FFmpeg  4.4.6
g723_1dec.c
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1 /*
2  * G.723.1 compatible decoder
3  * Copyright (c) 2006 Benjamin Larsson
4  * Copyright (c) 2010 Mohamed Naufal Basheer
5  *
6  * This file is part of FFmpeg.
7  *
8  * FFmpeg is free software; you can redistribute it and/or
9  * modify it under the terms of the GNU Lesser General Public
10  * License as published by the Free Software Foundation; either
11  * version 2.1 of the License, or (at your option) any later version.
12  *
13  * FFmpeg is distributed in the hope that it will be useful,
14  * but WITHOUT ANY WARRANTY; without even the implied warranty of
15  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
16  * Lesser General Public License for more details.
17  *
18  * You should have received a copy of the GNU Lesser General Public
19  * License along with FFmpeg; if not, write to the Free Software
20  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21  */
22 
23 /**
24  * @file
25  * G.723.1 compatible decoder
26  */
27 
29 #include "libavutil/mem.h"
30 #include "libavutil/opt.h"
31 
32 #define BITSTREAM_READER_LE
33 #include "acelp_vectors.h"
34 #include "avcodec.h"
35 #include "celp_filters.h"
36 #include "celp_math.h"
37 #include "get_bits.h"
38 #include "internal.h"
39 #include "g723_1.h"
40 
41 #define CNG_RANDOM_SEED 12345
42 
43 /**
44  * Postfilter gain weighting factors scaled by 2^15
45  */
46 static const int16_t ppf_gain_weight[2] = {0x1800, 0x2000};
47 
48 static const int16_t pitch_contrib[340] = {
49  60, 0, 0, 2489, 60, 0, 0, 5217,
50  1, 6171, 0, 3953, 0, 10364, 1, 9357,
51  -1, 8843, 1, 9396, 0, 5794, -1, 10816,
52  2, 11606, -2, 12072, 0, 8616, 1, 12170,
53  0, 14440, 0, 7787, -1, 13721, 0, 18205,
54  0, 14471, 0, 15807, 1, 15275, 0, 13480,
55  -1, 18375, -1, 0, 1, 11194, -1, 13010,
56  1, 18836, -2, 20354, 1, 16233, -1, 0,
57  60, 0, 0, 12130, 0, 13385, 1, 17834,
58  1, 20875, 0, 21996, 1, 0, 1, 18277,
59  -1, 21321, 1, 13738, -1, 19094, -1, 20387,
60  -1, 0, 0, 21008, 60, 0, -2, 22807,
61  0, 15900, 1, 0, 0, 17989, -1, 22259,
62  1, 24395, 1, 23138, 0, 23948, 1, 22997,
63  2, 22604, -1, 25942, 0, 26246, 1, 25321,
64  0, 26423, 0, 24061, 0, 27247, 60, 0,
65  -1, 25572, 1, 23918, 1, 25930, 2, 26408,
66  -1, 19049, 1, 27357, -1, 24538, 60, 0,
67  -1, 25093, 0, 28549, 1, 0, 0, 22793,
68  -1, 25659, 0, 29377, 0, 30276, 0, 26198,
69  1, 22521, -1, 28919, 0, 27384, 1, 30162,
70  -1, 0, 0, 24237, -1, 30062, 0, 21763,
71  1, 30917, 60, 0, 0, 31284, 0, 29433,
72  1, 26821, 1, 28655, 0, 31327, 2, 30799,
73  1, 31389, 0, 32322, 1, 31760, -2, 31830,
74  0, 26936, -1, 31180, 1, 30875, 0, 27873,
75  -1, 30429, 1, 31050, 0, 0, 0, 31912,
76  1, 31611, 0, 31565, 0, 25557, 0, 31357,
77  60, 0, 1, 29536, 1, 28985, -1, 26984,
78  -1, 31587, 2, 30836, -2, 31133, 0, 30243,
79  -1, 30742, -1, 32090, 60, 0, 2, 30902,
80  60, 0, 0, 30027, 0, 29042, 60, 0,
81  0, 31756, 0, 24553, 0, 25636, -2, 30501,
82  60, 0, -1, 29617, 0, 30649, 60, 0,
83  0, 29274, 2, 30415, 0, 27480, 0, 31213,
84  -1, 28147, 0, 30600, 1, 31652, 2, 29068,
85  60, 0, 1, 28571, 1, 28730, 1, 31422,
86  0, 28257, 0, 24797, 60, 0, 0, 0,
87  60, 0, 0, 22105, 0, 27852, 60, 0,
88  60, 0, -1, 24214, 0, 24642, 0, 23305,
89  60, 0, 60, 0, 1, 22883, 0, 21601,
90  60, 0, 2, 25650, 60, 0, -2, 31253,
91  -2, 25144, 0, 17998
92 };
93 
94 /**
95  * Size of the MP-MLQ fixed excitation codebooks
96  */
97 static const int32_t max_pos[4] = {593775, 142506, 593775, 142506};
98 
99 /**
100  * 0.65^i (Zero part) and 0.75^i (Pole part) scaled by 2^15
101  */
102 static const int16_t postfilter_tbl[2][LPC_ORDER] = {
103  /* Zero */
104  {21299, 13844, 8999, 5849, 3802, 2471, 1606, 1044, 679, 441},
105  /* Pole */
106  {24576, 18432, 13824, 10368, 7776, 5832, 4374, 3281, 2460, 1845}
107 };
108 
109 static const int cng_adaptive_cb_lag[4] = { 1, 0, 1, 3 };
110 
111 static const int cng_filt[4] = { 273, 998, 499, 333 };
112 
113 static const int cng_bseg[3] = { 2048, 18432, 231233 };
114 
116 {
117  G723_1_Context *s = avctx->priv_data;
118 
120  if (avctx->channels < 1 || avctx->channels > 2) {
121  av_log(avctx, AV_LOG_ERROR, "Only mono and stereo are supported (requested channels: %d).\n", avctx->channels);
122  return AVERROR(EINVAL);
123  }
125  for (int ch = 0; ch < avctx->channels; ch++) {
126  G723_1_ChannelContext *p = &s->ch[ch];
127 
128  p->pf_gain = 1 << 12;
129 
130  memcpy(p->prev_lsp, dc_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
131  memcpy(p->sid_lsp, dc_lsp, LPC_ORDER * sizeof(*p->sid_lsp));
132 
135  }
136 
137  return 0;
138 }
139 
140 /**
141  * Unpack the frame into parameters.
