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sipr.c
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1 /*
2  * SIPR / ACELP.NET decoder
3  *
4  * Copyright (c) 2008 Vladimir Voroshilov
5  * Copyright (c) 2009 Vitor Sessak
6  *
7  * This file is part of FFmpeg.
8  *
9  * FFmpeg is free software; you can redistribute it and/or
10  * modify it under the terms of the GNU Lesser General Public
11  * License as published by the Free Software Foundation; either
12  * version 2.1 of the License, or (at your option) any later version.
13  *
14  * FFmpeg is distributed in the hope that it will be useful,
15  * but WITHOUT ANY WARRANTY; without even the implied warranty of
16  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
17  * Lesser General Public License for more details.
18  *
19  * You should have received a copy of the GNU Lesser General Public
20  * License along with FFmpeg; if not, write to the Free Software
21  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22  */
23 
24 #include <math.h>
25 #include <stdint.h>
26 #include <string.h>
27 
29 #include "libavutil/float_dsp.h"
30 #include "libavutil/mathematics.h"
31 #include "avcodec.h"
32 #define BITSTREAM_READER_LE
33 #include "get_bits.h"
34 #include "internal.h"
35 
36 #include "lsp.h"
37 #include "acelp_vectors.h"
38 #include "acelp_pitch_delay.h"
39 #include "acelp_filters.h"
40 #include "celp_filters.h"
41 
42 #define MAX_SUBFRAME_COUNT 5
43 
44 #include "sipr.h"
45 #include "siprdata.h"
46 
47 typedef struct SiprModeParam {
48  const char *mode_name;
49  uint16_t bits_per_frame;
53 
54  /* bitstream parameters */
56  uint8_t ma_predictor_bits; ///< size in bits of the switched MA predictor
57 
58  /** size in bits of the i-th stage vector of quantizer */
60 
61  /** size in bits of the adaptive-codebook index for every subframe */
63 
65  uint8_t fc_index_bits[10]; ///< size in bits of the fixed codebook indexes
66  uint8_t gc_index_bits; ///< size in bits of the gain codebook indexes
68 
69 static const SiprModeParam modes[MODE_COUNT] = {
70  [MODE_16k] = {
71  .mode_name = "16k",
72  .bits_per_frame = 160,
73  .subframe_count = SUBFRAME_COUNT_16k,
74  .frames_per_packet = 1,
75  .pitch_sharp_factor = 0.00,
76 
77  .number_of_fc_indexes = 10,
78  .ma_predictor_bits = 1,
79  .vq_indexes_bits = {7, 8, 7, 7, 7},
80  .pitch_delay_bits = {9, 6},
81  .gp_index_bits = 4,
82  .fc_index_bits = {4, 5, 4, 5, 4, 5, 4, 5, 4, 5},
83  .gc_index_bits = 5
84  },
85 
86  [MODE_8k5] = {
87  .mode_name = "8k5",
88  .bits_per_frame = 152,
89  .subframe_count = 3,
90  .frames_per_packet = 1,
91  .pitch_sharp_factor = 0.8,
92 
93  .number_of_fc_indexes = 3,
94  .ma_predictor_bits = 0,
95  .vq_indexes_bits = {6, 7, 7, 7, 5},
96  .pitch_delay_bits = {8, 5, 5},
97  .gp_index_bits = 0,
98  .fc_index_bits = {9, 9, 9},
99  .gc_index_bits = 7
100  },
101 
102  [MODE_6k5] = {
103  .mode_name = "6k5",
104  .bits_per_frame = 232,
105  .subframe_count = 3,
106  .frames_per_packet = 2,
107  .pitch_sharp_factor = 0.8,
108 
109  .number_of_fc_indexes = 3,
110  .ma_predictor_bits = 0,
111  .vq_indexes_bits = {6, 7, 7, 7, 5},
112  .pitch_delay_bits = {8, 5, 5},
113  .gp_index_bits = 0,
114  .fc_index_bits = {5, 5, 5},
115  .gc_index_bits = 7
116  },
117 
118  [MODE_5k0] = {
119  .mode_name = "5k0",
120  .bits_per_frame = 296,
121  .subframe_count = 5,
122  .frames_per_packet = 2,
123  .pitch_sharp_factor = 0.85,
