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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
33#include "libavfilter/bufferqueue.h"
34
35#include "audio.h"
36#include "avfilter.h"
37#include "internal.h"
38
39typedef struct cqueue {
40 double *elements;
41 int size;
42 int nb_elements;
43 int first;
44} cqueue;
45
46typedef struct DynamicAudioNormalizerContext {
47 const AVClass *class;
48
49 struct FFBufQueue queue;
50
51 int frame_len;
52 int frame_len_msec;
53 int filter_size;
54 int dc_correction;
55 int channels_coupled;
56 int alt_boundary_mode;
57
58 double peak_value;
59 double max_amplification;
60 double target_rms;
61 double compress_factor;
62 double *prev_amplification_factor;
63 double *dc_correction_value;
64 double *compress_threshold;
65 double *fade_factors[2];
66 double *weights;
67
68 int channels;
69 int delay;
70
71 cqueue **gain_history_original;
72 cqueue **gain_history_minimum;
73 cqueue **gain_history_smoothed;
74} DynamicAudioNormalizerContext;
75
76#define OFFSET(x) offsetof(DynamicAudioNormalizerContext, x)
77#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
78
79static const AVOption dynaudnorm_options[] = {
80 { "f", "set the frame length in msec", OFFSET(frame_len_msec), AV_OPT_TYPE_INT, {.i64 = 500}, 10, 8000, FLAGS },
81 { "g", "set the filter size", OFFSET(filter_size), AV_OPT_TYPE_INT, {.i64 = 31}, 3, 301, FLAGS },
82 { "p", "set the peak value", OFFSET(peak_value), AV_OPT_TYPE_DOUBLE, {.dbl = 0.95}, 0.0, 1.0, FLAGS },
83 { "m", "set the max amplification", OFFSET(max_amplification), AV_OPT_TYPE_DOUBLE, {.dbl = 10.0}, 1.0, 100.0, FLAGS },
84 { "r", "set the target RMS", OFFSET(target_rms), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 1.0, FLAGS },
85 { "n", "set channel coupling", OFFSET(channels_coupled), AV_OPT_TYPE_BOOL, {.i64 = 1}, 0, 1, FLAGS },
86 { "c", "set DC correction", OFFSET(dc_correction), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, FLAGS },
87 { "b", "set alternative boundary mode", OFFSET(alt_boundary_mode), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, FLAGS },
88 { "s", "set the compress factor", OFFSET(compress_factor), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 30.0, FLAGS },
89 { NULL }
90};
91
92AVFILTER_DEFINE_CLASS(dynaudnorm);
93
94static av_cold int init(AVFilterContext *ctx)
95{
96 DynamicAudioNormalizerContext *s = ctx->priv;
97
98 if (!(s->filter_size & 1)) {
99 av_log(ctx, AV_LOG_ERROR, "filter size %d is invalid. Must be an odd value.\n", s->filter_size);
100 return AVERROR(EINVAL);
101 }
102
103 return 0;
104}
105
106static int query_formats(AVFilterContext *ctx)
107{
108 AVFilterFormats *formats;
109 AVFilterChannelLayouts *layouts;
110 static const enum AVSampleFormat sample_fmts[] = {
111 AV_SAMPLE_FMT_DBLP,
112 AV_SAMPLE_FMT_NONE
113 };
114 int ret;
115
116 layouts = ff_all_channel_counts();
117 if (!layouts)
118 return AVERROR(ENOMEM);
119 ret = ff_set_common_channel_layouts(ctx, layouts);
120 if (ret < 0)
121 return ret;
122
123 formats = ff_make_format_list(sample_fmts);
124 if (!formats)
125 return AVERROR(ENOMEM);
126 ret = ff_set_common_formats(ctx, formats);
127 if (ret < 0)
128 return ret;
129
130 formats = ff_all_samplerates();
131 if (!formats)
132 return AVERROR(ENOMEM);
133 return ff_set_common_samplerates(ctx, formats);
134}
135
136static inline int frame_size(int sample_rate, int frame_len_msec)
137{
138 const int frame_size = lrint((double)sample_rate * (frame_len_msec / 1000.0));
139 return frame_size + (frame_size % 2);
140}
141
142static void precalculate_fade_factors(double *fade_factors[2], int frame_len)
143{
144 const double step_size = 1.0 / frame_len;
145 int pos;
146
