FFmpeg
af_afftdn.c
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1 /*
2  * Copyright (c) 2018 The FFmpeg Project
3  *
4  * This file is part of FFmpeg.
5  *
6  * FFmpeg is free software; you can redistribute it and/or
7  * modify it under the terms of the GNU Lesser General Public
8  * License as published by the Free Software Foundation; either
9  * version 2.1 of the License, or (at your option) any later version.
10  *
11  * FFmpeg is distributed in the hope that it will be useful,
12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14  * Lesser General Public License for more details.
15  *
16  * You should have received a copy of the GNU Lesser General Public
17  * License along with FFmpeg; if not, write to the Free Software
18  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19  */
20 
21 #include <float.h>
22 
23 #include "libavutil/avstring.h"
25 #include "libavutil/opt.h"
26 #include "libavutil/tx.h"
27 #include "avfilter.h"
28 #include "audio.h"
29 #include "formats.h"
30 #include "filters.h"
31 
32 #define C (M_LN10 * 0.1)
33 #define SOLVE_SIZE (5)
34 #define NB_PROFILE_BANDS (15)
35 
41 };
42 
43 enum OutModes {
48 };
49 
56 };
57 
58 enum NoiseType {
64 };
65 
66 typedef struct DeNoiseChannel {
70  double *amt;
71  double *band_amt;
72  double *band_excit;
73  double *gain;
74  double *smoothed_gain;
75  double *prior;
77  double *clean_data;
78  double *noisy_data;
79  double *out_samples;
80  double *spread_function;
81  double *abs_var;
82  double *rel_var;
83  double *min_abs_var;
84  float *fft_in;
88 
93 
96  double noise_floor;
100  double max_gain;
101  double max_var;
102  double gain_scale;
104 
105 typedef struct AudioFFTDeNoiseContext {
106  const AVClass *class;
107 
109  float noise_floor;
117  float ratio;
121 
122  int channels;
126  float sample_rate;
134 
136 
137  int *bin2band;
138  double *window;
139  double *band_alpha;
140  double *band_beta;
141 
143 
145 
147  double floor;
148  double sample_floor;
149 
157 
158 #define OFFSET(x) offsetof(AudioFFTDeNoiseContext, x)
159 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
160 #define AFR AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
161 
162 static const AVOption afftdn_options[] = {
163  { "noise_reduction", "set the noise reduction",OFFSET(noise_reduction), AV_OPT_TYPE_FLOAT,{.dbl = 12}, .01, 97, AFR },
164  { "nr", "set the noise reduction", OFFSET(noise_reduction), AV_OPT_TYPE_FLOAT, {.dbl = 12}, .01, 97, AFR },
165  { "noise_floor", "set the noise floor",OFFSET(noise_floor), AV_OPT_TYPE_FLOAT, {.dbl =-50}, -80,-20, AFR },
166  { "nf", "set the noise floor", OFFSET(noise_floor), AV_OPT_TYPE_FLOAT, {.dbl =-50}, -80,-20, AFR },
167  { "noise_type", "set the noise type", OFFSET(noise_type), AV_OPT_TYPE_INT, {.i64 = WHITE_NOISE}, WHITE_NOISE, NB_NOISE-1, AF, "type" },
168  { "nt", "set the noise type", OFFSET(noise_type), AV_OPT_TYPE_INT, {.i64 = WHITE_NOISE}, WHITE_NOISE, NB_NOISE-1, AF, "type" },
169  { "white", "white noise", 0, AV_OPT_TYPE_CONST, {.i64 = WHITE_NOISE}, 0, 0, AF, "type" },
170  { "w", "white noise", 0, AV_OPT_TYPE_CONST, {.i64 = WHITE_NOISE}, 0, 0, AF, "type" },
171  { "vinyl", "vinyl noise", 0, AV_OPT_TYPE_CONST, {.i64 = VINYL_NOISE}, 0, 0, AF, "type" },
172  { "v", "vinyl noise", 0, AV_OPT_TYPE_CONST, {.i64 = VINYL_NOISE}, 0, 0, AF, "type" },
173  { "shellac", "shellac noise", 0, AV_OPT_TYPE_CONST, {.i64 = SHELLAC_NOISE}, 0, 0, AF, "type" },
174  { "s", "shellac noise", 0, AV_OPT_TYPE_CONST, {.i64 = SHELLAC_NOISE}, 0, 0, AF, "type" },
175  { "custom", "custom noise", 0, AV_OPT_TYPE_CONST, {.i64 = CUSTOM_NOISE}, 0, 0, AF, "type" },
176  { "c", "custom noise", 0, AV_OPT_TYPE_CONST, {.i64 = CUSTOM_NOISE}, 0, 0, AF, "type" },
177  { "band_noise", "set the custom bands noise", OFFSET(band_noise_str), AV_OPT_TYPE_STRING, {.str = 0}, 0, 0, AF },
178  { "bn", "set the custom bands noise", OFFSET(band_noise_str), AV_OPT_TYPE_STRING, {.str = 0}, 0, 0, AF },
179  { "residual_floor", "set the residual floor",OFFSET(residual_floor), AV_OPT_TYPE_FLOAT, {.dbl =-38}, -80,-20, AFR },
180  { "rf", "set the residual floor", OFFSET(residual_floor), AV_OPT_TYPE_FLOAT, {.dbl =-38}, -80,-20, AFR },
181  { "track_noise", "track noise", OFFSET(track_noise), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, AFR },
182  { "tn", "track noise", OFFSET(track_noise), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, AFR },
183  { "track_residual", "track residual", OFFSET(track_residual), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, AFR },
184  { "tr", "track residual", OFFSET(track_residual), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, AFR },
185  { "output_mode", "set output mode", OFFSET(output_mode), AV_OPT_TYPE_INT, {.i64 = OUT_MODE}, 0, NB_MODES-1, AFR, "mode" },
186  { "om", "set output mode", OFFSET(output_mode), AV_OPT_TYPE_INT, {.i64 = OUT_MODE}, 0, NB_MODES-1, AFR, "mode" },
187  { "input", "input", 0, AV_OPT_TYPE_CONST, {.i64 = IN_MODE}, 0, 0, AFR, "mode" },
188  { "i", "input", 0, AV_OPT_TYPE_CONST, {.i64 = IN_MODE}, 0, 0, AFR, "mode" },
189  { "output", "output", 0, AV_OPT_TYPE_CONST, {.i64 = OUT_MODE}, 0, 0, AFR, "mode" },
190  { "o", "output", 0, AV_OPT_TYPE_CONST, {.i64 = OUT_MODE}, 0, 0, AFR, "mode" },
191  { "noise", "noise", 0, AV_OPT_TYPE_CONST, {.i64 = NOISE_MODE}, 0, 0, AFR, "mode" },
192  { "n", "noise", 0, AV_OPT_TYPE_CONST, {.i64 = NOISE_MODE}, 0, 0, AFR, "mode" },
193  { "adaptivity", "set adaptivity factor",OFFSET(ratio), AV_OPT_TYPE_FLOAT, {.dbl = 0.5}, 0, 1, AFR },
194  { "ad", "set adaptivity factor",OFFSET(ratio), AV_OPT_TYPE_FLOAT, {.dbl = 0.5}, 0, 1, AFR },
195  { "floor_offset", "set noise floor offset factor",OFFSET(floor_offset), AV_OPT_TYPE_FLOAT, {.dbl = 1.0}, -2, 2, AFR },
196  { "fo", "set noise floor offset factor",OFFSET(floor_offset), AV_OPT_TYPE_FLOAT, {.dbl = 1.0}, -2, 2, AFR },
197  { "noise_link", "set the noise floor link",OFFSET(noise_floor_link),AV_OPT_TYPE_INT,{.i64 = MIN_LINK}, 0, NB_LINK-1, AFR, "link" },
198  { "nl", "set the noise floor link", OFFSET(noise_floor_link),AV_OPT_TYPE_INT,{.i64 = MIN_LINK}, 0, NB_LINK-1, AFR, "link" },
199  { "none", "none", 0, AV_OPT_TYPE_CONST, {.i64 = NONE_LINK}, 0, 0, AFR, "link" },
200  { "min", "min", 0, AV_OPT_TYPE_CONST, {.i64 = MIN_LINK}, 0, 0, AFR, "link" },
201  { "max", "max", 0, AV_OPT_TYPE_CONST, {.i64 = MAX_LINK}, 0, 0, AFR, "link" },
202  { "average", "average", 0, AV_OPT_TYPE_CONST, {.i64 = AVERAGE_LINK}, 0, 0, AFR, "link" },
203  { "band_multiplier", "set band multiplier",OFFSET(band_multiplier), AV_OPT_TYPE_FLOAT,{.dbl = 1.25}, 0.2,5, AF },
204  { "bm", "set band multiplier", OFFSET(band_multiplier), AV_OPT_TYPE_FLOAT,{.dbl = 1.25}, 0.2,5, AF },
205  { "sample_noise", "set sample noise mode",OFFSET(sample_noise_mode),AV_OPT_TYPE_INT,{.i64 = SAMPLE_NONE}, 0, NB_SAMPLEMODES-1, AFR, "sample" },
206  { "sn", "set sample noise mode",OFFSET(sample_noise_mode),AV_OPT_TYPE_INT,{.i64 = SAMPLE_NONE}, 0, NB_SAMPLEMODES-1, AFR, "sample" },
207  { "none", "none", 0, AV_OPT_TYPE_CONST, {.i64 = SAMPLE_NONE}, 0, 0, AFR, "sample" },
