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dcaenc.c
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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 #include "libavutil/avassert.h"
26 #include "libavutil/common.h"
27 #include "avcodec.h"
28 #include "dca.h"
29 #include "dcadata.h"
30 #include "dcaenc.h"
31 #include "internal.h"
32 #include "mathops.h"
33 #include "put_bits.h"
34 
35 #define MAX_CHANNELS 6
36 #define DCA_MAX_FRAME_SIZE 16384
37 #define DCA_HEADER_SIZE 13
38 #define DCA_LFE_SAMPLES 8
39 
40 #define DCA_SUBBANDS 32
41 #define SUBFRAMES 1
42 #define SUBSUBFRAMES 2
43 #define SUBBAND_SAMPLES (SUBFRAMES * SUBSUBFRAMES * 8)
44 #define AUBANDS 25
45 
46 typedef struct DCAContext {
48  int frame_size;
51  int channels;
58  int lfe_scale_factor;
61 
62  int32_t history[512][MAX_CHANNELS]; /* This is a circular buffer */
76 } DCAContext;
77 
78 static int32_t cos_table[2048];
79 static int32_t band_interpolation[2][512];
80 static int32_t band_spectrum[2][8];
81 static int32_t auf[9][AUBANDS][256];
82 static int32_t cb_to_add[256];
83 static int32_t cb_to_level[2048];
84 static int32_t lfe_fir_64i[512];
85 
86 /* Transfer function of outer and middle ear, Hz -> dB */
87 static double hom(double f)
88 {
89  double f1 = f / 1000;
90 
91  return -3.64 * pow(f1, -0.8)
92  + 6.8 * exp(-0.6 * (f1 - 3.4) * (f1 - 3.4))
93  - 6.0 * exp(-0.15 * (f1 - 8.7) * (f1 - 8.7))
94  - 0.0006 * (f1 * f1) * (f1 * f1);
95 }
96 
97 static double gammafilter(int i, double f)
98 {
99  double h = (f - fc[i]) / erb[i];
100 
101  h = 1 + h * h;
102  h = 1 / (h * h);
103  return 20 * log10(h);
104 }
105 
106 static int encode_init(AVCodecContext *avctx)
107 {
108  DCAContext *c = avctx->priv_data;
109  uint64_t layout = avctx->channel_layout;
110  int i, min_frame_bits;
111 
112  c->fullband_channels = c->channels = avctx->channels;
113  c->lfe_channel = (avctx->channels == 3 || avctx->channels == 6);
116  c->worst_quantization_noise = -2047;
117  c->worst_noise_ever = -2047;
118 
119  if (!layout) {
120  av_log(avctx, AV_LOG_WARNING, "No channel layout specified. The "
121  "encoder will guess the layout, but it "
122  "might be incorrect.\n");
123  layout = av_get_default_channel_layout(avctx->channels);
124  }
125  switch (layout) {
126  case AV_CH_LAYOUT_MONO: c->channel_config = 0; break;
127  case AV_CH_LAYOUT_STEREO: c->channel_config = 2; break;
128  case AV_CH_LAYOUT_2_2: c->channel_config = 8; break;
129  case AV_CH_LAYOUT_5POINT0: c->channel_config = 9; break;
130  case AV_CH_LAYOUT_5POINT1: c->channel_config = 9; break;
131  default:
132  av_log(avctx, AV_LOG_ERROR, "Unsupported channel layout!\n");
133  return AVERROR_PATCHWELCOME;
134  }
135 
136  if (c->lfe_channel)
137  c->fullband_channels--;
138 
139  for (i = 0; i < 9; i++) {
140  if (sample_rates[i] == avctx->sample_rate)
141  break;
142  }
143  if (i == 9)
144  return AVERROR(EINVAL);
145  c->samplerate_index = i;
146 
147  if (avctx->bit_rate < 32000 || avctx->bit_rate > 3840000) {
148  av_log(avctx, AV_LOG_ERROR, "Bit rate %i not supported.", avctx->bit_rate);
149  return AVERROR(EINVAL);
150  }
151  for (i = 0; dca_bit_rates[i] < avctx->bit_rate; i++)
152  ;
153  c->bitrate_index = i;
154  avctx->bit_rate = dca_bit_rates[i];
155  c->frame_bits = FFALIGN((avctx->bit_rate * 512 + avctx->sample_rate - 1) / avctx->sample_rate, 32);