147 for (pos = 0; pos < frame_len; pos++) {
148 fade_factors[0][pos] = 1.0 - (step_size * (pos + 1.0));
149 fade_factors[1][pos] = 1.0 - fade_factors[0][pos];
150 }
151}
152
153static cqueue *cqueue_create(int size)
154{
155 cqueue *q;
156
157 q = av_malloc(sizeof(cqueue));
158 if (!q)
159 return NULL;
160
161 q->size = size;
162 q->nb_elements = 0;
163 q->first = 0;
164
165 q->elements = av_malloc_array(size, sizeof(double));
166 if (!q->elements) {
167 av_free(q);
168 return NULL;
169 }
170
171 return q;
172}
173
174static void cqueue_free(cqueue *q)
175{
176 if (q)
177 av_free(q->elements);
178 av_free(q);
179}
180
181static int cqueue_size(cqueue *q)
182{
183 return q->nb_elements;
184}
185
186static int cqueue_empty(cqueue *q)
187{
188 return !q->nb_elements;
189}
190
191static int cqueue_enqueue(cqueue *q, double element)
192{
193 int i;
194
195 av_assert2(q->nb_elements != q->size);
196
197 i = (q->first + q->nb_elements) % q->size;
198 q->elements[i] = element;
199 q->nb_elements++;
200
201 return 0;
202}
203
204static double cqueue_peek(cqueue *q, int index)
205{
206 av_assert2(index < q->nb_elements);
207 return q->elements[(q->first + index) % q->size];
208}
209
210static int cqueue_dequeue(cqueue *q, double *element)
211{
212 av_assert2(!cqueue_empty(q));
213
214 *element = q->elements[q->first];
215 q->first = (q->first + 1) % q->size;
216 q->nb_elements--;
217
218 return 0;
219}
220
221static int cqueue_pop(cqueue *q)
222{
223 av_assert2(!cqueue_empty(q));
224
225 q->first = (q->first + 1) % q->size;
226 q->nb_elements--;
227
228 return 0;
229}
230
231static void init_gaussian_filter(DynamicAudioNormalizerContext *s)
232{
233 double total_weight = 0.0;
234 const double sigma = (((s->filter_size / 2.0) - 1.0) / 3.0) + (1.0 / 3.0);
235 double adjust;
236 int i;
237
238 // Pre-compute constants
239 const int offset = s->filter_size / 2;
240 const double c1 = 1.0 / (sigma * sqrt(2.0 * M_PI));
241 const double c2 = 2.0 * sigma * sigma;
242
243 // Compute weights
244 for (i = 0; i < s->filter_size; i++) {
245 const int x = i - offset;
246
247 s->weights[i] = c1 * exp(-x * x / c2);
248 total_weight += s->weights[i];
249 }
250
251 // Adjust weights
252 adjust = 1.0 / total_weight;
253 for (i = 0; i < s->filter_size; i++) {
254 s->weights[i] *= adjust;
255 }
256}
257
258static av_cold void uninit(AVFilterContext *ctx)
259{
260 DynamicAudioNormalizerContext *s = ctx->priv;
261 int c;
262
263 av_freep(&s->prev_amplification_factor);
264 av_freep(&s->dc_correction_value);
265 av_freep(&s->compress_threshold);
266 av_freep(&s->fade_factors[0]);
267 av_freep(&s->fade_factors[1]);
268
269 for (c = 0; c < s->channels; c++) {
270 if (s->gain_history_original)
271 cqueue_free(s->gain_history_original[c]);
272 if (s->gain_history_minimum)
273 cqueue_free(s->gain_history_minimum[c]);
274 if (s->gain_history_smoothed)
275 cqueue_free(s->gain_history_smoothed[c]);
276 }
277
278 av_freep(&s->gain_history_original);
279 av_freep(&s->gain_history_minimum);
280 av_freep(&s->gain_history_smoothed);
281
282 av_freep(&s->weights);
283
284 ff_bufqueue_discard_all(&s->queue);
285}
286
287static int config_input(AVFilterLink *inlink)
288{
289 AVFilterContext *ctx = inlink->dst;
290 DynamicAudioNormalizerContext *s = ctx->priv;
291 int c;
292
293 uninit(ctx);
294
295 s->frame_len =
296 inlink->min_samples =
297 inlink->max_samples =
298 inlink->partial_buf_size = frame_size(inlink->sample_rate, s->frame_len_msec);
299 av_log(ctx, AV_LOG_DEBUG, "frame len %d\n", s->frame_len);
300
301 s->fade_factors[0] = av_malloc_array(s->frame_len, sizeof(*s->fade_factors[0]));
302 s->fade_factors[1] = av_malloc_array(s->frame_len, sizeof(*s->fade_factors[1]));
