FFmpeg  4.2.1
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 
44 {
45  G723_1_Context *s = avctx->priv_data;
46 
48  if (avctx->channels < 1 || avctx->channels > 2) {
49  av_log(avctx, AV_LOG_ERROR, "Only mono and stereo are supported (requested channels: %d).\n", avctx->channels);
50  return AVERROR(EINVAL);
51  }
53  for (int ch = 0; ch < avctx->channels; ch++) {
54  G723_1_ChannelContext *p = &s->ch[ch];
55 
56  p->pf_gain = 1 << 12;
57 
58  memcpy(p->prev_lsp, dc_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
59  memcpy(p->sid_lsp, dc_lsp, LPC_ORDER * sizeof(*p->sid_lsp));
60 
63  }
64 
65  return 0;
66 }
67 
68 /**
69  * Unpack the frame into parameters.
70  *
71  * @param p the context
72  * @param buf pointer to the input buffer
73  * @param buf_size size of the input buffer
74  */
76  int buf_size)
77 {
78  GetBitContext gb;
79  int ad_cb_len;
80  int temp, info_bits, i;
81  int ret;
82 
83  ret = init_get_bits8(&gb, buf, buf_size);
84  if (ret < 0)
85  return ret;
86 
87  /* Extract frame type and rate info */
88  info_bits = get_bits(&gb, 2);
89 
90  if (info_bits == 3) {
92  return 0;
93  }
94 
95  /* Extract 24 bit lsp indices, 8 bit for each band */
96  p->lsp_index[2] = get_bits(&gb, 8);
97  p->lsp_index[1] = get_bits(&gb, 8);
98  p->lsp_index[0] = get_bits(&gb, 8);
99 
100  if (info_bits == 2) {
102  p->subframe[0].amp_index = get_bits(&gb, 6);
103  return 0;
104  }
105 
106  /* Extract the info common to both rates */
107  p->cur_rate = info_bits ? RATE_5300 : RATE_6300;
109 
110  p->pitch_lag[0] = get_bits(&gb, 7);
111  if (p->pitch_lag[0] > 123) /* test if forbidden code */
112  return -1;
113  p->pitch_lag[0] += PITCH_MIN;
114  p->subframe[1].ad_cb_lag = get_bits(&gb, 2);
115 
116  p->pitch_lag[1] = get_bits(&gb, 7);
117  if (p->pitch_lag[1] > 123)
118  return -1;
119  p->pitch_lag[1] += PITCH_MIN;
120  p->subframe[3].ad_cb_lag = get_bits(&gb, 2);
121  p->subframe[0].ad_cb_lag = 1;
122  p->subframe[2].ad_cb_lag = 1;
123 
124  for (i = 0; i < SUBFRAMES; i++) {
125  /* Extract combined gain */
126  temp = get_bits(&gb, 12);
127  ad_cb_len = 170;
128  p->subframe[i].dirac_train = 0;
129  if (p->cur_rate == RATE_6300 && p->pitch_lag[i >> 1] < SUBFRAME_LEN - 2) {
130  p->subframe[i].dirac_train = temp >> 11;
131  temp &= 0x7FF;
132  ad_cb_len = 85;
133  }
134  p->subframe[i].ad_cb_gain = FASTDIV(temp, GAIN_LEVELS);
135  if (p->subframe[i].ad_cb_gain < ad_cb_len) {
136  p->subframe[i].amp_index = temp - p->subframe[i].ad_cb_gain *
137  GAIN_LEVELS;
138  } else {
139  return -1;
140  }
141  }
142 
143  p->subframe[0].grid_index = get_bits1(&gb);
144  p->subframe[1].grid_index = get_bits1(&gb);
145  p->subframe[2].grid_index = get_bits1(&gb);
146  p->subframe[3].grid_index = get_bits1(&gb);
147 
148  if (p->cur_rate == RATE_6300) {
149  skip_bits1(&gb); /* skip reserved bit */
150 
151  /* Compute pulse_pos index using the 13-bit combined position index */
152  temp = get_bits(&gb, 13);
153  p->subframe[0].pulse_pos = temp / 810;
154 
155  temp -= p->subframe[0].pulse_pos * 810;
156  p->subframe[1].pulse_pos = FASTDIV(temp, 90);
157 
158  temp -= p->subframe[1].pulse_pos * 90;
159  p->subframe[2].pulse_pos = FASTDIV(temp, 9);
160  p->subframe[3].pulse_pos = temp - p->subframe[2].pulse_pos * 9;
161 
162  p->subframe[0].pulse_pos = (p->subframe[0].pulse_pos << 16) +
163  get_bits(&gb, 16);
164  p->subframe[1].pulse_pos = (p->subframe[1].pulse_pos << 14) +
165  get_bits(&gb, 14);
166  p->subframe[2].pulse_pos = (p->subframe[2].pulse_pos << 16) +
167  get_bits(&gb, 16);
168  p->subframe[3].pulse_pos = (p->subframe[3].pulse_pos << 14) +
169  get_bits(&gb, 14);
170 
171  p->subframe[0].pulse_sign = get_bits(&gb, 6);
172  p->subframe[1].pulse_sign = get_bits(&gb, 5);
173  p->subframe[2].pulse_sign = get_bits(&gb, 6);
174  p->subframe[3].pulse_sign = get_bits(&gb, 5);
175  } else { /* 5300 bps */
176  p->subframe[0].pulse_pos = get_bits(&gb, 12);
177  p->subframe[1].pulse_pos = get_bits(&gb, 12);
178  p->subframe[2].pulse_pos = get_bits(&gb, 12);
179  p->subframe[3].pulse_pos = get_bits(&gb, 12);
180 
181  p->subframe[0].pulse_sign = get_bits(&gb, 4);
182  p->subframe[1].pulse_sign = get_bits(&gb, 4);
183  p->subframe[2].pulse_sign = get_bits(&gb, 4);
184  p->subframe[3].pulse_sign = get_bits(&gb, 4);
185  }
186 
187  return 0;
188 }
189 
190 /**
191  * Bitexact implementation of sqrt(val/2).
