Libav
amrwbdec.c
Go to the documentation of this file.
1 /*
2  * AMR wideband decoder
3  * Copyright (c) 2010 Marcelo Galvao Povoa
4  *
5  * This file is part of Libav.
6  *
7  * Libav is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * Libav is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A particular PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with Libav; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
28 #include "libavutil/common.h"
29 #include "libavutil/float_dsp.h"
30 #include "libavutil/lfg.h"
31 
32 #include "avcodec.h"
33 #include "lsp.h"
34 #include "celp_filters.h"
35 #include "acelp_filters.h"
36 #include "acelp_vectors.h"
37 #include "acelp_pitch_delay.h"
38 #include "internal.h"
39 
40 #define AMR_USE_16BIT_TABLES
41 #include "amr.h"
42 
43 #include "amrwbdata.h"
44 
45 typedef struct {
47  enum Mode fr_cur_mode;
49  float isf_cur[LP_ORDER];
50  float isf_q_past[LP_ORDER];
51  float isf_past_final[LP_ORDER];
52  double isp[4][LP_ORDER];
53  double isp_sub4_past[LP_ORDER];
54 
55  float lp_coef[4][LP_ORDER];
56 
59 
60  float excitation_buf[AMRWB_P_DELAY_MAX + LP_ORDER + 2 + AMRWB_SFR_SIZE];
61  float *excitation;
62 
63  float pitch_vector[AMRWB_SFR_SIZE];
64  float fixed_vector[AMRWB_SFR_SIZE];
65 
66  float prediction_error[4];
67  float pitch_gain[6];
68  float fixed_gain[2];
69 
70  float tilt_coef;
71 
74  float prev_tr_gain;
75 
76  float samples_az[LP_ORDER + AMRWB_SFR_SIZE];
77  float samples_up[UPS_MEM_SIZE + AMRWB_SFR_SIZE];
78  float samples_hb[LP_ORDER_16k + AMRWB_SFR_SIZE_16k];
79 
80  float hpf_31_mem[2], hpf_400_mem[2];
81  float demph_mem[1];
82  float bpf_6_7_mem[HB_FIR_SIZE];
83  float lpf_7_mem[HB_FIR_SIZE];
84 
87 } AMRWBContext;
88 
90 {
91  AMRWBContext *ctx = avctx->priv_data;
92  int i;
93 
94  if (avctx->channels > 1) {
95  avpriv_report_missing_feature(avctx, "multi-channel AMR");
96  return AVERROR_PATCHWELCOME;
97  }
98 
99  avctx->channels = 1;
101  avctx->sample_rate = 16000;
102  avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
103 
104  av_lfg_init(&ctx->prng, 1);
105 
107  ctx->first_frame = 1;
108 
109  for (i = 0; i < LP_ORDER; i++)
110  ctx->isf_past_final[i] = isf_init[i] * (1.0f / (1 << 15));
111 
112  for (i = 0; i < 4; i++)
113  ctx->prediction_error[i] = MIN_ENERGY;
114 
115  return 0;
116 }
117 
127 static int decode_mime_header(AMRWBContext *ctx, const uint8_t *buf)
128 {
129  /* Decode frame header (1st octet) */
130  ctx->fr_cur_mode = buf[0] >> 3 & 0x0F;
131  ctx->fr_quality = (buf[0] & 0x4) == 0x4;
132 
133  return 1;
134 }
135 
143 static void decode_isf_indices_36b(uint16_t *ind, float *isf_q)
144 {
145  int i;
146 
147  for (i = 0; i < 9; i++)
148  isf_q[i] = dico1_isf[ind[0]][i] * (1.0f / (1 << 15));
149 
150  for (i = 0; i < 7; i++)
151  isf_q[i + 9] = dico2_isf[ind[1]][i] * (1.0f / (1 << 15));
152 
153  for (i = 0; i < 5; i++)
154  isf_q[i] += dico21_isf_36b[ind[2]][i] * (1.0f / (1 << 15));
155 
156  for (i = 0; i < 4; i++)
157  isf_q[i + 5] += dico22_isf_36b[ind[3]][i] * (1.0f / (1 << 15));
158 
159  for (i = 0; i < 7; i++)
160  isf_q[i + 9] += dico23_isf_36b[ind[4]][i] * (1.0f / (1 << 15));
161 }
162 
170 static void decode_isf_indices_46b(uint16_t *ind, float *isf_q)
171 {
172  int i;
173 
174  for (i = 0; i < 9; i++)
175  isf_q[i] = dico1_isf[ind[0]][i] * (1.0f / (1 << 15));
176 
177  for (i = 0; i < 7; i++)
178  isf_q[i + 9] = dico2_isf[ind[1]][i] * (1.0f / (1 << 15));
179 
180  for (i = 0; i < 3; i++)
181  isf_q[i] += dico21_isf[ind[2]][i] * (1.0f / (1 << 15));
182 
183  for (i = 0; i < 3; i++)
184  isf_q[i + 3] += dico22_isf[ind[3]][i] * (1.0f / (1 << 15));
185 
186  for (i = 0; i < 3; i++)
187  isf_q[i + 6] += dico23_isf[ind[4]][i] * (1.0f / (1 << 15));
188 
189  for (i = 0; i < 3; i++)
190  isf_q[i + 9] += dico24_isf[ind[5]][i] * (1.0f / (1 << 15));
191 
192  for (i = 0; i < 4; i++)
193  isf_q[i + 12] += dico25_isf[ind[6]][i] * (1.0f / (1 << 15));
194 }
195 
204 static void isf_add_mean_and_past(float *isf_q, float *isf_past)
205 {
206  int i;
207  float tmp;
208 
209  for (i = 0; i < LP_ORDER; i++) {
210  tmp = isf_q[i];
211  isf_q[i] += isf_mean[i] * (1.0f / (1 << 15));
212  isf_q[i] += PRED_FACTOR * isf_past[i];
213  isf_past[i] = tmp;
214  }
215 }
216 
224 static void interpolate_isp(double isp_q[4][LP_ORDER], const double *isp4_past)
225 {
226  int i, k;
227 
228  for (k = 0; k < 3; k++) {
229  float c = isfp_inter[k];
230  for (i = 0; i < LP_ORDER; i++)
231  isp_q[k][i] = (1.0 - c) * isp4_past[i] + c * isp_q[3][i];
232  }
233 }
234 
246 static void decode_pitch_lag_high(int *lag_int, int *lag_frac, int pitch_index,
247  uint8_t *base_lag_int, int subframe)
248 {
249  if (subframe == 0 || subframe == 2) {
250  if (pitch_index < 376) {
251  *lag_int = (pitch_index + 137) >> 2;
252  *lag_frac = pitch_index - (*lag_int << 2) + 136;
253  } else if (pitch_index < 440) {
254  *lag_int = (pitch_index + 257 - 376) >> 1;
255  *lag_frac = (pitch_index - (*lag_int << 1) + 256 - 376) << 1;
256  /* the actual resolution is 1/2 but expressed as 1/4 */
257  } else {
258  *lag_int = pitch_index - 280;
259  *lag_frac = 0;
260  }
261  /* minimum lag for next subframe */
262  *base_lag_int = av_clip(*lag_int - 8 - (*lag_frac < 0),
264  // XXX: the spec states clearly that *base_lag_int should be
265  // the nearest integer to *lag_int (minus 8), but the ref code
266  // actually always uses its floor, I'm following the latter
267  } else {
268  *lag_int = (pitch_index + 1) >> 2;
269  *lag_frac = pitch_index - (*lag_int << 2);
270  *lag_int += *base_lag_int;
271  }
272 }
273 
279 static void decode_pitch_lag_low(int *lag_int, int *lag_frac, int pitch_index,
280  uint8_t *base_lag_int, int subframe, enum Mode mode)
281 {
282  if (subframe == 0 || (subframe == 2 && mode != MODE_6k60)) {
283  if (pitch_index < 116) {
284  *lag_int = (pitch_index + 69) >> 1;
285  *lag_frac = (pitch_index - (*lag_int << 1) + 68) << 1;
286  } else {
287  *lag_int = pitch_index - 24;
288  *lag_frac = 0;
289  }
290  // XXX: same problem as before
291  *base_lag_int = av_clip(*lag_int - 8 - (*lag_frac < 0),
293  } else {
294  *lag_int = (pitch_index + 1) >> 1;
