2 * DCA compatible decoder
3 * Copyright (C) 2004 Gildas Bazin
4 * Copyright (C) 2004 Benjamin Zores
5 * Copyright (C) 2006 Benjamin Larsson
6 * Copyright (C) 2007 Konstantin Shishkov
7 * Copyright (C) 2012 Paul B Mahol
8 * Copyright (C) 2014 Niels Möller
10 * This file is part of FFmpeg.
12 * FFmpeg is free software; you can redistribute it and/or
13 * modify it under the terms of the GNU Lesser General Public
14 * License as published by the Free Software Foundation; either
15 * version 2.1 of the License, or (at your option) any later version.
17 * FFmpeg is distributed in the hope that it will be useful,
18 * but WITHOUT ANY WARRANTY; without even the implied warranty of
19 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
20 * Lesser General Public License for more details.
22 * You should have received a copy of the GNU Lesser General Public
23 * License along with FFmpeg; if not, write to the Free Software
24 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
31 #include "libavutil/attributes.h"
32 #include "libavutil/channel_layout.h"
33 #include "libavutil/common.h"
34 #include "libavutil/float_dsp.h"
35 #include "libavutil/internal.h"
36 #include "libavutil/intreadwrite.h"
37 #include "libavutil/mathematics.h"
38 #include "libavutil/opt.h"
39 #include "libavutil/samplefmt.h"
43 #include "dca_syncwords.h"
48 #include "fmtconvert.h"
53 #include "synth_filter.h"
74 enum DCAXxchSpeakerMask {
75 DCA_XXCH_FRONT_CENTER = 0x0000001,
76 DCA_XXCH_FRONT_LEFT = 0x0000002,
77 DCA_XXCH_FRONT_RIGHT = 0x0000004,
78 DCA_XXCH_SIDE_REAR_LEFT = 0x0000008,
79 DCA_XXCH_SIDE_REAR_RIGHT = 0x0000010,
80 DCA_XXCH_LFE1 = 0x0000020,
81 DCA_XXCH_REAR_CENTER = 0x0000040,
82 DCA_XXCH_SURROUND_REAR_LEFT = 0x0000080,
83 DCA_XXCH_SURROUND_REAR_RIGHT = 0x0000100,
84 DCA_XXCH_SIDE_SURROUND_LEFT = 0x0000200,
85 DCA_XXCH_SIDE_SURROUND_RIGHT = 0x0000400,
86 DCA_XXCH_FRONT_CENTER_LEFT = 0x0000800,
87 DCA_XXCH_FRONT_CENTER_RIGHT = 0x0001000,
88 DCA_XXCH_FRONT_HIGH_LEFT = 0x0002000,
89 DCA_XXCH_FRONT_HIGH_CENTER = 0x0004000,
90 DCA_XXCH_FRONT_HIGH_RIGHT = 0x0008000,
91 DCA_XXCH_LFE2 = 0x0010000,
92 DCA_XXCH_SIDE_FRONT_LEFT = 0x0020000,
93 DCA_XXCH_SIDE_FRONT_RIGHT = 0x0040000,
94 DCA_XXCH_OVERHEAD = 0x0080000,
95 DCA_XXCH_SIDE_HIGH_LEFT = 0x0100000,
96 DCA_XXCH_SIDE_HIGH_RIGHT = 0x0200000,
97 DCA_XXCH_REAR_HIGH_CENTER = 0x0400000,
98 DCA_XXCH_REAR_HIGH_LEFT = 0x0800000,
99 DCA_XXCH_REAR_HIGH_RIGHT = 0x1000000,
100 DCA_XXCH_REAR_LOW_CENTER = 0x2000000,
101 DCA_XXCH_REAR_LOW_LEFT = 0x4000000,
102 DCA_XXCH_REAR_LOW_RIGHT = 0x8000000,
105 #define DCA_DOLBY 101 /* FIXME */
107 #define DCA_CHANNEL_BITS 6
108 #define DCA_CHANNEL_MASK 0x3F
112 #define HEADER_SIZE 14
114 #define DCA_NSYNCAUX 0x9A1105A0
116 /** Bit allocation */
117 typedef struct BitAlloc {
118 int offset; ///< code values offset
119 int maxbits[8]; ///< max bits in VLC
120 int wrap; ///< wrap for get_vlc2()
121 VLC vlc[8]; ///< actual codes
124 static BitAlloc dca_bitalloc_index; ///< indexes for samples VLC select
125 static BitAlloc dca_tmode; ///< transition mode VLCs
126 static BitAlloc dca_scalefactor; ///< scalefactor VLCs
127 static BitAlloc dca_smpl_bitalloc[11]; ///< samples VLCs
129 static av_always_inline int get_bitalloc(GetBitContext *gb, BitAlloc *ba,
132 return get_vlc2(gb, ba->vlc[idx].table, ba->vlc[idx].bits, ba->wrap) +
136 static float dca_dmix_code(unsigned code);
138 static av_cold void dca_init_vlcs(void)
140 static int vlcs_initialized = 0;
142 static VLC_TYPE dca_table[23622][2];
144 if (vlcs_initialized)
147 dca_bitalloc_index.offset = 1;
148 dca_bitalloc_index.wrap = 2;
149 for (i = 0; i < 5; i++) {
150 dca_bitalloc_index.vlc[i].table = &dca_table[ff_dca_vlc_offs[i]];
151 dca_bitalloc_index.vlc[i].table_allocated = ff_dca_vlc_offs[i + 1] - ff_dca_vlc_offs[i];
152 init_vlc(&dca_bitalloc_index.vlc[i], bitalloc_12_vlc_bits[i], 12,
153 bitalloc_12_bits[i], 1, 1,
154 bitalloc_12_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
156 dca_scalefactor.offset = -64;
157 dca_scalefactor.wrap = 2;
158 for (i = 0; i < 5; i++) {
159 dca_scalefactor.vlc[i].table = &dca_table[ff_dca_vlc_offs[i + 5]];
160 dca_scalefactor.vlc[i].table_allocated = ff_dca_vlc_offs[i + 6] - ff_dca_vlc_offs[i + 5];
161 init_vlc(&dca_scalefactor.vlc[i], SCALES_VLC_BITS, 129,
162 scales_bits[i], 1, 1,
163 scales_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
165 dca_tmode.offset = 0;
167 for (i = 0; i < 4; i++) {
168 dca_tmode.vlc[i].table = &dca_table[ff_dca_vlc_offs[i + 10]];
169 dca_tmode.vlc[i].table_allocated = ff_dca_vlc_offs[i + 11] - ff_dca_vlc_offs[i + 10];
170 init_vlc(&dca_tmode.vlc[i], tmode_vlc_bits[i], 4,
172 tmode_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
175 for (i = 0; i < 10; i++)
176 for (j = 0; j < 7; j++) {
177 if (!bitalloc_codes[i][j])
179 dca_smpl_bitalloc[i + 1].offset = bitalloc_offsets[i];
180 dca_smpl_bitalloc[i + 1].wrap = 1 + (j > 4);
181 dca_smpl_bitalloc[i + 1].vlc[j].table = &dca_table[ff_dca_vlc_offs[c]];
182 dca_smpl_bitalloc[i + 1].vlc[j].table_allocated = ff_dca_vlc_offs[c + 1] - ff_dca_vlc_offs[c];
184 init_vlc(&dca_smpl_bitalloc[i + 1].vlc[j], bitalloc_maxbits[i][j],
186 bitalloc_bits[i][j], 1, 1,
187 bitalloc_codes[i][j], 2, 2, INIT_VLC_USE_NEW_STATIC);
190 vlcs_initialized = 1;
193 static inline void get_array(GetBitContext *gb, int *dst, int len, int bits)
196 *dst++ = get_bits(gb, bits);
199 static inline int dca_xxch2index(DCAContext *s, int xxch_ch)
203 /* locate channel set containing the channel */
204 for (i = -1, base = 0, mask = (s->xxch_core_spkmask & ~DCA_XXCH_LFE1);
205 i <= s->xxch_chset && !(mask & xxch_ch); mask = s->xxch_spk_masks[++i])
206 base += av_popcount(mask);
208 return base + av_popcount(mask & (xxch_ch - 1));
211 static int dca_parse_audio_coding_header(DCAContext *s, int base_channel,
215 static const uint8_t adj_table[4] = { 16, 18, 20, 23 };
216 static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };
217 static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };
218 int hdr_pos = 0, hdr_size = 0;
220 int this_chans, acc_mask;
221 int embedded_downmix;
225 /* xxch has arbitrary sized audio coding headers */
227 hdr_pos = get_bits_count(&s->gb);
228 hdr_size = get_bits(&s->gb, 7) + 1;
231 nchans = get_bits(&s->gb, 3) + 1;
232 if (xxch && nchans >= 3) {
233 av_log(s->avctx, AV_LOG_ERROR, "nchans %d is too large\n", nchans);
234 return AVERROR_INVALIDDATA;
235 } else if (nchans + base_channel > DCA_PRIM_CHANNELS_MAX) {
236 av_log(s->avctx, AV_LOG_ERROR, "channel sum %d + %d is too large\n", nchans, base_channel);
237 return AVERROR_INVALIDDATA;
240 s->audio_header.total_channels = nchans + base_channel;
241 s->audio_header.prim_channels = s->audio_header.total_channels;
243 /* obtain speaker layout mask & downmix coefficients for XXCH */
245 acc_mask = s->xxch_core_spkmask;
247 this_chans = get_bits(&s->gb, s->xxch_nbits_spk_mask - 6) << 6;
248 s->xxch_spk_masks[s->xxch_chset] = this_chans;
249 s->xxch_chset_nch[s->xxch_chset] = nchans;
251 for (i = 0; i <= s->xxch_chset; i++)
252 acc_mask |= s->xxch_spk_masks[i];
254 /* check for downmixing information */
255 if (get_bits1(&s->gb)) {
256 embedded_downmix = get_bits1(&s->gb);
257 coeff = get_bits(&s->gb, 6);
