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107 #define WMAPRO_MAX_CHANNELS 8
108 #define MAX_SUBFRAMES 32
110 #define MAX_FRAMESIZE 32768
111 #define XMA_MAX_STREAMS 8
112 #define XMA_MAX_CHANNELS_STREAM 2
113 #define XMA_MAX_CHANNELS (XMA_MAX_STREAMS * XMA_MAX_CHANNELS_STREAM)
115 #define WMAPRO_BLOCK_MIN_BITS 6
116 #define WMAPRO_BLOCK_MAX_BITS 13
117 #define WMAPRO_BLOCK_MIN_SIZE (1 << WMAPRO_BLOCK_MIN_BITS)
118 #define WMAPRO_BLOCK_MAX_SIZE (1 << WMAPRO_BLOCK_MAX_BITS)
119 #define WMAPRO_BLOCK_SIZES (WMAPRO_BLOCK_MAX_BITS - WMAPRO_BLOCK_MIN_BITS + 1)
123 #define SCALEVLCBITS 8
124 #define VEC4MAXDEPTH ((HUFF_VEC4_MAXBITS+VLCBITS-1)/VLCBITS)
125 #define VEC2MAXDEPTH ((HUFF_VEC2_MAXBITS+VLCBITS-1)/VLCBITS)
126 #define VEC1MAXDEPTH ((HUFF_VEC1_MAXBITS+VLCBITS-1)/VLCBITS)
127 #define SCALEMAXDEPTH ((HUFF_SCALE_MAXBITS+SCALEVLCBITS-1)/SCALEVLCBITS)
128 #define SCALERLMAXDEPTH ((HUFF_SCALE_RL_MAXBITS+VLCBITS-1)/VLCBITS)
259 #define PRINT(a, b) av_log(s->avctx, AV_LOG_DEBUG, " %s = %d\n", a, b);
260 #define PRINT_HEX(a, b) av_log(s->avctx, AV_LOG_DEBUG, " %s = %"PRIx32"\n", a, b);
262 PRINT(
"ed sample bit depth",
s->bits_per_sample);
263 PRINT_HEX(
"ed decode flags",
s->decode_flags);
264 PRINT(
"samples per frame",
s->samples_per_frame);
265 PRINT(
"log2 frame size",
s->log2_frame_size);
266 PRINT(
"max num subframes",
s->max_num_subframes);
267 PRINT(
"len prefix",
s->len_prefix);
268 PRINT(
"num channels",
s->nb_channels);
320 unsigned int channel_mask;
322 int log2_max_num_subframes;
323 int num_possible_block_sizes;
346 s->decode_flags = 0x10d6;
347 s->bits_per_sample = 16;
354 s->decode_flags = 0x10d6;
355 s->bits_per_sample = 16;
357 s->nb_channels = edata_ptr[32 + ((edata_ptr[0]==3)?0:8) + 4*num_stream + 0];
359 s->decode_flags = 0x10d6;
360 s->bits_per_sample = 16;
362 s->nb_channels = edata_ptr[8 + 20*num_stream + 17];
364 s->decode_flags =
AV_RL16(edata_ptr+14);
365 channel_mask =
AV_RL32(edata_ptr+2);
366 s->bits_per_sample =
AV_RL16(edata_ptr);
369 if (
s->bits_per_sample > 32 ||
s->bits_per_sample < 1) {
380 if (
s->log2_frame_size > 25) {
392 s->len_prefix = (
s->decode_flags & 0x40);
401 s->samples_per_frame = 1 <<
bits;
403 s->samples_per_frame = 512;
407 log2_max_num_subframes = ((
s->decode_flags & 0x38) >> 3);
408 s->max_num_subframes = 1 << log2_max_num_subframes;
409 if (
s->max_num_subframes == 16 ||
s->max_num_subframes == 4)
410 s->max_subframe_len_bit = 1;
411 s->subframe_len_bits =
av_log2(log2_max_num_subframes) + 1;
413 num_possible_block_sizes = log2_max_num_subframes + 1;
414 s->min_samples_per_subframe =
s->samples_per_frame /
s->max_num_subframes;
415 s->dynamic_range_compression = (
s->decode_flags & 0x80);
419 s->max_num_subframes);
425 s->min_samples_per_subframe);
429 if (
s->avctx->sample_rate <= 0) {
434 if (
s->nb_channels <= 0) {
449 for (
i = 0;
i <
s->nb_channels;
i++)
450 s->channel[
i].prev_block_len =
s->samples_per_frame;
455 if (channel_mask & 8) {
458 if (channel_mask &
mask)
493 for (
i = 0;
i < num_possible_block_sizes;
i++) {
494 int subframe_len =
s->samples_per_frame >>
i;
499 s->sfb_offsets[
i][0] = 0;
501 for (x = 0; x <
MAX_BANDS-1 &&
s->sfb_offsets[
i][band - 1] < subframe_len; x++) {
504 if (
offset >
s->sfb_offsets[
i][band - 1])
507 if (
offset >= subframe_len)
510 s->sfb_offsets[
i][band - 1] = subframe_len;
511 s->num_sfb[
i] = band - 1;
512 if (
s->num_sfb[
i] <= 0) {
524 for (
i = 0;
i < num_possible_block_sizes;
i++) {
526 for (
b = 0;
b <
s->num_sfb[
i];
b++) {
529 +
s->sfb_offsets[
i][
b + 1] - 1) <<
i) >> 1;
530 for (x = 0; x < num_possible_block_sizes; x++) {
532 while (
s->sfb_offsets[x][v + 1] << x <
offset) {
536 s->sf_offsets[
i][x][
b] = v;
549 / (1ll << (
s->bits_per_sample - 1)));
559 for (
i = 0;
i < num_possible_block_sizes;
i++) {
560 int block_size =
s->samples_per_frame >>
i;
561 int cutoff = (440*block_size + 3LL * (
s->avctx->sample_rate >> 1) - 1)
562 /
s->avctx->sample_rate;
563 s->subwoofer_cutoffs[
i] =
