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 #define SAMPLES_PER_SUBBAND 8 // number of samples per subband per subsubframe
118 /** Bit allocation */
119 typedef struct BitAlloc {
120 int offset; ///< code values offset
121 int maxbits[8]; ///< max bits in VLC
122 int wrap; ///< wrap for get_vlc2()
123 VLC vlc[8]; ///< actual codes
126 static BitAlloc dca_bitalloc_index; ///< indexes for samples VLC select
127 static BitAlloc dca_tmode; ///< transition mode VLCs
128 static BitAlloc dca_scalefactor; ///< scalefactor VLCs
129 static BitAlloc dca_smpl_bitalloc[11]; ///< samples VLCs
131 static av_always_inline int get_bitalloc(GetBitContext *gb, BitAlloc *ba,
134 return get_vlc2(gb, ba->vlc[idx].table, ba->vlc[idx].bits, ba->wrap) +
138 static float dca_dmix_code(unsigned code);
140 static av_cold void dca_init_vlcs(void)
142 static int vlcs_initialized = 0;
144 static VLC_TYPE dca_table[23622][2];
146 if (vlcs_initialized)
149 dca_bitalloc_index.offset = 1;
150 dca_bitalloc_index.wrap = 2;
151 for (i = 0; i < 5; i++) {
152 dca_bitalloc_index.vlc[i].table = &dca_table[ff_dca_vlc_offs[i]];
153 dca_bitalloc_index.vlc[i].table_allocated = ff_dca_vlc_offs[i + 1] - ff_dca_vlc_offs[i];
154 init_vlc(&dca_bitalloc_index.vlc[i], bitalloc_12_vlc_bits[i], 12,
155 bitalloc_12_bits[i], 1, 1,
156 bitalloc_12_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
158 dca_scalefactor.offset = -64;
159 dca_scalefactor.wrap = 2;
160 for (i = 0; i < 5; i++) {
161 dca_scalefactor.vlc[i].table = &dca_table[ff_dca_vlc_offs[i + 5]];
162 dca_scalefactor.vlc[i].table_allocated = ff_dca_vlc_offs[i + 6] - ff_dca_vlc_offs[i + 5];
163 init_vlc(&dca_scalefactor.vlc[i], SCALES_VLC_BITS, 129,
164 scales_bits[i], 1, 1,
165 scales_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
167 dca_tmode.offset = 0;
169 for (i = 0; i < 4; i++) {
170 dca_tmode.vlc[i].table = &dca_table[ff_dca_vlc_offs[i + 10]];
171 dca_tmode.vlc[i].table_allocated = ff_dca_vlc_offs[i + 11] - ff_dca_vlc_offs[i + 10];
172 init_vlc(&dca_tmode.vlc[i], tmode_vlc_bits[i], 4,
174 tmode_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
177 for (i = 0; i < 10; i++)
178 for (j = 0; j < 7; j++) {
179 if (!bitalloc_codes[i][j])
181 dca_smpl_bitalloc[i + 1].offset = bitalloc_offsets[i];
182 dca_smpl_bitalloc[i + 1].wrap = 1 + (j > 4);
183 dca_smpl_bitalloc[i + 1].vlc[j].table = &dca_table[ff_dca_vlc_offs[c]];
184 dca_smpl_bitalloc[i + 1].vlc[j].table_allocated = ff_dca_vlc_offs[c + 1] - ff_dca_vlc_offs[c];
186 init_vlc(&dca_smpl_bitalloc[i + 1].vlc[j], bitalloc_maxbits[i][j],
188 bitalloc_bits[i][j], 1, 1,
189 bitalloc_codes[i][j], 2, 2, INIT_VLC_USE_NEW_STATIC);
192 vlcs_initialized = 1;
195 static inline void get_array(GetBitContext *gb, int *dst, int len, int bits)
198 *dst++ = get_bits(gb, bits);
201 static inline int dca_xxch2index(DCAContext *s, int xxch_ch)
205 /* locate channel set containing the channel */
206 for (i = -1, base = 0, mask = (s->xxch_core_spkmask & ~DCA_XXCH_LFE1);
207 i <= s->xxch_chset && !(mask & xxch_ch); mask = s->xxch_spk_masks[++i])
208 base += av_popcount(mask);
210 return base + av_popcount(mask & (xxch_ch - 1));
213 static int dca_parse_audio_coding_header(DCAContext *s, int base_channel,
217 static const float adj_table[4] = { 1.0, 1.1250, 1.2500, 1.4375 };
218 static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };
219 static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };
220 int hdr_pos = 0, hdr_size = 0;
222 int this_chans, acc_mask;
223 int embedded_downmix;
227 /* xxch has arbitrary sized audio coding headers */
229 hdr_pos = get_bits_count(&s->gb);
230 hdr_size = get_bits(&s->gb, 7) + 1;
233 nchans = get_bits(&s->gb, 3) + 1;
234 if (xxch && nchans >= 3) {
235 av_log(s->avctx, AV_LOG_ERROR, "nchans %d is too large\n", nchans);
236 return AVERROR_INVALIDDATA;
237 } else if (nchans + base_channel > DCA_PRIM_CHANNELS_MAX) {
238 av_log(s->avctx, AV_LOG_ERROR, "channel sum %d + %d is too large\n", nchans, base_channel);
239 return AVERROR_INVALIDDATA;
242 s->audio_header.total_channels = nchans + base_channel;
243 s->audio_header.prim_channels = s->audio_header.total_channels;
245 /* obtain speaker layout mask & downmix coefficients for XXCH */
247 acc_mask = s->xxch_core_spkmask;
249 this_chans = get_bits(&s->gb, s->xxch_nbits_spk_mask - 6) << 6;
250 s->xxch_spk_masks[s->xxch_chset] = this_chans;
251 s->xxch_chset_nch[s->xxch_chset] = nchans;
253 for (i = 0; i <= s->xxch_chset; i++)
254 acc_mask |= s->xxch_spk_masks[i];
256 /* check for downmixing information */
257 if (get_bits1(&s->gb)) {
258 embedded_downmix = get_bits1(&s->gb);
259 coeff = get_bits(&s->gb, 6);
261 if (coeff<1 || coeff>61) {
262 av_log(s->avctx, AV_LOG_ERROR, "6bit coeff %d is out of range\n", coeff);
263 return AVERROR_INVALIDDATA;
266 scale_factor = -1.0f / dca_dmix_code((coeff<<2)-3);
268 s->xxch_dmix_sf[s->xxch_chset] = scale_factor;
270 for (i = base_channel; i < s->audio_header.prim_channels; i++) {
271 mask[i] = get_bits(&s->gb, s->xxch_nbits_spk_mask);
274 for (j = base_channel; j < s->audio_header.prim_channels; j++) {
275 memset(s->xxch_dmix_coeff[j], 0, sizeof(s->xxch_dmix_coeff[0]));
276 s->xxch_dmix_embedded |= (embedded_downmix << j);
277 for (i = 0; i < s->xxch_nbits_spk_mask; i++) {
278 if (mask[j] & (1 << i)) {
279 if ((1 << i) == DCA_XXCH_LFE1) {
280 av_log(s->avctx, AV_LOG_WARNING,
281 "DCA-XXCH: dmix to LFE1 not supported.\n");
285 coeff = get_bits(&s->gb, 7);
286 ichan = dca_xxch2index(s, 1 << i);
287 if ((coeff&63)<1 || (coeff&63)>61) {
288 av_log(s->avctx, AV_LOG_ERROR, "7bit coeff %d is out of range\n", coeff);
289 return AVERROR_INVALIDDATA;
291 s->xxch_dmix_coeff[j][ichan] = dca_dmix_code((coeff<<2)-3);
298 if (s->audio_header.prim_channels > DCA_PRIM_CHANNELS_MAX)
299 s->audio_header.prim_channels = DCA_PRIM_CHANNELS_MAX;
301 for (i = base_channel; i < s->audio_header.prim_channels; i++) {
302 s->audio_header.subband_activity[i] = get_bits(&s->gb, 5) + 2;
303 if (s->audio_header.subband_activity[i] > DCA_SUBBANDS)
304 s->audio_header.subband_activity[i] = DCA_SUBBANDS;
306 for (i = base_channel; i < s->audio_header.prim_channels; i++) {
307 s->audio_header.vq_start_subband[i] = get_bits(&s->gb, 5) + 1;
308 if (s->audio_header.vq_start_subband[i] > DCA_SUBBANDS)
309 s->audio_header.vq_start_subband[i] = DCA_SUBBANDS;
311 get_array(&s->gb, s->audio_header.joint_intensity + base_channel,
