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opus_silk.c
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1 /*
2  * Copyright (c) 2012 Andrew D'Addesio
3  * Copyright (c) 2013-2014 Mozilla Corporation
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * FFmpeg is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
22 /**
23  * @file
24  * Opus SILK decoder
25  */
26 
27 #include <stdint.h>
28 
29 #include "opus.h"
30 #include "opustab.h"
31 
32 typedef struct SilkFrame {
33  int coded;
34  int log_gain;
35  int16_t nlsf[16];
36  float lpc[16];
37 
38  float output [2 * SILK_HISTORY];
41 
43 } SilkFrame;
44 
45 struct SilkContext {
48 
49  int midonly;
50  int subframes;
51  int sflength;
52  int flength;
54 
56  int wb;
57 
60  float stereo_weights[2];
61 
63 };
64 
65 static inline void silk_stabilize_lsf(int16_t nlsf[16], int order, const uint16_t min_delta[17])
66 {
67  int pass, i;
68  for (pass = 0; pass < 20; pass++) {
69  int k, min_diff = 0;
70  for (i = 0; i < order+1; i++) {
71  int low = i != 0 ? nlsf[i-1] : 0;
72  int high = i != order ? nlsf[i] : 32768;
73  int diff = (high - low) - (min_delta[i]);
74 
75  if (diff < min_diff) {
76  min_diff = diff;
77  k = i;
78 
79  if (pass == 20)
80  break;
81  }
82  }
83  if (min_diff == 0) /* no issues; stabilized */
84  return;
85 
86  /* wiggle one or two LSFs */
87  if (k == 0) {
88  /* repel away from lower bound */
89  nlsf[0] = min_delta[0];
90  } else if (k == order) {
91  /* repel away from higher bound */
92  nlsf[order-1] = 32768 - min_delta[order];
93  } else {
94  /* repel away from current position */
95  int min_center = 0, max_center = 32768, center_val;
96 
97  /* lower extent */
98  for (i = 0; i < k; i++)
99  min_center += min_delta[i];
100  min_center += min_delta[k] >> 1;
101 
102  /* upper extent */
103  for (i = order; i > k; i--)
104  max_center -= min_delta[i];
105  max_center -= min_delta[k] >> 1;
106 
107  /* move apart */
108  center_val = nlsf[k - 1] + nlsf[k];
109  center_val = (center_val >> 1) + (center_val & 1); // rounded divide by 2
110  center_val = FFMIN(max_center, FFMAX(min_center, center_val));
111 
112  nlsf[k - 1] = center_val - (min_delta[k] >> 1);
113  nlsf[k] = nlsf[k - 1] + min_delta[k];
114  }
115  }
116 
117  /* resort to the fall-back method, the standard method for LSF stabilization */
118 
119  /* sort; as the LSFs should be nearly sorted, use insertion sort */
120  for (i = 1; i < order; i++) {
121  int j, value = nlsf[i];
122  for (j = i - 1; j >= 0 && nlsf[j] > value; j--)
123  nlsf[j + 1] = nlsf[j];
124  nlsf[j + 1] = value;
125  }
126 
127  /* push forwards to increase distance */
128  if (nlsf[0] < min_delta[0])
129  nlsf[0] = min_delta[0];
130  for (i = 1; i < order; i++)
131  nlsf[i] = FFMAX(nlsf[i], FFMIN(nlsf[i - 1] + min_delta[i], 32767));
132 
133  /* push backwards to increase distance */
134  if (nlsf[order-1] > 32768 - min_delta[order])
135  nlsf[order-1] = 32768 - min_delta[order];
136  for (i = order-2; i >= 0; i--)
137  if (nlsf[i] > nlsf[i + 1] - min_delta[i+1])
138  nlsf[i] = nlsf[i + 1] - min_delta[i+1];
139 
140  return;
141 }
142 
143 static inline int silk_is_lpc_stable(const int16_t lpc[16], int order)
144 {
145  int k, j, DC_resp = 0;
146  int32_t lpc32[2][16]; // Q24
147  int totalinvgain = 1 << 30; // 1.0 in Q30
148  int32_t *row = lpc32[0], *prevrow;
149 
150  /* initialize the first row for the Levinson recursion */
151  for (k = 0; k < order; k++) {
152  DC_resp += lpc[k];
153  row[k] = lpc[k] * 4096;
154  }
155 
156  if (DC_resp >= 4096)
157  return 0;
158 
159  /* check if prediction gain pushes any coefficients too far */
160  for (k = order - 1; 1; k--) {
161  int rc; // Q31; reflection coefficient
162  int gaindiv; // Q30; inverse of the gain (the divisor)
163  int gain; // gain for this reflection coefficient
164  int fbits; // fractional bits used for the gain
165  int error; // Q29; estimate of the error of our partial estimate of 1/gaindiv
166 
167  if (FFABS(row[k]) > 16773022)
168  return 0;
169 
170  rc = -(row[k] * 128);
171  gaindiv = (1 << 30) - MULH(rc, rc);
172 
173  totalinvgain = MULH(totalinvgain, gaindiv) << 2;
174  if (k == 0)
175  return (totalinvgain >= 107374);
176 
177  /* approximate 1.0/gaindiv */
178  fbits = opus_ilog(gaindiv);
179  gain = ((1 << 29) - 1) / (gaindiv >> (fbits + 1 - 16)); // Q<fbits-16>
180  error = (1 << 29) - MULL(gaindiv << (15 + 16 - fbits), gain, 16);
181  gain = ((gain << 16) + (error * gain >> 13));