156  min_frame_bits = 132 + (493 + 28 * 32) * c->fullband_channels + c->lfe_channel * 72;
157  if (c->frame_bits < min_frame_bits || c->frame_bits > (DCA_MAX_FRAME_SIZE << 3))
158  return AVERROR(EINVAL);
159 
160  c->frame_size = (c->frame_bits + 7) / 8;
161 
162  avctx->frame_size = 32 * SUBBAND_SAMPLES;
163 
164  if (!cos_table[0]) {
165  int j, k;
166 
167  for (i = 0; i < 2048; i++) {
168  cos_table[i] = (int32_t)(0x7fffffff * cos(M_PI * i / 1024));
169  cb_to_level[i] = (int32_t)(0x7fffffff * pow(10, -0.005 * i));
170  }
171 
172  /* FIXME: probably incorrect */
173  for (i = 0; i < 256; i++) {
174  lfe_fir_64i[i] = (int32_t)(0x01ffffff * lfe_fir_64[i]);
175  lfe_fir_64i[511 - i] = (int32_t)(0x01ffffff * lfe_fir_64[i]);
176  }
177 
178  for (i = 0; i < 512; i++) {
179  band_interpolation[0][i] = (int32_t)(0x1000000000ULL * fir_32bands_perfect[i]);
180  band_interpolation[1][i] = (int32_t)(0x1000000000ULL * fir_32bands_nonperfect[i]);
181  }
182 
183  for (i = 0; i < 9; i++) {
184  for (j = 0; j < AUBANDS; j++) {
185  for (k = 0; k < 256; k++) {
186  double freq = sample_rates[i] * (k + 0.5) / 512;
187 
188  auf[i][j][k] = (int32_t)(10 * (hom(freq) + gammafilter(j, freq)));
189  }
190  }
191  }
192 
193  for (i = 0; i < 256; i++) {
194  double add = 1 + pow(10, -0.01 * i);
195  cb_to_add[i] = (int32_t)(100 * log10(add));
196  }
197  for (j = 0; j < 8; j++) {
198  double accum = 0;
199  for (i = 0; i < 512; i++) {
200  double reconst = fir_32bands_perfect[i] * ((i & 64) ? (-1) : 1);
201  accum += reconst * cos(2 * M_PI * (i + 0.5 - 256) * (j + 0.5) / 512);
202  }
203  band_spectrum[0][j] = (int32_t)(200 * log10(accum));
204  }
205  for (j = 0; j < 8; j++) {
206  double accum = 0;
207  for (i = 0; i < 512; i++) {
208  double reconst = fir_32bands_nonperfect[i] * ((i & 64) ? (-1) : 1);
209  accum += reconst * cos(2 * M_PI * (i + 0.5 - 256) * (j + 0.5) / 512);
210  }
211  band_spectrum[1][j] = (int32_t)(200 * log10(accum));
212  }
213  }
214  return 0;
215 }
216 
217 static inline int32_t cos_t(int x)
218 {
219  return cos_table[x & 2047];
220 }
221 
222 static inline int32_t sin_t(int x)
223 {
224  return cos_t(x - 512);
225 }
226 
227 static inline int32_t half32(int32_t a)
228 {
229  return (a + 1) >> 1;
230 }
231 
232 static inline int32_t mul32(int32_t a, int32_t b)
233 {
234  int64_t r = (int64_t)a * b + 0x80000000ULL;
235  return r >> 32;
236 }
237 
238 static void subband_transform(DCAContext *c, const int32_t *input)
239 {
240  int ch, subs, i, k, j;
241 
242  for (ch = 0; ch < c->fullband_channels; ch++) {
243  /* History is copied because it is also needed for PSY */
244  int32_t hist[512];
245  int hist_start = 0;
246 
247  for (i = 0; i < 512; i++)
248  hist[i] = c->history[i][ch];
249 
250  for (subs = 0; subs < SUBBAND_SAMPLES; subs++) {
251  int32_t accum[64];
252  int32_t resp;
253  int band;
254 
255  /* Calculate the convolutions at once */
256  for (i = 0; i < 64; i++)
257  accum[i] = 0;
258 
259  for (k = 0, i = hist_start, j = 0;
260  i < 512; k = (k + 1) & 63, i++, j++)
261  accum[k] += mul32(hist[i], c->band_interpolation[j]);
262  for (i = 0; i < hist_start; k = (k + 1) & 63, i++, j++)
263  accum[k] += mul32(hist[i], c->band_interpolation[j]);
264 
265  for (k = 16; k < 32; k++)
266  accum[k] = accum[k] - accum[31 - k];
267  for (k = 32; k < 48; k++)
268  accum[k] = accum[k] + accum[95 - k];
269 
