FFmpeg coverage

Line	Branch	Exec	Source
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/channel_layout.h"
31			#include "libavutil/eval.h"
32			#include "libavutil/mem.h"
33			#include "libavutil/opt.h"
34
35			#define MIN_FILTER_SIZE 3
36			#define MAX_FILTER_SIZE 301
37
38			#define FF_BUFQUEUE_SIZE (MAX_FILTER_SIZE + 1)
39			#include "libavfilter/bufferqueue.h"
40
41			#include "audio.h"
42			#include "avfilter.h"
43			#include "filters.h"
44
45			static const char * const var_names[] = {
46			"ch", ///< the value of the current channel
47			"sn", ///< number of samples
48			"nb_channels",
49			"t", ///< timestamp expressed in seconds
50			"sr", ///< sample rate
51			"p", ///< peak value
52			NULL
53			};
54
55			enum var_name {
56			VAR_CH,
57			VAR_SN,
58			VAR_NB_CHANNELS,
59			VAR_T,
60			VAR_SR,
61			VAR_P,
62			VAR_VARS_NB
63			};
64
65			typedef struct local_gain {
66			double max_gain;
67			double threshold;
68			} local_gain;
69
70			typedef struct cqueue {
71			double *elements;
72			int size;
73			int max_size;
74			int nb_elements;
75			} cqueue;
76
77			typedef struct DynamicAudioNormalizerContext {
78			const AVClass *class;
79
80			struct FFBufQueue queue;
81
82			int frame_len;
83			int frame_len_msec;
84			int filter_size;
85			int dc_correction;
86			int channels_coupled;
87			int alt_boundary_mode;
88			double overlap;
89			char *expr_str;
90
91			double peak_value;
92			double max_amplification;
93			double target_rms;
94			double compress_factor;
95			double threshold;
96			double *prev_amplification_factor;
97			double *dc_correction_value;
98			double *compress_threshold;
99			double *weights;
100
101			int channels;
102			int sample_advance;
103			int eof;
104			char *channels_to_filter;
105			AVChannelLayout ch_layout;
106			int64_t pts;
107
108			cqueue **gain_history_original;
109			cqueue **gain_history_minimum;
110			cqueue **gain_history_smoothed;
111			cqueue **threshold_history;
112
113			cqueue *is_enabled;
114
115			AVFrame *window;
116
117			AVExpr *expr;
118			double var_values[VAR_VARS_NB];
119			} DynamicAudioNormalizerContext;
120
121			typedef struct ThreadData {
122			AVFrame *in, *out;
123			int enabled;
124			} ThreadData;
125
126			#define OFFSET(x) offsetof(DynamicAudioNormalizerContext, x)
127			#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
128
129			static const AVOption dynaudnorm_options[] = {
130			{ "framelen", "set the frame length in msec", OFFSET(frame_len_msec), AV_OPT_TYPE_INT, {.i64 = 500}, 10, 8000, FLAGS },
131			{ "f", "set the frame length in msec", OFFSET(frame_len_msec), AV_OPT_TYPE_INT, {.i64 = 500}, 10, 8000, FLAGS },
132			{ "gausssize", "set the filter size", OFFSET(filter_size), AV_OPT_TYPE_INT, {.i64 = 31}, 3, 301, FLAGS },
133			{ "g", "set the filter size", OFFSET(filter_size), AV_OPT_TYPE_INT, {.i64 = 31}, 3, 301, FLAGS },
134			{ "peak", "set the peak value", OFFSET(peak_value), AV_OPT_TYPE_DOUBLE, {.dbl = 0.95}, 0.0, 1.0, FLAGS },
135			{ "p", "set the peak value", OFFSET(peak_value), AV_OPT_TYPE_DOUBLE, {.dbl = 0.95}, 0.0, 1.0, FLAGS },
136			{ "maxgain", "set the max amplification", OFFSET(max_amplification), AV_OPT_TYPE_DOUBLE, {.dbl = 10.0}, 1.0, 100.0, FLAGS },
137			{ "m", "set the max amplification", OFFSET(max_amplification), AV_OPT_TYPE_DOUBLE, {.dbl = 10.0}, 1.0, 100.0, FLAGS },
138			{ "targetrms", "set the target RMS", OFFSET(target_rms), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 1.0, FLAGS },
139			{ "r", "set the target RMS", OFFSET(target_rms), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 1.0, FLAGS },
140			{ "coupling", "set channel coupling", OFFSET(channels_coupled), AV_OPT_TYPE_BOOL, {.i64 = 1}, 0, 1, FLAGS },
141			{ "n", "set channel coupling", OFFSET(channels_coupled), AV_OPT_TYPE_BOOL, {.i64 = 1}, 0, 1, FLAGS },
142			{ "correctdc", "set DC correction", OFFSET(dc_correction), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, FLAGS },
143			{ "c", "set DC correction", OFFSET(dc_correction), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, FLAGS },
144			{ "altboundary", "set alternative boundary mode", OFFSET(alt_boundary_mode), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, FLAGS },
145			{ "b", "set alternative boundary mode", OFFSET(alt_boundary_mode), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, FLAGS },
