FFmpeg
sipr.c
Go to the documentation of this file.
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 
32 #define BITSTREAM_READER_LE
33 #include "avcodec.h"
34 #include "codec_internal.h"
35 #include "decode.h"
36 #include "get_bits.h"
37 #include "lsp.h"
38 #include "acelp_vectors.h"
39 #include "acelp_pitch_delay.h"
40 #include "acelp_filters.h"
41 #include "celp_filters.h"
42 
43 #define MAX_SUBFRAME_COUNT 5
44 
45 #include "sipr.h"
46 #include "siprdata.h"
47 
48 typedef struct SiprModeParam {
49  const char *mode_name;
50  uint16_t bits_per_frame;
51  uint8_t subframe_count;
54 
55  /* bitstream parameters */
57  uint8_t ma_predictor_bits; ///< size in bits of the switched MA predictor
58 
59  /** size in bits of the i-th stage vector of quantizer */
60  uint8_t vq_indexes_bits[5];
61 
62  /** size in bits of the adaptive-codebook index for every subframe */
63  uint8_t pitch_delay_bits[5];
64 
65  uint8_t gp_index_bits;
66  uint8_t fc_index_bits[10]; ///< size in bits of the fixed codebook indexes
67  uint8_t gc_index_bits; ///< size in bits of the gain codebook indexes
69 
70 static const SiprModeParam modes[MODE_COUNT] = {
71  [MODE_16k] = {
72  .mode_name = "16k",
73  .bits_per_frame = 160,
74  .subframe_count = SUBFRAME_COUNT_16k,
75  .frames_per_packet = 1,
76  .pitch_sharp_factor = 0.00,
77 
78  .number_of_fc_indexes = 10,
79  .ma_predictor_bits = 1,
80  .vq_indexes_bits = {7, 8, 7, 7, 7},
81  .pitch_delay_bits = {9, 6},
82  .gp_index_bits = 4,
83  .fc_index_bits = {4, 5, 4, 5, 4, 5, 4, 5, 4, 5},
84  .gc_index_bits = 5
85  },
86 
87  [MODE_8k5] = {
88  .mode_name = "8k5",
89  .bits_per_frame = 152,
90  .subframe_count = 3,
91  .frames_per_packet = 1,
92  .pitch_sharp_factor = 0.8,
93 
94  .number_of_fc_indexes = 3,
95  .ma_predictor_bits = 0,
96  .vq_indexes_bits = {6, 7, 7, 7, 5},
97  .pitch_delay_bits = {8, 5, 5},
98  .gp_index_bits = 0,
99  .fc_index_bits = {9, 9, 9},
100  .gc_index_bits = 7
101  },
102 
103  [MODE_6k5] = {
104  .mode_name = "6k5",
105  .bits_per_frame = 232,
106  .subframe_count = 3,
107  .frames_per_packet = 2,
108  .pitch_sharp_factor = 0.8,
109 
110  .number_of_fc_indexes = 3,
111  .ma_predictor_bits = 0,
112  .vq_indexes_bits = {6, 7, 7, 7, 5},
113  .pitch_delay_bits = {8, 5, 5},
114  .gp_index_bits = 0,
115  .fc_index_bits = {5, 5, 5},
116  .gc_index_bits = 7
117  },
118 
119  [MODE_5k0] = {
120  .mode_name = "5k0",
121  .bits_per_frame = 296,
122  .subframe_count = 5,
123  .frames_per_packet = 2,
124  .pitch_sharp_factor = 0.85,
125 
126  .number_of_fc_indexes = 1,
127  .ma_predictor_bits = 0,
128  .vq_indexes_bits = {6, 7, 7, 7, 5},
129  .pitch_delay_bits = {8, 5, 8, 5, 5},
130  .gp_index_bits = 0,
131  .fc_index_bits = {10},
132  .gc_index_bits = 7
133  }
134 };
135 
136 const float ff_pow_0_5[] = {
137  1.0/(1 << 1), 1.0/(1 << 2), 1.0/(1 << 3), 1.0/(1 << 4),
138  1.0/(1 << 5), 1.0/(1 << 6), 1.0/(1 << 7), 1.0/(1 << 8),
