FFmpeg
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 "libavutil/ffmath.h"
28 #include "libavutil/mem_internal.h"
29 #include "libavutil/opt.h"
30 #include "libavutil/thread.h"
31 #include "libavutil/tx.h"
32 #include "avcodec.h"
33 #include "codec_internal.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 "encode.h"
41 #include "put_bits.h"
42 
43 #define MAX_CHANNELS 6
44 #define DCA_MAX_FRAME_SIZE 16384
45 #define DCA_HEADER_SIZE 13
46 #define DCA_LFE_SAMPLES 8
47 
48 #define DCAENC_SUBBANDS 32
49 #define SUBFRAMES 1
50 #define SUBSUBFRAMES 2
51 #define SUBBAND_SAMPLES (SUBFRAMES * SUBSUBFRAMES * 8)
52 #define AUBANDS 25
53 
54 #define COS_T(x) (c->cos_table[(x) & 2047])
55 
56 typedef struct CompressionOptions {
59 
60 typedef struct DCAEncContext {
61  AVClass *class;
70  int channels;
80  const int8_t *channel_order_tab; ///< channel reordering table, lfe and non lfe
81 
84  int32_t history[MAX_CHANNELS][512]; /* This is a circular buffer */
88  int32_t diff_peak_cb[MAX_CHANNELS][DCAENC_SUBBANDS]; ///< expected peak of residual signal
101  int consumed_adpcm_bits; ///< Number of bits to transmit ADPCM related info
102 
106  int32_t auf[9][AUBANDS][256];
110 } DCAEncContext;
111 
112 /* Transfer function of outer and middle ear, Hz -> dB */
113 static double hom(double f)
114 {
115  double f1 = f / 1000;
116 
117  return -3.64 * pow(f1, -0.8)
118  + 6.8 * exp(-0.6 * (f1 - 3.4) * (f1 - 3.4))
119  - 6.0 * exp(-0.15 * (f1 - 8.7) * (f1 - 8.7))
120  - 0.0006 * (f1 * f1) * (f1 * f1);
121 }
122 
123 static double gammafilter(int i, double f)
124 {
125  double h = (f - fc[i]) / erb[i];
126 
127  h = 1 + h * h;
128  h = 1 / (h * h);
129  return 20 * log10(h);
130 }
131 
133 {
134  int ch, band;
137  sizeof(int32_t));
138  if (!bufer)
139  return AVERROR(ENOMEM);
140 
141  /* we need a place for DCA_ADPCM_COEFF samples from previous frame
142  * to calc prediction coefficients for each subband */
143  for (ch = 0; ch < MAX_CHANNELS; ch++) {
144  for (band = 0; band < DCAENC_SUBBANDS; band++) {
145  c->subband[ch][band] = bufer +
148  }
149  }
150  return 0;
151 }
152 
154 {
155  if (c->subband[0][0]) {
156  int32_t *bufer = c->subband[0][0] - DCA_ADPCM_COEFFS;
157  av_free(bufer);
158  c->subband[0][0] = NULL;
159  }
160 }
161 
162 static uint16_t bitalloc_12_table[DCA_BITALLOC_12_COUNT][12 + 1][2];
163 
165 static const uint16_t (*bitalloc_tables[DCA_CODE_BOOKS][8])[2];
166 
167 static av_cold void create_enc_table(uint16_t dst[][2], unsigned count,
168  const uint8_t (**src_tablep)[2])
169 {
170  const uint8_t (*src_table)[2] = *src_tablep;
171  uint16_t code = 0;
172 
173  for (unsigned i = 0; i < count; i++) {
174  unsigned dst_idx = src_table[i][0];
175 
176  dst[dst_idx][0] = code >> (16 - src_table[i][1]);
177  dst[dst_idx][1] = src_table[i][1];
178 
179  code += 1 << (16 - src_table[i][1]);
180  }
181  *src_tablep += count;
182 }
183 
185 {
186  uint16_t (*bitalloc_dst)[2] = bitalloc_table;
187  const uint8_t (*src_table)[2] = ff_dca_vlc_src_tables;
188 
189  for (unsigned i = 0; i < DCA_CODE_BOOKS; i++) {
190  for (unsigned j = 0; j < ff_dca_quant_index_group_size[i]; j++) {
192  &src_table);
193  bitalloc_tables[i][j] = bitalloc_dst - ff_dca_bitalloc_offsets[i];
194  bitalloc_dst += ff_dca_bitalloc_sizes[i];
195  }
196  }
197 
198  for (unsigned i = 0; i < DCA_BITALLOC_12_COUNT; i++)
199  create_enc_table(&bitalloc_12_table[i][1], 12, &src_table);
200 }
201 
202 static int encode_init(AVCodecContext *avctx)
203 {
204  static AVOnce init_static_once = AV_ONCE_INIT;
205  DCAEncContext *c = avctx->priv_data;
207  int i, j, k, min_frame_bits;
208  float scale = 1.0f;
209  int ret;
210 
211  if ((ret = subband_bufer_alloc(c)) < 0)
212  return ret;
213 
214  c->fullband_channels = c->channels = layout.nb_channels;
215  c->lfe_channel = (c->channels == 3 || c->channels == 6);
216  c->band_interpolation = c->band_interpolation_tab[1];
217  c->band_spectrum = c->band_spectrum_tab[1];
218  c->worst_quantization_noise = -2047;
219  c->worst_noise_ever = -2047;
220  c->consumed_adpcm_bits = 0;
221 
222  if (ff_dcaadpcm_init(&c->adpcm_ctx))
223  return AVERROR(ENOMEM);
224 
225  switch (layout.nb_channels) {
226  case 1: /* mono */
227  c->channel_config = 0;
228  break;
229  case 2: /* stereo */
230  c->channel_config = 2;
231  break;
232  case 4: /* 2.2 */
233  c->channel_config = 8;
234  break;
235  case 5: /* 5.0 */
236  c->channel_config = 9;
237  break;
238  case 6: /* 5.1 */
239  c->channel_config = 9;
240  break;
241  default:
242  av_assert1(!"impossible channel layout");
243  }
244 
245  if (c->lfe_channel) {
246  c->fullband_channels--;
247  c->channel_order_tab = channel_reorder_lfe[c->channel_config];
248  } else {
249  c->channel_order_tab = channel_reorder_nolfe[c->channel_config];
250  }
251 
252  for (i = 0; i < MAX_CHANNELS; i++) {
253  for (j = 0; j < DCA_CODE_BOOKS; j++) {
254  c->quant_index_sel[i][j] = ff_dca_quant_index_group_size[j];
255  }
256  /* 6 - no Huffman */
257  c->bit_allocation_sel[i] = 6;
258 
259  for (j = 0; j < DCAENC_SUBBANDS; j++) {
260  /* -1 - no ADPCM */
261  c->prediction_mode[i][j] = -1;
262  memset(c->adpcm_history[i][j], 0, sizeof(int32_t)*DCA_ADPCM_COEFFS);
263  }
264  }
265 
266  for (i = 0; i < 9; i++) {
267  if (sample_rates[i] == avctx->sample_rate)
268  break;
269  }
270  if (i == 9)
271  return AVERROR(EINVAL);
272  c->samplerate_index = i;
273 
274  if (avctx->bit_rate < 32000 || avctx->bit_rate > 3840000) {
275  av_log(avctx, AV_LOG_ERROR, "Bit rate %"PRId64" not supported.", avctx->bit_rate);
276  return AVERROR(EINVAL);
277  }
278  for (i = 0; ff_dca_bit_rates[i] < avctx->bit_rate; i++)
279  ;
280  c->bitrate_index = i;
281  c->frame_bits = FFALIGN((avctx->bit_rate * 512 + avctx->sample_rate - 1) / avctx->sample_rate, 32);
282  min_frame_bits = 132 + (493 + 28 * 32) * c->fullband_channels + c->lfe_channel * 72;
283  if (c->frame_bits < min_frame_bits || c->frame_bits > (DCA_MAX_FRAME_SIZE << 3))
284  return AVERROR(EINVAL);
285 
286  c->frame_size = (c->frame_bits + 7) / 8;
287 
288  avctx->frame_size = 32 * SUBBAND_SAMPLES;
289 
290  if ((ret = av_tx_init(&c->mdct, &c->mdct_fn, AV_TX_INT32_MDCT, 0, 256, &scale, 0)) < 0)
291  return ret;
292 
293  /* Init all tables */
294  c->cos_table[0] = 0x7fffffff;
295  c->cos_table[512] = 0;
296  c->cos_table[1024] = -c->cos_table[0];
297  for (i = 1; i < 512; i++) {
298  c->cos_table[i] = (int32_t)(0x7fffffff * cos(M_PI * i / 1024));
299  c->cos_table[1024-i] = -c->cos_table[i];
300  c->cos_table[1024+i] = -c->cos_table[i];
301  c->cos_table[2048-i] = +c->cos_table[i];
302  }
303 
304  for (i = 0; i < 2048; i++)
305  c->cb_to_level[i] = (int32_t)(0x7fffffff * ff_exp10(-0.005 * i));
306 
307  for (k = 0; k < 32; k++) {
308  for (j = 0; j < 8; j++) {
309  c->lfe_fir_64i[64 * j + k] = (int32_t)(0xffffff800000ULL * ff_dca_lfe_fir_64[8 * k + j]);
310  c->lfe_fir_64i[64 * (7-j) + (63 - k)] = (int32_t)(0xffffff800000ULL * ff_dca_lfe_fir_64[8 * k + j]);
311  }
312  }
313 
314  for (i = 0; i < 512; i++) {
315  c->band_interpolation_tab[0][i] = (int32_t)(0x1000000000ULL * ff_dca_fir_32bands_perfect[i]);
316  c->band_interpolation_tab[1][i] = (int32_t)(0x1000000000ULL * ff_dca_fir_32bands_nonperfect[i]);
317  }
318 
319  for (i = 0; i < 9; i++) {
320  for (j = 0; j < AUBANDS; j++) {
321  for (k = 0; k < 256; k++) {
322  double freq = sample_rates[i] * (k + 0.5) / 512;
323 
324  c->auf[i][j][k] = (int32_t)(10 * (hom(freq) + gammafilter(j, freq)));
325  }
326  }
327  }
328 
329  for (i = 0; i < 256; i++) {
330  double add = 1 + ff_exp10(-0.01 * i);
331  c->cb_to_add[i] = (int32_t)(100 * log10(add));
332  }
333  for (j = 0; j < 8; j++) {
334  double accum = 0;
335  for (i = 0; i < 512; i++) {
336  double reconst = ff_dca_fir_32bands_perfect[i] * ((i & 64) ? (-1) : 1);
