64 5, 6, 7, 8, 9, 10, 11, 12, 14, 16
88 3.162275, 2.818382, 2.511886, 2.238719, 1.995261, 1.778278, 1.584893,
89 1.412536, 1.258924, 1.122018, 1.000000, 0.891251, 0.794328, 0.707946,
90 0.630957, 0.562341, 0.501187, 0.446683, 0.398107, 0.354813, 0.316227,
91 0.281838, 0.251188, 0.223872, 0.199526, 0.177828, 0.158489, 0.141253,
92 0.125892, 0.112201, 0.100000, 0.089125
100 { { 2, 7 }, { 7, 2 }, },
102 { { 2, 7 }, { 7, 2 }, },
103 { { 2, 7 }, { 5, 5 }, { 7, 2 }, },
104 { { 2, 7 }, { 7, 2 }, { 6, 6 }, },
105 { { 2, 7 }, { 5, 5 }, { 7, 2 }, { 8, 8 }, },
106 { { 2, 7 }, { 7, 2 }, { 6, 7 }, { 7, 6 }, },
107 { { 2, 7 }, { 5, 5 }, { 7, 2 }, { 6, 7 }, { 7, 6 }, },
118 return ((
code - (levels >> 1)) * (1 << 24)) / levels;
130 for (
i = 0;
i < 128;
i++) {
138 for (
i = 0;
i < 32;
i++) {
144 for (
i = 0;
i < 128;
i++) {
156 for (
i = 0;
i < 7;
i++) {
160 for (
i = 0;
i < 15;
i++) {
168 for (
i = 0;
i < 256;
i++) {
169 int v = (
i >> 5) - ((
i >> 7) << 3) - 5;
175 for (
i = 0;
i < 256;
i++) {
176 int v = (
i >> 4) - ((
i >> 7) << 4) - 4;
226 s->xcfptr[
i] =
s->transform_coeffs[
i];
227 s->dlyptr[
i] =
s->delay[
i];
246 i = !
s->channel_mode;
248 s->dialog_normalization[(!
s->channel_mode)-
i] = -
get_bits(gbc, 5);
249 if (
s->dialog_normalization[(!
s->channel_mode)-
i] == 0) {
250 s->dialog_normalization[(!
s->channel_mode)-
i] = -31;
252 if (
s->target_level != 0) {
253 s->level_gain[(!
s->channel_mode)-
i] =
powf(2.0f,
254 (
float)(
s->target_level -
255 s->dialog_normalization[(!
s->channel_mode)-
i])/6.0f);
257 if (
s->compression_exists[(!
s->channel_mode)-
i] =
get_bits1(gbc)) {
258 s->heavy_dynamic_range[(!
s->channel_mode)-
i] =
270 if (
s->bitstream_id != 6) {
278 s->center_mix_level_ltrt =
get_bits(gbc, 3);
284 s->dolby_surround_ex_mode =
get_bits(gbc, 2);
285 s->dolby_headphone_mode =
get_bits(gbc, 2);
314 s->bit_alloc_params.sr_code = hdr.
sr_code;
319 s->bit_alloc_params.sr_shift = hdr.
sr_shift;
323 s->fbw_channels =
s->channels -
s->lfe_on;
324 s->lfe_ch =
s->fbw_channels + 1;
329 s->center_mix_level_ltrt = 4;
331 s->surround_mix_level_ltrt = 4;
332 s->lfe_mix_level_exists = 0;
341 s->start_freq[
s->lfe_ch] = 0;
342 s->end_freq[
s->lfe_ch] = 7;
343 s->num_exp_groups[
s->lfe_ch] = 2;
344 s->channel_in_cpl[
s->lfe_ch] = 0;
347 if (
s->bitstream_id <= 10) {
349 s->snr_offset_strategy = 2;
350 s->block_switch_syntax = 1;
351 s->dither_flag_syntax = 1;
352 s->bit_allocation_syntax = 1;
353 s->fast_gain_syntax = 0;
354 s->first_cpl_leak = 0;
357 memset(
s->channel_uses_aht, 0,
sizeof(
s->channel_uses_aht));
380 if (!
s->downmix_coeffs[0]) {
382 sizeof(**
s->downmix_coeffs));
383 if (!
