
It has been in the current encoder in its current implementation
for quite some time now, so enable it by default. Will increase
quality at all bitrates, especially at low ones.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Purely a cosmetic change, most of the zeroing of encoder resources
should happen at the top of the main loop.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This commit reworks the TNS implementation to a hybrid between what
the specifications say, what the decoder does and what's the best
thing to do.
The filter application function was copied from the decoder and
modified such that it applies the inverse AR filter to the
coefficients. The LPC coefficients themselves are fed into the
same quantization expression that the specifications say should
be used however further processing is not done, instead they're
converted to the form that the decoder expects them to be in
and are sent off to the compute_lpc_coeffs function exactly the
way the decoder does. This function does all conversions and will
return the exact coefficients that the decoder will generate, which
are then applied to the coefficients.
Having the exact same coefficients on both the encoder and decoder
is a must since otherwise the entire sfb's over which the filter
is applied will be attenuated.
Despite this major rework, TNS might not work fine on some audio
types at very low bitrates (e.g. sub 90kbps) as it can attenuate
some coefficients too much. Users are advised to experiment with
TNS at higher bitrates if they wish to use this tool or simply
wait for the implementation to be improved.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Turns out autocorrelating more than 750 coefficients at once
will cause a segfault, despite there being enough space to
hold an entire frame of samples into the buffer.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
The order should never go above TNS_MAX_ORDER (and thus cause
the context to be reinitialized) but this is just in case.
Also fix a comparison, since the coefficients are zero-indexed.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Pulses are already on the way so expect to see the list
gone in the close future.
TNS is already of sufficiently high quality to be enabled
by default (but isn't yet, so you too can help by testing!).
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This commit abandons the way the specifications state to
quantize the coefficients, makes use of the new LPC float
functions and is much better.
The original way of converting non-normalized float samples
to int32_t which out LPC system expects was wrong and it was
wrong to assume the coefficients that are generated are also
valid. It was essentially a full garbage-in, garbage-out
system and it definitely shows when looking at spectrals
and listening. The high frequencies were very overattenuated.
The new LPC function performs the analysis directly.
The specifications state to quantize the coefficients into
four bit index values using an asin() function which of course
had to have ugly ternary operators because the function turns
negative if the coefficients are negative which when encoding
causes invalid bitstream to get generated.
This deviates from this by using the direct TNS tables, which
are fairly small since you only have 4 bits at most for index
values. The LPC values are directly quantized against the tables
and are then used to perform filtering after the requantization,
which simply fetches the array values.
The end result is that TNS works much better now and doesn't
attenuate anything but the actual signal, e.g. TNS removes
quantization errors and does it's job correctly now.
It might be enabled by default soon since it doesn't hurt and
helps reduce nastyness at low bitrates.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This commit completely alters the algorithm of prediction.
The original commit which introduced prediction was completely
incorrect to even remotely care about what the actual coefficients
contain or whether any options were enabled. Not my actual fault.
This commit treats prediction the way the decoder does and expects
to do: like lossy encryption. Everything related to prediction now
happens at the very end but just before quantization and encoding
of coefficients. On the decoder side, prediction happens before
anything has had a chance to even access the coefficients.
Also the original implementation had problems because it actually
touched the band_type of special bands which already had their
scalefactor indices marked and it's a wonder the asserion wasn't
triggered when transmitting those.
Overall, this now drastically increases audio quality and you should
think about enabling it if you don't plan on playing anything encoded
on really old low power ultra-embedded devices since they might not
support decoding of prediction or AAC-Main. Though the specifications
were written ages ago and as times change so do the FLOPS.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This was missed when the original commits were done. FF_PROFILE_UNKNOWN
is what's in avctx->profile when no audio profile is specified.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
When the encoder is ran without specifying -profile:a
the default avctx->profile value is -99 (FF_PROFILE_UKNOWN),
which used to be treated as AAC-LC.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This commit finalizes AAC-Main profile encoding support
by implementing all mandatory and optional tools available
in the specifications and current decoders.
The AAC-Main profile reqires that prediction support be
present (although decoders don't require it to be enabled)
for an encoder to be deemed capable of AAC-Main encoding,
as well as TNS, PNS and IS, all of which were implemented
with previous commits or earlier of this year.
Users are encouraged to test the new functionality using either
-profile:a aac_main or -aac_pred 1, the former of which will enable
the prediction option by default and the latter will change the
profile to AAC-Main. No other options shall be changed by enabling
either, it's currently up to the users to decide what's best.
The current implementation works best using M/S and/or IS,
so users are also welcome to enable both options and any
other options (TNS, PNS) for maximum quality.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This commit implements temporal noise shaping support in the
encoder, along with an -aac_tns option to toggle it on or off
(off by default for now). TNS will increase audio quality
and reduce quantization noise by applying a multitap FIR filter
across allowed coefficients and transmit side information to the
decoder so it could create an inverse filter.