337  accum += reconst * cos(2 * M_PI * (i + 0.5 - 256) * (j + 0.5) / 512);
338  }
339  c->band_spectrum_tab[0][j] = (int32_t)(200 * log10(accum));
340  }
341  for (j = 0; j < 8; j++) {
342  double accum = 0;
343  for (i = 0; i < 512; i++) {
344  double reconst = ff_dca_fir_32bands_nonperfect[i] * ((i & 64) ? (-1) : 1);
345  accum += reconst * cos(2 * M_PI * (i + 0.5 - 256) * (j + 0.5) / 512);
346  }
347  c->band_spectrum_tab[1][j] = (int32_t)(200 * log10(accum));
348  }
349 
350  ff_thread_once(&init_static_once, dcaenc_init_static_tables);
351  return 0;
352 }
353 
355 {
356  DCAEncContext *c = avctx->priv_data;
357  av_tx_uninit(&c->mdct);
359  ff_dcaadpcm_free(&c->adpcm_ctx);
360 
361  return 0;
362 }
363 
365 {
366  int ch, subs, i, k, j;
367 
368  for (ch = 0; ch < c->fullband_channels; ch++) {
369  /* History is copied because it is also needed for PSY */
370  int32_t hist[512];
371  int hist_start = 0;
372  const int chi = c->channel_order_tab[ch];
373 
374  memcpy(hist, &c->history[ch][0], 512 * sizeof(int32_t));
375 
376  for (subs = 0; subs < SUBBAND_SAMPLES; subs++) {
377  int32_t accum[64];
378  int32_t resp;
379  int band;
380 
381  /* Calculate the convolutions at once */
382  memset(accum, 0, 64 * sizeof(int32_t));
383 
384  for (k = 0, i = hist_start, j = 0;
385  i < 512; k = (k + 1) & 63, i++, j++)
386  accum[k] += mul32(hist[i], c->band_interpolation[j]);
387  for (i = 0; i < hist_start; k = (k + 1) & 63, i++, j++)
388  accum[k] += mul32(hist[i], c->band_interpolation[j]);
389 
390  for (k = 16; k < 32; k++)
391  accum[k] = accum[k] - accum[31 - k];
392  for (k = 32; k < 48; k++)
393  accum[k] = accum[k] + accum[95 - k];
394 
395  for (band = 0; band < 32; band++) {
396  resp = 0;
397  for (i = 16; i < 48; i++) {
398  int s = (2 * band + 1) * (2 * (i + 16) + 1);
399  resp += mul32(accum[i], COS_T(s << 3)) >> 3;
400  }
401 
402  c->subband[ch][band][subs] = ((band + 1) & 2) ? -resp : resp;
403  }
404 
405  /* Copy in 32 new samples from input */
406  for (i = 0; i < 32; i++)
407  hist[i + hist_start] = input[(subs * 32 + i) * c->channels + chi];
408 
409  hist_start = (hist_start + 32) & 511;
410  }
411  }
412 }
413 
415 {
416  /* FIXME: make 128x LFE downsampling possible */
417  const int lfech = lfe_index[c->channel_config];
418  int i, j, lfes;
419  int32_t hist[512];
420  int32_t accum;
421  int hist_start = 0;
422 
423  memcpy(hist, &c->history[c->channels - 1][0], 512 * sizeof(int32_t));
424 
425  for (lfes = 0; lfes < DCA_LFE_SAMPLES; lfes++) {
426  /* Calculate the convolution */
427  accum = 0;
428 
429  for (i = hist_start, j = 0; i < 512; i++, j++)
430  accum += mul32(hist[i], c->lfe_fir_64i[j]);
431  for (i = 0; i < hist_start; i++, j++)
432  accum += mul32(hist[i], c->lfe_fir_64i[j]);
433 
434  c->downsampled_lfe[lfes] = accum;
435 
436  /* Copy in 64 new samples from input */
437  for (i = 0; i < 64; i++)
438  hist[i + hist_start] = input[(lfes * 64 + i) * c->channels + lfech];
439 
440  hist_start = (hist_start + 64) & 511;
441  }
442 }
443 
444 static uint32_t dca_vlc_calc_alloc_bits(const int values[], uint8_t n, uint8_t sel)
445 {
446  uint32_t sum = 0;
447  for (unsigned i = 0; i < n; i++)
448  sum += bitalloc_12_table[sel][values[i]][1];
449  return sum;
450 }
451 
452 static void dca_vlc_enc_alloc(PutBitContext *pb, const int values[],
453  uint8_t n, uint8_t sel)
454 {
455  for (unsigned i = 0; i < n; i++)
456  put_bits(pb, bitalloc_12_table[sel][values[i]][1],
457  bitalloc_12_table[sel][values[i]][0]);
458 }
459 
460 static uint32_t dca_vlc_calc_quant_bits(const int values[], uint8_t n,
461  uint8_t sel, uint8_t table)
462 {
463  uint32_t sum = 0;
464  for (unsigned i = 0; i < n; i++)
465  sum += bitalloc_tables[table][sel][values[i]][1];
466  return sum;
467 }
468 
469 static void dca_vlc_enc_quant(PutBitContext *pb, const int values[],
470  uint8_t n, uint8_t sel, uint8_t table)
471 {
472  for (unsigned i = 0; i < n; i++)
473  put_bits(pb, bitalloc_tables[table][sel][values[i]][1],
474  bitalloc_tables[table][sel][values[i]][0]);
475 }
476 
478 {
479  int i, res = 0;
480  in = FFABS(in);
481 
482  for (i = 1024; i > 0; i >>= 1) {
483  if (c->cb_to_level[i + res] >= in)
484  res += i;
485  }
486  return -res;
487 }
488 
490 {
491  if (a < b)
492  FFSWAP(int32_t, a, b);
493 
494  if (a - b >= 256)
495  return a;
496  return a + c->cb_to_add[a - b];
497 }
498 
500  const int32_t in[2 * 256], int32_t power[256])
501 {
502  int i;
503  LOCAL_ALIGNED_32(int32_t, data, [512]);
504  LOCAL_ALIGNED_32(int32_t, coeff, [256]);
505 
506  for (i = 0; i < 512; i++)
507  data[i] = norm__(mul32(in[i], 0x3fffffff - (COS_T(4 * i + 2) >> 1)), 4);
508 
509  c->mdct_fn(c->mdct, coeff, data, sizeof(int32_t));
510  for (i = 0; i < 256; i++) {
511  const int32_t cb = get_cb(c, coeff[i]);
512  power[i] = add_cb(c, cb, cb);
513  }
514 }
515 
517  const int32_t in[512], int32_t out_cb[256])
518 {
519  int32_t power[256];
520  int32_t out_cb_unnorm[256];
521  int32_t denom;
522  const int32_t ca_cb = -1114;
523  const int32_t cs_cb = 928;
524  const int samplerate_index = c->samplerate_index;
525  int i, j;
526 
527  calc_power(c, in, power);
528 
529  for (j = 0; j < 256; j++)
530  out_cb_unnorm[j] = -2047; /* and can only grow */
531 
532  for (i = 0; i < AUBANDS; i++) {
533  denom = ca_cb; /* and can only grow */
534  for (j = 0; j < 256; j++)
535  denom = add_cb(c, denom, power[j] + c->auf[samplerate_index][i][j]);
536  for (j = 0; j < 256; j++)
537  out_cb_unnorm[j] = add_cb(c, out_cb_unnorm[j],
538  -denom + c->auf[samplerate_index][i][j]);
539  }
540 
541  for (j = 0; j < 256; j++)
542  out_cb[j] = add_cb(c, out_cb[j], -out_cb_unnorm[j] - ca_cb - cs_cb);
543 }
544 
545 typedef void (*walk_band_t)(DCAEncContext *c, int band1, int band2, int f,
546  int32_t spectrum1, int32_t spectrum2, int channel,
547  int32_t * arg);
548 
549 static void walk_band_low(DCAEncContext *c, int band, int channel,
550  walk_band_t walk, int32_t *arg)
551 {
552  int f;
553 
554  if (band == 0) {
555  for (f = 0; f < 4; f++)
556  walk(c, 0, 0, f, 0, -2047, channel, arg);
557  } else {
558  for (f = 0; f < 8; f++)
559  walk(c, band, band - 1, 8 * band - 4 + f,
560  c->band_spectrum[7 - f], c->band_spectrum[f], channel, arg);
561  }
562 }
563 
564 static void walk_band_high(DCAEncContext *c, int band, int channel,
565  walk_band_t walk, int32_t *arg)
566 {
567  int f;
568 
569  if (band == 31) {
570  for (f = 0; f < 4; f++)
571  walk(c, 31, 31, 256 - 4 + f, 0, -2047, channel, arg);
572  } else {
573  for (f = 0; f < 8; f++)
574  walk(c, band, band + 1, 8 * band + 4 + f,
575  c->band_spectrum[f], c->band_spectrum[7 - f], channel, arg);
576  }
577 }
578 
579 static void update_band_masking(DCAEncContext *c, int band1, int band2,
580  int f, int32_t spectrum1, int32_t spectrum2,
581  int channel, int32_t * arg)
582 {
583  int32_t value = c->eff_masking_curve_cb[f] - spectrum1;
584 
585  if (value < c->band_masking_cb[band1])
586  c->band_masking_cb[band1] = value;
587 }
588 
589 static void calc_masking(DCAEncContext *c, const int32_t *input)
590 {
591  int i, k, band, ch, ssf;
592  int32_t data[512];
593 
594  for (i = 0; i < 256; i++)
595  for (ssf = 0; ssf < SUBSUBFRAMES; ssf++)
596  c->masking_curve_cb[ssf][i] = -2047;
597 
598  for (ssf = 0; ssf < SUBSUBFRAMES; ssf++)
599  for (ch = 0; ch < c->fullband_channels; ch++) {
600  const int chi = c->channel_order_tab[ch];
601 
602  for (i = 0, k = 128 + 256 * ssf; k < 512; i++, k++)
603  data[i] = c->history[ch][k];
604  for (k -= 512; i < 512; i++, k++)
605  data[i] = input[k * c->channels + chi];
606  adjust_jnd(c, data, c->masking_curve_cb[ssf]);
607  }
608  for (i = 0; i < 256; i++) {
609  int32_t m = 2048;
610 
611  for (ssf = 0; ssf < SUBSUBFRAMES; ssf++)
612  if (c->masking_curve_cb[ssf][i] < m)