124 
125  .number_of_fc_indexes = 1,
126  .ma_predictor_bits = 0,
127  .vq_indexes_bits = {6, 7, 7, 7, 5},
128  .pitch_delay_bits = {8, 5, 8, 5, 5},
129  .gp_index_bits = 0,
130  .fc_index_bits = {10},
131  .gc_index_bits = 7
132  }
133 };
134 
135 const float ff_pow_0_5[] = {
136  1.0/(1 << 1), 1.0/(1 << 2), 1.0/(1 << 3), 1.0/(1 << 4),
137  1.0/(1 << 5), 1.0/(1 << 6), 1.0/(1 << 7), 1.0/(1 << 8),
138  1.0/(1 << 9), 1.0/(1 << 10), 1.0/(1 << 11), 1.0/(1 << 12),
139  1.0/(1 << 13), 1.0/(1 << 14), 1.0/(1 << 15), 1.0/(1 << 16)
140 };
141 
142 static void dequant(float *out, const int *idx, const float * const cbs[])
143 {
144  int i;
145  int stride = 2;
146  int num_vec = 5;
147 
148  for (i = 0; i < num_vec; i++)
149  memcpy(out + stride*i, cbs[i] + stride*idx[i], stride*sizeof(float));
150 
151 }
152 
153 static void lsf_decode_fp(float *lsfnew, float *lsf_history,
154  const SiprParameters *parm)
155 {
156  int i;
157  float lsf_tmp[LP_FILTER_ORDER];
158 
159  dequant(lsf_tmp, parm->vq_indexes, lsf_codebooks);
160 
161  for (i = 0; i < LP_FILTER_ORDER; i++)
162  lsfnew[i] = lsf_history[i] * 0.33 + lsf_tmp[i] + mean_lsf[i];
163 
164  ff_sort_nearly_sorted_floats(lsfnew, LP_FILTER_ORDER - 1);
165 
166  /* Note that a minimum distance is not enforced between the last value and
167  the previous one, contrary to what is done in ff_acelp_reorder_lsf() */
168  ff_set_min_dist_lsf(lsfnew, LSFQ_DIFF_MIN, LP_FILTER_ORDER - 1);
169  lsfnew[9] = FFMIN(lsfnew[LP_FILTER_ORDER - 1], 1.3 * M_PI);
170 
171  memcpy(lsf_history, lsf_tmp, LP_FILTER_ORDER * sizeof(*lsf_history));
172 
173  for (i = 0; i < LP_FILTER_ORDER - 1; i++)
174  lsfnew[i] = cos(lsfnew[i]);
175  lsfnew[LP_FILTER_ORDER - 1] *= 6.153848 / M_PI;
176 }
177 
178 /** Apply pitch lag to the fixed vector (AMR section 6.1.2). */
179 static void pitch_sharpening(int pitch_lag_int, float beta,
180  float *fixed_vector)
181 {
182  int i;
183 
184  for (i = pitch_lag_int; i < SUBFR_SIZE; i++)
185  fixed_vector[i] += beta * fixed_vector[i - pitch_lag_int];
186 }
187 
188 /**
189  * Extract decoding parameters from the input bitstream.
190  * @param parms parameters structure
191  * @param pgb pointer to initialized GetBitContext structure
192  */
194  const SiprModeParam *p)
195 {
196  int i, j;
197 
198  if (p->ma_predictor_bits)
199  parms->ma_pred_switch = get_bits(pgb, p->ma_predictor_bits);
200 
201  for (i = 0; i < 5; i++)
202  parms->vq_indexes[i] = get_bits(pgb, p->vq_indexes_bits[i]);
203 
204  for (i = 0; i < p->subframe_count; i++) {
205  parms->pitch_delay[i] = get_bits(pgb, p->pitch_delay_bits[i]);
206  if (p->gp_index_bits)
207  parms->gp_index[i] = get_bits(pgb, p->gp_index_bits);
208 
209  for (j = 0; j < p->number_of_fc_indexes; j++)
210  parms->fc_indexes[i][j] = get_bits(pgb, p->fc_index_bits[j]);
211 
212  parms->gc_index[i] = get_bits(pgb, p->gc_index_bits);
213  }
214 }
215 
216 static void sipr_decode_lp(float *lsfnew, const float *lsfold, float *Az,
217  int num_subfr)
218 {
219  double lsfint[LP_FILTER_ORDER];
220  int i,j;
221  float t, t0 = 1.0 / num_subfr;
222 
223  t = t0 * 0.5;
224  for (i = 0; i < num_subfr; i++) {
225  for (j = 0; j < LP_FILTER_ORDER; j++)
226  lsfint[j] = lsfold[j] * (1 - t) + t * lsfnew[j];
227 
228  ff_amrwb_lsp2lpc(lsfint, Az, LP_FILTER_ORDER);
229  Az += LP_FILTER_ORDER;
230  t += t0;
231  }
232 }
233 
234 /**
235  * Evaluate the adaptive impulse response.
