625 lines
23 KiB
C
625 lines
23 KiB
C
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/*
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* E-AC-3 decoder
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* Copyright (c) 2007 Bartlomiej Wolowiec <bartek.wolowiec@gmail.com>
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* Copyright (c) 2008 Justin Ruggles
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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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* There are several features of E-AC-3 that this decoder does not yet support.
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*
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* Enhanced Coupling
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* No known samples exist. If any ever surface, this feature should not be
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* too difficult to implement.
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*
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* Reduced Sample Rates
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* No known samples exist. The spec also does not give clear information
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* on how this is to be implemented.
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*
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* Transient Pre-noise Processing
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* This is side information which a decoder should use to reduce artifacts
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* caused by transients. There are samples which are known to have this
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* information, but this decoder currently ignores it.
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*/
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#include "avcodec.h"
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#include "internal.h"
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#include "aac_ac3_parser.h"
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#include "ac3.h"
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#include "ac3dec.h"
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#include "ac3dec_data.h"
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#include "eac3_data.h"
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/** gain adaptive quantization mode */
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typedef enum {
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EAC3_GAQ_NO =0,
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EAC3_GAQ_12,
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EAC3_GAQ_14,
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EAC3_GAQ_124
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} EAC3GaqMode;
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#define EAC3_SR_CODE_REDUCED 3
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static void ff_eac3_apply_spectral_extension(AC3DecodeContext *s)
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{
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int bin, bnd, ch, i;
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uint8_t wrapflag[SPX_MAX_BANDS]={1,0,}, num_copy_sections, copy_sizes[SPX_MAX_BANDS];
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float rms_energy[SPX_MAX_BANDS];
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/* Set copy index mapping table. Set wrap flags to apply a notch filter at
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wrap points later on. */
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bin = s->spx_dst_start_freq;
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num_copy_sections = 0;
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for (bnd = 0; bnd < s->num_spx_bands; bnd++) {
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int copysize;
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int bandsize = s->spx_band_sizes[bnd];
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if (bin + bandsize > s->spx_src_start_freq) {
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copy_sizes[num_copy_sections++] = bin - s->spx_dst_start_freq;
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bin = s->spx_dst_start_freq;
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wrapflag[bnd] = 1;
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}
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for (i = 0; i < bandsize; i += copysize) {
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if (bin == s->spx_src_start_freq) {
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copy_sizes[num_copy_sections++] = bin - s->spx_dst_start_freq;
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bin = s->spx_dst_start_freq;
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}
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copysize = FFMIN(bandsize - i, s->spx_src_start_freq - bin);
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bin += copysize;
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}
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}
