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FFmpeg/libavcodec/aaccoder.c

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/*
* AAC coefficients encoder
* Copyright (C) 2008-2009 Konstantin Shishkov
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* AAC coefficients encoder
*/
/***********************************
* TODOs:
* speedup quantizer selection
* add sane pulse detection
***********************************/
#include "libavutil/libm.h" // brought forward to work around cygwin header breakage
#include <float.h>
#include "libavutil/mathematics.h"
#include "avcodec.h"
#include "put_bits.h"
#include "aac.h"
#include "aacenc.h"
#include "aactab.h"
#include "aacenctab.h"
#include "aacenc_utils.h"
#include "aacenc_quantization.h"
#include "aac_tablegen_decl.h"
#include "aacenc_is.h"
#include "aacenc_tns.h"
#include "aacenc_pred.h"
#include "libavcodec/aaccoder_twoloop.h"
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 13:18:42 +02:00
/** Frequency in Hz for lower limit of noise substitution **/
#define NOISE_LOW_LIMIT 4000
/* Parameter of f(x) = a*(lambda/100), defines the maximum fourier spread
* beyond which no PNS is used (since the SFBs contain tone rather than noise) */
#define NOISE_SPREAD_THRESHOLD 0.5073f
/* Parameter of f(x) = a*(100/lambda), defines how much PNS is allowed to
* replace low energy non zero bands */
#define NOISE_LAMBDA_REPLACE 1.948f
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 13:18:42 +02:00
#include "libavcodec/aaccoder_trellis.h"
/**
* structure used in optimal codebook search
*/
typedef struct BandCodingPath {
int prev_idx; ///< pointer to the previous path point
float cost; ///< path cost
int run;
} BandCodingPath;
/**
* Encode band info for single window group bands.
*/
static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce,
int win, int group_len, const float lambda)
{
BandCodingPath path[120][CB_TOT_ALL];
2011-06-01 19:26:27 +03:00
int w, swb, cb, start, size;
int i, j;
const int max_sfb = sce->ics.max_sfb;
const int run_bits = sce->ics.num_windows == 1 ? 5 : 3;
const int run_esc = (1 << run_bits) - 1;
int idx, ppos, count;
int stackrun[120], stackcb[120], stack_len;
float next_minrd = INFINITY;
int next_mincb = 0;
abs_pow34_v(s->scoefs, sce->coeffs, 1024);
start = win*128;
for (cb = 0; cb < CB_TOT_ALL; cb++) {
path[0][cb].cost = 0.0f;
path[0][cb].prev_idx = -1;
path[0][cb].run = 0;
}
for (swb = 0; swb < max_sfb; swb++) {
size = sce->ics.swb_sizes[swb];
if (sce->zeroes[win*16 + swb]) {
for (cb = 0; cb < CB_TOT_ALL; cb++) {
path[swb+1][cb].prev_idx = cb;
path[swb+1][cb].cost = path[swb][cb].cost;
path[swb+1][cb].run = path[swb][cb].run + 1;
}
} else {
float minrd = next_minrd;
int mincb = next_mincb;
next_minrd = INFINITY;
next_mincb = 0;
for (cb = 0; cb < CB_TOT_ALL; cb++) {
float cost_stay_here, cost_get_here;
float rd = 0.0f;
