FFmpeg
 All Data Structures Namespaces Files Functions Variables Typedefs Enumerations Enumerator Macros Groups Pages
g723_1dec.c
Go to the documentation of this file.
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 
44 {
45  G723_1_Context *p = avctx->priv_data;
46 
49  avctx->channels = 1;
50  p->pf_gain = 1 << 12;
51 
52  memcpy(p->prev_lsp, dc_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
53  memcpy(p->sid_lsp, dc_lsp, LPC_ORDER * sizeof(*p->sid_lsp));
54 
57 
58  return 0;
59 }
60 
61 /**
62  * Unpack the frame into parameters.
63  *
64  * @param p the context
65  * @param buf pointer to the input buffer
66  * @param buf_size size of the input buffer
67  */
69  int buf_size)
70 {
71  GetBitContext gb;
72  int ad_cb_len;
73  int temp, info_bits, i;
74 
75  init_get_bits(&gb, buf, buf_size * 8);
76 
77  /* Extract frame type and rate info */
78  info_bits = get_bits(&gb, 2);
79 
80  if (info_bits == 3) {
82  return 0;
83  }
84 
85  /* Extract 24 bit lsp indices, 8 bit for each band */
86  p->lsp_index[2] = get_bits(&gb, 8);
87  p->lsp_index[1] = get_bits(&gb, 8);
88  p->lsp_index[0] = get_bits(&gb, 8);
89 
90  if (info_bits == 2) {
92  p->subframe[0].amp_index = get_bits(&gb, 6);
93  return 0;
94  }
95 
96  /* Extract the info common to both rates */
97  p->cur_rate = info_bits ? RATE_5300 : RATE_6300;
99 
100  p->pitch_lag[0] = get_bits(&gb, 7);
101  if (p->pitch_lag[0] > 123) /* test if forbidden code */
102  return -1;
103  p->pitch_lag[0] += PITCH_MIN;
104  p->subframe[1].ad_cb_lag = get_bits(&gb, 2);
105 
106  p->pitch_lag[1] = get_bits(&gb, 7);
107  if (p->pitch_lag[1] > 123)
108  return -1;
109  p->pitch_lag[1] += PITCH_MIN;
110  p->subframe[3].ad_cb_lag = get_bits(&gb, 2);
111  p->subframe[0].ad_cb_lag = 1;
112  p->subframe[2].ad_cb_lag = 1;
113 
114  for (i = 0; i < SUBFRAMES; i++) {
115  /* Extract combined gain */
116  temp = get_bits(&gb, 12);
117  ad_cb_len = 170;
118  p->subframe[i].dirac_train = 0;
119  if (p->cur_rate == RATE_6300 && p->pitch_lag[i >> 1] < SUBFRAME_LEN - 2) {
120  p->subframe[i].dirac_train = temp >> 11;
121  temp &= 0x7FF;
122  ad_cb_len = 85;
123  }
124  p->subframe[i].ad_cb_gain = FASTDIV(temp, GAIN_LEVELS);
125  if (p->subframe[i].ad_cb_gain < ad_cb_len) {
126  p->subframe[i].amp_index = temp - p->subframe[i].ad_cb_gain *
127  GAIN_LEVELS;
128  } else {
129  return -1;
130  }
131  }
132 
133  p->subframe[0].grid_index = get_bits1(&gb);
134  p->subframe[1].grid_index = get_bits1(&gb);
135  p->subframe[2].grid_index = get_bits1(&gb);
136  p->subframe[3].grid_index = get_bits1(&gb);
137 
138  if (p->cur_rate == RATE_6300) {
139  skip_bits1(&gb); /* skip reserved bit */
140 
141  /* Compute pulse_pos index using the 13-bit combined position index */
142  temp = get_bits(&gb, 13);
143  p->subframe[0].pulse_pos = temp / 810;
144 
145  temp -= p->subframe[0].pulse_pos * 810;
146  p->subframe[1].pulse_pos = FASTDIV(temp, 90);
147 
148  temp -= p->subframe[1].pulse_pos * 90;
149  p->subframe[2].pulse_pos = FASTDIV(temp, 9);
150  p->subframe[3].pulse_pos = temp - p->subframe[2].pulse_pos * 9;
151 
152  p->subframe[0].pulse_pos = (p->subframe[0].pulse_pos << 16) +
153  get_bits(&gb, 16);
154  p->subframe[1].pulse_pos = (p->subframe[1].pulse_pos << 14) +
155  get_bits(&gb, 14);
156  p->subframe[2].pulse_pos = (p->subframe[2].pulse_pos << 16) +
157  get_bits(&gb, 16);
158  p->subframe[3].pulse_pos = (p->subframe[3].pulse_pos << 14) +
159  get_bits(&gb, 14);
160 
161  p->subframe[0].pulse_sign = get_bits(&gb, 6);
162  p->subframe[1].pulse_sign = get_bits(&gb, 5);
163  p->subframe[2].pulse_sign = get_bits(&gb, 6);
164  p->subframe[3].pulse_sign = get_bits(&gb, 5);
165  } else { /* 5300 bps */
166  p->subframe[0].pulse_pos = get_bits(&gb, 12);
167  p->subframe[1].pulse_pos = get_bits(&gb, 12);
168  p->subframe[2].pulse_pos = get_bits(&gb, 12);
169  p->subframe[3].pulse_pos = get_bits(&gb, 12);
170 
171  p->subframe[0].pulse_sign = get_bits(&gb, 4);
172  p->subframe[1].pulse_sign = get_bits(&gb, 4);
173  p->subframe[2].pulse_sign = get_bits(&gb, 4);
174  p->subframe[3].pulse_sign = get_bits(&gb, 4);
175  }
176 
177  return 0;
178 }
179 
180 /**
181  * Bitexact implementation of sqrt(val/2).
