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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 
28 #define BITSTREAM_READER_LE
30 #include "libavutil/mem.h"
31 #include "libavutil/opt.h"
32 #include "avcodec.h"
33 #include "get_bits.h"
34 #include "acelp_vectors.h"
35 #include "celp_filters.h"
36 #include "celp_math.h"
37 #include "g723_1.h"
38 #include "internal.h"
39 
40 #define CNG_RANDOM_SEED 12345
41 
43 {
44  G723_1_Context *p = avctx->priv_data;
45 
48  avctx->channels = 1;
49  p->pf_gain = 1 << 12;
50 
51  memcpy(p->prev_lsp, dc_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
52  memcpy(p->sid_lsp, dc_lsp, LPC_ORDER * sizeof(*p->sid_lsp));
53 
56 
57  return 0;
58 }
59 
60 /**
61  * Unpack the frame into parameters.
62  *
63  * @param p the context
64  * @param buf pointer to the input buffer
65  * @param buf_size size of the input buffer
66  */
68  int buf_size)
69 {
70  GetBitContext gb;
71  int ad_cb_len;
72  int temp, info_bits, i;
73 
74  init_get_bits(&gb, buf, buf_size * 8);
75 
76  /* Extract frame type and rate info */
77  info_bits = get_bits(&gb, 2);
78 
79  if (info_bits == 3) {
81  return 0;
82  }
83 
84  /* Extract 24 bit lsp indices, 8 bit for each band */
85  p->lsp_index[2] = get_bits(&gb, 8);
86  p->lsp_index[1] = get_bits(&gb, 8);
87  p->lsp_index[0] = get_bits(&gb, 8);
88 
89  if (info_bits == 2) {
91  p->subframe[0].amp_index = get_bits(&gb, 6);
92  return 0;
93  }
94 
95  /* Extract the info common to both rates */
96  p->cur_rate = info_bits ? RATE_5300 : RATE_6300;
98 
99  p->pitch_lag[0] = get_bits(&gb, 7);
100  if (p->pitch_lag[0] > 123) /* test if forbidden code */
101  return -1;
102  p->pitch_lag[0] += PITCH_MIN;
103  p->subframe[1].ad_cb_lag = get_bits(&gb, 2);
104 
105  p->pitch_lag[1] = get_bits(&gb, 7);
106  if (p->pitch_lag[1] > 123)
107  return -1;
108  p->pitch_lag[1] += PITCH_MIN;
109  p->subframe[3].ad_cb_lag = get_bits(&gb, 2);
110  p->subframe[0].ad_cb_lag = 1;
111  p->subframe[2].ad_cb_lag = 1;
112 
113  for (i = 0; i < SUBFRAMES; i++) {
114  /* Extract combined gain */
115  temp = get_bits(&gb, 12);
116  ad_cb_len = 170;
117  p->subframe[i].dirac_train = 0;
118  if (p->cur_rate == RATE_6300 && p->pitch_lag[i >> 1] < SUBFRAME_LEN - 2) {
119  p->subframe[i].dirac_train = temp >> 11;
120  temp &= 0x7FF;
121  ad_cb_len = 85;
122  }
123  p->subframe[i].ad_cb_gain = FASTDIV(temp, GAIN_LEVELS);
124  if (p->subframe[i].ad_cb_gain < ad_cb_len) {
125  p->subframe[i].amp_index = temp - p->subframe[i].ad_cb_gain *
126  GAIN_LEVELS;
127  } else {
128  return -1;
129  }
130  }
131 
132  p->subframe[0].grid_index = get_bits1(&gb);
133  p->subframe[1].grid_index = get_bits1(&gb);
134  p->subframe[2].grid_index = get_bits1(&gb);
135  p->subframe[3].grid_index = get_bits1(&gb);
136 
137  if (p->cur_rate == RATE_6300) {
138  skip_bits1(&gb); /* skip reserved bit */
139 
140  /* Compute pulse_pos index using the 13-bit combined position index */
141  temp = get_bits(&gb, 13);
142  p->subframe[0].pulse_pos = temp / 810;
143 
144  temp -= p->subframe[0].pulse_pos * 810;
145  p->subframe[1].pulse_pos = FASTDIV(temp, 90);
146 
147  temp -= p->subframe[1].pulse_pos * 90;
148  p->subframe[2].pulse_pos = FASTDIV(temp, 9);
149  p->subframe[3].pulse_pos = temp - p->subframe[2].pulse_pos * 9;
150 
151  p->subframe[0].pulse_pos = (p->subframe[0].pulse_pos << 16) +
152  get_bits(&gb, 16);
153  p->subframe[1].pulse_pos = (p->subframe[1].pulse_pos << 14) +
154  get_bits(&gb, 14);
155  p->subframe[2].pulse_pos = (p->subframe[2].pulse_pos << 16) +
156  get_bits(&gb, 16);
157  p->subframe[3].pulse_pos = (p->subframe[3].pulse_pos << 14) +
158  get_bits(&gb, 14);
159 
160  p->subframe[0].pulse_sign = get_bits(&gb, 6);
161  p->subframe[1].pulse_sign = get_bits(&gb, 5);
162  p->subframe[2].pulse_sign = get_bits(&gb, 6);
163  p->subframe[3].pulse_sign = get_bits(&gb, 5);
164  } else { /* 5300 bps */
165  p->subframe[0].pulse_pos = get_bits(&gb, 12);
166  p->subframe[1].pulse_pos = get_bits(&gb, 12);
167  p->subframe[2].pulse_pos = get_bits(&gb, 12);
168  p->subframe[3].pulse_pos = get_bits(&gb, 12);
169 
170  p->subframe[0].pulse_sign = get_bits(&gb, 4);
171  p->subframe[1].pulse_sign = get_bits(&gb, 4);
172  p->subframe[2].pulse_sign = get_bits(&gb, 4);
173  p->subframe[3].pulse_sign = get_bits(&gb, 4);
174  }
175 
176  return 0;
177 }
178 
179 /**
180  * Bitexact implementation of sqrt(val/2).