if (cb >= 12 && sce->band_type[win*16+swb] < aac_cb_out_map[cb] ||
cb < aac_cb_in_map[sce->band_type[win*16+swb]] && sce->band_type[win*16+swb] > aac_cb_out_map[cb]) {
path[swb+1][cb].prev_idx = -1;
path[swb+1][cb].cost = INFINITY;
path[swb+1][cb].run = path[swb][cb].run + 1;
continue;
}
for (w = 0; w < group_len; w++) {
FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[(win+w)*16+swb];
rd += quantize_band_cost(s, &sce->coeffs[start + w*128],
&s->scoefs[start + w*128], size,
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 13:18:42 +02:00
sce->sf_idx[(win+w)*16+swb], aac_cb_out_map[cb],
lambda / band->threshold, INFINITY, NULL, 0);
}
cost_stay_here = path[swb][cb].cost + rd;
cost_get_here = minrd + rd + run_bits + 4;
if ( run_value_bits[sce->ics.num_windows == 8][path[swb][cb].run]
!= run_value_bits[sce->ics.num_windows == 8][path[swb][cb].run+1])
cost_stay_here += run_bits;
if (cost_get_here < cost_stay_here) {
path[swb+1][cb].prev_idx = mincb;
path[swb+1][cb].cost = cost_get_here;
path[swb+1][cb].run = 1;
} else {
path[swb+1][cb].prev_idx = cb;
path[swb+1][cb].cost = cost_stay_here;
path[swb+1][cb].run = path[swb][cb].run + 1;
}
if (path[swb+1][cb].cost < next_minrd) {
next_minrd = path[swb+1][cb].cost;
next_mincb = cb;
}
}
}
start += sce->ics.swb_sizes[swb];
}
//convert resulting path from backward-linked list
stack_len = 0;
idx = 0;
for (cb = 1; cb < CB_TOT_ALL; cb++)
if (path[max_sfb][cb].cost < path[max_sfb][idx].cost)
idx = cb;
ppos = max_sfb;
while (ppos > 0) {
av_assert1(idx >= 0);
cb = idx;
stackrun[stack_len] = path[ppos][cb].run;
stackcb [stack_len] = cb;
idx = path[ppos-path[ppos][cb].run+1][cb].prev_idx;
ppos -= path[ppos][cb].run;
stack_len++;
}
//perform actual band info encoding
start = 0;
for (i = stack_len - 1; i >= 0; i--) {
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 13:18:42 +02:00
cb = aac_cb_out_map[stackcb[i]];
put_bits(&s->pb, 4, cb);
count = stackrun[i];
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 13:18:42 +02:00
memset(sce->zeroes + win*16 + start, !cb, count);
//XXX: memset when band_type is also uint8_t
for (j = 0; j < count; j++) {
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 13:18:42 +02:00
sce->band_type[win*16 + start] = cb;
start++;
}
while (count >= run_esc) {
put_bits(&s->pb, run_bits, run_esc);
count -= run_esc;
}
put_bits(&s->pb, run_bits, count);
}
}
typedef struct TrellisPath {
float cost;
int prev;
} TrellisPath;
#define TRELLIS_STAGES 121
#define TRELLIS_STATES (SCALE_MAX_DIFF+1)
static void set_special_band_scalefactors(AACEncContext *s, SingleChannelElement *sce)
{
int w, g, start = 0;
int minscaler_n = sce->sf_idx[0], minscaler_i = sce->sf_idx[0];
int bands = 0;
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
start = 0;
for (g = 0; g < sce->ics.num_swb; g++) {
if (sce->band_type[w*16+g] == INTENSITY_BT || sce->band_type[w*16+g] == INTENSITY_BT2) {