s->downmix_coeffs[0])
388 for (
i = 0;
i <
s->fbw_channels;
i++) {
392 if (
s->channel_mode > 1 &&
s->channel_mode & 1) {
393 downmix_coeffs[0][1] = downmix_coeffs[1][1] = cmix;
396 int nf =
s->channel_mode - 2;
397 downmix_coeffs[0][nf] = downmix_coeffs[1][nf] = smix *
LEVEL_MINUS_3DB;
400 int nf =
s->channel_mode - 4;
401 downmix_coeffs[0][nf] = downmix_coeffs[1][nf+1] = smix;
406 for (
i = 0;
i <
s->fbw_channels;
i++) {
407 norm0 += downmix_coeffs[0][
i];
408 norm1 += downmix_coeffs[1][
i];
410 norm0 = 1.0f / norm0;
411 norm1 = 1.0f / norm1;
412 for (
i = 0;
i <
s->fbw_channels;
i++) {
413 downmix_coeffs[0][
i] *= norm0;
414 downmix_coeffs[1][
i] *= norm1;
418 for (
i = 0;
i <
s->fbw_channels;
i++)
419 downmix_coeffs[0][
i] = (downmix_coeffs[0][
i] +
422 for (
i = 0;
i <
s->fbw_channels;
i++) {
423 s->downmix_coeffs[0][
i] =
FIXR12(downmix_coeffs[0][
i]);
424 s->downmix_coeffs[1][
i] =
FIXR12(downmix_coeffs[1][
i]);
438 int i, j, grp, group_size;
443 group_size = exp_strategy + (exp_strategy ==
EXP_D45);
444 for (grp = 0,
i = 0; grp < ngrps; grp++) {
457 for (
i = 0, j = 0;
i < ngrps * 3;
i++) {
458 prevexp += dexp[
i] - 2;
463 switch (group_size) {
464 case 4: dexps[j++] = prevexp;
465 dexps[j++] = prevexp;
466 case 2: dexps[j++] = prevexp;
467 case 1: dexps[j++] = prevexp;
483 for (band = 0; band <
s->num_cpl_bands; band++) {
484 int band_start = bin;
485 int band_end = bin +
s->cpl_band_sizes[band];
486 for (ch = 1; ch <=
s->fbw_channels; ch++) {
487 if (
s->channel_in_cpl[ch]) {
488 int cpl_coord =
s->cpl_coords[ch][band] << 5;
489 for (bin = band_start; bin < band_end; bin++) {
490 s->fixed_coeffs[ch][bin] =
491 MULH(
s->fixed_coeffs[
CPL_CH][bin] * (1 << 4), cpl_coord);
493 if (ch == 2 &&
s->phase_flags[band]) {
494 for (bin = band_start; bin < band_end; bin++)
495 s->fixed_coeffs[2][bin] = -
s->fixed_coeffs[2][bin];
521 int start_freq =
s->start_freq[ch_index];
522 int end_freq =
s->end_freq[ch_index];
524 int8_t *exps =
s->dexps[ch_index];
525 int32_t *coeffs =
s->fixed_coeffs[ch_index];
526 int dither = (ch_index ==
CPL_CH) ||
s->dither_flag[ch_index];
530 for (freq = start_freq; freq < end_freq; freq++) {
531 int bap = baps[freq];
537 mantissa = (((
av_lfg_get(&
s->dith_state)>>8)*181)>>8) - 5931008;
591 coeffs[freq] = mantissa >> exps[freq];
603 for (ch = 1; ch <=
s->fbw_channels; ch++) {
604 if (!
s->dither_flag[ch] &&
s->channel_in_cpl[ch]) {
607 s->fixed_coeffs[ch][
i] = 0;
616 if (!
s->channel_uses_aht[ch]) {
624 for (bin =
s->start_freq[ch]; bin < s->end_freq[ch]; bin++) {
625 s->fixed_coeffs[ch][bin] =
s->pre_mantissa[ch][bin][
blk] >>
s->dexps[ch][bin];
641 for (ch = 1; ch <=
s->channels; ch++) {
646 if (
s->channel_in_cpl[ch]) {
654 end =
s->end_freq[ch];
657 s->fixed_coeffs[ch][end] = 0;
674 end =
FFMIN(
s->end_freq[1],
s->end_freq[2]);
676 for (bnd = 0; bnd <
s->num_rematrixing_bands; bnd++) {
677 if (
s->rematrixing_flags[bnd]) {
680 int tmp0 =
s->fixed_coeffs[1][
i];
681 s->fixed_coeffs[1][
i] +=
s->fixed_coeffs[2][
i];
682 s->fixed_coeffs[2][
i] = tmp0 -
s->fixed_coeffs[2][
i];
697 for (ch = 1; ch <=
channels; ch++) {
698 if (
s->block_switch[ch]) {
701 for (
i = 0;
i < 128;
i++)
702 x[
i] =
s->transform_coeffs[ch][2 *
i];
703 s->imdct_256.imdct_half(&
s->imdct_256,
s->tmp_output, x);
705 s->fdsp->vector_fmul_window_scaled(
s->outptr[ch - 1],
s->delay[ch - 1 +
offset],
706 s->tmp_output,
s->window, 128, 8);
708 s->fdsp->vector_fmul_window(
s->outptr[ch - 1],
s->delay[ch - 1 +
offset],
709 s->tmp_output,
s->window, 128);
711 for (
i = 0;
i < 128;
i++)
712 x[
i] =
s->transform_coeffs[ch][2 *
i + 1];
713 s->imdct_256.imdct_half(&
s->imdct_256,
s->delay[ch - 1 +
offset], x);
715 s->imdct_512.imdct_half(&
s->imdct_512,
s->tmp_output,
s->transform_coeffs[ch]);
717 s->fdsp->vector_fmul_window_scaled(
s->outptr[ch - 1],
s->delay[ch - 1 +
offset],
718 s->tmp_output,
s->window, 128, 8);
720 s->fdsp->vector_fmul_window(
s->outptr[ch - 1],
s->delay[ch - 1 +
offset],
721 s->tmp_output,
s->window, 128);
733 int channel_data_size =
sizeof(
s->delay[0]);
734 switch (
s->channel_mode) {
738 memcpy(
s->delay[1],
s->delay[0], channel_data_size);
741 memset(
s->delay[3], 0, channel_data_size);
743 memset(
s->delay[2], 0, channel_data_size);
746 memset(
s->delay[4], 0, channel_data_size);
748 memset(
s->delay[3], 0, channel_data_size);
750 memcpy(
s->delay[2],
s->delay[1], channel_data_size);
751 memset(
s->delay[1], 0, channel_data_size);
774 int ecpl,
int start_subband,
int end_subband,
775 const uint8_t *default_band_struct,
776 int *num_bands,
uint8_t *band_sizes,
777 uint8_t *band_struct,
int band_struct_size)
779 int subbnd, bnd, n_subbands, n_bands=0;
782 n_subbands = end_subband - start_subband;