613  m = c->masking_curve_cb[ssf][i];
614  c->eff_masking_curve_cb[i] = m;
615  }
616 
617  for (band = 0; band < 32; band++) {
618  c->band_masking_cb[band] = 2048;
621  }
622 }
623 
624 static inline int32_t find_peak(DCAEncContext *c, const int32_t *in, int len)
625 {
626  int sample;
627  int32_t m = 0;
628  for (sample = 0; sample < len; sample++) {
629  int32_t s = abs(in[sample]);
630  if (m < s)
631  m = s;
632  }
633  return get_cb(c, m);
634 }
635 
637 {
638  int band, ch;
639 
640  for (ch = 0; ch < c->fullband_channels; ch++) {
641  for (band = 0; band < 32; band++)
642  c->peak_cb[ch][band] = find_peak(c, c->subband[ch][band],
644  }
645 
646  if (c->lfe_channel)
647  c->lfe_peak_cb = find_peak(c, c->downsampled_lfe, DCA_LFE_SAMPLES);
648 }
649 
651 {
652  int ch, band;
653  int pred_vq_id;
654  int32_t *samples;
655  int32_t estimated_diff[SUBBAND_SAMPLES];
656 
657  c->consumed_adpcm_bits = 0;
658  for (ch = 0; ch < c->fullband_channels; ch++) {
659  for (band = 0; band < 32; band++) {
660  samples = c->subband[ch][band] - DCA_ADPCM_COEFFS;
661  pred_vq_id = ff_dcaadpcm_subband_analysis(&c->adpcm_ctx, samples,
662  SUBBAND_SAMPLES, estimated_diff);
663  if (pred_vq_id >= 0) {
664  c->prediction_mode[ch][band] = pred_vq_id;
665  c->consumed_adpcm_bits += 12; //12 bits to transmit prediction vq index
666  c->diff_peak_cb[ch][band] = find_peak(c, estimated_diff, 16);
667  } else {
668  c->prediction_mode[ch][band] = -1;
669  }
670  }
671  }
672 }
673 
674 static const int snr_fudge = 128;
675 #define USED_1ABITS 1
676 #define USED_26ABITS 4
677 
678 static inline int32_t get_step_size(DCAEncContext *c, int ch, int band)
679 {
680  int32_t step_size;
681 
682  if (c->bitrate_index == 3)
683  step_size = ff_dca_lossless_quant[c->abits[ch][band]];
684  else
685  step_size = ff_dca_lossy_quant[c->abits[ch][band]];
686 
687  return step_size;
688 }
689 
690 static int calc_one_scale(DCAEncContext *c, int32_t peak_cb, int abits,
691  softfloat *quant)
692 {
693  int32_t peak;
694  int our_nscale, try_remove;
695  softfloat our_quant;
696 
697  av_assert0(peak_cb <= 0);
698  av_assert0(peak_cb >= -2047);
699 
700  our_nscale = 127;
701  peak = c->cb_to_level[-peak_cb];
702 
703  for (try_remove = 64; try_remove > 0; try_remove >>= 1) {
704  if (scalefactor_inv[our_nscale - try_remove].e + stepsize_inv[abits].e <= 17)
705  continue;
706  our_quant.m = mul32(scalefactor_inv[our_nscale - try_remove].m, stepsize_inv[abits].m);
707  our_quant.e = scalefactor_inv[our_nscale - try_remove].e + stepsize_inv[abits].e - 17;
708  if ((ff_dca_quant_levels[abits] - 1) / 2 < quantize_value(peak, our_quant))
709  continue;
710  our_nscale -= try_remove;
711  }
712 
713  if (our_nscale >= 125)
714  our_nscale = 124;
715 
716  quant->m = mul32(scalefactor_inv[our_nscale].m, stepsize_inv[abits].m);
717  quant->e = scalefactor_inv[our_nscale].e + stepsize_inv[abits].e - 17;
718  av_assert0((ff_dca_quant_levels[abits] - 1) / 2 >= quantize_value(peak, *quant));
719 
720  return our_nscale;
721 }
722 
723 static inline void quantize_adpcm_subband(DCAEncContext *c, int ch, int band)
724 {
725  int32_t step_size;
726  int32_t diff_peak_cb = c->diff_peak_cb[ch][band];
727  c->scale_factor[ch][band] = calc_one_scale(c, diff_peak_cb,
728  c->abits[ch][band],
729  &c->quant[ch][band]);
730 
731  step_size = get_step_size(c, ch, band);
732  ff_dcaadpcm_do_real(c->prediction_mode[ch][band],
733  c->quant[ch][band],
734  ff_dca_scale_factor_quant7[c->scale_factor[ch][band]],
735  step_size, c->adpcm_history[ch][band], c->subband[ch][band],
736  c->adpcm_history[ch][band] + 4, c->quantized[ch][band],
737  SUBBAND_SAMPLES, c->cb_to_level[-diff_peak_cb]);
738 }
739 
741 {
742  int band, ch;
743 
744  for (ch = 0; ch < c->fullband_channels; ch++)
745  for (band = 0; band < 32; band++)
746  if (c->prediction_mode[ch][band] >= 0)
747  quantize_adpcm_subband(c, ch, band);
748 }
749 
751 {
752  int sample, band, ch;
753 
754  for (ch = 0; ch < c->fullband_channels; ch++) {
755  for (band = 0; band < 32; band++) {
756  if (c->prediction_mode[ch][band] == -1) {
757  for (sample = 0; sample < SUBBAND_SAMPLES; sample++) {
758  int32_t val = quantize_value(c->subband[ch][band][sample],
759  c->quant[ch][band]);
760  c->quantized[ch][band][sample] = val;
761  }
762  }
763  }
764  }
765 }
766 
767 static void accumulate_huff_bit_consumption(int abits, int32_t *quantized,
768  uint32_t *result)
769 {
770  uint8_t sel, id = abits - 1;
771  for (sel = 0; sel < ff_dca_quant_index_group_size[id]; sel++)
773  sel, id);
774 }
775 
776 static uint32_t set_best_code(uint32_t vlc_bits[DCA_CODE_BOOKS][7],
777  uint32_t clc_bits[DCA_CODE_BOOKS],
778  int32_t res[DCA_CODE_BOOKS])
779 {
780  uint8_t i, sel;
781  uint32_t best_sel_bits[DCA_CODE_BOOKS];
782  int32_t best_sel_id[DCA_CODE_BOOKS];
783  uint32_t t, bits = 0;
784 
785  for (i = 0; i < DCA_CODE_BOOKS; i++) {
786 
787  av_assert0(!((!!vlc_bits[i][0]) ^ (!!clc_bits[i])));
788  if (vlc_bits[i][0] == 0) {
789  /* do not transmit adjustment index for empty codebooks */
791  /* and skip it */
792  continue;
793  }
794 
795  best_sel_bits[i] = vlc_bits[i][0];
796  best_sel_id[i] = 0;
797  for (sel = 0; sel < ff_dca_quant_index_group_size[i]; sel++) {
798  if (best_sel_bits[i] > vlc_bits[i][sel] && vlc_bits[i][sel]) {
799  best_sel_bits[i] = vlc_bits[i][sel];
800  best_sel_id[i] = sel;
801  }
802  }
803 
804  /* 2 bits to transmit scale factor adjustment index */
805  t = best_sel_bits[i] + 2;
806  if (t < clc_bits[i]) {
807  res[i] = best_sel_id[i];
808  bits += t;
809  } else {
811  bits += clc_bits[i];
812  }
813  }
814  return bits;
815 }
816 
817 static uint32_t set_best_abits_code(int abits[DCAENC_SUBBANDS], int bands,
818  int32_t *res)
819 {
820  uint8_t i;
821  uint32_t t;
822  int32_t best_sel = 6;
823  int32_t best_bits = bands * 5;
824 
825  /* Check do we have subband which cannot be encoded by Huffman tables */
826  for (i = 0; i < bands; i++) {
827  if (abits[i] > 12 || abits[i] == 0) {
828  *res = best_sel;
829  return best_bits;
830  }
831  }
832 
833  for (i = 0; i < DCA_BITALLOC_12_COUNT; i++) {
834  t = dca_vlc_calc_alloc_bits(abits, bands, i);
835  if (t < best_bits) {
836  best_bits = t;
837  best_sel = i;
838  }
839  }
840 
841  *res = best_sel;
842  return best_bits;
843 }
844 
845 static int init_quantization_noise(DCAEncContext *c, int noise, int forbid_zero)
846 {
847  int ch, band, ret = USED_26ABITS | USED_1ABITS;
848  uint32_t huff_bit_count_accum[MAX_CHANNELS][DCA_CODE_BOOKS][7];
849  uint32_t clc_bit_count_accum[MAX_CHANNELS][DCA_CODE_BOOKS];
850  uint32_t bits_counter = 0;
851 
852  c->consumed_bits = 132 + 333 * c->fullband_channels;
853  c->consumed_bits += c->consumed_adpcm_bits;
854  if (c->lfe_channel)
855  c->consumed_bits += 72;
856 
857  /* attempt to guess the bit distribution based on the prevoius frame */
858  for (ch = 0; ch < c->fullband_channels; ch++) {
859  for (band = 0; band < 32; band++) {
860  int snr_cb = c->peak_cb[ch][band] - c->band_masking_cb[band] - noise;
861 
862  if (snr_cb >= 1312) {
863  c->abits[ch][band] = 26;
864  ret &= ~USED_1ABITS;
865  } else if (snr_cb >= 222) {
866  c->abits[ch][band] = 8 + mul32(snr_cb - 222, 69000000);
867  ret &= ~(USED_26ABITS | USED_1ABITS);
868  } else if (snr_cb >= 0) {
869  c->abits[ch][band] = 2 + mul32(snr_cb, 106000000);
870  ret &= ~(USED_26ABITS | USED_1ABITS);
871  } else if (forbid_zero || snr_cb >= -140) {
872  c->abits[ch][band] = 1;
873  ret &= ~USED_26ABITS;
874  } else {
875  c->abits[ch][band] = 0;
876  ret &= ~(USED_26ABITS | USED_1ABITS);
877  }
878  }
879  c->consumed_bits += set_best_abits_code(c->abits[ch], 32,
880  &c->bit_allocation_sel[ch]);
881  }
882 
883  /* Recalc scale_factor each time to get bits consumption in case of Huffman coding.