182  */
183 static int16_t square_root(unsigned val)
184 {
185  av_assert2(!(val & 0x80000000));
186 
187  return (ff_sqrt(val << 1) >> 1) & (~1);
188 }
189 
190 /**
191  * Generate fixed codebook excitation vector.
192  *
193  * @param vector decoded excitation vector
194  * @param subfrm current subframe
195  * @param cur_rate current bitrate
196  * @param pitch_lag closed loop pitch lag
197  * @param index current subframe index
198  */
199 static void gen_fcb_excitation(int16_t *vector, G723_1_Subframe *subfrm,
200  enum Rate cur_rate, int pitch_lag, int index)
201 {
202  int temp, i, j;
203 
204  memset(vector, 0, SUBFRAME_LEN * sizeof(*vector));
205 
206  if (cur_rate == RATE_6300) {
207  if (subfrm->pulse_pos >= max_pos[index])
208  return;
209 
210  /* Decode amplitudes and positions */
211  j = PULSE_MAX - pulses[index];
212  temp = subfrm->pulse_pos;
213  for (i = 0; i < SUBFRAME_LEN / GRID_SIZE; i++) {
214  temp -= combinatorial_table[j][i];
215  if (temp >= 0)
216  continue;
217  temp += combinatorial_table[j++][i];
218  if (subfrm->pulse_sign & (1 << (PULSE_MAX - j))) {
219  vector[subfrm->grid_index + GRID_SIZE * i] =
220  -fixed_cb_gain[subfrm->amp_index];
221  } else {
222  vector[subfrm->grid_index + GRID_SIZE * i] =
223  fixed_cb_gain[subfrm->amp_index];
224  }
225  if (j == PULSE_MAX)
226  break;
227  }
228  if (subfrm->dirac_train == 1)
229  ff_g723_1_gen_dirac_train(vector, pitch_lag);
230  } else { /* 5300 bps */
231  int cb_gain = fixed_cb_gain[subfrm->amp_index];
232  int cb_shift = subfrm->grid_index;
233  int cb_sign = subfrm->pulse_sign;
234  int cb_pos = subfrm->pulse_pos;
235  int offset, beta, lag;
236 
237  for (i = 0; i < 8; i += 2) {
238  offset = ((cb_pos & 7) << 3) + cb_shift + i;
239  vector[offset] = (cb_sign & 1) ? cb_gain : -cb_gain;
240  cb_pos >>= 3;
241  cb_sign >>= 1;
242  }
243 
244  /* Enhance harmonic components */
245  lag = pitch_contrib[subfrm->ad_cb_gain << 1] + pitch_lag +
246  subfrm->ad_cb_lag - 1;
247  beta = pitch_contrib[(subfrm->ad_cb_gain << 1) + 1];
248 
249  if (lag < SUBFRAME_LEN - 2) {
250  for (i = lag; i < SUBFRAME_LEN; i++)
251  vector[i] += beta * vector[i - lag] >> 15;
252  }
253  }
254 }
255 
256 /**
257  * Estimate maximum auto-correlation around pitch lag.
258  *
259  * @param buf buffer with offset applied
260  * @param offset offset of the excitation vector
261  * @param ccr_max pointer to the maximum auto-correlation
262  * @param pitch_lag decoded pitch lag
263  * @param length length of autocorrelation
264  * @param dir forward lag(1) / backward lag(-1)
265  */
266 static int autocorr_max(const int16_t *buf, int offset, int *ccr_max,
267  int pitch_lag, int length, int dir)
268 {
269  int limit, ccr, lag = 0;
270  int i;
271 
272  pitch_lag = FFMIN(PITCH_MAX - 3, pitch_lag);
273  if (dir > 0)
274  limit = FFMIN(FRAME_LEN + PITCH_MAX - offset - length, pitch_lag + 3);
275  else
276  limit = pitch_lag + 3;
277 
278  for (i = pitch_lag - 3; i <= limit; i++) {
279  ccr = ff_g723_1_dot_product(buf, buf + dir * i, length);
280 
281  if (ccr > *ccr_max) {
282  *ccr_max = ccr;
283  lag = i;
284  }
285  }
286  return lag;
287 }
288 
289 /**
290  * Calculate pitch postfilter optimal and scaling gains.
