aboutsummaryrefslogtreecommitdiffstats
path: root/libavcodec/dcaenc.c
blob: d6efd250d5389679f480054f304c93ebd8401ae8 (plain) (blame)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
397
398
399
400
401
402
403
404
405
406
407
408
409
410
411
412
413
414
415
416
417
418
419
420
421
422
423
424
425
426
427
428
429
430
431
432
433
434
435
436
437
438
439
440
441
442
443
444
445
446
447
448
449
450
451
452
453
454
455
456
457
458
459
460
461
462
463
464
465
466
467
468
469
470
471
472
473
474
475
476
477
478
479
480
481
482
483
484
485
486
487
488
489
490
491
492
493
494
495
496
497
498
499
500
501
502
503
504
505
506
507
508
509
510
511
512
513
514
515
516
517
518
519
520
521
522
523
524
525
526
527
528
529
530
531
532
533
534
535
536
537
538
539
540
541
542
543
544
545
546
547
548
549
550
551
552
553
554
555
556
557
558
559
560
561
562
563
564
565
566
567
568
569
570
571
572
573
574
575
576
577
578
579
580
581
582
583
584
585
586
587
588
589
590
591
592
593
594
595
596
597
598
599
600
601
/*
 * DCA encoder
 * Copyright (C) 2008 Alexander E. Patrakov
 *               2010 Benjamin Larsson
 *               2011 Xiang Wang
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

#include "libavutil/common.h"
#include "libavutil/avassert.h"
#include "libavutil/audioconvert.h"
#include "avcodec.h"
#include "get_bits.h"
#include "internal.h"
#include "put_bits.h"
#include "dcaenc.h"
#include "dcadata.h"

#undef NDEBUG

#define MAX_CHANNELS 6
#define DCA_SUBBANDS_32 32
#define DCA_MAX_FRAME_SIZE 16383
#define DCA_HEADER_SIZE 13

#define DCA_SUBBANDS 32 ///< Subband activity count
#define QUANTIZER_BITS 16
#define SUBFRAMES 1
#define SUBSUBFRAMES 4
#define PCM_SAMPLES (SUBFRAMES*SUBSUBFRAMES*8)
#define LFE_BITS 8
#define LFE_INTERPOLATION 64
#define LFE_PRESENT 2
#define LFE_MISSING 0

static const int8_t dca_lfe_index[] = {
    1,2,2,2,2,3,2,3,2,3,2,3,1,3,2,3
};

static const int8_t dca_channel_reorder_lfe[][9] = {
    { 0, -1, -1, -1, -1, -1, -1, -1, -1 },
    { 0,  1, -1, -1, -1, -1, -1, -1, -1 },
    { 0,  1, -1, -1, -1, -1, -1, -1, -1 },
    { 0,  1, -1, -1, -1, -1, -1, -1, -1 },
    { 0,  1, -1, -1, -1, -1, -1, -1, -1 },
    { 1,  2,  0, -1, -1, -1, -1, -1, -1 },
    { 0,  1, -1,  2, -1, -1, -1, -1, -1 },
    { 1,  2,  0, -1,  3, -1, -1, -1, -1 },
    { 0,  1, -1,  2,  3, -1, -1, -1, -1 },
    { 1,  2,  0, -1,  3,  4, -1, -1, -1 },
    { 2,  3, -1,  0,  1,  4,  5, -1, -1 },
    { 1,  2,  0, -1,  3,  4,  5, -1, -1 },
    { 0, -1,  4,  5,  2,  3,  1, -1, -1 },
    { 3,  4,  1, -1,  0,  2,  5,  6, -1 },
    { 2,  3, -1,  5,  7,  0,  1,  4,  6 },
    { 3,  4,  1, -1,  0,  2,  5,  7,  6 },
};

