1 /*
2 * DCA encoder
3 * Copyright (C) 2008-2012 Alexander E. Patrakov
4 * 2010 Benjamin Larsson
5 * 2011 Xiang Wang
6 *
7 * This file is part of FFmpeg.
8 *
9 * FFmpeg is free software; you can redistribute it and/or
10 * modify it under the terms of the GNU Lesser General Public
11 * License as published by the Free Software Foundation; either
12 * version 2.1 of the License, or (at your option) any later version.
13 *
14 * FFmpeg is distributed in the hope that it will be useful,
15 * but WITHOUT ANY WARRANTY; without even the implied warranty of
16 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
17 * Lesser General Public License for more details.
18 *
19 * You should have received a copy of the GNU Lesser General Public
20 * License along with FFmpeg; if not, write to the Free Software
21 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22 */
23
24 #define FFT_FLOAT 0
25 #define FFT_FIXED_32 1
26
27 #include "libavutil/avassert.h"
28 #include "libavutil/channel_layout.h"
29 #include "libavutil/common.h"
30 #include "libavutil/ffmath.h"
31 #include "libavutil/mem_internal.h"
32 #include "libavutil/opt.h"
33 #include "avcodec.h"
34 #include "dca.h"
35 #include "dcaadpcm.h"
36 #include "dcamath.h"
37 #include "dca_core.h"
38 #include "dcadata.h"
39 #include "dcaenc.h"
40 #include "fft.h"
41 #include "internal.h"
42 #include "mathops.h"
43 #include "put_bits.h"
44
45 #define MAX_CHANNELS 6
46 #define DCA_MAX_FRAME_SIZE 16384
47 #define DCA_HEADER_SIZE 13
48 #define DCA_LFE_SAMPLES 8
49
50 #define DCAENC_SUBBANDS 32
51 #define SUBFRAMES 1
52 #define SUBSUBFRAMES 2
53 #define SUBBAND_SAMPLES (SUBFRAMES * SUBSUBFRAMES * 8)
54 #define AUBANDS 25
55
56 #define COS_T(x) (c->cos_table[(x) & 2047])
57
58 typedef struct CompressionOptions {
59 int adpcm_mode;
60 } CompressionOptions;
61
62 typedef struct DCAEncContext {
63 AVClass *class;
64 PutBitContext pb;
65 DCAADPCMEncContext adpcm_ctx;
66 FFTContext mdct;
67 CompressionOptions options;
68 int frame_size;
69 int frame_bits;
70 int fullband_channels;
71 int channels;
72 int lfe_channel;
73 int samplerate_index;
74 int bitrate_index;
75 int channel_config;
76 const int32_t *band_interpolation;
77 const int32_t *band_spectrum;
78 int lfe_scale_factor;
79 softfloat lfe_quant;
80 int32_t lfe_peak_cb;
81 const int8_t *channel_order_tab; ///< channel reordering table, lfe and non lfe
82
83 int32_t prediction_mode[MAX_CHANNELS][DCAENC_SUBBANDS];
84 int32_t adpcm_history[MAX_CHANNELS][DCAENC_SUBBANDS][DCA_ADPCM_COEFFS * 2];
85 int32_t history[MAX_CHANNELS][512]; /* This is a circular buffer */
86 int32_t *subband[MAX_CHANNELS][DCAENC_SUBBANDS];
87 int32_t quantized[MAX_CHANNELS][DCAENC_SUBBANDS][SUBBAND_SAMPLES];
88 int32_t peak_cb[MAX_CHANNELS][DCAENC_SUBBANDS];
89 int32_t diff_peak_cb[MAX_CHANNELS][DCAENC_SUBBANDS]; ///< expected peak of residual signal
90 int32_t downsampled_lfe[DCA_LFE_SAMPLES];
91 int32_t masking_curve_cb[SUBSUBFRAMES][256];
92 int32_t bit_allocation_sel[MAX_CHANNELS];
93 int abits[MAX_CHANNELS][DCAENC_SUBBANDS];
94 int scale_factor[MAX_CHANNELS][DCAENC_SUBBANDS];
95 softfloat quant[MAX_CHANNELS][DCAENC_SUBBANDS];
96 int32_t quant_index_sel[MAX_CHANNELS][DCA_CODE_BOOKS];
97 int32_t eff_masking_curve_cb[256];
98 int32_t band_masking_cb[32];
99 int32_t worst_quantization_noise;
100 int32_t worst_noise_ever;
101 int consumed_bits;
102 int consumed_adpcm_bits; ///< Number of bits to transmit ADPCM related info
103
104 int32_t cos_table[2048];
105 int32_t band_interpolation_tab[2][512];
106 int32_t band_spectrum_tab[2][8];
107 int32_t auf[9][AUBANDS][256];
108 int32_t cb_to_add[256];
109 int32_t cb_to_level[2048];
110 int32_t lfe_fir_64i[512];
111 } DCAEncContext;
112
113 /* Transfer function of outer and middle ear, Hz -> dB */
hom(double f)114 static double hom(double f)
115 {
116 double f1 = f / 1000;
117
118 return -3.64 * pow(f1, -0.8)
119 + 6.8 * exp(-0.6 * (f1 - 3.4) * (f1 - 3.4))
120 - 6.0 * exp(-0.15 * (f1 - 8.7) * (f1 - 8.7))
121 - 0.0006 * (f1 * f1) * (f1 * f1);
122 }
123
gammafilter(int i,double f)124 static double gammafilter(int i, double f)
125 {
126 double h = (f - fc[i]) / erb[i];
127
128 h = 1 + h * h;
129 h = 1 / (h * h);
130 return 20 * log10(h);
131 }
132
subband_bufer_alloc(DCAEncContext * c)133 static int subband_bufer_alloc(DCAEncContext *c)
134 {
135 int ch, band;
136 int32_t *bufer = av_calloc(MAX_CHANNELS * DCAENC_SUBBANDS *
137 (SUBBAND_SAMPLES + DCA_ADPCM_COEFFS),
138 sizeof(int32_t));
139 if (!bufer)
140 return AVERROR(ENOMEM);
141
142 /* we need a place for DCA_ADPCM_COEFF samples from previous frame
143 * to calc prediction coefficients for each subband */
144 for (ch = 0; ch < MAX_CHANNELS; ch++) {
145 for (band = 0; band < DCAENC_SUBBANDS; band++) {
146 c->subband[ch][band] = bufer +
147 ch * DCAENC_SUBBANDS * (SUBBAND_SAMPLES + DCA_ADPCM_COEFFS) +
148 band * (SUBBAND_SAMPLES + DCA_ADPCM_COEFFS) + DCA_ADPCM_COEFFS;
149 }
150 }
151 return 0;
152 }
153
subband_bufer_free(DCAEncContext * c)154 static void subband_bufer_free(DCAEncContext *c)
155 {
156 if (c->subband[0][0]) {
157 int32_t *bufer = c->subband[0][0] - DCA_ADPCM_COEFFS;
158 av_free(bufer);
159 c->subband[0][0] = NULL;
160 }
161 }
162