312 s->audio_header.prim_channels - base_channel, 3);
313 get_array(&s->gb, s->audio_header.transient_huffman + base_channel,
314 s->audio_header.prim_channels - base_channel, 2);
315 get_array(&s->gb, s->audio_header.scalefactor_huffman + base_channel,
316 s->audio_header.prim_channels - base_channel, 3);
317 get_array(&s->gb, s->audio_header.bitalloc_huffman + base_channel,
318 s->audio_header.prim_channels - base_channel, 3);
320 /* Get codebooks quantization indexes */
322 memset(s->audio_header.quant_index_huffman, 0, sizeof(s->audio_header.quant_index_huffman));
323 for (j = 1; j < 11; j++)
324 for (i = base_channel; i < s->audio_header.prim_channels; i++)
325 s->audio_header.quant_index_huffman[i][j] = get_bits(&s->gb, bitlen[j]);
327 /* Get scale factor adjustment */
328 for (j = 0; j < 11; j++)
329 for (i = base_channel; i < s->audio_header.prim_channels; i++)
330 s->audio_header.scalefactor_adj[i][j] = 1;
332 for (j = 1; j < 11; j++)
333 for (i = base_channel; i < s->audio_header.prim_channels; i++)
334 if (s->audio_header.quant_index_huffman[i][j] < thr[j])
335 s->audio_header.scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)];
338 if (s->crc_present) {
339 /* Audio header CRC check */
340 get_bits(&s->gb, 16);
343 /* Skip to the end of the header, also ignore CRC if present */
344 i = get_bits_count(&s->gb);
345 if (hdr_pos + 8 * hdr_size > i)
346 skip_bits_long(&s->gb, hdr_pos + 8 * hdr_size - i);
349 s->current_subframe = 0;
350 s->current_subsubframe = 0;
355 static int dca_parse_frame_header(DCAContext *s)
357 init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
360 skip_bits_long(&s->gb, 32);
363 s->frame_type = get_bits(&s->gb, 1);
364 s->samples_deficit = get_bits(&s->gb, 5) + 1;
365 s->crc_present = get_bits(&s->gb, 1);
366 s->sample_blocks = get_bits(&s->gb, 7) + 1;
367 s->frame_size = get_bits(&s->gb, 14) + 1;
368 if (s->frame_size < 95)
369 return AVERROR_INVALIDDATA;
370 s->amode = get_bits(&s->gb, 6);
371 s->sample_rate = avpriv_dca_sample_rates[get_bits(&s->gb, 4)];
373 return AVERROR_INVALIDDATA;
374 s->bit_rate_index = get_bits(&s->gb, 5);
375 s->bit_rate = ff_dca_bit_rates[s->bit_rate_index];
377 return AVERROR_INVALIDDATA;
379 skip_bits1(&s->gb); // always 0 (reserved, cf. ETSI TS 102 114 V1.4.1)
380 s->dynrange = get_bits(&s->gb, 1);
381 s->timestamp = get_bits(&s->gb, 1);
382 s->aux_data = get_bits(&s->gb, 1);
383 s->hdcd = get_bits(&s->gb, 1);
384 s->ext_descr = get_bits(&s->gb, 3);
385 s->ext_coding = get_bits(&s->gb, 1);
386 s->aspf = get_bits(&s->gb, 1);
387 s->lfe = get_bits(&s->gb, 2);
388 s->predictor_history = get_bits(&s->gb, 1);
392 av_log(s->avctx, AV_LOG_ERROR, "Invalid LFE value: %d\n", s->lfe);
393 return AVERROR_INVALIDDATA;
396 /* TODO: check CRC */
398 s->header_crc = get_bits(&s->gb, 16);
400 s->multirate_inter = get_bits(&s->gb, 1);
401 s->version = get_bits(&s->gb, 4);
402 s->copy_history = get_bits(&s->gb, 2);
403 s->source_pcm_res = get_bits(&s->gb, 3);
404 s->front_sum = get_bits(&s->gb, 1);
405 s->surround_sum = get_bits(&s->gb, 1);
406 s->dialog_norm = get_bits(&s->gb, 4);
408 /* FIXME: channels mixing levels */
409 s->output = s->amode;
411 s->output |= DCA_LFE;
413 /* Primary audio coding header */
414 s->audio_header.subframes = get_bits(&s->gb, 4) + 1;
416 return dca_parse_audio_coding_header(s, 0, 0);
419 static inline int get_scale(GetBitContext *gb, int level, int value, int log2range)
422 /* huffman encoded */
423 value += get_bitalloc(gb, &dca_scalefactor, level);
424 value = av_clip(value, 0, (1 << log2range) - 1);
425 } else if (level < 8) {
426 if (level + 1 > log2range) {
427 skip_bits(gb, level + 1 - log2range);
428 value = get_bits(gb, log2range);
430 value = get_bits(gb, level + 1);
436 static int dca_subframe_header(DCAContext *s, int base_channel, int block_index)
438 /* Primary audio coding side information */
441 if (get_bits_left(&s->gb) < 0)
442 return AVERROR_INVALIDDATA;
445 s->subsubframes[s->current_subframe] = get_bits(&s->gb, 2) + 1;
446 if (block_index + s->subsubframes[s->current_subframe] > (s->sample_blocks / SAMPLES_PER_SUBBAND)) {
447 s->subsubframes[s->current_subframe] = 1;
448 return AVERROR_INVALIDDATA;
450 s->partial_samples[s->current_subframe] = get_bits(&s->gb, 3);
453 for (j = base_channel; j < s->audio_header.prim_channels; j++) {
454 for (k = 0; k < s->audio_header.subband_activity[j]; k++)
455 s->dca_chan[j].prediction_mode[k] = get_bits(&s->gb, 1);
458 /* Get prediction codebook */
459 for (j = base_channel; j < s->audio_header.prim_channels; j++) {
460 for (k = 0; k < s->audio_header.subband_activity[j]; k++) {
461 if (s->dca_chan[j].prediction_mode[k] > 0) {
462 /* (Prediction coefficient VQ address) */
463 s->dca_chan[j].prediction_vq[k] = get_bits(&s->gb, 12);
468 /* Bit allocation index */
469 for (j = base_channel; j < s->audio_header.prim_channels; j++) {
470 for (k = 0; k < s->audio_header.vq_start_subband[j]; k++) {
471 if (s->audio_header.bitalloc_huffman[j] == 6)
472 s->dca_chan[j].bitalloc[k] = get_bits(&s->gb, 5);
473 else if (s->audio_header.bitalloc_huffman[j] == 5)
474 s->dca_chan[j].bitalloc[k] = get_bits(&s->gb, 4);
475 else if (s->audio_header.bitalloc_huffman[j] == 7) {
476 av_log(s->avctx, AV_LOG_ERROR,
477 "Invalid bit allocation index\n");
478 return AVERROR_INVALIDDATA;
480 s->dca_chan[j].bitalloc[k] =
481 get_bitalloc(&s->gb, &dca_bitalloc_index, s->audio_header.bitalloc_huffman[j]);
484 if (s->dca_chan[j].bitalloc[k] > 26) {
485 ff_dlog(s->avctx, "bitalloc index [%i][%i] too big (%i)\n",
486 j, k, s->dca_chan[j].bitalloc[k]);
487 return AVERROR_INVALIDDATA;
492 /* Transition mode */
493 for (j = base_channel; j < s->audio_header.prim_channels; j++) {
494 for (k = 0; k < s->audio_header.subband_activity[j]; k++) {
495 s->dca_chan[j].transition_mode[k] = 0;
496 if (s->subsubframes[s->current_subframe] > 1 &&
497 k < s->audio_header.vq_start_subband[j] && s->dca_chan[j].bitalloc[k] > 0) {
498 s->dca_chan[j].transition_mode[k] =
499 get_bitalloc(&s->gb, &dca_tmode, s->audio_header.transient_huffman[j]);
504 if (get_bits_left(&s->gb) < 0)
505 return AVERROR_INVALIDDATA;
507 for (j = base_channel; j < s->audio_header.prim_channels; j++) {
508 const uint32_t *scale_table;
509 int scale_sum, log_size;
511 memset(s->dca_chan[j].scale_factor, 0,
512 s->audio_header.subband_activity[j] * sizeof(s->dca_chan[j].scale_factor[0][0]) * 2);