sce->sf_idx[w*16+g] = av_clip(roundf(log2f(sce->is_ener[w*16+g])*2), -155, 100);
minscaler_i = FFMIN(minscaler_i, sce->sf_idx[w*16+g]);
bands++;
} else if (sce->band_type[w*16+g] == NOISE_BT) {
sce->sf_idx[w*16+g] = av_clip(3+ceilf(log2f(sce->pns_ener[w*16+g])*2), -100, 155);
minscaler_n = FFMIN(minscaler_n, sce->sf_idx[w*16+g]);
bands++;
}
start += sce->ics.swb_sizes[g];
}
}
if (!bands)
return;
/* Clip the scalefactor indices */
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
for (g = 0; g < sce->ics.num_swb; g++) {
if (sce->band_type[w*16+g] == INTENSITY_BT || sce->band_type[w*16+g] == INTENSITY_BT2) {
sce->sf_idx[w*16+g] = av_clip(sce->sf_idx[w*16+g], minscaler_i, minscaler_i + SCALE_MAX_DIFF);
} else if (sce->band_type[w*16+g] == NOISE_BT) {
sce->sf_idx[w*16+g] = av_clip(sce->sf_idx[w*16+g], minscaler_n, minscaler_n + SCALE_MAX_DIFF);
}
}
}
}
static void search_for_quantizers_anmr(AVCodecContext *avctx, AACEncContext *s,
SingleChannelElement *sce,
const float lambda)
{
int q, w, w2, g, start = 0;
int i, j;
int idx;
TrellisPath paths[TRELLIS_STAGES][TRELLIS_STATES];
int bandaddr[TRELLIS_STAGES];
int minq;
float mincost;
float q0f = FLT_MAX, q1f = 0.0f, qnrgf = 0.0f;
int q0, q1, qcnt = 0;
for (i = 0; i < 1024; i++) {
float t = fabsf(sce->coeffs[i]);
if (t > 0.0f) {
q0f = FFMIN(q0f, t);
q1f = FFMAX(q1f, t);
qnrgf += t*t;
qcnt++;
}
}
if (!qcnt) {
memset(sce->sf_idx, 0, sizeof(sce->sf_idx));
memset(sce->zeroes, 1, sizeof(sce->zeroes));
return;
}
//minimum scalefactor index is when minimum nonzero coefficient after quantizing is not clipped
q0 = coef2minsf(q0f);
//maximum scalefactor index is when maximum coefficient after quantizing is still not zero
q1 = coef2maxsf(q1f);
if (q1 - q0 > 60) {
int q0low = q0;
int q1high = q1;
//minimum scalefactor index is when maximum nonzero coefficient after quantizing is not clipped
int qnrg = av_clip_uint8(log2f(sqrtf(qnrgf/qcnt))*4 - 31 + SCALE_ONE_POS - SCALE_DIV_512);
q1 = qnrg + 30;
q0 = qnrg - 30;
if (q0 < q0low) {
q1 += q0low - q0;
q0 = q0low;
} else if (q1 > q1high) {
q0 -= q1 - q1high;
q1 = q1high;
}
}
for (i = 0; i < TRELLIS_STATES; i++) {
paths[0][i].cost = 0.0f;
paths[0][i].prev = -1;
}
for (j = 1; j < TRELLIS_STAGES; j++) {
for (i = 0; i < TRELLIS_STATES; i++) {
paths[j][i].cost = INFINITY;
paths[j][i].prev = -2;
}
}
idx = 1;
abs_pow34_v(s->scoefs, sce->coeffs, 1024);
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
start = w*128;
for (g = 0; g < sce->ics.num_swb; g++) {
const float *coefs = &sce->coeffs[start];
float qmin, qmax;
int nz = 0;
bandaddr[idx] = w * 16 + g;
qmin = INT_MAX;
qmax = 0.0f;
for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[(w+w2)*16+g];
if (band->energy <= band->threshold || band->threshold == 0.0f) {
sce->zeroes[(w+w2)*16+g] = 1;
continue;
}
sce->zeroes[(w+w2)*16+g] = 0;
nz = 1;
for (i = 0; i < sce->ics.swb_sizes[g]; i++) {