139  1.0/(1 << 9), 1.0/(1 << 10), 1.0/(1 << 11), 1.0/(1 << 12),
140  1.0/(1 << 13), 1.0/(1 << 14), 1.0/(1 << 15), 1.0/(1 << 16)
141 };
142 
143 static void dequant(float *out, const int *idx, const float * const cbs[])
144 {
145  int i;
146  int stride = 2;
147  int num_vec = 5;
148 
149  for (i = 0; i < num_vec; i++)
150  memcpy(out + stride*i, cbs[i] + stride*idx[i], stride*sizeof(float));
151 
152 }
153 
154 static void lsf_decode_fp(float *lsfnew, float *lsf_history,
155  const SiprParameters *parm)
156 {
157  int i;
158  float lsf_tmp[LP_FILTER_ORDER];
159 
160  dequant(lsf_tmp, parm->vq_indexes, lsf_codebooks);
161 
162  for (i = 0; i < LP_FILTER_ORDER; i++)
163  lsfnew[i] = lsf_history[i] * 0.33 + lsf_tmp[i] + mean_lsf[i];
164 
166 
167  /* Note that a minimum distance is not enforced between the last value and
168  the previous one, contrary to what is done in ff_acelp_reorder_lsf() */
170  lsfnew[9] = FFMIN(lsfnew[LP_FILTER_ORDER - 1], 1.3 * M_PI);
171 
172  memcpy(lsf_history, lsf_tmp, LP_FILTER_ORDER * sizeof(*lsf_history));
173 
174  for (i = 0; i < LP_FILTER_ORDER - 1; i++)
175  lsfnew[i] = cos(lsfnew[i]);
176  lsfnew[LP_FILTER_ORDER - 1] *= 6.153848 / M_PI;
177 }
178 
179 /** Apply pitch lag to the fixed vector (AMR section 6.1.2). */
180 static void pitch_sharpening(int pitch_lag_int, float beta,
181  float *fixed_vector)
182 {
183  int i;
184 
185  for (i = pitch_lag_int; i < SUBFR_SIZE; i++)
186  fixed_vector[i] += beta * fixed_vector[i - pitch_lag_int];
187 }
188 
189 /**
190  * Extract decoding parameters from the input bitstream.
191  * @param parms parameters structure
192  * @param pgb pointer to initialized GetBitContext structure
193  */
195  const SiprModeParam *p)
196 {
197  int i, j;
198 
199  if (p->ma_predictor_bits)
200  parms->ma_pred_switch = get_bits(pgb, p->ma_predictor_bits);
201 
202  for (i = 0; i < 5; i++)
203  parms->vq_indexes[i] = get_bits(pgb, p->vq_indexes_bits[i]);
204 
205  for (i = 0; i < p->subframe_count; i++) {
206  parms->pitch_delay[i] = get_bits(pgb, p->pitch_delay_bits[i]);
207  if (p->gp_index_bits)
208  parms->gp_index[i] = get_bits(pgb, p->gp_index_bits);
209 
210  for (j = 0; j < p->number_of_fc_indexes; j++)
211  parms->fc_indexes[i][j] = get_bits(pgb, p->fc_index_bits[j]);
212 
213  parms->gc_index[i] = get_bits(pgb, p->gc_index_bits);
214  }
215 }
216 
217 static void sipr_decode_lp(float *lsfnew, const float *lsfold, float *Az,
218  int num_subfr)
219 {
220  double lsfint[LP_FILTER_ORDER];
221  int i,j;
222  float t, t0 = 1.0 / num_subfr;
223 
224  t = t0 * 0.5;
225  for (i = 0; i < num_subfr; i++) {
226  for (j = 0; j < LP_FILTER_ORDER; j++)
227  lsfint[j] = lsfold[j] * (1 - t) + t * lsfnew[j];
228 
229  ff_amrwb_lsp2lpc(lsfint, Az, LP_FILTER_ORDER);
230  Az += LP_FILTER_ORDER;
231  t += t0;
232  }
233 }
234 
235 /**
236  * Evaluate the adaptive impulse response.