259 if (coeff<1 || coeff>61) {
260 av_log(s->avctx, AV_LOG_ERROR, "6bit coeff %d is out of range\n", coeff);
261 return AVERROR_INVALIDDATA;
264 scale_factor = -1.0f / dca_dmix_code((coeff<<2)-3);
266 s->xxch_dmix_sf[s->xxch_chset] = scale_factor;
268 for (i = base_channel; i < s->audio_header.prim_channels; i++) {
269 mask[i] = get_bits(&s->gb, s->xxch_nbits_spk_mask);
272 for (j = base_channel; j < s->audio_header.prim_channels; j++) {
273 memset(s->xxch_dmix_coeff[j], 0, sizeof(s->xxch_dmix_coeff[0]));
274 s->xxch_dmix_embedded |= (embedded_downmix << j);
275 for (i = 0; i < s->xxch_nbits_spk_mask; i++) {
276 if (mask[j] & (1 << i)) {
277 if ((1 << i) == DCA_XXCH_LFE1) {
278 av_log(s->avctx, AV_LOG_WARNING,
279 "DCA-XXCH: dmix to LFE1 not supported.\n");
283 coeff = get_bits(&s->gb, 7);
284 ichan = dca_xxch2index(s, 1 << i);
285 if ((coeff&63)<1 || (coeff&63)>61) {
286 av_log(s->avctx, AV_LOG_ERROR, "7bit coeff %d is out of range\n", coeff);
287 return AVERROR_INVALIDDATA;
289 s->xxch_dmix_coeff[j][ichan] = dca_dmix_code((coeff<<2)-3);
296 if (s->audio_header.prim_channels > DCA_PRIM_CHANNELS_MAX)
297 s->audio_header.prim_channels = DCA_PRIM_CHANNELS_MAX;
299 for (i = base_channel; i < s->audio_header.prim_channels; i++) {
300 s->audio_header.subband_activity[i] = get_bits(&s->gb, 5) + 2;
301 if (s->audio_header.subband_activity[i] > DCA_SUBBANDS)
302 s->audio_header.subband_activity[i] = DCA_SUBBANDS;
304 for (i = base_channel; i < s->audio_header.prim_channels; i++) {
305 s->audio_header.vq_start_subband[i] = get_bits(&s->gb, 5) + 1;
306 if (s->audio_header.vq_start_subband[i] > DCA_SUBBANDS)
307 s->audio_header.vq_start_subband[i] = DCA_SUBBANDS;
309 get_array(&s->gb, s->audio_header.joint_intensity + base_channel,
310 s->audio_header.prim_channels - base_channel, 3);
311 get_array(&s->gb, s->audio_header.transient_huffman + base_channel,
312 s->audio_header.prim_channels - base_channel, 2);
313 get_array(&s->gb, s->audio_header.scalefactor_huffman + base_channel,
314 s->audio_header.prim_channels - base_channel, 3);
315 get_array(&s->gb, s->audio_header.bitalloc_huffman + base_channel,
316 s->audio_header.prim_channels - base_channel, 3);
318 /* Get codebooks quantization indexes */
320 memset(s->audio_header.quant_index_huffman, 0, sizeof(s->audio_header.quant_index_huffman));
321 for (j = 1; j < 11; j++)
322 for (i = base_channel; i < s->audio_header.prim_channels; i++)
323 s->audio_header.quant_index_huffman[i][j] = get_bits(&s->gb, bitlen[j]);
325 /* Get scale factor adjustment */
326 for (j = 0; j < 11; j++)
327 for (i = base_channel; i < s->audio_header.prim_channels; i++)
328 s->audio_header.scalefactor_adj[i][j] = 16;
330 for (j = 1; j < 11; j++)
331 for (i = base_channel; i < s->audio_header.prim_channels; i++)
332 if (s->audio_header.quant_index_huffman[i][j] < thr[j])
333 s->audio_header.scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)];
336 if (s->crc_present) {
337 /* Audio header CRC check */
338 get_bits(&s->gb, 16);
341 /* Skip to the end of the header, also ignore CRC if present */
342 i = get_bits_count(&s->gb);
343 if (hdr_pos + 8 * hdr_size > i)
344 skip_bits_long(&s->gb, hdr_pos + 8 * hdr_size - i);
347 s->current_subframe = 0;
348 s->current_subsubframe = 0;
353 static int dca_parse_frame_header(DCAContext *s)
355 init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
358 skip_bits_long(&s->gb, 32);
361 s->frame_type = get_bits(&s->gb, 1);
362 s->samples_deficit = get_bits(&s->gb, 5) + 1;
363 s->crc_present = get_bits(&s->gb, 1);
364 s->sample_blocks = get_bits(&s->gb, 7) + 1;
365 s->frame_size = get_bits(&s->gb, 14) + 1;
366 if (s->frame_size < 95)
367 return AVERROR_INVALIDDATA;
368 s->amode = get_bits(&s->gb, 6);
369 s->sample_rate = avpriv_dca_sample_rates[get_bits(&s->gb, 4)];
371 return AVERROR_INVALIDDATA;
372 s->bit_rate_index = get_bits(&s->gb, 5);
373 s->bit_rate = ff_dca_bit_rates[s->bit_rate_index];
375 return AVERROR_INVALIDDATA;
377 skip_bits1(&s->gb); // always 0 (reserved, cf. ETSI TS 102 114 V1.4.1)
378 s->dynrange = get_bits(&s->gb, 1);
379 s->timestamp = get_bits(&s->gb, 1);
380 s->aux_data = get_bits(&s->gb, 1);
381 s->hdcd = get_bits(&s->gb, 1);
382 s->ext_descr = get_bits(&s->gb, 3);
383 s->ext_coding = get_bits(&s->gb, 1);
384 s->aspf = get_bits(&s->gb, 1);
385 s->lfe = get_bits(&s->gb, 2);
386 s->predictor_history = get_bits(&s->gb, 1);
390 av_log(s->avctx, AV_LOG_ERROR, "Invalid LFE value: %d\n", s->lfe);
391 return AVERROR_INVALIDDATA;
394 /* TODO: check CRC */
396 s->header_crc = get_bits(&s->gb, 16);
398 s->multirate_inter = get_bits(&s->gb, 1);
399 s->version = get_bits(&s->gb, 4);
400 s->copy_history = get_bits(&s->gb, 2);
401 s->source_pcm_res = get_bits(&s->gb, 3);
402 s->front_sum = get_bits(&s->gb, 1);
403 s->surround_sum = get_bits(&s->gb, 1);
404 s->dialog_norm = get_bits(&s->gb, 4);
406 /* FIXME: channels mixing levels */
407 s->output = s->amode;
409 s->output |= DCA_LFE;
411 /* Primary audio coding header */
412 s->audio_header.subframes = get_bits(&s->gb, 4) + 1;
414 return dca_parse_audio_coding_header(s, 0, 0);
417 static inline int get_scale(GetBitContext *gb, int level, int value, int log2range)
420 /* huffman encoded */
421 value += get_bitalloc(gb, &dca_scalefactor, level);
422 value = av_clip(value, 0, (1 << log2range) - 1);
423 } else if (level < 8) {
424 if (level + 1 > log2range) {
425 skip_bits(gb, level + 1 - log2range);
426 value = get_bits(gb, log2range);
428 value = get_bits(gb, level + 1);
434 static int dca_subframe_header(DCAContext *s, int base_channel, int block_index)
436 /* Primary audio coding side information */
439 if (get_bits_left(&s->gb) < 0)
440 return AVERROR_INVALIDDATA;
443 s->subsubframes[s->current_subframe] = get_bits(&s->gb, 2) + 1;
444 if (block_index + s->subsubframes[s->current_subframe] > (s->sample_blocks / SAMPLES_PER_SUBBAND)) {
445 s->subsubframes[s->current_subframe] = 1;
446 return AVERROR_INVALIDDATA;
448 s->partial_samples[s->current_subframe] = get_bits(&s->gb, 3);
451 for (j = base_channel; j < s->audio_header.prim_channels; j++) {
452 for (k = 0; k < s->audio_header.subband_activity[j]; k++)
453 s->dca_chan[j].prediction_mode[k] = get_bits(&s->gb, 1);
456 /* Get prediction codebook */
457 for (j = base_channel; j < s->audio_header.prim_channels; j++) {
458 for (k = 0; k < s->audio_header.subband_activity[j]; k++) {
459 if (s->dca_chan[j].prediction_mode[k] > 0) {
460 /* (Prediction coefficient VQ address) */
461 s->dca_chan[j].prediction_vq[k] = get_bits(&s->gb, 12);
466 /* Bit allocation index */
467 for (j = base_channel; j < s->audio_header.prim_channels; j++) {
468 for (k = 0; k < s->audio_header.vq_start_subband[j]; k++) {
469 if (s->audio_header.bitalloc_huffman[j] == 6)
470 s->dca_chan[j].bitalloc[k] = get_bits(&s->gb, 5);
471 else if (s->audio_header.bitalloc_huffman[j] == 5)
472 s->dca_chan[j].bitalloc[k] = get_bits(&s->gb, 4);
473 else if (s->audio_header.bitalloc_huffman[j] == 7) {
474 av_log(s->avctx, AV_LOG_ERROR,
475 "Invalid bit allocation index\n");
476 return AVERROR_INVALIDDATA;
478 s->dca_chan[j].bitalloc[k] =
479 get_bitalloc(&s->gb, &dca_bitalloc_index, s->audio_header.bitalloc_huffman[j]);
482 if (s->dca_chan[j].bitalloc[k] > 26) {