181  */
182 static int16_t square_root(unsigned val)
183 {
184  av_assert2(!(val & 0x80000000));
185 
186  return (ff_sqrt(val << 1) >> 1) & (~1);
187 }
188 
189 /**
190  * Generate fixed codebook excitation vector.
191  *
192  * @param vector decoded excitation vector
193  * @param subfrm current subframe
194  * @param cur_rate current bitrate
195  * @param pitch_lag closed loop pitch lag
196  * @param index current subframe index
197  */
198 static void gen_fcb_excitation(int16_t *vector, G723_1_Subframe *subfrm,
199  enum Rate cur_rate, int pitch_lag, int index)
200 {
201  int temp, i, j;
202 
203  memset(vector, 0, SUBFRAME_LEN * sizeof(*vector));
204 
205  if (cur_rate == RATE_6300) {
206  if (subfrm->pulse_pos >= max_pos[index])
207  return;
208 
209  /* Decode amplitudes and positions */
210  j = PULSE_MAX - pulses[index];
211  temp = subfrm->pulse_pos;
212  for (i = 0; i < SUBFRAME_LEN / GRID_SIZE; i++) {
213  temp -= combinatorial_table[j][i];
214  if (temp >= 0)
215  continue;
216  temp += combinatorial_table[j++][i];
217  if (subfrm->pulse_sign & (1 << (PULSE_MAX - j))) {
218  vector[subfrm->grid_index + GRID_SIZE * i] =
219  -fixed_cb_gain[subfrm->amp_index];
220  } else {
221  vector[subfrm->grid_index + GRID_SIZE * i] =
222  fixed_cb_gain[subfrm->amp_index];
223  }
224  if (j == PULSE_MAX)
225  break;
226  }
227  if (subfrm->dirac_train == 1)
228  ff_g723_1_gen_dirac_train(vector, pitch_lag);
229  } else { /* 5300 bps */
230  int cb_gain = fixed_cb_gain[subfrm->amp_index];
231  int cb_shift = subfrm->grid_index;
232  int cb_sign = subfrm->pulse_sign;
233  int cb_pos = subfrm->pulse_pos;
234  int offset, beta, lag;
235 
236  for (i = 0; i < 8; i += 2) {
237  offset = ((cb_pos & 7) << 3) + cb_shift + i;
238  vector[offset] = (cb_sign & 1) ? cb_gain : -cb_gain;
239  cb_pos >>= 3;
240  cb_sign >>= 1;
241  }
242 
243  /* Enhance harmonic components */
244  lag = pitch_contrib[subfrm->ad_cb_gain << 1] + pitch_lag +
245  subfrm->ad_cb_lag - 1;
246  beta = pitch_contrib[(subfrm->ad_cb_gain << 1) + 1];
247 
248  if (lag < SUBFRAME_LEN - 2) {
249  for (i = lag; i < SUBFRAME_LEN; i++)
250  vector[i] += beta * vector[i - lag] >> 15;
251  }
252  }
253 }
254 
255 /**
256  * Estimate maximum auto-correlation around pitch lag.
257  *
258  * @param buf buffer with offset applied
259  * @param offset offset of the excitation vector
260  * @param ccr_max pointer to the maximum auto-correlation
261  * @param pitch_lag decoded pitch lag
262  * @param length length of autocorrelation
263  * @param dir forward lag(1) / backward lag(-1)
264  */
265 static int autocorr_max(const int16_t *buf, int offset, int *ccr_max,
266  int pitch_lag, int length, int dir)
267 {
268  int limit, ccr, lag = 0;
269  int i;
270 
271  pitch_lag = FFMIN(PITCH_MAX - 3, pitch_lag);
272  if (dir > 0)
273  limit = FFMIN(FRAME_LEN + PITCH_MAX - offset - length, pitch_lag + 3);
274  else
275  limit = pitch_lag + 3;
276 
277  for (i = pitch_lag - 3; i <= limit; i++) {
278  ccr = ff_g723_1_dot_product(buf, buf + dir * i, length);
279 
280  if (ccr > *ccr_max) {
281  *ccr_max = ccr;
282  lag = i;
283  }
284  }
285  return lag;
286 }
287 
288 /**
289  * Calculate pitch postfilter optimal and scaling gains.
290  *
291  * @param lag pitch postfilter forward/backward lag
292  * @param ppf pitch postfilter parameters
293  * @param cur_rate current bitrate
294  * @param tgt_eng target energy
295  * @param ccr cross-correlation
296  * @param res_eng residual energy
297  */
298 static void comp_ppf_gains(int lag, PPFParam *ppf, enum Rate cur_rate,
299  int tgt_eng, int ccr, int res_eng)
300 {
301  int pf_residual; /* square of postfiltered residual */
302  int temp1, temp2;
303 
304  ppf->index = lag;
305 
306  temp1 = tgt_eng * res_eng >> 1;
307  temp2 = ccr * ccr << 1;
308 
309  if (temp2 > temp1) {
310  if (ccr >= res_eng) {
311  ppf->opt_gain = ppf_gain_weight[cur_rate];
312  } else {
313  ppf->opt_gain = (ccr << 15) / res_eng *
314  ppf_gain_weight[cur_rate] >> 15;
315  }
316  /* pf_res^2 = tgt_eng + 2*ccr*gain + res_eng*gain^2 */
317  temp1 = (tgt_eng << 15) + (ccr * ppf->opt_gain << 1);
318  temp2 = (ppf->opt_gain * ppf->opt_gain >> 15) * res_eng;
319  pf_residual = av_sat_add32(temp1, temp2 + (1 << 15)) >> 16;
320 
321  if (tgt_eng >= pf_residual << 1) {
322  temp1 = 0x7fff;
323  } else {
324  temp1 = (tgt_eng << 14) / pf_residual;
325  }
326 
327  /* scaling_gain = sqrt(tgt_eng/pf_res^2) */
328  ppf->sc_gain = square_root(temp1 << 16);
329  } else {
330  ppf->opt_gain = 0;
331  ppf->sc_gain = 0x7fff;
332  }
333 
334  ppf->opt_gain = av_clip_int16(ppf->opt_gain * ppf->sc_gain >> 15);
335 }
336 
337 /**
338  * Calculate pitch postfilter parameters.