av_clip(cutoff, 4, block_size);
567 for (
i = 0;
i < 33;
i++)
598 int frame_len_shift = 0;
602 if (
offset ==
s->samples_per_frame -
s->min_samples_per_subframe)
603 return s->min_samples_per_subframe;
609 if (
s->max_subframe_len_bit) {
611 frame_len_shift = 1 +
get_bits(&
s->gb,
s->subframe_len_bits-1);
613 frame_len_shift =
get_bits(&
s->gb,
s->subframe_len_bits);
615 subframe_len =
s->samples_per_frame >> frame_len_shift;
618 if (subframe_len < s->min_samples_per_subframe ||
619 subframe_len >
s->samples_per_frame) {
651 int channels_for_cur_subframe =
s->nb_channels;
652 int fixed_channel_layout = 0;
653 int min_channel_len = 0;
663 for (
c = 0;
c <
s->nb_channels;
c++)
664 s->channel[
c].num_subframes = 0;
667 fixed_channel_layout = 1;
674 for (
c = 0;
c <
s->nb_channels;
c++) {
675 if (num_samples[
c] == min_channel_len) {
676 if (fixed_channel_layout || channels_for_cur_subframe == 1 ||
677 (min_channel_len ==
s->samples_per_frame -
s->min_samples_per_subframe))
678 contains_subframe[
c] = 1;
682 contains_subframe[
c] = 0;
690 min_channel_len += subframe_len;
691 for (
c = 0;
c <
s->nb_channels;
c++) {
694 if (contains_subframe[
c]) {
697 "broken frame: num subframes > 31\n");
701 num_samples[
c] += subframe_len;
703 if (num_samples[
c] >
s->samples_per_frame) {
705 "channel len > samples_per_frame\n");
708 }
else if (num_samples[
c] <= min_channel_len) {
709 if (num_samples[
c] < min_channel_len) {
710 channels_for_cur_subframe = 0;
711 min_channel_len = num_samples[
c];
713 ++channels_for_cur_subframe;
716 }
while (min_channel_len < s->samples_per_frame);
718 for (
c = 0;
c <
s->nb_channels;
c++) {
721 for (
i = 0;
i <
s->channel[
c].num_subframes;
i++) {
722 ff_dlog(
s->avctx,
"frame[%"PRIu32
"] channel[%i] subframe[%i]"
723 " len %i\n",
s->frame_num,
c,
i,
724 s->channel[
c].subframe_len[
i]);
725 s->channel[
c].subframe_offset[
i] =
offset;
756 for (x = 0; x <
i; x++) {
758 for (y = 0; y <
i + 1; y++) {
761 int n = rotation_offset[
offset + x];
767 cosv =
sin64[32 - n];
769 sinv =
sin64[64 - n];
770 cosv = -
sin64[n - 32];
774 (v1 * sinv) - (v2 * cosv);
776 (v1 * cosv) + (v2 * sinv);
798 if (
s->nb_channels > 1) {
799 int remaining_channels =
s->channels_for_cur_subframe;
803 "Channel transform bit");
807 for (
s->num_chgroups = 0; remaining_channels &&
808 s->num_chgroups <
s->channels_for_cur_subframe;
s->num_chgroups++) {
815 if (remaining_channels > 2) {
816 for (
i = 0;
i <
s->channels_for_cur_subframe;
i++) {
817 int channel_idx =
s->channel_indexes_for_cur_subframe[
i];
818 if (!
s->channel[channel_idx].grouped
821 s->channel[channel_idx].grouped = 1;
822 *channel_data++ =
s->channel[channel_idx].coeffs;
827 for (
i = 0;
i <
s->channels_for_cur_subframe;
i++) {
828 int channel_idx =
s->channel_indexes_for_cur_subframe[
i];
829 if (!
s->channel[channel_idx].grouped)
830 *channel_data++ =
s->channel[channel_idx].coeffs;
831 s->channel[channel_idx].grouped = 1;
840 "Unknown channel transform type");
845 if (
s->nb_channels == 2) {
867 "Coupled channels > 6");
883 for (
i = 0;
i <
s->num_bands;
i++) {
907 static const uint32_t fval_tab[16] = {
908 0x00000000, 0x3f800000, 0x40000000, 0x40400000,
909 0x40800000, 0x40a00000, 0x40c00000, 0x40e00000,
910 0x41000000, 0x41100000, 0x41200000, 0x41300000,
911 0x41400000, 0x41500000, 0x41600000, 0x41700000,
922 ff_dlog(
s->avctx,
"decode coefficients for channel %i\n",
c);
937 while ((
s->transmit_num_vec_coeffs || !rl_mode) &&
946 for (
i = 0;
i < 4;
i += 2) {
971 for (
i = 0;
i < 4;
i++) {
977 ci->
coeffs[cur_coeff] = 0;
980 rl_mode |= (++num_zeros >
s->subframe_len >> 8);
987 if (cur_coeff < s->subframe_len) {
990 memset(&ci->
coeffs[cur_coeff], 0,
991 sizeof(*ci->
coeffs) * (
s->subframe_len - cur_coeff));
994 cur_coeff,
s->subframe_len,
995 s->subframe_len,
s->esc_len, 0);
1016 for (
i = 0;
i <
s->channels_for_cur_subframe;
i++) {
1017 int c =
s->channel_indexes_for_cur_subframe[
i];
1020 s->channel[
c].scale_factors =
s->channel[
c].saved_scale_factors[!