270  for (band = 0; band < 32; band++) {
271  resp = 0;
272  for (i = 16; i < 48; i++) {
273  int s = (2 * band + 1) * (2 * (i + 16) + 1);
274  resp += mul32(accum[i], cos_t(s << 3)) >> 3;
275  }
276 
277  c->subband[subs][band][ch] = ((band + 1) & 2) ? -resp : resp;
278  }
279 
280  /* Copy in 32 new samples from input */
281  for (i = 0; i < 32; i++)
282  hist[i + hist_start] = input[(subs * 32 + i) * c->channels + ch];
283  hist_start = (hist_start + 32) & 511;
284  }
285  }
286 }
287 
288 static void lfe_downsample(DCAContext *c, const int32_t *input)
289 {
290  /* FIXME: make 128x LFE downsampling possible */
291  int i, j, lfes;
292  int32_t hist[512];
293  int32_t accum;
294  int hist_start = 0;
295 
296  for (i = 0; i < 512; i++)
297  hist[i] = c->history[i][c->channels - 1];
298 
299  for (lfes = 0; lfes < DCA_LFE_SAMPLES; lfes++) {
300  /* Calculate the convolution */
301  accum = 0;
302 
303  for (i = hist_start, j = 0; i < 512; i++, j++)
304  accum += mul32(hist[i], lfe_fir_64i[j]);
305  for (i = 0; i < hist_start; i++, j++)
306  accum += mul32(hist[i], lfe_fir_64i[j]);
307 
308  c->downsampled_lfe[lfes] = accum;
309 
310  /* Copy in 64 new samples from input */
311  for (i = 0; i < 64; i++)
312  hist[i + hist_start] = input[(lfes * 64 + i) * c->channels + c->channels - 1];
313 
314  hist_start = (hist_start + 64) & 511;
315  }
316 }
317 
318 typedef struct {
321 } cplx32;
322 
323 static void fft(const int32_t in[2 * 256], cplx32 out[256])
324 {
325  cplx32 buf[256], rin[256], rout[256];
326  int i, j, k, l;
327 
328  /* do two transforms in parallel */
329  for (i = 0; i < 256; i++) {
330  /* Apply the Hann window */
331  rin[i].re = mul32(in[2 * i], 0x3fffffff - (cos_t(8 * i + 2) >> 1));
332  rin[i].im = mul32(in[2 * i + 1], 0x3fffffff - (cos_t(8 * i + 6) >> 1));
333  }
334  /* pre-rotation */
335  for (i = 0; i < 256; i++) {
336  buf[i].re = mul32(cos_t(4 * i + 2), rin[i].re)
337  - mul32(sin_t(4 * i + 2), rin[i].im);
338  buf[i].im = mul32(cos_t(4 * i + 2), rin[i].im)
339  + mul32(sin_t(4 * i + 2), rin[i].re);
340  }
341 
342  for (j = 256, l = 1; j != 1; j >>= 1, l <<= 1) {
343  for (k = 0; k < 256; k += j) {
344  for (i = k; i < k + j / 2; i++) {
345  cplx32 sum, diff;
346  int t = 8 * l * i;
347 
348  sum.re = buf[i].re + buf[i + j / 2].re;
349  sum.im = buf[i].im + buf[i + j / 2].im;
350 
351  diff.re = buf[i].re - buf[i + j / 2].re;
352  diff.im = buf[i].im - buf[i + j / 2].im;
353 
354  buf[i].re = half32(sum.re);
355  buf[i].im = half32(sum.im);
356 
357  buf[i + j / 2].re = mul32(diff.re, cos_t(t))
358  - mul32(diff.im, sin_t(t));
359  buf[i + j / 2].im = mul32(diff.im, cos_t(t))
360  + mul32(diff.re, sin_t(t));
361  }
362  }
363  }
364  /* post-rotation */
365  for (i = 0; i < 256; i++) {
366  int b = ff_reverse[i];
367  rout[i].re = mul32(buf[b].re, cos_t(4 * i))
368  - mul32(buf[b].im, sin_t(4 * i));
369  rout[i].im = mul32(buf[b].im, cos_t(4 * i))
370  + mul32(buf[b].re, sin_t(4 * i));
371  }
372  for (i = 0; i < 256; i++) {
373  /* separate the results of the two transforms */
374  cplx32 o1, o2;
375 
376  o1.re = rout[i].re - rout[255 - i].re;
377  o1.im = rout[i].im + rout[255 - i].im;
378 
379  o2.re = rout[i].im - rout[255 - i].im;
380  o2.im = -rout[i].re - rout[255 - i].re;
381 
382  /* combine them into one long transform */
383  out[i].re = mul32( o1.re + o2.re, cos_t(2 * i + 1))