146			{ "compress", "set the compress factor", OFFSET(compress_factor), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 30.0, FLAGS },
147			{ "s", "set the compress factor", OFFSET(compress_factor), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 30.0, FLAGS },
148			{ "threshold", "set the threshold value", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 1.0, FLAGS },
149			{ "t", "set the threshold value", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 1.0, FLAGS },
150			{ "channels", "set channels to filter", OFFSET(channels_to_filter),AV_OPT_TYPE_STRING, {.str="all"}, 0, 0, FLAGS },
151			{ "h", "set channels to filter", OFFSET(channels_to_filter),AV_OPT_TYPE_STRING, {.str="all"}, 0, 0, FLAGS },
152			{ "overlap", "set the frame overlap", OFFSET(overlap), AV_OPT_TYPE_DOUBLE, {.dbl=.0}, 0.0, 1.0, FLAGS },
153			{ "o", "set the frame overlap", OFFSET(overlap), AV_OPT_TYPE_DOUBLE, {.dbl=.0}, 0.0, 1.0, FLAGS },
154			{ "curve", "set the custom peak mapping curve",OFFSET(expr_str), AV_OPT_TYPE_STRING, {.str=NULL}, .flags = FLAGS },
155			{ "v", "set the custom peak mapping curve",OFFSET(expr_str), AV_OPT_TYPE_STRING, {.str=NULL}, .flags = FLAGS },
156			{ NULL }
157			};
158
159			AVFILTER_DEFINE_CLASS(dynaudnorm);
160
161		✗	static av_cold int init(AVFilterContext *ctx)
162			{
163		✗	DynamicAudioNormalizerContext *s = ctx->priv;
164
165		✗	if (!(s->filter_size & 1)) {
166		✗	av_log(ctx, AV_LOG_WARNING, "filter size %d is invalid. Changing to an odd value.\n", s->filter_size);
167		✗	s->filter_size |= 1;
168			}
169
170		✗	return 0;
171			}
172
173		✗	static inline int frame_size(int sample_rate, int frame_len_msec)
174			{
175		✗	const int frame_size = lrint((double)sample_rate * (frame_len_msec / 1000.0));
176		✗	return frame_size + (frame_size % 2);
177			}
178
179		✗	static cqueue *cqueue_create(int size, int max_size)
180			{
181			cqueue *q;
182
183		✗	if (max_size < size)
184		✗	return NULL;
185
186		✗	q = av_malloc(sizeof(cqueue));
187		✗	if (!q)
188		✗	return NULL;
189
190		✗	q->max_size = max_size;
191		✗	q->size = size;
192		✗	q->nb_elements = 0;
193
194		✗	q->elements = av_malloc_array(max_size, sizeof(double));
195		✗	if (!q->elements) {
196		✗	av_free(q);
197		✗	return NULL;
198			}
199
200		✗	return q;
201			}
202
203		✗	static void cqueue_free(cqueue *q)
204			{
205		✗	if (q)
206		✗	av_free(q->elements);
207		✗	av_free(q);
208		✗	}
209
210		✗	static int cqueue_size(cqueue *q)
211			{
212		✗	return q->nb_elements;
213			}
214
215		✗	static int cqueue_empty(cqueue *q)
216			{
217		✗	return q->nb_elements <= 0;
218			}
219
220		✗	static int cqueue_enqueue(cqueue *q, double element)
221			{
222			av_assert2(q->nb_elements < q->max_size);
223
224		✗	q->elements[q->nb_elements] = element;
225		✗	q->nb_elements++;
226
227		✗	return 0;
228			}
229
230		✗	static double cqueue_peek(cqueue *q, int index)
231			{
232			av_assert2(index < q->nb_elements);
233		✗	return q->elements[index];
234			}
235
236		✗	static int cqueue_dequeue(cqueue *q, double *element)
237			{
238			av_assert2(!cqueue_empty(q));
239
240		✗	*element = q->elements[0];
241		✗	memmove(&q->elements[0], &q->elements[1], (q->nb_elements - 1) * sizeof(double));
242		✗	q->nb_elements--;
243
244		✗	return 0;
245			}
246
247		✗	static int cqueue_pop(cqueue *q)
248			{
249			av_assert2(!cqueue_empty(q));
250
251		✗	memmove(&q->elements[0], &q->elements[1], (q->nb_elements - 1) * sizeof(double));
252		✗	q->nb_elements--;
253
254		✗	return 0;
255			}
256
257		✗	static void cqueue_resize(cqueue *q, int new_size)
258			{
259			av_assert2(q->max_size >= new_size);
260			av_assert2(MIN_FILTER_SIZE <= new_size);
261
262		✗	if (new_size > q->nb_elements) {
263		✗	const int side = (new_size - q->nb_elements) / 2;
264
265		✗	memmove(q->elements + side, q->elements, sizeof(double) * q->nb_elements);
266		✗	for (int i = 0; i < side; i++)
267		✗	q->elements[i] = q->elements[side];
268		✗	q->nb_elements = new_size - 1 - side;
269			} else {
270		✗	int count = (q->size - new_size + 1) / 2;
271
272		✗	while (count-- > 0)
273		✗	cqueue_pop(q);
274			}
275
276		✗	q->size = new_size;
277		✗	}
278
279		✗	static void init_gaussian_filter(DynamicAudioNormalizerContext *s)
280			{
281		✗	double total_weight = 0.0;
282		✗	const double sigma = (((s->filter_size / 2.0) - 1.0) / 3.0) + (1.0 / 3.0);
283			double adjust;
284
285			// Pre-compute constants