295  *lag_frac = (pitch_index - (*lag_int << 1)) << 1;
296  *lag_int += *base_lag_int;
297  }
298 }
299 
309  const AMRWBSubFrame *amr_subframe,
310  const int subframe)
311 {
312  int pitch_lag_int, pitch_lag_frac;
313  int i;
314  float *exc = ctx->excitation;
315  enum Mode mode = ctx->fr_cur_mode;
316 
317  if (mode <= MODE_8k85) {
318  decode_pitch_lag_low(&pitch_lag_int, &pitch_lag_frac, amr_subframe->adap,
319  &ctx->base_pitch_lag, subframe, mode);
320  } else
321  decode_pitch_lag_high(&pitch_lag_int, &pitch_lag_frac, amr_subframe->adap,
322  &ctx->base_pitch_lag, subframe);
323 
324  ctx->pitch_lag_int = pitch_lag_int;
325  pitch_lag_int += pitch_lag_frac > 0;
326 
327  /* Calculate the pitch vector by interpolating the past excitation at the
328  pitch lag using a hamming windowed sinc function */
329  ff_acelp_interpolatef(exc, exc + 1 - pitch_lag_int,
330  ac_inter, 4,
331  pitch_lag_frac + (pitch_lag_frac > 0 ? 0 : 4),
332  LP_ORDER, AMRWB_SFR_SIZE + 1);
333 
334  /* Check which pitch signal path should be used
335  * 6k60 and 8k85 modes have the ltp flag set to 0 */
336  if (amr_subframe->ltp) {
337  memcpy(ctx->pitch_vector, exc, AMRWB_SFR_SIZE * sizeof(float));
338  } else {
339  for (i = 0; i < AMRWB_SFR_SIZE; i++)
340  ctx->pitch_vector[i] = 0.18 * exc[i - 1] + 0.64 * exc[i] +
341  0.18 * exc[i + 1];
342  memcpy(exc, ctx->pitch_vector, AMRWB_SFR_SIZE * sizeof(float));
343  }
344 }
345 
347 #define BIT_STR(x,lsb,len) (((x) >> (lsb)) & ((1 << (len)) - 1))
348 
350 #define BIT_POS(x, p) (((x) >> (p)) & 1)
351 
365 static inline void decode_1p_track(int *out, int code, int m, int off)
366 {
367  int pos = BIT_STR(code, 0, m) + off;
368 
369  out[0] = BIT_POS(code, m) ? -pos : pos;
370 }
371 
372 static inline void decode_2p_track(int *out, int code, int m, int off)
373 {
374  int pos0 = BIT_STR(code, m, m) + off;
375  int pos1 = BIT_STR(code, 0, m) + off;
376 
377  out[0] = BIT_POS(code, 2*m) ? -pos0 : pos0;
378  out[1] = BIT_POS(code, 2*m) ? -pos1 : pos1;
379  out[1] = pos0 > pos1 ? -out[1] : out[1];
380 }
381 
382 static void decode_3p_track(int *out, int code, int m, int off)
383 {
384  int half_2p = BIT_POS(code, 2*m - 1) << (m - 1);
385 
386  decode_2p_track(out, BIT_STR(code, 0, 2*m - 1),
387  m - 1, off + half_2p);
388  decode_1p_track(out + 2, BIT_STR(code, 2*m, m + 1), m, off);
389 }
390 
391 static void decode_4p_track(int *out, int code, int m, int off)
392 {
393  int half_4p, subhalf_2p;
394  int b_offset = 1 << (m - 1);
395 
396  switch (BIT_STR(code, 4*m - 2, 2)) { /* case ID (2 bits) */
397  case 0: /* 0 pulses in A, 4 pulses in B or vice versa */
398  half_4p = BIT_POS(code, 4*m - 3) << (m - 1); // which has 4 pulses
399  subhalf_2p = BIT_POS(code, 2*m - 3) << (m - 2);
400 
401  decode_2p_track(out, BIT_STR(code, 0, 2*m - 3),
402  m - 2, off + half_4p + subhalf_2p);
403  decode_2p_track(out + 2, BIT_STR(code, 2*m - 2, 2*m - 1),
404  m - 1, off + half_4p);
405  break;
406  case 1: /* 1 pulse in A, 3 pulses in B */
407  decode_1p_track(out, BIT_STR(code, 3*m - 2, m),
408  m - 1, off);
409  decode_3p_track(out + 1, BIT_STR(code, 0, 3*m - 2),
410  m - 1, off + b_offset);
411  break;
412  case 2: /* 2 pulses in each half */
413  decode_2p_track(out, BIT_STR(code, 2*m - 1, 2*m - 1),
414  m - 1, off);
415  decode_2p_track(out + 2, BIT_STR(code, 0, 2*m - 1),
416  m - 1, off + b_offset);
417  break;
418  case 3: /* 3 pulses in A, 1 pulse in B */
419  decode_3p_track(out, BIT_STR(code, m, 3*m - 2),
420  m - 1, off);
421  decode_1p_track(out + 3, BIT_STR(code, 0, m),
422  m - 1, off + b_offset);
423  break;
424  }
425 }
426 
427 static void decode_5p_track(int *out, int code, int m, int off)
428 {
429  int half_3p = BIT_POS(code, 5*m - 1) << (m - 1);
430 
431  decode_3p_track(out, BIT_STR(code, 2*m + 1, 3*m - 2),
432  m - 1, off + half_3p);
433 
434  decode_2p_track(out + 3, BIT_STR(code, 0, 2*m + 1), m, off);
435 }
436 
437 static void decode_6p_track(int *out, int code, int m, int off)
438 {
439  int b_offset = 1 << (m - 1);
440  /* which half has more pulses in cases 0 to 2 */
441  int half_more = BIT_POS(code, 6*m - 5) << (m - 1);
442  int half_other = b_offset - half_more;
443 
444  switch (BIT_STR(code, 6*m - 4, 2)) { /* case ID (2 bits) */
445  case 0: /* 0 pulses in A, 6 pulses in B or vice versa */
446  decode_1p_track(out, BIT_STR(code, 0, m),
447  m - 1, off + half_more);
448  decode_5p_track(out + 1, BIT_STR(code, m, 5*m - 5),
449  m - 1, off + half_more);
450  break;
451  case 1: /* 1 pulse in A, 5 pulses in B or vice versa */
452  decode_1p_track(out, BIT_STR(code, 0, m),
453  m - 1, off + half_other);
454  decode_5p_track(out + 1, BIT_STR(code, m, 5*m - 5),
455  m - 1, off + half_more);
456  break;
457  case 2: /* 2 pulses in A, 4 pulses in B or vice versa */
458  decode_2p_track(out, BIT_STR(code, 0, 2*m - 1),
459  m - 1, off + half_other);
460  decode_4p_track(out + 2, BIT_STR(code, 2*m - 1, 4*m - 4),
461  m - 1, off + half_more);
462  break;
463  case 3: /* 3 pulses in A, 3 pulses in B */
464  decode_3p_track(out, BIT_STR(code, 3*m - 2, 3*m - 2),
465  m - 1, off);
466  decode_3p_track(out + 3, BIT_STR(code, 0, 3*m - 2),
467  m - 1, off + b_offset);
468  break;
469  }
470 }
471 
481 static void decode_fixed_vector(float *fixed_vector, const uint16_t *pulse_hi,
482  const uint16_t *pulse_lo, const enum Mode mode)
483 {
484  /* sig_pos stores for each track the decoded pulse position indexes
485  * (1-based) multiplied by its corresponding amplitude (+1 or -1) */
486  int sig_pos[4][6];
487  int spacing = (mode == MODE_6k60) ? 2 : 4;
488  int i, j;
489 
490  switch (mode) {
491  case MODE_6k60:
492  for (i = 0; i < 2; i++)
493  decode_1p_track(sig_pos[i], pulse_lo[i], 5, 1);
494  break;
495  case MODE_8k85:
496  for (i = 0; i < 4; i++)
497  decode_1p_track(sig_pos[i], pulse_lo[i], 4, 1);
498  break;
499  case MODE_12k65:
500  for (i = 0; i < 4; i++)
501  decode_2p_track(sig_pos[i], pulse_lo[i], 4, 1);
502  break;
503  case MODE_14k25:
504  for (i = 0; i < 2; i++)
505  decode_3p_track(sig_pos[i], pulse_lo[i], 4, 1);
506  for (i = 2; i < 4; i++)
507  decode_2p_track(sig_pos[i], pulse_lo[i], 4, 1);
