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 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 
30 #include <stddef.h>
31 
33 #include "avcodec.h"
34 #include "internal.h"
35 #include "get_bits.h"
36 #include "dsputil.h"
37 #include "qcelpdata.h"
38 #include "celp_filters.h"
39 #include "acelp_filters.h"
40 #include "acelp_vectors.h"
41 #include "lsp.h"
42 
43 #undef NDEBUG
44 #include <assert.h>
45 
46 typedef enum {
47  I_F_Q = -1,
54 
55 typedef struct {
63  float prev_lspf[10];
64  float predictor_lspf[10];
65  float pitch_synthesis_filter_mem[303];
66  float pitch_pre_filter_mem[303];
67  float rnd_fir_filter_mem[180];
68  float formant_mem[170];
70  int prev_g1[2];
72  float pitch_gain[4];
73  uint8_t pitch_lag[4];
74  uint16_t first16bits;
76 
77  /* postfilter */
78  float postfilter_synth_mem[10];
81 } QCELPContext;
82 
89 {
90  QCELPContext *q = avctx->priv_data;
91  int i;
92 
93  avctx->channels = 1;
96 
97  for (i = 0; i < 10; i++)
98  q->prev_lspf[i] = (i + 1) / 11.;
99 
101  avctx->coded_frame = &q->avframe;
102 
103  return 0;
104 }
105 
117 static int decode_lspf(QCELPContext *q, float *lspf)
118 {
119  int i;
120  float tmp_lspf, smooth, erasure_coeff;
121  const float *predictors;
122 
123  if (q->bitrate == RATE_OCTAVE || q->bitrate == I_F_Q) {
124  predictors = q->prev_bitrate != RATE_OCTAVE &&
125  q->prev_bitrate != I_F_Q ? q->prev_lspf
126  : q->predictor_lspf;
127 
128  if (q->bitrate == RATE_OCTAVE) {
129  q->octave_count++;
130 
131  for (i = 0; i < 10; i++) {
132  q->predictor_lspf[i] =
133  lspf[i] = (q->frame.lspv[i] ? QCELP_LSP_SPREAD_FACTOR
135  predictors[i] * QCELP_LSP_OCTAVE_PREDICTOR +
136  (i + 1) * ((1 - QCELP_LSP_OCTAVE_PREDICTOR) / 11);
137  }
138  smooth = q->octave_count < 10 ? .875 : 0.1;
139  } else {
140  erasure_coeff = QCELP_LSP_OCTAVE_PREDICTOR;
141 
142  assert(q->bitrate == I_F_Q);
143 
144  if (q->erasure_count > 1)
145  erasure_coeff *= q->erasure_count < 4 ? 0.9 : 0.7;
146 
147  for (i = 0; i < 10; i++) {
148  q->predictor_lspf[i] =
149  lspf[i] = (i + 1) * (1 - erasure_coeff) / 11 +
150  erasure_coeff * predictors[i];
151  }
152  smooth = 0.125;
153  }
154 
155  // Check the stability of the LSP frequencies.
156  lspf[0] = FFMAX(lspf[0], QCELP_LSP_SPREAD_FACTOR);
157  for (i = 1; i < 10; i++)
158  lspf[i] = FFMAX(lspf[i], lspf[i - 1] + QCELP_LSP_SPREAD_FACTOR);
159 
160  lspf[9] = FFMIN(lspf[9], 1.0 - QCELP_LSP_SPREAD_FACTOR);
161  for (i = 9; i > 0; i--)
162  lspf[i - 1] = FFMIN(lspf[i - 1], lspf[i] - QCELP_LSP_SPREAD_FACTOR);
163 
164  // Low-pass filter the LSP frequencies.
165  ff_weighted_vector_sumf(lspf, lspf, q->prev_lspf, smooth, 1.0 - smooth, 10);
166  } else {
167  q->octave_count = 0;
168 
169  tmp_lspf = 0.;
170  for (i = 0; i < 5; i++) {
171  lspf[2 * i + 0] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][0] * 0.0001;
172  lspf[2 * i + 1] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][1] * 0.0001;
173  }
174 
175  // Check for badly received packets.