142  *
143  * @param p the context
144  * @param buf pointer to the input buffer
145  * @param buf_size size of the input buffer
146  */
148  int buf_size)
149 {
150  GetBitContext gb;
151  int ad_cb_len;
152  int temp, info_bits, i;
153  int ret;
154 
155  ret = init_get_bits8(&gb, buf, buf_size);
156  if (ret < 0)
157  return ret;
158 
159  /* Extract frame type and rate info */
160  info_bits = get_bits(&gb, 2);
161 
162  if (info_bits == 3) {
164  return 0;
165  }
166 
167  /* Extract 24 bit lsp indices, 8 bit for each band */
168  p->lsp_index[2] = get_bits(&gb, 8);
169  p->lsp_index[1] = get_bits(&gb, 8);
170  p->lsp_index[0] = get_bits(&gb, 8);
171 
172  if (info_bits == 2) {
174  p->subframe[0].amp_index = get_bits(&gb, 6);
175  return 0;
176  }
177 
178  /* Extract the info common to both rates */
179  p->cur_rate = info_bits ? RATE_5300 : RATE_6300;
181 
182  p->pitch_lag[0] = get_bits(&gb, 7);
183  if (p->pitch_lag[0] > 123) /* test if forbidden code */
184  return -1;
185  p->pitch_lag[0] += PITCH_MIN;
186  p->subframe[1].ad_cb_lag = get_bits(&gb, 2);
187 
188  p->pitch_lag[1] = get_bits(&gb, 7);
189  if (p->pitch_lag[1] > 123)
190  return -1;
191  p->pitch_lag[1] += PITCH_MIN;
192  p->subframe[3].ad_cb_lag = get_bits(&gb, 2);
193  p->subframe[0].ad_cb_lag = 1;
194  p->subframe[2].ad_cb_lag = 1;
195 
196  for (i = 0; i < SUBFRAMES; i++) {
197  /* Extract combined gain */
198  temp = get_bits(&gb, 12);
199  ad_cb_len = 170;
200  p->subframe[i].dirac_train = 0;
201  if (p->cur_rate == RATE_6300 && p->pitch_lag[i >> 1] < SUBFRAME_LEN - 2) {
202  p->subframe[i].dirac_train = temp >> 11;
203  temp &= 0x7FF;
204  ad_cb_len = 85;
205  }
207  if (p->subframe[i].ad_cb_gain < ad_cb_len) {
209  GAIN_LEVELS;
210  } else {
211  return -1;
212  }
213  }
214 
215  p->subframe[0].grid_index = get_bits1(&gb);
216  p->subframe[1].grid_index = get_bits1(&gb);
217  p->subframe[2].grid_index = get_bits1(&gb);
218  p->subframe[3].grid_index = get_bits1(&gb);
219 
220  if (p->cur_rate == RATE_6300) {
221  skip_bits1(&gb); /* skip reserved bit */
222 
223  /* Compute pulse_pos index using the 13-bit combined position index */
224  temp = get_bits(&gb, 13);
225  p->subframe[0].pulse_pos = temp / 810;
226 
227  temp -= p->subframe[0].pulse_pos * 810;
228  p->subframe[1].pulse_pos = FASTDIV(temp, 90);
229 
230  temp -= p->subframe[1].pulse_pos * 90;
231  p->subframe[2].pulse_pos = FASTDIV(temp, 9);
232  p->subframe[3].pulse_pos = temp - p->subframe[2].pulse_pos * 9;
233 
234  p->subframe[0].pulse_pos = (p->subframe[0].pulse_pos << 16) +
235  get_bits(&gb, 16);
236  p->subframe[1].pulse_pos = (p->subframe[1].pulse_pos << 14) +
237  get_bits(&gb, 14);
238  p->subframe[2].pulse_pos = (p->subframe[2].pulse_pos << 16) +
239  get_bits(&gb, 16);
240  p->subframe[3].pulse_pos = (p->subframe[3].pulse_pos << 14) +
241  get_bits(&gb, 14);
242 
243  p->subframe[0].pulse_sign = get_bits(&gb, 6);
244  p->subframe[1].pulse_sign = get_bits(&gb, 5);
245  p->subframe[2].pulse_sign = get_bits(&gb, 6);
246  p->subframe[3].pulse_sign = get_bits(&gb, 5);
247  } else { /* 5300 bps */
248  p->subframe[0].pulse_pos = get_bits(&gb, 12);
249  p->subframe[1].pulse_pos = get_bits(&gb, 12);
250  p->subframe[2].pulse_pos = get_bits(&gb, 12);
251  p->subframe[3].pulse_pos = get_bits(&gb, 12);
252 
253  p->subframe[0].pulse_sign = get_bits(&gb, 4);
254  p->subframe[1].pulse_sign = get_bits(&gb, 4);
255  p->subframe[2].pulse_sign = get_bits(&gb, 4);
256  p->subframe[3].pulse_sign = get_bits(&gb, 4);
257  }
258 
259  return 0;
260 }
261 
262 /**
263  * Bitexact implementation of sqrt(val/2).
264  */
265 static int16_t square_root(unsigned val)
266 {
267  av_assert2(!(val & 0x80000000));
268 
269  return (ff_sqrt(val << 1) >> 1) & (~1);
270 }
271 
272 /**
273  * Generate fixed codebook excitation vector.
274  *
275  * @param vector decoded excitation vector
276  * @param subfrm current subframe
277  * @param cur_rate current bitrate
278  * @param pitch_lag closed loop pitch lag
279  * @param index current subframe index
280  */
281 static void gen_fcb_excitation(int16_t *vector, G723_1_Subframe *subfrm,
282  enum Rate cur_rate, int pitch_lag, int index)
283 {
284  int temp, i, j;
285 
286  memset(vector, 0, SUBFRAME_LEN * sizeof(*vector));
287 
288  if (cur_rate == RATE_6300) {
289  if (subfrm->pulse_pos >= max_pos[index])
290  return;
291 
292  /* Decode amplitudes and positions */
293  j = PULSE_MAX - pulses[index];
294  temp = subfrm->pulse_pos;
295  for (i = 0; i < SUBFRAME_LEN / GRID_SIZE; i++) {
297  if (temp >= 0)
298  continue;
300  if (subfrm->pulse_sign & (1 << (PULSE_MAX - j))) {
301  vector[subfrm->grid_index + GRID_SIZE * i] =
303  } else {
304  vector[subfrm->grid_index + GRID_SIZE * i] =
306  }
307  if (j == PULSE_MAX)
308  break;
309  }
310  if (subfrm->dirac_train == 1)
311  ff_g723_1_gen_dirac_train(vector, pitch_lag);
312  } else { /* 5300 bps */
313  int cb_gain = ff_g723_1_fixed_cb_gain[subfrm->amp_index];
314  int cb_shift = subfrm->grid_index;
315  int cb_sign = subfrm->pulse_sign;
316  int cb_pos = subfrm->pulse_pos;
317  int offset, beta, lag;
318 
319  for (i = 0; i < 8; i += 2) {
320  offset = ((cb_pos & 7) << 3) + cb_shift + i;
321  vector[offset] = (cb_sign & 1) ? cb_gain : -cb_gain;
322  cb_pos >>= 3;
323  cb_sign >>= 1;
324  }
325 
326  /* Enhance harmonic components */
327  lag = pitch_contrib[subfrm->ad_cb_gain << 1] + pitch_lag +
328  subfrm->ad_cb_lag - 1;
329  beta = pitch_contrib[(subfrm->ad_cb_gain << 1) + 1];
330 
331  if (lag < SUBFRAME_LEN - 2) {
332  for (i = lag; i < SUBFRAME_LEN; i++)
333  vector[i] += beta * vector[i - lag] >> 15;
334  }
335  }
336 }
337 
338 /**
339  * Estimate maximum auto-correlation around pitch lag.
