Go to the documentation of this file.
63 5, 6, 7, 8, 9, 10, 11, 12, 14, 16
87 3.162275, 2.818382, 2.511886, 2.238719, 1.995261, 1.778278, 1.584893,
88 1.412536, 1.258924, 1.122018, 1.000000, 0.891251, 0.794328, 0.707946,
89 0.630957, 0.562341, 0.501187, 0.446683, 0.398107, 0.354813, 0.316227,
90 0.281838, 0.251188, 0.223872, 0.199526, 0.177828, 0.158489, 0.141253,
91 0.125892, 0.112201, 0.100000, 0.089125
99 { { 2, 7 }, { 7, 2 }, },
101 { { 2, 7 }, { 7, 2 }, },
102 { { 2, 7 }, { 5, 5 }, { 7, 2 }, },
103 { { 2, 7 }, { 7, 2 }, { 6, 6 }, },
104 { { 2, 7 }, { 5, 5 }, { 7, 2 }, { 8, 8 }, },
105 { { 2, 7 }, { 7, 2 }, { 6, 7 }, { 7, 6 }, },
106 { { 2, 7 }, { 5, 5 }, { 7, 2 }, { 6, 7 }, { 7, 6 }, },
117 return ((
code - (levels >> 1)) * (1 << 24)) / levels;
129 for (
i = 0;
i < 128;
i++) {
137 for (
i = 0;
i < 32;
i++) {
143 for (
i = 0;
i < 128;
i++) {
155 for (
i = 0;
i < 7;
i++) {
159 for (
i = 0;
i < 15;
i++) {
167 for (
i = 0;
i < 256;
i++) {
168 int v = (
i >> 5) - ((
i >> 7) << 3) - 5;
174 for (
i = 0;
i < 256;
i++) {
175 int v = (
i >> 4) - ((
i >> 7) << 4) - 4;
222 s->xcfptr[
i] =
s->transform_coeffs[
i];
223 s->dlyptr[
i] =
s->delay[
i];
240 i = !
s->channel_mode;
242 s->dialog_normalization[(!
s->channel_mode)-
i] = -
get_bits(gbc, 5);
243 if (
s->dialog_normalization[(!
s->channel_mode)-
i] == 0) {
244 s->dialog_normalization[(!
s->channel_mode)-
i] = -31;
246 if (
s->target_level != 0) {
247 s->level_gain[(!
s->channel_mode)-
i] =
powf(2.0
f,
248 (
float)(
s->target_level -
249 s->dialog_normalization[(!
s->channel_mode)-
i])/6.0f);
251 if (
s->compression_exists[(!
s->channel_mode)-
i] =
get_bits1(gbc)) {
252 s->heavy_dynamic_range[(!
s->channel_mode)-
i] =
264 if (
s->bitstream_id != 6) {
272 s->center_mix_level_ltrt =
get_bits(gbc, 3);
273 s->surround_mix_level_ltrt = av_clip(
get_bits(gbc, 3), 3, 7);
275 s->surround_mix_level = av_clip(
get_bits(gbc, 3), 3, 7);
278 s->dolby_surround_ex_mode =
get_bits(gbc, 2);
279 s->dolby_headphone_mode =
get_bits(gbc, 2);
308 s->bit_alloc_params.sr_code = hdr.
sr_code;
313 s->bit_alloc_params.sr_shift = hdr.
sr_shift;
317 s->fbw_channels =
s->channels -
s->lfe_on;
318 s->lfe_ch =
s->fbw_channels + 1;
323 s->center_mix_level_ltrt = 4;
325 s->surround_mix_level_ltrt = 4;
326 s->lfe_mix_level_exists = 0;
335 s->start_freq[
s->lfe_ch] = 0;
336 s->end_freq[
s->lfe_ch] = 7;
337 s->num_exp_groups[
s->lfe_ch] = 2;
338 s->channel_in_cpl[
s->lfe_ch] = 0;
341 if (
s->bitstream_id <= 10) {
343 s->snr_offset_strategy = 2;
344 s->block_switch_syntax = 1;
345 s->dither_flag_syntax = 1;
346 s->bit_allocation_syntax = 1;
347 s->fast_gain_syntax = 0;
348 s->first_cpl_leak = 0;
351 memset(
s->channel_uses_aht, 0,
sizeof(
s->channel_uses_aht));
353 }
else if (CONFIG_EAC3_DECODER) {
374 if (!
s->downmix_coeffs[0]) {
376 sizeof(**
s->downmix_coeffs));
377 if (!