182 
183  /* switch to the next row of the LPC coefficients */
184  prevrow = row;
185  row = lpc32[k & 1];
186 
187  for (j = 0; j < k; j++) {
188  int x = av_sat_sub32(prevrow[j], ROUND_MULL(prevrow[k - j - 1], rc, 31));
189  int64_t tmp = ROUND_MULL(x, gain, fbits);
190 
191  /* per RFC 8251 section 6, if this calculation overflows, the filter
192  is considered unstable. */
193  if (tmp < INT32_MIN || tmp > INT32_MAX)
194  return 0;
195 
196  row[j] = (int32_t)tmp;
197  }
198  }
199 }
200 
201 static void silk_lsp2poly(const int32_t lsp[16], int32_t pol[16], int half_order)
202 {
203  int i, j;
204 
205  pol[0] = 65536; // 1.0 in Q16
206  pol[1] = -lsp[0];
207 
208  for (i = 1; i < half_order; i++) {
209  pol[i + 1] = pol[i - 1] * 2 - ROUND_MULL(lsp[2 * i], pol[i], 16);
210  for (j = i; j > 1; j--)
211  pol[j] += pol[j - 2] - ROUND_MULL(lsp[2 * i], pol[j - 1], 16);
212 
213  pol[1] -= lsp[2 * i];
214  }
215 }
216 
217 static void silk_lsf2lpc(const int16_t nlsf[16], float lpcf[16], int order)
218 {
219  int i, k;
220  int32_t lsp[16]; // Q17; 2*cos(LSF)
221  int32_t p[9], q[9]; // Q16
222  int32_t lpc32[16]; // Q17
223  int16_t lpc[16]; // Q12
224 
225  /* convert the LSFs to LSPs, i.e. 2*cos(LSF) */
226  for (k = 0; k < order; k++) {
227  int index = nlsf[k] >> 8;
228  int offset = nlsf[k] & 255;
229  int k2 = (order == 10) ? ff_silk_lsf_ordering_nbmb[k] : ff_silk_lsf_ordering_wb[k];
230 
231  /* interpolate and round */
232  lsp[k2] = ff_silk_cosine[index] * 256;
233  lsp[k2] += (ff_silk_cosine[index + 1] - ff_silk_cosine[index]) * offset;
234  lsp[k2] = (lsp[k2] + 4) >> 3;
235  }
236 
237  silk_lsp2poly(lsp , p, order >> 1);
238  silk_lsp2poly(lsp + 1, q, order >> 1);
239 
240  /* reconstruct A(z) */
241  for (k = 0; k < order>>1; k++) {
242  lpc32[k] = -p[k + 1] - p[k] - q[k + 1] + q[k];
243  lpc32[order-k-1] = -p[k + 1] - p[k] + q[k + 1] - q[k];
244  }
245 
246  /* limit the range of the LPC coefficients to each fit within an int16_t */
247  for (i = 0; i < 10; i++) {
248  int j;
249  unsigned int maxabs = 0;
250  for (j = 0, k = 0; j < order; j++) {
251  unsigned int x = FFABS(lpc32[k]);
252  if (x > maxabs) {
253  maxabs = x; // Q17
254  k = j;
255  }
256  }
257 
258  maxabs = (maxabs + 16) >> 5; // convert to Q12
259 
260  if (maxabs > 32767) {
261  /* perform bandwidth expansion */
262  unsigned int chirp, chirp_base; // Q16
263  maxabs = FFMIN(maxabs, 163838); // anything above this overflows chirp's numerator
264  chirp_base = chirp = 65470 - ((maxabs - 32767) << 14) / ((maxabs * (k+1)) >> 2);
265 
266  for (k = 0; k < order; k++) {
267  lpc32[k] = ROUND_MULL(lpc32[k], chirp, 16);
268  chirp = (chirp_base * chirp + 32768) >> 16;
269  }
270  } else break;
271  }
272 
273  if (i == 10) {
274  /* time's up: just clamp */
275  for (k = 0; k < order; k++) {
276  int x = (lpc32[k] + 16) >> 5;
277  lpc[k] = av_clip_int16(x);
278  lpc32[k] = lpc[k] << 5; // shortcut mandated by the spec; drops lower 5 bits
279  }
280  } else {
281  for (k = 0; k < order; k++)
282  lpc[k] = (lpc32[k] + 16) >> 5;
283  }
284 
285  /* if the prediction gain causes the LPC filter to become unstable,
286  apply further bandwidth expansion on the Q17 coefficients */
287  for (i = 1; i <= 16 && !silk_is_lpc_stable(lpc, order); i++) {
288  unsigned int chirp, chirp_base;
289  chirp_base = chirp = 65536 - (1 << i);
290 
291  for (k = 0; k < order; k++) {
292  lpc32[k] = ROUND_MULL(lpc32[k], chirp, 16);
293  lpc[k] = (lpc32[k] + 16) >> 5;
294  chirp = (chirp_base * chirp + 32768) >> 16;
295  }
296  }
297 
298  for (i = 0; i < order; i++)
299  lpcf[i] = lpc[i] / 4096.0f;
300 }
301 
303  OpusRangeCoder *rc,
304  float lpc_leadin[16], float lpc[16],
305  int *lpc_order, int *has_lpc_leadin, int voiced)
306 {
307  int i;
308  int order; // order of the LP polynomial; 10 for NB/MB and 16 for WB
309  int8_t lsf_i1, lsf_i2[16]; // stage-1 and stage-2 codebook indices
310  int16_t lsf_res[16]; // residual as a Q10 value
311  int16_t nlsf[16]; // Q15
312 
313  *lpc_order = order = s->wb ? 16 : 10;
314 
315  /* obtain LSF stage-1 and stage-2 indices */
316  lsf_i1 = ff_opus_rc_dec_cdf(rc, ff_silk_model_lsf_s1[s->wb][voiced]);
317  for (i = 0; i < order; i++) {
318  int index = s->wb ? ff_silk_lsf_s2_model_sel_wb [lsf_i1][i] :
320  lsf_i2[i] = ff_opus_rc_dec_cdf(rc, ff_silk_model_lsf_s2[index]) - 4;
321  if (lsf_i2[i] == -4)
323  else if (lsf_i2[i] == 4)
325  }
326 
327  /* reverse the backwards-prediction step */
