mirror of
https://github.com/xenia-project/FFmpeg.git
synced 2024-11-24 12:09:55 +00:00
764 lines
35 KiB
C
764 lines
35 KiB
C
/*
|
|
* AAC encoder twoloop coder
|
|
* 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 encoder twoloop coder
|
|
* @author Konstantin Shishkov, Claudio Freire
|
|
*/
|
|
|
|
/**
|
|
* This file contains a template for the twoloop coder function.
|
|
* It needs to be provided, externally, as an already included declaration,
|
|
* the following functions from aacenc_quantization/util.h. They're not included
|
|
* explicitly here to make it possible to provide alternative implementations:
|
|
* - quantize_band_cost
|
|
* - abs_pow34_v
|
|
* - find_max_val
|
|
* - find_min_book
|
|
* - find_form_factor
|
|
*/
|
|
|
|
#ifndef AVCODEC_AACCODER_TWOLOOP_H
|
|
#define AVCODEC_AACCODER_TWOLOOP_H
|
|
|
|
#include <float.h>
|
|
#include "libavutil/mathematics.h"
|
|
#include "mathops.h"
|
|
#include "avcodec.h"
|
|
#include "put_bits.h"
|
|
#include "aac.h"
|
|
#include "aacenc.h"
|
|
#include "aactab.h"
|
|
#include "aacenctab.h"
|
|
|
|
/** Frequency in Hz for lower limit of noise substitution **/
|
|
#define NOISE_LOW_LIMIT 4000
|
|
|
|
#define sclip(x) av_clip(x,60,218)
|
|
|
|
/* Reflects the cost to change codebooks */
|
|
static inline int ff_pns_bits(SingleChannelElement *sce, int w, int g)
|
|
{
|
|
return (!g || !sce->zeroes[w*16+g-1] || !sce->can_pns[w*16+g-1]) ? 9 : 5;
|
|
}
|
|
|
|
/**
|
|
* two-loop quantizers search taken from ISO 13818-7 Appendix C
|
|
*/
|
|
static void search_for_quantizers_twoloop(AVCodecContext *avctx,
|
|
AACEncContext *s,
|
|
SingleChannelElement *sce,
|
|
const float lambda)
|
|
{
|
|
int start = 0, i, w, w2, g, recomprd;
|
|
int destbits = avctx->bit_rate * 1024.0 / avctx->sample_rate
|
|
/ ((avctx->flags & CODEC_FLAG_QSCALE) ? 2.0f : avctx->channels)
|
|
* (lambda / 120.f);
|
|
int refbits = destbits;
|
|
int toomanybits, toofewbits;
|
|
char nzs[128];
|
|
uint8_t nextband[128];
|
|
int maxsf[128], minsf[128];
|
|
float dists[128] = { 0 }, qenergies[128] = { 0 }, uplims[128], euplims[128], energies[128];
|
|
float maxvals[128], spread_thr_r[128];
|
|
float min_spread_thr_r, max_spread_thr_r;
|
|
|
|
/**
|
|
* rdlambda controls the maximum tolerated distortion. Twoloop
|
|
* will keep iterating until it fails to lower it or it reaches
|
|
* ulimit * rdlambda. Keeping it low increases quality on difficult
|
|
* signals, but lower it too much, and bits will be taken from weak
|
|
* signals, creating "holes". A balance is necesary.
