Go to the documentation of this file.
112 #define WMAPRO_MAX_CHANNELS 8
113 #define MAX_SUBFRAMES 32
115 #define MAX_FRAMESIZE 32768
116 #define XMA_MAX_STREAMS 8
117 #define XMA_MAX_CHANNELS_STREAM 2
118 #define XMA_MAX_CHANNELS (XMA_MAX_STREAMS * XMA_MAX_CHANNELS_STREAM)
120 #define WMAPRO_BLOCK_MIN_BITS 6
121 #define WMAPRO_BLOCK_MAX_BITS 13
122 #define WMAPRO_BLOCK_MIN_SIZE (1 << WMAPRO_BLOCK_MIN_BITS)
123 #define WMAPRO_BLOCK_MAX_SIZE (1 << WMAPRO_BLOCK_MAX_BITS)
124 #define WMAPRO_BLOCK_SIZES (WMAPRO_BLOCK_MAX_BITS - WMAPRO_BLOCK_MIN_BITS + 1)
128 #define SCALEVLCBITS 8
129 #define VEC4MAXDEPTH ((HUFF_VEC4_MAXBITS+VLCBITS-1)/VLCBITS)
130 #define VEC2MAXDEPTH ((HUFF_VEC2_MAXBITS+VLCBITS-1)/VLCBITS)
131 #define VEC1MAXDEPTH ((HUFF_VEC1_MAXBITS+VLCBITS-1)/VLCBITS)
132 #define SCALEMAXDEPTH ((HUFF_SCALE_MAXBITS+SCALEVLCBITS-1)/SCALEVLCBITS)
133 #define SCALERLMAXDEPTH ((HUFF_SCALE_RL_MAXBITS+VLCBITS-1)/VLCBITS)
268 #define PRINT(a, b) av_log(s->avctx, AV_LOG_DEBUG, " %s = %d\n", a, b);
269 #define PRINT_HEX(a, b) av_log(s->avctx, AV_LOG_DEBUG, " %s = %"PRIx32"\n", a, b);
271 PRINT(
"ed sample bit depth",
s->bits_per_sample);
272 PRINT_HEX(
"ed decode flags",
s->decode_flags);
273 PRINT(
"samples per frame",
s->samples_per_frame);
274 PRINT(
"log2 frame size",
s->log2_frame_size);
275 PRINT(
"max num subframes",
s->max_num_subframes);
276 PRINT(
"len prefix",
s->len_prefix);
277 PRINT(
"num channels",
s->nb_channels);
323 static VLCElem vlc_buf[2108 + 3912];
351 for (
int i = 0;
i < 33;
i++)
367 unsigned int channel_mask;
369 int log2_max_num_subframes;
370 int num_possible_block_sizes;
393 s->decode_flags = 0x10d6;
394 s->bits_per_sample = 16;
401 s->decode_flags = 0x10d6;
402 s->bits_per_sample = 16;
404 s->nb_channels = edata_ptr[32 + ((edata_ptr[0]==3)?0:8) + 4*num_stream + 0];
406 s->decode_flags = 0x10d6;
407 s->bits_per_sample = 16;
409 s->nb_channels = edata_ptr[8 + 20*num_stream + 17];
411 s->decode_flags =
AV_RL16(edata_ptr+14);
412 channel_mask =
AV_RL32(edata_ptr+2);
413 s->bits_per_sample =
AV_RL16(edata_ptr);
416 if (
s->bits_per_sample > 32 ||
s->bits_per_sample < 1) {
427 if (
s->log2_frame_size > 25) {
436 s->len_prefix = (
s->decode_flags & 0x40);
445 s->samples_per_frame = 1 <<
bits;
447 s->samples_per_frame = 512;
451 log2_max_num_subframes = ((
s->decode_flags & 0x38) >> 3);
452 s->max_num_subframes = 1 << log2_max_num_subframes;
453 if (
s->max_num_subframes == 16 ||
s->max_num_subframes == 4)
454 s->max_subframe_len_bit = 1;
455 s->subframe_len_bits =
av_log2(log2_max_num_subframes) + 1;
457 num_possible_block_sizes = log2_max_num_subframes + 1;
458 s->min_samples_per_subframe =
s->samples_per_frame /
s->max_num_subframes;
459 s->dynamic_range_compression = (
s->decode_flags & 0x80);
463 s->max_num_subframes);
469 s->min_samples_per_subframe);
473 if (
s->avctx->sample_rate <= 0) {
478 if (
s->nb_channels <= 0) {
493 for (
i = 0;
i <
s->nb_channels;
i++)
494 s->channel[
i].prev_block_len =
s->samples_per_frame;
499 if (channel_mask & 8) {
502 if (channel_mask &
mask)
509 for (
i = 0;
i < num_possible_block_sizes;
i++) {
510 int subframe_len =
s->samples_per_frame >>
i;
515 s->sfb_offsets[
i][0] = 0;
517 for (x = 0; x <
MAX_BANDS-1 &&
s->sfb_offsets[
i][band - 1] < subframe_len; x++) {
520 if (
offset >
s->sfb_offsets[
i][band - 1])
523 if (
offset >= subframe_len)
526 s->sfb_offsets[
i][band - 1] = subframe_len;
527 s->num_sfb[
i] = band - 1;
528 if (
s->num_sfb[
i] <= 0) {
540 for (
i = 0;
i < num_possible_block_sizes;
i++) {
542 for (
b = 0;
b <
s->num_sfb[
i];
b++) {
545 +
s->sfb_offsets[
i][
b + 1] - 1) <<
i) >> 1;
546 for (x = 0; x < num_possible_block_sizes; x++) {
548 while (
s->sfb_offsets[x][v + 1] << x <
offset) {
552 s->sf_offsets[
i][x][
b] = v;
564 / (1ll << (
s->bits_per_sample - 1));
578 for (
i = 0;
i < num_possible_block_sizes;
i++) {
579 int block_size =
s->samples_per_frame >>
i;
580 int cutoff = (440*block_size + 3LL * (
s->avctx->sample_rate >> 1) - 1)
581 /
s->avctx->sample_rate;
582 s->subwoofer_cutoffs[
i] =
av_clip(cutoff, 4, block_size);
621 int frame_len_shift = 0;
625 if (
offset ==