508  break;
509  case MODE_15k85:
510  for (i = 0; i < 4; i++)
511  decode_3p_track(sig_pos[i], pulse_lo[i], 4, 1);
512  break;
513  case MODE_18k25:
514  for (i = 0; i < 4; i++)
515  decode_4p_track(sig_pos[i], (int) pulse_lo[i] +
516  ((int) pulse_hi[i] << 14), 4, 1);
517  break;
518  case MODE_19k85:
519  for (i = 0; i < 2; i++)
520  decode_5p_track(sig_pos[i], (int) pulse_lo[i] +
521  ((int) pulse_hi[i] << 10), 4, 1);
522  for (i = 2; i < 4; i++)
523  decode_4p_track(sig_pos[i], (int) pulse_lo[i] +
524  ((int) pulse_hi[i] << 14), 4, 1);
525  break;
526  case MODE_23k05:
527  case MODE_23k85:
528  for (i = 0; i < 4; i++)
529  decode_6p_track(sig_pos[i], (int) pulse_lo[i] +
530  ((int) pulse_hi[i] << 11), 4, 1);
531  break;
532  }
533 
534  memset(fixed_vector, 0, sizeof(float) * AMRWB_SFR_SIZE);
535 
536  for (i = 0; i < 4; i++)
537  for (j = 0; j < pulses_nb_per_mode_tr[mode][i]; j++) {
538  int pos = (FFABS(sig_pos[i][j]) - 1) * spacing + i;
539 
540  fixed_vector[pos] += sig_pos[i][j] < 0 ? -1.0 : 1.0;
541  }
542 }
543 
552 static void decode_gains(const uint8_t vq_gain, const enum Mode mode,
553  float *fixed_gain_factor, float *pitch_gain)
554 {
555  const int16_t *gains = (mode <= MODE_8k85 ? qua_gain_6b[vq_gain] :
556  qua_gain_7b[vq_gain]);
557 
558  *pitch_gain = gains[0] * (1.0f / (1 << 14));
559  *fixed_gain_factor = gains[1] * (1.0f / (1 << 11));
560 }
561 
568 // XXX: Spec states this procedure should be applied when the pitch
569 // lag is less than 64, but this checking seems absent in reference and AMR-NB
570 static void pitch_sharpening(AMRWBContext *ctx, float *fixed_vector)
571 {
572  int i;
573 
574  /* Tilt part */
575  for (i = AMRWB_SFR_SIZE - 1; i != 0; i--)
576  fixed_vector[i] -= fixed_vector[i - 1] * ctx->tilt_coef;
577 
578  /* Periodicity enhancement part */
579  for (i = ctx->pitch_lag_int; i < AMRWB_SFR_SIZE; i++)
580  fixed_vector[i] += fixed_vector[i - ctx->pitch_lag_int] * 0.85;
581 }
582 
589 // XXX: There is something wrong with the precision here! The magnitudes
590 // of the energies are not correct. Please check the reference code carefully
591 static float voice_factor(float *p_vector, float p_gain,
592  float *f_vector, float f_gain)
593 {
594  double p_ener = (double) avpriv_scalarproduct_float_c(p_vector, p_vector,
595  AMRWB_SFR_SIZE) *
596  p_gain * p_gain;
597  double f_ener = (double) avpriv_scalarproduct_float_c(f_vector, f_vector,
598  AMRWB_SFR_SIZE) *
599  f_gain * f_gain;
600 
601  return (p_ener - f_ener) / (p_ener + f_ener);
602 }
603 
614 static float *anti_sparseness(AMRWBContext *ctx,
615  float *fixed_vector, float *buf)
616 {
617  int ir_filter_nr;
618 
619  if (ctx->fr_cur_mode > MODE_8k85) // no filtering in higher modes
620  return fixed_vector;
621 
622  if (ctx->pitch_gain[0] < 0.6) {
623  ir_filter_nr = 0; // strong filtering
624  } else if (ctx->pitch_gain[0] < 0.9) {
625  ir_filter_nr = 1; // medium filtering
626  } else
627  ir_filter_nr = 2; // no filtering
628 
629  /* detect 'onset' */
630  if (ctx->fixed_gain[0] > 3.0 * ctx->fixed_gain[1]) {
631  if (ir_filter_nr < 2)
632  ir_filter_nr++;
633  } else {
634  int i, count = 0;
635 
636  for (i = 0; i < 6; i++)
637  if (ctx->pitch_gain[i] < 0.6)
638  count++;
639 
640  if (count > 2)
641  ir_filter_nr = 0;
642 
643  if (ir_filter_nr > ctx->prev_ir_filter_nr + 1)
644  ir_filter_nr--;
645  }
646 
647  /* update ir filter strength history */
648  ctx->prev_ir_filter_nr = ir_filter_nr;
649 
650  ir_filter_nr += (ctx->fr_cur_mode == MODE_8k85);
651 
652  if (ir_filter_nr < 2) {
653  int i;
654  const float *coef = ir_filters_lookup[ir_filter_nr];
655 
656  /* Circular convolution code in the reference
657  * decoder was modified to avoid using one
658  * extra array. The filtered vector is given by:
659  *
660  * c2(n) = sum(i,0,len-1){ c(i) * coef( (n - i + len) % len ) }
661  */
662 
663  memset(buf, 0, sizeof(float) * AMRWB_SFR_SIZE);
664  for (i = 0; i < AMRWB_SFR_SIZE; i++)
665  if (fixed_vector[i])
666  ff_celp_circ_addf(buf, buf, coef, i, fixed_vector[i],
667  AMRWB_SFR_SIZE);
668  fixed_vector = buf;
669  }
670 
671  return fixed_vector;
672 }
673 
678 static float stability_factor(const float *isf, const float *isf_past)
679 {
680  int i;
681  float acc = 0.0;
682 
683  for (i = 0; i < LP_ORDER - 1; i++)
684  acc += (isf[i] - isf_past[i]) * (isf[i] - isf_past[i]);
685 
686  // XXX: This part is not so clear from the reference code
687  // the result is more accurate changing the "/ 256" to "* 512"
688  return FFMAX(0.0, 1.25 - acc * 0.8 * 512);
689 }
690 
702 static float noise_enhancer(float fixed_gain, float *prev_tr_gain,
703  float voice_fac, float stab_fac)
704 {
705  float sm_fac = 0.5 * (1 - voice_fac) * stab_fac;
706  float g0;
707 
708  // XXX: the following fixed-point constants used to in(de)crement
709  // gain by 1.5dB were taken from the reference code, maybe it could
710  // be simpler
711  if (fixed_gain < *prev_tr_gain) {
712  g0 = FFMIN(*prev_tr_gain, fixed_gain + fixed_gain *
713  (6226 * (1.0f / (1 << 15)))); // +1.5 dB
714  } else
715  g0 = FFMAX(*prev_tr_gain, fixed_gain *
716  (27536 * (1.0f / (1 << 15)))); // -1.5 dB
717 
718  *prev_tr_gain = g0; // update next frame threshold
719 
720  return sm_fac * g0 + (1 - sm_fac) * fixed_gain;
721 }
722 
729 static void pitch_enhancer(float *fixed_vector, float voice_fac)
730 {
731  int i;
732  float cpe = 0.125 * (1 + voice_fac);
733  float last = fixed_vector[0]; // holds c(i - 1)
734 
735  fixed_vector[0] -= cpe * fixed_vector[1];
736 
737  for (i = 1; i < AMRWB_SFR_SIZE - 1; i++) {
738  float cur = fixed_vector[i];
739 
740  fixed_vector[i] -= cpe * (last + fixed_vector[i + 1]);
741  last = cur;
742  }
743 
744  fixed_vector[AMRWB_SFR_SIZE - 1] -= cpe * last;
745 }
746 
757 static void synthesis(AMRWBContext *ctx, float *lpc, float *excitation,
758  float fixed_gain, const float *fixed_vector,
759  float *samples)
760 {
761  ff_weighted_vector_sumf(excitation, ctx->pitch_vector, fixed_vector,
762  ctx->pitch_gain[0], fixed_gain, AMRWB_SFR_SIZE);
763 
764  /* emphasize pitch vector contribution in low bitrate modes */