|
|
* rdmax and rdmin specify the relative deviation from rdlambda
|
|
* allowed for tonality compensation
|
|
*/
|
|
float rdlambda = av_clipf(2.0f * 120.f / lambda, 0.0625f, 16.0f);
|
|
const float nzslope = 1.5f;
|
|
float rdmin = 0.03125f;
|
|
float rdmax = 1.0f;
|
|
|
|
/**
|
|
* sfoffs controls an offset of optmium allocation that will be
|
|
* applied based on lambda. Keep it real and modest, the loop
|
|
* will take care of the rest, this just accelerates convergence
|
|
*/
|
|
float sfoffs = av_clipf(log2f(120.0f / lambda) * 4.0f, -5, 10);
|
|
|
|
int fflag, minscaler, maxscaler, nminscaler;
|
|
int its = 0;
|
|
int maxits = 30;
|
|
int allz = 0;
|
|
int tbits;
|
|
int cutoff = 1024;
|
|
int pns_start_pos;
|
|
int prev;
|
|
|
|
/**
|
|
* zeroscale controls a multiplier of the threshold, if band energy
|
|
* is below this, a zero is forced. Keep it lower than 1, unless
|
|
* low lambda is used, because energy < threshold doesn't mean there's
|
|
* no audible signal outright, it's just energy. Also make it rise
|
|
* slower than rdlambda, as rdscale has due compensation with
|
|
* noisy band depriorization below, whereas zeroing logic is rather dumb
|
|
*/
|
|
float zeroscale;
|
|
if (lambda > 120.f) {
|
|
zeroscale = av_clipf(powf(120.f / lambda, 0.25f), 0.0625f, 1.0f);
|
|
} else {
|
|
zeroscale = 1.f;
|
|
}
|
|
|
|
if (s->psy.bitres.alloc >= 0) {
|
|
/**
|
|
* Psy granted us extra bits to use, from the reservoire
|
|
* adjust for lambda except what psy already did
|
|
*/
|
|
destbits = s->psy.bitres.alloc
|
|
* (lambda / (avctx->global_quality ? avctx->global_quality : 120));
|
|
}
|
|
|
|
if (avctx->flags & CODEC_FLAG_QSCALE) {
|
|
/**
|
|
* Constant Q-scale doesn't compensate MS coding on its own
|
|
* No need to be overly precise, this only controls RD
|
|
* adjustment CB limits when going overboard
|
|
*/
|
|
if (s->options.mid_side && s->cur_type == TYPE_CPE)
|
|
destbits *= 2;
|
|
|
|
/**
|
|
* When using a constant Q-scale, don't adjust bits, just use RD
|
|
* Don't let it go overboard, though... 8x psy target is enough
|
|
*/
|
|
toomanybits = 5800;
|
|
toofewbits = destbits / 16;
|
|
|
|
/** Don't offset scalers, just RD */
|
|
sfoffs = sce->ics.num_windows - 1;
|
|
rdlambda = sqrtf(rdlambda);
|
|
|
|
/** search further */
|
|
maxits *= 2;
|
|
} else {
|
|
/* When using ABR, be strict, but a reasonable leeway is
|
|
* critical to allow RC to smoothly track desired bitrate
|
|
* without sudden quality drops that cause audible artifacts.
|
|
* Symmetry is also desirable, to avoid systematic bias.
|
|
*/
|
|
toomanybits = destbits + destbits/8;
|
|
toofewbits = destbits - destbits/8;
|
|
|
|
sfoffs = 0;
|
|
rdlambda = sqrtf(rdlambda);
|
|
}
|
|
|
|
/** and zero out above cutoff frequency */
|
|
{
|
|
int wlen = 1024 / sce->ics.num_windows;
|
|
int bandwidth;
|
|
|
|
/**
|
|
* Scale, psy gives us constant quality, this LP only scales
|
|
* bitrate by lambda, so we save bits on subjectively unimportant HF
|
|
* rather than increase quantization noise. Adjust nominal bitrate
|
|
* to effective bitrate according to encoding parameters,
|
|
* AAC_CUTOFF_FROM_BITRATE is calibrated for effective bitrate.
|
|
*/
|
|
float rate_bandwidth_multiplier = 1.5f;
|
|
int frame_bit_rate = (avctx->flags & CODEC_FLAG_QSCALE)
|
|
? (refbits * rate_bandwidth_multiplier * avctx->sample_rate / 1024)
|
|
: (avctx->bit_rate / avctx->channels);
|
|
|
|
/** Compensate for extensions that increase efficiency */
|
|
if (s->options.pns || s->options.intensity_stereo)
|
|
frame_bit_rate *= 1.15f;
|
|
|
|
if (avctx->cutoff > 0) {
|
|
bandwidth = avctx->cutoff;
|
|
} else {
|
|
bandwidth = FFMAX(3000, AAC_CUTOFF_FROM_BITRATE(frame_bit_rate, 1, avctx->sample_rate));
|
|
s->psy.cutoff = bandwidth;
|
|
}
|
|
|
|
cutoff = bandwidth * 2 * wlen / avctx->sample_rate;
|
|
pns_start_pos = NOISE_LOW_LIMIT * 2 * wlen / avctx->sample_rate;
|
|
}
|
|
|
|
/**
|
|
* for values above this the decoder might end up in an endless loop
|
|
* due to always having more bits than what can be encoded.