s->samples_per_frame -
s->min_samples_per_subframe)
626 return s->min_samples_per_subframe;
632 if (
s->max_subframe_len_bit) {
634 frame_len_shift = 1 +
get_bits(&
s->gb,
s->subframe_len_bits-1);
636 frame_len_shift =
get_bits(&
s->gb,
s->subframe_len_bits);
638 subframe_len =
s->samples_per_frame >> frame_len_shift;
641 if (subframe_len < s->min_samples_per_subframe ||
642 subframe_len >
s->samples_per_frame) {
674 int channels_for_cur_subframe =
s->nb_channels;
675 int fixed_channel_layout = 0;
676 int min_channel_len = 0;
686 for (
c = 0;
c <
s->nb_channels;
c++)
687 s->channel[
c].num_subframes = 0;
690 fixed_channel_layout = 1;
697 for (
c = 0;
c <
s->nb_channels;
c++) {
698 if (num_samples[
c] == min_channel_len) {
699 if (fixed_channel_layout || channels_for_cur_subframe == 1 ||
700 (min_channel_len ==
s->samples_per_frame -
s->min_samples_per_subframe))
701 contains_subframe[
c] = 1;
705 contains_subframe[
c] = 0;
713 min_channel_len += subframe_len;
714 for (
c = 0;
c <
s->nb_channels;
c++) {
717 if (contains_subframe[
c]) {
720 "broken frame: num subframes > 31\n");
724 num_samples[
c] += subframe_len;
726 if (num_samples[
c] >
s->samples_per_frame) {
728 "channel len > samples_per_frame\n");
731 }
else if (num_samples[
c] <= min_channel_len) {
732 if (num_samples[
c] < min_channel_len) {
733 channels_for_cur_subframe = 0;
734 min_channel_len = num_samples[
c];
736 ++channels_for_cur_subframe;
739 }
while (min_channel_len < s->samples_per_frame);
741 for (
c = 0;
c <
s->nb_channels;
c++) {
744 for (
i = 0;
i <
s->channel[
c].num_subframes;
i++) {
745 ff_dlog(
s->avctx,
"frame[%"PRIu32
"] channel[%i] subframe[%i]"
746 " len %i\n",
s->frame_num,
c,
i,
747 s->channel[
c].subframe_len[
i]);
748 s->channel[
c].subframe_offset[
i] =
offset;
779 for (x = 0; x <
i; x++) {
781 for (y = 0; y <
i + 1; y++) {
784 int n = rotation_offset[
offset + x];
790 cosv =
sin64[32 - n];
792 sinv =
sin64[64 - n];
793 cosv = -
sin64[n - 32];
797 (v1 * sinv) - (v2 * cosv);
799 (v1 * cosv) + (v2 * sinv);
821 if (
s->nb_channels > 1) {
822 int remaining_channels =
s->channels_for_cur_subframe;
826 "Channel transform bit");
830 for (
s->num_chgroups = 0; remaining_channels &&
831 s->num_chgroups <
s->channels_for_cur_subframe;
s->num_chgroups++) {
838 if (remaining_channels > 2) {
839 for (
i = 0;
i <
s->channels_for_cur_subframe;
i++) {
840 int channel_idx =
s->channel_indexes_for_cur_subframe[
i];
841 if (!
s->channel[channel_idx].grouped
844 s->channel[channel_idx].grouped = 1;
845 *channel_data++ =
s->channel[channel_idx].coeffs;
850 for (
i = 0;
i <
s->channels_for_cur_subframe;
i++) {
851 int channel_idx =
s->channel_indexes_for_cur_subframe[
i];
852 if (!
s->channel[channel_idx].grouped)
853 *channel_data++ =
s->channel[channel_idx].coeffs;
854 s->channel[channel_idx].grouped = 1;
863 "Unknown channel transform type");
868 if (
s->nb_channels == 2) {
890 "Coupled channels > 6");
906 for (
i = 0;
i <
s->num_bands;
i++) {
930 static const uint32_t fval_tab[16] = {
931 0x00000000, 0x3f800000, 0x40000000, 0x40400000,
932 0x40800000, 0x40a00000, 0x40c00000, 0x40e00000,
933 0x41000000, 0x41100000, 0x41200000, 0x41300000,
934 0x41400000, 0x41500000, 0x41600000, 0x41700000,
945 ff_dlog(
s->avctx,
"decode coefficients for channel %i\n",
c);
960 while ((
s->transmit_num_vec_coeffs || !rl_mode) &&
969 for (
i = 0;
i < 4;
i += 2) {
982 vals[
i] = fval_tab[idx >> 4 ];
983 vals[
i+1] = fval_tab[idx & 0xF];
987 vals[0] = fval_tab[ idx >> 12 ];
988 vals[1] = fval_tab[(idx >> 8) & 0xF];
989 vals[2] = fval_tab[(idx >> 4) & 0xF];
990 vals[3] = fval_tab[ idx & 0xF];
994 for (
i = 0;
i < 4;
i++) {
1000 ci->
coeffs[cur_coeff] = 0;
1003 rl_mode |= (++num_zeros >
s->subframe_len >> 8);
1010 if (cur_coeff < s->subframe_len) {
1013 memset(&ci->
coeffs[cur_coeff], 0,
1014 sizeof(*ci->
coeffs) * (
s->subframe_len - cur_coeff));
1017 cur_coeff,
s->subframe_len,
1018 s->subframe_len,
s->esc_len, 0);
1039 for (
i = 0;
i <
s->channels_for_cur_subframe;
i++) {
1040 int c =
s->channel_indexes_for_cur_subframe[
i];
1043 s->channel[
c].scale_factors =
s->channel[
c].saved_scale_factors[!
