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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 DCAENC_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 DCAEncContext {
51  int channels;
61 
62  int32_t history[512][MAX_CHANNELS]; /* This is a circular buffer */
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  DCAEncContext *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; ff_dca_bit_rates[i] < avctx->bit_rate; i++)
152  ;
153  c->bitrate_index = i;
154  avctx->bit_rate = ff_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 * ff_dca_lfe_fir_64[i]);
175  lfe_fir_64i[511 - i] = (int32_t)(0x01ffffff * ff_dca_lfe_fir_64[i]);
176  }
177 
178  for (i = 0; i < 512; i++) {
179  band_interpolation[0][i] = (int32_t)(0x1000000000ULL * ff_dca_fir_32bands_perfect[i]);
180  band_interpolation[1][i] = (int32_t)(0x1000000000ULL * ff_dca_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 = ff_dca_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 = ff_dca_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(DCAEncContext *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(DCAEncContext *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)(DCAEncContext *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(DCAEncContext *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(DCAEncContext *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(DCAEncContext *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(DCAEncContext *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);
519  }
520 }
521 
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 
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(DCAEncContext *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 
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 
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, DCAENC_SUBBANDS - 2);
802 
803  /* High frequency VQ start subband */
804  for (ch = 0; ch < c->fullband_channels; ch++)
805  put_bits(&c->pb, 5, DCAENC_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(DCAEncContext *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  put_sbits(&c->pb, bits, c->quantized[ss * 8 + i][band][ch]);
851  }
852  }
853 }
854 
855 static void put_subframe(DCAEncContext *c, int subframe)
856 {
857  int i, band, ss, ch;
858 
859  /* Subsubframes count */
860  put_bits(&c->pb, 2, SUBSUBFRAMES -1);
861 
862  /* Partial subsubframe sample count: dummy */
863  put_bits(&c->pb, 3, 0);
864 
865  /* Prediction mode: no ADPCM, in each channel and subband */
866  for (ch = 0; ch < c->fullband_channels; ch++)
867  for (band = 0; band < DCAENC_SUBBANDS; band++)
868  put_bits(&c->pb, 1, 0);
869 
870  /* Prediction VQ address: not transmitted */
871  /* Bit allocation index */
872  for (ch = 0; ch < c->fullband_channels; ch++)
873  for (band = 0; band < DCAENC_SUBBANDS; band++)
874  put_bits(&c->pb, 5, c->abits[band][ch]);
875 
876  if (SUBSUBFRAMES > 1) {
877  /* Transition mode: none for each channel and subband */
878  for (ch = 0; ch < c->fullband_channels; ch++)
879  for (band = 0; band < DCAENC_SUBBANDS; band++)
880  put_bits(&c->pb, 1, 0); /* codebook A4 */
881  }
882 
883  /* Scale factors */
884  for (ch = 0; ch < c->fullband_channels; ch++)
885  for (band = 0; band < DCAENC_SUBBANDS; band++)
886  put_bits(&c->pb, 7, c->scale_factor[band][ch]);
887 
888  /* Joint subband scale factor codebook select: not transmitted */
889  /* Scale factors for joint subband coding: not transmitted */
890  /* Stereo down-mix coefficients: not transmitted */
891  /* Dynamic range coefficient: not transmitted */
892  /* Stde information CRC check word: not transmitted */
893  /* VQ encoded high frequency subbands: not transmitted */
894 
895  /* LFE data: 8 samples and scalefactor */
896  if (c->lfe_channel) {
897  for (i = 0; i < DCA_LFE_SAMPLES; i++)
898  put_bits(&c->pb, 8, quantize_value(c->downsampled_lfe[i], c->lfe_quant) & 0xff);
899  put_bits(&c->pb, 8, c->lfe_scale_factor);
900  }
901 
902  /* Audio data (subsubframes) */
903  for (ss = 0; ss < SUBSUBFRAMES ; ss++)
904  for (ch = 0; ch < c->fullband_channels; ch++)
905  for (band = 0; band < DCAENC_SUBBANDS; band++)
906  put_subframe_samples(c, ss, band, ch);
907 
908  /* DSYNC */
909  put_bits(&c->pb, 16, 0xffff);
910 }
911 
912 static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
913  const AVFrame *frame, int *got_packet_ptr)
914 {
915  DCAEncContext *c = avctx->priv_data;
916  const int32_t *samples;
917  int ret, i;
918 
919  if ((ret = ff_alloc_packet2(avctx, avpkt, c->frame_size , 0)) < 0)
920  return ret;
921 
922  samples = (const int32_t *)frame->data[0];
923 
924  subband_transform(c, samples);
925  if (c->lfe_channel)
926  lfe_downsample(c, samples);
927 
928  calc_masking(c, samples);
929  find_peaks(c);
930  assign_bits(c);
931  calc_scales(c);
932  quantize_all(c);
933  shift_history(c, samples);
934 
935  init_put_bits(&c->pb, avpkt->data, avpkt->size);
936  put_frame_header(c);
938  for (i = 0; i < SUBFRAMES; i++)
939  put_subframe(c, i);
940 
941 
942  for (i = put_bits_count(&c->pb); i < 8*c->frame_size; i++)
943  put_bits(&c->pb, 1, 0);
944 
945  flush_put_bits(&c->pb);
946 
947  avpkt->pts = frame->pts;
948  avpkt->duration = ff_samples_to_time_base(avctx, frame->nb_samples);
949  avpkt->size = c->frame_size + 1;
950  *got_packet_ptr = 1;
951  return 0;
952 }
953 
954 static const AVCodecDefault defaults[] = {
955  { "b", "1411200" },
956  { NULL },
957 };
958 
960  .name = "dca",
961  .long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
962  .type = AVMEDIA_TYPE_AUDIO,
963  .id = AV_CODEC_ID_DTS,
964  .priv_data_size = sizeof(DCAEncContext),
965  .init = encode_init,
966  .encode2 = encode_frame,
967  .capabilities = AV_CODEC_CAP_EXPERIMENTAL,
968  .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S32,
970  .supported_samplerates = sample_rates,
971  .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO,
976  0 },
977  .defaults = defaults,
978 };
#define MAX_CHANNELS
Definition: dcaenc.c:35
#define NULL
Definition: coverity.c:32
const char * s
Definition: avisynth_c.h:631
int32_t m
Definition: dcaenc.h:28
This structure describes decoded (raw) audio or video data.
Definition: frame.h:171
static int32_t cb_to_add[256]
Definition: dcaenc.c:82
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:101
int32_t eff_masking_curve_cb[256]
Definition: dcaenc.c:71
static void put_sbits(PutBitContext *pb, int n, int32_t value)
Definition: put_bits.h:192
static int32_t auf[9][AUBANDS][256]
Definition: dcaenc.c:81
static void put_bits(Jpeg2000EncoderContext *s, int val, int n)
put n times val bit
Definition: j2kenc.c:167
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
static void put_frame_header(DCAEncContext *c)
Definition: dcaenc.c:704
#define AUBANDS
Definition: dcaenc.c:44
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
static int32_t band_spectrum[2][8]
Definition: dcaenc.c:80
int size
Definition: avcodec.h:1423
const char * b
Definition: vf_curves.c:109
static const uint8_t bitstream_sfreq[]
Definition: dcaenc.h:36
static const uint16_t erb[]
Definition: dcaenc.h:46
#define AV_CODEC_CAP_EXPERIMENTAL
Codec is experimental and is thus avoided in favor of non experimental encoders.
Definition: avcodec.h:911
static void shift_history(DCAEncContext *c, const int32_t *input)
Definition: dcaenc.c:630
softfloat lfe_quant
Definition: dcaenc.c:59
#define AV_CH_LAYOUT_STEREO
static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: dcaenc.c:912
#define sample
static void walk_band_high(DCAEncContext *c, int band, int channel, walk_band_t walk, int32_t *arg)
Definition: dcaenc.c:464
AVCodec.
Definition: avcodec.h:3471
#define AV_CH_LAYOUT_5POINT0
int32_t subband[SUBBAND_SAMPLES][DCAENC_SUBBANDS][MAX_CHANNELS]
Definition: dcaenc.c:63
#define USED_NABITS
Definition: dcaenc.c:552
int frame_size
Definition: dcaenc.c:48
const float ff_dca_fir_32bands_nonperfect[512]
Definition: dcadata.c:6785
#define FFALIGN(x, a)
Definition: common.h:71
static void walk_band_low(DCAEncContext *c, int band, int channel, walk_band_t walk, int32_t *arg)
Definition: dcaenc.c:449
static const int quant_levels[27]
Definition: dcaenc.h:107
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
#define SUBSUBFRAMES
Definition: dcaenc.c:42
static void calc_masking(DCAEncContext *c, const int32_t *input)
Definition: dcaenc.c:489
static const softfloat stepsize_inv[27]
Definition: dcaenc.h:51
const uint32_t ff_dca_bit_rates[32]
Definition: dcadata.c:33
uint8_t bits
Definition: crc.c:295
static int32_t lfe_fir_64i[512]
Definition: dcaenc.c:84
int32_t im
Definition: dcaenc.c:320
static int32_t cos_table[2048]
Definition: dcaenc.c:78
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:257
static int calc_one_scale(int32_t peak_cb, int abits, softfloat *quant)
Definition: dcaenc.c:648
int scale_factor[DCAENC_SUBBANDS][MAX_CHANNELS]
Definition: dcaenc.c:69
int32_t history[512][MAX_CHANNELS]
Definition: dcaenc.c:62
int32_t masking_curve_cb[SUBSUBFRAMES][256]
Definition: dcaenc.c:67
static AVFrame * frame
uint8_t * data
Definition: avcodec.h:1422
static void quantize_all(DCAEncContext *c)
Definition: dcaenc.c:694
static int32_t sin_t(int x)
Definition: dcaenc.c:222
int frame_bits
Definition: dcaenc.c:49
int lfe_channel
Definition: dcaenc.c:52
int32_t re
Definition: dcaenc.c:319
signed 32 bits
Definition: samplefmt.h:63
int duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
Definition: avcodec.h:1440
#define av_log(a,...)
unsigned m
Definition: audioconvert.c:187
#define DCA_LFE_SAMPLES
Definition: dcaenc.c:38
#define AV_CH_LAYOUT_5POINT1
static int32_t add_cb(int32_t a, int32_t b)
Definition: dcaenc.c:404
#define USED_1ABITS
Definition: dcaenc.c:551
static const softfloat scalefactor_inv[128]
Definition: dcaenc.h:61
static void lfe_downsample(DCAEncContext *c, const int32_t *input)
Definition: dcaenc.c:288
softfloat quant[DCAENC_SUBBANDS][MAX_CHANNELS]
Definition: dcaenc.c:70
static double hom(double f)
Definition: dcaenc.c:87
int32_t band_masking_cb[32]
Definition: dcaenc.c:72
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
static void put_subframe(DCAEncContext *c, int subframe)
Definition: dcaenc.c:855
Definition: dcaenc.c:318
int samplerate_index
Definition: dcaenc.c:53
static const int snr_fudge
Definition: dcaenc.c:550
#define AVERROR(e)
Definition: error.h:43
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:175
const uint8_t ff_reverse[256]
Definition: reverse.c:23
int channels
Definition: dcaenc.c:51
int abits[DCAENC_SUBBANDS][MAX_CHANNELS]
Definition: dcaenc.c:68
const char * r
Definition: vf_curves.c:107
const float ff_dca_lfe_fir_64[256]
Definition: dcadata.c:7316
const char * arg
Definition: jacosubdec.c:66
static void update_band_masking(DCAEncContext *c, int band1, int band2, int f, int32_t spectrum1, int32_t spectrum2, int channel, int32_t *arg)
Definition: dcaenc.c:479
simple assert() macros that are a bit more flexible than ISO C assert().
const char * name
Name of the codec implementation.
Definition: avcodec.h:3478
static const uint8_t offset[127][2]
Definition: vf_spp.c:92
Libavcodec external API header.
static int32_t quantize_value(int32_t value, softfloat quant)
Definition: dcaenc.c:639
static const int sample_rates[]
Definition: dcaenc.h:32
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:2322
static int put_bits_count(PutBitContext *s)
Definition: put_bits.h:85
int32_t e
Definition: dcaenc.h:29
#define AV_CH_LAYOUT_2_2
static const uint16_t fc[]
Definition: dcaenc.h:41
int bit_rate
the average bitrate
Definition: avcodec.h:1566
static void assign_bits(DCAEncContext *c)
Definition: dcaenc.c:592
audio channel layout utility functions
int fullband_channels
Definition: dcaenc.c:50
static int init_quantization_noise(DCAEncContext *c, int noise)
Definition: dcaenc.c:555
typedef void(APIENTRY *FF_PFNGLACTIVETEXTUREPROC)(GLenum texture)
GLsizei GLboolean const GLfloat * value
Definition: opengl_enc.c:109
int32_t
int32_t worst_noise_ever
Definition: dcaenc.c:74
#define DCA_MAX_FRAME_SIZE
Definition: dcaenc.c:36
#define DCAENC_SUBBANDS
Definition: dcaenc.c:40
int32_t lfe_peak_cb
Definition: dcaenc.c:60
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Definition: error.h:62
int32_t quantized[SUBBAND_SAMPLES][DCAENC_SUBBANDS][MAX_CHANNELS]
Definition: dcaenc.c:64
int32_t peak_cb[DCAENC_SUBBANDS][MAX_CHANNELS]
Definition: dcaenc.c:65
#define SUBBAND_SAMPLES
Definition: dcaenc.c:43
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:2281
static int noise(AVBitStreamFilterContext *bsfc, AVCodecContext *avctx, const char *args, uint8_t **poutbuf, int *poutbuf_size, const uint8_t *buf, int buf_size, int keyframe)
Definition: noise_bsf.c:28
int bitrate_index
Definition: dcaenc.c:54
static void put_primary_audio_header(DCAEncContext *c)
Definition: dcaenc.c:787
int frame_size
Definition: mxfenc.c:1803
static void find_peaks(DCAEncContext *c)
Definition: dcaenc.c:522
static int32_t mul32(int32_t a, int32_t b)
Definition: dcaenc.c:232
const int32_t * band_spectrum
Definition: dcaenc.c:57
float im
Definition: fft-test.c:73
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:59
static int32_t cb_to_level[2048]
Definition: dcaenc.c:83
int sample_rate
samples per second
Definition: avcodec.h:2261
main external API structure.
Definition: avcodec.h:1501
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt)>2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);returnNULL;}returnac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> in
void * buf
Definition: avisynth_c.h:553
static void put_subframe_samples(DCAEncContext *c, int ss, int band, int ch)
Definition: dcaenc.c:833
static void adjust_jnd(int samplerate_index, const int32_t in[512], int32_t out_cb[256])
Definition: dcaenc.c:414
int ff_alloc_packet2(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int64_t min_size)
Check AVPacket size and/or allocate data.
Definition: utils.c:1782
int32_t worst_quantization_noise
Definition: dcaenc.c:73
static int encode_init(AVCodecContext *avctx)
Definition: dcaenc.c:106
static int32_t band_interpolation[2][512]
Definition: dcaenc.c:79
static int32_t cos_t(int x)
Definition: dcaenc.c:217
static void calc_scales(DCAEncContext *c)
Definition: dcaenc.c:680
const uint8_t * quant
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:182
#define SUBFRAMES
Definition: dcaenc.c:41
static void subband_transform(DCAEncContext *c, const int32_t *input)
Definition: dcaenc.c:238
int channel_config
Definition: dcaenc.c:55
AVCodec ff_dca_encoder
Definition: dcaenc.c:959
PutBitContext pb
Definition: dcaenc.c:47
static void fft(const int32_t in[2 *256], cplx32 out[256])
Definition: dcaenc.c:323
common internal api header.
static void flush_put_bits(PutBitContext *s)
Pad the end of the output stream with zeros.
Definition: put_bits.h:101
common internal and external API header
if(ret< 0)
Definition: vf_mcdeint.c:280
#define USED_26ABITS
Definition: dcaenc.c:553
static double c[64]
static int32_t get_cb(int32_t in)
Definition: dcaenc.c:390
int lfe_scale_factor
Definition: dcaenc.c:58
static void init_put_bits(PutBitContext *s, uint8_t *buffer, int buffer_size)
Initialize the PutBitContext s.
Definition: put_bits.h:48
void * priv_data
Definition: avcodec.h:1543
static av_always_inline int diff(const uint32_t a, const uint32_t b)
float re
Definition: fft-test.c:73
int consumed_bits
Definition: dcaenc.c:75
int channels
number of audio channels
Definition: avcodec.h:2262
void(* walk_band_t)(DCAEncContext *c, int band1, int band2, int f, int32_t spectrum1, int32_t spectrum2, int channel, int32_t *arg)
Definition: dcaenc.c:445
static int32_t half32(int32_t a)
Definition: dcaenc.c:227
uint64_t layout
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt)>2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);returnNULL;}returnac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> out
int64_t av_get_default_channel_layout(int nb_channels)
Return default channel layout for a given number of channels.
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:701
const int32_t * band_interpolation
Definition: dcaenc.c:56
int32_t downsampled_lfe[DCA_LFE_SAMPLES]
Definition: dcaenc.c:66
#define M_PI
Definition: mathematics.h:46
static av_always_inline int64_t ff_samples_to_time_base(AVCodecContext *avctx, int64_t samples)
Rescale from sample rate to AVCodecContext.time_base.
Definition: internal.h:232
#define FFSWAP(type, a, b)
Definition: common.h:69
static const AVCodecDefault defaults[]
Definition: dcaenc.c:954
static const int bit_consumption[27]
Definition: dcaenc.h:100
const float ff_dca_fir_32bands_perfect[512]
Definition: dcadata.c:6270
#define AV_CH_LAYOUT_MONO
This structure stores compressed data.
Definition: avcodec.h:1399
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:225
static double gammafilter(int i, double f)
Definition: dcaenc.c:97
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
Definition: avcodec.h:1415
for(j=16;j >0;--j)
bitstream writer API