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qcelpdec.c
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1 /*
2  * QCELP decoder
3  * Copyright (c) 2007 Reynaldo H. Verdejo Pinochet
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg 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  * FFmpeg 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 FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
22 /**
23  * @file
24  * QCELP decoder
25  * @author Reynaldo H. Verdejo Pinochet
26  * @remark FFmpeg merging spearheaded by Kenan Gillet
27  * @remark Development mentored by Benjamin Larson
28  */
29 
30 #include <stddef.h>
31 
32 #include "libavutil/avassert.h"
34 #include "libavutil/float_dsp.h"
35 #include "avcodec.h"
36 #include "internal.h"
37 #include "get_bits.h"
38 #include "qcelpdata.h"
39 #include "celp_filters.h"
40 #include "acelp_filters.h"
41 #include "acelp_vectors.h"
42 #include "lsp.h"
43 
44 typedef enum {
45  I_F_Q = -1, /**< insufficient frame quality */
52 
53 typedef struct QCELPContext {
56  QCELPFrame frame; /**< unpacked data frame */
57 
59  uint8_t octave_count; /**< count the consecutive RATE_OCTAVE frames */
60  float prev_lspf[10];
61  float predictor_lspf[10];/**< LSP predictor for RATE_OCTAVE and I_F_Q */
64  float rnd_fir_filter_mem[180];
65  float formant_mem[170];
67  int prev_g1[2];
69  float pitch_gain[4];
71  uint16_t first16bits;
73 
74  /* postfilter */
78 } QCELPContext;
79 
80 /**
81  * Initialize the speech codec according to the specification.
82  *
83  * TIA/EIA/IS-733 2.4.9
84  */
86 {
87  QCELPContext *q = avctx->priv_data;
88  int i;
89 
90  avctx->channels = 1;
93 
94  for (i = 0; i < 10; i++)
95  q->prev_lspf[i] = (i + 1) / 11.0;
96 
97  return 0;
98 }
99 
100 /**
101  * Decode the 10 quantized LSP frequencies from the LSPV/LSP
102  * transmission codes of any bitrate and check for badly received packets.
103  *
104  * @param q the context
105  * @param lspf line spectral pair frequencies
106  *
107  * @return 0 on success, -1 if the packet is badly received
108  *
109  * TIA/EIA/IS-733 2.4.3.2.6.2-2, 2.4.8.7.3
110  */
111 static int decode_lspf(QCELPContext *q, float *lspf)
112 {
113  int i;
114  float tmp_lspf, smooth, erasure_coeff;
115  const float *predictors;
116 
117  if (q->bitrate == RATE_OCTAVE || q->bitrate == I_F_Q) {
118  predictors = q->prev_bitrate != RATE_OCTAVE &&
119  q->prev_bitrate != I_F_Q ? q->prev_lspf
120  : q->predictor_lspf;
121 
122  if (q->bitrate == RATE_OCTAVE) {
123  q->octave_count++;
124 
125  for (i = 0; i < 10; i++) {
126  q->predictor_lspf[i] =
127  lspf[i] = (q->frame.lspv[i] ? QCELP_LSP_SPREAD_FACTOR
129  predictors[i] * QCELP_LSP_OCTAVE_PREDICTOR +
130  (i + 1) * ((1 - QCELP_LSP_OCTAVE_PREDICTOR) / 11);
131  }
132  smooth = q->octave_count < 10 ? .875 : 0.1;
133  } else {
134  erasure_coeff = QCELP_LSP_OCTAVE_PREDICTOR;
135 
136  av_assert2(q->bitrate == I_F_Q);
137 
138  if (q->erasure_count > 1)
139  erasure_coeff *= q->erasure_count < 4 ? 0.9 : 0.7;
140 
141  for (i = 0; i < 10; i++) {
142  q->predictor_lspf[i] =
143  lspf[i] = (i + 1) * (1 - erasure_coeff) / 11 +
144  erasure_coeff * predictors[i];
145  }
146  smooth = 0.125;
147  }
148 
149  // Check the stability of the LSP frequencies.
150  lspf[0] = FFMAX(lspf[0], QCELP_LSP_SPREAD_FACTOR);
151  for (i = 1; i < 10; i++)
152  lspf[i] = FFMAX(lspf[i], lspf[i - 1] + QCELP_LSP_SPREAD_FACTOR);
153 
154  lspf[9] = FFMIN(lspf[9], 1.0 - QCELP_LSP_SPREAD_FACTOR);
155  for (i = 9; i > 0; i--)
156  lspf[i - 1] = FFMIN(lspf[i - 1], lspf[i] - QCELP_LSP_SPREAD_FACTOR);
157 
158  // Low-pass filter the LSP frequencies.
159  ff_weighted_vector_sumf(lspf, lspf, q->prev_lspf, smooth, 1.0 - smooth, 10);
160  } else {
161  q->octave_count = 0;
162 
163  tmp_lspf = 0.0;
164  for (i = 0; i < 5; i++) {
165  lspf[2 * i + 0] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][0] * 0.0001;
166  lspf[2 * i + 1] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][1] * 0.0001;
167  }
168 
169  // Check for badly received packets.
