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Commit Graph

75 Commits

Author SHA1 Message Date
Andreas Rheinhardt
c488ee6182 avcodec/aacenc: Move aac_pce_configs to its only user
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2022-08-05 03:28:45 +02:00
Anton Khirnov
494760f971 aac: convert to new channel layout API
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: James Almer <jamrial@gmail.com>
2022-03-15 09:42:39 -03:00
Andreas Rheinhardt
1be3d8a0cb avcodec/avcodec: Stop including channel_layout.h in avcodec.h
Also include channel_layout.h directly wherever used.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2021-07-22 11:14:31 +02:00
Anton Khirnov
e15371061d lavu/mem: move the DECLARE_ALIGNED macro family to mem_internal on next+1 bump
They are not properly namespaced and not intended for public use.
2021-01-01 14:14:57 +01:00
Rostislav Pehlivanov
fbf295e2bd aacenc: support extended channel layouts using PCEs
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.
2017-11-09 03:37:48 +00:00
James Almer
318778de9e Merge commit 'fd9212f2edfe9b107c3c08ba2df5fd2cba5ab9e3'
* commit 'fd9212f2edfe9b107c3c08ba2df5fd2cba5ab9e3':
  Mark some arrays that never change as const.

Merged-by: James Almer <jamrial@gmail.com>
2017-09-26 16:02:40 -03:00
Anton Khirnov
fd9212f2ed Mark some arrays that never change as const. 2017-02-01 10:42:59 +01:00
Rostislav Pehlivanov
0cf6853804 aacenc: quit when the audio queue reaches 0 rather than keeping track of empty frames
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>
2016-11-08 00:50:51 +00:00
Rostislav Pehlivanov
d2ae5f77c6 aacenc: add SIMD optimizations for abs_pow34 and quantization
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>
2016-10-18 21:41:18 +01:00
Rostislav Pehlivanov
230178dfe2 aacenc: use the decoder's lcg PRNG
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>
2016-10-12 11:15:49 +01:00
Reimar Döffinger
b91e376390 aacenc: use generational cache instead of resetting.
Approximately 11% faster transcoding from mp3 with
default settings.

Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
2016-03-08 23:56:51 +01:00
Rostislav Pehlivanov
6a505e955b aacenc: remove FAAC-like coder
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>
2016-01-20 16:56:53 +00:00
Rostislav Pehlivanov
4386f17bbd acenc: remove deprecated avctx->frame_bits use
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>
2015-12-18 14:28:40 +00:00
Rostislav Pehlivanov
ade31b9424 aacenc: switch to using the RNG from libavutil
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>
2015-12-14 18:53:09 +00:00
Rostislav Pehlivanov
27d23ae074 aacenc: add support for encoding files using Long Term Prediction
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.
2015-10-17 02:31:20 +01:00
Rostislav Pehlivanov
93e6b23c9f aacenc: shorten name of ff_aac_adjust_common_prediction
To keep it similar to the other functions which are all named *_pred.
2015-10-12 23:33:07 +01:00
Rostislav Pehlivanov
65f5b96dd8 aacenc: increase size of s->planar_samples[] from 6 to 8
Left out of last commit which added support for eight channel audio.
2015-10-12 23:25:45 +01:00
Rostislav Pehlivanov
0f4334df45 aacenc: add support for changing options based on a profile
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.
2015-10-12 16:57:56 +01:00
Claudio Freire
b629c67ddf AAC encoder: memoize quantize_band_cost
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.
2015-10-12 03:56:22 -03:00
Claudio Freire
01ecb7172b AAC encoder: Extensive improvements
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.
2015-10-11 17:29:50 -03:00
Claudio Freire
7ec74ae4aa AAC encoder: tweak rate-distortion logic
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).
2015-09-23 02:33:44 -03:00
Rostislav Pehlivanov
92aa3e7fb2 aacenc: copy PRNG from the decoder
Needed for the following PNS commits.

Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
2015-09-06 15:30:25 +01:00
Rostislav Pehlivanov
f3f6c6b928 aacenc_tns: rework coefficient quantization and filter application
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>
2015-09-01 06:44:07 +01:00
Rostislav Pehlivanov
f20b67173c aacenc_tns: rework the way coefficients are calculated
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>
2015-08-29 06:47:31 +01:00
Rostislav Pehlivanov
44ddee945a aacenc_pred: rework the way prediction is done
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>
2015-08-29 06:34:08 +01:00
Rostislav Pehlivanov
76b81b10d9 aacenc: implement the complete AAC-Main profile
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>
2015-08-21 19:38:05 +01:00
Rostislav Pehlivanov
a1c487e921 aacenc_tns: implement temporal noise shaping
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>
2015-08-21 19:27:38 +01:00
Rostislav Pehlivanov
eab12d072e aacenc: do not reject AAC-Main profile
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>
2015-08-21 19:20:22 +01:00
Rostislav Pehlivanov
43b378a0d3 aaccoder: move the quantization functions to a separate file
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>
2015-08-21 18:53:14 +01:00
Rostislav Pehlivanov
b47a1e5c5f aacenc: create and initialize an LTP context
This commit only creates and initializes an LTP
context which is needed for upcoming commits (TNS).

Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
2015-08-21 18:43:09 +01:00
Rostislav Pehlivanov
6d175158e9 aacenc: remove redundant argument from coder functions
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>
2015-08-01 02:54:35 +02:00
Claudio Freire
59216e0525 AAC Encoder: clipping avoidance
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>
2015-07-27 19:13:48 +02:00
Rostislav Pehlivanov
331c1e7494 aacenc: move the generation of ff_aac_pow34sf_tab[]
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>
2015-07-21 13:53:04 +02:00
Rostislav Pehlivanov
e8576dc8df aacenc: implement Intensity Stereo encoding support
This commit implements intensity stereo coding support
to the native aac encoder. This is a way to increase the efficiency
of the encoder by zeroing the right channel's spectral coefficients
(in a channel pair) and rederiving them in the decoder using information
from the scalefactor indices of special band types. This commit
confomrs to the official ISO 13818-7 specifications, although due to
their ambiguity certain deviations have been taken to ensure maximum
sound quality. This commit has been extensively tested and has shown
to not result in audiable audio artifacts unless in extreme cases.
This commit also adds an option, aac_is, which has the value of
0 by default. Intensity Stereo is part of the scalable aac profile
and is thus non-default.

The way IS coding works is that it rederives the right channel's
spectral coefficients from the left channel via the scalefactor
index values left in the right channel. Since an entire band's
spectral coefficients do not need to be coded, the encoder's
efficiency jumps up and it unzeroes some high frequency values
which it previously did not have enough bits to encode. That way
less information is lost than the information lost by rederiving
the spectral coefficients with some error. This is why the
filesize of files encoded with IS do not decrease significantly.
Users wishing that IS coding should reduce filesize are expected
to reduce their encoding bitrates appropriately.

This is V2 of the commit. The old version did not mark ms_mask as
0 since M/S and IS coding are incompactible, which resulted in
distortions with M/S coding enabled. This version also improves
phase detection by measuring it for every spectral coefficient in
the band and using a simple majority rule to determine whether the
coefficients are in or out of phase. Also, the energy values per
spectral coefficient were changed as to reflect the
official specifications.

Reviewed-by: Claudio Freire <klaussfreire@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-07-05 16:59:26 +02:00
Rostislav Pehlivanov
38fd4c2e66 aaccoder: add a new perceptual noise substitution implementation
This commit finalizes the PNS implementation previously added to the encoder
by moving it to a seperate function search_for_pns() and thus making it
coder-generic. This new implementation makes use of the spread field of
the psy bands and the lambda quality feedback paremeter. The spread of the
spectrum in a band prevents PNS from being used excessively and thus preserve
more phase information in high frequencies.  The lambda parameter allows
the number of PNS-marked bands to vary based on the lambda parameter and the
amount of bits available, making better choices on which bands are to be marked
as noise. Comparisons with the previous PNS implementation can be found
here: https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/

This is V2 of the patch, the changes from the previous version being that this
version uses the new band->spread metric from aacpsy and normalizes the
energy using the group size. These changes were suggested by Claudio Freire
on the mailing list. Another change is the use of lambda to alter the
frequency threshold. This change makes the actual threshold frequencies
vary between +-2Khz of what's specified, depending on frame encoding performance.

Reviewed-by: Claudio Freire <klaussfreire@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-07-05 16:39:06 +02:00
Rostislav Pehlivanov
e06578e392 aacenc: use the new function for setting special band scalefactor indices
This commit enables the function added with commit 7c10b87 and uses that
new function for setting any special scalefactor indices. This commit does
not change the behaviour of the encoder since no bands are being marked as
either NOISE_BT(due to the previous PNS implementation removed in the
previous commit) or INTENSITY_BT2/INTENSITY_BT.

Reviewed-by: Claudio Freire <klaussfreire@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-07-05 16:36:38 +02:00
Rostislav Pehlivanov
c5d4f87e81 aaccoder: Implement Perceptual Noise Substitution for AAC
This commit implements the perceptual noise substitution AAC extension. This is a proof of concept
implementation, and as such, is not enabled by default. This is the fourth revision of this patch,
made after some problems were noted out. Any changes made since the previous revisions have been indicated.

In order to extend the encoder to use an additional codebook, the array holding each codebook has been
modified with two additional entries - 13 for the NOISE_BT codebook and 12 which has a placeholder function.
The cost system was modified to skip the 12th entry using an array to map the input and outputs it has. It
also does not accept using the 13th codebook for any band which is not marked as containing noise, thereby
restricting its ability to arbitrarily choose it for bands. The use of arrays allows the system to be easily
extended to allow for intensity stereo encoding, which uses additional codebooks.

