summaryrefslogtreecommitdiffstats
path: root/libavcodec/aaccoder_twoloop.h
diff options
context:
space:
mode:
Diffstat (limited to 'libavcodec/aaccoder_twoloop.h')
-rw-r--r--libavcodec/aaccoder_twoloop.h763
1 files changed, 763 insertions, 0 deletions
diff --git a/libavcodec/aaccoder_twoloop.h b/libavcodec/aaccoder_twoloop.h
new file mode 100644
index 0000000..8e1bc88
--- /dev/null
+++ b/libavcodec/aaccoder_twoloop.h
@@ -0,0 +1,763 @@
+/*
+ * 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 & AV_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 necessary.
+ * 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 & AV_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 & AV_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;
+ s->abs_pow34(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 & AV_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 & AV_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 */
OpenPOWER on IntegriCloud