176  if (q->bitrate == RATE_QUARTER) {
177  if (lspf[9] <= .70 || lspf[9] >= .97)
178  return -1;
179  for (i = 3; i < 10; i++)
180  if (fabs(lspf[i] - lspf[i - 2]) < .08)
181  return -1;
182  } else {
183  if (lspf[9] <= .66 || lspf[9] >= .985)
184  return -1;
185  for (i = 4; i < 10; i++)
186  if (fabs(lspf[i] - lspf[i - 4]) < .0931)
187  return -1;
188  }
189  }
190  return 0;
191 }
192 
201 static void decode_gain_and_index(QCELPContext *q, float *gain)
202 {
203  int i, subframes_count, g1[16];
204  float slope;
205 
206  if (q->bitrate >= RATE_QUARTER) {
207  switch (q->bitrate) {
208  case RATE_FULL: subframes_count = 16; break;
209  case RATE_HALF: subframes_count = 4; break;
210  default: subframes_count = 5;
211  }
212  for (i = 0; i < subframes_count; i++) {
213  g1[i] = 4 * q->frame.cbgain[i];
214  if (q->bitrate == RATE_FULL && !((i + 1) & 3)) {
215  g1[i] += av_clip((g1[i - 1] + g1[i - 2] + g1[i - 3]) / 3 - 6, 0, 32);
216  }
217 
218  gain[i] = qcelp_g12ga[g1[i]];
219 
220  if (q->frame.cbsign[i]) {
221  gain[i] = -gain[i];
222  q->frame.cindex[i] = (q->frame.cindex[i] - 89) & 127;
223  }
224  }
225 
226  q->prev_g1[0] = g1[i - 2];
227  q->prev_g1[1] = g1[i - 1];
228  q->last_codebook_gain = qcelp_g12ga[g1[i - 1]];
229 
230  if (q->bitrate == RATE_QUARTER) {
231  // Provide smoothing of the unvoiced excitation energy.
232  gain[7] = gain[4];
233  gain[6] = 0.4 * gain[3] + 0.6 * gain[4];
234  gain[5] = gain[3];
235  gain[4] = 0.8 * gain[2] + 0.2 * gain[3];
236  gain[3] = 0.2 * gain[1] + 0.8 * gain[2];
237  gain[2] = gain[1];
238  gain[1] = 0.6 * gain[0] + 0.4 * gain[1];
239  }
240  } else if (q->bitrate != SILENCE) {
241  if (q->bitrate == RATE_OCTAVE) {
242  g1[0] = 2 * q->frame.cbgain[0] +
243  av_clip((q->prev_g1[0] + q->prev_g1[1]) / 2 - 5, 0, 54);
244  subframes_count = 8;
245  } else {
246  assert(q->bitrate == I_F_Q);
247 
248  g1[0] = q->prev_g1[1];
249  switch (q->erasure_count) {
250  case 1 : break;
251  case 2 : g1[0] -= 1; break;
252  case 3 : g1[0] -= 2; break;
253  default: g1[0] -= 6;
254  }
255  if (g1[0] < 0)
256  g1[0] = 0;
257  subframes_count = 4;
258  }
259  // This interpolation is done to produce smoother background noise.
260  slope = 0.5 * (qcelp_g12ga[g1[0]] - q->last_codebook_gain) / subframes_count;
261  for (i = 1; i <= subframes_count; i++)
262  gain[i - 1] = q->last_codebook_gain + slope * i;
263 
264  q->last_codebook_gain = gain[i - 2];
265  q->prev_g1[0] = q->prev_g1[1];
266  q->prev_g1[1] = g1[0];
267  }
268 }
269 
280 {
281  int i, diff, prev_diff = 0;
282 
283  for (i = 1; i < 5; i++) {
284  diff = cbgain[i] - cbgain[i-1];
285  if (FFABS(diff) > 10)
286  return -1;
287  else if (FFABS(diff - prev_diff) > 12)
288  return -1;
289  prev_diff = diff;
290  }
291  return 0;
292 }
293 
315 static void compute_svector(QCELPContext *q, const float *gain,
316  float *cdn_vector)
317 {
318  int i, j, k;
319  uint16_t cbseed, cindex;
320  float *rnd, tmp_gain, fir_filter_value;
321 
322  switch (q->bitrate) {
323  case RATE_FULL:
324  for (i = 0; i < 16; i++) {
325  tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
326  cindex = -q->frame.cindex[i];
327  for (j = 0; j < 10; j++)
328  *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cindex++ & 127];
329  }
330  break;
331  case RATE_HALF:
332  for (i = 0; i < 4; i++) {