|
|
*/
|
|
destbits = FFMIN(destbits, 5800);
|
|
toomanybits = FFMIN(toomanybits, 5800);
|
|
toofewbits = FFMIN(toofewbits, 5800);
|
|
/**
|
|
* XXX: some heuristic to determine initial quantizers will reduce search time
|
|
* determine zero bands and upper distortion limits
|
|
*/
|
|
min_spread_thr_r = -1;
|
|
max_spread_thr_r = -1;
|
|
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
|
|
for (g = start = 0; g < sce->ics.num_swb; start += sce->ics.swb_sizes[g++]) {
|
|
int nz = 0;
|
|
float uplim = 0.0f, energy = 0.0f, spread = 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 (start >= cutoff || band->energy <= (band->threshold * zeroscale) || band->threshold == 0.0f) {
|
|
sce->zeroes[(w+w2)*16+g] = 1;
|
|
continue;
|
|
}
|
|
nz = 1;
|
|
}
|
|
if (!nz) {
|
|
uplim = 0.0f;
|
|
} else {
|
|
nz = 0;
|
|
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 * zeroscale) || band->threshold == 0.0f)
|
|
continue;
|
|
uplim += band->threshold;
|
|
energy += band->energy;
|
|
spread += band->spread;
|
|
nz++;
|
|
}
|
|
}
|
|
uplims[w*16+g] = uplim;
|
|
energies[w*16+g] = energy;
|
|
nzs[w*16+g] = nz;
|
|
sce->zeroes[w*16+g] = !nz;
|
|
allz |= nz;
|
|
if (nz && sce->can_pns[w*16+g]) {
|
|
spread_thr_r[w*16+g] = energy * nz / (uplim * spread);
|
|
if (min_spread_thr_r < 0) {
|
|
min_spread_thr_r = max_spread_thr_r = spread_thr_r[w*16+g];
|
|
} else {
|
|
min_spread_thr_r = FFMIN(min_spread_thr_r, spread_thr_r[w*16+g]);
|
|
max_spread_thr_r = FFMAX(max_spread_thr_r, spread_thr_r[w*16+g]);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
/** Compute initial scalers */
|
|
minscaler = 65535;
|
|
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->zeroes[w*16+g]) {
|
|
sce->sf_idx[w*16+g] = SCALE_ONE_POS;
|
|
continue;
|
|
}
|
|
/**
|
|
* log2f-to-distortion ratio is, technically, 2 (1.5db = 4, but it's power vs level so it's 2).
|
|
* But, as offsets are applied, low-frequency signals are too sensitive to the induced distortion,
|
|
* so we make scaling more conservative by choosing a lower log2f-to-distortion ratio, and thus
|
|
* more robust.
|
|
*/
|
|
sce->sf_idx[w*16+g] = av_clip(
|
|
SCALE_ONE_POS
|
|
+ 1.75*log2f(FFMAX(0.00125f,uplims[w*16+g]) / sce->ics.swb_sizes[g])
|
|
+ sfoffs,
|
|
60, SCALE_MAX_POS);
|
|
minscaler = FFMIN(minscaler, sce->sf_idx[w*16+g]);
|
|
}
|
|
}
|
|
|
|
/** Clip */
|
|
minscaler = av_clip(minscaler, SCALE_ONE_POS - SCALE_DIV_512, SCALE_MAX_POS - SCALE_DIV_512);
|
|
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->zeroes[w*16+g])
|
|
sce->sf_idx[w*16+g] = av_clip(sce->sf_idx[w*16+g], minscaler, minscaler + SCALE_MAX_DIFF - 1);
|
|
|
|
if (!allz)
|
|
return;
|
|
abs_pow34_v(s->scoefs, sce->coeffs, 1024);
|
|
ff_quantize_band_cost_cache_init(s);
|
|
|
|
for (i = 0; i < sizeof(minsf) / sizeof(minsf[0]); ++i)
|
|
minsf[i] = 0;
|
|
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 *scaled = s->scoefs + start;
|
|
int minsfidx;
|
|
maxvals[w*16+g] = find_max_val(sce->ics.group_len[w], sce->ics.swb_sizes[g], scaled);
|
|
if (maxvals[w*16+g] > 0) {
|
|
minsfidx = coef2minsf(maxvals[w*16+g]);
|
|
for (w2 = 0; w2 < sce->ics.group_len[w]; w2++)
|
|
minsf[(w+w2)*16+g] = minsfidx;
|
|
}
|
|
start += sce->ics.swb_sizes[g];
|
|
}
|
|
}
|
|
|
|
/**
|
|
* Scale uplims to match rate distortion to quality
|
|
* bu applying noisy band depriorization and tonal band priorization.
