FFmpeg  4.2.3
af_dynaudnorm.c
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1 /*
2  * Dynamic Audio Normalizer
3  * Copyright (c) 2015 LoRd_MuldeR <mulder2@gmx.de>. Some rights reserved.
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * FFmpeg is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
22 /**
23  * @file
24  * Dynamic Audio Normalizer
25  */
26 
27 #include <float.h>
28 
29 #include "libavutil/avassert.h"
30 #include "libavutil/opt.h"
31 
32 #define FF_BUFQUEUE_SIZE 302
34 
35 #include "audio.h"
36 #include "avfilter.h"
37 #include "filters.h"
38 #include "internal.h"
39 
40 typedef struct cqueue {
41  double *elements;
42  int size;
44  int first;
45 } cqueue;
46 
48  const AVClass *class;
49 
50  struct FFBufQueue queue;
51 
52  int frame_len;
58 
59  double peak_value;
61  double target_rms;
66  double *fade_factors[2];
67  double *weights;
68 
69  int channels;
70  int delay;
71  int eof;
72  int64_t pts;
73 
77 
80 
81 #define OFFSET(x) offsetof(DynamicAudioNormalizerContext, x)
82 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
83 
84 static const AVOption dynaudnorm_options[] = {
85  { "f", "set the frame length in msec", OFFSET(frame_len_msec), AV_OPT_TYPE_INT, {.i64 = 500}, 10, 8000, FLAGS },
86  { "g", "set the filter size", OFFSET(filter_size), AV_OPT_TYPE_INT, {.i64 = 31}, 3, 301, FLAGS },
87  { "p", "set the peak value", OFFSET(peak_value), AV_OPT_TYPE_DOUBLE, {.dbl = 0.95}, 0.0, 1.0, FLAGS },
88  { "m", "set the max amplification", OFFSET(max_amplification), AV_OPT_TYPE_DOUBLE, {.dbl = 10.0}, 1.0, 100.0, FLAGS },
89  { "r", "set the target RMS", OFFSET(target_rms), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 1.0, FLAGS },
90  { "n", "set channel coupling", OFFSET(channels_coupled), AV_OPT_TYPE_BOOL, {.i64 = 1}, 0, 1, FLAGS },
91  { "c", "set DC correction", OFFSET(dc_correction), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, FLAGS },
92  { "b", "set alternative boundary mode", OFFSET(alt_boundary_mode), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, FLAGS },
93  { "s", "set the compress factor", OFFSET(compress_factor), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 30.0, FLAGS },
94  { NULL }
95 };
96 
97 AVFILTER_DEFINE_CLASS(dynaudnorm);
98 
100 {
102 
103  if (!(s->filter_size & 1)) {
104  av_log(ctx, AV_LOG_ERROR, "filter size %d is invalid. Must be an odd value.\n", s->filter_size);
105  return AVERROR(EINVAL);
106  }
107 
108  return 0;
109 }
110 
112 {
115  static const enum AVSampleFormat sample_fmts[] = {
118  };
119  int ret;
120 
121  layouts = ff_all_channel_counts();
122  if (!layouts)
123  return AVERROR(ENOMEM);
124  ret = ff_set_common_channel_layouts(ctx, layouts);
125  if (ret < 0)
126  return ret;
127 
128  formats = ff_make_format_list(sample_fmts);
129  if (!formats)
130  return AVERROR(ENOMEM);
131  ret = ff_set_common_formats(ctx, formats);
132  if (ret < 0)
133  return ret;
134 
135  formats = ff_all_samplerates();
136  if (!formats)
137  return AVERROR(ENOMEM);
138  return ff_set_common_samplerates(ctx, formats);
139 }
140 
141 static inline int frame_size(int sample_rate, int frame_len_msec)
142 {
143  const int frame_size = lrint((double)sample_rate * (frame_len_msec / 1000.0));
144  return frame_size + (frame_size % 2);
145 }
146 
147 static void precalculate_fade_factors(double *fade_factors[2], int frame_len)
148 {
149  const double step_size = 1.0 / frame_len;
150  int pos;
151 
152  for (pos = 0; pos < frame_len; pos++) {
153  fade_factors[0][pos] = 1.0 - (step_size * (pos + 1.0));