328  for (i = order - 1; i >= 0; i--) {
329  int qstep = s->wb ? 9830 : 11796;
330 
331  lsf_res[i] = lsf_i2[i] * 1024;
332  if (lsf_i2[i] < 0) lsf_res[i] += 102;
333  else if (lsf_i2[i] > 0) lsf_res[i] -= 102;
334  lsf_res[i] = (lsf_res[i] * qstep) >> 16;
335 
336  if (i + 1 < order) {
337  int weight = s->wb ? ff_silk_lsf_pred_weights_wb [ff_silk_lsf_weight_sel_wb [lsf_i1][i]][i] :
339  lsf_res[i] += (lsf_res[i+1] * weight) >> 8;
340  }
341  }
342 
343  /* reconstruct the NLSF coefficients from the supplied indices */
344  for (i = 0; i < order; i++) {
345  const uint8_t * codebook = s->wb ? ff_silk_lsf_codebook_wb [lsf_i1] :
347  int cur, prev, next, weight_sq, weight, ipart, fpart, y, value;
348 
349  /* find the weight of the residual */
350  /* TODO: precompute */
351  cur = codebook[i];
352  prev = i ? codebook[i - 1] : 0;
353  next = i + 1 < order ? codebook[i + 1] : 256;
354  weight_sq = (1024 / (cur - prev) + 1024 / (next - cur)) << 16;
355 
356  /* approximate square-root with mandated fixed-point arithmetic */
357  ipart = opus_ilog(weight_sq);
358  fpart = (weight_sq >> (ipart-8)) & 127;
359  y = ((ipart & 1) ? 32768 : 46214) >> ((32 - ipart)>>1);
360  weight = y + ((213 * fpart * y) >> 16);
361 
362  value = cur * 128 + (lsf_res[i] * 16384) / weight;
363  nlsf[i] = av_clip_uintp2(value, 15);
364  }
365 
366  /* stabilize the NLSF coefficients */
369 
370  /* produce an interpolation for the first 2 subframes, */
371  /* and then convert both sets of NLSFs to LPC coefficients */
372  *has_lpc_leadin = 0;
373  if (s->subframes == 4) {
375  if (offset != 4 && frame->coded) {
376  *has_lpc_leadin = 1;
377  if (offset != 0) {
378  int16_t nlsf_leadin[16];
379  for (i = 0; i < order; i++)
380  nlsf_leadin[i] = frame->nlsf[i] +
381  ((nlsf[i] - frame->nlsf[i]) * offset >> 2);
382  silk_lsf2lpc(nlsf_leadin, lpc_leadin, order);
383  } else /* avoid re-computation for a (roughly) 1-in-4 occurrence */
384  memcpy(lpc_leadin, frame->lpc, 16 * sizeof(float));
385  } else
386  offset = 4;
388 
389  silk_lsf2lpc(nlsf, lpc, order);
390  } else {
391  s->nlsf_interp_factor = 4;
392  silk_lsf2lpc(nlsf, lpc, order);
393  }
394 
395  memcpy(frame->nlsf, nlsf, order * sizeof(nlsf[0]));
396  memcpy(frame->lpc, lpc, order * sizeof(lpc[0]));
397 }
398 
399 static inline void silk_count_children(OpusRangeCoder *rc, int model, int32_t total,
400  int32_t child[2])
401 {
402  if (total != 0) {
403  child[0] = ff_opus_rc_dec_cdf(rc,
404  ff_silk_model_pulse_location[model] + (((total - 1 + 5) * (total - 1)) >> 1));
405  child[1] = total - child[0];
406  } else {
407  child[0] = 0;
408  child[1] = 0;
409  }
410 }
411 
413  float* excitationf,
414  int qoffset_high, int active, int voiced)
415 {
416  int i;
417  uint32_t seed;
418  int shellblocks;
419  int ratelevel;
420  uint8_t pulsecount[20]; // total pulses in each shell block
421  uint8_t lsbcount[20] = {0}; // raw lsbits defined for each pulse in each shell block
422  int32_t excitation[320]; // Q23
423 
424  /* excitation parameters */
426  shellblocks = ff_silk_shell_blocks[s->bandwidth][s->subframes >> 2];
427  ratelevel = ff_opus_rc_dec_cdf(rc, ff_silk_model_exc_rate[voiced]);
428 
429  for (i = 0; i < shellblocks; i++) {
430  pulsecount[i] = ff_opus_rc_dec_cdf(rc, ff_silk_model_pulse_count[ratelevel]);
431  if (pulsecount[i] == 17) {
432  while (pulsecount[i] == 17 && ++lsbcount[i] != 10)
433  pulsecount[i] = ff_opus_rc_dec_cdf(rc, ff_silk_model_pulse_count[9]);
434  if (lsbcount[i] == 10)
435  pulsecount[i] = ff_opus_rc_dec_cdf(rc, ff_silk_model_pulse_count[10]);
436  }
437  }
438 
439  /* decode pulse locations using PVQ */
440  for (i = 0; i < shellblocks; i++) {
441  if (pulsecount[i] != 0) {
442  int a, b, c, d;
443  int32_t * location = excitation + 16*i;
444  int32_t branch[4][2];
445  branch[0][0] = pulsecount[i];
446 
447  /* unrolled tail recursion */
448  for (a = 0; a < 1; a++) {
449  silk_count_children(rc, 0, branch[0][a], branch[1]);
450  for (b = 0; b < 2; b++) {
451  silk_count_children(rc, 1, branch[1][b], branch[2]);
452  for (c = 0; c < 2; c++) {
453  silk_count_children(rc, 2, branch[2][c], branch[3]);
454  for (d = 0; d < 2; d++) {
455  silk_count_children(rc, 3, branch[3][d], location);
456  location += 2;
457  }
458  }
459  }
460  }
461  } else
462  memset(excitation + 16*i, 0, 16*sizeof(int32_t));
463  }
464 
465  /* decode least significant bits */
466  for (i = 0; i < shellblocks << 4; i++) {
467  int bit;