encode_init(AVCodecContext * avctx)163 static int encode_init(AVCodecContext *avctx)
164 {
165 DCAEncContext *c = avctx->priv_data;
166 uint64_t layout = avctx->channel_layout;
167 int i, j, k, min_frame_bits;
168 int ret;
169
170 if ((ret = subband_bufer_alloc(c)) < 0)
171 return ret;
172
173 c->fullband_channels = c->channels = avctx->channels;
174 c->lfe_channel = (avctx->channels == 3 || avctx->channels == 6);
175 c->band_interpolation = c->band_interpolation_tab[1];
176 c->band_spectrum = c->band_spectrum_tab[1];
177 c->worst_quantization_noise = -2047;
178 c->worst_noise_ever = -2047;
179 c->consumed_adpcm_bits = 0;
180
181 if (ff_dcaadpcm_init(&c->adpcm_ctx))
182 return AVERROR(ENOMEM);
183
184 if (!layout) {
185 av_log(avctx, AV_LOG_WARNING, "No channel layout specified. The "
186 "encoder will guess the layout, but it "
187 "might be incorrect.\n");
188 layout = av_get_default_channel_layout(avctx->channels);
189 }
190 switch (layout) {
191 case AV_CH_LAYOUT_MONO: c->channel_config = 0; break;
192 case AV_CH_LAYOUT_STEREO: c->channel_config = 2; break;
193 case AV_CH_LAYOUT_2_2: c->channel_config = 8; break;
194 case AV_CH_LAYOUT_5POINT0: c->channel_config = 9; break;
195 case AV_CH_LAYOUT_5POINT1: c->channel_config = 9; break;
196 default:
197 av_log(avctx, AV_LOG_ERROR, "Unsupported channel layout!\n");
198 return AVERROR_PATCHWELCOME;
199 }
200
201 if (c->lfe_channel) {
202 c->fullband_channels--;
203 c->channel_order_tab = channel_reorder_lfe[c->channel_config];
204 } else {
205 c->channel_order_tab = channel_reorder_nolfe[c->channel_config];
206 }
207
208 for (i = 0; i < MAX_CHANNELS; i++) {
209 for (j = 0; j < DCA_CODE_BOOKS; j++) {
210 c->quant_index_sel[i][j] = ff_dca_quant_index_group_size[j];
211 }
212 /* 6 - no Huffman */
213 c->bit_allocation_sel[i] = 6;
214
215 for (j = 0; j < DCAENC_SUBBANDS; j++) {
216 /* -1 - no ADPCM */
217 c->prediction_mode[i][j] = -1;
218 memset(c->adpcm_history[i][j], 0, sizeof(int32_t)*DCA_ADPCM_COEFFS);
219 }
220 }
221
222 for (i = 0; i < 9; i++) {
223 if (sample_rates[i] == avctx->sample_rate)
224 break;
225 }
226 if (i == 9)
227 return AVERROR(EINVAL);
228 c->samplerate_index = i;
229
230 if (avctx->bit_rate < 32000 || avctx->bit_rate > 3840000) {
231 av_log(avctx, AV_LOG_ERROR, "Bit rate %"PRId64" not supported.", avctx->bit_rate);
232 return AVERROR(EINVAL);
233 }
234 for (i = 0; ff_dca_bit_rates[i] < avctx->bit_rate; i++)
235 ;
236 c->bitrate_index = i;
237 c->frame_bits = FFALIGN((avctx->bit_rate * 512 + avctx->sample_rate - 1) / avctx->sample_rate, 32);
238 min_frame_bits = 132 + (493 + 28 * 32) * c->fullband_channels + c->lfe_channel * 72;
239 if (c->frame_bits < min_frame_bits || c->frame_bits > (DCA_MAX_FRAME_SIZE << 3))
240 return AVERROR(EINVAL);
241
242 c->frame_size = (c->frame_bits + 7) / 8;
243
244 avctx->frame_size = 32 * SUBBAND_SAMPLES;
245
246 if ((ret = ff_mdct_init(&c->mdct, 9, 0, 1.0)) < 0)
247 return ret;
248
249 /* Init all tables */
250 c->cos_table[0] = 0x7fffffff;
251 c->cos_table[512] = 0;
252 c->cos_table[1024] = -c->cos_table[0];
253 for (i = 1; i < 512; i++) {
254 c->cos_table[i] = (int32_t)(0x7fffffff * cos(M_PI * i / 1024));
255 c->cos_table[1024-i] = -c->cos_table[i];
256 c->cos_table[1024+i] = -c->cos_table[i];
257 c->cos_table[2048-i] = +c->cos_table[i];
258 }
259
260 for (i = 0; i < 2048; i++)
261 c->cb_to_level[i] = (int32_t)(0x7fffffff * ff_exp10(-0.005 * i));
262
263 for (k = 0; k < 32; k++) {
264 for (j = 0; j < 8; j++) {
265 c->lfe_fir_64i[64 * j + k] = (int32_t)(0xffffff800000ULL * ff_dca_lfe_fir_64[8 * k + j]);
266 c->lfe_fir_64i[64 * (7-j) + (63 - k)] = (int32_t)(0xffffff800000ULL * ff_dca_lfe_fir_64[8 * k + j]);
267 }
268 }
269
270 for (i = 0; i < 512; i++) {
271 c->band_interpolation_tab[0][i] = (int32_t)(0x1000000000ULL * ff_dca_fir_32bands_perfect[i]);
272 c->band_interpolation_tab[1][i] = (int32_t)(0x1000000000ULL * ff_dca_fir_32bands_nonperfect[i]);
273 }
274
275 for (i = 0; i < 9; i++) {
276 for (j = 0; j < AUBANDS; j++) {
277 for (k = 0; k < 256; k++) {
278 double freq = sample_rates[i] * (k + 0.5) / 512;
279
280 c->auf[i][j][k] = (int32_t)(10 * (hom(freq) + gammafilter(j, freq)));
281 }
282 }
283 }
284
285 for (i = 0; i < 256; i++) {
286 double add = 1 + ff_exp10(-0.01 * i);
287 c->cb_to_add[i] = (int32_t)(100 * log10(add));
288 }
289 for (j = 0; j < 8; j++) {
290 double accum = 0;
291 for (i = 0; i < 512; i++) {
292 double reconst = ff_dca_fir_32bands_perfect[i] * ((i & 64) ? (-1) : 1);
293 accum += reconst * cos(2 * M_PI * (i + 0.5 - 256) * (j + 0.5) / 512);
294 }
295 c->band_spectrum_tab[0][j] = (int32_t)(200 * log10(accum));
296 }
297 for (j = 0; j < 8; j++) {
298 double accum = 0;
299 for (i = 0; i < 512; i++) {
300 double reconst = ff_dca_fir_32bands_nonperfect[i] * ((i & 64) ? (-1) : 1);
301 accum += reconst * cos(2 * M_PI * (i + 0.5 - 256) * (j + 0.5) / 512);
302 }
303 c->band_spectrum_tab[1][j] = (int32_t)(200 * log10(accum));
304 }
305
306 return 0;
307 }
308
encode_close(AVCodecContext * avctx)309 static av_cold int encode_close(AVCodecContext *avctx)
310 {
311 DCAEncContext *c = avctx->priv_data;
312 ff_mdct_end(&c->mdct);
313 subband_bufer_free(c);
314 ff_dcaadpcm_free(&c->adpcm_ctx);
315
316 return 0;
317 }
318