514 if (s->audio_header.scalefactor_huffman[j] == 6) {
515 scale_table = ff_dca_scale_factor_quant7;
518 scale_table = ff_dca_scale_factor_quant6;
522 /* When huffman coded, only the difference is encoded */
525 for (k = 0; k < s->audio_header.subband_activity[j]; k++) {
526 if (k >= s->audio_header.vq_start_subband[j] || s->dca_chan[j].bitalloc[k] > 0) {
527 scale_sum = get_scale(&s->gb, s->audio_header.scalefactor_huffman[j], scale_sum, log_size);
528 s->dca_chan[j].scale_factor[k][0] = scale_table[scale_sum];
531 if (k < s->audio_header.vq_start_subband[j] && s->dca_chan[j].transition_mode[k]) {
532 /* Get second scale factor */
533 scale_sum = get_scale(&s->gb, s->audio_header.scalefactor_huffman[j], scale_sum, log_size);
534 s->dca_chan[j].scale_factor[k][1] = scale_table[scale_sum];
539 /* Joint subband scale factor codebook select */
540 for (j = base_channel; j < s->audio_header.prim_channels; j++) {
541 /* Transmitted only if joint subband coding enabled */
542 if (s->audio_header.joint_intensity[j] > 0)
543 s->dca_chan[j].joint_huff = get_bits(&s->gb, 3);
546 if (get_bits_left(&s->gb) < 0)
547 return AVERROR_INVALIDDATA;
549 /* Scale factors for joint subband coding */
550 for (j = base_channel; j < s->audio_header.prim_channels; j++) {
553 /* Transmitted only if joint subband coding enabled */
554 if (s->audio_header.joint_intensity[j] > 0) {
556 source_channel = s->audio_header.joint_intensity[j] - 1;
558 /* When huffman coded, only the difference is encoded
559 * (is this valid as well for joint scales ???) */
561 for (k = s->audio_header.subband_activity[j];
562 k < s->audio_header.subband_activity[source_channel]; k++) {
563 scale = get_scale(&s->gb, s->dca_chan[j].joint_huff, 64 /* bias */, 7);
564 s->dca_chan[j].joint_scale_factor[k] = scale; /*joint_scale_table[scale]; */
567 if (!(s->debug_flag & 0x02)) {
568 av_log(s->avctx, AV_LOG_DEBUG,
569 "Joint stereo coding not supported\n");
570 s->debug_flag |= 0x02;
575 /* Dynamic range coefficient */
576 if (!base_channel && s->dynrange)
577 s->dynrange_coef = get_bits(&s->gb, 8);
579 /* Side information CRC check word */
580 if (s->crc_present) {
581 get_bits(&s->gb, 16);
585 * Primary audio data arrays
588 /* VQ encoded high frequency subbands */
589 for (j = base_channel; j < s->audio_header.prim_channels; j++)
590 for (k = s->audio_header.vq_start_subband[j]; k < s->audio_header.subband_activity[j]; k++)
591 /* 1 vector -> 32 samples */
592 s->dca_chan[j].high_freq_vq[k] = get_bits(&s->gb, 10);
594 /* Low frequency effect data */
595 if (!base_channel && s->lfe) {
598 int lfe_samples = 2 * s->lfe * (4 + block_index);
599 int lfe_end_sample = 2 * s->lfe * (4 + block_index + s->subsubframes[s->current_subframe]);
602 for (j = lfe_samples; j < lfe_end_sample; j++) {
603 /* Signed 8 bits int */
604 s->lfe_data[j] = get_sbits(&s->gb, 8);
607 /* Scale factor index */
608 quant7 = get_bits(&s->gb, 8);
610 avpriv_request_sample(s->avctx, "LFEScaleIndex larger than 127");
611 return AVERROR_INVALIDDATA;
613 s->lfe_scale_factor = ff_dca_scale_factor_quant7[quant7];
615 /* Quantization step size * scale factor */
616 lfe_scale = 0.035 * s->lfe_scale_factor;
618 for (j = lfe_samples; j < lfe_end_sample; j++)
619 s->lfe_data[j] *= lfe_scale;
625 static void qmf_32_subbands(DCAContext *s, int chans,
626 float samples_in[32][SAMPLES_PER_SUBBAND], float *samples_out,
629 const float *prCoeff;
631 int sb_act = s->audio_header.subband_activity[chans];
633 scale *= sqrt(1 / 8.0);
636 if (!s->multirate_inter) /* Non-perfect reconstruction */
637 prCoeff = ff_dca_fir_32bands_nonperfect;
638 else /* Perfect reconstruction */
639 prCoeff = ff_dca_fir_32bands_perfect;
641 s->dcadsp.qmf_32_subbands(samples_in, sb_act, &s->synth, &s->imdct,
642 s->dca_chan[chans].subband_fir_hist,
643 &s->dca_chan[chans].hist_index,
644 s->dca_chan[chans].subband_fir_noidea, prCoeff,
645 samples_out, s->raXin, scale);
648 static QMF64_table *qmf64_precompute(void)
651 QMF64_table *table = av_malloc(sizeof(*table));
655 for (i = 0; i < 32; i++)
656 for (j = 0; j < 32; j++)
657 table->dct4_coeff[i][j] = cos((2 * i + 1) * (2 * j + 1) * M_PI / 128);
658 for (i = 0; i < 32; i++)
659 for (j = 0; j < 32; j++)
660 table->dct2_coeff[i][j] = cos((2 * i + 1) * j * M_PI / 64);
662 /* FIXME: Is the factor 0.125 = 1/8 right? */
663 for (i = 0; i < 32; i++)
664 table->rcos[i] = 0.125 / cos((2 * i + 1) * M_PI / 256);
665 for (i = 0; i < 32; i++)
666 table->rsin[i] = -0.125 / sin((2 * i + 1) * M_PI / 256);
671 /* FIXME: Totally unoptimized. Based on the reference code and
672 * http://multimedia.cx/mirror/dca-transform.pdf, with guessed tweaks
673 * for doubling the size. */
674 static void qmf_64_subbands(DCAContext *s, int chans, float samples_in[64][SAMPLES_PER_SUBBAND],
675 float *samples_out, float scale)
679 float *raX = s->dca_chan[chans].subband_fir_hist;
680 float *raZ = s->dca_chan[chans].subband_fir_noidea;
681 unsigned i, j, k, subindex;
683 for (i = s->audio_header.subband_activity[chans]; i < 64; i++)
685 for (subindex = 0; subindex < SAMPLES_PER_SUBBAND; subindex++) {
686 for (i = 0; i < s->audio_header.subband_activity[chans]; i++)
687 raXin[i] = samples_in[i][subindex];
689 for (k = 0; k < 32; k++) {
691 for (i = 0; i < 32; i++)
692 A[k] += (raXin[2 * i] + raXin[2 * i + 1]) * s->qmf64_table->dct4_coeff[k][i];
694 for (k = 0; k < 32; k++) {
695 B[k] = raXin[0] * s->qmf64_table->dct2_coeff[k][0];
696 for (i = 1; i < 32; i++)
697 B[k] += (raXin[2 * i] + raXin[2 * i - 1]) * s->qmf64_table->dct2_coeff[k][i];
699 for (k = 0; k < 32; k++) {
700 raX[k] = s->qmf64_table->rcos[k] * (A[k] + B[k]);
701 raX[63 - k] = s->qmf64_table->rsin[k] * (A[k] - B[k]);
704 for (i = 0; i < 64; i++) {
706 for (j = 0; j < 1024; j += 128)
707 out += ff_dca_fir_64bands[j + i] * (raX[j + i] - raX[j + 63 - i]);
708 *samples_out++ = out * scale;
711 for (i = 0; i < 64; i++) {
713 for (j = 0; j < 1024; j += 128)
714 hist += ff_dca_fir_64bands[64 + j + i] * (-raX[i + j] - raX[j + 63 - i]);
719 /* FIXME: Make buffer circular, to avoid this move. */
720 memmove(raX + 64, raX, (1024 - 64) * sizeof(*raX));
724 static void lfe_interpolation_fir(DCAContext *s, const float *samples_in,
727 /* samples_in: An array holding decimated samples.
728 * Samples in current subframe starts from samples_in[0],
729 * while samples_in[-1], samples_in[-2], ..., stores samples
730 * from last subframe as history.