483 ff_dlog(s->avctx, "bitalloc index [%i][%i] too big (%i)\n",
484 j, k, s->dca_chan[j].bitalloc[k]);
485 return AVERROR_INVALIDDATA;
490 /* Transition mode */
491 for (j = base_channel; j < s->audio_header.prim_channels; j++) {
492 for (k = 0; k < s->audio_header.subband_activity[j]; k++) {
493 s->dca_chan[j].transition_mode[k] = 0;
494 if (s->subsubframes[s->current_subframe] > 1 &&
495 k < s->audio_header.vq_start_subband[j] && s->dca_chan[j].bitalloc[k] > 0) {
496 s->dca_chan[j].transition_mode[k] =
497 get_bitalloc(&s->gb, &dca_tmode, s->audio_header.transient_huffman[j]);
502 if (get_bits_left(&s->gb) < 0)
503 return AVERROR_INVALIDDATA;
505 for (j = base_channel; j < s->audio_header.prim_channels; j++) {
506 const uint32_t *scale_table;
507 int scale_sum, log_size;
509 memset(s->dca_chan[j].scale_factor, 0,
510 s->audio_header.subband_activity[j] * sizeof(s->dca_chan[j].scale_factor[0][0]) * 2);
512 if (s->audio_header.scalefactor_huffman[j] == 6) {
513 scale_table = ff_dca_scale_factor_quant7;
516 scale_table = ff_dca_scale_factor_quant6;
520 /* When huffman coded, only the difference is encoded */
523 for (k = 0; k < s->audio_header.subband_activity[j]; k++) {
524 if (k >= s->audio_header.vq_start_subband[j] || s->dca_chan[j].bitalloc[k] > 0) {
525 scale_sum = get_scale(&s->gb, s->audio_header.scalefactor_huffman[j], scale_sum, log_size);
526 s->dca_chan[j].scale_factor[k][0] = scale_table[scale_sum];
529 if (k < s->audio_header.vq_start_subband[j] && s->dca_chan[j].transition_mode[k]) {
530 /* Get second scale factor */
531 scale_sum = get_scale(&s->gb, s->audio_header.scalefactor_huffman[j], scale_sum, log_size);
532 s->dca_chan[j].scale_factor[k][1] = scale_table[scale_sum];
537 /* Joint subband scale factor codebook select */
538 for (j = base_channel; j < s->audio_header.prim_channels; j++) {
539 /* Transmitted only if joint subband coding enabled */
540 if (s->audio_header.joint_intensity[j] > 0)
541 s->dca_chan[j].joint_huff = get_bits(&s->gb, 3);
544 if (get_bits_left(&s->gb) < 0)
545 return AVERROR_INVALIDDATA;
547 /* Scale factors for joint subband coding */
548 for (j = base_channel; j < s->audio_header.prim_channels; j++) {
551 /* Transmitted only if joint subband coding enabled */
552 if (s->audio_header.joint_intensity[j] > 0) {
554 source_channel = s->audio_header.joint_intensity[j] - 1;
556 /* When huffman coded, only the difference is encoded
557 * (is this valid as well for joint scales ???) */
559 for (k = s->audio_header.subband_activity[j];
560 k < s->audio_header.subband_activity[source_channel]; k++) {
561 scale = get_scale(&s->gb, s->dca_chan[j].joint_huff, 64 /* bias */, 7);
562 s->dca_chan[j].joint_scale_factor[k] = scale; /*joint_scale_table[scale]; */
565 if (!(s->debug_flag & 0x02)) {
566 av_log(s->avctx, AV_LOG_DEBUG,
567 "Joint stereo coding not supported\n");
568 s->debug_flag |= 0x02;
573 /* Dynamic range coefficient */
574 if (!base_channel && s->dynrange)
575 s->dynrange_coef = get_bits(&s->gb, 8);
577 /* Side information CRC check word */
578 if (s->crc_present) {
579 get_bits(&s->gb, 16);
583 * Primary audio data arrays
586 /* VQ encoded high frequency subbands */
587 for (j = base_channel; j < s->audio_header.prim_channels; j++)
588 for (k = s->audio_header.vq_start_subband[j]; k < s->audio_header.subband_activity[j]; k++)
589 /* 1 vector -> 32 samples */
590 s->dca_chan[j].high_freq_vq[k] = get_bits(&s->gb, 10);
592 /* Low frequency effect data */
593 if (!base_channel && s->lfe) {
596 int lfe_samples = 2 * s->lfe * (4 + block_index);
597 int lfe_end_sample = 2 * s->lfe * (4 + block_index + s->subsubframes[s->current_subframe]);
600 for (j = lfe_samples; j < lfe_end_sample; j++) {
601 /* Signed 8 bits int */
602 s->lfe_data[j] = get_sbits(&s->gb, 8);
605 /* Scale factor index */
606 quant7 = get_bits(&s->gb, 8);
608 avpriv_request_sample(s->avctx, "LFEScaleIndex larger than 127");
609 return AVERROR_INVALIDDATA;
611 s->lfe_scale_factor = ff_dca_scale_factor_quant7[quant7];
613 /* Quantization step size * scale factor */
614 lfe_scale = 0.035 * s->lfe_scale_factor;
616 for (j = lfe_samples; j < lfe_end_sample; j++)
617 s->lfe_data[j] *= lfe_scale;
623 static void qmf_32_subbands(DCAContext *s, int chans,
624 float samples_in[DCA_SUBBANDS][SAMPLES_PER_SUBBAND], float *samples_out,
627 const float *prCoeff;
629 int sb_act = s->audio_header.subband_activity[chans];
631 scale *= sqrt(1 / 8.0);
634 if (!s->multirate_inter) /* Non-perfect reconstruction */
635 prCoeff = ff_dca_fir_32bands_nonperfect;
636 else /* Perfect reconstruction */
637 prCoeff = ff_dca_fir_32bands_perfect;
639 s->dcadsp.qmf_32_subbands(samples_in, sb_act, &s->synth, &s->imdct,
640 s->dca_chan[chans].subband_fir_hist,
641 &s->dca_chan[chans].hist_index,
642 s->dca_chan[chans].subband_fir_noidea, prCoeff,
643 samples_out, s->raXin, scale);
646 static QMF64_table *qmf64_precompute(void)
649 QMF64_table *table = av_malloc(sizeof(*table));
653 for (i = 0; i < 32; i++)
654 for (j = 0; j < 32; j++)
655 table->dct4_coeff[i][j] = cos((2 * i + 1) * (2 * j + 1) * M_PI / 128);
656 for (i = 0; i < 32; i++)
657 for (j = 0; j < 32; j++)
658 table->dct2_coeff[i][j] = cos((2 * i + 1) * j * M_PI / 64);
660 /* FIXME: Is the factor 0.125 = 1/8 right? */
661 for (i = 0; i < 32; i++)
662 table->rcos[i] = 0.125 / cos((2 * i + 1) * M_PI / 256);
663 for (i = 0; i < 32; i++)
664 table->rsin[i] = -0.125 / sin((2 * i + 1) * M_PI / 256);
669 /* FIXME: Totally unoptimized. Based on the reference code and
670 * http://multimedia.cx/mirror/dca-transform.pdf, with guessed tweaks
671 * for doubling the size. */
672 static void qmf_64_subbands(DCAContext *s, int chans,
673 float samples_in[DCA_SUBBANDS_X96K][SAMPLES_PER_SUBBAND],
674 float *samples_out, float scale)
678 float *raX = s->dca_chan[chans].subband_fir_hist;
679 float *raZ = s->dca_chan[chans].subband_fir_noidea;
680 unsigned i, j, k, subindex;
682 for (i = s->audio_header.subband_activity[chans]; i < DCA_SUBBANDS_X96K; i++)
684 for (subindex = 0; subindex < SAMPLES_PER_SUBBAND; subindex++) {
685 for (i = 0; i < s->audio_header.subband_activity[chans]; i++)
686 raXin[i] = samples_in[i][subindex];
688 for (k = 0; k < 32; k++) {
690 for (i = 0; i < 32; i++)
691 A[k] += (raXin[2 * i] + raXin[2 * i + 1]) * s->qmf64_table->dct4_coeff[k][i];
693 for (k = 0; k < 32; k++) {
694 B[k] = raXin[0] * s->qmf64_table->dct2_coeff[k][0];
695 for (i = 1; i < 32; i++)
696 B[k] += (raXin[2 * i] + raXin[2 * i - 1]) * s->qmf64_table->dct2_coeff[k][i];
698 for (k = 0; k < 32; k++) {
699 raX[k] = s->qmf64_table->rcos[k] * (A[k] + B[k]);
700 raX[63 - k] = s->qmf64_table->rsin[k] * (A[k] - B[k]);
703 for (i = 0; i < DCA_SUBBANDS_X96K; i++) {
705 for (j = 0; j < 1024; j += 128)
706 out += ff_dca_fir_64bands[j + i] * (raX[j + i] - raX[j + 63 - i]);
707 *samples_out++ = out * scale;
710 for (i = 0; i < DCA_SUBBANDS_X96K; i++) {
712 for (j = 0; j < 1024; j += 128)
713 hist += ff_dca_fir_64bands[64 + j + i] * (-raX[i + j] - raX[j + 63 - i]);
718 /* FIXME: Make buffer circular, to avoid this move. */
719 memmove(raX + 64, raX, (1024 - 64) * sizeof(*raX));
723 static void lfe_interpolation_fir(DCAContext *s, const float *samples_in,
726 /* samples_in: An array holding decimated samples.