340  *
341  * @param buf buffer with offset applied
342  * @param offset offset of the excitation vector
343  * @param ccr_max pointer to the maximum auto-correlation
344  * @param pitch_lag decoded pitch lag
345  * @param length length of autocorrelation
346  * @param dir forward lag(1) / backward lag(-1)
347  */
348 static int autocorr_max(const int16_t *buf, int offset, int *ccr_max,
349  int pitch_lag, int length, int dir)
350 {
351  int limit, ccr, lag = 0;
352  int i;
353 
354  pitch_lag = FFMIN(PITCH_MAX - 3, pitch_lag);
355  if (dir > 0)
356  limit = FFMIN(FRAME_LEN + PITCH_MAX - offset - length, pitch_lag + 3);
357  else
358  limit = pitch_lag + 3;
359 
360  for (i = pitch_lag - 3; i <= limit; i++) {
361  ccr = ff_g723_1_dot_product(buf, buf + dir * i, length);
362 
363  if (ccr > *ccr_max) {
364  *ccr_max = ccr;
365  lag = i;
366  }
367  }
368  return lag;
369 }
370 
371 /**
372  * Calculate pitch postfilter optimal and scaling gains.
373  *
374  * @param lag pitch postfilter forward/backward lag
375  * @param ppf pitch postfilter parameters
376  * @param cur_rate current bitrate
377  * @param tgt_eng target energy
378  * @param ccr cross-correlation
379  * @param res_eng residual energy
380  */
381 static void comp_ppf_gains(int lag, PPFParam *ppf, enum Rate cur_rate,
382  int tgt_eng, int ccr, int res_eng)
383 {
384  int pf_residual; /* square of postfiltered residual */
385  int temp1, temp2;
386 
387  ppf->index = lag;
388 
389  temp1 = tgt_eng * res_eng >> 1;
390  temp2 = ccr * ccr << 1;
391 
392  if (temp2 > temp1) {
393  if (ccr >= res_eng) {
394  ppf->opt_gain = ppf_gain_weight[cur_rate];
395  } else {
396  ppf->opt_gain = (ccr << 15) / res_eng *
397  ppf_gain_weight[cur_rate] >> 15;
398  }
399  /* pf_res^2 = tgt_eng + 2*ccr*gain + res_eng*gain^2 */
400  temp1 = (tgt_eng << 15) + (ccr * ppf->opt_gain << 1);
401  temp2 = (ppf->opt_gain * ppf->opt_gain >> 15) * res_eng;
402  pf_residual = av_sat_add32(temp1, temp2 + (1 << 15)) >> 16;
403 
404  if (tgt_eng >= pf_residual << 1) {
405  temp1 = 0x7fff;
406  } else {
407  temp1 = (tgt_eng << 14) / pf_residual;
408  }
409 
410  /* scaling_gain = sqrt(tgt_eng/pf_res^2) */
411  ppf->sc_gain = square_root(temp1 << 16);
412  } else {
413  ppf->opt_gain = 0;
414  ppf->sc_gain = 0x7fff;
415  }
416 
417  ppf->opt_gain = av_clip_int16(ppf->opt_gain * ppf->sc_gain >> 15);
418 }
419 
420 /**
421  * Calculate pitch postfilter parameters.
422  *
423  * @param p the context
424  * @param offset offset of the excitation vector
425  * @param pitch_lag decoded pitch lag
426  * @param ppf pitch postfilter parameters
427  * @param cur_rate current bitrate
428  */
429 static void comp_ppf_coeff(G723_1_ChannelContext *p, int offset, int pitch_lag,
430  PPFParam *ppf, enum Rate cur_rate)
431 {
432 
433  int16_t scale;
434  int i;
435  int temp1, temp2;
436 
437  /*
438  * 0 - target energy
439  * 1 - forward cross-correlation
440  * 2 - forward residual energy
441  * 3 - backward cross-correlation
442  * 4 - backward residual energy
443  */
444  int energy[5] = {0, 0, 0, 0, 0};
445  int16_t *buf = p->audio + LPC_ORDER + offset;
446  int fwd_lag = autocorr_max(buf, offset, &energy[1], pitch_lag,
447  SUBFRAME_LEN, 1);
448  int back_lag = autocorr_max(buf, offset, &energy[3], pitch_lag,
449  SUBFRAME_LEN, -1);
450 
451  ppf->index = 0;
452  ppf->opt_gain = 0;
453  ppf->sc_gain = 0x7fff;
454 
455  /* Case 0, Section 3.6 */
456  if (!back_lag && !fwd_lag)
457  return;
458 
459  /* Compute target energy */
460  energy[0] = ff_g723_1_dot_product(buf, buf, SUBFRAME_LEN);
461 
462  /* Compute forward residual energy */
463  if (fwd_lag)
464  energy[2] = ff_g723_1_dot_product(buf + fwd_lag, buf + fwd_lag,
465  SUBFRAME_LEN);
466 
467  /* Compute backward residual energy */
468  if (back_lag)
469  energy[4] = ff_g723_1_dot_product(buf - back_lag, buf - back_lag,
470  SUBFRAME_LEN);
471 
472  /* Normalize and shorten */
473  temp1 = 0;
474  for (i = 0; i < 5; i++)
475  temp1 = FFMAX(energy[i], temp1);
476 
477  scale = ff_g723_1_normalize_bits(temp1, 31);
478  for (i = 0; i < 5; i++)
479  energy[i] = (energy[i] << scale) >> 16;
480 
481  if (fwd_lag && !back_lag) { /* Case 1 */
482  comp_ppf_gains(fwd_lag, ppf, cur_rate, energy[0], energy[1],
483  energy[2]);
484  } else if (!fwd_lag) { /* Case 2 */
485  comp_ppf_gains(-back_lag, ppf, cur_rate, energy[0], energy[3],
486  energy[4]);
487  } else { /* Case 3 */
488 
489  /*
490  * Select the largest of energy[1]^2/energy[2]
491  * and energy[3]^2/energy[4]
492  */
493  temp1 = energy[4] * ((energy[1] * energy[1] + (1 << 14)) >> 15);
494  temp2 = energy[2] * ((energy[3] * energy[3] + (1 << 14)) >> 15);
495  if (temp1 >= temp2) {
496  comp_ppf_gains(fwd_lag, ppf, cur_rate, energy[0], energy[1],
497  energy[2]);
498  } else {
499  comp_ppf_gains(-back_lag, ppf, cur_rate, energy[0], energy[3],
500  energy[4]);
501  }
502  }
503 }
504 
505 /**
506  * Classify frames as voiced/unvoiced.