333  tmp_gain = gain[i] * QCELP_RATE_HALF_CODEBOOK_RATIO;
334  cindex = -q->frame.cindex[i];
335  for (j = 0; j < 40; j++)
336  *cdn_vector++ = tmp_gain * qcelp_rate_half_codebook[cindex++ & 127];
337  }
338  break;
339  case RATE_QUARTER:
340  cbseed = (0x0003 & q->frame.lspv[4]) << 14 |
341  (0x003F & q->frame.lspv[3]) << 8 |
342  (0x0060 & q->frame.lspv[2]) << 1 |
343  (0x0007 & q->frame.lspv[1]) << 3 |
344  (0x0038 & q->frame.lspv[0]) >> 3;
345  rnd = q->rnd_fir_filter_mem + 20;
346  for (i = 0; i < 8; i++) {
347  tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
348  for (k = 0; k < 20; k++) {
349  cbseed = 521 * cbseed + 259;
350  *rnd = (int16_t) cbseed;
351 
352  // FIR filter
353  fir_filter_value = 0.0;
354  for (j = 0; j < 10; j++)
355  fir_filter_value += qcelp_rnd_fir_coefs[j] *
356  (rnd[-j] + rnd[-20+j]);
357 
358  fir_filter_value += qcelp_rnd_fir_coefs[10] * rnd[-10];
359  *cdn_vector++ = tmp_gain * fir_filter_value;
360  rnd++;
361  }
362  }
363  memcpy(q->rnd_fir_filter_mem, q->rnd_fir_filter_mem + 160,
364  20 * sizeof(float));
365  break;
366  case RATE_OCTAVE:
367  cbseed = q->first16bits;
368  for (i = 0; i < 8; i++) {
369  tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
370  for (j = 0; j < 20; j++) {
371  cbseed = 521 * cbseed + 259;
372  *cdn_vector++ = tmp_gain * (int16_t) cbseed;
373  }
374  }
375  break;
376  case I_F_Q:
377  cbseed = -44; // random codebook index
378  for (i = 0; i < 4; i++) {
379  tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
380  for (j = 0; j < 40; j++)
381  *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cbseed++ & 127];
382  }
383  break;
384  case SILENCE:
385  memset(cdn_vector, 0, 160 * sizeof(float));
386  break;
387  }
388 }
389 
399 static void apply_gain_ctrl(float *v_out, const float *v_ref, const float *v_in)
400 {
401  int i;
402 
403  for (i = 0; i < 160; i += 40)
404  ff_scale_vector_to_given_sum_of_squares(v_out + i, v_in + i,
405  ff_scalarproduct_float_c(v_ref + i,
406  v_ref + i,
407  40),
408  40);
409 }
410 
428 static const float *do_pitchfilter(float memory[303], const float v_in[160],
429  const float gain[4], const uint8_t *lag,
430  const uint8_t pfrac[4])
431 {
432  int i, j;
433  float *v_lag, *v_out;
434  const float *v_len;
435 
436  v_out = memory + 143; // Output vector starts at memory[143].
437 
438  for (i = 0; i < 4; i++) {
439  if (gain[i]) {
440  v_lag = memory + 143 + 40 * i - lag[i];
441  for (v_len = v_in + 40; v_in < v_len; v_in++) {
442  if (pfrac[i]) { // If it is a fractional lag...
443  for (j = 0, *v_out = 0.; j < 4; j++)
444  *v_out += qcelp_hammsinc_table[j] * (v_lag[j - 4] + v_lag[3 - j]);
445  } else
446  *v_out = *v_lag;
447 
448  *v_out = *v_in + gain[i] * *v_out;
449 
450  v_lag++;
451  v_out++;
452  }
453  } else {
454  memcpy(v_out, v_in, 40 * sizeof(float));
455  v_in += 40;
456  v_out += 40;
457  }
458  }
459 
460  memmove(memory, memory + 160, 143 * sizeof(float));
461  return memory + 143;
462 }
463 
471 static void apply_pitch_filters(QCELPContext *q, float *cdn_vector)
472 {
473  int i;
474  const float *v_synthesis_filtered, *v_pre_filtered;
475 
476  if (q->bitrate >= RATE_HALF || q->bitrate == SILENCE ||
477  (q->bitrate == I_F_Q && (q->prev_bitrate >= RATE_HALF))) {
478 
479  if (q->bitrate >= RATE_HALF) {
480  // Compute gain & lag for the whole frame.
