dca: Move all tables into dcadata.h
[libav.git] / libavcodec / dcadec.c
1 /*
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 *
8 * This file is part of Libav.
9 *
10 * Libav is free software; you can redistribute it and/or
11 * modify it under the terms of the GNU Lesser General Public
12 * License as published by the Free Software Foundation; either
13 * version 2.1 of the License, or (at your option) any later version.
14 *
15 * Libav is distributed in the hope that it will be useful,
16 * but WITHOUT ANY WARRANTY; without even the implied warranty of
17 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
18 * Lesser General Public License for more details.
19 *
20 * You should have received a copy of the GNU Lesser General Public
21 * License along with Libav; if not, write to the Free Software
22 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
23 */
24
25 #include <math.h>
26 #include <stddef.h>
27 #include <stdio.h>
28
29 #include "libavutil/channel_layout.h"
30 #include "libavutil/common.h"
31 #include "libavutil/float_dsp.h"
32 #include "libavutil/internal.h"
33 #include "libavutil/intreadwrite.h"
34 #include "libavutil/mathematics.h"
35 #include "libavutil/opt.h"
36 #include "libavutil/samplefmt.h"
37
38 #include "avcodec.h"
39 #include "dca.h"
40 #include "dcadata.h"
41 #include "dcadsp.h"
42 #include "dcahuff.h"
43 #include "dca_exss.h"
44 #include "fft.h"
45 #include "fmtconvert.h"
46 #include "get_bits.h"
47 #include "internal.h"
48 #include "mathops.h"
49 #include "put_bits.h"
50 #include "synth_filter.h"
51
52 #if ARCH_ARM
53 # include "arm/dca.h"
54 #endif
55
56 enum DCAMode {
57 DCA_MONO = 0,
58 DCA_CHANNEL,
59 DCA_STEREO,
60 DCA_STEREO_SUMDIFF,
61 DCA_STEREO_TOTAL,
62 DCA_3F,
63 DCA_2F1R,
64 DCA_3F1R,
65 DCA_2F2R,
66 DCA_3F2R,
67 DCA_4F2R
68 };
69
70 /* -1 are reserved or unknown */
71 static const int dca_ext_audio_descr_mask[] = {
72 DCA_EXT_XCH,
73 -1,
74 DCA_EXT_X96,
75 DCA_EXT_XCH | DCA_EXT_X96,
76 -1,
77 -1,
78 DCA_EXT_XXCH,
79 -1,
80 };
81
82 /* Tables for mapping dts channel configurations to libavcodec multichannel api.
83 * Some compromises have been made for special configurations. Most configurations
84 * are never used so complete accuracy is not needed.
85 *
86 * L = left, R = right, C = center, S = surround, F = front, R = rear, T = total, OV = overhead.
87 * S -> side, when both rear and back are configured move one of them to the side channel
88 * OV -> center back
89 * All 2 channel configurations -> AV_CH_LAYOUT_STEREO
90 */
91 static const uint64_t dca_core_channel_layout[] = {
92 AV_CH_FRONT_CENTER, ///< 1, A
93 AV_CH_LAYOUT_STEREO, ///< 2, A + B (dual mono)
94 AV_CH_LAYOUT_STEREO, ///< 2, L + R (stereo)
95 AV_CH_LAYOUT_STEREO, ///< 2, (L + R) + (L - R) (sum-difference)
96 AV_CH_LAYOUT_STEREO, ///< 2, LT + RT (left and right total)
97 AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER, ///< 3, C + L + R
98 AV_CH_LAYOUT_STEREO | AV_CH_BACK_CENTER, ///< 3, L + R + S
99 AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER | AV_CH_BACK_CENTER, ///< 4, C + L + R + S
100 AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT, ///< 4, L + R + SL + SR
101
102 AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER | AV_CH_SIDE_LEFT |
103 AV_CH_SIDE_RIGHT, ///< 5, C + L + R + SL + SR
104
105 AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT |
106 AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER, ///< 6, CL + CR + L + R + SL + SR
107
108 AV_CH_LAYOUT_STEREO | AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT |
109 AV_CH_FRONT_CENTER | AV_CH_BACK_CENTER, ///< 6, C + L + R + LR + RR + OV
110
111 AV_CH_FRONT_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER |
112 AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_BACK_CENTER |
113 AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT, ///< 6, CF + CR + LF + RF + LR + RR
114
115 AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_CENTER |
116 AV_CH_FRONT_RIGHT_OF_CENTER | AV_CH_LAYOUT_STEREO |
117 AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT, ///< 7, CL + C + CR + L + R + SL + SR
118
119 AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER |
120 AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT |
121 AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT, ///< 8, CL + CR + L + R + SL1 + SL2 + SR1 + SR2
122
123 AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_CENTER |
124 AV_CH_FRONT_RIGHT_OF_CENTER | AV_CH_LAYOUT_STEREO |
125 AV_CH_SIDE_LEFT | AV_CH_BACK_CENTER | AV_CH_SIDE_RIGHT, ///< 8, CL + C + CR + L + R + SL + S + SR
126 };
127
128 #define DCA_DOLBY 101 /* FIXME */
129
130 #define DCA_CHANNEL_BITS 6
131 #define DCA_CHANNEL_MASK 0x3F
132
133 #define DCA_LFE 0x80
134
135 #define HEADER_SIZE 14
136
137 #define DCA_NSYNCAUX 0x9A1105A0
138
139 /** Bit allocation */
140 typedef struct BitAlloc {
141 int offset; ///< code values offset
142 int maxbits[8]; ///< max bits in VLC
143 int wrap; ///< wrap for get_vlc2()
144 VLC vlc[8]; ///< actual codes
145 } BitAlloc;
146
147 static BitAlloc dca_bitalloc_index; ///< indexes for samples VLC select
148 static BitAlloc dca_tmode; ///< transition mode VLCs
149 static BitAlloc dca_scalefactor; ///< scalefactor VLCs
150 static BitAlloc dca_smpl_bitalloc[11]; ///< samples VLCs
151
152 static av_always_inline int get_bitalloc(GetBitContext *gb, BitAlloc *ba,
153 int idx)
154 {
155 return get_vlc2(gb, ba->vlc[idx].table, ba->vlc[idx].bits, ba->wrap) +
156 ba->offset;
157 }
158
159 static av_cold void dca_init_vlcs(void)
160 {
161 static int vlcs_initialized = 0;
162 int i, j, c = 14;
163 static VLC_TYPE dca_table[23622][2];
164
165 if (vlcs_initialized)
166 return;
167
168 dca_bitalloc_index.offset = 1;
169 dca_bitalloc_index.wrap = 2;
170 for (i = 0; i < 5; i++) {
171 dca_bitalloc_index.vlc[i].table = &dca_table[dca_vlc_offs[i]];
172 dca_bitalloc_index.vlc[i].table_allocated = dca_vlc_offs[i + 1] - dca_vlc_offs[i];
173 init_vlc(&dca_bitalloc_index.vlc[i], bitalloc_12_vlc_bits[i], 12,
174 bitalloc_12_bits[i], 1, 1,
175 bitalloc_12_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
176 }
177 dca_scalefactor.offset = -64;
178 dca_scalefactor.wrap = 2;
179 for (i = 0; i < 5; i++) {