732 * samples_out: An array holding interpolated samples
736 const float *prCoeff;
739 /* Select decimation filter */
742 prCoeff = ff_dca_lfe_fir_128;
745 if (s->exss_ext_mask & DCA_EXT_EXSS_XLL)
746 prCoeff = ff_dca_lfe_xll_fir_64;
748 prCoeff = ff_dca_lfe_fir_64;
751 for (deciindex = 0; deciindex < 2 * s->lfe; deciindex++) {
752 s->dcadsp.lfe_fir[idx](samples_out, samples_in, prCoeff);
754 samples_out += 2 * 32 * (1 + idx);
758 /* downmixing routines */
759 #define MIX_REAR1(samples, s1, rs, coef) \
760 samples[0][i] += samples[s1][i] * coef[rs][0]; \
761 samples[1][i] += samples[s1][i] * coef[rs][1];
763 #define MIX_REAR2(samples, s1, s2, rs, coef) \
764 samples[0][i] += samples[s1][i] * coef[rs][0] + samples[s2][i] * coef[rs + 1][0]; \
765 samples[1][i] += samples[s1][i] * coef[rs][1] + samples[s2][i] * coef[rs + 1][1];
767 #define MIX_FRONT3(samples, coef) \
771 samples[0][i] = t * coef[0][0] + u * coef[1][0] + v * coef[2][0]; \
772 samples[1][i] = t * coef[0][1] + u * coef[1][1] + v * coef[2][1];
774 #define DOWNMIX_TO_STEREO(op1, op2) \
775 for (i = 0; i < 256; i++) { \
780 static void dca_downmix(float **samples, int srcfmt, int lfe_present,
781 float coef[DCA_PRIM_CHANNELS_MAX + 1][2],
782 const int8_t *channel_mapping)
784 int c, l, r, sl, sr, s;
791 av_log(NULL, AV_LOG_ERROR, "Not implemented!\n");
795 case DCA_STEREO_TOTAL:
796 case DCA_STEREO_SUMDIFF:
799 c = channel_mapping[0];
800 l = channel_mapping[1];
801 r = channel_mapping[2];
802 DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), );
805 s = channel_mapping[2];
806 DOWNMIX_TO_STEREO(MIX_REAR1(samples, s, 2, coef), );
809 c = channel_mapping[0];
810 l = channel_mapping[1];
811 r = channel_mapping[2];
812 s = channel_mapping[3];
813 DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
814 MIX_REAR1(samples, s, 3, coef));
817 sl = channel_mapping[2];
818 sr = channel_mapping[3];
819 DOWNMIX_TO_STEREO(MIX_REAR2(samples, sl, sr, 2, coef), );
822 c = channel_mapping[0];
823 l = channel_mapping[1];
824 r = channel_mapping[2];
825 sl = channel_mapping[3];
826 sr = channel_mapping[4];
827 DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
828 MIX_REAR2(samples, sl, sr, 3, coef));
832 int lf_buf = ff_dca_lfe_index[srcfmt];
833 int lf_idx = ff_dca_channels[srcfmt];
834 for (i = 0; i < 256; i++) {
835 samples[0][i] += samples[lf_buf][i] * coef[lf_idx][0];
836 samples[1][i] += samples[lf_buf][i] * coef[lf_idx][1];
841 #ifndef decode_blockcodes
842 /* Very compact version of the block code decoder that does not use table
843 * look-up but is slightly slower */
844 static int decode_blockcode(int code, int levels, int32_t *values)
847 int offset = (levels - 1) >> 1;
849 for (i = 0; i < 4; i++) {
850 int div = FASTDIV(code, levels);
851 values[i] = code - offset - div * levels;
858 static int decode_blockcodes(int code1, int code2, int levels, int32_t *values)
860 return decode_blockcode(code1, levels, values) |
861 decode_blockcode(code2, levels, values + 4);
865 static const uint8_t abits_sizes[7] = { 7, 10, 12, 13, 15, 17, 19 };
866 static const uint8_t abits_levels[7] = { 3, 5, 7, 9, 13, 17, 25 };
868 static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
871 int subsubframe = s->current_subsubframe;
873 const float *quant_step_table;
875 LOCAL_ALIGNED_16(int32_t, block, [SAMPLES_PER_SUBBAND * DCA_SUBBANDS]);
881 /* Select quantization step size table */
882 if (s->bit_rate_index == 0x1f)
883 quant_step_table = ff_dca_lossless_quant_d;
885 quant_step_table = ff_dca_lossy_quant_d;
887 for (k = base_channel; k < s->audio_header.prim_channels; k++) {
888 float (*subband_samples)[8] = s->dca_chan[k].subband_samples[block_index];
889 float rscale[DCA_SUBBANDS];
891 if (get_bits_left(&s->gb) < 0)
892 return AVERROR_INVALIDDATA;
894 for (l = 0; l < s->audio_header.vq_start_subband[k]; l++) {
897 /* Select the mid-tread linear quantizer */
898 int abits = s->dca_chan[k].bitalloc[l];
900 float quant_step_size = quant_step_table[abits];
903 * Determine quantization index code book and its type
906 /* Select quantization index code book */
907 int sel = s->audio_header.quant_index_huffman[k][abits];
910 * Extract bits from the bit stream
914 memset(block + SAMPLES_PER_SUBBAND * l, 0, SAMPLES_PER_SUBBAND * sizeof(block[0]));
916 /* Deal with transients */
917 int sfi = s->dca_chan[k].transition_mode[l] &&
918 subsubframe >= s->dca_chan[k].transition_mode[l];
919 rscale[l] = quant_step_size * s->dca_chan[k].scale_factor[l][sfi] *
920 s->audio_header.scalefactor_adj[k][sel];
922 if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table) {
925 int block_code1, block_code2, size, levels, err;
927 size = abits_sizes[abits - 1];
928 levels = abits_levels[abits - 1];
930 block_code1 = get_bits(&s->gb, size);
931 block_code2 = get_bits(&s->gb, size);
932 err = decode_blockcodes(block_code1, block_code2,
933 levels, block + SAMPLES_PER_SUBBAND * l);
935 av_log(s->avctx, AV_LOG_ERROR,
936 "ERROR: block code look-up failed\n");
937 return AVERROR_INVALIDDATA;
941 for (m = 0; m < SAMPLES_PER_SUBBAND; m++)
942 block[SAMPLES_PER_SUBBAND * l + m] = get_sbits(&s->gb, abits - 3);
946 for (m = 0; m < SAMPLES_PER_SUBBAND; m++)
947 block[SAMPLES_PER_SUBBAND * l + m] = get_bitalloc(&s->gb,
948 &dca_smpl_bitalloc[abits], sel);
953 s->fmt_conv.int32_to_float_fmul_array8(&s->fmt_conv, subband_samples[0],
954 block, rscale, SAMPLES_PER_SUBBAND * s->audio_header.vq_start_subband[k]);
956 for (l = 0; l < s->audio_header.vq_start_subband[k]; l++) {
959 * Inverse ADPCM if in prediction mode
961 if (s->dca_chan[k].prediction_mode[l]) {
963 if (s->predictor_history)
964 subband_samples[l][0] += (ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][0] *
965 s->dca_chan[k].subband_samples_hist[l][3] +
966 ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][1] *
967 s->dca_chan[k].subband_samples_hist[l][2] +
968 ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][2] *
969 s->dca_chan[k].subband_samples_hist[l][1] +
970 ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][3] *
971 s->dca_chan[k].subband_samples_hist[l][0]) *
973 for (m = 1; m < SAMPLES_PER_SUBBAND; m++) {
974 float sum = ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][0] *
975 subband_samples[l][m - 1];
976 for (n = 2; n <= 4; n++)
978 sum += ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][n - 1] *
979 subband_samples[l][m - n];
980 else if (s->predictor_history)
981 sum += ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][n - 1] *
982 s->dca_chan[k].subband_samples_hist[l][m - n + 4];
983 subband_samples[l][m] += sum * (1.0f / 8192);
988 /* Backup predictor history for adpcm */
989 for (l = 0; l < DCA_SUBBANDS; l++)
990 AV_COPY128(s->dca_chan[k].subband_samples_hist[l], &subband_samples[l][4]);
994 * Decode VQ encoded high frequencies
996 if (s->audio_header.subband_activity[k] > s->audio_header.vq_start_subband[k]) {
997 if (!(s->debug_flag & 0x01)) {
998 av_log(s->avctx, AV_LOG_DEBUG,
999 "Stream with high frequencies VQ coding\n");
1000 s->debug_flag |= 0x01;
1003 s->dcadsp.decode_hf(subband_samples, s->dca_chan[k].high_freq_vq,
1004 ff_dca_high_freq_vq, subsubframe * SAMPLES_PER_SUBBAND,
1005 s->dca_chan[k].scale_factor,
1006 s->audio_header.vq_start_subband[k],
1007 s->audio_header.subband_activity[k]);
1011 /* Check for DSYNC after subsubframe */
1012 if (s->aspf || subsubframe == s->subsubframes[s->current_subframe] - 1) {
1013 if (get_bits(&s->gb, 16) != 0xFFFF) {
1014 av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n");
1015 return AVERROR_INVALIDDATA;
1022 static int dca_filter_channels(DCAContext *s, int block_index, int upsample)
1027 if (!s->qmf64_table) {
1028 s->qmf64_table = qmf64_precompute();
1029 if (!s->qmf64_table)
1030 return AVERROR(ENOMEM);
1033 /* 64 subbands QMF */
1034 for (k = 0; k < s->audio_header.prim_channels; k++) {
1035 float (*subband_samples)[SAMPLES_PER_SUBBAND] = s->dca_chan[k].subband_samples[block_index];
1037 if (s->channel_order_tab[k] >= 0)
1038 qmf_64_subbands(s, k, subband_samples,
1039 s->samples_chanptr[s->channel_order_tab[k]],
1040 /* Upsampling needs a factor 2 here. */
1044 /* 32 subbands QMF */
1045 for (k = 0; k < s->audio_header.prim_channels; k++) {
1046 float (*subband_samples)[SAMPLES_PER_SUBBAND] = s->dca_chan[k].subband_samples[block_index];
1048 if (s->channel_order_tab[k] >= 0)
1049 qmf_32_subbands(s, k, subband_samples,
1050 s->samples_chanptr[s->channel_order_tab[k]],
1051 M_SQRT1_2 / 32768.0);
1055 /* Generate LFE samples for this subsubframe FIXME!!! */
1057 float *samples = s->samples_chanptr[s->lfe_index];
1058 lfe_interpolation_fir(s,
1059 s->lfe_data + 2 * s->lfe * (block_index + 4),
1063 /* Should apply the filter in Table 6-11 when upsampling. For
1064 * now, just duplicate. */
1065 for (i = 255; i > 0; i--) {
1067 samples[2 * i + 1] = samples[i];
1069 samples[1] = samples[0];
1073 /* FIXME: This downmixing is probably broken with upsample.