468  for (bit = 0; bit < lsbcount[i >> 4]; bit++)
469  excitation[i] = (excitation[i] << 1) |
471  }
472 
473  /* decode signs */
474  for (i = 0; i < shellblocks << 4; i++) {
475  if (excitation[i] != 0) {
476  int sign = ff_opus_rc_dec_cdf(rc, ff_silk_model_excitation_sign[active +
477  voiced][qoffset_high][FFMIN(pulsecount[i >> 4], 6)]);
478  if (sign == 0)
479  excitation[i] *= -1;
480  }
481  }
482 
483  /* assemble the excitation */
484  for (i = 0; i < shellblocks << 4; i++) {
485  int value = excitation[i];
486  excitation[i] = value * 256 | ff_silk_quant_offset[voiced][qoffset_high];
487  if (value < 0) excitation[i] += 20;
488  else if (value > 0) excitation[i] -= 20;
489 
490  /* invert samples pseudorandomly */
491  seed = 196314165 * seed + 907633515;
492  if (seed & 0x80000000)
493  excitation[i] *= -1;
494  seed += value;
495 
496  excitationf[i] = excitation[i] / 8388608.0f;
497  }
498 }
499 
500 /** Maximum residual history according to 4.2.7.6.1 */
501 #define SILK_MAX_LAG (288 + LTP_ORDER / 2)
502 
503 /** Order of the LTP filter */
504 #define LTP_ORDER 5
505 
507  int frame_num, int channel, int coded_channels, int active, int active1)
508 {
509  /* per frame */
510  int voiced; // combines with active to indicate inactive, active, or active+voiced
511  int qoffset_high;
512  int order; // order of the LPC coefficients
513  float lpc_leadin[16], lpc_body[16], residual[SILK_MAX_LAG + SILK_HISTORY];
514  int has_lpc_leadin;
515  float ltpscale;
516 
517  /* per subframe */
518  struct {
519  float gain;
520  int pitchlag;
521  float ltptaps[5];
522  } sf[4];
523 
524  SilkFrame * const frame = s->frame + channel;
525 
526  int i;
527 
528  /* obtain stereo weights */
529  if (coded_channels == 2 && channel == 0) {
530  int n, wi[2], ws[2], w[2];
532  wi[0] = ff_opus_rc_dec_cdf(rc, ff_silk_model_stereo_s2) + 3 * (n / 5);
534  wi[1] = ff_opus_rc_dec_cdf(rc, ff_silk_model_stereo_s2) + 3 * (n % 5);
536 
537  for (i = 0; i < 2; i++)
538  w[i] = ff_silk_stereo_weights[wi[i]] +
539  (((ff_silk_stereo_weights[wi[i] + 1] - ff_silk_stereo_weights[wi[i]]) * 6554) >> 16)
540  * (ws[i]*2 + 1);
541 
542  s->stereo_weights[0] = (w[0] - w[1]) / 8192.0;
543  s->stereo_weights[1] = w[1] / 8192.0;
544 
545  /* and read the mid-only flag */
546  s->midonly = active1 ? 0 : ff_opus_rc_dec_cdf(rc, ff_silk_model_mid_only);
547  }
548 
549  /* obtain frame type */
550  if (!active) {
552  voiced = 0;
553  } else {
555  qoffset_high = type & 1;
556  voiced = type >> 1;
557  }
558 
559  /* obtain subframe quantization gains */
560  for (i = 0; i < s->subframes; i++) {
561  int log_gain; //Q7
562  int ipart, fpart, lingain;
563 
564  if (i == 0 && (frame_num == 0 || !frame->coded)) {
565  /* gain is coded absolute */
566  int x = ff_opus_rc_dec_cdf(rc, ff_silk_model_gain_highbits[active + voiced]);
567  log_gain = (x<<3) | ff_opus_rc_dec_cdf(rc, ff_silk_model_gain_lowbits);
568 
569  if (frame->coded)
570  log_gain = FFMAX(log_gain, frame->log_gain - 16);
571  } else {
572  /* gain is coded relative */
573  int delta_gain = ff_opus_rc_dec_cdf(rc, ff_silk_model_gain_delta);
574  log_gain = av_clip_uintp2(FFMAX((delta_gain<<1) - 16,
575  frame->log_gain + delta_gain - 4), 6);
576  }
577 
578  frame->log_gain = log_gain;
579 
580  /* approximate 2**(x/128) with a Q7 (i.e. non-integer) input */
581  log_gain = (log_gain * 0x1D1C71 >> 16) + 2090;
582  ipart = log_gain >> 7;
583  fpart = log_gain & 127;
584  lingain = (1 << ipart) + ((-174 * fpart * (128-fpart) >>16) + fpart) * ((1<<ipart) >> 7);
585  sf[i].gain = lingain / 65536.0f;
586  }
587 
588  /* obtain LPC filter coefficients */
589  silk_decode_lpc(s, frame, rc, lpc_leadin, lpc_body, &order, &has_lpc_leadin, voiced);
590 
591  /* obtain pitch lags, if this is a voiced frame */
592  if (voiced) {
593  int lag_absolute = (!frame_num || !frame->prev_voiced);
594  int primarylag; // primary pitch lag for the entire SILK frame
595  int ltpfilter;
596  const int8_t * offsets;
597 
598  if (!lag_absolute) {
600  if (delta)
601  primarylag = frame->primarylag + delta - 9;
602  else
603  lag_absolute = 1;
604  }
605 
606  if (lag_absolute) {
607  /* primary lag is coded absolute */
608  int highbits, lowbits;
609  static const uint16_t * const model[] = {
612  };
614  lowbits = ff_opus_rc_dec_cdf(rc, model[s->bandwidth]);
615 
616  primarylag = ff_silk_pitch_min_lag[s->bandwidth] +
617  highbits*ff_silk_pitch_scale[s->bandwidth] + lowbits;
618  }
619  frame->primarylag = primarylag;