765  if (ctx->pitch_gain[0] > 0.5 && ctx->fr_cur_mode <= MODE_8k85) {
766  int i;
767  float energy = avpriv_scalarproduct_float_c(excitation, excitation,
769 
770  // XXX: Weird part in both ref code and spec. A unknown parameter
771  // {beta} seems to be identical to the current pitch gain
772  float pitch_factor = 0.25 * ctx->pitch_gain[0] * ctx->pitch_gain[0];
773 
774  for (i = 0; i < AMRWB_SFR_SIZE; i++)
775  excitation[i] += pitch_factor * ctx->pitch_vector[i];
776 
777  ff_scale_vector_to_given_sum_of_squares(excitation, excitation,
778  energy, AMRWB_SFR_SIZE);
779  }
780 
781  ff_celp_lp_synthesis_filterf(samples, lpc, excitation,
783 }
784 
794 static void de_emphasis(float *out, float *in, float m, float mem[1])
795 {
796  int i;
797 
798  out[0] = in[0] + m * mem[0];
799 
800  for (i = 1; i < AMRWB_SFR_SIZE; i++)
801  out[i] = in[i] + out[i - 1] * m;
802 
803  mem[0] = out[AMRWB_SFR_SIZE - 1];
804 }
805 
814 static void upsample_5_4(float *out, const float *in, int o_size)
815 {
816  const float *in0 = in - UPS_FIR_SIZE + 1;
817  int i, j, k;
818  int int_part = 0, frac_part;
819 
820  i = 0;
821  for (j = 0; j < o_size / 5; j++) {
822  out[i] = in[int_part];
823  frac_part = 4;
824  i++;
825 
826  for (k = 1; k < 5; k++) {
827  out[i] = avpriv_scalarproduct_float_c(in0 + int_part,
828  upsample_fir[4 - frac_part],
829  UPS_MEM_SIZE);
830  int_part++;
831  frac_part--;
832  i++;
833  }
834  }
835 }
836 
846 static float find_hb_gain(AMRWBContext *ctx, const float *synth,
847  uint16_t hb_idx, uint8_t vad)
848 {
849  int wsp = (vad > 0);
850  float tilt;
851 
852  if (ctx->fr_cur_mode == MODE_23k85)
853  return qua_hb_gain[hb_idx] * (1.0f / (1 << 14));
854 
855  tilt = avpriv_scalarproduct_float_c(synth, synth + 1, AMRWB_SFR_SIZE - 1) /
857 
858  /* return gain bounded by [0.1, 1.0] */
859  return av_clipf((1.0 - FFMAX(0.0, tilt)) * (1.25 - 0.25 * wsp), 0.1, 1.0);
860 }
861 
871 static void scaled_hb_excitation(AMRWBContext *ctx, float *hb_exc,
872  const float *synth_exc, float hb_gain)
873 {
874  int i;
875  float energy = avpriv_scalarproduct_float_c(synth_exc, synth_exc,
877 
878  /* Generate a white-noise excitation */
879  for (i = 0; i < AMRWB_SFR_SIZE_16k; i++)
880  hb_exc[i] = 32768.0 - (uint16_t) av_lfg_get(&ctx->prng);
881 
883  energy * hb_gain * hb_gain,
884  AMRWB_SFR_SIZE_16k);
885 }
886 
890 static float auto_correlation(float *diff_isf, float mean, int lag)
891 {
892  int i;
893  float sum = 0.0;
894 
895  for (i = 7; i < LP_ORDER - 2; i++) {
896  float prod = (diff_isf[i] - mean) * (diff_isf[i - lag] - mean);
897  sum += prod * prod;
898  }
899  return sum;
900 }
901 
909 static void extrapolate_isf(float isf[LP_ORDER_16k])
910 {
911  float diff_isf[LP_ORDER - 2], diff_mean;
912  float corr_lag[3];
913  float est, scale;
914  int i, j, i_max_corr;
915 
916  isf[LP_ORDER_16k - 1] = isf[LP_ORDER - 1];
917 
918  /* Calculate the difference vector */
919  for (i = 0; i < LP_ORDER - 2; i++)
920  diff_isf[i] = isf[i + 1] - isf[i];
921 
922  diff_mean = 0.0;
923  for (i = 2; i < LP_ORDER - 2; i++)
924  diff_mean += diff_isf[i] * (1.0f / (LP_ORDER - 4));
925 
926  /* Find which is the maximum autocorrelation */
927  i_max_corr = 0;
928  for (i = 0; i < 3; i++) {
929  corr_lag[i] = auto_correlation(diff_isf, diff_mean, i + 2);
930 
931  if (corr_lag[i] > corr_lag[i_max_corr])
932  i_max_corr = i;
933  }
934  i_max_corr++;
935 
936  for (i = LP_ORDER - 1; i < LP_ORDER_16k - 1; i++)
937  isf[i] = isf[i - 1] + isf[i - 1 - i_max_corr]
938  - isf[i - 2 - i_max_corr];
939 
940  /* Calculate an estimate for ISF(18) and scale ISF based on the error */
941  est = 7965 + (isf[2] - isf[3] - isf[4]) / 6.0;
942  scale = 0.5 * (FFMIN(est, 7600) - isf[LP_ORDER - 2]) /
943  (isf[LP_ORDER_16k - 2] - isf[LP_ORDER - 2]);
944 
945  for (i = LP_ORDER - 1, j = 0; i < LP_ORDER_16k - 1; i++, j++)
946  diff_isf[j] = scale * (isf[i] - isf[i - 1]);
947 
948  /* Stability insurance */
949  for (i = 1; i < LP_ORDER_16k - LP_ORDER; i++)
950  if (diff_isf[i] + diff_isf[i - 1] < 5.0) {
951  if (diff_isf[i] > diff_isf[i - 1]) {
952  diff_isf[i - 1] = 5.0 - diff_isf[i];
953  } else
954  diff_isf[i] = 5.0 - diff_isf[i - 1];
955  }
956 
957  for (i = LP_ORDER - 1, j = 0; i < LP_ORDER_16k - 1; i++, j++)
958  isf[i] = isf[i - 1] + diff_isf[j] * (1.0f / (1 << 15));
959 
960  /* Scale the ISF vector for 16000 Hz */
961  for (i = 0; i < LP_ORDER_16k - 1; i++)
962  isf[i] *= 0.8;
963 }
964 
974 static void lpc_weighting(float *out, const float *lpc, float gamma, int size)
975 {
976  int i;
977  float fac = gamma;
978 
979  for (i = 0; i < size; i++) {
980  out[i] = lpc[i] * fac;
981  fac *= gamma;
982  }
983 }
984 
996 static void hb_synthesis(AMRWBContext *ctx, int subframe, float *samples,
997  const float *exc, const float *isf, const float *isf_past)
998 {
999  float hb_lpc[LP_ORDER_16k];
1000  enum Mode mode = ctx->fr_cur_mode;
1001 
1002  if (mode == MODE_6k60) {
1003  float e_isf[LP_ORDER_16k]; // ISF vector for extrapolation
1004  double e_isp[LP_ORDER_16k];
1005 
1006  ff_weighted_vector_sumf(e_isf, isf_past, isf, isfp_inter[subframe],
1007  1.0 - isfp_inter[subframe], LP_ORDER);
1008 
1009  extrapolate_isf(e_isf);
1010 
1011  e_isf[LP_ORDER_16k - 1] *= 2.0;
1012  ff_acelp_lsf2lspd(e_isp, e_isf, LP_ORDER_16k);
1013  ff_amrwb_lsp2lpc(e_isp, hb_lpc, LP_ORDER_16k);
1014 
1015  lpc_weighting(hb_lpc, hb_lpc, 0.9, LP_ORDER_16k);
1016  } else {
1017  lpc_weighting(hb_lpc, ctx->lp_coef[subframe], 0.6, LP_ORDER);
1018  }
1019 
1020  ff_celp_lp_synthesis_filterf(samples, hb_lpc, exc, AMRWB_SFR_SIZE_16k,
1021  (mode == MODE_6k60) ? LP_ORDER_16k : LP_ORDER);
1022 }
1023 
1035 static void hb_fir_filter(float *out, const float fir_coef[HB_FIR_SIZE + 1],
1036  float mem[HB_FIR_SIZE], const float *in)
1037 {
1038  int i, j;
1039  float data[AMRWB_SFR_SIZE_16k + HB_FIR_SIZE]; // past and current samples
1040 
1041  memcpy(data, mem, HB_FIR_SIZE * sizeof(float));
1042  memcpy(data + HB_FIR_SIZE, in, AMRWB_SFR_SIZE_16k * sizeof(float));
1043 
1044  for (i = 0; i < AMRWB_SFR_SIZE_16k; i++) {
1045  out[i] = 0.0;
1046  for (j = 0; j <= HB_FIR_SIZE; j++)