s->downmix_coeffs[0])
382 for (
i = 0;
i <
s->fbw_channels;
i++) {
386 if (
s->channel_mode > 1 &&
s->channel_mode & 1) {
387 downmix_coeffs[0][1] = downmix_coeffs[1][1] = cmix;
390 int nf =
s->channel_mode - 2;
391 downmix_coeffs[0][nf] = downmix_coeffs[1][nf] = smix *
LEVEL_MINUS_3DB;
394 int nf =
s->channel_mode - 4;
395 downmix_coeffs[0][nf] = downmix_coeffs[1][nf+1] = smix;
400 for (
i = 0;
i <
s->fbw_channels;
i++) {
401 norm0 += downmix_coeffs[0][
i];
402 norm1 += downmix_coeffs[1][
i];
404 norm0 = 1.0f / norm0;
405 norm1 = 1.0f / norm1;
406 for (
i = 0;
i <
s->fbw_channels;
i++) {
407 downmix_coeffs[0][
i] *= norm0;
408 downmix_coeffs[1][
i] *= norm1;
412 for (
i = 0;
i <
s->fbw_channels;
i++)
413 downmix_coeffs[0][
i] = (downmix_coeffs[0][
i] +
416 for (
i = 0;
i <
s->fbw_channels;
i++) {
417 s->downmix_coeffs[0][
i] =
FIXR12(downmix_coeffs[0][
i]);
418 s->downmix_coeffs[1][
i] =
FIXR12(downmix_coeffs[1][
i]);
432 int i, j, grp, group_size;
437 group_size = exp_strategy + (exp_strategy ==
EXP_D45);
438 for (grp = 0,
i = 0; grp < ngrps; grp++) {
451 for (
i = 0, j = 0;
i < ngrps * 3;
i++) {
452 prevexp += dexp[
i] - 2;
457 switch (group_size) {
458 case 4: dexps[j++] = prevexp;
459 dexps[j++] = prevexp;
460 case 2: dexps[j++] = prevexp;
461 case 1: dexps[j++] = prevexp;
477 for (band = 0; band <
s->num_cpl_bands; band++) {
478 int band_start = bin;
479 int band_end = bin +
s->cpl_band_sizes[band];
480 for (
ch = 1;
ch <=
s->fbw_channels;
ch++) {
481 if (
s->channel_in_cpl[
ch]) {
482 int cpl_coord =
s->cpl_coords[
ch][band] << 5;
483 for (bin = band_start; bin < band_end; bin++) {
484 s->fixed_coeffs[
ch][bin] =
485 MULH(
s->fixed_coeffs[
CPL_CH][bin] * (1 << 4), cpl_coord);
487 if (
ch == 2 &&
s->phase_flags[band]) {
488 for (bin = band_start; bin < band_end; bin++)
489 s->fixed_coeffs[2][bin] = -
s->fixed_coeffs[2][bin];
515 int start_freq =
s->start_freq[ch_index];
516 int end_freq =
s->end_freq[ch_index];
518 int8_t *exps =
s->dexps[ch_index];
519 int32_t *coeffs =
s->fixed_coeffs[ch_index];
520 int dither = (ch_index ==
CPL_CH) ||
s->dither_flag[ch_index];
524 for (freq = start_freq; freq < end_freq; freq++) {
525 int bap = baps[freq];
531 mantissa = (((
av_lfg_get(&
s->dith_state)>>8)*181)>>8) - 5931008;
585 coeffs[freq] = mantissa >> exps[freq];
597 for (
ch = 1;
ch <=
s->fbw_channels;
ch++) {
598 if (!
s->dither_flag[
ch] &&
s->channel_in_cpl[
ch]) {
601 s->fixed_coeffs[
ch][
i] = 0;
610 if (!
s->channel_uses_aht[
ch]) {
616 if (CONFIG_EAC3_DECODER && !
blk)
618 for (bin =
s->start_freq[
ch]; bin < s->end_freq[
ch]; bin++) {
619 s->fixed_coeffs[
ch][bin] =
s->pre_mantissa[
ch][bin][
blk] >>
s->dexps[
ch][bin];
635 for (
ch = 1;
ch <=
s->channels;
ch++) {
640 if (
s->channel_in_cpl[
ch]) {
651 s->fixed_coeffs[
ch][
end] = 0;
670 for (bnd = 0; bnd <
s->num_rematrixing_bands; bnd++) {
671 if (
s->rematrixing_flags[bnd]) {
674 int tmp0 =
s->fixed_coeffs[1][
i];
675 s->fixed_coeffs[1][
i] +=
s->fixed_coeffs[2][
i];
676 s->fixed_coeffs[2][
i] = tmp0 -
s->fixed_coeffs[2][
i];
692 if (
s->block_switch[
ch]) {
695 for (
i = 0;
i < 128;
i++)
696 x[
i] =
s->transform_coeffs[
ch][2 *
i];
697 s->imdct_256.imdct_half(&
s->imdct_256,
s->tmp_output, x);
699 s->fdsp->vector_fmul_window_scaled(
s->outptr[
ch - 1],
s->delay[
ch - 1 +
offset],
700 s->tmp_output,
s->window, 128, 8);
702 s->fdsp->vector_fmul_window(
s->outptr[
ch - 1],
s->delay[
ch - 1 +
offset],
703 s->tmp_output,
s->window, 128);
705 for (
i = 0;
i < 128;
i++)
706 x[
i] =
s->transform_coeffs[
ch][2 *
i + 1];
707 s->imdct_256.imdct_half(&
s->imdct_256,
s->delay[
ch - 1 +
offset], x);
709 s->imdct_512.imdct_half(&
s->imdct_512,
s->tmp_output,
s->transform_coeffs[
ch]);
711 s->fdsp->vector_fmul_window_scaled(
s->outptr[
ch - 1],
s->delay[
ch - 1 +
offset],
712 s->tmp_output,
s->window, 128, 8);
714 s->fdsp->vector_fmul_window(
s->outptr[
ch - 1],
s->delay[
ch - 1 +
offset],
715 s->tmp_output,
s->window, 128);
727 int channel_data_size =
sizeof(
s->delay[0]);
728 switch (
s->channel_mode) {
732 memcpy(
s->delay[1],
s->delay[0], channel_data_size);
735 memset(
s->delay[3], 0, channel_data_size);
737 memset(
s->delay[2], 0, channel_data_size);
740 memset(
s->delay[4], 0, channel_data_size);
742 memset(
s->delay[3], 0, channel_data_size);
744 memcpy(
s->delay[2],
s->delay[1], channel_data_size);
745 memset(
s->delay[1], 0, channel_data_size);
768 int ecpl,
int start_subband,
int end_subband,
769 const uint8_t *default_band_struct,
770 int *num_bands,
uint8_t *band_sizes,
771 uint8_t *band_struct,
int band_struct_size)
773 int subbnd, bnd, n_subbands, n_bands=0;
776 n_subbands = end_subband - start_subband;