s->channel[
c].scale_factor_idx];
1021 sf_end =
s->channel[
c].scale_factors +
s->num_bands;
1028 if (
s->channel[
c].reuse_sf) {
1029 const int8_t* sf_offsets =
s->sf_offsets[
s->table_idx][
s->channel[
c].table_idx];
1031 for (
b = 0;
b <
s->num_bands;
b++)
1032 s->channel[
c].scale_factors[
b] =
1033 s->channel[
c].saved_scale_factors[
s->channel[
c].scale_factor_idx][*sf_offsets++];
1036 if (!
s->channel[
c].cur_subframe ||
get_bits1(&
s->gb)) {
1038 if (!
s->channel[
c].reuse_sf) {
1041 s->channel[
c].scale_factor_step =
get_bits(&
s->gb, 2) + 1;
1042 val = 45 /
s->channel[
c].scale_factor_step;
1043 for (sf =
s->channel[
c].scale_factors; sf < sf_end; sf++) {
1050 for (
i = 0;
i <
s->num_bands;
i++) {
1061 sign = (
code & 1) - 1;
1062 skip = (
code & 0x3f) >> 1;
1063 }
else if (idx == 1) {
1072 if (
i >=
s->num_bands) {
1074 "invalid scale factor coding\n");
1077 s->channel[
c].scale_factors[
i] += (
val ^ sign) - sign;
1081 s->channel[
c].scale_factor_idx = !
s->channel[
c].scale_factor_idx;
1082 s->channel[
c].table_idx =
s->table_idx;
1083 s->channel[
c].reuse_sf = 1;
1087 s->channel[
c].max_scale_factor =
s->channel[
c].scale_factors[0];
1088 for (sf =
s->channel[
c].scale_factors + 1; sf < sf_end; sf++) {
1089 s->channel[
c].max_scale_factor =
1090 FFMAX(
s->channel[
c].max_scale_factor, *sf);
1105 for (
i = 0;
i <
s->num_chgroups;
i++) {
1106 if (
s->chgroup[
i].transform) {
1108 const int num_channels =
s->chgroup[
i].num_channels;
1109 float** ch_data =
s->chgroup[
i].channel_data;
1110 float** ch_end = ch_data + num_channels;
1111 const int8_t*
tb =
s->chgroup[
i].transform_band;
1115 for (sfb =
s->cur_sfb_offsets;
1116 sfb < s->cur_sfb_offsets +
s->num_bands; sfb++) {
1120 for (y = sfb[0]; y <
FFMIN(sfb[1],
s->subframe_len); y++) {
1121 const float* mat =
s->chgroup[
i].decorrelation_matrix;
1122 const float* data_end =
data + num_channels;
1123 float* data_ptr =
data;
1126 for (ch = ch_data; ch < ch_end; ch++)
1127 *data_ptr++ = (*ch)[y];
1129 for (ch = ch_data; ch < ch_end; ch++) {
1132 while (data_ptr < data_end)
1133 sum += *data_ptr++ * *mat++;
1138 }
else if (
s->nb_channels == 2) {
1139 int len =
FFMIN(sfb[1],
s->subframe_len) - sfb[0];
1140 s->fdsp->vector_fmul_scalar(ch_data[0] + sfb[0],
1141 ch_data[0] + sfb[0],
1143 s->fdsp->vector_fmul_scalar(ch_data[1] + sfb[0],
1144 ch_data[1] + sfb[0],
1159 for (
i = 0;
i <
s->channels_for_cur_subframe;
i++) {
1160 int c =
s->channel_indexes_for_cur_subframe[
i];
1162 int winlen =
s->channel[
c].prev_block_len;
1163 float* start =
s->channel[
c].coeffs - (winlen >> 1);
1165 if (
s->subframe_len < winlen) {
1166 start += (winlen -
s->subframe_len) >> 1;
1167 winlen =
s->subframe_len;
1174 s->fdsp->vector_fmul_window(start, start, start + winlen,
1177 s->channel[
c].prev_block_len =
s->subframe_len;
1188 int offset =
s->samples_per_frame;
1189 int subframe_len =
s->samples_per_frame;
1191 int total_samples =
s->samples_per_frame *
s->nb_channels;
1192 int transmit_coeffs = 0;
1193 int cur_subwoofer_cutoff;
1201 for (
i = 0;
i <
s->nb_channels;
i++) {
1202 s->channel[
i].grouped = 0;
1203 if (
offset >
s->channel[
i].decoded_samples) {
1204 offset =
s->channel[
i].decoded_samples;
1206 s->channel[
i].subframe_len[
s->channel[
i].cur_subframe];
1211 "processing subframe with offset %i len %i\n",
offset, subframe_len);
1214 s->channels_for_cur_subframe = 0;
1215 for (
i = 0;
i <
s->nb_channels;
i++) {
1216 const int cur_subframe =
s->channel[
i].cur_subframe;
1218 total_samples -=
s->channel[
i].decoded_samples;
1221 if (
offset ==
s->channel[
i].decoded_samples &&
1222 subframe_len ==
s->channel[
i].subframe_len[cur_subframe]) {
1223 total_samples -=
s->channel[
i].subframe_len[cur_subframe];
1224 s->channel[
i].decoded_samples +=
1225 s->channel[
i].subframe_len[cur_subframe];
1226 s->channel_indexes_for_cur_subframe[
s->channels_for_cur_subframe] =
i;
1227 ++
s->channels_for_cur_subframe;
1234 s->parsed_all_subframes = 1;
1237 ff_dlog(
s->avctx,