1047  out[i] += data[i + j] * fir_coef[j];
1048  }
1049 
1050  memcpy(mem, data + AMRWB_SFR_SIZE_16k, HB_FIR_SIZE * sizeof(float));
1051 }
1052 
1057 {
1058  memmove(&ctx->excitation_buf[0], &ctx->excitation_buf[AMRWB_SFR_SIZE],
1059  (AMRWB_P_DELAY_MAX + LP_ORDER + 1) * sizeof(float));
1060 
1061  memmove(&ctx->pitch_gain[1], &ctx->pitch_gain[0], 5 * sizeof(float));
1062  memmove(&ctx->fixed_gain[1], &ctx->fixed_gain[0], 1 * sizeof(float));
1063 
1064  memmove(&ctx->samples_az[0], &ctx->samples_az[AMRWB_SFR_SIZE],
1065  LP_ORDER * sizeof(float));
1066  memmove(&ctx->samples_up[0], &ctx->samples_up[AMRWB_SFR_SIZE],
1067  UPS_MEM_SIZE * sizeof(float));
1068  memmove(&ctx->samples_hb[0], &ctx->samples_hb[AMRWB_SFR_SIZE_16k],
1069  LP_ORDER_16k * sizeof(float));
1070 }
1071 
1072 static int amrwb_decode_frame(AVCodecContext *avctx, void *data,
1073  int *got_frame_ptr, AVPacket *avpkt)
1074 {
1075  AMRWBContext *ctx = avctx->priv_data;
1076  AVFrame *frame = data;
1077  AMRWBFrame *cf = &ctx->frame;
1078  const uint8_t *buf = avpkt->data;
1079  int buf_size = avpkt->size;
1080  int expected_fr_size, header_size;
1081  float *buf_out;
1082  float spare_vector[AMRWB_SFR_SIZE]; // extra stack space to hold result from anti-sparseness processing
1083  float fixed_gain_factor; // fixed gain correction factor (gamma)
1084  float *synth_fixed_vector; // pointer to the fixed vector that synthesis should use
1085  float synth_fixed_gain; // the fixed gain that synthesis should use
1086  float voice_fac, stab_fac; // parameters used for gain smoothing
1087  float synth_exc[AMRWB_SFR_SIZE]; // post-processed excitation for synthesis
1088  float hb_exc[AMRWB_SFR_SIZE_16k]; // excitation for the high frequency band
1089  float hb_samples[AMRWB_SFR_SIZE_16k]; // filtered high-band samples from synthesis
1090  float hb_gain;
1091  int sub, i, ret;
1092 
1093  /* get output buffer */
1094  frame->nb_samples = 4 * AMRWB_SFR_SIZE_16k;
1095  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
1096  av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
1097  return ret;
1098  }
1099  buf_out = (float *)frame->data[0];
1100 
1101  header_size = decode_mime_header(ctx, buf);
1102  if (ctx->fr_cur_mode > MODE_SID) {
1103  av_log(avctx, AV_LOG_ERROR,
1104  "Invalid mode %d\n", ctx->fr_cur_mode);
1105  return AVERROR_INVALIDDATA;
1106  }
1107  expected_fr_size = ((cf_sizes_wb[ctx->fr_cur_mode] + 7) >> 3) + 1;
1108 
1109  if (buf_size < expected_fr_size) {
1110  av_log(avctx, AV_LOG_ERROR,
1111  "Frame too small (%d bytes). Truncated file?\n", buf_size);
1112  *got_frame_ptr = 0;
1113  return AVERROR_INVALIDDATA;
1114  }
1115 
1116  if (!ctx->fr_quality || ctx->fr_cur_mode > MODE_SID)
1117  av_log(avctx, AV_LOG_ERROR, "Encountered a bad or corrupted frame\n");
1118 
1119  if (ctx->fr_cur_mode == MODE_SID) { /* Comfort noise frame */
1120  avpriv_request_sample(avctx, "SID mode");
1121  return AVERROR_PATCHWELCOME;
1122  }
1123 
1124  ff_amr_bit_reorder((uint16_t *) &ctx->frame, sizeof(AMRWBFrame),
1125  buf + header_size, amr_bit_orderings_by_mode[ctx->fr_cur_mode]);
1126 
1127  /* Decode the quantized ISF vector */
1128  if (ctx->fr_cur_mode == MODE_6k60) {
1130  } else {
1132  }
1133 
1136 
1137  stab_fac = stability_factor(ctx->isf_cur, ctx->isf_past_final);
1138 
1139  ctx->isf_cur[LP_ORDER - 1] *= 2.0;
1140  ff_acelp_lsf2lspd(ctx->isp[3], ctx->isf_cur, LP_ORDER);
1141 
1142  /* Generate a ISP vector for each subframe */
1143  if (ctx->first_frame) {
1144  ctx->first_frame = 0;
1145  memcpy(ctx->isp_sub4_past, ctx->isp[3], LP_ORDER * sizeof(double));
1146  }
1147  interpolate_isp(ctx->isp, ctx->isp_sub4_past);
1148 
1149  for (sub = 0; sub < 4; sub++)
1150  ff_amrwb_lsp2lpc(ctx->isp[sub], ctx->lp_coef[sub], LP_ORDER);
1151 
1152  for (sub = 0; sub < 4; sub++) {
1153  const AMRWBSubFrame *cur_subframe = &cf->subframe[sub];
1154  float *sub_buf = buf_out + sub * AMRWB_SFR_SIZE_16k;
1155 
1156  /* Decode adaptive codebook (pitch vector) */
1157  decode_pitch_vector(ctx, cur_subframe, sub);
1158  /* Decode innovative codebook (fixed vector) */
1159  decode_fixed_vector(ctx->fixed_vector, cur_subframe->pul_ih,
1160  cur_subframe->pul_il, ctx->fr_cur_mode);
1161 
1162  pitch_sharpening(ctx, ctx->fixed_vector);
1163 
1164  decode_gains(cur_subframe->vq_gain, ctx->fr_cur_mode,
1165  &fixed_gain_factor, &ctx->pitch_gain[0]);
1166 
1167  ctx->fixed_gain[0] =
1168  ff_amr_set_fixed_gain(fixed_gain_factor,
1170  ctx->fixed_vector,
1171  AMRWB_SFR_SIZE) /
1173  ctx->prediction_error,
1175 
1176  /* Calculate voice factor and store tilt for next subframe */
1177  voice_fac = voice_factor(ctx->pitch_vector, ctx->pitch_gain[0],
1178  ctx->fixed_vector, ctx->fixed_gain[0]);
1179  ctx->tilt_coef = voice_fac * 0.25 + 0.25;
1180 
1181  /* Construct current excitation */
1182  for (i = 0; i < AMRWB_SFR_SIZE; i++) {
1183  ctx->excitation[i] *= ctx->pitch_gain[0];
1184  ctx->excitation[i] += ctx->fixed_gain[0] * ctx->fixed_vector[i];
1185  ctx->excitation[i] = truncf(ctx->excitation[i]);
1186  }
1187 
1188  /* Post-processing of excitation elements */
1189  synth_fixed_gain = noise_enhancer(ctx->fixed_gain[0], &ctx->prev_tr_gain,
1190  voice_fac, stab_fac);
1191 
1192  synth_fixed_vector = anti_sparseness(ctx, ctx->fixed_vector,
1193  spare_vector);
1194 
1195  pitch_enhancer(synth_fixed_vector, voice_fac);
1196 
1197  synthesis(ctx, ctx->lp_coef[sub], synth_exc, synth_fixed_gain,
1198  synth_fixed_vector, &ctx->samples_az[LP_ORDER]);
1199 
1200  /* Synthesis speech post-processing */
1202  &ctx->samples_az[LP_ORDER], PREEMPH_FAC, ctx->demph_mem);
1203 
1206  hpf_31_gain, ctx->hpf_31_mem, AMRWB_SFR_SIZE);
1207 
1208  upsample_5_4(sub_buf, &ctx->samples_up[UPS_FIR_SIZE],
1209  AMRWB_SFR_SIZE_16k);
1210 
1211  /* High frequency band (6.4 - 7.0 kHz) generation part */
1214  hpf_400_gain, ctx->hpf_400_mem, AMRWB_SFR_SIZE);
1215 
1216  hb_gain = find_hb_gain(ctx, hb_samples,
1217  cur_subframe->hb_gain, cf->vad);
1218 
1219  scaled_hb_excitation(ctx, hb_exc, synth_exc, hb_gain);
1220 
1221  hb_synthesis(ctx, sub, &ctx->samples_hb[LP_ORDER_16k],
1222  hb_exc, ctx->isf_cur, ctx->isf_past_final);