384  + mul32( o1.im - o2.im, sin_t(2 * i + 1));
385  out[i].im = mul32( o1.im + o2.im, cos_t(2 * i + 1))
386  + mul32(-o1.re + o2.re, sin_t(2 * i + 1));
387  }
388 }
389 
391 {
392  int i, res;
393 
394  res = 0;
395  if (in < 0)
396  in = -in;
397  for (i = 1024; i > 0; i >>= 1) {
398  if (cb_to_level[i + res] >= in)
399  res += i;
400  }
401  return -res;
402 }
403 
405 {
406  if (a < b)
407  FFSWAP(int32_t, a, b);
408 
409  if (a - b >= 256)
410  return a;
411  return a + cb_to_add[a - b];
412 }
413 
414 static void adjust_jnd(int samplerate_index,
415  const int32_t in[512], int32_t out_cb[256])
416 {
417  int32_t power[256];
418  cplx32 out[256];
419  int32_t out_cb_unnorm[256];
420  int32_t denom;
421  const int32_t ca_cb = -1114;
422  const int32_t cs_cb = 928;
423  int i, j;
424 
425  fft(in, out);
426 
427  for (j = 0; j < 256; j++) {
428  power[j] = add_cb(get_cb(out[j].re), get_cb(out[j].im));
429  out_cb_unnorm[j] = -2047; /* and can only grow */
430  }
431 
432  for (i = 0; i < AUBANDS; i++) {
433  denom = ca_cb; /* and can only grow */
434  for (j = 0; j < 256; j++)
435  denom = add_cb(denom, power[j] + auf[samplerate_index][i][j]);
436  for (j = 0; j < 256; j++)
437  out_cb_unnorm[j] = add_cb(out_cb_unnorm[j],
438  -denom + auf[samplerate_index][i][j]);
439  }
440 
441  for (j = 0; j < 256; j++)
442  out_cb[j] = add_cb(out_cb[j], -out_cb_unnorm[j] - ca_cb - cs_cb);
443 }
444 
445 typedef void (*walk_band_t)(DCAContext *c, int band1, int band2, int f,
446  int32_t spectrum1, int32_t spectrum2, int channel,
447  int32_t * arg);
448 
449 static void walk_band_low(DCAContext *c, int band, int channel,
450  walk_band_t walk, int32_t *arg)
451 {
452  int f;
453 
454  if (band == 0) {
455  for (f = 0; f < 4; f++)
456  walk(c, 0, 0, f, 0, -2047, channel, arg);
457  } else {
458  for (f = 0; f < 8; f++)
459  walk(c, band, band - 1, 8 * band - 4 + f,
460  c->band_spectrum[7 - f], c->band_spectrum[f], channel, arg);
461  }
462 }
463 
464 static void walk_band_high(DCAContext *c, int band, int channel,
465  walk_band_t walk, int32_t *arg)
466 {
467  int f;
468 
469  if (band == 31) {
470  for (f = 0; f < 4; f++)
471  walk(c, 31, 31, 256 - 4 + f, 0, -2047, channel, arg);
472  } else {
473  for (f = 0; f < 8; f++)
474  walk(c, band, band + 1, 8 * band + 4 + f,
475  c->band_spectrum[f], c->band_spectrum[7 - f], channel, arg);
476  }
477 }
478 
479 static void update_band_masking(DCAContext *c, int band1, int band2,
480  int f, int32_t spectrum1, int32_t spectrum2,
481  int channel, int32_t * arg)
482 {
483  int32_t value = c->eff_masking_curve_cb[f] - spectrum1;
484 
485  if (value < c->band_masking_cb[band1])
486  c->band_masking_cb[band1] = value;
487 }
488 
489 static void calc_masking(DCAContext *c, const int32_t *input)
490 {
491  int i, k, band, ch, ssf;
492  int32_t data[512];
493 
494  for (i = 0; i < 256; i++)
495  for (ssf = 0; ssf < SUBSUBFRAMES; ssf++)
496  c->masking_curve_cb[ssf][i] = -2047;
497 
498  for (ssf = 0; ssf < SUBSUBFRAMES; ssf++)
499  for (ch = 0; ch < c->fullband_channels; ch++) {
500  for (i = 0, k = 128 + 256 * ssf; k < 512; i++, k++)
501  data[i] = c->history[k][ch];
502  for (k -= 512; i < 512; i++, k++)
503  data[i] = input[k * c->channels + ch];
504  adjust_jnd(c->samplerate_index, data, c->masking_curve_cb[ssf]);
505  }
506  for (i = 0; i < 256; i++) {
507  int32_t m = 2048;
508 