"subframe is part of %i channels\n",
1238 s->channels_for_cur_subframe);
1241 s->table_idx =
av_log2(
s->samples_per_frame/subframe_len);
1242 s->num_bands =
s->num_sfb[
s->table_idx];
1243 s->cur_sfb_offsets =
s->sfb_offsets[
s->table_idx];
1244 cur_subwoofer_cutoff =
s->subwoofer_cutoffs[
s->table_idx];
1247 offset +=
s->samples_per_frame >> 1;
1249 for (
i = 0;
i <
s->channels_for_cur_subframe;
i++) {
1250 int c =
s->channel_indexes_for_cur_subframe[
i];
1252 s->channel[
c].coeffs = &
s->channel[
c].out[
offset];
1255 s->subframe_len = subframe_len;
1256 s->esc_len =
av_log2(
s->subframe_len - 1) + 1;
1261 if (!(num_fill_bits =
get_bits(&
s->gb, 2))) {
1266 if (num_fill_bits >= 0) {
1287 for (
i = 0;
i <
s->channels_for_cur_subframe;
i++) {
1288 int c =
s->channel_indexes_for_cur_subframe[
i];
1289 if ((
s->channel[
c].transmit_coefs =
get_bits1(&
s->gb)))
1290 transmit_coeffs = 1;
1294 if (transmit_coeffs) {
1296 int quant_step = 90 *
s->bits_per_sample >> 4;
1299 if ((
s->transmit_num_vec_coeffs =
get_bits1(&
s->gb))) {
1300 int num_bits =
av_log2((
s->subframe_len + 3)/4) + 1;
1301 for (
i = 0;
i <
s->channels_for_cur_subframe;
i++) {
1302 int c =
s->channel_indexes_for_cur_subframe[
i];
1303 int num_vec_coeffs =
get_bits(&
s->gb, num_bits) << 2;
1304 if (num_vec_coeffs >
s->subframe_len) {
1309 s->channel[
c].num_vec_coeffs = num_vec_coeffs;
1312 for (
i = 0;
i <
s->channels_for_cur_subframe;
i++) {
1313 int c =
s->channel_indexes_for_cur_subframe[
i];
1314 s->channel[
c].num_vec_coeffs =
s->subframe_len;
1321 const int sign = (
step == 31) - 1;
1327 quant_step += ((
quant +
step) ^ sign) - sign;
1329 if (quant_step < 0) {
1335 if (
s->channels_for_cur_subframe == 1) {
1336 s->channel[
s->channel_indexes_for_cur_subframe[0]].quant_step = quant_step;
1339 for (
i = 0;
i <
s->channels_for_cur_subframe;
i++) {
1340 int c =
s->channel_indexes_for_cur_subframe[
i];
1341 s->channel[
c].quant_step = quant_step;
1344 s->channel[
c].quant_step +=
get_bits(&
s->gb, modifier_len) + 1;
1346 ++
s->channel[
c].quant_step;
1356 ff_dlog(
s->avctx,
"BITSTREAM: subframe header length was %i\n",
1360 for (
i = 0;
i <
s->channels_for_cur_subframe;
i++) {
1361 int c =
s->channel_indexes_for_cur_subframe[
i];
1362 if (
s->channel[
c].transmit_coefs &&
1366 memset(
s->channel[
c].coeffs, 0,
1367 sizeof(*
s->channel[
c].coeffs) * subframe_len);
1370 ff_dlog(
s->avctx,
"BITSTREAM: subframe length was %i\n",
1373 if (transmit_coeffs) {
1377 for (
i = 0;
i <
s->channels_for_cur_subframe;
i++) {
1378 int c =
s->channel_indexes_for_cur_subframe[
i];
1379 const int* sf =
s->channel[
c].scale_factors;
1382 if (
c ==
s->lfe_channel)
1383 memset(&
s->tmp[cur_subwoofer_cutoff], 0,
sizeof(*
s->tmp) *
1384 (subframe_len - cur_subwoofer_cutoff));
1387 for (
b = 0;
b <
s->num_bands;
b++) {
1388 const int end =
FFMIN(
s->cur_sfb_offsets[
b+1],
s->subframe_len);
1389 const int exp =
s->channel[
c].quant_step -
1390 (
s->channel[
c].max_scale_factor - *sf++) *
1391 s->channel[
c].scale_factor_step;
1393 int start =
s->cur_sfb_offsets[
b];
1394 s->fdsp->vector_fmul_scalar(
s->tmp + start,
1395 s->channel[
c].coeffs + start,
1396 quant, end - start);
1408 for (
i = 0;
i <
s->channels_for_cur_subframe;
i++) {
1409 int c =
s->channel_indexes_for_cur_subframe[
i];
1410 if (
s->channel[
c].cur_subframe >=
s->channel[
c].num_subframes) {
1414 ++
s->channel[
c].cur_subframe;
1429 int more_frames = 0;
1437 ff_dlog(
s->avctx,
"decoding frame with length %x\n",
len);
1448 for (
i = 0;
i <
s->nb_channels *
s->nb_channels;
i++)
1454 if (
s->dynamic_range_compression) {
1456 ff_dlog(
s->avctx,
"drc_gain %i\n",
s->drc_gain);
1467 ff_dlog(
s->avctx,
"start skip: %i\n", skip);
1473 ff_dlog(
s->avctx,
"end skip: %i\n", skip);
1478 ff_dlog(
s->avctx,
"BITSTREAM: frame header length was %i\n",
1482 s->parsed_all_subframes = 0;
1483 for (
i = 0;
i <
s->nb_channels;
i++) {
1484 s->channel[
i].decoded_samples = 0;
1485 s->channel[
i].cur_subframe = 0;
1486 s->channel[
i].reuse_sf = 0;
1490 while (!