180 dca_scalefactor.vlc[i].table = &dca_table[dca_vlc_offs[i + 5]];
181 dca_scalefactor.vlc[i].table_allocated = dca_vlc_offs[i + 6] - dca_vlc_offs[i + 5];
182 init_vlc(&dca_scalefactor.vlc[i], SCALES_VLC_BITS, 129,
183 scales_bits[i], 1, 1,
184 scales_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
185 }
186 dca_tmode.offset = 0;
187 dca_tmode.wrap = 1;
188 for (i = 0; i < 4; i++) {
189 dca_tmode.vlc[i].table = &dca_table[dca_vlc_offs[i + 10]];
190 dca_tmode.vlc[i].table_allocated = dca_vlc_offs[i + 11] - dca_vlc_offs[i + 10];
191 init_vlc(&dca_tmode.vlc[i], tmode_vlc_bits[i], 4,
192 tmode_bits[i], 1, 1,
193 tmode_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
194 }
195
196 for (i = 0; i < 10; i++)
197 for (j = 0; j < 7; j++) {
198 if (!bitalloc_codes[i][j])
199 break;
200 dca_smpl_bitalloc[i + 1].offset = bitalloc_offsets[i];
201 dca_smpl_bitalloc[i + 1].wrap = 1 + (j > 4);
202 dca_smpl_bitalloc[i + 1].vlc[j].table = &dca_table[dca_vlc_offs[c]];
203 dca_smpl_bitalloc[i + 1].vlc[j].table_allocated = dca_vlc_offs[c + 1] - dca_vlc_offs[c];
204
205 init_vlc(&dca_smpl_bitalloc[i + 1].vlc[j], bitalloc_maxbits[i][j],
206 bitalloc_sizes[i],
207 bitalloc_bits[i][j], 1, 1,
208 bitalloc_codes[i][j], 2, 2, INIT_VLC_USE_NEW_STATIC);
209 c++;
210 }
211 vlcs_initialized = 1;
212 }
213
214 static inline void get_array(GetBitContext *gb, int *dst, int len, int bits)
215 {
216 while (len--)
217 *dst++ = get_bits(gb, bits);
218 }
219
220 static int dca_parse_audio_coding_header(DCAContext *s, int base_channel)
221 {
222 int i, j;
223 static const float adj_table[4] = { 1.0, 1.1250, 1.2500, 1.4375 };
224 static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };
225 static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };
226
227 s->total_channels = get_bits(&s->gb, 3) + 1 + base_channel;
228 s->prim_channels = s->total_channels;
229
230 if (s->prim_channels > DCA_PRIM_CHANNELS_MAX)
231 s->prim_channels = DCA_PRIM_CHANNELS_MAX;
232
233 for (i = base_channel; i < s->prim_channels; i++) {
234 s->subband_activity[i] = get_bits(&s->gb, 5) + 2;
235 if (s->subband_activity[i] > DCA_SUBBANDS)
236 s->subband_activity[i] = DCA_SUBBANDS;
237 }
238 for (i = base_channel; i < s->prim_channels; i++) {
239 s->vq_start_subband[i] = get_bits(&s->gb, 5) + 1;
240 if (s->vq_start_subband[i] > DCA_SUBBANDS)
241 s->vq_start_subband[i] = DCA_SUBBANDS;
242 }
243 get_array(&s->gb, s->joint_intensity + base_channel, s->prim_channels - base_channel, 3);
244 get_array(&s->gb, s->transient_huffman + base_channel, s->prim_channels - base_channel, 2);
245 get_array(&s->gb, s->scalefactor_huffman + base_channel, s->prim_channels - base_channel, 3);
246 get_array(&s->gb, s->bitalloc_huffman + base_channel, s->prim_channels - base_channel, 3);
247
248 /* Get codebooks quantization indexes */
249 if (!base_channel)
250 memset(s->quant_index_huffman, 0, sizeof(s->quant_index_huffman));
251 for (j = 1; j < 11; j++)
252 for (i = base_channel; i < s->prim_channels; i++)
253 s->quant_index_huffman[i][j] = get_bits(&s->gb, bitlen[j]);
254
255 /* Get scale factor adjustment */
256 for (j = 0; j < 11; j++)
257 for (i = base_channel; i < s->prim_channels; i++)
258 s->scalefactor_adj[i][j] = 1;
259
260 for (j = 1; j < 11; j++)
261 for (i = base_channel; i < s->prim_channels; i++)
262 if (s->quant_index_huffman[i][j] < thr[j])
263 s->scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)];
264
265 if (s->crc_present) {
266 /* Audio header CRC check */
267 get_bits(&s->gb, 16);
268 }
269
270 s->current_subframe = 0;
271 s->current_subsubframe = 0;
272
273 return 0;
274 }
275
276 static int dca_parse_frame_header(DCAContext *s)
277 {
278 init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
279
280 /* Sync code */
281 skip_bits_long(&s->gb, 32);
282
283 /* Frame header */
284 s->frame_type = get_bits(&s->gb, 1);
285 s->samples_deficit = get_bits(&s->gb, 5) + 1;
286 s->crc_present = get_bits(&s->gb, 1);
287 s->sample_blocks = get_bits(&s->gb, 7) + 1;
288 s->frame_size = get_bits(&s->gb, 14) + 1;
289 if (s->frame_size < 95)
290 return AVERROR_INVALIDDATA;
291 s->amode = get_bits(&s->gb, 6);
292 s->sample_rate = avpriv_dca_sample_rates[get_bits(&s->gb, 4)];
293 if (!s->sample_rate)
294 return AVERROR_INVALIDDATA;
295 s->bit_rate_index = get_bits(&s->gb, 5);
296 s->bit_rate = dca_bit_rates[s->bit_rate_index];
297 if (!s->bit_rate)
298 return AVERROR_INVALIDDATA;
299
300 skip_bits1(&s->gb); // always 0 (reserved, cf. ETSI TS 102 114 V1.4.1)
301 s->dynrange = get_bits(&s->gb, 1);
302 s->timestamp = get_bits(&s->gb, 1);
303 s->aux_data = get_bits(&s->gb, 1);
304 s->hdcd = get_bits(&s->gb, 1);
305 s->ext_descr = get_bits(&s->gb, 3);
306 s->ext_coding = get_bits(&s->gb, 1);
307 s->aspf = get_bits(&s->gb, 1);
308 s->lfe = get_bits(&s->gb, 2);
309 s->predictor_history = get_bits(&s->gb, 1);
310
311 if (s->lfe > 2) {
312 av_log(s->avctx, AV_LOG_ERROR, "Invalid LFE value: %d\n", s->lfe);
313 return AVERROR_INVALIDDATA;
314 }
315
316 /* TODO: check CRC */
317 if (s->crc_present)
318 s->header_crc = get_bits(&s->gb, 16);
319
320 s->multirate_inter = get_bits(&s->gb, 1);
321 s->version = get_bits(&s->gb, 4);
322 s->copy_history = get_bits(&s->gb, 2);
323 s->source_pcm_res = get_bits(&s->gb, 3);
324 s->front_sum = get_bits(&s->gb, 1);
325 s->surround_sum = get_bits(&s->gb, 1);
326 s->dialog_norm = get_bits(&s->gb, 4);
327
328 /* FIXME: channels mixing levels */
329 s->output = s->amode;
330 if (s->lfe)
331 s->output |= DCA_LFE;
332
333 /* Primary audio coding header */
334 s->subframes = get_bits(&s->gb, 4) + 1;
335
336 return dca_parse_audio_coding_header(s, 0);
337 }
338
339 static inline int get_scale(GetBitContext *gb, int level, int value, int log2range)
340 {
341 if (level < 5) {
342 /* huffman encoded */
343 value += get_bitalloc(gb, &dca_scalefactor, level);
344 value = av_clip(value, 0, (1 << log2range) - 1);
345 } else if (level < 8) {
346 if (level + 1 > log2range) {
347 skip_bits(gb, level + 1 - log2range);
348 value = get_bits(gb, log2range);
349 } else {
350 value = get_bits(gb, level + 1);
351 }
352 }
353 return value;
354 }
355
356 static int dca_subframe_header(DCAContext *s, int base_channel, int block_index)
357 {
358 /* Primary audio coding side information */