291  *
292  * @param lag pitch postfilter forward/backward lag
293  * @param ppf pitch postfilter parameters
294  * @param cur_rate current bitrate
295  * @param tgt_eng target energy
296  * @param ccr cross-correlation
297  * @param res_eng residual energy
298  */
299 static void comp_ppf_gains(int lag, PPFParam *ppf, enum Rate cur_rate,
300  int tgt_eng, int ccr, int res_eng)
301 {
302  int pf_residual; /* square of postfiltered residual */
303  int temp1, temp2;
304 
305  ppf->index = lag;
306 
307  temp1 = tgt_eng * res_eng >> 1;
308  temp2 = ccr * ccr << 1;
309 
310  if (temp2 > temp1) {
311  if (ccr >= res_eng) {
312  ppf->opt_gain = ppf_gain_weight[cur_rate];
313  } else {
314  ppf->opt_gain = (ccr << 15) / res_eng *
315  ppf_gain_weight[cur_rate] >> 15;
316  }
317  /* pf_res^2 = tgt_eng + 2*ccr*gain + res_eng*gain^2 */
318  temp1 = (tgt_eng << 15) + (ccr * ppf->opt_gain << 1);
319  temp2 = (ppf->opt_gain * ppf->opt_gain >> 15) * res_eng;
320  pf_residual = av_sat_add32(temp1, temp2 + (1 << 15)) >> 16;
321 
322  if (tgt_eng >= pf_residual << 1) {
323  temp1 = 0x7fff;
324  } else {
325  temp1 = (tgt_eng << 14) / pf_residual;
326  }
327 
328  /* scaling_gain = sqrt(tgt_eng/pf_res^2) */
329  ppf->sc_gain = square_root(temp1 << 16);
330  } else {
331  ppf->opt_gain = 0;
332  ppf->sc_gain = 0x7fff;
333  }
334 
335  ppf->opt_gain = av_clip_int16(ppf->opt_gain * ppf->sc_gain >> 15);
336 }
337 
338 /**
339  * Calculate pitch postfilter parameters.
340  *
341  * @param p the context
342  * @param offset offset of the excitation vector
343  * @param pitch_lag decoded pitch lag
344  * @param ppf pitch postfilter parameters
345  * @param cur_rate current bitrate
346  */
347 static void comp_ppf_coeff(G723_1_Context *p, int offset, int pitch_lag,
348  PPFParam *ppf, enum Rate cur_rate)
349 {
350 
351  int16_t scale;
352  int i;
353  int temp1, temp2;
354 
355  /*
356  * 0 - target energy
357  * 1 - forward cross-correlation
358  * 2 - forward residual energy
359  * 3 - backward cross-correlation
360  * 4 - backward residual energy
361  */
362  int energy[5] = {0, 0, 0, 0, 0};
363  int16_t *buf = p->audio + LPC_ORDER + offset;
364  int fwd_lag = autocorr_max(buf, offset, &energy[1], pitch_lag,
365  SUBFRAME_LEN, 1);
366  int back_lag = autocorr_max(buf, offset, &energy[3], pitch_lag,
367  SUBFRAME_LEN, -1);
368 
369  ppf->index = 0;
370  ppf->opt_gain = 0;
371  ppf->sc_gain = 0x7fff;
372 
373  /* Case 0, Section 3.6 */
374  if (!back_lag && !fwd_lag)
375  return;
376 
377  /* Compute target energy */
378  energy[0] = ff_g723_1_dot_product(buf, buf, SUBFRAME_LEN);
379 
380  /* Compute forward residual energy */
381  if (fwd_lag)
382  energy[2] = ff_g723_1_dot_product(buf + fwd_lag, buf + fwd_lag,
383  SUBFRAME_LEN);
384 
385  /* Compute backward residual energy */
386  if (back_lag)
387  energy[4] = ff_g723_1_dot_product(buf - back_lag, buf - back_lag,
388  SUBFRAME_LEN);
389 
390  /* Normalize and shorten */
391  temp1 = 0;
392  for (i = 0; i < 5; i++)
393  temp1 = FFMAX(energy[i], temp1);
394 
395  scale = ff_g723_1_normalize_bits(temp1, 31);
396  for (i = 0; i < 5; i++)
397  energy[i] = (energy[i] << scale) >> 16;
398 
399  if (fwd_lag && !back_lag) { /* Case 1 */
400  comp_ppf_gains(fwd_lag, ppf, cur_rate, energy[0], energy[1],
401  energy[2]);
402  } else if (!fwd_lag) { /* Case 2 */
403  comp_ppf_gains(-back_lag, ppf, cur_rate, energy[0], energy[3],
404  energy[4]);
405  } else { /* Case 3 */
406 
407  /*
408  * Select the largest of energy[1]^2/energy[2]
409  * and energy[3]^2/energy[4]
410  */
411  temp1 = energy[4] * ((energy[1] * energy[1] + (1 << 14)) >> 15);
412  temp2 = energy[2] * ((energy[3] * energy[3] + (1 << 14)) >> 15);
413  if (temp1 >= temp2) {
414  comp_ppf_gains(fwd_lag, ppf, cur_rate, energy[0], energy[1],
415  energy[2]);
416  } else {
417  comp_ppf_gains(-back_lag, ppf, cur_rate, energy[0], energy[3],
418  energy[4]);
419  }
420  }
421 }
422 
423 /**
424  * Classify frames as voiced/unvoiced.
