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/*
* 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
*/
#ifndef AVCODEC_AACCODER_TWOLOOP_H
#define AVCODEC_AACCODER_TWOLOOP_H
#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 "aac_tablegen_decl.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)
{
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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];
int maxsf[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, minrdsf;
int tbits;
int cutoff = 1024;
int pns_start_pos;
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/**
* 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)
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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 */
toomanybits = destbits + destbits/16;
toofewbits = destbits - destbits/4;
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));
}
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++]) {
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) {
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 (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;
maxvals[w*16+g] = find_max_val(sce->ics.group_len[w], sce->ics.swb_sizes[g], scaled);
start += sce->ics.swb_sizes[g];
}
}
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/**
* 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;
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;
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;
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,
}
dists[w*16+g] = dist - bits;
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 = FFMAX(SCALE_ONE_POS, sce->sf_idx[i] - qstep);
if (new_sf != sce->sf_idx[i]) {
sce->sf_idx[i] = new_sf;
changed = 1;
}
}
}
if (!qstep && tbits > toomanybits && sce->sf_idx[0] < 217 && changed)
qstep = 1;
} while (qstep);
overdist = 1;
for (i = 0; i < 2 && (overdist || recomprd); ++i) {
if (recomprd) {
/** Must recompute distortion */
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);
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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) {
float maxoverdist = 0.0f;
overdist = recomprd = 0;
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
float ovrfactor = 2.f+(maxits-its)*16.f/maxits;
for (g = start = 0; g < sce->ics.num_swb; start += sce->ics.swb_sizes[g++]) {
if (!sce->zeroes[w*16+g] && 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;
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;
zspread = FFMIN(maxoverdist, zspread);
maxzeroed = zeroable * its / (2 * maxits);
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->zeroes[w*16+g] = 1;
sce->band_type[w*16+g] = 0;
zeroed++;
}
}
}
if (zeroed)
recomprd = 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]);
}
}
}
fflag = 0;
minscaler = nminscaler = av_clip(minscaler, SCALE_ONE_POS - SCALE_DIV_512, SCALE_MAX_POS - SCALE_DIV_512);
minrdsf = FFMAX3(60, minscaler - 1, maxscaler - SCALE_MAX_DIFF - 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];
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int minrdsfboost = (sce->ics.num_windows > 1) ? av_clip(g-4, -2, 0) : av_clip(g-16, -4, 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]);
if ((!cmb || dists[w*16+g] > uplims[w*16+g]) && sce->sf_idx[w*16+g] > minrdsf) {
/* 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; ++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;
}
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,
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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] < (minrdsf+minrdsfboost) || (
(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] < maxscaler
&& (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; ++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;
}
}
}
sce->sf_idx[w*16+g] = av_clip(sce->sf_idx[w*16+g], minrdsf, minscaler + SCALE_MAX_DIFF);
sce->sf_idx[w*16+g] = FFMIN(sce->sf_idx[w*16+g], SCALE_MAX_POS - SCALE_DIV_512);
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];
}
}
if (nminscaler < minscaler || sce->ics.num_windows > 1) {
/** SF difference limit violation risk. Must re-clamp. */
minscaler = nminscaler;
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
for (g = 0; g < sce->ics.num_swb; g++) {
sce->sf_idx[w*16+g] = av_clip(sce->sf_idx[w*16+g], minscaler, minscaler + SCALE_MAX_DIFF);
sce->band_type[w*16+g] = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]);
}
} while (fflag && its < maxits);
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
/** Make sure proper codebooks are set */
for (g = start = 0; g < sce->ics.num_swb; start += sce->ics.swb_sizes[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) {
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) {
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 */