1223 
1224  /* High-band post-processing filters */
1225  hb_fir_filter(hb_samples, bpf_6_7_coef, ctx->bpf_6_7_mem,
1226  &ctx->samples_hb[LP_ORDER_16k]);
1227 
1228  if (ctx->fr_cur_mode == MODE_23k85)
1229  hb_fir_filter(hb_samples, lpf_7_coef, ctx->lpf_7_mem,
1230  hb_samples);
1231 
1232  /* Add the low and high frequency bands */
1233  for (i = 0; i < AMRWB_SFR_SIZE_16k; i++)
1234  sub_buf[i] = (sub_buf[i] + hb_samples[i]) * (1.0f / (1 << 15));
1235 
1236  /* Update buffers and history */
1237  update_sub_state(ctx);
1238  }
1239 
1240  /* update state for next frame */
1241  memcpy(ctx->isp_sub4_past, ctx->isp[3], LP_ORDER * sizeof(ctx->isp[3][0]));
1242  memcpy(ctx->isf_past_final, ctx->isf_cur, LP_ORDER * sizeof(float));
1243 
1244  *got_frame_ptr = 1;
1245 
1246  return expected_fr_size;
1247 }
1248 
1250  .name = "amrwb",
1251  .long_name = NULL_IF_CONFIG_SMALL("AMR-WB (Adaptive Multi-Rate WideBand)"),
1252  .type = AVMEDIA_TYPE_AUDIO,
1253  .id = AV_CODEC_ID_AMR_WB,
1254  .priv_data_size = sizeof(AMRWBContext),
1257  .capabilities = CODEC_CAP_DR1,
1258  .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,
1260 };
AMRWBSubFrame subframe[4]
data for subframes
Definition: amrwbdata.h:81
Definition: lfg.h:25
AMRWBFrame frame
AMRWB parameters decoded from bitstream.
Definition: amrwbdec.c:46
static const int16_t dico2_isf[256][7]
Definition: amrwbdata.h:951
float samples_up[UPS_MEM_SIZE+AMRWB_SFR_SIZE]
low-band samples and memory processed for upsampling
Definition: amrwbdec.c:77
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:54
float hpf_31_mem[2]
Definition: amrwbdec.c:80
void ff_celp_lp_synthesis_filterf(float *out, const float *filter_coeffs, const float *in, int buffer_length, int filter_length)
LP synthesis filter.
Definition: celp_filters.c:83
AVLFG prng
random number generator for white noise excitation
Definition: amrwbdec.c:85
int size
static const uint8_t pulses_nb_per_mode_tr[][4]
[i][j] is the number of pulses present in track j at mode i
Definition: amrwbdata.h:1656
This structure describes decoded (raw) audio or video data.
Definition: frame.h:135
static const int16_t qua_gain_6b[64][2]
Tables for decoding quantized gains { pitch (Q14), fixed factor (Q11) }.
Definition: amrwbdata.h:1663
static const float lpf_7_coef[31]
Definition: amrwbdata.h:1870
float * excitation
points to current excitation in excitation_buf[]
Definition: amrwbdec.c:61
23.05 kbit/s
Definition: amrwbdata.h:59
void ff_acelp_apply_order_2_transfer_function(float *out, const float *in, const float zero_coeffs[2], const float pole_coeffs[2], float gain, float mem[2], int n)
Apply an order 2 rational transfer function in-place.
static void hb_fir_filter(float *out, const float fir_coef[HB_FIR_SIZE+1], float mem[HB_FIR_SIZE], const float *in)
Apply a 15th order filter to high-band samples.
Definition: amrwbdec.c:1035
float fixed_gain[2]
quantified fixed gains for the current and previous subframes
Definition: amrwbdec.c:68
void ff_weighted_vector_sumf(float *out, const float *in_a, const float *in_b, float weight_coeff_a, float weight_coeff_b, int length)
float implementation of weighted sum of two vectors.
static void decode_pitch_lag_low(int *lag_int, int *lag_frac, int pitch_index, uint8_t *base_lag_int, int subframe, enum Mode mode)
Decode an adaptive codebook index into pitch lag for 8k85 and 6k60 modes.
Definition: amrwbdec.c:279
float pitch_vector[AMRWB_SFR_SIZE]
adaptive codebook (pitch) vector for current subframe
Definition: amrwbdec.c:63
int acc
Definition: yuv2rgb.c:471
float prev_tr_gain
previous initial gain used by noise enhancer for threshold
Definition: amrwbdec.c:74
#define UPS_FIR_SIZE
upsampling filter size
Definition: amrwbdata.h:36
static void decode_5p_track(int *out, int code, int m, int off)
code: 5m bits
Definition: amrwbdec.c:427
#define AMRWB_P_DELAY_MAX
maximum pitch delay value
Definition: amrwbdata.h:47
int size
Definition: avcodec.h:968
static void extrapolate_isf(float isf[LP_ORDER_16k])
Extrapolate a ISF vector to the 16kHz range (20th order LP) used at mode 6k60 LP filter for the high ...
Definition: amrwbdec.c:909
static void decode_6p_track(int *out, int code, int m, int off)
code: 6m-2 bits
Definition: amrwbdec.c:437
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt)>2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);returnNULL;}returnac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_dlog(ac->avr,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> in
static float stability_factor(const float *isf, const float *isf_past)
Calculate a stability factor {teta} based on distance between current and past isf.
Definition: amrwbdec.c:678
static const int16_t dico24_isf[32][3]
Definition: amrwbdata.h:1379
static const int16_t dico23_isf[128][3]
Definition: amrwbdata.h:1312
static void isf_add_mean_and_past(float *isf_q, float *isf_past)
Apply mean and past ISF values using the prediction factor.
Definition: amrwbdec.c:204
float isf_past_final[LP_ORDER]
final processed ISF vector of the previous frame
Definition: amrwbdec.c:51
static const int16_t dico22_isf[128][3]
Definition: amrwbdata.h:1245
enum Mode fr_cur_mode
mode index of current frame
Definition: amrwbdec.c:47
static void lpc_weighting(float *out, const float *lpc, float gamma, int size)
Spectral expand the LP coefficients using the equation: y[i] = x[i] * (gamma ** i) ...
Definition: amrwbdec.c:974
uint8_t first_frame
flag active during decoding of the first frame
Definition: amrwbdec.c:86
static void pitch_enhancer(float *fixed_vector, float voice_fac)
Filter the fixed_vector to emphasize the higher frequencies.
Definition: amrwbdec.c:729
AVCodec.
Definition: avcodec.h:2790
float tilt_coef
{beta_1} related to the voicing of the previous subframe
Definition: amrwbdec.c:70
static const int16_t dico23_isf_36b[64][7]
Definition: amrwbdata.h:1551
static void scaled_hb_excitation(AMRWBContext *ctx, float *hb_exc, const float *synth_exc, float hb_gain)
Generate the high-band excitation with the same energy from the lower one and scaled by the given gai...
Definition: amrwbdec.c:871
uint16_t vq_gain
VQ adaptive and innovative gains.