float t = fabsf(coefs[w2*128+i]);
if (t > 0.0f)
qmin = FFMIN(qmin, t);
qmax = FFMAX(qmax, t);
}
}
if (nz) {
int minscale, maxscale;
float minrd = INFINITY;
float maxval;
//minimum scalefactor index is when minimum nonzero coefficient after quantizing is not clipped
minscale = coef2minsf(qmin);
//maximum scalefactor index is when maximum coefficient after quantizing is still not zero
maxscale = coef2maxsf(qmax);
minscale = av_clip(minscale - q0, 0, TRELLIS_STATES - 1);
maxscale = av_clip(maxscale - q0, 0, TRELLIS_STATES);
maxval = find_max_val(sce->ics.group_len[w], sce->ics.swb_sizes[g], s->scoefs+start);
for (q = minscale; q < maxscale; q++) {
float dist = 0;
int cb = find_min_book(maxval, sce->sf_idx[w*16+g]);
for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[(w+w2)*16+g];
dist += quantize_band_cost(s, coefs + w2*128, s->scoefs + start + w2*128, sce->ics.swb_sizes[g],
q + q0, cb, lambda / band->threshold, INFINITY, NULL, 0);
}
minrd = FFMIN(minrd, dist);
for (i = 0; i < q1 - q0; i++) {
float cost;
cost = paths[idx - 1][i].cost + dist
+ ff_aac_scalefactor_bits[q - i + SCALE_DIFF_ZERO];
if (cost < paths[idx][q].cost) {
paths[idx][q].cost = cost;
paths[idx][q].prev = i;
}
}
}
} else {
for (q = 0; q < q1 - q0; q++) {
paths[idx][q].cost = paths[idx - 1][q].cost + 1;
paths[idx][q].prev = q;
}
}
sce->zeroes[w*16+g] = !nz;
start += sce->ics.swb_sizes[g];
idx++;
}
}
idx--;
mincost = paths[idx][0].cost;
minq = 0;
for (i = 1; i < TRELLIS_STATES; i++) {
if (paths[idx][i].cost < mincost) {
mincost = paths[idx][i].cost;
minq = i;
}
}
while (idx) {
sce->sf_idx[bandaddr[idx]] = minq + q0;
minq = paths[idx][minq].prev;
idx--;
}
//set the same quantizers inside window groups
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
for (g = 0; g < sce->ics.num_swb; g++)
for (w2 = 1; w2 < sce->ics.group_len[w]; w2++)
sce->sf_idx[(w+w2)*16+g] = sce->sf_idx[w*16+g];
}
static void search_for_quantizers_faac(AVCodecContext *avctx, AACEncContext *s,
SingleChannelElement *sce,
const float lambda)
{
int start = 0, i, w, w2, g;
float uplim[128], maxq[128];
int minq, maxsf;
float distfact = ((sce->ics.num_windows > 1) ? 85.80 : 147.84) / lambda;
int last = 0, lastband = 0, curband = 0;
float avg_energy = 0.0;
if (sce->ics.num_windows == 1) {
start = 0;
for (i = 0; i < 1024; i++) {
if (i - start >= sce->ics.swb_sizes[curband]) {
start += sce->ics.swb_sizes[curband];
curband++;
}
if (sce->coeffs[i]) {
avg_energy += sce->coeffs[i] * sce->coeffs[i];
last = i;
lastband = curband;
}
}
} else {
for (w = 0; w < 8; w++) {
const float *coeffs = &sce->coeffs[w*128];
curband = start = 0;
for (i = 0; i < 128; i++) {
if (i - start >= sce->ics.swb_sizes[curband]) {
start += sce->ics.swb_sizes[curband];
curband++;
}
if (coeffs[i]) {
avg_energy += coeffs[i] * coeffs[i];
last = FFMAX(last, i);
lastband = FFMAX(lastband, curband);
}
}
}
}
last++;
avg_energy /= last;