359 int j, k;
360
361 if (get_bits_left(&s->gb) < 0)
362 return AVERROR_INVALIDDATA;
363
364 if (!base_channel) {
365 s->subsubframes[s->current_subframe] = get_bits(&s->gb, 2) + 1;
366 s->partial_samples[s->current_subframe] = get_bits(&s->gb, 3);
367 }
368
369 for (j = base_channel; j < s->prim_channels; j++) {
370 for (k = 0; k < s->subband_activity[j]; k++)
371 s->prediction_mode[j][k] = get_bits(&s->gb, 1);
372 }
373
374 /* Get prediction codebook */
375 for (j = base_channel; j < s->prim_channels; j++) {
376 for (k = 0; k < s->subband_activity[j]; k++) {
377 if (s->prediction_mode[j][k] > 0) {
378 /* (Prediction coefficient VQ address) */
379 s->prediction_vq[j][k] = get_bits(&s->gb, 12);
380 }
381 }
382 }
383
384 /* Bit allocation index */
385 for (j = base_channel; j < s->prim_channels; j++) {
386 for (k = 0; k < s->vq_start_subband[j]; k++) {
387 if (s->bitalloc_huffman[j] == 6)
388 s->bitalloc[j][k] = get_bits(&s->gb, 5);
389 else if (s->bitalloc_huffman[j] == 5)
390 s->bitalloc[j][k] = get_bits(&s->gb, 4);
391 else if (s->bitalloc_huffman[j] == 7) {
392 av_log(s->avctx, AV_LOG_ERROR,
393 "Invalid bit allocation index\n");
394 return AVERROR_INVALIDDATA;
395 } else {
396 s->bitalloc[j][k] =
397 get_bitalloc(&s->gb, &dca_bitalloc_index, s->bitalloc_huffman[j]);
398 }
399
400 if (s->bitalloc[j][k] > 26) {
401 av_dlog(s->avctx, "bitalloc index [%i][%i] too big (%i)\n",
402 j, k, s->bitalloc[j][k]);
403 return AVERROR_INVALIDDATA;
404 }
405 }
406 }
407
408 /* Transition mode */
409 for (j = base_channel; j < s->prim_channels; j++) {
410 for (k = 0; k < s->subband_activity[j]; k++) {
411 s->transition_mode[j][k] = 0;
412 if (s->subsubframes[s->current_subframe] > 1 &&
413 k < s->vq_start_subband[j] && s->bitalloc[j][k] > 0) {
414 s->transition_mode[j][k] =
415 get_bitalloc(&s->gb, &dca_tmode, s->transient_huffman[j]);
416 }
417 }
418 }
419
420 if (get_bits_left(&s->gb) < 0)
421 return AVERROR_INVALIDDATA;
422
423 for (j = base_channel; j < s->prim_channels; j++) {
424 const uint32_t *scale_table;
425 int scale_sum, log_size;
426
427 memset(s->scale_factor[j], 0,
428 s->subband_activity[j] * sizeof(s->scale_factor[0][0][0]) * 2);
429
430 if (s->scalefactor_huffman[j] == 6) {
431 scale_table = scale_factor_quant7;
432 log_size = 7;
433 } else {
434 scale_table = scale_factor_quant6;
435 log_size = 6;
436 }
437
438 /* When huffman coded, only the difference is encoded */
439 scale_sum = 0;
440
441 for (k = 0; k < s->subband_activity[j]; k++) {
442 if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0) {
443 scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum, log_size);
444 s->scale_factor[j][k][0] = scale_table[scale_sum];
445 }
446
447 if (k < s->vq_start_subband[j] && s->transition_mode[j][k]) {
448 /* Get second scale factor */
449 scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum, log_size);
450 s->scale_factor[j][k][1] = scale_table[scale_sum];
451 }
452 }
453 }
454
455 /* Joint subband scale factor codebook select */
456 for (j = base_channel; j < s->prim_channels; j++) {
457 /* Transmitted only if joint subband coding enabled */
458 if (s->joint_intensity[j] > 0)
459 s->joint_huff[j] = get_bits(&s->gb, 3);
460 }
461
462 if (get_bits_left(&s->gb) < 0)
463 return AVERROR_INVALIDDATA;
464
465 /* Scale factors for joint subband coding */
466 for (j = base_channel; j < s->prim_channels; j++) {
467 int source_channel;
468
469 /* Transmitted only if joint subband coding enabled */
470 if (s->joint_intensity[j] > 0) {
471 int scale = 0;
472 source_channel = s->joint_intensity[j] - 1;
473
474 /* When huffman coded, only the difference is encoded
475 * (is this valid as well for joint scales ???) */
476
477 for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++) {
478 scale = get_scale(&s->gb, s->joint_huff[j], 64 /* bias */, 7);
479 s->joint_scale_factor[j][k] = scale; /*joint_scale_table[scale]; */
480 }
481
482 if (!(s->debug_flag & 0x02)) {
483 av_log(s->avctx, AV_LOG_DEBUG,
484 "Joint stereo coding not supported\n");
485 s->debug_flag |= 0x02;
486 }
487 }
488 }
489
490 /* Dynamic range coefficient */
491 if (!base_channel && s->dynrange)
492 s->dynrange_coef = get_bits(&s->gb, 8);
493
494 /* Side information CRC check word */
495 if (s->crc_present) {
496 get_bits(&s->gb, 16);
497 }
498
499 /*
500 * Primary audio data arrays
501 */
502
503 /* VQ encoded high frequency subbands */
504 for (j = base_channel; j < s->prim_channels; j++)
505 for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++)
506 /* 1 vector -> 32 samples */
507 s->high_freq_vq[j][k] = get_bits(&s->gb, 10);
508
509 /* Low frequency effect data */
510 if (!base_channel && s->lfe) {
511 /* LFE samples */
512 int lfe_samples = 2 * s->lfe * (4 + block_index);
513 int lfe_end_sample = 2 * s->lfe * (4 + block_index + s->subsubframes[s->current_subframe]);
514 float lfe_scale;
515
516 for (j = lfe_samples; j < lfe_end_sample; j++) {
517 /* Signed 8 bits int */
518 s->lfe_data[j] = get_sbits(&s->gb, 8);
519 }
520
521 /* Scale factor index */
522 skip_bits(&s->gb, 1);
523 s->lfe_scale_factor = scale_factor_quant7[get_bits(&s->gb, 7)];
524
525 /* Quantization step size * scale factor */
526 lfe_scale = 0.035 * s->lfe_scale_factor;
527
528 for (j = lfe_samples; j < lfe_end_sample; j++)
529 s->lfe_data[j] *= lfe_scale;
530 }
531
532 return 0;
533 }
534
535 static void qmf_32_subbands(DCAContext *s, int chans,
536 float samples_in[32][8], float *samples_out,
537 float scale)
538 {
539 const float *prCoeff;
540
541 int sb_act = s->subband_activity[chans];
542
543 scale *= sqrt(1 / 8.0);
544
545 /* Select filter */
546 if (!s->multirate_inter) /* Non-perfect reconstruction */
547 prCoeff = fir_32bands_nonperfect;
548 else /* Perfect reconstruction */
549 prCoeff = fir_32bands_perfect;
550
551 s->dcadsp.qmf_32_subbands(samples_in, sb_act, &s->synth, &s->imdct,
552 s->subband_fir_hist[chans],
553 &s->hist_index[chans],
554 s->subband_fir_noidea[chans], prCoeff,
555 samples_out, s->raXin, scale);
556 }
557
558 static void lfe_interpolation_fir(DCAContext *s, int decimation_select,
559 int num_deci_sample, float *samples_in,
560 float *samples_out)
561 {
562 /* samples_in: An array holding decimated samples.
