Go to the documentation of this file.
32 return (v >> 1) ^ -(v & 1);
44 return (v >> 1) ^ -(v & 1);
77 int i, j, *coeff_ptr =
c->dmix_coeff;
79 for (
i = 0;
i < m;
i++) {
84 if (!
c->primary_chset) {
86 sign = (
code >> 8) - 1;
94 c->dmix_scale[
i] = (
scale ^ sign) - sign;
95 c->dmix_scale_inv[
i] = (scale_inv ^ sign) - sign;
99 for (j = 0; j <
c->nchannels; j++) {
101 sign = (
code >> 8) - 1;
108 if (!
c->primary_chset)
111 *coeff_ptr++ = (
coeff ^ sign) - sign;
141 c->residual_encode =
get_bits(&
s->gb,
c->nchannels);
148 if (
c->storage_bit_res != 16 &&
c->storage_bit_res != 20 &&
c->storage_bit_res != 24) {
153 if (
c->pcm_bit_res >
c->storage_bit_res) {
154 av_log(
s->avctx,
AV_LOG_ERROR,
"Invalid PCM bit resolution for XLL channel set (%d > %d)\n",
c->pcm_bit_res,
c->storage_bit_res);
160 if (
c->freq > 192000) {
180 if (
c->primary_chset != (
c == p)) {
189 c->dmix_embedded =
c->dmix_coeffs_present &&
get_bits1(&
s->gb);
192 if (
c->dmix_coeffs_present &&
c->primary_chset) {
202 if (!
c->hier_chset &&
s->nchsets != 1) {
225 for (
i = 0, j = 0;
i <
s->ch_mask_nbits;
i++)
226 if (
c->ch_mask & (1
U <<
i))
227 c->ch_remap[j++] =
i;
230 if (
c->nchannels != 2 ||
s->nchsets != 1 ||
get_bits1(&
s->gb)) {
236 c->primary_chset = 1;
237 c->dmix_coeffs_present = 0;
238 c->dmix_embedded = 0;
245 if (
c->freq > 96000) {
258 c->freq >>=
c->nfreqbands - 1;
269 if (
c->storage_bit_res > 16)
271 else if (
c->storage_bit_res > 8)
277 if ((
s->nchsets > 1 ||
c->nfreqbands > 1) &&
c->nabits < 5)
280 for (band = 0,
b =
c->bands; band < c->nfreqbands; band++,
b++) {
282 if ((
b->decor_enabled =
get_bits1(&
s->gb)) &&
c->nchannels > 1) {
286 for (
i = 0;
i <
c->nchannels;
i++) {
288 if (
b->orig_order[
i] >=
c->nchannels) {
295 for (
i = 0;
i <
c->nchannels / 2;
i++)
298 for (
i = 0;
i <
c->nchannels;
i++)
299 b->orig_order[
i] =
i;
300 for (
i = 0;
i <
c->nchannels / 2;
i++)
301 b->decor_coeff[
i] = 0;
305 b->highest_pred_order = 0;
306 for (
i = 0;
i <
c->nchannels;
i++) {
308 if (
b->adapt_pred_order[
i] >
b->highest_pred_order)
309 b->highest_pred_order =
b->adapt_pred_order[
i];
311 if (
b->highest_pred_order >
s->nsegsamples) {
317 for (
i = 0;
i <
c->nchannels;
i++)
318 b->fixed_pred_order[
i] =
b->adapt_pred_order[
i] ? 0 :
get_bits(&
s->gb, 2);
321 for (
i = 0;
i <
c->nchannels;
i++) {
322 for (j = 0; j <
b->adapt_pred_order[
i]; j++) {
336 b->dmix_embedded =
c->dmix_embedded && (band == 0 ||
get_bits1(&
s->gb));
339 if ((band == 0 &&
s->scalable_lsbs) || (band != 0 &&
get_bits1(&
s->gb))) {
342 if (
b->lsb_section_size < 0 ||
b->lsb_section_size >
s->frame_size) {
348 if (
b->lsb_section_size && (
s->band_crc_present > 2 ||
349 (band == 0 &&
s->band_crc_present > 1)))
350 b->lsb_section_size += 2;
353 for (
i = 0;
i <
c->nchannels;
i++) {
355 if (
b->nscalablelsbs[
i] && !