286		✗	const int offset = s->filter_size / 2;
287		✗	const double c1 = 1.0 / (sigma * sqrt(2.0 * M_PI));
288		✗	const double c2 = 2.0 * sigma * sigma;
289
290			// Compute weights
291		✗	for (int i = 0; i < s->filter_size; i++) {
292		✗	const int x = i - offset;
293
294		✗	s->weights[i] = c1 * exp(-x * x / c2);
295		✗	total_weight += s->weights[i];
296			}
297
298			// Adjust weights
299		✗	adjust = 1.0 / total_weight;
300		✗	for (int i = 0; i < s->filter_size; i++) {
301		✗	s->weights[i] *= adjust;
302			}
303		✗	}
304
305		✗	static av_cold void uninit(AVFilterContext *ctx)
306			{
307		✗	DynamicAudioNormalizerContext *s = ctx->priv;
308
309		✗	av_freep(&s->prev_amplification_factor);
310		✗	av_freep(&s->dc_correction_value);
311		✗	av_freep(&s->compress_threshold);
312
313		✗	for (int c = 0; c < s->channels; c++) {
314		✗	if (s->gain_history_original)
315		✗	cqueue_free(s->gain_history_original[c]);
316		✗	if (s->gain_history_minimum)
317		✗	cqueue_free(s->gain_history_minimum[c]);
318		✗	if (s->gain_history_smoothed)
319		✗	cqueue_free(s->gain_history_smoothed[c]);
320		✗	if (s->threshold_history)
321		✗	cqueue_free(s->threshold_history[c]);
322			}
323
324		✗	av_freep(&s->gain_history_original);
325		✗	av_freep(&s->gain_history_minimum);
326		✗	av_freep(&s->gain_history_smoothed);
327		✗	av_freep(&s->threshold_history);
328
329		✗	cqueue_free(s->is_enabled);
330		✗	s->is_enabled = NULL;
331
332		✗	av_freep(&s->weights);
333
334		✗	av_channel_layout_uninit(&s->ch_layout);
335
336		✗	ff_bufqueue_discard_all(&s->queue);
337
338		✗	av_frame_free(&s->window);
339		✗	av_expr_free(s->expr);
340		✗	s->expr = NULL;
341		✗	}
342
343		✗	static int config_input(AVFilterLink *inlink)
344			{
345		✗	AVFilterContext *ctx = inlink->dst;
346		✗	DynamicAudioNormalizerContext *s = ctx->priv;
347		✗	int ret = 0;
348
349		✗	uninit(ctx);
350
351		✗	s->channels = inlink->ch_layout.nb_channels;
352		✗	s->frame_len = frame_size(inlink->sample_rate, s->frame_len_msec);
353		✗	av_log(ctx, AV_LOG_DEBUG, "frame len %d\n", s->frame_len);
354
355		✗	s->prev_amplification_factor = av_malloc_array(inlink->ch_layout.nb_channels, sizeof(*s->prev_amplification_factor));
356		✗	s->dc_correction_value = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->dc_correction_value));
357		✗	s->compress_threshold = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->compress_threshold));
358		✗	s->gain_history_original = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->gain_history_original));
359		✗	s->gain_history_minimum = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->gain_history_minimum));
360		✗	s->gain_history_smoothed = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->gain_history_smoothed));
361		✗	s->threshold_history = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->threshold_history));
362		✗	s->weights = av_malloc_array(MAX_FILTER_SIZE, sizeof(*s->weights));
363		✗	s->is_enabled = cqueue_create(s->filter_size, MAX_FILTER_SIZE);
364		✗	if (!s->prev_amplification_factor || !s->dc_correction_value ||
365		✗	!s->compress_threshold ||
366		✗	!s->gain_history_original || !s->gain_history_minimum ||
367		✗	!s->gain_history_smoothed || !s->threshold_history ||
368		✗	!s->is_enabled || !s->weights)
369		✗	return AVERROR(ENOMEM);
370
371		✗	for (int c = 0; c < inlink->ch_layout.nb_channels; c++) {
372		✗	s->prev_amplification_factor[c] = 1.0;
373
374		✗	s->gain_history_original[c] = cqueue_create(s->filter_size, MAX_FILTER_SIZE);
375		✗	s->gain_history_minimum[c] = cqueue_create(s->filter_size, MAX_FILTER_SIZE);
376		✗	s->gain_history_smoothed[c] = cqueue_create(s->filter_size, MAX_FILTER_SIZE);
377		✗	s->threshold_history[c] = cqueue_create(s->filter_size, MAX_FILTER_SIZE);
378
379		✗	if (!s->gain_history_original[c] || !s->gain_history_minimum[c] ||
380		✗	!s->gain_history_smoothed[c] || !s->threshold_history[c])
381		✗	return AVERROR(ENOMEM);
382			}
383
384		✗	init_gaussian_filter(s);
385
386		✗	s->window = ff_get_audio_buffer(ctx->outputs[0], s->frame_len * 2);
387		✗	if (!s->window)
388		✗	return AVERROR(ENOMEM);
389		✗	s->sample_advance = FFMAX(1, lrint(s->frame_len * (1. - s->overlap)));
390
391		✗	s->var_values[VAR_SR] = inlink->sample_rate;
392		✗	s->var_values[VAR_NB_CHANNELS] = s->channels;
393
394		✗	if (s->expr_str)
395		✗	ret = av_expr_parse(&s->expr, s->expr_str, var_names, NULL, NULL,