303
304 s->prev_amplification_factor = av_malloc_array(inlink->channels, sizeof(*s->prev_amplification_factor));
305 s->dc_correction_value = av_calloc(inlink->channels, sizeof(*s->dc_correction_value));
306 s->compress_threshold = av_calloc(inlink->channels, sizeof(*s->compress_threshold));
307 s->gain_history_original = av_calloc(inlink->channels, sizeof(*s->gain_history_original));
308 s->gain_history_minimum = av_calloc(inlink->channels, sizeof(*s->gain_history_minimum));
309 s->gain_history_smoothed = av_calloc(inlink->channels, sizeof(*s->gain_history_smoothed));
310 s->weights = av_malloc_array(s->filter_size, sizeof(*s->weights));
311 if (!s->prev_amplification_factor || !s->dc_correction_value ||
312 !s->compress_threshold || !s->fade_factors[0] || !s->fade_factors[1] ||
313 !s->gain_history_original || !s->gain_history_minimum ||
314 !s->gain_history_smoothed || !s->weights)
315 return AVERROR(ENOMEM);
316
317 for (c = 0; c < inlink->channels; c++) {
318 s->prev_amplification_factor[c] = 1.0;
319
320 s->gain_history_original[c] = cqueue_create(s->filter_size);
321 s->gain_history_minimum[c] = cqueue_create(s->filter_size);
322 s->gain_history_smoothed[c] = cqueue_create(s->filter_size);
323
324 if (!s->gain_history_original[c] || !s->gain_history_minimum[c] ||
325 !s->gain_history_smoothed[c])
326 return AVERROR(ENOMEM);
327 }
328
329 precalculate_fade_factors(s->fade_factors, s->frame_len);
330 init_gaussian_filter(s);
331
332 s->channels = inlink->channels;
333 s->delay = s->filter_size;
334
335 return 0;
336}
337
338static inline double fade(double prev, double next, int pos,
339 double *fade_factors[2])
340{
341 return fade_factors[0][pos] * prev + fade_factors[1][pos] * next;
342}
343
344static inline double pow_2(const double value)
345{
346 return value * value;
347}
348
349static inline double bound(const double threshold, const double val)
350{
351 const double CONST = 0.8862269254527580136490837416705725913987747280611935; //sqrt(PI) / 2.0
352 return erf(CONST * (val / threshold)) * threshold;
353}
354
355static double find_peak_magnitude(AVFrame *frame, int channel)
356{
357 double max = DBL_EPSILON;
358 int c, i;
359
360 if (channel == -1) {
361 for (c = 0; c < av_frame_get_channels(frame); c++) {
362 double *data_ptr = (double *)frame->extended_data[c];
363
364 for (i = 0; i < frame->nb_samples; i++)
365 max = FFMAX(max, fabs(data_ptr[i]));
366 }
367 } else {
368 double *data_ptr = (double *)frame->extended_data[channel];
369
370 for (i = 0; i < frame->nb_samples; i++)
371 max = FFMAX(max, fabs(data_ptr[i]));
372 }
373
374 return max;
375}
376
377static double compute_frame_rms(AVFrame *frame, int channel)
378{
379 double rms_value = 0.0;
380 int c, i;
381
382 if (channel == -1) {
383 for (c = 0; c < av_frame_get_channels(frame); c++) {
384 const double *data_ptr = (double *)frame->extended_data[c];
385
386 for (i = 0; i < frame->nb_samples; i++) {
387 rms_value += pow_2(data_ptr[i]);
388 }
389 }
390
391 rms_value /= frame->nb_samples * av_frame_get_channels(frame);
392 } else {
393 const double *data_ptr = (double *)frame->extended_data[channel];
394 for (i = 0; i < frame->nb_samples; i++) {
395 rms_value += pow_2(data_ptr[i]);
396 }
397
398 rms_value /= frame->nb_samples;
399 }
400
401 return FFMAX(sqrt(rms_value), DBL_EPSILON);
402}
403
404static double get_max_local_gain(DynamicAudioNormalizerContext *s, AVFrame *frame,
405 int channel)
406{
407 const double maximum_gain = s->peak_value / find_peak_magnitude(frame, channel);
408 const double rms_gain = s->target_rms > DBL_EPSILON ? (s->target_rms / compute_frame_rms(frame, channel)) : DBL_MAX;
409 return bound(s->max_amplification, FFMIN(maximum_gain, rms_gain));