563 * Samples in current subframe starts from samples_in[0],
564 * while samples_in[-1], samples_in[-2], ..., stores samples
565 * from last subframe as history.
566 *
567 * samples_out: An array holding interpolated samples
568 */
569
570 int idx;
571 const float *prCoeff;
572 int deciindex;
573
574 /* Select decimation filter */
575 if (decimation_select == 1) {
576 idx = 1;
577 prCoeff = lfe_fir_128;
578 } else {
579 idx = 0;
580 prCoeff = lfe_fir_64;
581 }
582 /* Interpolation */
583 for (deciindex = 0; deciindex < num_deci_sample; deciindex++) {
584 s->dcadsp.lfe_fir[idx](samples_out, samples_in, prCoeff);
585 samples_in++;
586 samples_out += 2 * 32 * (1 + idx);
587 }
588 }
589
590 /* downmixing routines */
591 #define MIX_REAR1(samples, s1, rs, coef) \
592 samples[0][i] += samples[s1][i] * coef[rs][0]; \
593 samples[1][i] += samples[s1][i] * coef[rs][1];
594
595 #define MIX_REAR2(samples, s1, s2, rs, coef) \
596 samples[0][i] += samples[s1][i] * coef[rs][0] + samples[s2][i] * coef[rs + 1][0]; \
597 samples[1][i] += samples[s1][i] * coef[rs][1] + samples[s2][i] * coef[rs + 1][1];
598
599 #define MIX_FRONT3(samples, coef) \
600 t = samples[c][i]; \
601 u = samples[l][i]; \
602 v = samples[r][i]; \
603 samples[0][i] = t * coef[0][0] + u * coef[1][0] + v * coef[2][0]; \
604 samples[1][i] = t * coef[0][1] + u * coef[1][1] + v * coef[2][1];
605
606 #define DOWNMIX_TO_STEREO(op1, op2) \
607 for (i = 0; i < 256; i++) { \
608 op1 \
609 op2 \
610 }
611
612 static void dca_downmix(float **samples, int srcfmt, int lfe_present,
613 float coef[DCA_PRIM_CHANNELS_MAX + 1][2],
614 const int8_t *channel_mapping)
615 {
616 int c, l, r, sl, sr, s;
617 int i;
618 float t, u, v;
619
620 switch (srcfmt) {
621 case DCA_MONO:
622 case DCA_4F2R:
623 av_log(NULL, 0, "Not implemented!\n");
624 break;
625 case DCA_CHANNEL:
626 case DCA_STEREO:
627 case DCA_STEREO_TOTAL:
628 case DCA_STEREO_SUMDIFF:
629 break;
630 case DCA_3F:
631 c = channel_mapping[0];
632 l = channel_mapping[1];
633 r = channel_mapping[2];
634 DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), );
635 break;
636 case DCA_2F1R:
637 s = channel_mapping[2];
638 DOWNMIX_TO_STEREO(MIX_REAR1(samples, s, 2, coef), );
639 break;
640 case DCA_3F1R:
641 c = channel_mapping[0];
642 l = channel_mapping[1];
643 r = channel_mapping[2];
644 s = channel_mapping[3];
645 DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
646 MIX_REAR1(samples, s, 3, coef));
647 break;
648 case DCA_2F2R:
649 sl = channel_mapping[2];
650 sr = channel_mapping[3];
651 DOWNMIX_TO_STEREO(MIX_REAR2(samples, sl, sr, 2, coef), );
652 break;
653 case DCA_3F2R:
654 c = channel_mapping[0];
655 l = channel_mapping[1];
656 r = channel_mapping[2];
657 sl = channel_mapping[3];
658 sr = channel_mapping[4];
659 DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
660 MIX_REAR2(samples, sl, sr, 3, coef));
661 break;
662 }
663 if (lfe_present) {
664 int lf_buf = dca_lfe_index[srcfmt];
665 int lf_idx = dca_channels[srcfmt];
666 for (i = 0; i < 256; i++) {
667 samples[0][i] += samples[lf_buf][i] * coef[lf_idx][0];
668 samples[1][i] += samples[lf_buf][i] * coef[lf_idx][1];
669 }
670 }
671 }
672
673 #ifndef decode_blockcodes
674 /* Very compact version of the block code decoder that does not use table
675 * look-up but is slightly slower */
676 static int decode_blockcode(int code, int levels, int32_t *values)
677 {
678 int i;
679 int offset = (levels - 1) >> 1;
680
681 for (i = 0; i < 4; i++) {
682 int div = FASTDIV(code, levels);
683 values[i] = code - offset - div * levels;
684 code = div;
685 }
686
687 return code;
688 }
689
690 static int decode_blockcodes(int code1, int code2, int levels, int32_t *values)
691 {
692 return decode_blockcode(code1, levels, values) |
693 decode_blockcode(code2, levels, values + 4);
694 }
695 #endif
696
697 static const uint8_t abits_sizes[7] = { 7, 10, 12, 13, 15, 17, 19 };
698 static const uint8_t abits_levels[7] = { 3, 5, 7, 9, 13, 17, 25 };
699
700 static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
701 {
702 int k, l;
703 int subsubframe = s->current_subsubframe;
704
705 const float *quant_step_table;
706
707 /* FIXME */
708 float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];
709 LOCAL_ALIGNED_16(int32_t, block, [8 * DCA_SUBBANDS]);
710
711 /*
712 * Audio data
713 */
714
715 /* Select quantization step size table */
716 if (s->bit_rate_index == 0x1f)
717 quant_step_table = lossless_quant_d;
718 else
719 quant_step_table = lossy_quant_d;
720
721 for (k = base_channel; k < s->prim_channels; k++) {
722 float rscale[DCA_SUBBANDS];
723
724 if (get_bits_left(&s->gb) < 0)
725 return AVERROR_INVALIDDATA;
726
727 for (l = 0; l < s->vq_start_subband[k]; l++) {
728 int m;
729
730 /* Select the mid-tread linear quantizer */
731 int abits = s->bitalloc[k][l];
732
733 float quant_step_size = quant_step_table[abits];
734
735 /*
736 * Determine quantization index code book and its type
737 */
738
739 /* Select quantization index code book */
740 int sel = s->quant_index_huffman[k][abits];
741
742 /*
743 * Extract bits from the bit stream