779 memcpy(band_struct, default_band_struct, band_struct_size);
781 av_assert0(band_struct_size >= start_subband + n_subbands);
783 band_struct += start_subband + 1;
787 for (subbnd = 0; subbnd < n_subbands - 1; subbnd++) {
795 if (num_bands || band_sizes ) {
796 n_bands = n_subbands;
797 bnd_sz[0] = ecpl ? 6 : 12;
798 for (bnd = 0, subbnd = 1; subbnd < n_subbands; subbnd++) {
799 int subbnd_size = (ecpl && subbnd < 4) ? 6 : 12;
800 if (band_struct[subbnd - 1]) {
802 bnd_sz[bnd] += subbnd_size;
804 bnd_sz[++bnd] = subbnd_size;
811 *num_bands = n_bands;
813 memcpy(band_sizes, bnd_sz, n_bands);
819 int fbw_channels =
s->fbw_channels;
820 int dst_start_freq, dst_end_freq, src_start_freq,
821 start_subband, end_subband,
ch;
825 s->channel_uses_spx[1] = 1;
827 for (
ch = 1;
ch <= fbw_channels;
ch++)
834 start_subband =
get_bits(bc, 3) + 2;
835 if (start_subband > 7)
836 start_subband += start_subband - 7;
842 end_subband += end_subband - 7;
843 dst_start_freq = dst_start_freq * 12 + 25;
844 src_start_freq = start_subband * 12 + 25;
845 dst_end_freq = end_subband * 12 + 25;
848 if (start_subband >= end_subband) {
850 "range (%d >= %d)\n", start_subband, end_subband);
853 if (dst_start_freq >= src_start_freq) {
855 "copy start bin (%d >= %d)\n", dst_start_freq, src_start_freq);
859 s->spx_dst_start_freq = dst_start_freq;
860 s->spx_src_start_freq = src_start_freq;
862 s->spx_dst_end_freq = dst_end_freq;
865 start_subband, end_subband,
869 s->spx_band_struct,
sizeof(
s->spx_band_struct));
876 int fbw_channels =
s->fbw_channels;
879 for (
ch = 1;
ch <= fbw_channels;
ch++) {
880 if (
s->channel_uses_spx[
ch]) {
883 int bin, master_spx_coord;
885 s->first_spx_coords[
ch] = 0;
887 master_spx_coord =
get_bits(bc, 2) * 3;
889 bin =
s->spx_src_start_freq;
890 for (bnd = 0; bnd <
s->num_spx_bands; bnd++) {
891 int bandsize =
s->spx_band_sizes[bnd];
892 int spx_coord_exp, spx_coord_mant;
896 int64_t accu = ((bin << 23) + (bandsize << 22))
897 * (int64_t)
s->spx_dst_end_freq;
898 nratio = (
int)(accu >> 32);
899 nratio -= spx_blend << 18;
904 }
else if (nratio > 0x7fffff) {
909 accu = (int64_t)nblend * 1859775393;
910 nblend = (
int)((accu + (1<<29)) >> 30);
917 nratio = ((float)((bin + (bandsize >> 1))) /
s->spx_dst_end_freq) - spx_blend;
918 nratio = av_clipf(nratio, 0.0
f, 1.0
f);
919 nblend = sqrtf(3.0
f * nratio);
921 sblend = sqrtf(1.0
f - nratio);
928 if (spx_coord_exp == 15) spx_coord_mant <<= 1;
929 else spx_coord_mant += 4;
930 spx_coord_mant <<= (25 - spx_coord_exp - master_spx_coord);
934 accu = (int64_t)nblend * spx_coord_mant;
935 s->spx_noise_blend[
ch][bnd] = (
int)((accu + (1<<22)) >> 23);
936 accu = (int64_t)sblend * spx_coord_mant;
937 s->spx_signal_blend[
ch][bnd] = (
int)((accu + (1<<22)) >> 23);
939 spx_coord = spx_coord_mant * (1.0f / (1 << 23));
940 s->spx_noise_blend [
ch][bnd] = nblend * spx_coord;
941 s->spx_signal_blend[
ch][bnd] = sblend * spx_coord;
946 s->first_spx_coords[
ch] = 1;
955 int fbw_channels =
s->fbw_channels;
956 int channel_mode =
s->channel_mode;
962 if (
s->cpl_in_use[
blk]) {
964 int cpl_start_subband, cpl_end_subband;
980 s->channel_in_cpl[1] = 1;
981 s->channel_in_cpl[2] = 1;
983 for (
ch = 1;
ch <= fbw_channels;
ch++)
992 cpl_start_subband =
get_bits(bc, 4);
993 cpl_end_subband =
s->spx_in_use ? (
s->spx_src_start_freq - 37) / 12 :
995 if (cpl_start_subband >= cpl_end_subband) {
997 cpl_start_subband, cpl_end_subband);
1000 s->start_freq[
CPL_CH] = cpl_start_subband * 12 + 37;
1001 s->end_freq[
CPL_CH] = cpl_end_subband * 12 + 37;
1006 &
s->num_cpl_bands,
s->cpl_band_sizes,
1007 s->cpl_band_struct,
sizeof(
s->cpl_band_struct));
1010 for (
ch = 1;
ch <= fbw_channels;
ch++) {
1011 s->channel_in_cpl[
ch] = 0;
1012 s->first_cpl_coords[
ch] = 1;
1014 s->first_cpl_leak =
s->eac3;
1015 s->phase_flags_in_use = 0;
1024 int fbw_channels =
s->fbw_channels;
1026 int cpl_coords_exist = 0;
1028 for (
ch = 1;
ch <= fbw_channels;
ch++) {
1029 if (
s->channel_in_cpl[
ch]) {
1030 if ((
s->eac3 &&
s->first_cpl_coords[
ch]) ||
get_bits1(bc)) {
1031 int master_cpl_coord, cpl_coord_exp, cpl_coord_mant;
1032 s->first_cpl_coords[
ch] = 0;
1033 cpl_coords_exist = 1;
1034 master_cpl_coord = 3 *
get_bits(bc, 2);
1035 for (bnd = 0; bnd <
s->num_cpl_bands; bnd++) {
1038 if (cpl_coord_exp == 15)
1039 s->cpl_coords[
ch][bnd] = cpl_coord_mant << 22;
1041 s->cpl_coords[
ch][bnd] = (cpl_coord_mant + 16) << 21;
1042 s->cpl_coords[
ch][bnd] >>= (cpl_coord_exp + master_cpl_coord);
1046 "be present in block 0\n");
1051 s->first_cpl_coords[
ch] = 1;
1056 for (bnd = 0; bnd <
s->num_cpl_bands; bnd++) {
1057 s->phase_flags[bnd] =
s->phase_flags_in_use ?