410}
411
412static double minimum_filter(cqueue *q)
413{
414 double min = DBL_MAX;
415 int i;
416
417 for (i = 0; i < cqueue_size(q); i++) {
418 min = FFMIN(min, cqueue_peek(q, i));
419 }
420
421 return min;
422}
423
424static double gaussian_filter(DynamicAudioNormalizerContext *s, cqueue *q)
425{
426 double result = 0.0;
427 int i;
428
429 for (i = 0; i < cqueue_size(q); i++) {
430 result += cqueue_peek(q, i) * s->weights[i];
431 }
432
433 return result;
434}
435
436static void update_gain_history(DynamicAudioNormalizerContext *s, int channel,
437 double current_gain_factor)
438{
439 if (cqueue_empty(s->gain_history_original[channel]) ||
440 cqueue_empty(s->gain_history_minimum[channel])) {
441 const int pre_fill_size = s->filter_size / 2;
442 const double initial_value = s->alt_boundary_mode ? current_gain_factor : 1.0;
443
444 s->prev_amplification_factor[channel] = initial_value;
445
446 while (cqueue_size(s->gain_history_original[channel]) < pre_fill_size) {
447 cqueue_enqueue(s->gain_history_original[channel], initial_value);
448 }
449 }
450
451 cqueue_enqueue(s->gain_history_original[channel], current_gain_factor);
452
453 while (cqueue_size(s->gain_history_original[channel]) >= s->filter_size) {
454 double minimum;
455 av_assert0(cqueue_size(s->gain_history_original[channel]) == s->filter_size);
456
457 if (cqueue_empty(s->gain_history_minimum[channel])) {
458 const int pre_fill_size = s->filter_size / 2;
459 double initial_value = s->alt_boundary_mode ? cqueue_peek(s->gain_history_original[channel], 0) : 1.0;
460 int input = pre_fill_size;
461
462 while (cqueue_size(s->gain_history_minimum[channel]) < pre_fill_size) {
463 initial_value = FFMIN(initial_value, cqueue_peek(s->gain_history_original[channel], ++input));
464 cqueue_enqueue(s->gain_history_minimum[channel], initial_value);
465 }
466 }
467
468 minimum = minimum_filter(s->gain_history_original[channel]);
469
470 cqueue_enqueue(s->gain_history_minimum[channel], minimum);
471
472 cqueue_pop(s->gain_history_original[channel]);
473 }
474
475 while (cqueue_size(s->gain_history_minimum[channel]) >= s->filter_size) {
476 double smoothed;
477 av_assert0(cqueue_size(s->gain_history_minimum[channel]) == s->filter_size);
478 smoothed = gaussian_filter(s, s->gain_history_minimum[channel]);
479
480 cqueue_enqueue(s->gain_history_smoothed[channel], smoothed);
481
482 cqueue_pop(s->gain_history_minimum[channel]);
483 }
484}
485
486static inline double update_value(double new, double old, double aggressiveness)
487{
488 av_assert0((aggressiveness >= 0.0) && (aggressiveness <= 1.0));
489 return aggressiveness * new + (1.0 - aggressiveness) * old;
490}
491
492static void perform_dc_correction(DynamicAudioNormalizerContext *s, AVFrame *frame)
493{
494 const double diff = 1.0 / frame->nb_samples;
495 int is_first_frame = cqueue_empty(s->gain_history_original[0]);
496 int c, i;
497
498 for (c = 0; c < s->channels; c++) {
499 double *dst_ptr = (double *)frame->extended_data[c];
500 double current_average_value = 0.0;
501 double prev_value;
502
503 for (i = 0; i < frame->nb_samples; i++)
504 current_average_value += dst_ptr[i] * diff;
505
506 prev_value = is_first_frame ? current_average_value : s->dc_correction_value[c];
507 s->dc_correction_value[c] = is_first_frame ? current_average_value : update_value(current_average_value, s->dc_correction_value[c], 0.1);
508
509 for (i = 0; i < frame->nb_samples; i++) {
510 dst_ptr[i] -= fade(prev_value, s->dc_correction_value[c], i, s->fade_factors);
511 }
512 }
513}
514
515static double setup_compress_thresh(double threshold)
516{
517 if ((threshold > DBL_EPSILON) && (threshold < (1.0 - DBL_EPSILON))) {