170  if (q->bitrate == RATE_QUARTER) {
171  if (lspf[9] <= .70 || lspf[9] >= .97)
172  return -1;
173  for (i = 3; i < 10; i++)
174  if (fabs(lspf[i] - lspf[i - 2]) < .08)
175  return -1;
176  } else {
177  if (lspf[9] <= .66 || lspf[9] >= .985)
178  return -1;
179  for (i = 4; i < 10; i++)
180  if (fabs(lspf[i] - lspf[i - 4]) < .0931)
181  return -1;
182  }
183  }
184  return 0;
185 }
186 
187 /**
188  * Convert codebook transmission codes to GAIN and INDEX.
189  *
190  * @param q the context
191  * @param gain array holding the decoded gain
192  *
193  * TIA/EIA/IS-733 2.4.6.2
194  */
195 static void decode_gain_and_index(QCELPContext *q, float *gain)
196 {
197  int i, subframes_count, g1[16];
198  float slope;
199 
200  if (q->bitrate >= RATE_QUARTER) {
201  switch (q->bitrate) {
202  case RATE_FULL: subframes_count = 16; break;
203  case RATE_HALF: subframes_count = 4; break;
204  default: subframes_count = 5;
205  }
206  for (i = 0; i < subframes_count; i++) {
207  g1[i] = 4 * q->frame.cbgain[i];
208  if (q->bitrate == RATE_FULL && !((i + 1) & 3)) {
209  g1[i] += av_clip((g1[i - 1] + g1[i - 2] + g1[i - 3]) / 3 - 6, 0, 32);
210  }
211 
212  gain[i] = qcelp_g12ga[g1[i]];
213 
214  if (q->frame.cbsign[i]) {
215  gain[i] = -gain[i];
216  q->frame.cindex[i] = (q->frame.cindex[i] - 89) & 127;
217  }
218  }
219 
220  q->prev_g1[0] = g1[i - 2];
221  q->prev_g1[1] = g1[i - 1];
222  q->last_codebook_gain = qcelp_g12ga[g1[i - 1]];
223 
224  if (q->bitrate == RATE_QUARTER) {
225  // Provide smoothing of the unvoiced excitation energy.
226  gain[7] = gain[4];
227  gain[6] = 0.4 * gain[3] + 0.6 * gain[4];
228  gain[5] = gain[3];
229  gain[4] = 0.8 * gain[2] + 0.2 * gain[3];
230  gain[3] = 0.2 * gain[1] + 0.8 * gain[2];
231  gain[2] = gain[1];
232  gain[1] = 0.6 * gain[0] + 0.4 * gain[1];
233  }
234  } else if (q->bitrate != SILENCE) {
235  if (q->bitrate == RATE_OCTAVE) {
236  g1[0] = 2 * q->frame.cbgain[0] +
237  av_clip((q->prev_g1[0] + q->prev_g1[1]) / 2 - 5, 0, 54);
238  subframes_count = 8;
239  } else {
240  av_assert2(q->bitrate == I_F_Q);
241 
242  g1[0] = q->prev_g1[1];
243  switch (q->erasure_count) {
244  case 1 : break;
245  case 2 : g1[0] -= 1; break;
246  case 3 : g1[0] -= 2; break;
247  default: g1[0] -= 6;
248  }
249  if (g1[0] < 0)
250  g1[0] = 0;
251  subframes_count = 4;
252  }
253  // This interpolation is done to produce smoother background noise.
254  slope = 0.5 * (qcelp_g12ga[g1[0]] - q->last_codebook_gain) / subframes_count;
255  for (i = 1; i <= subframes_count; i++)
256  gain[i - 1] = q->last_codebook_gain + slope * i;
257 
258  q->last_codebook_gain = gain[i - 2];
259  q->prev_g1[0] = q->prev_g1[1];
260  q->prev_g1[1] = g1[0];
261  }
262 }
263 
264 /**
265  * If the received packet is Rate 1/4 a further sanity check is made of the
266  * codebook gain.
267  *
268  * @param cbgain the unpacked cbgain array
269  * @return -1 if the sanity check fails, 0 otherwise
270  *
271  * TIA/EIA/IS-733 2.4.8.7.3
272  */
274 {
275  int i, diff, prev_diff = 0;
276 
277  for (i = 1; i < 5; i++) {
278  diff = cbgain[i] - cbgain[i-1];
279  if (FFABS(diff) > 10)
280  return -1;
281  else if (FFABS(diff - prev_diff) > 12)
282  return -1;
283  prev_diff = diff;
284  }
285  return 0;
286 }
287 
288 /**
289  * Compute the scaled codebook vector Cdn From INDEX and GAIN
290  * for all rates.
291  *
292  * The specification lacks some information here.
293  *
294  * TIA/EIA/IS-733 has an omission on the codebook index determination
295  * formula for RATE_FULL and RATE_HALF frames at section 2.4.8.1.1. It says
296  * you have to subtract the decoded index parameter from the given scaled
297  * codebook vector index 'n' to get the desired circular codebook index, but
298  * it does not mention that you have to clamp 'n' to [0-9] in order to get
299  * RI-compliant results.