237  */
238 static void eval_ir(const float *Az, int pitch_lag, float *freq,
239  float pitch_sharp_factor)
240 {
241  float tmp1[SUBFR_SIZE+1], tmp2[LP_FILTER_ORDER+1];
242  int i;
243 
244  tmp1[0] = 1.0;
245  for (i = 0; i < LP_FILTER_ORDER; i++) {
246  tmp1[i+1] = Az[i] * ff_pow_0_55[i];
247  tmp2[i ] = Az[i] * ff_pow_0_7 [i];
248  }
249  memset(tmp1 + 11, 0, 37 * sizeof(float));
250 
251  ff_celp_lp_synthesis_filterf(freq, tmp2, tmp1, SUBFR_SIZE,
253 
254  pitch_sharpening(pitch_lag, pitch_sharp_factor, freq);
255 }
256 
257 /**
258  * Evaluate the convolution of a vector with a sparse vector.
259  */
260 static void convolute_with_sparse(float *out, const AMRFixed *pulses,
261  const float *shape, int length)
262 {
263  int i, j;
264 
265  memset(out, 0, length*sizeof(float));
266  for (i = 0; i < pulses->n; i++)
267  for (j = pulses->x[i]; j < length; j++)
268  out[j] += pulses->y[i] * shape[j - pulses->x[i]];
269 }
270 
271 /**
272  * Apply postfilter, very similar to AMR one.
273  */
274 static void postfilter_5k0(SiprContext *ctx, const float *lpc, float *samples)
275 {
276  float buf[SUBFR_SIZE + LP_FILTER_ORDER];
277  float *pole_out = buf + LP_FILTER_ORDER;
278  float lpc_n[LP_FILTER_ORDER];
279  float lpc_d[LP_FILTER_ORDER];
280  int i;
281 
282  for (i = 0; i < LP_FILTER_ORDER; i++) {
283  lpc_d[i] = lpc[i] * ff_pow_0_75[i];
284  lpc_n[i] = lpc[i] * ff_pow_0_5 [i];
285  };
286 
287  memcpy(pole_out - LP_FILTER_ORDER, ctx->postfilter_mem,
288  LP_FILTER_ORDER*sizeof(float));
289 
292 
293  memcpy(ctx->postfilter_mem, pole_out + SUBFR_SIZE - LP_FILTER_ORDER,
294  LP_FILTER_ORDER*sizeof(float));
295 
296  ff_tilt_compensation(&ctx->tilt_mem, 0.4, pole_out, SUBFR_SIZE);
297 
298  memcpy(pole_out - LP_FILTER_ORDER, ctx->postfilter_mem5k0,
299  LP_FILTER_ORDER*sizeof(*pole_out));
300 
301  memcpy(ctx->postfilter_mem5k0, pole_out + SUBFR_SIZE - LP_FILTER_ORDER,
302  LP_FILTER_ORDER*sizeof(*pole_out));
303 
306 
307 }
308 
309 static void decode_fixed_sparse(AMRFixed *fixed_sparse, const int16_t *pulses,
310  SiprMode mode, int low_gain)
311 {
312  int i;
313 
314  switch (mode) {
315  case MODE_6k5:
316  for (i = 0; i < 3; i++) {
317  fixed_sparse->x[i] = 3 * (pulses[i] & 0xf) + i;
318  fixed_sparse->y[i] = pulses[i] & 0x10 ? -1 : 1;
319  }
320  fixed_sparse->n = 3;
321  break;
322  case MODE_8k5:
323  for (i = 0; i < 3; i++) {
324  fixed_sparse->x[2*i ] = 3 * ((pulses[i] >> 4) & 0xf) + i;
325  fixed_sparse->x[2*i + 1] = 3 * ( pulses[i] & 0xf) + i;
326 
327  fixed_sparse->y[2*i ] = (pulses[i] & 0x100) ? -1.0: 1.0;
328 
329  fixed_sparse->y[2*i + 1] =
330  (fixed_sparse->x[2*i + 1] < fixed_sparse->x[2*i]) ?