192  */
193 static int16_t square_root(unsigned val)
194 {
195  av_assert2(!(val & 0x80000000));
196 
197  return (ff_sqrt(val << 1) >> 1) & (~1);
198 }
199 
200 /**
201  * Generate fixed codebook excitation vector.
202  *
203  * @param vector decoded excitation vector
204  * @param subfrm current subframe
205  * @param cur_rate current bitrate
206  * @param pitch_lag closed loop pitch lag
207  * @param index current subframe index
208  */
209 static void gen_fcb_excitation(int16_t *vector, G723_1_Subframe *subfrm,
210  enum Rate cur_rate, int pitch_lag, int index)
211 {
212  int temp, i, j;
213 
214  memset(vector, 0, SUBFRAME_LEN * sizeof(*vector));
215 
216  if (cur_rate == RATE_6300) {
217  if (subfrm->pulse_pos >= max_pos[index])
218  return;
219 
220  /* Decode amplitudes and positions */
221  j = PULSE_MAX - pulses[index];
222  temp = subfrm->pulse_pos;
223  for (i = 0; i < SUBFRAME_LEN / GRID_SIZE; i++) {
224  temp -= combinatorial_table[j][i];
225  if (temp >= 0)
226  continue;
227  temp += combinatorial_table[j++][i];
228  if (subfrm->pulse_sign & (1 << (PULSE_MAX - j))) {
229  vector[subfrm->grid_index + GRID_SIZE * i] =
230  -fixed_cb_gain[subfrm->amp_index];
231  } else {
232  vector[subfrm->grid_index + GRID_SIZE * i] =
233  fixed_cb_gain[subfrm->amp_index];
234  }
235  if (j == PULSE_MAX)
236  break;
237  }
238  if (subfrm->dirac_train == 1)
239  ff_g723_1_gen_dirac_train(vector, pitch_lag);
240  } else { /* 5300 bps */
241  int cb_gain = fixed_cb_gain[subfrm->amp_index];
242  int cb_shift = subfrm->grid_index;
243  int cb_sign = subfrm->pulse_sign;
244  int cb_pos = subfrm->pulse_pos;
245  int offset, beta, lag;
246 
247  for (i = 0; i < 8; i += 2) {
248  offset = ((cb_pos & 7) << 3) + cb_shift + i;
249  vector[offset] = (cb_sign & 1) ? cb_gain : -cb_gain;
250  cb_pos >>= 3;
251  cb_sign >>= 1;
252  }
253 
254  /* Enhance harmonic components */
255  lag = pitch_contrib[subfrm->ad_cb_gain << 1] + pitch_lag +
256  subfrm->ad_cb_lag - 1;
257  beta = pitch_contrib[(subfrm->ad_cb_gain << 1) + 1];
258 
259  if (lag < SUBFRAME_LEN - 2) {
260  for (i = lag; i < SUBFRAME_LEN; i++)
261  vector[i] += beta * vector[i - lag] >> 15;
262  }
263  }
264 }
265 
266 /**
267  * Estimate maximum auto-correlation around pitch lag.
268  *
269  * @param buf buffer with offset applied
270  * @param offset offset of the excitation vector
271  * @param ccr_max pointer to the maximum auto-correlation
272  * @param pitch_lag decoded pitch lag
273  * @param length length of autocorrelation
274  * @param dir forward lag(1) / backward lag(-1)
275  */
276 static int autocorr_max(const int16_t *buf, int offset, int *ccr_max,
277  int pitch_lag, int length, int dir)
278 {
279  int limit, ccr, lag = 0;
280  int i;
281 
282  pitch_lag = FFMIN(PITCH_MAX - 3, pitch_lag);
283  if (dir > 0)
284  limit = FFMIN(FRAME_LEN + PITCH_MAX - offset - length, pitch_lag + 3);
285  else
286  limit = pitch_lag + 3;
287 
288  for (i = pitch_lag - 3; i <= limit; i++) {
289  ccr = ff_g723_1_dot_product(buf, buf + dir * i, length);
290 
291  if (ccr > *ccr_max) {
292  *ccr_max = ccr;
293  lag = i;
294  }
295  }
296  return lag;
297 }
298 
299 /**
300  * Calculate pitch postfilter optimal and scaling gains.
301  *
302  * @param lag pitch postfilter forward/backward lag
303  * @param ppf pitch postfilter parameters
304  * @param cur_rate current bitrate
305  * @param tgt_eng target energy
306  * @param ccr cross-correlation
307  * @param res_eng residual energy
308  */
309 static void comp_ppf_gains(int lag, PPFParam *ppf, enum Rate cur_rate,
310  int tgt_eng, int ccr, int res_eng)
311 {
312  int pf_residual; /* square of postfiltered residual */
313  int temp1, temp2;
314 
315  ppf->index = lag;
316 
317  temp1 = tgt_eng * res_eng >> 1;
318  temp2 = ccr * ccr << 1;
319 
320  if (temp2 > temp1) {
321  if (ccr >= res_eng) {
322  ppf->opt_gain = ppf_gain_weight[cur_rate];
323  } else {
324  ppf->opt_gain = (ccr << 15) / res_eng *
325  ppf_gain_weight[cur_rate] >> 15;
326  }
327  /* pf_res^2 = tgt_eng + 2*ccr*gain + res_eng*gain^2 */
328  temp1 = (tgt_eng << 15) + (ccr * ppf->opt_gain << 1);
329  temp2 = (ppf->opt_gain * ppf->opt_gain >> 15) * res_eng;
330  pf_residual = av_sat_add32(temp1, temp2 + (1 << 15)) >> 16;
331 
332  if (tgt_eng >= pf_residual << 1) {
333  temp1 = 0x7fff;
334  } else {
335  temp1 = (tgt_eng << 14) / pf_residual;
336  }
337 
338  /* scaling_gain = sqrt(tgt_eng/pf_res^2) */
339  ppf->sc_gain = square_root(temp1 << 16);
340  } else {
341  ppf->opt_gain = 0;
342  ppf->sc_gain = 0x7fff;
343  }
344 
345  ppf->opt_gain = av_clip_int16(ppf->opt_gain * ppf->sc_gain >> 15);
346 }
347 
348 /**
349  * Calculate pitch postfilter parameters.