Definition: amrwbdata.h:72
float avpriv_scalarproduct_float_c(const float *v1, const float *v2, int len)
Return the scalar product of two vectors.
Definition: float_dsp.c:104
static int decode(MimicContext *ctx, int quality, int num_coeffs, int is_iframe)
Definition: mimic.c:275
void void avpriv_request_sample(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1793
float lpf_7_mem[HB_FIR_SIZE]
previous values in the high-band low pass filter
Definition: amrwbdec.c:83
uint8_t
#define av_cold
Definition: attributes.h:66
static int amrwb_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: amrwbdec.c:1072
void ff_amrwb_lsp2lpc(const double *lsp, float *lp, int lp_order)
LSP to LP conversion (5.2.4 of AMR-WB)
Definition: lsp.c:143
static const int16_t isf_mean[LP_ORDER]
Means of ISF vectors in Q15.
Definition: amrwbdata.h:1619
Mode
Frame type (Table 1a in 3GPP TS 26.101)
Definition: amrnbdata.h:39
18.25 kbit/s
Definition: amrwbdata.h:57
14.25 kbit/s
Definition: amrwbdata.h:55
static const float energy_pred_fac[4]
4-tap moving average prediction coefficients in reverse order
Definition: amrnbdata.h:1463
#define CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: avcodec.h:684
uint16_t isp_id[7]
index of ISP subvectors
Definition: amrwbdata.h:80
const char data[16]
Definition: mxf.c:70
#define MIN_ISF_SPACING
minimum isf gap
Definition: amrwbdata.h:39
static const float hpf_31_gain
Definition: amrwbdata.h:1815
uint8_t * data
Definition: avcodec.h:967
#define UPS_MEM_SIZE
Definition: amrwbdata.h:37
static const float hpf_zeros[2]
High-pass filters coefficients for 31 Hz and 400 Hz cutoff.
Definition: amrwbdata.h:1813
static const float ac_inter[65]
Coefficients for FIR interpolation of excitation vector at pitch lag resulting the adaptive codebook ...
Definition: amrwbdata.h:1635
float bpf_6_7_mem[HB_FIR_SIZE]
previous values in the high-band band pass filter
Definition: amrwbdec.c:82
static const float bpf_6_7_coef[31]
High-band post-processing FIR filters coefficients from Q15.
Definition: amrwbdata.h:1856
static void ff_amr_bit_reorder(uint16_t *out, int size, const uint8_t *data, const R_TABLE_TYPE *ord_table)
Fill the frame structure variables from bitstream by parsing the given reordering table that uses the...
Definition: amr.h:51
float isf_cur[LP_ORDER]
working ISF vector from current frame
Definition: amrwbdec.c:49
static void decode_3p_track(int *out, int code, int m, int off)
code: 3m+1 bits
Definition: amrwbdec.c:382
static const float hpf_31_poles[2]
Definition: amrwbdata.h:1814
uint8_t prev_ir_filter_nr
previous impulse response filter "impNr": 0 - strong, 1 - medium, 2 - none
Definition: amrwbdec.c:73
static const float isfp_inter[4]
ISF/ISP interpolation coefficients for each subframe.
Definition: amrwbdata.h:1631
static void synthesis(AMRWBContext *ctx, float *lpc, float *excitation, float fixed_gain, const float *fixed_vector, float *samples)
Conduct 16th order linear predictive coding synthesis from excitation.
Definition: amrwbdec.c:757
static void de_emphasis(float *out, float *in, float m, float mem[1])
Apply to synthesis a de-emphasis filter of the form: H(z) = 1 / (1 - m * z^-1)
Definition: amrwbdec.c:794
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:123
static float * anti_sparseness(AMRWBContext *ctx, float *fixed_vector, float *buf)
Reduce fixed vector sparseness by smoothing with one of three IR filters, also known as "adaptive pha...
Definition: amrwbdec.c:614
6.60 kbit/s
Definition: amrwbdata.h:52
#define AMRWB_SFR_SIZE
samples per subframe at 12.8 kHz
Definition: amrwbdata.h:45
sample_fmts
Definition: avconv_filter.c:68
static void decode_1p_track(int *out, int code, int m, int off)
The next six functions decode_[i]p_track decode exactly i pulses positions and amplitudes (-1 or 1) i...
Definition: amrwbdec.c:365
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:150
static float voice_factor(float *p_vector, float p_gain, float *f_vector, float f_gain)
Calculate the voicing factor (-1.0 = unvoiced to 1.0 = voiced).
Definition: amrwbdec.c:591
float prev_sparse_fixed_gain
previous fixed gain; used by anti-sparseness to determine "onset"
Definition: amrwbdec.c:72
float isf_q_past[LP_ORDER]
quantized ISF vector of the previous frame
Definition: amrwbdec.c:50
void av_log(void *avcl, int level, const char *fmt,...)
Definition: log.c:168
const char * name
Name of the codec implementation.
Definition: avcodec.h:2797
void ff_scale_vector_to_given_sum_of_squares(float *out, const float *in, float sum_of_squares, const int n)
Set the sum of squares of a signal by scaling.
#define FFMAX(a, b)
Definition: common.h:55
static const int16_t dico21_isf_36b[128][5]
Definition: amrwbdata.h:1417
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:1846
static void interpolate_isp(double isp_q[4][LP_ORDER], const double *isp4_past)
Interpolate the fourth ISP vector from current and past frames to obtain an ISP vector for each subfr...
Definition: amrwbdec.c:224
static void decode_pitch_vector(AMRWBContext *ctx, const AMRWBSubFrame *amr_subframe, const int subframe)
Find the pitch vector by interpolating the past excitation at the pitch delay, which is obtained in t...
Definition: amrwbdec.c:308
audio channel layout utility functions
#define MIN_ENERGY
Initial energy in dB.
Definition: amrnbdec.c:82
#define FFMIN(a, b)
Definition: common.h:57
float demph_mem[1]
previous value in the de-emphasis filter
Definition: amrwbdec.c:81
double isp_sub4_past[LP_ORDER]
ISP vector for the 4th subframe of the previous frame.
Definition: amrwbdec.c:53
static const int16_t dico21_isf[64][3]
Definition: amrwbdata.h:1210
#define FFABS(a)
Definition: common.h:52
static const float * ir_filters_lookup[2]
Definition: amrnbdata.h:1658
uint16_t pul_il[4]
LSBs part of codebook index.
Definition: amrwbdata.h:75
static av_always_inline av_const float truncf(float x)
Definition: libm.h:172
static const int16_t dico25_isf[32][4]
Definition: amrwbdata.h:1398
float samples_az[LP_ORDER+AMRWB_SFR_SIZE]
low-band samples and memory from synthesis at 12.8kHz
Definition: amrwbdec.c:76
float prediction_error[4]
quantified prediction errors {20log10(^gamma_gc)} for previous four subframes
Definition: amrwbdec.c:66
static void upsample_5_4(float *out, const float *in, int o_size)
Upsample a signal by 5/4 ratio (from 12.8kHz to 16kHz) using a FIR interpolation filter.
Definition: amrwbdec.c:814
static void hb_synthesis(AMRWBContext *ctx, int subframe, float *samples, const float *exc, const float *isf, const float *isf_past)
Conduct 20th order linear predictive coding synthesis for the high frequency band excitation at 16kHz...
Definition: amrwbdec.c:996
static void decode_2p_track(int *out, int code, int m, int off)
code: 2m+1 bits
Definition: amrwbdec.c:372
float lp_coef[4][LP_ORDER]
Linear Prediction Coefficients from ISP vector.
Definition: amrwbdec.c:55
float pitch_gain[6]
quantified pitch gains for the current and previous five subframes
Definition: amrwbdec.c:67
#define LP_ORDER
linear predictive coding filter order
Definition: amrwbdata.h:33
static const uint16_t * amr_bit_orderings_by_mode[]
Reordering array addresses for each mode.
Definition: amrwbdata.h:676
#define AVERROR_PATCHWELCOME
Not yet implemented in Libav, patches welcome.
Definition: error.h:57
uint16_t pul_ih[4]
MSBs part of codebook index (high modes only)
Definition: amrwbdata.h:74
static void decode_isf_indices_46b(uint16_t *ind, float *isf_q)
Decode quantized ISF vectors using 46-bit indexes (except 6K60 mode).
