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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
+ */
+
+/**
+ * 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"
+
+
+/**
+ * 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;
+ int destbits = avctx->bit_rate * 1024.0 / avctx->sample_rate / avctx->channels * (lambda / 120.f);
+ float dists[128] = { 0 }, uplims[128] = { 0 };
+ float maxvals[128];
+ int fflag, minscaler;
+ int its = 0;
+ int allz = 0;
+ float minthr = INFINITY;
+
+ // 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);
+ //XXX: some heuristic to determine initial quantizers will reduce search time
+ //determine zero bands and upper limits
+ for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
+ for (g = 0; g < sce->ics.num_swb; g++) {
+ int nz = 0;
+ float uplim = 0.0f, energy = 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];
+ uplim += band->threshold;
+ energy += band->energy;
+ if (band->energy <= band->threshold || band->threshold == 0.0f) {
+ sce->zeroes[(w+w2)*16+g] = 1;
+ continue;
+ }
+ nz = 1;
+ }
+ uplims[w*16+g] = uplim *512;
+ sce->zeroes[w*16+g] = !nz;
+ if (nz)
+ minthr = FFMIN(minthr, uplim);
+ allz |= nz;
+ }
+ }
+ 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;
+ }
+ sce->sf_idx[w*16+g] = SCALE_ONE_POS + FFMIN(log2f(uplims[w*16+g]/minthr)*4,59);
+ }
+ }
+
+ if (!allz)
+ return;
+ abs_pow34_v(s->scoefs, sce->coeffs, 1024);
+
+ 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];
+ }
+ }
+
+ //perform two-loop search
+ //outer loop - improve quality
+ do {
+ int tbits, qstep;
+ minscaler = sce->sf_idx[0];
+ //inner loop - quantize spectrum to fit into given number of bits
+ qstep = its ? 1 : 32;
+ do {
+ int 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;
+
+ if (sce->zeroes[w*16+g] || sce->sf_idx[w*16+g] >= 218) {
+ start += sce->ics.swb_sizes[g];
+ continue;
+ }
+ minscaler = FFMIN(minscaler, sce->sf_idx[w*16+g]);
+ 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(s, coefs + w2*128,
+ scaled + w2*128,
+ sce->ics.swb_sizes[g],
+ sce->sf_idx[w*16+g],
+ cb,
+ 1.0f,
+ INFINITY,
+ &b,
+ 0);
+ bits += b;
+ }
+ dists[w*16+g] = dist - bits;
+ if (prev != -1) {
+ bits += ff_aac_scalefactor_bits[sce->sf_idx[w*16+g] - prev + SCALE_DIFF_ZERO];
+ }
+ tbits += bits;
+ start += sce->ics.swb_sizes[g];
+ prev = sce->sf_idx[w*16+g];
+ }
+ }
+ if (tbits > destbits) {
+ for (i = 0; i < 128; i++)
+ if (sce->sf_idx[i] < 218 - qstep)
+ sce->sf_idx[i] += qstep;
+ } else {
+ for (i = 0; i < 128; i++)
+ if (sce->sf_idx[i] > 60 - qstep)
+ sce->sf_idx[i] -= qstep;
+ }
+ qstep >>= 1;
+ if (!qstep && tbits > destbits*1.02 && sce->sf_idx[0] < 217)
+ qstep = 1;
+ } while (qstep);
+
+ fflag = 0;
+ minscaler = av_clip(minscaler, 60, 255 - SCALE_MAX_DIFF);
+
+ for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
+ for (g = 0; g < sce->ics.num_swb; g++) {
+ int prevsc = sce->sf_idx[w*16+g];
+ if (dists[w*16+g] > uplims[w*16+g] && sce->sf_idx[w*16+g] > 60) {
+ if (find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]-1))
+ sce->sf_idx[w*16+g]--;
+ else //Try to make sure there is some energy in every band
+ sce->sf_idx[w*16+g]-=2;
+ }
+ sce->sf_idx[w*16+g] = av_clip(sce->sf_idx[w*16+g], minscaler, minscaler + SCALE_MAX_DIFF);
+ sce->sf_idx[w*16+g] = FFMIN(sce->sf_idx[w*16+g], 219);
+ if (sce->sf_idx[w*16+g] != prevsc)
+ fflag = 1;
+ sce->band_type[w*16+g] = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]);
+ }
+ }
+ its++;
+ } while (fflag && its < 10);
+}
+
+#endif /* AVCODEC_AACCODER_TWOLOOP_H */
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