331  -fixed_sparse->y[2*i ] : fixed_sparse->y[2*i];
332  }
333 
334  fixed_sparse->n = 6;
335  break;
336  case MODE_5k0:
337  default:
338  if (low_gain) {
339  int offset = (pulses[0] & 0x200) ? 2 : 0;
340  int val = pulses[0];
341 
342  for (i = 0; i < 3; i++) {
343  int index = (val & 0x7) * 6 + 4 - i*2;
344 
345  fixed_sparse->y[i] = (offset + index) & 0x3 ? -1 : 1;
346  fixed_sparse->x[i] = index;
347 
348  val >>= 3;
349  }
350  fixed_sparse->n = 3;
351  } else {
352  int pulse_subset = (pulses[0] >> 8) & 1;
353 
354  fixed_sparse->x[0] = ((pulses[0] >> 4) & 15) * 3 + pulse_subset;
355  fixed_sparse->x[1] = ( pulses[0] & 15) * 3 + pulse_subset + 1;
356 
357  fixed_sparse->y[0] = pulses[0] & 0x200 ? -1 : 1;
358  fixed_sparse->y[1] = -fixed_sparse->y[0];
359  fixed_sparse->n = 2;
360  }
361  break;
362  }
363 }
364 
366  float *out_data)
367 {
368  int i, j;
369  int subframe_count = modes[ctx->mode].subframe_count;
370  int frame_size = subframe_count * SUBFR_SIZE;
372  float *excitation;
373  float ir_buf[SUBFR_SIZE + LP_FILTER_ORDER];
374  float lsf_new[LP_FILTER_ORDER];
375  float *impulse_response = ir_buf + LP_FILTER_ORDER;
376  float *synth = ctx->synth_buf + 16; // 16 instead of LP_FILTER_ORDER for
377  // memory alignment
378  int t0_first = 0;
379  AMRFixed fixed_cb;
380 
381  memset(ir_buf, 0, LP_FILTER_ORDER * sizeof(float));
382  lsf_decode_fp(lsf_new, ctx->lsf_history, params);
383 
384  sipr_decode_lp(lsf_new, ctx->lsp_history, Az, subframe_count);
385 
386  memcpy(ctx->lsp_history, lsf_new, LP_FILTER_ORDER * sizeof(float));
387 
388  excitation = ctx->excitation + PITCH_DELAY_MAX + L_INTERPOL;
389 
390  for (i = 0; i < subframe_count; i++) {
391  float *pAz = Az + i*LP_FILTER_ORDER;
392  float fixed_vector[SUBFR_SIZE];
393  int T0,T0_frac;
394  float pitch_gain, gain_code, avg_energy;
395 
396  ff_decode_pitch_lag(&T0, &T0_frac, params->pitch_delay[i], t0_first, i,
397  ctx->mode == MODE_5k0, 6);
398 
399  if (i == 0 || (i == 2 && ctx->mode == MODE_5k0))
400  t0_first = T0;
401 
402  ff_acelp_interpolatef(excitation, excitation - T0 + (T0_frac <= 0),
403  ff_b60_sinc, 6,
404  2 * ((2 + T0_frac)%3 + 1), LP_FILTER_ORDER,
405  SUBFR_SIZE);
406 
407  decode_fixed_sparse(&fixed_cb, params->fc_indexes[i], ctx->mode,
408  ctx->past_pitch_gain < 0.8);
409 
410  eval_ir(pAz, T0, impulse_response, modes[ctx->mode].pitch_sharp_factor);
411 
412  convolute_with_sparse(fixed_vector, &fixed_cb, impulse_response,
413  SUBFR_SIZE);
414 
415  avg_energy = (0.01 + avpriv_scalarproduct_float_c(fixed_vector,
416  fixed_vector,
417  SUBFR_SIZE)) /
418  SUBFR_SIZE;
419 
420  ctx->past_pitch_gain = pitch_gain = gain_cb[params->gc_index[i]][0];
421 
422  gain_code = ff_amr_set_fixed_gain(gain_cb[params->gc_index[i]][1],
423  avg_energy, ctx->energy_history,
424  34 - 15.0/(0.05*M_LN10/M_LN2),
425  pred);
426 
427  ff_weighted_vector_sumf(excitation, excitation, fixed_vector,
428  pitch_gain, gain_code, SUBFR_SIZE);
429 
430  pitch_gain *= 0.5 * pitch_gain;
431  pitch_gain = FFMIN(pitch_gain, 0.4);
432 
433  ctx->gain_mem = 0.7 * ctx->gain_mem + 0.3 * pitch_gain;
434  ctx->gain_mem = FFMIN(ctx->gain_mem, pitch_gain);
435  gain_code *= ctx->gain_mem;
436 
437  for (j = 0; j < SUBFR_SIZE; j++)
438  fixed_vector[j] = excitation[j] - gain_code * fixed_vector[j];
439 
440  if (ctx->mode == MODE_5k0) {
441  postfilter_5k0(ctx, pAz, fixed_vector);
442 
444  pAz, excitation, SUBFR_SIZE,
446  }
447 
448  ff_celp_lp_synthesis_filterf(synth + i*SUBFR_SIZE, pAz, fixed_vector,
450 
451  excitation += SUBFR_SIZE;
452  }
453 