727 * Samples in current subframe starts from samples_in[0],
728 * while samples_in[-1], samples_in[-2], ..., stores samples
729 * from last subframe as history.
731 * samples_out: An array holding interpolated samples
735 const float *prCoeff;
738 /* Select decimation filter */
741 prCoeff = ff_dca_lfe_fir_128;
744 if (s->exss_ext_mask & DCA_EXT_EXSS_XLL)
745 prCoeff = ff_dca_lfe_xll_fir_64;
747 prCoeff = ff_dca_lfe_fir_64;
750 for (deciindex = 0; deciindex < 2 * s->lfe; deciindex++) {
751 s->dcadsp.lfe_fir[idx](samples_out, samples_in, prCoeff);
753 samples_out += 2 * 32 * (1 + idx);
757 /* downmixing routines */
758 #define MIX_REAR1(samples, s1, rs, coef) \
759 samples[0][i] += samples[s1][i] * coef[rs][0]; \
760 samples[1][i] += samples[s1][i] * coef[rs][1];
762 #define MIX_REAR2(samples, s1, s2, rs, coef) \
763 samples[0][i] += samples[s1][i] * coef[rs][0] + samples[s2][i] * coef[rs + 1][0]; \
764 samples[1][i] += samples[s1][i] * coef[rs][1] + samples[s2][i] * coef[rs + 1][1];
766 #define MIX_FRONT3(samples, coef) \
770 samples[0][i] = t * coef[0][0] + u * coef[1][0] + v * coef[2][0]; \
771 samples[1][i] = t * coef[0][1] + u * coef[1][1] + v * coef[2][1];
773 #define DOWNMIX_TO_STEREO(op1, op2) \
774 for (i = 0; i < 256; i++) { \
779 static void dca_downmix(float **samples, int srcfmt, int lfe_present,
780 float coef[DCA_PRIM_CHANNELS_MAX + 1][2],
781 const int8_t *channel_mapping)
783 int c, l, r, sl, sr, s;
790 av_log(NULL, AV_LOG_ERROR, "Not implemented!\n");
794 case DCA_STEREO_TOTAL:
795 case DCA_STEREO_SUMDIFF:
798 c = channel_mapping[0];
799 l = channel_mapping[1];
800 r = channel_mapping[2];
801 DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), );
804 s = channel_mapping[2];
805 DOWNMIX_TO_STEREO(MIX_REAR1(samples, s, 2, coef), );
808 c = channel_mapping[0];
809 l = channel_mapping[1];
810 r = channel_mapping[2];
811 s = channel_mapping[3];
812 DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
813 MIX_REAR1(samples, s, 3, coef));
816 sl = channel_mapping[2];
817 sr = channel_mapping[3];
818 DOWNMIX_TO_STEREO(MIX_REAR2(samples, sl, sr, 2, coef), );
821 c = channel_mapping[0];
822 l = channel_mapping[1];
823 r = channel_mapping[2];
824 sl = channel_mapping[3];
825 sr = channel_mapping[4];
826 DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
827 MIX_REAR2(samples, sl, sr, 3, coef));
831 int lf_buf = ff_dca_lfe_index[srcfmt];
832 int lf_idx = ff_dca_channels[srcfmt];
833 for (i = 0; i < 256; i++) {
834 samples[0][i] += samples[lf_buf][i] * coef[lf_idx][0];
835 samples[1][i] += samples[lf_buf][i] * coef[lf_idx][1];
840 #ifndef decode_blockcodes
841 /* Very compact version of the block code decoder that does not use table
842 * look-up but is slightly slower */
843 static int decode_blockcode(int code, int levels, int32_t *values)
846 int offset = (levels - 1) >> 1;
848 for (i = 0; i < 4; i++) {
849 int div = FASTDIV(code, levels);
850 values[i] = code - offset - div * levels;
857 static int decode_blockcodes(int code1, int code2, int levels, int32_t *values)
859 return decode_blockcode(code1, levels, values) |
860 decode_blockcode(code2, levels, values + 4);
864 static const uint8_t abits_sizes[7] = { 7, 10, 12, 13, 15, 17, 19 };
865 static const uint8_t abits_levels[7] = { 3, 5, 7, 9, 13, 17, 25 };
867 static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
870 int subsubframe = s->current_subsubframe;
871 const uint32_t *quant_step_table;
877 /* Select quantization step size table */
878 if (s->bit_rate_index == 0x1f)
879 quant_step_table = ff_dca_lossless_quant;
881 quant_step_table = ff_dca_lossy_quant;
883 for (k = base_channel; k < s->audio_header.prim_channels; k++) {
884 int32_t (*subband_samples)[8] = s->dca_chan[k].subband_samples[block_index];
886 if (get_bits_left(&s->gb) < 0)
887 return AVERROR_INVALIDDATA;
889 for (l = 0; l < s->audio_header.vq_start_subband[k]; l++) {
892 /* Select the mid-tread linear quantizer */
893 int abits = s->dca_chan[k].bitalloc[l];
895 uint32_t quant_step_size = quant_step_table[abits];
898 * Extract bits from the bit stream
901 memset(subband_samples[l], 0, SAMPLES_PER_SUBBAND *
902 sizeof(subband_samples[l][0]));
905 /* Deal with transients */
906 int sfi = s->dca_chan[k].transition_mode[l] &&
907 subsubframe >= s->dca_chan[k].transition_mode[l];
908 /* Determine quantization index code book and its type.
909 Select quantization index code book */
910 int sel = s->audio_header.quant_index_huffman[k][abits];
912 rscale = (s->dca_chan[k].scale_factor[l][sfi] *
913 s->audio_header.scalefactor_adj[k][sel] + 8) >> 4;
915 if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table) {
918 int block_code1, block_code2, size, levels, err;
920 size = abits_sizes[abits - 1];
921 levels = abits_levels[abits - 1];
923 block_code1 = get_bits(&s->gb, size);
924 block_code2 = get_bits(&s->gb, size);
925 err = decode_blockcodes(block_code1, block_code2,
926 levels, subband_samples[l]);
928 av_log(s->avctx, AV_LOG_ERROR,
929 "ERROR: block code look-up failed\n");
930 return AVERROR_INVALIDDATA;
934 for (m = 0; m < SAMPLES_PER_SUBBAND; m++)
935 subband_samples[l][m] = get_sbits(&s->gb, abits - 3);
939 for (m = 0; m < SAMPLES_PER_SUBBAND; m++)
940 subband_samples[l][m] = get_bitalloc(&s->gb,
941 &dca_smpl_bitalloc[abits], sel);
943 s->dcadsp.dequantize(subband_samples[l], quant_step_size, rscale);
947 for (l = 0; l < s->audio_header.vq_start_subband[k]; l++) {
950 * Inverse ADPCM if in prediction mode
952 if (s->dca_chan[k].prediction_mode[l]) {
954 if (s->predictor_history)
955 subband_samples[l][0] += (ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][0] *
956 (int64_t)s->dca_chan[k].subband_samples_hist[l][3] +
957 ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][1] *
958 (int64_t)s->dca_chan[k].subband_samples_hist[l][2] +
959 ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][2] *
960 (int64_t)s->dca_chan[k].subband_samples_hist[l][1] +
961 ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][3] *
962 (int64_t)s->dca_chan[k].subband_samples_hist[l][0]) +
964 for (m = 1; m < SAMPLES_PER_SUBBAND; m++) {
965 int64_t sum = ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][0] *
966 (int64_t)subband_samples[l][m - 1];
967 for (n = 2; n <= 4; n++)
969 sum += ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][n - 1] *
970 (int64_t)subband_samples[l][m - n];
971 else if (s->predictor_history)
972 sum += ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][n - 1] *
973 (int64_t)s->dca_chan[k].subband_samples_hist[l][m - n + 4];
974 subband_samples[l][m] += (int32_t)(sum + (1 << 12) >> 13);
979 /* Backup predictor history for adpcm */
980 for (l = 0; l < DCA_SUBBANDS; l++)
981 AV_COPY128(s->dca_chan[k].subband_samples_hist[l], &subband_samples[l][4]);
985 * Decode VQ encoded high frequencies
987 if (s->audio_header.subband_activity[k] > s->audio_header.vq_start_subband[k]) {
988 if (!(s->debug_flag & 0x01)) {
989 av_log(s->avctx, AV_LOG_DEBUG,
990 "Stream with high frequencies VQ coding\n");
991 s->debug_flag |= 0x01;
994 s->dcadsp.decode_hf(subband_samples, s->dca_chan[k].high_freq_vq,
996 subsubframe * SAMPLES_PER_SUBBAND,
997 s->dca_chan[k].scale_factor,
998 s->audio_header.vq_start_subband[k],
999 s->audio_header.subband_activity[k]);
1003 /* Check for DSYNC after subsubframe */
1004 if (s->aspf || subsubframe == s->subsubframes[s->current_subframe] - 1) {
1005 if (get_bits(&s->gb, 16) != 0xFFFF) {
1006 av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n");
1007 return AVERROR_INVALIDDATA;
1014 static int dca_filter_channels(DCAContext *s, int block_index, int upsample)
1019 LOCAL_ALIGNED(32, float, samples, [DCA_SUBBANDS_X96K], [SAMPLES_PER_SUBBAND]);
1021 if (!s->qmf64_table) {
1022 s->qmf64_table = qmf64_precompute();
1023 if (!s->qmf64_table)
1024 return AVERROR(ENOMEM);
1027 /* 64 subbands QMF */
1028 for (k = 0; k < s->audio_header.prim_channels; k++) {
1029 int32_t (*subband_samples)[SAMPLES_PER_SUBBAND] =
1030 s->dca_chan[k].subband_samples[block_index];
1032 s->fmt_conv.int32_to_float(samples[0], subband_samples[0],
1033 DCA_SUBBANDS_X96K * SAMPLES_PER_SUBBAND);
1035 if (s->channel_order_tab[k] >= 0)
1036 qmf_64_subbands(s, k, samples,
1037 s->samples_chanptr[s->channel_order_tab[k]],
1038 /* Upsampling needs a factor 2 here. */
1042 /* 32 subbands QMF */
1043 LOCAL_ALIGNED(32, float, samples, [DCA_SUBBANDS], [SAMPLES_PER_SUBBAND]);
1045 for (k = 0; k < s->audio_header.prim_channels; k++) {
1046 int32_t (*subband_samples)[SAMPLES_PER_SUBBAND] =
1047 s->dca_chan[k].subband_samples[block_index];
1049 s->fmt_conv.int32_to_float(samples[0], subband_samples[0],
1050 DCA_SUBBANDS * SAMPLES_PER_SUBBAND);
1052 if (s->channel_order_tab[k] >= 0)
1053 qmf_32_subbands(s, k, samples,
1054 s->samples_chanptr[s->channel_order_tab[k]],
1055 M_SQRT1_2 / 32768.0);
1059 /* Generate LFE samples for this subsubframe FIXME!!! */
1061 float *samples = s->samples_chanptr[s->lfe_index];
1062 lfe_interpolation_fir(s,
1063 s->lfe_data + 2 * s->lfe * (block_index + 4),
1067 /* Should apply the filter in Table 6-11 when upsampling. For
1068 * now, just duplicate. */
1069 for (i = 255; i > 0; i--) {
1071 samples[2 * i + 1] = samples[i];
1073 samples[1] = samples[0];
1077 /* FIXME: This downmixing is probably broken with upsample.