785 memcpy(band_struct, default_band_struct, band_struct_size);
787 av_assert0(band_struct_size >= start_subband + n_subbands);
789 band_struct += start_subband + 1;
793 for (subbnd = 0; subbnd < n_subbands - 1; subbnd++) {
801 if (num_bands || band_sizes ) {
802 n_bands = n_subbands;
803 bnd_sz[0] = ecpl ? 6 : 12;
804 for (bnd = 0, subbnd = 1; subbnd < n_subbands; subbnd++) {
805 int subbnd_size = (ecpl && subbnd < 4) ? 6 : 12;
806 if (band_struct[subbnd - 1]) {
808 bnd_sz[bnd] += subbnd_size;
810 bnd_sz[++bnd] = subbnd_size;
817 *num_bands = n_bands;
819 memcpy(band_sizes, bnd_sz, n_bands);
825 int fbw_channels =
s->fbw_channels;
826 int dst_start_freq, dst_end_freq, src_start_freq,
827 start_subband, end_subband, ch;
831 s->channel_uses_spx[1] = 1;
833 for (ch = 1; ch <= fbw_channels; ch++)
840 start_subband =
get_bits(bc, 3) + 2;
841 if (start_subband > 7)
842 start_subband += start_subband - 7;
848 end_subband += end_subband - 7;
849 dst_start_freq = dst_start_freq * 12 + 25;
850 src_start_freq = start_subband * 12 + 25;
851 dst_end_freq = end_subband * 12 + 25;
854 if (start_subband >= end_subband) {
856 "range (%d >= %d)\n", start_subband, end_subband);
859 if (dst_start_freq >= src_start_freq) {
861 "copy start bin (%d >= %d)\n", dst_start_freq, src_start_freq);
865 s->spx_dst_start_freq = dst_start_freq;
866 s->spx_src_start_freq = src_start_freq;
868 s->spx_dst_end_freq = dst_end_freq;
871 start_subband, end_subband,
875 s->spx_band_struct,
sizeof(
s->spx_band_struct));
882 int fbw_channels =
s->fbw_channels;
885 for (ch = 1; ch <= fbw_channels; ch++) {
886 if (
s->channel_uses_spx[ch]) {
887 if (
s->first_spx_coords[ch] ||
get_bits1(bc)) {
889 int bin, master_spx_coord;
891 s->first_spx_coords[ch] = 0;
893 master_spx_coord =
get_bits(bc, 2) * 3;
895 bin =
s->spx_src_start_freq;
896 for (bnd = 0; bnd <
s->num_spx_bands; bnd++) {
897 int bandsize =
s->spx_band_sizes[bnd];
898 int spx_coord_exp, spx_coord_mant;
902 int64_t accu = ((bin << 23) + (bandsize << 22))
903 * (int64_t)
s->spx_dst_end_freq;
904 nratio = (
int)(accu >> 32);
905 nratio -= spx_blend << 18;
910 }
else if (nratio > 0x7fffff) {
915 accu = (int64_t)nblend * 1859775393;
916 nblend = (
int)((accu + (1<<29)) >> 30);
923 nratio = ((float)((bin + (bandsize >> 1))) /
s->spx_dst_end_freq) - spx_blend;
924 nratio =
av_clipf(nratio, 0.0f, 1.0f);
925 nblend = sqrtf(3.0f * nratio);
927 sblend = sqrtf(1.0f - nratio);
934 if (spx_coord_exp == 15) spx_coord_mant <<= 1;
935 else spx_coord_mant += 4;
936 spx_coord_mant <<= (25 - spx_coord_exp - master_spx_coord);
940 accu = (int64_t)nblend * spx_coord_mant;
941 s->spx_noise_blend[ch][bnd] = (
int)((accu + (1<<22)) >> 23);
942 accu = (int64_t)sblend * spx_coord_mant;
943 s->spx_signal_blend[ch][bnd] = (
int)((accu + (1<<22)) >> 23);
945 spx_coord = spx_coord_mant * (1.0f / (1 << 23));
946 s->spx_noise_blend [ch][bnd] = nblend * spx_coord;
947 s->spx_signal_blend[ch][bnd] = sblend * spx_coord;
952 s->first_spx_coords[ch] = 1;
961 int fbw_channels =
s->fbw_channels;
962 int channel_mode =
s->channel_mode;
968 if (
s->cpl_in_use[
blk]) {
970 int cpl_start_subband, cpl_end_subband;
986 s->channel_in_cpl[1] = 1;
987 s->channel_in_cpl[2] = 1;
989 for (ch = 1; ch <= fbw_channels; ch++)
998 cpl_start_subband =
get_bits(bc, 4);
999 cpl_end_subband =
s->spx_in_use ? (
s->spx_src_start_freq - 37) / 12 :
1001 if (cpl_start_subband >= cpl_end_subband) {
1003 cpl_start_subband, cpl_end_subband);
1006 s->start_freq[
CPL_CH] = cpl_start_subband * 12 + 37;
1007 s->end_freq[
CPL_CH] = cpl_end_subband * 12 + 37;
1012 &
s->num_cpl_bands,
s->cpl_band_sizes,
1013 s->cpl_band_struct,
sizeof(
s->cpl_band_struct));
1016 for (ch = 1; ch <= fbw_channels; ch++) {
1017 s->channel_in_cpl[ch] = 0;
1018 s->first_cpl_coords[ch] = 1;
1020 s->first_cpl_leak =
s->eac3;
1021 s->phase_flags_in_use = 0;
1030 int fbw_channels =
s->fbw_channels;
1032 int cpl_coords_exist = 0;
1034 for (ch = 1; ch <= fbw_channels; ch++) {
1035 if (
s->channel_in_cpl[ch]) {
1036 if ((
s->eac3 &&
s->first_cpl_coords[ch]) ||
get_bits1(bc)) {
1037 int master_cpl_coord, cpl_coord_exp, cpl_coord_mant;
1038 s->first_cpl_coords[ch] = 0;
1039 cpl_coords_exist = 1;
1040 master_cpl_coord = 3 *
get_bits(bc, 2);
1041 for (bnd = 0; bnd <
s->num_cpl_bands; bnd++) {
1044 if (cpl_coord_exp == 15)
1045 s->cpl_coords[ch][bnd] = cpl_coord_mant << 22;
1047 s->cpl_coords[ch][bnd] = (cpl_coord_mant + 16) << 21;
1048 s->cpl_coords[ch][bnd] >>= (cpl_coord_exp + master_cpl_coord);
1052 "be present in block 0\n");
1057 s->first_cpl_coords[ch] = 1;
1062 for (bnd = 0; bnd <
s->num_cpl_bands; bnd++) {
1063 s->phase_flags[bnd] =
s->phase_flags_in_use ?