481  for (i = 0; i < 4; i++) {
482  q->pitch_gain[i] = q->frame.plag[i] ? (q->frame.pgain[i] + 1) * 0.25 : 0.0;
483 
484  q->pitch_lag[i] = q->frame.plag[i] + 16;
485  }
486  } else {
487  float max_pitch_gain;
488 
489  if (q->bitrate == I_F_Q) {
490  if (q->erasure_count < 3)
491  max_pitch_gain = 0.9 - 0.3 * (q->erasure_count - 1);
492  else
493  max_pitch_gain = 0.0;
494  } else {
495  assert(q->bitrate == SILENCE);
496  max_pitch_gain = 1.0;
497  }
498  for (i = 0; i < 4; i++)
499  q->pitch_gain[i] = FFMIN(q->pitch_gain[i], max_pitch_gain);
500 
501  memset(q->frame.pfrac, 0, sizeof(q->frame.pfrac));
502  }
503 
504  // pitch synthesis filter
505  v_synthesis_filtered = do_pitchfilter(q->pitch_synthesis_filter_mem,
506  cdn_vector, q->pitch_gain,
507  q->pitch_lag, q->frame.pfrac);
508 
509  // pitch prefilter update
510  for (i = 0; i < 4; i++)
511  q->pitch_gain[i] = 0.5 * FFMIN(q->pitch_gain[i], 1.0);
512 
513  v_pre_filtered = do_pitchfilter(q->pitch_pre_filter_mem,
514  v_synthesis_filtered,
515  q->pitch_gain, q->pitch_lag,
516  q->frame.pfrac);
517 
518  apply_gain_ctrl(cdn_vector, v_synthesis_filtered, v_pre_filtered);
519  } else {
520  memcpy(q->pitch_synthesis_filter_mem, cdn_vector + 17, 143 * sizeof(float));
521  memcpy(q->pitch_pre_filter_mem, cdn_vector + 17, 143 * sizeof(float));
522  memset(q->pitch_gain, 0, sizeof(q->pitch_gain));
523  memset(q->pitch_lag, 0, sizeof(q->pitch_lag));
524  }
525 }
526 
539 static void lspf2lpc(const float *lspf, float *lpc)
540 {
541  double lsp[10];
542  double bandwidth_expansion_coeff = QCELP_BANDWIDTH_EXPANSION_COEFF;
543  int i;
544 
545  for (i = 0; i < 10; i++)
546  lsp[i] = cos(M_PI * lspf[i]);
547 
548  ff_acelp_lspd2lpc(lsp, lpc, 5);
549 
550  for (i = 0; i < 10; i++) {
551  lpc[i] *= bandwidth_expansion_coeff;
552  bandwidth_expansion_coeff *= QCELP_BANDWIDTH_EXPANSION_COEFF;
553  }
554 }
555 
567 static void interpolate_lpc(QCELPContext *q, const float *curr_lspf,
568  float *lpc, const int subframe_num)
569 {
570  float interpolated_lspf[10];
571  float weight;
572 
573  if (q->bitrate >= RATE_QUARTER)
574  weight = 0.25 * (subframe_num + 1);
575  else if (q->bitrate == RATE_OCTAVE && !subframe_num)
576  weight = 0.625;
577  else
578  weight = 1.0;
579 
580  if (weight != 1.0) {
581  ff_weighted_vector_sumf(interpolated_lspf, curr_lspf, q->prev_lspf,
582  weight, 1.0 - weight, 10);
583  lspf2lpc(interpolated_lspf, lpc);
584  } else if (q->bitrate >= RATE_QUARTER ||
585  (q->bitrate == I_F_Q && !subframe_num))
586  lspf2lpc(curr_lspf, lpc);
587  else if (q->bitrate == SILENCE && !subframe_num)
588  lspf2lpc(q->prev_lspf, lpc);
589 }
590 
591 static qcelp_packet_rate buf_size2bitrate(const int buf_size)
592 {
593  switch (buf_size) {
594  case 35: return RATE_FULL;
595  case 17: return RATE_HALF;
596  case 8: return RATE_QUARTER;
597  case 4: return RATE_OCTAVE;
598  case 1: return SILENCE;
599  }
600 
601  return I_F_Q;
602 }
603 
617  const int buf_size,
618  const uint8_t **buf)
619 {
620  qcelp_packet_rate bitrate;
621 
622  if ((bitrate = buf_size2bitrate(buf_size)) >= 0) {
623  if (bitrate > **buf) {
624  QCELPContext *q = avctx->priv_data;
625  if (!q->warned_buf_mismatch_bitrate) {
626  av_log(avctx, AV_LOG_WARNING,
627  "Claimed bitrate and buffer size mismatch.\n");
629  }
630  bitrate = **buf;
631  } else if (bitrate < **buf) {
632  av_log(avctx, AV_LOG_ERROR,
633  "Buffer is too small for the claimed bitrate.\n");
634  return I_F_Q;
635  }
636  (*buf)++;
637  } else if ((bitrate = buf_size2bitrate(buf_size + 1)) >= 0) {
638  av_log(avctx, AV_LOG_WARNING,
639  "Bitrate byte is missing, guessing the bitrate from packet size.\n");
640  } else
641  return I_F_Q;
642 
643  if (bitrate == SILENCE) {
644  //FIXME: Remove experimental warning when tested with samples.