339  *
340  * @param p the context
341  * @param offset offset of the excitation vector
342  * @param pitch_lag decoded pitch lag
343  * @param ppf pitch postfilter parameters
344  * @param cur_rate current bitrate
345  */
346 static void comp_ppf_coeff(G723_1_Context *p, int offset, int pitch_lag,
347  PPFParam *ppf, enum Rate cur_rate)
348 {
349 
350  int16_t scale;
351  int i;
352  int temp1, temp2;
353 
354  /*
355  * 0 - target energy
356  * 1 - forward cross-correlation
357  * 2 - forward residual energy
358  * 3 - backward cross-correlation
359  * 4 - backward residual energy
360  */
361  int energy[5] = {0, 0, 0, 0, 0};
362  int16_t *buf = p->audio + LPC_ORDER + offset;
363  int fwd_lag = autocorr_max(buf, offset, &energy[1], pitch_lag,
364  SUBFRAME_LEN, 1);
365  int back_lag = autocorr_max(buf, offset, &energy[3], pitch_lag,
366  SUBFRAME_LEN, -1);
367 
368  ppf->index = 0;
369  ppf->opt_gain = 0;
370  ppf->sc_gain = 0x7fff;
371 
372  /* Case 0, Section 3.6 */
373  if (!back_lag && !fwd_lag)
374  return;
375 
376  /* Compute target energy */
377  energy[0] = ff_g723_1_dot_product(buf, buf, SUBFRAME_LEN);
378 
379  /* Compute forward residual energy */
380  if (fwd_lag)
381  energy[2] = ff_g723_1_dot_product(buf + fwd_lag, buf + fwd_lag,
382  SUBFRAME_LEN);
383 
384  /* Compute backward residual energy */
385  if (back_lag)
386  energy[4] = ff_g723_1_dot_product(buf - back_lag, buf - back_lag,
387  SUBFRAME_LEN);
388 
389  /* Normalize and shorten */
390  temp1 = 0;
391  for (i = 0; i < 5; i++)
392  temp1 = FFMAX(energy[i], temp1);
393 
394  scale = ff_g723_1_normalize_bits(temp1, 31);
395  for (i = 0; i < 5; i++)
396  energy[i] = (energy[i] << scale) >> 16;
397 
398  if (fwd_lag && !back_lag) { /* Case 1 */
399  comp_ppf_gains(fwd_lag, ppf, cur_rate, energy[0], energy[1],
400  energy[2]);
401  } else if (!fwd_lag) { /* Case 2 */
402  comp_ppf_gains(-back_lag, ppf, cur_rate, energy[0], energy[3],
403  energy[4]);
404  } else { /* Case 3 */
405 
406  /*
407  * Select the largest of energy[1]^2/energy[2]
408  * and energy[3]^2/energy[4]
409  */
410  temp1 = energy[4] * ((energy[1] * energy[1] + (1 << 14)) >> 15);
411  temp2 = energy[2] * ((energy[3] * energy[3] + (1 << 14)) >> 15);
412  if (temp1 >= temp2) {
413  comp_ppf_gains(fwd_lag, ppf, cur_rate, energy[0], energy[1],
414  energy[2]);
415  } else {
416  comp_ppf_gains(-back_lag, ppf, cur_rate, energy[0], energy[3],
417  energy[4]);
418  }
419  }
420 }
421 
422 /**
423  * Classify frames as voiced/unvoiced.
424  *
425  * @param p the context
426  * @param pitch_lag decoded pitch_lag
427  * @param exc_eng excitation energy estimation
428  * @param scale scaling factor of exc_eng
429  *
430  * @return residual interpolation index if voiced, 0 otherwise
431  */
432 static int comp_interp_index(G723_1_Context *p, int pitch_lag,
433  int *exc_eng, int *scale)
434 {
435  int offset = PITCH_MAX + 2 * SUBFRAME_LEN;
436  int16_t *buf = p->audio + LPC_ORDER;
437 
438  int index, ccr, tgt_eng, best_eng, temp;
439 
441  buf += offset;
442 
443  /* Compute maximum backward cross-correlation */
444  ccr = 0;
445  index = autocorr_max(buf, offset, &ccr, pitch_lag, SUBFRAME_LEN * 2, -1);
446  ccr = av_sat_add32(ccr, 1 << 15) >> 16;
447 
448  /* Compute target energy */
449  tgt_eng = ff_g723_1_dot_product(buf, buf, SUBFRAME_LEN * 2);
450  *exc_eng = av_sat_add32(tgt_eng, 1 << 15) >> 16;
451 
452  if (ccr <= 0)
453  return 0;
454 
455  /* Compute best energy */
456  best_eng = ff_g723_1_dot_product(buf - index, buf - index,
457  SUBFRAME_LEN * 2);
458  best_eng = av_sat_add32(best_eng, 1 << 15) >> 16;
459 
460  temp = best_eng * *exc_eng >> 3;
461 
462  if (temp < ccr * ccr) {
463  return index;
464  } else
465  return 0;
466 }
467 
468 /**
469  * Perform residual interpolation based on frame classification.
470  *
471  * @param buf decoded excitation vector
472  * @param out output vector
473  * @param lag decoded pitch lag
474  * @param gain interpolated gain
475  * @param rseed seed for random number generator
476  */
477 static void residual_interp(int16_t *buf, int16_t *out, int lag,
478  int gain, int *rseed)
479 {
480  int i;
481  if (lag) { /* Voiced */
482  int16_t *vector_ptr = buf + PITCH_MAX;
483  /* Attenuate */
484  for (i = 0; i < lag; i++)
485  out[i] = vector_ptr[i - lag] * 3 >> 2;
486  av_memcpy_backptr((uint8_t*)(out + lag), lag * sizeof(*out),
487  (FRAME_LEN - lag) * sizeof(*out));
488  } else { /* Unvoiced */
489  for (i = 0; i < FRAME_LEN; i++) {
490  *rseed = *rseed * 521 + 259;
491  out[i] = gain * *rseed >> 15;
492  }
493  memset(buf, 0, (FRAME_LEN + PITCH_MAX) * sizeof(*buf));
494  }
495 }
496 
497 /**
498  * Perform IIR filtering.