get_bits1(bc) : 0;
1069 int fbw_channels =
s->fbw_channels;
1070 int channel_mode =
s->channel_mode;
1071 int i, bnd, seg,
ch,
ret;
1072 int different_transforms;
1079 different_transforms = 0;
1080 if (
s->block_switch_syntax) {
1081 for (
ch = 1;
ch <= fbw_channels;
ch++) {
1083 if (
ch > 1 &&
s->block_switch[
ch] !=
s->block_switch[1])
1084 different_transforms = 1;
1089 if (
s->dither_flag_syntax) {
1090 for (
ch = 1;
ch <= fbw_channels;
ch++) {
1096 i = !
s->channel_mode;
1103 if (range_bits <= 127 || s->drc_scale <= 1.0)
1106 s->dynamic_range[
i] = range;
1107 }
else if (
blk == 0) {
1115 if (
s->spx_in_use) {
1120 if (!
s->eac3 || !
s->spx_in_use) {
1122 for (
ch = 1;
ch <= fbw_channels;
ch++) {
1123 s->channel_uses_spx[
ch] = 0;
1124 s->first_spx_coords[
ch] = 1;
1136 }
else if (!
s->eac3) {
1139 "be present in block 0\n");
1142 s->cpl_in_use[
blk] =
s->cpl_in_use[
blk-1];
1145 cpl_in_use =
s->cpl_in_use[
blk];
1156 s->num_rematrixing_bands = 4;
1157 if (cpl_in_use &&
s->start_freq[
CPL_CH] <= 61) {
1158 s->num_rematrixing_bands -= 1 + (
s->start_freq[
CPL_CH] == 37);
1159 }
else if (
s->spx_in_use &&
s->spx_src_start_freq <= 61) {
1160 s->num_rematrixing_bands--;
1162 for (bnd = 0; bnd <
s->num_rematrixing_bands; bnd++)
1166 "new rematrixing strategy not present in block 0\n");
1167 s->num_rematrixing_bands = 0;
1172 for (
ch = !cpl_in_use;
ch <=
s->channels;
ch++) {
1176 bit_alloc_stages[
ch] = 3;
1180 for (
ch = 1;
ch <= fbw_channels;
ch++) {
1181 s->start_freq[
ch] = 0;
1184 int prev =
s->end_freq[
ch];
1185 if (
s->channel_in_cpl[
ch])
1187 else if (
s->channel_uses_spx[
ch])
1188 s->end_freq[
ch] =
s->spx_src_start_freq;
1190 int bandwidth_code =
get_bits(gbc, 6);
1191 if (bandwidth_code > 60) {
1195 s->end_freq[
ch] = bandwidth_code * 3 + 73;
1197 group_size = 3 << (
s->exp_strategy[
blk][
ch] - 1);
1198 s->num_exp_groups[
ch] = (
s->end_freq[
ch] + group_size-4) / group_size;
1199 if (
blk > 0 &&
s->end_freq[
ch] != prev)
1209 for (
ch = !cpl_in_use;
ch <=
s->channels;
ch++) {
1213 s->num_exp_groups[
ch],
s->dexps[
ch][0],
1214 &
s->dexps[
ch][
s->start_freq[
ch]+!!
ch])) {
1223 if (
s->bit_allocation_syntax) {
1230 for (
ch = !cpl_in_use;
ch <=
s->channels;
ch++)
1231 bit_alloc_stages[
ch] =
FFMAX(bit_alloc_stages[
ch], 2);
1234 "be present in block 0\n");
1240 if (!
s->eac3 || !
blk) {
1241 if (
s->snr_offset_strategy &&
get_bits1(gbc)) {
1244 csnr = (
get_bits(gbc, 6) - 15) << 4;
1245 for (
i =
ch = !cpl_in_use;
ch <=
s->channels;
ch++) {
1247 if (
ch ==
i ||
s->snr_offset_strategy == 2)
1248 snr = (csnr +
get_bits(gbc, 4)) << 2;
1250 if (
blk &&
s->snr_offset[
ch] != snr) {
1251 bit_alloc_stages[
ch] =
FFMAX(bit_alloc_stages[
ch], 1);
1253 s->snr_offset[
ch] = snr;
1257 int prev =
s->fast_gain[
ch];
1260 if (
blk && prev !=
s->fast_gain[
ch])
1261 bit_alloc_stages[
ch] =
FFMAX(bit_alloc_stages[
ch], 2);
1264 }
else if (!
s->eac3 && !
blk) {
1272 for (
ch = !cpl_in_use;
ch <=
s->channels;
ch++) {
1273 int prev =
s->fast_gain[
ch];
1276 if (
blk && prev !=
s->fast_gain[
ch])
1277 bit_alloc_stages[
ch] =
FFMAX(bit_alloc_stages[
ch], 2);
1279 }
else if (
s->eac3 && !