|
|
* Maxval-energy ratio gives us an idea of how noisy/tonal the band is.
|
|
* If maxval^2 ~ energy, then that band is mostly noise, and we can relax
|
|
* rate distortion requirements.
|
|
*/
|
|
memcpy(euplims, uplims, sizeof(euplims));
|
|
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
|
|
/** psy already priorizes transients to some extent */
|
|
float de_psy_factor = (sce->ics.num_windows > 1) ? 8.0f / sce->ics.group_len[w] : 1.0f;
|
|
start = w*128;
|
|
for (g = 0; g < sce->ics.num_swb; g++) {
|
|
if (nzs[g] > 0) {
|
|
float cleanup_factor = ff_sqrf(av_clipf(start / (cutoff * 0.75f), 1.0f, 2.0f));
|
|
float energy2uplim = find_form_factor(
|
|
sce->ics.group_len[w], sce->ics.swb_sizes[g],
|
|
uplims[w*16+g] / (nzs[g] * sce->ics.swb_sizes[w]),
|
|
sce->coeffs + start,
|
|
nzslope * cleanup_factor);
|
|
energy2uplim *= de_psy_factor;
|
|
if (!(avctx->flags & CODEC_FLAG_QSCALE)) {
|
|
/** In ABR, we need to priorize less and let rate control do its thing */
|
|
energy2uplim = sqrtf(energy2uplim);
|
|
}
|
|
energy2uplim = FFMAX(0.015625f, FFMIN(1.0f, energy2uplim));
|
|
uplims[w*16+g] *= av_clipf(rdlambda * energy2uplim, rdmin, rdmax)
|
|
* sce->ics.group_len[w];
|
|
|
|
energy2uplim = find_form_factor(
|
|
sce->ics.group_len[w], sce->ics.swb_sizes[g],
|
|
uplims[w*16+g] / (nzs[g] * sce->ics.swb_sizes[w]),
|
|
sce->coeffs + start,
|
|
2.0f);
|
|
energy2uplim *= de_psy_factor;
|
|
if (!(avctx->flags & CODEC_FLAG_QSCALE)) {
|
|
/** In ABR, we need to priorize less and let rate control do its thing */
|
|
energy2uplim = sqrtf(energy2uplim);
|
|
}
|
|
energy2uplim = FFMAX(0.015625f, FFMIN(1.0f, energy2uplim));
|
|
euplims[w*16+g] *= av_clipf(rdlambda * energy2uplim * sce->ics.group_len[w],
|
|
0.5f, 1.0f);
|
|
}
|
|
start += sce->ics.swb_sizes[g];
|
|
}
|
|
}
|
|
|
|
for (i = 0; i < sizeof(maxsf) / sizeof(maxsf[0]); ++i)
|
|
maxsf[i] = SCALE_MAX_POS;
|
|
|
|
//perform two-loop search
|
|
//outer loop - improve quality
|
|
do {
|
|
//inner loop - quantize spectrum to fit into given number of bits
|
|
int overdist;
|
|
int qstep = its ? 1 : 32;
|
|
do {
|
|
int changed = 0;
|
|
prev = -1;
|
|
recomprd = 0;
|
|
tbits = 0;
|
|
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];
|
|
int bits = 0;
|
|
int cb;
|
|
float dist = 0.0f;
|
|
float qenergy = 0.0f;
|
|
|
|
if (sce->zeroes[w*16+g] || sce->sf_idx[w*16+g] >= 218) {
|
|
start += sce->ics.swb_sizes[g];
|
|
if (sce->can_pns[w*16+g]) {
|
|
/** PNS isn't free */
|
|
tbits += ff_pns_bits(sce, w, g);
|
|
}
|
|
continue;
|
|
}
|
|
cb = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]);
|
|
for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
|
|
int b;
|
|
float sqenergy;
|
|
dist += quantize_band_cost_cached(s, w + w2, g, coefs + w2*128,
|
|
scaled + w2*128,
|
|
sce->ics.swb_sizes[g],
|
|
sce->sf_idx[w*16+g],
|
|
cb,
|
|
1.0f,
|
|
INFINITY,
|
|
&b, &sqenergy,
|
|
0);
|
|
bits += b;
|
|
qenergy += sqenergy;
|
|
}
|
|
dists[w*16+g] = dist - bits;
|
|
qenergies[w*16+g] = qenergy;
|
|
if (prev != -1) {