236  */
237 static void eval_ir(const float *Az, int pitch_lag, float *freq,
238  float pitch_sharp_factor)
239 {
240  float tmp1[SUBFR_SIZE+1], tmp2[LP_FILTER_ORDER+1];
241  int i;
242 
243  tmp1[0] = 1.0;
244  for (i = 0; i < LP_FILTER_ORDER; i++) {
245  tmp1[i+1] = Az[i] * ff_pow_0_55[i];
246  tmp2[i ] = Az[i] * ff_pow_0_7 [i];
247  }
248  memset(tmp1 + 11, 0, 37 * sizeof(float));
249 
250  ff_celp_lp_synthesis_filterf(freq, tmp2, tmp1, SUBFR_SIZE,
251  LP_FILTER_ORDER);
252 
253  pitch_sharpening(pitch_lag, pitch_sharp_factor, freq);
254 }
255 
256 /**
257  * Evaluate the convolution of a vector with a sparse vector.
258  */
259 static void convolute_with_sparse(float *out, const AMRFixed *pulses,
260  const float *shape, int length)
261 {
262  int i, j;
263 
264  memset(out, 0, length*sizeof(float));
265  for (i = 0; i < pulses->n; i++)
266  for (j = pulses->x[i]; j < length; j++)
267  out[j] += pulses->y[i] * shape[j - pulses->x[i]];
268 }
269 
270 /**
271  * Apply postfilter, very similar to AMR one.
272  */
273 static void postfilter_5k0(SiprContext *ctx, const float *lpc, float *samples)
274 {
275  float buf[SUBFR_SIZE + LP_FILTER_ORDER];
276  float *pole_out = buf + LP_FILTER_ORDER;
277  float lpc_n[LP_FILTER_ORDER];
278  float lpc_d[LP_FILTER_ORDER];
279  int i;
280 
281  for (i = 0; i < LP_FILTER_ORDER; i++) {
282  lpc_d[i] = lpc[i] * ff_pow_0_75[i];
283  lpc_n[i] = lpc[i] * ff_pow_0_5 [i];
284  };
285 
286  memcpy(pole_out - LP_FILTER_ORDER, ctx->postfilter_mem,
287  LP_FILTER_ORDER*sizeof(float));
288 
289  ff_celp_lp_synthesis_filterf(pole_out, lpc_d, samples, SUBFR_SIZE,
290  LP_FILTER_ORDER);
291 
292  memcpy(ctx->postfilter_mem, pole_out + SUBFR_SIZE - LP_FILTER_ORDER,
293  LP_FILTER_ORDER*sizeof(float));
294 
295  ff_tilt_compensation(&ctx->tilt_mem, 0.4, pole_out, SUBFR_SIZE);
296 
297  memcpy(pole_out - LP_FILTER_ORDER, ctx->postfilter_mem5k0,
298  LP_FILTER_ORDER*sizeof(*pole_out));
299 
300  memcpy(ctx->postfilter_mem5k0, pole_out + SUBFR_SIZE - LP_FILTER_ORDER,
301  LP_FILTER_ORDER*sizeof(*pole_out));
302 
303  ff_celp_lp_zero_synthesis_filterf(samples, lpc_n, pole_out, SUBFR_SIZE,
304  LP_FILTER_ORDER);
305 
306 }
307 
308 static void decode_fixed_sparse(AMRFixed *fixed_sparse, const int16_t *pulses,
309  SiprMode mode, int low_gain)
310 {
311  int i;
312 
313  switch (mode) {
314  case MODE_6k5:
315  for (i = 0; i < 3; i++) {
316  fixed_sparse->x[i] = 3 * (pulses[i] & 0xf) + i;
317  fixed_sparse->y[i] = pulses[i] & 0x10 ? -1 : 1;
318  }
319  fixed_sparse->n = 3;
320  break;
321  case MODE_8k5:
322  for (i = 0; i < 3; i++) {
323  fixed_sparse->x[2*i ] = 3 * ((pulses[i] >> 4) & 0xf) + i;
324  fixed_sparse->x[2*i + 1] = 3 * ( pulses[i] & 0xf) + i;
325 
326  fixed_sparse->y[2*i ] = (pulses[i] & 0x100) ? -1.0: 1.0;
327 
328  fixed_sparse->y[2*i + 1] =
329  (fixed_sparse->x[2*i + 1] < fixed_sparse->x[2*i]) ?