s->channel[
c].scale_factor_idx];
1044 sf_end =
s->channel[
c].scale_factors +
s->num_bands;
1051 if (
s->channel[
c].reuse_sf) {
1052 const int8_t* sf_offsets =
s->sf_offsets[
s->table_idx][
s->channel[
c].table_idx];
1054 for (
b = 0;
b <
s->num_bands;
b++)
1055 s->channel[
c].scale_factors[
b] =
1056 s->channel[
c].saved_scale_factors[
s->channel[
c].scale_factor_idx][*sf_offsets++];
1059 if (!
s->channel[
c].cur_subframe ||
get_bits1(&
s->gb)) {
1061 if (!
s->channel[
c].reuse_sf) {
1064 s->channel[
c].scale_factor_step =
get_bits(&
s->gb, 2) + 1;
1065 val = 45 /
s->channel[
c].scale_factor_step;
1066 for (sf =
s->channel[
c].scale_factors; sf < sf_end; sf++) {
1073 for (
i = 0;
i <
s->num_bands;
i++) {
1084 sign = (
code & 1) - 1;
1086 }
else if (idx == 1) {
1095 if (
i >=
s->num_bands) {
1097 "invalid scale factor coding\n");
1100 s->channel[
c].scale_factors[
i] += (
val ^ sign) - sign;
1104 s->channel[
c].scale_factor_idx = !
s->channel[
c].scale_factor_idx;
1105 s->channel[
c].table_idx =
s->table_idx;
1106 s->channel[
c].reuse_sf = 1;
1110 s->channel[
c].max_scale_factor =
s->channel[
c].scale_factors[0];
1111 for (sf =
s->channel[
c].scale_factors + 1; sf < sf_end; sf++) {
1112 s->channel[
c].max_scale_factor =
1113 FFMAX(
s->channel[
c].max_scale_factor, *sf);
1128 for (
i = 0;
i <
s->num_chgroups;
i++) {
1129 if (
s->chgroup[
i].transform) {
1131 const int num_channels =
s->chgroup[
i].num_channels;
1132 float** ch_data =
s->chgroup[
i].channel_data;
1133 float** ch_end = ch_data + num_channels;
1134 const int8_t*
tb =
s->chgroup[
i].transform_band;
1138 for (sfb =
s->cur_sfb_offsets;
1139 sfb < s->cur_sfb_offsets +
s->num_bands; sfb++) {
1143 for (y = sfb[0]; y <
FFMIN(sfb[1],
s->subframe_len); y++) {
1144 const float* mat =
s->chgroup[
i].decorrelation_matrix;
1145 const float* data_end =
data + num_channels;
1146 float* data_ptr =
data;
1149 for (ch = ch_data; ch < ch_end; ch++)
1150 *data_ptr++ = (*ch)[y];
1152 for (ch = ch_data; ch < ch_end; ch++) {
1155 while (data_ptr < data_end)
1156 sum += *data_ptr++ * *mat++;
1161 }
else if (
s->nb_channels == 2) {
1162 int len =
FFMIN(sfb[1],
s->subframe_len) - sfb[0];
1163 s->fdsp->vector_fmul_scalar(ch_data[0] + sfb[0],
1164 ch_data[0] + sfb[0],
1166 s->fdsp->vector_fmul_scalar(ch_data[1] + sfb[0],
1167 ch_data[1] + sfb[0],
1182 for (
i = 0;
i <
s->channels_for_cur_subframe;
i++) {
1183 int c =
s->channel_indexes_for_cur_subframe[
i];
1185 int winlen =
s->channel[
c].prev_block_len;
1186 float* start =
s->channel[
c].coeffs - (winlen >> 1);
1188 if (
s->subframe_len < winlen) {
1189 start += (winlen -
s->subframe_len) >> 1;
1190 winlen =
s->subframe_len;
1197 s->fdsp->vector_fmul_window(start, start, start + winlen,
1200 s->channel[
c].prev_block_len =
s->subframe_len;
1211 int offset =
s->samples_per_frame;
1212 int subframe_len =
s->samples_per_frame;
1214 int total_samples =
s->samples_per_frame *
s->nb_channels;
1215 int transmit_coeffs = 0;
1216 int cur_subwoofer_cutoff;
1224 for (
i = 0;
i <
s->nb_channels;
i++) {
1225 s->channel[
i].grouped = 0;
1226 if (
offset >
s->channel[
i].decoded_samples) {
1227 offset =
s->channel[
i].decoded_samples;
1229 s->channel[
i].subframe_len[
s->channel[
i].cur_subframe];
1234 "processing subframe with offset %i len %i\n",
offset, subframe_len);
1237 s->channels_for_cur_subframe = 0;
1238 for (
i = 0;
i <
s->nb_channels;
i++) {
1239 const int cur_subframe =
s->channel[
i].cur_subframe;
1241 total_samples -=
s->channel[
i].decoded_samples;
1244 if (
offset ==
s->channel[
i].decoded_samples &&
1245 subframe_len ==
s->channel[
i].subframe_len[cur_subframe]) {
1246 total_samples -=
s->channel[
i].subframe_len[cur_subframe];
1247 s->channel[
i].decoded_samples +=
1248 s->channel[
i].subframe_len[cur_subframe];
1249 s->channel_indexes_for_cur_subframe[
s->channels_for_cur_subframe] =
i;
1250 ++
s->channels_for_cur_subframe;
1257 s->parsed_all_subframes = 1;
1260 ff_dlog(
s->avctx,
"subframe is part of %i channels\n",