499  *
500  * @param fir_coef FIR coefficients
501  * @param iir_coef IIR coefficients
502  * @param src source vector
503  * @param dest destination vector
504  * @param width width of the output, 16 bits(0) / 32 bits(1)
505  */
506 #define iir_filter(fir_coef, iir_coef, src, dest, width)\
507 {\
508  int m, n;\
509  int res_shift = 16 & ~-(width);\
510  int in_shift = 16 - res_shift;\
511 \
512  for (m = 0; m < SUBFRAME_LEN; m++) {\
513  int64_t filter = 0;\
514  for (n = 1; n <= LPC_ORDER; n++) {\
515  filter -= (fir_coef)[n - 1] * (src)[m - n] -\
516  (iir_coef)[n - 1] * ((dest)[m - n] >> in_shift);\
517  }\
518 \
519  (dest)[m] = av_clipl_int32(((src)[m] << 16) + (filter << 3) +\
520  (1 << 15)) >> res_shift;\
521  }\
522 }
523 
524 /**
525  * Adjust gain of postfiltered signal.
526  *
527  * @param p the context
528  * @param buf postfiltered output vector
529  * @param energy input energy coefficient
530  */
531 static void gain_scale(G723_1_Context *p, int16_t * buf, int energy)
532 {
533  int num, denom, gain, bits1, bits2;
534  int i;
535 
536  num = energy;
537  denom = 0;
538  for (i = 0; i < SUBFRAME_LEN; i++) {
539  int temp = buf[i] >> 2;
540  temp *= temp;
541  denom = av_sat_dadd32(denom, temp);
542  }
543 
544  if (num && denom) {
545  bits1 = ff_g723_1_normalize_bits(num, 31);
546  bits2 = ff_g723_1_normalize_bits(denom, 31);
547  num = num << bits1 >> 1;
548  denom <<= bits2;
549 
550  bits2 = 5 + bits1 - bits2;
551  bits2 = FFMAX(0, bits2);
552 
553  gain = (num >> 1) / (denom >> 16);
554  gain = square_root(gain << 16 >> bits2);
555  } else {
556  gain = 1 << 12;
557  }
558 
559  for (i = 0; i < SUBFRAME_LEN; i++) {
560  p->pf_gain = (15 * p->pf_gain + gain + (1 << 3)) >> 4;
561  buf[i] = av_clip_int16((buf[i] * (p->pf_gain + (p->pf_gain >> 4)) +
562  (1 << 10)) >> 11);
563  }
564 }
565 
566 /**
567  * Perform formant filtering.
568  *
569  * @param p the context
570  * @param lpc quantized lpc coefficients
571  * @param buf input buffer
572  * @param dst output buffer
573  */
574 static void formant_postfilter(G723_1_Context *p, int16_t *lpc,
575  int16_t *buf, int16_t *dst)
576 {
577  int16_t filter_coef[2][LPC_ORDER];
578  int filter_signal[LPC_ORDER + FRAME_LEN], *signal_ptr;
579  int i, j, k;
580 
581  memcpy(buf, p->fir_mem, LPC_ORDER * sizeof(*buf));
582  memcpy(filter_signal, p->iir_mem, LPC_ORDER * sizeof(*filter_signal));
583 
584  for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) {
585  for (k = 0; k < LPC_ORDER; k++) {
586  filter_coef[0][k] = (-lpc[k] * postfilter_tbl[0][k] +
587  (1 << 14)) >> 15;
588  filter_coef[1][k] = (-lpc[k] * postfilter_tbl[1][k] +
589  (1 << 14)) >> 15;
590  }
591  iir_filter(filter_coef[0], filter_coef[1], buf + i, filter_signal + i, 1);
592  lpc += LPC_ORDER;
593  }
594 
595  memcpy(p->fir_mem, buf + FRAME_LEN, LPC_ORDER * sizeof(int16_t));
596  memcpy(p->iir_mem, filter_signal + FRAME_LEN, LPC_ORDER * sizeof(int));
597 
598  buf += LPC_ORDER;
599  signal_ptr = filter_signal + LPC_ORDER;
600  for (i = 0; i < SUBFRAMES; i++) {
601  int temp;
602  int auto_corr[2];
603  int scale, energy;
604 
605  /* Normalize */
606  scale = ff_g723_1_scale_vector(dst, buf, SUBFRAME_LEN);
607 
608  /* Compute auto correlation coefficients */
609  auto_corr[0] = ff_g723_1_dot_product(dst, dst + 1, SUBFRAME_LEN - 1);
610  auto_corr[1] = ff_g723_1_dot_product(dst, dst, SUBFRAME_LEN);
611 
612  /* Compute reflection coefficient */
613  temp = auto_corr[1] >> 16;
614  if (temp) {
615  temp = (auto_corr[0] >> 2) / temp;
616  }
617  p->reflection_coef = (3 * p->reflection_coef + temp + 2) >> 2;
618  temp = -p->reflection_coef >> 1 & ~3;
619 
620  /* Compensation filter */
621  for (j = 0; j < SUBFRAME_LEN; j++) {
622  dst[j] = av_sat_dadd32(signal_ptr[j],
623  (signal_ptr[j - 1] >> 16) * temp) >> 16;
624  }
625 
626  /* Compute normalized signal energy */
627  temp = 2 * scale + 4;
628  if (temp < 0) {
629  energy = av_clipl_int32((int64_t)auto_corr[1] << -temp);
630  } else
631  energy = auto_corr[1] >> temp;
632 
633  gain_scale(p, dst, energy);
634 
635  buf += SUBFRAME_LEN;
636  signal_ptr += SUBFRAME_LEN;
637  dst += SUBFRAME_LEN;
638  }
639 }
640 
641 static int sid_gain_to_lsp_index(int gain)
642 {
643  if (gain < 0x10)
644  return gain << 6;
645  else if (gain < 0x20)
646  return gain - 8 << 7;
647  else
648  return gain - 20 << 8;
649 }
650 
651 static inline int cng_rand(int *state, int base)
652 {
653  *state = (*state * 521 + 259) & 0xFFFF;
654  return (*state & 0x7FFF) * base >> 15;
655 }
656 
658 {
659  int i, shift, seg, seg2, t, val, val_add, x, y;