507  *
508  * @param p the context
509  * @param pitch_lag decoded pitch_lag
510  * @param exc_eng excitation energy estimation
511  * @param scale scaling factor of exc_eng
512  *
513  * @return residual interpolation index if voiced, 0 otherwise
514  */
515 static int comp_interp_index(G723_1_ChannelContext *p, int pitch_lag,
516  int *exc_eng, int *scale)
517 {
518  int offset = PITCH_MAX + 2 * SUBFRAME_LEN;
519  int16_t *buf = p->audio + LPC_ORDER;
520 
521  int index, ccr, tgt_eng, best_eng, temp;
522 
524  buf += offset;
525 
526  /* Compute maximum backward cross-correlation */
527  ccr = 0;
528  index = autocorr_max(buf, offset, &ccr, pitch_lag, SUBFRAME_LEN * 2, -1);
529  ccr = av_sat_add32(ccr, 1 << 15) >> 16;
530 
531  /* Compute target energy */
532  tgt_eng = ff_g723_1_dot_product(buf, buf, SUBFRAME_LEN * 2);
533  *exc_eng = av_sat_add32(tgt_eng, 1 << 15) >> 16;
534 
535  if (ccr <= 0)
536  return 0;
537 
538  /* Compute best energy */
539  best_eng = ff_g723_1_dot_product(buf - index, buf - index,
540  SUBFRAME_LEN * 2);
541  best_eng = av_sat_add32(best_eng, 1 << 15) >> 16;
542 
543  temp = best_eng * *exc_eng >> 3;
544 
545  if (temp < ccr * ccr) {
546  return index;
547  } else
548  return 0;
549 }
550 
551 /**
552  * Perform residual interpolation based on frame classification.
553  *
554  * @param buf decoded excitation vector
555  * @param out output vector
556  * @param lag decoded pitch lag
557  * @param gain interpolated gain
558  * @param rseed seed for random number generator
559  */
560 static void residual_interp(int16_t *buf, int16_t *out, int lag,
561  int gain, int *rseed)
562 {
563  int i;
564  if (lag) { /* Voiced */
565  int16_t *vector_ptr = buf + PITCH_MAX;
566  /* Attenuate */
567  for (i = 0; i < lag; i++)
568  out[i] = vector_ptr[i - lag] * 3 >> 2;
569  av_memcpy_backptr((uint8_t*)(out + lag), lag * sizeof(*out),
570  (FRAME_LEN - lag) * sizeof(*out));
571  } else { /* Unvoiced */
572  for (i = 0; i < FRAME_LEN; i++) {
573  *rseed = (int16_t)(*rseed * 521 + 259);
574  out[i] = gain * *rseed >> 15;
575  }
576  memset(buf, 0, (FRAME_LEN + PITCH_MAX) * sizeof(*buf));
577  }
578 }
579 
580 /**
581  * Perform IIR filtering.
582  *
583  * @param fir_coef FIR coefficients
584  * @param iir_coef IIR coefficients
585  * @param src source vector
586  * @param dest destination vector
587  * @param width width of the output, 16 bits(0) / 32 bits(1)
588  */
589 #define iir_filter(fir_coef, iir_coef, src, dest, width)\
590 {\
591  int m, n;\
592  int res_shift = 16 & ~-(width);\
593  int in_shift = 16 - res_shift;\
594 \
595  for (m = 0; m < SUBFRAME_LEN; m++) {\
596  int64_t filter = 0;\
597  for (n = 1; n <= LPC_ORDER; n++) {\
598  filter -= (fir_coef)[n - 1] * (src)[m - n] -\
599  (iir_coef)[n - 1] * ((dest)[m - n] >> in_shift);\
600  }\
601 \
602  (dest)[m] = av_clipl_int32(((src)[m] * 65536) + (filter * 8) +\
603  (1 << 15)) >> res_shift;\
604  }\
605 }
606 
607 /**
608  * Adjust gain of postfiltered signal.
609  *
610  * @param p the context
611  * @param buf postfiltered output vector
612  * @param energy input energy coefficient
613  */
614 static void gain_scale(G723_1_ChannelContext *p, int16_t * buf, int energy)
615 {
616  int num, denom, gain, bits1, bits2;
617  int i;
618 
619  num = energy;
620  denom = 0;
621  for (i = 0; i < SUBFRAME_LEN; i++) {
622  int temp = buf[i] >> 2;
623  temp *= temp;
624  denom = av_sat_dadd32(denom, temp);
625  }
626 
627  if (num && denom) {
628  bits1 = ff_g723_1_normalize_bits(num, 31);
629  bits2 = ff_g723_1_normalize_bits(denom, 31);
630  num = num << bits1 >> 1;
631  denom <<= bits2;
632 
633  bits2 = 5 + bits1 - bits2;
634  bits2 = av_clip_uintp2(bits2, 5);
635 
636  gain = (num >> 1) / (denom >> 16);
637  gain = square_root(gain << 16 >> bits2);
638  } else {
639  gain = 1 << 12;
640  }
641 
642  for (i = 0; i < SUBFRAME_LEN; i++) {
643  p->pf_gain = (15 * p->pf_gain + gain + (1 << 3)) >> 4;
644  buf[i] = av_clip_int16((buf[i] * (p->pf_gain + (p->pf_gain >> 4)) +
645  (1 << 10)) >> 11);
646  }
647 }
648 
649 /**
650  * Perform formant filtering.
651  *
652  * @param p the context
653  * @param lpc quantized lpc coefficients
654  * @param buf input buffer
655  * @param dst output buffer
656  */
657 static void formant_postfilter(G723_1_ChannelContext *p, int16_t *lpc,
658  int16_t *buf, int16_t *dst)
659 {
660  int16_t filter_coef[2][LPC_ORDER];
661  int filter_signal[LPC_ORDER + FRAME_LEN], *signal_ptr;
662  int i, j, k;
663 
664  memcpy(buf, p->fir_mem, LPC_ORDER * sizeof(*buf));
665  memcpy(filter_signal, p->iir_mem, LPC_ORDER * sizeof(*filter_signal));
666 
667  for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) {
668  for (k = 0; k < LPC_ORDER; k++) {
669  filter_coef[0][k] = (-lpc[k] * postfilter_tbl[0][k] +
670  (1 << 14)) >> 15;
671  filter_coef[1][k] = (-lpc[k] * postfilter_tbl[1][k] +
672  (1 << 14)) >> 15;
673  }
674  iir_filter(filter_coef[0], filter_coef[1], buf + i, filter_signal + i, 1);
675  lpc += LPC_ORDER;
676  }
677 
678  memcpy(p->fir_mem, buf + FRAME_LEN, LPC_ORDER * sizeof(int16_t));
679  memcpy(p->iir_mem, filter_signal + FRAME_LEN, LPC_ORDER * sizeof(int));
680 
681  buf += LPC_ORDER;
682  signal_ptr = filter_signal + LPC_ORDER;
683  for (i = 0; i < SUBFRAMES; i++) {
684  int temp;
685  int auto_corr[2];
686  int scale, energy;
687 
688  /* Normalize */
689  scale = ff_g723_1_scale_vector(dst, buf, SUBFRAME_LEN);
690 
691  /* Compute auto correlation coefficients */
692  auto_corr[0] = ff_g723_1_dot_product(dst, dst + 1, SUBFRAME_LEN - 1);
693  auto_corr[1] = ff_g723_1_dot_product(dst, dst, SUBFRAME_LEN);
694 
695  /* Compute reflection coefficient */
696  temp = auto_corr[1] >> 16;
697  if (temp) {
698  temp = (auto_corr[0] >> 2) / temp;
699  }
700  p->reflection_coef = (3 * p->reflection_coef + temp + 2) >> 2;
701  temp = -p->reflection_coef >> 1 & ~3;
702 
703  /* Compensation filter */