subband_transform(DCAEncContext * c,const int32_t * input)319 static void subband_transform(DCAEncContext *c, const int32_t *input)
320 {
321 int ch, subs, i, k, j;
322
323 for (ch = 0; ch < c->fullband_channels; ch++) {
324 /* History is copied because it is also needed for PSY */
325 int32_t hist[512];
326 int hist_start = 0;
327 const int chi = c->channel_order_tab[ch];
328
329 memcpy(hist, &c->history[ch][0], 512 * sizeof(int32_t));
330
331 for (subs = 0; subs < SUBBAND_SAMPLES; subs++) {
332 int32_t accum[64];
333 int32_t resp;
334 int band;
335
336 /* Calculate the convolutions at once */
337 memset(accum, 0, 64 * sizeof(int32_t));
338
339 for (k = 0, i = hist_start, j = 0;
340 i < 512; k = (k + 1) & 63, i++, j++)
341 accum[k] += mul32(hist[i], c->band_interpolation[j]);
342 for (i = 0; i < hist_start; k = (k + 1) & 63, i++, j++)
343 accum[k] += mul32(hist[i], c->band_interpolation[j]);
344
345 for (k = 16; k < 32; k++)
346 accum[k] = accum[k] - accum[31 - k];
347 for (k = 32; k < 48; k++)
348 accum[k] = accum[k] + accum[95 - k];
349
350 for (band = 0; band < 32; band++) {
351 resp = 0;
352 for (i = 16; i < 48; i++) {
353 int s = (2 * band + 1) * (2 * (i + 16) + 1);
354 resp += mul32(accum[i], COS_T(s << 3)) >> 3;
355 }
356
357 c->subband[ch][band][subs] = ((band + 1) & 2) ? -resp : resp;
358 }
359
360 /* Copy in 32 new samples from input */
361 for (i = 0; i < 32; i++)
362 hist[i + hist_start] = input[(subs * 32 + i) * c->channels + chi];
363
364 hist_start = (hist_start + 32) & 511;
365 }
366 }
367 }
368
lfe_downsample(DCAEncContext * c,const int32_t * input)369 static void lfe_downsample(DCAEncContext *c, const int32_t *input)
370 {
371 /* FIXME: make 128x LFE downsampling possible */
372 const int lfech = lfe_index[c->channel_config];
373 int i, j, lfes;
374 int32_t hist[512];
375 int32_t accum;
376 int hist_start = 0;
377
378 memcpy(hist, &c->history[c->channels - 1][0], 512 * sizeof(int32_t));
379
380 for (lfes = 0; lfes < DCA_LFE_SAMPLES; lfes++) {
381 /* Calculate the convolution */
382 accum = 0;
383
384 for (i = hist_start, j = 0; i < 512; i++, j++)
385 accum += mul32(hist[i], c->lfe_fir_64i[j]);
386 for (i = 0; i < hist_start; i++, j++)
387 accum += mul32(hist[i], c->lfe_fir_64i[j]);
388
389 c->downsampled_lfe[lfes] = accum;
390
391 /* Copy in 64 new samples from input */
392 for (i = 0; i < 64; i++)
393 hist[i + hist_start] = input[(lfes * 64 + i) * c->channels + lfech];
394
395 hist_start = (hist_start + 64) & 511;
396 }
397 }
398
get_cb(DCAEncContext * c,int32_t in)399 static int32_t get_cb(DCAEncContext *c, int32_t in)
400 {
401 int i, res = 0;
402 in = FFABS(in);
403
404 for (i = 1024; i > 0; i >>= 1) {
405 if (c->cb_to_level[i + res] >= in)
406 res += i;
407 }
408 return -res;
409 }
410
add_cb(DCAEncContext * c,int32_t a,int32_t b)411 static int32_t add_cb(DCAEncContext *c, int32_t a, int32_t b)
412 {
413 if (a < b)
414 FFSWAP(int32_t, a, b);
415
416 if (a - b >= 256)
417 return a;
418 return a + c->cb_to_add[a - b];
419 }
420
calc_power(DCAEncContext * c,const int32_t in[2* 256],int32_t power[256])421 static void calc_power(DCAEncContext *c,
422 const int32_t in[2 * 256], int32_t power[256])
423 {
424 int i;
425 LOCAL_ALIGNED_32(int32_t, data, [512]);
426 LOCAL_ALIGNED_32(int32_t, coeff, [256]);
427
428 for (i = 0; i < 512; i++)
429 data[i] = norm__(mul32(in[i], 0x3fffffff - (COS_T(4 * i + 2) >> 1)), 4);
430
431 c->mdct.mdct_calc(&c->mdct, coeff, data);
432 for (i = 0; i < 256; i++) {
433 const int32_t cb = get_cb(c, coeff[i]);
434 power[i] = add_cb(c, cb, cb);
435 }
436 }
437
adjust_jnd(DCAEncContext * c,const int32_t in[512],int32_t out_cb[256])438 static void adjust_jnd(DCAEncContext *c,
439 const int32_t in[512], int32_t out_cb[256])
440 {
441 int32_t power[256];
442 int32_t out_cb_unnorm[256];
443 int32_t denom;
444 const int32_t ca_cb = -1114;
445 const int32_t cs_cb = 928;
446 const int samplerate_index = c->samplerate_index;
447 int i, j;
448
449 calc_power(c, in, power);
450
451 for (j = 0; j < 256; j++)
452 out_cb_unnorm[j] = -2047; /* and can only grow */
453
454 for (i = 0; i < AUBANDS; i++) {
455 denom = ca_cb; /* and can only grow */
456 for (j = 0; j < 256; j++)
457 denom = add_cb(c, denom, power[j] + c->auf[samplerate_index][i][j]);
458 for (j = 0; j < 256; j++)
459 out_cb_unnorm[j] = add_cb(c, out_cb_unnorm[j],
460 -denom + c->auf[samplerate_index][i][j]);
461 }
462
463 for (j = 0; j < 256; j++)
464 out_cb[j] = add_cb(c, out_cb[j], -out_cb_unnorm[j] - ca_cb - cs_cb);
465 }
466
467 typedef void (*walk_band_t)(DCAEncContext *c, int band1, int band2, int f,
468 int32_t spectrum1, int32_t spectrum2, int channel,
469 int32_t * arg);
470
walk_band_low(DCAEncContext * c,int band,int channel,walk_band_t walk,int32_t * arg)471 static void walk_band_low(DCAEncContext *c, int band, int channel,
472 walk_band_t walk, int32_t *arg)
473 {
474 int f;
475
476 if (band == 0) {
477 for (f = 0; f < 4; f++)
478 walk(c, 0, 0, f, 0, -2047, channel, arg);
479 } else {
480 for (f = 0; f < 8; f++)
481 walk(c, band, band - 1, 8 * band - 4 + f,
482 c->band_spectrum[7 - f], c->band_spectrum[f], channel, arg);
483 }
484 }
485
walk_band_high(DCAEncContext * c,int band,int channel,walk_band_t walk,int32_t * arg)486 static void walk_band_high(DCAEncContext *c, int band, int channel,