if (avg_energy == 0.0f) {
for (i = 0; i < FF_ARRAY_ELEMS(sce->sf_idx); i++)
sce->sf_idx[i] = SCALE_ONE_POS;
return;
}
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
start = w*128;
for (g = 0; g < sce->ics.num_swb; g++) {
float *coefs = &sce->coeffs[start];
const int size = sce->ics.swb_sizes[g];
int start2 = start, end2 = start + size, peakpos = start;
float maxval = -1, thr = 0.0f, t;
maxq[w*16+g] = 0.0f;
if (g > lastband) {
maxq[w*16+g] = 0.0f;
start += size;
for (w2 = 0; w2 < sce->ics.group_len[w]; w2++)
memset(coefs + w2*128, 0, sizeof(coefs[0])*size);
continue;
}
for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
for (i = 0; i < size; i++) {
float t = coefs[w2*128+i]*coefs[w2*128+i];
maxq[w*16+g] = FFMAX(maxq[w*16+g], fabsf(coefs[w2*128 + i]));
thr += t;
if (sce->ics.num_windows == 1 && maxval < t) {
maxval = t;
peakpos = start+i;
}
}
}
if (sce->ics.num_windows == 1) {
start2 = FFMAX(peakpos - 2, start2);
end2 = FFMIN(peakpos + 3, end2);
} else {
start2 -= start;
end2 -= start;
}
start += size;
thr = pow(thr / (avg_energy * (end2 - start2)), 0.3 + 0.1*(lastband - g) / lastband);
t = 1.0 - (1.0 * start2 / last);
uplim[w*16+g] = distfact / (1.4 * thr + t*t*t + 0.075);
}
}
memset(sce->sf_idx, 0, sizeof(sce->sf_idx));
abs_pow34_v(s->scoefs, sce->coeffs, 1024);
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
start = w*128;
for (g = 0; g < sce->ics.num_swb; g++) {
const float *coefs = &sce->coeffs[start];
const float *scaled = &s->scoefs[start];
const int size = sce->ics.swb_sizes[g];
int scf, prev_scf, step;
int min_scf = -1, max_scf = 256;
float curdiff;
if (maxq[w*16+g] < 21.544) {
sce->zeroes[w*16+g] = 1;
start += size;
continue;
}
sce->zeroes[w*16+g] = 0;
scf = prev_scf = av_clip(SCALE_ONE_POS - SCALE_DIV_512 - log2f(1/maxq[w*16+g])*16/3, 60, 218);
for (;;) {
float dist = 0.0f;
int quant_max;
for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
int b;
dist += quantize_band_cost(s, coefs + w2*128,
scaled + w2*128,
sce->ics.swb_sizes[g],
scf,
ESC_BT,
lambda,
INFINITY,
&b,
0);
dist -= b;
}
dist *= 1.0f / 512.0f / lambda;
quant_max = quant(maxq[w*16+g], ff_aac_pow2sf_tab[POW_SF2_ZERO - scf + SCALE_ONE_POS - SCALE_DIV_512], ROUND_STANDARD);
if (quant_max >= 8191) { // too much, return to the previous quantizer
sce->sf_idx[w*16+g] = prev_scf;
break;
}
prev_scf = scf;
curdiff = fabsf(dist - uplim[w*16+g]);
if (curdiff <= 1.0f)
step = 0;
else
step = log2f(curdiff);
if (dist > uplim[w*16+g])
step = -step;
scf += step;
scf = av_clip_uint8(scf);
step = scf - prev_scf;
if (FFABS(step) <= 1 || (step > 0 && scf >= max_scf) || (step < 0 && scf <= min_scf)) {
sce->sf_idx[w*16+g] = av_clip(scf, min_scf, max_scf);
break;
}
if (step > 0)
min_scf = prev_scf;
else
max_scf = prev_scf;
}
start += size;
}
}
minq = sce->sf_idx[0] ? sce->sf_idx[0] : INT_MAX;
for (i = 1; i < 128; i++) {
if (!sce->sf_idx[i])
sce->sf_idx[i] = sce->sf_idx[i-1];
