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ffmpeg/libavcodec/aacenc_is.c
Claudio Freire ca203e9985 AAC encoder: improve SF range utilization 
This patch does 4 things, all of which interact and thus it
woudln't be possible to commit them separately without causing
either quality regressions or assertion failures.

Fate comparison targets don't all reflect improvements in
quality, yet listening tests show substantially improved quality
and stability.

1. Increase SF range utilization.

The spec requires SF delta values to be constrained within the
range -60..60. The previous code was applying that range to
the whole SF array and not only the deltas of consecutive values,
because doing so requires smarter code: zeroing or otherwise
skipping a band may invalidate lots of SF choices.

This patch implements that logic to allow the coders to utilize
the full dynamic range of scalefactors, increasing quality quite
considerably, and fixing delta-SF-related assertion failures,
since now the limitation is enforced rather than asserted.

2. PNS tweaks

The previous modification makes big improvements in twoloop's
efficiency, and every time that happens PNS logic needs to be
tweaked accordingly to avoid it from stepping all over twoloop's
decisions. This patch includes modifications of the sort.

3. Account for lowpass cutoff during PSY analysis

The closer PSY's allocation is to final allocation the better
the quality is, and given these modifications, twoloop is now
very efficient at avoiding holes. Thus, to compute accurate
thresholds, PSY needs to account for the lowpass applied
implicitly during twoloop (by zeroing high bands).

This patch makes twoloop set the cutoff in psymodel's context
the first time it runs, and makes PSY account for it during
threshold computation, making PE and threshold computations
closer to the final allocation and thus achieving better
subjective quality.

4. Tweaks to RC lambda tracking loop in relation to PNS

Without this tweak some corner cases cause quality regressions.
Basically, lambda needs to react faster to overall bitrate
efficiency changes since now PNS can be quite successful in
enforcing maximum bitrates, when PSY allocates too many bits
to the lower bands, suppressing the signals RC logic uses to
lower lambda in those cases and causing aggressive PNS.

This tweak makes PNS much less aggressive, though it can still
use some further tweaks.

Also update MIPS specializations and adjust fuzz

Also in lavc/mips/aacpsy_mips.h: remove trailing whitespace
2015-12-02 07:47:37 -03:00

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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};
if (ener01 <= 0 || ener0 <= 0) {
is_error.pass = 0;
return is_error;
}
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, 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, 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, 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;
is_error.ener01 = ener01;
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, prev_sf1 = -1;
const float freq_mult = avctx->sample_rate/(1024.0f/sce0->ics.num_windows)/2.0f;
uint8_t nextband1[128];
if (!cpe->common_window)
return;
/** Scout out next nonzero bands */
ff_init_nextband_map(sce1, nextband1);
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] &&
ff_sfdelta_can_remove_band(sce1, nextband1, prev_sf1, w*16+g)) {
float ener0 = 0.0f, ener1 = 0.0f, ener01 = 0.0f, ener01p = 0.0f;
struct AACISError ph_err1, ph_err2, *erf;
if (sce0->band_type[w*16+g] == NOISE_BT ||
sce1->band_type[w*16+g] == NOISE_BT) {
start += sce0->ics.swb_sizes[g];
continue;
}
for (w2 = 0; w2 < sce0->ics.group_len[w]; w2++) {
for (i = 0; i < sce0->ics.swb_sizes[g]; i++) {
float coef0 = fabsf(sce0->coeffs[start+(w+w2)*128+i]);
float coef1 = fabsf(sce1->coeffs[start+(w+w2)*128+i]);
ener0 += coef0*coef0;
ener1 += coef1*coef1;
ener01 += (coef0 + coef1)*(coef0 + coef1);
ener01p += (coef0 - coef1)*(coef0 - coef1);
}
}
ph_err1 = ff_aac_is_encoding_err(s, cpe, start, w, g,
ener0, ener1, ener01p, 0, -1);
ph_err2 = ff_aac_is_encoding_err(s, cpe, start, w, g,
ener0, ener1, ener01, 0, +1);
erf = (ph_err1.pass && ph_err1.error < ph_err2.error) ? &ph_err1 : &ph_err2;
if (erf->pass) {
cpe->is_mask[w*16+g] = 1;
cpe->ms_mask[w*16+g] = 0;
cpe->ch[0].is_ener[w*16+g] = sqrt(ener0 / erf->ener01);
cpe->ch[1].is_ener[w*16+g] = ener0/ener1;
cpe->ch[1].band_type[w*16+g] = (erf->phase > 0) ? INTENSITY_BT : INTENSITY_BT2;
count++;
}
}
if (!sce1->zeroes[w*16+g] && sce1->band_type[w*16+g] < RESERVED_BT)
prev_sf1 = sce1->sf_idx[w*16+g];
start += sce0->ics.swb_sizes[g];
}
}
cpe->is_mode = !!count;
}
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