509  for (ssf = 0; ssf < SUBSUBFRAMES; ssf++)
510  if (c->masking_curve_cb[ssf][i] < m)
511  m = c->masking_curve_cb[ssf][i];
512  c->eff_masking_curve_cb[i] = m;
513  }
514 
515  for (band = 0; band < 32; band++) {
516  c->band_masking_cb[band] = 2048;
517  walk_band_low(c, band, 0, update_band_masking, NULL);
518  walk_band_high(c, band, 0, update_band_masking, NULL);
519  }
520 }
521 
522 static void find_peaks(DCAContext *c)
523 {
524  int band, ch;
525 
526  for (band = 0; band < 32; band++)
527  for (ch = 0; ch < c->fullband_channels; ch++) {
528  int sample;
529  int32_t m = 0;
530 
531  for (sample = 0; sample < SUBBAND_SAMPLES; sample++) {
532  int32_t s = abs(c->subband[sample][band][ch]);
533  if (m < s)
534  m = s;
535  }
536  c->peak_cb[band][ch] = get_cb(m);
537  }
538 
539  if (c->lfe_channel) {
540  int sample;
541  int32_t m = 0;
542 
543  for (sample = 0; sample < DCA_LFE_SAMPLES; sample++)
544  if (m < abs(c->downsampled_lfe[sample]))
545  m = abs(c->downsampled_lfe[sample]);
546  c->lfe_peak_cb = get_cb(m);
547  }
548 }
549 
550 static const int snr_fudge = 128;
551 #define USED_1ABITS 1
552 #define USED_NABITS 2
553 #define USED_26ABITS 4
554 
556 {
557  int ch, band, ret = 0;
558 
559  c->consumed_bits = 132 + 493 * c->fullband_channels;
560  if (c->lfe_channel)
561  c->consumed_bits += 72;
562 
563  /* attempt to guess the bit distribution based on the prevoius frame */
564  for (ch = 0; ch < c->fullband_channels; ch++) {
565  for (band = 0; band < 32; band++) {
566  int snr_cb = c->peak_cb[band][ch] - c->band_masking_cb[band] - noise;
567 
568  if (snr_cb >= 1312) {
569  c->abits[band][ch] = 26;
570  ret |= USED_26ABITS;
571  } else if (snr_cb >= 222) {
572  c->abits[band][ch] = 8 + mul32(snr_cb - 222, 69000000);
573  ret |= USED_NABITS;
574  } else if (snr_cb >= 0) {
575  c->abits[band][ch] = 2 + mul32(snr_cb, 106000000);
576  ret |= USED_NABITS;
577  } else {
578  c->abits[band][ch] = 1;
579  ret |= USED_1ABITS;
580  }
581  }
582  }
583 
584  for (band = 0; band < 32; band++)
585  for (ch = 0; ch < c->fullband_channels; ch++) {
586  c->consumed_bits += bit_consumption[c->abits[band][ch]];
587  }
588 
589  return ret;
590 }
591 
592 static void assign_bits(DCAContext *c)
593 {
594  /* Find the bounds where the binary search should work */
595  int low, high, down;
596  int used_abits = 0;
597 
599  low = high = c->worst_quantization_noise;
600  if (c->consumed_bits > c->frame_bits) {
601  while (c->consumed_bits > c->frame_bits) {
602  av_assert0(used_abits != USED_1ABITS);
603  low = high;
604  high += snr_fudge;
605  used_abits = init_quantization_noise(c, high);
606  }
607  } else {
608  while (c->consumed_bits <= c->frame_bits) {
609  high = low;
610  if (used_abits == USED_26ABITS)
611  goto out; /* The requested bitrate is too high, pad with zeros */
612  low -= snr_fudge;
613  used_abits = init_quantization_noise(c, low);
614  }
615  }
616 
617  /* Now do a binary search between low and high to see what fits */
618  for (down = snr_fudge >> 1; down; down >>= 1) {
619  init_quantization_noise(c, high - down);
620  if (c->consumed_bits <= c->frame_bits)
621  high -= down;
622  }
623  init_quantization_noise(c, high);
624 out:
625  c->worst_quantization_noise = high;
626  if (high > c->worst_noise_ever)
627  c->worst_noise_ever = high;
628 }
629 
630 static void shift_history(DCAContext *c, const int32_t *input)