get_bits1(bc) : 0;
1075 int fbw_channels =
s->fbw_channels;
1076 int channel_mode =
s->channel_mode;
1077 int i, bnd, seg, ch, ret;
1078 int different_transforms;
1085 different_transforms = 0;
1086 if (
s->block_switch_syntax) {
1087 for (ch = 1; ch <= fbw_channels; ch++) {
1089 if (ch > 1 &&
s->block_switch[ch] !=
s->block_switch[1])
1090 different_transforms = 1;
1095 if (
s->dither_flag_syntax) {
1096 for (ch = 1; ch <= fbw_channels; ch++) {
1102 i = !
s->channel_mode;
1109 if (range_bits <= 127 || s->drc_scale <= 1.0)
1112 s->dynamic_range[
i] = range;
1113 }
else if (
blk == 0) {
1121 if (
s->spx_in_use) {
1126 if (!
s->eac3 || !
s->spx_in_use) {
1128 for (ch = 1; ch <= fbw_channels; ch++) {
1129 s->channel_uses_spx[ch] = 0;
1130 s->first_spx_coords[ch] = 1;
1142 }
else if (!
s->eac3) {
1145 "be present in block 0\n");
1148 s->cpl_in_use[
blk] =
s->cpl_in_use[
blk-1];
1151 cpl_in_use =
s->cpl_in_use[
blk];
1162 s->num_rematrixing_bands = 4;
1163 if (cpl_in_use &&
s->start_freq[
CPL_CH] <= 61) {
1164 s->num_rematrixing_bands -= 1 + (
s->start_freq[
CPL_CH] == 37);
1165 }
else if (
s->spx_in_use &&
s->spx_src_start_freq <= 61) {
1166 s->num_rematrixing_bands--;
1168 for (bnd = 0; bnd <
s->num_rematrixing_bands; bnd++)
1172 "new rematrixing strategy not present in block 0\n");
1173 s->num_rematrixing_bands = 0;
1178 for (ch = !cpl_in_use; ch <=
s->channels; ch++) {
1180 s->exp_strategy[
blk][ch] =
get_bits(gbc, 2 - (ch ==
s->lfe_ch));
1182 bit_alloc_stages[ch] = 3;
1186 for (ch = 1; ch <= fbw_channels; ch++) {
1187 s->start_freq[ch] = 0;
1190 int prev =
s->end_freq[ch];
1191 if (
s->channel_in_cpl[ch])
1192 s->end_freq[ch] =
s->start_freq[
CPL_CH];
1193 else if (
s->channel_uses_spx[ch])
1194 s->end_freq[ch] =
s->spx_src_start_freq;
1196 int bandwidth_code =
get_bits(gbc, 6);
1197 if (bandwidth_code > 60) {
1201 s->end_freq[ch] = bandwidth_code * 3 + 73;
1203 group_size = 3 << (
s->exp_strategy[
blk][ch] - 1);
1204 s->num_exp_groups[ch] = (
s->end_freq[ch] + group_size-4) / group_size;
1205 if (
blk > 0 &&
s->end_freq[ch] != prev)
1215 for (ch = !cpl_in_use; ch <=
s->channels; ch++) {
1217 s->dexps[ch][0] =
get_bits(gbc, 4) << !ch;
1219 s->num_exp_groups[ch],
s->dexps[ch][0],
1220 &
s->dexps[ch][
s->start_freq[ch]+!!ch])) {
1223 if (ch !=
CPL_CH && ch !=
s->lfe_ch)
1229 if (
s->bit_allocation_syntax) {
1236 for (ch = !cpl_in_use; ch <=
s->channels; ch++)
1237 bit_alloc_stages[ch] =
FFMAX(bit_alloc_stages[ch], 2);
1240 "be present in block 0\n");
1246 if (!
s->eac3 || !
blk) {
1247 if (
s->snr_offset_strategy &&
get_bits1(gbc)) {
1250 csnr = (
get_bits(gbc, 6) - 15) << 4;
1251 for (
i = ch = !cpl_in_use; ch <=
s->channels; ch++) {
1253 if (ch ==
i ||
s->snr_offset_strategy == 2)
1254 snr = (csnr +
get_bits(gbc, 4)) << 2;
1256 if (
blk &&
s->snr_offset[ch] != snr) {
1257 bit_alloc_stages[ch] =
FFMAX(bit_alloc_stages[ch], 1);
1259 s->snr_offset[ch] = snr;
1263 int prev =
s->fast_gain[ch];
1266 if (
blk && prev !=
s->fast_gain[ch])
1267 bit_alloc_stages[ch] =
FFMAX(bit_alloc_stages[ch], 2);
1270 }
else if (!
s->eac3 && !