1078 * Probably totally broken also with XLL in general. */
1079 /* Downmixing to Stereo */
1080 if (s->audio_header.prim_channels + !!s->lfe > 2 &&
1081 s->avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
1082 dca_downmix(s->samples_chanptr, s->amode, !!s->lfe, s->downmix_coef,
1083 s->channel_order_tab);
1089 static int dca_subframe_footer(DCAContext *s, int base_channel)
1091 int in, out, aux_data_count, aux_data_end, reserved;
1095 * Unpack optional information
1098 /* presumably optional information only appears in the core? */
1099 if (!base_channel) {
1101 skip_bits_long(&s->gb, 32);
1104 aux_data_count = get_bits(&s->gb, 6);
1107 skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
1109 aux_data_end = 8 * aux_data_count + get_bits_count(&s->gb);
1111 if ((nsyncaux = get_bits_long(&s->gb, 32)) != DCA_NSYNCAUX) {
1112 av_log(s->avctx, AV_LOG_ERROR, "nSYNCAUX mismatch %#"PRIx32"\n",
1114 return AVERROR_INVALIDDATA;
1117 if (get_bits1(&s->gb)) { // bAUXTimeStampFlag
1118 avpriv_request_sample(s->avctx,
1119 "Auxiliary Decode Time Stamp Flag");
1121 skip_bits(&s->gb, (-get_bits_count(&s->gb)) & 4);
1122 // 44 bits: nMSByte (8), nMarker (4), nLSByte (28), nMarker (4)
1123 skip_bits_long(&s->gb, 44);
1126 if ((s->core_downmix = get_bits1(&s->gb))) {
1127 int am = get_bits(&s->gb, 3);
1130 s->core_downmix_amode = DCA_MONO;
1133 s->core_downmix_amode = DCA_STEREO;
1136 s->core_downmix_amode = DCA_STEREO_TOTAL;
1139 s->core_downmix_amode = DCA_3F;
1142 s->core_downmix_amode = DCA_2F1R;
1145 s->core_downmix_amode = DCA_2F2R;
1148 s->core_downmix_amode = DCA_3F1R;
1151 av_log(s->avctx, AV_LOG_ERROR,
1152 "Invalid mode %d for embedded downmix coefficients\n",
1154 return AVERROR_INVALIDDATA;
1156 for (out = 0; out < ff_dca_channels[s->core_downmix_amode]; out++) {
1157 for (in = 0; in < s->audio_header.prim_channels + !!s->lfe; in++) {
1158 uint16_t tmp = get_bits(&s->gb, 9);
1159 if ((tmp & 0xFF) > 241) {
1160 av_log(s->avctx, AV_LOG_ERROR,
1161 "Invalid downmix coefficient code %"PRIu16"\n",
1163 return AVERROR_INVALIDDATA;
1165 s->core_downmix_codes[in][out] = tmp;
1170 align_get_bits(&s->gb); // byte align
1171 skip_bits(&s->gb, 16); // nAUXCRC16
1174 * additional data (reserved, cf. ETSI TS 102 114 V1.4.1)
1176 * Note: don't check for overreads, aux_data_count can't be trusted.
1178 if ((reserved = (aux_data_end - get_bits_count(&s->gb))) > 0) {
1179 avpriv_request_sample(s->avctx,
1180 "Core auxiliary data reserved content");
1181 skip_bits_long(&s->gb, reserved);
1185 if (s->crc_present && s->dynrange)
1186 get_bits(&s->gb, 16);
1193 * Decode a dca frame block
1195 * @param s pointer to the DCAContext
1198 static int dca_decode_block(DCAContext *s, int base_channel, int block_index)
1203 if (s->current_subframe >= s->audio_header.subframes) {
1204 av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i",
1205 s->current_subframe, s->audio_header.subframes);
1206 return AVERROR_INVALIDDATA;
1209 if (!s->current_subsubframe) {
1210 /* Read subframe header */
1211 if ((ret = dca_subframe_header(s, base_channel, block_index)))
1215 /* Read subsubframe */
1216 if ((ret = dca_subsubframe(s, base_channel, block_index)))
1220 s->current_subsubframe++;
1221 if (s->current_subsubframe >= s->subsubframes[s->current_subframe]) {
1222 s->current_subsubframe = 0;
1223 s->current_subframe++;
1225 if (s->current_subframe >= s->audio_header.subframes) {
1226 /* Read subframe footer */
1227 if ((ret = dca_subframe_footer(s, base_channel)))
1234 int ff_dca_xbr_parse_frame(DCAContext *s)
1236 int scale_table_high[DCA_CHSET_CHANS_MAX][DCA_SUBBANDS][2];
1237 int active_bands[DCA_CHSETS_MAX][DCA_CHSET_CHANS_MAX];
1238 int abits_high[DCA_CHSET_CHANS_MAX][DCA_SUBBANDS];
1239 int anctemp[DCA_CHSET_CHANS_MAX];
1240 int chset_fsize[DCA_CHSETS_MAX];
1241 int n_xbr_ch[DCA_CHSETS_MAX];
1242 int hdr_size, num_chsets, xbr_tmode, hdr_pos;
1243 int i, j, k, l, chset, chan_base;
1245 av_log(s->avctx, AV_LOG_DEBUG, "DTS-XBR: decoding XBR extension\n");
1247 /* get bit position of sync header */
1248 hdr_pos = get_bits_count(&s->gb) - 32;
1250 hdr_size = get_bits(&s->gb, 6) + 1;
1251 num_chsets = get_bits(&s->gb, 2) + 1;
1253 for(i = 0; i < num_chsets; i++)
1254 chset_fsize[i] = get_bits(&s->gb, 14) + 1;
1256 xbr_tmode = get_bits1(&s->gb);
1258 for(i = 0; i < num_chsets; i++) {
1259 n_xbr_ch[i] = get_bits(&s->gb, 3) + 1;
1260 k = get_bits(&s->gb, 2) + 5;
1261 for(j = 0; j < n_xbr_ch[i]; j++) {
1262 active_bands[i][j] = get_bits(&s->gb, k) + 1;
1263 if (active_bands[i][j] > DCA_SUBBANDS) {
1264 av_log(s->avctx, AV_LOG_ERROR, "too many active subbands (%d)\n", active_bands[i][j]);
1265 return AVERROR_INVALIDDATA;
1270 /* skip to the end of the header */
1271 i = get_bits_count(&s->gb);
1272 if(hdr_pos + hdr_size * 8 > i)
1273 skip_bits_long(&s->gb, hdr_pos + hdr_size * 8 - i);
1275 /* loop over the channel data sets */
1276 /* only decode as many channels as we've decoded base data for */
1277 for(chset = 0, chan_base = 0;
1278 chset < num_chsets && chan_base + n_xbr_ch[chset] <= s->audio_header.prim_channels;
1279 chan_base += n_xbr_ch[chset++]) {
1280 int start_posn = get_bits_count(&s->gb);
1281 int subsubframe = 0;
1284 /* loop over subframes */
1285 for (k = 0; k < (s->sample_blocks / 8); k++) {
1286 /* parse header if we're on first subsubframe of a block */
1287 if(subsubframe == 0) {
1288 /* Parse subframe header */
1289 for(i = 0; i < n_xbr_ch[chset]; i++) {
1290 anctemp[i] = get_bits(&s->gb, 2) + 2;
1293 for(i = 0; i < n_xbr_ch[chset]; i++) {
1294 get_array(&s->gb, abits_high[i], active_bands[chset][i], anctemp[i]);
1297 for(i = 0; i < n_xbr_ch[chset]; i++) {
1298 anctemp[i] = get_bits(&s->gb, 3);
1299 if(anctemp[i] < 1) {
1300 av_log(s->avctx, AV_LOG_ERROR, "DTS-XBR: SYNC ERROR\n");
1301 return AVERROR_INVALIDDATA;
1305 /* generate scale factors */
1306 for(i = 0; i < n_xbr_ch[chset]; i++) {
1307 const uint32_t *scale_table;
1309 int scale_table_size;
1311 if (s->audio_header.scalefactor_huffman[chan_base+i] == 6) {
1312 scale_table = ff_dca_scale_factor_quant7;
1313 scale_table_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant7);
1315 scale_table = ff_dca_scale_factor_quant6;
1316 scale_table_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant6);
1321 for(j = 0; j < active_bands[chset][i]; j++) {
1322 if(abits_high[i][j] > 0) {
1323 int index = get_bits(&s->gb, nbits);
1324 if (index >= scale_table_size) {
1325 av_log(s->avctx, AV_LOG_ERROR, "scale table index %d invalid\n", index);
1326 return AVERROR_INVALIDDATA;
1328 scale_table_high[i][j][0] = scale_table[index];
1330 if(xbr_tmode && s->dca_chan[i].transition_mode[j]) {
1331 int index = get_bits(&s->gb, nbits);
1332 if (index >= scale_table_size) {
1333 av_log(s->avctx, AV_LOG_ERROR, "scale table index %d invalid\n", index);
1334 return AVERROR_INVALIDDATA;
1336 scale_table_high[i][j][1] = scale_table[index];
1343 /* decode audio array for this block */
1344 for(i = 0; i < n_xbr_ch[chset]; i++) {
1345 for(j = 0; j < active_bands[chset][i]; j++) {
1346 const int xbr_abits = abits_high[i][j];
1347 const uint32_t quant_step_size = ff_dca_lossless_quant[xbr_abits];
1348 const int sfi = xbr_tmode && s->dca_chan[i].transition_mode[j] && subsubframe >= s->dca_chan[i].transition_mode[j];
1349 const uint32_t rscale = scale_table_high[i][j][sfi];