396			NULL, NULL, 0, ctx);
397		✗	return ret;
398			}
399
400		✗	static inline double fade(double prev, double next, int pos, int length)
401			{
402		✗	const double step_size = 1.0 / length;
403		✗	const double f0 = 1.0 - (step_size * (pos + 1.0));
404		✗	const double f1 = 1.0 - f0;
405		✗	return f0 * prev + f1 * next;
406			}
407
408		✗	static inline double pow_2(const double value)
409			{
410		✗	return value * value;
411			}
412
413		✗	static inline double bound(const double threshold, const double val)
414			{
415		✗	const double CONST = 0.8862269254527580136490837416705725913987747280611935; //sqrt(PI) / 2.0
416		✗	return erf(CONST * (val / threshold)) * threshold;
417			}
418
419		✗	static double find_peak_magnitude(AVFrame *frame, int channel)
420			{
421		✗	double max = DBL_EPSILON;
422
423		✗	if (channel == -1) {
424		✗	for (int c = 0; c < frame->ch_layout.nb_channels; c++) {
425		✗	double *data_ptr = (double *)frame->extended_data[c];
426
427		✗	for (int i = 0; i < frame->nb_samples; i++)
428		✗	max = fmax(max, fabs(data_ptr[i]));
429			}
430			} else {
431		✗	double *data_ptr = (double *)frame->extended_data[channel];
432
433		✗	for (int i = 0; i < frame->nb_samples; i++)
434		✗	max = fmax(max, fabs(data_ptr[i]));
435			}
436
437		✗	return max;
438			}
439
440		✗	static double compute_frame_rms(AVFrame *frame, int channel)
441			{
442		✗	double rms_value = 0.0;
443
444		✗	if (channel == -1) {
445		✗	for (int c = 0; c < frame->ch_layout.nb_channels; c++) {
446		✗	const double *data_ptr = (double *)frame->extended_data[c];
447
448		✗	for (int i = 0; i < frame->nb_samples; i++) {
449		✗	rms_value += pow_2(data_ptr[i]);
450			}
451			}
452
453		✗	rms_value /= frame->nb_samples * frame->ch_layout.nb_channels;
454			} else {
455		✗	const double *data_ptr = (double *)frame->extended_data[channel];
456		✗	for (int i = 0; i < frame->nb_samples; i++) {
457		✗	rms_value += pow_2(data_ptr[i]);
458			}
459
460		✗	rms_value /= frame->nb_samples;
461			}
462
463		✗	return fmax(sqrt(rms_value), DBL_EPSILON);
464			}
465
466		✗	static local_gain get_max_local_gain(DynamicAudioNormalizerContext *s, AVFrame *frame,
467			int channel)
468			{
469		✗	const double peak_magnitude = find_peak_magnitude(frame, channel);
470		✗	const double maximum_gain = s->peak_value / peak_magnitude;
471		✗	const double rms_gain = s->target_rms > DBL_EPSILON ? (s->target_rms / compute_frame_rms(frame, channel)) : DBL_MAX;
472		✗	double target_gain = DBL_MAX;
473			local_gain gain;
474
475		✗	if (s->expr_str) {
476			double var_values[VAR_VARS_NB];
477
478		✗	memcpy(var_values, s->var_values, sizeof(var_values));
479
480		✗	var_values[VAR_CH] = channel;
481		✗	var_values[VAR_P] = peak_magnitude;
482
483		✗	target_gain = av_expr_eval(s->expr, var_values, s) / peak_magnitude;
484			}
485
486		✗	gain.threshold = peak_magnitude > s->threshold;
487		✗	gain.max_gain = bound(s->max_amplification, fmin(target_gain, fmin(maximum_gain, rms_gain)));
488
489		✗	return gain;
490			}
491
492		✗	static double minimum_filter(cqueue *q)
493			{
494		✗	double min = DBL_MAX;
495
496		✗	for (int i = 0; i < cqueue_size(q); i++) {
497		✗	min = fmin(min, cqueue_peek(q, i));
498			}
499
500		✗	return min;
501			}
502
503		✗	static double gaussian_filter(DynamicAudioNormalizerContext *s, cqueue *q, cqueue *tq)
504			{
505		✗	const double *weights = s->weights;
506		✗	double result = 0.0, tsum = 0.0;
507
508		✗	for (int i = 0; i < cqueue_size(q); i++) {
509		✗	double tq_item = cqueue_peek(tq, i);
510		✗	double q_item = cqueue_peek(q, i);
511
512		✗	tsum += tq_item * weights[i];
513		✗	result += tq_item * weights[i] * q_item;
514			}
515
516		✗	if (tsum == 0.0)
517		✗	result = 1.0;
518
519		✗	return result;
520			}
521
522		✗	static void update_gain_history(DynamicAudioNormalizerContext *s, int channel,
523			local_gain gain)
524			{
525		✗	if (cqueue_empty(s->gain_history_original[channel])) {
526		✗	const int pre_fill_size = s->filter_size / 2;
527		✗	const double initial_value = s->alt_boundary_mode ? gain.max_gain : fmin(1.0, gain.max_gain);
528
529		✗	s->prev_amplification_factor[channel] = initial_value;
530
531		✗	while (cqueue_size(s->gain_history_original[channel]) < pre_fill_size) {
532		✗	cqueue_enqueue(s->gain_history_original[channel], initial_value);
533		✗	cqueue_enqueue(s->threshold_history[channel], gain.threshold);
534			}
535			}
536