350  *
351  * @param p the context
352  * @param offset offset of the excitation vector
353  * @param pitch_lag decoded pitch lag
354  * @param ppf pitch postfilter parameters
355  * @param cur_rate current bitrate
356  */
357 static void comp_ppf_coeff(G723_1_ChannelContext *p, int offset, int pitch_lag,
358  PPFParam *ppf, enum Rate cur_rate)
359 {
360 
361  int16_t scale;
362  int i;
363  int temp1, temp2;
364 
365  /*
366  * 0 - target energy
367  * 1 - forward cross-correlation
368  * 2 - forward residual energy
369  * 3 - backward cross-correlation
370  * 4 - backward residual energy
371  */
372  int energy[5] = {0, 0, 0, 0, 0};
373  int16_t *buf = p->audio + LPC_ORDER + offset;
374  int fwd_lag = autocorr_max(buf, offset, &energy[1], pitch_lag,
375  SUBFRAME_LEN, 1);
376  int back_lag = autocorr_max(buf, offset, &energy[3], pitch_lag,
377  SUBFRAME_LEN, -1);
378 
379  ppf->index = 0;
380  ppf->opt_gain = 0;
381  ppf->sc_gain = 0x7fff;
382 
383  /* Case 0, Section 3.6 */
384  if (!back_lag && !fwd_lag)
385  return;
386 
387  /* Compute target energy */
388  energy[0] = ff_g723_1_dot_product(buf, buf, SUBFRAME_LEN);
389 
390  /* Compute forward residual energy */
391  if (fwd_lag)
392  energy[2] = ff_g723_1_dot_product(buf + fwd_lag, buf + fwd_lag,
393  SUBFRAME_LEN);
394 
395  /* Compute backward residual energy */
396  if (back_lag)
397  energy[4] = ff_g723_1_dot_product(buf - back_lag, buf - back_lag,
398  SUBFRAME_LEN);
399 
400  /* Normalize and shorten */
401  temp1 = 0;
402  for (i = 0; i < 5; i++)
403  temp1 = FFMAX(energy[i], temp1);
404 
405  scale = ff_g723_1_normalize_bits(temp1, 31);
406  for (i = 0; i < 5; i++)
407  energy[i] = (energy[i] << scale) >> 16;
408 
409  if (fwd_lag && !back_lag) { /* Case 1 */
410  comp_ppf_gains(fwd_lag, ppf, cur_rate, energy[0], energy[1],
411  energy[2]);
412  } else if (!fwd_lag) { /* Case 2 */
413  comp_ppf_gains(-back_lag, ppf, cur_rate, energy[0], energy[3],
414  energy[4]);
415  } else { /* Case 3 */
416 
417  /*
418  * Select the largest of energy[1]^2/energy[2]
419  * and energy[3]^2/energy[4]
420  */
421  temp1 = energy[4] * ((energy[1] * energy[1] + (1 << 14)) >> 15);
422  temp2 = energy[2] * ((energy[3] * energy[3] + (1 << 14)) >> 15);
423  if (temp1 >= temp2) {
424  comp_ppf_gains(fwd_lag, ppf, cur_rate, energy[0], energy[1],
425  energy[2]);
426  } else {
427  comp_ppf_gains(-back_lag, ppf, cur_rate, energy[0], energy[3],
428  energy[4]);
429  }
430  }
431 }
432 
433 /**
434  * Classify frames as voiced/unvoiced.
435  *
436  * @param p the context
437  * @param pitch_lag decoded pitch_lag
438  * @param exc_eng excitation energy estimation
439  * @param scale scaling factor of exc_eng
440  *
441  * @return residual interpolation index if voiced, 0 otherwise
442  */
443 static int comp_interp_index(G723_1_ChannelContext *p, int pitch_lag,
444  int *exc_eng, int *scale)
445 {
446  int offset = PITCH_MAX + 2 * SUBFRAME_LEN;
447  int16_t *buf = p->audio + LPC_ORDER;
448 
449  int index, ccr, tgt_eng, best_eng, temp;
450 
452  buf += offset;
453 
454  /* Compute maximum backward cross-correlation */
455  ccr = 0;
456  index = autocorr_max(buf, offset, &ccr, pitch_lag, SUBFRAME_LEN * 2, -1);
457  ccr = av_sat_add32(ccr, 1 << 15) >> 16;
458 
459  /* Compute target energy */
460  tgt_eng = ff_g723_1_dot_product(buf, buf, SUBFRAME_LEN * 2);
461  *exc_eng = av_sat_add32(tgt_eng, 1 << 15) >> 16;
462 
463  if (ccr <= 0)
464  return 0;
465 
466  /* Compute best energy */
467  best_eng = ff_g723_1_dot_product(buf - index, buf - index,
468  SUBFRAME_LEN * 2);
469  best_eng = av_sat_add32(best_eng, 1 << 15) >> 16;
470 
471  temp = best_eng * *exc_eng >> 3;
472 
473  if (temp < ccr * ccr) {
474  return index;
475  } else
476  return 0;
477 }
478 
479 /**
480  * Perform residual interpolation based on frame classification.
481  *
482  * @param buf decoded excitation vector
483  * @param out output vector
484  * @param lag decoded pitch lag
485  * @param gain interpolated gain
486  * @param rseed seed for random number generator
487  */
488 static void residual_interp(int16_t *buf, int16_t *out, int lag,
489  int gain, int *rseed)
490 {
491  int i;
492  if (lag) { /* Voiced */
493  int16_t *vector_ptr = buf + PITCH_MAX;
494  /* Attenuate */
495  for (i = 0; i < lag; i++)
496  out[i] = vector_ptr[i - lag] * 3 >> 2;
497  av_memcpy_backptr((uint8_t*)(out + lag), lag * sizeof(*out),
498  (FRAME_LEN - lag) * sizeof(*out));
499  } else { /* Unvoiced */
500  for (i = 0; i < FRAME_LEN; i++) {
501  *rseed = (int16_t)(*rseed * 521 + 259);
502  out[i] = gain * *rseed >> 15;
503  }
504  memset(buf, 0, (FRAME_LEN + PITCH_MAX) * sizeof(*buf));
505  }
506 }
507 
508 /**
509  * Perform IIR filtering.