487 walk_band_t walk, int32_t *arg)
488 {
489 int f;
490
491 if (band == 31) {
492 for (f = 0; f < 4; f++)
493 walk(c, 31, 31, 256 - 4 + f, 0, -2047, channel, arg);
494 } else {
495 for (f = 0; f < 8; f++)
496 walk(c, band, band + 1, 8 * band + 4 + f,
497 c->band_spectrum[f], c->band_spectrum[7 - f], channel, arg);
498 }
499 }
500
update_band_masking(DCAEncContext * c,int band1,int band2,int f,int32_t spectrum1,int32_t spectrum2,int channel,int32_t * arg)501 static void update_band_masking(DCAEncContext *c, int band1, int band2,
502 int f, int32_t spectrum1, int32_t spectrum2,
503 int channel, int32_t * arg)
504 {
505 int32_t value = c->eff_masking_curve_cb[f] - spectrum1;
506
507 if (value < c->band_masking_cb[band1])
508 c->band_masking_cb[band1] = value;
509 }
510
calc_masking(DCAEncContext * c,const int32_t * input)511 static void calc_masking(DCAEncContext *c, const int32_t *input)
512 {
513 int i, k, band, ch, ssf;
514 int32_t data[512];
515
516 for (i = 0; i < 256; i++)
517 for (ssf = 0; ssf < SUBSUBFRAMES; ssf++)
518 c->masking_curve_cb[ssf][i] = -2047;
519
520 for (ssf = 0; ssf < SUBSUBFRAMES; ssf++)
521 for (ch = 0; ch < c->fullband_channels; ch++) {
522 const int chi = c->channel_order_tab[ch];
523
524 for (i = 0, k = 128 + 256 * ssf; k < 512; i++, k++)
525 data[i] = c->history[ch][k];
526 for (k -= 512; i < 512; i++, k++)
527 data[i] = input[k * c->channels + chi];
528 adjust_jnd(c, data, c->masking_curve_cb[ssf]);
529 }
530 for (i = 0; i < 256; i++) {
531 int32_t m = 2048;
532
533 for (ssf = 0; ssf < SUBSUBFRAMES; ssf++)
534 if (c->masking_curve_cb[ssf][i] < m)
535 m = c->masking_curve_cb[ssf][i];
536 c->eff_masking_curve_cb[i] = m;
537 }
538
539 for (band = 0; band < 32; band++) {
540 c->band_masking_cb[band] = 2048;
541 walk_band_low(c, band, 0, update_band_masking, NULL);
542 walk_band_high(c, band, 0, update_band_masking, NULL);
543 }
544 }
545
find_peak(DCAEncContext * c,const int32_t * in,int len)546 static inline int32_t find_peak(DCAEncContext *c, const int32_t *in, int len)
547 {
548 int sample;
549 int32_t m = 0;
550 for (sample = 0; sample < len; sample++) {
551 int32_t s = abs(in[sample]);
552 if (m < s)
553 m = s;
554 }
555 return get_cb(c, m);
556 }
557
find_peaks(DCAEncContext * c)558 static void find_peaks(DCAEncContext *c)
559 {
560 int band, ch;
561
562 for (ch = 0; ch < c->fullband_channels; ch++) {
563 for (band = 0; band < 32; band++)
564 c->peak_cb[ch][band] = find_peak(c, c->subband[ch][band],
565 SUBBAND_SAMPLES);
566 }
567
568 if (c->lfe_channel)
569 c->lfe_peak_cb = find_peak(c, c->downsampled_lfe, DCA_LFE_SAMPLES);
570 }
571
adpcm_analysis(DCAEncContext * c)572 static void adpcm_analysis(DCAEncContext *c)
573 {
574 int ch, band;
575 int pred_vq_id;
576 int32_t *samples;
577 int32_t estimated_diff[SUBBAND_SAMPLES];
578
579 c->consumed_adpcm_bits = 0;
580 for (ch = 0; ch < c->fullband_channels; ch++) {
581 for (band = 0; band < 32; band++) {
582 samples = c->subband[ch][band] - DCA_ADPCM_COEFFS;
583 pred_vq_id = ff_dcaadpcm_subband_analysis(&c->adpcm_ctx, samples,
584 SUBBAND_SAMPLES, estimated_diff);
585 if (pred_vq_id >= 0) {
586 c->prediction_mode[ch][band] = pred_vq_id;
587 c->consumed_adpcm_bits += 12; //12 bits to transmit prediction vq index
588 c->diff_peak_cb[ch][band] = find_peak(c, estimated_diff, 16);
589 } else {
590 c->prediction_mode[ch][band] = -1;
591 }
592 }
593 }
594 }
595
596 static const int snr_fudge = 128;
597 #define USED_1ABITS 1
598 #define USED_26ABITS 4
599
get_step_size(DCAEncContext * c,int ch,int band)600 static inline int32_t get_step_size(DCAEncContext *c, int ch, int band)
601 {
602 int32_t step_size;
603
604 if (c->bitrate_index == 3)
605 step_size = ff_dca_lossless_quant[c->abits[ch][band]];
606 else
607 step_size = ff_dca_lossy_quant[c->abits[ch][band]];
608
609 return step_size;
610 }
611
calc_one_scale(DCAEncContext * c,int32_t peak_cb,int abits,softfloat * quant)612 static int calc_one_scale(DCAEncContext *c, int32_t peak_cb, int abits,
613 softfloat *quant)
614 {
615 int32_t peak;
616 int our_nscale, try_remove;
617 softfloat our_quant;
618
619 av_assert0(peak_cb <= 0);
620 av_assert0(peak_cb >= -2047);
621
622 our_nscale = 127;
623 peak = c->cb_to_level[-peak_cb];
624
625 for (try_remove = 64; try_remove > 0; try_remove >>= 1) {
626 if (scalefactor_inv[our_nscale - try_remove].e + stepsize_inv[abits].e <= 17)
627 continue;
628 our_quant.m = mul32(scalefactor_inv[our_nscale - try_remove].m, stepsize_inv[abits].m);
629 our_quant.e = scalefactor_inv[our_nscale - try_remove].e + stepsize_inv[abits].e - 17;
630 if ((ff_dca_quant_levels[abits] - 1) / 2 < quantize_value(peak, our_quant))
631 continue;
632 our_nscale -= try_remove;
633 }
634
635 if (our_nscale >= 125)
636 our_nscale = 124;
637
638 quant->m = mul32(scalefactor_inv[our_nscale].m, stepsize_inv[abits].m);
639 quant->e = scalefactor_inv[our_nscale].e + stepsize_inv[abits].e - 17;
640 av_assert0((ff_dca_quant_levels[abits] - 1) / 2 >= quantize_value(peak, *quant));
641
642 return our_nscale;
643 }
644
quantize_adpcm_subband(DCAEncContext * c,int ch,int band)645 static inline void quantize_adpcm_subband(DCAEncContext *c, int ch, int band)