static const int8_t dca_channel_reorder_nolfe[][9] = {
    { 0, -1, -1, -1, -1, -1, -1, -1, -1 },
    { 0,  1, -1, -1, -1, -1, -1, -1, -1 },
    { 0,  1, -1, -1, -1, -1, -1, -1, -1 },
    { 0,  1, -1, -1, -1, -1, -1, -1, -1 },
    { 0,  1, -1, -1, -1, -1, -1, -1, -1 },
    { 1,  2,  0, -1, -1, -1, -1, -1, -1 },
    { 0,  1,  2, -1, -1, -1, -1, -1, -1 },
    { 1,  2,  0,  3, -1, -1, -1, -1, -1 },
    { 0,  1,  2,  3, -1, -1, -1, -1, -1 },
    { 1,  2,  0,  3,  4, -1, -1, -1, -1 },
    { 2,  3,  0,  1,  4,  5, -1, -1, -1 },
    { 1,  2,  0,  3,  4,  5, -1, -1, -1 },
    { 0,  4,  5,  2,  3,  1, -1, -1, -1 },
    { 3,  4,  1,  0,  2,  5,  6, -1, -1 },
    { 2,  3,  5,  7,  0,  1,  4,  6, -1 },
    { 3,  4,  1,  0,  2,  5,  7,  6, -1 },
};

typedef struct {
    PutBitContext pb;
    int32_t history[MAX_CHANNELS][512]; /* This is a circular buffer */
    int start[MAX_CHANNELS];
    int frame_size;
    int prim_channels;
    int lfe_channel;
    int sample_rate_code;
    int scale_factor[MAX_CHANNELS][DCA_SUBBANDS_32];
    int lfe_scale_factor;
    int lfe_data[SUBFRAMES*SUBSUBFRAMES*4];

    int a_mode;                         ///< audio channels arrangement
    int num_channel;
    int lfe_state;
    int lfe_offset;
    const int8_t *channel_order_tab;    ///< channel reordering table, lfe and non lfe

    int32_t pcm[FFMAX(LFE_INTERPOLATION, DCA_SUBBANDS_32)];
    int32_t subband[PCM_SAMPLES][MAX_CHANNELS][DCA_SUBBANDS_32]; /* [sample][channel][subband] */
} DCAContext;

static int32_t cos_table[128];

static inline int32_t mul32(int32_t a, int32_t b)
{
    int64_t r = (int64_t) a * b;
    /* round the result before truncating - improves accuracy */
    return (r + 0x80000000) >> 32;
}

/* Integer version of the cosine modulated Pseudo QMF */

static void qmf_init(void)
{
    int i;
    int32_t c[17], s[17];
    s[0] = 0;           /* sin(index * PI / 64) * 0x7fffffff */
    c[0] = 0x7fffffff;  /* cos(index * PI / 64) * 0x7fffffff */

    for (i = 1; i <= 16; i++) {
        s[i] = 2 * (mul32(c[i - 1], 105372028)  + mul32(s[i - 1], 2144896908));
        c[i] = 2 * (mul32(c[i - 1], 2144896908) - mul32(s[i - 1], 105372028));
    }

    for (i = 0; i < 16; i++) {
        cos_table[i      ]  =  c[i]      >> 3; /* avoid output overflow */
        cos_table[i +  16]  =  s[16 - i] >> 3;
        cos_table[i +  32]  = -s[i]      >> 3;
        cos_table[i +  48]  = -c[16 - i] >> 3;
        cos_table[i +  64]  = -c[i]      >> 3;
        cos_table[i +  80]  = -s[16 - i] >> 3;
        cos_table[i +  96]  =  s[i]      >> 3;
        cos_table[i + 112]  =  c[16 - i] >> 3;
    }
}

static int32_t band_delta_factor(int band, int sample_num)
{
    int index = band * (2 * sample_num + 1);
    if (band == 0)
        return 0x07ffffff;
    else
        return cos_table[index & 127];
}

static void add_new_samples(DCAContext *c, const int32_t *in,
                            int count, int channel)
{
    int i;

    /* Place new samples into the history buffer */
    for (i = 0; i < count; i++) {
        c->history[channel][c->start[channel] + i] = in[i];
        av_assert0(c->start[channel] + i < 512);
    }
    c->start[channel] += count;
    if (c->start[channel] == 512)
        c->start[channel] = 0;
    av_assert0(c->start[channel] < 512);
}

static void qmf_decompose(DCAContext *c, int32_t in[32], int32_t out[32],
                          int channel)
{
    int band, i, j, k;
    int32_t resp;
    int32_t accum[DCA_SUBBANDS_32] = {0};

    add_new_samples(c, in, DCA_SUBBANDS_32, channel);