510  *
511  * @param fir_coef FIR coefficients
512  * @param iir_coef IIR coefficients
513  * @param src source vector
514  * @param dest destination vector
515  * @param width width of the output, 16 bits(0) / 32 bits(1)
516  */
517 #define iir_filter(fir_coef, iir_coef, src, dest, width)\
518 {\
519  int m, n;\
520  int res_shift = 16 & ~-(width);\
521  int in_shift = 16 - res_shift;\
522 \
523  for (m = 0; m < SUBFRAME_LEN; m++) {\
524  int64_t filter = 0;\
525  for (n = 1; n <= LPC_ORDER; n++) {\
526  filter -= (fir_coef)[n - 1] * (src)[m - n] -\
527  (iir_coef)[n - 1] * ((dest)[m - n] >> in_shift);\
528  }\
529 \
530  (dest)[m] = av_clipl_int32(((src)[m] * 65536) + (filter * 8) +\
531  (1 << 15)) >> res_shift;\
532  }\
533 }
534 
535 /**
536  * Adjust gain of postfiltered signal.
537  *
538  * @param p the context
539  * @param buf postfiltered output vector
540  * @param energy input energy coefficient
541  */
542 static void gain_scale(G723_1_ChannelContext *p, int16_t * buf, int energy)
543 {
544  int num, denom, gain, bits1, bits2;
545  int i;
546 
547  num = energy;
548  denom = 0;
549  for (i = 0; i < SUBFRAME_LEN; i++) {
550  int temp = buf[i] >> 2;
551  temp *= temp;
552  denom = av_sat_dadd32(denom, temp);
553  }
554 
555  if (num && denom) {
556  bits1 = ff_g723_1_normalize_bits(num, 31);
557  bits2 = ff_g723_1_normalize_bits(denom, 31);
558  num = num << bits1 >> 1;
559  denom <<= bits2;
560 
561  bits2 = 5 + bits1 - bits2;
562  bits2 = av_clip_uintp2(bits2, 5);
563 
564  gain = (num >> 1) / (denom >> 16);
565  gain = square_root(gain << 16 >> bits2);
566  } else {
567  gain = 1 << 12;
568  }
569 
570  for (i = 0; i < SUBFRAME_LEN; i++) {
571  p->pf_gain = (15 * p->pf_gain + gain + (1 << 3)) >> 4;
572  buf[i] = av_clip_int16((buf[i] * (p->pf_gain + (p->pf_gain >> 4)) +
573  (1 << 10)) >> 11);
574  }
575 }
576 
577 /**
578  * Perform formant filtering.
579  *
580  * @param p the context
581  * @param lpc quantized lpc coefficients
582  * @param buf input buffer
583  * @param dst output buffer
584  */
585 static void formant_postfilter(G723_1_ChannelContext *p, int16_t *lpc,
586  int16_t *buf, int16_t *dst)
587 {
588  int16_t filter_coef[2][LPC_ORDER];
589  int filter_signal[LPC_ORDER + FRAME_LEN], *signal_ptr;
590  int i, j, k;
591 
592  memcpy(buf, p->fir_mem, LPC_ORDER * sizeof(*buf));
593  memcpy(filter_signal, p->iir_mem, LPC_ORDER * sizeof(*filter_signal));
594 
595  for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) {
596  for (k = 0; k < LPC_ORDER; k++) {
597  filter_coef[0][k] = (-lpc[k] * postfilter_tbl[0][k] +
598  (1 << 14)) >> 15;
599  filter_coef[1][k] = (-lpc[k] * postfilter_tbl[1][k] +
600  (1 << 14)) >> 15;
601  }
602  iir_filter(filter_coef[0], filter_coef[1], buf + i, filter_signal + i, 1);
603  lpc += LPC_ORDER;
604  }
605 
606  memcpy(p->fir_mem, buf + FRAME_LEN, LPC_ORDER * sizeof(int16_t));
607  memcpy(p->iir_mem, filter_signal + FRAME_LEN, LPC_ORDER * sizeof(int));
608 
609  buf += LPC_ORDER;
610  signal_ptr = filter_signal + LPC_ORDER;
611  for (i = 0; i < SUBFRAMES; i++) {
612  int temp;
613  int auto_corr[2];
614  int scale, energy;
615 
616  /* Normalize */
617  scale = ff_g723_1_scale_vector(dst, buf, SUBFRAME_LEN);
618 
619  /* Compute auto correlation coefficients */
620  auto_corr[0] = ff_g723_1_dot_product(dst, dst + 1, SUBFRAME_LEN - 1);
621  auto_corr[1] = ff_g723_1_dot_product(dst, dst, SUBFRAME_LEN);
622 
623  /* Compute reflection coefficient */
624  temp = auto_corr[1] >> 16;
625  if (temp) {
626  temp = (auto_corr[0] >> 2) / temp;
627  }
628  p->reflection_coef = (3 * p->reflection_coef + temp + 2) >> 2;
629  temp = -p->reflection_coef >> 1 & ~3;
630 
631  /* Compensation filter */
632  for (j = 0; j < SUBFRAME_LEN; j++) {
633  dst[j] = av_sat_dadd32(signal_ptr[j],
634  (signal_ptr[j - 1] >> 16) * temp) >> 16;
635  }
636 
637  /* Compute normalized signal energy */
638  temp = 2 * scale + 4;
639  if (temp < 0) {
640  energy = av_clipl_int32((int64_t)auto_corr[1] << -temp);
641  } else
642  energy = auto_corr[1] >> temp;
643 
644  gain_scale(p, dst, energy);
645 
646  buf += SUBFRAME_LEN;
647  signal_ptr += SUBFRAME_LEN;
648  dst += SUBFRAME_LEN;
649  }
650 }
651 
652 static int sid_gain_to_lsp_index(int gain)
653 {
654  if (gain < 0x10)
655  return gain << 6;
656  else if (gain < 0x20)
657  return gain - 8 << 7;
658  else
659  return gain - 20 << 8;
660 }
661 
662 static inline int cng_rand(int *state, int base)
663 {
664  *state = (*state * 521 + 259) & 0xFFFF;
665  return (*state & 0x7FFF) * base >> 15;
666 }
667 
669 {
670  int i, shift, seg, seg2, t, val, val_add, x, y;
671 
672  shift = 16 - p->cur_gain * 2;
673  if (shift > 0) {
674  if (p->sid_gain == 0) {
675  t = 0;
676  } else if (shift >= 31 || (int32_t)((uint32_t)p->sid_gain << shift) >> shift != p->sid_gain) {
677  if (p->sid_gain < 0) t = INT32_MIN;
678  else t = INT32_MAX;
679  } else
680  t = p->sid_gain << shift;