744 */
745 if (!abits) {
746 rscale[l] = 0;
747 memset(block + 8 * l, 0, 8 * sizeof(block[0]));
748 } else {
749 /* Deal with transients */
750 int sfi = s->transition_mode[k][l] && subsubframe >= s->transition_mode[k][l];
751 rscale[l] = quant_step_size * s->scale_factor[k][l][sfi] *
752 s->scalefactor_adj[k][sel];
753
754 if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table) {
755 if (abits <= 7) {
756 /* Block code */
757 int block_code1, block_code2, size, levels, err;
758
759 size = abits_sizes[abits - 1];
760 levels = abits_levels[abits - 1];
761
762 block_code1 = get_bits(&s->gb, size);
763 block_code2 = get_bits(&s->gb, size);
764 err = decode_blockcodes(block_code1, block_code2,
765 levels, block + 8 * l);
766 if (err) {
767 av_log(s->avctx, AV_LOG_ERROR,
768 "ERROR: block code look-up failed\n");
769 return AVERROR_INVALIDDATA;
770 }
771 } else {
772 /* no coding */
773 for (m = 0; m < 8; m++)
774 block[8 * l + m] = get_sbits(&s->gb, abits - 3);
775 }
776 } else {
777 /* Huffman coded */
778 for (m = 0; m < 8; m++)
779 block[8 * l + m] = get_bitalloc(&s->gb,
780 &dca_smpl_bitalloc[abits], sel);
781 }
782 }
783 }
784
785 s->fmt_conv.int32_to_float_fmul_array8(&s->fmt_conv, subband_samples[k][0],
786 block, rscale, 8 * s->vq_start_subband[k]);
787
788 for (l = 0; l < s->vq_start_subband[k]; l++) {
789 int m;
790 /*
791 * Inverse ADPCM if in prediction mode
792 */
793 if (s->prediction_mode[k][l]) {
794 int n;
795 if (s->predictor_history)
796 subband_samples[k][l][0] += (adpcm_vb[s->prediction_vq[k][l]][0] *
797 s->subband_samples_hist[k][l][3] +
798 adpcm_vb[s->prediction_vq[k][l]][1] *
799 s->subband_samples_hist[k][l][2] +
800 adpcm_vb[s->prediction_vq[k][l]][2] *
801 s->subband_samples_hist[k][l][1] +
802 adpcm_vb[s->prediction_vq[k][l]][3] *
803 s->subband_samples_hist[k][l][0]) *
804 (1.0f / 8192);
805 for (m = 1; m < 8; m++) {
806 float sum = adpcm_vb[s->prediction_vq[k][l]][0] *
807 subband_samples[k][l][m - 1];
808 for (n = 2; n <= 4; n++)
809 if (m >= n)
810 sum += adpcm_vb[s->prediction_vq[k][l]][n - 1] *
811 subband_samples[k][l][m - n];
812 else if (s->predictor_history)
813 sum += adpcm_vb[s->prediction_vq[k][l]][n - 1] *
814 s->subband_samples_hist[k][l][m - n + 4];
815 subband_samples[k][l][m] += sum * 1.0f / 8192;
816 }
817 }
818 }
819
820 /*
821 * Decode VQ encoded high frequencies
822 */
823 if (s->subband_activity[k] > s->vq_start_subband[k]) {
824 if (!s->debug_flag & 0x01) {
825 av_log(s->avctx, AV_LOG_DEBUG,
826 "Stream with high frequencies VQ coding\n");
827 s->debug_flag |= 0x01;
828 }
829 s->dcadsp.decode_hf(subband_samples[k], s->high_freq_vq[k],
830 high_freq_vq, subsubframe * 8,
831 s->scale_factor[k], s->vq_start_subband[k],
832 s->subband_activity[k]);
833 }
834 }
835
836 /* Check for DSYNC after subsubframe */
837 if (s->aspf || subsubframe == s->subsubframes[s->current_subframe] - 1) {
838 if (get_bits(&s->gb, 16) != 0xFFFF) {
839 av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n");
840 return AVERROR_INVALIDDATA;
841 }
842 }
843
844 /* Backup predictor history for adpcm */
845 for (k = base_channel; k < s->prim_channels; k++)
846 for (l = 0; l < s->vq_start_subband[k]; l++)
847 AV_COPY128(s->subband_samples_hist[k][l], &subband_samples[k][l][4]);
848
849 return 0;
850 }
851
852 static int dca_filter_channels(DCAContext *s, int block_index)
853 {
854 float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];
855 int k;
856
857 /* 32 subbands QMF */
858 for (k = 0; k < s->prim_channels; k++) {
859 if (s->channel_order_tab[k] >= 0)
860 qmf_32_subbands(s, k, subband_samples[k],
861 s->samples_chanptr[s->channel_order_tab[k]],
862 M_SQRT1_2 / 32768.0);
863 }
864
865 /* Generate LFE samples for this subsubframe FIXME!!! */
866 if (s->lfe) {
867 lfe_interpolation_fir(s, s->lfe, 2 * s->lfe,
868 s->lfe_data + 2 * s->lfe * (block_index + 4),
869 s->samples_chanptr[dca_lfe_index[s->amode]]);
870 /* Outputs 20bits pcm samples */
871 }
872
873 /* Downmixing to Stereo */
874 if (s->prim_channels + !!s->lfe > 2 &&
875 s->avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
876 dca_downmix(s->samples_chanptr, s->amode, !!s->lfe, s->downmix_coef,
877 s->channel_order_tab);
878 }
879
880 return 0;
881 }
882
883 static int dca_subframe_footer(DCAContext *s, int base_channel)
884 {
885 int in, out, aux_data_count, aux_data_end, reserved;
886 uint32_t nsyncaux;
887
888 /*
889 * Unpack optional information
890 */
891
892 /* presumably optional information only appears in the core? */
893 if (!base_channel) {
894 if (s->timestamp)
895 skip_bits_long(&s->gb, 32);
896
897 if (s->aux_data) {
898 aux_data_count = get_bits(&s->gb, 6);
899
900 // align (32-bit)
901 skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
902
903 aux_data_end = 8 * aux_data_count + get_bits_count(&s->gb);
904
905 if ((nsyncaux = get_bits_long(&s->gb, 32)) != DCA_NSYNCAUX) {
906 av_log(s->avctx, AV_LOG_ERROR, "nSYNCAUX mismatch %#"PRIx32"\n",
907 nsyncaux);
908 return AVERROR_INVALIDDATA;
909 }
910
911 if (get_bits1(&s->gb)) { // bAUXTimeStampFlag