else
minq = FFMIN(minq, sce->sf_idx[i]);
}
if (minq == INT_MAX)
minq = 0;
minq = FFMIN(minq, SCALE_MAX_POS);
maxsf = FFMIN(minq + SCALE_MAX_DIFF, SCALE_MAX_POS);
for (i = 126; i >= 0; i--) {
if (!sce->sf_idx[i])
sce->sf_idx[i] = sce->sf_idx[i+1];
sce->sf_idx[i] = av_clip(sce->sf_idx[i], minq, maxsf);
}
}
static void search_for_quantizers_fast(AVCodecContext *avctx, AACEncContext *s,
SingleChannelElement *sce,
const float lambda)
{
2011-06-01 19:26:27 +03:00
int i, w, w2, g;
int minq = 255;
memset(sce->sf_idx, 0, sizeof(sce->sf_idx));
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
for (g = 0; g < sce->ics.num_swb; g++) {
for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[(w+w2)*16+g];
if (band->energy <= band->threshold) {
sce->sf_idx[(w+w2)*16+g] = 218;
sce->zeroes[(w+w2)*16+g] = 1;
} else {
sce->sf_idx[(w+w2)*16+g] = av_clip(SCALE_ONE_POS - SCALE_DIV_512 + log2f(band->threshold), 80, 218);
sce->zeroes[(w+w2)*16+g] = 0;
}
minq = FFMIN(minq, sce->sf_idx[(w+w2)*16+g]);
}
}
}
for (i = 0; i < 128; i++) {
sce->sf_idx[i] = 140;
//av_clip(sce->sf_idx[i], minq, minq + SCALE_MAX_DIFF - 1);
}
//set the same quantizers inside window groups
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
for (g = 0; g < sce->ics.num_swb; g++)
for (w2 = 1; w2 < sce->ics.group_len[w]; w2++)
sce->sf_idx[(w+w2)*16+g] = sce->sf_idx[w*16+g];
}
static void search_for_pns(AACEncContext *s, AVCodecContext *avctx, SingleChannelElement *sce)
{
FFPsyBand *band;
int w, g, w2, i;
float *PNS = &s->scoefs[0*128], *PNS34 = &s->scoefs[1*128];
float *NOR34 = &s->scoefs[3*128];
const float lambda = s->lambda;
const float freq_mult = avctx->sample_rate/(1024.0f/sce->ics.num_windows)/2.0f;
const float thr_mult = NOISE_LAMBDA_REPLACE*(100.0f/lambda);
const float spread_threshold = NOISE_SPREAD_THRESHOLD*FFMAX(0.5f, lambda/100.f);
memcpy(sce->band_alt, sce->band_type, sizeof(sce->band_type));
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
int wstart = w*128;
for (g = 0; g < sce->ics.num_swb; g++) {
int noise_sfi;
float dist1 = 0.0f, dist2 = 0.0f, noise_amp;
float pns_energy = 0.0f, pns_tgt_energy, energy_ratio, dist_thresh;
float sfb_energy = 0.0f, threshold = 0.0f, spread = 0.0f;
const int start = wstart+sce->ics.swb_offset[g];
const float freq = (start-wstart)*freq_mult;
const float freq_boost = FFMAX(0.88f*freq/NOISE_LOW_LIMIT, 1.0f);
if (freq < NOISE_LOW_LIMIT || avctx->cutoff && freq >= avctx->cutoff)
continue;
for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
band = &s->psy.ch[s->cur_channel].psy_bands[(w+w2)*16+g];
sfb_energy += band->energy;
spread += band->spread;
threshold += band->threshold;
}
/* Ramps down at ~8000Hz and loosens the dist threshold */
dist_thresh = FFMIN(2.5f*NOISE_LOW_LIMIT/freq, 2.5f);
/* zero and energy close to threshold usually means hole avoidance,
* we do want to remain avoiding holes with PNS
*/