330  -fixed_sparse->y[2*i ] : fixed_sparse->y[2*i];
331  }
332 
333  fixed_sparse->n = 6;
334  break;
335  case MODE_5k0:
336  default:
337  if (low_gain) {
338  int offset = (pulses[0] & 0x200) ? 2 : 0;
339  int val = pulses[0];
340 
341  for (i = 0; i < 3; i++) {
342  int index = (val & 0x7) * 6 + 4 - i*2;
343 
344  fixed_sparse->y[i] = (offset + index) & 0x3 ? -1 : 1;
345  fixed_sparse->x[i] = index;
346 
347  val >>= 3;
348  }
349  fixed_sparse->n = 3;
350  } else {
351  int pulse_subset = (pulses[0] >> 8) & 1;
352 
353  fixed_sparse->x[0] = ((pulses[0] >> 4) & 15) * 3 + pulse_subset;
354  fixed_sparse->x[1] = ( pulses[0] & 15) * 3 + pulse_subset + 1;
355 
356  fixed_sparse->y[0] = pulses[0] & 0x200 ? -1 : 1;
357  fixed_sparse->y[1] = -fixed_sparse->y[0];
358  fixed_sparse->n = 2;
359  }
360  break;
361  }
362 }
363 
365  float *out_data)
366 {
367  int i, j;
368  int subframe_count = modes[ctx->mode].subframe_count;
369  int frame_size = subframe_count * SUBFR_SIZE;
371  float *excitation;
372  float ir_buf[SUBFR_SIZE + LP_FILTER_ORDER];
373  float lsf_new[LP_FILTER_ORDER];
374  float *impulse_response = ir_buf + LP_FILTER_ORDER;
375  float *synth = ctx->synth_buf + 16; // 16 instead of LP_FILTER_ORDER for
376  // memory alignment
377  int t0_first = 0;
378  AMRFixed fixed_cb;
379 
380  memset(ir_buf, 0, LP_FILTER_ORDER * sizeof(float));
381  lsf_decode_fp(lsf_new, ctx->lsf_history, params);
382 
383  sipr_decode_lp(lsf_new, ctx->lsp_history, Az, subframe_count);
384 
385  memcpy(ctx->lsp_history, lsf_new, LP_FILTER_ORDER * sizeof(float));
386 
387  excitation = ctx->excitation + PITCH_DELAY_MAX + L_INTERPOL;
388 
389  for (i = 0; i < subframe_count; i++) {
390  float *pAz = Az + i*LP_FILTER_ORDER;
391  float fixed_vector[SUBFR_SIZE];
392  int T0,T0_frac;
393  float pitch_gain, gain_code, avg_energy;
394 
395  ff_decode_pitch_lag(&T0, &T0_frac, params->pitch_delay[i], t0_first, i,
396  ctx->mode == MODE_5k0, 6);
397 
398  if (i == 0 || (i == 2 && ctx->mode == MODE_5k0))
399  t0_first = T0;
400 
401  ff_acelp_interpolatef(excitation, excitation - T0 + (T0_frac <= 0),
402  ff_b60_sinc, 6,
403  2 * ((2 + T0_frac)%3 + 1), LP_FILTER_ORDER,
404  SUBFR_SIZE);
405 
406  decode_fixed_sparse(&fixed_cb, params->fc_indexes[i], ctx->mode,
407  ctx->past_pitch_gain < 0.8);
408 
409  eval_ir(pAz, T0, impulse_response, modes[ctx->mode].pitch_sharp_factor);
410 
411  convolute_with_sparse(fixed_vector, &fixed_cb, impulse_response,
412  SUBFR_SIZE);
413 
414  avg_energy = (0.01 + avpriv_scalarproduct_float_c(fixed_vector,
415  fixed_vector,
416  SUBFR_SIZE)) /
417  SUBFR_SIZE;
418 
419  ctx->past_pitch_gain = pitch_gain = gain_cb[params->gc_index[i]][0];
420 
421  gain_code = ff_amr_set_fixed_gain(gain_cb[params->gc_index[i]][1],