b->lsb_section_size) {
361 b->lsb_section_size = 0;
362 for (
i = 0;
i <
c->nchannels;
i++)
363 b->nscalablelsbs[
i] = 0;
367 if ((band == 0 &&
s->scalable_lsbs) || (band != 0 &&
get_bits1(&
s->gb))) {
369 for (
i = 0;
i <
c->nchannels;
i++)
372 for (
i = 0;
i <
c->nchannels;
i++)
373 b->bit_width_adjust[
i] = 0;
391 int nchsamples =
s->nframesamples + ndecisamples;
392 int i, j, nsamples = nchsamples *
c->nchannels *
c->nfreqbands;
397 if (!
c->sample_buffer[0])
400 ptr =
c->sample_buffer[0] + ndecisamples;
401 for (
i = 0;
i <
c->nfreqbands;
i++) {
402 for (j = 0; j <
c->nchannels; j++) {
403 c->bands[
i].msb_sample_buffer[j] = ptr;
413 int i, j, nsamples = 0;
417 for (
i = 0;
i <
c->nfreqbands;
i++)
418 if (
c->bands[
i].lsb_section_size)
419 nsamples +=
s->nframesamples *
c->nchannels;
425 if (!
c->sample_buffer[1])
428 ptr =
c->sample_buffer[1];
429 for (
i = 0;
i <
c->nfreqbands;
i++) {
430 if (
c->bands[
i].lsb_section_size) {
431 for (j = 0; j <
c->nchannels; j++) {
432 c->bands[
i].lsb_sample_buffer[j] = ptr;
433 ptr +=
s->nframesamples;
436 for (j = 0; j <
c->nchannels; j++)
437 c->bands[
i].lsb_sample_buffer[j] =
NULL;
457 k =
c->seg_common ? 1 :
c->nchannels;
460 for (
i = 0;
i < k;
i++) {
466 if (!
c->seg_common &&
c->rice_code_flag[
i] &&
get_bits1(&
s->gb))
468 c->bitalloc_hybrid_linear[
i] =
get_bits(&
s->gb,
c->nabits) + 1;
471 c->bitalloc_hybrid_linear[
i] = 0;
475 for (
i = 0;
i < k;
i++) {
481 if (!
c->rice_code_flag[
i] &&
c->bitalloc_part_a[
i])
482 c->bitalloc_part_a[
i]++;
485 c->nsamples_part_a[
i] =
b->adapt_pred_order[
i];
487 c->nsamples_part_a[
i] =
b->highest_pred_order;
489 c->bitalloc_part_a[
i] = 0;
490 c->nsamples_part_a[
i] = 0;
497 if (!
c->rice_code_flag[
i] &&
c->bitalloc_part_b[
i])
498 c->bitalloc_part_b[
i]++;
503 for (
i = 0;
i <
c->nchannels;
i++) {
508 k =
c->seg_common ? 0 :
i;
511 part_a =
b->msb_sample_buffer[
i] + seg *
s->nsegsamples;
512 part_b = part_a +
c->nsamples_part_a[k];
513 nsamples_part_b =
s->nsegsamples -
c->nsamples_part_a[k];
518 if (!