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copy_sizes[num_copy_sections++] = bin - s->spx_dst_start_freq;
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for (ch = 1; ch <= s->fbw_channels; ch++) {
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if (!s->channel_uses_spx[ch])
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continue;
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/* Copy coeffs from normal bands to extension bands */
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bin = s->spx_src_start_freq;
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for (i = 0; i < num_copy_sections; i++) {
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memcpy(&s->transform_coeffs[ch][bin],
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&s->transform_coeffs[ch][s->spx_dst_start_freq],
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copy_sizes[i]*sizeof(INTFLOAT));
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bin += copy_sizes[i];
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}
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/* Calculate RMS energy for each SPX band. */
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bin = s->spx_src_start_freq;
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for (bnd = 0; bnd < s->num_spx_bands; bnd++) {
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int bandsize = s->spx_band_sizes[bnd];
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float accum = 0.0f;
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for (i = 0; i < bandsize; i++) {
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float coeff = s->transform_coeffs[ch][bin++];
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accum += coeff * coeff;
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}
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rms_energy[bnd] = sqrtf(accum / bandsize);
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}
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/* Apply a notch filter at transitions between normal and extension
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bands and at all wrap points. */
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if (s->spx_atten_code[ch] >= 0) {
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const float *atten_tab = ff_eac3_spx_atten_tab[s->spx_atten_code[ch]];
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bin = s->spx_src_start_freq - 2;
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for (bnd = 0; bnd < s->num_spx_bands; bnd++) {
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if (wrapflag[bnd]) {
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INTFLOAT *coeffs = &s->transform_coeffs[ch][bin];
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coeffs[0] *= atten_tab[0];
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coeffs[1] *= atten_tab[1];
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coeffs[2] *= atten_tab[2];
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coeffs[3] *= atten_tab[1];
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coeffs[4] *= atten_tab[0];
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}
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bin += s->spx_band_sizes[bnd];
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}
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}
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/* Apply noise-blended coefficient scaling based on previously
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calculated RMS energy, blending factors, and SPX coordinates for
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each band. */
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bin = s->spx_src_start_freq;
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for (bnd = 0; bnd < s->num_spx_bands; bnd++) {
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float nscale = s->spx_noise_blend[ch][bnd] * rms_energy[bnd] * (1.0f / INT32_MIN);
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float sscale = s->spx_signal_blend[ch][bnd];
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#if USE_FIXED
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// spx_noise_blend and spx_signal_blend are both FP.23
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nscale *= 1.0 / (1<<23);
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sscale *= 1.0 / (1<<23);
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#endif
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for (i = 0; i < s->spx_band_sizes[bnd]; i++) {
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float noise = nscale * (int32_t)av_lfg_get(&s->dith_state);
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s->transform_coeffs[ch][bin] *= sscale;
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s->transform_coeffs[ch][bin++] += noise;
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}
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}
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}
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}
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/** lrint(M_SQRT2*cos(2*M_PI/12)*(1<<23)) */