518 double current_threshold = threshold;
519 double step_size = 1.0;
520
521 while (step_size > DBL_EPSILON) {
522 while ((llrint((current_threshold + step_size) * (UINT64_C(1) << 63)) >
523 llrint(current_threshold * (UINT64_C(1) << 63))) &&
524 (bound(current_threshold + step_size, 1.0) <= threshold)) {
525 current_threshold += step_size;
526 }
527
528 step_size /= 2.0;
529 }
530
531 return current_threshold;
532 } else {
533 return threshold;
534 }
535}
536
537static double compute_frame_std_dev(DynamicAudioNormalizerContext *s,
538 AVFrame *frame, int channel)
539{
540 double variance = 0.0;
541 int i, c;
542
543 if (channel == -1) {
544 for (c = 0; c < s->channels; c++) {
545 const double *data_ptr = (double *)frame->extended_data[c];
546
547 for (i = 0; i < frame->nb_samples; i++) {
548 variance += pow_2(data_ptr[i]); // Assume that MEAN is *zero*
549 }
550 }
551 variance /= (s->channels * frame->nb_samples) - 1;
552 } else {
553 const double *data_ptr = (double *)frame->extended_data[channel];
554
555 for (i = 0; i < frame->nb_samples; i++) {
556 variance += pow_2(data_ptr[i]); // Assume that MEAN is *zero*
557 }
558 variance /= frame->nb_samples - 1;
559 }
560
561 return FFMAX(sqrt(variance), DBL_EPSILON);
562}
563
564static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame)
565{
566 int is_first_frame = cqueue_empty(s->gain_history_original[0]);
567 int c, i;
568
569 if (s->channels_coupled) {
570 const double standard_deviation = compute_frame_std_dev(s, frame, -1);
571 const double current_threshold = FFMIN(1.0, s->compress_factor * standard_deviation);
572
573 const double prev_value = is_first_frame ? current_threshold : s->compress_threshold[0];
574 double prev_actual_thresh, curr_actual_thresh;
575 s->compress_threshold[0] = is_first_frame ? current_threshold : update_value(current_threshold, s->compress_threshold[0], (1.0/3.0));
576
577 prev_actual_thresh = setup_compress_thresh(prev_value);
578 curr_actual_thresh = setup_compress_thresh(s->compress_threshold[0]);
579
580 for (c = 0; c < s->channels; c++) {
581 double *const dst_ptr = (double *)frame->extended_data[c];
582 for (i = 0; i < frame->nb_samples; i++) {
583 const double localThresh = fade(prev_actual_thresh, curr_actual_thresh, i, s->fade_factors);
584 dst_ptr[i] = copysign(bound(localThresh, fabs(dst_ptr[i])), dst_ptr[i]);
585 }
586 }
587 } else {
588 for (c = 0; c < s->channels; c++) {
589 const double standard_deviation = compute_frame_std_dev(s, frame, c);
590 const double current_threshold = setup_compress_thresh(FFMIN(1.0, s->compress_factor * standard_deviation));
591
592 const double prev_value = is_first_frame ? current_threshold : s->compress_threshold[c];
593 double prev_actual_thresh, curr_actual_thresh;
594 double *dst_ptr;
595 s->compress_threshold[c] = is_first_frame ? current_threshold : update_value(current_threshold, s->compress_threshold[c], 1.0/3.0);
596
597 prev_actual_thresh = setup_compress_thresh(prev_value);
598 curr_actual_thresh = setup_compress_thresh(s->compress_threshold[c]);
599
600 dst_ptr = (double *)frame->extended_data[c];
601 for (i = 0; i < frame->nb_samples; i++) {
602 const double localThresh = fade(prev_actual_thresh, curr_actual_thresh, i, s->fade_factors);
603 dst_ptr[i] = copysign(bound(localThresh, fabs(dst_ptr[i])), dst_ptr[i]);
604 }
605 }
606 }
607}
608
609static void analyze_frame(DynamicAudioNormalizerContext *s, AVFrame *frame)
610{
611 if (s->dc_correction) {
612 perform_dc_correction(s, frame);
613 }
614
615 if (s->compress_factor > DBL_EPSILON) {
616 perform_compression(s, frame);
617 }
618
619 if (s->channels_coupled) {
620 const double current_gain_factor = get_max_local_gain(s, frame, -1);