blk) {
1280 for (
ch = !cpl_in_use;
ch <=
s->channels;
ch++)
1296 if (
blk && (fl !=
s->bit_alloc_params.cpl_fast_leak ||
1297 sl !=
s->bit_alloc_params.cpl_slow_leak)) {
1300 s->bit_alloc_params.cpl_fast_leak = fl;
1301 s->bit_alloc_params.cpl_slow_leak = sl;
1302 }
else if (!
s->eac3 && !
blk) {
1304 "be present in block 0\n");
1307 s->first_cpl_leak = 0;
1313 for (
ch = !cpl_in_use;
ch <= fbw_channels;
ch++) {
1319 bit_alloc_stages[
ch] =
FFMAX(bit_alloc_stages[
ch], 2);
1322 for (
ch = !cpl_in_use;
ch <= fbw_channels;
ch++) {
1325 for (seg = 0; seg <
s->dba_nsegs[
ch]; seg++) {
1331 bit_alloc_stages[
ch] =
FFMAX(bit_alloc_stages[
ch], 2);
1334 }
else if (
blk == 0) {
1335 for (
ch = 0;
ch <=
s->channels;
ch++) {
1341 for (
ch = !cpl_in_use;
ch <=
s->channels;
ch++) {
1342 if (bit_alloc_stages[
ch] > 2) {
1345 s->start_freq[
ch],
s->end_freq[
ch],
1346 s->psd[
ch],
s->band_psd[
ch]);
1348 if (bit_alloc_stages[
ch] > 1) {
1352 s->start_freq[
ch],
s->end_freq[
ch],
1353 s->fast_gain[
ch], (
ch ==
s->lfe_ch),
1354 s->dba_mode[
ch],
s->dba_nsegs[
ch],
1355 s->dba_offsets[
ch],
s->dba_lengths[
ch],
1356 s->dba_values[
ch],
s->mask[
ch])) {
1361 if (bit_alloc_stages[
ch] > 0) {
1365 s->ac3dsp.bit_alloc_calc_bap(
s->mask[
ch],
s->psd[
ch],
1366 s->start_freq[
ch],
s->end_freq[
ch],
1368 s->bit_alloc_params.floor,
1390 for (
ch = 1;
ch <=
s->channels;
ch++) {
1391 int audio_channel = 0;
1394 audio_channel = 2-
ch;
1395 if (
s->heavy_compression &&
s->compression_exists[audio_channel])
1396 gain =
s->heavy_dynamic_range[audio_channel];
1398 gain =
s->dynamic_range[audio_channel];
1403 if (
s->target_level != 0)
1404 gain = gain *
s->level_gain[audio_channel];
1405 gain *= 1.0 / 4194304.0f;
1406 s->fmt_conv.int32_to_float_fmul_scalar(
s->transform_coeffs[
ch],
1407 s->fixed_coeffs[
ch], gain, 256);
1412 if (CONFIG_EAC3_DECODER &&
s->spx_in_use) {
1419 downmix_output =
s->channels !=
s->out_channels &&
1420 !((
s->output_mode & AC3_OUTPUT_LFEON) &&
1421 s->fbw_channels ==
s->out_channels);
1422 if (different_transforms) {
1432 if (downmix_output) {
1435 s->out_channels,
s->fbw_channels, 256);
1438 s->out_channels,
s->fbw_channels, 256);
1442 if (downmix_output) {
1444 s->out_channels,
s->fbw_channels, 256);
1447 if (downmix_output && !
s->downmixed) {
1450 s->out_channels,
s->fbw_channels, 128);
1463 int *got_frame_ptr,
AVPacket *avpkt)
1467 int buf_size, full_buf_size = avpkt->
size;
1471 int skip = 0, got_independent_frame = 0;
1478 s->superframe_size = 0;
1480 buf_size = full_buf_size;
1481 for (
i = 1;
i < buf_size;
i += 2) {
1482 if (
buf[
i] == 0x77 ||
buf[
i] == 0x0B) {
1483 if ((
buf[
i] ^
buf[
i-1]) == (0x77 ^ 0x0B)) {
1486 }
else if ((
buf[
i] ^
buf[
i+1]) == (0x77 ^ 0x0B)) {
1500 if (buf_size >= 2 &&
AV_RB16(
buf) == 0x770B) {
1502 int cnt =
FFMIN(buf_size, AC3_FRAME_BUFFER_SIZE) >> 1;
1503 s->bdsp.bswap16_buf((uint16_t *)
s->input_buffer,
1504 (
const uint16_t *)
buf, cnt);
1506 memcpy(
s->input_buffer,
buf,
FFMIN(buf_size, AC3_FRAME_BUFFER_SIZE));
1511 if (
s->consistent_noise_generation)
1514 buf =
s->input_buffer;
1540 if (
s->substreamid) {
1542 "unsupported substream %d: skipping frame\n",
1559 if (
s->frame_size > buf_size) {
1565 s->frame_size - 2)) {
1577 return FFMIN(full_buf_size,
s->frame_size);
1581 if (!err || (
s->channels &&
s->out_channels !=
s->channels)) {
1582 s->out_channels =
s->channels;
1583 s->output_mode =
s->channel_mode;
1585 s->output_mode |= AC3_OUTPUT_LFEON;
1586 if (
s->channels > 1 &&
1588 s->out_channels = 1;
1590 }
else if (
s->channels > 2 &&
1592 s->out_channels = 2;
1596 s->loro_center_mix_level =
gain_levels[
s-> center_mix_level];
1597 s->loro_surround_mix_level =
gain_levels[
s->surround_mix_level];
1601 if (
s->channels !=
s->out_channels && !((
s->output_mode & AC3_OUTPUT_LFEON) &&
1602 s->fbw_channels ==
s->out_channels)) {
1608 }
else if (!