300  *
301  * The reason for this mistake seems to be the fact they forgot to mention you
302  * have to do these calculations per codebook subframe and adjust given
303  * equation values accordingly.
304  *
305  * @param q the context
306  * @param gain array holding the 4 pitch subframe gain values
307  * @param cdn_vector array for the generated scaled codebook vector
308  */
309 static void compute_svector(QCELPContext *q, const float *gain,
310  float *cdn_vector)
311 {
312  int i, j, k;
313  uint16_t cbseed, cindex;
314  float *rnd, tmp_gain, fir_filter_value;
315 
316  switch (q->bitrate) {
317  case RATE_FULL:
318  for (i = 0; i < 16; i++) {
319  tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
320  cindex = -q->frame.cindex[i];
321  for (j = 0; j < 10; j++)
322  *cdn_vector++ = tmp_gain *
323  qcelp_rate_full_codebook[cindex++ & 127];
324  }
325  break;
326  case RATE_HALF:
327  for (i = 0; i < 4; i++) {
328  tmp_gain = gain[i] * QCELP_RATE_HALF_CODEBOOK_RATIO;
329  cindex = -q->frame.cindex[i];
330  for (j = 0; j < 40; j++)
331  *cdn_vector++ = tmp_gain *
332  qcelp_rate_half_codebook[cindex++ & 127];
333  }
334  break;
335  case RATE_QUARTER:
336  cbseed = (0x0003 & q->frame.lspv[4]) << 14 |
337  (0x003F & q->frame.lspv[3]) << 8 |
338  (0x0060 & q->frame.lspv[2]) << 1 |
339  (0x0007 & q->frame.lspv[1]) << 3 |
340  (0x0038 & q->frame.lspv[0]) >> 3;
341  rnd = q->rnd_fir_filter_mem + 20;
342  for (i = 0; i < 8; i++) {
343  tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
344  for (k = 0; k < 20; k++) {
345  cbseed = 521 * cbseed + 259;
346  *rnd = (int16_t) cbseed;
347 
348  // FIR filter
349  fir_filter_value = 0.0;
350  for (j = 0; j < 10; j++)
351  fir_filter_value += qcelp_rnd_fir_coefs[j] *
352  (rnd[-j] + rnd[-20+j]);
353 
354  fir_filter_value += qcelp_rnd_fir_coefs[10] * rnd[-10];
355  *cdn_vector++ = tmp_gain * fir_filter_value;
356  rnd++;
357  }
358  }
359  memcpy(q->rnd_fir_filter_mem, q->rnd_fir_filter_mem + 160,
360  20 * sizeof(float));
361  break;
362  case RATE_OCTAVE:
363  cbseed = q->first16bits;
364  for (i = 0; i < 8; i++) {
365  tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
366  for (j = 0; j < 20; j++) {
367  cbseed = 521 * cbseed + 259;
368  *cdn_vector++ = tmp_gain * (int16_t) cbseed;
369  }
370  }
371  break;
372  case I_F_Q:
373  cbseed = -44; // random codebook index
374  for (i = 0; i < 4; i++) {
375  tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
376  for (j = 0; j < 40; j++)
377  *cdn_vector++ = tmp_gain *
378  qcelp_rate_full_codebook[cbseed++ & 127];
379  }
380  break;
381  case SILENCE:
382  memset(cdn_vector, 0, 160 * sizeof(float));
383  break;
384  }
385 }
386 
387 /**
388  * Apply generic gain control.
389  *
390  * @param v_out output vector
391  * @param v_in gain-controlled vector
392  * @param v_ref vector to control gain of
393  *
394  * TIA/EIA/IS-733 2.4.8.3, 2.4.8.6
395  */
396 static void apply_gain_ctrl(float *v_out, const float *v_ref, const float *v_in)
397 {
398  int i;
399 
400  for (i = 0; i < 160; i += 40) {
401  float res = avpriv_scalarproduct_float_c(v_ref + i, v_ref + i, 40);
402  ff_scale_vector_to_given_sum_of_squares(v_out + i, v_in + i, res, 40);
403  }
404 }
405 
406 /**
407  * Apply filter in pitch-subframe steps.
408  *
409  * @param memory buffer for the previous state of the filter
410  * - must be able to contain 303 elements
411  * - the 143 first elements are from the previous state
412  * - the next 160 are for output
413  * @param v_in input filter vector
414  * @param gain per-subframe gain array, each element is between 0.0 and 2.0
415  * @param lag per-subframe lag array, each element is
416  * - between 16 and 143 if its corresponding pfrac is 0,
417  * - between 16 and 139 otherwise
418  * @param pfrac per-subframe boolean array, 1 if the lag is fractional, 0
419  * otherwise
420  *
421  * @return filter output vector
422  */
423 static const float *do_pitchfilter(float memory[303], const float v_in[160],
424  const float gain[4], const uint8_t *lag,
425  const uint8_t pfrac[4])
426 {
427  int i, j;
428  float *v_lag, *v_out;
429  const float *v_len;
430 
431  v_out = memory + 143; // Output vector starts at memory[143].