1261 s->channels_for_cur_subframe);
1264 s->table_idx =
av_log2(
s->samples_per_frame/subframe_len);
1265 s->num_bands =
s->num_sfb[
s->table_idx];
1266 s->cur_sfb_offsets =
s->sfb_offsets[
s->table_idx];
1267 cur_subwoofer_cutoff =
s->subwoofer_cutoffs[
s->table_idx];
1270 offset +=
s->samples_per_frame >> 1;
1272 for (
i = 0;
i <
s->channels_for_cur_subframe;
i++) {
1273 int c =
s->channel_indexes_for_cur_subframe[
i];
1275 s->channel[
c].coeffs = &
s->channel[
c].out[
offset];
1278 s->subframe_len = subframe_len;
1279 s->esc_len =
av_log2(
s->subframe_len - 1) + 1;
1284 if (!(num_fill_bits =
get_bits(&
s->gb, 2))) {
1289 if (num_fill_bits >= 0) {
1310 for (
i = 0;
i <
s->channels_for_cur_subframe;
i++) {
1311 int c =
s->channel_indexes_for_cur_subframe[
i];
1312 if ((
s->channel[
c].transmit_coefs =
get_bits1(&
s->gb)))
1313 transmit_coeffs = 1;
1317 if (transmit_coeffs) {
1319 int quant_step = 90 *
s->bits_per_sample >> 4;
1322 if ((
s->transmit_num_vec_coeffs =
get_bits1(&
s->gb))) {
1323 int num_bits =
av_log2((
s->subframe_len + 3)/4) + 1;
1324 for (
i = 0;
i <
s->channels_for_cur_subframe;
i++) {
1325 int c =
s->channel_indexes_for_cur_subframe[
i];
1326 int num_vec_coeffs =
get_bits(&
s->gb, num_bits) << 2;
1327 if (num_vec_coeffs >
s->subframe_len) {
1332 s->channel[
c].num_vec_coeffs = num_vec_coeffs;
1335 for (
i = 0;
i <
s->channels_for_cur_subframe;
i++) {
1336 int c =
s->channel_indexes_for_cur_subframe[
i];
1337 s->channel[
c].num_vec_coeffs =
s->subframe_len;
1344 const int sign = (
step == 31) - 1;
1350 quant_step += ((
quant +
step) ^ sign) - sign;
1352 if (quant_step < 0) {
1358 if (
s->channels_for_cur_subframe == 1) {
1359 s->channel[
s->channel_indexes_for_cur_subframe[0]].quant_step = quant_step;
1362 for (
i = 0;
i <
s->channels_for_cur_subframe;
i++) {
1363 int c =
s->channel_indexes_for_cur_subframe[
i];
1364 s->channel[
c].quant_step = quant_step;
1367 s->channel[
c].quant_step +=
get_bits(&
s->gb, modifier_len) + 1;
1369 ++
s->channel[
c].quant_step;
1379 ff_dlog(
s->avctx,
"BITSTREAM: subframe header length was %i\n",
1383 for (
i = 0;
i <
s->channels_for_cur_subframe;
i++) {
1384 int c =
s->channel_indexes_for_cur_subframe[
i];
1385 if (
s->channel[
c].transmit_coefs &&
1389 memset(
s->channel[
c].coeffs, 0,
1390 sizeof(*
s->channel[
c].coeffs) * subframe_len);
1393 ff_dlog(
s->avctx,
"BITSTREAM: subframe length was %i\n",
1396 if (transmit_coeffs) {
1401 for (
i = 0;
i <
s->channels_for_cur_subframe;
i++) {
1402 int c =
s->channel_indexes_for_cur_subframe[
i];
1403 const int* sf =
s->channel[
c].scale_factors;
1406 if (
c ==
s->lfe_channel)
1407 memset(&
s->tmp[cur_subwoofer_cutoff], 0,
sizeof(*
s->tmp) *
1408 (subframe_len - cur_subwoofer_cutoff));
1411 for (
b = 0;
b <
s->num_bands;
b++) {
1412 const int end =
FFMIN(
s->cur_sfb_offsets[
b+1],
s->subframe_len);
1413 const int exp =
s->channel[
c].quant_step -
1414 (
s->channel[
c].max_scale_factor - *sf++) *
1415 s->channel[
c].scale_factor_step;
1417 int start =
s->cur_sfb_offsets[
b];
1418 s->fdsp->vector_fmul_scalar(
s->tmp + start,
1419 s->channel[
c].coeffs + start,
1420 quant, end - start);
1424 tx_fn(tx,
s->channel[
c].coeffs,
s->tmp,
sizeof(
float));
1432 for (
i = 0;
i <
s->channels_for_cur_subframe;
i++) {
1433 int c =
s->channel_indexes_for_cur_subframe[
i];
1434 if (
s->channel[
c].cur_subframe >=
s->channel[
c].num_subframes) {
1438 ++
s->channel[
c].cur_subframe;
1453 int more_frames = 0;
1461 ff_dlog(
s->avctx,
"decoding frame with length %x\n",
len);
1472 for (
i = 0;
i <
s->nb_channels *
s->nb_channels;
i++)
1478 if (
s->dynamic_range_compression) {
1480 ff_dlog(
s->avctx,
"drc_gain %i\n",
s->drc_gain);
1490 s->trim_start =
s->trim_end = 0;
1493 ff_dlog(
s->avctx,
"BITSTREAM: frame header length was %i\n",
1497 s->parsed_all_subframes = 0;
1498 for (
i = 0;
i <
s->nb_channels;
i++) {
1499 s->channel[
i].decoded_samples = 0;
1500 s->channel[
i].cur_subframe = 0;
1501 s->channel[
i].reuse_sf = 0;
1505 while (!