The 12th entry in the codebook function array points to a function which stops the execution of the program
by calling an assert with an always 'false' argument. It was pointed out in an email discussion with
Claudio Freire that having a 'NULL' entry can result in unexpected behaviour and could be used as
a security hole. There is no danger of this function being called during encoding due to the codebook maps introduced.

Another change from version 1 of the patch is the addition of an argument to the encoder, '-aac_pns' to
enable and disable the PNS. This currently defaults to disable the PNS, as it is experimental.
The switch will be removed in the future, when the algorithm to select noise bands has been improved.
The current algorithm simply compares the energy to the threshold (multiplied by a constant) to determine
noise, however the FFPsyBand structure contains other useful figures to determine which bands carry noise more accurately.

Some of the sample files provided triggered an assertion when the parameter to tune the threshold was set to
a value of '2.2'. Claudio Freire reported the problem's source could be in the range of the scalefactor
indices for noise and advised to measure the minimal index and clip anything above the maximum allowed
value. This has been implemented and all the files which used to trigger the asserion now encode without error.

The third revision of the problem also removes unneded variabes and comparisons. All of them were
redundant and were of little use for when the PNS implementation would be extended.

The fourth revision moved the clipping of the noise scalefactors outside the second loop of the two-loop
algorithm in order to prevent their redundant calculations. Also, freq_mult has been changed to a float
variable due to the fact that rounding errors can prove to be a problem at low frequencies.
Considerations were taken whether the entire expression could be evaluated inside the expression
, but in the end it was decided that it would be for the best if just the type of the variable were
to change. Claudio Freire reported the two problems. There is no change of functionality
(except for low sampling frequencies) so the spectral demonstrations at the end of this commit's message were not updated.

Finally, the way energy values are converted to scalefactor indices has changed since the first commit,
as per the suggestion of Claudio Freire. This may still have some drawbacks, but unlike the first commit
it works without having redundant offsets and outputs what the decoder expects to have, in terms of the
ranges of the scalefactor indices.

Some spectral comparisons: https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/Original.png (original),
https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/PNS_NO.png (encoded without PNS),
https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/PNS1.2.png (encoded with PNS, const = 1.2),
https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/Difference1.png (spectral difference).
The constant is the value which multiplies the threshold when it gets compared to the energy, larger
values means more noise will be substituded by PNS values. Example when const = 2.2:
https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/PNS_2.2.png

Reviewed-by: Claudio Freire <klaussfreire@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-04-15 19:59:44 +02:00
Michael Niedermayer
14285c3331 avcodec/aacenc: Use avpriv_float_dsp_alloc()
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-11-29 18:58:13 +01:00
Michael Niedermayer
16837f9846 avcodec/aacenc: use enum for aac coder.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-09-12 17:25:30 +02:00
Timothy Gu
4bd910d83d aacenc: add AAC_CODER_(FAAC|ANMR|etc.) macros
Signed-off-by: Timothy Gu <timothygu99@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-09-12 17:22:36 +02:00
Bojan Zivkovic
26f3924d78 mips: Optimization of AAC coefficients encoder functions
Signed-off-by: Bojan Zivkovic <bojan@mips.com>
Reviewed-by: Nedeljko Babic <Nedeljko.Babic@imgtec.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-03-20 12:34:37 +01:00
Michael Niedermayer
a984efd104 Merge commit 'c242bbd8b6939507a1a6fb64101b0553d92d303f'
* commit 'c242bbd8b6939507a1a6fb64101b0553d92d303f':
  Remove unnecessary dsputil.h #includes

Conflicts:
	libavcodec/ffv1.c
	libavcodec/h261dec.c
	libavcodec/h261enc.c
	libavcodec/h264pred.c
	libavcodec/lpc.h
	libavcodec/mjpegdec.c
	libavcodec/rectangle.h
	libavcodec/x86/idct_sse2_xvid.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2013-02-26 13:05:10 +01:00
Diego Biurrun
c242bbd8b6 Remove unnecessary dsputil.h #includes 2013-02-26 00:51:34 +01:00
Michael Niedermayer
6e6e170898 Merge commit '42d324694883cdf1fff1612ac70fa403692a1ad4'
* commit '42d324694883cdf1fff1612ac70fa403692a1ad4':
  floatdsp: move vector_fmul_reverse from dsputil to avfloatdsp.