620 
621  if (s->subframes == 2)
622  offsets = (s->bandwidth == OPUS_BANDWIDTH_NARROWBAND)
627  else
628  offsets = (s->bandwidth == OPUS_BANDWIDTH_NARROWBAND)
633 
634  for (i = 0; i < s->subframes; i++)
635  sf[i].pitchlag = av_clip(primarylag + offsets[i],
638 
639  /* obtain LTP filter coefficients */
641  for (i = 0; i < s->subframes; i++) {
642  int index, j;
643  static const uint16_t * const filter_sel[] = {
646  };
647  static const int8_t (* const filter_taps[])[5] = {
649  };
650  index = ff_opus_rc_dec_cdf(rc, filter_sel[ltpfilter]);
651  for (j = 0; j < 5; j++)
652  sf[i].ltptaps[j] = filter_taps[ltpfilter][index][j] / 128.0f;
653  }
654  }
655 
656  /* obtain LTP scale factor */
657  if (voiced && frame_num == 0)
659  ff_silk_model_ltp_scale_index)] / 16384.0f;
660  else ltpscale = 15565.0f/16384.0f;
661 
662  /* generate the excitation signal for the entire frame */
663  silk_decode_excitation(s, rc, residual + SILK_MAX_LAG, qoffset_high,
664  active, voiced);
665 
666  /* skip synthesising the side channel if we want mono-only */
667  if (s->output_channels == channel)
668  return;
669 
670  /* generate the output signal */
671  for (i = 0; i < s->subframes; i++) {
672  const float * lpc_coeff = (i < 2 && has_lpc_leadin) ? lpc_leadin : lpc_body;
673  float *dst = frame->output + SILK_HISTORY + i * s->sflength;
674  float *resptr = residual + SILK_MAX_LAG + i * s->sflength;
675  float *lpc = frame->lpc_history + SILK_HISTORY + i * s->sflength;
676  float sum;
677  int j, k;
678 
679  if (voiced) {
680  int out_end;
681  float scale;
682 
683  if (i < 2 || s->nlsf_interp_factor == 4) {
684  out_end = -i * s->sflength;
685  scale = ltpscale;
686  } else {
687  out_end = -(i - 2) * s->sflength;
688  scale = 1.0f;
689  }
690 
691  /* when the LPC coefficients change, a re-whitening filter is used */
692  /* to produce a residual that accounts for the change */
693  for (j = - sf[i].pitchlag - LTP_ORDER/2; j < out_end; j++) {
694  sum = dst[j];
695  for (k = 0; k < order; k++)
696  sum -= lpc_coeff[k] * dst[j - k - 1];
697  resptr[j] = av_clipf(sum, -1.0f, 1.0f) * scale / sf[i].gain;
698  }
699 
700  if (out_end) {
701  float rescale = sf[i-1].gain / sf[i].gain;
702  for (j = out_end; j < 0; j++)
703  resptr[j] *= rescale;
704  }
705 
706  /* LTP synthesis */
707  for (j = 0; j < s->sflength; j++) {
708  sum = resptr[j];
709  for (k = 0; k < LTP_ORDER; k++)
710  sum += sf[i].ltptaps[k] * resptr[j - sf[i].pitchlag + LTP_ORDER/2 - k];
711  resptr[j] = sum;
712  }
713  }
714 
715  /* LPC synthesis */
716  for (j = 0; j < s->sflength; j++) {
717  sum = resptr[j] * sf[i].gain;
718  for (k = 1; k <= order; k++)
719  sum += lpc_coeff[k - 1] * lpc[j - k];
720 
721  lpc[j] = sum;
722  dst[j] = av_clipf(sum, -1.0f, 1.0f);
723  }
724  }
725 
726  frame->prev_voiced = voiced;
727  memmove(frame->lpc_history, frame->lpc_history + s->flength, SILK_HISTORY * sizeof(float));
728  memmove(frame->output, frame->output + s->flength, SILK_HISTORY * sizeof(float));
729 
730  frame->coded = 1;
731 }
732 
733 static void silk_unmix_ms(SilkContext *s, float *l, float *r)
734 {
735  float *mid = s->frame[0].output + SILK_HISTORY - s->flength;
736  float *side = s->frame[1].output + SILK_HISTORY - s->flength;
737  float w0_prev = s->prev_stereo_weights[0];
738  float w1_prev = s->prev_stereo_weights[1];
739  float w0 = s->stereo_weights[0];
740  float w1 = s->stereo_weights[1];
742  int i;
743 
744  for (i = 0; i < n1; i++) {
745  float interp0 = w0_prev + i * (w0 - w0_prev) / n1;
746  float interp1 = w1_prev + i * (w1 - w1_prev) / n1;
747  float p0 = 0.25 * (mid[i - 2] + 2 * mid[i - 1] + mid[i]);
748 
749  l[i] = av_clipf((1 + interp1) * mid[i - 1] + side[i - 1] + interp0 * p0, -1.0, 1.0);
750  r[i] = av_clipf((1 - interp1) * mid[i - 1] - side[i - 1] - interp0 * p0, -1.0, 1.0);
751  }
752 
753  for (; i < s->flength; i++) {
754  float p0 = 0.25 * (mid[i - 2] + 2 * mid[i - 1] + mid[i]);
755 
756  l[i] = av_clipf((1 + w1) * mid[i - 1] + side[i - 1] + w0 * p0, -1.0, 1.0);
757  r[i] = av_clipf((1 - w1) * mid[i - 1] - side[i - 1] - w0 * p0, -1.0, 1.0);
758  }
759 
760  memcpy(s->prev_stereo_weights, s->stereo_weights, sizeof(s->stereo_weights));
761 }
762 
764 {
765  if (!frame->coded)
766  return;
767 
768  memset(frame->output, 0, sizeof(frame->output));
769  memset(frame->lpc_history, 0, sizeof(frame->lpc_history));
770 
771  memset(frame->lpc, 0, sizeof(frame->lpc));
772  memset(frame->nlsf, 0, sizeof(frame->nlsf));