422  avg_energy, ctx->energy_history,
423  34 - 15.0/(0.05*M_LN10/M_LN2),
424  pred);
425 
426  ff_weighted_vector_sumf(excitation, excitation, fixed_vector,
427  pitch_gain, gain_code, SUBFR_SIZE);
428 
429  pitch_gain *= 0.5 * pitch_gain;
430  pitch_gain = FFMIN(pitch_gain, 0.4);
431 
432  ctx->gain_mem = 0.7 * ctx->gain_mem + 0.3 * pitch_gain;
433  ctx->gain_mem = FFMIN(ctx->gain_mem, pitch_gain);
434  gain_code *= ctx->gain_mem;
435 
436  for (j = 0; j < SUBFR_SIZE; j++)
437  fixed_vector[j] = excitation[j] - gain_code * fixed_vector[j];
438 
439  if (ctx->mode == MODE_5k0) {
440  postfilter_5k0(ctx, pAz, fixed_vector);
441 
442  ff_celp_lp_synthesis_filterf(ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i*SUBFR_SIZE,
443  pAz, excitation, SUBFR_SIZE,
444  LP_FILTER_ORDER);
445  }
446 
447  ff_celp_lp_synthesis_filterf(synth + i*SUBFR_SIZE, pAz, fixed_vector,
448  SUBFR_SIZE, LP_FILTER_ORDER);
449 
450  excitation += SUBFR_SIZE;
451  }
452 
453  memcpy(synth - LP_FILTER_ORDER, synth + frame_size - LP_FILTER_ORDER,
454  LP_FILTER_ORDER * sizeof(float));
455 
456  if (ctx->mode == MODE_5k0) {
457  for (i = 0; i < subframe_count; i++) {
458  float energy = avpriv_scalarproduct_float_c(ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i * SUBFR_SIZE,
459  ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i * SUBFR_SIZE,
460  SUBFR_SIZE);
461  ff_adaptive_gain_control(&synth[i * SUBFR_SIZE],
462  &synth[i * SUBFR_SIZE], energy,
463  SUBFR_SIZE, 0.9, &ctx->postfilter_agc);
464  }
465 
466  memcpy(ctx->postfilter_syn5k0, ctx->postfilter_syn5k0 + frame_size,
467  LP_FILTER_ORDER*sizeof(float));
468  }
469  memmove(ctx->excitation, excitation - PITCH_DELAY_MAX - L_INTERPOL,
470  (PITCH_DELAY_MAX + L_INTERPOL) * sizeof(float));
471 
473  (const float[2]) {-1.99997 , 1.000000000},
474  (const float[2]) {-1.93307352, 0.935891986},
475  0.939805806,
476  ctx->highpass_filt_mem,
477  frame_size);
478 }
479 
481 {
482  SiprContext *ctx = avctx->priv_data;
483  int i;
484 
485  switch (avctx->block_align) {
486  case 20: ctx->mode = MODE_16k; break;
487  case 19: ctx->mode = MODE_8k5; break;
488  case 29: ctx->mode = MODE_6k5; break;
489  case 37: ctx->mode = MODE_5k0; break;
490  default:
491  if (avctx->bit_rate > 12200) ctx->mode = MODE_16k;
492  else if (avctx->bit_rate > 7500 ) ctx->mode = MODE_8k5;
493  else if (avctx->bit_rate > 5750 ) ctx->mode = MODE_6k5;
494  else ctx->mode = MODE_5k0;
495  av_log(avctx, AV_LOG_WARNING,
496  "Invalid block_align: %d. Mode %s guessed based on bitrate: %d\n",
497  avctx->block_align, modes[ctx->mode].mode_name, avctx->bit_rate);
498  }
499 
500  av_log(avctx, AV_LOG_DEBUG, "Mode: %s\n", modes[ctx->mode].mode_name);
501 
502  if (ctx->mode == MODE_16k) {
503  ff_sipr_init_16k(ctx);
505  } else {