1074 * Probably totally broken also with XLL in general. */
1075 /* Downmixing to Stereo */
1076 if (s->audio_header.prim_channels + !!s->lfe > 2 &&
1077 s->avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
1078 dca_downmix(s->samples_chanptr, s->amode, !!s->lfe, s->downmix_coef,
1079 s->channel_order_tab);
1085 static int dca_subframe_footer(DCAContext *s, int base_channel)
1087 int in, out, aux_data_count, aux_data_end, reserved;
1091 * Unpack optional information
1094 /* presumably optional information only appears in the core? */
1095 if (!base_channel) {
1097 skip_bits_long(&s->gb, 32);
1100 aux_data_count = get_bits(&s->gb, 6);
1103 skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
1105 aux_data_end = 8 * aux_data_count + get_bits_count(&s->gb);
1107 if ((nsyncaux = get_bits_long(&s->gb, 32)) != DCA_NSYNCAUX) {
1108 av_log(s->avctx, AV_LOG_ERROR, "nSYNCAUX mismatch %#"PRIx32"\n",
1110 return AVERROR_INVALIDDATA;
1113 if (get_bits1(&s->gb)) { // bAUXTimeStampFlag
1114 avpriv_request_sample(s->avctx,
1115 "Auxiliary Decode Time Stamp Flag");
1117 skip_bits(&s->gb, (-get_bits_count(&s->gb)) & 4);
1118 // 44 bits: nMSByte (8), nMarker (4), nLSByte (28), nMarker (4)
1119 skip_bits_long(&s->gb, 44);
1122 if ((s->core_downmix = get_bits1(&s->gb))) {
1123 int am = get_bits(&s->gb, 3);
1126 s->core_downmix_amode = DCA_MONO;
1129 s->core_downmix_amode = DCA_STEREO;
1132 s->core_downmix_amode = DCA_STEREO_TOTAL;
1135 s->core_downmix_amode = DCA_3F;
1138 s->core_downmix_amode = DCA_2F1R;
1141 s->core_downmix_amode = DCA_2F2R;
1144 s->core_downmix_amode = DCA_3F1R;
1147 av_log(s->avctx, AV_LOG_ERROR,
1148 "Invalid mode %d for embedded downmix coefficients\n",
1150 return AVERROR_INVALIDDATA;
1152 for (out = 0; out < ff_dca_channels[s->core_downmix_amode]; out++) {
1153 for (in = 0; in < s->audio_header.prim_channels + !!s->lfe; in++) {
1154 uint16_t tmp = get_bits(&s->gb, 9);
1155 if ((tmp & 0xFF) > 241) {
1156 av_log(s->avctx, AV_LOG_ERROR,
1157 "Invalid downmix coefficient code %"PRIu16"\n",
1159 return AVERROR_INVALIDDATA;
1161 s->core_downmix_codes[in][out] = tmp;
1166 align_get_bits(&s->gb); // byte align
1167 skip_bits(&s->gb, 16); // nAUXCRC16
1170 * additional data (reserved, cf. ETSI TS 102 114 V1.4.1)
1172 * Note: don't check for overreads, aux_data_count can't be trusted.
1174 if ((reserved = (aux_data_end - get_bits_count(&s->gb))) > 0) {
1175 avpriv_request_sample(s->avctx,
1176 "Core auxiliary data reserved content");
1177 skip_bits_long(&s->gb, reserved);
1181 if (s->crc_present && s->dynrange)
1182 get_bits(&s->gb, 16);
1189 * Decode a dca frame block
1191 * @param s pointer to the DCAContext
1194 static int dca_decode_block(DCAContext *s, int base_channel, int block_index)
1199 if (s->current_subframe >= s->audio_header.subframes) {
1200 av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i",
1201 s->current_subframe, s->audio_header.subframes);
1202 return AVERROR_INVALIDDATA;
1205 if (!s->current_subsubframe) {
1206 /* Read subframe header */
1207 if ((ret = dca_subframe_header(s, base_channel, block_index)))
1211 /* Read subsubframe */
1212 if ((ret = dca_subsubframe(s, base_channel, block_index)))
1216 s->current_subsubframe++;
1217 if (s->current_subsubframe >= s->subsubframes[s->current_subframe]) {
1218 s->current_subsubframe = 0;
1219 s->current_subframe++;
1221 if (s->current_subframe >= s->audio_header.subframes) {
1222 /* Read subframe footer */
1223 if ((ret = dca_subframe_footer(s, base_channel)))
1230 int ff_dca_xbr_parse_frame(DCAContext *s)
1232 int scale_table_high[DCA_CHSET_CHANS_MAX][DCA_SUBBANDS][2];
1233 int active_bands[DCA_CHSETS_MAX][DCA_CHSET_CHANS_MAX];
1234 int abits_high[DCA_CHSET_CHANS_MAX][DCA_SUBBANDS];
1235 int anctemp[DCA_CHSET_CHANS_MAX];
1236 int chset_fsize[DCA_CHSETS_MAX];
1237 int n_xbr_ch[DCA_CHSETS_MAX];
1238 int hdr_size, num_chsets, xbr_tmode, hdr_pos;
1239 int i, j, k, l, chset, chan_base;
1241 av_log(s->avctx, AV_LOG_DEBUG, "DTS-XBR: decoding XBR extension\n");
1243 /* get bit position of sync header */
1244 hdr_pos = get_bits_count(&s->gb) - 32;
1246 hdr_size = get_bits(&s->gb, 6) + 1;
1247 num_chsets = get_bits(&s->gb, 2) + 1;
1249 for(i = 0; i < num_chsets; i++)
1250 chset_fsize[i] = get_bits(&s->gb, 14) + 1;
1252 xbr_tmode = get_bits1(&s->gb);
1254 for(i = 0; i < num_chsets; i++) {
1255 n_xbr_ch[i] = get_bits(&s->gb, 3) + 1;
1256 k = get_bits(&s->gb, 2) + 5;
1257 for(j = 0; j < n_xbr_ch[i]; j++) {
1258 active_bands[i][j] = get_bits(&s->gb, k) + 1;
1259 if (active_bands[i][j] > DCA_SUBBANDS) {
1260 av_log(s->avctx, AV_LOG_ERROR, "too many active subbands (%d)\n", active_bands[i][j]);
1261 return AVERROR_INVALIDDATA;
1266 /* skip to the end of the header */
1267 i = get_bits_count(&s->gb);
1268 if(hdr_pos + hdr_size * 8 > i)
1269 skip_bits_long(&s->gb, hdr_pos + hdr_size * 8 - i);
1271 /* loop over the channel data sets */
1272 /* only decode as many channels as we've decoded base data for */
1273 for(chset = 0, chan_base = 0;
1274 chset < num_chsets && chan_base + n_xbr_ch[chset] <= s->audio_header.prim_channels;
1275 chan_base += n_xbr_ch[chset++]) {
1276 int start_posn = get_bits_count(&s->gb);
1277 int subsubframe = 0;
1280 /* loop over subframes */
1281 for (k = 0; k < (s->sample_blocks / 8); k++) {
1282 /* parse header if we're on first subsubframe of a block */
1283 if(subsubframe == 0) {
1284 /* Parse subframe header */
1285 for(i = 0; i < n_xbr_ch[chset]; i++) {
1286 anctemp[i] = get_bits(&s->gb, 2) + 2;
1289 for(i = 0; i < n_xbr_ch[chset]; i++) {
1290 get_array(&s->gb, abits_high[i], active_bands[chset][i], anctemp[i]);
1293 for(i = 0; i < n_xbr_ch[chset]; i++) {
1294 anctemp[i] = get_bits(&s->gb, 3);
1295 if(anctemp[i] < 1) {
1296 av_log(s->avctx, AV_LOG_ERROR, "DTS-XBR: SYNC ERROR\n");
1297 return AVERROR_INVALIDDATA;
1301 /* generate scale factors */
1302 for(i = 0; i < n_xbr_ch[chset]; i++) {
1303 const uint32_t *scale_table;
1305 int scale_table_size;
1307 if (s->audio_header.scalefactor_huffman[chan_base+i] == 6) {
1308 scale_table = ff_dca_scale_factor_quant7;
1309 scale_table_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant7);
1311 scale_table = ff_dca_scale_factor_quant6;
1312 scale_table_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant6);
1317 for(j = 0; j < active_bands[chset][i]; j++) {
1318 if(abits_high[i][j] > 0) {
1319 int index = get_bits(&s->gb, nbits);
1320 if (index >= scale_table_size) {
1321 av_log(s->avctx, AV_LOG_ERROR, "scale table index %d invalid\n", index);
1322 return AVERROR_INVALIDDATA;
1324 scale_table_high[i][j][0] = scale_table[index];
1326 if(xbr_tmode && s->dca_chan[i].transition_mode[j]) {
1327 int index = get_bits(&s->gb, nbits);
1328 if (index >= scale_table_size) {
1329 av_log(s->avctx, AV_LOG_ERROR, "scale table index %d invalid\n", index);
1330 return AVERROR_INVALIDDATA;
1332 scale_table_high[i][j][1] = scale_table[index];
1339 /* decode audio array for this block */
1340 for(i = 0; i < n_xbr_ch[chset]; i++) {
1341 for(j = 0; j < active_bands[chset][i]; j++) {
1342 const int xbr_abits = abits_high[i][j];
1343 const float quant_step_size = ff_dca_lossless_quant_d[xbr_abits];
1344 const int sfi = xbr_tmode && s->dca_chan[i].transition_mode[j] && subsubframe >= s->dca_chan[i].transition_mode[j];