912 avpriv_request_sample(s->avctx,
913 "Auxiliary Decode Time Stamp Flag");
914 // align (4-bit)
915 skip_bits(&s->gb, (-get_bits_count(&s->gb)) & 4);
916 // 44 bits: nMSByte (8), nMarker (4), nLSByte (28), nMarker (4)
917 skip_bits_long(&s->gb, 44);
918 }
919
920 if ((s->core_downmix = get_bits1(&s->gb))) {
921 int am = get_bits(&s->gb, 3);
922 switch (am) {
923 case 0:
924 s->core_downmix_amode = DCA_MONO;
925 break;
926 case 1:
927 s->core_downmix_amode = DCA_STEREO;
928 break;
929 case 2:
930 s->core_downmix_amode = DCA_STEREO_TOTAL;
931 break;
932 case 3:
933 s->core_downmix_amode = DCA_3F;
934 break;
935 case 4:
936 s->core_downmix_amode = DCA_2F1R;
937 break;
938 case 5:
939 s->core_downmix_amode = DCA_2F2R;
940 break;
941 case 6:
942 s->core_downmix_amode = DCA_3F1R;
943 break;
944 default:
945 av_log(s->avctx, AV_LOG_ERROR,
946 "Invalid mode %d for embedded downmix coefficients\n",
947 am);
948 return AVERROR_INVALIDDATA;
949 }
950 for (out = 0; out < dca_channels[s->core_downmix_amode]; out++) {
951 for (in = 0; in < s->prim_channels + !!s->lfe; in++) {
952 uint16_t tmp = get_bits(&s->gb, 9);
953 if ((tmp & 0xFF) > 241) {
954 av_log(s->avctx, AV_LOG_ERROR,
955 "Invalid downmix coefficient code %"PRIu16"\n",
956 tmp);
957 return AVERROR_INVALIDDATA;
958 }
959 s->core_downmix_codes[in][out] = tmp;
960 }
961 }
962 }
963
964 align_get_bits(&s->gb); // byte align
965 skip_bits(&s->gb, 16); // nAUXCRC16
966
967 // additional data (reserved, cf. ETSI TS 102 114 V1.4.1)
968 if ((reserved = (aux_data_end - get_bits_count(&s->gb))) < 0) {
969 av_log(s->avctx, AV_LOG_ERROR,
970 "Overread auxiliary data by %d bits\n", -reserved);
971 return AVERROR_INVALIDDATA;
972 } else if (reserved) {
973 avpriv_request_sample(s->avctx,
974 "Core auxiliary data reserved content");
975 skip_bits_long(&s->gb, reserved);
976 }
977 }
978
979 if (s->crc_present && s->dynrange)
980 get_bits(&s->gb, 16);
981 }
982
983 return 0;
984 }
985
986 /**
987 * Decode a dca frame block
988 *
989 * @param s pointer to the DCAContext
990 */
991
992 static int dca_decode_block(DCAContext *s, int base_channel, int block_index)
993 {
994 int ret;
995
996 /* Sanity check */
997 if (s->current_subframe >= s->subframes) {
998 av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i",
999 s->current_subframe, s->subframes);
1000 return AVERROR_INVALIDDATA;
1001 }
1002
1003 if (!s->current_subsubframe) {
1004 /* Read subframe header */
1005 if ((ret = dca_subframe_header(s, base_channel, block_index)))
1006 return ret;
1007 }
1008
1009 /* Read subsubframe */
1010 if ((ret = dca_subsubframe(s, base_channel, block_index)))
1011 return ret;
1012
1013 /* Update state */
1014 s->current_subsubframe++;
1015 if (s->current_subsubframe >= s->subsubframes[s->current_subframe]) {
1016 s->current_subsubframe = 0;
1017 s->current_subframe++;
1018 }
1019 if (s->current_subframe >= s->subframes) {
1020 /* Read subframe footer */
1021 if ((ret = dca_subframe_footer(s, base_channel)))
1022 return ret;
1023 }
1024
1025 return 0;
1026 }
1027
1028 static float dca_dmix_code(unsigned code)
1029 {
1030 int sign = (code >> 8) - 1;
1031 code &= 0xff;
1032 return ((dca_dmixtable[code] ^ sign) - sign) * (1.0 / (1U << 15));
1033 }
1034
1035 /**
1036 * Main frame decoding function
1037 * FIXME add arguments
1038 */
1039 static int dca_decode_frame(AVCodecContext *avctx, void *data,
1040 int *got_frame_ptr, AVPacket *avpkt)
1041 {
1042 AVFrame *frame = data;
1043 const uint8_t *buf = avpkt->data;
1044 int buf_size = avpkt->size;
1045
1046 int lfe_samples;
1047 int num_core_channels = 0;
1048 int i, ret;
1049 float **samples_flt;
1050 DCAContext *s = avctx->priv_data;
1051 int channels, full_channels;
1052 int core_ss_end;
1053
1054 s->xch_present = 0;
1055
1056 s->dca_buffer_size = ff_dca_convert_bitstream(buf, buf_size, s->dca_buffer,
1057 DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE);
1058 if (s->dca_buffer_size == AVERROR_INVALIDDATA) {
1059 av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n");
1060 return AVERROR_INVALIDDATA;
1061 }
1062
1063 if ((ret = dca_parse_frame_header(s)) < 0) {
1064 // seems like the frame is corrupt, try with the next one
1065 return ret;
1066 }
1067 // set AVCodec values with parsed data
1068 avctx->sample_rate = s->sample_rate;
1069 avctx->bit_rate = s->bit_rate;
1070
1071 s->profile = FF_PROFILE_DTS;
1072
1073 for (i = 0; i < (s->sample_blocks / 8); i++) {
1074 if ((ret = dca_decode_block(s, 0, i))) {
1075 av_log(avctx, AV_LOG_ERROR, "error decoding block\n");
1076 return ret;
1077 }
1078 }
1079
1080 /* record number of core channels incase less than max channels are requested */
1081 num_core_channels = s->prim_channels;
1082
1083 if (s->ext_coding)
1084 s->core_ext_mask = dca_ext_audio_descr_mask[s->ext_descr];
1085 else
1086 s->core_ext_mask = 0;
1087
1088 core_ss_end = FFMIN(s->frame_size, s->dca_buffer_size) * 8;
1089
1090 /* only scan for extensions if ext_descr was unknown or indicated a
1091 * supported XCh extension */