1350 int32_t *subband_samples = s->dca_chan[chan_base+i].subband_samples[k][j];
1351 int32_t block[SAMPLES_PER_SUBBAND];
1357 get_array(&s->gb, block, SAMPLES_PER_SUBBAND, xbr_abits - 3);
1359 int block_code1, block_code2, size, levels, err;
1361 size = abits_sizes[xbr_abits - 1];
1362 levels = abits_levels[xbr_abits - 1];
1364 block_code1 = get_bits(&s->gb, size);
1365 block_code2 = get_bits(&s->gb, size);
1366 err = decode_blockcodes(block_code1, block_code2,
1369 av_log(s->avctx, AV_LOG_ERROR,
1370 "ERROR: DTS-XBR: block code look-up failed\n");
1371 return AVERROR_INVALIDDATA;
1375 /* scale & sum into subband */
1376 s->dcadsp.dequantize(block, quant_step_size, rscale);
1377 for(l = 0; l < SAMPLES_PER_SUBBAND; l++)
1378 subband_samples[l] += block[l];
1382 /* check DSYNC marker */
1383 if(s->aspf || subsubframe == s->subsubframes[subframe] - 1) {
1384 if(get_bits(&s->gb, 16) != 0xffff) {
1385 av_log(s->avctx, AV_LOG_ERROR, "DTS-XBR: Didn't get subframe DSYNC\n");
1386 return AVERROR_INVALIDDATA;
1390 /* advance sub-sub-frame index */
1391 if(++subsubframe >= s->subsubframes[subframe]) {
1397 /* skip to next channel set */
1398 i = get_bits_count(&s->gb);
1399 if(start_posn + chset_fsize[chset] * 8 != i) {
1400 j = start_posn + chset_fsize[chset] * 8 - i;
1402 av_log(s->avctx, AV_LOG_ERROR, "DTS-XBR: end of channel set,"
1403 " skipping further than expected (%d bits)\n", j);
1404 skip_bits_long(&s->gb, j);
1412 /* parse initial header for XXCH and dump details */
1413 int ff_dca_xxch_decode_frame(DCAContext *s)
1415 int hdr_size, spkmsk_bits, num_chsets, core_spk, hdr_pos;
1416 int i, chset, base_channel, chstart, fsize[8];
1418 /* assume header word has already been parsed */
1419 hdr_pos = get_bits_count(&s->gb) - 32;
1420 hdr_size = get_bits(&s->gb, 6) + 1;
1421 /*chhdr_crc =*/ skip_bits1(&s->gb);
1422 spkmsk_bits = get_bits(&s->gb, 5) + 1;
1423 num_chsets = get_bits(&s->gb, 2) + 1;
1425 for (i = 0; i < num_chsets; i++)
1426 fsize[i] = get_bits(&s->gb, 14) + 1;
1428 core_spk = get_bits(&s->gb, spkmsk_bits);
1429 s->xxch_core_spkmask = core_spk;
1430 s->xxch_nbits_spk_mask = spkmsk_bits;
1431 s->xxch_dmix_embedded = 0;
1433 /* skip to the end of the header */
1434 i = get_bits_count(&s->gb);
1435 if (hdr_pos + hdr_size * 8 > i)
1436 skip_bits_long(&s->gb, hdr_pos + hdr_size * 8 - i);
1438 for (chset = 0; chset < num_chsets; chset++) {
1439 chstart = get_bits_count(&s->gb);
1440 base_channel = s->audio_header.prim_channels;
1441 s->xxch_chset = chset;
1443 /* XXCH and Core headers differ, see 6.4.2 "XXCH Channel Set Header" vs.
1444 5.3.2 "Primary Audio Coding Header", DTS Spec 1.3.1 */
1445 dca_parse_audio_coding_header(s, base_channel, 1);
1447 /* decode channel data */
1448 for (i = 0; i < (s->sample_blocks / 8); i++) {
1449 if (dca_decode_block(s, base_channel, i)) {
1450 av_log(s->avctx, AV_LOG_ERROR,
1451 "Error decoding DTS-XXCH extension\n");
1456 /* skip to end of this section */
1457 i = get_bits_count(&s->gb);
1458 if (chstart + fsize[chset] * 8 > i)
1459 skip_bits_long(&s->gb, chstart + fsize[chset] * 8 - i);
1461 s->xxch_chset = num_chsets;
1466 static float dca_dmix_code(unsigned code)
1468 int sign = (code >> 8) - 1;
1470 return ((ff_dca_dmixtable[code] ^ sign) - sign) * (1.0 / (1 << 15));
1473 static int scan_for_extensions(AVCodecContext *avctx)
1475 DCAContext *s = avctx->priv_data;
1476 int core_ss_end, ret = 0;
1478 core_ss_end = FFMIN(s->frame_size, s->dca_buffer_size) * 8;
1480 /* only scan for extensions if ext_descr was unknown or indicated a
1481 * supported XCh extension */
1482 if (s->core_ext_mask < 0 || s->core_ext_mask & (DCA_EXT_XCH | DCA_EXT_XXCH)) {
1483 /* if ext_descr was unknown, clear s->core_ext_mask so that the
1484 * extensions scan can fill it up */
1485 s->core_ext_mask = FFMAX(s->core_ext_mask, 0);
1487 /* extensions start at 32-bit boundaries into bitstream */
1488 skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
1490 while (core_ss_end - get_bits_count(&s->gb) >= 32) {
1491 uint32_t bits = get_bits_long(&s->gb, 32);
1495 case DCA_SYNCWORD_XCH: {
1496 int ext_amode, xch_fsize;
1498 s->xch_base_channel = s->audio_header.prim_channels;
1500 /* validate sync word using XCHFSIZE field */
1501 xch_fsize = show_bits(&s->gb, 10);
1502 if ((s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize) &&
1503 (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize + 1))
1506 /* skip length-to-end-of-frame field for the moment */
1507 skip_bits(&s->gb, 10);
1509 s->core_ext_mask |= DCA_EXT_XCH;
1511 /* extension amode(number of channels in extension) should be 1 */
1512 /* AFAIK XCh is not used for more channels */
1513 if ((ext_amode = get_bits(&s->gb, 4)) != 1) {
1514 av_log(avctx, AV_LOG_ERROR,
1515 "XCh extension amode %d not supported!\n",
1520 if (s->xch_base_channel < 2) {
1521 avpriv_request_sample(avctx, "XCh with fewer than 2 base channels");
1525 /* much like core primary audio coding header */
1526 dca_parse_audio_coding_header(s, s->xch_base_channel, 0);
1528 for (i = 0; i < (s->sample_blocks / 8); i++)
1529 if ((ret = dca_decode_block(s, s->xch_base_channel, i))) {
1530 av_log(avctx, AV_LOG_ERROR, "error decoding XCh extension\n");
1537 case DCA_SYNCWORD_XXCH:
1538 /* XXCh: extended channels */
1539 /* usually found either in core or HD part in DTS-HD HRA streams,
1540 * but not in DTS-ES which contains XCh extensions instead */
1541 s->core_ext_mask |= DCA_EXT_XXCH;
1542 ff_dca_xxch_decode_frame(s);
1546 int fsize96 = show_bits(&s->gb, 12) + 1;
1547 if (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + fsize96)
1550 av_log(avctx, AV_LOG_DEBUG, "X96 extension found at %d bits\n",
1551 get_bits_count(&s->gb));
1552 skip_bits(&s->gb, 12);
1553 av_log(avctx, AV_LOG_DEBUG, "FSIZE96 = %d bytes\n", fsize96);
1554 av_log(avctx, AV_LOG_DEBUG, "REVNO = %d\n", get_bits(&s->gb, 4));
1556 s->core_ext_mask |= DCA_EXT_X96;
1561 skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
1564 /* no supported extensions, skip the rest of the core substream */
1565 skip_bits_long(&s->gb, core_ss_end - get_bits_count(&s->gb));
1568 if (s->core_ext_mask & DCA_EXT_X96)
1569 s->profile = FF_PROFILE_DTS_96_24;
1570 else if (s->core_ext_mask & (DCA_EXT_XCH | DCA_EXT_XXCH))
1571 s->profile = FF_PROFILE_DTS_ES;
1573 /* check for ExSS (HD part) */
1574 if (s->dca_buffer_size - s->frame_size > 32 &&
1575 get_bits_long(&s->gb, 32) == DCA_SYNCWORD_SUBSTREAM)
1576 ff_dca_exss_parse_header(s);
1581 static int set_channel_layout(AVCodecContext *avctx, int *channels, int num_core_channels)
1583 DCAContext *s = avctx->priv_data;
1584 int i, j, chset, mask;
1585 int channel_layout, channel_mask;
1588 /* If we have XXCH then the channel layout is managed differently */
1589 /* note that XLL will also have another way to do things */
1590 if (!(s->core_ext_mask & DCA_EXT_XXCH)) {
1591 /* xxx should also do MA extensions */
1592 if (s->amode < 16) {
1593 avctx->channel_layout = ff_dca_core_channel_layout[s->amode];
1595 if (s->audio_header.prim_channels + !!s->lfe > 2 &&
1596 avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
1598 * Neither the core's auxiliary data nor our default tables contain
1599 * downmix coefficients for the additional channel coded in the XCh
1600 * extension, so when we're doing a Stereo downmix, don't decode it.