if (((sce->zeroes[w*16+g] || !sce->band_alt[w*16+g]) && sfb_energy < threshold*sqrtf(1.5f/freq_boost)) || spread < spread_threshold ||
(sce->band_alt[w*16+g] && sfb_energy > threshold*thr_mult*freq_boost)) {
sce->pns_ener[w*16+g] = sfb_energy;
continue;
}
pns_tgt_energy = sfb_energy*spread*spread/sce->ics.group_len[w];
noise_sfi = av_clip(roundf(log2f(pns_tgt_energy)*2), -100, 155); /* Quantize */
noise_amp = -ff_aac_pow2sf_tab[noise_sfi + POW_SF2_ZERO]; /* Dequantize */
for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
float band_energy, scale, pns_senergy;
const int start_c = (w+w2)*128+sce->ics.swb_offset[g];
band = &s->psy.ch[s->cur_channel].psy_bands[(w+w2)*16+g];
for (i = 0; i < sce->ics.swb_sizes[g]; i++)
PNS[i] = s->random_state = lcg_random(s->random_state);
band_energy = s->fdsp->scalarproduct_float(PNS, PNS, sce->ics.swb_sizes[g]);
scale = noise_amp/sqrtf(band_energy);
s->fdsp->vector_fmul_scalar(PNS, PNS, scale, sce->ics.swb_sizes[g]);
pns_senergy = s->fdsp->scalarproduct_float(PNS, PNS, sce->ics.swb_sizes[g]);
pns_energy += pns_senergy;
abs_pow34_v(NOR34, &sce->coeffs[start_c], sce->ics.swb_sizes[g]);
abs_pow34_v(PNS34, PNS, sce->ics.swb_sizes[g]);
dist1 += quantize_band_cost(s, &sce->coeffs[start_c],
NOR34,
sce->ics.swb_sizes[g],
sce->sf_idx[(w+w2)*16+g],
sce->band_alt[(w+w2)*16+g],
lambda/band->threshold, INFINITY, NULL, 0);
/* Estimate rd on average as 9 bits for CB and sf + spread energy * lambda/thr */
dist2 += 9+band->energy/(band->spread*band->spread)*lambda/band->threshold;
}
energy_ratio = pns_tgt_energy/pns_energy; /* Compensates for quantization error */
sce->pns_ener[w*16+g] = energy_ratio*pns_tgt_energy;
if (energy_ratio > 0.85f && energy_ratio < 1.25f && (sce->zeroes[w*16+g] || !sce->band_alt[w*16+g] || dist2*dist_thresh < dist1)) {
sce->band_type[w*16+g] = NOISE_BT;
sce->zeroes[w*16+g] = 0;
}
}
}
}
static void search_for_ms(AACEncContext *s, ChannelElement *cpe)
{
int start = 0, i, w, w2, g;
float M[128], S[128];
float *L34 = s->scoefs, *R34 = s->scoefs + 128, *M34 = s->scoefs + 128*2, *S34 = s->scoefs + 128*3;
const float lambda = s->lambda;
SingleChannelElement *sce0 = &cpe->ch[0];
SingleChannelElement *sce1 = &cpe->ch[1];
if (!cpe->common_window)
return;
for (w = 0; w < sce0->ics.num_windows; w += sce0->ics.group_len[w]) {
start = 0;
for (g = 0; g < sce0->ics.num_swb; g++) {
if (!cpe->ch[0].zeroes[w*16+g] && !cpe->ch[1].zeroes[w*16+g]) {
float dist1 = 0.0f, dist2 = 0.0f;
for (w2 = 0; w2 < sce0->ics.group_len[w]; w2++) {
FFPsyBand *band0 = &s->psy.ch[s->cur_channel+0].psy_bands[(w+w2)*16+g];
FFPsyBand *band1 = &s->psy.ch[s->cur_channel+1].psy_bands[(w+w2)*16+g];
float minthr = FFMIN(band0->threshold, band1->threshold);
float maxthr = FFMAX(band0->threshold, band1->threshold);
for (i = 0; i < sce0->ics.swb_sizes[g]; i++) {
M[i] = (sce0->coeffs[start+(w+w2)*128+i]
+ sce1->coeffs[start+(w+w2)*128+i]) * 0.5;