1092 if (s->core_ext_mask < 0 || s->core_ext_mask & DCA_EXT_XCH) {
1093 /* if ext_descr was unknown, clear s->core_ext_mask so that the
1094 * extensions scan can fill it up */
1095 s->core_ext_mask = FFMAX(s->core_ext_mask, 0);
1096
1097 /* extensions start at 32-bit boundaries into bitstream */
1098 skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
1099
1100 while (core_ss_end - get_bits_count(&s->gb) >= 32) {
1101 uint32_t bits = get_bits_long(&s->gb, 32);
1102
1103 switch (bits) {
1104 case 0x5a5a5a5a: {
1105 int ext_amode, xch_fsize;
1106
1107 s->xch_base_channel = s->prim_channels;
1108
1109 /* validate sync word using XCHFSIZE field */
1110 xch_fsize = show_bits(&s->gb, 10);
1111 if ((s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize) &&
1112 (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize + 1))
1113 continue;
1114
1115 /* skip length-to-end-of-frame field for the moment */
1116 skip_bits(&s->gb, 10);
1117
1118 s->core_ext_mask |= DCA_EXT_XCH;
1119
1120 /* extension amode(number of channels in extension) should be 1 */
1121 /* AFAIK XCh is not used for more channels */
1122 if ((ext_amode = get_bits(&s->gb, 4)) != 1) {
1123 av_log(avctx, AV_LOG_ERROR,
1124 "XCh extension amode %d not supported!\n",
1125 ext_amode);
1126 continue;
1127 }
1128
1129 /* much like core primary audio coding header */
1130 dca_parse_audio_coding_header(s, s->xch_base_channel);
1131
1132 for (i = 0; i < (s->sample_blocks / 8); i++)
1133 if ((ret = dca_decode_block(s, s->xch_base_channel, i))) {
1134 av_log(avctx, AV_LOG_ERROR, "error decoding XCh extension\n");
1135 continue;
1136 }
1137
1138 s->xch_present = 1;
1139 break;
1140 }
1141 case 0x47004a03:
1142 /* XXCh: extended channels */
1143 /* usually found either in core or HD part in DTS-HD HRA streams,
1144 * but not in DTS-ES which contains XCh extensions instead */
1145 s->core_ext_mask |= DCA_EXT_XXCH;
1146 break;
1147
1148 case 0x1d95f262: {
1149 int fsize96 = show_bits(&s->gb, 12) + 1;
1150 if (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + fsize96)
1151 continue;
1152
1153 av_log(avctx, AV_LOG_DEBUG, "X96 extension found at %d bits\n",
1154 get_bits_count(&s->gb));
1155 skip_bits(&s->gb, 12);
1156 av_log(avctx, AV_LOG_DEBUG, "FSIZE96 = %d bytes\n", fsize96);
1157 av_log(avctx, AV_LOG_DEBUG, "REVNO = %d\n", get_bits(&s->gb, 4));
1158
1159 s->core_ext_mask |= DCA_EXT_X96;
1160 break;
1161 }
1162 }
1163
1164 skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
1165 }
1166 } else {
1167 /* no supported extensions, skip the rest of the core substream */
1168 skip_bits_long(&s->gb, core_ss_end - get_bits_count(&s->gb));
1169 }
1170
1171 if (s->core_ext_mask & DCA_EXT_X96)
1172 s->profile = FF_PROFILE_DTS_96_24;
1173 else if (s->core_ext_mask & (DCA_EXT_XCH | DCA_EXT_XXCH))
1174 s->profile = FF_PROFILE_DTS_ES;
1175
1176 /* check for ExSS (HD part) */
1177 if (s->dca_buffer_size - s->frame_size > 32 &&
1178 get_bits_long(&s->gb, 32) == DCA_HD_MARKER)
1179 ff_dca_exss_parse_header(s);
1180
1181 avctx->profile = s->profile;
1182
1183 full_channels = channels = s->prim_channels + !!s->lfe;
1184
1185 if (s->amode < 16) {
1186 avctx->channel_layout = dca_core_channel_layout[s->amode];
1187
1188 if (s->prim_channels + !!s->lfe > 2 &&
1189 avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
1190 /*
1191 * Neither the core's auxiliary data nor our default tables contain
1192 * downmix coefficients for the additional channel coded in the XCh
1193 * extension, so when we're doing a Stereo downmix, don't decode it.
1194 */
1195 s->xch_disable = 1;
1196 }
1197
1198 #if FF_API_REQUEST_CHANNELS
1199 FF_DISABLE_DEPRECATION_WARNINGS
1200 if (s->xch_present && !s->xch_disable &&
1201 (!avctx->request_channels ||
1202 avctx->request_channels > num_core_channels + !!s->lfe)) {
1203 FF_ENABLE_DEPRECATION_WARNINGS
1204 #else
1205 if (s->xch_present && !s->xch_disable) {
1206 #endif
1207 avctx->channel_layout |= AV_CH_BACK_CENTER;
1208 if (s->lfe) {
1209 avctx->channel_layout |= AV_CH_LOW_FREQUENCY;
1210 s->channel_order_tab = dca_channel_reorder_lfe_xch[s->amode];
1211 } else {
1212 s->channel_order_tab = dca_channel_reorder_nolfe_xch[s->amode];
1213 }
1214 } else {
1215 channels = num_core_channels + !!s->lfe;
1216 s->xch_present = 0; /* disable further xch processing */
1217 if (s->lfe) {
1218 avctx->channel_layout |= AV_CH_LOW_FREQUENCY;
1219 s->channel_order_tab = dca_channel_reorder_lfe[s->amode];
1220 } else
1221 s->channel_order_tab = dca_channel_reorder_nolfe[s->amode];
1222 }
1223
1224 if (channels > !!s->lfe &&
1225 s->channel_order_tab[channels - 1 - !!s->lfe] < 0)
1226 return AVERROR_INVALIDDATA;
1227
1228 if (num_core_channels + !!s->lfe > 2 &&
1229 avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
1230 channels = 2;
1231 s->output = s->prim_channels == 2 ? s->amode : DCA_STEREO;
1232 avctx->channel_layout = AV_CH_LAYOUT_STEREO;
1233
1234 /* Stereo downmix coefficients
1235 *
1236 * The decoder can only downmix to 2-channel, so we need to ensure
1237 * embedded downmix coefficients are actually targeting 2-channel.