1605 if (s->xch_present && !s->xch_disable) {
1606 if (avctx->channel_layout & AV_CH_BACK_CENTER) {
1607 avpriv_request_sample(avctx, "XCh with Back center channel");
1608 return AVERROR_INVALIDDATA;
1610 avctx->channel_layout |= AV_CH_BACK_CENTER;
1612 avctx->channel_layout |= AV_CH_LOW_FREQUENCY;
1613 s->channel_order_tab = ff_dca_channel_reorder_lfe_xch[s->amode];
1615 s->channel_order_tab = ff_dca_channel_reorder_nolfe_xch[s->amode];
1617 if (s->channel_order_tab[s->xch_base_channel] < 0)
1618 return AVERROR_INVALIDDATA;
1620 *channels = num_core_channels + !!s->lfe;
1621 s->xch_present = 0; /* disable further xch processing */
1623 avctx->channel_layout |= AV_CH_LOW_FREQUENCY;
1624 s->channel_order_tab = ff_dca_channel_reorder_lfe[s->amode];
1626 s->channel_order_tab = ff_dca_channel_reorder_nolfe[s->amode];
1629 if (*channels > !!s->lfe &&
1630 s->channel_order_tab[*channels - 1 - !!s->lfe] < 0)
1631 return AVERROR_INVALIDDATA;
1633 if (av_get_channel_layout_nb_channels(avctx->channel_layout) != *channels) {
1634 av_log(avctx, AV_LOG_ERROR, "Number of channels %d mismatches layout %d\n", *channels, av_get_channel_layout_nb_channels(avctx->channel_layout));
1635 return AVERROR_INVALIDDATA;
1638 if (num_core_channels + !!s->lfe > 2 &&
1639 avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
1641 s->output = s->audio_header.prim_channels == 2 ? s->amode : DCA_STEREO;
1642 avctx->channel_layout = AV_CH_LAYOUT_STEREO;
1644 else if (avctx->request_channel_layout & AV_CH_LAYOUT_NATIVE) {
1645 static const int8_t dca_channel_order_native[9] = { 0, 1, 2, 3, 4, 5, 6, 7, 8 };
1646 s->channel_order_tab = dca_channel_order_native;
1648 s->lfe_index = ff_dca_lfe_index[s->amode];
1650 av_log(avctx, AV_LOG_ERROR,
1651 "Non standard configuration %d !\n", s->amode);
1652 return AVERROR_INVALIDDATA;
1655 s->xxch_dmix_embedded = 0;
1657 /* we only get here if an XXCH channel set can be added to the mix */
1658 channel_mask = s->xxch_core_spkmask;
1661 *channels = s->audio_header.prim_channels + !!s->lfe;
1662 for (i = 0; i < s->xxch_chset; i++) {
1663 channel_mask |= s->xxch_spk_masks[i];
1667 /* Given the DTS spec'ed channel mask, generate an avcodec version */
1669 for (i = 0; i < s->xxch_nbits_spk_mask; ++i) {
1670 if (channel_mask & (1 << i)) {
1671 channel_layout |= ff_dca_map_xxch_to_native[i];
1675 /* make sure that we have managed to get equivalent dts/avcodec channel
1676 * masks in some sense -- unfortunately some channels could overlap */
1677 if (av_popcount(channel_mask) != av_popcount(channel_layout)) {
1678 av_log(avctx, AV_LOG_DEBUG,
1679 "DTS-XXCH: Inconsistent avcodec/dts channel layouts\n");
1680 return AVERROR_INVALIDDATA;
1683 avctx->channel_layout = channel_layout;
1685 if (!(avctx->request_channel_layout & AV_CH_LAYOUT_NATIVE)) {
1686 /* Estimate DTS --> avcodec ordering table */
1687 for (chset = -1, j = 0; chset < s->xxch_chset; ++chset) {
1688 mask = chset >= 0 ? s->xxch_spk_masks[chset]
1689 : s->xxch_core_spkmask;
1690 for (i = 0; i < s->xxch_nbits_spk_mask; i++) {
1691 if (mask & ~(DCA_XXCH_LFE1 | DCA_XXCH_LFE2) & (1 << i)) {
1692 lavc = ff_dca_map_xxch_to_native[i];
1693 posn = av_popcount(channel_layout & (lavc - 1));
1694 s->xxch_order_tab[j++] = posn;
1700 s->lfe_index = av_popcount(channel_layout & (AV_CH_LOW_FREQUENCY-1));
1701 } else { /* native ordering */
1702 for (i = 0; i < *channels; i++)
1703 s->xxch_order_tab[i] = i;
1705 s->lfe_index = *channels - 1;
1708 s->channel_order_tab = s->xxch_order_tab;
1715 * Main frame decoding function
1716 * FIXME add arguments
1718 static int dca_decode_frame(AVCodecContext *avctx, void *data,
1719 int *got_frame_ptr, AVPacket *avpkt)
1721 AVFrame *frame = data;
1722 const uint8_t *buf = avpkt->data;
1723 int buf_size = avpkt->size;
1725 int num_core_channels = 0;
1727 float **samples_flt;
1730 DCAContext *s = avctx->priv_data;
1731 int channels, full_channels;
1740 s->exss_ext_mask = 0;
1743 s->dca_buffer_size = AVERROR_INVALIDDATA;
1744 for (i = 0; i < buf_size - 3 && s->dca_buffer_size == AVERROR_INVALIDDATA; i++)
1745 s->dca_buffer_size = avpriv_dca_convert_bitstream(buf + i, buf_size - i, s->dca_buffer,
1746 DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE);
1748 if (s->dca_buffer_size == AVERROR_INVALIDDATA) {
1749 av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n");
1750 return AVERROR_INVALIDDATA;
1753 if ((ret = dca_parse_frame_header(s)) < 0) {
1754 // seems like the frame is corrupt, try with the next one
1757 // set AVCodec values with parsed data
1758 avctx->sample_rate = s->sample_rate;
1760 s->profile = FF_PROFILE_DTS;
1762 for (i = 0; i < (s->sample_blocks / SAMPLES_PER_SUBBAND); i++) {
1763 if ((ret = dca_decode_block(s, 0, i))) {
1764 av_log(avctx, AV_LOG_ERROR, "error decoding block\n");
1769 /* record number of core channels incase less than max channels are requested */
1770 num_core_channels = s->audio_header.prim_channels;
1772 if (s->audio_header.prim_channels + !!s->lfe > 2 &&
1773 avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
1774 /* Stereo downmix coefficients
1776 * The decoder can only downmix to 2-channel, so we need to ensure
1777 * embedded downmix coefficients are actually targeting 2-channel.