blk) {
1278 for (ch = !cpl_in_use; ch <=
s->channels; ch++) {
1279 int prev =
s->fast_gain[ch];
1282 if (
blk && prev !=
s->fast_gain[ch])
1283 bit_alloc_stages[ch] =
FFMAX(bit_alloc_stages[ch], 2);
1285 }
else if (
s->eac3 && !
blk) {
1286 for (ch = !cpl_in_use; ch <=
s->channels; ch++)
1302 if (
blk && (fl !=
s->bit_alloc_params.cpl_fast_leak ||
1303 sl !=
s->bit_alloc_params.cpl_slow_leak)) {
1306 s->bit_alloc_params.cpl_fast_leak = fl;
1307 s->bit_alloc_params.cpl_slow_leak = sl;
1308 }
else if (!
s->eac3 && !
blk) {
1310 "be present in block 0\n");
1313 s->first_cpl_leak = 0;
1319 for (ch = !cpl_in_use; ch <= fbw_channels; ch++) {
1325 bit_alloc_stages[ch] =
FFMAX(bit_alloc_stages[ch], 2);
1328 for (ch = !cpl_in_use; ch <= fbw_channels; ch++) {
1330 s->dba_nsegs[ch] =
get_bits(gbc, 3) + 1;
1331 for (seg = 0; seg <
s->dba_nsegs[ch]; seg++) {
1332 s->dba_offsets[ch][seg] =
get_bits(gbc, 5);
1333 s->dba_lengths[ch][seg] =
get_bits(gbc, 4);
1334 s->dba_values[ch][seg] =
get_bits(gbc, 3);
1337 bit_alloc_stages[ch] =
FFMAX(bit_alloc_stages[ch], 2);
1340 }
else if (
blk == 0) {
1341 for (ch = 0; ch <=
s->channels; ch++) {
1347 for (ch = !cpl_in_use; ch <=
s->channels; ch++) {
1348 if (bit_alloc_stages[ch] > 2) {
1351 s->start_freq[ch],
s->end_freq[ch],
1352 s->psd[ch],
s->band_psd[ch]);
1354 if (bit_alloc_stages[ch] > 1) {
1358 s->start_freq[ch],
s->end_freq[ch],
1359 s->fast_gain[ch], (ch ==
s->lfe_ch),
1360 s->dba_mode[ch],
s->dba_nsegs[ch],
1361 s->dba_offsets[ch],
s->dba_lengths[ch],
1362 s->dba_values[ch],
s->mask[ch])) {
1367 if (bit_alloc_stages[ch] > 0) {
1371 s->ac3dsp.bit_alloc_calc_bap(
s->mask[ch],
s->psd[ch],
1372 s->start_freq[ch],
s->end_freq[ch],
1374 s->bit_alloc_params.floor,
1396 for (ch = 1; ch <=
s->channels; ch++) {
1397 int audio_channel = 0;
1400 audio_channel = 2-ch;
1401 if (
s->heavy_compression &&
s->compression_exists[audio_channel])
1402 gain =
s->heavy_dynamic_range[audio_channel];
1404 gain =
s->dynamic_range[audio_channel];
1407 scale_coefs(
s->transform_coeffs[ch],
s->fixed_coeffs[ch], gain, 256);
1409 if (
s->target_level != 0)
1410 gain = gain *
s->level_gain[audio_channel];
1411 gain *= 1.0 / 4194304.0f;
1412 s->fmt_conv.int32_to_float_fmul_scalar(
s->transform_coeffs[ch],
1413 s->fixed_coeffs[ch], gain, 256);
1425 downmix_output =
s->channels !=
s->out_channels &&
1426 !((
s->output_mode & AC3_OUTPUT_LFEON) &&
1427 s->fbw_channels ==
s->out_channels);
1428 if (different_transforms) {
1438 if (downmix_output) {
1441 s->out_channels,
s->fbw_channels, 256);
1444 s->out_channels,
s->fbw_channels, 256);
1448 if (downmix_output) {
1450 s->out_channels,
s->fbw_channels, 256);
1453 if (downmix_output && !
s->downmixed) {
1456 s->out_channels,
s->fbw_channels, 128);
1469 int *got_frame_ptr,
AVPacket *avpkt)
1473 int buf_size, full_buf_size = avpkt->
size;
1477 int skip = 0, got_independent_frame = 0;
1484 s->superframe_size = 0;
1486 buf_size = full_buf_size;
1487 for (
i = 1;
i < buf_size;
i += 2) {
1488 if (buf[
i] == 0x77 || buf[
i] == 0x0B) {
1489 if ((buf[
i] ^ buf[
i-1]) == (0x77 ^ 0x0B)) {
1492 }
else if ((buf[
i] ^ buf[
i+1]) == (0x77 ^ 0x0B)) {
1506 if (buf_size >= 2 &&
AV_RB16(buf) == 0x770B) {
1508 int cnt =
FFMIN(buf_size, AC3_FRAME_BUFFER_SIZE) >> 1;
1509 s->bdsp.bswap16_buf((uint16_t *)
s->input_buffer,
1510 (
const uint16_t *) buf, cnt);
1512 memcpy(
s->input_buffer, buf,
FFMIN(buf_size, AC3_FRAME_BUFFER_SIZE));
1517 if (
s->consistent_noise_generation)
1520 buf =
s->input_buffer;
1546 if (
s->substreamid) {
1548 "unsupported substream %d: skipping frame\n",
1565 if (
s->frame_size > buf_size) {
1571 s->frame_size - 2)) {
1583 return FFMIN(full_buf_size,
s->frame_size);
1587 if (!err || (
s->channels &&
s->out_channels !=
s->channels)) {
1588 s->out_channels =
s->channels;
1589 s->output_mode =
s->channel_mode;
1591 s->output_mode |= AC3_OUTPUT_LFEON;
1592 if (
s->channels > 1 &&
1594 s->out_channels = 1;
1596 }
else if (
s->channels > 2 &&
1598 s->out_channels = 2;
1602 s->loro_center_mix_level =
gain_levels[
s-> center_mix_level];
1603 s->loro_surround_mix_level =
gain_levels[
s->surround_mix_level];
1607 if (
s->channels !=
s->out_channels && !((
s->output_mode & AC3_OUTPUT_LFEON) &&
1608 s->fbw_channels ==
s->out_channels)) {
1614 }
else if (!