1345 const float rscale = quant_step_size * scale_table_high[i][j][sfi];
1346 float *subband_samples = s->dca_chan[chan_base+i].subband_samples[k][j];
1353 get_array(&s->gb, block, 8, xbr_abits - 3);
1355 int block_code1, block_code2, size, levels, err;
1357 size = abits_sizes[xbr_abits - 1];
1358 levels = abits_levels[xbr_abits - 1];
1360 block_code1 = get_bits(&s->gb, size);
1361 block_code2 = get_bits(&s->gb, size);
1362 err = decode_blockcodes(block_code1, block_code2,
1365 av_log(s->avctx, AV_LOG_ERROR,
1366 "ERROR: DTS-XBR: block code look-up failed\n");
1367 return AVERROR_INVALIDDATA;
1371 /* scale & sum into subband */
1372 for(l = 0; l < 8; l++)
1373 subband_samples[l] += (float)block[l] * rscale;
1377 /* check DSYNC marker */
1378 if(s->aspf || subsubframe == s->subsubframes[subframe] - 1) {
1379 if(get_bits(&s->gb, 16) != 0xffff) {
1380 av_log(s->avctx, AV_LOG_ERROR, "DTS-XBR: Didn't get subframe DSYNC\n");
1381 return AVERROR_INVALIDDATA;
1385 /* advance sub-sub-frame index */
1386 if(++subsubframe >= s->subsubframes[subframe]) {
1392 /* skip to next channel set */
1393 i = get_bits_count(&s->gb);
1394 if(start_posn + chset_fsize[chset] * 8 != i) {
1395 j = start_posn + chset_fsize[chset] * 8 - i;
1397 av_log(s->avctx, AV_LOG_ERROR, "DTS-XBR: end of channel set,"
1398 " skipping further than expected (%d bits)\n", j);
1399 skip_bits_long(&s->gb, j);
1407 /* parse initial header for XXCH and dump details */
1408 int ff_dca_xxch_decode_frame(DCAContext *s)
1410 int hdr_size, spkmsk_bits, num_chsets, core_spk, hdr_pos;
1411 int i, chset, base_channel, chstart, fsize[8];
1413 /* assume header word has already been parsed */
1414 hdr_pos = get_bits_count(&s->gb) - 32;
1415 hdr_size = get_bits(&s->gb, 6) + 1;
1416 /*chhdr_crc =*/ skip_bits1(&s->gb);
1417 spkmsk_bits = get_bits(&s->gb, 5) + 1;
1418 num_chsets = get_bits(&s->gb, 2) + 1;
1420 for (i = 0; i < num_chsets; i++)
1421 fsize[i] = get_bits(&s->gb, 14) + 1;
1423 core_spk = get_bits(&s->gb, spkmsk_bits);
1424 s->xxch_core_spkmask = core_spk;
1425 s->xxch_nbits_spk_mask = spkmsk_bits;
1426 s->xxch_dmix_embedded = 0;
1428 /* skip to the end of the header */
1429 i = get_bits_count(&s->gb);
1430 if (hdr_pos + hdr_size * 8 > i)
1431 skip_bits_long(&s->gb, hdr_pos + hdr_size * 8 - i);
1433 for (chset = 0; chset < num_chsets; chset++) {
1434 chstart = get_bits_count(&s->gb);
1435 base_channel = s->audio_header.prim_channels;
1436 s->xxch_chset = chset;
1438 /* XXCH and Core headers differ, see 6.4.2 "XXCH Channel Set Header" vs.
1439 5.3.2 "Primary Audio Coding Header", DTS Spec 1.3.1 */
1440 dca_parse_audio_coding_header(s, base_channel, 1);
1442 /* decode channel data */
1443 for (i = 0; i < (s->sample_blocks / 8); i++) {
1444 if (dca_decode_block(s, base_channel, i)) {
1445 av_log(s->avctx, AV_LOG_ERROR,
1446 "Error decoding DTS-XXCH extension\n");
1451 /* skip to end of this section */
1452 i = get_bits_count(&s->gb);
1453 if (chstart + fsize[chset] * 8 > i)
1454 skip_bits_long(&s->gb, chstart + fsize[chset] * 8 - i);
1456 s->xxch_chset = num_chsets;
1461 static float dca_dmix_code(unsigned code)
1463 int sign = (code >> 8) - 1;
1465 return ((ff_dca_dmixtable[code] ^ sign) - sign) * (1.0 / (1 << 15));
1468 static int scan_for_extensions(AVCodecContext *avctx)
1470 DCAContext *s = avctx->priv_data;
1471 int core_ss_end, ret = 0;
1473 core_ss_end = FFMIN(s->frame_size, s->dca_buffer_size) * 8;
1475 /* only scan for extensions if ext_descr was unknown or indicated a
1476 * supported XCh extension */
1477 if (s->core_ext_mask < 0 || s->core_ext_mask & (DCA_EXT_XCH | DCA_EXT_XXCH)) {
1478 /* if ext_descr was unknown, clear s->core_ext_mask so that the
1479 * extensions scan can fill it up */
1480 s->core_ext_mask = FFMAX(s->core_ext_mask, 0);
1482 /* extensions start at 32-bit boundaries into bitstream */
1483 skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
1485 while (core_ss_end - get_bits_count(&s->gb) >= 32) {
1486 uint32_t bits = get_bits_long(&s->gb, 32);
1490 case DCA_SYNCWORD_XCH: {
1491 int ext_amode, xch_fsize;
1493 s->xch_base_channel = s->audio_header.prim_channels;
1495 /* validate sync word using XCHFSIZE field */
1496 xch_fsize = show_bits(&s->gb, 10);
1497 if ((s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize) &&
1498 (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize + 1))
1501 /* skip length-to-end-of-frame field for the moment */
1502 skip_bits(&s->gb, 10);
1504 s->core_ext_mask |= DCA_EXT_XCH;
1506 /* extension amode(number of channels in extension) should be 1 */
1507 /* AFAIK XCh is not used for more channels */
1508 if ((ext_amode = get_bits(&s->gb, 4)) != 1) {
1509 av_log(avctx, AV_LOG_ERROR,
1510 "XCh extension amode %d not supported!\n",
1515 if (s->xch_base_channel < 2) {
1516 avpriv_request_sample(avctx, "XCh with fewer than 2 base channels");
1520 /* much like core primary audio coding header */
1521 dca_parse_audio_coding_header(s, s->xch_base_channel, 0);
1523 for (i = 0; i < (s->sample_blocks / 8); i++)
1524 if ((ret = dca_decode_block(s, s->xch_base_channel, i))) {
1525 av_log(avctx, AV_LOG_ERROR, "error decoding XCh extension\n");
1532 case DCA_SYNCWORD_XXCH:
1533 /* XXCh: extended channels */
1534 /* usually found either in core or HD part in DTS-HD HRA streams,
1535 * but not in DTS-ES which contains XCh extensions instead */
1536 s->core_ext_mask |= DCA_EXT_XXCH;
1537 ff_dca_xxch_decode_frame(s);
1541 int fsize96 = show_bits(&s->gb, 12) + 1;
1542 if (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + fsize96)
1545 av_log(avctx, AV_LOG_DEBUG, "X96 extension found at %d bits\n",
1546 get_bits_count(&s->gb));
1547 skip_bits(&s->gb, 12);
1548 av_log(avctx, AV_LOG_DEBUG, "FSIZE96 = %d bytes\n", fsize96);
1549 av_log(avctx, AV_LOG_DEBUG, "REVNO = %d\n", get_bits(&s->gb, 4));
1551 s->core_ext_mask |= DCA_EXT_X96;
1556 skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
1559 /* no supported extensions, skip the rest of the core substream */
1560 skip_bits_long(&s->gb, core_ss_end - get_bits_count(&s->gb));
1563 if (s->core_ext_mask & DCA_EXT_X96)
1564 s->profile = FF_PROFILE_DTS_96_24;
1565 else if (s->core_ext_mask & (DCA_EXT_XCH | DCA_EXT_XXCH))
1566 s->profile = FF_PROFILE_DTS_ES;
1568 /* check for ExSS (HD part) */
1569 if (s->dca_buffer_size - s->frame_size > 32 &&
1570 get_bits_long(&s->gb, 32) == DCA_SYNCWORD_SUBSTREAM)
1571 ff_dca_exss_parse_header(s);
1576 static int set_channel_layout(AVCodecContext *avctx, int *channels, int num_core_channels)
1578 DCAContext *s = avctx->priv_data;
1579 int i, j, chset, mask;
1580 int channel_layout, channel_mask;
1583 /* If we have XXCH then the channel layout is managed differently */
1584 /* note that XLL will also have another way to do things */
1585 if (!(s->core_ext_mask & DCA_EXT_XXCH)) {
1586 /* xxx should also do MA extensions */
1587 if (s->amode < 16) {
1588 avctx->channel_layout = ff_dca_core_channel_layout[s->amode];
1590 if (s->audio_header.prim_channels + !!s->lfe > 2 &&
1591 avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
1593 * Neither the core's auxiliary data nor our default tables contain
1594 * downmix coefficients for the additional channel coded in the XCh
1595 * extension, so when we're doing a Stereo downmix, don't decode it.