646 {
647 int32_t step_size;
648 int32_t diff_peak_cb = c->diff_peak_cb[ch][band];
649 c->scale_factor[ch][band] = calc_one_scale(c, diff_peak_cb,
650 c->abits[ch][band],
651 &c->quant[ch][band]);
652
653 step_size = get_step_size(c, ch, band);
654 ff_dcaadpcm_do_real(c->prediction_mode[ch][band],
655 c->quant[ch][band],
656 ff_dca_scale_factor_quant7[c->scale_factor[ch][band]],
657 step_size, c->adpcm_history[ch][band], c->subband[ch][band],
658 c->adpcm_history[ch][band] + 4, c->quantized[ch][band],
659 SUBBAND_SAMPLES, c->cb_to_level[-diff_peak_cb]);
660 }
661
quantize_adpcm(DCAEncContext * c)662 static void quantize_adpcm(DCAEncContext *c)
663 {
664 int band, ch;
665
666 for (ch = 0; ch < c->fullband_channels; ch++)
667 for (band = 0; band < 32; band++)
668 if (c->prediction_mode[ch][band] >= 0)
669 quantize_adpcm_subband(c, ch, band);
670 }
671
quantize_pcm(DCAEncContext * c)672 static void quantize_pcm(DCAEncContext *c)
673 {
674 int sample, band, ch;
675
676 for (ch = 0; ch < c->fullband_channels; ch++) {
677 for (band = 0; band < 32; band++) {
678 if (c->prediction_mode[ch][band] == -1) {
679 for (sample = 0; sample < SUBBAND_SAMPLES; sample++) {
680 int32_t val = quantize_value(c->subband[ch][band][sample],
681 c->quant[ch][band]);
682 c->quantized[ch][band][sample] = val;
683 }
684 }
685 }
686 }
687 }
688
accumulate_huff_bit_consumption(int abits,int32_t * quantized,uint32_t * result)689 static void accumulate_huff_bit_consumption(int abits, int32_t *quantized,
690 uint32_t *result)
691 {
692 uint8_t sel, id = abits - 1;
693 for (sel = 0; sel < ff_dca_quant_index_group_size[id]; sel++)
694 result[sel] += ff_dca_vlc_calc_quant_bits(quantized, SUBBAND_SAMPLES,
695 sel, id);
696 }
697
set_best_code(uint32_t vlc_bits[DCA_CODE_BOOKS][7],uint32_t clc_bits[DCA_CODE_BOOKS],int32_t res[DCA_CODE_BOOKS])698 static uint32_t set_best_code(uint32_t vlc_bits[DCA_CODE_BOOKS][7],
699 uint32_t clc_bits[DCA_CODE_BOOKS],
700 int32_t res[DCA_CODE_BOOKS])
701 {
702 uint8_t i, sel;
703 uint32_t best_sel_bits[DCA_CODE_BOOKS];
704 int32_t best_sel_id[DCA_CODE_BOOKS];
705 uint32_t t, bits = 0;
706
707 for (i = 0; i < DCA_CODE_BOOKS; i++) {
708
709 av_assert0(!((!!vlc_bits[i][0]) ^ (!!clc_bits[i])));
710 if (vlc_bits[i][0] == 0) {
711 /* do not transmit adjustment index for empty codebooks */
712 res[i] = ff_dca_quant_index_group_size[i];
713 /* and skip it */
714 continue;
715 }
716
717 best_sel_bits[i] = vlc_bits[i][0];
718 best_sel_id[i] = 0;
719 for (sel = 0; sel < ff_dca_quant_index_group_size[i]; sel++) {
720 if (best_sel_bits[i] > vlc_bits[i][sel] && vlc_bits[i][sel]) {
721 best_sel_bits[i] = vlc_bits[i][sel];
722 best_sel_id[i] = sel;
723 }
724 }
725
726 /* 2 bits to transmit scale factor adjustment index */
727 t = best_sel_bits[i] + 2;
728 if (t < clc_bits[i]) {
729 res[i] = best_sel_id[i];
730 bits += t;
731 } else {
732 res[i] = ff_dca_quant_index_group_size[i];
733 bits += clc_bits[i];
734 }
735 }
736 return bits;
737 }
738
set_best_abits_code(int abits[DCAENC_SUBBANDS],int bands,int32_t * res)739 static uint32_t set_best_abits_code(int abits[DCAENC_SUBBANDS], int bands,
740 int32_t *res)
741 {
742 uint8_t i;
743 uint32_t t;
744 int32_t best_sel = 6;
745 int32_t best_bits = bands * 5;
746
747 /* Check do we have subband which cannot be encoded by Huffman tables */
748 for (i = 0; i < bands; i++) {
749 if (abits[i] > 12 || abits[i] == 0) {
750 *res = best_sel;
751 return best_bits;
752 }
753 }
754
755 for (i = 0; i < DCA_BITALLOC_12_COUNT; i++) {
756 t = ff_dca_vlc_calc_alloc_bits(abits, bands, i);
757 if (t < best_bits) {
758 best_bits = t;
759 best_sel = i;
760 }
761 }
762
763 *res = best_sel;
764 return best_bits;
765 }
766
init_quantization_noise(DCAEncContext * c,int noise,int forbid_zero)767 static int init_quantization_noise(DCAEncContext *c, int noise, int forbid_zero)
768 {
769 int ch, band, ret = USED_26ABITS | USED_1ABITS;
770 uint32_t huff_bit_count_accum[MAX_CHANNELS][DCA_CODE_BOOKS][7];
771 uint32_t clc_bit_count_accum[MAX_CHANNELS][DCA_CODE_BOOKS];
772 uint32_t bits_counter = 0;
773
774 c->consumed_bits = 132 + 333 * c->fullband_channels;
775 c->consumed_bits += c->consumed_adpcm_bits;
776 if (c->lfe_channel)
777 c->consumed_bits += 72;
778
779 /* attempt to guess the bit distribution based on the prevoius frame */
780 for (ch = 0; ch < c->fullband_channels; ch++) {
781 for (band = 0; band < 32; band++) {
782 int snr_cb = c->peak_cb[ch][band] - c->band_masking_cb[band] - noise;
783
784 if (snr_cb >= 1312) {
785 c->abits[ch][band] = 26;
786 ret &= ~USED_1ABITS;
787 } else if (snr_cb >= 222) {
788 c->abits[ch][band] = 8 + mul32(snr_cb - 222, 69000000);
789 ret &= ~(USED_26ABITS | USED_1ABITS);
790 } else if (snr_cb >= 0) {
791 c->abits[ch][band] = 2 + mul32(snr_cb, 106000000);
792 ret &= ~(USED_26ABITS | USED_1ABITS);
793 } else if (forbid_zero || snr_cb >= -140) {
794 c->abits[ch][band] = 1;
795 ret &= ~USED_26ABITS;
796 } else {
797 c->abits[ch][band] = 0;
798 ret &= ~(USED_26ABITS | USED_1ABITS);
799 }
800 }
801 c->consumed_bits += set_best_abits_code(c->abits[ch], 32,
802 &c->bit_allocation_sel[ch]);
803 }
804
805 /* Recalc scale_factor each time to get bits consumption in case of Huffman coding.