425  *
426  * @param p the context
427  * @param pitch_lag decoded pitch_lag
428  * @param exc_eng excitation energy estimation
429  * @param scale scaling factor of exc_eng
430  *
431  * @return residual interpolation index if voiced, 0 otherwise
432  */
433 static int comp_interp_index(G723_1_Context *p, int pitch_lag,
434  int *exc_eng, int *scale)
435 {
436  int offset = PITCH_MAX + 2 * SUBFRAME_LEN;
437  int16_t *buf = p->audio + LPC_ORDER;
438 
439  int index, ccr, tgt_eng, best_eng, temp;
440 
442  buf += offset;
443 
444  /* Compute maximum backward cross-correlation */
445  ccr = 0;
446  index = autocorr_max(buf, offset, &ccr, pitch_lag, SUBFRAME_LEN * 2, -1);
447  ccr = av_sat_add32(ccr, 1 << 15) >> 16;
448 
449  /* Compute target energy */
450  tgt_eng = ff_g723_1_dot_product(buf, buf, SUBFRAME_LEN * 2);
451  *exc_eng = av_sat_add32(tgt_eng, 1 << 15) >> 16;
452 
453  if (ccr <= 0)
454  return 0;
455 
456  /* Compute best energy */
457  best_eng = ff_g723_1_dot_product(buf - index, buf - index,
458  SUBFRAME_LEN * 2);
459  best_eng = av_sat_add32(best_eng, 1 << 15) >> 16;
460 
461  temp = best_eng * *exc_eng >> 3;
462 
463  if (temp < ccr * ccr) {
464  return index;
465  } else
466  return 0;
467 }
468 
469 /**
470  * Perform residual interpolation based on frame classification.
471  *
472  * @param buf decoded excitation vector
473  * @param out output vector
474  * @param lag decoded pitch lag
475  * @param gain interpolated gain
476  * @param rseed seed for random number generator
477  */
478 static void residual_interp(int16_t *buf, int16_t *out, int lag,
479  int gain, int *rseed)
480 {
481  int i;
482  if (lag) { /* Voiced */
483  int16_t *vector_ptr = buf + PITCH_MAX;
484  /* Attenuate */
485  for (i = 0; i < lag; i++)
486  out[i] = vector_ptr[i - lag] * 3 >> 2;
487  av_memcpy_backptr((uint8_t*)(out + lag), lag * sizeof(*out),
488  (FRAME_LEN - lag) * sizeof(*out));
489  } else { /* Unvoiced */
490  for (i = 0; i < FRAME_LEN; i++) {
491  *rseed = (int16_t)(*rseed * 521 + 259);
492  out[i] = gain * *rseed >> 15;
493  }
494  memset(buf, 0, (FRAME_LEN + PITCH_MAX) * sizeof(*buf));
495  }
496 }
497 
498 /**
499  * Perform IIR filtering.
500  *
501  * @param fir_coef FIR coefficients
502  * @param iir_coef IIR coefficients
503  * @param src source vector
504  * @param dest destination vector
505  * @param width width of the output, 16 bits(0) / 32 bits(1)
506  */
507 #define iir_filter(fir_coef, iir_coef, src, dest, width)\
508 {\
509  int m, n;\
510  int res_shift = 16 & ~-(width);\
511  int in_shift = 16 - res_shift;\
512 \
513  for (m = 0; m < SUBFRAME_LEN; m++) {\
514  int64_t filter = 0;\
515  for (n = 1; n <= LPC_ORDER; n++) {\
516  filter -= (fir_coef)[n - 1] * (src)[m - n] -\
517  (iir_coef)[n - 1] * ((dest)[m - n] >> in_shift);\
518  }\
519 \
520  (dest)[m] = av_clipl_int32(((src)[m] * 65536) + (filter * 8) +\
521  (1 << 15)) >> res_shift;\
522  }\
523 }
524 
525 /**
526  * Adjust gain of postfiltered signal.
527  *
528  * @param p the context
529  * @param buf postfiltered output vector
530  * @param energy input energy coefficient
531  */
532 static void gain_scale(G723_1_Context *p, int16_t * buf, int energy)
533 {
534  int num, denom, gain, bits1, bits2;
535  int i;
536 
537  num = energy;
538  denom = 0;
539  for (i = 0; i < SUBFRAME_LEN; i++) {
540  int temp = buf[i] >> 2;
541  temp *= temp;
542  denom = av_sat_dadd32(denom, temp);
543  }
544 
545  if (num && denom) {
546  bits1 = ff_g723_1_normalize_bits(num, 31);
547  bits2 = ff_g723_1_normalize_bits(denom, 31);
548  num = num << bits1 >> 1;
549  denom <<= bits2;
550 
551  bits2 = 5 + bits1 - bits2;
552  bits2 = FFMAX(0, bits2);
553 
554  gain = (num >> 1) / (denom >> 16);
555  gain = square_root(gain << 16 >> bits2);
556  } else {
557  gain = 1 << 12;
558  }
559 
560  for (i = 0; i < SUBFRAME_LEN; i++) {
561  p->pf_gain = (15 * p->pf_gain + gain + (1 << 3)) >> 4;
562  buf[i] = av_clip_int16((buf[i] * (p->pf_gain + (p->pf_gain >> 4)) +
563  (1 << 10)) >> 11);
564  }
565 }
566 
567 /**
568  * Perform formant filtering.