645  av_log_ask_for_sample(avctx, "'Blank frame handling is experimental.");
646  }
647  return bitrate;
648 }
649 
651  const char *message)
652 {
653  av_log(avctx, AV_LOG_WARNING, "Frame #%d, IFQ: %s\n",
654  avctx->frame_number, message);
655 }
656 
657 static void postfilter(QCELPContext *q, float *samples, float *lpc)
658 {
659  static const float pow_0_775[10] = {
660  0.775000, 0.600625, 0.465484, 0.360750, 0.279582,
661  0.216676, 0.167924, 0.130141, 0.100859, 0.078166
662  }, pow_0_625[10] = {
663  0.625000, 0.390625, 0.244141, 0.152588, 0.095367,
664  0.059605, 0.037253, 0.023283, 0.014552, 0.009095
665  };
666  float lpc_s[10], lpc_p[10], pole_out[170], zero_out[160];
667  int n;
668 
669  for (n = 0; n < 10; n++) {
670  lpc_s[n] = lpc[n] * pow_0_625[n];
671  lpc_p[n] = lpc[n] * pow_0_775[n];
672  }
673 
674  ff_celp_lp_zero_synthesis_filterf(zero_out, lpc_s,
675  q->formant_mem + 10, 160, 10);
676  memcpy(pole_out, q->postfilter_synth_mem, sizeof(float) * 10);
677  ff_celp_lp_synthesis_filterf(pole_out + 10, lpc_p, zero_out, 160, 10);
678  memcpy(q->postfilter_synth_mem, pole_out + 160, sizeof(float) * 10);
679 
680  ff_tilt_compensation(&q->postfilter_tilt_mem, 0.3, pole_out + 10, 160);
681 
682  ff_adaptive_gain_control(samples, pole_out + 10,
684  q->formant_mem + 10, 160),
685  160, 0.9375, &q->postfilter_agc_mem);
686 }
687 
688 static int qcelp_decode_frame(AVCodecContext *avctx, void *data,
689  int *got_frame_ptr, AVPacket *avpkt)
690 {
691  const uint8_t *buf = avpkt->data;
692  int buf_size = avpkt->size;
693  QCELPContext *q = avctx->priv_data;
694  float *outbuffer;
695  int i, ret;
696  float quantized_lspf[10], lpc[10];
697  float gain[16];
698  float *formant_mem;
699 
700  /* get output buffer */
701  q->avframe.nb_samples = 160;
702  if ((ret = ff_get_buffer(avctx, &q->avframe)) < 0) {
703  av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
704  return ret;
705  }
706  outbuffer = (float *)q->avframe.data[0];
707 
708  if ((q->bitrate = determine_bitrate(avctx, buf_size, &buf)) == I_F_Q) {
709  warn_insufficient_frame_quality(avctx, "bitrate cannot be determined.");
710  goto erasure;
711  }
712 
713  if (q->bitrate == RATE_OCTAVE &&
714  (q->first16bits = AV_RB16(buf)) == 0xFFFF) {
715  warn_insufficient_frame_quality(avctx, "Bitrate is 1/8 and first 16 bits are on.");
716  goto erasure;
717  }
718 
719  if (q->bitrate > SILENCE) {
721  const QCELPBitmap *bitmaps_end = qcelp_unpacking_bitmaps_per_rate[q->bitrate] +
723  uint8_t *unpacked_data = (uint8_t *)&q->frame;
724 
725  init_get_bits(&q->gb, buf, 8 * buf_size);
726 
727  memset(&q->frame, 0, sizeof(QCELPFrame));
728 
729  for (; bitmaps < bitmaps_end; bitmaps++)
730  unpacked_data[bitmaps->index] |= get_bits(&q->gb, bitmaps->bitlen) << bitmaps->bitpos;
731 
732  // Check for erasures/blanks on rates 1, 1/4 and 1/8.