1779 if (s->core_downmix && (s->core_downmix_amode == DCA_STEREO ||
1780 s->core_downmix_amode == DCA_STEREO_TOTAL)) {
1781 for (i = 0; i < num_core_channels + !!s->lfe; i++) {
1782 /* Range checked earlier */
1783 s->downmix_coef[i][0] = dca_dmix_code(s->core_downmix_codes[i][0]);
1784 s->downmix_coef[i][1] = dca_dmix_code(s->core_downmix_codes[i][1]);
1786 s->output = s->core_downmix_amode;
1788 int am = s->amode & DCA_CHANNEL_MASK;
1789 if (am >= FF_ARRAY_ELEMS(ff_dca_default_coeffs)) {
1790 av_log(s->avctx, AV_LOG_ERROR,
1791 "Invalid channel mode %d\n", am);
1792 return AVERROR_INVALIDDATA;
1794 if (num_core_channels + !!s->lfe >
1795 FF_ARRAY_ELEMS(ff_dca_default_coeffs[0])) {
1796 avpriv_request_sample(s->avctx, "Downmixing %d channels",
1797 s->audio_header.prim_channels + !!s->lfe);
1798 return AVERROR_PATCHWELCOME;
1800 for (i = 0; i < num_core_channels + !!s->lfe; i++) {
1801 s->downmix_coef[i][0] = ff_dca_default_coeffs[am][i][0];
1802 s->downmix_coef[i][1] = ff_dca_default_coeffs[am][i][1];
1805 ff_dlog(s->avctx, "Stereo downmix coeffs:\n");
1806 for (i = 0; i < num_core_channels + !!s->lfe; i++) {
1807 ff_dlog(s->avctx, "L, input channel %d = %f\n", i,
1808 s->downmix_coef[i][0]);
1809 ff_dlog(s->avctx, "R, input channel %d = %f\n", i,
1810 s->downmix_coef[i][1]);
1812 ff_dlog(s->avctx, "\n");
1816 s->core_ext_mask = ff_dca_ext_audio_descr_mask[s->ext_descr];
1818 s->core_ext_mask = 0;
1820 ret = scan_for_extensions(avctx);
1822 avctx->profile = s->profile;
1824 full_channels = channels = s->audio_header.prim_channels + !!s->lfe;
1826 ret = set_channel_layout(avctx, &channels, num_core_channels);
1830 /* get output buffer */
1831 frame->nb_samples = 256 * (s->sample_blocks / SAMPLES_PER_SUBBAND);
1832 if (s->exss_ext_mask & DCA_EXT_EXSS_XLL) {
1833 int xll_nb_samples = s->xll_segments * s->xll_smpl_in_seg;
1834 /* Check for invalid/unsupported conditions first */
1835 if (s->xll_residual_channels > channels) {
1836 av_log(s->avctx, AV_LOG_WARNING,
1837 "DCA: too many residual channels (%d, core channels %d). Disabling XLL\n",
1838 s->xll_residual_channels, channels);
1839 s->exss_ext_mask &= ~DCA_EXT_EXSS_XLL;
1840 } else if (xll_nb_samples != frame->nb_samples &&
1841 2 * frame->nb_samples != xll_nb_samples) {
1842 av_log(s->avctx, AV_LOG_WARNING,
1843 "DCA: unsupported upsampling (%d XLL samples, %d core samples). Disabling XLL\n",
1844 xll_nb_samples, frame->nb_samples);
1845 s->exss_ext_mask &= ~DCA_EXT_EXSS_XLL;
1847 if (2 * frame->nb_samples == xll_nb_samples) {
1848 av_log(s->avctx, AV_LOG_INFO,
1849 "XLL: upsampling core channels by a factor of 2\n");
1852 frame->nb_samples = xll_nb_samples;
1853 // FIXME: Is it good enough to copy from the first channel set?
1854 avctx->sample_rate = s->xll_chsets[0].sampling_frequency;
1856 /* If downmixing to stereo, don't decode additional channels.
1857 * FIXME: Using the xch_disable flag for this doesn't seem right. */
1858 if (!s->xch_disable)
1859 channels = s->xll_channels;
1863 if (avctx->channels != channels) {
1864 if (avctx->channels)
1865 av_log(avctx, AV_LOG_INFO, "Number of channels changed in DCA decoder (%d -> %d)\n", avctx->channels, channels);
1866 avctx->channels = channels;
1869 /* FIXME: This is an ugly hack, to just revert to the default
1870 * layout if we have additional channels. Need to convert the XLL
1871 * channel masks to ffmpeg channel_layout mask. */
1872 if (av_get_channel_layout_nb_channels(avctx->channel_layout) != avctx->channels)
1873 avctx->channel_layout = 0;
1875 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
1877 samples_flt = (float **) frame->extended_data;
1879 /* allocate buffer for extra channels if downmixing */
1880 if (avctx->channels < full_channels) {
1881 ret = av_samples_get_buffer_size(NULL, full_channels - channels,
1883 avctx->sample_fmt, 0);
1887 av_fast_malloc(&s->extra_channels_buffer,
1888 &s->extra_channels_buffer_size, ret);
1889 if (!s->extra_channels_buffer)
1890 return AVERROR(ENOMEM);
1892 ret = av_samples_fill_arrays((uint8_t **) s->extra_channels, NULL,
1893 s->extra_channels_buffer,
1894 full_channels - channels,
1895 frame->nb_samples, avctx->sample_fmt, 0);
1900 /* filter to get final output */
1901 for (i = 0; i < (s->sample_blocks / SAMPLES_PER_SUBBAND); i++) {
1903 unsigned block = upsample ? 512 : 256;
1904 for (ch = 0; ch < channels; ch++)
1905 s->samples_chanptr[ch] = samples_flt[ch] + i * block;
1906 for (; ch < full_channels; ch++)
1907 s->samples_chanptr[ch] = s->extra_channels[ch - channels] + i * block;
1909 dca_filter_channels(s, i, upsample);
1911 /* If this was marked as a DTS-ES stream we need to subtract back- */
1912 /* channel from SL & SR to remove matrixed back-channel signal */
1913 if ((s->source_pcm_res & 1) && s->xch_present) {
1914 float *back_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel]];
1915 float *lt_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 2]];
1916 float *rt_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 1]];
1917 s->fdsp->vector_fmac_scalar(lt_chan, back_chan, -M_SQRT1_2, 256);
1918 s->fdsp->vector_fmac_scalar(rt_chan, back_chan, -M_SQRT1_2, 256);
1921 /* If stream contains XXCH, we might need to undo an embedded downmix */
1922 if (s->xxch_dmix_embedded) {
1923 /* Loop over channel sets in turn */
1924 ch = num_core_channels;
1925 for (chset = 0; chset < s->xxch_chset; chset++) {
1926 endch = ch + s->xxch_chset_nch[chset];
1927 mask = s->xxch_dmix_embedded;
1930 for (j = ch; j < endch; j++) {
1931 if (mask & (1 << j)) { /* this channel has been mixed-out */
1932 src_chan = s->samples_chanptr[s->channel_order_tab[j]];
1933 for (k = 0; k < endch; k++) {
1934 achan = s->channel_order_tab[k];
1935 scale = s->xxch_dmix_coeff[j][k];
1937 dst_chan = s->samples_chanptr[achan];
1938 s->fdsp->vector_fmac_scalar(dst_chan, src_chan,
1945 /* if a downmix has been embedded then undo the pre-scaling */
1946 if ((mask & (1 << ch)) && s->xxch_dmix_sf[chset] != 1.0f) {
1947 scale = s->xxch_dmix_sf[chset];
1949 for (j = 0; j < ch; j++) {
1950 src_chan = s->samples_chanptr[s->channel_order_tab[j]];
1951 for (k = 0; k < 256; k++)
1952 src_chan[k] *= scale;
1955 /* LFE channel is always part of core, scale if it exists */
1957 src_chan = s->samples_chanptr[s->lfe_index];
1958 for (k = 0; k < 256; k++)
1959 src_chan[k] *= scale;
1969 /* update lfe history */
1970 lfe_samples = 2 * s->lfe * (s->sample_blocks / SAMPLES_PER_SUBBAND);
1971 for (i = 0; i < 2 * s->lfe * 4; i++)
1972 s->lfe_data[i] = s->lfe_data[i + lfe_samples];
1974 if (s->exss_ext_mask & DCA_EXT_EXSS_XLL) {
1975 ret = ff_dca_xll_decode_audio(s, frame);
1981 * DCA_STEREO_TOTAL (Lt/Rt) is equivalent to Dolby Surround */
1982 ret = ff_side_data_update_matrix_encoding(frame,
1983 (s->output & ~DCA_LFE) == DCA_STEREO_TOTAL ?
1984 AV_MATRIX_ENCODING_DOLBY : AV_MATRIX_ENCODING_NONE);
1988 if ( avctx->profile != FF_PROFILE_DTS_HD_MA
1989 && avctx->profile != FF_PROFILE_DTS_HD_HRA)
1990 avctx->bit_rate = s->bit_rate;
1997 * DCA initialization
1999 * @param avctx pointer to the AVCodecContext
2002 static av_cold int dca_decode_init(AVCodecContext *avctx)
2004 DCAContext *s = avctx->priv_data;
2009 s->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
2011 return AVERROR(ENOMEM);
2013 ff_mdct_init(&s->imdct, 6, 1, 1.0);
2014 ff_synth_filter_init(&s->synth);
2015 ff_dcadsp_init(&s->dcadsp);
2016 ff_fmt_convert_init(&s->fmt_conv, avctx);
2018 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
2020 /* allow downmixing to stereo */
2021 if (avctx->channels > 2 &&
2022 avctx->request_channel_layout == AV_CH_LAYOUT_STEREO)
2023 avctx->channels = 2;
2028 static av_cold int dca_decode_end(AVCodecContext *avctx)
2030 DCAContext *s = avctx->priv_data;
2031 ff_mdct_end(&s->imdct);
2032 av_freep(&s->extra_channels_buffer);
2034 av_freep(&s->xll_sample_buf);
2035 av_freep(&s->qmf64_table);
2039 static const AVOption options[] = {
2040 { "disable_xch", "disable decoding of the XCh extension", offsetof(DCAContext, xch_disable), AV_OPT_TYPE_BOOL, { .i64 = 0 }, 0, 1, AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM },
2041 { "disable_xll", "disable decoding of the XLL extension", offsetof(DCAContext, xll_disable), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM },
2045 static const AVClass dca_decoder_class = {
2046 .class_name = "DCA decoder",
2047 .item_name = av_default_item_name,
2049 .version = LIBAVUTIL_VERSION_INT,
2050 .category = AV_CLASS_CATEGORY_DECODER,
2053 AVCodec ff_dca_decoder = {
2055 .long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
2056 .type = AVMEDIA_TYPE_AUDIO,
2057 .id = AV_CODEC_ID_DTS,
2058 .priv_data_size = sizeof(DCAContext),
2059 .init = dca_decode_init,
2060 .decode = dca_decode_frame,
2061 .close = dca_decode_end,
2062 .capabilities = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
2063 .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
2064 AV_SAMPLE_FMT_NONE },
2065 .profiles = NULL_IF_CONFIG_SMALL(ff_dca_profiles),
2066 .priv_class = &dca_decoder_class,