537		✗	cqueue_enqueue(s->gain_history_original[channel], gain.max_gain);
538
539		✗	while (cqueue_size(s->gain_history_original[channel]) >= s->filter_size) {
540			double minimum;
541
542		✗	if (cqueue_empty(s->gain_history_minimum[channel])) {
543		✗	const int pre_fill_size = s->filter_size / 2;
544		✗	double initial_value = s->alt_boundary_mode ? cqueue_peek(s->gain_history_original[channel], 0) : 1.0;
545		✗	int input = pre_fill_size;
546
547		✗	while (cqueue_size(s->gain_history_minimum[channel]) < pre_fill_size) {
548		✗	input++;
549		✗	initial_value = fmin(initial_value, cqueue_peek(s->gain_history_original[channel], input));
550		✗	cqueue_enqueue(s->gain_history_minimum[channel], initial_value);
551			}
552			}
553
554		✗	minimum = minimum_filter(s->gain_history_original[channel]);
555
556		✗	cqueue_enqueue(s->gain_history_minimum[channel], minimum);
557
558		✗	cqueue_enqueue(s->threshold_history[channel], gain.threshold);
559
560		✗	cqueue_pop(s->gain_history_original[channel]);
561			}
562
563		✗	while (cqueue_size(s->gain_history_minimum[channel]) >= s->filter_size) {
564			double smoothed, limit;
565
566		✗	smoothed = gaussian_filter(s, s->gain_history_minimum[channel], s->threshold_history[channel]);
567		✗	limit = cqueue_peek(s->gain_history_original[channel], 0);
568		✗	smoothed = fmin(smoothed, limit);
569
570		✗	cqueue_enqueue(s->gain_history_smoothed[channel], smoothed);
571
572		✗	cqueue_pop(s->gain_history_minimum[channel]);
573		✗	cqueue_pop(s->threshold_history[channel]);
574			}
575		✗	}
576
577		✗	static int update_gain_histories(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
578			{
579		✗	DynamicAudioNormalizerContext *s = ctx->priv;
580		✗	AVFrame *analyze_frame = arg;
581		✗	const int channels = s->channels;
582		✗	const int start = ff_slice_pos(channels, jobnr, nb_jobs);
583		✗	const int end = ff_slice_pos(channels, jobnr + 1, nb_jobs);
584
585		✗	for (int c = start; c < end; c++)
586		✗	update_gain_history(s, c, get_max_local_gain(s, analyze_frame, c));
587
588		✗	return 0;
589			}
590
591		✗	static inline double update_value(double new, double old, double aggressiveness)
592			{
593		✗	av_assert0((aggressiveness >= 0.0) && (aggressiveness <= 1.0));
594		✗	return aggressiveness * new + (1.0 - aggressiveness) * old;
595			}
596
597		✗	static inline int bypass_channel(DynamicAudioNormalizerContext *s, AVFrame *frame, int ch)
598			{
599		✗	enum AVChannel channel = av_channel_layout_channel_from_index(&frame->ch_layout, ch);
600
601		✗	return av_channel_layout_index_from_channel(&s->ch_layout, channel) < 0;
602			}
603
604		✗	static void perform_dc_correction(DynamicAudioNormalizerContext *s, AVFrame *frame)
605			{
606		✗	const double diff = 1.0 / frame->nb_samples;
607		✗	int is_first_frame = cqueue_empty(s->gain_history_original[0]);
608
609		✗	for (int c = 0; c < s->channels; c++) {
610		✗	const int bypass = bypass_channel(s, frame, c);
611		✗	double *dst_ptr = (double *)frame->extended_data[c];
612		✗	double current_average_value = 0.0;
613			double prev_value;
614
615		✗	for (int i = 0; i < frame->nb_samples; i++)
616		✗	current_average_value += dst_ptr[i] * diff;
617
618		✗	prev_value = is_first_frame ? current_average_value : s->dc_correction_value[c];
619		✗	s->dc_correction_value[c] = is_first_frame ? current_average_value : update_value(current_average_value, s->dc_correction_value[c], 0.1);
620
621		✗	for (int i = 0; i < frame->nb_samples && !bypass; i++) {
622		✗	dst_ptr[i] -= fade(prev_value, s->dc_correction_value[c], i, frame->nb_samples);
623			}
624			}
625		✗	}
626
627		✗	static double setup_compress_thresh(double threshold)
628			{
629		✗	if ((threshold > DBL_EPSILON) && (threshold < (1.0 - DBL_EPSILON))) {
630		✗	double current_threshold = threshold;
631		✗	double step_size = 1.0;
632
633		✗	while (step_size > DBL_EPSILON) {
634		✗	while ((llrint((current_threshold + step_size) * (UINT64_C(1) << 63)) >
635		✗	llrint(current_threshold * (UINT64_C(1) << 63))) &&
636		✗	(bound(current_threshold + step_size, 1.0) <= threshold)) {
637		✗	current_threshold += step_size;
638			}
639
640		✗	step_size /= 2.0;
641			}
642
643		✗	return current_threshold;
644			} else {
645		✗	return threshold;
646			}
647			}
648
649		✗	static double compute_frame_std_dev(DynamicAudioNormalizerContext *s,
650			AVFrame *frame, int channel)
651			{
652		✗	double variance = 0.0;