773 
774  frame->log_gain = 0;
775 
776  frame->primarylag = 0;
777  frame->prev_voiced = 0;
778  frame->coded = 0;
779 }
780 
782  float *output[2],
783  enum OpusBandwidth bandwidth,
784  int coded_channels,
785  int duration_ms)
786 {
787  int active[2][6], redundancy[2];
788  int nb_frames, i, j;
789 
790  if (bandwidth > OPUS_BANDWIDTH_WIDEBAND ||
791  coded_channels > 2 || duration_ms > 60) {
792  av_log(s->avctx, AV_LOG_ERROR, "Invalid parameters passed "
793  "to the SILK decoder.\n");
794  return AVERROR(EINVAL);
795  }
796 
797  nb_frames = 1 + (duration_ms > 20) + (duration_ms > 40);
798  s->subframes = duration_ms / nb_frames / 5; // 5ms subframes
799  s->sflength = 20 * (bandwidth + 2);
800  s->flength = s->sflength * s->subframes;
801  s->bandwidth = bandwidth;
802  s->wb = bandwidth == OPUS_BANDWIDTH_WIDEBAND;
803 
804  /* make sure to flush the side channel when switching from mono to stereo */
805  if (coded_channels > s->prev_coded_channels)
806  silk_flush_frame(&s->frame[1]);
807  s->prev_coded_channels = coded_channels;
808 
809  /* read the LP-layer header bits */
810  for (i = 0; i < coded_channels; i++) {
811  for (j = 0; j < nb_frames; j++)
812  active[i][j] = ff_opus_rc_dec_log(rc, 1);
813 
814  redundancy[i] = ff_opus_rc_dec_log(rc, 1);
815  if (redundancy[i]) {
816  avpriv_report_missing_feature(s->avctx, "LBRR frames");
817  return AVERROR_PATCHWELCOME;
818  }
819  }
820 
821  for (i = 0; i < nb_frames; i++) {
822  for (j = 0; j < coded_channels && !s->midonly; j++)
823  silk_decode_frame(s, rc, i, j, coded_channels, active[j][i], active[1][i]);
824 
825  /* reset the side channel if it is not coded */
826  if (s->midonly && s->frame[1].coded)
827  silk_flush_frame(&s->frame[1]);
828 
829  if (coded_channels == 1 || s->output_channels == 1) {
830  for (j = 0; j < s->output_channels; j++) {
831  memcpy(output[j] + i * s->flength,
832  s->frame[0].output + SILK_HISTORY - s->flength - 2,
833  s->flength * sizeof(float));
834  }
835  } else {
836  silk_unmix_ms(s, output[0] + i * s->flength, output[1] + i * s->flength);
837  }
838 
839  s->midonly = 0;
840  }
841 
842  return nb_frames * s->flength;
843 }
844 
846 {
847  av_freep(ps);
848 }
849 
851 {
852  silk_flush_frame(&s->frame[0]);
853  silk_flush_frame(&s->frame[1]);
854 
855  memset(s->prev_stereo_weights, 0, sizeof(s->prev_stereo_weights));
856 }
857 
858 int ff_silk_init(AVCodecContext *avctx, SilkContext **ps, int output_channels)
859 {
860  SilkContext *s;
861 
862  if (output_channels != 1 && output_channels != 2) {
863  av_log(avctx, AV_LOG_ERROR, "Invalid number of output channels: %d\n",
864  output_channels);
865  return AVERROR(EINVAL);
866  }
867 
868  s = av_mallocz(sizeof(*s));
869  if (!s)
870  return AVERROR(ENOMEM);
871 
872  s->avctx = avctx;
873  s->output_channels = output_channels;
874 
875  ff_silk_flush(s);
876 
877  *ps = s;
878 
879  return 0;
880 }
static void silk_lsp2poly(const int32_t lsp[16], int32_t pol[16], int half_order)
Definition: opus_silk.c:201
int output_channels
Definition: opus_silk.c:47
const char * s
Definition: avisynth_c.h:768
static void silk_decode_frame(SilkContext *s, OpusRangeCoder *rc, int frame_num, int channel, int coded_channels, int active, int active1)
Definition: opus_silk.c:506
int nlsf_interp_factor
Definition: opus_silk.c:53
const int8_t ff_silk_ltp_filter1_taps[16][5]
Definition: opustab.c:682
const uint8_t ff_silk_shell_blocks[3][2]
Definition: opustab.c:738
const uint16_t ff_silk_model_ltp_scale_index[]
Definition: opustab.c:150
const uint8_t ff_silk_lsf_codebook_wb[32][16]
Definition: opustab.c:506
const uint16_t ff_silk_pitch_max_lag[]
Definition: opustab.c:597
const uint16_t ff_silk_model_exc_rate[2][10]
Definition: opustab.c:154
const uint8_t ff_silk_lsf_pred_weights_nbmb[2][9]
Definition: opustab.c:391
int ff_silk_init(AVCodecContext *avctx, SilkContext **ps, int output_channels)
Definition: opus_silk.c:858
const uint16_t ff_silk_model_pitch_contour_mbwb10ms[]
Definition: opustab.c:125
const uint16_t ff_silk_model_frame_type_active[]
Definition: opustab.c:42
const char * b
Definition: vf_curves.c:113
int sflength
Definition: opus_silk.c:51
const uint16_t ff_silk_model_mid_only[]
Definition: opustab.c:38
uint32_t ff_opus_rc_dec_log(OpusRangeCoder *rc, uint32_t bits)
Definition: opus_rc.c:114
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
Definition: mem.c:236
const uint16_t ff_silk_model_pitch_lowbits_wb[]
Definition: opustab.c:112