660 
661  shift = 16 - p->cur_gain * 2;
662  if (shift > 0)
663  t = p->sid_gain << shift;
664  else
665  t = p->sid_gain >> -shift;
666  x = t * cng_filt[0] >> 16;
667 
668  if (x >= cng_bseg[2])
669  return 0x3F;
670 
671  if (x >= cng_bseg[1]) {
672  shift = 4;
673  seg = 3;
674  } else {
675  shift = 3;
676  seg = (x >= cng_bseg[0]);
677  }
678  seg2 = FFMIN(seg, 3);
679 
680  val = 1 << shift;
681  val_add = val >> 1;
682  for (i = 0; i < shift; i++) {
683  t = seg * 32 + (val << seg2);
684  t *= t;
685  if (x >= t)
686  val += val_add;
687  else
688  val -= val_add;
689  val_add >>= 1;
690  }
691 
692  t = seg * 32 + (val << seg2);
693  y = t * t - x;
694  if (y <= 0) {
695  t = seg * 32 + (val + 1 << seg2);
696  t = t * t - x;
697  val = (seg2 - 1 << 4) + val;
698  if (t >= y)
699  val++;
700  } else {
701  t = seg * 32 + (val - 1 << seg2);
702  t = t * t - x;
703  val = (seg2 - 1 << 4) + val;
704  if (t >= y)
705  val--;
706  }
707 
708  return val;
709 }
710 
712 {
713  int i, j, idx, t;
714  int off[SUBFRAMES];
715  int signs[SUBFRAMES / 2 * 11], pos[SUBFRAMES / 2 * 11];
716  int tmp[SUBFRAME_LEN * 2];
717  int16_t *vector_ptr;
718  int64_t sum;
719  int b0, c, delta, x, shift;
720 
721  p->pitch_lag[0] = cng_rand(&p->cng_random_seed, 21) + 123;
722  p->pitch_lag[1] = cng_rand(&p->cng_random_seed, 19) + 123;
723 
724  for (i = 0; i < SUBFRAMES; i++) {
725  p->subframe[i].ad_cb_gain = cng_rand(&p->cng_random_seed, 50) + 1;
727  }
728 
729  for (i = 0; i < SUBFRAMES / 2; i++) {
730  t = cng_rand(&p->cng_random_seed, 1 << 13);
731  off[i * 2] = t & 1;
732  off[i * 2 + 1] = ((t >> 1) & 1) + SUBFRAME_LEN;
733  t >>= 2;
734  for (j = 0; j < 11; j++) {
735  signs[i * 11 + j] = (t & 1) * 2 - 1 << 14;
736  t >>= 1;
737  }
738  }
739 
740  idx = 0;
741  for (i = 0; i < SUBFRAMES; i++) {
742  for (j = 0; j < SUBFRAME_LEN / 2; j++)
743  tmp[j] = j;
744  t = SUBFRAME_LEN / 2;
745  for (j = 0; j < pulses[i]; j++, idx++) {
746  int idx2 = cng_rand(&p->cng_random_seed, t);
747 
748  pos[idx] = tmp[idx2] * 2 + off[i];
749  tmp[idx2] = tmp[--t];
750  }
751  }
752 
753  vector_ptr = p->audio + LPC_ORDER;
754  memcpy(vector_ptr, p->prev_excitation,
755  PITCH_MAX * sizeof(*p->excitation));
756  for (i = 0; i < SUBFRAMES; i += 2) {
757  ff_g723_1_gen_acb_excitation(vector_ptr, vector_ptr,
758  p->pitch_lag[i >> 1], &p->subframe[i],
759  p->cur_rate);
761  vector_ptr + SUBFRAME_LEN,
762  p->pitch_lag[i >> 1], &p->subframe[i + 1],
763  p->cur_rate);
764 
765  t = 0;
766  for (j = 0; j < SUBFRAME_LEN * 2; j++)
767  t |= FFABS(vector_ptr[j]);
768  t = FFMIN(t, 0x7FFF);
769  if (!t) {
770  shift = 0;
771  } else {
772  shift = -10 + av_log2(t);
773  if (shift < -2)
774  shift = -2;
775  }
776  sum = 0;
777  if (shift < 0) {
778  for (j = 0; j < SUBFRAME_LEN * 2; j++) {
779  t = vector_ptr[j] << -shift;
780  sum += t * t;
781  tmp[j] = t;
782  }
783  } else {
784  for (j = 0; j < SUBFRAME_LEN * 2; j++) {
785  t = vector_ptr[j] >> shift;
786  sum += t * t;
787  tmp[j] = t;
788  }
789  }
790 
791  b0 = 0;
792  for (j = 0; j < 11; j++)
793  b0 += tmp[pos[(i / 2) * 11 + j]] * signs[(i / 2) * 11 + j];
794  b0 = b0 * 2 * 2979LL + (1 << 29) >> 30; // approximated division by 11
795 
796  c = p->cur_gain * (p->cur_gain * SUBFRAME_LEN >> 5);
797  if (shift * 2 + 3 >= 0)
798  c >>= shift * 2 + 3;
799  else
800  c <<= -(shift * 2 + 3);
801  c = (av_clipl_int32(sum << 1) - c) * 2979LL >> 15;
802 
803  delta = b0 * b0 * 2 - c;
804  if (delta <= 0) {
805  x = -b0;
806  } else {
807  delta = square_root(delta);
808  x = delta - b0;
809  t = delta + b0;
810  if (FFABS(t) < FFABS(x))
811  x = -t;
812  }
813  shift++;
814  if (shift < 0)
815  x >>= -shift;
816  else
817  x <<= shift;
818  x = av_clip(x, -10000, 10000);
819 
820  for (j = 0; j < 11; j++) {
821  idx = (i / 2) * 11 + j;
822  vector_ptr[pos[idx]] = av_clip_int16(vector_ptr[pos[idx]] +
823  (x * signs[idx] >> 15));
824  }
825 
826  /* copy decoded data to serve as a history for the next decoded subframes */
827  memcpy(vector_ptr + PITCH_MAX, vector_ptr,
828  sizeof(*vector_ptr) * SUBFRAME_LEN * 2);
829  vector_ptr += SUBFRAME_LEN * 2;
830  }
831  /* Save the excitation for the next frame */
832  memcpy(p->prev_excitation, p->audio + LPC_ORDER + FRAME_LEN,
833  PITCH_MAX * sizeof(*p->excitation));
834 }
835 
836 static int g723_1_decode_frame(AVCodecContext *avctx, void *data,
837  int *got_frame_ptr, AVPacket *avpkt)
838 {
839  G723_1_Context *p = avctx->priv_data;
840  AVFrame *frame = data;
841  const uint8_t *buf = avpkt->data;
842  int buf_size = avpkt->size;