653
654		✗	if (channel == -1) {
655		✗	for (int c = 0; c < s->channels; c++) {
656		✗	const double *data_ptr = (double *)frame->extended_data[c];
657
658		✗	for (int i = 0; i < frame->nb_samples; i++) {
659		✗	variance += pow_2(data_ptr[i]); // Assume that MEAN is *zero*
660			}
661			}
662		✗	variance /= (s->channels * frame->nb_samples) - 1;
663			} else {
664		✗	const double *data_ptr = (double *)frame->extended_data[channel];
665
666		✗	for (int i = 0; i < frame->nb_samples; i++) {
667		✗	variance += pow_2(data_ptr[i]); // Assume that MEAN is *zero*
668			}
669		✗	variance /= frame->nb_samples - 1;
670			}
671
672		✗	return fmax(sqrt(variance), DBL_EPSILON);
673			}
674
675		✗	static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame)
676			{
677		✗	int is_first_frame = cqueue_empty(s->gain_history_original[0]);
678
679		✗	if (s->channels_coupled) {
680		✗	const double standard_deviation = compute_frame_std_dev(s, frame, -1);
681		✗	const double current_threshold = fmin(1.0, s->compress_factor * standard_deviation);
682
683		✗	const double prev_value = is_first_frame ? current_threshold : s->compress_threshold[0];
684			double prev_actual_thresh, curr_actual_thresh;
685		✗	s->compress_threshold[0] = is_first_frame ? current_threshold : update_value(current_threshold, s->compress_threshold[0], (1.0/3.0));
686
687		✗	prev_actual_thresh = setup_compress_thresh(prev_value);
688		✗	curr_actual_thresh = setup_compress_thresh(s->compress_threshold[0]);
689
690		✗	for (int c = 0; c < s->channels; c++) {
691		✗	double *const dst_ptr = (double *)frame->extended_data[c];
692		✗	const int bypass = bypass_channel(s, frame, c);
693
694		✗	if (bypass)
695		✗	continue;
696
697		✗	for (int i = 0; i < frame->nb_samples; i++) {
698		✗	const double localThresh = fade(prev_actual_thresh, curr_actual_thresh, i, frame->nb_samples);
699		✗	dst_ptr[i] = copysign(bound(localThresh, fabs(dst_ptr[i])), dst_ptr[i]);
700			}
701			}
702			} else {
703		✗	for (int c = 0; c < s->channels; c++) {
704		✗	const int bypass = bypass_channel(s, frame, c);
705		✗	const double standard_deviation = compute_frame_std_dev(s, frame, c);
706		✗	const double current_threshold = setup_compress_thresh(fmin(1.0, s->compress_factor * standard_deviation));
707		✗	const double prev_value = is_first_frame ? current_threshold : s->compress_threshold[c];
708			double prev_actual_thresh, curr_actual_thresh;
709			double *dst_ptr;
710
711		✗	s->compress_threshold[c] = is_first_frame ? current_threshold : update_value(current_threshold, s->compress_threshold[c], 1.0/3.0);
712
713		✗	prev_actual_thresh = setup_compress_thresh(prev_value);
714		✗	curr_actual_thresh = setup_compress_thresh(s->compress_threshold[c]);
715
716		✗	dst_ptr = (double *)frame->extended_data[c];
717		✗	for (int i = 0; i < frame->nb_samples && !bypass; i++) {
718		✗	const double localThresh = fade(prev_actual_thresh, curr_actual_thresh, i, frame->nb_samples);
719		✗	dst_ptr[i] = copysign(bound(localThresh, fabs(dst_ptr[i])), dst_ptr[i]);
720			}
721			}
722			}
723		✗	}
724
725		✗	static int analyze_frame(AVFilterContext *ctx, AVFilterLink *outlink, AVFrame **frame)
726			{
727		✗	FilterLink *outl = ff_filter_link(outlink);
728		✗	DynamicAudioNormalizerContext *s = ctx->priv;
729			AVFrame *analyze_frame;
730
731		✗	if (s->dc_correction || s->compress_factor > DBL_EPSILON) {
732			int ret;
733
734		✗	if (!av_frame_is_writable(*frame)) {
735		✗	AVFrame *out = ff_get_audio_buffer(outlink, (*frame)->nb_samples);
736
737		✗	if (!out) {
738		✗	av_frame_free(frame);
739		✗	return AVERROR(ENOMEM);
740			}
741		✗	ret = av_frame_copy_props(out, *frame);
742		✗	if (ret < 0) {
743		✗	av_frame_free(frame);
744		✗	av_frame_free(&out);
745		✗	return ret;
746			}
747		✗	ret = av_frame_copy(out, *frame);
748		✗	if (ret < 0) {
749		✗	av_frame_free(frame);
750		✗	av_frame_free(&out);
751		✗	return ret;
752			}
753
754		✗	av_frame_free(frame);
755		✗	*frame = out;
756			}
757			}
758
759		✗	if (s->dc_correction)
760		✗	perform_dc_correction(s, *frame);
761
762		✗	if (s->compress_factor > DBL_EPSILON)
763		✗	perform_compression(s, *frame);
764
765		✗	if (s->frame_len != s->sample_advance) {
766		✗	const int offset = s->frame_len - s->sample_advance;
767
768		✗	for (int c = 0; c < s->channels; c++) {
769		✗	double *src = (double *)s->window->extended_data[c];
770