s->channels) {
1614 if (
s->output_mode & AC3_OUTPUT_LFEON)
1619 if (
s->bitstream_mode == 0x7 &&
s->channels > 1)
1629 for (
ch = 0;
ch <
s->channels;
ch++) {
1630 if (ch < s->out_channels)
1631 s->outptr[channel_map[
ch]] =
s->output_buffer[
ch +
offset];
1639 for (
ch = 0;
ch <
s->out_channels;
ch++)
1641 for (
ch = 0;
ch <
s->out_channels;
ch++)
1644 if (!
ch || channel_map[
ch])
1650 for (
ch = 0;
ch <
s->out_channels;
ch++)
1654 if (buf_size >
s->frame_size) {
1658 if (buf_size -
s->frame_size <= 16) {
1659 skip = buf_size -
s->frame_size;
1674 buf +=
s->frame_size;
1675 buf_size -=
s->frame_size;
1676 s->prev_output_mode =
s->output_mode;
1677 s->prev_bit_rate =
s->bit_rate;
1678 got_independent_frame = 1;
1679 goto dependent_frame;
1690 avctx->
bit_rate =
s->bit_rate +
s->prev_bit_rate;
1694 extended_channel_map[
ch] =
ch;
1699 uint64_t channel_layout;
1702 if (
s->prev_output_mode & AC3_OUTPUT_LFEON)
1705 channel_layout = ich_layout;
1706 for (
ch = 0;
ch < 16;
ch++) {
1727 if (extend >= channel_map_size)
1730 extended_channel_map[
index] =
offset + channel_map[extend++];
1734 for (
i = 0;
i < 64;
i++) {
1740 if (extend >= channel_map_size)
1743 extended_channel_map[
index] =
offset + channel_map[extend++];
1757 int map = extended_channel_map[
ch];
1760 s->output_buffer[
map],
1772 s->channel_mode == (
s->output_mode & ~AC3_OUTPUT_LFEON)) {
1778 s->channel_mode == (
s->output_mode & ~AC3_OUTPUT_LFEON)) {
1779 switch (
s->dolby_surround_ex_mode) {
1795 switch (
s->preferred_downmix) {
1813 if (
s->lfe_mix_level_exists)
1822 if (!
s->superframe_size)
1823 return FFMIN(full_buf_size,
s->frame_size + skip);
1825 return FFMIN(full_buf_size,
s->superframe_size + skip);
1842 #define OFFSET(x) offsetof(AC3DecodeContext, x)
1843 #define PAR (AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM)
static int b3_mantissas[8]
static int b2_mantissas[128][3]
@ AV_SAMPLE_FMT_FLTP
float, planar
static void skip_bits_long(GetBitContext *s, int n)
Skips the specified number of bits.
#define AV_LOG_WARNING
Something somehow does not look correct.
static int set_downmix_coeffs(AC3DecodeContext *s)
Set stereo downmixing coefficients based on frame header info.
const uint8_t ff_ac3_fast_decay_tab[4]
static int ff_eac3_parse_header(AC3DecodeContext *s)
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.
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later. That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another. Frame references ownership and permissions
static int b4_mantissas[128][2]
enum AVAudioServiceType audio_service_type
Type of service that the audio stream conveys.
uint64_t channel_layout
Audio channel layout.
@ EAC3_FRAME_TYPE_INDEPENDENT
int sample_rate
samples per second
av_cold void av_lfg_init(AVLFG *c, unsigned int seed)
#define AV_CH_LAYOUT_MONO
int err_recognition
Error recognition; may misdetect some more or less valid parts as errors.
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(const uint8_t *) pi - 0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(const int16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(const int16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(const int32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(const int32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(const int64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64, *(const int64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64, *(const int64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(const float *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(const double *) pi *(INT64_C(1)<< 63))) #define FMT_PAIR_FUNC(out, in) static conv_func_type *const fmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB *AV_SAMPLE_FMT_NB]={ FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64), };static void cpy1(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, len);} static void cpy2(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 2 *len);} static void cpy4(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 4 *len);} static void cpy8(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 8 *len);} AudioConvert *swri_audio_convert_alloc(enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, const int *ch_map, int flags) { AudioConvert *ctx;conv_func_type *f=fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt)+AV_SAMPLE_FMT_NB *av_get_packed_sample_fmt(in_fmt)];if(!f) return NULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) return NULL;if(channels==1){ in_fmt=av_get_planar_sample_fmt(in_fmt);out_fmt=av_get_planar_sample_fmt(out_fmt);} ctx->channels=channels;ctx->conv_f=f;ctx->ch_map=ch_map;if(in_fmt==AV_SAMPLE_FMT_U8||in_fmt==AV_SAMPLE_FMT_U8P) memset(ctx->silence, 0x80, sizeof(ctx->silence));if(out_fmt==in_fmt &&!ch_map) { switch(av_get_bytes_per_sample(in_fmt)){ case 1:ctx->simd_f=cpy1;break;case 2:ctx->simd_f=cpy2;break;case 4:ctx->simd_f=cpy4;break;case 8:ctx->simd_f=cpy8;break;} } if(HAVE_X86ASM &&1) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);if(ARCH_AARCH64) swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels);return ctx;} void swri_audio_convert_free(AudioConvert **ctx) { av_freep(ctx);} int swri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, int len) { int ch;int off=0;const int os=(out->planar ? 1 :out->ch_count) *out->bps;unsigned misaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask) { int planes=in->planar ? in->ch_count :1;unsigned m=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) in->ch[ch];misaligned|=m &ctx->in_simd_align_mask;} if(ctx->out_simd_align_mask) { int planes=out->planar ? out->ch_count :1;unsigned m=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) out->ch[ch];misaligned|=m &ctx->out_simd_align_mask;} if(ctx->simd_f &&!ctx->ch_map &&!misaligned){ off=len &~15;av_assert1(off >=0);av_assert1(off<=len);av_assert2(ctx->channels==SWR_CH_MAX||!in->ch[ctx->channels]);if(off >0){ if(out->planar==in->planar){ int planes=out->planar ? out->ch_count :1;for(ch=0;ch< planes;ch++){ ctx->simd_f(out-> ch ch
filter_frame For filters that do not use the this method is called when a frame is pushed to the filter s input It can be called at any time except in a reentrant way If the input frame is enough to produce output
#define AC3_DYNAMIC_RANGE(x)
static av_cold int end(AVCodecContext *avctx)
This structure describes decoded (raw) audio or video data.
static const float gain_levels[9]
Adjustments in dB gain.