1238 */
1239 if (s->core_downmix && (s->core_downmix_amode == DCA_STEREO ||
1240 s->core_downmix_amode == DCA_STEREO_TOTAL)) {
1241 for (i = 0; i < num_core_channels + !!s->lfe; i++) {
1242 /* Range checked earlier */
1243 s->downmix_coef[i][0] = dca_dmix_code(s->core_downmix_codes[i][0]);
1244 s->downmix_coef[i][1] = dca_dmix_code(s->core_downmix_codes[i][1]);
1245 }
1246 s->output = s->core_downmix_amode;
1247 } else {
1248 int am = s->amode & DCA_CHANNEL_MASK;
1249 if (am >= FF_ARRAY_ELEMS(dca_default_coeffs)) {
1250 av_log(s->avctx, AV_LOG_ERROR,
1251 "Invalid channel mode %d\n", am);
1252 return AVERROR_INVALIDDATA;
1253 }
1254 if (num_core_channels + !!s->lfe >
1255 FF_ARRAY_ELEMS(dca_default_coeffs[0])) {
1256 avpriv_request_sample(s->avctx, "Downmixing %d channels",
1257 s->prim_channels + !!s->lfe);
1258 return AVERROR_PATCHWELCOME;
1259 }
1260 for (i = 0; i < num_core_channels + !!s->lfe; i++) {
1261 s->downmix_coef[i][0] = dca_default_coeffs[am][i][0];
1262 s->downmix_coef[i][1] = dca_default_coeffs[am][i][1];
1263 }
1264 }
1265 av_dlog(s->avctx, "Stereo downmix coeffs:\n");
1266 for (i = 0; i < num_core_channels + !!s->lfe; i++) {
1267 av_dlog(s->avctx, "L, input channel %d = %f\n", i,
1268 s->downmix_coef[i][0]);
1269 av_dlog(s->avctx, "R, input channel %d = %f\n", i,
1270 s->downmix_coef[i][1]);
1271 }
1272 av_dlog(s->avctx, "\n");
1273 }
1274 } else {
1275 av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n", s->amode);
1276 return AVERROR_INVALIDDATA;
1277 }
1278 avctx->channels = channels;
1279
1280 /* get output buffer */
1281 frame->nb_samples = 256 * (s->sample_blocks / 8);
1282 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
1283 av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
1284 return ret;
1285 }
1286 samples_flt = (float **) frame->extended_data;
1287
1288 /* allocate buffer for extra channels if downmixing */
1289 if (avctx->channels < full_channels) {
1290 ret = av_samples_get_buffer_size(NULL, full_channels - channels,
1291 frame->nb_samples,
1292 avctx->sample_fmt, 0);
1293 if (ret < 0)
1294 return ret;
1295
1296 av_fast_malloc(&s->extra_channels_buffer,
1297 &s->extra_channels_buffer_size, ret);
1298 if (!s->extra_channels_buffer)
1299 return AVERROR(ENOMEM);
1300
1301 ret = av_samples_fill_arrays((uint8_t **) s->extra_channels, NULL,
1302 s->extra_channels_buffer,
1303 full_channels - channels,
1304 frame->nb_samples, avctx->sample_fmt, 0);
1305 if (ret < 0)
1306 return ret;
1307 }
1308
1309 /* filter to get final output */
1310 for (i = 0; i < (s->sample_blocks / 8); i++) {
1311 int ch;
1312
1313 for (ch = 0; ch < channels; ch++)
1314 s->samples_chanptr[ch] = samples_flt[ch] + i * 256;
1315 for (; ch < full_channels; ch++)
1316 s->samples_chanptr[ch] = s->extra_channels[ch - channels] + i * 256;
1317
1318 dca_filter_channels(s, i);
1319
1320 /* If this was marked as a DTS-ES stream we need to subtract back- */
1321 /* channel from SL & SR to remove matrixed back-channel signal */
1322 if ((s->source_pcm_res & 1) && s->xch_present) {
1323 float *back_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel]];
1324 float *lt_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 2]];
1325 float *rt_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 1]];
1326 s->fdsp.vector_fmac_scalar(lt_chan, back_chan, -M_SQRT1_2, 256);
1327 s->fdsp.vector_fmac_scalar(rt_chan, back_chan, -M_SQRT1_2, 256);
1328 }
1329 }
1330
1331 /* update lfe history */
1332 lfe_samples = 2 * s->lfe * (s->sample_blocks / 8);
1333 for (i = 0; i < 2 * s->lfe * 4; i++)
1334 s->lfe_data[i] = s->lfe_data[i + lfe_samples];
1335
1336 /* AVMatrixEncoding
1337 *
1338 * DCA_STEREO_TOTAL (Lt/Rt) is equivalent to Dolby Surround */
1339 ret = ff_side_data_update_matrix_encoding(frame,
1340 (s->output & ~DCA_LFE) == DCA_STEREO_TOTAL ?
1341 AV_MATRIX_ENCODING_DOLBY : AV_MATRIX_ENCODING_NONE);
1342 if (ret < 0)
1343 return ret;
1344
1345 *got_frame_ptr = 1;
1346
1347 return buf_size;
1348 }
1349
1350 /**
1351 * DCA initialization
1352 *
1353 * @param avctx pointer to the AVCodecContext
1354 */
1355
1356 static av_cold int dca_decode_init(AVCodecContext *avctx)
1357 {
1358 DCAContext *s = avctx->priv_data;
1359
1360 s->avctx = avctx;
1361 dca_init_vlcs();
1362
1363 avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
1364 ff_mdct_init(&s->imdct, 6, 1, 1.0);
1365 ff_synth_filter_init(&s->synth);
1366 ff_dcadsp_init(&s->dcadsp);
1367 ff_fmt_convert_init(&s->fmt_conv, avctx);
1368
1369 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
1370
1371 /* allow downmixing to stereo */
1372 #if FF_API_REQUEST_CHANNELS
1373 FF_DISABLE_DEPRECATION_WARNINGS
1374 if (avctx->request_channels == 2)
1375 avctx->request_channel_layout = AV_CH_LAYOUT_STEREO;
1376 FF_ENABLE_DEPRECATION_WARNINGS
1377 #endif
1378 if (avctx->channels > 2 &&
1379 avctx->request_channel_layout == AV_CH_LAYOUT_STEREO)
1380 avctx->channels = 2;
1381
1382 return 0;
1383 }
1384
1385 static av_cold int dca_decode_end(AVCodecContext *avctx)
1386 {
1387 DCAContext *s = avctx->priv_data;
1388 ff_mdct_end(&s->imdct);
1389 av_freep(&s->extra_channels_buffer);
1390 return 0;
1391 }
1392
1393 static const AVProfile profiles[] = {
1394 { FF_PROFILE_DTS, "DTS" },
1395 { FF_PROFILE_DTS_ES, "DTS-ES" },
1396 { FF_PROFILE_DTS_96_24, "DTS 96/24" },
1397 { FF_PROFILE_DTS_HD_HRA, "DTS-HD HRA" },
1398 { FF_PROFILE_DTS_HD_MA, "DTS-HD MA" },
1399 { FF_PROFILE_UNKNOWN },
1400 };
1401
1402 static const AVOption options[] = {
1403 { "disable_xch", "disable decoding of the XCh extension", offsetof(DCAContext, xch_disable), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM },
1404 { NULL },
1405 };
1406
1407 static const AVClass dca_decoder_class = {
1408 .class_name = "DCA decoder",
1409 .item_name = av_default_item_name,
1410 .option = options,
1411 .version = LIBAVUTIL_VERSION_INT,
1412 };
1413
1414 AVCodec ff_dca_decoder = {
1415 .name = "dca",
1416 .long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
1417 .type = AVMEDIA_TYPE_AUDIO,
1418 .id = AV_CODEC_ID_DTS,
1419 .priv_data_size = sizeof(DCAContext),
1420 .init = dca_decode_init,
1421 .decode = dca_decode_frame,
1422 .close = dca_decode_end,
1423 .capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
1424 .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
1425 AV_SAMPLE_FMT_NONE },
1426 .profiles = NULL_IF_CONFIG_SMALL(profiles),
1427 .priv_class = &dca_decoder_class,
1428 };