454  memcpy(synth - LP_FILTER_ORDER, synth + frame_size - LP_FILTER_ORDER,
455  LP_FILTER_ORDER * sizeof(float));
456 
457  if (ctx->mode == MODE_5k0) {
458  for (i = 0; i < subframe_count; i++) {
459  float energy = avpriv_scalarproduct_float_c(ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i * SUBFR_SIZE,
460  ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i * SUBFR_SIZE,
461  SUBFR_SIZE);
463  &synth[i * SUBFR_SIZE], energy,
464  SUBFR_SIZE, 0.9, &ctx->postfilter_agc);
465  }
466 
467  memcpy(ctx->postfilter_syn5k0, ctx->postfilter_syn5k0 + frame_size,
468  LP_FILTER_ORDER*sizeof(float));
469  }
470  memmove(ctx->excitation, excitation - PITCH_DELAY_MAX - L_INTERPOL,
471  (PITCH_DELAY_MAX + L_INTERPOL) * sizeof(float));
472 
474  (const float[2]) {-1.99997 , 1.000000000},
475  (const float[2]) {-1.93307352, 0.935891986},
476  0.939805806,
477  ctx->highpass_filt_mem,
478  frame_size);
479 }
480 
482 {
483  SiprContext *ctx = avctx->priv_data;
484  int i;
485 
486  switch (avctx->block_align) {
487  case 20: ctx->mode = MODE_16k; break;
488  case 19: ctx->mode = MODE_8k5; break;
489  case 29: ctx->mode = MODE_6k5; break;
490  case 37: ctx->mode = MODE_5k0; break;
491  default:
492  if (avctx->bit_rate > 12200) ctx->mode = MODE_16k;
493  else if (avctx->bit_rate > 7500 ) ctx->mode = MODE_8k5;
494  else if (avctx->bit_rate > 5750 ) ctx->mode = MODE_6k5;
495  else ctx->mode = MODE_5k0;
496  av_log(avctx, AV_LOG_WARNING,
497  "Invalid block_align: %d. Mode %s guessed based on bitrate: %"PRId64"\n",
498  avctx->block_align, modes[ctx->mode].mode_name, avctx->bit_rate);
499  }
500 
501  av_log(avctx, AV_LOG_DEBUG, "Mode: %s\n", modes[ctx->mode].mode_name);
502 
503  if (ctx->mode == MODE_16k) {
505  ctx->decode_frame = ff_sipr_decode_frame_16k;
506  } else {
507  ctx->decode_frame = decode_frame;
508  }
509 
510  for (i = 0; i < LP_FILTER_ORDER; i++)
511  ctx->lsp_history[i] = cos((i+1) * M_PI / (LP_FILTER_ORDER + 1));
512 
513  for (i = 0; i < 4; i++)
514  ctx->energy_history[i] = -14;
515 
518  avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
519 
520  return 0;
521 }
522 
524  int *got_frame_ptr, AVPacket *avpkt)
525 {
526  SiprContext *ctx = avctx->priv_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  if (avpkt->size < (mode_par->bits_per_frame >> 3)) {
536  av_log(avctx, AV_LOG_ERROR,
537  "Error processing packet: packet size (%d) too small\n",
538  avpkt->size);
539  return AVERROR_INVALIDDATA;
540  }
541 
542  /* get output buffer */
543  frame->nb_samples = mode_par->frames_per_packet * subframe_size *
544  mode_par->subframe_count;
545  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
546  return ret;
547  samples = (float *)frame->data[0];
548 
549  init_get_bits(&gb, buf, mode_par->bits_per_frame);
550 
551  for (i = 0; i < mode_par->frames_per_packet; i++) {
552  decode_parameters(&parm, &gb, mode_par);
553 
554  ctx->decode_frame(ctx, &parm, samples);
555 
556  samples += subframe_size * mode_par->subframe_count;
557  }
558 
559  *got_frame_ptr = 1;
560 
561  return mode_par->bits_per_frame >> 3;
562 }
563 
565  .p.name = "sipr",
566  CODEC_LONG_NAME("RealAudio SIPR / ACELP.NET"),
567  .p.type = AVMEDIA_TYPE_AUDIO,
568  .p.id = AV_CODEC_ID_SIPR,
569  .priv_data_size = sizeof(SiprContext),
572  .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF,
573 };
AV_LOG_WARNING
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:186
AMRFixed::x
int x[10]
Definition: acelp_vectors.h:55
acelp_vectors.h
eval_ir
static void eval_ir(const float *Az, int pitch_lag, float *freq, float pitch_sharp_factor)
Evaluate the adaptive impulse response.