s->parsed_all_subframes) {
1498 for (
i = 0;
i <
s->nb_channels;
i++)
1499 memcpy(
frame->extended_data[
i],
s->channel[
i].out,
1500 s->samples_per_frame *
sizeof(*
s->channel[
i].out));
1502 for (
i = 0;
i <
s->nb_channels;
i++) {
1504 memcpy(&
s->channel[
i].out[0],
1505 &
s->channel[
i].out[
s->samples_per_frame],
1506 s->samples_per_frame *
sizeof(*
s->channel[
i].out) >> 1);
1509 if (
s->skip_frame) {
1517 if (
s->len_prefix) {
1521 "frame[%"PRIu32
"] would have to skip %i bits\n",
1571 s->num_saved_bits =
s->frame_offset;
1573 buflen = (
s->num_saved_bits +
len + 7) >> 3;
1585 s->num_saved_bits +=
len;
1612 int buf_size = avpkt->
size;
1613 int num_bits_prev_frame;
1614 int packet_sequence_number;
1629 for (
i = 0;
i <
s->nb_channels;
i++) {
1630 memset(
frame->extended_data[
i], 0,
1631 s->samples_per_frame *
sizeof(*
s->channel[
i].out));
1633 memcpy(
frame->extended_data[
i],
s->channel[
i].out,
1634 s->samples_per_frame *
sizeof(*
s->channel[
i].out) >> 1);
1645 else if (
s->packet_done ||
s->packet_loss) {
1663 s->buf_bit_size = buf_size << 3;
1668 packet_sequence_number =
get_bits(gb, 4);
1673 packet_sequence_number = 0;
1677 num_bits_prev_frame =
get_bits(gb,
s->log2_frame_size);
1685 num_bits_prev_frame);
1689 ((
s->packet_sequence_number + 1) & 0xF) != packet_sequence_number) {
1692 "Packet loss detected! seq %"PRIx8
" vs %x\n",
1693 s->packet_sequence_number, packet_sequence_number);
1695 s->packet_sequence_number = packet_sequence_number;
1697 if (num_bits_prev_frame > 0) {
1699 if (num_bits_prev_frame >= remaining_packet_bits) {
1700 num_bits_prev_frame = remaining_packet_bits;
1707 ff_dlog(avctx,
"accumulated %x bits of frame data\n",
1708 s->num_saved_bits -
s->frame_offset);
1711 if (!
s->packet_loss)
1713 }
else if (
s->num_saved_bits -
s->frame_offset) {
1714 ff_dlog(avctx,
"ignoring %x previously saved bits\n",
1715 s->num_saved_bits -
s->frame_offset);
1718 if (
s->packet_loss) {
1722 s->num_saved_bits = 0;
1728 if (avpkt->
size <
s->next_packet_start) {
1733 s->buf_bit_size = (avpkt->
size -
s->next_packet_start) << 3;
1740 if (!
s->packet_loss)
1742 }
else if (!
s->len_prefix
1762 if (
s->packet_done && !
s->packet_loss &&
1784 int *got_frame_ptr,
AVPacket *avpkt)
1791 frame->nb_samples =
s->samples_per_frame;
1801 int *got_frame_ptr,
AVPacket *avpkt)
1804 int got_stream_frame_ptr = 0;
1808 if (!
s->frames[
s->current_stream]->data[0]) {
1809 s->frames[
s->current_stream]->nb_samples = 512;
1816 &got_stream_frame_ptr, avpkt);
1818 if (got_stream_frame_ptr &&
s->offset[
s->current_stream] >= 64) {
1819 got_stream_frame_ptr = 0;
1824 if (got_stream_frame_ptr) {
1825 int start_ch =
s->start_channel[
s->current_stream];
1826 memcpy(&
s->samples[start_ch + 0][
s->offset[
s->current_stream] * 512],
1827 s->frames[
s->current_stream]->extended_data[0], 512 * 4);
1828 if (
s->xma[
s->current_stream].nb_channels > 1)
1829 memcpy(&
s->samples[start_ch + 1][
s->offset[
s->current_stream] * 512],
1830 s->frames[
s->current_stream]->extended_data[1], 512 * 4);
1831 s->offset[
s->current_stream]++;
1832 }
else if (
ret < 0) {
1833 memset(
s->offset, 0,
sizeof(
s->offset));
1834 s->current_stream = 0;
1841 if (
s->xma[
s->current_stream].packet_done ||
1842 s->xma[
s->current_stream].packet_loss) {
1845 if (
s->xma[
s->current_stream].skip_packets != 0) {
1848 min[0] =
s->xma[0].skip_packets;
1851 for (
i = 1;
i <
s->num_streams;
i++) {
1852 if (
s->xma[
i].skip_packets <
min[0]) {
1853 min[0] =
s->xma[
i].skip_packets;
1858 s->current_stream =
min[1];
1862 for (
i = 0;
i <
s->num_streams;
i++) {
1863 s->xma[
i].skip_packets =
FFMAX(0,
s->xma[
i].skip_packets - 1);
1867 for (
i = 0;
i <
s->num_streams;
i++) {
1878 for (
i = 0;
i <
s->num_streams;
i++) {
1879 int start_ch =
s->start_channel[
i];
1880 memcpy(
frame->extended_data[start_ch + 0],
s->samples[start_ch + 0],
frame->nb_samples * 4);
1881 if (
s->xma[
i].nb_channels > 1)
1882 memcpy(
frame->extended_data[start_ch + 1],
s->samples[start_ch + 1],
frame->nb_samples * 4);
1886 memmove(
s->samples[start_ch + 0],
s->samples[start_ch + 0] +