884  It is suboptimal solution */
885  /* TODO: May be cache scaled values */
886  for (ch = 0; ch < c->fullband_channels; ch++) {
887  for (band = 0; band < 32; band++) {
888  if (c->prediction_mode[ch][band] == -1) {
889  c->scale_factor[ch][band] = calc_one_scale(c, c->peak_cb[ch][band],
890  c->abits[ch][band],
891  &c->quant[ch][band]);
892  }
893  }
894  }
895  quantize_adpcm(c);
896  quantize_pcm(c);
897 
898  memset(huff_bit_count_accum, 0, MAX_CHANNELS * DCA_CODE_BOOKS * 7 * sizeof(uint32_t));
899  memset(clc_bit_count_accum, 0, MAX_CHANNELS * DCA_CODE_BOOKS * sizeof(uint32_t));
900  for (ch = 0; ch < c->fullband_channels; ch++) {
901  for (band = 0; band < 32; band++) {
902  if (c->abits[ch][band] && c->abits[ch][band] <= DCA_CODE_BOOKS) {
903  accumulate_huff_bit_consumption(c->abits[ch][band],
904  c->quantized[ch][band],
905  huff_bit_count_accum[ch][c->abits[ch][band] - 1]);
906  clc_bit_count_accum[ch][c->abits[ch][band] - 1] += bit_consumption[c->abits[ch][band]];
907  } else {
908  bits_counter += bit_consumption[c->abits[ch][band]];
909  }
910  }
911  }
912 
913  for (ch = 0; ch < c->fullband_channels; ch++) {
914  bits_counter += set_best_code(huff_bit_count_accum[ch],
915  clc_bit_count_accum[ch],
916  c->quant_index_sel[ch]);
917  }
918 
919  c->consumed_bits += bits_counter;
920 
921  return ret;
922 }
923 
925 {
926  /* Find the bounds where the binary search should work */
927  int low, high, down;
928  int used_abits = 0;
929  int forbid_zero = 1;
930 restart:
931  init_quantization_noise(c, c->worst_quantization_noise, forbid_zero);
932  low = high = c->worst_quantization_noise;
933  if (c->consumed_bits > c->frame_bits) {
934  while (c->consumed_bits > c->frame_bits) {
935  if (used_abits == USED_1ABITS && forbid_zero) {
936  forbid_zero = 0;
937  goto restart;
938  }
939  low = high;
940  high += snr_fudge;
941  used_abits = init_quantization_noise(c, high, forbid_zero);
942  }
943  } else {
944  while (c->consumed_bits <= c->frame_bits) {
945  high = low;
946  if (used_abits == USED_26ABITS)
947  goto out; /* The requested bitrate is too high, pad with zeros */
948  low -= snr_fudge;
949  used_abits = init_quantization_noise(c, low, forbid_zero);
950  }
951  }
952 
953  /* Now do a binary search between low and high to see what fits */
954  for (down = snr_fudge >> 1; down; down >>= 1) {
955  init_quantization_noise(c, high - down, forbid_zero);
956  if (c->consumed_bits <= c->frame_bits)
957  high -= down;
958  }
959  init_quantization_noise(c, high, forbid_zero);
960 out:
961  c->worst_quantization_noise = high;
962  if (high > c->worst_noise_ever)
963  c->worst_noise_ever = high;
964 }
965 
967 {
968  int k, ch;
969 
970  for (k = 0; k < 512; k++)
971  for (ch = 0; ch < c->channels; ch++) {
972  const int chi = c->channel_order_tab[ch];
973 
974  c->history[ch][k] = input[k * c->channels + chi];
975  }
976 }
977 
979 {
980  int ch, band;
981  int32_t step_size;
982  /* We fill in ADPCM work buffer for subbands which hasn't been ADPCM coded
983  * in current frame - we need this data if subband of next frame is
984  * ADPCM
985  */
986  for (ch = 0; ch < c->channels; ch++) {
987  for (band = 0; band < 32; band++) {
988  int32_t *samples = c->subband[ch][band] - DCA_ADPCM_COEFFS;
989  if (c->prediction_mode[ch][band] == -1) {
990  step_size = get_step_size(c, ch, band);
991 
992  ff_dca_core_dequantize(c->adpcm_history[ch][band],
993  c->quantized[ch][band]+12, step_size,
994  ff_dca_scale_factor_quant7[c->scale_factor[ch][band]], 0, 4);
995  } else {
996  AV_COPY128U(c->adpcm_history[ch][band], c->adpcm_history[ch][band]+4);
997  }
998  /* Copy dequantized values for LPC analysis.
999  * It reduces artifacts in case of extreme quantization,
1000  * example: in current frame abits is 1 and has no prediction flag,
1001  * but end of this frame is sine like signal. In this case, if LPC analysis uses
1002  * original values, likely LPC analysis returns good prediction gain, and sets prediction flag.
1003  * But there are no proper value in decoder history, so likely result will be no good.
1004  * Bitstream has "Predictor history flag switch", but this flag disables history for all subbands
1005  */
1006  samples[0] = c->adpcm_history[ch][band][0] * (1 << 7);
1007  samples[1] = c->adpcm_history[ch][band][1] * (1 << 7);
1008  samples[2] = c->adpcm_history[ch][band][2] * (1 << 7);
1009  samples[3] = c->adpcm_history[ch][band][3] * (1 << 7);
1010  }
1011  }
1012 }
1013 
1015 {
1016  if (c->lfe_channel)
1017  c->lfe_scale_factor = calc_one_scale(c, c->lfe_peak_cb, 11, &c->lfe_quant);
1018 }
1019 
1021 {
1022  /* SYNC */
1023  put_bits(&c->pb, 16, 0x7ffe);
1024  put_bits(&c->pb, 16, 0x8001);
1025 
1026  /* Frame type: normal */
1027  put_bits(&c->pb, 1, 1);
1028 
1029  /* Deficit sample count: none */
1030  put_bits(&c->pb, 5, 31);
1031 
1032  /* CRC is not present */
1033  put_bits(&c->pb, 1, 0);
1034 
1035  /* Number of PCM sample blocks */
1036  put_bits(&c->pb, 7, SUBBAND_SAMPLES - 1);
1037 
1038  /* Primary frame byte size */
1039  put_bits(&c->pb, 14, c->frame_size - 1);
1040 
1041  /* Audio channel arrangement */
1042  put_bits(&c->pb, 6, c->channel_config);
1043 
1044  /* Core audio sampling frequency */
1045  put_bits(&c->pb, 4, bitstream_sfreq[c->samplerate_index]);
1046 
1047  /* Transmission bit rate */
1048  put_bits(&c->pb, 5, c->bitrate_index);
1049 
1050  /* Embedded down mix: disabled */
1051  put_bits(&c->pb, 1, 0);
1052 
1053  /* Embedded dynamic range flag: not present */
1054  put_bits(&c->pb, 1, 0);
1055 
1056  /* Embedded time stamp flag: not present */
1057  put_bits(&c->pb, 1, 0);
1058 
1059  /* Auxiliary data flag: not present */
1060  put_bits(&c->pb, 1, 0);
1061 
1062  /* HDCD source: no */
1063  put_bits(&c->pb, 1, 0);
1064 
1065  /* Extension audio ID: N/A */
1066  put_bits(&c->pb, 3, 0);
1067 
1068  /* Extended audio data: not present */
1069  put_bits(&c->pb, 1, 0);
1070 
1071  /* Audio sync word insertion flag: after each sub-frame */
1072  put_bits(&c->pb, 1, 0);
1073 
1074  /* Low frequency effects flag: not present or 64x subsampling */
1075  put_bits(&c->pb, 2, c->lfe_channel ? 2 : 0);
1076 
1077  /* Predictor history switch flag: on */
1078  put_bits(&c->pb, 1, 1);
1079 
1080  /* No CRC */
1081  /* Multirate interpolator switch: non-perfect reconstruction */
1082  put_bits(&c->pb, 1, 0);
1083 
1084  /* Encoder software revision: 7 */
1085  put_bits(&c->pb, 4, 7);
1086 
1087  /* Copy history: 0 */
1088  put_bits(&c->pb, 2, 0);
1089 
1090  /* Source PCM resolution: 16 bits, not DTS ES */
1091  put_bits(&c->pb, 3, 0);
1092 
1093  /* Front sum/difference coding: no */
1094  put_bits(&c->pb, 1, 0);
1095 
1096  /* Surrounds sum/difference coding: no */
1097  put_bits(&c->pb, 1, 0);
1098 
1099  /* Dialog normalization: 0 dB */
1100  put_bits(&c->pb, 4, 0);
1101 }
1102 
1104 {
1105  int ch, i;
1106  /* Number of subframes */
1107  put_bits(&c->pb, 4, SUBFRAMES - 1);
1108 
1109  /* Number of primary audio channels */
1110  put_bits(&c->pb, 3, c->fullband_channels - 1);
1111 
1112  /* Subband activity count */
1113  for (ch = 0; ch < c->fullband_channels; ch++)
1114  put_bits(&c->pb, 5, DCAENC_SUBBANDS - 2);
1115 
1116  /* High frequency VQ start subband */
1117  for (ch = 0; ch < c->fullband_channels; ch++)
1118  put_bits(&c->pb, 5, DCAENC_SUBBANDS - 1);
1119 
1120  /* Joint intensity coding index: 0, 0 */