154  fade_factors[1][pos] = 1.0 - fade_factors[0][pos];
155  }
156 }
157 
159 {
160  cqueue *q;
161 
162  q = av_malloc(sizeof(cqueue));
163  if (!q)
164  return NULL;
165 
166  q->size = size;
167  q->nb_elements = 0;
168  q->first = 0;
169 
170  q->elements = av_malloc_array(size, sizeof(double));
171  if (!q->elements) {
172  av_free(q);
173  return NULL;
174  }
175 
176  return q;
177 }
178 
179 static void cqueue_free(cqueue *q)
180 {
181  if (q)
182  av_free(q->elements);
183  av_free(q);
184 }
185 
186 static int cqueue_size(cqueue *q)
187 {
188  return q->nb_elements;
189 }
190 
191 static int cqueue_empty(cqueue *q)
192 {
193  return !q->nb_elements;
194 }
195 
196 static int cqueue_enqueue(cqueue *q, double element)
197 {
198  int i;
199 
200  av_assert2(q->nb_elements != q->size);
201 
202  i = (q->first + q->nb_elements) % q->size;
203  q->elements[i] = element;
204  q->nb_elements++;
205 
206  return 0;
207 }
208 
209 static double cqueue_peek(cqueue *q, int index)
210 {
211  av_assert2(index < q->nb_elements);
212  return q->elements[(q->first + index) % q->size];
213 }
214 
215 static int cqueue_dequeue(cqueue *q, double *element)
216 {
218 
219  *element = q->elements[q->first];
220  q->first = (q->first + 1) % q->size;
221  q->nb_elements--;
222 
223  return 0;
224 }
225 
226 static int cqueue_pop(cqueue *q)
227 {
229 
230  q->first = (q->first + 1) % q->size;
231  q->nb_elements--;
232 
233  return 0;
234 }
235 
237 {
238  double total_weight = 0.0;
239  const double sigma = (((s->filter_size / 2.0) - 1.0) / 3.0) + (1.0 / 3.0);
240  double adjust;
241  int i;
242 
243  // Pre-compute constants
244  const int offset = s->filter_size / 2;
245  const double c1 = 1.0 / (sigma * sqrt(2.0 * M_PI));
246  const double c2 = 2.0 * sigma * sigma;
247 
248  // Compute weights
249  for (i = 0; i < s->filter_size; i++) {
250  const int x = i - offset;
251 
252  s->weights[i] = c1 * exp(-x * x / c2);
253  total_weight += s->weights[i];
254  }
255 
256  // Adjust weights
257  adjust = 1.0 / total_weight;
258  for (i = 0; i < s->filter_size; i++) {
259  s->weights[i] *= adjust;
260  }
261 }
262 
264 {
266  int c;
267 
271  av_freep(&s->fade_factors[0]);
272  av_freep(&s->fade_factors[1]);
273 
274  for (c = 0; c < s->channels; c++) {
275  if (s->gain_history_original)
277  if (s->gain_history_minimum)
279  if (s->gain_history_smoothed)
281  }
282 
286 
288  s->is_enabled = NULL;
289 
290  av_freep(&s->weights);
291 
293 }
294 
295 static int config_input(AVFilterLink *inlink)
296 {
297  AVFilterContext *ctx = inlink->dst;
299  int c;
300 
301  uninit(ctx);
302 
304  av_log(ctx, AV_LOG_DEBUG, "frame len %d\n", s->frame_len);
305 
306  s->fade_factors[0] = av_malloc_array(s->frame_len, sizeof(*s->fade_factors[0]));
307  s->fade_factors[1] = av_malloc_array(s->frame_len, sizeof(*s->fade_factors[1]));
308 
310  s->dc_correction_value = av_calloc(inlink->channels, sizeof(*s->dc_correction_value));
311  s->compress_threshold = av_calloc(inlink->channels, sizeof(*s->compress_threshold));
313  s->gain_history_minimum = av_calloc(inlink->channels, sizeof(*s->gain_history_minimum));
315  s->weights = av_malloc_array(s->filter_size, sizeof(*s->weights));
318  !s->compress_threshold || !s->fade_factors[0] || !s->fade_factors[1] ||
320  !s->gain_history_smoothed || !s->is_enabled || !s->weights)
321  return AVERROR(ENOMEM);
322 
323  for (c = 0; c < inlink->channels; c++) {
324  s->prev_amplification_factor[c] = 1.0;
325 
329 
330  if (!s->gain_history_original[c] || !s->gain_history_minimum[c] ||