806 It is suboptimal solution */
807 /* TODO: May be cache scaled values */
808 for (ch = 0; ch < c->fullband_channels; ch++) {
809 for (band = 0; band < 32; band++) {
810 if (c->prediction_mode[ch][band] == -1) {
811 c->scale_factor[ch][band] = calc_one_scale(c, c->peak_cb[ch][band],
812 c->abits[ch][band],
813 &c->quant[ch][band]);
814 }
815 }
816 }
817 quantize_adpcm(c);
818 quantize_pcm(c);
819
820 memset(huff_bit_count_accum, 0, MAX_CHANNELS * DCA_CODE_BOOKS * 7 * sizeof(uint32_t));
821 memset(clc_bit_count_accum, 0, MAX_CHANNELS * DCA_CODE_BOOKS * sizeof(uint32_t));
822 for (ch = 0; ch < c->fullband_channels; ch++) {
823 for (band = 0; band < 32; band++) {
824 if (c->abits[ch][band] && c->abits[ch][band] <= DCA_CODE_BOOKS) {
825 accumulate_huff_bit_consumption(c->abits[ch][band],
826 c->quantized[ch][band],
827 huff_bit_count_accum[ch][c->abits[ch][band] - 1]);
828 clc_bit_count_accum[ch][c->abits[ch][band] - 1] += bit_consumption[c->abits[ch][band]];
829 } else {
830 bits_counter += bit_consumption[c->abits[ch][band]];
831 }
832 }
833 }
834
835 for (ch = 0; ch < c->fullband_channels; ch++) {
836 bits_counter += set_best_code(huff_bit_count_accum[ch],
837 clc_bit_count_accum[ch],
838 c->quant_index_sel[ch]);
839 }
840
841 c->consumed_bits += bits_counter;
842
843 return ret;
844 }
845
assign_bits(DCAEncContext * c)846 static void assign_bits(DCAEncContext *c)
847 {
848 /* Find the bounds where the binary search should work */
849 int low, high, down;
850 int used_abits = 0;
851 int forbid_zero = 1;
852 restart:
853 init_quantization_noise(c, c->worst_quantization_noise, forbid_zero);
854 low = high = c->worst_quantization_noise;
855 if (c->consumed_bits > c->frame_bits) {
856 while (c->consumed_bits > c->frame_bits) {
857 if (used_abits == USED_1ABITS && forbid_zero) {
858 forbid_zero = 0;
859 goto restart;
860 }
861 low = high;
862 high += snr_fudge;
863 used_abits = init_quantization_noise(c, high, forbid_zero);
864 }
865 } else {
866 while (c->consumed_bits <= c->frame_bits) {
867 high = low;
868 if (used_abits == USED_26ABITS)
869 goto out; /* The requested bitrate is too high, pad with zeros */
870 low -= snr_fudge;
871 used_abits = init_quantization_noise(c, low, forbid_zero);
872 }
873 }
874
875 /* Now do a binary search between low and high to see what fits */
876 for (down = snr_fudge >> 1; down; down >>= 1) {
877 init_quantization_noise(c, high - down, forbid_zero);
878 if (c->consumed_bits <= c->frame_bits)
879 high -= down;
880 }
881 init_quantization_noise(c, high, forbid_zero);
882 out:
883 c->worst_quantization_noise = high;
884 if (high > c->worst_noise_ever)
885 c->worst_noise_ever = high;
886 }
887
shift_history(DCAEncContext * c,const int32_t * input)888 static void shift_history(DCAEncContext *c, const int32_t *input)
889 {
890 int k, ch;
891
892 for (k = 0; k < 512; k++)
893 for (ch = 0; ch < c->channels; ch++) {
894 const int chi = c->channel_order_tab[ch];
895
896 c->history[ch][k] = input[k * c->channels + chi];
897 }
898 }
899
fill_in_adpcm_bufer(DCAEncContext * c)900 static void fill_in_adpcm_bufer(DCAEncContext *c)
901 {
902 int ch, band;
903 int32_t step_size;
904 /* We fill in ADPCM work buffer for subbands which hasn't been ADPCM coded
905 * in current frame - we need this data if subband of next frame is
906 * ADPCM
907 */
908 for (ch = 0; ch < c->channels; ch++) {
909 for (band = 0; band < 32; band++) {
910 int32_t *samples = c->subband[ch][band] - DCA_ADPCM_COEFFS;
911 if (c->prediction_mode[ch][band] == -1) {
912 step_size = get_step_size(c, ch, band);
913
914 ff_dca_core_dequantize(c->adpcm_history[ch][band],
915 c->quantized[ch][band]+12, step_size,
916 ff_dca_scale_factor_quant7[c->scale_factor[ch][band]], 0, 4);
917 } else {
918 AV_COPY128U(c->adpcm_history[ch][band], c->adpcm_history[ch][band]+4);
919 }
920 /* Copy dequantized values for LPC analysis.
921 * It reduces artifacts in case of extreme quantization,
922 * example: in current frame abits is 1 and has no prediction flag,
923 * but end of this frame is sine like signal. In this case, if LPC analysis uses
924 * original values, likely LPC analysis returns good prediction gain, and sets prediction flag.
925 * But there are no proper value in decoder history, so likely result will be no good.