Conflicts:
	libavcodec/arm/dsputil_init_vfp.c
	libavcodec/arm/dsputil_vfp.S
	libavcodec/dsputil.c
	libavcodec/ppc/float_altivec.c
	libavcodec/x86/dsputil.asm
	libavutil/x86/float_dsp.asm

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2013-01-23 14:04:50 +01:00
Ronald S. Bultje
42d3246948 floatdsp: move vector_fmul_reverse from dsputil to avfloatdsp.
Now, nellymoserenc and aacenc no longer depends on dsputil. Independent
of this patch, wmaprodec also does not depend on dsputil, so I removed
it from there also.
2013-01-22 11:55:42 -08:00
Michael Niedermayer
7e22514d98 Merge remote-tracking branch 'qatar/master'
* qatar/master:
  float_dsp: ppc: add a separate header for Altivec function prototypes
  ARM: fix float_dsp breakage from d5a7229
  Add a float DSP framework to libavutil
  PPC: Move types_altivec.h and util_altivec.h from libavcodec to libavutil
  ARM: Move asm.S from libavcodec to libavutil
  vc1dsp: mark put/avg_vc1_mspel_mc() always_inline

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-06-08 23:59:09 +02:00
Justin Ruggles
d5a7229ba4 Add a float DSP framework to libavutil
Move vector_fmul() from DSPContext to AVFloatDSPContext.
2012-06-08 13:14:38 -04:00
Michael Niedermayer
967facb695 Merge remote-tracking branch 'qatar/master'
* qatar/master: (26 commits)
  adxenc: use AVCodec.encode2()
  adxenc: Use the AVFrame in ADXContext for coded_frame
  indeo4: fix out-of-bounds function call.
  configure: Restructure help output.
  configure: Internal-only components should not be command-line selectable.
  vorbisenc: use AVCodec.encode2()
  libvorbis: use AVCodec.encode2()
  libopencore-amrnbenc: use AVCodec.encode2()
  ra144enc: use AVCodec.encode2()
  nellymoserenc: use AVCodec.encode2()
  roqaudioenc: use AVCodec.encode2()
  libspeex: use AVCodec.encode2()
  libvo_amrwbenc: use AVCodec.encode2()
  libvo_aacenc: use AVCodec.encode2()
  wmaenc: use AVCodec.encode2()
  mpegaudioenc: use AVCodec.encode2()
  libmp3lame: use AVCodec.encode2()
  libgsmenc: use AVCodec.encode2()
  libfaac: use AVCodec.encode2()
  g726enc: use AVCodec.encode2()
  ...

Conflicts:
	configure
	libavcodec/Makefile
	libavcodec/ac3enc.c
	libavcodec/adxenc.c
	libavcodec/libgsm.c
	libavcodec/libvorbis.c
	libavcodec/vorbisenc.c
	libavcodec/wmaenc.c
	tests/ref/acodec/g722
	tests/ref/lavf/asf
	tests/ref/lavf/ffm
	tests/ref/lavf/mkv
	tests/ref/lavf/mpg
	tests/ref/lavf/rm
	tests/ref/lavf/ts
	tests/ref/seek/lavf_asf
	tests/ref/seek/lavf_ffm
	tests/ref/seek/lavf_mkv
	tests/ref/seek/lavf_mpg
	tests/ref/seek/lavf_rm
	tests/ref/seek/lavf_ts

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-03-22 00:40:11 +01:00
Justin Ruggles
ad95307f92 aacenc: use AVCodec.encode2() 2012-03-20 18:46:49 -04:00
Michael Niedermayer
0bb57f8bf0 Merge remote-tracking branch 'qatar/master'
* qatar/master:
  Remove ffmpeg.
  aacenc: Simplify windowing
  aacenc: Move saved overlap samples to the beginning of the same buffer as incoming samples.
  aacenc: Deinterleave input samples before processing.
  aacenc: Store channel count in AACEncContext.
  aacenc: Move Q^3/4 calculation to it's own table
  aacenc: Request normalized float samples instead of converting s16 samples to float.
  aacpsy: Replace an if with FFMAX in LAME windowing.
  aacenc: cosmetics, replace 'rd' with 'bits' in codebook_trellis_rate to make it more clear what is being calculated.
  aacpsy: cosmetics, change a FIXME to a NOTE about subshort comparisons
  aacenc: cosmetics: move init() and end() to the bottom of the file.
  aacenc: aac_encode_init() cleanup
  XWD encoder and decoder
  vc1: don't read the interpfrm and bfraction elements for interlaced frames
  mxfdec: fix memleak on mxf_read_close()
  westwood: split the AUD and VQA demuxers into separate files.

Conflicts:
	.gitignore
	Changelog
	Makefile
	configure
	doc/ffmpeg.texi
	ffmpeg.c
	libavcodec/Makefile
	libavcodec/aacenc.c
	libavcodec/allcodecs.c
	libavcodec/avcodec.h
	libavcodec/version.h
	libavformat/Makefile
	libavformat/img2.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-24 02:41:53 +01:00