569  *
570  * @param p the context
571  * @param lpc quantized lpc coefficients
572  * @param buf input buffer
573  * @param dst output buffer
574  */
575 static void formant_postfilter(G723_1_Context *p, int16_t *lpc,
576  int16_t *buf, int16_t *dst)
577 {
578  int16_t filter_coef[2][LPC_ORDER];
579  int filter_signal[LPC_ORDER + FRAME_LEN], *signal_ptr;
580  int i, j, k;
581 
582  memcpy(buf, p->fir_mem, LPC_ORDER * sizeof(*buf));
583  memcpy(filter_signal, p->iir_mem, LPC_ORDER * sizeof(*filter_signal));
584 
585  for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) {
586  for (k = 0; k < LPC_ORDER; k++) {
587  filter_coef[0][k] = (-lpc[k] * postfilter_tbl[0][k] +
588  (1 << 14)) >> 15;
589  filter_coef[1][k] = (-lpc[k] * postfilter_tbl[1][k] +
590  (1 << 14)) >> 15;
591  }
592  iir_filter(filter_coef[0], filter_coef[1], buf + i, filter_signal + i, 1);
593  lpc += LPC_ORDER;
594  }
595 
596  memcpy(p->fir_mem, buf + FRAME_LEN, LPC_ORDER * sizeof(int16_t));
597  memcpy(p->iir_mem, filter_signal + FRAME_LEN, LPC_ORDER * sizeof(int));
598 
599  buf += LPC_ORDER;
600  signal_ptr = filter_signal + LPC_ORDER;
601  for (i = 0; i < SUBFRAMES; i++) {
602  int temp;
603  int auto_corr[2];
604  int scale, energy;
605 
606  /* Normalize */
607  scale = ff_g723_1_scale_vector(dst, buf, SUBFRAME_LEN);
608 
609  /* Compute auto correlation coefficients */
610  auto_corr[0] = ff_g723_1_dot_product(dst, dst + 1, SUBFRAME_LEN - 1);
611  auto_corr[1] = ff_g723_1_dot_product(dst, dst, SUBFRAME_LEN);
612 
613  /* Compute reflection coefficient */
614  temp = auto_corr[1] >> 16;
615  if (temp) {
616  temp = (auto_corr[0] >> 2) / temp;
617  }
618  p->reflection_coef = (3 * p->reflection_coef + temp + 2) >> 2;
619  temp = -p->reflection_coef >> 1 & ~3;
620 
621  /* Compensation filter */
622  for (j = 0; j < SUBFRAME_LEN; j++) {
623  dst[j] = av_sat_dadd32(signal_ptr[j],
624  (signal_ptr[j - 1] >> 16) * temp) >> 16;
625  }
626 
627  /* Compute normalized signal energy */
628  temp = 2 * scale + 4;
629  if (temp < 0) {
630  energy = av_clipl_int32((int64_t)auto_corr[1] << -temp);
631  } else
632  energy = auto_corr[1] >> temp;
633 
634  gain_scale(p, dst, energy);
635 
636  buf += SUBFRAME_LEN;
637  signal_ptr += SUBFRAME_LEN;
638  dst += SUBFRAME_LEN;
639  }
640 }
641 
642 static int sid_gain_to_lsp_index(int gain)
643 {
644  if (gain < 0x10)
645  return gain << 6;
646  else if (gain < 0x20)
647  return gain - 8 << 7;
648  else
649  return gain - 20 << 8;
650 }
651 
652 static inline int cng_rand(int *state, int base)
653 {
654  *state = (*state * 521 + 259) & 0xFFFF;
655  return (*state & 0x7FFF) * base >> 15;
656 }
657 
659 {
660  int i, shift, seg, seg2, t, val, val_add, x, y;
661 
662  shift = 16 - p->cur_gain * 2;
663  if (shift > 0) {
664  if (p->sid_gain == 0) {
665  t = 0;
666  } else if (shift >= 31 || (int32_t)((uint32_t)p->sid_gain << shift) >> shift != p->sid_gain) {
667  if (p->sid_gain < 0) t = INT32_MIN;
668  else t = INT32_MAX;
669  } else
670  t = p->sid_gain << shift;
671  }else
672  t = p->sid_gain >> -shift;
673  x = av_clipl_int32(t * (int64_t)cng_filt[0] >> 16);
674 
675  if (x >= cng_bseg[2])
676  return 0x3F;
677 
678  if (x >= cng_bseg[1]) {
679  shift = 4;
680  seg = 3;
681  } else {
682  shift = 3;
683  seg = (x >= cng_bseg[0]);
684  }
685  seg2 = FFMIN(seg, 3);
686 
687  val = 1 << shift;
688  val_add = val >> 1;
689  for (i = 0; i < shift; i++) {
690  t = seg * 32 + (val << seg2);
691  t *= t;
692  if (x >= t)
693  val += val_add;
694  else
695  val -= val_add;
696  val_add >>= 1;
697  }
698 
699  t = seg * 32 + (val << seg2);
700  y = t * t - x;
701  if (y <= 0) {
702  t = seg * 32 + (val + 1 << seg2);
703  t = t * t - x;
704  val = (seg2 - 1) * 16 + val;
705  if (t >= y)
706  val++;
707  } else {
708  t = seg * 32 + (val - 1 << seg2);
709  t = t * t - x;
710  val = (seg2 - 1) * 16 + val;
711  if (t >= y)
712  val--;
713  }
714 
715  return val;
716 }
717 
719 {
720  int i, j, idx, t;
721  int off[SUBFRAMES];
722  int signs[SUBFRAMES / 2 * 11], pos[SUBFRAMES / 2 * 11];
723  int tmp[SUBFRAME_LEN * 2];
724  int16_t *vector_ptr;
725  int64_t sum;
726  int b0, c, delta, x, shift;
727 
728  p->pitch_lag[0] = cng_rand(&p->cng_random_seed, 21) + 123;
729  p->pitch_lag[1] = cng_rand(&p->cng_random_seed, 19) + 123;
730 
731  for (i = 0; i < SUBFRAMES; i++) {
732  p->subframe[i].ad_cb_gain = cng_rand(&p->cng_random_seed, 50) + 1;