208  { "start", "start", 0, AV_OPT_TYPE_CONST, {.i64 = SAMPLE_START}, 0, 0, AFR, "sample" },
209  { "begin", "start", 0, AV_OPT_TYPE_CONST, {.i64 = SAMPLE_START}, 0, 0, AFR, "sample" },
210  { "stop", "stop", 0, AV_OPT_TYPE_CONST, {.i64 = SAMPLE_STOP}, 0, 0, AFR, "sample" },
211  { "end", "stop", 0, AV_OPT_TYPE_CONST, {.i64 = SAMPLE_STOP}, 0, 0, AFR, "sample" },
212  { "gain_smooth", "set gain smooth radius",OFFSET(gain_smooth), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 50, AFR },
213  { "gs", "set gain smooth radius",OFFSET(gain_smooth), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 50, AFR },
214  { NULL }
215 };
216 
217 AVFILTER_DEFINE_CLASS(afftdn);
218 
220  int band, double a,
221  double b, double c)
222 {
223  double d1, d2, d3;
224 
225  d1 = a / s->band_centre[band];
226  d1 = 10.0 * log(1.0 + d1 * d1) / M_LN10;
227  d2 = b / s->band_centre[band];
228  d2 = 10.0 * log(1.0 + d2 * d2) / M_LN10;
229  d3 = s->band_centre[band] / c;
230  d3 = 10.0 * log(1.0 + d3 * d3) / M_LN10;
231 
232  return -d1 + d2 - d3;
233 }
234 
235 static void factor(double *array, int size)
236 {
237  for (int i = 0; i < size - 1; i++) {
238  for (int j = i + 1; j < size; j++) {
239  double d = array[j + i * size] / array[i + i * size];
240 
241  array[j + i * size] = d;
242  for (int k = i + 1; k < size; k++) {
243  array[j + k * size] -= d * array[i + k * size];
244  }
245  }
246  }
247 }
248 
249 static void solve(double *matrix, double *vector, int size)
250 {
251  for (int i = 0; i < size - 1; i++) {
252  for (int j = i + 1; j < size; j++) {
253  double d = matrix[j + i * size];
254  vector[j] -= d * vector[i];
255  }
256  }
257 
258  vector[size - 1] /= matrix[size * size - 1];
259 
260  for (int i = size - 2; i >= 0; i--) {
261  double d = vector[i];
262  for (int j = i + 1; j < size; j++)
263  d -= matrix[i + j * size] * vector[j];
264  vector[i] = d / matrix[i + i * size];
265  }
266 }
267 
269  DeNoiseChannel *dnch,
270  int band)
271 {
272  double product, sum, f;
273  int i = 0;
274 
275  if (band < NB_PROFILE_BANDS)
276  return dnch->band_noise[band];
277 
278  for (int j = 0; j < SOLVE_SIZE; j++) {
279  sum = 0.0;
280  for (int k = 0; k < NB_PROFILE_BANDS; k++)
281  sum += s->matrix_b[i++] * dnch->band_noise[k];
282  s->vector_b[j] = sum;
283  }
284 
285  solve(s->matrix_a, s->vector_b, SOLVE_SIZE);
286  f = (0.5 * s->sample_rate) / s->band_centre[NB_PROFILE_BANDS-1];
287  f = 15.0 + log(f / 1.5) / log(1.5);
288  sum = 0.0;
289  product = 1.0;
290  for (int j = 0; j < SOLVE_SIZE; j++) {
291  sum += product * s->vector_b[j];
292  product *= f;
293  }
294 
295  return sum;
296 }
297 
298 static double limit_gain(double a, double b)
299 {
300  if (a > 1.0)
301  return (b * a - 1.0) / (b + a - 2.0);
302  if (a < 1.0)
303  return (b * a - 2.0 * a + 1.0) / (b - a);
304  return 1.0;
305 }
306 
307 static void spectral_flatness(AudioFFTDeNoiseContext *s, const double *const spectral,
308  double floor, int len, double *rnum, double *rden)
309 {
310  double num = 0., den = 0.;
311  int size = 0;
312 
313  for (int n = 0; n < len; n++) {
314  const double v = spectral[n];
315  if (v > floor) {
316  num += log(v);
317  den += v;
318  size++;
319  }
320  }
321 
322  size = FFMAX(size, 1);
323 
324  num /= size;
325  den /= size;
326 
327  num = exp(num);
328 
329  *rnum = num;
330  *rden = den;
331 }
332 
333 static void set_parameters(AudioFFTDeNoiseContext *s, DeNoiseChannel *dnch, int update_var, int update_auto_var);
334 
335 static double floor_offset(const double *S, int size, double mean)
336 {
337  double offset = 0.0;
338 
339  for (int n = 0; n < size; n++) {
340  const double p = S[n] - mean;
341 
342  offset = fmax(offset, fabs(p));
343  }
344 
345  return offset / mean;
346 }
347 
350  AVComplexFloat *fft_data,
351  double *prior, double *prior_band_excit, int track_noise)
352 {
353  AVFilterLink *outlink = ctx->outputs[0];
354  const double *abs_var = dnch->abs_var;
355  const double ratio = outlink->frame_count_out ? s->ratio : 1.0;
356  const double rratio = 1. - ratio;
357  const int *bin2band = s->bin2band;
358  double *noisy_data = dnch->noisy_data;
359  double *band_excit = dnch->band_excit;
360  double *band_amt = dnch->band_amt;
361  double *smoothed_gain = dnch->smoothed_gain;
362  double *gain = dnch->gain;
363 
364  for (int i = 0; i < s->bin_count; i++) {
365  double sqr_new_gain, new_gain, power, mag, mag_abs_var, new_mag_abs_var;
366 
367  noisy_data[i] = mag = hypot(fft_data[i].re, fft_data[i].im);
368  power = mag * mag;
369  mag_abs_var = power / abs_var[i];
370  new_mag_abs_var = ratio * prior[i] + rratio * fmax(mag_abs_var - 1.0, 0.0);
371  new_gain = new_mag_abs_var / (1.0 + new_mag_abs_var);
372  sqr_new_gain = new_gain * new_gain;
373  prior[i] = mag_abs_var * sqr_new_gain;
374  dnch->clean_data[i] = power * sqr_new_gain;
375  gain[i] = new_gain;
376  }
377 
378  if (track_noise) {
379  double flatness, num, den;
380 
381  spectral_flatness(s, noisy_data, s->floor, s->bin_count, &num, &den);
382 
383  flatness = num / den;
384  if (flatness > 0.8) {
385  const double offset = s->floor_offset * floor_offset(noisy_data, s->bin_count, den);
386  const double new_floor = av_clipd(10.0 * log10(den) - 100.0 + offset, -90., -20.);
387 
388  dnch->noise_floor = 0.1 * new_floor + dnch->noise_floor * 0.9;
389  set_parameters(s, dnch, 1, 1);
390  }
391  }
392 
393  for (int i = 0; i < s->number_of_bands; i++) {
394  band_excit[i] = 0.0;
395  band_amt[i] = 0.0;
396  }
397 
398  for (int i = 0; i < s->bin_count; i++)
399  band_excit[bin2band[i]] += dnch->clean_data[i];
400 
401  for (int i = 0; i < s->number_of_bands; i++) {
402  band_excit[i] = fmax(band_excit[i],
403  s->band_alpha[i] * band_excit[i] +
404  s->band_beta[i] * prior_band_excit[i]);
405  prior_band_excit[i] = band_excit[i];
406  }
407 
408  for (int j = 0, i = 0; j < s->number_of_bands; j++) {
409  for (int k = 0; k < s->number_of_bands; k++) {
410  band_amt[j] += dnch->spread_function[i++] * band_excit[k];
411  }
412  }
413 
414  for (int i = 0; i < s->bin_count; i++)
415  dnch->amt[i] = band_amt[bin2band[i]];
416 
417  for (int i = 0; i < s->bin_count; i++) {
418  if (dnch->amt[i] > abs_var[i]) {
419  gain[i] = 1.0;
420  } else if (dnch->amt[i] > dnch->min_abs_var[i]) {
421  const double limit = sqrt(abs_var[i] / dnch->amt[i]);
422 
423  gain[i] = limit_gain(gain[i], limit);
424  } else {
425  gain[i] = limit_gain(gain[i], dnch->max_gain);
426  }
427  }
428 
429  memcpy(smoothed_gain, gain, s->bin_count * sizeof(*smoothed_gain));
430  if (s->gain_smooth > 0) {
431  const int r = s->gain_smooth;
432 
433  for (int i = r; i < s->bin_count - r; i++) {
434  const double gc = gain[i];
435  double num = 0., den = 0.;
436 
437  for (int j = -r; j <= r; j++) {
438  const double g = gain[i + j];
439  const double d = 1. - fabs(g - gc);
440 
441  num += g * d;
442  den += d;
443  }
444 
445  smoothed_gain[i] = num / den;
446  }
447  }
448 
449  for (int i = 0; i < s->bin_count; i++) {
450  const double new_gain = smoothed_gain[i];
451 
452  fft_data[i].re *= new_gain;
453  fft_data[i].im *= new_gain;
454  }
455 }
456 
457 static double freq2bark(double x)
458 {
459  double d = x / 7500.0;
460 
461  return 13.0 * atan(7.6E-4 * x) + 3.5 * atan(d * d);
462 }
463 
465 {
466  if (band == -1)
467  return lrint(s->band_centre[0] / 1.5);
468 
469  return s->band_centre[band];
470 }
471 
472 static int get_band_edge(AudioFFTDeNoiseContext *s, int band)
473 {
474  int i;
475 
476  if (band == NB_PROFILE_BANDS) {
477  i = lrint(s->band_centre[NB_PROFILE_BANDS - 1] * 1.224745);
478  } else {
479  i = lrint(s->band_centre[band] / 1.224745);
480  }
481 
482  return FFMIN(i, s->sample_rate / 2);
483 }
484 
486  DeNoiseChannel *dnch)
487 {
488  double band_noise, d2, d3, d4, d5;
489  int i = 0, j = 0, k = 0;
490 
491  d5 = 0.0;
492  band_noise = process_get_band_noise(s, dnch, 0);
493  for (int m = j; m < s->bin_count; m++) {
494  if (m == j) {
495  i = j;
496  d5 = band_noise;
497  if (k >= NB_PROFILE_BANDS) {
498  j = s->bin_count;
499  } else {
500  j = s->fft_length * get_band_centre(s, k) / s->sample_rate;
501  }
502  d2 = j - i;
503  band_noise = process_get_band_noise(s, dnch, k);
504  k++;
505  }
506  d3 = (j - m) / d2;
507  d4 = (m - i) / d2;
508  dnch->rel_var[m] = exp((d5 * d3 + band_noise * d4) * C);
509  }
510 
511  for (i = 0; i < NB_PROFILE_BANDS; i++)
512  dnch->noise_band_auto_var[i] = dnch->max_var * exp((process_get_band_noise(s, dnch, i) - 2.0) * C);
513 }
514 
516 {
517  DeNoiseChannel *dnch = &s->dnch[ch];
518  char *custom_noise_str, *p, *arg, *saveptr = NULL;
519  double band_noise[NB_PROFILE_BANDS] = { 0.f };
520  int ret;
521 
522  if (!s->band_noise_str)
523  return;
524 
525  custom_noise_str = p = av_strdup(s->band_noise_str);
526  if (!p)
527  return;
528 
529  for (int i = 0; i < NB_PROFILE_BANDS; i++) {
530  float noise;
531 
532  if (!(arg = av_strtok(p, "| ", &saveptr)))
533  break;
534 
535  p = NULL;
536 
537  ret = av_sscanf(arg, "%f", &noise);
538  if (ret != 1) {
539  av_log(s, AV_LOG_ERROR, "Custom band noise must be float.\n");
540  break;
541  }
542 
543  band_noise[i] = av_clipd(noise, -24., 24.);
544  }
545 
546  av_free(custom_noise_str);
547  memcpy(dnch->band_noise, band_noise, sizeof(band_noise));
548 }
549 
550 static void set_parameters(AudioFFTDeNoiseContext *s, DeNoiseChannel *dnch, int update_var, int update_auto_var)
551 {
552  if (dnch->last_noise_floor != dnch->noise_floor)
553  dnch->last_noise_floor = dnch->noise_floor;
554 
555  if (s->track_residual)
557 
558  dnch->max_var = s->floor * exp((100.0 + dnch->last_noise_floor) * C);
559  if (update_auto_var) {
560  for (int i = 0; i < NB_PROFILE_BANDS; i++)
561  dnch->noise_band_auto_var[i] = dnch->max_var * exp((process_get_band_noise(s, dnch, i) - 2.0) * C);
562  }
563 
564  if (s->track_residual) {
565  if (update_var || dnch->last_residual_floor != dnch->residual_floor) {
566  update_var = 1;
567  dnch->last_residual_floor = dnch->residual_floor;
568  dnch->last_noise_reduction = fmax(dnch->last_noise_floor - dnch->last_residual_floor + 100., 0);
569  dnch->max_gain = exp(dnch->last_noise_reduction * (0.5 * C));
570  }
571  } else if (update_var || dnch->noise_reduction != dnch->last_noise_reduction) {
572  update_var = 1;
574  dnch->last_residual_floor = av_clipd(dnch->last_noise_floor - dnch->last_noise_reduction, -80, -20);
575  dnch->max_gain = exp(dnch->last_noise_reduction * (0.5 * C));
576  }
577 
578  dnch->gain_scale = 1.0 / (dnch->max_gain * dnch->max_gain);
579 
580  if (update_var) {
581  set_band_parameters(s, dnch);
582 
583  for (int i = 0; i < s->bin_count; i++) {
584  dnch->abs_var[i] = fmax(dnch->max_var * dnch->rel_var[i], 1.0);
585  dnch->min_abs_var[i] = dnch->gain_scale * dnch->abs_var[i];
586  }
587  }
588 }
589 
590 static void reduce_mean(double *band_noise)
591 {
592  double mean = 0.f;
593 
594  for (int i = 0; i < NB_PROFILE_BANDS; i++)
595  mean += band_noise[i];
597 
598  for (int i = 0; i < NB_PROFILE_BANDS; i++)
599  band_noise[i] -= mean;
600 }
601 
603 {
604  AVFilterContext *ctx = inlink->dst;
605  AudioFFTDeNoiseContext *s = ctx->priv;
606  double wscale, sar, sum, sdiv;
607  int i, j, k, m, n, ret;
608 
609  s->dnch = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->dnch));
610  if (!s->dnch)
611  return AVERROR(ENOMEM);
612 
613  s->channels = inlink->ch_layout.nb_channels;
614  s->sample_rate = inlink->sample_rate;
615  s->sample_advance = s->sample_rate / 80;
616  s->window_length = 3 * s->sample_advance;
617  s->fft_length2 = 1 << (32 - ff_clz(s->window_length));
618  s->fft_length = s->fft_length2;
619  s->buffer_length = s->fft_length * 2;
620  s->bin_count = s->fft_length2 / 2 + 1;
621 
622  s->band_centre[0] = 80;
623  for (i = 1; i < NB_PROFILE_BANDS; i++) {
624  s->band_centre[i] = lrint(1.5 * s->band_centre[i - 1] + 5.0);
625  if (s->band_centre[i] < 1000) {
626  s->band_centre[i] = 10 * (s->band_centre[i] / 10);
627  } else if (s->band_centre[i] < 5000) {
628  s->band_centre[i] = 50 * ((s->band_centre[i] + 20) / 50);
629  } else if (s->band_centre[i] < 15000) {
630  s->band_centre[i] = 100 * ((s->band_centre[i] + 45) / 100);
631  } else {
632  s->band_centre[i] = 1000 * ((s->band_centre[i] + 495) / 1000);
633  }
634  }
635 
636  for (j = 0; j < SOLVE_SIZE; j++) {
637  for (k = 0; k < SOLVE_SIZE; k++) {
638  s->matrix_a[j + k * SOLVE_SIZE] = 0.0;
639  for (m = 0; m < NB_PROFILE_BANDS; m++)
640  s->matrix_a[j + k * SOLVE_SIZE] += pow(m, j + k);
641  }
642  }
643 
644  factor(s->matrix_a, SOLVE_SIZE);
645 
646  i = 0;
647  for (j = 0; j < SOLVE_SIZE; j++)
648  for (k = 0; k < NB_PROFILE_BANDS; k++)
649  s->matrix_b[i++] = pow(k, j);
650 
651  i = 0;
652  for (j = 0; j < NB_PROFILE_BANDS; j++)
653  for (k = 0; k < SOLVE_SIZE; k++)
654  s->matrix_c[i++] = pow(j, k);
655 
656  s->window = av_calloc(s->window_length, sizeof(*s->window));
657  s->bin2band = av_calloc(s->bin_count, sizeof(*s->bin2band));
658  if (!s->window || !s->bin2band)
659  return AVERROR(ENOMEM);
660 
661  sdiv = s->band_multiplier;
662  for (i = 0; i < s->bin_count; i++)
663  s->bin2band[i] = lrint(sdiv * freq2bark((0.5 * i * s->sample_rate) / s->fft_length2));
664 
665  s->number_of_bands = s->bin2band[s->bin_count - 1] + 1;
666 
667  s->band_alpha = av_calloc(s->number_of_bands, sizeof(*s->band_alpha));
668  s->band_beta = av_calloc(s->number_of_bands, sizeof(*s->band_beta));
669  if (!s->band_alpha || !s->band_beta)
670  return AVERROR(ENOMEM);
671 
672  for (int ch = 0; ch < inlink->ch_layout.nb_channels; ch++) {
673  DeNoiseChannel *dnch = &s->dnch[ch];
674  float scale = 1.f;
675 
676  switch (s->noise_type) {
677  case WHITE_NOISE:
678  for (i = 0; i < NB_PROFILE_BANDS; i++)
679  dnch->band_noise[i] = 0.;
680  break;
681  case VINYL_NOISE:
682  for (i = 0; i < NB_PROFILE_BANDS; i++)
683  dnch->band_noise[i] = get_band_noise(s, i, 50.0, 500.5, 2125.0);
684  break;
685  case SHELLAC_NOISE:
686  for (i = 0; i < NB_PROFILE_BANDS; i++)
687  dnch->band_noise[i] = get_band_noise(s, i, 1.0, 500.0, 1.0E10);
688  break;
689  case CUSTOM_NOISE:
690  read_custom_noise(s, ch);
691  break;
692  default:
693  return AVERROR_BUG;
694  }
695 
696  reduce_mean(dnch->band_noise);
697 
698  dnch->amt = av_calloc(s->bin_count, sizeof(*dnch->amt));
699  dnch->band_amt = av_calloc(s->number_of_bands, sizeof(*dnch->band_amt));
700  dnch->band_excit = av_calloc(s->number_of_bands, sizeof(*dnch->band_excit));
701  dnch->gain = av_calloc(s->bin_count, sizeof(*dnch->gain));
702  dnch->smoothed_gain = av_calloc(s->bin_count, sizeof(*dnch->smoothed_gain));
703  dnch->prior = av_calloc(s->bin_count, sizeof(*dnch->prior));
704  dnch->prior_band_excit = av_calloc(s->number_of_bands, sizeof(*dnch->prior_band_excit));
705  dnch->clean_data = av_calloc(s->bin_count, sizeof(*dnch->clean_data));
706  dnch->noisy_data = av_calloc(s->bin_count, sizeof(*dnch->noisy_data));
707  dnch->out_samples = av_calloc(s->buffer_length, sizeof(*dnch->out_samples));
708  dnch->abs_var = av_calloc(s->bin_count, sizeof(*dnch->abs_var));
709  dnch->rel_var = av_calloc(s->bin_count, sizeof(*dnch->rel_var));
710  dnch->min_abs_var = av_calloc(s->bin_count, sizeof(*dnch->min_abs_var));
711  dnch->fft_in = av_calloc(s->fft_length2, sizeof(*dnch->fft_in));
712  dnch->fft_out = av_calloc(s->fft_length2 + 1, sizeof(*dnch->fft_out));
713  ret = av_tx_init(&dnch->fft, &dnch->tx_fn, AV_TX_FLOAT_RDFT, 0, s->fft_length2, &scale, 0);
714  if (ret < 0)
715  return ret;
716  ret = av_tx_init(&dnch->ifft, &dnch->itx_fn, AV_TX_FLOAT_RDFT, 1, s->fft_length2, &scale, 0);
717  if (ret < 0)
718  return ret;
719  dnch->spread_function = av_calloc(s->number_of_bands * s->number_of_bands,
720  sizeof(*dnch->spread_function));
721 
722  if (!dnch->amt ||
723  !dnch->band_amt ||
724  !dnch->band_excit ||
725  !dnch->gain ||
726  !dnch->smoothed_gain ||
727  !dnch->prior ||
728  !dnch->prior_band_excit ||
729  !dnch->clean_data ||
730  !dnch->noisy_data ||
731  !dnch->out_samples ||
732  !dnch->fft_in ||
733  !dnch->fft_out ||
734  !dnch->abs_var ||
735  !dnch->rel_var ||
736  !dnch->min_abs_var ||
737  !dnch->spread_function ||
738  !dnch->fft ||
739  !dnch->ifft)
740  return AVERROR(ENOMEM);
741  }
742 
743  for (int ch = 0; ch < inlink->ch_layout.nb_channels; ch++) {
744  DeNoiseChannel *dnch = &s->dnch[ch];
745  double *prior_band_excit = dnch->prior_band_excit;
746  double min, max;
747  double p1, p2;
748 
749  p1 = pow(0.1, 2.5 / sdiv);
750  p2 = pow(0.1, 1.0 / sdiv);
751  j = 0;
752  for (m = 0; m < s->number_of_bands; m++) {
753  for (n = 0; n < s->number_of_bands; n++) {
754  if (n < m) {
755  dnch->spread_function[j++] = pow(p2, m - n);
756  } else if (n > m) {
757  dnch->spread_function[j++] = pow(p1, n - m);
758  } else {
759  dnch->spread_function[j++] = 1.0;
760  }
761  }
762  }
763 
764  for (m = 0; m < s->number_of_bands; m++) {
765  dnch->band_excit[m] = 0.0;
766  prior_band_excit[m] = 0.0;
767  }
768 
769  for (m = 0; m < s->bin_count; m++)
770  dnch->band_excit[s->bin2band[m]] += 1.0;
771 
772  j = 0;
773  for (m = 0; m < s->number_of_bands; m++) {
774  for (n = 0; n < s->number_of_bands; n++)
775  prior_band_excit[m] += dnch->spread_function[j++] * dnch->band_excit[n];
776  }
777 
778  min = pow(0.1, 2.5);
779  max = pow(0.1, 1.0);
780  for (int i = 0; i < s->number_of_bands; i++) {
781  if (i < lrint(12.0 * sdiv)) {
782  dnch->band_excit[i] = pow(0.1, 1.45 + 0.1 * i / sdiv);
783  } else {
784  dnch->band_excit[i] = pow(0.1, 2.5 - 0.2 * (i / sdiv - 14.0));
785  }
786  dnch->band_excit[i] = av_clipd(dnch->band_excit[i], min, max);
787  }
788 
789  for (int i = 0; i < s->buffer_length; i++)
790  dnch->out_samples[i] = 0;
791 
792  j = 0;
793  for (int i = 0; i < s->number_of_bands; i++)
794  for (int k = 0; k < s->number_of_bands; k++)
795  dnch->spread_function[j++] *= dnch->band_excit[i] / prior_band_excit[i];
796  }
797 
798  j = 0;
799  sar = s->sample_advance / s->sample_rate;
800  for (int i = 0; i < s->bin_count; i++) {
801  if ((i == s->fft_length2) || (s->bin2band[i] > j)) {
802  double d6 = (i - 1) * s->sample_rate / s->fft_length;
803  double d7 = fmin(0.008 + 2.2 / d6, 0.03);
804  s->band_alpha[j] = exp(-sar / d7);
805  s->band_beta[j] = 1.0 - s->band_alpha[j];
806  j = s->bin2band[i];
807  }
808  }
809 
810  s->winframe = ff_get_audio_buffer(inlink, s->window_length);
811  if (!s->winframe)
812  return AVERROR(ENOMEM);
813 
814  wscale = sqrt(8.0 / (9.0 * s->fft_length));
815  sum = 0.0;
816  for (int i = 0; i < s->window_length; i++) {
817  double d10 = sin(i * M_PI / s->window_length);
818  d10 *= wscale * d10;
819  s->window[i] = d10;
820  sum += d10 * d10;
821  }
822 
823  s->window_weight = 0.5 * sum;
824  s->floor = (1LL << 48) * exp(-23.025558369790467) * s->window_weight;
825  s->sample_floor = s->floor * exp(4.144600506562284);
826 
827  for (int ch = 0; ch < inlink->ch_layout.nb_channels; ch++) {
828  DeNoiseChannel *dnch = &s->dnch[ch];
829 
830  dnch->noise_reduction = s->noise_reduction;
831  dnch->noise_floor = s->noise_floor;
832  dnch->residual_floor = s->residual_floor;
833 
834  set_parameters(s, dnch, 1, 1);
835  }
836 
837  s->noise_band_edge[0] = FFMIN(s->fft_length2, s->fft_length * get_band_edge(s, 0) / s->sample_rate);
838  i = 0;
839  for (int j = 1; j < NB_PROFILE_BANDS + 1; j++) {
840  s->noise_band_edge[j] = FFMIN(s->fft_length2, s->fft_length * get_band_edge(s, j) / s->sample_rate);
841  if (s->noise_band_edge[j] > lrint(1.1 * s->noise_band_edge[j - 1]))
842  i++;
843  s->noise_band_edge[NB_PROFILE_BANDS + 1] = i;
844  }
845  s->noise_band_count = s->noise_band_edge[NB_PROFILE_BANDS + 1];
846 
847  return 0;
848 }
849 
851 {
852  for (int i = 0; i < NB_PROFILE_BANDS; i++) {
853  dnch->noise_band_norm[i] = 0.0;
854  dnch->noise_band_avr[i] = 0.0;
855  dnch->noise_band_avi[i] = 0.0;
856  dnch->noise_band_var[i] = 0.0;
857  }
858 }
859 
861  DeNoiseChannel *dnch,
862  AVFrame *in, int ch)
863 {
864  float *src = (float *)in->extended_data[ch];
865  double mag2, var = 0.0, avr = 0.0, avi = 0.0;
866  int edge, j, k, n, edgemax;
867 
868  for (int i = 0; i < s->window_length; i++)
869  dnch->fft_in[i] = s->window[i] * src[i] * (1LL << 23);
870 
871  for (int i = s->window_length; i < s->fft_length2; i++)
872  dnch->fft_in[i] = 0.0;
873 
874  dnch->tx_fn(dnch->fft, dnch->fft_out, dnch->fft_in, sizeof(float));
875 
876  edge = s->noise_band_edge[0];
877  j = edge;
878  k = 0;
879  n = j;
880  edgemax = fmin(s->fft_length2, s->noise_band_edge[NB_PROFILE_BANDS]);
881  for (int i = j; i <= edgemax; i++) {
882  if ((i == j) && (i < edgemax)) {
883  if (j > edge) {
884  dnch->noise_band_norm[k - 1] += j - edge;
885  dnch->noise_band_avr[k - 1] += avr;
886  dnch->noise_band_avi[k - 1] += avi;
887  dnch->noise_band_var[k - 1] += var;
888  }
889  k++;
890  edge = j;
891  j = s->noise_band_edge[k];
892  if (k == NB_PROFILE_BANDS) {
893  j++;
894  }
895  var = 0.0;
896  avr = 0.0;
897  avi = 0.0;
898  }
899  avr += dnch->fft_out[n].re;
900  avi += dnch->fft_out[n].im;
901  mag2 = dnch->fft_out[n].re * dnch->fft_out[n].re +
902  dnch->fft_out[n].im * dnch->fft_out[n].im;
903 
904  mag2 = fmax(mag2, s->sample_floor);
905 
906  var += mag2;
907  n++;
908  }
909 
910  dnch->noise_band_norm[k - 1] += j - edge;
911  dnch->noise_band_avr[k - 1] += avr;
912  dnch->noise_band_avi[k - 1] += avi;
913  dnch->noise_band_var[k - 1] += var;
914 }
915 
917  DeNoiseChannel *dnch,
918  double *sample_noise)
919 {
920  for (int i = 0; i < s->noise_band_count; i++) {
921  dnch->noise_band_avr[i] /= dnch->noise_band_norm[i];
922  dnch->noise_band_avi[i] /= dnch->noise_band_norm[i];
923  dnch->noise_band_var[i] /= dnch->noise_band_norm[i];
924  dnch->noise_band_var[i] -= dnch->noise_band_avr[i] * dnch->noise_band_avr[i] +
925  dnch->noise_band_avi[i] * dnch->noise_band_avi[i];
926  dnch->noise_band_auto_var[i] = dnch->noise_band_var[i];
927  sample_noise[i] = 10.0 * log10(dnch->noise_band_var[i] / s->floor) - 100.0;
928  }
929  if (s->noise_band_count < NB_PROFILE_BANDS) {
930  for (int i = s->noise_band_count; i < NB_PROFILE_BANDS; i++)
931  sample_noise[i] = sample_noise[i - 1];
932  }
933 }
934 
936  DeNoiseChannel *dnch,
937  double *sample_noise)
938 {
939  double new_band_noise[NB_PROFILE_BANDS];
940  double temp[NB_PROFILE_BANDS];
941  double sum = 0.0;
942 
943  for (int m = 0; m < NB_PROFILE_BANDS; m++)
944  temp[m] = sample_noise[m];
945 
946  for (int m = 0, i = 0; m < SOLVE_SIZE; m++) {
947  sum = 0.0;
948  for (int n = 0; n < NB_PROFILE_BANDS; n++)
949  sum += s->matrix_b[i++] * temp[n];
950  s->vector_b[m] = sum;
951  }
952  solve(s->matrix_a, s->vector_b, SOLVE_SIZE);
953  for (int m = 0, i = 0; m < NB_PROFILE_BANDS; m++) {
954  sum = 0.0;
955  for (int n = 0; n < SOLVE_SIZE; n++)
956  sum += s->matrix_c[i++] * s->vector_b[n];
957  temp[m] = sum;
958  }
959 
960  reduce_mean(temp);
961 
962  av_log(s, AV_LOG_INFO, "bn=");
963  for (int m = 0; m < NB_PROFILE_BANDS; m++) {
964  new_band_noise[m] = temp[m];
965  new_band_noise[m] = av_clipd(new_band_noise[m], -24.0, 24.0);
966  av_log(s, AV_LOG_INFO, "%f ", new_band_noise[m]);
967  }
968  av_log(s, AV_LOG_INFO, "\n");
969  memcpy(dnch->band_noise, new_band_noise, sizeof(new_band_noise));
970 }
971 
972 static int filter_channel(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
973 {
974  AudioFFTDeNoiseContext *s = ctx->priv;
975  AVFrame *in = arg;
976  const int start = (in->ch_layout.nb_channels * jobnr) / nb_jobs;
977  const int end = (in->ch_layout.nb_channels * (jobnr+1)) / nb_jobs;
978  const int window_length = s->window_length;
979  const double *window = s->window;
980 
981  for (int ch = start; ch < end; ch++) {
982  DeNoiseChannel *dnch = &s->dnch[ch];
983  const float *src = (const float *)in->extended_data[ch];
984  double *dst = dnch->out_samples;
985  float *fft_in = dnch->fft_in;
986 
987  for (int m = 0; m < window_length; m++)
988  fft_in[m] = window[m] * src[m] * (1LL << 23);
989 
990  for (int m = window_length; m < s->fft_length2; m++)
991  fft_in[m] = 0;
992 
993  dnch->tx_fn(dnch->fft, dnch->fft_out, fft_in, sizeof(float));
994 
995  process_frame(ctx, s, dnch, dnch->fft_out,
996  dnch->prior,
997  dnch->prior_band_excit,
998  s->track_noise);
999 
1000  dnch->itx_fn(dnch->ifft, fft_in, dnch->fft_out, sizeof(float));
1001 
1002  for (int m = 0; m < window_length; m++)
1003  dst[m] += s->window[m] * fft_in[m] / (1LL << 23);
1004  }
1005 
1006  return 0;
1007 }
1008 
1010 {
1011  AVFilterContext *ctx = inlink->dst;
1012  AVFilterLink *outlink = ctx->outputs[0];
1013  AudioFFTDeNoiseContext *s = ctx->priv;
1014  const int output_mode = ctx->is_disabled ? IN_MODE : s->output_mode;
1015  const int offset = s->window_length - s->sample_advance;
1016  AVFrame *out;
1017 
1018  for (int ch = 0; ch < s->channels; ch++) {
1019  float *src = (float *)s->winframe->extended_data[ch];
1020 
1021  memmove(src, &src[s->sample_advance], offset * sizeof(float));
1022  memcpy(&src[offset], in->extended_data[ch], in->nb_samples * sizeof(float));
1023  memset(&src[offset + in->nb_samples], 0, (s->sample_advance - in->nb_samples) * sizeof(float));
1024  }
1025 
1026  if (s->track_noise) {
1027  double average = 0.0, min = DBL_MAX, max = -DBL_MAX;
1028 
1029  for (int ch = 0; ch < inlink->ch_layout.nb_channels; ch++) {
1030  DeNoiseChannel *dnch = &s->dnch[ch];
1031 
1032  average += dnch->noise_floor;
1033  max = fmax(max, dnch->noise_floor);
1034  min = fmin(min, dnch->noise_floor);
1035  }
1036 
1037  average /= inlink->ch_layout.nb_channels;
1038 
1039  for (int ch = 0; ch < inlink->ch_layout.nb_channels; ch++) {
1040  DeNoiseChannel *dnch = &s->dnch[ch];
1041 
1042  switch (s->noise_floor_link) {
1043  case MIN_LINK: dnch->noise_floor = min; break;
1044  case MAX_LINK: dnch->noise_floor = max; break;
1045  case AVERAGE_LINK: dnch->noise_floor = average; break;
1046  case NONE_LINK:
1047  default:
1048  break;
1049  }
1050 
1051  if (dnch->noise_floor != dnch->last_noise_floor)
1052  set_parameters(s, dnch, 1, 0);
1053  }
1054  }
1055 
1056  if (s->sample_noise_mode == SAMPLE_START) {
1057  for (int ch = 0; ch < inlink->ch_layout.nb_channels; ch++) {
1058  DeNoiseChannel *dnch = &s->dnch[ch];
1059 
1060  init_sample_noise(dnch);
1061  }
1062  s->sample_noise_mode = SAMPLE_NONE;
1063  s->sample_noise = 1;
1064  s->sample_noise_blocks = 0;
1065  }
1066 
1067  if (s->sample_noise) {
1068  for (int ch = 0; ch < inlink->ch_layout.nb_channels; ch++) {
1069  DeNoiseChannel *dnch = &s->dnch[ch];
1070 
1071  sample_noise_block(s, dnch, s->winframe, ch);
1072  }
1073  s->sample_noise_blocks++;
1074  }
1075 
1076  if (s->sample_noise_mode == SAMPLE_STOP) {
1077  for (int ch = 0; ch < inlink->ch_layout.nb_channels; ch++) {
1078  DeNoiseChannel *dnch = &s->dnch[ch];
1079  double sample_noise[NB_PROFILE_BANDS];
1080 
1081  if (s->sample_noise_blocks <= 0)
1082  break;
1083  finish_sample_noise(s, dnch, sample_noise);
1084  set_noise_profile(s, dnch, sample_noise);
1085  set_parameters(s, dnch, 1, 1);
1086  }
1087  s->sample_noise = 0;
1088  s->sample_noise_blocks = 0;
1089  s->sample_noise_mode = SAMPLE_NONE;
1090  }
1091 
1092  ff_filter_execute(ctx, filter_channel, s->winframe, NULL,
1094 
1095  if (av_frame_is_writable(in)) {
1096  out = in;
1097  } else {
1098  out = ff_get_audio_buffer(outlink, in->nb_samples);
1099  if (!out) {
1100  av_frame_free(&in);
1101  return AVERROR(ENOMEM);
1102  }
1103 
1104  out->pts = in->pts;
1105  }
1106 
1107  for (int ch = 0; ch < inlink->ch_layout.nb_channels; ch++) {
1108  DeNoiseChannel *dnch = &s->dnch[ch];
1109  double *src = dnch->out_samples;
1110  const float *orig = (const float *)s->winframe->extended_data[ch];
1111  float *dst = (float *)out->extended_data[ch];
1112 
1113  switch (output_mode) {
1114  case IN_MODE:
1115  for (int m = 0; m < out->nb_samples; m++)
1116  dst[m] = orig[m];
1117  break;
1118  case OUT_MODE:
1119  for (int m = 0; m < out->nb_samples; m++)
1120  dst[m] = src[m];
1121  break;
1122  case NOISE_MODE:
1123  for (int m = 0; m < out->nb_samples; m++)
1124  dst[m] = orig[m] - src[m];
1125  break;
1126  default:
1127  if (in != out)
1128  av_frame_free(&in);
1129  av_frame_free(&out);
1130  return AVERROR_BUG;
1131  }
1132  memmove(src, src + s->sample_advance, (s->window_length - s->sample_advance) * sizeof(*src));
1133  memset(src + (s->window_length - s->sample_advance), 0, s->sample_advance * sizeof(*src));
1134  }
1135 
1136  if (out != in)
1137  av_frame_free(&in);
1138  return ff_filter_frame(outlink, out);
1139 }
1140 
1142 {
1143  AVFilterLink *inlink = ctx->inputs[0];
1144  AVFilterLink *outlink = ctx->outputs[0];
1145  AudioFFTDeNoiseContext *s = ctx->priv;
1146  AVFrame *in = NULL;
1147  int ret;
1148 
1150 
1151  ret = ff_inlink_consume_samples(inlink, s->sample_advance, s->sample_advance, &in);
1152  if (ret < 0)
1153  return ret;
1154  if (ret > 0)
1155  return output_frame(inlink, in);
1156 
1157  if (ff_inlink_queued_samples(inlink) >= s->sample_advance) {
1158  ff_filter_set_ready(ctx, 10);
1159  return 0;
1160  }
1161 
1162  FF_FILTER_FORWARD_STATUS(inlink, outlink);
1163  FF_FILTER_FORWARD_WANTED(outlink, inlink);
1164 
1165  return FFERROR_NOT_READY;
1166 }
1167 
1169 {
1170  AudioFFTDeNoiseContext *s = ctx->priv;
1171 
1172  av_freep(&s->window);
1173  av_freep(&s->bin2band);
1174  av_freep(&s->band_alpha);
1175  av_freep(&s->band_beta);
1176  av_frame_free(&s->winframe);
1177 
1178  if (s->dnch) {
1179  for (int ch = 0; ch < s->channels; ch++) {
1180  DeNoiseChannel *dnch = &s->dnch[ch];
1181  av_freep(&dnch->amt);
1182  av_freep(&dnch->band_amt);
1183  av_freep(&dnch->band_excit);
1184  av_freep(&dnch->gain);
1185  av_freep(&dnch->smoothed_gain);
1186  av_freep(&dnch->prior);
1187  av_freep(&dnch->prior_band_excit);
1188  av_freep(&dnch->clean_data);
1189  av_freep(&dnch->noisy_data);
1190  av_freep(&dnch->out_samples);
1191  av_freep(&dnch->spread_function);
1192  av_freep(&dnch->abs_var);
1193  av_freep(&dnch->rel_var);
1194  av_freep(&dnch->min_abs_var);
1195  av_freep(&dnch->fft_in);
1196  av_freep(&dnch->fft_out);
1197  av_tx_uninit(&dnch->fft);
1198  av_tx_uninit(&dnch->ifft);
1199  }
1200  av_freep(&s->dnch);
1201  }
1202 }
1203 
1204 static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
1205  char *res, int res_len, int flags)
1206 {
1207  AudioFFTDeNoiseContext *s = ctx->priv;
1208  int ret = 0;
1209 
1210  ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
1211  if (ret < 0)
1212  return ret;
1213 
1214  if (!strcmp(cmd, "sample_noise") || !strcmp(cmd, "sn"))
1215  return 0;
1216 
1217  for (int ch = 0; ch < s->channels; ch++) {
1218  DeNoiseChannel *dnch = &s->dnch[ch];
1219 
1220  dnch->noise_reduction = s->noise_reduction;
1221  dnch->noise_floor = s->noise_floor;
1222  dnch->residual_floor = s->residual_floor;
1223 
1224  set_parameters(s, dnch, 1, 1);
1225  }
1226 
1227  return 0;
1228 }
1229 
1230 static const AVFilterPad inputs[] = {
1231  {
1232  .name = "default",
1233  .type = AVMEDIA_TYPE_AUDIO,
1234  .config_props = config_input,
1235  },
1236 };
1237 
1238 static const AVFilterPad outputs[] = {
1239  {
1240  .name = "default",
1241  .type = AVMEDIA_TYPE_AUDIO,
1242  },
1243 };
1244 
1246  .name = "afftdn",
1247  .description = NULL_IF_CONFIG_SMALL("Denoise audio samples using FFT."),
1248  .priv_size = sizeof(AudioFFTDeNoiseContext),
1249  .priv_class = &afftdn_class,
1250  .activate = activate,
1251  .uninit = uninit,
1255  .process_command = process_command,
1258 };
NB_PROFILE_BANDS
#define NB_PROFILE_BANDS
Definition: af_afftdn.c:34
ff_get_audio_buffer
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
Definition: audio.c:100
AV_SAMPLE_FMT_FLTP
@ AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:66
NB_MODES
@ NB_MODES
Definition: af_afftdn.c:47
DeNoiseChannel::noise_floor
double noise_floor
Definition: af_afftdn.c:96
AudioFFTDeNoiseContext::floor
double floor
Definition: af_afftdn.c:147
DeNoiseChannel::clean_data
double * clean_data
Definition: af_afftdn.c:77
DeNoiseChannel::noise_band_auto_var
double noise_band_auto_var[NB_PROFILE_BANDS]
Definition: af_afftdn.c:68
C
#define C
Definition: af_afftdn.c:32
DeNoiseChannel::ifft
AVTXContext * ifft
Definition: af_afftdn.c:86
r
const char * r
Definition: vf_curves.c:116
AVERROR
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later. That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another. Frame references ownership and permissions
opt.h
process_frame
static void process_frame(AVFilterContext *ctx, AudioFFTDeNoiseContext *s, DeNoiseChannel *dnch, AVComplexFloat *fft_data, double *prior, double *prior_band_excit, int track_noise)
Definition: af_afftdn.c:348
out
FILE * out
Definition: movenc.c:54
ff_filter_frame
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:999
activate
static int activate(AVFilterContext *ctx)
Definition: af_afftdn.c:1141
inputs
static const AVFilterPad inputs[]
Definition: af_afftdn.c:1230
FFERROR_NOT_READY
return FFERROR_NOT_READY
Definition: filter_design.txt:204
matrix
Definition: vc1dsp.c:42
AVTXContext
Definition: tx_priv.h:201
AudioFFTDeNoiseContext::window_weight
double window_weight
Definition: af_afftdn.c:146
FILTER_SINGLE_SAMPLEFMT
#define FILTER_SINGLE_SAMPLEFMT(sample_fmt_)
Definition: internal.h:183
inlink
The exact code depends on how similar the blocks are and how related they are to the and needs to apply these operations to the correct inlink or outlink if there are several Macros are available to factor that when no extra processing is inlink
Definition: filter_design.txt:212
ff_clz
#define ff_clz
Definition: intmath.h:142
solve
static void solve(double *matrix, double *vector, int size)
Definition: af_afftdn.c:249
av_frame_free
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:111
im
float im
Definition: fft.c:79
OUT_MODE
@ OUT_MODE
Definition: af_afftdn.c:45
SOLVE_SIZE
#define SOLVE_SIZE
Definition: af_afftdn.c:33
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:325
AVFrame::pts
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:432
sample_noise_block
static void sample_noise_block(AudioFFTDeNoiseContext *s, DeNoiseChannel *dnch, AVFrame *in, int ch)
Definition: af_afftdn.c:860
AVOption
AVOption.
Definition: opt.h:251
b
#define b
Definition: input.c:34
NONE_LINK
@ NONE_LINK
Definition: af_afftdn.c:51
float.h
AVComplexFloat
Definition: tx.h:27
max
#define max(a, b)
Definition: cuda_runtime.h:33
FFMAX
#define FFMAX(a, b)
Definition: macros.h:47
AVFilter::name
const char * name
Filter name.
Definition: avfilter.h:175
DeNoiseChannel::max_var
double max_var
Definition: af_afftdn.c:101
AVChannelLayout::nb_channels
int nb_channels
Number of channels in this layout.
Definition: channel_layout.h:300
AudioFFTDeNoiseContext::fft_length2
int fft_length2
Definition: af_afftdn.c:129
AudioFFTDeNoiseContext::vector_b
double vector_b[SOLVE_SIZE]
Definition: af_afftdn.c:153
DeNoiseChannel::band_amt
double * band_amt
Definition: af_afftdn.c:71
FF_FILTER_FORWARD_STATUS_BACK
#define FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink)
Forward the status on an output link to an input link.
Definition: filters.h:199
AudioFFTDeNoiseContext::noise_floor_link
int noise_floor_link
Definition: af_afftdn.c:116
AudioFFTDeNoiseContext::sample_noise_mode
int sample_noise_mode
Definition: af_afftdn.c:125
av_tx_init
av_cold int av_tx_init(AVTXContext **ctx, av_tx_fn *tx, enum AVTXType type, int inv, int len, const void *scale, uint64_t flags)
Initialize a transform context with the given configuration (i)MDCTs with an odd length are currently...
Definition: tx.c:638
NoiseType
NoiseType
Definition: af_afftdn.c:58
formats.h
S
#define S(s, c, i)
Definition: flacdsp_template.c:46
DeNoiseChannel::noise_band_norm
double noise_band_norm[NB_PROFILE_BANDS]
Definition: af_afftdn.c:89
factor
static void factor(double *array, int size)
Definition: af_afftdn.c:235
DeNoiseChannel::noise_band_avr
double noise_band_avr[NB_PROFILE_BANDS]
Definition: af_afftdn.c:90
config_input
static int config_input(AVFilterLink *inlink)
Definition: af_afftdn.c:602
AudioFFTDeNoiseContext::residual_floor
float residual_floor
Definition: af_afftdn.c:112
AudioFFTDeNoiseContext::buffer_length
int buffer_length
Definition: af_afftdn.c:127
AVComplexFloat::im
float im
Definition: tx.h:28
window
static SDL_Window * window
Definition: ffplay.c:365
freq2bark
static double freq2bark(double x)
Definition: af_afftdn.c:457
AudioFFTDeNoiseContext::number_of_bands
int number_of_bands
Definition: af_afftdn.c:133
DeNoiseChannel::band_noise
double band_noise[NB_PROFILE_BANDS]
Definition: af_afftdn.c:67
AVFrame::ch_layout
AVChannelLayout ch_layout
Channel layout of the audio data.
Definition: frame.h:704
scale
static av_always_inline float scale(float x, float s)
Definition: vf_v360.c:1388
AudioFFTDeNoiseContext::window_length
int window_length
Definition: af_afftdn.c:131
AVFilterPad
A filter pad used for either input or output.
Definition: internal.h:49
lrint
#define lrint
Definition: tablegen.h:53
AudioFFTDeNoiseContext::dnch
DeNoiseChannel * dnch
Definition: af_afftdn.c:142
DeNoiseChannel::spread_function
double * spread_function
Definition: af_afftdn.c:80
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
OFFSET
#define OFFSET(x)
Definition: af_afftdn.c:158
NoiseLinkType
NoiseLinkType
Definition: af_afftdn.c:50
av_tx_fn
void(* av_tx_fn)(AVTXContext *s, void *out, void *in, ptrdiff_t stride)
Function pointer to a function to perform the transform.
Definition: tx.h:111
DeNoiseChannel::last_noise_reduction
double last_noise_reduction
Definition: af_afftdn.c:95
afftdn_options
static const AVOption afftdn_options[]
Definition: af_afftdn.c:162
s
#define s(width, name)
Definition: cbs_vp9.c:256
AudioFFTDeNoiseContext::winframe
AVFrame * winframe
Definition: af_afftdn.c:144
NB_NOISE
@ NB_NOISE
Definition: af_afftdn.c:63
AudioFFTDeNoiseContext::matrix_a
double matrix_a[SOLVE_SIZE *SOLVE_SIZE]
Definition: af_afftdn.c:152
floor
static __device__ float floor(float a)
Definition: cuda_runtime.h:173
g
const char * g
Definition: vf_curves.c:117
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
av_strtok
char * av_strtok(char *s, const char *delim, char **saveptr)
Split the string into several tokens which can be accessed by successive calls to av_strtok().
Definition: avstring.c:189
AudioFFTDeNoiseContext
Definition: af_afftdn.c:105
DeNoiseChannel::gain
double * gain
Definition: af_afftdn.c:73
filters.h
DeNoiseChannel::rel_var
double * rel_var
Definition: af_afftdn.c:82
DeNoiseChannel
Definition: af_afftdn.c:66
DeNoiseChannel::min_abs_var
double * min_abs_var
Definition: af_afftdn.c:83
ctx
AVFormatContext * ctx
Definition: movenc.c:48
DeNoiseChannel::noise_band_sample
double noise_band_sample[NB_PROFILE_BANDS]
Definition: af_afftdn.c:69
AudioFFTDeNoiseContext::sample_noise_blocks
int sample_noise_blocks
Definition: af_afftdn.c:124
FILTER_INPUTS
#define FILTER_INPUTS(array)
Definition: internal.h:190
E
#define E
Definition: avdct.c:32
get_band_edge
static int get_band_edge(AudioFFTDeNoiseContext *s, int band)
Definition: af_afftdn.c:472
arg
const char * arg
Definition: jacosubdec.c:67
AF
#define AF
Definition: af_afftdn.c:159
av_sscanf
int av_sscanf(const char *string, const char *format,...)
See libc sscanf manual for more information.
Definition: avsscanf.c:962
AVClass
Describe the class of an AVClass context structure.
Definition: log.h:66
fabs
static __device__ float fabs(float a)
Definition: cuda_runtime.h:182
ff_inlink_consume_samples
int ff_inlink_consume_samples(AVFilterLink *link, unsigned min, unsigned max, AVFrame **rframe)
Take samples from the link's FIFO and update the link's stats.
Definition: avfilter.c:1413
NULL
#define NULL
Definition: coverity.c:32
DeNoiseChannel::last_noise_floor
double last_noise_floor
Definition: af_afftdn.c:97
DeNoiseChannel::max_gain
double max_gain
Definition: af_afftdn.c:100
DeNoiseChannel::residual_floor
double residual_floor
Definition: af_afftdn.c:98
AudioFFTDeNoiseContext::matrix_b
double matrix_b[SOLVE_SIZE *NB_PROFILE_BANDS]
Definition: af_afftdn.c:154
SampleNoiseModes
SampleNoiseModes
Definition: af_afftdn.c:36
AudioFFTDeNoiseContext::ratio
float ratio
Definition: af_afftdn.c:117
filter_channel
static int filter_channel(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
Definition: af_afftdn.c:972
AudioFFTDeNoiseContext::noise_type
int noise_type
Definition: af_afftdn.c:110
NB_SAMPLEMODES
@ NB_SAMPLEMODES
Definition: af_afftdn.c:40
set_band_parameters
static void set_band_parameters(AudioFFTDeNoiseContext *s, DeNoiseChannel *dnch)
Definition: af_afftdn.c:485
WHITE_NOISE
@ WHITE_NOISE
Definition: af_afftdn.c:59
exp
int8_t exp
Definition: eval.c:72
c
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
Definition: undefined.txt:32
init_sample_noise
static void init_sample_noise(DeNoiseChannel *dnch)
Definition: af_afftdn.c:850
NOISE_MODE
@ NOISE_MODE
Definition: af_afftdn.c:46
DeNoiseChannel::fft_in
float * fft_in
Definition: af_afftdn.c:84
read_custom_noise
static void read_custom_noise(AudioFFTDeNoiseContext *s, int ch)
Definition: af_afftdn.c:515
AudioFFTDeNoiseContext::sample_advance
int sample_advance
Definition: af_afftdn.c:132
OutModes
OutModes
Definition: af_afftdn.c:43
DeNoiseChannel::last_residual_floor
double last_residual_floor
Definition: af_afftdn.c:99
f
f
Definition: af_crystalizer.c:122
AudioFFTDeNoiseContext::output_mode
int output_mode
Definition: af_afftdn.c:115
NULL_IF_CONFIG_SMALL
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:116
NB_LINK
@ NB_LINK
Definition: af_afftdn.c:55
outputs
static const AVFilterPad outputs[]
Definition: af_afftdn.c:1238
for
for(k=2;k<=8;++k)
Definition: h264pred_template.c:425
output_frame
static int output_frame(AVFilterLink *inlink, AVFrame *in)
Definition: af_afftdn.c:1009
fmin
double fmin(double, double)
hypot
static av_const double hypot(double x, double y)
Definition: libm.h:366
size
int size
Definition: twinvq_data.h:10344
AVComplexFloat::re
float re
Definition: tx.h:28
AudioFFTDeNoiseContext::matrix_c
double matrix_c[SOLVE_SIZE *NB_PROFILE_BANDS]
Definition: af_afftdn.c:155
av_frame_is_writable
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
Definition: frame.c:523
AudioFFTDeNoiseContext::track_noise
int track_noise
Definition: af_afftdn.c:113
AudioFFTDeNoiseContext::noise_reduction
float noise_reduction
Definition: af_afftdn.c:108
ff_filter_process_command
int ff_filter_process_command(AVFilterContext *ctx, const char *cmd, const char *arg, char *res, int res_len, int flags)
Generic processing of user supplied commands that are set in the same way as the filter options.
Definition: avfilter.c:863
AudioFFTDeNoiseContext::window
double * window
Definition: af_afftdn.c:138
a
The reader does not expect b to be semantically here and if the code is changed by maybe adding a a division or other the signedness will almost certainly be mistaken To avoid this confusion a new type was SUINT is the C unsigned type but it holds a signed int to use the same example SUINT a
Definition: undefined.txt:41
finish_sample_noise
static void finish_sample_noise(AudioFFTDeNoiseContext *s, DeNoiseChannel *dnch, double *sample_noise)
Definition: af_afftdn.c:916
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_FILTER_FORWARD_WANTED
FF_FILTER_FORWARD_WANTED(outlink, inlink)
VINYL_NOISE
@ VINYL_NOISE
Definition: af_afftdn.c:60
AudioFFTDeNoiseContext::band_alpha
double * band_alpha
Definition: af_afftdn.c:139
limit_gain
static double limit_gain(double a, double b)
Definition: af_afftdn.c:298
MIN_LINK
@ MIN_LINK
Definition: af_afftdn.c:52
AudioFFTDeNoiseContext::channels
int channels
Definition: af_afftdn.c:122
M_PI
#define M_PI
Definition: mathematics.h:52
av_tx_uninit
av_cold void av_tx_uninit(AVTXContext **ctx)
Frees a context and sets *ctx to NULL, does nothing when *ctx == NULL.
Definition: tx.c:251
AV_LOG_INFO
#define AV_LOG_INFO
Standard information.