432 
433  for (i = 0; i < 4; i++) {
434  if (gain[i]) {
435  v_lag = memory + 143 + 40 * i - lag[i];
436  for (v_len = v_in + 40; v_in < v_len; v_in++) {
437  if (pfrac[i]) { // If it is a fractional lag...
438  for (j = 0, *v_out = 0.0; j < 4; j++)
439  *v_out += qcelp_hammsinc_table[j] *
440  (v_lag[j - 4] + v_lag[3 - j]);
441  } else
442  *v_out = *v_lag;
443 
444  *v_out = *v_in + gain[i] * *v_out;
445 
446  v_lag++;
447  v_out++;
448  }
449  } else {
450  memcpy(v_out, v_in, 40 * sizeof(float));
451  v_in += 40;
452  v_out += 40;
453  }
454  }
455 
456  memmove(memory, memory + 160, 143 * sizeof(float));
457  return memory + 143;
458 }
459 
460 /**
461  * Apply pitch synthesis filter and pitch prefilter to the scaled codebook vector.
462  * TIA/EIA/IS-733 2.4.5.2, 2.4.8.7.2
463  *
464  * @param q the context
465  * @param cdn_vector the scaled codebook vector
466  */
467 static void apply_pitch_filters(QCELPContext *q, float *cdn_vector)
468 {
469  int i;
470  const float *v_synthesis_filtered, *v_pre_filtered;
471 
472  if (q->bitrate >= RATE_HALF || q->bitrate == SILENCE ||
473  (q->bitrate == I_F_Q && (q->prev_bitrate >= RATE_HALF))) {
474 
475  if (q->bitrate >= RATE_HALF) {
476  // Compute gain & lag for the whole frame.
477  for (i = 0; i < 4; i++) {
478  q->pitch_gain[i] = q->frame.plag[i] ? (q->frame.pgain[i] + 1) * 0.25 : 0.0;
479 
480  q->pitch_lag[i] = q->frame.plag[i] + 16;
481  }
482  } else {
483  float max_pitch_gain;
484 
485  if (q->bitrate == I_F_Q) {
486  if (q->erasure_count < 3)
487  max_pitch_gain = 0.9 - 0.3 * (q->erasure_count - 1);
488  else
489  max_pitch_gain = 0.0;
490  } else {
491  av_assert2(q->bitrate == SILENCE);
492  max_pitch_gain = 1.0;
493  }
494  for (i = 0; i < 4; i++)
495  q->pitch_gain[i] = FFMIN(q->pitch_gain[i], max_pitch_gain);
496 
497  memset(q->frame.pfrac, 0, sizeof(q->frame.pfrac));
498  }
499 
500  // pitch synthesis filter
501  v_synthesis_filtered = do_pitchfilter(q->pitch_synthesis_filter_mem,
502  cdn_vector, q->pitch_gain,
503  q->pitch_lag, q->frame.pfrac);
504 
505  // pitch prefilter update
506  for (i = 0; i < 4; i++)
507  q->pitch_gain[i] = 0.5 * FFMIN(q->pitch_gain[i], 1.0);
508 
509  v_pre_filtered = do_pitchfilter(q->pitch_pre_filter_mem,
510  v_synthesis_filtered,
511  q->pitch_gain, q->pitch_lag,
512  q->frame.pfrac);
513 
514  apply_gain_ctrl(cdn_vector, v_synthesis_filtered, v_pre_filtered);
515  } else {
516  memcpy(q->pitch_synthesis_filter_mem,
517  cdn_vector + 17, 143 * sizeof(float));
518  memcpy(q->pitch_pre_filter_mem, cdn_vector + 17, 143 * sizeof(float));
519  memset(q->pitch_gain, 0, sizeof(q->pitch_gain));
520  memset(q->pitch_lag, 0, sizeof(q->pitch_lag));
521  }
522 }
523 
524 /**
525  * Reconstruct LPC coefficients from the line spectral pair frequencies
526  * and perform bandwidth expansion.
527  *
528  * @param lspf line spectral pair frequencies
529  * @param lpc linear predictive coding coefficients
530  *
531  * @note: bandwidth_expansion_coeff could be precalculated into a table
532  * but it seems to be slower on x86
533  *
534  * TIA/EIA/IS-733 2.4.3.3.5
535  */
536 static void lspf2lpc(const float *lspf, float *lpc)
537 {
538  double lsp[10];
539  double bandwidth_expansion_coeff = QCELP_BANDWIDTH_EXPANSION_COEFF;
540  int i;
541 
542  for (i = 0; i < 10; i++)
543  lsp[i] = cos(M_PI * lspf[i]);
544 
545  ff_acelp_lspd2lpc(lsp, lpc, 5);
546 
547  for (i = 0; i < 10; i++) {
548  lpc[i] *= bandwidth_expansion_coeff;
549  bandwidth_expansion_coeff *= QCELP_BANDWIDTH_EXPANSION_COEFF;
550  }
551 }
552 
553 /**
554  * Interpolate LSP frequencies and compute LPC coefficients
555  * for a given bitrate & pitch subframe.