843  int dec_mode = buf[0] & 3;
844 
845  PPFParam ppf[SUBFRAMES];
846  int16_t cur_lsp[LPC_ORDER];
847  int16_t lpc[SUBFRAMES * LPC_ORDER];
848  int16_t acb_vector[SUBFRAME_LEN];
849  int16_t *out;
850  int bad_frame = 0, i, j, ret;
851  int16_t *audio = p->audio;
852 
853  if (buf_size < frame_size[dec_mode]) {
854  if (buf_size)
855  av_log(avctx, AV_LOG_WARNING,
856  "Expected %d bytes, got %d - skipping packet\n",
857  frame_size[dec_mode], buf_size);
858  *got_frame_ptr = 0;
859  return buf_size;
860  }
861 
862  if (unpack_bitstream(p, buf, buf_size) < 0) {
863  bad_frame = 1;
864  if (p->past_frame_type == ACTIVE_FRAME)
866  else
868  }
869 
870  frame->nb_samples = FRAME_LEN;
871  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
872  return ret;
873 
874  out = (int16_t *)frame->data[0];
875 
876  if (p->cur_frame_type == ACTIVE_FRAME) {
877  if (!bad_frame)
878  p->erased_frames = 0;
879  else if (p->erased_frames != 3)
880  p->erased_frames++;
881 
882  ff_g723_1_inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, bad_frame);
883  ff_g723_1_lsp_interpolate(lpc, cur_lsp, p->prev_lsp);
884 
885  /* Save the lsp_vector for the next frame */
886  memcpy(p->prev_lsp, cur_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
887 
888  /* Generate the excitation for the frame */
889  memcpy(p->excitation, p->prev_excitation,
890  PITCH_MAX * sizeof(*p->excitation));
891  if (!p->erased_frames) {
892  int16_t *vector_ptr = p->excitation + PITCH_MAX;
893 
894  /* Update interpolation gain memory */
896  p->subframe[3].amp_index) >> 1];
897  for (i = 0; i < SUBFRAMES; i++) {
898  gen_fcb_excitation(vector_ptr, &p->subframe[i], p->cur_rate,
899  p->pitch_lag[i >> 1], i);
900  ff_g723_1_gen_acb_excitation(acb_vector,
901  &p->excitation[SUBFRAME_LEN * i],
902  p->pitch_lag[i >> 1],
903  &p->subframe[i], p->cur_rate);
904  /* Get the total excitation */
905  for (j = 0; j < SUBFRAME_LEN; j++) {
906  int v = av_clip_int16(vector_ptr[j] << 1);
907  vector_ptr[j] = av_clip_int16(v + acb_vector[j]);
908  }
909  vector_ptr += SUBFRAME_LEN;
910  }
911 
912  vector_ptr = p->excitation + PITCH_MAX;
913 
915  &p->sid_gain, &p->cur_gain);
916 
917  /* Perform pitch postfiltering */
918  if (p->postfilter) {
919  i = PITCH_MAX;
920  for (j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
921  comp_ppf_coeff(p, i, p->pitch_lag[j >> 1],
922  ppf + j, p->cur_rate);
923 
924  for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
926  vector_ptr + i,
927  vector_ptr + i + ppf[j].index,
928  ppf[j].sc_gain,
929  ppf[j].opt_gain,
930  1 << 14, 15, SUBFRAME_LEN);
931  } else {
932  audio = vector_ptr - LPC_ORDER;
933  }
934 
935  /* Save the excitation for the next frame */
936  memcpy(p->prev_excitation, p->excitation + FRAME_LEN,
937  PITCH_MAX * sizeof(*p->excitation));
938  } else {
939  p->interp_gain = (p->interp_gain * 3 + 2) >> 2;
940  if (p->erased_frames == 3) {
941  /* Mute output */
942  memset(p->excitation, 0,
943  (FRAME_LEN + PITCH_MAX) * sizeof(*p->excitation));
944  memset(p->prev_excitation, 0,
945  PITCH_MAX * sizeof(*p->excitation));
946  memset(frame->data[0], 0,
947  (FRAME_LEN + LPC_ORDER) * sizeof(int16_t));
948  } else {
949  int16_t *buf = p->audio + LPC_ORDER;
950 
951  /* Regenerate frame */
953  p->interp_gain, &p->random_seed);
954 
955  /* Save the excitation for the next frame */
956  memcpy(p->prev_excitation, buf + (FRAME_LEN - PITCH_MAX),
957  PITCH_MAX * sizeof(*p->excitation));
958  }
959  }
961  } else {
962  if (p->cur_frame_type == SID_FRAME) {
965  } else if (p->past_frame_type == ACTIVE_FRAME) {
966  p->sid_gain = estimate_sid_gain(p);
967  }
968 
969  if (p->past_frame_type == ACTIVE_FRAME)
970  p->cur_gain = p->sid_gain;
971  else
972  p->cur_gain = (p->cur_gain * 7 + p->sid_gain) >> 3;
973  generate_noise(p);
975  /* Save the lsp_vector for the next frame */
976  memcpy(p->prev_lsp, p->sid_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
977  }
978 
980 
981  memcpy(p->audio, p->synth_mem, LPC_ORDER * sizeof(*p->audio));
982  for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
983  ff_celp_lp_synthesis_filter(p->audio + i, &lpc[j * LPC_ORDER],
984  audio + i, SUBFRAME_LEN, LPC_ORDER,
985  0, 1, 1 << 12);
986  memcpy(p->synth_mem, p->audio + FRAME_LEN, LPC_ORDER * sizeof(*p->audio));
987 
988  if (p->postfilter) {
989  formant_postfilter(p, lpc, p->audio, out);
990  } else { // if output is not postfiltered it should be scaled by 2
991  for (i = 0; i < FRAME_LEN; i++)
992  out[i] = av_clip_int16(p->audio[LPC_ORDER + i] << 1);
993  }
994 
995  *got_frame_ptr = 1;