Definition: sipr.c:238
SiprModeParam::gp_index_bits
uint8_t gp_index_bits
Definition: sipr.c:65
ff_amr_set_fixed_gain
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)
Definition: acelp_pitch_delay.c:84
SiprModeParam::mode_name
const char * mode_name
Definition: sipr.c:49
out
FILE * out
Definition: movenc.c:54
ff_decode_pitch_lag
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...
Definition: acelp_pitch_delay.c:105
postfilter_5k0
static void postfilter_5k0(SiprContext *ctx, const float *lpc, float *samples)
Apply postfilter, very similar to AMR one.
Definition: sipr.c:274
ff_amrwb_lsp2lpc
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:175
ff_b60_sinc
const float ff_b60_sinc[61]
b60 hamming windowed sinc function coefficients
Definition: acelp_vectors.c:103
siprdata.h
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:340
MODE_5k0
@ MODE_5k0
Definition: sipr.h:51
ff_acelp_apply_order_2_transfer_function
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.
Definition: acelp_filters.c:121
AVPacket::data
uint8_t * data
Definition: packet.h:522
ff_sipr_decode_frame_16k
void ff_sipr_decode_frame_16k(SiprContext *ctx, SiprParameters *params, float *out_data)
Definition: sipr16k.c:176
t0
#define t0
Definition: regdef.h:28
FFCodec
Definition: codec_internal.h:127
mathematics.h
ff_sort_nearly_sorted_floats
void ff_sort_nearly_sorted_floats(float *vals, int len)
Sort values in ascending order.
Definition: lsp.c:239
LP_FILTER_ORDER
#define LP_FILTER_ORDER
linear predictive coding filter order
Definition: amrnbdata.h:53
MAX_SUBFRAME_COUNT
#define MAX_SUBFRAME_COUNT
Definition: sipr.c:43
ff_celp_lp_synthesis_filterf
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:85
ff_sipr_decoder
const FFCodec ff_sipr_decoder
Definition: sipr.c:564
init_get_bits
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
Definition: get_bits.h:514
AVFrame::data
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:361
LSFQ_DIFF_MIN
#define LSFQ_DIFF_MIN
minimum LSF distance (3.2.4) 0.0391 in Q13
Definition: g729dec.c:58
MODE_8k5
@ MODE_8k5
Definition: sipr.h:49
M_LN2
#define M_LN2
Definition: mathematics.h:43
SiprModeParam::frames_per_packet
uint8_t frames_per_packet
Definition: sipr.c:52
SUBFR_SIZE
#define SUBFR_SIZE
Subframe size for all modes except 16k.
Definition: sipr.h:43
get_bits
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:335
FFCodec::p
AVCodec p
The public AVCodec.
Definition: codec_internal.h:131
PITCH_DELAY_MAX
#define PITCH_DELAY_MAX
Definition: acelp_pitch_delay.h:31
ff_pow_0_55
const float ff_pow_0_55[10]
Table of pow(0.55,n)
Definition: acelp_vectors.c:98
AVCodecContext::ch_layout
AVChannelLayout ch_layout
Audio channel layout.
Definition: avcodec.h:1065
lsf_decode_fp
static void lsf_decode_fp(float *lsfnew, float *lsf_history, const SiprParameters *parm)
Definition: sipr.c:154
convolute_with_sparse
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:260
SiprModeParam::number_of_fc_indexes
uint8_t number_of_fc_indexes
Definition: sipr.c:56
GetBitContext
Definition: get_bits.h:108
SiprParameters
Definition: sipr.h:55
val
static double val(void *priv, double ch)
Definition: aeval.c:78
SiprParameters::gc_index
int gc_index[5]
fixed-codebook gain indexes
Definition: sipr.h:61
SiprModeParam::pitch_sharp_factor
float pitch_sharp_factor
Definition: sipr.c:53
ff_adaptive_gain_control
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)
Definition: acelp_vectors.c:192
decode_parameters
static void decode_parameters(SiprParameters *parms, GetBitContext *pgb, const SiprModeParam *p)
Extract decoding parameters from the input bitstream.