331  !s->gain_history_smoothed[c])
332  return AVERROR(ENOMEM);
333  }
334 
337 
338  s->channels = inlink->channels;
339  s->delay = s->filter_size;
340 
341  return 0;
342 }
343 
344 static inline double fade(double prev, double next, int pos,
345  double *fade_factors[2])
346 {
347  return fade_factors[0][pos] * prev + fade_factors[1][pos] * next;
348 }
349 
350 static inline double pow_2(const double value)
351 {
352  return value * value;
353 }
354 
355 static inline double bound(const double threshold, const double val)
356 {
357  const double CONST = 0.8862269254527580136490837416705725913987747280611935; //sqrt(PI) / 2.0
358  return erf(CONST * (val / threshold)) * threshold;
359 }
360 
362 {
363  double max = DBL_EPSILON;
364  int c, i;
365 
366  if (channel == -1) {
367  for (c = 0; c < frame->channels; c++) {
368  double *data_ptr = (double *)frame->extended_data[c];
369 
370  for (i = 0; i < frame->nb_samples; i++)
371  max = FFMAX(max, fabs(data_ptr[i]));
372  }
373  } else {
374  double *data_ptr = (double *)frame->extended_data[channel];
375 
376  for (i = 0; i < frame->nb_samples; i++)
377  max = FFMAX(max, fabs(data_ptr[i]));
378  }
379 
380  return max;
381 }
382 
384 {
385  double rms_value = 0.0;
386  int c, i;
387 
388  if (channel == -1) {
389  for (c = 0; c < frame->channels; c++) {
390  const double *data_ptr = (double *)frame->extended_data[c];
391 
392  for (i = 0; i < frame->nb_samples; i++) {
393  rms_value += pow_2(data_ptr[i]);
394  }
395  }
396 
397  rms_value /= frame->nb_samples * frame->channels;
398  } else {
399  const double *data_ptr = (double *)frame->extended_data[channel];
400  for (i = 0; i < frame->nb_samples; i++) {
401  rms_value += pow_2(data_ptr[i]);
402  }
403 
404  rms_value /= frame->nb_samples;
405  }
406 
407  return FFMAX(sqrt(rms_value), DBL_EPSILON);
408 }
409 
411  int channel)
412 {
413  const double maximum_gain = s->peak_value / find_peak_magnitude(frame, channel);
414  const double rms_gain = s->target_rms > DBL_EPSILON ? (s->target_rms / compute_frame_rms(frame, channel)) : DBL_MAX;
415  return bound(s->max_amplification, FFMIN(maximum_gain, rms_gain));
416 }
417 
418 static double minimum_filter(cqueue *q)
419 {
420  double min = DBL_MAX;
421  int i;
422 
423  for (i = 0; i < cqueue_size(q); i++) {
424  min = FFMIN(min, cqueue_peek(q, i));
425  }
426 
427  return min;
428 }
429 
431 {
432  double result = 0.0;
433  int i;
434 
435  for (i = 0; i < cqueue_size(q); i++) {
436  result += cqueue_peek(q, i) * s->weights[i];
437  }
438 
439  return result;
440 }
441 
443  double current_gain_factor)
444 {
445  if (cqueue_empty(s->gain_history_original[channel]) ||
446  cqueue_empty(s->gain_history_minimum[channel])) {
447  const int pre_fill_size = s->filter_size / 2;
448  const double initial_value = s->alt_boundary_mode ? current_gain_factor : 1.0;
449 
450  s->prev_amplification_factor[channel] = initial_value;
451 
452  while (cqueue_size(s->gain_history_original[channel]) < pre_fill_size) {
453  cqueue_enqueue(s->gain_history_original[channel], initial_value);
454  }
455  }
456 
457  cqueue_enqueue(s->gain_history_original[channel], current_gain_factor);
458 
459  while (cqueue_size(s->gain_history_original[channel]) >= s->filter_size) {
460  double minimum;
462 
463  if (cqueue_empty(s->gain_history_minimum[channel])) {
464  const int pre_fill_size = s->filter_size / 2;
465  double initial_value = s->alt_boundary_mode ? cqueue_peek(s->gain_history_original[channel], 0) : 1.0;
466  int input = pre_fill_size;
467 