996 
997  return frame_size[dec_mode];
998 }
999 
1000 #define OFFSET(x) offsetof(G723_1_Context, x)
1001 #define AD AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM
1002 
1003 static const AVOption options[] = {
1004  { "postfilter", "enable postfilter", OFFSET(postfilter), AV_OPT_TYPE_BOOL,
1005  { .i64 = 1 }, 0, 1, AD },
1006  { NULL }
1007 };
1008 
1009 
1010 static const AVClass g723_1dec_class = {
1011  .class_name = "G.723.1 decoder",
1012  .item_name = av_default_item_name,
1013  .option = options,
1014  .version = LIBAVUTIL_VERSION_INT,
1015 };
1016 
1018  .name = "g723_1",
1019  .long_name = NULL_IF_CONFIG_SMALL("G.723.1"),
1020  .type = AVMEDIA_TYPE_AUDIO,
1021  .id = AV_CODEC_ID_G723_1,
1022  .priv_data_size = sizeof(G723_1_Context),
1025  .capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
1026  .priv_class = &g723_1dec_class,
1027 };
int16_t audio[FRAME_LEN+LPC_ORDER+PITCH_MAX+4]
Definition: g723_1.h:149
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:298
#define NULL
Definition: coverity.c:32
const char const char void * val
Definition: avisynth_c.h:634
int cur_gain
Definition: g723_1.h:143
static int shift(int a, int b)
Definition: sonic.c:82
int erased_frames
Definition: g723_1.h:128
int dirac_train
Definition: g723_1.h:83
This structure describes decoded (raw) audio or video data.
Definition: frame.h:184
int reflection_coef
Definition: g723_1.h:144
int ad_cb_gain
Definition: g723_1.h:82
AVOption.
Definition: opt.h:245
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:101
int pitch_lag[2]
Definition: g723_1.h:127
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:198
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:247
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
static const int32_t max_pos[4]
Size of the MP-MLQ fixed excitation codebooks.
Definition: g723_1.h:725
#define LIBAVUTIL_VERSION_INT
Definition: version.h:70
memory handling functions
else temp
Definition: vf_mcdeint.c:259
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
G723.1 unpacked data subframe.
Definition: g723_1.h:80
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
int16_t fir_mem[LPC_ORDER]
Definition: g723_1.h:135
int16_t excitation[PITCH_MAX+FRAME_LEN+4]
Definition: g723_1.h:133
static const AVClass g723_1dec_class
Definition: g723_1dec.c:1010
static const int8_t pulses[4]
Number of non-zero pulses in the MP-MLQ excitation.
Definition: g723_1.h:720
int size
Definition: avcodec.h:1581
int av_log2(unsigned v)
Definition: intmath.c:26
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:477
AVCodec.
Definition: avcodec.h:3542
#define PITCH_MIN
Definition: g723_1.h:43
static int unpack_bitstream(G723_1_Context *p, const uint8_t *buf, int buf_size)
Unpack the frame into parameters.
Definition: g723_1dec.c:67
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
enum FrameType past_frame_type
Definition: g723_1.h:124
#define FRAME_LEN
Definition: g723_1.h:37
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
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:201
static const int cng_filt[4]
Definition: g723_1.h:1437
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2418
uint8_t
#define av_cold
Definition: attributes.h:82
#define av_assert2(cond)
assert() equivalent, that does lie in speed critical code.
Definition: avassert.h:63
float delta
static int comp_interp_index(G723_1_Context *p, int pitch_lag, int *exc_eng, int *scale)
Classify frames as voiced/unvoiced.
Definition: g723_1dec.c:432
AVOptions.
#define LPC_ORDER
Definition: g723_1.h:40
Rate
G723.1 rate values.
Definition: g723_1.h:72
static AVFrame * frame
int pulse_sign
Definition: g723_1.h:84
static void comp_ppf_coeff(G723_1_Context *p, int offset, int pitch_lag, PPFParam *ppf, enum Rate cur_rate)
Calculate pitch postfilter parameters.
Definition: g723_1dec.c:346
uint8_t * data
Definition: avcodec.h:1580
static const uint8_t bits2[81]
Definition: aactab.c:130
bitstream reader API header.
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:180
#define GRID_SIZE
Definition: g723_1.h:46
#define av_log(a,...)
#define ff_sqrt
Definition: mathops.h:215
static const int32_t 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.h:627
int16_t sid_lsp[LPC_ORDER]
Definition: g723_1.h:131
av_default_item_name
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:49
int amp_index
Definition: g723_1.h:86
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:176
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:74
static void gain_scale(G723_1_Context *p, int16_t *buf, int energy)
Adjust gain of postfiltered signal.