S[i] = M[i]
- sce1->coeffs[start+(w+w2)*128+i];
}
abs_pow34_v(L34, sce0->coeffs+start+(w+w2)*128, sce0->ics.swb_sizes[g]);
abs_pow34_v(R34, sce1->coeffs+start+(w+w2)*128, sce0->ics.swb_sizes[g]);
abs_pow34_v(M34, M, sce0->ics.swb_sizes[g]);
abs_pow34_v(S34, S, sce0->ics.swb_sizes[g]);
dist1 += quantize_band_cost(s, &sce0->coeffs[start + (w+w2)*128],
L34,
sce0->ics.swb_sizes[g],
sce0->sf_idx[(w+w2)*16+g],
sce0->band_type[(w+w2)*16+g],
lambda / band0->threshold, INFINITY, NULL, 0);
dist1 += quantize_band_cost(s, &sce1->coeffs[start + (w+w2)*128],
R34,
sce1->ics.swb_sizes[g],
sce1->sf_idx[(w+w2)*16+g],
sce1->band_type[(w+w2)*16+g],
lambda / band1->threshold, INFINITY, NULL, 0);
dist2 += quantize_band_cost(s, M,
M34,
sce0->ics.swb_sizes[g],
sce0->sf_idx[(w+w2)*16+g],
sce0->band_type[(w+w2)*16+g],
lambda / maxthr, INFINITY, NULL, 0);
dist2 += quantize_band_cost(s, S,
S34,
sce1->ics.swb_sizes[g],
sce1->sf_idx[(w+w2)*16+g],
sce1->band_type[(w+w2)*16+g],
lambda / minthr, INFINITY, NULL, 0);
}
cpe->ms_mask[w*16+g] = dist2 < dist1;
}
start += sce0->ics.swb_sizes[g];
}
}
}
AACCoefficientsEncoder ff_aac_coders[AAC_CODER_NB] = {
[AAC_CODER_FAAC] = {
search_for_quantizers_faac,
encode_window_bands_info,
quantize_and_encode_band,
ff_aac_encode_tns_info,
ff_aac_encode_main_pred,
ff_aac_adjust_common_prediction,
ff_aac_apply_main_pred,
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 07:47:31 +02:00
ff_aac_apply_tns,
set_special_band_scalefactors,
search_for_pns,
ff_aac_search_for_tns,
search_for_ms,
ff_aac_search_for_is,
ff_aac_search_for_pred,
},
[AAC_CODER_ANMR] = {
search_for_quantizers_anmr,
encode_window_bands_info,
quantize_and_encode_band,
ff_aac_encode_tns_info,
ff_aac_encode_main_pred,
ff_aac_adjust_common_prediction,
ff_aac_apply_main_pred,
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 07:47:31 +02:00
ff_aac_apply_tns,
set_special_band_scalefactors,
search_for_pns,
ff_aac_search_for_tns,
search_for_ms,
ff_aac_search_for_is,
ff_aac_search_for_pred,
},
[AAC_CODER_TWOLOOP] = {
search_for_quantizers_twoloop,
codebook_trellis_rate,
quantize_and_encode_band,
ff_aac_encode_tns_info,
ff_aac_encode_main_pred,
ff_aac_adjust_common_prediction,
ff_aac_apply_main_pred,
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 07:47:31 +02:00
ff_aac_apply_tns,
set_special_band_scalefactors,
search_for_pns,
ff_aac_search_for_tns,
search_for_ms,
ff_aac_search_for_is,
ff_aac_search_for_pred,
},
[AAC_CODER_FAST] = {
search_for_quantizers_fast,
encode_window_bands_info,
quantize_and_encode_band,
ff_aac_encode_tns_info,
ff_aac_encode_main_pred,
ff_aac_adjust_common_prediction,
ff_aac_apply_main_pred,
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 07:47:31 +02:00
ff_aac_apply_tns,
set_special_band_scalefactors,
search_for_pns,
ff_aac_search_for_tns,
search_for_ms,
ff_aac_search_for_is,
ff_aac_search_for_pred,
},
};