468  while (cqueue_size(s->gain_history_minimum[channel]) < pre_fill_size) {
469  input++;
470  initial_value = FFMIN(initial_value, cqueue_peek(s->gain_history_original[channel], input));
471  cqueue_enqueue(s->gain_history_minimum[channel], initial_value);
472  }
473  }
474 
475  minimum = minimum_filter(s->gain_history_original[channel]);
476 
477  cqueue_enqueue(s->gain_history_minimum[channel], minimum);
478 
479  cqueue_pop(s->gain_history_original[channel]);
480  }
481 
482  while (cqueue_size(s->gain_history_minimum[channel]) >= s->filter_size) {
483  double smoothed;
485  smoothed = gaussian_filter(s, s->gain_history_minimum[channel]);
486 
487  cqueue_enqueue(s->gain_history_smoothed[channel], smoothed);
488 
489  cqueue_pop(s->gain_history_minimum[channel]);
490  }
491 }
492 
493 static inline double update_value(double new, double old, double aggressiveness)
494 {
495  av_assert0((aggressiveness >= 0.0) && (aggressiveness <= 1.0));
496  return aggressiveness * new + (1.0 - aggressiveness) * old;
497 }
498 
500 {
501  const double diff = 1.0 / frame->nb_samples;
502  int is_first_frame = cqueue_empty(s->gain_history_original[0]);
503  int c, i;
504 
505  for (c = 0; c < s->channels; c++) {
506  double *dst_ptr = (double *)frame->extended_data[c];
507  double current_average_value = 0.0;
508  double prev_value;
509 
510  for (i = 0; i < frame->nb_samples; i++)
511  current_average_value += dst_ptr[i] * diff;
512 
513  prev_value = is_first_frame ? current_average_value : s->dc_correction_value[c];
514  s->dc_correction_value[c] = is_first_frame ? current_average_value : update_value(current_average_value, s->dc_correction_value[c], 0.1);
515 
516  for (i = 0; i < frame->nb_samples; i++) {
517  dst_ptr[i] -= fade(prev_value, s->dc_correction_value[c], i, s->fade_factors);
518  }
519  }
520 }
521 
522 static double setup_compress_thresh(double threshold)
523 {
524  if ((threshold > DBL_EPSILON) && (threshold < (1.0 - DBL_EPSILON))) {
525  double current_threshold = threshold;
526  double step_size = 1.0;
527 
528  while (step_size > DBL_EPSILON) {
529  while ((llrint((current_threshold + step_size) * (UINT64_C(1) << 63)) >
530  llrint(current_threshold * (UINT64_C(1) << 63))) &&
531  (bound(current_threshold + step_size, 1.0) <= threshold)) {
532  current_threshold += step_size;
533  }
534 
535  step_size /= 2.0;
536  }
537 
538  return current_threshold;
539  } else {
540  return threshold;
541  }
542 }
543 
545  AVFrame *frame, int channel)
546 {
547  double variance = 0.0;
548  int i, c;
549 
550  if (channel == -1) {
551  for (c = 0; c < s->channels; c++) {
552  const double *data_ptr = (double *)frame->extended_data[c];
553 
554  for (i = 0; i < frame->nb_samples; i++) {
555  variance += pow_2(data_ptr[i]); // Assume that MEAN is *zero*
556  }
557  }
558  variance /= (s->channels * frame->nb_samples) - 1;
559  } else {
560  const double *data_ptr = (double *)frame->extended_data[channel];
561 
562  for (i = 0; i < frame->nb_samples; i++) {
563  variance += pow_2(data_ptr[i]); // Assume that MEAN is *zero*
564  }
565  variance /= frame->nb_samples - 1;
566  }
567 
568  return FFMAX(sqrt(variance), DBL_EPSILON);
569 }
570 
572 {
573  int is_first_frame = cqueue_empty(s->gain_history_original[0]);
574  int c, i;
575 
576  if (s->channels_coupled) {
577  const double standard_deviation = compute_frame_std_dev(s, frame, -1);
578  const double current_threshold = FFMIN(1.0, s->compress_factor * standard_deviation);
579 
580  const double prev_value = is_first_frame ? current_threshold : s->compress_threshold[0];