926 * Bitstream has "Predictor history flag switch", but this flag disables history for all subbands
927 */
928 samples[0] = c->adpcm_history[ch][band][0] * (1 << 7);
929 samples[1] = c->adpcm_history[ch][band][1] * (1 << 7);
930 samples[2] = c->adpcm_history[ch][band][2] * (1 << 7);
931 samples[3] = c->adpcm_history[ch][band][3] * (1 << 7);
932 }
933 }
934 }
935
calc_lfe_scales(DCAEncContext * c)936 static void calc_lfe_scales(DCAEncContext *c)
937 {
938 if (c->lfe_channel)
939 c->lfe_scale_factor = calc_one_scale(c, c->lfe_peak_cb, 11, &c->lfe_quant);
940 }
941
put_frame_header(DCAEncContext * c)942 static void put_frame_header(DCAEncContext *c)
943 {
944 /* SYNC */
945 put_bits(&c->pb, 16, 0x7ffe);
946 put_bits(&c->pb, 16, 0x8001);
947
948 /* Frame type: normal */
949 put_bits(&c->pb, 1, 1);
950
951 /* Deficit sample count: none */
952 put_bits(&c->pb, 5, 31);
953
954 /* CRC is not present */
955 put_bits(&c->pb, 1, 0);
956
957 /* Number of PCM sample blocks */
958 put_bits(&c->pb, 7, SUBBAND_SAMPLES - 1);
959
960 /* Primary frame byte size */
961 put_bits(&c->pb, 14, c->frame_size - 1);
962
963 /* Audio channel arrangement */
964 put_bits(&c->pb, 6, c->channel_config);
965
966 /* Core audio sampling frequency */
967 put_bits(&c->pb, 4, bitstream_sfreq[c->samplerate_index]);
968
969 /* Transmission bit rate */
970 put_bits(&c->pb, 5, c->bitrate_index);
971
972 /* Embedded down mix: disabled */
973 put_bits(&c->pb, 1, 0);
974
975 /* Embedded dynamic range flag: not present */
976 put_bits(&c->pb, 1, 0);
977
978 /* Embedded time stamp flag: not present */
979 put_bits(&c->pb, 1, 0);
980
981 /* Auxiliary data flag: not present */
982 put_bits(&c->pb, 1, 0);
983
984 /* HDCD source: no */
985 put_bits(&c->pb, 1, 0);
986
987 /* Extension audio ID: N/A */
988 put_bits(&c->pb, 3, 0);
989
990 /* Extended audio data: not present */
991 put_bits(&c->pb, 1, 0);
992
993 /* Audio sync word insertion flag: after each sub-frame */
994 put_bits(&c->pb, 1, 0);
995
996 /* Low frequency effects flag: not present or 64x subsampling */
997 put_bits(&c->pb, 2, c->lfe_channel ? 2 : 0);
998
999 /* Predictor history switch flag: on */
1000 put_bits(&c->pb, 1, 1);
1001
1002 /* No CRC */
1003 /* Multirate interpolator switch: non-perfect reconstruction */
1004 put_bits(&c->pb, 1, 0);
1005
1006 /* Encoder software revision: 7 */
1007 put_bits(&c->pb, 4, 7);
1008
1009 /* Copy history: 0 */
1010 put_bits(&c->pb, 2, 0);
1011
1012 /* Source PCM resolution: 16 bits, not DTS ES */
1013 put_bits(&c->pb, 3, 0);
1014
1015 /* Front sum/difference coding: no */
1016 put_bits(&c->pb, 1, 0);
1017
1018 /* Surrounds sum/difference coding: no */
1019 put_bits(&c->pb, 1, 0);
1020
1021 /* Dialog normalization: 0 dB */
1022 put_bits(&c->pb, 4, 0);
1023 }
1024
put_primary_audio_header(DCAEncContext * c)1025 static void put_primary_audio_header(DCAEncContext *c)
1026 {
1027 int ch, i;
1028 /* Number of subframes */
1029 put_bits(&c->pb, 4, SUBFRAMES - 1);
1030
1031 /* Number of primary audio channels */
1032 put_bits(&c->pb, 3, c->fullband_channels - 1);
1033
1034 /* Subband activity count */
1035 for (ch = 0; ch < c->fullband_channels; ch++)
1036 put_bits(&c->pb, 5, DCAENC_SUBBANDS - 2);
1037
1038 /* High frequency VQ start subband */
1039 for (ch = 0; ch < c->fullband_channels; ch++)
1040 put_bits(&c->pb, 5, DCAENC_SUBBANDS - 1);
1041
1042 /* Joint intensity coding index: 0, 0 */
1043 for (ch = 0; ch < c->fullband_channels; ch++)
1044 put_bits(&c->pb, 3, 0);
1045
1046 /* Transient mode codebook: A4, A4 (arbitrary) */
1047 for (ch = 0; ch < c->fullband_channels; ch++)
1048 put_bits(&c->pb, 2, 0);
1049
1050 /* Scale factor code book: 7 bit linear, 7-bit sqrt table (for each channel) */
1051 for (ch = 0; ch < c->fullband_channels; ch++)
1052 put_bits(&c->pb, 3, 6);
1053
1054 /* Bit allocation quantizer select: linear 5-bit */
1055 for (ch = 0; ch < c->fullband_channels; ch++)
1056 put_bits(&c->pb, 3, c->bit_allocation_sel[ch]);
1057
1058 /* Quantization index codebook select */
1059 for (i = 0; i < DCA_CODE_BOOKS; i++)
1060 for (ch = 0; ch < c->fullband_channels; ch++)
1061 put_bits(&c->pb, ff_dca_quant_index_sel_nbits[i], c->quant_index_sel[ch][i]);
1062
1063 /* Scale factor adjustment index: transmitted in case of Huffman coding */
1064 for (i = 0; i < DCA_CODE_BOOKS; i++)
1065 for (ch = 0; ch < c->fullband_channels; ch++)
1066 if (c->quant_index_sel[ch][i] < ff_dca_quant_index_group_size[i])
1067 put_bits(&c->pb, 2, 0);
1068
1069 /* Audio header CRC check word: not transmitted */
1070 }
1071
put_subframe_samples(DCAEncContext * c,int ss,int band,int ch)1072 static void put_subframe_samples(DCAEncContext *c, int ss, int band, int ch)
1073 {
1074 int i, j, sum, bits, sel;
1075 if (c->abits[ch][band] <= DCA_CODE_BOOKS) {
1076 av_assert0(c->abits[ch][band] > 0);
1077 sel = c->quant_index_sel[ch][c->abits[ch][band] - 1];
1078 // Huffman codes
1079 if (sel < ff_dca_quant_index_group_size[c->abits[ch][band] - 1]) {
1080 ff_dca_vlc_enc_quant(&c->pb, &c->quantized[ch][band][ss * 8], 8,
1081 sel, c->abits[ch][band] - 1);
1082 return;
1083 }
1084
1085 // Block codes
1086 if (c->abits[ch][band] <= 7) {
1087 for (i = 0; i < 8; i += 4) {
1088 sum = 0;
1089 for (j = 3; j >= 0; j--) {
1090 sum *= ff_dca_quant_levels[c->abits[ch][band]];
1091 sum += c->quantized[ch][band][ss * 8 + i + j];
1092 sum += (ff_dca_quant_levels[c->abits[ch][band]] - 1) / 2;
1093 }
1094 put_bits(&c->pb, bit_consumption[c->abits[ch][band]] / 4, sum);
1095 }
1096 return;
1097 }
1098 }