c->rice_code_flag[k]) {
522 c->bitalloc_part_a[k]);
526 c->bitalloc_part_b[k]);
531 c->bitalloc_part_a[k]);
533 if (
c->bitalloc_hybrid_linear[k]) {
536 int nisosamples =
get_bits(&
s->gb,
s->nsegsamples_log2);
539 memset(part_b, 0,
sizeof(*part_b) * nsamples_part_b);
542 for (j = 0; j < nisosamples; j++) {
543 int loc =
get_bits(&
s->gb,
s->nsegsamples_log2);
544 if (loc >= nsamples_part_b) {
552 for (j = 0; j < nsamples_part_b; j++) {
554 part_b[j] =
get_linear(&
s->gb,
c->bitalloc_hybrid_linear[k]);
556 part_b[j] =
get_rice(&
s->gb,
c->bitalloc_part_b[k]);
567 if (seg == 0 && band == 1) {
569 for (
i = 0;
i <
c->nchannels;
i++)
575 if (
b->lsb_section_size) {
583 for (
i = 0;
i <
c->nchannels;
i++) {
584 if (
b->nscalablelsbs[
i]) {
586 b->lsb_sample_buffer[
i] + seg *
s->nsegsamples,
587 s->nsegsamples,
b->nscalablelsbs[
i]);
608 nsamples =
s->nframesamples;
611 nsamples =
s->nsegsamples;
614 for (
i = 0;
i <
c->nchannels;
i++) {
615 memset(
b->msb_sample_buffer[
i] +
offset, 0, nsamples *
sizeof(
int32_t));
616 if (
b->lsb_section_size)
617 memset(
b->lsb_sample_buffer[
i] +
offset, 0, nsamples *
sizeof(
int32_t));
620 if (seg <= 0 && band)
621 memset(
c->deci_history, 0,
sizeof(
c->deci_history));
624 memset(
b->nscalablelsbs, 0,
sizeof(
b->nscalablelsbs));
625 memset(
b->bit_width_adjust, 0,
sizeof(
b->bit_width_adjust));
632 int nsamples =
s->nframesamples;
636 for (
i = 0;
i <
c->nchannels;
i++) {
638 int order =
b->adapt_pred_order[
i];
642 for (j = 0; j < order; j++) {
643 int rc =
b->adapt_refl_coeff[
i][j];
644 for (k = 0; k < (j + 1) / 2; k++) {
645 int tmp1 =
coeff[ k ];
646 int tmp2 =
coeff[j - k - 1];
653 for (j = 0; j < nsamples - order; j++) {
655 for (k = 0; k < order; k++)
656 err += (int64_t)buf[j + k] *
coeff[order - k - 1];
661 for (j = 0; j <
b->fixed_pred_order[
i]; j++)
662 for (k = 1; k < nsamples; k++)
663 buf[k] += (
unsigned)buf[k - 1];
668 if (
b->decor_enabled) {
671 for (
i = 0;
i <
c->nchannels / 2;
i++) {
674 s->dcadsp->decor(
b->msb_sample_buffer[
i * 2 + 1],
675 b->msb_sample_buffer[
i * 2 ],
681 for (
i = 0;
i <
c->nchannels;
i++)
682 tmp[
i] =
b->msb_sample_buffer[
i];
684 for (
i = 0;
i <
c->nchannels;
i++)
685 b->msb_sample_buffer[
b->orig_order[
i]] =
tmp[
i];
689 if (
c->nfreqbands == 1)
690 for (
i = 0;
i <
c->nchannels;
i++)
691 s->output_samples[
c->ch_remap[
i]] =
b->msb_sample_buffer[
i];
696 int adj =
c->bands[band].bit_width_adjust[ch];
697 int shift =
c->bands[band].nscalablelsbs[ch];
699 if (
s->fixed_lsb_width)
700 shift =
s->fixed_lsb_width;
701 else if (
shift && adj)
712 int n, ch, nsamples =
s->nframesamples;
714 for (ch = 0; ch <
c->nchannels; ch++) {
717 int32_t *msb =
b->msb_sample_buffer[ch];
718 if (
b->nscalablelsbs[ch]) {
719 int32_t *lsb =
b->lsb_sample_buffer[ch];
720 int adj =
b->bit_width_adjust[ch];
721 for (n = 0; n < nsamples; n++)
722 msb[n] = msb[n] * (
SUINT)(1 <<
shift) + (lsb[n] << adj);
724 for (n = 0; n < nsamples; n++)
733 int ch, nsamples =
s->nframesamples;
740 2 * nsamples *
c->nchannels *
sizeof(
int32_t));
741 if (!
c->sample_buffer[2])
745 ptr =
c->sample_buffer[2];
746 for (ch = 0; ch <
c->nchannels; ch++) {
747 int32_t *band0 =
c->bands[0].msb_sample_buffer[ch];
748 int32_t *band1 =
c->bands[1].msb_sample_buffer[ch];
752 c->deci_history[ch],
sizeof(
c->deci_history[0]));
755 s->dcadsp->assemble_freq_bands(ptr, band0, band1,
760 s->output_samples[
c->ch_remap[ch]] = ptr;
769 int stream_ver, header_size, frame_size_nbits, nframesegs_log2;
779 if (stream_ver > 1) {
794 frame_size_nbits =
get_bits(&
s->gb, 5) + 1;
813 s->nframesegs = 1 << nframesegs_log2;
814 if (
s->nframesegs > 1024) {
823 if (!