631 {
632  int k, ch;
633 
634  for (k = 0; k < 512; k++)
635  for (ch = 0; ch < c->channels; ch++)
636  c->history[k][ch] = input[k * c->channels + ch];
637 }
638 
640 {
641  int32_t offset = 1 << (quant.e - 1);
642 
643  value = mul32(value, quant.m) + offset;
644  value = value >> quant.e;
645  return value;
646 }
647 
648 static int calc_one_scale(int32_t peak_cb, int abits, softfloat *quant)
649 {
650  int32_t peak;
651  int our_nscale, try_remove;
652  softfloat our_quant;
653 
654  av_assert0(peak_cb <= 0);
655  av_assert0(peak_cb >= -2047);
656 
657  our_nscale = 127;
658  peak = cb_to_level[-peak_cb];
659 
660  for (try_remove = 64; try_remove > 0; try_remove >>= 1) {
661  if (scalefactor_inv[our_nscale - try_remove].e + stepsize_inv[abits].e <= 17)
662  continue;
663  our_quant.m = mul32(scalefactor_inv[our_nscale - try_remove].m, stepsize_inv[abits].m);
664  our_quant.e = scalefactor_inv[our_nscale - try_remove].e + stepsize_inv[abits].e - 17;
665  if ((quant_levels[abits] - 1) / 2 < quantize_value(peak, our_quant))
666  continue;
667  our_nscale -= try_remove;
668  }
669 
670  if (our_nscale >= 125)
671  our_nscale = 124;
672 
673  quant->m = mul32(scalefactor_inv[our_nscale].m, stepsize_inv[abits].m);
674  quant->e = scalefactor_inv[our_nscale].e + stepsize_inv[abits].e - 17;
675  av_assert0((quant_levels[abits] - 1) / 2 >= quantize_value(peak, *quant));
676 
677  return our_nscale;
678 }
679 
680 static void calc_scales(DCAContext *c)
681 {
682  int band, ch;
683 
684  for (band = 0; band < 32; band++)
685  for (ch = 0; ch < c->fullband_channels; ch++)
686  c->scale_factor[band][ch] = calc_one_scale(c->peak_cb[band][ch],
687  c->abits[band][ch],
688  &c->quant[band][ch]);
689 
690  if (c->lfe_channel)
692 }
693 
694 static void quantize_all(DCAContext *c)
695 {
696  int sample, band, ch;
697 
698  for (sample = 0; sample < SUBBAND_SAMPLES; sample++)
699  for (band = 0; band < 32; band++)
700  for (ch = 0; ch < c->fullband_channels; ch++)
701  c->quantized[sample][band][ch] = quantize_value(c->subband[sample][band][ch], c->quant[band][ch]);
702 }
703 
705 {
706  /* SYNC */
707  put_bits(&c->pb, 16, 0x7ffe);
708  put_bits(&c->pb, 16, 0x8001);
709 
710  /* Frame type: normal */
711  put_bits(&c->pb, 1, 1);
712 
713  /* Deficit sample count: none */
714  put_bits(&c->pb, 5, 31);
715 
716  /* CRC is not present */
717  put_bits(&c->pb, 1, 0);
718 
719  /* Number of PCM sample blocks */
720  put_bits(&c->pb, 7, SUBBAND_SAMPLES - 1);
721 
722  /* Primary frame byte size */
723  put_bits(&c->pb, 14, c->frame_size - 1);
724 
725  /* Audio channel arrangement */
726  put_bits(&c->pb, 6, c->channel_config);
727 
728  /* Core audio sampling frequency */
730 
731  /* Transmission bit rate */
732  put_bits(&c->pb, 5, c->bitrate_index);
733 
734  /* Embedded down mix: disabled */
735  put_bits(&c->pb, 1, 0);
736 
737  /* Embedded dynamic range flag: not present */
738  put_bits(&c->pb, 1, 0);
739 
740  /* Embedded time stamp flag: not present */
741  put_bits(&c->pb, 1, 0);
742 
743  /* Auxiliary data flag: not present */
744  put_bits(&c->pb, 1, 0);
745 
746  /* HDCD source: no */
747  put_bits(&c->pb, 1, 0);
748 
749  /* Extension audio ID: N/A */
750  put_bits(&c->pb, 3, 0);
751 
752  /* Extended audio data: not present */
753  put_bits(&c->pb, 1, 0);
754 
755  /* Audio sync word insertion flag: after each sub-frame */