704  for (j = 0; j < SUBFRAME_LEN; j++) {
705  dst[j] = av_sat_dadd32(signal_ptr[j],
706  (signal_ptr[j - 1] >> 16) * temp) >> 16;
707  }
708 
709  /* Compute normalized signal energy */
710  temp = 2 * scale + 4;
711  if (temp < 0) {
712  energy = av_clipl_int32((int64_t)auto_corr[1] << -temp);
713  } else
714  energy = auto_corr[1] >> temp;
715 
716  gain_scale(p, dst, energy);
717 
718  buf += SUBFRAME_LEN;
719  signal_ptr += SUBFRAME_LEN;
720  dst += SUBFRAME_LEN;
721  }
722 }
723 
724 static int sid_gain_to_lsp_index(int gain)
725 {
726  if (gain < 0x10)
727  return gain << 6;
728  else if (gain < 0x20)
729  return gain - 8 << 7;
730  else
731  return gain - 20 << 8;
732 }
733 
734 static inline int cng_rand(int *state, int base)
735 {
736  *state = (*state * 521 + 259) & 0xFFFF;
737  return (*state & 0x7FFF) * base >> 15;
738 }
739 
741 {
742  int i, shift, seg, seg2, t, val, val_add, x, y;
743 
744  shift = 16 - p->cur_gain * 2;
745  if (shift > 0) {
746  if (p->sid_gain == 0) {
747  t = 0;
748  } else if (shift >= 31 || (int32_t)((uint32_t)p->sid_gain << shift) >> shift != p->sid_gain) {
749  if (p->sid_gain < 0) t = INT32_MIN;
750  else t = INT32_MAX;
751  } else
752  t = p->sid_gain * (1 << shift);
753  } else if(shift < -31) {
754  t = (p->sid_gain < 0) ? -1 : 0;
755  }else
756  t = p->sid_gain >> -shift;
757  x = av_clipl_int32(t * (int64_t)cng_filt[0] >> 16);
758 
759  if (x >= cng_bseg[2])
760  return 0x3F;
761 
762  if (x >= cng_bseg[1]) {
763  shift = 4;
764  seg = 3;
765  } else {
766  shift = 3;
767  seg = (x >= cng_bseg[0]);
768  }
769  seg2 = FFMIN(seg, 3);
770 
771  val = 1 << shift;
772  val_add = val >> 1;
773  for (i = 0; i < shift; i++) {
774  t = seg * 32 + (val << seg2);
775  t *= t;
776  if (x >= t)
777  val += val_add;
778  else
779  val -= val_add;
780  val_add >>= 1;
781  }
782 
783  t = seg * 32 + (val << seg2);
784  y = t * t - x;
785  if (y <= 0) {
786  t = seg * 32 + (val + 1 << seg2);
787  t = t * t - x;
788  val = (seg2 - 1) * 16 + val;
789  if (t >= y)
790  val++;
791  } else {
792  t = seg * 32 + (val - 1 << seg2);
793  t = t * t - x;
794  val = (seg2 - 1) * 16 + val;
795  if (t >= y)
796  val--;
797  }
798 
799  return val;
800 }
801 
803 {
804  int i, j, idx, t;
805  int off[SUBFRAMES];
806  int signs[SUBFRAMES / 2 * 11], pos[SUBFRAMES / 2 * 11];
807  int tmp[SUBFRAME_LEN * 2];
808  int16_t *vector_ptr;
809  int64_t sum;
810  int b0, c, delta, x, shift;
811 
812  p->pitch_lag[0] = cng_rand(&p->cng_random_seed, 21) + 123;
813  p->pitch_lag[1] = cng_rand(&p->cng_random_seed, 19) + 123;
814 
815  for (i = 0; i < SUBFRAMES; i++) {
816  p->subframe[i].ad_cb_gain = cng_rand(&p->cng_random_seed, 50) + 1;
818  }
819 
820  for (i = 0; i < SUBFRAMES / 2; i++) {
821  t = cng_rand(&p->cng_random_seed, 1 << 13);
822  off[i * 2] = t & 1;
823  off[i * 2 + 1] = ((t >> 1) & 1) + SUBFRAME_LEN;
824  t >>= 2;
825  for (j = 0; j < 11; j++) {
826  signs[i * 11 + j] = ((t & 1) * 2 - 1) * (1 << 14);
827  t >>= 1;
828  }
829  }
830 
831  idx = 0;
832  for (i = 0; i < SUBFRAMES; i++) {
833  for (j = 0; j < SUBFRAME_LEN / 2; j++)
834  tmp[j] = j;
835  t = SUBFRAME_LEN / 2;
836  for (j = 0; j < pulses[i]; j++, idx++) {
837  int idx2 = cng_rand(&p->cng_random_seed, t);
838 
839  pos[idx] = tmp[idx2] * 2 + off[i];
840  tmp[idx2] = tmp[--t];
841  }
842  }
843 
844  vector_ptr = p->audio + LPC_ORDER;
845  memcpy(vector_ptr, p->prev_excitation,
846  PITCH_MAX * sizeof(*p->excitation));
847  for (i = 0; i < SUBFRAMES; i += 2) {
848  ff_g723_1_gen_acb_excitation(vector_ptr, vector_ptr,
849  p->pitch_lag[i >> 1], &p->subframe[i],
850  p->cur_rate);
852  vector_ptr + SUBFRAME_LEN,
853  p->pitch_lag[i >> 1], &p->subframe[i + 1],
854  p->cur_rate);
855 
856  t = 0;
857  for (j = 0; j < SUBFRAME_LEN * 2; j++)
858  t |= FFABS(vector_ptr[j]);
859  t = FFMIN(t, 0x7FFF);
860  if (!t) {
861  shift = 0;
862  } else {
863  shift = -10 + av_log2(t);
864  if (shift < -2)
865  shift = -2;
866  }
867  sum = 0;
868  if (shift < 0) {
869  for (j = 0; j < SUBFRAME_LEN * 2; j++) {
870  t = vector_ptr[j] * (1 << -shift);
871  sum += t * t;
872  tmp[j] = t;
873  }
874  } else {
875  for (j = 0; j < SUBFRAME_LEN * 2; j++) {
876  t = vector_ptr[j] >> shift;
877  sum += t * t;
878  tmp[j] = t;
879  }
880  }
881 
882  b0 = 0;
883  for (j = 0; j < 11; j++)
884  b0 += tmp[pos[(i / 2) * 11 + j]] * signs[(i / 2) * 11 + j];
885  b0 = b0 * 2 * 2979LL + (1 << 29) >> 30; // approximated division by 11
886 
887  c = p->cur_gain * (p->cur_gain * SUBFRAME_LEN >> 5);
888  if (shift * 2 + 3 >= 0)
889  c >>= shift * 2 + 3;
890  else
891  c <<= -(shift * 2 + 3);
892  c = (av_clipl_int32(sum << 1) - c) * 2979LL >> 15;
893 
894  delta = b0 * b0 * 2 - c;
895  if (delta <= 0) {
896  x = -b0;
897  } else {
899  x = delta - b0;
900  t = delta + b0;
901  if (FFABS(t) < FFABS(x))
902  x = -t;
903  }
904  shift++;
905  if (shift < 0)
906  x >>= -shift;
907  else
908  x *= 1 << shift;
909  x = av_clip(x, -10000, 10000);
910 
911  for (j = 0; j < 11; j++) {
912  idx = (i / 2) * 11 + j;
913  vector_ptr[pos[idx]] = av_clip_int16(vector_ptr[pos[idx]] +
914  (x * signs[idx] >> 15));
915  }
916 
917  /* copy decoded data to serve as a history for the next decoded subframes */
918  memcpy(vector_ptr + PITCH_MAX, vector_ptr,
919  sizeof(*vector_ptr) * SUBFRAME_LEN * 2);
920  vector_ptr += SUBFRAME_LEN * 2;
921  }
922  /* Save the excitation for the next frame */
923  memcpy(p->prev_excitation, p->audio + LPC_ORDER + FRAME_LEN,
924  PITCH_MAX * sizeof(*p->excitation));
925 }
926 
927 static int g723_1_decode_frame(AVCodecContext *avctx, void *data,
928  int *got_frame_ptr, AVPacket *avpkt)
929 {
930  G723_1_Context *s = avctx->priv_data;
931  AVFrame *frame = data;
932  const uint8_t *buf = avpkt->data;
933  int buf_size = avpkt->size;
934  int dec_mode = buf[0] & 3;
935 
936  PPFParam ppf[SUBFRAMES];
937  int16_t cur_lsp[LPC_ORDER];
938  int16_t lpc[SUBFRAMES * LPC_ORDER];