s->nsegsamples_log2) {
827 s->nsegsamples = 1 <<
s->nsegsamples_log2;
828 if (
s->nsegsamples > 512) {
834 s->nframesamples_log2 =
s->nsegsamples_log2 + nframesegs_log2;
835 s->nframesamples = 1 <<
s->nframesamples_log2;
836 if (
s->nframesamples > 65536) {
858 if (
s->scalable_lsbs)
861 s->fixed_lsb_width = 0;
876 return !
c->primary_chset &&
c->dmix_embedded &&
c->hier_chset;
882 while (++c < &s->chset[
s->nchsets])
891 int i, j, *coeff_ptr =
c->dmix_coeff;
893 for (
i = 0;
i <
c->hier_ofs;
i++) {
897 c->dmix_scale_inv[
i] =
mul16(
c->dmix_scale_inv[
i], scale_inv);
898 for (j = 0; j <
c->nchannels; j++) {
915 for (
i = 0,
c =
s->chset; i < s->nchsets;
i++,
c++) {
916 c->hier_ofs =
s->nchannels;
919 if (
c->nfreqbands >
s->nfreqbands)
920 s->nfreqbands =
c->nfreqbands;
922 s->nchannels +=
c->nchannels;
923 if (
c->residual_encode != (1 <<
c->nchannels) - 1)
928 for (
i =
s->nchsets - 1,
c = &
s->chset[
i];
i > 0;
i--,
c--) {
939 s->nactivechsets = 1;
943 s->nactivechsets = (
s->chset[0].nchannels < 5 &&
s->nchsets > 1) ? 2 : 1;
946 s->nactivechsets =
s->nchsets;
955 int chs, seg, band, navi_nb, navi_pos, *navi_ptr;
959 navi_nb =
s->nfreqbands *
s->nframesegs *
s->nchsets;
960 if (navi_nb > 1024) {
973 for (band = 0; band <
s->nfreqbands; band++) {
974 for (seg = 0; seg <
s->nframesegs; seg++) {
975 for (chs = 0,
c =
s->chset; chs < s->nchsets; chs++,
c++) {
977 if (
c->nfreqbands > band) {
979 if (size < 0 || size >=
s->frame_size) {
1006 int ret, chs, seg, band, navi_pos, *navi_ptr;
1009 for (chs = 0,
c =
s->chset; chs < s->nactivechsets; chs++,
c++) {
1018 for (band = 0; band <
s->nfreqbands; band++) {
1019 for (seg = 0; seg <
s->nframesegs; seg++) {
1020 for (chs = 0,
c =
s->chset; chs < s->nchsets; chs++,
c++) {
1021 if (
c->nfreqbands > band) {
1022 navi_pos += *navi_ptr * 8;
1023 if (navi_pos >
s->gb.size_in_bits) {
1027 if (chs < s->nactivechsets &&
1079 s->pbr_length =
size;
1080 s->pbr_delay = delay;
1111 if (
s->frame_size >
size)
1115 if (
s->frame_size <
size)
1131 memcpy(
s->pbr_buffer +
s->pbr_length,
data,
size);
1132 s->pbr_length +=
size;
1135 if (
s->pbr_delay > 0 && --
s->pbr_delay)
1141 if (
s->frame_size >
s->pbr_length) {
1146 if (
s->frame_size ==
s->pbr_length) {
1150 s->pbr_length -=
s->frame_size;
1151 memmove(
s->pbr_buffer,
s->pbr_buffer +
s->frame_size,
s->pbr_length);
1182 int i, j, k, nchannels = 0, *coeff_ptr = o->
dmix_coeff;
1185 for (
i = 0,
c =
s->chset; i < s->nactivechsets;
i++,
c++) {
1190 for (j = 0; j <
c->nchannels; j++) {
1192 int coeff = *coeff_ptr++;
1194 s->dcadsp->dmix_sub(
c->bands[band].msb_sample_buffer[j],
1196 coeff,
s->nframesamples);
1198 s->dcadsp->dmix_sub(
c->deci_history[j],
1205 nchannels +=
c->nchannels;
1213 int i, j, nchannels = 0;
1216 for (
i = 0,
c =
s->chset; i < s->nactivechsets;
i++,
c++) {
1221 for (j = 0; j <
c->nchannels; j++) {