756  put_bits(&c->pb, 1, 0);
757 
758  /* Low frequency effects flag: not present or 64x subsampling */
759  put_bits(&c->pb, 2, c->lfe_channel ? 2 : 0);
760 
761  /* Predictor history switch flag: on */
762  put_bits(&c->pb, 1, 1);
763 
764  /* No CRC */
765  /* Multirate interpolator switch: non-perfect reconstruction */
766  put_bits(&c->pb, 1, 0);
767 
768  /* Encoder software revision: 7 */
769  put_bits(&c->pb, 4, 7);
770 
771  /* Copy history: 0 */
772  put_bits(&c->pb, 2, 0);
773 
774  /* Source PCM resolution: 16 bits, not DTS ES */
775  put_bits(&c->pb, 3, 0);
776 
777  /* Front sum/difference coding: no */
778  put_bits(&c->pb, 1, 0);
779 
780  /* Surrounds sum/difference coding: no */
781  put_bits(&c->pb, 1, 0);
782 
783  /* Dialog normalization: 0 dB */
784  put_bits(&c->pb, 4, 0);
785 }
786 
788 {
789  static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };
790  static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };
791 
792  int ch, i;
793  /* Number of subframes */
794  put_bits(&c->pb, 4, SUBFRAMES - 1);
795 
796  /* Number of primary audio channels */
797  put_bits(&c->pb, 3, c->fullband_channels - 1);
798 
799  /* Subband activity count */
800  for (ch = 0; ch < c->fullband_channels; ch++)
801  put_bits(&c->pb, 5, DCA_SUBBANDS - 2);
802 
803  /* High frequency VQ start subband */
804  for (ch = 0; ch < c->fullband_channels; ch++)
805  put_bits(&c->pb, 5, DCA_SUBBANDS - 1);
806 
807  /* Joint intensity coding index: 0, 0 */
808  for (ch = 0; ch < c->fullband_channels; ch++)
809  put_bits(&c->pb, 3, 0);
810 
811  /* Transient mode codebook: A4, A4 (arbitrary) */
812  for (ch = 0; ch < c->fullband_channels; ch++)
813  put_bits(&c->pb, 2, 0);
814 
815  /* Scale factor code book: 7 bit linear, 7-bit sqrt table (for each channel) */
816  for (ch = 0; ch < c->fullband_channels; ch++)
817  put_bits(&c->pb, 3, 6);
818 
819  /* Bit allocation quantizer select: linear 5-bit */
820  for (ch = 0; ch < c->fullband_channels; ch++)
821  put_bits(&c->pb, 3, 6);
822 
823  /* Quantization index codebook select: dummy data
824  to avoid transmission of scale factor adjustment */
825  for (i = 1; i < 11; i++)
826  for (ch = 0; ch < c->fullband_channels; ch++)
827  put_bits(&c->pb, bitlen[i], thr[i]);
828 
829  /* Scale factor adjustment index: not transmitted */
830  /* Audio header CRC check word: not transmitted */
831 }
832 
833 static void put_subframe_samples(DCAContext *c, int ss, int band, int ch)
834 {
835  if (c->abits[band][ch] <= 7) {
836  int sum, i, j;
837  for (i = 0; i < 8; i += 4) {
838  sum = 0;
839  for (j = 3; j >= 0; j--) {
840  sum *= quant_levels[c->abits[band][ch]];
841  sum += c->quantized[ss * 8 + i + j][band][ch];
842  sum += (quant_levels[c->abits[band][ch]] - 1) / 2;
843  }
844  put_bits(&c->pb, bit_consumption[c->abits[band][ch]] / 4, sum);
845  }
846  } else {
847  int i;
848  for (i = 0; i < 8; i++) {
849  int bits = bit_consumption[c->abits[band][ch]] / 16;
850  int32_t mask = (1 << bits) - 1;
851  put_bits(&c->pb, bits, c->quantized[ss * 8 + i][band][ch] & mask);
852  }
853  }
854 }
855 
856 static void put_subframe(DCAContext *c, int subframe)
857 {
858  int i, band, ss, ch;
859 
860  /* Subsubframes count */
861  put_bits(&c->pb, 2, SUBSUBFRAMES -1);
862 
863  /* Partial subsubframe sample count: dummy */
864  put_bits(&c->pb, 3, 0);
865 