556  *
557  * TIA/EIA/IS-733 2.4.3.3.4, 2.4.8.7.2
558  *
559  * @param q the context
560  * @param curr_lspf LSP frequencies vector of the current frame
561  * @param lpc float vector for the resulting LPC
562  * @param subframe_num frame number in decoded stream
563  */
564 static void interpolate_lpc(QCELPContext *q, const float *curr_lspf,
565  float *lpc, const int subframe_num)
566 {
567  float interpolated_lspf[10];
568  float weight;
569 
570  if (q->bitrate >= RATE_QUARTER)
571  weight = 0.25 * (subframe_num + 1);
572  else if (q->bitrate == RATE_OCTAVE && !subframe_num)
573  weight = 0.625;
574  else
575  weight = 1.0;
576 
577  if (weight != 1.0) {
578  ff_weighted_vector_sumf(interpolated_lspf, curr_lspf, q->prev_lspf,
579  weight, 1.0 - weight, 10);
580  lspf2lpc(interpolated_lspf, lpc);
581  } else if (q->bitrate >= RATE_QUARTER ||
582  (q->bitrate == I_F_Q && !subframe_num))
583  lspf2lpc(curr_lspf, lpc);
584  else if (q->bitrate == SILENCE && !subframe_num)
585  lspf2lpc(q->prev_lspf, lpc);
586 }
587 
588 static qcelp_packet_rate buf_size2bitrate(const int buf_size)
589 {
590  switch (buf_size) {
591  case 35: return RATE_FULL;
592  case 17: return RATE_HALF;
593  case 8: return RATE_QUARTER;
594  case 4: return RATE_OCTAVE;
595  case 1: return SILENCE;
596  }
597 
598  return I_F_Q;
599 }
600 
601 /**
602  * Determine the bitrate from the frame size and/or the first byte of the frame.
603  *
604  * @param avctx the AV codec context
605  * @param buf_size length of the buffer
606  * @param buf the bufffer
607  *
608  * @return the bitrate on success,
609  * I_F_Q if the bitrate cannot be satisfactorily determined
610  *
611  * TIA/EIA/IS-733 2.4.8.7.1
612  */
614  const int buf_size,
615  const uint8_t **buf)
616 {
617  qcelp_packet_rate bitrate;
618 
619  if ((bitrate = buf_size2bitrate(buf_size)) >= 0) {
620  if (bitrate > **buf) {
621  QCELPContext *q = avctx->priv_data;
622  if (!q->warned_buf_mismatch_bitrate) {
623  av_log(avctx, AV_LOG_WARNING,
624  "Claimed bitrate and buffer size mismatch.\n");
626  }
627  bitrate = **buf;
628  } else if (bitrate < **buf) {
629  av_log(avctx, AV_LOG_ERROR,
630  "Buffer is too small for the claimed bitrate.\n");
631  return I_F_Q;
632  }
633  (*buf)++;
634  } else if ((bitrate = buf_size2bitrate(buf_size + 1)) >= 0) {
635  av_log(avctx, AV_LOG_WARNING,
636  "Bitrate byte missing, guessing bitrate from packet size.\n");
637  } else
638  return I_F_Q;
639 
640  if (bitrate == SILENCE) {
641  // FIXME: Remove this warning when tested with samples.
642  avpriv_request_sample(avctx, "Blank frame handling");
643  }
644  return bitrate;
645 }
646 
648  const char *message)
649 {
650  av_log(avctx, AV_LOG_WARNING, "Frame #%d, IFQ: %s\n",
651  avctx->frame_number, message);
652 }
653 
654 static void postfilter(QCELPContext *q, float *samples, float *lpc)
655 {
656  static const float pow_0_775[10] = {
657  0.775000, 0.600625, 0.465484, 0.360750, 0.279582,
658  0.216676, 0.167924, 0.130141, 0.100859, 0.078166
659  }, pow_0_625[10] = {
660  0.625000, 0.390625, 0.244141, 0.152588, 0.095367,
661  0.059605, 0.037253, 0.023283, 0.014552, 0.009095
662  };
663  float lpc_s[10], lpc_p[10], pole_out[170], zero_out[160];
664  int n;
665 
666  for (n = 0; n < 10; n++) {
667  lpc_s[n] = lpc[n] * pow_0_625[n];
668  lpc_p[n] = lpc[n] * pow_0_775[n];
669  }
670 
671  ff_celp_lp_zero_synthesis_filterf(zero_out, lpc_s,
672  q->formant_mem + 10, 160, 10);
673  memcpy(pole_out, q->postfilter_synth_mem, sizeof(float) * 10);
674  ff_celp_lp_synthesis_filterf(pole_out + 10, lpc_p, zero_out, 160, 10);
675  memcpy(q->postfilter_synth_mem, pole_out + 160, sizeof(float) * 10);
676 
677  ff_tilt_compensation(&q->postfilter_tilt_mem, 0.3, pole_out + 10, 160);
678 
679  ff_adaptive_gain_control(samples, pole_out + 10,
681  q->formant_mem + 10,
682  160),
683  160, 0.9375, &q->postfilter_agc_mem);
684 }
685 
686 static int qcelp_decode_frame(AVCodecContext *avctx, void *data,