581  double prev_actual_thresh, curr_actual_thresh;
582  s->compress_threshold[0] = is_first_frame ? current_threshold : update_value(current_threshold, s->compress_threshold[0], (1.0/3.0));
583 
584  prev_actual_thresh = setup_compress_thresh(prev_value);
585  curr_actual_thresh = setup_compress_thresh(s->compress_threshold[0]);
586 
587  for (c = 0; c < s->channels; c++) {
588  double *const dst_ptr = (double *)frame->extended_data[c];
589  for (i = 0; i < frame->nb_samples; i++) {
590  const double localThresh = fade(prev_actual_thresh, curr_actual_thresh, i, s->fade_factors);
591  dst_ptr[i] = copysign(bound(localThresh, fabs(dst_ptr[i])), dst_ptr[i]);
592  }
593  }
594  } else {
595  for (c = 0; c < s->channels; c++) {
596  const double standard_deviation = compute_frame_std_dev(s, frame, c);
597  const double current_threshold = setup_compress_thresh(FFMIN(1.0, s->compress_factor * standard_deviation));
598 
599  const double prev_value = is_first_frame ? current_threshold : s->compress_threshold[c];
600  double prev_actual_thresh, curr_actual_thresh;
601  double *dst_ptr;
602  s->compress_threshold[c] = is_first_frame ? current_threshold : update_value(current_threshold, s->compress_threshold[c], 1.0/3.0);
603 
604  prev_actual_thresh = setup_compress_thresh(prev_value);
605  curr_actual_thresh = setup_compress_thresh(s->compress_threshold[c]);
606 
607  dst_ptr = (double *)frame->extended_data[c];
608  for (i = 0; i < frame->nb_samples; i++) {
609  const double localThresh = fade(prev_actual_thresh, curr_actual_thresh, i, s->fade_factors);
610  dst_ptr[i] = copysign(bound(localThresh, fabs(dst_ptr[i])), dst_ptr[i]);
611  }
612  }
613  }
614 }
615 
617 {
618  if (s->dc_correction) {
619  perform_dc_correction(s, frame);
620  }
621 
622  if (s->compress_factor > DBL_EPSILON) {
623  perform_compression(s, frame);
624  }
625 
626  if (s->channels_coupled) {
627  const double current_gain_factor = get_max_local_gain(s, frame, -1);
628  int c;
629 
630  for (c = 0; c < s->channels; c++)
631  update_gain_history(s, c, current_gain_factor);
632  } else {
633  int c;
634 
635  for (c = 0; c < s->channels; c++)
636  update_gain_history(s, c, get_max_local_gain(s, frame, c));
637  }
638 }
639 
641 {
642  int c, i;
643 
644  for (c = 0; c < s->channels; c++) {
645  double *dst_ptr = (double *)frame->extended_data[c];
646  double current_amplification_factor;
647 
648  cqueue_dequeue(s->gain_history_smoothed[c], &current_amplification_factor);
649 
650  for (i = 0; i < frame->nb_samples && enabled; i++) {
651  const double amplification_factor = fade(s->prev_amplification_factor[c],
652  current_amplification_factor, i,
653  s->fade_factors);
654 
655  dst_ptr[i] *= amplification_factor;
656 
657  if (fabs(dst_ptr[i]) > s->peak_value)
658  dst_ptr[i] = copysign(s->peak_value, dst_ptr[i]);
659  }
660 
661  s->prev_amplification_factor[c] = current_amplification_factor;
662  }
663 }
664 
665 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
666 {
667  AVFilterContext *ctx = inlink->dst;
669  AVFilterLink *outlink = inlink->dst->outputs[0];
670  int ret = 1;
671 
672  if (!cqueue_empty(s->gain_history_smoothed[0])) {
673  double is_enabled;
675 
676  cqueue_dequeue(s->is_enabled, &is_enabled);
677 
678  amplify_frame(s, out, is_enabled > 0.);
679  ret = ff_filter_frame(outlink, out);
680  }
681 
684  analyze_frame(s, in);
685  ff_bufqueue_add(ctx, &s->queue, in);
686 
687  return ret;
688 }
689 
691  AVFilterLink *outlink)
692 {