1223 if (
scale != (1 << 15)) {
1224 s->dcadsp->dmix_scale(
c->bands[band].msb_sample_buffer[j],
1225 scale,
s->nframesamples);
1227 s->dcadsp->dmix_scale(
c->deci_history[j],
1244 for (band = 0; band <
c->nfreqbands; band++)
1247 for (ch = 0; ch <
c->nchannels; ch++) {
1248 if (!(
c->residual_encode & (1 << ch)))
1252 c->residual_encode &= ~(1 << ch);
1259 int ch, nsamples =
s->nframesamples;
1284 for (ch = 0; ch <
c->nchannels; ch++) {
1288 if (
c->residual_encode & (1 << ch))
1294 av_log(
s->avctx,
AV_LOG_WARNING,
"Residual encoded channel (%d) references unavailable core channel\n",
c->ch_remap[ch]);
1308 dst =
c->bands[0].msb_sample_buffer[ch];
1312 for (n = 0; n < nsamples; n++)
1316 for (n = 0; n < nsamples; n++)
1331 int i, j, k,
ret,
shift, nsamples, request_mask;
1336 for (
i = 0,
c =
s->chset; i < s->nchsets;
i++,
c++) {
1337 if (i < s->nactivechsets)
1340 if (!
c->primary_chset)
1341 c->dmix_embedded = 0;
1344 s->scalable_lsbs = 0;
1345 s->fixed_lsb_width = 0;
1350 for (
i = 0,
c =
s->chset; i < s->nactivechsets;
i++,
c++) {
1353 if (
c->residual_encode != (1 <<
c->nchannels) - 1
1357 if (
s->scalable_lsbs)
1360 if (
c->nfreqbands > 1) {
1365 s->output_mask |=
c->ch_mask;
1369 for (
i = 1,
c = &
s->chset[1]; i < s->nchsets;
i++,
c++) {
1373 if (
i >=
s->nactivechsets) {
1374 for (j = 0; j <
c->nfreqbands; j++)
1375 if (
c->bands[j].dmix_embedded)
1380 for (j = 0; j <
c->nfreqbands; j++)
1381 if (
c->bands[j].dmix_embedded)
1386 if (
s->nfreqbands > 1) {
1387 for (
i = 0;
i <
s->nactivechsets;
i++)
1411 request_mask =
s->output_mask;
1435 frame->nb_samples = nsamples =
s->nframesamples << (
s->nfreqbands - 1);
1440 if (request_mask !=
s->output_mask) {
1449 int16_t *plane = (int16_t *)
frame->extended_data[
i];
1450 for (k = 0; k < nsamples; k++)
1454 for (k = 0; k < nsamples; k++)
1486 c->sample_size[j] = 0;
static DCAXllChSet * find_next_hier_dmix_chset(DCAXllDecoder *s, DCAXllChSet *c)
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.
int ff_dca_xll_parse(DCAXllDecoder *s, const uint8_t *data, DCAExssAsset *asset)
static int get_bits_left(GetBitContext *gb)
static int parse_dmix_coeffs(DCAXllDecoder *s, DCAXllChSet *c)
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
#define DCA_XLL_DECI_HISTORY_MAX
int sample_rate
samples per second
static int32_t norm16(int64_t a)
#define DCA_SPEAKER_LAYOUT_5POINT0
int npcmsamples
Number of PCM samples per channel.
static int32_t mul15(int32_t a, int32_t b)
static unsigned int get_bits_long(GetBitContext *s, int n)
Read 0-32 bits.
static int chs_assemble_freq_bands(DCAXllDecoder *s, DCAXllChSet *c)
static void get_array(GetBitContext *gb, int32_t *array, int size, int n)
static int get_bits_count(const GetBitContext *s)
static void scale_down_mix(DCAXllDecoder *s, DCAXllChSet *o, int band)
This structure describes decoded (raw) audio or video data.