const uint16_t ff_silk_model_ltp_filter1_sel[]
Definition: opustab.c:141
const int16_t ff_silk_stereo_weights[]
Definition: opustab.c:316
#define SILK_HISTORY
Definition: opus.h:47
int subframes
Definition: opus_silk.c:50
static void silk_count_children(OpusRangeCoder *rc, int model, int32_t total, int32_t child[2])
Definition: opus_silk.c:399
const uint16_t ff_silk_model_lsf_interpolation_offset[]
Definition: opustab.c:101
float output[2 *SILK_HISTORY]
Definition: opus_silk.c:38
const uint16_t ff_silk_model_lsf_s2[32][10]
Definition: opustab.c:77
#define opus_ilog(i)
Definition: opus_rc.h:31
const uint16_t ff_silk_model_pulse_location[4][168]
Definition: opustab.c:184
const int8_t ff_silk_pitch_offset_mbwb20ms[34][4]
Definition: opustab.c:634
const uint16_t ff_silk_model_gain_delta[]
Definition: opustab.c:52
uint8_t
float delta
const uint8_t ff_silk_lsf_ordering_nbmb[]
Definition: opustab.c:549
AVS_FilterInfo AVS_Value child
Definition: avisynth_c.h:731
int log_gain
Definition: opus_silk.c:34
static AVFrame * frame
const int8_t ff_silk_pitch_offset_nb20ms[11][4]
Definition: opustab.c:605
int16_t nlsf[16]
Definition: opus_silk.c:35
const uint16_t ff_silk_model_excitation_lsb[]
Definition: opustab.c:256
#define av_log(a,...)
static void silk_stabilize_lsf(int16_t nlsf[16], int order, const uint16_t min_delta[17])
Definition: opus_silk.c:65
const uint16_t ff_silk_model_lsf_s1[2][2][33]
Definition: opustab.c:57
const uint8_t ff_silk_lsf_s2_model_sel_nbmb[32][10]
Definition: opustab.c:321
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
const uint16_t ff_silk_model_pitch_contour_nb10ms[]
Definition: opustab.c:119
const uint16_t ff_silk_model_pitch_delta[]
Definition: opustab.c:114
#define AVERROR(e)
Definition: error.h:43
const uint8_t ff_silk_lsf_codebook_nbmb[32][10]
Definition: opustab.c:471
const uint16_t ff_silk_model_gain_lowbits[]
Definition: opustab.c:50
const char * r
Definition: vf_curves.c:111
const uint8_t ff_silk_lsf_pred_weights_wb[2][15]
Definition: opustab.c:396
const uint8_t ff_silk_lsf_weight_sel_wb[32][15]
Definition: opustab.c:436
const int8_t ff_silk_pitch_offset_mbwb10ms[12][2]
Definition: opustab.c:619
static const uint8_t offset[127][2]
Definition: vf_spp.c:92
#define FFMAX(a, b)
Definition: common.h:94
static void silk_unmix_ms(SilkContext *s, float *l, float *r)
Definition: opus_silk.c:733
const uint16_t ff_silk_model_ltp_filter2_sel[]
Definition: opustab.c:145
const uint8_t ff_silk_lsf_s2_model_sel_wb[32][16]
Definition: opustab.c:356
const int8_t ff_silk_pitch_offset_nb10ms[3][2]
Definition: opustab.c:599
uint32_t ff_opus_rc_dec_cdf(OpusRangeCoder *rc, const uint16_t *cdf)
Definition: opus_rc.c:90
const uint16_t ff_silk_model_excitation_sign[3][2][7][3]
Definition: opustab.c:258
#define pass
Definition: fft_template.c:593
const uint8_t ff_silk_quant_offset[2][2]
Definition: opustab.c:744
static void silk_decode_excitation(SilkContext *s, OpusRangeCoder *rc, float *excitationf, int qoffset_high, int active, int voiced)
Definition: opus_silk.c:412
uint64_t residual
Definition: dirac_vlc.h:29
const uint16_t ff_silk_lsf_min_spacing_nbmb[]
Definition: opustab.c:541
const uint16_t ff_silk_model_pitch_contour_mbwb20ms[]
Definition: opustab.c:129
static void rescale(GDVContext *gdv, uint8_t *dst, int w, int h, int scale_v, int scale_h)
Definition: gdv.c:75
const uint16_t ff_silk_model_stereo_s3[]
Definition: opustab.c:36
float lpc[16]
Definition: opus_silk.c:36
const uint16_t ff_silk_model_ltp_filter0_sel[]
Definition: opustab.c:137
#define FFMIN(a, b)
Definition: common.h:96
const uint16_t ff_silk_model_stereo_s1[]
Definition: opustab.c:29
enum OpusBandwidth bandwidth
Definition: opus_silk.c:55
int midonly
Definition: opus_silk.c:49
GLsizei GLboolean const GLfloat * value
Definition: opengl_enc.c:109
const uint16_t ff_silk_model_pitch_highbits[]
Definition: opustab.c:103
int32_t
const int8_t ff_silk_ltp_filter0_taps[8][5]
Definition: opustab.c:671
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
Definition: common.h:72
int coded
Definition: opus_silk.c:33
const uint16_t ff_silk_model_pulse_count[11][19]
Definition: opustab.c:159
int n
Definition: avisynth_c.h:684
const uint16_t ff_silk_pitch_min_lag[]
Definition: opustab.c:595
int ff_silk_decode_superframe(SilkContext *s, OpusRangeCoder *rc, float *output[2], enum OpusBandwidth bandwidth, int coded_channels, int duration_ms)
Decode the LP layer of one Opus frame (which may correspond to several SILK frames).