s->parsed_all_subframes) {
1513 for (
i = 0;
i <
s->nb_channels;
i++)
1515 s->samples_per_frame *
sizeof(*
s->channel[
i].out));
1517 for (
i = 0;
i <
s->nb_channels;
i++) {
1519 memcpy(&
s->channel[
i].out[0],
1520 &
s->channel[
i].out[
s->samples_per_frame],
1521 s->samples_per_frame *
sizeof(*
s->channel[
i].out) >> 1);
1524 if (
s->skip_frame) {
1532 if (
s->len_prefix) {
1536 "frame[%"PRIu32
"] would have to skip %i bits\n",
1586 s->num_saved_bits =
s->frame_offset;
1588 buflen = (
s->num_saved_bits +
len + 7) >> 3;
1600 s->num_saved_bits +=
len;
1626 const uint8_t* buf = avpkt->
data;
1627 int buf_size = avpkt->
size;
1628 int num_bits_prev_frame;
1629 int packet_sequence_number;
1644 for (
i = 0;
i <
s->nb_channels;
i++) {
1646 s->samples_per_frame *
sizeof(*
s->channel[
i].out));
1649 s->samples_per_frame *
sizeof(*
s->channel[
i].out) >> 1);
1657 else if (
s->packet_done ||
s->packet_loss) {
1675 s->buf_bit_size = buf_size << 3;
1682 packet_sequence_number =
get_bits(gb, 4);
1686 ff_dlog(avctx,
"packet[%"PRId64
"]: number of frames %d\n", avctx->
frame_num, num_frames);
1687 packet_sequence_number = 0;
1691 num_bits_prev_frame =
get_bits(gb,
s->log2_frame_size);
1695 ff_dlog(avctx,
"packet[%"PRId64
"]: skip packets %d\n", avctx->
frame_num,
s->skip_packets);
1699 num_bits_prev_frame);
1703 ((
s->packet_sequence_number + 1) & 0xF) != packet_sequence_number) {
1706 "Packet loss detected! seq %"PRIx8
" vs %x\n",
1707 s->packet_sequence_number, packet_sequence_number);
1709 s->packet_sequence_number = packet_sequence_number;
1711 if (num_bits_prev_frame > 0) {
1713 if (num_bits_prev_frame >= remaining_packet_bits) {
1714 num_bits_prev_frame = remaining_packet_bits;
1721 ff_dlog(avctx,
"accumulated %x bits of frame data\n",
1722 s->num_saved_bits -
s->frame_offset);
1725 if (!
s->packet_loss)
1727 }
else if (
s->num_saved_bits -
s->frame_offset) {
1728 ff_dlog(avctx,
"ignoring %x previously saved bits\n",
1729 s->num_saved_bits -
s->frame_offset);
1732 if (
s->packet_loss) {
1736 s->num_saved_bits = 0;
1742 if (avpkt->
size <
s->next_packet_start) {
1747 s->buf_bit_size = (avpkt->
size -
s->next_packet_start) << 3;
1756 if (!
s->packet_loss)
1758 }
else if (!
s->len_prefix
1778 if (
s->packet_done && !
s->packet_loss &&
1823 int *got_frame_ptr,
AVPacket *avpkt)
1839 int *got_frame_ptr,
AVPacket *avpkt)
1842 int got_stream_frame_ptr = 0;
1843 int i,
ret = 0, eof = 0;
1845 if (!
s->frames[
s->current_stream]->data[0]) {
1847 s->frames[
s->current_stream]->nb_samples = 512;
1850 }
else if (
s->frames[
s->current_stream]->nb_samples != 512) {
1853 s->frames[
s->current_stream]->nb_samples = 512;
1858 if (!
s->xma[
s->current_stream].eof_done) {
1860 &got_stream_frame_ptr, avpkt);
1866 for (
i = 0;
i <
s->num_streams;
i++) {
1867 if (!
s->xma[
i].eof_done &&
s->frames[
i]->data[0]) {
1869 &got_stream_frame_ptr, avpkt);
1872 eof &=
s->xma[
i].eof_done;
1876 if (
s->xma[0].trim_start)
1877 s->trim_start =
s->xma[0].trim_start;
1878 if (
s->xma[0].trim_end)
1879 s->trim_end =
s->xma[0].trim_end;
1882 if (got_stream_frame_ptr) {
1883 const int nb_samples =
s->frames[
s->current_stream]->nb_samples;
1884 void *
left[1] = {
s->frames[
s->current_stream]->extended_data[0] };
1885 void *right[1] = {
s->frames[
s->current_stream]->extended_data[1] };
1888 if (
s->xma[
s->current_stream].nb_channels > 1)
1890 }
else if (
ret < 0) {
1891 s->current_stream = 0;
1898 if (
s->xma[
s->current_stream].packet_done ||
1899 s->xma[
s->current_stream].packet_loss) {
1900 int nb_samples = INT_MAX;
1903 if (
s->xma[
s->current_stream].skip_packets != 0) {
1906 min[0] =
s->xma[0].skip_packets;
1909 for (
i = 1;
i <
s->num_streams;
i++) {
1910 if (
s->xma[
i].skip_packets <
min[0]) {
1911 min[0] =
s->xma[
i].skip_packets;
1916 s->current_stream =
min[1];
1920 for (
i = 0;
i <
s->num_streams;
i++) {
1921 s->xma[
i].skip_packets =
FFMAX(0,
s->xma[
i].skip_packets - 1);
1925 if (!eof && avpkt->
size)
1926 nb_samples -=
FFMIN(nb_samples, 4096);
1929 if ((nb_samples > 0 || eof || !avpkt->
size) && !