s->channels) {
1620 if (
s->output_mode & AC3_OUTPUT_LFEON)
1625 if (
s->bitstream_mode == 0x7 &&
s->channels > 1)
1632 output[ch] =
s->output[ch +
offset];
1633 s->outptr[ch] =
s->output[ch +
offset];
1635 for (ch = 0; ch <
s->channels; ch++) {
1636 if (ch < s->out_channels)
1637 s->outptr[channel_map[ch]] =
s->output_buffer[ch +
offset];
1645 for (ch = 0; ch <
s->out_channels; ch++)
1647 for (ch = 0; ch <
s->out_channels; ch++)
1648 output[ch] =
s->outptr[channel_map[ch]];
1649 for (ch = 0; ch <
s->out_channels; ch++) {
1650 if (!ch || channel_map[ch])
1656 for (ch = 0; ch <
s->out_channels; ch++)
1660 if (buf_size >
s->frame_size) {
1664 if (buf_size -
s->frame_size <= 16) {
1665 skip = buf_size -
s->frame_size;
1669 if ((ret =
init_get_bits8(&
s->gbc, buf +
s->frame_size, buf_size -
s->frame_size)) < 0)
1680 buf +=
s->frame_size;
1681 buf_size -=
s->frame_size;
1682 s->prev_output_mode =
s->output_mode;
1683 s->prev_bit_rate =
s->bit_rate;
1684 got_independent_frame = 1;
1685 goto dependent_frame;
1696 avctx->
bit_rate =
s->bit_rate +
s->prev_bit_rate;
1700 extended_channel_map[ch] = ch;
1705 uint64_t channel_layout;
1708 if (
s->prev_output_mode & AC3_OUTPUT_LFEON)
1711 channel_layout = ich_layout;
1712 for (ch = 0; ch < 16; ch++) {
1733 if (extend >= channel_map_size)
1736 extended_channel_map[
index] =
offset + channel_map[extend++];
1740 for (
i = 0;
i < 64;
i++) {
1746 if (extend >= channel_map_size)
1749 extended_channel_map[
index] =
offset + channel_map[extend++];
1762 for (ch = 0; ch < avctx->
channels; ch++) {
1763 int map = extended_channel_map[ch];
1766 s->output_buffer[
map],
1778 s->channel_mode == (
s->output_mode & ~AC3_OUTPUT_LFEON)) {
1784 s->channel_mode == (
s->output_mode & ~AC3_OUTPUT_LFEON)) {
1785 switch (
s->dolby_surround_ex_mode) {
1801 switch (
s->preferred_downmix) {
1819 if (
s->lfe_mix_level_exists)
1828 if (!
s->superframe_size)
1829 return FFMIN(full_buf_size,
s->frame_size + skip);
1831 return FFMIN(full_buf_size,
s->superframe_size + skip);
1848 #define OFFSET(x) offsetof(AC3DecodeContext, x)
1849 #define PAR (AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM)
@ AAC_AC3_PARSE_ERROR_CHANNEL_CFG
@ AAC_AC3_PARSE_ERROR_FRAME_TYPE
@ AAC_AC3_PARSE_ERROR_SYNC
@ AAC_AC3_PARSE_ERROR_CRC
@ AAC_AC3_PARSE_ERROR_BSID
@ AAC_AC3_PARSE_ERROR_SAMPLE_RATE
@ AAC_AC3_PARSE_ERROR_FRAME_SIZE
int ff_ac3_bit_alloc_calc_mask(AC3BitAllocParameters *s, int16_t *band_psd, int start, int end, int fast_gain, int is_lfe, int dba_mode, int dba_nsegs, uint8_t *dba_offsets, uint8_t *dba_lengths, uint8_t *dba_values, int16_t *mask)
Calculate the masking curve.
void ff_ac3_bit_alloc_calc_psd(int8_t *exp, int start, int end, int16_t *psd, int16_t *band_psd)
Calculate the log power-spectral density of the input signal.
#define LEVEL_PLUS_1POINT5DB
@ AC3_DHEADPHONMOD_NOTINDICATED
#define CPL_CH
coupling channel index
#define EAC3_MAX_CHANNELS
maximum number of channels in EAC3
#define AC3_MAX_CHANNELS
maximum number of channels, including coupling channel
@ AC3_DSUREXMOD_NOTINDICATED
#define LEVEL_MINUS_1POINT5DB
@ EAC3_FRAME_TYPE_DEPENDENT
@ EAC3_FRAME_TYPE_INDEPENDENT
@ AC3_DMIXMOD_NOTINDICATED
#define AC3_DYNAMIC_RANGE(x)
#define LEVEL_MINUS_4POINT5DB
#define AC3_DYNAMIC_RANGE1
#define AC3_HEAVY_RANGE(x)
int ff_ac3_parse_header(GetBitContext *gbc, AC3HeaderInfo *hdr)
Parse AC-3 frame header.
const uint8_t ff_ac3_ungroup_3_in_5_bits_tab[32][3]
Table used to ungroup 3 values stored in 5 bits.
const uint8_t ff_eac3_default_spx_band_struct[17]
Table E2.15 Default Spectral Extension Banding Structure.