733  if (q->frame.reserved) {
734  warn_insufficient_frame_quality(avctx, "Wrong data in reserved frame area.");
735  goto erasure;
736  }
737  if (q->bitrate == RATE_QUARTER &&
739  warn_insufficient_frame_quality(avctx, "Codebook gain sanity check failed.");
740  goto erasure;
741  }
742 
743  if (q->bitrate >= RATE_HALF) {
744  for (i = 0; i < 4; i++) {
745  if (q->frame.pfrac[i] && q->frame.plag[i] >= 124) {
746  warn_insufficient_frame_quality(avctx, "Cannot initialize pitch filter.");
747  goto erasure;
748  }
749  }
750  }
751  }
752 
753  decode_gain_and_index(q, gain);
754  compute_svector(q, gain, outbuffer);
755 
756  if (decode_lspf(q, quantized_lspf) < 0) {
757  warn_insufficient_frame_quality(avctx, "Badly received packets in frame.");
758  goto erasure;
759  }
760 
761  apply_pitch_filters(q, outbuffer);
762 
763  if (q->bitrate == I_F_Q) {
764 erasure:
765  q->bitrate = I_F_Q;
766  q->erasure_count++;
767  decode_gain_and_index(q, gain);
768  compute_svector(q, gain, outbuffer);
769  decode_lspf(q, quantized_lspf);
770  apply_pitch_filters(q, outbuffer);
771  } else
772  q->erasure_count = 0;
773 
774  formant_mem = q->formant_mem + 10;
775  for (i = 0; i < 4; i++) {
776  interpolate_lpc(q, quantized_lspf, lpc, i);
777  ff_celp_lp_synthesis_filterf(formant_mem, lpc, outbuffer + i * 40, 40, 10);
778  formant_mem += 40;
779  }
780 
781  // postfilter, as per TIA/EIA/IS-733 2.4.8.6
782  postfilter(q, outbuffer, lpc);
783 
784  memcpy(q->formant_mem, q->formant_mem + 160, 10 * sizeof(float));
785 
786  memcpy(q->prev_lspf, quantized_lspf, sizeof(q->prev_lspf));
787  q->prev_bitrate = q->bitrate;
788 
789  *got_frame_ptr = 1;
790  *(AVFrame *)data = q->avframe;
791 
792  return buf_size;
793 }
794 
796  .name = "qcelp",
797  .type = AVMEDIA_TYPE_AUDIO,
798  .id = AV_CODEC_ID_QCELP,
799  .init = qcelp_decode_init,
800  .decode = qcelp_decode_frame,
801  .capabilities = CODEC_CAP_DR1,
802  .priv_data_size = sizeof(QCELPContext),
803  .long_name = NULL_IF_CONFIG_SMALL("QCELP / PureVoice"),
804 };
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
static int16_t * samples
This structure describes decoded (raw) audio or video data.
Definition: avcodec.h:989
float formant_mem[170]
Definition: qcelpdec.c:68
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:237
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.
AVFrame * coded_frame
the picture in the bitstream
Definition: avcodec.h:2725
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:471
uint8_t pfrac[4]
fractional pitch lag for each pitch subframe
Definition: qcelpdata.h:51
int size
Definition: avcodec.h:916
#define QCELP_RATE_FULL_CODEBOOK_RATIO
Definition: qcelpdata.h:477
static void warn_insufficient_frame_quality(AVCodecContext *avctx, const char *message)
Definition: qcelpdec.c:650
static qcelp_packet_rate buf_size2bitrate(const int buf_size)
Definition: qcelpdec.c:591
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:2960
static const float qcelp_hammsinc_table[4]
pre-calculated table for hammsinc function Only half of the table is needed because of symmetry...
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:117
uint8_t index
index into the QCELPContext structure
Definition: qcelpdata.h:77
insufficient frame quality
Definition: qcelpdec.c:47
uint8_t warned_buf_mismatch_bitrate
Definition: qcelpdec.c:75
uint8_t octave_count
count the consecutive RATE_OCTAVE frames
Definition: qcelpdec.c:62
QCELP unpacked data frame.