687  int *got_frame_ptr, AVPacket *avpkt)
688 {
689  const uint8_t *buf = avpkt->data;
690  int buf_size = avpkt->size;
691  QCELPContext *q = avctx->priv_data;
692  AVFrame *frame = data;
693  float *outbuffer;
694  int i, ret;
695  float quantized_lspf[10], lpc[10];
696  float gain[16];
697  float *formant_mem;
698 
699  /* get output buffer */
700  frame->nb_samples = 160;
701  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
702  return ret;
703  outbuffer = (float *)frame->data[0];
704 
705  if ((q->bitrate = determine_bitrate(avctx, buf_size, &buf)) == I_F_Q) {
706  warn_insufficient_frame_quality(avctx, "Bitrate cannot be determined.");
707  goto erasure;
708  }
709 
710  if (q->bitrate == RATE_OCTAVE &&
711  (q->first16bits = AV_RB16(buf)) == 0xFFFF) {
712  warn_insufficient_frame_quality(avctx, "Bitrate is 1/8 and first 16 bits are on.");
713  goto erasure;
714  }
715 
716  if (q->bitrate > SILENCE) {
718  const QCELPBitmap *bitmaps_end = qcelp_unpacking_bitmaps_per_rate[q->bitrate] +
720  uint8_t *unpacked_data = (uint8_t *)&q->frame;
721 
722  if ((ret = init_get_bits8(&q->gb, buf, buf_size)) < 0)
723  return ret;
724 
725  memset(&q->frame, 0, sizeof(QCELPFrame));
726 
727  for (; bitmaps < bitmaps_end; bitmaps++)
728  unpacked_data[bitmaps->index] |= get_bits(&q->gb, bitmaps->bitlen) << bitmaps->bitpos;
729 
730  // Check for erasures/blanks on rates 1, 1/4 and 1/8.
731  if (q->frame.reserved) {
732  warn_insufficient_frame_quality(avctx, "Wrong data in reserved frame area.");
733  goto erasure;
734  }
735  if (q->bitrate == RATE_QUARTER &&
737  warn_insufficient_frame_quality(avctx, "Codebook gain sanity check failed.");
738  goto erasure;
739  }
740 
741  if (q->bitrate >= RATE_HALF) {
742  for (i = 0; i < 4; i++) {
743  if (q->frame.pfrac[i] && q->frame.plag[i] >= 124) {
744  warn_insufficient_frame_quality(avctx, "Cannot initialize pitch filter.");
745  goto erasure;
746  }
747  }
748  }
749  }
750 
751  decode_gain_and_index(q, gain);
752  compute_svector(q, gain, outbuffer);
753 
754  if (decode_lspf(q, quantized_lspf) < 0) {
755  warn_insufficient_frame_quality(avctx, "Badly received packets in frame.");
756  goto erasure;
757  }
758 
759  apply_pitch_filters(q, outbuffer);
760 
761  if (q->bitrate == I_F_Q) {
762 erasure:
763  q->bitrate = I_F_Q;
764  q->erasure_count++;
765  decode_gain_and_index(q, gain);
766  compute_svector(q, gain, outbuffer);
767  decode_lspf(q, quantized_lspf);
768  apply_pitch_filters(q, outbuffer);
769  } else
770  q->erasure_count = 0;
771 
772  formant_mem = q->formant_mem + 10;
773  for (i = 0; i < 4; i++) {
774  interpolate_lpc(q, quantized_lspf, lpc, i);
775  ff_celp_lp_synthesis_filterf(formant_mem, lpc,
776  outbuffer + i * 40, 40, 10);
777  formant_mem += 40;
778  }
779 
780  // postfilter, as per TIA/EIA/IS-733 2.4.8.6
781  postfilter(q, outbuffer, lpc);
782 
783  memcpy(q->formant_mem, q->formant_mem + 160, 10 * sizeof(float));
784 
785  memcpy(q->prev_lspf, quantized_lspf, sizeof(q->prev_lspf));
786  q->prev_bitrate = q->bitrate;
787 
788  *got_frame_ptr = 1;
789 
790  return buf_size;
791 }
792 
794  .name = "qcelp",
795  .long_name = NULL_IF_CONFIG_SMALL("QCELP / PureVoice"),
796  .type = AVMEDIA_TYPE_AUDIO,
797  .id = AV_CODEC_ID_QCELP,
798  .init = qcelp_decode_init,
799  .decode = qcelp_decode_frame,
800  .capabilities = AV_CODEC_CAP_DR1,
801  .priv_data_size = sizeof(QCELPContext),
802 };
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:84
This structure describes decoded (raw) audio or video data.
Definition: frame.h:171
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:101
float formant_mem[170]
Definition: qcelpdec.c:65
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:260
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
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 apply_pitch_filters(QCELPContext *q, float *cdn_vector)
Apply pitch synthesis filter and pitch prefilter to the scaled codebook vector.