const uint8_t ff_eac3_hebap_tab[64]
static void ac3_downmix_c_fixed16(int16_t **samples, int16_t **matrix, int out_ch, int in_ch, int len)
Downmix samples from original signal to stereo or mono (this is for 16-bit samples and fixed point de...
static const int end_freq_inv_tab[8]
static void scale_coefs(int32_t *dst, const int32_t *src, int dynrng, int len)
av_cold void ff_ac3dsp_init(AC3DSPContext *c, int bit_exact)
void ff_ac3dsp_downmix(AC3DSPContext *c, float **samples, float **matrix, int out_ch, int in_ch, int len)
const uint16_t avpriv_ac3_channel_layout_tab[8]
Map audio coding mode (acmod) to channel layout mask.
const uint8_t ff_ac3_rematrix_band_tab[5]
Table of bin locations for rematrixing bands reference: Section 7.5.2 Rematrixing : Frequency Band De...
const uint8_t ff_ac3_channels_tab[8]
Map audio coding mode (acmod) to number of full-bandwidth channels.
const uint8_t ff_ac3_dec_channel_map[8][2][6]
Table to remap channels from AC-3 order to SMPTE order.
const uint8_t ff_ac3_fast_decay_tab[4]
const uint16_t ff_ac3_fast_gain_tab[8]
const uint16_t ff_ac3_slow_gain_tab[4]
const uint8_t ff_eac3_default_cpl_band_struct[18]
Table E2.16 Default Coupling Banding Structure.
const uint8_t ff_ac3_slow_decay_tab[4]
const uint64_t ff_eac3_custom_channel_map_locations[16][2]
const int16_t ff_ac3_floor_tab[8]
const uint16_t ff_ac3_db_per_bit_tab[4]
const uint8_t ff_ac3_bap_tab[64]
#define av_assert0(cond)
assert() equivalent, that is always enabled.
#define AV_EF_CRCCHECK
Verify checksums embedded in the bitstream (could be of either encoded or decoded data,...
#define AV_EF_EXPLODE
abort decoding on minor error detection
#define AV_EF_CAREFUL
consider things that violate the spec, are fast to calculate and have not been seen in the wild as er...
audio channel layout utility functions
#define CONFIG_EAC3_DECODER
Public header for CRC hash function implementation.
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
static const uint8_t bap_tab[64]
static void ff_eac3_decode_transform_coeffs_aht_ch(AC3DecodeContext *s, int ch)
static int ff_eac3_parse_header(AC3DecodeContext *s)
static void ff_eac3_apply_spectral_extension(AC3DecodeContext *s)
static av_always_inline int fixed_sqrt(int x, int bits)
Calculate the square root.
#define FF_DECODE_ERROR_INVALID_BITSTREAM
#define AV_NUM_DATA_POINTERS
static int get_sbits(GetBitContext *s, int n)
static void skip_bits_long(GetBitContext *s, int n)
Skips the specified number of bits.
static unsigned int get_bits1(GetBitContext *s)
static void skip_bits(GetBitContext *s, int n)
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
int av_get_channel_layout_channel_index(uint64_t channel_layout, uint64_t channel)
Get the index of a channel in channel_layout.
#define AV_CH_LAYOUT_MONO
#define AV_CH_LAYOUT_STEREO
int av_get_channel_layout_nb_channels(uint64_t channel_layout)
Return the number of channels in the channel layout.
@ AV_MATRIX_ENCODING_NONE
@ AV_MATRIX_ENCODING_DOLBY
@ AV_MATRIX_ENCODING_DPLIIZ
@ AV_MATRIX_ENCODING_DOLBYEX
@ AV_MATRIX_ENCODING_DOLBYHEADPHONE
#define AV_CH_LOW_FREQUENCY
AVDownmixInfo * av_downmix_info_update_side_data(AVFrame *frame)
Get a frame's AV_FRAME_DATA_DOWNMIX_INFO side data for editing.
@ AV_DOWNMIX_TYPE_UNKNOWN
Not indicated.
@ AV_DOWNMIX_TYPE_LTRT
Lt/Rt 2-channel downmix, Dolby Surround compatible.
@ AV_DOWNMIX_TYPE_LORO
Lo/Ro 2-channel downmix (Stereo).
@ AV_DOWNMIX_TYPE_DPLII
Lt/Rt 2-channel downmix, Dolby Pro Logic II compatible.
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
@ AV_AUDIO_SERVICE_TYPE_KARAOKE
const AVCRC * av_crc_get_table(AVCRCId crc_id)
Get an initialized standard CRC table.
uint32_t av_crc(const AVCRC *ctx, uint32_t crc, const uint8_t *buffer, size_t length)
Calculate the CRC of a block.
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
#define AV_LOG_WARNING
Something somehow does not look correct.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
@ AV_SAMPLE_FMT_FLTP
float, planar
@ AV_SAMPLE_FMT_S16P
signed 16 bits, planar
const VDPAUPixFmtMap * map
av_cold void ff_kbd_window_init(float *window, float alpha, int n)
Generate a Kaiser-Bessel Derived Window.
av_cold void av_lfg_init(AVLFG *c, unsigned int seed)
int av_lfg_init_from_data(AVLFG *c, const uint8_t *data, unsigned int length)
Seed the state of the ALFG using binary data.
static unsigned int av_lfg_get(AVLFG *c)
Get the next random unsigned 32-bit number using an ALFG.
static const float gain_levels[9]
Adjustments in dB gain.
static const float gain_levels_lfe[32]
Adjustments in dB gain (LFE, +10 to -21 dB)
float ff_ac3_heavy_dynamic_range_tab[256]
static int coupling_coordinates(AC3DecodeContext *s, int blk)
static void remove_dithering(AC3DecodeContext *s)
Remove random dithering from coupling range coefficients with zero-bit mantissas for coupled channels...