734  }
735 
736  for (i = 0; i < SUBFRAMES / 2; i++) {
737  t = cng_rand(&p->cng_random_seed, 1 << 13);
738  off[i * 2] = t & 1;
739  off[i * 2 + 1] = ((t >> 1) & 1) + SUBFRAME_LEN;
740  t >>= 2;
741  for (j = 0; j < 11; j++) {
742  signs[i * 11 + j] = ((t & 1) * 2 - 1) * (1 << 14);
743  t >>= 1;
744  }
745  }
746 
747  idx = 0;
748  for (i = 0; i < SUBFRAMES; i++) {
749  for (j = 0; j < SUBFRAME_LEN / 2; j++)
750  tmp[j] = j;
751  t = SUBFRAME_LEN / 2;
752  for (j = 0; j < pulses[i]; j++, idx++) {
753  int idx2 = cng_rand(&p->cng_random_seed, t);
754 
755  pos[idx] = tmp[idx2] * 2 + off[i];
756  tmp[idx2] = tmp[--t];
757  }
758  }
759 
760  vector_ptr = p->audio + LPC_ORDER;
761  memcpy(vector_ptr, p->prev_excitation,
762  PITCH_MAX * sizeof(*p->excitation));
763  for (i = 0; i < SUBFRAMES; i += 2) {
764  ff_g723_1_gen_acb_excitation(vector_ptr, vector_ptr,
765  p->pitch_lag[i >> 1], &p->subframe[i],
766  p->cur_rate);
768  vector_ptr + SUBFRAME_LEN,
769  p->pitch_lag[i >> 1], &p->subframe[i + 1],
770  p->cur_rate);
771 
772  t = 0;
773  for (j = 0; j < SUBFRAME_LEN * 2; j++)
774  t |= FFABS(vector_ptr[j]);
775  t = FFMIN(t, 0x7FFF);
776  if (!t) {
777  shift = 0;
778  } else {
779  shift = -10 + av_log2(t);
780  if (shift < -2)
781  shift = -2;
782  }
783  sum = 0;
784  if (shift < 0) {
785  for (j = 0; j < SUBFRAME_LEN * 2; j++) {
786  t = vector_ptr[j] * (1 << -shift);
787  sum += t * t;
788  tmp[j] = t;
789  }
790  } else {
791  for (j = 0; j < SUBFRAME_LEN * 2; j++) {
792  t = vector_ptr[j] >> shift;
793  sum += t * t;
794  tmp[j] = t;
795  }
796  }
797 
798  b0 = 0;
799  for (j = 0; j < 11; j++)
800  b0 += tmp[pos[(i / 2) * 11 + j]] * signs[(i / 2) * 11 + j];
801  b0 = b0 * 2 * 2979LL + (1 << 29) >> 30; // approximated division by 11
802 
803  c = p->cur_gain * (p->cur_gain * SUBFRAME_LEN >> 5);
804  if (shift * 2 + 3 >= 0)
805  c >>= shift * 2 + 3;
806  else
807  c <<= -(shift * 2 + 3);
808  c = (av_clipl_int32(sum << 1) - c) * 2979LL >> 15;
809 
810  delta = b0 * b0 * 2 - c;
811  if (delta <= 0) {
812  x = -b0;
813  } else {
814  delta = square_root(delta);
815  x = delta - b0;
816  t = delta + b0;
817  if (FFABS(t) < FFABS(x))
818  x = -t;
819  }
820  shift++;
821  if (shift < 0)
822  x >>= -shift;
823  else
824  x *= 1 << shift;
825  x = av_clip(x, -10000, 10000);
826 
827  for (j = 0; j < 11; j++) {
828  idx = (i / 2) * 11 + j;
829  vector_ptr[pos[idx]] = av_clip_int16(vector_ptr[pos[idx]] +
830  (x * signs[idx] >> 15));
831  }
832 
833  /* copy decoded data to serve as a history for the next decoded subframes */
834  memcpy(vector_ptr + PITCH_MAX, vector_ptr,
835  sizeof(*vector_ptr) * SUBFRAME_LEN * 2);
836  vector_ptr += SUBFRAME_LEN * 2;
837  }
838  /* Save the excitation for the next frame */
839  memcpy(p->prev_excitation, p->audio + LPC_ORDER + FRAME_LEN,
840  PITCH_MAX * sizeof(*p->excitation));
841 }
842 
843 static int g723_1_decode_frame(AVCodecContext *avctx, void *data,
844  int *got_frame_ptr, AVPacket *avpkt)
845 {
846  G723_1_Context *p = avctx->priv_data;
847  AVFrame *frame = data;
848  const uint8_t *buf = avpkt->data;
849  int buf_size = avpkt->size;
850  int dec_mode = buf[0] & 3;
851 
852  PPFParam ppf[SUBFRAMES];
853  int16_t cur_lsp[LPC_ORDER];
854  int16_t lpc[SUBFRAMES * LPC_ORDER];
855  int16_t acb_vector[SUBFRAME_LEN];
856  int16_t *out;
857  int bad_frame = 0, i, j, ret;
858  int16_t *audio = p->audio;
859 
860  if (buf_size < frame_size[dec_mode]) {
861  if (buf_size)
862  av_log(avctx, AV_LOG_WARNING,
863  "Expected %d bytes, got %d - skipping packet\n",
864  frame_size[dec_mode], buf_size);
865  *got_frame_ptr = 0;
866  return buf_size;
867  }
868 
869  if (unpack_bitstream(p, buf, buf_size) < 0) {
870  bad_frame = 1;
871  if (p->past_frame_type == ACTIVE_FRAME)
873  else
875  }
876 
877  frame->nb_samples = FRAME_LEN;
878  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
879  return ret;
880 
881  out = (int16_t *)frame->data[0];
882 
883  if (p->cur_frame_type == ACTIVE_FRAME) {
884  if (!bad_frame)
885  p->erased_frames = 0;
886  else if (p->erased_frames != 3)
887  p->erased_frames++;
888 
889  ff_g723_1_inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, bad_frame);
890  ff_g723_1_lsp_interpolate(lpc, cur_lsp, p->prev_lsp);
891 
892  /* Save the lsp_vector for the next frame */
893  memcpy(p->prev_lsp, cur_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
894 