    /* Calculate the dot product of the signal with the (possibly inverted)
       reference decoder's response to this vector:
       (0.0, 0.0, ..., 0.0, -1.0, 1.0, 0.0, ..., 0.0)
       so that -1.0 cancels 1.0 from the previous step */

    for (k = 48, j = 0, i = c->start[channel]; i < 512; k++, j++, i++)
        accum[(k & 32) ? (31 - (k & 31)) : (k & 31)] += mul32(c->history[channel][i], UnQMF[j]);
    for (i = 0; i < c->start[channel]; k++, j++, i++)
        accum[(k & 32) ? (31 - (k & 31)) : (k & 31)] += mul32(c->history[channel][i], UnQMF[j]);

    resp = 0;
    /* TODO: implement FFT instead of this naive calculation */
    for (band = 0; band < DCA_SUBBANDS_32; band++) {
        for (j = 0; j < 32; j++)
            resp += mul32(accum[j], band_delta_factor(band, j));

        out[band] = (band & 2) ? (-resp) : resp;
    }
}

static int32_t lfe_fir_64i[512];
static int lfe_downsample(DCAContext *c, int32_t in[LFE_INTERPOLATION])
{
    int i, j;
    int channel = c->prim_channels;
    int32_t accum = 0;

    add_new_samples(c, in, LFE_INTERPOLATION, channel);
    for (i = c->start[channel], j = 0; i < 512; i++, j++)
        accum += mul32(c->history[channel][i], lfe_fir_64i[j]);
    for (i = 0; i < c->start[channel]; i++, j++)
        accum += mul32(c->history[channel][i], lfe_fir_64i[j]);
    return accum;
}

static void init_lfe_fir(void)
{
    static int initialized = 0;
    int i;
    if (initialized)
        return;

    for (i = 0; i < 512; i++)
        lfe_fir_64i[i] = lfe_fir_64[i] * (1 << 25); //float -> int32_t
    initialized = 1;
}

static void put_frame_header(DCAContext *c)
{
    /* SYNC */
    put_bits(&c->pb, 16, 0x7ffe);
    put_bits(&c->pb, 16, 0x8001);

    /* Frame type: normal */
    put_bits(&c->pb, 1, 1);

    /* Deficit sample count: none */
    put_bits(&c->pb, 5, 31);

    /* CRC is not present */
    put_bits(&c->pb, 1, 0);

    /* Number of PCM sample blocks */
    put_bits(&c->pb, 7, PCM_SAMPLES-1);

    /* Primary frame byte size */
    put_bits(&c->pb, 14, c->frame_size-1);

    /* Audio channel arrangement: L + R (stereo) */
    put_bits(&c->pb, 6, c->num_channel);

    /* Core audio sampling frequency */
    put_bits(&c->pb, 4, c->sample_rate_code);

    /* Transmission bit rate: 1411.2 kbps */
    put_bits(&c->pb, 5, 0x16); /* FIXME: magic number */

    /* Embedded down mix: disabled */
    put_bits(&c->pb, 1, 0);

    /* Embedded dynamic range flag: not present */
    put_bits(&c->pb, 1, 0);

    /* Embedded time stamp flag: not present */
    put_bits(&c->pb, 1, 0);

    /* Auxiliary data flag: not present */
    put_bits(&c->pb, 1, 0);

    /* HDCD source: no */
    put_bits(&c->pb, 1, 0);

    /* Extension audio ID: N/A */
    put_bits(&c->pb, 3, 0);

    /* Extended audio data: not present */
    put_bits(&c->pb, 1, 0);

    /* Audio sync word insertion flag: after each sub-frame */
    put_bits(&c->pb, 1, 0);

    /* Low frequency effects flag: not present or interpolation factor=64 */
    put_bits(&c->pb, 2, c->lfe_state);

    /* Predictor history switch flag: on */
    put_bits(&c->pb, 1, 1);

    /* No CRC */
    /* Multirate interpolator switch: non-perfect reconstruction */
    put_bits(&c->pb, 1, 0);