Users are encouraged to test the new functionality by enabling
-aac_tns 1 during encoding.
No major bugs are observable at this time so after a while if no
new problems appear and if the current implementation is deemed
of high enough quality and stability it will be enabled by default,
possibly at the same time the encoder has its experimental flag
removed and becomes the standard aac encoder in ffmpeg.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This commit permits for the use of the Main profile
in encoding. The functionality of that profile will
be added in the commits following. By itself, this
commit does not alter anything.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This commit moves the intensity stereo implementation
out from aaccoder and into a separate file. This was
possible using the previous commits.
This commit also drastically improves the IS implementation
by making it phase invariant e.g. it will always choose the
best possible phase regardless of whether M/S coding is on
or most of the coefficients have identical phases.
This also increases the quality and reduces any distortions
introduced by enablind intensity stereo.
Users are encouraged to test it out using the -aac_is 1
parameter as it has always been.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This commit moves the quantizer to a separate header file.
This allows the quantizer to be used from a separate files outside
of aaccoder without having to put another function pointer and will
result in a slight speedup as the compiler can do more optimizations.
This is required for commits following.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This commit only creates and initializes an LTP
context which is needed for upcoming commits (TNS).
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This commit simply populates the table pointer which is needed
for upcoming commits (TNS, prediction, etc.). Copied from
the decoder.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This commit moves the resetting of special bands (above RESERVED_BT)
to the main frame encoding function rather than the way it was done
previously in their corresponding search_for_... functions.
The reason why special bands need to be reset is that while normal
bands get chosen for every frame by the coder (twoloop by default)
the coders do not touch any special sfbs and will therefore
make them persist throughout the file.
If we zero them out any bands left unmarked will be chosen by
the second part of the coder (the trellis function in aaccoder.c).
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This commit only changes the coding style to a saner way
of accessing coefficients (makes more sense to get the
memory address of a coefficients and start from there
rather than adding arbitrary numbers to offset a pointer).
Some compilers might detect an out of bounds access easier.
Also the way M/S and IS coefficients are calculated has been
changed, but should still have the same result (with the exception
that IS now applies from the normal coefficients rather than the
pristine ones, this is needed for upcoming commits).
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
As well as tables littered everywhere, functions were spread
out all across the encoder's files. This moves them to a single
place where they can be used by either the encoder's main files
or additional encoder files. Additionally, it changes the type
of some to 'inline' to enable us to simply put them in a header
file and possibly gain some speed due to compiler optimizations.
Signed-off-by: Claudio Freire <klaussfreire@gmail.com>
This commit moves any tables specific to the encoder from aacenc
and aaccoder to a separate file called 'aacenctab.c/.h'.
This was done as a clean up attempt as the encoder was filled with
tables pasted in between functions which made it confusing to follow
and track where each table and definition had been used.
This commit solves this by simply exporting the smaller tables out to
the aacenctab.h while the larger ones are compiled using aacenctab.c
and are referenced from the header file.
Signed-off-by: Claudio Freire <klaussfreire@gmail.com>
This commit adds a short description to the aac_coder option of the
AAC encoder in order to be consistent with the other options.
Generally, right now, the 'FAAC' method works fine with speech and
low broadband spectrum audio. 'Fast' is just as the name suggests.
'ANMR' still needs work and 'Twoloop', the default, works well with
every type of audio.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This commit removes a redundant argument from the functions in aaccoder.
The argument lambda was redundant as it was just a copy of s->lambda,
to which all functions have access to anyway. This cleans up the function
pointers a bit which is helpful as there are a lot of other search_for_*
functions under development and with them populated it gets messy.
Reviewed-by: Claudio Freire <klaussfreire@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This parameter can be used to inform the allocation code about how much
downsizing might occur, and can be used to optimize how to allocate the
packet
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Avoid clipping due to quantization noise to produce audible
artifacts, by detecting near-clipping signals and both attenuating
them a little and encoding escape-encoded bands (usually the
loudest) rounding towards zero instead of nearest, which tends to
decrease overall energy and thus clipping.
Currently fate tests measure numerical error so this change makes
tests using asynth (which are near clipping) report higher error
not less, because of window attenuation. Yet, they sound better,
not worse (albeit subtle, other samples aren't subtle at all).
Only measuring psychoacoustically weighted error would make for
a representative test, so that will be left for a future patch.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This commit moves the generation of ff_aac_pow34sf_tab[] out of the
encoder and into the table generator. The original commit log for
this table in 2011 actually mentions that it should be moved outside
but this never happened.
This is the first commit which cleans up the encoder a little.
Reviewed-by: Claudio Freire <klaussfreire@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