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#define COEFF_0 10273905LL
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/** lrint(M_SQRT2*cos(0*M_PI/12)*(1<<23)) = lrint(M_SQRT2*(1<<23)) */
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#define COEFF_1 11863283LL
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/** lrint(M_SQRT2*cos(5*M_PI/12)*(1<<23)) */
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#define COEFF_2 3070444LL
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/**
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* Calculate 6-point IDCT of the pre-mantissas.
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* All calculations are 24-bit fixed-point.
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*/
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static void idct6(int pre_mant[6])
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{
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int tmp;
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int even0, even1, even2, odd0, odd1, odd2;
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odd1 = pre_mant[1] - pre_mant[3] - pre_mant[5];
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even2 = ( pre_mant[2] * COEFF_0) >> 23;
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tmp = ( pre_mant[4] * COEFF_1) >> 23;
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odd0 = ((pre_mant[1] + pre_mant[5]) * COEFF_2) >> 23;
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even0 = pre_mant[0] + (tmp >> 1);
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even1 = pre_mant[0] - tmp;
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tmp = even0;
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even0 = tmp + even2;
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even2 = tmp - even2;
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tmp = odd0;
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odd0 = tmp + pre_mant[1] + pre_mant[3];
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odd2 = tmp + pre_mant[5] - pre_mant[3];
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pre_mant[0] = even0 + odd0;
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pre_mant[1] = even1 + odd1;
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pre_mant[2] = even2 + odd2;
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pre_mant[3] = even2 - odd2;
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pre_mant[4] = even1 - odd1;
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pre_mant[5] = even0 - odd0;
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}
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static void ff_eac3_decode_transform_coeffs_aht_ch(AC3DecodeContext *s, int ch)
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{
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int bin, blk, gs;
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int end_bap, gaq_mode;
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GetBitContext *gbc = &s->gbc;
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int gaq_gain[AC3_MAX_COEFS];
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gaq_mode = get_bits(gbc, 2);
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end_bap = (gaq_mode < 2) ? 12 : 17;
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/* if GAQ gain is used, decode gain codes for bins with hebap between
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8 and end_bap */
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gs = 0;
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if (gaq_mode == EAC3_GAQ_12 || gaq_mode == EAC3_GAQ_14) {
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/* read 1-bit GAQ gain codes */
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for (bin = s->start_freq[ch]; bin < s->end_freq[ch]; bin++) {
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if (s->bap[ch][bin] > 7 && s->bap[ch][bin] < end_bap)
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gaq_gain[gs++] = get_bits1(gbc) << (gaq_mode-1);
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}
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} else if (gaq_mode == EAC3_GAQ_124) {
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/* read 1.67-bit GAQ gain codes (3 codes in 5 bits) */
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int gc = 2;
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for (bin = s->start_freq[ch]; bin < s->end_freq[ch]; bin++) {
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if (s->bap[ch][bin] > 7 && s->bap[ch][bin] < 17) {
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if (gc++ == 2) {
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int group_code = get_bits(gbc, 5);
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if (group_code > 26) {
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av_log(s->avctx, AV_LOG_WARNING, "GAQ gain group code out-of-range\n");
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group_code = 26;
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}
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gaq_gain[gs++] = ff_ac3_ungroup_3_in_5_bits_tab[group_code][0];