@ AV_SAMPLE_FMT_S32P
signed 32 bits, planar
#define DCA_SPEAKER_LAYOUT_STEREO
static int is_hier_dmix_chset(DCAXllChSet *c)
int request_channel_layout
Converted from avctx.request_channel_layout.
static int ff_dca_seek_bits(GetBitContext *s, int p)
#define AV_LOG_VERBOSE
Detailed information.
int xll_size
Size of XLL data in extension substream.
static void chs_assemble_msbs_lsbs(DCAXllDecoder *s, DCAXllChSet *c, int band)
static int ff_dca_check_crc(AVCodecContext *avctx, GetBitContext *s, int p1, int p2)
int nb_channels
Number of channels in this layout.
static int copy_to_pbr(DCAXllDecoder *s, const uint8_t *data, int size, int delay)
int hier_ofs
Number of preceding channels in a hierarchy (M)
#define FF_DCA_DMIXTABLE_OFFSET
int one_to_one_map_ch_to_spkr
One to one channel to speaker mapping flag.
static int parse_navi_table(DCAXllDecoder *s)
#define DCA_XLL_PBR_BUFFER_MAX
#define DCA_PACKET_RECOVERY
Sync error recovery flag.
static void skip_bits(GetBitContext *s, int n)
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
static void undo_down_mix(DCAXllDecoder *s, DCAXllChSet *o, int band)
int dmix_coeff[DCA_XLL_DMIX_COEFFS_MAX]
Downmixing coefficients.
AVChannelLayout ch_layout
Audio channel layout.
static av_always_inline float scale(float x, float s)
static int parse_common_header(DCAXllDecoder *s)
static int chs_alloc_msb_band_data(DCAXllDecoder *s, DCAXllChSet *c)
static int32_t clip23(int32_t a)
int nchannels
Number of channels in the channel set (N)
DCAExssParser exss
EXSS parser context.
#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.
int output_rate
Output sample rate (1x or 2x header rate)
void ff_dca_downmix_to_stereo_fixed(DCADSPContext *dcadsp, int32_t **samples, int *coeff_l, int nsamples, int ch_mask)
#define DCA_XLL_ADAPT_PRED_ORDER_MAX
#define DCA_SPEAKER_LAYOUT_5POINT1
static int chs_get_lsb_width(DCAXllDecoder *s, DCAXllChSet *c, int band, int ch)
const uint16_t ff_dca_xll_refl_coeff[128]
int bits_per_raw_sample
Bits per sample/pixel of internal libavcodec pixel/sample format.
int storage_bit_res
Storage bit resolution (16 or 24)
static int32_t mul16(int32_t a, int32_t b)
int dmix_type
Primary channel set downmix type.
int dmix_embedded
Downmix already performed by encoder.
static int chs_alloc_lsb_band_data(DCAXllDecoder *s, DCAXllChSet *c)
int ff_dca_set_channel_layout(AVCodecContext *avctx, int *ch_remap, int dca_mask)
@ AV_MATRIX_ENCODING_DOLBYHEADPHONE
static void clear_pbr(DCAXllDecoder *s)
#define DCA_HAS_STEREO(mask)
int32_t * output_samples[DCA_SPEAKER_COUNT]
PCM output for fixed point mode.
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
const uint8_t ff_dca_dmix_primary_nch[8]
int64_t bit_rate
the average bitrate
static unsigned int get_bits1(GetBitContext *s)
#define DCA_XLL_SAMPLE_BUFFERS_MAX
static int get_rice(GetBitContext *gb, int k)
#define AV_EF_EXPLODE
abort decoding on minor error detection
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 int get_unary(GetBitContext *gb, int stop, int len)
Get unary code of limited length.
int xll_offset
Offset to XLL data from start of substream.
const uint32_t ff_dca_inv_dmixtable[FF_DCA_INV_DMIXTABLE_SIZE]
static int combine_residual_frame(DCAXllDecoder *s, DCAXllChSet *c)
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
const uint16_t ff_dca_dmixtable[FF_DCA_DMIXTABLE_SIZE]
const int32_t ff_dca_xll_band_coeff[20]
@ AV_MATRIX_ENCODING_NONE
enum AVSampleFormat sample_fmt
audio sample format
static int chs_parse_band_data(DCAXllDecoder *s, DCAXllChSet *c, int band, int seg, int band_data_end)
int xll_delay_nframes
Initial XLL decoding delay in frames.