Definition: g723_1dec.c:531
GLsizei GLsizei * length
Definition: opengl_enc.c:115
int grid_index
Definition: g723_1.h:85
const char * name
Name of the codec implementation.
Definition: avcodec.h:3549
int16_t prev_excitation[PITCH_MAX]
Definition: g723_1.h:132
static const uint8_t offset[127][2]
Definition: vf_spp.c:92
#define FFMAX(a, b)
Definition: common.h:94
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:2461
int interp_index
Definition: g723_1.h:140
static int estimate_sid_gain(G723_1_Context *p)
Definition: g723_1dec.c:657
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:86
G723_1_Subframe subframe[4]
Definition: g723_1.h:122
#define PITCH_MAX
Definition: g723_1.h:44
static const int16_t fixed_cb_gain[GAIN_LEVELS]
Definition: g723_1.h:727
enum Rate cur_rate
Definition: g723_1.h:125
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 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_1.h:1380
audio channel layout utility functions
int16_t synth_mem[LPC_ORDER]
Definition: g723_1.h:134
#define FFMIN(a, b)
Definition: common.h:96
AVCodec ff_g723_1_decoder
Definition: g723_1dec.c:1017
static const int cng_adaptive_cb_lag[4]
Definition: g723_1.h:1435
int ff_g723_1_dot_product(const int16_t *a, const int16_t *b, int length)
Definition: g723_1.c:54
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
Definition: common.h:72
#define OFFSET(x)
Definition: g723_1dec.c:1000
int index
postfilter backward/forward lag
Definition: g723_1.h:94
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:265
int sid_gain
Definition: g723_1.h:142
#define GAIN_LEVELS
Definition: g723_1.h:48
#define iir_filter(fir_coef, iir_coef, src, dest, width)
Perform IIR filtering.
Definition: g723_1dec.c:506
int16_t opt_gain
optimal gain
Definition: g723_1.h:95
int postfilter
Definition: g723_1.h:147
int frame_size
Definition: mxfenc.c:1821
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:32
Libavcodec external API header.
static const int16_t dc_lsp[LPC_ORDER]
LSP DC component.
Definition: g723_1.h:229
static const int16_t pitch_contrib[340]
Definition: g723_1.h:671
main external API structure.
Definition: avcodec.h:1649
static const int16_t ppf_gain_weight[2]
Postfilter gain weighting factors scaled by 2^15.
Definition: g723_1.h:224
static int sid_gain_to_lsp_index(int gain)
Definition: g723_1dec.c:641
#define FASTDIV(a, b)
Definition: mathops.h:211
Silence Insertion Descriptor frame.
Definition: g723_1.h:65
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: utils.c:928
G.723.1 types, functions and data tables.
void * buf
Definition: avisynth_c.h:553
static void formant_postfilter(G723_1_Context *p, int16_t *lpc, int16_t *buf, int16_t *dst)
Perform formant filtering.
Definition: g723_1dec.c:574
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:299
static void skip_bits1(GetBitContext *s)
Definition: get_bits.h:324
Describe the class of an AVClass context structure.
Definition: log.h:67
#define PULSE_MAX
Definition: dss_sp.c:32
int16_t sc_gain
scaling gain
Definition: g723_1.h:96
int index
Definition: gxfenc.c:89
#define AV_CODEC_CAP_SUBFRAMES
Codec can output multiple frames per AVPacket Normally demuxers return one frame at a time...
Definition: avcodec.h:1006
int cng_random_seed
Definition: g723_1.h:139
int random_seed
Definition: g723_1.h:138
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
Definition: get_bits.h:406
Active speech.
Definition: g723_1.h:64
#define CNG_RANDOM_SEED
Definition: g723_1dec.c:40
enum FrameType cur_frame_type
Definition: g723_1.h:123
#define SUBFRAME_LEN
Definition: g723_1.h:36
static struct @228 state
int16_t prev_lsp[LPC_ORDER]
Definition: g723_1.h:130
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:198
int pf_gain
formant postfilter gain scaling unit memory
Definition: g723_1.h:145
#define SUBFRAMES
Definition: dcaenc.c:42
#define AD
Definition: g723_1dec.c:1001
static int decode(AVCodecContext *avctx, void *data, int *got_sub, AVPacket *avpkt)
Definition: ccaption_dec.c:722
common internal api header.
if(ret< 0)
Definition: vf_mcdeint.c:282
Pitch postfilter parameters.
Definition: g723_1.h:93
signed 16 bits
Definition: samplefmt.h:61
static double c[64]
void * priv_data
Definition: avcodec.h:1691
static const int cng_bseg[3]
Definition: g723_1.h:1439
int channels
number of audio channels
Definition: avcodec.h:2411
static int16_t square_root(unsigned val)
Bitexact implementation of sqrt(val/2).
Definition: g723_1dec.c:182
static uint8_t tmp[8]
Definition: des.c:38
static const AVOption options[]
Definition: g723_1dec.c:1003
uint8_t lsp_index[LSP_BANDS]
Definition: g723_1.h:126
int pulse_pos
Definition: g723_1.h:87
int iir_mem[LPC_ORDER]
Definition: g723_1.h:136
static av_cold int g723_1_decode_init(AVCodecContext *avctx)
Definition: g723_1dec.c:42
FILE * out
Definition: movenc.c:54
void av_memcpy_backptr(uint8_t *dst, int back, int cnt)
deliberately overlapping memcpy implementation
Definition: mem.c:430
static int cng_rand(int *state, int base)
Definition: g723_1dec.c:651
int interp_gain
Definition: g723_1.h:141
static void generate_noise(G723_1_Context *p)
Definition: g723_1dec.c:711
#define AV_CH_LAYOUT_MONO
This structure stores compressed data.
Definition: avcodec.h:1557
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:241
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: avcodec.h:956
static int g723_1_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: g723_1dec.c:836
static const uint8_t bits1[81]
Definition: aactab.c:107
int ad_cb_lag
adaptive codebook lag
Definition: g723_1.h:81