Definition: sipr.c:194
AV_CODEC_ID_SIPR
@ AV_CODEC_ID_SIPR
Definition: codec_id.h:481
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:180
av_cold
#define av_cold
Definition: attributes.h:90
FF_CODEC_DECODE_CB
#define FF_CODEC_DECODE_CB(func)
Definition: codec_internal.h:287
frame_size
int frame_size
Definition: mxfenc.c:2422
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
dequant
static void dequant(float *out, const int *idx, const float *const cbs[])
Definition: sipr.c:143
AMRFixed
Sparse representation for the algebraic codebook (fixed) vector.
Definition: acelp_vectors.h:53
sipr_decode_lp
static void sipr_decode_lp(float *lsfnew, const float *lsfold, float *Az, int num_subfr)
Definition: sipr.c:217
AV_LOG_DEBUG
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:201
ctx
AVFormatContext * ctx
Definition: movenc.c:48
decode.h
get_bits.h
AMRFixed::y
float y[10]
Definition: acelp_vectors.h:56
CODEC_LONG_NAME
#define CODEC_LONG_NAME(str)
Definition: codec_internal.h:272
frame
static AVFrame * frame
Definition: demux_decode.c:54
decode_fixed_sparse
static void decode_fixed_sparse(AMRFixed *fixed_sparse, const int16_t *pulses, SiprMode mode, int low_gain)
Definition: sipr.c:309
SiprMode
SiprMode
Definition: sipr.h:47
SiprParameters::ma_pred_switch
int ma_pred_switch
switched moving average predictor
Definition: sipr.h:56
L_INTERPOL
#define L_INTERPOL
Number of past samples needed for excitation interpolation.
Definition: sipr.h:40
SiprModeParam::vq_indexes_bits
uint8_t vq_indexes_bits[5]
size in bits of the i-th stage vector of quantizer
Definition: sipr.c:60
MODE_16k
@ MODE_16k
Definition: sipr.h:48
AVCodecContext::bit_rate
int64_t bit_rate
the average bitrate
Definition: avcodec.h:495
SiprParameters::fc_indexes
int16_t fc_indexes[5][10]
fixed-codebook indexes
Definition: sipr.h:60
SiprParameters::pitch_delay
int pitch_delay[5]
pitch delay
Definition: sipr.h:58
celp_filters.h
index
int index
Definition: gxfenc.c:89
float_dsp.h
AV_CODEC_CAP_CHANNEL_CONF
#define AV_CODEC_CAP_CHANNEL_CONF
Codec should fill in channel configuration and samplerate instead of container.
Definition: codec.h:106
ff_get_buffer
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1568
init
int(* init)(AVBSFContext *ctx)
Definition: dts2pts.c:365
AV_CODEC_CAP_DR1
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
Definition: codec.h:52
AVPacket::size
int size
Definition: packet.h:523
AVChannelLayout
An AVChannelLayout holds information about the channel layout of audio data.
Definition: channel_layout.h:303
codec_internal.h
AVCodecContext::sample_fmt
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1057
ff_pow_0_7
const float ff_pow_0_7[10]
Table of pow(0.7,n)
Definition: acelp_vectors.c:88
sipr_decode_frame
static int sipr_decode_frame(AVCodecContext *avctx, AVFrame *frame, int *got_frame_ptr, AVPacket *avpkt)
Definition: sipr.c:523
SiprModeParam::bits_per_frame
uint16_t bits_per_frame
Definition: sipr.c:50
offset
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf offset
Definition: writing_filters.txt:86
ff_pow_0_5
const float ff_pow_0_5[]
Definition: sipr.c:136
lsf_codebooks
static const float *const lsf_codebooks[]
Definition: siprdata.h:209
M_PI
#define M_PI
Definition: mathematics.h:67
ff_tilt_compensation
void ff_tilt_compensation(float *mem, float tilt, float *samples, int size)
Apply tilt compensation filter, 1 - tilt * z-1.