621 int c;
622
623 for (c = 0; c < s->channels; c++)
624 update_gain_history(s, c, current_gain_factor);
625 } else {
626 int c;
627
628 for (c = 0; c < s->channels; c++)
629 update_gain_history(s, c, get_max_local_gain(s, frame, c));
630 }
631}
632
633static void amplify_frame(DynamicAudioNormalizerContext *s, AVFrame *frame)
634{
635 int c, i;
636
637 for (c = 0; c < s->channels; c++) {
638 double *dst_ptr = (double *)frame->extended_data[c];
639 double current_amplification_factor;
640
641 cqueue_dequeue(s->gain_history_smoothed[c], &current_amplification_factor);
642
643 for (i = 0; i < frame->nb_samples; i++) {
644 const double amplification_factor = fade(s->prev_amplification_factor[c],
645 current_amplification_factor, i,
646 s->fade_factors);
647
648 dst_ptr[i] *= amplification_factor;
649
650 if (fabs(dst_ptr[i]) > s->peak_value)
651 dst_ptr[i] = copysign(s->peak_value, dst_ptr[i]);
652 }
653
654 s->prev_amplification_factor[c] = current_amplification_factor;
655 }
656}
657
658static int filter_frame(AVFilterLink *inlink, AVFrame *in)
659{
660 AVFilterContext *ctx = inlink->dst;
661 DynamicAudioNormalizerContext *s = ctx->priv;
662 AVFilterLink *outlink = inlink->dst->outputs[0];
663 int ret = 0;
664
665 if (!cqueue_empty(s->gain_history_smoothed[0])) {
666 AVFrame *out = ff_bufqueue_get(&s->queue);
667
668 amplify_frame(s, out);
669 ret = ff_filter_frame(outlink, out);
670 }
671
672 analyze_frame(s, in);
673 ff_bufqueue_add(ctx, &s->queue, in);
674
675 return ret;
676}
677
678static int flush_buffer(DynamicAudioNormalizerContext *s, AVFilterLink *inlink,
679 AVFilterLink *outlink)
680{
681 AVFrame *out = ff_get_audio_buffer(outlink, s->frame_len);
682 int c, i;
683
684 if (!out)
685 return AVERROR(ENOMEM);
686
687 for (c = 0; c < s->channels; c++) {
688 double *dst_ptr = (double *)out->extended_data[c];
689
690 for (i = 0; i < out->nb_samples; i++) {
691 dst_ptr[i] = s->alt_boundary_mode ? DBL_EPSILON : ((s->target_rms > DBL_EPSILON) ? FFMIN(s->peak_value, s->target_rms) : s->peak_value);
692 if (s->dc_correction) {
693 dst_ptr[i] *= ((i % 2) == 1) ? -1 : 1;
694 dst_ptr[i] += s->dc_correction_value[c];
695 }
696 }
697 }
698
699 s->delay--;
700 return filter_frame(inlink, out);
701}
702
703static int request_frame(AVFilterLink *outlink)
704{
705 AVFilterContext *ctx = outlink->src;
706 DynamicAudioNormalizerContext *s = ctx->priv;
707 int ret = 0;
708
709 ret = ff_request_frame(ctx->inputs[0]);
710
711 if (ret == AVERROR_EOF && !ctx->is_disabled && s->delay) {
712 if (!cqueue_empty(s->gain_history_smoothed[0])) {
713 ret = flush_buffer(s, ctx->inputs[0], outlink);
714 } else if (s->queue.available) {
715 AVFrame *out = ff_bufqueue_get(&s->queue);
716
717 ret = ff_filter_frame(outlink, out);
718 }
719 }
720
721 return ret;
722}
723
724static const AVFilterPad avfilter_af_dynaudnorm_inputs[] = {
725 {
726 .name = "default",
727 .type = AVMEDIA_TYPE_AUDIO,
728 .filter_frame = filter_frame,
729 .config_props = config_input,
730 .needs_writable = 1,
731 },
732 { NULL }
733};
734
735static const AVFilterPad avfilter_af_dynaudnorm_outputs[] = {
736 {
737 .name = "default",
738 .type = AVMEDIA_TYPE_AUDIO,
739 .request_frame = request_frame,
740 },
741 { NULL }
742};
743
744AVFilter ff_af_dynaudnorm = {
745 .name = "dynaudnorm",
746 .description = NULL_IF_CONFIG_SMALL("Dynamic Audio Normalizer."),
747 .query_formats = query_formats,
748 .priv_size = sizeof(DynamicAudioNormalizerContext),
749 .init = init,
750 .uninit = uninit,
751 .inputs = avfilter_af_dynaudnorm_inputs,
752 .outputs = avfilter_af_dynaudnorm_outputs,
753 .priv_class = &dynaudnorm_class,
754};
755
generated by cgit at 2024-05-17 02:05:24 (GMT)