1099
1100 for (i = 0; i < 8; i++) {
1101 bits = bit_consumption[c->abits[ch][band]] / 16;
1102 put_sbits(&c->pb, bits, c->quantized[ch][band][ss * 8 + i]);
1103 }
1104 }
1105
put_subframe(DCAEncContext * c,int subframe)1106 static void put_subframe(DCAEncContext *c, int subframe)
1107 {
1108 int i, band, ss, ch;
1109
1110 /* Subsubframes count */
1111 put_bits(&c->pb, 2, SUBSUBFRAMES -1);
1112
1113 /* Partial subsubframe sample count: dummy */
1114 put_bits(&c->pb, 3, 0);
1115
1116 /* Prediction mode: no ADPCM, in each channel and subband */
1117 for (ch = 0; ch < c->fullband_channels; ch++)
1118 for (band = 0; band < DCAENC_SUBBANDS; band++)
1119 put_bits(&c->pb, 1, !(c->prediction_mode[ch][band] == -1));
1120
1121 /* Prediction VQ address */
1122 for (ch = 0; ch < c->fullband_channels; ch++)
1123 for (band = 0; band < DCAENC_SUBBANDS; band++)
1124 if (c->prediction_mode[ch][band] >= 0)
1125 put_bits(&c->pb, 12, c->prediction_mode[ch][band]);
1126
1127 /* Bit allocation index */
1128 for (ch = 0; ch < c->fullband_channels; ch++) {
1129 if (c->bit_allocation_sel[ch] == 6) {
1130 for (band = 0; band < DCAENC_SUBBANDS; band++) {
1131 put_bits(&c->pb, 5, c->abits[ch][band]);
1132 }
1133 } else {
1134 ff_dca_vlc_enc_alloc(&c->pb, c->abits[ch], DCAENC_SUBBANDS,
1135 c->bit_allocation_sel[ch]);
1136 }
1137 }
1138
1139 if (SUBSUBFRAMES > 1) {
1140 /* Transition mode: none for each channel and subband */
1141 for (ch = 0; ch < c->fullband_channels; ch++)
1142 for (band = 0; band < DCAENC_SUBBANDS; band++)
1143 if (c->abits[ch][band])
1144 put_bits(&c->pb, 1, 0); /* codebook A4 */
1145 }
1146
1147 /* Scale factors */
1148 for (ch = 0; ch < c->fullband_channels; ch++)
1149 for (band = 0; band < DCAENC_SUBBANDS; band++)
1150 if (c->abits[ch][band])
1151 put_bits(&c->pb, 7, c->scale_factor[ch][band]);
1152
1153 /* Joint subband scale factor codebook select: not transmitted */
1154 /* Scale factors for joint subband coding: not transmitted */
1155 /* Stereo down-mix coefficients: not transmitted */
1156 /* Dynamic range coefficient: not transmitted */
1157 /* Stde information CRC check word: not transmitted */
1158 /* VQ encoded high frequency subbands: not transmitted */
1159
1160 /* LFE data: 8 samples and scalefactor */
1161 if (c->lfe_channel) {
1162 for (i = 0; i < DCA_LFE_SAMPLES; i++)
1163 put_bits(&c->pb, 8, quantize_value(c->downsampled_lfe[i], c->lfe_quant) & 0xff);
1164 put_bits(&c->pb, 8, c->lfe_scale_factor);
1165 }
1166
1167 /* Audio data (subsubframes) */
1168 for (ss = 0; ss < SUBSUBFRAMES ; ss++)
1169 for (ch = 0; ch < c->fullband_channels; ch++)
1170 for (band = 0; band < DCAENC_SUBBANDS; band++)
1171 if (c->abits[ch][band])
1172 put_subframe_samples(c, ss, band, ch);
1173
1174 /* DSYNC */
1175 put_bits(&c->pb, 16, 0xffff);
1176 }
1177
encode_frame(AVCodecContext * avctx,AVPacket * avpkt,const AVFrame * frame,int * got_packet_ptr)1178 static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
1179 const AVFrame *frame, int *got_packet_ptr)
1180 {
1181 DCAEncContext *c = avctx->priv_data;
1182 const int32_t *samples;
1183 int ret, i;
1184
1185 if ((ret = ff_alloc_packet2(avctx, avpkt, c->frame_size, 0)) < 0)
1186 return ret;
1187
1188 samples = (const int32_t *)frame->data[0];
1189
1190 subband_transform(c, samples);
1191 if (c->lfe_channel)
1192 lfe_downsample(c, samples);
1193
1194 calc_masking(c, samples);
1195 if (c->options.adpcm_mode)
1196 adpcm_analysis(c);
1197 find_peaks(c);
1198 assign_bits(c);
1199 calc_lfe_scales(c);
1200 shift_history(c, samples);
1201
1202 init_put_bits(&c->pb, avpkt->data, avpkt->size);
1203 fill_in_adpcm_bufer(c);
1204 put_frame_header(c);
1205 put_primary_audio_header(c);
1206 for (i = 0; i < SUBFRAMES; i++)
1207 put_subframe(c, i);
1208
1209
1210 for (i = put_bits_count(&c->pb); i < 8*c->frame_size; i++)
1211 put_bits(&c->pb, 1, 0);
1212
1213 flush_put_bits(&c->pb);
1214
1215 avpkt->pts = frame->pts;
1216 avpkt->duration = ff_samples_to_time_base(avctx, frame->nb_samples);
1217 avpkt->size = put_bits_count(&c->pb) >> 3;
1218 *got_packet_ptr = 1;
1219 return 0;
1220 }
1221
1222 #define DCAENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
1223
1224 static const AVOption options[] = {
1225 { "dca_adpcm", "Use ADPCM encoding", offsetof(DCAEncContext, options.adpcm_mode), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, DCAENC_FLAGS },
1226 { NULL },
1227 };
1228
1229 static const AVClass dcaenc_class = {
1230 .class_name = "DCA (DTS Coherent Acoustics)",
1231 .item_name = av_default_item_name,
1232 .option = options,
1233 .version = LIBAVUTIL_VERSION_INT,
1234 };
1235
1236 static const AVCodecDefault defaults[] = {
1237 { "b", "1411200" },
1238 { NULL },
1239 };
1240
1241 AVCodec ff_dca_encoder = {
1242 .name = "dca",
1243 .long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
1244 .type = AVMEDIA_TYPE_AUDIO,
1245 .id = AV_CODEC_ID_DTS,
1246 .priv_data_size = sizeof(DCAEncContext),
1247 .init = encode_init,
1248 .close = encode_close,
1249 .encode2 = encode_frame,
1250 .capabilities = AV_CODEC_CAP_EXPERIMENTAL,
1251 .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
1252 .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S32,
1253 AV_SAMPLE_FMT_NONE },
1254 .supported_samplerates = sample_rates,
1255 .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO,
1256 AV_CH_LAYOUT_STEREO,
1257 AV_CH_LAYOUT_2_2,
1258 AV_CH_LAYOUT_5POINT0,
1259 AV_CH_LAYOUT_5POINT1,
1260 0 },
1261 .defaults = defaults,
1262 .priv_class = &dcaenc_class,
1263 };
1264