Definition: qcelpdata.h:40
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:279
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2112
float prev_lspf[10]
Definition: qcelpdec.c:63
uint8_t
static const qcelp_vector *const qcelp_lspvq[5]
Definition: qcelpdata.h:414
const char data[16]
Definition: mxf.c:66
uint8_t * data
Definition: avcodec.h:915
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 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:616
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:567
uint8_t erasure_count
Definition: qcelpdec.c:61
void av_log_ask_for_sample(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message asking for a sample.
static void decode_gain_and_index(QCELPContext *q, float *gain)
Convert codebook transmission codes to GAIN and INDEX.
Definition: qcelpdec.c:201
float postfilter_agc_mem
Definition: qcelpdec.c:79
#define AV_RB16
Definition: intreadwrite.h:53
float pitch_pre_filter_mem[303]
Definition: qcelpdec.c:66
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:88
void av_log(void *avcl, int level, const char *fmt,...)
Definition: log.c:146
float ff_scalarproduct_float_c(const float *v1, const float *v2, int len)
Return the scalar product of two vectors.
Definition: dsputil.c:2418
const char * name
Name of the codec implementation.
Definition: avcodec.h:2967
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.
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:2165
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:315
#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:657
static int qcelp_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: qcelpdec.c:688
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:88
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame)
Get a buffer for a frame.
Definition: utils.c:464
int prev_g1[2]
Definition: qcelpdec.c:70
QCELPFrame frame
unpacked data frame
Definition: qcelpdec.c:59
#define QCELP_LSP_SPREAD_FACTOR
This spread factor is used, for bitrate 1/8 and I_F_Q, to force the LSP frequencies to be at least 80...
Definition: qcelpdata.h:533
GetBitContext gb
Definition: qcelpdec.c:57
#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:201
float predictor_lspf[10]
LSP predictor for RATE_OCTAVE and I_F_Q.
Definition: qcelpdec.c:64
Data tables for the QCELP decoder.
external API header
uint8_t bitpos
position of the lowest bit in the value's byte
Definition: qcelpdata.h:78
AVCodec ff_qcelp_decoder
Definition: qcelpdec.c:795
main external API structure.
Definition: avcodec.h:1339
float pitch_synthesis_filter_mem[303]
Definition: qcelpdec.c:65
void avcodec_get_frame_defaults(AVFrame *frame)
Set the fields of the given AVFrame to default values.
Definition: utils.c:602
static const uint16_t qcelp_unpacking_bitmaps_lengths[5]
Definition: qcelpdata.h:276
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
Definition: get_bits.h:372
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:196
#define QCELP_RATE_HALF_CODEBOOK_RATIO
Definition: qcelpdata.h:502
static const QCELPBitmap *const qcelp_unpacking_bitmaps_per_rate[5]
position of the bitmapping data for each packet type in the QCELPContext
Definition: qcelpdata.h:268
uint8_t pitch_lag[4]
Definition: qcelpdec.c:73
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:67
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: avcodec.h:997
static void lspf2lpc(const float *lspf, float *lpc)
Reconstruct LPC coefficients from the line spectral pair frequencies and perform bandwidth expansion...
Definition: qcelpdec.c:539
common internal api header.
int prev_bitrate
Definition: qcelpdec.c:71
float last_codebook_gain
Definition: qcelpdec.c:69
float postfilter_synth_mem[10]
Definition: qcelpdec.c:78
DSP utils.
static void apply_gain_ctrl(float *v_out, const float *v_ref, const float *v_in)
Apply generic gain control.
Definition: qcelpdec.c:399
void * priv_data
Definition: avcodec.h:1382
qcelp_packet_rate bitrate
Definition: qcelpdec.c:58
int channels
number of audio channels
Definition: avcodec.h:2105
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:46
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:428
uint16_t first16bits
Definition: qcelpdec.c:74
float pitch_gain[4]
Definition: qcelpdec.c:72
int frame_number
Frame counter, set by libavcodec.
Definition: avcodec.h:2135
float postfilter_tilt_mem
Definition: qcelpdec.c:80
uint8_t cbgain[16]
unsigned codebook gain for each codebook subframe
Definition: qcelpdata.h:44
AVFrame avframe
Definition: qcelpdec.c:56
#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:898
int nb_samples
number of audio samples (per channel) described by this frame
Definition: avcodec.h:1042
if(!(ptr_align%ac->ptr_align)&&samples_align >=aligned_len)