681  }else
682  t = p->sid_gain >> -shift;
683  x = av_clipl_int32(t * (int64_t)cng_filt[0] >> 16);
684 
685  if (x >= cng_bseg[2])
686  return 0x3F;
687 
688  if (x >= cng_bseg[1]) {
689  shift = 4;
690  seg = 3;
691  } else {
692  shift = 3;
693  seg = (x >= cng_bseg[0]);
694  }
695  seg2 = FFMIN(seg, 3);
696 
697  val = 1 << shift;
698  val_add = val >> 1;
699  for (i = 0; i < shift; i++) {
700  t = seg * 32 + (val << seg2);
701  t *= t;
702  if (x >= t)
703  val += val_add;
704  else
705  val -= val_add;
706  val_add >>= 1;
707  }
708 
709  t = seg * 32 + (val << seg2);
710  y = t * t - x;
711  if (y <= 0) {
712  t = seg * 32 + (val + 1 << seg2);
713  t = t * t - x;
714  val = (seg2 - 1) * 16 + val;
715  if (t >= y)
716  val++;
717  } else {
718  t = seg * 32 + (val - 1 << seg2);
719  t = t * t - x;
720  val = (seg2 - 1) * 16 + val;
721  if (t >= y)
722  val--;
723  }
724 
725  return val;
726 }
727 
729 {
730  int i, j, idx, t;
731  int off[SUBFRAMES];
732  int signs[SUBFRAMES / 2 * 11], pos[SUBFRAMES / 2 * 11];
733  int tmp[SUBFRAME_LEN * 2];
734  int16_t *vector_ptr;
735  int64_t sum;
736  int b0, c, delta, x, shift;
737 
738  p->pitch_lag[0] = cng_rand(&p->cng_random_seed, 21) + 123;
739  p->pitch_lag[1] = cng_rand(&p->cng_random_seed, 19) + 123;
740 
741  for (i = 0; i < SUBFRAMES; i++) {
742  p->subframe[i].ad_cb_gain = cng_rand(&p->cng_random_seed, 50) + 1;
744  }
745 
746  for (i = 0; i < SUBFRAMES / 2; i++) {
747  t = cng_rand(&p->cng_random_seed, 1 << 13);
748  off[i * 2] = t & 1;
749  off[i * 2 + 1] = ((t >> 1) & 1) + SUBFRAME_LEN;
750  t >>= 2;
751  for (j = 0; j < 11; j++) {
752  signs[i * 11 + j] = ((t & 1) * 2 - 1) * (1 << 14);
753  t >>= 1;
754  }
755  }
756 
757  idx = 0;
758  for (i = 0; i < SUBFRAMES; i++) {
759  for (j = 0; j < SUBFRAME_LEN / 2; j++)
760  tmp[j] = j;
761  t = SUBFRAME_LEN / 2;
762  for (j = 0; j < pulses[i]; j++, idx++) {
763  int idx2 = cng_rand(&p->cng_random_seed, t);
764 
765  pos[idx] = tmp[idx2] * 2 + off[i];
766  tmp[idx2] = tmp[--t];
767  }
768  }
769 
770  vector_ptr = p->audio + LPC_ORDER;
771  memcpy(vector_ptr, p->prev_excitation,
772  PITCH_MAX * sizeof(*p->excitation));
773  for (i = 0; i < SUBFRAMES; i += 2) {
774  ff_g723_1_gen_acb_excitation(vector_ptr, vector_ptr,
775  p->pitch_lag[i >> 1], &p->subframe[i],
776  p->cur_rate);
778  vector_ptr + SUBFRAME_LEN,
779  p->pitch_lag[i >> 1], &p->subframe[i + 1],
780  p->cur_rate);
781 
782  t = 0;
783  for (j = 0; j < SUBFRAME_LEN * 2; j++)
784  t |= FFABS(vector_ptr[j]);
785  t = FFMIN(t, 0x7FFF);
786  if (!t) {
787  shift = 0;
788  } else {
789  shift = -10 + av_log2(t);
790  if (shift < -2)
791  shift = -2;
792  }
793  sum = 0;
794  if (shift < 0) {
795  for (j = 0; j < SUBFRAME_LEN * 2; j++) {
796  t = vector_ptr[j] * (1 << -shift);
797  sum += t * t;
798  tmp[j] = t;
799  }
800  } else {
801  for (j = 0; j < SUBFRAME_LEN * 2; j++) {
802  t = vector_ptr[j] >> shift;
803  sum += t * t;
804  tmp[j] = t;
805  }
806  }
807 
808  b0 = 0;
809  for (j = 0; j < 11; j++)
810  b0 += tmp[pos[(i / 2) * 11 + j]] * signs[(i / 2) * 11 + j];
811  b0 = b0 * 2 * 2979LL + (1 << 29) >> 30; // approximated division by 11
812 
813  c = p->cur_gain * (p->cur_gain * SUBFRAME_LEN >> 5);
814  if (shift * 2 + 3 >= 0)
815  c >>= shift * 2 + 3;
816  else
817  c <<= -(shift * 2 + 3);
818  c = (av_clipl_int32(sum << 1) - c) * 2979LL >> 15;
819 
820  delta = b0 * b0 * 2 - c;
821  if (delta <= 0) {
822  x = -b0;
823  } else {
824  delta = square_root(delta);
825  x = delta - b0;
826  t = delta + b0;
827  if (FFABS(t) < FFABS(x))
828  x = -t;
829  }
830  shift++;
831  if (shift < 0)
832  x >>= -shift;
833  else
834  x *= 1 << shift;
835  x = av_clip(x, -10000, 10000);
836 
837  for (j = 0; j < 11; j++) {
838  idx = (i / 2) * 11 + j;
839  vector_ptr[pos[idx]] = av_clip_int16(vector_ptr[pos[idx]] +
840  (x * signs[idx] >> 15));
841  }
842 
843  /* copy decoded data to serve as a history for the next decoded subframes */
844  memcpy(vector_ptr + PITCH_MAX, vector_ptr,
845  sizeof(*vector_ptr) * SUBFRAME_LEN * 2);
846  vector_ptr += SUBFRAME_LEN * 2;
847  }
848  /* Save the excitation for the next frame */
849  memcpy(p->prev_excitation, p->audio + LPC_ORDER + FRAME_LEN,
850  PITCH_MAX * sizeof(*p->excitation));
851 }
852 
853 static int g723_1_decode_frame(AVCodecContext *avctx, void *data,
854  int *got_frame_ptr, AVPacket *avpkt)
855 {
856  G723_1_Context *s = avctx->priv_data;
857  AVFrame *frame = data;
858  const uint8_t *buf = avpkt->data;
859  int buf_size = avpkt->size;
860  int dec_mode = buf[0] & 3;
861 
862  PPFParam ppf[SUBFRAMES];
863  int16_t cur_lsp[LPC_ORDER];
864  int16_t lpc[SUBFRAMES * LPC_ORDER];
865  int16_t acb_vector[SUBFRAME_LEN];
866  int16_t *out;
867  int bad_frame = 0, i, j, ret;
868 
869  if (buf_size < frame_size[dec_mode] * avctx->channels) {
870  if (buf_size)
871  av_log(avctx, AV_LOG_WARNING,
872  "Expected %d bytes, got %d - skipping packet\n",