693  AVFrame *out = ff_get_audio_buffer(outlink, s->frame_len);
694  int c, i;
695 
696  if (!out)
697  return AVERROR(ENOMEM);
698 
699  for (c = 0; c < s->channels; c++) {
700  double *dst_ptr = (double *)out->extended_data[c];
701 
702  for (i = 0; i < out->nb_samples; i++) {
703  dst_ptr[i] = s->alt_boundary_mode ? DBL_EPSILON : ((s->target_rms > DBL_EPSILON) ? FFMIN(s->peak_value, s->target_rms) : s->peak_value);
704  if (s->dc_correction) {
705  dst_ptr[i] *= ((i % 2) == 1) ? -1 : 1;
706  dst_ptr[i] += s->dc_correction_value[c];
707  }
708  }
709  }
710 
711  s->delay--;
712  return filter_frame(inlink, out);
713 }
714 
715 static int flush(AVFilterLink *outlink)
716 {
717  AVFilterContext *ctx = outlink->src;
719  int ret = 0;
720 
721  if (!cqueue_empty(s->gain_history_smoothed[0])) {
722  ret = flush_buffer(s, ctx->inputs[0], outlink);
723  } else if (s->queue.available) {
725 
726  s->pts = out->pts;
727  ret = ff_filter_frame(outlink, out);
728  s->delay = s->queue.available;
729  }
730 
731  return ret;
732 }
733 
735 {
736  AVFilterLink *inlink = ctx->inputs[0];
737  AVFilterLink *outlink = ctx->outputs[0];
739  AVFrame *in = NULL;
740  int ret = 0, status;
741  int64_t pts;
742 
743  FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
744 
745  if (!s->eof) {
746  ret = ff_inlink_consume_samples(inlink, s->frame_len, s->frame_len, &in);
747  if (ret < 0)
748  return ret;
749  if (ret > 0) {
750  ret = filter_frame(inlink, in);
751  if (ret <= 0)
752  return ret;
753  }
754 
755  if (ff_inlink_queued_samples(inlink) >= s->frame_len) {
756  ff_filter_set_ready(ctx, 10);
757  return 0;
758  }
759  }
760 
761  if (!s->eof && ff_inlink_acknowledge_status(inlink, &status, &pts)) {
762  if (status == AVERROR_EOF)
763  s->eof = 1;
764  }
765 
766  if (s->eof && s->delay > 0)
767  return flush(outlink);
768 
769  if (s->eof && s->delay <= 0) {
770  ff_outlink_set_status(outlink, AVERROR_EOF, s->pts);
771  return 0;
772  }
773 
774  if (!s->eof)
775  FF_FILTER_FORWARD_WANTED(outlink, inlink);
776 
777  return FFERROR_NOT_READY;
778 }
779 
781  {
782  .name = "default",
783  .type = AVMEDIA_TYPE_AUDIO,
784  .config_props = config_input,
785  },
786  { NULL }
787 };
788 
790  {
791  .name = "default",
792  .type = AVMEDIA_TYPE_AUDIO,
793  },
794  { NULL }
795 };
796 
798  .name = "dynaudnorm",
799  .description = NULL_IF_CONFIG_SMALL("Dynamic Audio Normalizer."),
800  .query_formats = query_formats,
801  .priv_size = sizeof(DynamicAudioNormalizerContext),
802  .init = init,
803  .uninit = uninit,
804  .activate = activate,
805  .inputs = avfilter_af_dynaudnorm_inputs,
806  .outputs = avfilter_af_dynaudnorm_outputs,
807  .priv_class = &dynaudnorm_class,
809 };
static AVFrame * ff_bufqueue_get(struct FFBufQueue *queue)
Get the first buffer from the queue and remove it.
Definition: bufferqueue.h:98
static const AVFilterPad avfilter_af_dynaudnorm_inputs[]
#define FLAGS
Definition: af_dynaudnorm.c:82
static double bound(const double threshold, const double val)
#define NULL
Definition: coverity.c:32
int ff_set_common_channel_layouts(AVFilterContext *ctx, AVFilterChannelLayouts *layouts)
A helper for query_formats() which sets all links to the same list of channel layouts/sample rates...
Definition: formats.c:549
const char const char void * val
Definition: avisynth_c.h:863
static double compute_frame_rms(AVFrame *frame, int channel)
static void amplify_frame(DynamicAudioNormalizerContext *s, AVFrame *frame, int enabled)
This structure describes decoded (raw) audio or video data.
Definition: frame.h:295
AVOption.