frame->nb_samples,
s->offset[
i] * 4 * 512);
1887 if (
s->xma[
i].nb_channels > 1)
1888 memmove(
s->samples[start_ch + 1],
s->samples[start_ch + 1] +
frame->nb_samples,
s->offset[
i] * 4 * 512);
1902 int i,
ret, start_channels = 0;
1909 s->num_streams = (avctx->
channels + 1) / 2;
1939 for (
i = 0;
i <
s->num_streams;
i++) {
1947 s->start_channel[
i] = start_channels;
1948 start_channels +=
s->xma[
i].nb_channels;
1950 if (start_channels != avctx->
channels)
1961 for (
i = 0;
i <
s->num_streams;
i++) {
1975 for (
i = 0;
i <
s->nb_channels;
i++)
1976 memset(
s->channel[
i].out, 0,
s->samples_per_frame *
1977 sizeof(*
s->channel[
i].out));
1979 s->skip_packets = 0;
2000 for (
i = 0;
i <
s->num_streams;
i++)
2003 memset(
s->offset, 0,
sizeof(
s->offset));
2004 s->current_stream = 0;
uint16_t num_vec_coeffs
number of vector coded coefficients
static const float *const default_decorrelation[]
default decorrelation matrix offsets
static av_cold int xma_decode_init(AVCodecContext *avctx)
int subframe_offset
subframe offset in the bit reservoir
@ AV_SAMPLE_FMT_FLTP
float, planar
static void skip_bits_long(GetBitContext *s, int n)
Skips the specified number of bits.
static av_always_inline double ff_exp10(double x)
Compute 10^x for floating point values.
static av_cold int init(AVCodecContext *avctx)
static int get_bits_left(GetBitContext *gb)
static int decode_subframe(WMAProDecodeCtx *s)
Decode a single subframe (block).
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 uint8_t * append(uint8_t *buf, const uint8_t *src, int size)
GetBitContext gb
bitstream reader context
uint16_t samples_per_frame
number of samples to output
SINETABLE_CONST float *const ff_sine_windows[]
uint64_t channel_layout
Audio channel layout.
int8_t scale_factor_step
scaling step for the current subframe
static const uint8_t scale_huffbits[HUFF_SCALE_SIZE]
static void wmapro_window(WMAProDecodeCtx *s)
Apply sine window and reconstruct the output buffer.
#define WMAPRO_BLOCK_MAX_BITS
log2 of max block size
uint16_t min_samples_per_subframe
int sample_rate
samples per second
static enum AVSampleFormat sample_fmts[]
uint16_t subframe_offset[MAX_SUBFRAMES]
subframe positions in the current frame
static int decode_tilehdr(WMAProDecodeCtx *s)
Decode how the data in the frame is split into subframes.
#define INIT_VLC_STATIC(vlc, bits, a, b, c, d, e, f, g, static_size)
static void init_put_bits(PutBitContext *s, uint8_t *buffer, int buffer_size)
Initialize the PutBitContext s.
static int get_bits_count(const GetBitContext *s)
static const uint16_t coef0_run[HUFF_COEF0_SIZE]
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
AVCodecContext * avctx
codec context for av_log
static VLC sf_rl_vlc
scale factor run length vlc
This structure describes decoded (raw) audio or video data.
static void put_bits(Jpeg2000EncoderContext *s, int val, int n)
put n times val bit
trying all byte sequences megabyte in length and selecting the best looking sequence will yield cases to try But a word about which is also called distortion Distortion can be quantified by almost any quality measurement one chooses the sum of squared differences is used but more complex methods that consider psychovisual effects can be used as well It makes no difference in this discussion First step
static av_cold int wmapro_decode_init(AVCodecContext *avctx)
Initialize the decoder.
static void flush(WMAProDecodeCtx *s)
static int decode_packet(AVCodecContext *avctx, WMAProDecodeCtx *s, void *data, int *got_frame_ptr, AVPacket *avpkt)
static av_cold int get_rate(AVCodecContext *avctx)
#define WMAPRO_BLOCK_MIN_SIZE
minimum block size
static int decode_scale_factors(WMAProDecodeCtx *s)
Extract scale factors from the bitstream.
static av_always_inline int get_vlc2(GetBitContext *s, VLC_TYPE(*table)[2], int bits, int max_depth)
Parse a vlc code.
static const uint8_t scale_rl_huffbits[HUFF_SCALE_RL_SIZE]
#define WMAPRO_BLOCK_MAX_SIZE
maximum block size
float samples[XMA_MAX_CHANNELS][512 *64]
static av_always_inline uint32_t av_float2int(float f)
Reinterpret a float as a 32-bit integer.