506  ctx->decode_frame = decode_frame;
507  }
508 
509  for (i = 0; i < LP_FILTER_ORDER; i++)
510  ctx->lsp_history[i] = cos((i+1) * M_PI / (LP_FILTER_ORDER + 1));
511 
512  for (i = 0; i < 4; i++)
513  ctx->energy_history[i] = -14;
514 
515  avctx->channels = 1;
517  avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
518 
519  return 0;
520 }
521 
522 static int sipr_decode_frame(AVCodecContext *avctx, void *data,
523  int *got_frame_ptr, AVPacket *avpkt)
524 {
525  SiprContext *ctx = avctx->priv_data;
526  AVFrame *frame = data;
527  const uint8_t *buf=avpkt->data;
528  SiprParameters parm;
529  const SiprModeParam *mode_par = &modes[ctx->mode];
530  GetBitContext gb;
531  float *samples;
532  int subframe_size = ctx->mode == MODE_16k ? L_SUBFR_16k : SUBFR_SIZE;
533  int i, ret;
534 
535  ctx->avctx = avctx;
536  if (avpkt->size < (mode_par->bits_per_frame >> 3)) {
537  av_log(avctx, AV_LOG_ERROR,
538  "Error processing packet: packet size (%d) too small\n",
539  avpkt->size);
540  return -1;
541  }
542 
543  /* get output buffer */
544  frame->nb_samples = mode_par->frames_per_packet * subframe_size *
545  mode_par->subframe_count;
546  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
547  return ret;
548  samples = (float *)frame->data[0];
549 
550  init_get_bits(&gb, buf, mode_par->bits_per_frame);
551 
552  for (i = 0; i < mode_par->frames_per_packet; i++) {
553  decode_parameters(&parm, &gb, mode_par);
554 
555  ctx->decode_frame(ctx, &parm, samples);
556 
557  samples += subframe_size * mode_par->subframe_count;
558  }
559 
560  *got_frame_ptr = 1;
561 
562  return mode_par->bits_per_frame >> 3;
563 }
564 
566  .name = "sipr",
567  .long_name = NULL_IF_CONFIG_SMALL("RealAudio SIPR / ACELP.NET"),
568  .type = AVMEDIA_TYPE_AUDIO,
569  .id = AV_CODEC_ID_SIPR,
570  .priv_data_size = sizeof(SiprContext),
573  .capabilities = AV_CODEC_CAP_DR1,
574 };
int gp_index[5]
adaptive-codebook gain indexes
Definition: sipr.h:60
void ff_decode_pitch_lag(int *lag_int, int *lag_frac, int pitch_index, const int prev_lag_int, const int subframe, int third_as_first, int resolution)
Decode the adaptive codebook index to the integer and fractional parts of the pitch lag for one subfr...
const char const char void * val
Definition: avisynth_c.h:634
int pitch_delay[5]
pitch delay
Definition: sipr.h:59
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
int vq_indexes[5]
Definition: sipr.h:58
This structure describes decoded (raw) audio or video data.
Definition: frame.h:171
uint8_t gp_index_bits
Definition: sipr.c:64
uint8_t vq_indexes_bits[5]
size in bits of the i-th stage vector of quantizer
Definition: sipr.c:59
Definition: sipr.h:50
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:101
#define SUBFR_SIZE
Subframe size for all modes except 16k.