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gaq_gain[gs++] = ff_ac3_ungroup_3_in_5_bits_tab[group_code][1];
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gaq_gain[gs++] = ff_ac3_ungroup_3_in_5_bits_tab[group_code][2];
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gc = 0;
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}
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}
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}
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}
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gs=0;
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for (bin = s->start_freq[ch]; bin < s->end_freq[ch]; bin++) {
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int hebap = s->bap[ch][bin];
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int bits = ff_eac3_bits_vs_hebap[hebap];
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if (!hebap) {
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/* zero-mantissa dithering */
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for (blk = 0; blk < 6; blk++) {
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s->pre_mantissa[ch][bin][blk] = (av_lfg_get(&s->dith_state) & 0x7FFFFF) - 0x400000;
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}
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} else if (hebap < 8) {
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/* Vector Quantization */
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int v = get_bits(gbc, bits);
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for (blk = 0; blk < 6; blk++) {
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s->pre_mantissa[ch][bin][blk] = ff_eac3_mantissa_vq[hebap][v][blk] * (1 << 8);
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}
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} else {
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/* Gain Adaptive Quantization */
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int gbits, log_gain;
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if (gaq_mode != EAC3_GAQ_NO && hebap < end_bap) {
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log_gain = gaq_gain[gs++];
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} else {
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log_gain = 0;
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}
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gbits = bits - log_gain;
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for (blk = 0; blk < 6; blk++) {
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int mant = get_sbits(gbc, gbits);
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if (log_gain && mant == -(1 << (gbits-1))) {
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/* large mantissa */
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int b;
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int mbits = bits - (2 - log_gain);
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mant = get_sbits(gbc, mbits);
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mant = ((unsigned)mant) << (23 - (mbits - 1));
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/* remap mantissa value to correct for asymmetric quantization */
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if (mant >= 0)
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b = 1 << (23 - log_gain);
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else
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b = ff_eac3_gaq_remap_2_4_b[hebap-8][log_gain-1] * (1 << 8);
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mant += ((ff_eac3_gaq_remap_2_4_a[hebap-8][log_gain-1] * (int64_t)mant) >> 15) + b;
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} else {
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/* small mantissa, no GAQ, or Gk=1 */
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mant *= (1 << 24 - bits);
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if (!log_gain) {
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/* remap mantissa value for no GAQ or Gk=1 */
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mant += (ff_eac3_gaq_remap_1[hebap-8] * (int64_t)mant) >> 15;
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}
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}
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s->pre_mantissa[ch][bin][blk] = mant;
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}
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}
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idct6(s->pre_mantissa[ch][bin]);
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}
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}
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static int ff_eac3_parse_header(AC3DecodeContext *s)
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{
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int i, blk, ch;
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int ac3_exponent_strategy, parse_aht_info, parse_spx_atten_data;
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int parse_transient_proc_info;