#define LEVEL_MINUS_4POINT5DB
const uint8_t ff_ac3_channels_tab[8]
Map audio coding mode (acmod) to number of full-bandwidth channels.
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.
uint64_t request_channel_layout
Request decoder to use this channel layout if it can (0 for default)
static void ff_eac3_decode_transform_coeffs_aht_ch(AC3DecodeContext *s, int ch)
double surround_mix_level_ltrt
Absolute scale factor representing the nominal level of the surround channels during an Lt/Rt compati...
static int ac3_parse_header(AC3DecodeContext *s)
Parse the 'sync info' and 'bit stream info' from the AC-3 bitstream.
@ EAC3_FRAME_TYPE_DEPENDENT
static void skip_bits(GetBitContext *s, int n)
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
@ AV_DOWNMIX_TYPE_UNKNOWN
Not indicated.
static const uint8_t quantization_tab[16]
Quantization table: levels for symmetric.
Grouped mantissas for 3-level 5-level and 11-level quantization.
int flags
AV_CODEC_FLAG_*.
#define AC3_MAX_CHANNELS
maximum number of channels, including coupling channel
This structure describes optional metadata relevant to a downmix procedure.
static void scale_coefs(int32_t *dst, const int32_t *src, int dynrng, int len)
const uint8_t ff_ac3_bap_tab[64]
const uint8_t ff_ac3_dec_channel_map[8][2][6]
Table to remap channels from AC-3 order to SMPTE order.
AVFixedDSPContext * avpriv_alloc_fixed_dsp(int bit_exact)
Allocate and initialize a fixed DSP context.
#define AV_CH_LAYOUT_STEREO
static int coupling_coordinates(AC3DecodeContext *s, int blk)
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
@ AV_MATRIX_ENCODING_DOLBY
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
#define AV_CH_LOW_FREQUENCY
static int parse_frame_header(AC3DecodeContext *s)
Common function to parse AC-3 or E-AC-3 frame header.
static void calc_transform_coeffs_cpl(AC3DecodeContext *s)
Generate transform coefficients for each coupled channel in the coupling range using the coupling coe...
static unsigned int av_lfg_get(AVLFG *c)
Get the next random unsigned 32-bit number using an ALFG.
const int16_t ff_ac3_floor_tab[8]
#define av_assert0(cond)
assert() equivalent, that is always enabled.
@ AAC_AC3_PARSE_ERROR_SYNC
@ AAC_AC3_PARSE_ERROR_BSID
static int get_sbits(GetBitContext *s, int n)
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
static av_cold int ac3_decode_init(AVCodecContext *avctx)
AVCodec initialization.
static av_cold void ac3_tables_init(void)
static void remove_dithering(AC3DecodeContext *s)
Remove random dithering from coupling range coefficients with zero-bit mantissas for coupled channels...
float ff_ac3_heavy_dynamic_range_tab[256]
@ AV_MATRIX_ENCODING_DOLBYHEADPHONE
av_cold void ff_bswapdsp_init(BswapDSPContext *c)
#define FF_DECODE_ERROR_INVALID_BITSTREAM
double surround_mix_level
Absolute scale factor representing the nominal level of the surround channels during a regular downmi...
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
static const float gain_levels_lfe[32]
Adjustments in dB gain (LFE, +10 to -21 dB)
#define AC3_DYNAMIC_RANGE1
static int symmetric_dequant(int code, int levels)
Symmetrical Dequantization reference: Section 7.3.3 Expansion of Mantissas for Symmetrical Quantizati...
int64_t bit_rate
the average bitrate
#define CPL_CH
coupling channel index
static unsigned int get_bits1(GetBitContext *s)
static void ac3_upmix_delay(AC3DecodeContext *s)
Upmix delay samples from stereo to original channel layout.
static int spx_strategy(AC3DecodeContext *s, int blk)
@ AC3_DHEADPHONMOD_NOTINDICATED
#define AV_EF_EXPLODE
abort decoding on minor error detection
int av_get_channel_layout_nb_channels(uint64_t channel_layout)
Return the number of channels in the channel layout.
const uint64_t ff_eac3_custom_channel_map_locations[16][2]
#define AV_EF_CAREFUL
consider things that violate the spec, are fast to calculate and have not been seen in the wild as er...
@ AV_AUDIO_SERVICE_TYPE_KARAOKE
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
static int coupling_strategy(AC3DecodeContext *s, int blk, uint8_t *bit_alloc_stages)
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...
@ AV_MATRIX_ENCODING_NONE
const uint16_t ff_ac3_db_per_bit_tab[4]
enum AVSampleFormat sample_fmt
audio sample format
#define LEVEL_MINUS_1POINT5DB
@ AAC_AC3_PARSE_ERROR_SAMPLE_RATE
#define AV_NUM_DATA_POINTERS
av_cold void ff_fmt_convert_init(FmtConvertContext *c, AVCodecContext *avctx)
double center_mix_level_ltrt
Absolute scale factor representing the nominal level of the center channel during an Lt/Rt compatible...