1121  for (ch = 0; ch < c->fullband_channels; ch++)
1122  put_bits(&c->pb, 3, 0);
1123 
1124  /* Transient mode codebook: A4, A4 (arbitrary) */
1125  for (ch = 0; ch < c->fullband_channels; ch++)
1126  put_bits(&c->pb, 2, 0);
1127 
1128  /* Scale factor code book: 7 bit linear, 7-bit sqrt table (for each channel) */
1129  for (ch = 0; ch < c->fullband_channels; ch++)
1130  put_bits(&c->pb, 3, 6);
1131 
1132  /* Bit allocation quantizer select: linear 5-bit */
1133  for (ch = 0; ch < c->fullband_channels; ch++)
1134  put_bits(&c->pb, 3, c->bit_allocation_sel[ch]);
1135 
1136  /* Quantization index codebook select */
1137  for (i = 0; i < DCA_CODE_BOOKS; i++)
1138  for (ch = 0; ch < c->fullband_channels; ch++)
1139  put_bits(&c->pb, ff_dca_quant_index_sel_nbits[i], c->quant_index_sel[ch][i]);
1140 
1141  /* Scale factor adjustment index: transmitted in case of Huffman coding */
1142  for (i = 0; i < DCA_CODE_BOOKS; i++)
1143  for (ch = 0; ch < c->fullband_channels; ch++)
1144  if (c->quant_index_sel[ch][i] < ff_dca_quant_index_group_size[i])
1145  put_bits(&c->pb, 2, 0);
1146 
1147  /* Audio header CRC check word: not transmitted */
1148 }
1149 
1150 static void put_subframe_samples(DCAEncContext *c, int ss, int band, int ch)
1151 {
1152  int i, j, sum, bits, sel;
1153  if (c->abits[ch][band] <= DCA_CODE_BOOKS) {
1154  av_assert0(c->abits[ch][band] > 0);
1155  sel = c->quant_index_sel[ch][c->abits[ch][band] - 1];
1156  // Huffman codes
1157  if (sel < ff_dca_quant_index_group_size[c->abits[ch][band] - 1]) {
1158  dca_vlc_enc_quant(&c->pb, &c->quantized[ch][band][ss * 8], 8,
1159  sel, c->abits[ch][band] - 1);
1160  return;
1161  }
1162 
1163  // Block codes
1164  if (c->abits[ch][band] <= 7) {
1165  for (i = 0; i < 8; i += 4) {
1166  sum = 0;
1167  for (j = 3; j >= 0; j--) {
1168  sum *= ff_dca_quant_levels[c->abits[ch][band]];
1169  sum += c->quantized[ch][band][ss * 8 + i + j];
1170  sum += (ff_dca_quant_levels[c->abits[ch][band]] - 1) / 2;
1171  }
1172  put_bits(&c->pb, bit_consumption[c->abits[ch][band]] / 4, sum);
1173  }
1174  return;
1175  }
1176  }
1177 
1178  for (i = 0; i < 8; i++) {
1179  bits = bit_consumption[c->abits[ch][band]] / 16;
1180  put_sbits(&c->pb, bits, c->quantized[ch][band][ss * 8 + i]);
1181  }
1182 }
1183 
1184 static void put_subframe(DCAEncContext *c, int subframe)
1185 {
1186  int i, band, ss, ch;
1187 
1188  /* Subsubframes count */
1189  put_bits(&c->pb, 2, SUBSUBFRAMES -1);
1190 
1191  /* Partial subsubframe sample count: dummy */
1192  put_bits(&c->pb, 3, 0);
1193 
1194  /* Prediction mode: no ADPCM, in each channel and subband */
1195  for (ch = 0; ch < c->fullband_channels; ch++)
1196  for (band = 0; band < DCAENC_SUBBANDS; band++)
1197  put_bits(&c->pb, 1, !(c->prediction_mode[ch][band] == -1));
1198 
1199  /* Prediction VQ address */
1200  for (ch = 0; ch < c->fullband_channels; ch++)
1201  for (band = 0; band < DCAENC_SUBBANDS; band++)
1202  if (c->prediction_mode[ch][band] >= 0)
1203  put_bits(&c->pb, 12, c->prediction_mode[ch][band]);
1204 
1205  /* Bit allocation index */
1206  for (ch = 0; ch < c->fullband_channels; ch++) {
1207  if (c->bit_allocation_sel[ch] == 6) {
1208  for (band = 0; band < DCAENC_SUBBANDS; band++) {
1209  put_bits(&c->pb, 5, c->abits[ch][band]);
1210  }
1211  } else {
1212  dca_vlc_enc_alloc(&c->pb, c->abits[ch], DCAENC_SUBBANDS,
1213  c->bit_allocation_sel[ch]);
1214  }
1215  }
1216 
1217  if (SUBSUBFRAMES > 1) {
1218  /* Transition mode: none for each channel and subband */
1219  for (ch = 0; ch < c->fullband_channels; ch++)
1220  for (band = 0; band < DCAENC_SUBBANDS; band++)
1221  if (c->abits[ch][band])
1222  put_bits(&c->pb, 1, 0); /* codebook A4 */
1223  }
1224 
1225  /* Scale factors */
1226  for (ch = 0; ch < c->fullband_channels; ch++)
1227  for (band = 0; band < DCAENC_SUBBANDS; band++)
1228  if (c->abits[ch][band])
1229  put_bits(&c->pb, 7, c->scale_factor[ch][band]);
1230 
1231  /* Joint subband scale factor codebook select: not transmitted */
1232  /* Scale factors for joint subband coding: not transmitted */
1233  /* Stereo down-mix coefficients: not transmitted */
1234  /* Dynamic range coefficient: not transmitted */
1235  /* Stde information CRC check word: not transmitted */
1236  /* VQ encoded high frequency subbands: not transmitted */
1237 
1238  /* LFE data: 8 samples and scalefactor */
1239  if (c->lfe_channel) {
1240  for (i = 0; i < DCA_LFE_SAMPLES; i++)
1241  put_bits(&c->pb, 8, quantize_value(c->downsampled_lfe[i], c->lfe_quant) & 0xff);
1242  put_bits(&c->pb, 8, c->lfe_scale_factor);
1243  }
1244 
1245  /* Audio data (subsubframes) */
1246  for (ss = 0; ss < SUBSUBFRAMES ; ss++)
1247  for (ch = 0; ch < c->fullband_channels; ch++)
1248  for (band = 0; band < DCAENC_SUBBANDS; band++)
1249  if (c->abits[ch][band])
1250  put_subframe_samples(c, ss, band, ch);
1251 
1252  /* DSYNC */
1253  put_bits(&c->pb, 16, 0xffff);
1254 }
1255 
1256 static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
1257  const AVFrame *frame, int *got_packet_ptr)
1258 {
1259  DCAEncContext *c = avctx->priv_data;
1260  const int32_t *samples;
1261  int ret, i;
1262 
1263  if ((ret = ff_get_encode_buffer(avctx, avpkt, c->frame_size, 0)) < 0)
1264  return ret;
1265 
1266  samples = (const int32_t *)frame->data[0];
1267 
1269  if (c->lfe_channel)
1271 
1273  if (c->options.adpcm_mode)
1274  adpcm_analysis(c);
1275  find_peaks(c);
1276  assign_bits(c);
1277  calc_lfe_scales(c);
1279 
1280  init_put_bits(&c->pb, avpkt->data, avpkt->size);
1284  for (i = 0; i < SUBFRAMES; i++)
1285  put_subframe(c, i);
1286 
1287  flush_put_bits(&c->pb);
1288  memset(put_bits_ptr(&c->pb), 0, put_bytes_left(&c->pb, 0));
1289 
1290  *got_packet_ptr = 1;
1291  return 0;
1292 }
1293 
1294 #define DCAENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
1295 
1296 static const AVOption options[] = {
1297  { "dca_adpcm", "Use ADPCM encoding", offsetof(DCAEncContext, options.adpcm_mode), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, DCAENC_FLAGS },
1298  { NULL },
1299 };
1300 
1301 static const AVClass dcaenc_class = {
1302  .class_name = "DCA (DTS Coherent Acoustics)",
1303  .item_name = av_default_item_name,
1304  .option = options,
1305  .version = LIBAVUTIL_VERSION_INT,
1306 };
1307 
1308 static const FFCodecDefault defaults[] = {
1309  { "b", "1411200" },
1310  { NULL },
1311 };
1312 
1314  .p.name = "dca",
1315  CODEC_LONG_NAME("DCA (DTS Coherent Acoustics)"),
1316  .p.type = AVMEDIA_TYPE_AUDIO,
1317  .p.id = AV_CODEC_ID_DTS,
1318  .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_EXPERIMENTAL,
1319  .priv_data_size = sizeof(DCAEncContext),
1320  .init = encode_init,
1321  .close = encode_close,
1323  .caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
1324  .p.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S32,
1326  .p.supported_samplerates = sample_rates,
1330  .p.ch_layouts = (const AVChannelLayout[]){
1336  { 0 },
1337  },
1338  .defaults = defaults,
1339  .p.priv_class = &dcaenc_class,
1340 };
dcamath.h
DCAEncContext::pb
PutBitContext pb
Definition: dcaenc.c:62
lfe_downsample
static void lfe_downsample(DCAEncContext *c, const int32_t *input)
Definition: dcaenc.c:414
AVCodecContext::frame_size
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:1035
ff_exp10
static av_always_inline double ff_exp10(double x)
Compute 10^x for floating point values.