1600 if (s->xch_present && !s->xch_disable) {
1601 if (avctx->channel_layout & AV_CH_BACK_CENTER) {
1602 avpriv_request_sample(avctx, "XCh with Back center channel");
1603 return AVERROR_INVALIDDATA;
1605 avctx->channel_layout |= AV_CH_BACK_CENTER;
1607 avctx->channel_layout |= AV_CH_LOW_FREQUENCY;
1608 s->channel_order_tab = ff_dca_channel_reorder_lfe_xch[s->amode];
1610 s->channel_order_tab = ff_dca_channel_reorder_nolfe_xch[s->amode];
1612 if (s->channel_order_tab[s->xch_base_channel] < 0)
1613 return AVERROR_INVALIDDATA;
1615 *channels = num_core_channels + !!s->lfe;
1616 s->xch_present = 0; /* disable further xch processing */
1618 avctx->channel_layout |= AV_CH_LOW_FREQUENCY;
1619 s->channel_order_tab = ff_dca_channel_reorder_lfe[s->amode];
1621 s->channel_order_tab = ff_dca_channel_reorder_nolfe[s->amode];
1624 if (*channels > !!s->lfe &&
1625 s->channel_order_tab[*channels - 1 - !!s->lfe] < 0)
1626 return AVERROR_INVALIDDATA;
1628 if (av_get_channel_layout_nb_channels(avctx->channel_layout) != *channels) {
1629 av_log(avctx, AV_LOG_ERROR, "Number of channels %d mismatches layout %d\n", *channels, av_get_channel_layout_nb_channels(avctx->channel_layout));
1630 return AVERROR_INVALIDDATA;
1633 if (num_core_channels + !!s->lfe > 2 &&
1634 avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
1636 s->output = s->audio_header.prim_channels == 2 ? s->amode : DCA_STEREO;
1637 avctx->channel_layout = AV_CH_LAYOUT_STEREO;
1639 else if (avctx->request_channel_layout & AV_CH_LAYOUT_NATIVE) {
1640 static const int8_t dca_channel_order_native[9] = { 0, 1, 2, 3, 4, 5, 6, 7, 8 };
1641 s->channel_order_tab = dca_channel_order_native;
1643 s->lfe_index = ff_dca_lfe_index[s->amode];
1645 av_log(avctx, AV_LOG_ERROR,
1646 "Non standard configuration %d !\n", s->amode);
1647 return AVERROR_INVALIDDATA;
1650 s->xxch_dmix_embedded = 0;
1652 /* we only get here if an XXCH channel set can be added to the mix */
1653 channel_mask = s->xxch_core_spkmask;
1656 *channels = s->audio_header.prim_channels + !!s->lfe;
1657 for (i = 0; i < s->xxch_chset; i++) {
1658 channel_mask |= s->xxch_spk_masks[i];
1662 /* Given the DTS spec'ed channel mask, generate an avcodec version */
1664 for (i = 0; i < s->xxch_nbits_spk_mask; ++i) {
1665 if (channel_mask & (1 << i)) {
1666 channel_layout |= ff_dca_map_xxch_to_native[i];
1670 /* make sure that we have managed to get equivalent dts/avcodec channel
1671 * masks in some sense -- unfortunately some channels could overlap */
1672 if (av_popcount(channel_mask) != av_popcount(channel_layout)) {
1673 av_log(avctx, AV_LOG_DEBUG,
1674 "DTS-XXCH: Inconsistent avcodec/dts channel layouts\n");
1675 return AVERROR_INVALIDDATA;
1678 avctx->channel_layout = channel_layout;
1680 if (!(avctx->request_channel_layout & AV_CH_LAYOUT_NATIVE)) {
1681 /* Estimate DTS --> avcodec ordering table */
1682 for (chset = -1, j = 0; chset < s->xxch_chset; ++chset) {
1683 mask = chset >= 0 ? s->xxch_spk_masks[chset]
1684 : s->xxch_core_spkmask;
1685 for (i = 0; i < s->xxch_nbits_spk_mask; i++) {
1686 if (mask & ~(DCA_XXCH_LFE1 | DCA_XXCH_LFE2) & (1 << i)) {
1687 lavc = ff_dca_map_xxch_to_native[i];
1688 posn = av_popcount(channel_layout & (lavc - 1));
1689 s->xxch_order_tab[j++] = posn;
1695 s->lfe_index = av_popcount(channel_layout & (AV_CH_LOW_FREQUENCY-1));
1696 } else { /* native ordering */
1697 for (i = 0; i < *channels; i++)
1698 s->xxch_order_tab[i] = i;
1700 s->lfe_index = *channels - 1;
1703 s->channel_order_tab = s->xxch_order_tab;
1710 * Main frame decoding function
1711 * FIXME add arguments
1713 static int dca_decode_frame(AVCodecContext *avctx, void *data,
1714 int *got_frame_ptr, AVPacket *avpkt)
1716 AVFrame *frame = data;
1717 const uint8_t *buf = avpkt->data;
1718 int buf_size = avpkt->size;
1720 int num_core_channels = 0;
1722 float **samples_flt;
1725 DCAContext *s = avctx->priv_data;
1726 int channels, full_channels;
1735 s->exss_ext_mask = 0;
1738 s->dca_buffer_size = AVERROR_INVALIDDATA;
1739 for (i = 0; i < buf_size - 3 && s->dca_buffer_size == AVERROR_INVALIDDATA; i++)
1740 s->dca_buffer_size = avpriv_dca_convert_bitstream(buf + i, buf_size - i, s->dca_buffer,
1741 DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE);
1743 if (s->dca_buffer_size == AVERROR_INVALIDDATA) {
1744 av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n");
1745 return AVERROR_INVALIDDATA;
1748 if ((ret = dca_parse_frame_header(s)) < 0) {
1749 // seems like the frame is corrupt, try with the next one
1752 // set AVCodec values with parsed data
1753 avctx->sample_rate = s->sample_rate;
1755 s->profile = FF_PROFILE_DTS;
1757 for (i = 0; i < (s->sample_blocks / SAMPLES_PER_SUBBAND); i++) {
1758 if ((ret = dca_decode_block(s, 0, i))) {
1759 av_log(avctx, AV_LOG_ERROR, "error decoding block\n");
1764 /* record number of core channels incase less than max channels are requested */
1765 num_core_channels = s->audio_header.prim_channels;
1767 if (s->audio_header.prim_channels + !!s->lfe > 2 &&
1768 avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
1769 /* Stereo downmix coefficients
1771 * The decoder can only downmix to 2-channel, so we need to ensure
1772 * embedded downmix coefficients are actually targeting 2-channel.
1774 if (s->core_downmix && (s->core_downmix_amode == DCA_STEREO ||
1775 s->core_downmix_amode == DCA_STEREO_TOTAL)) {
1776 for (i = 0; i < num_core_channels + !!s->lfe; i++) {
1777 /* Range checked earlier */
1778 s->downmix_coef[i][0] = dca_dmix_code(s->core_downmix_codes[i][0]);
1779 s->downmix_coef[i][1] = dca_dmix_code(s->core_downmix_codes[i][1]);
1781 s->output = s->core_downmix_amode;
1783 int am = s->amode & DCA_CHANNEL_MASK;
1784 if (am >= FF_ARRAY_ELEMS(ff_dca_default_coeffs)) {
1785 av_log(s->avctx, AV_LOG_ERROR,
1786 "Invalid channel mode %d\n", am);
1787 return AVERROR_INVALIDDATA;
1789 if (num_core_channels + !!s->lfe >
1790 FF_ARRAY_ELEMS(ff_dca_default_coeffs[0])) {
1791 avpriv_request_sample(s->avctx, "Downmixing %d channels",
1792 s->audio_header.prim_channels + !!s->lfe);
1793 return AVERROR_PATCHWELCOME;
1795 for (i = 0; i < num_core_channels + !!s->lfe; i++) {
1796 s->downmix_coef[i][0] = ff_dca_default_coeffs[am][i][0];
1797 s->downmix_coef[i][1] = ff_dca_default_coeffs[am][i][1];
1800 ff_dlog(s->avctx, "Stereo downmix coeffs:\n");
1801 for (i = 0; i < num_core_channels + !!s->lfe; i++) {
1802 ff_dlog(s->avctx, "L, input channel %d = %f\n", i,
1803 s->downmix_coef[i][0]);
1804 ff_dlog(s->avctx, "R, input channel %d = %f\n", i,
1805 s->downmix_coef[i][1]);
1807 ff_dlog(s->avctx, "\n");
1811 s->core_ext_mask = ff_dca_ext_audio_descr_mask[s->ext_descr];
1813 s->core_ext_mask = 0;
1815 ret = scan_for_extensions(avctx);
1817 avctx->profile = s->profile;
1819 full_channels = channels = s->audio_header.prim_channels + !!s->lfe;
1821 ret = set_channel_layout(avctx, &channels, num_core_channels);
1825 /* get output buffer */
1826 frame->nb_samples = 256 * (s->sample_blocks / SAMPLES_PER_SUBBAND);
1827 if (s->exss_ext_mask & DCA_EXT_EXSS_XLL) {
1828 int xll_nb_samples = s->xll_segments * s->xll_smpl_in_seg;
1829 /* Check for invalid/unsupported conditions first */
1830 if (s->xll_residual_channels > channels) {
1831 av_log(s->avctx, AV_LOG_WARNING,
1832 "DCA: too many residual channels (%d, core channels %d). Disabling XLL\n",
1833 s->xll_residual_channels, channels);
1834 s->exss_ext_mask &= ~DCA_EXT_EXSS_XLL;
1835 } else if (xll_nb_samples != frame->nb_samples &&
1836 2 * frame->nb_samples != xll_nb_samples) {
1837 av_log(s->avctx, AV_LOG_WARNING,
1838 "DCA: unsupported upsampling (%d XLL samples, %d core samples). Disabling XLL\n",
1839 xll_nb_samples, frame->nb_samples);
1840 s->exss_ext_mask &= ~DCA_EXT_EXSS_XLL;
1842 if (2 * frame->nb_samples == xll_nb_samples) {
1843 av_log(s->avctx, AV_LOG_INFO,
1844 "XLL: upsampling core channels by a factor of 2\n");
1847 frame->nb_samples = xll_nb_samples;
1848 // FIXME: Is it good enough to copy from the first channel set?