static int decode_audio_block(AC3DecodeContext *s, int blk, int offset)
Decode a single audio block from the AC-3 bitstream.
static uint8_t ungroup_3_in_7_bits_tab[128][3]
table for ungrouping 3 values in 7 bits.
static int b5_mantissas[16]
static float dynamic_range_tab[256]
dynamic range table.
static av_cold void ac3_tables_init(void)
static int coupling_strategy(AC3DecodeContext *s, int blk, uint8_t *bit_alloc_stages)
static int set_downmix_coeffs(AC3DecodeContext *s)
Set stereo downmixing coefficients based on frame header info.
static int b1_mantissas[32][3]
tables for ungrouping mantissas
static void decode_transform_coeffs_ch(AC3DecodeContext *s, int blk, int ch, mant_groups *m)
static int b3_mantissas[8]
static int symmetric_dequant(int code, int levels)
Symmetrical Dequantization reference: Section 7.3.3 Expansion of Mantissas for Symmetrical Quantizati...
static av_cold int ac3_decode_end(AVCodecContext *avctx)
Uninitialize the AC-3 decoder.
static void ac3_upmix_delay(AC3DecodeContext *s)
Upmix delay samples from stereo to original channel layout.
static const uint8_t ac3_default_coeffs[8][5][2]
Table for default stereo downmixing coefficients reference: Section 7.8.2 Downmixing Into Two Channel...
static int b4_mantissas[128][2]
static void spx_coordinates(AC3DecodeContext *s)
static void decode_transform_coeffs(AC3DecodeContext *s, int blk)
Decode the transform coefficients.
static int decode_exponents(AC3DecodeContext *s, GetBitContext *gbc, int exp_strategy, int ngrps, uint8_t absexp, int8_t *dexps)
Decode the grouped exponents according to exponent strategy.
static const uint8_t quantization_tab[16]
Quantization table: levels for symmetric.
static int ac3_parse_header(AC3DecodeContext *s)
Parse the 'sync info' and 'bit stream info' from the AC-3 bitstream.
static int ac3_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Decode a single AC-3 frame.
static int b2_mantissas[128][3]
static void calc_transform_coeffs_cpl(AC3DecodeContext *s)
Generate transform coefficients for each coupled channel in the coupling range using the coupling coe...
static void decode_band_structure(GetBitContext *gbc, int blk, int eac3, int ecpl, int start_subband, int end_subband, const uint8_t *default_band_struct, int *num_bands, uint8_t *band_sizes, uint8_t *band_struct, int band_struct_size)
Decode band structure for coupling, spectral extension, or enhanced coupling.
static int parse_frame_header(AC3DecodeContext *s)
Common function to parse AC-3 or E-AC-3 frame header.
static void do_rematrixing(AC3DecodeContext *s)
Stereo rematrixing.
static int spx_strategy(AC3DecodeContext *s, int blk)
static void ac3_decode_transform_coeffs_ch(AC3DecodeContext *s, int ch_index, mant_groups *m)
Decode the transform coefficients for a particular channel reference: Section 7.3 Quantization and De...
static void do_imdct(AC3DecodeContext *s, int channels, int offset)
Inverse MDCT Transform.
static av_cold int ac3_decode_init(AVCodecContext *avctx)
AVCodec initialization.
av_cold void ff_bswapdsp_init(BswapDSPContext *c)
av_cold void ff_fmt_convert_init(FmtConvertContext *c, AVCodecContext *avctx)
int ff_side_data_update_matrix_encoding(AVFrame *frame, enum AVMatrixEncoding matrix_encoding)
Add or update AV_FRAME_DATA_MATRIXENCODING side data.
AVFixedDSPContext * avpriv_alloc_fixed_dsp(int bit_exact)
Allocate and initialize a fixed DSP context.
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
common internal API header
static int ff_thread_once(char *control, void(*routine)(void))
main external API structure.
enum AVSampleFormat sample_fmt
audio sample format
enum AVAudioServiceType audio_service_type
Type of service that the audio stream conveys.
int64_t bit_rate
the average bitrate
int sample_rate
samples per second
uint64_t request_channel_layout
Request decoder to use this channel layout if it can (0 for default)
int flags
AV_CODEC_FLAG_*.
int channels
number of audio channels
uint64_t channel_layout
Audio channel layout.
int err_recognition
Error recognition; may misdetect some more or less valid parts as errors.
This structure describes optional metadata relevant to a downmix procedure.
double lfe_mix_level
Absolute scale factor representing the level at which the LFE data is mixed into L/R channels during ...
double surround_mix_level_ltrt
Absolute scale factor representing the nominal level of the surround channels during an Lt/Rt compati...
double surround_mix_level
Absolute scale factor representing the nominal level of the surround channels during a regular downmi...
double center_mix_level
Absolute scale factor representing the nominal level of the center channel during a regular downmix.
enum AVDownmixType preferred_downmix_type
Type of downmix preferred by the mastering engineer.
double center_mix_level_ltrt
Absolute scale factor representing the nominal level of the center channel during an Lt/Rt compatible...
This structure describes decoded (raw) audio or video data.
int nb_samples
number of audio samples (per channel) described by this frame
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
int decode_error_flags
decode error flags of the frame, set to a combination of FF_DECODE_ERROR_xxx flags if the decoder pro...
uint8_t ** extended_data
pointers to the data planes/channels.
This structure stores compressed data.
Grouped mantissas for 3-level 5-level and 11-level quantization.
#define av_malloc_array(a, b)
#define avpriv_request_sample(...)
static const uint8_t dither[8][8]
static const uint8_t offset[127][2]