873  frame_size[dec_mode], buf_size);
874  *got_frame_ptr = 0;
875  return buf_size;
876  }
877 
878  frame->nb_samples = FRAME_LEN;
879  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
880  return ret;
881 
882  for (int ch = 0; ch < avctx->channels; ch++) {
883  G723_1_ChannelContext *p = &s->ch[ch];
884  int16_t *audio = p->audio;
885 
886  if (unpack_bitstream(p, buf + ch * (buf_size / avctx->channels),
887  buf_size / avctx->channels) < 0) {
888  bad_frame = 1;
889  if (p->past_frame_type == ACTIVE_FRAME)
891  else
893  }
894 
895  out = (int16_t *)frame->extended_data[ch];
896 
897  if (p->cur_frame_type == ACTIVE_FRAME) {
898  if (!bad_frame)
899  p->erased_frames = 0;
900  else if (p->erased_frames != 3)
901  p->erased_frames++;
902 
903  ff_g723_1_inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, bad_frame);
904  ff_g723_1_lsp_interpolate(lpc, cur_lsp, p->prev_lsp);
905 
906  /* Save the lsp_vector for the next frame */
907  memcpy(p->prev_lsp, cur_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
908 
909  /* Generate the excitation for the frame */
910  memcpy(p->excitation, p->prev_excitation,
911  PITCH_MAX * sizeof(*p->excitation));
912  if (!p->erased_frames) {
913  int16_t *vector_ptr = p->excitation + PITCH_MAX;
914 
915  /* Update interpolation gain memory */
917  p->subframe[3].amp_index) >> 1];
918  for (i = 0; i < SUBFRAMES; i++) {
919  gen_fcb_excitation(vector_ptr, &p->subframe[i], p->cur_rate,
920  p->pitch_lag[i >> 1], i);
921  ff_g723_1_gen_acb_excitation(acb_vector,
922  &p->excitation[SUBFRAME_LEN * i],
923  p->pitch_lag[i >> 1],
924  &p->subframe[i], p->cur_rate);
925  /* Get the total excitation */
926  for (j = 0; j < SUBFRAME_LEN; j++) {
927  int v = av_clip_int16(vector_ptr[j] * 2);
928  vector_ptr[j] = av_clip_int16(v + acb_vector[j]);
929  }
930  vector_ptr += SUBFRAME_LEN;
931  }
932 
933  vector_ptr = p->excitation + PITCH_MAX;
934 
936  &p->sid_gain, &p->cur_gain);
937 
938  /* Perform pitch postfiltering */
939  if (s->postfilter) {
940  i = PITCH_MAX;
941  for (j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
942  comp_ppf_coeff(p, i, p->pitch_lag[j >> 1],
943  ppf + j, p->cur_rate);
944 
945  for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
947  vector_ptr + i,
948  vector_ptr + i + ppf[j].index,
949  ppf[j].sc_gain,
950  ppf[j].opt_gain,
951  1 << 14, 15, SUBFRAME_LEN);
952  } else {
953  audio = vector_ptr - LPC_ORDER;
954  }
955 
956  /* Save the excitation for the next frame */
957  memcpy(p->prev_excitation, p->excitation + FRAME_LEN,
958  PITCH_MAX * sizeof(*p->excitation));
959  } else {
960  p->interp_gain = (p->interp_gain * 3 + 2) >> 2;
961  if (p->erased_frames == 3) {
962  /* Mute output */
963  memset(p->excitation, 0,
964  (FRAME_LEN + PITCH_MAX) * sizeof(*p->excitation));
965  memset(p->prev_excitation, 0,
966  PITCH_MAX * sizeof(*p->excitation));
967  memset(frame->data[0], 0,
968  (FRAME_LEN + LPC_ORDER) * sizeof(int16_t));
969  } else {
970  int16_t *buf = p->audio + LPC_ORDER;
971 
972  /* Regenerate frame */
974  p->interp_gain, &p->random_seed);
975 
976  /* Save the excitation for the next frame */
977  memcpy(p->prev_excitation, buf + (FRAME_LEN - PITCH_MAX),
978  PITCH_MAX * sizeof(*p->excitation));
979  }
980  }
982  } else {
983  if (p->cur_frame_type == SID_FRAME) {
986  } else if (p->past_frame_type == ACTIVE_FRAME) {
987  p->sid_gain = estimate_sid_gain(p);
988  }
989 
990  if (p->past_frame_type == ACTIVE_FRAME)
991  p->cur_gain = p->sid_gain;
992  else
993  p->cur_gain = (p->cur_gain * 7 + p->sid_gain) >> 3;
994  generate_noise(p);
996  /* Save the lsp_vector for the next frame */
997  memcpy(p->prev_lsp, p->sid_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
998  }
999 
1001 
1002  memcpy(p->audio, p->synth_mem, LPC_ORDER * sizeof(*p->audio));
1003  for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
1004  ff_celp_lp_synthesis_filter(p->audio + i, &lpc[j * LPC_ORDER],
1005  audio + i, SUBFRAME_LEN, LPC_ORDER,
1006  0, 1, 1 << 12);
1007  memcpy(p->synth_mem, p->audio + FRAME_LEN, LPC_ORDER * sizeof(*p->audio));
1008 
1009  if (s->postfilter) {
1010  formant_postfilter(p, lpc, p->audio, out);
1011  } else { // if output is not postfiltered it should be scaled by 2
1012  for (i = 0; i < FRAME_LEN; i++)
1013  out[i] = av_clip_int16(p->audio[LPC_ORDER + i] << 1);
1014  }
1015  }
1016 
1017  *got_frame_ptr = 1;
1018 
1019  return frame_size[dec_mode] * avctx->channels;
1020 }
1021 
1022 #define OFFSET(x) offsetof(G723_1_Context, x)
1023 #define AD AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM
1024 
1025 static const AVOption options[] = {
1026  { "postfilter", "enable postfilter", OFFSET(postfilter), AV_OPT_TYPE_BOOL,
1027  { .i64 = 1 }, 0, 1, AD },
1028  { NULL }
1029 };
1030 
1031 
1032 static const AVClass g723_1dec_class = {
1033  .class_name = "G.723.1 decoder",
1034  .item_name = av_default_item_name,
1035  .option = options,