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int num_cpl_blocks;
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GetBitContext *gbc = &s->gbc;
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/* An E-AC-3 stream can have multiple independent streams which the
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application can select from. each independent stream can also contain
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dependent streams which are used to add or replace channels. */
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if (s->frame_type == EAC3_FRAME_TYPE_RESERVED) {
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av_log(s->avctx, AV_LOG_ERROR, "Reserved frame type\n");
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return AAC_AC3_PARSE_ERROR_FRAME_TYPE;
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}
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/* The substream id indicates which substream this frame belongs to. each
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independent stream has its own substream id, and the dependent streams
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associated to an independent stream have matching substream id's. */
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if (s->substreamid) {
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/* only decode substream with id=0. skip any additional substreams. */
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if (!s->eac3_subsbtreamid_found) {
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s->eac3_subsbtreamid_found = 1;
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avpriv_request_sample(s->avctx, "Additional substreams");
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}
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return AAC_AC3_PARSE_ERROR_FRAME_TYPE;
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}
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if (s->bit_alloc_params.sr_code == EAC3_SR_CODE_REDUCED) {
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/* The E-AC-3 specification does not tell how to handle reduced sample
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rates in bit allocation. The best assumption would be that it is
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handled like AC-3 DolbyNet, but we cannot be sure until we have a
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sample which utilizes this feature. */
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avpriv_request_sample(s->avctx, "Reduced sampling rate");
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return AVERROR_PATCHWELCOME;
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}
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skip_bits(gbc, 5); // skip bitstream id
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/* volume control params */
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for (i = 0; i < (s->channel_mode ? 1 : 2); i++) {
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s->dialog_normalization[i] = -get_bits(gbc, 5);
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if (s->dialog_normalization[i] == 0) {
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s->dialog_normalization[i] = -31;
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}
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if (s->target_level != 0) {
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s->level_gain[i] = powf(2.0f,
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(float)(s->target_level - s->dialog_normalization[i])/6.0f);
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}
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s->compression_exists[i] = get_bits1(gbc);
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if (s->compression_exists[i]) {
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s->heavy_dynamic_range[i] = AC3_HEAVY_RANGE(get_bits(gbc, 8));
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}
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}
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/* dependent stream channel map */
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if (s->frame_type == EAC3_FRAME_TYPE_DEPENDENT) {
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if (get_bits1(gbc)) {
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int64_t channel_layout = 0;
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int channel_map = get_bits(gbc, 16);
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av_log(s->avctx, AV_LOG_DEBUG, "channel_map: %0X\n", channel_map);
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for (i = 0; i < 16; i++)
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if (channel_map & (1 << (EAC3_MAX_CHANNELS - i - 1)))
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channel_layout |= ff_eac3_custom_channel_map_locations[i][1];
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if (av_popcount64(channel_layout) > EAC3_MAX_CHANNELS) {
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return AVERROR_INVALIDDATA;
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}