    /* Encoder software revision: 7 */
    put_bits(&c->pb, 4, 7);

    /* Copy history: 0 */
    put_bits(&c->pb, 2, 0);

    /* Source PCM resolution: 16 bits, not DTS ES */
    put_bits(&c->pb, 3, 0);

    /* Front sum/difference coding: no */
    put_bits(&c->pb, 1, 0);

    /* Surrounds sum/difference coding: no */
    put_bits(&c->pb, 1, 0);

    /* Dialog normalization: 0 dB */
    put_bits(&c->pb, 4, 0);
}

static void put_primary_audio_header(DCAContext *c)
{
    static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };
    static const int thr[11]    = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };

    int ch, i;
    /* Number of subframes */
    put_bits(&c->pb, 4, SUBFRAMES - 1);

    /* Number of primary audio channels */
    put_bits(&c->pb, 3, c->prim_channels - 1);

    /* Subband activity count */
    for (ch = 0; ch < c->prim_channels; ch++)
        put_bits(&c->pb, 5, DCA_SUBBANDS - 2);

    /* High frequency VQ start subband */
    for (ch = 0; ch < c->prim_channels; ch++)
        put_bits(&c->pb, 5, DCA_SUBBANDS - 1);

    /* Joint intensity coding index: 0, 0 */
    for (ch = 0; ch < c->prim_channels; ch++)
        put_bits(&c->pb, 3, 0);

    /* Transient mode codebook: A4, A4 (arbitrary) */
    for (ch = 0; ch < c->prim_channels; ch++)
        put_bits(&c->pb, 2, 0);

    /* Scale factor code book: 7 bit linear, 7-bit sqrt table (for each channel) */
    for (ch = 0; ch < c->prim_channels; ch++)
        put_bits(&c->pb, 3, 6);

    /* Bit allocation quantizer select: linear 5-bit */
    for (ch = 0; ch < c->prim_channels; ch++)
        put_bits(&c->pb, 3, 6);

    /* Quantization index codebook select: dummy data
       to avoid transmission of scale factor adjustment */

    for (i = 1; i < 11; i++)
        for (ch = 0; ch < c->prim_channels; ch++)
            put_bits(&c->pb, bitlen[i], thr[i]);

    /* Scale factor adjustment index: not transmitted */
}

/**
 * 8-23 bits quantization
 * @param sample
 * @param bits
 */
static inline uint32_t quantize(int32_t sample, int bits)
{
    av_assert0(sample <    1 << (bits - 1));
    av_assert0(sample >= -(1 << (bits - 1)));
    return sample & ((1 << bits) - 1);
}

static inline int find_scale_factor7(int64_t max_value, int bits)
{
    int i = 0, j = 128, q;
    max_value = ((max_value << 15) / lossy_quant[bits + 3]) >> (bits - 1);
    while (i < j) {
        q = (i + j) >> 1;
        if (max_value < scale_factor_quant7[q])
            j = q;
        else
            i = q + 1;
    }
    av_assert1(i < 128);
    return i;
}

static inline void put_sample7(DCAContext *c, int64_t sample, int bits,
                               int scale_factor)
{
    sample = (sample << 15) / ((int64_t) lossy_quant[bits + 3] * scale_factor_quant7[scale_factor]);
    put_bits(&c->pb, bits, quantize((int) sample, bits));
}

static void put_subframe(DCAContext *c,
                         int32_t subband_data[8 * SUBSUBFRAMES][MAX_CHANNELS][32],
                         int subframe)
{
    int i, sub, ss, ch, max_value;
    int32_t *lfe_data = c->lfe_data + 4 * SUBSUBFRAMES * subframe;

    /* Subsubframes count */
    put_bits(&c->pb, 2, SUBSUBFRAMES -1);

    /* Partial subsubframe sample count: dummy */
    put_bits(&c->pb, 3, 0);

    /* Prediction mode: no ADPCM, in each channel and subband */
    for (ch = 0; ch < c->prim_channels; ch++)
        for (sub = 0; sub < DCA_SUBBANDS; sub++)
            put_bits(&c->pb, 1, 0);

