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https://github.com/FFmpeg/FFmpeg.git
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204 lines
7.7 KiB
C
204 lines
7.7 KiB
C
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/*
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* AAC encoder twoloop coder
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* Copyright (C) 2008-2009 Konstantin Shishkov
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* AAC encoder twoloop coder
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* @author Konstantin Shishkov
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*/
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/**
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* This file contains a template for the twoloop coder function.
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* It needs to be provided, externally, as an already included declaration,
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* the following functions from aacenc_quantization/util.h. They're not included
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* explicitly here to make it possible to provide alternative implementations:
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* - quantize_band_cost
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* - abs_pow34_v
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* - find_max_val
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* - find_min_book
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*/
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#ifndef AVCODEC_AACCODER_TWOLOOP_H
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#define AVCODEC_AACCODER_TWOLOOP_H
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#include <float.h>
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#include "libavutil/mathematics.h"
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#include "avcodec.h"
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#include "put_bits.h"
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#include "aac.h"
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#include "aacenc.h"
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#include "aactab.h"
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#include "aacenctab.h"
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#include "aac_tablegen_decl.h"
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/**
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* two-loop quantizers search taken from ISO 13818-7 Appendix C
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*/
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static void search_for_quantizers_twoloop(AVCodecContext *avctx,
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AACEncContext *s,
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SingleChannelElement *sce,
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const float lambda)
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{
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int start = 0, i, w, w2, g;
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int destbits = avctx->bit_rate * 1024.0 / avctx->sample_rate / avctx->channels * (lambda / 120.f);
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float dists[128] = { 0 }, uplims[128] = { 0 };
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float maxvals[128];
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int fflag, minscaler;
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int its = 0;
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int allz = 0;
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float minthr = INFINITY;
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// for values above this the decoder might end up in an endless loop
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// due to always having more bits than what can be encoded.
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destbits = FFMIN(destbits, 5800);
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//XXX: some heuristic to determine initial quantizers will reduce search time
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//determine zero bands and upper limits
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for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
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for (g = 0; g < sce->ics.num_swb; g++) {
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int nz = 0;
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float uplim = 0.0f, energy = 0.0f;
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for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
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FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[(w+w2)*16+g];
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uplim += band->threshold;
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energy += band->energy;
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if (band->energy <= band->threshold || band->threshold == 0.0f) {
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sce->zeroes[(w+w2)*16+g] = 1;
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continue;
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}
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nz = 1;
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}
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uplims[w*16+g] = uplim *512;
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sce->zeroes[w*16+g] = !nz;
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if (nz)
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minthr = FFMIN(minthr, uplim);
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allz |= nz;
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}
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}
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for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
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for (g = 0; g < sce->ics.num_swb; g++) {
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if (sce->zeroes[w*16+g]) {
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sce->sf_idx[w*16+g] = SCALE_ONE_POS;
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continue;
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}
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sce->sf_idx[w*16+g] = SCALE_ONE_POS + FFMIN(log2f(uplims[w*16+g]/minthr)*4,59);
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}
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}
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if (!allz)
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return;
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abs_pow34_v(s->scoefs, sce->coeffs, 1024);
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for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
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start = w*128;
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for (g = 0; g < sce->ics.num_swb; g++) {
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const float *scaled = s->scoefs + start;
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maxvals[w*16+g] = find_max_val(sce->ics.group_len[w], sce->ics.swb_sizes[g], scaled);
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start += sce->ics.swb_sizes[g];
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}
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}
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//perform two-loop search
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//outer loop - improve quality
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do {
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int tbits, qstep;
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minscaler = sce->sf_idx[0];
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//inner loop - quantize spectrum to fit into given number of bits
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qstep = its ? 1 : 32;
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do {
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int prev = -1;
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tbits = 0;
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for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
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start = w*128;
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for (g = 0; g < sce->ics.num_swb; g++) {
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const float *coefs = &sce->coeffs[start];
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const float *scaled = &s->scoefs[start];
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int bits = 0;
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int cb;
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float dist = 0.0f;
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if (sce->zeroes[w*16+g] || sce->sf_idx[w*16+g] >= 218) {
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start += sce->ics.swb_sizes[g];
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continue;
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}
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minscaler = FFMIN(minscaler, sce->sf_idx[w*16+g]);
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cb = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]);
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for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
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int b;
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dist += quantize_band_cost(s, coefs + w2*128,
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scaled + w2*128,
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sce->ics.swb_sizes[g],
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sce->sf_idx[w*16+g],
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cb,
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1.0f,
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INFINITY,
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&b,
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0);
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bits += b;
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}
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dists[w*16+g] = dist - bits;
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if (prev != -1) {
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bits += ff_aac_scalefactor_bits[sce->sf_idx[w*16+g] - prev + SCALE_DIFF_ZERO];
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}
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tbits += bits;
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start += sce->ics.swb_sizes[g];
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prev = sce->sf_idx[w*16+g];
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}
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}
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if (tbits > destbits) {
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for (i = 0; i < 128; i++)
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if (sce->sf_idx[i] < 218 - qstep)
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sce->sf_idx[i] += qstep;
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} else {
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for (i = 0; i < 128; i++)
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if (sce->sf_idx[i] > 60 - qstep)
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sce->sf_idx[i] -= qstep;
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}
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qstep >>= 1;
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if (!qstep && tbits > destbits*1.02 && sce->sf_idx[0] < 217)
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qstep = 1;
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} while (qstep);
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fflag = 0;
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minscaler = av_clip(minscaler, 60, 255 - SCALE_MAX_DIFF);
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for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
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for (g = 0; g < sce->ics.num_swb; g++) {
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int prevsc = sce->sf_idx[w*16+g];
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if (dists[w*16+g] > uplims[w*16+g] && sce->sf_idx[w*16+g] > 60) {
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if (find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]-1))
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sce->sf_idx[w*16+g]--;
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else //Try to make sure there is some energy in every band
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sce->sf_idx[w*16+g]-=2;
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}
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sce->sf_idx[w*16+g] = av_clip(sce->sf_idx[w*16+g], minscaler, minscaler + SCALE_MAX_DIFF);
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sce->sf_idx[w*16+g] = FFMIN(sce->sf_idx[w*16+g], 219);
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if (sce->sf_idx[w*16+g] != prevsc)
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fflag = 1;
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sce->band_type[w*16+g] = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]);
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}
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}
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its++;
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} while (fflag && its < 10);
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}
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#endif /* AVCODEC_AACCODER_TWOLOOP_H */
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