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s->channel_map = channel_map;
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}
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}
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/* mixing metadata */
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if (get_bits1(gbc)) {
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/* center and surround mix levels */
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if (s->channel_mode > AC3_CHMODE_STEREO) {
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s->preferred_downmix = get_bits(gbc, 2);
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if (s->channel_mode & 1) {
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/* if three front channels exist */
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s->center_mix_level_ltrt = get_bits(gbc, 3);
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s->center_mix_level = get_bits(gbc, 3);
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}
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if (s->channel_mode & 4) {
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|
/* if a surround channel exists */
|
||
|
s->surround_mix_level_ltrt = av_clip(get_bits(gbc, 3), 3, 7);
|
||
|
s->surround_mix_level = av_clip(get_bits(gbc, 3), 3, 7);
|
||
|
}
|
||
|
}
|
||
|
|
||
|
/* lfe mix level */
|
||
|
if (s->lfe_on && (s->lfe_mix_level_exists = get_bits1(gbc))) {
|
||
|
s->lfe_mix_level = get_bits(gbc, 5);
|
||
|
}
|
||
|
|
||
|
/* info for mixing with other streams and substreams */
|
||
|
if (s->frame_type == EAC3_FRAME_TYPE_INDEPENDENT) {
|
||
|
for (i = 0; i < (s->channel_mode ? 1 : 2); i++) {
|
||
|
// TODO: apply program scale factor
|
||
|
if (get_bits1(gbc)) {
|
||
|
skip_bits(gbc, 6); // skip program scale factor
|
||
|
}
|
||
|
}
|
||
|
if (get_bits1(gbc)) {
|
||
|
skip_bits(gbc, 6); // skip external program scale factor
|
||
|
}
|
||
|
/* skip mixing parameter data */
|
||
|
switch(get_bits(gbc, 2)) {
|
||
|
case 1: skip_bits(gbc, 5); break;
|
||
|
case 2: skip_bits(gbc, 12); break;
|
||
|
case 3: {
|
||
|
int mix_data_size = (get_bits(gbc, 5) + 2) << 3;
|
||
|
skip_bits_long(gbc, mix_data_size);
|
||
|
break;
|
||
|
}
|
||
|
}
|
||
|
/* skip pan information for mono or dual mono source */
|
||
|
if (s->channel_mode < AC3_CHMODE_STEREO) {
|
||
|
for (i = 0; i < (s->channel_mode ? 1 : 2); i++) {
|
||
|
if (get_bits1(gbc)) {
|
||
|
/* note: this is not in the ATSC A/52B specification
|
||
|
reference: ETSI TS 102 366 V1.1.1
|
||
|
section: E.1.3.1.25 */
|
||
|
skip_bits(gbc, 8); // skip pan mean direction index
|
||
|
skip_bits(gbc, 6); // skip reserved paninfo bits
|
||
|
}
|
||
|
}
|
||
|
}
|
||
|
/* skip mixing configuration information */
|
||
|
if (get_bits1(gbc)) {
|
||
|
for (blk = 0; blk < s->num_blocks; blk++) {
|
||
|
if (s->num_blocks == 1 || get_bits1(gbc)) {
|
||
|
skip_bits(gbc, 5);
|
||
|
}
|
||
|
}
|
||
|
}
|
||
|
}
|
||
|
}
|
||
|
|
||
|
/* informational metadata */
|
||
|
if (get_bits1(gbc)) {
|
||
|
s->bitstream_mode = get_bits(gbc, 3);
|
||
|
skip_bits(gbc, 2); // skip copyright bit and original bitstream bit
|
||
|
if (s->channel_mode == AC3_CHMODE_STEREO) {
|
||
|
s->dolby_surround_mode = get_bits(gbc, 2);
|
||
|
s->dolby_headphone_mode = get_bits(gbc, 2);
|
||
|
}
|
||
|
if (s->channel_mode >= AC3_CHMODE_2F2R) {
|
||
|
s->dolby_surround_ex_mode = get_bits(gbc, 2);
|
||
|
}
|
||
|
for (i = 0; i < (s->channel_mode ? 1 : 2); i++) {
|
||
|
if (get_bits1(gbc)) {
|
||
|
skip_bits(gbc, 8); // skip mix level, room type, and A/D converter type
|
||
|
}
|
||
|
}
|
||
|
if (s->bit_alloc_params.sr_code != EAC3_SR_CODE_REDUCED) {
|
||
|
skip_bits1(gbc); // skip source sample rate code
|
||
|
}
|
||
|
}
|
||
|
|
||
|
/* converter synchronization flag
|
||
|
If frames are less than six blocks, this bit should be turned on
|
||
|
once every 6 blocks to indicate the start of a frame set.
|
||
|
reference: RFC 4598, Section 2.1.3 Frame Sets */
|
||
|
if (s->frame_type == EAC3_FRAME_TYPE_INDEPENDENT && s->num_blocks != 6) {
|
||
|
skip_bits1(gbc); // skip converter synchronization flag
|
||
|
}
|
||
|
|
||
|
/* original frame size code if this stream was converted from AC-3 */
|
||
|
if (s->frame_type == EAC3_FRAME_TYPE_AC3_CONVERT &&
|
||
|
(s->num_blocks == 6 || get_bits1(gbc))) {
|
||
|
skip_bits(gbc, 6); // skip frame size code
|
||
|
}
|
||
|
|
||
|
/* additional bitstream info */
|
||
|
if (get_bits1(gbc)) {
|
||
|
int addbsil = get_bits(gbc, 6);
|
||
|
for (i = 0; i < addbsil + 1; i++) {
|
||
|
skip_bits(gbc, 8); // skip additional bit stream info
|
||
|
}
|
||
|
}
|
||
|
|
||
|
/* audio frame syntax flags, strategy data, and per-frame data */
|
||
|
|
||
|
if (s->num_blocks == 6) {
|
||
|
ac3_exponent_strategy = get_bits1(gbc);
|
||
|
parse_aht_info = get_bits1(gbc);
|
||
|
} else {
|
||
|
/* less than 6 blocks, so use AC-3-style exponent strategy syntax, and
|
||
|
do not use AHT */
|
||
|
ac3_exponent_strategy = 1;
|
||
|
parse_aht_info = 0;
|
||
|
}
|
||
|
|
||
|
s->snr_offset_strategy = get_bits(gbc, 2);
|
||
|
parse_transient_proc_info = get_bits1(gbc);
|
||
|
|
||
|