s->flushed) {
1933 nb_samples -=
av_clip(
s->trim_end +
s->trim_start - 128 - 64, 0, nb_samples);
1941 for (
i = 0;
i <
s->num_streams;
i++) {
1942 const int start_ch =
s->start_channel[
i];
1946 if (
s->xma[
i].nb_channels > 1) {
1952 *got_frame_ptr = nb_samples > 0;
1962 int i,
ret, start_channels = 0;
2005 for (
i = 0;
i <
s->num_streams;
i++) {
2013 s->start_channel[
i] = start_channels;
2014 start_channels +=
s->xma[
i].nb_channels;
2022 if (!
s->samples[0][
i] || !
s->samples[1][
i])
2034 for (
i = 0;
i <
s->num_streams;
i++) {
2053 for (
i = 0;
i <
s->nb_channels;
i++)
2054 memset(
s->channel[
i].out, 0,
s->samples_per_frame *
2055 sizeof(*
s->channel[
i].out));
2057 s->skip_packets = 0;
2083 for (
i = 0;
i <
s->num_streams;
i++)
2086 s->current_stream = 0;
2103 #if FF_API_SUBFRAMES
2104 AV_CODEC_CAP_SUBFRAMES |
2124 #if FF_API_SUBFRAMES
2125 AV_CODEC_CAP_SUBFRAMES |
2144 #if FF_API_SUBFRAMES
2145 AV_CODEC_CAP_SUBFRAMES |
uint16_t num_vec_coeffs
number of vector coded coefficients
void av_audio_fifo_free(AVAudioFifo *af)
Free an AVAudioFifo.
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.
#define FF_CODEC_CAP_INIT_CLEANUP
The codec allows calling the close function for deallocation even if the init function returned a fai...
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
GetBitContext gb
bitstream reader context
uint16_t samples_per_frame
number of samples to output
SINETABLE_CONST float *const ff_sine_windows[]
int8_t scale_factor_step
scaling step for the current subframe
static const uint8_t vec4_lens[HUFF_VEC4_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
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.
int av_audio_fifo_write(AVAudioFifo *af, void *const *data, int nb_samples)
Write data to an AVAudioFifo.
static VLCElem vec2_vlc[562]
2 coefficients per symbol
static const uint8_t vec2_table[HUFF_VEC2_SIZE][2]
int skip_samples
Number of audio samples to skip at the start of the next decoded frame.
static void init_put_bits(PutBitContext *s, uint8_t *buffer, int buffer_size)
Initialize the PutBitContext s.
static uint8_t * append(uint8_t *buf, const uint8_t *src, int size)
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
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 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.
int ff_wma_run_level_decode(AVCodecContext *avctx, GetBitContext *gb, const VLCElem *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.
#define WMAPRO_BLOCK_MAX_SIZE
maximum block size
enum AVChannelOrder order
Channel order used in this layout.
static av_always_inline uint32_t av_float2int(float f)
Reinterpret a float as a 32-bit integer.
static VLCElem vec4_vlc[604]
4 coefficients per symbol
int nb_channels
Number of channels in this layout.
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
av_cold int av_tx_init(AVTXContext **ctx, av_tx_fn *tx, enum AVTXType type, int inv, int len, const void *scale, uint64_t flags)
Initialize a transform context with the given configuration (i)MDCTs with an odd length are currently...
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.
static const VLCElem * coef_vlc[2]
coefficient run length vlc codes
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.
static SDL_Window * window
Context for an Audio FIFO Buffer.
AVCodec p
The public AVCodec.
static VLCElem vec1_vlc[562]
1 coefficient per symbol
AVChannelLayout ch_layout
Audio channel layout.
static const uint16_t coef0_syms[HUFF_COEF0_SIZE]
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.
AVChannelLayout ch_layout
Channel layout of the audio data.
int8_t sf_offsets[WMAPRO_BLOCK_SIZES][WMAPRO_BLOCK_SIZES][MAX_BANDS]
scale factor resample matrix
WMAProChannelGrp chgroup[WMAPRO_MAX_CHANNELS]
channel group information
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.
static VLCElem sf_vlc[616]
scale factor DPCM vlc
static const uint8_t quant[64]
float out[WMAPRO_BLOCK_MAX_SIZE+WMAPRO_BLOCK_MAX_SIZE/2]
output buffer
static int ff_thread_once(char *control, void(*routine)(void))
#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 int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
const FFCodec ff_xma1_decoder
void(* av_tx_fn)(AVTXContext *s, void *out, void *in, ptrdiff_t stride)
Function pointer to a function to perform the transform.
uint8_t subframe_len_bits
number of bits used for the subframe length
For static VLCs, the number of bits can often be hardcoded at each get_vlc2() callsite.
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
@ AV_TX_FLOAT_MDCT
Standard MDCT with a sample data type of float, double or int32_t, respecively.
int * scale_factors
pointer to the scale factor values used for decoding
#define FF_CODEC_DECODE_CB(func)
int8_t skip_frame
skip output step
int16_t subwoofer_cutoffs[WMAPRO_BLOCK_SIZES]
subwoofer cutoff values
uint32_t decode_flags
used compression features
const FFCodec ff_xma2_decoder
@ AV_CHANNEL_ORDER_UNSPEC
Only the channel count is specified, without any further information about the channel order.
uint8_t packet_loss
set in case of bitstream error
int av_channel_layout_from_mask(AVChannelLayout *channel_layout, uint64_t mask)
Initialize a native channel layout from a bitmask indicating which channels are present.