Definition: amrwbdec.c:170
uint16_t vad
voice activity detection flag
Definition: amrwbdata.h:79
Libavcodec external API header.
AVSampleFormat
Audio Sample Formats.
Definition: samplefmt.h:61
void ff_celp_circ_addf(float *out, const float *in, const float *lagged, int lag, float fac, int n)
Add an array to a rotated array.
Definition: celp_filters.c:49
#define LP_ORDER_16k
lpc filter order at 16kHz
Definition: amrwbdata.h:34
AVCodec ff_amrwb_decoder
Definition: amrwbdec.c:1249
uint16_t adap
adaptive codebook index
Definition: amrwbdata.h:70
AV_SAMPLE_FMT_NONE
Definition: avconv_filter.c:68
int sample_rate
samples per second
Definition: avcodec.h:1785
main external API structure.
Definition: avcodec.h:1044
static void decode_fixed_vector(float *fixed_vector, const uint16_t *pulse_hi, const uint16_t *pulse_lo, const enum Mode mode)
Decode the algebraic codebook index to pulse positions and signs, then construct the algebraic codebo...
Definition: amrwbdec.c:481
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: utils.c:603
#define PRED_FACTOR
Definition: amrwbdata.h:40
static unsigned int av_lfg_get(AVLFG *c)
Get the next random unsigned 32-bit number using an ALFG.
Definition: lfg.h:38
float excitation_buf[AMRWB_P_DELAY_MAX+LP_ORDER+2+AMRWB_SFR_SIZE]
current excitation and all necessary excitation history
Definition: amrwbdec.c:60
static const float hpf_400_poles[2]
Definition: amrwbdata.h:1817
static av_cold int amrwb_decode_init(AVCodecContext *avctx)
Definition: amrwbdec.c:89
static const int16_t qua_gain_7b[128][2]
Definition: amrwbdata.h:1698
static const float hpf_400_gain
Definition: amrwbdata.h:1818
void ff_acelp_lsf2lspd(double *lsp, const float *lsf, int lp_order)
Floating point version of ff_acelp_lsf2lsp()
Definition: lsp.c:91
uint8_t pitch_lag_int
integer part of pitch lag of the previous subframe
Definition: amrwbdec.c:58
static float noise_enhancer(float fixed_gain, float *prev_tr_gain, float voice_fac, float stab_fac)
Apply a non-linear fixed gain smoothing in order to reduce fluctuation in the energy of excitation...
Definition: amrwbdec.c:702
av_cold void av_lfg_init(AVLFG *c, unsigned int seed)
Definition: lfg.c:30
static float auto_correlation(float *diff_isf, float mean, int lag)
Calculate the auto-correlation for the ISF difference vector.
Definition: amrwbdec.c:890
static void update_sub_state(AMRWBContext *ctx)
Update context state before the next subframe.
Definition: amrwbdec.c:1056
15.85 kbit/s
Definition: amrwbdata.h:56
#define AMRWB_SFR_SIZE_16k
samples per subframe at 16 kHz
Definition: amrwbdata.h:46
static const uint16_t cf_sizes_wb[]
Core frame sizes in each mode.
Definition: amrwbdata.h:1885
void avpriv_report_missing_feature(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
uint8_t fr_quality
frame quality index (FQI)
Definition: amrwbdec.c:48
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:141
static void decode_pitch_lag_high(int *lag_int, int *lag_frac, int pitch_index, uint8_t *base_lag_int, int subframe)
Decode an adaptive codebook index into pitch lag (except 6k60, 8k85 modes).
Definition: amrwbdec.c:246
static float find_hb_gain(AMRWBContext *ctx, const float *synth, uint16_t hb_idx, uint8_t vad)
Calculate the high-band gain based on encoded index (23k85 mode) or on the low-band speech signal and...
Definition: amrwbdec.c:846
float samples_hb[LP_ORDER_16k+AMRWB_SFR_SIZE_16k]
high-band samples and memory from synthesis at 16kHz
Definition: amrwbdec.c:78
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt)>2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);returnNULL;}returnac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_dlog(ac->avr,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> out
static const float upsample_fir[4][24]
Interpolation coefficients for 5/4 signal upsampling Table from the reference source was reordered fo...
Definition: amrwbdata.h:1822
uint8_t base_pitch_lag
integer part of pitch lag for the next relative subframe
Definition: amrwbdec.c:57
comfort noise frame
Definition: amrwbdata.h:61
static int decode_mime_header(AMRWBContext *ctx, const uint8_t *buf)
Decode the frame header in the "MIME/storage" format.
Definition: amrwbdec.c:127
23.85 kbit/s
Definition: amrwbdata.h:60
common internal api header.
common internal and external API header
#define HB_FIR_SIZE
amount of past data needed by HB filters
Definition: amrwbdata.h:35
uint16_t hb_gain
high-band energy index (mode 23k85 only)
Definition: amrwbdata.h:73
void ff_set_min_dist_lsf(float *lsf, double min_spacing, int size)
Adjust the quantized LSFs so they are increasing and not too close.
Definition: lsp.c:49
#define BIT_STR(x, lsb, len)
Get x bits in the index interval [lsb,lsb+len-1] inclusive.
Definition: amrwbdec.c:347
8.85 kbit/s
Definition: amrwbdata.h:53
static const int16_t dico1_isf[256][9]
Indexed tables for retrieval of quantized ISF vectors in Q15.
Definition: amrwbdata.h:692
static void decode_gains(const uint8_t vq_gain, const enum Mode mode, float *fixed_gain_factor, float *pitch_gain)
Decode pitch gain and fixed gain correction factor.
Definition: amrwbdec.c:552
static av_cold int init(AVCodecParserContext *s)
Definition: h264_parser.c:499
float fixed_vector[AMRWB_SFR_SIZE]
algebraic codebook (fixed) vector for current subframe
Definition: amrwbdec.c:64
void * priv_data
Definition: avcodec.h:1086
#define ENERGY_MEAN
mean innovation energy (dB) in all modes
Definition: amrwbdata.h:42
#define PREEMPH_FAC
factor used to de-emphasize synthesis
Definition: amrwbdata.h:43
static const int16_t dico22_isf_36b[128][4]
Definition: amrwbdata.h:1484
int channels
number of audio channels
Definition: avcodec.h:1786
AMR wideband data and definitions.
19.85 kbit/s
Definition: amrwbdata.h:58
void ff_acelp_interpolatef(float *out, const float *in, const float *filter_coeffs, int precision, int frac_pos, int filter_length, int length)
Floating point version of ff_acelp_interpolate()
Definition: acelp_filters.c:77
float hpf_400_mem[2]
previous values in the high pass filters
Definition: amrwbdec.c:80
static void pitch_sharpening(AMRWBContext *ctx, float *fixed_vector)
Apply pitch sharpening filters to the fixed codebook vector.
Definition: amrwbdec.c:570
static const int16_t isf_init[LP_ORDER]
Initialization tables for the processed ISF vector in Q15.
Definition: amrwbdata.h:1625
#define BIT_POS(x, p)
Get the bit at specified position.
Definition: amrwbdec.c:350
static void decode_isf_indices_36b(uint16_t *ind, float *isf_q)
Decode quantized ISF vectors using 36-bit indexes (6K60 mode only).
Definition: amrwbdec.c:143
static const uint16_t qua_hb_gain[16]
High band quantized gains for 23k85 in Q14.
Definition: amrwbdata.h:1850
#define AV_CH_LAYOUT_MONO
static void decode_4p_track(int *out, int code, int m, int off)
code: 4m bits
Definition: amrwbdec.c:391
This structure stores compressed data.
Definition: avcodec.h:944
uint16_t ltp
ltp-filtering flag
Definition: amrwbdata.h:71
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:179
double isp[4][LP_ORDER]
ISP vectors from current frame.
Definition: amrwbdec.c:52
for(j=16;j >0;--j)
float ff_amr_set_fixed_gain(float fixed_gain_factor, float fixed_mean_energy, float *prediction_error, float energy_mean, const float *pred_table)
Calculate fixed gain (part of section 6.1.3 of AMR spec)
12.65 kbit/s
Definition: amrwbdata.h:54
#define AMRWB_P_DELAY_MIN
Definition: amrwbdata.h:48