939  int16_t acb_vector[SUBFRAME_LEN];
940  int16_t *out;
941  int bad_frame = 0, i, j, ret;
942 
943  if (buf_size < frame_size[dec_mode] * avctx->channels) {
944  if (buf_size)
945  av_log(avctx, AV_LOG_WARNING,
946  "Expected %d bytes, got %d - skipping packet\n",
947  frame_size[dec_mode], buf_size);
948  *got_frame_ptr = 0;
949  return buf_size;
950  }
951 
953  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
954  return ret;
955 
956  for (int ch = 0; ch < avctx->channels; ch++) {
957  G723_1_ChannelContext *p = &s->ch[ch];
958  int16_t *audio = p->audio;
959 
960  if (unpack_bitstream(p, buf + ch * (buf_size / avctx->channels),
961  buf_size / avctx->channels) < 0) {
962  bad_frame = 1;
963  if (p->past_frame_type == ACTIVE_FRAME)
965  else
967  }
968 
969  out = (int16_t *)frame->extended_data[ch];
970 
971  if (p->cur_frame_type == ACTIVE_FRAME) {
972  if (!bad_frame)
973  p->erased_frames = 0;
974  else if (p->erased_frames != 3)
975  p->erased_frames++;
976 
977  ff_g723_1_inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, bad_frame);
978  ff_g723_1_lsp_interpolate(lpc, cur_lsp, p->prev_lsp);
979 
980  /* Save the lsp_vector for the next frame */
981  memcpy(p->prev_lsp, cur_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
982 
983  /* Generate the excitation for the frame */
984  memcpy(p->excitation, p->prev_excitation,
985  PITCH_MAX * sizeof(*p->excitation));
986  if (!p->erased_frames) {
987  int16_t *vector_ptr = p->excitation + PITCH_MAX;
988 
989  /* Update interpolation gain memory */
991  p->subframe[3].amp_index) >> 1];
992  for (i = 0; i < SUBFRAMES; i++) {
993  gen_fcb_excitation(vector_ptr, &p->subframe[i], p->cur_rate,
994  p->pitch_lag[i >> 1], i);
995  ff_g723_1_gen_acb_excitation(acb_vector,
996  &p->excitation[SUBFRAME_LEN * i],
997  p->pitch_lag[i >> 1],
998  &p->subframe[i], p->cur_rate);
999  /* Get the total excitation */
1000  for (j = 0; j < SUBFRAME_LEN; j++) {
1001  int v = av_clip_int16(vector_ptr[j] * 2);
1002  vector_ptr[j] = av_clip_int16(v + acb_vector[j]);
1003  }
1004  vector_ptr += SUBFRAME_LEN;
1005  }
1006 
1007  vector_ptr = p->excitation + PITCH_MAX;
1008 
1010  &p->sid_gain, &p->cur_gain);
1011 
1012  /* Perform pitch postfiltering */
1013  if (s->postfilter) {
1014  i = PITCH_MAX;
1015  for (j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
1016  comp_ppf_coeff(p, i, p->pitch_lag[j >> 1],
1017  ppf + j, p->cur_rate);
1018 
1019  for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
1021  vector_ptr + i,
1022  vector_ptr + i + ppf[j].index,
1023  ppf[j].sc_gain,
1024  ppf[j].opt_gain,
1025  1 << 14, 15, SUBFRAME_LEN);
1026  } else {
1027  audio = vector_ptr - LPC_ORDER;
1028  }
1029 
1030  /* Save the excitation for the next frame */
1031  memcpy(p->prev_excitation, p->excitation + FRAME_LEN,
1032  PITCH_MAX * sizeof(*p->excitation));
1033  } else {
1034  p->interp_gain = (p->interp_gain * 3 + 2) >> 2;
1035  if (p->erased_frames == 3) {
1036  /* Mute output */
1037  memset(p->excitation, 0,
1038  (FRAME_LEN + PITCH_MAX) * sizeof(*p->excitation));
1039  memset(p->prev_excitation, 0,
1040  PITCH_MAX * sizeof(*p->excitation));
1041  memset(frame->data[0], 0,
1042  (FRAME_LEN + LPC_ORDER) * sizeof(int16_t));
1043  } else {
1044  int16_t *buf = p->audio + LPC_ORDER;
1045 
1046  /* Regenerate frame */
1048  p->interp_gain, &p->random_seed);
1049 
1050  /* Save the excitation for the next frame */
1051  memcpy(p->prev_excitation, buf + (FRAME_LEN - PITCH_MAX),
1052  PITCH_MAX * sizeof(*p->excitation));
1053  }
1054  }
1056  } else {
1057  if (p->cur_frame_type == SID_FRAME) {
1060  } else if (p->past_frame_type == ACTIVE_FRAME) {
1061  p->sid_gain = estimate_sid_gain(p);
1062  }
1063 
1064  if (p->past_frame_type == ACTIVE_FRAME)
1065  p->cur_gain = p->sid_gain;
1066  else
1067  p->cur_gain = (p->cur_gain * 7 + p->sid_gain) >> 3;
1068  generate_noise(p);
1070  /* Save the lsp_vector for the next frame */
1071  memcpy(p->prev_lsp, p->sid_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
1072  }
1073 
1075 
1076  memcpy(p->audio, p->synth_mem, LPC_ORDER * sizeof(*p->audio));
1077  for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
1078  ff_celp_lp_synthesis_filter(p->audio + i, &lpc[j * LPC_ORDER],
1079  audio + i, SUBFRAME_LEN, LPC_ORDER,
1080  0, 1, 1 << 12);
1081  memcpy(p->synth_mem, p->audio + FRAME_LEN, LPC_ORDER * sizeof(*p->audio));
1082 
1083  if (s->postfilter) {
1084  formant_postfilter(p, lpc, p->audio, out);
1085  } else { // if output is not postfiltered it should be scaled by 2
1086  for (i = 0; i < FRAME_LEN; i++)
1087  out[i] = av_clip_int16(2 * p->audio[LPC_ORDER + i]);
1088  }
1089  }
1090 
1091  *got_frame_ptr = 1;
1092 
1093  return frame_size[dec_mode] * avctx->channels;
1094 }
1095 
1096 #define OFFSET(x) offsetof(G723_1_Context, x)
1097 #define AD AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM
1098 
1099 static const AVOption options[] = {
1100  { "postfilter", "enable postfilter", OFFSET(postfilter), AV_OPT_TYPE_BOOL,
1101  { .i64 = 1 }, 0, 1, AD },
1102  { NULL }
1103 };
1104 
1105 
1106 static const AVClass g723_1dec_class = {
1107  .class_name = "G.723.1 decoder",
1108  .item_name = av_default_item_name,
1109  .option = options,
1110  .version = LIBAVUTIL_VERSION_INT,
1111 };
1112 
1114  .name = "g723_1",
1115  .long_name = NULL_IF_CONFIG_SMALL("G.723.1"),
1116  .type = AVMEDIA_TYPE_AUDIO,
1117  .id = AV_CODEC_ID_G723_1,
1118  .priv_data_size = sizeof(G723_1_Context),
1121  .capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
1122  .priv_class = &g723_1dec_class,
1123 };
static const uint8_t bits2[81]
Definition: aactab.c:159
static const uint8_t bits1[81]
Definition: aactab.c:136
void ff_acelp_weighted_vector_sum(int16_t *out, const int16_t *in_a, const int16_t *in_b, int16_t weight_coeff_a, int16_t weight_coeff_b, int16_t rounder, int shift, int length)
weighted sum of two vectors with rounding.