895  /* Generate the excitation for the frame */
896  memcpy(p->excitation, p->prev_excitation,
897  PITCH_MAX * sizeof(*p->excitation));
898  if (!p->erased_frames) {
899  int16_t *vector_ptr = p->excitation + PITCH_MAX;
900 
901  /* Update interpolation gain memory */
903  p->subframe[3].amp_index) >> 1];
904  for (i = 0; i < SUBFRAMES; i++) {
905  gen_fcb_excitation(vector_ptr, &p->subframe[i], p->cur_rate,
906  p->pitch_lag[i >> 1], i);
907  ff_g723_1_gen_acb_excitation(acb_vector,
908  &p->excitation[SUBFRAME_LEN * i],
909  p->pitch_lag[i >> 1],
910  &p->subframe[i], p->cur_rate);
911  /* Get the total excitation */
912  for (j = 0; j < SUBFRAME_LEN; j++) {
913  int v = av_clip_int16(vector_ptr[j] * 2);
914  vector_ptr[j] = av_clip_int16(v + acb_vector[j]);
915  }
916  vector_ptr += SUBFRAME_LEN;
917  }
918 
919  vector_ptr = p->excitation + PITCH_MAX;
920 
922  &p->sid_gain, &p->cur_gain);
923 
924  /* Perform pitch postfiltering */
925  if (p->postfilter) {
926  i = PITCH_MAX;
927  for (j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
928  comp_ppf_coeff(p, i, p->pitch_lag[j >> 1],
929  ppf + j, p->cur_rate);
930 
931  for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
933  vector_ptr + i,
934  vector_ptr + i + ppf[j].index,
935  ppf[j].sc_gain,
936  ppf[j].opt_gain,
937  1 << 14, 15, SUBFRAME_LEN);
938  } else {
939  audio = vector_ptr - LPC_ORDER;
940  }
941 
942  /* Save the excitation for the next frame */
943  memcpy(p->prev_excitation, p->excitation + FRAME_LEN,
944  PITCH_MAX * sizeof(*p->excitation));
945  } else {
946  p->interp_gain = (p->interp_gain * 3 + 2) >> 2;
947  if (p->erased_frames == 3) {
948  /* Mute output */
949  memset(p->excitation, 0,
950  (FRAME_LEN + PITCH_MAX) * sizeof(*p->excitation));
951  memset(p->prev_excitation, 0,
952  PITCH_MAX * sizeof(*p->excitation));
953  memset(frame->data[0], 0,
954  (FRAME_LEN + LPC_ORDER) * sizeof(int16_t));
955  } else {
956  int16_t *buf = p->audio + LPC_ORDER;
957 
958  /* Regenerate frame */
960  p->interp_gain, &p->random_seed);
961 
962  /* Save the excitation for the next frame */
963  memcpy(p->prev_excitation, buf + (FRAME_LEN - PITCH_MAX),
964  PITCH_MAX * sizeof(*p->excitation));
965  }
966  }
968  } else {
969  if (p->cur_frame_type == SID_FRAME) {
972  } else if (p->past_frame_type == ACTIVE_FRAME) {
973  p->sid_gain = estimate_sid_gain(p);
974  }
975 
976  if (p->past_frame_type == ACTIVE_FRAME)
977  p->cur_gain = p->sid_gain;
978  else
979  p->cur_gain = (p->cur_gain * 7 + p->sid_gain) >> 3;
980  generate_noise(p);
982  /* Save the lsp_vector for the next frame */
983  memcpy(p->prev_lsp, p->sid_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
984  }
985 
987 
988  memcpy(p->audio, p->synth_mem, LPC_ORDER * sizeof(*p->audio));
989  for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
990  ff_celp_lp_synthesis_filter(p->audio + i, &lpc[j * LPC_ORDER],
991  audio + i, SUBFRAME_LEN, LPC_ORDER,
992  0, 1, 1 << 12);
993  memcpy(p->synth_mem, p->audio + FRAME_LEN, LPC_ORDER * sizeof(*p->audio));
994 
995  if (p->postfilter) {
996  formant_postfilter(p, lpc, p->audio, out);
997  } else { // if output is not postfiltered it should be scaled by 2
998  for (i = 0; i < FRAME_LEN; i++)
999  out[i] = av_clip_int16(p->audio[LPC_ORDER + i] << 1);
1000  }
1001 
1002  *got_frame_ptr = 1;
1003 
1004  return frame_size[dec_mode];
1005 }
1006 
1007 #define OFFSET(x) offsetof(G723_1_Context, x)
1008 #define AD AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM
1009 
1010 static const AVOption options[] = {
1011  { "postfilter", "enable postfilter", OFFSET(postfilter), AV_OPT_TYPE_BOOL,
1012  { .i64 = 1 }, 0, 1, AD },
1013  { NULL }
1014 };
1015 
1016 
1017 static const AVClass g723_1dec_class = {
1018  .class_name = "G.723.1 decoder",
1019  .item_name = av_default_item_name,
1020  .option = options,
1021  .version = LIBAVUTIL_VERSION_INT,
1022 };
1023 
1025  .name = "g723_1",
1026  .long_name = NULL_IF_CONFIG_SMALL("G.723.1"),
1027  .type = AVMEDIA_TYPE_AUDIO,
1028  .id = AV_CODEC_ID_G723_1,
1029  .priv_data_size = sizeof(G723_1_Context),
1032  .capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
1033  .priv_class = &g723_1dec_class,
1034 };
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:299
#define NULL
Definition: coverity.c:32
const char const char void * val
Definition: avisynth_c.h:771
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:201