1036  .version = LIBAVUTIL_VERSION_INT,
1037 };
1038 
1040  .name = "g723_1",
1041  .long_name = NULL_IF_CONFIG_SMALL("G.723.1"),
1042  .type = AVMEDIA_TYPE_AUDIO,
1043  .id = AV_CODEC_ID_G723_1,
1044  .priv_data_size = sizeof(G723_1_Context),
1047  .capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
1048  .priv_class = &g723_1dec_class,
1049 };
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:309
int16_t excitation[PITCH_MAX+FRAME_LEN+4]
Definition: g723_1.h:131
#define NULL
Definition: coverity.c:32
const char const char void * val
Definition: avisynth_c.h:863
static int shift(int a, int b)
Definition: sonic.c:82
int dirac_train
Definition: g723_1.h:83
This structure describes decoded (raw) audio or video data.
Definition: frame.h:295
int ad_cb_gain
Definition: g723_1.h:82
AVOption.
Definition: opt.h:246
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:209
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:379
#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:728
#define LIBAVUTIL_VERSION_INT
Definition: version.h:85
Memory handling functions.
else temp
Definition: vf_mcdeint.c:256
G723_1_Subframe subframe[4]
Definition: g723_1.h:120
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
static const AVClass g723_1dec_class
Definition: g723_1dec.c:1032
static const int8_t pulses[4]
Number of non-zero pulses in the MP-MLQ excitation.
Definition: g723_1.h:723
int size
Definition: avcodec.h:1478
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(const uint8_t *) pi - 0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(const int16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(const int16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(const int32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(const int32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(const int64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64, *(const int64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64, *(const int64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(const float *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(const double *) pi *(INT64_C(1)<< 63))) #define FMT_PAIR_FUNC(out, in) static conv_func_type *const fmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB *AV_SAMPLE_FMT_NB]={ FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64), };static void cpy1(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, len);} static void cpy2(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 2 *len);} static void cpy4(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 4 *len);} static void cpy8(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 8 *len);} AudioConvert *swri_audio_convert_alloc(enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, const int *ch_map, int flags) { AudioConvert *ctx;conv_func_type *f=fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt)+AV_SAMPLE_FMT_NB *av_get_packed_sample_fmt(in_fmt)];if(!f) return NULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) return NULL;if(channels==1){ in_fmt=av_get_planar_sample_fmt(in_fmt);out_fmt=av_get_planar_sample_fmt(out_fmt);} ctx->channels=channels;ctx->conv_f=f;ctx->ch_map=ch_map;if(in_fmt==AV_SAMPLE_FMT_U8||in_fmt==AV_SAMPLE_FMT_U8P) memset(ctx->silence, 0x80, sizeof(ctx->silence));if(out_fmt==in_fmt &&!ch_map) { switch(av_get_bytes_per_sample(in_fmt)){ case 1:ctx->simd_f=cpy1;break;case 2:ctx->simd_f=cpy2;break;case 4:ctx->simd_f=cpy4;break;case 8:ctx->simd_f=cpy8;break;} } if(HAVE_X86ASM &&HAVE_MMX) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);if(ARCH_AARCH64) swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels);return ctx;} void swri_audio_convert_free(AudioConvert **ctx) { av_freep(ctx);} int swri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, int len) { int ch;int off=0;const int os=(out->planar ? 1 :out->ch_count) *out->bps;unsigned misaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask) { int planes=in->planar ? in->ch_count :1;unsigned m=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) in->ch[ch];misaligned|=m &ctx->in_simd_align_mask;} if(ctx->out_simd_align_mask) { int planes=out->planar ? out->ch_count :1;unsigned m=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) out->ch[ch];misaligned|=m &ctx->out_simd_align_mask;} if(ctx->simd_f &&!ctx->ch_map &&!misaligned){ off=len &~15;av_assert1(off >=0);av_assert1(off<=len);av_assert2(ctx->channels==SWR_CH_MAX||!in->ch[ctx->channels]);if(off >0){ if(out->planar==in->planar){ int planes=out->planar ? out->ch_count :1;for(ch=0;ch< planes;ch++){ ctx->simd_f(out-> ch ch
Definition: audioconvert.c:56
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:191
int16_t audio[FRAME_LEN+LPC_ORDER+PITCH_MAX+4]
Definition: g723_1.h:145
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:488
static void formant_postfilter(G723_1_ChannelContext *p, int16_t *lpc, int16_t *buf, int16_t *dst)
Perform formant filtering.