static double val(void *priv, double ch)
Definition: aeval.c:76
static void postfilter(AMRContext *p, float *lpc, float *buf_out)
Perform adaptive post-filtering to enhance the quality of the speech.
Definition: amrnbdec.c:904
#define av_cold
Definition: attributes.h:88
uint8_t
int32_t
#define av_assert2(cond)
assert() equivalent, that does lie in speed critical code.
Definition: avassert.h:64
Libavcodec external API header.
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:31
#define s(width, name)
Definition: cbs_vp9.c:257
int ff_celp_lp_synthesis_filter(int16_t *out, const int16_t *filter_coeffs, const int16_t *in, int buffer_length, int filter_length, int stop_on_overflow, int shift, int rounder)
LP synthesis filter.
Definition: celp_filters.c:60
audio channel layout utility functions
static struct @321 state
#define FFMIN(a, b)
Definition: common.h:105
#define av_sat_dadd32
Definition: common.h:155
#define av_clip
Definition: common.h:122
#define av_clipl_int32
Definition: common.h:140
#define av_sat_add32
Definition: common.h:152
#define av_clip_int16
Definition: common.h:137
#define FFMAX(a, b)
Definition: common.h:103
#define av_clip_uintp2
Definition: common.h:146
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
Definition: common.h:72
#define NULL
Definition: coverity.c:32
long long int64_t
Definition: coverity.c:34
#define SUBFRAMES
Definition: dcaenc.c:51
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1900
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
Definition: decode_audio.c:71
static AVFrame * frame
#define PULSE_MAX
Definition: dss_sp.c:33
G.723.1 types, functions and data tables.
#define GRID_SIZE
Definition: g723_1.h:46
Rate
G723.1 rate values.
Definition: g723_1.h:72
@ RATE_6300
Definition: g723_1.h:73
@ RATE_5300
Definition: g723_1.h:74
#define PITCH_MAX
Definition: g723_1.h:44
static const int16_t dc_lsp[LPC_ORDER]
LSP DC component.
Definition: g723_1.h:227
#define FRAME_LEN
Definition: g723_1.h:37
#define SUBFRAME_LEN
Definition: g723_1.h:36
#define LPC_ORDER
Definition: g723_1.h:40
@ ACTIVE_FRAME
Active speech.
Definition: g723_1.h:64
@ UNTRANSMITTED_FRAME
Definition: g723_1.h:66
@ SID_FRAME
Silence Insertion Descriptor frame.
Definition: g723_1.h:65
#define PITCH_MIN
Definition: g723_1.h:43
static const int8_t pulses[4]
Number of non-zero pulses in the MP-MLQ excitation.
Definition: g723_1.h:260
#define GAIN_LEVELS
Definition: g723_1.h:48
static void formant_postfilter(G723_1_ChannelContext *p, int16_t *lpc, int16_t *buf, int16_t *dst)
Perform formant filtering.
Definition: g723_1dec.c:657
static const int32_t max_pos[4]
Size of the MP-MLQ fixed excitation codebooks.
Definition: g723_1dec.c:97
static const AVClass g723_1dec_class
Definition: g723_1dec.c:1106
static const int16_t postfilter_tbl[2][LPC_ORDER]
0.65^i (Zero part) and 0.75^i (Pole part) scaled by 2^15
Definition: g723_1dec.c:102
static int cng_rand(int *state, int base)
Definition: g723_1dec.c:734
static void gen_fcb_excitation(int16_t *vector, G723_1_Subframe *subfrm, enum Rate cur_rate, int pitch_lag, int index)
Generate fixed codebook excitation vector.
Definition: g723_1dec.c:281
static void gain_scale(G723_1_ChannelContext *p, int16_t *buf, int energy)
Adjust gain of postfiltered signal.
Definition: g723_1dec.c:614
static const int cng_adaptive_cb_lag[4]
Definition: g723_1dec.c:109
static int16_t square_root(unsigned val)
Bitexact implementation of sqrt(val/2).
Definition: g723_1dec.c:265
static const AVOption options[]
Definition: g723_1dec.c:1099
static void generate_noise(G723_1_ChannelContext *p)
Definition: g723_1dec.c:802
static int sid_gain_to_lsp_index(int gain)
Definition: g723_1dec.c:724
AVCodec ff_g723_1_decoder
Definition: g723_1dec.c:1113
#define CNG_RANDOM_SEED
Definition: g723_1dec.c:41
static void residual_interp(int16_t *buf, int16_t *out, int lag, int gain, int *rseed)
Perform residual interpolation based on frame classification.
Definition: g723_1dec.c:560
static void comp_ppf_gains(int lag, PPFParam *ppf, enum Rate cur_rate, int tgt_eng, int ccr, int res_eng)
Calculate pitch postfilter optimal and scaling gains.
Definition: g723_1dec.c:381
static int unpack_bitstream(G723_1_ChannelContext *p, const uint8_t *buf, int buf_size)
Unpack the frame into parameters.
Definition: g723_1dec.c:147
static int autocorr_max(const int16_t *buf, int offset, int *ccr_max, int pitch_lag, int length, int dir)
Estimate maximum auto-correlation around pitch lag.
Definition: g723_1dec.c:348
#define AD
Definition: g723_1dec.c:1097
static int estimate_sid_gain(G723_1_ChannelContext *p)
Definition: g723_1dec.c:740
static const int16_t pitch_contrib[340]
Definition: g723_1dec.c:48
static const int cng_bseg[3]
Definition: g723_1dec.c:113
#define OFFSET(x)
Definition: g723_1dec.c:1096
static int comp_interp_index(G723_1_ChannelContext *p, int pitch_lag, int *exc_eng, int *scale)
Classify frames as voiced/unvoiced.