|
|
int sfdiff = av_clip(sce->sf_idx[w*16+g] - prev + SCALE_DIFF_ZERO, 0, 2*SCALE_MAX_DIFF);
|
|
bits += ff_aac_scalefactor_bits[sfdiff];
|
|
}
|
|
tbits += bits;
|
|
start += sce->ics.swb_sizes[g];
|
|
prev = sce->sf_idx[w*16+g];
|
|
}
|
|
}
|
|
if (tbits > toomanybits) {
|
|
recomprd = 1;
|
|
for (i = 0; i < 128; i++) {
|
|
if (sce->sf_idx[i] < (SCALE_MAX_POS - SCALE_DIV_512)) {
|
|
int maxsf_i = (tbits > 5800) ? SCALE_MAX_POS : maxsf[i];
|
|
int new_sf = FFMIN(maxsf_i, sce->sf_idx[i] + qstep);
|
|
if (new_sf != sce->sf_idx[i]) {
|
|
sce->sf_idx[i] = new_sf;
|
|
changed = 1;
|
|
}
|
|
}
|
|
}
|
|
} else if (tbits < toofewbits) {
|
|
recomprd = 1;
|
|
for (i = 0; i < 128; i++) {
|
|
if (sce->sf_idx[i] > SCALE_ONE_POS) {
|
|
int new_sf = FFMAX3(minsf[i], SCALE_ONE_POS, sce->sf_idx[i] - qstep);
|
|
if (new_sf != sce->sf_idx[i]) {
|
|
sce->sf_idx[i] = new_sf;
|
|
changed = 1;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
qstep >>= 1;
|
|
if (!qstep && tbits > toomanybits && sce->sf_idx[0] < 217 && changed)
|
|
qstep = 1;
|
|
} while (qstep);
|
|
|
|
overdist = 1;
|
|
fflag = tbits < toofewbits;
|
|
for (i = 0; i < 2 && (overdist || recomprd); ++i) {
|
|
if (recomprd) {
|
|
/** Must recompute distortion */
|
|
prev = -1;
|
|
tbits = 0;
|
|
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;
|
|
int bits = 0;
|
|
int cb;
|
|
float dist = 0.0f;
|
|
float qenergy = 0.0f;
|
|
|
|
if (sce->zeroes[w*16+g] || sce->sf_idx[w*16+g] >= 218) {
|
|
start += sce->ics.swb_sizes[g];
|
|
if (sce->can_pns[w*16+g]) {
|
|
/** PNS isn't free */
|
|
tbits += ff_pns_bits(sce, w, g);
|
|
}
|
|
continue;
|
|
}
|
|
cb = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]);
|
|
for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
|
|
int b;
|
|
float sqenergy;
|
|
dist += quantize_band_cost_cached(s, w + w2, g, coefs + w2*128,
|
|
scaled + w2*128,
|
|
sce->ics.swb_sizes[g],
|
|
sce->sf_idx[w*16+g],
|
|
cb,
|
|
1.0f,
|
|
INFINITY,
|
|
&b, &sqenergy,
|
|
0);
|
|
bits += b;
|
|
qenergy += sqenergy;
|
|
}
|
|
dists[w*16+g] = dist - bits;
|
|
qenergies[w*16+g] = qenergy;
|
|
if (prev != -1) {
|
|
int sfdiff = av_clip(sce->sf_idx[w*16+g] - prev + SCALE_DIFF_ZERO, 0, 2*SCALE_MAX_DIFF);
|
|
bits += ff_aac_scalefactor_bits[sfdiff];
|
|
}
|
|
tbits += bits;
|
|
start += sce->ics.swb_sizes[g];
|
|
prev = sce->sf_idx[w*16+g];
|
|
}
|
|
}
|
|
}
|
|
if (!i && s->options.pns && its > maxits/2 && tbits > toofewbits) {
|
|
float maxoverdist = 0.0f;
|
|
float ovrfactor = 1.f+(maxits-its)*16.f/maxits;
|
|
overdist = recomprd = 0;
|
|
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
|
|
for (g = start = 0; g < sce->ics.num_swb; start += sce->ics.swb_sizes[g++]) {
|
|
if (!sce->zeroes[w*16+g] && sce->sf_idx[w*16+g] > SCALE_ONE_POS && dists[w*16+g] > uplims[w*16+g]*ovrfactor) {
|
|
float ovrdist = dists[w*16+g] / FFMAX(uplims[w*16+g],euplims[w*16+g]);
|
|
maxoverdist = FFMAX(maxoverdist, ovrdist);
|
|
overdist++;
|
|
}
|
|
}
|
|
}
|
|
if (overdist) {
|
|