Definition: opt.h:246
#define CONST(name, help, val, unit)
Definition: vf_bwdif.c:373
static int cqueue_empty(cqueue *q)
static const AVFilterPad avfilter_af_dynaudnorm_outputs[]
static double pow_2(const double value)
static double erf(double z)
erf function Algorithm taken from the Boost project, source: http://www.boost.org/doc/libs/1_46_1/boo...
Definition: libm.h:121
Main libavfilter public API header.
static int cqueue_size(cqueue *q)
int first
Definition: af_dynaudnorm.c:44
double, planar
Definition: samplefmt.h:70
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
static double get_max_local_gain(DynamicAudioNormalizerContext *s, AVFrame *frame, int channel)
#define FFERROR_NOT_READY
Filters implementation helper functions.
Definition: filters.h:34
static void analyze_frame(DynamicAudioNormalizerContext *s, AVFrame *frame)
static void precalculate_fade_factors(double *fade_factors[2], int frame_len)
int is_disabled
the enabled state from the last expression evaluation
Definition: avfilter.h:385
static void ff_outlink_set_status(AVFilterLink *link, int status, int64_t pts)
Set the status field of a link from the source filter.
Definition: filters.h:189
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
Definition: mem.c:244
static int config_input(AVFilterLink *inlink)
AVFilterFormats * ff_make_format_list(const int *fmts)
Create a list of supported formats.
Definition: formats.c:283
Structure holding the queue.
Definition: bufferqueue.h:49
const char * name
Pad name.
Definition: internal.h:60
AVFilterLink ** inputs
array of pointers to input links
Definition: avfilter.h:346
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1080
#define av_cold
Definition: attributes.h:82
#define av_malloc(s)
#define av_assert2(cond)
assert() equivalent, that does lie in speed critical code.
Definition: avassert.h:64
AVOptions.
static double gaussian_filter(DynamicAudioNormalizerContext *s, cqueue *q)
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:388
double * elements
Definition: af_dynaudnorm.c:41
static AVFrame * frame
static const uint64_t c1
Definition: murmur3.c:49
#define AVERROR_EOF
End of file.
Definition: error.h:55
#define max(a, b)
Definition: cuda_runtime.h:33
static av_cold void uninit(AVFilterContext *ctx)
static void cqueue_free(cqueue *q)
#define av_log(a,...)
#define FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink)
Forward the status on an output link to an input link.
Definition: filters.h:199
A filter pad used for either input or output.
Definition: internal.h:54
int ff_inlink_acknowledge_status(AVFilterLink *link, int *rstatus, int64_t *rpts)
Test and acknowledge the change of status on the link.
Definition: avfilter.c:1436
static int flush_buffer(DynamicAudioNormalizerContext *s, AVFilterLink *inlink, AVFilterLink *outlink)
static int query_formats(AVFilterContext *ctx)
static double cqueue_peek(cqueue *q, int index)
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:259
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
int ff_set_common_formats(AVFilterContext *ctx, AVFilterFormats *formats)
A helper for query_formats() which sets all links to the same list of formats.
Definition: formats.c:568
static void init_gaussian_filter(DynamicAudioNormalizerContext *s)
AVFILTER_DEFINE_CLASS(dynaudnorm)
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
Definition: audio.c:86
#define AVERROR(e)
Definition: error.h:43
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:186
void * priv
private data for use by the filter
Definition: avfilter.h:353
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:197
simple assert() macros that are a bit more flexible than ISO C assert().
#define OFFSET(x)
Definition: af_dynaudnorm.c:81
static const uint8_t offset[127][2]
Definition: vf_spp.c:92
#define FFMAX(a, b)
Definition: common.h:94
int8_t exp
Definition: eval.c:72
AVFrame * queue[FF_BUFQUEUE_SIZE]
Definition: bufferqueue.h:50
#define FF_FILTER_FORWARD_WANTED(outlink, inlink)
Forward the frame_wanted_out flag from an output link to an input link.
Definition: filters.h:254
int channels
number of audio channels, only used for audio.
Definition: frame.h:601
#define FFMIN(a, b)
Definition: common.h:96
int ff_inlink_queued_samples(AVFilterLink *link)
Definition: avfilter.c:1461
static void perform_dc_correction(DynamicAudioNormalizerContext *s, AVFrame *frame)
static int flush(AVFilterLink *outlink)
AVFormatContext * ctx
Definition: movenc.c:48
#define s(width, name)
Definition: cbs_vp9.c:257
int size
Definition: af_dynaudnorm.c:42
static void ff_bufqueue_discard_all(struct FFBufQueue *queue)
Unref and remove all buffers from the queue.