    /* Prediction VQ addres: not transmitted */
    /* Bit allocation index */
    for (ch = 0; ch < c->prim_channels; ch++)
        for (sub = 0; sub < DCA_SUBBANDS; sub++)
            put_bits(&c->pb, 5, QUANTIZER_BITS+3);

    if (SUBSUBFRAMES > 1) {
        /* Transition mode: none for each channel and subband */
        for (ch = 0; ch < c->prim_channels; ch++)
            for (sub = 0; sub < DCA_SUBBANDS; sub++)
                put_bits(&c->pb, 1, 0); /* codebook A4 */
    }

    /* Determine scale_factor */
    for (ch = 0; ch < c->prim_channels; ch++)
        for (sub = 0; sub < DCA_SUBBANDS; sub++) {
            max_value = 0;
            for (i = 0; i < 8 * SUBSUBFRAMES; i++)
                max_value = FFMAX(max_value, FFABS(subband_data[i][ch][sub]));
            c->scale_factor[ch][sub] = find_scale_factor7(max_value, QUANTIZER_BITS);
        }

    if (c->lfe_channel) {
        max_value = 0;
        for (i = 0; i < 4 * SUBSUBFRAMES; i++)
            max_value = FFMAX(max_value, FFABS(lfe_data[i]));
        c->lfe_scale_factor = find_scale_factor7(max_value, LFE_BITS);
    }

    /* Scale factors: the same for each channel and subband,
       encoded according to Table D.1.2 */
    for (ch = 0; ch < c->prim_channels; ch++)
        for (sub = 0; sub < DCA_SUBBANDS; sub++)
            put_bits(&c->pb, 7, c->scale_factor[ch][sub]);

    /* Joint subband scale factor codebook select: not transmitted */
    /* Scale factors for joint subband coding: not transmitted */
    /* Stereo down-mix coefficients: not transmitted */
    /* Dynamic range coefficient: not transmitted */
    /* Stde information CRC check word: not transmitted */
    /* VQ encoded high frequency subbands: not transmitted */

    /* LFE data */
    if (c->lfe_channel) {
        for (i = 0; i < 4 * SUBSUBFRAMES; i++)
            put_sample7(c, lfe_data[i], LFE_BITS, c->lfe_scale_factor);
        put_bits(&c->pb, 8, c->lfe_scale_factor);
    }

    /* Audio data (subsubframes) */

    for (ss = 0; ss < SUBSUBFRAMES ; ss++)
        for (ch = 0; ch < c->prim_channels; ch++)
            for (sub = 0; sub < DCA_SUBBANDS; sub++)
                for (i = 0; i < 8; i++)
                    put_sample7(c, subband_data[ss * 8 + i][ch][sub], QUANTIZER_BITS, c->scale_factor[ch][sub]);

    /* DSYNC */
    put_bits(&c->pb, 16, 0xffff);
}

static void put_frame(DCAContext *c,
                      int32_t subband_data[PCM_SAMPLES][MAX_CHANNELS][32],
                      uint8_t *frame)
{
    int i;
    init_put_bits(&c->pb, frame + DCA_HEADER_SIZE, DCA_MAX_FRAME_SIZE-DCA_HEADER_SIZE);

    put_primary_audio_header(c);
    for (i = 0; i < SUBFRAMES; i++)
        put_subframe(c, &subband_data[SUBSUBFRAMES * 8 * i], i);

    flush_put_bits(&c->pb);
    c->frame_size = (put_bits_count(&c->pb) >> 3) + DCA_HEADER_SIZE;

    init_put_bits(&c->pb, frame, DCA_HEADER_SIZE);
    put_frame_header(c);
    flush_put_bits(&c->pb);
}

static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
                        const AVFrame *frame, int *got_packet_ptr)
{
    int i, k, channel;
    DCAContext *c = avctx->priv_data;
    const int16_t *samples;
    int ret, real_channel = 0;

    if ((ret = ff_alloc_packet2(avctx, avpkt, DCA_MAX_FRAME_SIZE + DCA_HEADER_SIZE)))
        return ret;