Definition: sipr.h:44
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 av_cold int sipr_decoder_init(AVCodecContext *avctx)
Definition: sipr.c:480
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 av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
static void lsf_decode_fp(float *lsfnew, float *lsf_history, const SiprParameters *parm)
Definition: sipr.c:153
int x[10]
Definition: acelp_vectors.h:55
int size
Definition: avcodec.h:1423
#define SUBFRAME_COUNT_16k
Definition: sipr.h:46
#define LSFQ_DIFF_MIN
minimum LSF distance (3.2.4) 0.0391 in Q13
Definition: g729dec.c:58
AVCodec.
Definition: avcodec.h:3471
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs...
Definition: avcodec.h:2298
static const float *const lsf_codebooks[]
Definition: siprdata.h:209
float synth_buf[LP_FILTER_ORDER+5 *SUBFR_SIZE+6]
Definition: sipr.h:75
float postfilter_syn5k0[LP_FILTER_ORDER+SUBFR_SIZE *5]
Definition: sipr.h:87
uint8_t number_of_fc_indexes
Definition: sipr.c:55
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
float lsf_history[LP_FILTER_ORDER_16k]
Definition: sipr.h:71
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2269
uint8_t
#define av_cold
Definition: attributes.h:74
Sparse representation for the algebraic codebook (fixed) vector.
Definition: acelp_vectors.h:53
uint8_t fc_index_bits[10]
size in bits of the fixed codebook indexes
Definition: sipr.c:65
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:145
mode
Definition: f_perms.c:27
static const float gain_cb[128][2]
Definition: siprdata.h:213
SiprMode
Definition: sipr.h:48
#define PITCH_DELAY_MAX
#define L_INTERPOL
Number of past samples needed for excitation interpolation.
Definition: sipr.h:41
#define t0
Definition: regdef.h:28
static AVFrame * frame
static void eval_ir(const float *Az, int pitch_lag, float *freq, float pitch_sharp_factor)
Evaluate the adaptive impulse response.
Definition: sipr.c:237
uint8_t * data
Definition: avcodec.h:1422
Definition: sipr.h:49
float highpass_filt_mem[2]
Definition: sipr.h:80
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.
float pitch_sharp_factor
Definition: sipr.c:52
#define av_log(a,...)
#define MAX_SUBFRAME_COUNT
Definition: sipr.c:42
uint8_t ma_predictor_bits
size in bits of the switched MA predictor
Definition: sipr.c:56
float lsp_history[LP_FILTER_ORDER]
Definition: sipr.h:77
uint16_t bits_per_frame
Definition: sipr.c:49
Definition: sipr.h:52
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
uint8_t pitch_delay_bits[5]
size in bits of the adaptive-codebook index for every subframe
Definition: sipr.c:62
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:175
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:197
AVCodecContext * avctx
Definition: sipr.h:66
GLenum GLint * params
Definition: opengl_enc.c:114
const float ff_pow_0_7[10]
Table of pow(0.7,n)
Definition: acelp_vectors.c:99
GLsizei GLsizei * length
Definition: opengl_enc.c:115
const char * name
Name of the codec implementation.
Definition: avcodec.h:3478
const float ff_pow_0_75[10]
Table of pow(0.75,n)
#define LP_FILTER_ORDER
linear predictive coding filter order
Definition: amrnbdata.h:53
static const uint8_t offset[127][2]
Definition: vf_spp.c:92
Libavcodec external API header.
static void decode_parameters(SiprParameters *parms, GetBitContext *pgb, const SiprModeParam *p)
Extract decoding parameters from the input bitstream.
Definition: sipr.c:193
static void postfilter_5k0(SiprContext *ctx, const float *lpc, float *samples)
Apply postfilter, very similar to AMR one.