Definition: log.h:191
uninit
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_afftdn.c:1168
AV_OPT_TYPE_FLOAT
@ AV_OPT_TYPE_FLOAT
Definition: opt.h:228
DeNoiseChannel::out_samples
double * out_samples
Definition: af_afftdn.c:79
AVFrame::nb_samples
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:405
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:271
DeNoiseChannel::itx_fn
av_tx_fn itx_fn
Definition: af_afftdn.c:87
DeNoiseChannel::fft
AVTXContext * fft
Definition: af_afftdn.c:86
AudioFFTDeNoiseContext::gain_smooth
int gain_smooth
Definition: af_afftdn.c:118
SAMPLE_START
@ SAMPLE_START
Definition: af_afftdn.c:38
DeNoiseChannel::noise_band_avi
double noise_band_avi[NB_PROFILE_BANDS]
Definition: af_afftdn.c:91
AVFrame::extended_data
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:386
DeNoiseChannel::prior_band_excit
double * prior_band_excit
Definition: af_afftdn.c:76
SAMPLE_NONE
@ SAMPLE_NONE
Definition: af_afftdn.c:37
ff_filter_get_nb_threads
int ff_filter_get_nb_threads(AVFilterContext *ctx)
Get number of threads for current filter instance.
Definition: avfilter.c:783
process_command
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, char *res, int res_len, int flags)
Definition: af_afftdn.c:1204
SAMPLE_STOP
@ SAMPLE_STOP
Definition: af_afftdn.c:39
DeNoiseChannel::smoothed_gain
double * smoothed_gain
Definition: af_afftdn.c:74
FFMIN
#define FFMIN(a, b)
Definition: macros.h:49
SHELLAC_NOISE
@ SHELLAC_NOISE
Definition: af_afftdn.c:61
DeNoiseChannel::fft_out
AVComplexFloat * fft_out
Definition: af_afftdn.c:85
len
int len
Definition: vorbis_enc_data.h:426
AVFilterPad::name
const char * name
Pad name.
Definition: internal.h:55
ff_inlink_queued_samples
int ff_inlink_queued_samples(AVFilterLink *link)
Definition: avfilter.c:1373
av_calloc
void * av_calloc(size_t nmemb, size_t size)
Definition: mem.c:272
AudioFFTDeNoiseContext::track_residual
int track_residual
Definition: af_afftdn.c:114
AudioFFTDeNoiseContext::noise_floor
float noise_floor
Definition: af_afftdn.c:109
limit
static double limit(double x)
Definition: vf_pseudocolor.c:128
AVFilter
Filter definition.
Definition: avfilter.h:171
DeNoiseChannel::amt
double * amt
Definition: af_afftdn.c:70
AVERAGE_LINK
@ AVERAGE_LINK
Definition: af_afftdn.c:54
array
static int array[MAX_W *MAX_W]
Definition: jpeg2000dwt.c:111
ret
ret
Definition: filter_design.txt:187
MAX_LINK
@ MAX_LINK
Definition: af_afftdn.c:53
get_band_noise
static double get_band_noise(AudioFFTDeNoiseContext *s, int band, double a, double b, double c)
Definition: af_afftdn.c:219
AudioFFTDeNoiseContext::bin2band
int * bin2band
Definition: af_afftdn.c:137
CUSTOM_NOISE
@ CUSTOM_NOISE
Definition: af_afftdn.c:62
IN_MODE
@ IN_MODE
Definition: af_afftdn.c:44
DeNoiseChannel::noisy_data
double * noisy_data
Definition: af_afftdn.c:78
AudioFFTDeNoiseContext::floor_offset
float floor_offset
Definition: af_afftdn.c:120
AV_TX_FLOAT_RDFT
@ AV_TX_FLOAT_RDFT
Real to complex and complex to real DFTs.
Definition: tx.h:88
fmax
double fmax(double, double)
AudioFFTDeNoiseContext::noise_band_count
int noise_band_count
Definition: af_afftdn.c:151
AFR
#define AFR
Definition: af_afftdn.c:160
AudioFFTDeNoiseContext::band_noise_str
char * band_noise_str
Definition: af_afftdn.c:111
DeNoiseChannel::tx_fn
av_tx_fn tx_fn
Definition: af_afftdn.c:87
power
static float power(float r, float g, float b, float max)
Definition: preserve_color.h:45
channel_layout.h
set_parameters
static void set_parameters(AudioFFTDeNoiseContext *s, DeNoiseChannel *dnch, int update_var, int update_auto_var)
Definition: af_afftdn.c:550
process_get_band_noise
static double process_get_band_noise(AudioFFTDeNoiseContext *s, DeNoiseChannel *dnch, int band)
Definition: af_afftdn.c:268
noise
static int noise(AVBSFContext *ctx, AVPacket *pkt)
Definition: noise_bsf.c:120
spectral_flatness
static void spectral_flatness(AudioFFTDeNoiseContext *s, const double *const spectral, double floor, int len, double *rnum, double *rden)
Definition: af_afftdn.c:307
AV_OPT_TYPE_INT
@ AV_OPT_TYPE_INT
Definition: opt.h:225
avfilter.h
DeNoiseChannel::band_excit
double * band_excit
Definition: af_afftdn.c:72
temp
else temp
Definition: vf_mcdeint.c:248
mean
static float mean(const float *input, int size)
Definition: vf_nnedi.c:857
DeNoiseChannel::noise_reduction
double noise_reduction
Definition: af_afftdn.c:94
AudioFFTDeNoiseContext::band_beta
double * band_beta
Definition: af_afftdn.c:140
AVFilterContext
An instance of a filter.
Definition: avfilter.h:408
DeNoiseChannel::noise_band_var
double noise_band_var[NB_PROFILE_BANDS]
Definition: af_afftdn.c:92
AudioFFTDeNoiseContext::band_centre
int band_centre[NB_PROFILE_BANDS]
Definition: af_afftdn.c:135
AVFILTER_FLAG_SLICE_THREADS
#define AVFILTER_FLAG_SLICE_THREADS
The filter supports multithreading by splitting frames into multiple parts and processing them concur...
Definition: avfilter.h:127
av_strdup
char * av_strdup(const char *s)
Duplicate a string.
Definition: mem.c:280
AVFILTER_DEFINE_CLASS
AVFILTER_DEFINE_CLASS(afftdn)
audio.h
M_LN10
#define M_LN10
Definition: mathematics.h:43
DeNoiseChannel::prior
double * prior
Definition: af_afftdn.c:75
av_free
#define av_free(p)
Definition: tableprint_vlc.h:33
FF_FILTER_FORWARD_STATUS
FF_FILTER_FORWARD_STATUS(inlink, outlink)
AV_OPT_TYPE_BOOL
@ AV_OPT_TYPE_BOOL
Definition: opt.h:244
FILTER_OUTPUTS
#define FILTER_OUTPUTS(array)
Definition: internal.h:191
av_freep
#define av_freep(p)
Definition: tableprint_vlc.h:34
src
INIT_CLIP pixel * src
Definition: h264pred_template.c:418
floor_offset
static double floor_offset(const double *S, int size, double mean)
Definition: af_afftdn.c:335
AudioFFTDeNoiseContext::noise_band_edge
int noise_band_edge[NB_PROFILE_BANDS+2]
Definition: af_afftdn.c:150
d
d
Definition: ffmpeg_filter.c:153
AudioFFTDeNoiseContext::bin_count
int bin_count
Definition: af_afftdn.c:130
reduce_mean
static void reduce_mean(double *band_noise)
Definition: af_afftdn.c:590
AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL
#define AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL
Same as AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC, except that the filter will have its filter_frame() c...
Definition: avfilter.h:160
AudioFFTDeNoiseContext::sample_noise
int sample_noise
Definition: af_afftdn.c:123
flags
#define flags(name, subs,...)
Definition: cbs_av1.c:561
AVERROR_BUG
#define AVERROR_BUG
Internal bug, also see AVERROR_BUG2.
Definition: error.h:52
AudioFFTDeNoiseContext::fft_length
int fft_length
Definition: af_afftdn.c:128
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:27
AudioFFTDeNoiseContext::sample_floor
double sample_floor
Definition: af_afftdn.c:148
set_noise_profile
static void set_noise_profile(AudioFFTDeNoiseContext *s, DeNoiseChannel *dnch, double *sample_noise)
Definition: af_afftdn.c:935
DeNoiseChannel::abs_var
double * abs_var
Definition: af_afftdn.c:81
avstring.h
AV_OPT_TYPE_STRING
@ AV_OPT_TYPE_STRING
Definition: opt.h:229
ff_filter_execute
static av_always_inline int ff_filter_execute(AVFilterContext *ctx, avfilter_action_func *func, void *arg, int *ret, int nb_jobs)
Definition: internal.h:142
get_band_centre
static int get_band_centre(AudioFFTDeNoiseContext *s, int band)
Definition: af_afftdn.c:464
AudioFFTDeNoiseContext::sample_rate
float sample_rate
Definition: af_afftdn.c:126
DeNoiseChannel::gain_scale
double gain_scale
Definition: af_afftdn.c:102
AV_OPT_TYPE_CONST
@ AV_OPT_TYPE_CONST
Definition: opt.h:234
ff_af_afftdn
const AVFilter ff_af_afftdn
Definition: af_afftdn.c:1245
AudioFFTDeNoiseContext::band_multiplier
float band_multiplier
Definition: af_afftdn.c:119
av_clipd
av_clipd
Definition: af_crystalizer.c:132
ff_filter_set_ready
void ff_filter_set_ready(AVFilterContext *filter, unsigned priority)
Mark a filter ready and schedule it for activation.
Definition: avfilter.c:191
tx.h
re
float re
Definition: fft.c:79
min
float min
Definition: vorbis_enc_data.h:429