/* We have overdistorted bands, trade for zeroes (that can be noise)
|
|
* Zero the bands in the lowest 1.25% spread-energy-threshold ranking
|
|
*/
|
|
float minspread = max_spread_thr_r;
|
|
float maxspread = min_spread_thr_r;
|
|
float zspread;
|
|
int zeroable = 0;
|
|
int zeroed = 0;
|
|
int maxzeroed, zloop;
|
|
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
|
|
for (g = start = 0; g < sce->ics.num_swb; start += sce->ics.swb_sizes[g++]) {
|
|
if (start >= pns_start_pos && !sce->zeroes[w*16+g] && sce->can_pns[w*16+g]) {
|
|
minspread = FFMIN(minspread, spread_thr_r[w*16+g]);
|
|
maxspread = FFMAX(maxspread, spread_thr_r[w*16+g]);
|
|
zeroable++;
|
|
}
|
|
}
|
|
}
|
|
zspread = (maxspread-minspread) * 0.0125f + minspread;
|
|
/* Don't PNS everything even if allowed. It suppresses bit starvation signals from RC,
|
|
* and forced the hand of the later search_for_pns step.
|
|
* Instead, PNS a fraction of the spread_thr_r range depending on how starved for bits we are,
|
|
* and leave further PNSing to search_for_pns if worthwhile.
|
|
*/
|
|
zspread = FFMIN3(min_spread_thr_r * 8.f, zspread,
|
|
((toomanybits - tbits) * min_spread_thr_r + (tbits - toofewbits) * max_spread_thr_r) / (toomanybits - toofewbits + 1));
|
|
maxzeroed = FFMIN(zeroable, FFMAX(1, (zeroable * its + maxits - 1) / (2 * maxits)));
|
|
for (zloop = 0; zloop < 2; zloop++) {
|
|
/* Two passes: first distorted stuff - two birds in one shot and all that,
|
|
* then anything viable. Viable means not zero, but either CB=zero-able
|
|
* (too high SF), not SF <= 1 (that means we'd be operating at very high
|
|
* quality, we don't want PNS when doing VHQ), PNS allowed, and within
|
|
* the lowest ranking percentile.
|
|
*/
|
|
float loopovrfactor = (zloop) ? 1.0f : ovrfactor;
|
|
int loopminsf = (zloop) ? (SCALE_ONE_POS - SCALE_DIV_512) : SCALE_ONE_POS;
|
|
int mcb;
|
|
for (g = sce->ics.num_swb-1; g > 0 && zeroed < maxzeroed; g--) {
|
|
if (sce->ics.swb_offset[g] < pns_start_pos)
|
|
continue;
|
|
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
|
|
if (!sce->zeroes[w*16+g] && sce->can_pns[w*16+g] && spread_thr_r[w*16+g] <= zspread
|
|
&& sce->sf_idx[w*16+g] > loopminsf
|
|
&& (dists[w*16+g] > loopovrfactor*uplims[w*16+g] || !(mcb = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]))
|
|
|| (mcb <= 1 && dists[w*16+g] > FFMIN(uplims[w*16+g], euplims[w*16+g]))) ) {
|
|
sce->zeroes[w*16+g] = 1;
|
|
sce->band_type[w*16+g] = 0;
|
|
zeroed++;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
if (zeroed)
|
|
recomprd = fflag = 1;
|
|
} else {
|
|
overdist = 0;
|
|
}
|
|
}
|
|
}
|
|
|
|
minscaler = SCALE_MAX_POS;
|
|
maxscaler = 0;
|
|
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->zeroes[w*16+g]) {
|
|
minscaler = FFMIN(minscaler, sce->sf_idx[w*16+g]);
|
|
maxscaler = FFMAX(maxscaler, sce->sf_idx[w*16+g]);
|
|
}
|
|
}
|
|
}
|
|
|
|
minscaler = nminscaler = av_clip(minscaler, SCALE_ONE_POS - SCALE_DIV_512, SCALE_MAX_POS - SCALE_DIV_512);
|
|
prev = -1;
|
|
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
|
|
/** Start with big steps, end up fine-tunning */
|
|
int depth = (its > maxits/2) ? ((its > maxits*2/3) ? 1 : 3) : 10;
|
|
int edepth = depth+2;
|
|
float uplmax = its / (maxits*0.25f) + 1.0f;
|
|
uplmax *= (tbits > destbits) ? FFMIN(2.0f, tbits / (float)FFMAX(1,destbits)) : 1.0f;
|
|
start = w * 128;
|
|
for (g = 0; g < sce->ics.num_swb; g++) {
|
|
int prevsc = sce->sf_idx[w*16+g];
|
|
if (prev < 0 && !sce->zeroes[w*16+g])
|
|
prev = sce->sf_idx[0];
|
|
if (!sce->zeroes[w*16+g]) {
|
|
const float *coefs = sce->coeffs + start;
|
|
const float *scaled = s->scoefs + start;
|
|
int cmb = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]);
|
|
int mindeltasf = FFMAX(0, prev - SCALE_MAX_DIFF);
|
|
int maxdeltasf = FFMIN(SCALE_MAX_POS - SCALE_DIV_512, prev + SCALE_MAX_DIFF);
|
|
if ((!cmb || dists[w*16+g] > uplims[w*16+g]) && sce->sf_idx[w*16+g] > FFMAX(mindeltasf, minsf[w*16+g])) {
|
|
/* Try to make sure there is some energy in every nonzero band
|
|
* NOTE: This algorithm must be forcibly imbalanced, pushing harder
|
|
* on holes or more distorted bands at first, otherwise there's
|
|
* no net gain (since the next iteration will offset all bands
|
|
* on the opposite direction to compensate for extra bits)
|
|
*/
|
|
for (i = 0; i < edepth && sce->sf_idx[w*16+g] > mindeltasf; ++i) {
|
|
int cb, bits;
|
|
float dist, qenergy;
|
|
int mb = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]-1);
|
|
cb = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]);
|
|
dist = qenergy = 0.f;
|
|
bits = 0;
|
|
if (!cb) {
|
|
maxsf[w*16+g] = FFMIN(sce->sf_idx[w*16+g]-1, maxsf[w*16+g]);
|
|
} else if (i >= depth && dists[w*16+g] < euplims[w*16+g]) {
|
|
break;
|
|
}
|
|
/* !g is the DC band, it's important, since quantization error here
|
|
* applies to less than a cycle, it creates horrible intermodulation
|
|
* distortion if it doesn't stick to what psy requests
|
|
*/
|
|
if (!g && sce->ics.num_windows > 1 && dists[w*16+g] >= euplims[w*16+g])
|
|
maxsf[w*16+g] = FFMIN(sce->sf_idx[w*16+g], maxsf[w*16+g]);
|
|
for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
|
|
int b;
|
|
float sqenergy;
|
|
dist += quantize_band_cost_cached(s, w + w2, g, coefs + w2*128,
|
|
scaled + w2*128,
|
|
sce->ics.swb_sizes[g],
|
|
sce->sf_idx[w*16+g]-1,
|
|
cb,
|
|
1.0f,
|
|
INFINITY,
|
|
&b, &sqenergy,
|
|
0);
|
|
bits += b;
|
|
qenergy += sqenergy;
|
|
}
|
|
sce->sf_idx[w*16+g]--;
|
|
dists[w*16+g] = dist - bits;
|
|
qenergies[w*16+g] = qenergy;
|
|
if (mb && (sce->sf_idx[w*16+g] < mindeltasf || (
|
|
(dists[w*16+g] < FFMIN(uplmax*uplims[w*16+g], euplims[w*16+g]))
|
|
&& (fabsf(qenergies[w*16+g]-energies[w*16+g]) < euplims[w*16+g])
|
|
) )) {
|
|
break;
|
|
}
|
|
}
|
|
} else if (tbits > toofewbits && sce->sf_idx[w*16+g] < FFMIN(maxdeltasf, maxsf[w*16+g])
|
|
&& (dists[w*16+g] < FFMIN(euplims[w*16+g], uplims[w*16+g]))
|
|
&& (fabsf(qenergies[w*16+g]-energies[w*16+g]) < euplims[w*16+g])
|
|
) {
|
|
/** Um... over target. Save bits for more important stuff. */
|
|
for (i = 0; i < depth && sce->sf_idx[w*16+g] < maxdeltasf; ++i) {
|
|
int cb, bits;
|
|
float dist, qenergy;
|
|
cb = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]+1);
|
|
if (cb > 0) {
|
|
dist = qenergy = 0.f;
|
|
bits = 0;
|
|
for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
|
|
int b;
|
|
float sqenergy;
|
|
dist += quantize_band_cost_cached(s, w + w2, g, coefs + w2*128,
|
|
scaled + w2*128,
|
|
sce->ics.swb_sizes[g],
|
|
sce->sf_idx[w*16+g]+1,
|
|
cb,
|
|
1.0f,
|
|
INFINITY,
|
|
&b, &sqenergy,
|
|
0);
|
|
bits += b;
|
|
qenergy += sqenergy;
|
|
}
|
|
dist -= bits;
|
|
if (dist < FFMIN(euplims[w*16+g], uplims[w*16+g])) {
|
|
sce->sf_idx[w*16+g]++;
|
|
dists[w*16+g] = dist;
|
|
qenergies[w*16+g] = qenergy;
|
|
} else {
|
|
break;
|
|
}
|
|
} else {
|
|
maxsf[w*16+g] = FFMIN(sce->sf_idx[w*16+g], maxsf[w*16+g]);
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
prev = sce->sf_idx[w*16+g] = av_clip(sce->sf_idx[w*16+g], mindeltasf, maxdeltasf);
|
|
if (sce->sf_idx[w*16+g] != prevsc)
|
|
fflag = 1;
|
|
nminscaler = FFMIN(nminscaler, sce->sf_idx[w*16+g]);
|
|
sce->band_type[w*16+g] = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]);
|
|
}
|
|
start += sce->ics.swb_sizes[g];
|
|
}
|
|
}
|
|
|
|
/** SF difference limit violation risk. Must re-clamp. */
|
|
prev = -1;
|
|
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->zeroes[w*16+g]) {
|
|
int prevsf = sce->sf_idx[w*16+g];
|
|
if (prev < 0)
|
|
prev = prevsf;
|
|
sce->sf_idx[w*16+g] = av_clip(sce->sf_idx[w*16+g], prev - SCALE_MAX_DIFF, prev + SCALE_MAX_DIFF);
|
|
sce->band_type[w*16+g] = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]);
|
|
prev = sce->sf_idx[w*16+g];
|
|
if (!fflag && prevsf != sce->sf_idx[w*16+g])
|
|
fflag = 1;
|
|
}
|
|
}
|
|
}
|
|
|
|
its++;
|
|
} while (fflag && its < maxits);
|
|
|
|
/** Scout out next nonzero bands */
|
|
ff_init_nextband_map(sce, nextband);
|
|
|
|
prev = -1;
|
|
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
|
|
/** Make sure proper codebooks are set */
|
|
for (g = 0; g < sce->ics.num_swb; g++) {
|
|
if (!sce->zeroes[w*16+g]) {
|
|
sce->band_type[w*16+g] = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]);
|
|
if (sce->band_type[w*16+g] <= 0) {
|
|
if (!ff_sfdelta_can_remove_band(sce, nextband, prev, w*16+g)) {
|
|
/** Cannot zero out, make sure it's not attempted */
|
|
sce->band_type[w*16+g] = 1;
|
|
} else {
|
|
sce->zeroes[w*16+g] = 1;
|
|
sce->band_type[w*16+g] = 0;
|
|
}
|
|
}
|
|
} else {
|
|
sce->band_type[w*16+g] = 0;
|
|
}
|
|
/** Check that there's no SF delta range violations */
|
|
if (!sce->zeroes[w*16+g]) {
|
|
if (prev != -1) {
|
|
av_unused int sfdiff = sce->sf_idx[w*16+g] - prev + SCALE_DIFF_ZERO;
|
|
av_assert1(sfdiff >= 0 && sfdiff <= 2*SCALE_MAX_DIFF);
|
|
} else if (sce->zeroes[0]) {
|
|
/** Set global gain to something useful */
|
|
sce->sf_idx[0] = sce->sf_idx[w*16+g];
|
|
}
|
|
prev = sce->sf_idx[w*16+g];
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
#endif /* AVCODEC_AACCODER_TWOLOOP_H */
|