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
PutBitContext pb
context for filling the frame_data buffer
static av_cold int decode_init(WMAProDecodeCtx *s, AVCodecContext *avctx, int num_stream)
Initialize the decoder.
static av_cold int decode_end(WMAProDecodeCtx *s)
Uninitialize the decoder and free all resources.
int16_t sfb_offsets[WMAPRO_BLOCK_SIZES][MAX_BANDS]
scale factor band offsets (multiples of 4)
static void skip_bits(GetBitContext *s, int n)
static float sin64[33]
sine table for decorrelation
#define HUFF_SCALE_RL_SIZE
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
AVCodec ff_wmapro_decoder
wmapro decoder
void ff_copy_bits(PutBitContext *pb, const uint8_t *src, int length)
Copy the content of src to the bitstream.
static SDL_Window * window
static VLC vec2_vlc
2 coefficients per symbol
static av_cold int wmapro_decode_end(AVCodecContext *avctx)
uint8_t num_chgroups
number of channel groups
uint8_t drc_gain
gain for the DRC tool
static int put_bits_left(PutBitContext *s)
int flags
AV_CODEC_FLAG_*.
static double val(void *priv, double ch)
int8_t num_bands
number of scale factor bands
float tmp[WMAPRO_BLOCK_MAX_SIZE]
IMDCT output buffer.
static const uint8_t coef1_huffbits[555]
int8_t sf_offsets[WMAPRO_BLOCK_SIZES][WMAPRO_BLOCK_SIZES][MAX_BANDS]
scale factor resample matrix
WMAProChannelGrp chgroup[WMAPRO_MAX_CHANNELS]
channel group information
static const uint32_t coef1_huffcodes[555]
int max_scale_factor
maximum scale factor for the current subframe
int quant_step
quantization step for the current subframe
AVFrame * av_frame_alloc(void)
Allocate an AVFrame and set its fields to default values.
uint8_t table_idx
index in sf_offsets for the scale factor reference block
static int decode_subframe_length(WMAProDecodeCtx *s, int offset)
Decode the subframe length.
float out[WMAPRO_BLOCK_MAX_SIZE+WMAPRO_BLOCK_MAX_SIZE/2]
output buffer
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
int buf_bit_size
buffer size in bits
#define FF_ARRAY_ELEMS(a)
static const uint16_t symbol_to_vec4[HUFF_VEC4_SIZE]
uint8_t subframe_len_bits
number of bits used for the subframe length
static const uint16_t mask[17]
static void decode_decorrelation_matrix(WMAProDecodeCtx *s, WMAProChannelGrp *chgroup)
Calculate a decorrelation matrix from the bitstream parameters.
frame specific decoder context for a single channel
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
int * scale_factors
pointer to the scale factor values used for decoding
int8_t skip_frame
skip output step
int16_t subwoofer_cutoffs[WMAPRO_BLOCK_SIZES]
subwoofer cutoff values
uint32_t decode_flags
used compression features
static const uint16_t vec2_huffcodes[HUFF_VEC2_SIZE]
uint8_t packet_loss
set in case of bitstream error
static const uint8_t symbol_to_vec2[HUFF_VEC2_SIZE]
static const uint16_t vec4_huffcodes[HUFF_VEC4_SIZE]
#define av_assert0(cond)
assert() equivalent, that is always enabled.
static void inverse_channel_transform(WMAProDecodeCtx *s)
Reconstruct the individual channel data.
static int get_sbits(GetBitContext *s, int n)
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_RL16
WMAProDecodeCtx xma[XMA_MAX_STREAMS]
static int decode_coeffs(WMAProDecodeCtx *s, int c)
Extract the coefficients from the bitstream.
#define XMA_MAX_CHANNELS_STREAM
int16_t prev_block_len
length of the previous block
int8_t transmit_num_vec_coeffs
number of vector coded coefficients is part of the bitstream
int8_t channel_indexes_for_cur_subframe[WMAPRO_MAX_CHANNELS]
uint8_t grouped
channel is part of a group
int start_channel[XMA_MAX_STREAMS]
static const uint8_t vec4_huffbits[HUFF_VEC4_SIZE]
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
static void wmapro_flush(AVCodecContext *avctx)
Clear decoder buffers (for seeking).
const float * windows[WMAPRO_BLOCK_SIZES]
windows for the different block sizes
int8_t transform
transform on / off
static unsigned int get_bits1(GetBitContext *s)
int8_t nb_channels
number of channels in stream (XMA1/2)
static void xma_flush(AVCodecContext *avctx)
#define WMAPRO_MAX_CHANNELS
current decoder limitations
channel group for channel transformations
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
static const uint8_t scale_rl_level[HUFF_SCALE_RL_SIZE]
uint8_t eof_done
set when EOF reached and extra subframe is written (XMA1/2)
void(* imdct_half)(struct FFTContext *s, FFTSample *output, const FFTSample *input)
uint32_t frame_num
current frame number (not used for decoding)
static VLC sf_vlc
scale factor DPCM vlc
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
float * coeffs
pointer to the subframe decode buffer
uint8_t len_prefix
frame is prefixed with its length
static const uint16_t critical_freq[]
frequencies to divide the frequency spectrum into scale factor bands
#define WMAPRO_BLOCK_SIZES
possible block sizes
enum AVSampleFormat sample_fmt
audio sample format
uint8_t frame_data[MAX_FRAMESIZE+AV_INPUT_BUFFER_PADDING_SIZE]
compressed frame data
static const uint8_t coef0_huffbits[666]
int8_t scale_factor_idx
index for the transmitted scale factor values (used for resampling)
#define MAX_SUBFRAMES
max number of subframes per channel
static const uint8_t vec1_huffbits[HUFF_VEC1_SIZE]
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
FFTContext mdct_ctx[WMAPRO_BLOCK_SIZES]
MDCT context per block size.
int8_t transform_band[MAX_BANDS]
controls if the transform is enabled for a certain band
uint8_t max_num_subframes
int8_t reuse_sf
share scale factors between subframes
int channels
number of audio channels
#define DECLARE_ALIGNED(n, t, v)
int next_packet_start
start offset of the next wma packet in the demuxer packet
static const uint32_t scale_rl_huffcodes[HUFF_SCALE_RL_SIZE]
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
static int put_bits_count(PutBitContext *s)
uint8_t cur_subframe
current subframe number
static const uint8_t scale_rl_run[HUFF_SCALE_RL_SIZE]
uint16_t decoded_samples
number of already processed samples
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
static unsigned int show_bits(GetBitContext *s, int n)
Show 1-25 bits.