Definition: sipr.c:273
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:2322
Definition: sipr.h:51
int bit_rate
the average bitrate
Definition: avcodec.h:1566
audio channel layout utility functions
#define M_LN2
Definition: mathematics.h:34
#define FFMIN(a, b)
Definition: common.h:66
const char * mode_name
Definition: sipr.c:48
float y[10]
Definition: acelp_vectors.h:56
static void decode_frame(SiprContext *ctx, SiprParameters *params, float *out_data)
Definition: sipr.c:364
#define L_SUBFR_16k
Definition: sipr.h:32
int16_t fc_indexes[5][10]
fixed-codebook indexes
Definition: sipr.h:61
float energy_history[4]
Definition: sipr.h:79
void ff_tilt_compensation(float *mem, float tilt, float *samples, int size)
Apply tilt compensation filter, 1 - tilt * z-1.
static const float pred[4]
Definition: siprdata.h:259
AVCodec ff_sipr_decoder
Definition: sipr.c:565
int frame_size
Definition: mxfenc.c:1803
main external API structure.
Definition: avcodec.h:1501
uint8_t subframe_count
Definition: sipr.c:50
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
void(* decode_frame)(struct SiprContext *ctx, SiprParameters *params, float *out_data)
Definition: sipr.h:98
int index
Definition: gxfenc.c:89
float postfilter_mem5k0[PITCH_DELAY_MAX+LP_FILTER_ORDER]
Definition: sipr.h:86
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
Definition: get_bits.h:410
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
static const float mean_lsf[10]
Definition: siprdata.h:27
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:182
float postfilter_mem[PITCH_DELAY_MAX+LP_FILTER_ORDER]
Definition: sipr.h:81
static int sipr_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: sipr.c:522
#define M_LN10
Definition: mathematics.h:37
static int decode(AVCodecContext *avctx, void *data, int *got_sub, AVPacket *avpkt)
Definition: ccaption_dec.c:523
GLint GLenum GLboolean GLsizei stride
Definition: opengl_enc.c:105
common internal api header.
int gc_index[5]
fixed-codebook gain indexes
Definition: sipr.h:62
static void pitch_sharpening(int pitch_lag_int, float beta, float *fixed_vector)
Apply pitch lag to the fixed vector (AMR section 6.1.2).
Definition: sipr.c:179
static void decode_fixed_sparse(AMRFixed *fixed_sparse, const int16_t *pulses, SiprMode mode, int low_gain)
Definition: sipr.c:308
float gain_mem
Definition: sipr.h:78
float excitation[L_INTERPOL+PITCH_MAX+2 *L_SUBFR_16k]
Definition: sipr.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:51
void ff_sipr_decode_frame_16k(SiprContext *ctx, SiprParameters *params, float *out_data)
Definition: sipr16k.c:176
void ff_sort_nearly_sorted_floats(float *vals, int len)
Sort values in ascending order.
Definition: lsp.c:228
float postfilter_agc
Definition: sipr.h:85
static const SiprModeParam modes[MODE_COUNT]
Definition: sipr.c:69
uint8_t frames_per_packet
Definition: sipr.c:51
void * priv_data
Definition: avcodec.h:1543
const float ff_b60_sinc[61]
b60 hamming windowed sinc function coefficients
int channels
number of audio channels
Definition: avcodec.h:2262
const float ff_pow_0_5[]
Definition: sipr.c:135
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: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_log(ac->avr, AV_LOG_TRACE,"%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 int8_t pulses[4]
Number of non-zero pulses in the MP-MLQ excitation.
Definition: g723_1_data.h:608
int ma_pred_switch
switched moving average predictor
Definition: sipr.h:57
float past_pitch_gain
Definition: sipr.h:70
#define M_PI
Definition: mathematics.h:46
SiprMode mode
Definition: sipr.h:68
#define AV_CH_LAYOUT_MONO
static void convolute_with_sparse(float *out, const AMRFixed *pulses, const float *shape, int length)
Evaluate the convolution of a vector with a sparse vector.
Definition: sipr.c:259
This structure stores compressed data.
Definition: avcodec.h:1399
uint8_t gc_index_bits
size in bits of the gain codebook indexes
Definition: sipr.c:66
const float ff_pow_0_55[10]
Table of pow(0.55,n)
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
float tilt_mem
Definition: sipr.h:84
static void dequant(float *out, const int *idx, const float *const cbs[])
Definition: sipr.c:142
static void sipr_decode_lp(float *lsfnew, const float *lsfold, float *Az, int num_subfr)
Definition: sipr.c:216
void ff_sipr_init_16k(SiprContext *ctx)
Definition: sipr16k.c:271
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)