866  /* Prediction mode: no ADPCM, in each channel and subband */
867  for (ch = 0; ch < c->fullband_channels; ch++)
868  for (band = 0; band < DCA_SUBBANDS; band++)
869  put_bits(&c->pb, 1, 0);
870 
871  /* Prediction VQ address: not transmitted */
872  /* Bit allocation index */
873  for (ch = 0; ch < c->fullband_channels; ch++)
874  for (band = 0; band < DCA_SUBBANDS; band++)
875  put_bits(&c->pb, 5, c->abits[band][ch]);
876 
877  if (SUBSUBFRAMES > 1) {
878  /* Transition mode: none for each channel and subband */
879  for (ch = 0; ch < c->fullband_channels; ch++)
880  for (band = 0; band < DCA_SUBBANDS; band++)
881  put_bits(&c->pb, 1, 0); /* codebook A4 */
882  }
883 
884  /* Scale factors */
885  for (ch = 0; ch < c->fullband_channels; ch++)
886  for (band = 0; band < DCA_SUBBANDS; band++)
887  put_bits(&c->pb, 7, c->scale_factor[band][ch]);
888 
889  /* Joint subband scale factor codebook select: not transmitted */
890  /* Scale factors for joint subband coding: not transmitted */
891  /* Stereo down-mix coefficients: not transmitted */
892  /* Dynamic range coefficient: not transmitted */
893  /* Stde information CRC check word: not transmitted */
894  /* VQ encoded high frequency subbands: not transmitted */
895 
896  /* LFE data: 8 samples and scalefactor */
897  if (c->lfe_channel) {
898  for (i = 0; i < DCA_LFE_SAMPLES; i++)
899  put_bits(&c->pb, 8, quantize_value(c->downsampled_lfe[i], c->lfe_quant) & 0xff);
900  put_bits(&c->pb, 8, c->lfe_scale_factor);
901  }
902 
903  /* Audio data (subsubframes) */
904  for (ss = 0; ss < SUBSUBFRAMES ; ss++)
905  for (ch = 0; ch < c->fullband_channels; ch++)
906  for (band = 0; band < DCA_SUBBANDS; band++)
907  put_subframe_samples(c, ss, band, ch);
908 
909  /* DSYNC */
910  put_bits(&c->pb, 16, 0xffff);
911 }
912 
913 static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
914  const AVFrame *frame, int *got_packet_ptr)
915 {
916  DCAContext *c = avctx->priv_data;
917  const int32_t *samples;
918  int ret, i;
919 
920  if ((ret = ff_alloc_packet2(avctx, avpkt, c->frame_size )) < 0)
921  return ret;
922 
923  samples = (const int32_t *)frame->data[0];
924 
925  subband_transform(c, samples);
926  if (c->lfe_channel)
927  lfe_downsample(c, samples);
928 
929  calc_masking(c, samples);
930  find_peaks(c);
931  assign_bits(c);
932  calc_scales(c);
933  quantize_all(c);
934  shift_history(c, samples);
935 
936  init_put_bits(&c->pb, avpkt->data, avpkt->size);
937  put_frame_header(c);
939  for (i = 0; i < SUBFRAMES; i++)
940  put_subframe(c, i);
941 
942  flush_put_bits(&c->pb);
943 
944  avpkt->pts = frame->pts;
945  avpkt->duration = ff_samples_to_time_base(avctx, frame->nb_samples);
946  avpkt->size = c->frame_size + 1;
947  *got_packet_ptr = 1;
948  return 0;
949 }
950 
951 static const AVCodecDefault defaults[] = {
952  { "b", "1411200" },
953  { NULL },
954 };
955 
957  .name = "dca",
958  .long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
959  .type = AVMEDIA_TYPE_AUDIO,
960  .id = AV_CODEC_ID_DTS,
961  .priv_data_size = sizeof(DCAContext),
962  .init = encode_init,
963  .encode2 = encode_frame,
964  .capabilities = CODEC_CAP_EXPERIMENTAL,
965  .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S32,
967  .supported_samplerates = sample_rates,
968  .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO,
973  0 },
974  .defaults = defaults,
975 };