#define FF_CODEC_CAP_INIT_CLEANUP
The codec allows calling the close function for deallocation even if the init function returned a fai...
static const float coef1_level[HUFF_COEF1_SIZE]
static VLC vec1_vlc
1 coefficient per symbol
AVSampleFormat
Audio sample formats.
#define MAX_BANDS
max number of scale factor bands
static const uint16_t coef1_run[HUFF_COEF1_SIZE]
void av_frame_unref(AVFrame *frame)
Unreference all the buffers referenced by frame and reset the frame fields.
const char * name
Name of the codec implementation.
static VLC coef_vlc[2]
coefficient run length vlc codes
tables for wmapro decoding
static int xma_decode_packet(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
GetBitContext pgb
bitstream reader context for the packet
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs.
int offset[XMA_MAX_STREAMS]
static void save_bits(WMAProDecodeCtx *s, GetBitContext *gb, int len, int append)
Fill the bit reservoir with a (partial) frame.
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
uint8_t num_channels
number of channels in the group
static const uint32_t coef0_huffcodes[666]
static av_cold void dump_context(WMAProDecodeCtx *s)
helper function to print the most important members of the context
#define AV_INPUT_BUFFER_PADDING_SIZE
static int decode_frame(WMAProDecodeCtx *s, AVFrame *frame, int *got_frame_ptr)
Decode one WMA frame.
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_RL32
static const uint16_t scale_huffcodes[HUFF_SCALE_SIZE]
int8_t channels_for_cur_subframe
number of channels that contain the subframe
main external API structure.
av_cold int ff_wma_get_frame_len_bits(int sample_rate, int version, unsigned int decode_flags)
Get the samples per frame for this stream.
int8_t esc_len
length of escaped coefficients
uint8_t table_idx
index for the num_sfb, sfb_offsets, sf_offsets and subwoofer_cutoffs tables
int8_t num_sfb[WMAPRO_BLOCK_SIZES]
scale factor bands per block size
static const uint8_t vec2_huffbits[HUFF_VEC2_SIZE]
uint16_t subframe_len[MAX_SUBFRAMES]
subframe length in samples
void ff_init_ff_sine_windows(int index)
initialize the specified entry of ff_sine_windows
uint8_t bits_per_sample
integer audio sample size for the unscaled IMDCT output (used to scale to [-1.0, 1....
uint8_t max_subframe_len_bit
flag indicating that the subframe is of maximum size when the first subframe length bit is 1
#define AV_CODEC_CAP_DELAY
Encoder or decoder requires flushing with NULL input at the end in order to give the complete and cor...
uint8_t packet_offset
frame offset in the packet
uint8_t skip_packets
packets to skip to find next packet in a stream (XMA1/2)
float * channel_data[WMAPRO_MAX_CHANNELS]
transformation coefficients
int saved_scale_factors[2][MAX_BANDS]
resampled and (previously) transmitted scale factor values
int frame_offset
frame offset in the bit reservoir
AVFrame * frames[XMA_MAX_STREAMS]
static av_always_inline int get_bitsz(GetBitContext *s, int n)
Read 0-25 bits.
#define AV_CODEC_CAP_SUBFRAMES
Codec can output multiple frames per AVPacket Normally demuxers return one frame at a time,...
uint8_t packet_done
set when a packet is fully decoded
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
int frame_number
Frame counter, set by libavcodec.
#define avpriv_request_sample(...)
static av_cold int xma_decode_end(AVCodecContext *avctx)
static int wmapro_decode_packet(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Decode a single WMA packet.
static void flush_put_bits(PutBitContext *s)
Pad the end of the output stream with zeros.
int8_t lfe_channel
lfe channel index
int ff_wma_run_level_decode(AVCodecContext *avctx, GetBitContext *gb, VLC *vlc, const float *level_table, const uint16_t *run_table, int version, WMACoef *ptr, int offset, int num_coefs, int block_len, int frame_len_bits, int coef_nb_bits)
Decode run level compressed coefficients.
int16_t * cur_sfb_offsets
sfb offsets for the current block
static int decode_channel_transform(WMAProDecodeCtx *s)
Decode channel transformation parameters.
int16_t subframe_len
current subframe length
int8_t parsed_all_subframes
all subframes decoded?
This structure stores compressed data.
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
static const float coef0_level[HUFF_COEF0_SIZE]
#define MAX_FRAMESIZE
maximum compressed frame size
uint8_t packet_sequence_number
current packet number
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
uint8_t dynamic_range_compression
frame contains DRC data
static int remaining_bits(WMAProDecodeCtx *s, GetBitContext *gb)
Calculate remaining input buffer length.
unsigned int ff_wma_get_large_val(GetBitContext *gb)
Decode an uncompressed coefficient.
#define FF_DEBUG_BITSTREAM
int num_saved_bits
saved number of bits
VLC_TYPE(* table)[2]
code, bits
float decorrelation_matrix[WMAPRO_MAX_CHANNELS *WMAPRO_MAX_CHANNELS]
static VLC vec4_vlc
4 coefficients per symbol
static const uint16_t vec1_huffcodes[HUFF_VEC1_SIZE]
#define WMAPRO_BLOCK_MIN_BITS
log2 of min block size