1849 avctx->sample_rate = s->xll_chsets[0].sampling_frequency;
1851 /* If downmixing to stereo, don't decode additional channels.
1852 * FIXME: Using the xch_disable flag for this doesn't seem right. */
1853 if (!s->xch_disable)
1854 channels = s->xll_channels;
1858 if (avctx->channels != channels) {
1859 if (avctx->channels)
1860 av_log(avctx, AV_LOG_INFO, "Number of channels changed in DCA decoder (%d -> %d)\n", avctx->channels, channels);
1861 avctx->channels = channels;
1864 /* FIXME: This is an ugly hack, to just revert to the default
1865 * layout if we have additional channels. Need to convert the XLL
1866 * channel masks to ffmpeg channel_layout mask. */
1867 if (av_get_channel_layout_nb_channels(avctx->channel_layout) != avctx->channels)
1868 avctx->channel_layout = 0;
1870 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
1872 samples_flt = (float **) frame->extended_data;
1874 /* allocate buffer for extra channels if downmixing */
1875 if (avctx->channels < full_channels) {
1876 ret = av_samples_get_buffer_size(NULL, full_channels - channels,
1878 avctx->sample_fmt, 0);
1882 av_fast_malloc(&s->extra_channels_buffer,
1883 &s->extra_channels_buffer_size, ret);
1884 if (!s->extra_channels_buffer)
1885 return AVERROR(ENOMEM);
1887 ret = av_samples_fill_arrays((uint8_t **) s->extra_channels, NULL,
1888 s->extra_channels_buffer,
1889 full_channels - channels,
1890 frame->nb_samples, avctx->sample_fmt, 0);
1895 /* filter to get final output */
1896 for (i = 0; i < (s->sample_blocks / SAMPLES_PER_SUBBAND); i++) {
1898 unsigned block = upsample ? 512 : 256;
1899 for (ch = 0; ch < channels; ch++)
1900 s->samples_chanptr[ch] = samples_flt[ch] + i * block;
1901 for (; ch < full_channels; ch++)
1902 s->samples_chanptr[ch] = s->extra_channels[ch - channels] + i * block;
1904 dca_filter_channels(s, i, upsample);
1906 /* If this was marked as a DTS-ES stream we need to subtract back- */
1907 /* channel from SL & SR to remove matrixed back-channel signal */
1908 if ((s->source_pcm_res & 1) && s->xch_present) {
1909 float *back_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel]];
1910 float *lt_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 2]];
1911 float *rt_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 1]];
1912 s->fdsp->vector_fmac_scalar(lt_chan, back_chan, -M_SQRT1_2, 256);
1913 s->fdsp->vector_fmac_scalar(rt_chan, back_chan, -M_SQRT1_2, 256);
1916 /* If stream contains XXCH, we might need to undo an embedded downmix */
1917 if (s->xxch_dmix_embedded) {
1918 /* Loop over channel sets in turn */
1919 ch = num_core_channels;
1920 for (chset = 0; chset < s->xxch_chset; chset++) {
1921 endch = ch + s->xxch_chset_nch[chset];
1922 mask = s->xxch_dmix_embedded;
1925 for (j = ch; j < endch; j++) {
1926 if (mask & (1 << j)) { /* this channel has been mixed-out */
1927 src_chan = s->samples_chanptr[s->channel_order_tab[j]];
1928 for (k = 0; k < endch; k++) {
1929 achan = s->channel_order_tab[k];
1930 scale = s->xxch_dmix_coeff[j][k];
1932 dst_chan = s->samples_chanptr[achan];
1933 s->fdsp->vector_fmac_scalar(dst_chan, src_chan,
1940 /* if a downmix has been embedded then undo the pre-scaling */
1941 if ((mask & (1 << ch)) && s->xxch_dmix_sf[chset] != 1.0f) {
1942 scale = s->xxch_dmix_sf[chset];
1944 for (j = 0; j < ch; j++) {
1945 src_chan = s->samples_chanptr[s->channel_order_tab[j]];
1946 for (k = 0; k < 256; k++)
1947 src_chan[k] *= scale;
1950 /* LFE channel is always part of core, scale if it exists */
1952 src_chan = s->samples_chanptr[s->lfe_index];
1953 for (k = 0; k < 256; k++)
1954 src_chan[k] *= scale;
1964 /* update lfe history */
1965 lfe_samples = 2 * s->lfe * (s->sample_blocks / SAMPLES_PER_SUBBAND);
1966 for (i = 0; i < 2 * s->lfe * 4; i++)
1967 s->lfe_data[i] = s->lfe_data[i + lfe_samples];
1969 if (s->exss_ext_mask & DCA_EXT_EXSS_XLL) {
1970 ret = ff_dca_xll_decode_audio(s, frame);
1976 * DCA_STEREO_TOTAL (Lt/Rt) is equivalent to Dolby Surround */
1977 ret = ff_side_data_update_matrix_encoding(frame,
1978 (s->output & ~DCA_LFE) == DCA_STEREO_TOTAL ?
1979 AV_MATRIX_ENCODING_DOLBY : AV_MATRIX_ENCODING_NONE);
1983 if ( avctx->profile != FF_PROFILE_DTS_HD_MA
1984 && avctx->profile != FF_PROFILE_DTS_HD_HRA)
1985 avctx->bit_rate = s->bit_rate;
1992 * DCA initialization
1994 * @param avctx pointer to the AVCodecContext
1997 static av_cold int dca_decode_init(AVCodecContext *avctx)
1999 DCAContext *s = avctx->priv_data;
2004 s->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
2006 return AVERROR(ENOMEM);
2008 ff_mdct_init(&s->imdct, 6, 1, 1.0);
2009 ff_synth_filter_init(&s->synth);
2010 ff_dcadsp_init(&s->dcadsp);
2011 ff_fmt_convert_init(&s->fmt_conv, avctx);
2013 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
2015 /* allow downmixing to stereo */
2016 if (avctx->channels > 2 &&
2017 avctx->request_channel_layout == AV_CH_LAYOUT_STEREO)
2018 avctx->channels = 2;
2023 static av_cold int dca_decode_end(AVCodecContext *avctx)
2025 DCAContext *s = avctx->priv_data;
2026 ff_mdct_end(&s->imdct);
2027 av_freep(&s->extra_channels_buffer);
2029 av_freep(&s->xll_sample_buf);
2030 av_freep(&s->qmf64_table);
2034 static const AVOption options[] = {
2035 { "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 },
2036 { "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 },
2040 static const AVClass dca_decoder_class = {
2041 .class_name = "DCA decoder",
2042 .item_name = av_default_item_name,
2044 .version = LIBAVUTIL_VERSION_INT,
2045 .category = AV_CLASS_CATEGORY_DECODER,
2048 AVCodec ff_dca_decoder = {
2050 .long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
2051 .type = AVMEDIA_TYPE_AUDIO,
2052 .id = AV_CODEC_ID_DTS,
2053 .priv_data_size = sizeof(DCAContext),
2054 .init = dca_decode_init,
2055 .decode = dca_decode_frame,
2056 .close = dca_decode_end,
2057 .capabilities = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
2058 .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
2059 AV_SAMPLE_FMT_NONE },
2060 .profiles = NULL_IF_CONFIG_SMALL(ff_dca_profiles),
2061 .priv_class = &dca_decoder_class,