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ffmpeg/libavcodec/aacenc_is.c
Rostislav Pehlivanov 4565611b04 aacenc_is: add a flag to use pure coefficients instead 
This commit adds a flag to use the pure coefficients instead
of the processed ones (sce->coeffs). This is needed because
IS will apply the changes to the coefficients immediately
before the adjust_common_prediction function and it doesn't
make sense to measure stereo channel coefficient difference
when one of the channels coefficients are all zero.

Therefore add a flag to use pure coefficients in that case.
TNS is the only thing touching the coefficients before IS
so common window prediction will not take that into account
but the effect of the TNS filter per coefficient can be small
(a few percent) so to some approximation it's fine to just
ignore that.

Also fixed a small error which doesn't alter the results
that much. pow(sqrt(number), 3.0/4.0) == pow(number, 3.0/8.0) !=
pow(number, 3.0/4.0).

Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
2015-09-05 08:32:09 +01:00

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/*
* AAC encoder intensity stereo
* Copyright (C) 2015 Rostislav Pehlivanov
*
* 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 encoder Intensity Stereo
* @author Rostislav Pehlivanov ( atomnuker gmail com )
*/
#include "aacenc.h"
#include "aacenc_utils.h"
#include "aacenc_is.h"
#include "aacenc_quantization.h"
struct AACISError ff_aac_is_encoding_err(AACEncContext *s, ChannelElement *cpe,
int start, int w, int g, float ener0,
float ener1, float ener01,
int use_pcoeffs, int phase)
{
int i, w2;
SingleChannelElement *sce0 = &cpe->ch[0];
SingleChannelElement *sce1 = &cpe->ch[1];
float *L = use_pcoeffs ? sce0->pcoeffs : sce0->coeffs;
float *R = use_pcoeffs ? sce1->pcoeffs : sce1->coeffs;
float *L34 = &s->scoefs[256*0], *R34 = &s->scoefs[256*1];
float *IS = &s->scoefs[256*2], *I34 = &s->scoefs[256*3];
float dist1 = 0.0f, dist2 = 0.0f;
struct AACISError is_error = {0};
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];
int is_band_type, is_sf_idx = FFMAX(1, sce0->sf_idx[(w+w2)*16+g]-4);
float e01_34 = phase*pow(ener1/ener0, 3.0/4.0);
float maxval, dist_spec_err = 0.0f;
float minthr = FFMIN(band0->threshold, band1->threshold);
for (i = 0; i < sce0->ics.swb_sizes[g]; i++)
IS[i] = (L[start+(w+w2)*128+i] + phase*R[start+(w+w2)*128+i])*sqrt(ener0/ener01);
abs_pow34_v(L34, &L[start+(w+w2)*128], sce0->ics.swb_sizes[g]);
abs_pow34_v(R34, &R[start+(w+w2)*128], sce0->ics.swb_sizes[g]);
abs_pow34_v(I34, IS, sce0->ics.swb_sizes[g]);
maxval = find_max_val(1, sce0->ics.swb_sizes[g], I34);
is_band_type = find_min_book(maxval, is_sf_idx);
dist1 += quantize_band_cost(s, &L[start + (w+w2)*128], L34,
sce0->ics.swb_sizes[g],
sce0->sf_idx[(w+w2)*16+g],
sce0->band_type[(w+w2)*16+g],
s->lambda / band0->threshold, INFINITY, NULL, 0);
dist1 += quantize_band_cost(s, &R[start + (w+w2)*128], R34,
sce1->ics.swb_sizes[g],
sce1->sf_idx[(w+w2)*16+g],
sce1->band_type[(w+w2)*16+g],
s->lambda / band1->threshold, INFINITY, NULL, 0);
dist2 += quantize_band_cost(s, IS, I34, sce0->ics.swb_sizes[g],
is_sf_idx, is_band_type,
s->lambda / minthr, INFINITY, NULL, 0);
for (i = 0; i < sce0->ics.swb_sizes[g]; i++) {
dist_spec_err += (L34[i] - I34[i])*(L34[i] - I34[i]);
dist_spec_err += (R34[i] - I34[i]*e01_34)*(R34[i] - I34[i]*e01_34);
}
dist_spec_err *= s->lambda / minthr;
dist2 += dist_spec_err;
}
is_error.pass = dist2 <= dist1;
is_error.phase = phase;
is_error.error = fabsf(dist1 - dist2);
is_error.dist1 = dist1;
is_error.dist2 = dist2;
return is_error;
}
void ff_aac_search_for_is(AACEncContext *s, AVCodecContext *avctx, ChannelElement *cpe)
{
SingleChannelElement *sce0 = &cpe->ch[0];
SingleChannelElement *sce1 = &cpe->ch[1];
int start = 0, count = 0, w, w2, g, i;
const float freq_mult = avctx->sample_rate/(1024.0f/sce0->ics.num_windows)/2.0f;
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 (start*freq_mult > INT_STEREO_LOW_LIMIT*(s->lambda/170.0f) &&
cpe->ch[0].band_type[w*16+g] != NOISE_BT && !cpe->ch[0].zeroes[w*16+g] &&
cpe->ch[1].band_type[w*16+g] != NOISE_BT && !cpe->ch[1].zeroes[w*16+g]) {
float ener0 = 0.0f, ener1 = 0.0f, ener01 = 0.0f;
struct AACISError ph_err1, ph_err2, *erf;
for (w2 = 0; w2 < sce0->ics.group_len[w]; w2++) {
for (i = 0; i < sce0->ics.swb_sizes[g]; i++) {
float coef0 = sce0->pcoeffs[start+(w+w2)*128+i];
float coef1 = sce1->pcoeffs[start+(w+w2)*128+i];
ener0 += coef0*coef0;
ener1 += coef1*coef1;
ener01 += (coef0 + coef1)*(coef0 + coef1);
}
}
ph_err1 = ff_aac_is_encoding_err(s, cpe, start, w, g,
ener0, ener1, ener01, 0, -1);
ph_err2 = ff_aac_is_encoding_err(s, cpe, start, w, g,
ener0, ener1, ener01, 0, +1);
erf = ph_err1.error < ph_err2.error ? &ph_err1 : &ph_err2;
if (erf->pass) {
cpe->is_mask[w*16+g] = 1;
cpe->ch[0].is_ener[w*16+g] = sqrt(ener0/ener01);
cpe->ch[1].is_ener[w*16+g] = ener0/ener1;
cpe->ch[1].band_type[w*16+g] = erf->phase ? INTENSITY_BT : INTENSITY_BT2;
count++;
}
}
start += sce0->ics.swb_sizes[g];
}
}
cpe->is_mode = !!count;
}
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