Definition: g723_1dec.c:515
static av_cold int g723_1_decode_init(AVCodecContext *avctx)
Definition: g723_1dec.c:115
static void comp_ppf_coeff(G723_1_ChannelContext *p, int offset, int pitch_lag, PPFParam *ppf, enum Rate cur_rate)
Calculate pitch postfilter parameters.
Definition: g723_1dec.c:429
static const int16_t ppf_gain_weight[2]
Postfilter gain weighting factors scaled by 2^15.
Definition: g723_1dec.c:46
static int g723_1_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: g723_1dec.c:927
static const int cng_filt[4]
Definition: g723_1dec.c:111
#define iir_filter(fir_coef, iir_coef, src, dest, width)
Perform IIR filtering.
Definition: g723_1dec.c:589
bitstream reader API header.
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:498
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
Definition: get_bits.h:677
static void skip_bits1(GetBitContext *s)
Definition: get_bits.h:538
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:379
@ AV_OPT_TYPE_BOOL
Definition: opt.h:242
#define AV_CH_LAYOUT_MONO
#define AV_CH_LAYOUT_STEREO
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
Definition: codec.h:52
#define AV_CODEC_CAP_SUBFRAMES
Codec can output multiple frames per AVPacket Normally demuxers return one frame at a time,...
Definition: codec.h:95
@ AV_CODEC_ID_G723_1
Definition: codec_id.h:476
#define AVERROR(e)
Definition: error.h:43
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:200
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:194
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:235
void av_memcpy_backptr(uint8_t *dst, int back, int cnt)
Overlapping memcpy() implementation.
Definition: mem.c:428
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
@ AV_SAMPLE_FMT_S16P
signed 16 bits, planar
Definition: samplefmt.h:67
#define LIBAVUTIL_VERSION_INT
Definition: version.h:85
int index
Definition: gxfenc.c:89
int i
Definition: input.c:407
#define av_log2
Definition: intmath.h:83
int ff_g723_1_dot_product(const int16_t *a, const int16_t *b, int length)
Definition: g723_1.c:1125
void ff_g723_1_lsp_interpolate(int16_t *lpc, int16_t *cur_lsp, int16_t *prev_lsp)
Quantize LSP frequencies by interpolation and convert them to the corresponding LPC coefficients.
Definition: g723_1.c:1251
int ff_g723_1_normalize_bits(int num, int width)
Calculate the number of left-shifts required for normalizing the input.
Definition: g723_1.c:1120
void ff_g723_1_gen_acb_excitation(int16_t *vector, int16_t *prev_excitation, int pitch_lag, G723_1_Subframe *subfrm, enum Rate cur_rate)
Generate adaptive codebook excitation.
Definition: g723_1.c:1157
void ff_g723_1_gen_dirac_train(int16_t *buf, int pitch_lag)
Generate a train of dirac functions with period as pitch lag.
Definition: g723_1.c:1145
void ff_g723_1_inverse_quant(int16_t *cur_lsp, int16_t *prev_lsp, uint8_t *lsp_index, int bad_frame)
Perform inverse quantization of LSP frequencies.
Definition: g723_1.c:1272
int ff_g723_1_scale_vector(int16_t *dst, const int16_t *vector, int length)
Scale vector contents based on the largest of their absolutes.
Definition: g723_1.c:1103
const int32_t ff_g723_1_combinatorial_table[PULSE_MAX][SUBFRAME_LEN/GRID_SIZE]
Used for the coding/decoding of the pulses positions for the MP-MLQ codebook.
Definition: g723_1.c:409
const int16_t ff_g723_1_fixed_cb_gain[GAIN_LEVELS]
Definition: g723_1.c:453
common internal API header
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:117
#define FASTDIV(a, b)
Definition: mathops.h:202
#define ff_sqrt
Definition: mathops.h:206
Memory handling functions.
const char data[16]
Definition: mxf.c:142
int frame_size
Definition: mxfenc.c:2206
AVOptions.
static int shift(int a, int b)
Definition: sonic.c:82
unsigned int pos
Definition: spdifenc.c:412
Describe the class of an AVClass context structure.
Definition: log.h:67
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:72
main external API structure.
Definition: avcodec.h:536
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1204
int channels
number of audio channels
Definition: avcodec.h:1197
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:1247
void * priv_data
Definition: avcodec.h:563
AVCodec.
Definition: codec.h:197
const char * name
Name of the codec implementation.
Definition: codec.h:204
This structure describes decoded (raw) audio or video data.
Definition: frame.h:318
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:384
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:332
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:365
AVOption.
Definition: opt.h:248
This structure stores compressed data.
Definition: packet.h:346
int size
Definition: packet.h:370
uint8_t * data
Definition: packet.h:369
int pf_gain
formant postfilter gain scaling unit memory
Definition: g723_1.h:143
int16_t excitation[PITCH_MAX+FRAME_LEN+4]
Definition: g723_1.h:131
enum FrameType past_frame_type
Definition: g723_1.h:122
int iir_mem[LPC_ORDER]
Definition: g723_1.h:134
int16_t synth_mem[LPC_ORDER]
Definition: g723_1.h:132
int16_t prev_lsp[LPC_ORDER]
Definition: g723_1.h:128
uint8_t lsp_index[LSP_BANDS]
Definition: g723_1.h:124
int16_t sid_lsp[LPC_ORDER]
Definition: g723_1.h:129
int16_t audio[FRAME_LEN+LPC_ORDER+PITCH_MAX+4]
Definition: g723_1.h:145
enum Rate cur_rate
Definition: g723_1.h:123
enum FrameType cur_frame_type
Definition: g723_1.h:121
int16_t prev_excitation[PITCH_MAX]
Definition: g723_1.h:130
int16_t fir_mem[LPC_ORDER]
Definition: g723_1.h:133
G723_1_Subframe subframe[4]
Definition: g723_1.h:120
G723.1 unpacked data subframe.
Definition: g723_1.h:80
int ad_cb_gain
Definition: g723_1.h:82
int amp_index
Definition: g723_1.h:86
int grid_index
Definition: g723_1.h:85
int dirac_train
Definition: g723_1.h:83
int pulse_sign
Definition: g723_1.h:84
int ad_cb_lag
adaptive codebook lag
Definition: g723_1.h:81
int pulse_pos
Definition: g723_1.h:87
Pitch postfilter parameters.
Definition: g723_1.h:93
int16_t opt_gain
optimal gain
Definition: g723_1.h:95
int16_t sc_gain
scaling gain
Definition: g723_1.h:96
int index
postfilter backward/forward lag
Definition: g723_1.h:94
#define av_log(a,...)
static uint8_t tmp[11]
Definition: aes_ctr.c:27
FILE * out
Definition: movenc.c:54
else temp
Definition: vf_mcdeint.c:259
if(ret< 0)
Definition: vf_mcdeint.c:282
static const uint8_t offset[127][2]
Definition: vf_spp.c:107
static double b0(void *priv, double x, double y)
Definition: vf_xfade.c:1664
float delta
uint8_t base
Definition: vp3data.h:141
static double c[64]