Definition: g723_1dec.c:585
#define AV_CH_LAYOUT_STEREO
AVCodec.
Definition: avcodec.h:3481
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
Definition: decode_audio.c:42
static int estimate_sid_gain(G723_1_ChannelContext *p)
Definition: g723_1dec.c:668
#define PITCH_MIN
Definition: g723_1.h:43
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
uint8_t base
Definition: vp3data.h:202
#define FRAME_LEN
Definition: g723_1.h:37
uint8_t lsp_index[LSP_BANDS]
Definition: g723_1.h:124
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:1440
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2233
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
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 generate_noise(G723_1_ChannelContext *p)
Definition: g723_1dec.c:728
const char data[16]
Definition: mxf.c:91
uint8_t * data
Definition: avcodec.h:1477
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:426
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
int pf_gain
formant postfilter gain scaling unit memory
Definition: g723_1.h:143
#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:630
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:259
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
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
#define AVERROR(e)
Definition: error.h:43
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
int grid_index
Definition: g723_1.h:85
const char * name
Name of the codec implementation.
Definition: avcodec.h:3488
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:2276
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
enum FrameType past_frame_type
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:730
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:1383
audio channel layout utility functions
#define FFMIN(a, b)
Definition: common.h:96
AVCodec ff_g723_1_decoder
Definition: g723_1dec.c:1039
static const int cng_adaptive_cb_lag[4]
Definition: g723_1.h:1438
static struct @313 state
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 s(width, name)
Definition: cbs_vp9.c:257
#define OFFSET(x)
Definition: g723_1dec.c:1022
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:276
if(ret< 0)
Definition: vf_mcdeint.c:279
#define av_log2
Definition: intmath.h:83
int iir_mem[LPC_ORDER]
Definition: g723_1.h:134
#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:517
int16_t opt_gain
optimal gain
Definition: g723_1.h:95
int postfilter
Definition: g723_1.h:161
int frame_size
Definition: mxfenc.c:2215
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:357
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:232
static const int16_t pitch_contrib[340]
Definition: g723_1.h:674
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
Definition: get_bits.h:677
static void gain_scale(G723_1_ChannelContext *p, int16_t *buf, int energy)
Adjust gain of postfiltered signal.
Definition: g723_1dec.c:542
main external API structure.
Definition: avcodec.h:1565
static const int16_t ppf_gain_weight[2]
Postfilter gain weighting factors scaled by 2^15.
Definition: g723_1.h:227
static int sid_gain_to_lsp_index(int gain)
Definition: g723_1dec.c:652
#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:1964
G.723.1 types, functions and data tables.
void * buf
Definition: avisynth_c.h:766
int16_t fir_mem[LPC_ORDER]
Definition: g723_1.h:133
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:498
static void skip_bits1(GetBitContext *s)
Definition: get_bits.h:538
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:1024
G723_1_ChannelContext ch[2]
Definition: g723_1.h:163
Active speech.
Definition: g723_1.h:64
int16_t prev_lsp[LPC_ORDER]
Definition: g723_1.h:128
#define CNG_RANDOM_SEED
Definition: g723_1dec.c:41
#define SUBFRAME_LEN
Definition: g723_1.h:36
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:309
#define SUBFRAMES
Definition: dcaenc.c:50
int16_t synth_mem[LPC_ORDER]
Definition: g723_1.h:132
#define AD
Definition: g723_1dec.c:1023
common internal api header.
Pitch postfilter parameters.
Definition: g723_1.h:93
static double c[64]
static int unpack_bitstream(G723_1_ChannelContext *p, const uint8_t *buf, int buf_size)
Unpack the frame into parameters.
Definition: g723_1dec.c:75
void * priv_data
Definition: avcodec.h:1592
enum FrameType cur_frame_type
Definition: g723_1.h:121
static const int cng_bseg[3]
Definition: g723_1.h:1442
int channels
number of audio channels
Definition: avcodec.h:2226
enum Rate cur_rate
Definition: g723_1.h:123
static int16_t square_root(unsigned val)
Bitexact implementation of sqrt(val/2).
Definition: g723_1dec.c:193
static const AVOption options[]
Definition: g723_1dec.c:1025
int pulse_pos
Definition: g723_1.h:87
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:443
static av_cold int g723_1_decode_init(AVCodecContext *avctx)
Definition: g723_1dec.c:43
FILE * out
Definition: movenc.c:54
signed 16 bits, planar
Definition: samplefmt.h:67
static int cng_rand(int *state, int base)
Definition: g723_1dec.c:662
const char int length
Definition: avisynth_c.h:860
int16_t sid_lsp[LPC_ORDER]
Definition: g723_1.h:129
int16_t prev_excitation[PITCH_MAX]
Definition: g723_1.h:130
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:342
#define AV_CH_LAYOUT_MONO
This structure stores compressed data.
Definition: avcodec.h:1454
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:361
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: avcodec.h:981
static int g723_1_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: g723_1dec.c:853
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