#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]
#define CODEC_LONG_NAME(str)
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]
#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
AVAudioFifo * av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels, int nb_samples)
Allocate an AVAudioFifo.
int8_t transform
transform on / off
static const uint8_t vec1_table[HUFF_VEC1_SIZE][2]
struct AVCodecInternal * internal
Private context used for internal data.
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
static av_always_inline int get_vlc2(GetBitContext *s, const VLCElem *table, int bits, int max_depth)
Parse a vlc code.
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)
uint32_t frame_num
current frame number (not used for decoding)
static int decode_packet(AVCodecContext *avctx, WMAProDecodeCtx *s, AVFrame *frame, int *got_frame_ptr, AVPacket *avpkt)
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
int(* init)(AVBSFContext *ctx)
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
static void scale(int *out, const int *in, const int w, const int h, const int shift)
float * coeffs
pointer to the subframe decode buffer
#define DECLARE_ALIGNED(n, t, v)
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
static const uint8_t scale_table[]
#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
int av_audio_fifo_read(AVAudioFifo *af, void *const *data, int nb_samples)
Read data from an AVAudioFifo.
static av_cold void decode_init_static(void)
int8_t scale_factor_idx
index for the transmitted scale factor values (used for resampling)
#define MAX_SUBFRAMES
max number of subframes per channel
AVAudioFifo * samples[2][XMA_MAX_STREAMS]
static const uint8_t *BS_FUNC() align(BSCTX *bc)
Skip bits to a byte boundary.
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
int av_audio_fifo_size(AVAudioFifo *af)
Get the current number of samples in the AVAudioFifo available for reading.
int8_t transform_band[MAX_BANDS]
controls if the transform is enabled for a certain band
av_cold void av_tx_uninit(AVTXContext **ctx)
Frees a context and sets *ctx to NULL, does nothing when *ctx == NULL.
uint8_t max_num_subframes
int8_t reuse_sf
share scale factors between subframes
int next_packet_start
start offset of the next wma packet in the demuxer packet
int nb_samples
number of audio samples (per channel) described by this frame
#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
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.
static int xma_decode_packet(AVCodecContext *avctx, AVFrame *frame, int *got_frame_ptr, AVPacket *avpkt)
uint8_t ** extended_data
pointers to the data planes/channels.
static const float coef1_level[HUFF_COEF1_SIZE]
av_tx_fn tx_fn[WMAPRO_BLOCK_SIZES]
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.
uint16_t trim_start
number of samples to skip at start
tables for wmapro decoding
GetBitContext pgb
bitstream reader context for the packet
int64_t frame_num
Frame counter, set by libavcodec.
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs.
static void save_bits(WMAProDecodeCtx *s, GetBitContext *gb, int len, int append)
Fill the bit reservoir with a (partial) frame.
uint8_t num_channels
number of channels in the group
static int wmapro_decode_packet(AVCodecContext *avctx, AVFrame *frame, int *got_frame_ptr, AVPacket *avpkt)
Decode a single WMA packet.
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.
Tag MUST be and< 10hcoeff half pel interpolation filter coefficients, hcoeff[0] are the 2 middle coefficients[1] are the next outer ones and so on, resulting in a filter like:...eff[2], hcoeff[1], hcoeff[0], hcoeff[0], hcoeff[1], hcoeff[2] ... the sign of the coefficients is not explicitly stored but alternates after each coeff and coeff[0] is positive, so ...,+,-,+,-,+,+,-,+,-,+,... hcoeff[0] is not explicitly stored but found by subtracting the sum of all stored coefficients with signs from 32 hcoeff[0]=32 - hcoeff[1] - hcoeff[2] - ... a good choice for hcoeff and htaps is htaps=6 hcoeff={40,-10, 2} an alternative which requires more computations at both encoder and decoder side and may or may not be better is htaps=8 hcoeff={42,-14, 6,-2}ref_frames minimum of the number of available reference frames and max_ref_frames for example the first frame after a key frame always has ref_frames=1spatial_decomposition_type wavelet type 0 is a 9/7 symmetric compact integer wavelet 1 is a 5/3 symmetric compact integer wavelet others are reserved stored as delta from last, last is reset to 0 if always_reset||keyframeqlog quality(logarithmic quantizer scale) stored as delta from last, last is reset to 0 if always_reset||keyframemv_scale stored as delta from last, last is reset to 0 if always_reset||keyframe FIXME check that everything works fine if this changes between framesqbias dequantization bias stored as delta from last, last is reset to 0 if always_reset||keyframeblock_max_depth maximum depth of the block tree stored as delta from last, last is reset to 0 if always_reset||keyframequant_table quantization tableHighlevel bitstream structure:==============================--------------------------------------------|Header|--------------------------------------------|------------------------------------|||Block0||||split?||||yes no||||......... intra?||||:Block01 :yes no||||:Block02 :....... ..........||||:Block03 ::y DC ::ref index:||||:Block04 ::cb DC ::motion x :||||......... :cr DC ::motion y :||||....... ..........|||------------------------------------||------------------------------------|||Block1|||...