Definition: ffmath.h:42
FF_CODEC_CAP_INIT_CLEANUP
#define FF_CODEC_CAP_INIT_CLEANUP
The codec allows calling the close function for deallocation even if the init function returned a fai...
Definition: codec_internal.h:42
dca_vlc_calc_quant_bits
static uint32_t dca_vlc_calc_quant_bits(const int values[], uint8_t n, uint8_t sel, uint8_t table)
Definition: dcaenc.c:460
AVERROR
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later. That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another. Frame references ownership and permissions
opt.h
ff_dcaadpcm_do_real
int ff_dcaadpcm_do_real(int pred_vq_index, softfloat quant, int32_t scale_factor, int32_t step_size, const int32_t *prev_hist, const int32_t *in, int32_t *next_hist, int32_t *out, int len, int32_t peak)
Definition: dcaadpcm.c:183
mem_internal.h
DCAADPCMEncContext
Definition: dcaadpcm.h:29
DCAEncContext::subband
int32_t * subband[MAX_CHANNELS][DCAENC_SUBBANDS]
Definition: dcaenc.c:85
out
FILE * out
Definition: movenc.c:54
AV_CHANNEL_LAYOUT_STEREO
#define AV_CHANNEL_LAYOUT_STEREO
Definition: channel_layout.h:369
AVCodecContext::sample_rate
int sample_rate
samples per second
Definition: avcodec.h:1007
cb
static double cb(void *priv, double x, double y)
Definition: vf_geq.c:239
dca.h
options
static const AVOption options[]
Definition: dcaenc.c:1296
DCAEncContext::frame_size
int frame_size
Definition: dcaenc.c:67
thread.h
walk_band_low
static void walk_band_low(DCAEncContext *c, int band, int channel, walk_band_t walk, int32_t *arg)
Definition: dcaenc.c:549
DCAEncContext::mdct_fn
av_tx_fn mdct_fn
Definition: dcaenc.c:65
DCAEncContext::worst_noise_ever
int32_t worst_noise_ever
Definition: dcaenc.c:99
ff_dca_bit_rates
const uint32_t ff_dca_bit_rates[32]
Definition: dcadata.c:32
AV_CH_LAYOUT_MONO
#define AV_CH_LAYOUT_MONO
Definition: channel_layout.h:210
SUBSUBFRAMES
#define SUBSUBFRAMES
Definition: dcaenc.c:50
AVTXContext
Definition: tx_priv.h:228
ff_dcaadpcm_free
av_cold void ff_dcaadpcm_free(DCAADPCMEncContext *s)
Definition: dcaadpcm.c:225
scalefactor_inv
static const softfloat scalefactor_inv[128]
Definition: dcaenc.h:63
put_sbits
static void put_sbits(PutBitContext *pb, int n, int32_t value)
Definition: put_bits.h:281
init_put_bits
static void init_put_bits(PutBitContext *s, uint8_t *buffer, int buffer_size)
Initialize the PutBitContext s.
Definition: put_bits.h:62
erb
static const uint16_t erb[]
Definition: dcaenc.h:48
AV_CHANNEL_LAYOUT_2_2
#define AV_CHANNEL_LAYOUT_2_2
Definition: channel_layout.h:376
lfe_index
static const uint8_t lfe_index[7]
Definition: dca_lbr.c:107
put_subframe
static void put_subframe(DCAEncContext *c, int subframe)
Definition: dcaenc.c:1184
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:330
put_bits
static void put_bits(Jpeg2000EncoderContext *s, int val, int n)
put n times val bit
Definition: j2kenc.c:221
DCAEncContext::options
CompressionOptions options
Definition: dcaenc.c:66
AVPacket::data
uint8_t * data
Definition: packet.h:374
get_step_size
static int32_t get_step_size(DCAEncContext *c, int ch, int band)
Definition: dcaenc.c:678
ff_dca_lossy_quant
const uint32_t ff_dca_lossy_quant[32]
Definition: dcadata.c:4223
AVOption
AVOption.
Definition: opt.h:251
encode.h
b
#define b
Definition: input.c:41
table
static const uint16_t table[]
Definition: prosumer.c:205
MAX_CHANNELS
#define MAX_CHANNELS
Definition: dcaenc.c:43
data
const char data[16]
Definition: mxf.c:146
calc_lfe_scales
static void calc_lfe_scales(DCAEncContext *c)
Definition: dcaenc.c:1014
FFCodec
Definition: codec_internal.h:119
DCAEncContext::adpcm_history
int32_t adpcm_history[MAX_CHANNELS][DCAENC_SUBBANDS][DCA_ADPCM_COEFFS *2]
Definition: dcaenc.c:83
fc
#define fc(width, name, range_min, range_max)
Definition: cbs_av1.c:551
update_band_masking
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:579
calc_one_scale
static int calc_one_scale(DCAEncContext *c, int32_t peak_cb, int abits, softfloat *quant)
Definition: dcaenc.c:690
DCAEncContext::lfe_quant
softfloat lfe_quant
Definition: dcaenc.c:78
quantize_value
static int32_t quantize_value(int32_t value, softfloat quant)
Definition: dcaenc.h:149
DCAEncContext::band_interpolation
const int32_t * band_interpolation
Definition: dcaenc.c:75
encode_frame
static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: dcaenc.c:1256
put_frame_header
static void put_frame_header(DCAEncContext *c)
Definition: dcaenc.c:1020
DCAEncContext::adpcm_ctx
DCAADPCMEncContext adpcm_ctx
Definition: dcaenc.c:63
av_tx_init
av_cold int av_tx_init(AVTXContext **ctx, av_tx_fn *tx, enum AVTXType type, int inv, int len, const void *scale, uint64_t flags)
Initialize a transform context with the given configuration (i)MDCTs with an odd length are currently...
Definition: tx.c:883
DCAEncContext::history
int32_t history[MAX_CHANNELS][512]
Definition: dcaenc.c:84
calc_masking
static void calc_masking(DCAEncContext *c, const int32_t *input)
Definition: dcaenc.c:589
dca_vlc_enc_quant
static void dca_vlc_enc_quant(PutBitContext *pb, const int values[], uint8_t n, uint8_t sel, uint8_t table)
Definition: dcaenc.c:469
DCAEncContext::fullband_channels
int fullband_channels
Definition: dcaenc.c:69
FFCodecDefault
Definition: codec_internal.h:89
FFCodec::p
AVCodec p
The public AVCodec.
Definition: codec_internal.h:123
adpcm_analysis
static void adpcm_analysis(DCAEncContext *c)
Definition: dcaenc.c:650
ff_dca_fir_32bands_nonperfect
const float ff_dca_fir_32bands_nonperfect[512]
Definition: dcadata.c:6808
AVCodecContext::ch_layout
AVChannelLayout ch_layout
Audio channel layout.
Definition: avcodec.h:2059
ff_dca_quant_index_group_size
const uint8_t ff_dca_quant_index_group_size[DCA_CODE_BOOKS]
Definition: dcadata.c:53
CompressionOptions::adpcm_mode
int adpcm_mode
Definition: dcaenc.c:57
accumulate_huff_bit_consumption
static void accumulate_huff_bit_consumption(int abits, int32_t *quantized, uint32_t *result)
Definition: dcaenc.c:767
init_quantization_noise
static int init_quantization_noise(DCAEncContext *c, int noise, int forbid_zero)
Definition: dcaenc.c:845
val
static double val(void *priv, double ch)
Definition: aeval.c:77
scale
static av_always_inline float scale(float x, float s)
Definition: vf_v360.c:1389
ff_dca_quant_levels
const uint32_t ff_dca_quant_levels[32]
Definition: dcadata.c:4215
ss
#define ss(width, name, subs,...)
Definition: cbs_vp9.c:260
dcadata.h
FF_CODEC_ENCODE_CB
#define FF_CODEC_ENCODE_CB(func)
Definition: codec_internal.h:307
dca_core.h
DCAEncContext::auf
int32_t auf[9][AUBANDS][256]
Definition: dcaenc.c:106
DCAEncContext::consumed_bits
int consumed_bits
Definition: dcaenc.c:100
AV_CH_LAYOUT_STEREO
#define AV_CH_LAYOUT_STEREO
Definition: channel_layout.h:211
quant
static int quant(float coef, const float Q, const float rounding)
Quantize one coefficient.
Definition: aacenc_utils.h:59
put_bytes_left
static int put_bytes_left(const PutBitContext *s, int round_up)
Definition: put_bits.h:135
create_enc_table
static av_cold void create_enc_table(uint16_t dst[][2], unsigned count, const uint8_t(**src_tablep)[2])
Definition: dcaenc.c:167
bit_consumption
static const int bit_consumption[27]
Definition: dcaenc.h:101
walk_band_high
static void walk_band_high(DCAEncContext *c, int band, int channel, walk_band_t walk, int32_t *arg)
Definition: dcaenc.c:564
quantize_adpcm_subband
static void quantize_adpcm_subband(DCAEncContext *c, int ch, int band)
Definition: dcaenc.c:723
DCAEncContext::quantized
int32_t quantized[MAX_CHANNELS][DCAENC_SUBBANDS][SUBBAND_SAMPLES]
Definition: dcaenc.c:86
avassert.h
DCAEncContext::lfe_fir_64i
int32_t lfe_fir_64i[512]
Definition: dcaenc.c:109
ff_thread_once
static int ff_thread_once(char *control, void(*routine)(void))
Definition: thread.h:184
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:180
AV_CODEC_CAP_EXPERIMENTAL
#define AV_CODEC_CAP_EXPERIMENTAL
Codec is experimental and is thus avoided in favor of non experimental encoders.