    samples = (const int16_t *)frame->data[0];
    for (i = 0; i < PCM_SAMPLES; i ++) { /* i is the decimated sample number */
        for (channel = 0; channel < c->prim_channels + 1; channel++) {
            /* Get 32 PCM samples */
            for (k = 0; k < 32; k++) { /* k is the sample number in a 32-sample block */
                c->pcm[k] = samples[avctx->channels * (32 * i + k) + channel] << 16;
            }
            /* Put subband samples into the proper place */
            real_channel = c->channel_order_tab[channel];
            if (real_channel >= 0) {
                qmf_decompose(c, c->pcm, &c->subband[i][real_channel][0], real_channel);
            }
        }
    }

    if (c->lfe_channel) {
        for (i = 0; i < PCM_SAMPLES / 2; i++) {
            for (k = 0; k < LFE_INTERPOLATION; k++) /* k is the sample number in a 32-sample block */
                c->pcm[k] = samples[avctx->channels * (LFE_INTERPOLATION*i+k) + c->lfe_offset] << 16;
            c->lfe_data[i] = lfe_downsample(c, c->pcm);
        }
    }

    put_frame(c, c->subband, avpkt->data);

    avpkt->size     = c->frame_size;
    *got_packet_ptr = 1;
    return 0;
}

static int encode_init(AVCodecContext *avctx)
{
    DCAContext *c = avctx->priv_data;
    int i;
    uint64_t layout = avctx->channel_layout;

    c->prim_channels = avctx->channels;
    c->lfe_channel   = (avctx->channels == 3 || avctx->channels == 6);

    if (!layout) {
        av_log(avctx, AV_LOG_WARNING, "No channel layout specified. The "
                                      "encoder will guess the layout, but it "
                                      "might be incorrect.\n");
        layout = av_get_default_channel_layout(avctx->channels);
    }
    switch (layout) {
    case AV_CH_LAYOUT_STEREO:       c->a_mode = 2; c->num_channel = 2; break;
    case AV_CH_LAYOUT_5POINT0:      c->a_mode = 9; c->num_channel = 9; break;
    case AV_CH_LAYOUT_5POINT1:      c->a_mode = 9; c->num_channel = 9; break;
    case AV_CH_LAYOUT_5POINT0_BACK: c->a_mode = 9; c->num_channel = 9; break;
    case AV_CH_LAYOUT_5POINT1_BACK: c->a_mode = 9; c->num_channel = 9; break;
    default:
    av_log(avctx, AV_LOG_ERROR,
           "Only stereo, 5.0, 5.1 channel layouts supported at the moment!\n");
    return AVERROR_PATCHWELCOME;
    }

    if (c->lfe_channel) {
        init_lfe_fir();
        c->prim_channels--;
        c->channel_order_tab = dca_channel_reorder_lfe[c->a_mode];
        c->lfe_state         = LFE_PRESENT;
        c->lfe_offset        = dca_lfe_index[c->a_mode];
    } else {
        c->channel_order_tab = dca_channel_reorder_nolfe[c->a_mode];
        c->lfe_state         = LFE_MISSING;
    }

    for (i = 0; i < 16; i++) {
        if (dca_sample_rates[i] && (dca_sample_rates[i] == avctx->sample_rate))
            break;
    }
    if (i == 16) {
        av_log(avctx, AV_LOG_ERROR, "Sample rate %iHz not supported, only ", avctx->sample_rate);
        for (i = 0; i < 16; i++)
            av_log(avctx, AV_LOG_ERROR, "%d, ", dca_sample_rates[i]);
        av_log(avctx, AV_LOG_ERROR, "supported.\n");
        return -1;
    }
    c->sample_rate_code = i;

    avctx->frame_size = 32 * PCM_SAMPLES;

    if (!cos_table[127])
        qmf_init();
    return 0;
}

AVCodec ff_dca_encoder = {
    .name           = "dca",
    .type           = AVMEDIA_TYPE_AUDIO,
    .id             = CODEC_ID_DTS,
    .priv_data_size = sizeof(DCAContext),
    .init           = encode_init,
    .encode2        = encode_frame,
    .capabilities   = CODEC_CAP_EXPERIMENTAL,
    .sample_fmts    = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
    .long_name      = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
};