s->block_switch_syntax = get_bits1(gbc);
|
||
|
if (!s->block_switch_syntax)
|
||
|
memset(s->block_switch, 0, sizeof(s->block_switch));
|
||
|
|
||
|
s->dither_flag_syntax = get_bits1(gbc);
|
||
|
if (!s->dither_flag_syntax) {
|
||
|
for (ch = 1; ch <= s->fbw_channels; ch++)
|
||
|
s->dither_flag[ch] = 1;
|
||
|
}
|
||
|
s->dither_flag[CPL_CH] = s->dither_flag[s->lfe_ch] = 0;
|
||
|
|
||
|
s->bit_allocation_syntax = get_bits1(gbc);
|
||
|
if (!s->bit_allocation_syntax) {
|
||
|
/* set default bit allocation parameters */
|
||
|
s->bit_alloc_params.slow_decay = ff_ac3_slow_decay_tab[2];
|
||
|
s->bit_alloc_params.fast_decay = ff_ac3_fast_decay_tab[1];
|
||
|
s->bit_alloc_params.slow_gain = ff_ac3_slow_gain_tab [1];
|
||
|
s->bit_alloc_params.db_per_bit = ff_ac3_db_per_bit_tab[2];
|
||
|
s->bit_alloc_params.floor = ff_ac3_floor_tab [7];
|
||
|
}
|
||
|
|
||
|
s->fast_gain_syntax = get_bits1(gbc);
|
||
|
s->dba_syntax = get_bits1(gbc);
|
||
|
s->skip_syntax = get_bits1(gbc);
|
||
|
parse_spx_atten_data = get_bits1(gbc);
|
||
|
|
||
|
/* coupling strategy occurrence and coupling use per block */
|
||
|
num_cpl_blocks = 0;
|
||
|
if (s->channel_mode > 1) {
|
||
|
for (blk = 0; blk < s->num_blocks; blk++) {
|
||
|
s->cpl_strategy_exists[blk] = (!blk || get_bits1(gbc));
|
||
|
if (s->cpl_strategy_exists[blk]) {
|
||
|
s->cpl_in_use[blk] = get_bits1(gbc);
|
||
|
} else {
|
||
|
s->cpl_in_use[blk] = s->cpl_in_use[blk-1];
|
||
|
}
|
||
|
num_cpl_blocks += s->cpl_in_use[blk];
|
||
|
}
|
||
|
} else {
|
||
|
memset(s->cpl_in_use, 0, sizeof(s->cpl_in_use));
|
||
|
}
|
||
|
|
||
|
/* exponent strategy data */
|
||
|
if (ac3_exponent_strategy) {
|
||
|
/* AC-3-style exponent strategy syntax */
|
||
|
for (blk = 0; blk < s->num_blocks; blk++) {
|
||
|
for (ch = !s->cpl_in_use[blk]; ch <= s->fbw_channels; ch++) {
|
||
|
s->exp_strategy[blk][ch] = get_bits(gbc, 2);
|
||
|
}
|
||
|
}
|
||
|
} else {
|
||
|
/* LUT-based exponent strategy syntax */
|
||
|
for (ch = !((s->channel_mode > 1) && num_cpl_blocks); ch <= s->fbw_channels; ch++) {
|
||
|
int frmchexpstr = get_bits(gbc, 5);
|
||
|
for (blk = 0; blk < 6; blk++) {
|
||
|
s->exp_strategy[blk][ch] = ff_eac3_frm_expstr[frmchexpstr][blk];
|
||
|
}
|
||
|
}
|
||
|
}
|
||
|
/* LFE exponent strategy */
|
||
|
if (s->lfe_on) {
|
||
|
for (blk = 0; blk < s->num_blocks; blk++) {
|
||
|
s->exp_strategy[blk][s->lfe_ch] = get_bits1(gbc);
|
||
|
}
|
||
|
}
|
||
|
/* original exponent strategies if this stream was converted from AC-3 */
|
||
|
if (s->frame_type == EAC3_FRAME_TYPE_INDEPENDENT &&
|
||
|
(s->num_blocks == 6 || get_bits1(gbc))) {
|
||
|
skip_bits(gbc, 5 * s->fbw_channels); // skip converter channel exponent strategy
|
||
|
}
|
||
|
|
||
|
/* determine which channels use AHT */
|
||
|
if (parse_aht_info) {
|
||
|
/* For AHT to be used, all non-zero blocks must reuse exponents from
|
||
|
the first block. Furthermore, for AHT to be used in the coupling
|
||
|
channel, all blocks must use coupling and use the same coupling
|
||
|
strategy. */
|
||
|
s->channel_uses_aht[CPL_CH]=0;
|
||
|
for (ch = (num_cpl_blocks != 6); ch <= s->channels; ch++) {
|
||
|
int use_aht = 1;
|
||
|
for (blk = 1; blk < 6; blk++) {
|
||
|
if ((s->exp_strategy[blk][ch] != EXP_REUSE) ||
|
||
|
(!ch && s->cpl_strategy_exists[blk])) {
|
||
|
use_aht = 0;
|
||
|
break;
|
||
|
}
|
||
|
}
|
||
|
s->channel_uses_aht[ch] = use_aht && get_bits1(gbc);
|
||
|
}
|
||
|
} else {
|
||
|
memset(s->channel_uses_aht, 0, sizeof(s->channel_uses_aht));
|
||
|
}
|
||
|
|
||
|
/* per-frame SNR offset */
|
||
|
if (!s->snr_offset_strategy) {
|
||
|
int csnroffst = (get_bits(gbc, 6) - 15) << 4;
|
||
|
int snroffst = (csnroffst + get_bits(gbc, 4)) << 2;
|
||
|
for (ch = 0; ch <= s->channels; ch++)
|
||
|
s->snr_offset[ch] = snroffst;
|
||
|
}
|
||
|
|
||
|
/* transient pre-noise processing data */
|
||
|
if (parse_transient_proc_info) {
|
||
|
for (ch = 1; ch <= s->fbw_channels; ch++) {
|
||
|
if (get_bits1(gbc)) { // channel in transient processing
|
||
|
skip_bits(gbc, 10); // skip transient processing location
|
||
|
skip_bits(gbc, 8); // skip transient processing length
|
||
|
}
|
||
|
}
|
||
|
}
|
||
|
|
||
|
/* spectral extension attenuation data */
|
||
|
for (ch = 1; ch <= s->fbw_channels; ch++) {
|
||
|
if (parse_spx_atten_data && get_bits1(gbc)) {
|
||
|
s->spx_atten_code[ch] = get_bits(gbc, 5);
|
||
|
} else {
|
||
|
s->spx_atten_code[ch] = -1;
|
||
|
}
|
||
|
}
|
||
|
|
||
|
/* block start information */
|
||
|
if (s->num_blocks > 1 && get_bits1(gbc)) {
|
||
|
/* reference: Section E2.3.2.27
|
||
|
nblkstrtbits = (numblks - 1) * (4 + ceiling(log2(words_per_frame)))
|
||
|
The spec does not say what this data is or what it's used for.
|
||
|
It is likely the offset of each block within the frame. */
|
||
|
int block_start_bits = (s->num_blocks-1) * (4 + av_log2(s->frame_size-2));
|
||
|
skip_bits_long(gbc, block_start_bits);
|
||
|
avpriv_request_sample(s->avctx, "Block start info");
|
||
|
}
|
||
|
|
||
|
/* syntax state initialization */
|
||
|
for (ch = 1; ch <= s->fbw_channels; ch++) {
|
||
|
s->first_spx_coords[ch] = 1;
|
||
|
s->first_cpl_coords[ch] = 1;
|
||
|
}
|
||
|
s->first_cpl_leak = 1;
|
||
|
|
||
|
return 0;
|
||
|
}
|