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
@ AV_SAMPLE_FMT_S16P
signed 16 bits, planar
static void chs_filter_band_data(DCAXllDecoder *s, DCAXllChSet *c, int band)
#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 av_always_inline av_const double round(double x)
#define DCA_XLL_CHANNELS_MAX
static int parse_frame_pbr(DCAXllDecoder *s, const uint8_t *data, int size, DCAExssAsset *asset)
int32_t * msb_sample_buffer[DCA_XLL_CHANNELS_MAX]
MSB sample buffer pointers.
#define av_assert1(cond)
assert() equivalent, that does not lie in speed critical code.
int pcm_bit_res
PCM bit resolution (variable)
int xll_sync_offset
Number of bytes offset to XLL sync.
DCAExssAsset assets[1]
Audio asset descriptors.
int32_t deci_history[DCA_XLL_CHANNELS_MAX][DCA_XLL_DECI_HISTORY_MAX]
Decimator history for frequency band 1.
int ff_dca_xll_filter_frame(DCAXllDecoder *s, AVFrame *frame)
static int ff_dca_core_map_spkr(DCACoreDecoder *core, int spkr)
int dmix_scale_inv[DCA_XLL_DMIX_SCALES_MAX]
Inverse downmixing scales.
static int array[MAX_W *MAX_W]
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
int nfreqbands
Number of frequency bands (1 or 2)
DCAXllBand bands[DCA_XLL_BANDS_MAX]
Frequency bands.
#define DCA_XLL_CHSETS_MAX
static int parse_frame_no_pbr(DCAXllDecoder *s, const uint8_t *data, int size, DCAExssAsset *asset)
#define AV_INPUT_BUFFER_PADDING_SIZE
main external API structure.
int xll_sync_present
XLL sync word present flag.
#define FF_DCA_DMIXTABLE_SIZE
int dmix_scale[DCA_XLL_DMIX_SCALES_MAX]
Downmixing scales.
const uint32_t ff_dca_sampling_freqs[16]
static int get_rice_un(GetBitContext *gb, int k)
#define FF_DCA_INV_DMIXTABLE_SIZE
static void prescale_down_mix(DCAXllChSet *c, DCAXllChSet *o)
static int chs_parse_header(DCAXllDecoder *s, DCAXllChSet *c, DCAExssAsset *asset)
static void get_linear_array(GetBitContext *gb, int32_t *array, int size, int n)
Filter the word “frame” indicates either a video frame or a group of audio samples
static int parse_sub_headers(DCAXllDecoder *s, DCAExssAsset *asset)
static int parse_frame(DCAXllDecoder *s, const uint8_t *data, int size, DCAExssAsset *asset)
int freq
Original sampling frequency (max. 96000 Hz)
static int shift(int a, int b)
static void get_rice_array(GetBitContext *gb, int32_t *array, int size, int k)
int ff_side_data_update_matrix_encoding(AVFrame *frame, enum AVMatrixEncoding matrix_encoding)
Add or update AV_FRAME_DATA_MATRIXENCODING side data.
#define avpriv_request_sample(...)
#define FF_PROFILE_DTS_HD_MA
void av_fast_malloc(void *ptr, unsigned int *size, size_t min_size)
Allocate a buffer, reusing the given one if large enough.
av_cold void ff_dca_xll_flush(DCAXllDecoder *s)
static av_cold void force_lossy_output(DCAXllDecoder *s, DCAXllChSet *c)
DCACoreDecoder core
Core decoder context.
static const double coeff[2][5]
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
static int get_sbits_long(GetBitContext *s, int n)
Read 0-32 bits as a signed integer.
static av_cold void chs_clear_band_data(DCAXllDecoder *s, DCAXllChSet *c, int band, int seg)
static int get_linear(GetBitContext *gb, int n)
int representation_type
Representation type.
int hd_stream_id
DTS-HD stream ID.
static int parse_band_data(DCAXllDecoder *s)
av_cold void ff_dca_xll_close(DCAXllDecoder *s)