771		✗	memmove(src, &src[s->sample_advance], offset * sizeof(double));
772		✗	memcpy(&src[offset], (*frame)->extended_data[c], (*frame)->nb_samples * sizeof(double));
773		✗	memset(&src[offset + (*frame)->nb_samples], 0, (s->sample_advance - (*frame)->nb_samples) * sizeof(double));
774			}
775
776		✗	analyze_frame = s->window;
777			} else {
778		✗	av_samples_copy(s->window->extended_data, (*frame)->extended_data, 0, 0,
779		✗	FFMIN(s->frame_len, (*frame)->nb_samples), (*frame)->ch_layout.nb_channels, (*frame)->format);
780		✗	analyze_frame = *frame;
781			}
782
783		✗	s->var_values[VAR_SN] = outl->sample_count_in;
784		✗	s->var_values[VAR_T] = s->var_values[VAR_SN] * (double)1/outlink->sample_rate;
785
786		✗	if (s->channels_coupled) {
787		✗	const local_gain gain = get_max_local_gain(s, analyze_frame, -1);
788		✗	for (int c = 0; c < s->channels; c++)
789		✗	update_gain_history(s, c, gain);
790			} else {
791		✗	ff_filter_execute(ctx, update_gain_histories, analyze_frame, NULL,
792		✗	FFMIN(s->channels, ff_filter_get_nb_threads(ctx)));
793			}
794
795		✗	return 0;
796			}
797
798		✗	static void amplify_channel(DynamicAudioNormalizerContext *s, AVFrame *in,
799			AVFrame *frame, int enabled, int c)
800			{
801		✗	const int bypass = bypass_channel(s, frame, c);
802		✗	const double *src_ptr = (const double *)in->extended_data[c];
803		✗	double *dst_ptr = (double *)frame->extended_data[c];
804			double current_amplification_factor;
805
806		✗	cqueue_dequeue(s->gain_history_smoothed[c], &current_amplification_factor);
807
808		✗	for (int i = 0; i < frame->nb_samples && enabled && !bypass; i++) {
809		✗	const double amplification_factor = fade(s->prev_amplification_factor[c],
810			current_amplification_factor, i,
811			frame->nb_samples);
812
813		✗	dst_ptr[i] = src_ptr[i] * amplification_factor;
814			}
815
816		✗	s->prev_amplification_factor[c] = current_amplification_factor;
817		✗	}
818
819		✗	static int amplify_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
820			{
821		✗	DynamicAudioNormalizerContext *s = ctx->priv;
822		✗	ThreadData *td = arg;
823		✗	AVFrame *out = td->out;
824		✗	AVFrame *in = td->in;
825		✗	const int enabled = td->enabled;
826		✗	const int channels = s->channels;
827		✗	const int start = ff_slice_pos(channels, jobnr, nb_jobs);
828		✗	const int end = ff_slice_pos(channels, jobnr + 1, nb_jobs);
829
830		✗	for (int ch = start; ch < end; ch++)
831		✗	amplify_channel(s, in, out, enabled, ch);
832
833		✗	return 0;
834			}
835
836		✗	static int filter_frame(AVFilterLink *inlink, AVFrame *in)
837			{
838		✗	AVFilterContext *ctx = inlink->dst;
839		✗	DynamicAudioNormalizerContext *s = ctx->priv;
840		✗	AVFilterLink *outlink = ctx->outputs[0];
841			ThreadData td;
842			int ret;
843
844		✗	while (((s->queue.available >= s->filter_size) ||
845		✗	(s->eof && s->queue.available)) &&
846		✗	!cqueue_empty(s->gain_history_smoothed[0])) {
847		✗	AVFrame *in = ff_bufqueue_get(&s->queue);
848			AVFrame *out;
849			double is_enabled;
850
851		✗	cqueue_dequeue(s->is_enabled, &is_enabled);
852
853		✗	if (av_frame_is_writable(in)) {
854		✗	out = in;
855			} else {
856		✗	out = ff_get_audio_buffer(outlink, in->nb_samples);
857		✗	if (!out) {
858		✗	av_frame_free(&in);
859		✗	return AVERROR(ENOMEM);
860			}
861		✗	av_frame_copy_props(out, in);
862			}
863
864		✗	td.in = in;
865		✗	td.out = out;
866		✗	td.enabled = is_enabled > 0.;
867		✗	ff_filter_execute(ctx, amplify_channels, &td, NULL,
868		✗	FFMIN(s->channels, ff_filter_get_nb_threads(ctx)));
869
870		✗	s->pts = out->pts + av_rescale_q(out->nb_samples, av_make_q(1, outlink->sample_rate),
871			outlink->time_base);
872		✗	if (out != in)
873		✗	av_frame_free(&in);
874		✗	ret = ff_filter_frame(outlink, out);
875		✗	if (ret < 0)
876		✗	return ret;
877			}
878
879		✗	ret = analyze_frame(ctx, outlink, &in);
880		✗	if (ret < 0)
881		✗	return ret;
882		✗	if (!s->eof) {
883		✗	ff_bufqueue_add(ctx, &s->queue, in);
884		✗	cqueue_enqueue(s->is_enabled, !ctx->is_disabled);
885			} else {
886		✗	av_frame_free(&in);
887			}
888
889		✗	return 1;
890			}
891
892		✗	static int flush_buffer(DynamicAudioNormalizerContext *s, AVFilterLink *inlink,
893			AVFilterLink *outlink)
894			{
895		✗	AVFrame *out = ff_get_audio_buffer(outlink, s->sample_advance);
896
897		✗	if (!out)
898		✗	return AVERROR(ENOMEM);
899
900		✗	for (int c = 0; c < s->channels; c++) {
901		✗	double *dst_ptr = (double *)out->extended_data[c];