#define EAC3_MAX_CHANNELS
maximum number of channels in EAC3
const uint8_t ff_ac3_ungroup_3_in_5_bits_tab[32][3]
Table used to ungroup 3 values stored in 5 bits.
const AVCRC * av_crc_get_table(AVCRCId crc_id)
Get an initialized standard CRC table.
@ AC3_DSUREXMOD_NOTINDICATED
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf offset
static const int end_freq_inv_tab[8]
static void spx_coordinates(AC3DecodeContext *s)
@ AV_SAMPLE_FMT_S16P
signed 16 bits, planar
double lfe_mix_level
Absolute scale factor representing the level at which the LFE data is mixed into L/R channels during ...
int channels
number of audio channels
int av_lfg_init_from_data(AVLFG *c, const uint8_t *data, unsigned int length)
Seed the state of the ALFG using binary data.
#define LEVEL_PLUS_1POINT5DB
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...
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 decode_transform_coeffs_ch(AC3DecodeContext *s, int blk, int ch, mant_groups *m)
double center_mix_level
Absolute scale factor representing the nominal level of the center channel during a regular downmix.
#define i(width, name, range_min, range_max)
and forward the test the status of outputs and forward it to the corresponding return FFERROR_NOT_READY If the filters stores internally one or a few frame for some it can consider them to be part of the FIFO and delay acknowledging a status change accordingly Example code
@ AAC_AC3_PARSE_ERROR_FRAME_SIZE
#define av_malloc_array(a, b)
const uint16_t avpriv_ac3_channel_layout_tab[8]
Map audio coding mode (acmod) to channel layout mask.
const uint8_t ff_ac3_slow_decay_tab[4]
int av_get_channel_layout_channel_index(uint64_t channel_layout, uint64_t channel)
Get the index of a channel in channel_layout.
static av_cold int ac3_decode_end(AVCodecContext *avctx)
Uninitialize the AC-3 decoder.
static void ff_eac3_apply_spectral_extension(AC3DecodeContext *s)
enum AVDownmixType preferred_downmix_type
Type of downmix preferred by the mastering engineer.
#define AC3_HEAVY_RANGE(x)
static uint8_t ungroup_3_in_7_bits_tab[128][3]
table for ungrouping 3 values in 7 bits.
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...
@ AV_DOWNMIX_TYPE_LORO
Lo/Ro 2-channel downmix (Stereo).
@ AAC_AC3_PARSE_ERROR_FRAME_TYPE
void ff_ac3dsp_downmix(AC3DSPContext *c, float **samples, float **matrix, int out_ch, int in_ch, int len)
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
#define AV_EF_CRCCHECK
Verify checksums embedded in the bitstream (could be of either encoded or decoded data,...
int ff_ac3_parse_header(GetBitContext *gbc, AC3HeaderInfo *hdr)
Parse AC-3 frame header.
main external API structure.
@ AAC_AC3_PARSE_ERROR_CHANNEL_CFG
@ AV_MATRIX_ENCODING_DOLBYEX
uint32_t av_crc(const AVCRC *ctx, uint32_t crc, const uint8_t *buffer, size_t length)
Calculate the CRC of a block.
@ AV_DOWNMIX_TYPE_DPLII
Lt/Rt 2-channel downmix, Dolby Pro Logic II compatible.
av_cold void ff_kbd_window_init(float *window, float alpha, int n)
Generate a Kaiser-Bessel Derived Window.
static const uint8_t bap_tab[64]
static int b1_mantissas[32][3]
tables for ungrouping mantissas
static void decode_transform_coeffs(AC3DecodeContext *s, int blk)
Decode the transform coefficients.
AVDownmixInfo * av_downmix_info_update_side_data(AVFrame *frame)
Get a frame's AV_FRAME_DATA_DOWNMIX_INFO side data for editing.
av_cold void ff_ac3dsp_init(AC3DSPContext *c, int bit_exact)
int ff_side_data_update_matrix_encoding(AVFrame *frame, enum AVMatrixEncoding matrix_encoding)
Add or update AV_FRAME_DATA_MATRIXENCODING side data.
const uint8_t ff_eac3_hebap_tab[64]
@ AC3_DMIXMOD_NOTINDICATED
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
#define avpriv_request_sample(...)
const VDPAUPixFmtMap * map
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.
const uint16_t ff_ac3_slow_gain_tab[4]
This structure stores compressed data.
static int decode_audio_block(AC3DecodeContext *s, int blk, int offset)
Decode a single audio block from the AC-3 bitstream.
static void do_imdct(AC3DecodeContext *s, int channels, int offset)
Inverse MDCT Transform.
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
const uint16_t ff_ac3_fast_gain_tab[8]
@ AV_MATRIX_ENCODING_DPLIIZ
static float dynamic_range_tab[256]
dynamic range table.
@ AAC_AC3_PARSE_ERROR_CRC
static void do_rematrixing(AC3DecodeContext *s)
Stereo rematrixing.
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
static av_always_inline int fixed_sqrt(int x, int bits)
Calculate the square root.
static int b5_mantissas[16]
@ AV_DOWNMIX_TYPE_LTRT
Lt/Rt 2-channel downmix, Dolby Surround compatible.
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_WB64 unsigned int_TMPL AV_WB32 unsigned int_TMPL AV_WB24 unsigned int_TMPL AV_RB16
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.
static int ac3_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Decode a single AC-3 frame.
const uint8_t ff_eac3_default_cpl_band_struct[18]
Table E2.16 Default Coupling Banding Structure.
const uint8_t ff_eac3_default_spx_band_struct[17]
Table E2.15 Default Spectral Extension Banding Structure.
static const uint8_t dither[8][8]