Definition: qcelpdec.c:467
uint8_t pfrac[4]
fractional pitch lag for each pitch subframe
Definition: qcelpdata.h:51
int size
Definition: avcodec.h:1423
#define QCELP_RATE_FULL_CODEBOOK_RATIO
Definition: qcelpdata.h:477
static void warn_insufficient_frame_quality(AVCodecContext *avctx, const char *message)
Definition: qcelpdec.c:647
static qcelp_packet_rate buf_size2bitrate(const int buf_size)
Definition: qcelpdec.c:588
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_WB64 unsigned int_TMPL AV_WB32 unsigned int_TMPL AV_WB24 unsigned int_TMPL AV_RB16
Definition: bytestream.h:87
static const int8_t qcelp_rate_half_codebook[128]
Circular codebook for rate 1/2 frames in x*2 form.
Definition: qcelpdata.h:484
AVCodec.
Definition: avcodec.h:3471
static const float qcelp_hammsinc_table[4]
Pre-calculated table for hammsinc function.
Definition: qcelpdata.h:74
static int decode_lspf(QCELPContext *q, float *lspf)
Decode the 10 quantized LSP frequencies from the LSPV/LSP transmission codes of any bitrate and check...
Definition: qcelpdec.c:111
uint8_t index
index into the QCELPContext structure
Definition: qcelpdata.h:77
insufficient frame quality
Definition: qcelpdec.c:45
uint8_t warned_buf_mismatch_bitrate
Definition: qcelpdec.c:72
uint8_t octave_count
count the consecutive RATE_OCTAVE frames
Definition: qcelpdec.c:59
QCELP unpacked data frame.
Definition: qcelpdata.h:40
float avpriv_scalarproduct_float_c(const float *v1, const float *v2, int len)
Return the scalar product of two vectors.
Definition: float_dsp.c:108
void void avpriv_request_sample(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
uint8_t cindex[16]
codebook index for each codebook subframe
Definition: qcelpdata.h:45
static int codebook_sanity_check_for_rate_quarter(const uint8_t *cbgain)
If the received packet is Rate 1/4 a further sanity check is made of the codebook gain...
Definition: qcelpdec.c:273
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2269
float prev_lspf[10]
Definition: qcelpdec.c:60
uint8_t
#define av_cold
Definition: attributes.h:74
#define av_assert2(cond)
assert() equivalent, that does lie in speed critical code.
Definition: avassert.h:63
static AVFrame * frame
static const qcelp_vector *const qcelp_lspvq[5]
Definition: qcelpdata.h:414
uint8_t * data
Definition: avcodec.h:1422
void ff_adaptive_gain_control(float *out, const float *in, float speech_energ, int size, float alpha, float *gain_mem)
Adaptive gain control (as used in AMR postfiltering)
bitstream reader API header.
uint8_t plag[4]
pitch lag for each pitch subframe
Definition: qcelpdata.h:50
uint8_t cbsign[16]
sign of the codebook gain for each codebook subframe
Definition: qcelpdata.h:43
#define av_log(a,...)
#define QCELP_LSP_OCTAVE_PREDICTOR
Predictor coefficient for the conversion of LSP codes to LSP frequencies for 1/8 and I_F_Q...
Definition: qcelpdata.h:541
uint8_t lspv[10]
line spectral pair frequencies (LSP) for RATE_OCTAVE, line spectral pair frequencies grouped into fiv...
Definition: qcelpdata.h:60
static qcelp_packet_rate determine_bitrate(AVCodecContext *avctx, const int buf_size, const uint8_t **buf)
Determine the bitrate from the frame size and/or the first byte of the frame.
Definition: qcelpdec.c:613
static void interpolate_lpc(QCELPContext *q, const float *curr_lspf, float *lpc, const int subframe_num)
Interpolate LSP frequencies and compute LPC coefficients for a given bitrate & pitch subframe...
Definition: qcelpdec.c:564
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
uint8_t erasure_count
Definition: qcelpdec.c:58
static void decode_gain_and_index(QCELPContext *q, float *gain)
Convert codebook transmission codes to GAIN and INDEX.
Definition: qcelpdec.c:195
float postfilter_agc_mem
Definition: qcelpdec.c:76
float pitch_pre_filter_mem[303]
Definition: qcelpdec.c:63
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:175
simple assert() macros that are a bit more flexible than ISO C assert().
const char * name
Name of the codec implementation.
Definition: avcodec.h:3478
static const int16_t qcelp_rate_full_codebook[128]
Circular codebook for rate 1 frames in x*100 form.
Definition: qcelpdata.h:459
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:64
Libavcodec external API header.
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:2322
static void compute_svector(QCELPContext *q, const float *gain, float *cdn_vector)
Compute the scaled codebook vector Cdn From INDEX and GAIN for all rates.
Definition: qcelpdec.c:309
#define QCELP_SQRT1887
sqrt(1.887) is the maximum of the pseudorandom white sequence used to generate the scaled codebook ve...
Definition: qcelpdata.h:511
audio channel layout utility functions
static void postfilter(QCELPContext *q, float *samples, float *lpc)
Definition: qcelpdec.c:654
#define FFMIN(a, b)
Definition: common.h:66
static int qcelp_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: qcelpdec.c:686
uint8_t pgain[4]
pitch gain for each pitch subframe
Definition: qcelpdata.h:52
static av_cold int qcelp_decode_init(AVCodecContext *avctx)
Initialize the speech codec according to the specification.