int reflection_coef
Definition: g723_1.h:144
int ad_cb_gain
Definition: g723_1.h:82
AVOption.
Definition: opt.h:246
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:199
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:261
#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:85
Memory handling functions.
else temp
Definition: vf_mcdeint.c:256
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:1017
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:1401
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:193
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:478
AVCodec.
Definition: avcodec.h:3351
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
Definition: decode_audio.c:42
#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:68
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:2151
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:64
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:433
AVOptions.
#define LPC_ORDER
Definition: g723_1.h:40
Rate
G723.1 rate values.
Definition: g723_1.h:72
static struct @281 state
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:347
uint8_t * data
Definition: avcodec.h:1400
static const uint8_t bits2[81]
Definition: aactab.c:140
void av_memcpy_backptr(uint8_t *dst, int back, int cnt)
Overlapping memcpy() implementation.
Definition: mem.c:414
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:206
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
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:186
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:532
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:3358
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:2194
int interp_index
Definition: g723_1.h:140
static int estimate_sid_gain(G723_1_Context *p)
Definition: g723_1dec.c:658
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:1024
static const int cng_adaptive_cb_lag[4]
Definition: g723_1.h:1435
int32_t
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:1007
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:266
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:507
int16_t opt_gain
optimal gain
Definition: g723_1.h:95
int postfilter
Definition: g723_1.h:147
int frame_size
Definition: mxfenc.c:1947
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:1488
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:642
#define FASTDIV(a, b)
Definition: mathops.h:202
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: decode.c:1886
G.723.1 types, functions and data tables.
void * buf
Definition: avisynth_c.h:690
static void formant_postfilter(G723_1_Context *p, int16_t *lpc, int16_t *buf, int16_t *dst)
Perform formant filtering.
Definition: g723_1dec.c:575
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:313
static void skip_bits1(GetBitContext *s)
Definition: get_bits.h:338
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:999
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:425
Active speech.
Definition: g723_1.h:64
#define CNG_RANDOM_SEED
Definition: g723_1dec.c:41
enum FrameType cur_frame_type
Definition: g723_1.h:123
#define SUBFRAME_LEN
Definition: g723_1.h:36
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:215
int pf_gain
formant postfilter gain scaling unit memory
Definition: g723_1.h:145
#define SUBFRAMES
Definition: dcaenc.c:46
#define AD
Definition: g723_1dec.c:1008
common internal api header.
if(ret< 0)
Definition: vf_mcdeint.c:279
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:1515
static const int cng_bseg[3]
Definition: g723_1.h:1439
int channels
number of audio channels
Definition: avcodec.h:2144
static int16_t square_root(unsigned val)
Bitexact implementation of sqrt(val/2).
Definition: g723_1dec.c:183
static const AVOption options[]
Definition: g723_1dec.c:1010
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:43
FILE * out
Definition: movenc.c:54
static int cng_rand(int *state, int base)
Definition: g723_1dec.c:652
int interp_gain
Definition: g723_1.h:141
static void generate_noise(G723_1_Context *p)
Definition: g723_1dec.c:718
#define AV_CH_LAYOUT_MONO
This structure stores compressed data.
Definition: avcodec.h:1377
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:267
#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:843
static const uint8_t bits1[81]
Definition: aactab.c:117
int ad_cb_lag
adaptive codebook lag
Definition: g723_1.h:81
static uint8_t tmp[11]
Definition: aes_ctr.c:26