Definition: acelp_filters.c:138
pitch_sharpening
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:180
sipr.h
AVFrame::nb_samples
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:420
SiprParameters::vq_indexes
int vq_indexes[5]
Definition: sipr.h:57
SiprContext
Definition: sipr.h:64
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:255
ff_sipr_init_16k
void ff_sipr_init_16k(SiprContext *ctx)
Definition: sipr16k.c:271
modes
static const SiprModeParam modes[MODE_COUNT]
Definition: sipr.c:70
decode_frame
static void decode_frame(SiprContext *ctx, SiprParameters *params, float *out_data)
Definition: sipr.c:365
xf
#define xf(width, name, var, range_min, range_max, subs,...)
Definition: cbs_av1.c:590
SiprModeParam::gc_index_bits
uint8_t gc_index_bits
size in bits of the gain codebook indexes
Definition: sipr.c:67
ff_pow_0_75
const float ff_pow_0_75[10]
Table of pow(0.75,n)
Definition: acelp_vectors.c:93
FFMIN
#define FFMIN(a, b)
Definition: macros.h:49
acelp_filters.h
ff_weighted_vector_sumf
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.
Definition: acelp_vectors.c:182
AVCodec::name
const char * name
Name of the codec implementation.
Definition: codec.h:194
gain_cb
static const float gain_cb[128][2]
Definition: siprdata.h:213
MODE_6k5
@ MODE_6k5
Definition: sipr.h:50
avcodec.h
stride
#define stride
Definition: h264pred_template.c:537
ret
ret
Definition: filter_design.txt:187
pred
static const float pred[4]
Definition: siprdata.h:259
AVCodecContext::block_align
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs.
Definition: avcodec.h:1083
lsp.h
ff_celp_lp_zero_synthesis_filterf
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:200
AMRFixed::n
int n
Definition: acelp_vectors.h:54
SiprModeParam::pitch_delay_bits
uint8_t pitch_delay_bits[5]
size in bits of the adaptive-codebook index for every subframe
Definition: sipr.c:63
AVCodecContext
main external API structure.
Definition: avcodec.h:445
channel_layout.h
mode
mode
Definition: ebur128.h:83
SiprModeParam
Definition: sipr.c:48
av_channel_layout_uninit
void av_channel_layout_uninit(AVChannelLayout *channel_layout)
Free any allocated data in the channel layout and reset the channel count to 0.
Definition: channel_layout.c:432
L_SUBFR_16k
#define L_SUBFR_16k
Definition: sipr.h:31
mean_lsf
static const float mean_lsf[10]
Definition: siprdata.h:27
samples
Filter the word “frame” indicates either a video frame or a group of audio samples
Definition: filter_design.txt:8
SUBFRAME_COUNT_16k
#define SUBFRAME_COUNT_16k
Definition: sipr.h:45
SiprModeParam::subframe_count
uint8_t subframe_count
Definition: sipr.c:51
avpriv_scalarproduct_float_c
float avpriv_scalarproduct_float_c(const float *v1, const float *v2, int len)
Return the scalar product of two vectors.
Definition: float_dsp.c:124
M_LN10
#define M_LN10
Definition: mathematics.h:49
AV_CHANNEL_LAYOUT_MONO
#define AV_CHANNEL_LAYOUT_MONO
Definition: channel_layout.h:378
SiprModeParam::fc_index_bits
uint8_t fc_index_bits[10]
size in bits of the fixed codebook indexes
Definition: sipr.c:66
AVPacket
This structure stores compressed data.
Definition: packet.h:499
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:472
sipr_decoder_init
static av_cold int sipr_decoder_init(AVCodecContext *avctx)
Definition: sipr.c:481
SiprParameters::gp_index
int gp_index[5]
adaptive-codebook gain indexes
Definition: sipr.h:59
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:27
MODE_COUNT
@ MODE_COUNT
Definition: cinepakenc.c:77
AVERROR_INVALIDDATA
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:61
pulses
static const int8_t pulses[4]
Number of non-zero pulses in the MP-MLQ excitation.
Definition: g723_1.h:260
ff_acelp_interpolatef
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:80
AV_SAMPLE_FMT_FLT
@ AV_SAMPLE_FMT_FLT
float
Definition: samplefmt.h:60
SiprModeParam::ma_predictor_bits
uint8_t ma_predictor_bits
size in bits of the switched MA predictor
Definition: sipr.c:57
ff_set_min_dist_lsf
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:55
acelp_pitch_delay.h