Definition: codec.h:105
av_cold
#define av_cold
Definition: attributes.h:90
DCA_LFE_SAMPLES
#define DCA_LFE_SAMPLES
Definition: dcaenc.c:46
ff_dca_encoder
const FFCodec ff_dca_encoder
Definition: dcaenc.c:1313
av_tx_fn
void(* av_tx_fn)(AVTXContext *s, void *out, void *in, ptrdiff_t stride)
Function pointer to a function to perform the transform.
Definition: tx.h:127
quantize_adpcm
static void quantize_adpcm(DCAEncContext *c)
Definition: dcaenc.c:740
DCAEncContext::lfe_channel
int lfe_channel
Definition: dcaenc.c:71
DCAEncContext::abits
int abits[MAX_CHANNELS][DCAENC_SUBBANDS]
Definition: dcaenc.c:92
DCAEncContext::peak_cb
int32_t peak_cb[MAX_CHANNELS][DCAENC_SUBBANDS]
Definition: dcaenc.c:87
DCAEncContext::band_spectrum
const int32_t * band_spectrum
Definition: dcaenc.c:76
DCAEncContext::lfe_scale_factor
int lfe_scale_factor
Definition: dcaenc.c:77
s
#define s(width, name)
Definition: cbs_vp9.c:256
DCA_ADPCM_COEFFS
#define DCA_ADPCM_COEFFS
Definition: dcadata.h:28
hom
static double hom(double f)
Definition: dcaenc.c:113
bitalloc_table
static uint16_t bitalloc_table[DCA_NUM_BITALLOC_CODES][2]
Definition: dcaenc.c:164
DCAEncContext::eff_masking_curve_cb
int32_t eff_masking_curve_cb[256]
Definition: dcaenc.c:96
DCAEncContext::downsampled_lfe
int32_t downsampled_lfe[DCA_LFE_SAMPLES]
Definition: dcaenc.c:89
AV_TX_INT32_MDCT
@ AV_TX_INT32_MDCT
Definition: tx.h:70
SUBBAND_SAMPLES
#define SUBBAND_SAMPLES
Definition: dcaenc.c:51
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
init
int(* init)(AVBSFContext *ctx)
Definition: dts2pts_bsf.c:365
DCAEncContext::samplerate_index
int samplerate_index
Definition: dcaenc.c:72
bits
uint8_t bits
Definition: vp3data.h:128
ff_dca_bitalloc_offsets
const int8_t ff_dca_bitalloc_offsets[DCA_CODE_BOOKS]
Definition: dcahuff.c:41
CODEC_OLD_CHANNEL_LAYOUTS
#define CODEC_OLD_CHANNEL_LAYOUTS(...)
Definition: codec_internal.h:294
set_best_abits_code
static uint32_t set_best_abits_code(int abits[DCAENC_SUBBANDS], int bands, int32_t *res)
Definition: dcaenc.c:817
av_assert0
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
dcaenc_init_static_tables
static av_cold void dcaenc_init_static_tables(void)
Definition: dcaenc.c:184
CompressionOptions
Definition: dcaenc.c:56
bands
static const float bands[]
Definition: af_superequalizer.c:56
PutBitContext
Definition: put_bits.h:50
CODEC_LONG_NAME
#define CODEC_LONG_NAME(str)
Definition: codec_internal.h:264
arg
const char * arg
Definition: jacosubdec.c:67
FFABS
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
Definition: common.h:64
if
if(ret)
Definition: filter_design.txt:179
softfloat
Definition: dcaenc.h:29
LIBAVUTIL_VERSION_INT
#define LIBAVUTIL_VERSION_INT
Definition: version.h:85
AV_ONCE_INIT
#define AV_ONCE_INIT
Definition: thread.h:182
AVClass
Describe the class of an AVClass context structure.
Definition: log.h:66
result
and forward the result(frame or status change) to the corresponding input. If nothing is possible
adjust_jnd
static void adjust_jnd(DCAEncContext *c, const int32_t in[512], int32_t out_cb[256])
Definition: dcaenc.c:516
NULL
#define NULL
Definition: coverity.c:32
LOCAL_ALIGNED_32
#define LOCAL_ALIGNED_32(t, v,...)
Definition: mem_internal.h:137
ff_dca_lossless_quant
const uint32_t ff_dca_lossless_quant[32]
Definition: dcadata.c:4231
mul32
static int32_t mul32(int32_t a, int32_t b)
Definition: dcamath.h:52
ff_dca_lfe_fir_64
const float ff_dca_lfe_fir_64[256]
Definition: dcadata.c:7339
AV_COPY128U
#define AV_COPY128U(d, s)
Definition: intreadwrite.h:580
USED_26ABITS
#define USED_26ABITS
Definition: dcaenc.c:676
AVCodecContext::bit_rate
int64_t bit_rate
the average bitrate
Definition: avcodec.h:448
bitalloc_12_table
static uint16_t bitalloc_12_table[DCA_BITALLOC_12_COUNT][12+1][2]
Definition: dcaenc.c:162
av_default_item_name
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:237
USED_1ABITS
#define USED_1ABITS
Definition: dcaenc.c:675
stepsize_inv
static const softfloat stepsize_inv[27]
Definition: dcaenc.h:53
AV_CH_LAYOUT_5POINT1
#define AV_CH_LAYOUT_5POINT1
Definition: channel_layout.h:221
softfloat::e
int32_t e
Definition: dcaenc.h:31
DCAEncContext::band_masking_cb
int32_t band_masking_cb[32]
Definition: dcaenc.c:97
defaults
static const FFCodecDefault defaults[]
Definition: dcaenc.c:1308
abs
#define abs(x)
Definition: cuda_runtime.h:35
ff_dcaadpcm_subband_analysis
int ff_dcaadpcm_subband_analysis(const DCAADPCMEncContext *s, const int32_t *in, int len, int *diff)
Definition: dcaadpcm.c:125
exp
int8_t exp
Definition: eval.c:72
AVOnce
#define AVOnce
Definition: thread.h:181
c
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
Definition: undefined.txt:32
DCAEncContext::channels
int channels
Definition: dcaenc.c:70
encode_close
static av_cold int encode_close(AVCodecContext *avctx)
Definition: dcaenc.c:354
SUBFRAMES
#define SUBFRAMES
Definition: dcaenc.c:49
DCAEncContext::worst_quantization_noise
int32_t worst_quantization_noise
Definition: dcaenc.c:98
DCAEncContext::band_interpolation_tab
int32_t band_interpolation_tab[2][512]
Definition: dcaenc.c:104
DCAEncContext::lfe_peak_cb
int32_t lfe_peak_cb
Definition: dcaenc.c:79
f
f
Definition: af_crystalizer.c:122
AV_CODEC_CAP_DR1
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
Definition: codec.h:52
ff_dca_scale_factor_quant7
const uint32_t ff_dca_scale_factor_quant7[128]
Definition: dcadata.c:4172
AVPacket::size
int size
Definition: packet.h:375
subband_bufer_free
static void subband_bufer_free(DCAEncContext *c)
Definition: dcaenc.c:153
DCAEncContext::quant
softfloat quant[MAX_CHANNELS][DCAENC_SUBBANDS]
Definition: dcaenc.c:94
AVChannelLayout
An AVChannelLayout holds information about the channel layout of audio data.
Definition: channel_layout.h:301
codec_internal.h
walk_band_t
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:545
add_cb
static int32_t add_cb(DCAEncContext *c, int32_t a, int32_t b)
Definition: dcaenc.c:489
for
for(k=2;k<=8;++k)
Definition: h264pred_template.c:425
AV_CODEC_ID_DTS
@ AV_CODEC_ID_DTS
Definition: codec_id.h:440
AV_SAMPLE_FMT_NONE
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:56
sample
#define sample
Definition: flacdsp_template.c:44
DCA_CODE_BOOKS
#define DCA_CODE_BOOKS
Definition: dcahuff.h:32
DCA_BITALLOC_12_COUNT
#define DCA_BITALLOC_12_COUNT
Definition: dcahuff.h:33
encode_init
static int encode_init(AVCodecContext *avctx)
Definition: dcaenc.c:202
fill_in_adpcm_bufer
static void fill_in_adpcm_bufer(DCAEncContext *c)
Definition: dcaenc.c:978
DCAEncContext::bitrate_index
int bitrate_index
Definition: dcaenc.c:73
dcaadpcm.h
DCA_MAX_FRAME_SIZE
#define DCA_MAX_FRAME_SIZE
Definition: dcaenc.c:44
quantize_pcm
static void quantize_pcm(DCAEncContext *c)
Definition: dcaenc.c:750
DCAEncContext::masking_curve_cb
int32_t masking_curve_cb[SUBSUBFRAMES][256]
Definition: dcaenc.c:90
dcaenc.h
a
The reader does not expect b to be semantically here and if the code is changed by maybe adding a a division or other the signedness will almost certainly be mistaken To avoid this confusion a new type was SUINT is the C unsigned type but it holds a signed int to use the same example SUINT a
Definition: undefined.txt:41
input
and forward the test the status of outputs and forward it to the corresponding return FFERROR_NOT_READY If the filters stores internally one or a few frame for some input
Definition: filter_design.txt:172
put_primary_audio_header
static void put_primary_audio_header(DCAEncContext *c)
Definition: dcaenc.c:1103
M_PI
#define M_PI
Definition: mathematics.h:52
av_tx_uninit
av_cold void av_tx_uninit(AVTXContext **ctx)
Frees a context and sets *ctx to NULL, does nothing when *ctx == NULL.