Definition: qcelpdec.c:85
#define FFABS(a)
Definition: common.h:61
int n
Definition: avisynth_c.h:547
int prev_g1[2]
Definition: qcelpdec.c:67
QCELPFrame frame
unpacked data frame
Definition: qcelpdec.c:56
#define QCELP_LSP_SPREAD_FACTOR
This spread factor is used, for bitrate 1/8 and I_F_Q, to force LSP frequencies to be at least 80 Hz ...
Definition: qcelpdata.h:533
GetBitContext gb
Definition: qcelpdec.c:54
#define QCELP_BANDWIDTH_EXPANSION_COEFF
Initial coefficient to perform bandwidth expansion on LPC.
Definition: qcelpdata.h:550
void ff_tilt_compensation(float *mem, float tilt, float *samples, int size)
Apply tilt compensation filter, 1 - tilt * z-1.
void ff_acelp_lspd2lpc(const double *lsp, float *lpc, int lp_half_order)
Reconstruct LPC coefficients from the line spectral pair frequencies.
Definition: lsp.c:209
float predictor_lspf[10]
LSP predictor for RATE_OCTAVE and I_F_Q.
Definition: qcelpdec.c:61
Data tables for the QCELP decoder.
uint8_t bitpos
position of the lowest bit in the value's byte
Definition: qcelpdata.h:78
AVCodec ff_qcelp_decoder
Definition: qcelpdec.c:793
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
Definition: get_bits.h:441
main external API structure.
Definition: avcodec.h:1501
float pitch_synthesis_filter_mem[303]
Definition: qcelpdec.c:62
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: utils.c:1040
void * buf
Definition: avisynth_c.h:553
static const uint16_t qcelp_unpacking_bitmaps_lengths[5]
Definition: qcelpdata.h:276
uint8_t bitlen
number of bits to read
Definition: qcelpdata.h:79
void ff_celp_lp_zero_synthesis_filterf(float *out, const float *filter_coeffs, const float *in, int buffer_length, int filter_length)
LP zero synthesis filter.
Definition: celp_filters.c:199
#define QCELP_RATE_HALF_CODEBOOK_RATIO
Definition: qcelpdata.h:502
static int weight(int i, int blen, int offset)
Definition: diracdec.c:1299
static const QCELPBitmap *const qcelp_unpacking_bitmaps_per_rate[5]
Bitmapping data position for each packet type in the QCELPContext.
Definition: qcelpdata.h:268
uint8_t pitch_lag[4]
Definition: qcelpdec.c:70
uint8_t reserved
reserved bits only present in bitrate 1, 1/4 and 1/8 packets
Definition: qcelpdata.h:65
float rnd_fir_filter_mem[180]
Definition: qcelpdec.c:64
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:182
static void lspf2lpc(const float *lspf, float *lpc)
Reconstruct LPC coefficients from the line spectral pair frequencies and perform bandwidth expansion...
Definition: qcelpdec.c:536
common internal api header.
if(ret< 0)
Definition: vf_mcdeint.c:280
int prev_bitrate
Definition: qcelpdec.c:68
#define rnd()
Definition: checkasm.h:43
float last_codebook_gain
Definition: qcelpdec.c:66
float postfilter_synth_mem[10]
Definition: qcelpdec.c:75
static void apply_gain_ctrl(float *v_out, const float *v_ref, const float *v_in)
Apply generic gain control.
Definition: qcelpdec.c:396
void * priv_data
Definition: avcodec.h:1543
static av_always_inline int diff(const uint32_t a, const uint32_t b)
qcelp_packet_rate bitrate
Definition: qcelpdec.c:55
int channels
number of audio channels
Definition: avcodec.h:2262
static const double qcelp_rnd_fir_coefs[11]
Table for impulse response of BPF used to filter the white excitation for bitrate 1/4 synthesis...
Definition: qcelpdata.h:521
qcelp_packet_rate
Definition: qcelpdec.c:44
static const float * do_pitchfilter(float memory[303], const float v_in[160], const float gain[4], const uint8_t *lag, const uint8_t pfrac[4])
Apply filter in pitch-subframe steps.
Definition: qcelpdec.c:423
uint16_t first16bits
Definition: qcelpdec.c:71
float pitch_gain[4]
Definition: qcelpdec.c:69
int frame_number
Frame counter, set by libavcodec.
Definition: avcodec.h:2292
float postfilter_tilt_mem
Definition: qcelpdec.c:77
#define M_PI
Definition: mathematics.h:46
uint8_t cbgain[16]
unsigned codebook gain for each codebook subframe
Definition: qcelpdata.h:44
#define AV_CH_LAYOUT_MONO
static const float qcelp_g12ga[61]
Table for computing Ga (decoded linear codebook gain magnitude)
Definition: qcelpdata.h:436
This structure stores compressed data.
Definition: avcodec.h:1399
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:225
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: avcodec.h:856