/* * 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 #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 */