902
903		✗	for (int i = 0; i < out->nb_samples; i++) {
904		✗	dst_ptr[i] = s->alt_boundary_mode ? DBL_EPSILON : ((s->target_rms > DBL_EPSILON) ? fmin(s->peak_value, s->target_rms) : s->peak_value);
905		✗	if (s->dc_correction) {
906		✗	dst_ptr[i] *= ((i % 2) == 1) ? -1 : 1;
907		✗	dst_ptr[i] += s->dc_correction_value[c];
908			}
909			}
910			}
911
912		✗	return filter_frame(inlink, out);
913			}
914
915		✗	static int flush(AVFilterLink *outlink)
916			{
917		✗	AVFilterContext *ctx = outlink->src;
918		✗	AVFilterLink *inlink = ctx->inputs[0];
919		✗	DynamicAudioNormalizerContext *s = ctx->priv;
920
921		✗	while (s->eof && cqueue_empty(s->gain_history_smoothed[0])) {
922		✗	for (int c = 0; c < s->channels; c++)
923		✗	update_gain_history(s, c, (local_gain){ cqueue_peek(s->gain_history_original[c], 0), 1.0});
924			}
925
926		✗	return flush_buffer(s, inlink, outlink);
927			}
928
929		✗	static int activate(AVFilterContext *ctx)
930			{
931		✗	AVFilterLink *inlink = ctx->inputs[0];
932		✗	AVFilterLink *outlink = ctx->outputs[0];
933		✗	DynamicAudioNormalizerContext *s = ctx->priv;
934		✗	AVFrame *in = NULL;
935		✗	int ret = 0, status;
936			int64_t pts;
937
938		✗	ret = av_channel_layout_copy(&s->ch_layout, &inlink->ch_layout);
939		✗	if (ret < 0)
940		✗	return ret;
941		✗	if (strcmp(s->channels_to_filter, "all"))
942		✗	av_channel_layout_from_string(&s->ch_layout, s->channels_to_filter);
943
944		✗	FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
945
946		✗	if (!s->eof) {
947		✗	ret = ff_inlink_consume_samples(inlink, s->sample_advance, s->sample_advance, &in);
948		✗	if (ret < 0)
949		✗	return ret;
950		✗	if (ret > 0) {
951		✗	ret = filter_frame(inlink, in);
952		✗	if (ret <= 0)
953		✗	return ret;
954			}
955
956		✗	if (ff_inlink_check_available_samples(inlink, s->sample_advance) > 0) {
957		✗	ff_filter_set_ready(ctx, 10);
958		✗	return 0;
959			}
960			}
961
962		✗	if (!s->eof && ff_inlink_acknowledge_status(inlink, &status, &pts)) {
963		✗	if (status == AVERROR_EOF)
964		✗	s->eof = 1;
965			}
966
967		✗	if (s->eof && s->queue.available)
968		✗	return flush(outlink);
969
970		✗	if (s->eof && !s->queue.available) {
971		✗	ff_outlink_set_status(outlink, AVERROR_EOF, s->pts);
972		✗	return 0;
973			}
974
975		✗	if (!s->eof)
976		✗	FF_FILTER_FORWARD_WANTED(outlink, inlink);
977
978		✗	return FFERROR_NOT_READY;
979			}
980
981		✗	static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
982			char *res, int res_len, int flags)
983			{
984		✗	DynamicAudioNormalizerContext *s = ctx->priv;
985		✗	AVFilterLink *inlink = ctx->inputs[0];
986		✗	int prev_filter_size = s->filter_size;
987			int ret;
988
989		✗	ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
990		✗	if (ret < 0)
991		✗	return ret;
992
993		✗	s->filter_size |= 1;
994		✗	if (prev_filter_size != s->filter_size) {
995		✗	init_gaussian_filter(s);
996
997		✗	for (int c = 0; c < s->channels; c++) {
998		✗	cqueue_resize(s->gain_history_original[c], s->filter_size);
999		✗	cqueue_resize(s->gain_history_minimum[c], s->filter_size);
1000		✗	cqueue_resize(s->threshold_history[c], s->filter_size);
1001			}
1002			}
1003
1004		✗	s->frame_len = frame_size(inlink->sample_rate, s->frame_len_msec);
1005		✗	s->sample_advance = FFMAX(1, lrint(s->frame_len * (1. - s->overlap)));
1006		✗	if (s->expr_str) {
1007		✗	ret = av_expr_parse(&s->expr, s->expr_str, var_names, NULL, NULL,
1008			NULL, NULL, 0, ctx);
1009		✗	if (ret < 0)
1010		✗	return ret;
1011			}
1012		✗	return 0;
1013			}
1014
1015			static const AVFilterPad avfilter_af_dynaudnorm_inputs[] = {
1016			{
1017			.name = "default",
1018			.type = AVMEDIA_TYPE_AUDIO,
1019			.config_props = config_input,
1020			},
1021			};
1022
1023			const FFFilter ff_af_dynaudnorm = {
1024			.p.name = "dynaudnorm",
1025			.p.description = NULL_IF_CONFIG_SMALL("Dynamic Audio Normalizer."),
1026			.p.priv_class = &dynaudnorm_class,
1027			.p.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
1028			AVFILTER_FLAG_SLICE_THREADS,
1029			.priv_size = sizeof(DynamicAudioNormalizerContext),
1030			.init = init,
1031			.uninit = uninit,
1032			.activate = activate,
1033			FILTER_INPUTS(avfilter_af_dynaudnorm_inputs),
1034			FILTER_OUTPUTS(ff_audio_default_filterpad),
1035			FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBLP),
1036			.process_command = process_command,
1037			};
1038