Definition: tx.c:294
bitalloc_tables
static const uint16_t(*[DCA_CODE_BOOKS][8] bitalloc_tables)[2]
Definition: dcaenc.c:165
sample_rates
sample_rates
Definition: ffmpeg_filter.c:156
DCAEncContext::quant_index_sel
int32_t quant_index_sel[MAX_CHANNELS][DCA_CODE_BOOKS]
Definition: dcaenc.c:95
dca_vlc_calc_alloc_bits
static uint32_t dca_vlc_calc_alloc_bits(const int values[], uint8_t n, uint8_t sel)
Definition: dcaenc.c:444
layout
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel layout
Definition: filter_design.txt:18
DCAEncContext::frame_bits
int frame_bits
Definition: dcaenc.c:68
AV_CH_LAYOUT_5POINT0
#define AV_CH_LAYOUT_5POINT0
Definition: channel_layout.h:220
find_peaks
static void find_peaks(DCAEncContext *c)
Definition: dcaenc.c:636
ff_dca_quant_index_sel_nbits
const uint8_t ff_dca_quant_index_sel_nbits[DCA_CODE_BOOKS]
Definition: dcadata.c:49
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:269
code
and forward the test the status of outputs and forward it to the corresponding return FFERROR_NOT_READY If the filters stores internally one or a few frame for some it can consider them to be part of the FIFO and delay acknowledging a status change accordingly Example code
Definition: filter_design.txt:178
common.h
av_assert1
#define av_assert1(cond)
assert() equivalent, that does not lie in speed critical code.
Definition: avassert.h:53
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:55
value
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf default value
Definition: writing_filters.txt:86
AUBANDS
#define AUBANDS
Definition: dcaenc.c:52
AVCodec::name
const char * name
Name of the codec implementation.
Definition: codec.h:211
len
int len
Definition: vorbis_enc_data.h:426
norm__
static int32_t norm__(int64_t a, int bits)
Definition: dcamath.h:27
DCAEncContext::channel_config
int channel_config
Definition: dcaenc.c:74
DCAEncContext::mdct
AVTXContext * mdct
Definition: dcaenc.c:64
av_calloc
void * av_calloc(size_t nmemb, size_t size)
Definition: mem.c:272
channel_reorder_nolfe
static const int8_t channel_reorder_nolfe[7][5]
Definition: dca_lbr.c:87
avcodec.h
snr_fudge
static const int snr_fudge
Definition: dcaenc.c:674
ret
ret
Definition: filter_design.txt:187
FFSWAP
#define FFSWAP(type, a, b)
Definition: macros.h:52
AVClass::class_name
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
Definition: log.h:71
subband_transform
static void subband_transform(DCAEncContext *c, const int32_t *input)
Definition: dcaenc.c:364
frame
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
Definition: filter_design.txt:264
DCAEncContext::consumed_adpcm_bits
int consumed_adpcm_bits
Number of bits to transmit ADPCM related info.
Definition: dcaenc.c:101
channel_reorder_lfe
static const int8_t channel_reorder_lfe[7][5]
Definition: dca_lbr.c:97
id
enum AVCodecID id
Definition: dts2pts_bsf.c:364
ff_dca_core_dequantize
static void ff_dca_core_dequantize(int32_t *output, const int32_t *input, int32_t step_size, int32_t scale, int residual, int len)
Definition: dca_core.h:226
ff_dcaadpcm_init
av_cold int ff_dcaadpcm_init(DCAADPCMEncContext *s)
Definition: dcaadpcm.c:212
AVCodecContext
main external API structure.
Definition: avcodec.h:398
power
static float power(float r, float g, float b, float max)
Definition: preserve_color.h:45
put_bits_ptr
static uint8_t * put_bits_ptr(PutBitContext *s)
Return the pointer to the byte where the bitstream writer will put the next bit.
Definition: put_bits.h:377
channel_layout.h
noise
static int noise(AVBSFContext *ctx, AVPacket *pkt)
Definition: noise_bsf.c:120
ff_get_encode_buffer
int ff_get_encode_buffer(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int flags)
Get a buffer for a packet.
Definition: encode.c:79
values
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return values
Definition: filter_design.txt:263
ff_dca_bitalloc_sizes
const uint8_t ff_dca_bitalloc_sizes[DCA_CODE_BOOKS]
Definition: dcahuff.c:37
samples
Filter the word “frame” indicates either a video frame or a group of audio samples
Definition: filter_design.txt:8
ff_dca_vlc_src_tables
const uint8_t ff_dca_vlc_src_tables[][2]
Definition: dcahuff.c:58
DCAEncContext
Definition: dcaenc.c:60
ffmath.h
subband_bufer_alloc
static int subband_bufer_alloc(DCAEncContext *c)
Definition: dcaenc.c:132
assign_bits
static void assign_bits(DCAEncContext *c)
Definition: dcaenc.c:924
DCAENC_SUBBANDS
#define DCAENC_SUBBANDS
Definition: dcaenc.c:48
get_cb
static int32_t get_cb(DCAEncContext *c, int32_t in)
Definition: dcaenc.c:477
DCAEncContext::cos_table
int32_t cos_table[2048]
Definition: dcaenc.c:103
bitstream_sfreq
static const uint8_t bitstream_sfreq[]
Definition: dcaenc.h:38
add
static float add(float src0, float src1)
Definition: dnn_backend_native_layer_mathbinary.c:35
find_peak
static int32_t find_peak(DCAEncContext *c, const int32_t *in, int len)
Definition: dcaenc.c:624
flush_put_bits
static void flush_put_bits(PutBitContext *s)
Pad the end of the output stream with zeros.
Definition: put_bits.h:143
AV_CHANNEL_LAYOUT_MONO
#define AV_CHANNEL_LAYOUT_MONO
Definition: channel_layout.h:368
av_free
#define av_free(p)
Definition: tableprint_vlc.h:33
FFALIGN
#define FFALIGN(x, a)
Definition: macros.h:78
AVPacket
This structure stores compressed data.
Definition: packet.h:351
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:425
AV_OPT_TYPE_BOOL
@ AV_OPT_TYPE_BOOL
Definition: opt.h:244
ff_dca_fir_32bands_perfect
const float ff_dca_fir_32bands_perfect[512]
Definition: dcadata.c:6293
shift_history
static void shift_history(DCAEncContext *c, const int32_t *input)
Definition: dcaenc.c:966
set_best_code
static uint32_t set_best_code(uint32_t vlc_bits[DCA_CODE_BOOKS][7], uint32_t clc_bits[DCA_CODE_BOOKS], int32_t res[DCA_CODE_BOOKS])
Definition: dcaenc.c:776
dca_vlc_enc_alloc
static void dca_vlc_enc_alloc(PutBitContext *pb, const int values[], uint8_t n, uint8_t sel)
Definition: dcaenc.c:452
softfloat::m
int32_t m
Definition: dcaenc.h:30
int32_t
int32_t
Definition: audioconvert.c:56
DCAEncContext::prediction_mode
int32_t prediction_mode[MAX_CHANNELS][DCAENC_SUBBANDS]
Definition: dcaenc.c:82
DCAEncContext::cb_to_add
int32_t cb_to_add[256]
Definition: dcaenc.c:107
coeff
static const double coeff[2][5]
Definition: vf_owdenoise.c:78
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:27
COS_T
#define COS_T(x)
Definition: dcaenc.c:54
h
h
Definition: vp9dsp_template.c:2038
AV_CHANNEL_LAYOUT_5POINT0
#define AV_CHANNEL_LAYOUT_5POINT0
Definition: channel_layout.h:378
DCAEncContext::cb_to_level
int32_t cb_to_level[2048]
Definition: dcaenc.c:108
put_subframe_samples
static void put_subframe_samples(DCAEncContext *c, int ss, int band, int ch)
Definition: dcaenc.c:1150
DCAEncContext::bit_allocation_sel
int32_t bit_allocation_sel[MAX_CHANNELS]
Definition: dcaenc.c:91
DCAEncContext::band_spectrum_tab
int32_t band_spectrum_tab[2][8]
Definition: dcaenc.c:105
AV_CHANNEL_LAYOUT_5POINT1
#define AV_CHANNEL_LAYOUT_5POINT1
Definition: channel_layout.h:379
put_bits.h
calc_power
static void calc_power(DCAEncContext *c, const int32_t in[2 *256], int32_t power[256])
Definition: dcaenc.c:499
AV_SAMPLE_FMT_S32
@ AV_SAMPLE_FMT_S32
signed 32 bits
Definition: samplefmt.h:59
AV_CH_LAYOUT_2_2
#define AV_CH_LAYOUT_2_2
Definition: channel_layout.h:218
channel
channel
Definition: ebur128.h:39
DCAEncContext::diff_peak_cb
int32_t diff_peak_cb[MAX_CHANNELS][DCAENC_SUBBANDS]
expected peak of residual signal
Definition: dcaenc.c:88
tx.h
DCAENC_FLAGS
#define DCAENC_FLAGS
Definition: dcaenc.c:1294
dcaenc_class
static const AVClass dcaenc_class
Definition: dcaenc.c:1301
DCAEncContext::scale_factor
int scale_factor[MAX_CHANNELS][DCAENC_SUBBANDS]
Definition: dcaenc.c:93
gammafilter
static double gammafilter(int i, double f)
Definition: dcaenc.c:123
DCAEncContext::channel_order_tab
const int8_t * channel_order_tab
channel reordering table, lfe and non lfe
Definition: dcaenc.c:80
DCA_NUM_BITALLOC_CODES
#define DCA_NUM_BITALLOC_CODES
Definition: dcahuff.h:34