|--------------------------------------------|------------ ------------ ------------|||Y subbands||Cb subbands||Cr subbands||||--- ---||--- ---||--- ---|||||LL0||HL0||||LL0||HL0||||LL0||HL0|||||--- ---||--- ---||--- ---||||--- ---||--- ---||--- ---|||||LH0||HH0||||LH0||HH0||||LH0||HH0|||||--- ---||--- ---||--- ---||||--- ---||--- ---||--- ---|||||HL1||LH1||||HL1||LH1||||HL1||LH1|||||--- ---||--- ---||--- ---||||--- ---||--- ---||--- ---|||||HH1||HL2||||HH1||HL2||||HH1||HL2|||||...||...||...|||------------ ------------ ------------|--------------------------------------------Decoding process:=================------------|||Subbands|------------||||------------|Intra DC||||LL0 subband prediction ------------|\ Dequantization ------------------- \||Reference frames|\ IDWT|------- -------|Motion \|||Frame 0||Frame 1||Compensation . OBMC v -------|------- -------|--------------. \------> Frame n output Frame Frame<----------------------------------/|...|------------------- Range Coder:============Binary Range Coder:------------------- The implemented range coder is an adapted version based upon "Range encoding: an algorithm for removing redundancy from a digitised message." by G. N. N. Martin. The symbols encoded by the Snow range coder are bits(0|1). The associated probabilities are not fix but change depending on the symbol mix seen so far. bit seen|new state ---------+----------------------------------------------- 0|256 - state_transition_table[256 - old_state];1|state_transition_table[old_state];state_transition_table={ 0, 0, 0, 0, 0, 0, 0, 0, 20, 21, 22, 23, 24, 25, 26, 27, 28, 29, 30, 31, 32, 33, 34, 35, 36, 37, 37, 38, 39, 40, 41, 42, 43, 44, 45, 46, 47, 48, 49, 50, 51, 52, 53, 54, 55, 56, 56, 57, 58, 59, 60, 61, 62, 63, 64, 65, 66, 67, 68, 69, 70, 71, 72, 73, 74, 75, 75, 76, 77, 78, 79, 80, 81, 82, 83, 84, 85, 86, 87, 88, 89, 90, 91, 92, 93, 94, 94, 95, 96, 97, 98, 99, 100, 101, 102, 103, 104, 105, 106, 107, 108, 109, 110, 111, 112, 113, 114, 114, 115, 116, 117, 118, 119, 120, 121, 122, 123, 124, 125, 126, 127, 128, 129, 130, 131, 132, 133, 133, 134, 135, 136, 137, 138, 139, 140, 141, 142, 143, 144, 145, 146, 147, 148, 149, 150, 151, 152, 152, 153, 154, 155, 156, 157, 158, 159, 160, 161, 162, 163, 164, 165, 166, 167, 168, 169, 170, 171, 171, 172, 173, 174, 175, 176, 177, 178, 179, 180, 181, 182, 183, 184, 185, 186, 187, 188, 189, 190, 190, 191, 192, 194, 194, 195, 196, 197, 198, 199, 200, 201, 202, 202, 204, 205, 206, 207, 208, 209, 209, 210, 211, 212, 213, 215, 215, 216, 217, 218, 219, 220, 220, 222, 223, 224, 225, 226, 227, 227, 229, 229, 230, 231, 232, 234, 234, 235, 236, 237, 238, 239, 240, 241, 242, 243, 244, 245, 246, 247, 248, 248, 0, 0, 0, 0, 0, 0, 0};FIXME Range Coding of integers:------------------------- FIXME Neighboring Blocks:===================left and top are set to the respective blocks unless they are outside of the image in which case they are set to the Null block top-left is set to the top left block unless it is outside of the image in which case it is set to the left block if this block has no larger parent block or it is at the left side of its parent block and the top right block is not outside of the image then the top right block is used for top-right else the top-left block is used Null block y, cb, cr are 128 level, ref, mx and my are 0 Motion Vector Prediction:=========================1. the motion vectors of all the neighboring blocks are scaled to compensate for the difference of reference frames scaled_mv=(mv *(256 *(current_reference+1)/(mv.reference+1))+128)> the median of the scaled left
void ff_copy_bits(PutBitContext *pb, const uint8_t *src, int length)
Copy the content of src to the bitstream.
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_RL32
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
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
void av_channel_layout_uninit(AVChannelLayout *channel_layout)
Free any allocated data in the channel layout and reset the channel count to 0.
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
const av_cold VLCElem * ff_vlc_init_tables_from_lengths(VLCInitState *state, int nb_bits, int nb_codes, const int8_t *lens, int lens_wrap, const void *symbols, int symbols_wrap, int symbols_size, int offset, int flags)
AVTXContext * tx[WMAPRO_BLOCK_SIZES]
MDCT context per block size.
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.
uint8_t packet_done
set when a packet is fully decoded
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
#define avpriv_request_sample(...)
static av_cold int xma_decode_end(AVCodecContext *avctx)
static void flush_put_bits(PutBitContext *s)
Pad the end of the output stream with zeros.
int8_t lfe_channel
lfe channel index
static const uint8_t scale_rl_table[HUFF_SCALE_RL_SIZE][2]
uint16_t trim_end
number of samples to skip at end
#define VLC_INIT_STATIC_TABLE_FROM_LENGTHS(vlc_table, nb_bits, nb_codes, lens, lens_wrap, syms, syms_wrap, syms_size, offset, flags)
static const uint16_t vec4_syms[HUFF_VEC4_SIZE]
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
#define VLC_INIT_STATE(_table)
int8_t parsed_all_subframes
all subframes decoded?
This structure stores compressed data.
static const uint8_t coef0_lens[HUFF_COEF0_SIZE]
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.
void av_audio_fifo_reset(AVAudioFifo *af)
Reset the AVAudioFifo buffer.
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
float decorrelation_matrix[WMAPRO_MAX_CHANNELS *WMAPRO_MAX_CHANNELS]
static void BS_FUNC() skip(BSCTX *bc, unsigned int n)
Skip n bits in the buffer.
const FFCodec ff_wmapro_decoder
wmapro decoder
static const uint8_t coef1_table[HUFF_COEF1_SIZE][2]
static VLCElem sf_rl_vlc[1406]
scale factor run length vlc
#define WMAPRO_BLOCK_MIN_BITS
log2 of min block size