Definition: opus_silk.c:781
static void error(const char *err)
const uint16_t ff_silk_model_pitch_lowbits_nb[]
Definition: opustab.c:108
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Definition: error.h:62
const uint8_t ff_silk_lsf_weight_sel_nbmb[32][9]
Definition: opustab.c:401
#define SILK_MAX_LAG
Maximum residual history according to 4.2.7.6.1.
Definition: opus_silk.c:501
int flength
Definition: opus_silk.c:52
const uint16_t ff_silk_model_frame_type_inactive[]
Definition: opustab.c:40
main external API structure.
Definition: avcodec.h:1488
static unsigned int seed
Definition: videogen.c:78
int primarylag
Definition: opus_silk.c:40
const int8_t ff_silk_ltp_filter2_taps[32][5]
Definition: opustab.c:701
GLint GLenum type
Definition: opengl_enc.c:105
float stereo_weights[2]
Definition: opus_silk.c:60
#define LTP_ORDER
Order of the LTP filter.
Definition: opus_silk.c:504
static void silk_flush_frame(SilkFrame *frame)
Definition: opus_silk.c:763
int index
Definition: gxfenc.c:89
const uint16_t ff_silk_lsf_min_spacing_wb[]
Definition: opustab.c:545
const int ff_silk_stereo_interp_len[3]
Definition: opustab.c:749
const uint8_t ff_silk_lsf_ordering_wb[]
Definition: opustab.c:553
int prev_coded_channels
Definition: opus_silk.c:62
const uint16_t ff_silk_model_pitch_contour_nb20ms[]
Definition: opustab.c:121
static void silk_decode_lpc(SilkContext *s, SilkFrame *frame, OpusRangeCoder *rc, float lpc_leadin[16], float lpc[16], int *lpc_order, int *has_lpc_leadin, int voiced)
Definition: opus_silk.c:302
AVCodecContext * avctx
Definition: opus_silk.c:46
#define ROUND_MULL(a, b, s)
Definition: opus.h:50
static int weight(int i, int blen, int offset)
Definition: diracdec.c:1523
void avpriv_report_missing_feature(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
const uint16_t ff_silk_model_ltp_filter[]
Definition: opustab.c:135
const int16_t ff_silk_cosine[]
Definition: opustab.c:557
OpusBandwidth
Definition: opus.h:70
static int silk_is_lpc_stable(const int16_t lpc[16], int order)
Definition: opus_silk.c:143
static double c[64]
channel
Use these values when setting the channel map with ebur128_set_channel().
Definition: ebur128.h:39
float prev_stereo_weights[2]
Definition: opus_silk.c:59
SilkFrame frame[2]
Definition: opus_silk.c:58
void ff_silk_free(SilkContext **ps)
Definition: opus_silk.c:845
const uint16_t ff_silk_pitch_scale[]
Definition: opustab.c:593
static av_always_inline int diff(const uint32_t a, const uint32_t b)
const uint16_t ff_silk_model_gain_highbits[3][9]
Definition: opustab.c:44
void ff_silk_flush(SilkContext *s)
Definition: opus_silk.c:850
int prev_voiced
Definition: opus_silk.c:42
const uint16_t ff_silk_model_pitch_lowbits_mb[]
Definition: opustab.c:110
#define MULL(a, b, s)
Definition: mathops.h:58
const uint16_t ff_silk_ltp_scale_factor[]
Definition: opustab.c:736
#define av_freep(p)
static void silk_lsf2lpc(const int16_t nlsf[16], float lpcf[16], int order)
Definition: opus_silk.c:217
float lpc_history[2 *SILK_HISTORY]
Definition: opus_silk.c:39
const uint16_t ff_silk_model_stereo_s2[]
Definition: opustab.c:34
const uint16_t ff_silk_model_lsf_s2_ext[]
Definition: opustab.c:99
#define MULH
Definition: mathops.h:42
static uint8_t tmp[11]
Definition: aes_ctr.c:26
const uint16_t ff_silk_model_lcg_seed[]
Definition: opustab.c:152