Definition: bufferqueue.h:111
static const AVFilterPad inputs[]
Definition: af_acontrast.c:193
static const AVFilterPad outputs[]
Definition: af_acontrast.c:203
A list of supported channel layouts.
Definition: formats.h:85
sample_rate
int nb_elements
Definition: af_dynaudnorm.c:43
AVFilter ff_af_dynaudnorm
static void update_gain_history(DynamicAudioNormalizerContext *s, int channel, double current_gain_factor)
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
unsigned short available
number of available buffers
Definition: bufferqueue.h:52
int ff_inlink_consume_samples(AVFilterLink *link, unsigned min, unsigned max, AVFrame **rframe)
Take samples from the link&#39;s FIFO and update the link&#39;s stats.
Definition: avfilter.c:1500
static double compute_frame_std_dev(DynamicAudioNormalizerContext *s, AVFrame *frame, int channel)
#define llrint(x)
Definition: libm.h:394
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) #define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac) { } void ff_audio_convert_free(AudioConvert **ac) { if(! *ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);} AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map) { AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;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);return NULL;} return ac;} 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;} else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->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);return ac;} int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in) { int use_generic=1;int len=in->nb_samples;int p;if(ac->dc) { av_log(ac->avr, AV_LOG_TRACE, "%d samples - audio_convert: %s to %s (dithered)\", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
static av_cold int init(AVFilterContext *ctx)
Definition: af_dynaudnorm.c:99
double value
Definition: eval.c:98
Describe the class of an AVClass context structure.
Definition: log.h:67
Filter definition.
Definition: avfilter.h:144
int index
Definition: gxfenc.c:89
const char * name
Filter name.
Definition: avfilter.h:148
static av_always_inline double copysign(double x, double y)
Definition: libm.h:68
static double setup_compress_thresh(double threshold)
#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:133
AVFilterLink ** outputs
array of pointers to output links
Definition: avfilter.h:350
enum MovChannelLayoutTag * layouts
Definition: mov_chan.c:434
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:395
static int64_t pts
int av_frame_make_writable(AVFrame *frame)
Ensure that the frame data is writable, avoiding data copy if possible.
Definition: frame.c:611
#define flags(name, subs,...)
Definition: cbs_av1.c:564
static double find_peak_magnitude(AVFrame *frame, int channel)
static int cqueue_pop(cqueue *q)
static double c[64]
static double minimum_filter(cqueue *q)
channel
Use these values when setting the channel map with ebur128_set_channel().
Definition: ebur128.h:39
void ff_filter_set_ready(AVFilterContext *filter, unsigned priority)
Mark a filter ready and schedule it for activation.
Definition: avfilter.c:193
static const uint64_t c2
Definition: murmur3.c:50
static int cqueue_enqueue(cqueue *q, double element)
static double fade(double prev, double next, int pos, double *fade_factors[2])
static av_always_inline int diff(const uint32_t a, const uint32_t b)
#define av_free(p)
static double update_value(double new, double old, double aggressiveness)
static int activate(AVFilterContext *ctx)
A list of supported formats for one end of a filter link.
Definition: formats.h:64
#define lrint
Definition: tablegen.h:53
An instance of a filter.
Definition: avfilter.h:338
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:701
FILE * out
Definition: movenc.c:54
#define av_freep(p)
#define M_PI
Definition: mathematics.h:52
static void ff_bufqueue_add(void *log, struct FFBufQueue *queue, AVFrame *buf)
Add a buffer to the queue.
Definition: bufferqueue.h:71
#define av_malloc_array(a, b)
formats
Definition: signature.h:48
static cqueue * cqueue_create(int size)
static const AVOption dynaudnorm_options[]
Definition: af_dynaudnorm.c:84
internal API functions
AVFilterChannelLayouts * ff_all_channel_counts(void)
Construct an AVFilterChannelLayouts coding for any channel layout, with known or unknown disposition...
Definition: formats.c:410
static int cqueue_dequeue(cqueue *q, double *element)
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:342
float min
static int frame_size(int sample_rate, int frame_len_msec)
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:361
for(j=16;j >0;--j)
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:556
static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame)