FFmpeg coverage

Line	Branch	Exec	Source
1			/*
2			* Copyright (c) 2014 - 2021 Jason Jang
3			* Copyright (c) 2021 Paul B Mahol
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 License
9			* 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
15			* GNU Lesser General Public License for more details.
16			*
17			* You should have received a copy of the GNU Lesser General Public License
18			* along with FFmpeg; if not, write to the Free Software Foundation, Inc.,
19			* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20			*/
21
22			#include "libavutil/mem.h"
23			#include "libavutil/opt.h"
24			#include "libavutil/tx.h"
25			#include "audio.h"
26			#include "avfilter.h"
27			#include "filters.h"
28			#include "internal.h"
29
30			typedef struct AudioPsyClipContext {
31			const AVClass *class;
32
33			double level_in;
34			double level_out;
35			double clip_level;
36			double adaptive;
37			int auto_level;
38			int diff_only;
39			int iterations;
40			char *protections_str;
41			double *protections;
42
43			int num_psy_bins;
44			int fft_size;
45			int overlap;
46			int channels;
47
48			int spread_table_rows;
49			int *spread_table_index;
50			int (*spread_table_range)[2];
51			float *window, *inv_window, *spread_table, *margin_curve;
52
53			AVFrame *in;
54			AVFrame *in_buffer;
55			AVFrame *in_frame;
56			AVFrame *out_dist_frame;
57			AVFrame *windowed_frame;
58			AVFrame *clipping_delta;
59			AVFrame *spectrum_buf;
60			AVFrame *mask_curve;
61
62			AVTXContext **tx_ctx;
63			av_tx_fn tx_fn;
64			AVTXContext **itx_ctx;
65			av_tx_fn itx_fn;
66			} AudioPsyClipContext;
67
68			#define OFFSET(x) offsetof(AudioPsyClipContext, x)
69			#define FLAGS AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM | AV_OPT_FLAG_RUNTIME_PARAM
70
71			static const AVOption apsyclip_options[] = {
72			{ "level_in", "set input level", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625, 64, FLAGS },
73			{ "level_out", "set output level", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625, 64, FLAGS },
74			{ "clip", "set clip level", OFFSET(clip_level), AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625, 1, FLAGS },
75			{ "diff", "enable difference", OFFSET(diff_only), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, FLAGS },
76			{ "adaptive", "set adaptive distortion", OFFSET(adaptive), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 1, FLAGS },
77			{ "iterations", "set iterations", OFFSET(iterations), AV_OPT_TYPE_INT, {.i64=10}, 1, 20, FLAGS },
78			{ "level", "set auto level", OFFSET(auto_level), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, FLAGS },
79			{NULL}
80			};
81
82			AVFILTER_DEFINE_CLASS(apsyclip);
83
84		✗	static void generate_hann_window(float *window, float *inv_window, int size)
85			{
86		✗	for (int i = 0; i < size; i++) {
87		✗	float value = 0.5f * (1.f - cosf(2.f * M_PI * i / size));
88
89		✗	window[i] = value;
90			// 1/window to calculate unwindowed peak.
91		✗	inv_window[i] = value > 0.1f ? 1.f / value : 0.f;
92			}
93		✗	}
94
95		✗	static void set_margin_curve(AudioPsyClipContext *s,
96			const int (*points)[2], int num_points, int sample_rate)
97			{
98		✗	int j = 0;
99
100		✗	s->margin_curve[0] = points[0][1];
101
102		✗	for (int i = 0; i < num_points - 1; i++) {
103		✗	while (j < s->fft_size / 2 + 1 && j * sample_rate / s->fft_size < points[i + 1][0]) {
104			// linearly interpolate between points
105		✗	int binHz = j * sample_rate / s->fft_size;
106		✗	s->margin_curve[j] = points[i][1] + (binHz - points[i][0]) * (points[i + 1][1] - points[i][1]) / (points[i + 1][0] - points[i][0]);
107		✗	j++;
108			}
109			}
110			// handle bins after the last point
111		✗	while (j < s->fft_size / 2 + 1) {
112		✗	s->margin_curve[j] = points[num_points - 1][1];
113		✗	j++;
114			}
115
116			// convert margin curve to linear amplitude scale
117		✗	for (j = 0; j < s->fft_size / 2 + 1; j++)
118		✗	s->margin_curve[j] = powf(10.f, s->margin_curve[j] / 20.f);
119		✗	}
120
121		✗	static void generate_spread_table(AudioPsyClipContext *s)
122			{
123			// Calculate tent-shape function in log-log scale.
124
125			// As an optimization, only consider bins close to "bin"
126			// This reduces the number of multiplications needed in calculate_mask_curve
127			// The masking contribution at faraway bins is negligeable
128
129			// Another optimization to save memory and speed up the calculation of the
130			// spread table is to calculate and store only 2 spread functions per
131			// octave, and reuse the same spread function for multiple bins.
132		✗	int table_index = 0;
133		✗	int bin = 0;
134		✗	int increment = 1;
135
136		✗	while (bin < s->num_psy_bins) {
137		✗	float sum = 0;
138		✗	int base_idx = table_index * s->num_psy_bins;
139		✗	int start_bin = bin * 3 / 4;
140		✗	int end_bin = FFMIN(s->num_psy_bins, ((bin + 1) * 4 + 2) / 3);
141			int next_bin;
142
143		✗	for (int j = start_bin; j < end_bin; j++) {
144			// add 0.5 so i=0 doesn't get log(0)
145		✗	float rel_idx_log = FFABS(logf((j + 0.5f) / (bin + 0.5f)));
146			float value;
147		✗	if (j >= bin) {
148			// mask up
149		✗	value = expf(-rel_idx_log * 40.f);
150			} else {
151			// mask down
152		✗	value = expf(-rel_idx_log * 80.f);
153			}
154			// the spreading function is centred in the row
155		✗	sum += value;
156		✗	s->spread_table[base_idx + s->num_psy_bins / 2 + j - bin] = value;
157			}
158			// now normalize it
159		✗	for (int j = start_bin; j < end_bin; j++) {
160		✗	s->spread_table[base_idx + s->num_psy_bins / 2 + j - bin] /= sum;
161			}
162
163		✗	s->spread_table_range[table_index][0] = start_bin - bin;
164		✗	s->spread_table_range[table_index][1] = end_bin - bin;
165
166		✗	if (bin <= 1) {
167		✗	next_bin = bin + 1;
168			} else {
169		✗	if ((bin & (bin - 1)) == 0) {
170			// power of 2
171		✗	increment = bin / 2;
172			}
173
174		✗	next_bin = bin + increment;
175			}
176
177			// set bins between "bin" and "next_bin" to use this table_index
178		✗	for (int i = bin; i < next_bin; i++)
179		✗	s->spread_table_index[i] = table_index;
180
181		✗	bin = next_bin;
182		✗	table_index++;
183			}
184		✗	}
185
186		✗	static int config_input(AVFilterLink *inlink)
187			{
188		✗	AVFilterContext *ctx = inlink->dst;
189		✗	AudioPsyClipContext *s = ctx->priv;
190			static const int points[][2] = { {0,14}, {125,14}, {250,16}, {500,18}, {1000,20}, {2000,20}, {4000,20}, {8000,17}, {16000,14}, {20000,-10} };
191			static const int num_points = 10;
192		✗	float scale = 1.f;
193			int ret;
194
195		✗	s->fft_size = inlink->sample_rate > 100000 ? 1024 : inlink->sample_rate > 50000 ? 512 : 256;
196		✗	s->overlap = s->fft_size / 4;
197
198			// The psy masking calculation is O(n^2),
199			// so skip it for frequencies not covered by base sampling rantes (i.e. 44k)
200		✗	if (inlink->sample_rate <= 50000) {
201		✗	s->num_psy_bins = s->fft_size / 2;
202		✗	} else if (inlink->sample_rate <= 100000) {
203		✗	s->num_psy_bins = s->fft_size / 4;
204			} else {
205		✗	s->num_psy_bins = s->fft_size / 8;
206			}
207
208		✗	s->window = av_calloc(s->fft_size, sizeof(*s->window));
209		✗	s->inv_window = av_calloc(s->fft_size, sizeof(*s->inv_window));
210		✗	if (!s->window || !s->inv_window)
211		✗	return AVERROR(ENOMEM);
212
213		✗	s->in_buffer = ff_get_audio_buffer(inlink, s->fft_size * 2);
214		✗	s->in_frame = ff_get_audio_buffer(inlink, s->fft_size * 2);
215		✗	s->out_dist_frame = ff_get_audio_buffer(inlink, s->fft_size * 2);
216		✗	s->windowed_frame = ff_get_audio_buffer(inlink, s->fft_size * 2);
217		✗	s->clipping_delta = ff_get_audio_buffer(inlink, s->fft_size * 2);
218		✗	s->spectrum_buf = ff_get_audio_buffer(inlink, s->fft_size * 2);
219		✗	s->mask_curve = ff_get_audio_buffer(inlink, s->fft_size / 2 + 1);
220		✗	if (!s->in_buffer || !s->in_frame ||
221		✗	!s->out_dist_frame || !s->windowed_frame ||
222		✗	!s->clipping_delta || !s->spectrum_buf || !s->mask_curve)
223		✗	return AVERROR(ENOMEM);
224
225		✗	generate_hann_window(s->window, s->inv_window, s->fft_size);
226
227		✗	s->margin_curve = av_calloc(s->fft_size / 2 + 1, sizeof(*s->margin_curve));
228		✗	if (!s->margin_curve)
229		✗	return AVERROR(ENOMEM);
230
231		✗	s->spread_table_rows = av_log2(s->num_psy_bins) * 2;
232		✗	s->spread_table = av_calloc(s->spread_table_rows * s->num_psy_bins, sizeof(*s->spread_table));
233		✗	if (!s->spread_table)
234		✗	return AVERROR(ENOMEM);
235
236		✗	s->spread_table_range = av_calloc(s->spread_table_rows * 2, sizeof(*s->spread_table_range));
237		✗	if (!s->spread_table_range)
238		✗	return AVERROR(ENOMEM);
239
240		✗	s->spread_table_index = av_calloc(s->num_psy_bins, sizeof(*s->spread_table_index));
241		✗	if (!s->spread_table_index)
242		✗	return AVERROR(ENOMEM);
243
244		✗	set_margin_curve(s, points, num_points, inlink->sample_rate);
245
246		✗	generate_spread_table(s);
247
248		✗	s->channels = inlink->ch_layout.nb_channels;
249
250		✗	s->tx_ctx = av_calloc(s->channels, sizeof(*s->tx_ctx));
251		✗	s->itx_ctx = av_calloc(s->channels, sizeof(*s->itx_ctx));
252		✗	if (!s->tx_ctx || !s->itx_ctx)
253		✗	return AVERROR(ENOMEM);
254
255		✗	for (int ch = 0; ch < s->channels; ch++) {
256		✗	ret = av_tx_init(&s->tx_ctx[ch], &s->tx_fn, AV_TX_FLOAT_FFT, 0, s->fft_size, &scale, 0);
257		✗	if (ret < 0)
258		✗	return ret;
259
260		✗	ret = av_tx_init(&s->itx_ctx[ch], &s->itx_fn, AV_TX_FLOAT_FFT, 1, s->fft_size, &scale, 0);
261		✗	if (ret < 0)
262		✗	return ret;
263			}
264
265		✗	return 0;
266			}
267
268		✗	static void apply_window(AudioPsyClipContext *s,
269			const float *in_frame, float *out_frame, const int add_to_out_frame)
270			{
271		✗	const float *window = s->window;
272
273		✗	for (int i = 0; i < s->fft_size; i++) {
274		✗	if (add_to_out_frame) {
275		✗	out_frame[i] += in_frame[i] * window[i];
276			} else {
277		✗	out_frame[i] = in_frame[i] * window[i];
278			}
279			}
280		✗	}
281
282		✗	static void calculate_mask_curve(AudioPsyClipContext *s,
283			const float *spectrum, float *mask_curve)
284			{
285		✗	for (int i = 0; i < s->fft_size / 2 + 1; i++)
286		✗	mask_curve[i] = 0;
287
288		✗	for (int i = 0; i < s->num_psy_bins; i++) {
289			int base_idx, start_bin, end_bin, table_idx;
290			float magnitude;
291			int range[2];
292
293		✗	if (i == 0) {
294		✗	magnitude = FFABS(spectrum[0]);
295		✗	} else if (i == s->fft_size / 2) {
296		✗	magnitude = FFABS(spectrum[s->fft_size]);
297			} else {
298			// Because the input signal is real, the + and - frequencies are redundant.
299			// Multiply the magnitude by 2 to simulate adding up the + and - frequencies.
300		✗	magnitude = hypotf(spectrum[2 * i], spectrum[2 * i + 1]) * 2;
301			}
302
303		✗	table_idx = s->spread_table_index[i];
304		✗	range[0] = s->spread_table_range[table_idx][0];
305		✗	range[1] = s->spread_table_range[table_idx][1];
306		✗	base_idx = table_idx * s->num_psy_bins;
307		✗	start_bin = FFMAX(0, i + range[0]);
308		✗	end_bin = FFMIN(s->num_psy_bins, i + range[1]);
309
310		✗	for (int j = start_bin; j < end_bin; j++)
311		✗	mask_curve[j] += s->spread_table[base_idx + s->num_psy_bins / 2 + j - i] * magnitude;
312			}
313
314			// for ultrasonic frequencies, skip the O(n^2) spread calculation and just copy the magnitude
315		✗	for (int i = s->num_psy_bins; i < s->fft_size / 2 + 1; i++) {
316			float magnitude;
317		✗	if (i == s->fft_size / 2) {
318		✗	magnitude = FFABS(spectrum[s->fft_size]);
319			} else {
320			// Because the input signal is real, the + and - frequencies are redundant.
321			// Multiply the magnitude by 2 to simulate adding up the + and - frequencies.
322		✗	magnitude = hypotf(spectrum[2 * i], spectrum[2 * i + 1]) * 2;
323			}
324
325		✗	mask_curve[i] = magnitude;
326			}
327
328		✗	for (int i = 0; i < s->fft_size / 2 + 1; i++)
329		✗	mask_curve[i] = mask_curve[i] / s->margin_curve[i];
330		✗	}
331
332		✗	static void clip_to_window(AudioPsyClipContext *s,
333			const float *windowed_frame, float *clipping_delta, float delta_boost)
334			{
335		✗	const float *window = s->window;
336
337		✗	for (int i = 0; i < s->fft_size; i++) {
338		✗	const float limit = s->clip_level * window[i];
339		✗	const float effective_value = windowed_frame[i] + clipping_delta[i];
340
341		✗	if (effective_value > limit) {
342		✗	clipping_delta[i] += (limit - effective_value) * delta_boost;
343		✗	} else if (effective_value < -limit) {
344		✗	clipping_delta[i] += (-limit - effective_value) * delta_boost;
345			}
346			}
347		✗	}
348
349		✗	static void limit_clip_spectrum(AudioPsyClipContext *s,
350			float *clip_spectrum, const float *mask_curve)
351			{
352			// bin 0
353		✗	float relative_distortion_level = FFABS(clip_spectrum[0]) / mask_curve[0];
354
355		✗	if (relative_distortion_level > 1.f)
356		✗	clip_spectrum[0] /= relative_distortion_level;
357
358			// bin 1..N/2-1
359		✗	for (int i = 1; i < s->fft_size / 2; i++) {
360		✗	float real = clip_spectrum[i * 2];
361		✗	float imag = clip_spectrum[i * 2 + 1];
362			// Because the input signal is real, the + and - frequencies are redundant.
363			// Multiply the magnitude by 2 to simulate adding up the + and - frequencies.
364		✗	relative_distortion_level = hypotf(real, imag) * 2 / mask_curve[i];
365		✗	if (relative_distortion_level > 1.0) {
366		✗	clip_spectrum[i * 2] /= relative_distortion_level;
367		✗	clip_spectrum[i * 2 + 1] /= relative_distortion_level;
368		✗	clip_spectrum[s->fft_size * 2 - i * 2] /= relative_distortion_level;
369		✗	clip_spectrum[s->fft_size * 2 - i * 2 + 1] /= relative_distortion_level;
370			}
371			}
372			// bin N/2
373		✗	relative_distortion_level = FFABS(clip_spectrum[s->fft_size]) / mask_curve[s->fft_size / 2];
374		✗	if (relative_distortion_level > 1.f)
375		✗	clip_spectrum[s->fft_size] /= relative_distortion_level;
376		✗	}
377
378		✗	static void r2c(float *buffer, int size)
379			{
380		✗	for (int i = size - 1; i >= 0; i--)
381		✗	buffer[2 * i] = buffer[i];
382
383		✗	for (int i = size - 1; i >= 0; i--)
384		✗	buffer[2 * i + 1] = 0.f;
385		✗	}
386
387		✗	static void c2r(float *buffer, int size)
388			{
389		✗	for (int i = 0; i < size; i++)
390		✗	buffer[i] = buffer[2 * i];
391
392		✗	for (int i = 0; i < size; i++)
393		✗	buffer[i + size] = 0.f;
394		✗	}
395
396		✗	static void feed(AVFilterContext *ctx, int ch,
397			const float *in_samples, float *out_samples, int diff_only,
398			float *in_frame, float *out_dist_frame,
399			float *windowed_frame, float *clipping_delta,
400			float *spectrum_buf, float *mask_curve)
401			{
402		✗	AudioPsyClipContext *s = ctx->priv;
403		✗	const float clip_level_inv = 1.f / s->clip_level;
404		✗	const float level_out = s->level_out;
405		✗	float orig_peak = 0;
406			float peak;
407
408			// shift in/out buffers
409		✗	for (int i = 0; i < s->fft_size - s->overlap; i++) {
410		✗	in_frame[i] = in_frame[i + s->overlap];
411		✗	out_dist_frame[i] = out_dist_frame[i + s->overlap];
412			}
413
414		✗	for (int i = 0; i < s->overlap; i++) {
415		✗	in_frame[i + s->fft_size - s->overlap] = in_samples[i];
416		✗	out_dist_frame[i + s->fft_size - s->overlap] = 0.f;
417			}
418
419		✗	apply_window(s, in_frame, windowed_frame, 0);
420		✗	r2c(windowed_frame, s->fft_size);
421		✗	s->tx_fn(s->tx_ctx[ch], spectrum_buf, windowed_frame, sizeof(AVComplexFloat));
422		✗	c2r(windowed_frame, s->fft_size);
423		✗	calculate_mask_curve(s, spectrum_buf, mask_curve);
424
425			// It would be easier to calculate the peak from the unwindowed input.
426			// This is just for consistency with the clipped peak calculateion
427			// because the inv_window zeros out samples on the edge of the window.
428		✗	for (int i = 0; i < s->fft_size; i++)
429		✗	orig_peak = FFMAX(orig_peak, FFABS(windowed_frame[i] * s->inv_window[i]));
430		✗	orig_peak *= clip_level_inv;
431		✗	peak = orig_peak;
432
433			// clear clipping_delta
434		✗	for (int i = 0; i < s->fft_size * 2; i++)
435		✗	clipping_delta[i] = 0.f;
436
437			// repeat clipping-filtering process a few times to control both the peaks and the spectrum
438		✗	for (int i = 0; i < s->iterations; i++) {
439		✗	float mask_curve_shift = 1.122f; // 1.122 is 1dB
440			// The last 1/3 of rounds have boosted delta to help reach the peak target faster
441		✗	float delta_boost = 1.f;
442		✗	if (i >= s->iterations - s->iterations / 3) {
443			// boosting the delta when largs peaks are still present is dangerous
444		✗	if (peak < 2.f)
445		✗	delta_boost = 2.f;
446			}
447
448		✗	clip_to_window(s, windowed_frame, clipping_delta, delta_boost);
449
450		✗	r2c(clipping_delta, s->fft_size);
451		✗	s->tx_fn(s->tx_ctx[ch], spectrum_buf, clipping_delta, sizeof(AVComplexFloat));
452
453		✗	limit_clip_spectrum(s, spectrum_buf, mask_curve);
454
455		✗	s->itx_fn(s->itx_ctx[ch], clipping_delta, spectrum_buf, sizeof(AVComplexFloat));
456		✗	c2r(clipping_delta, s->fft_size);
457
458		✗	for (int i = 0; i < s->fft_size; i++)
459		✗	clipping_delta[i] /= s->fft_size;
460
461		✗	peak = 0;
462		✗	for (int i = 0; i < s->fft_size; i++)
463		✗	peak = FFMAX(peak, FFABS((windowed_frame[i] + clipping_delta[i]) * s->inv_window[i]));
464		✗	peak *= clip_level_inv;
465
466			// Automatically adjust mask_curve as necessary to reach peak target
467		✗	if (orig_peak > 1.f && peak > 1.f) {
468		✗	float diff_achieved = orig_peak - peak;
469		✗	if (i + 1 < s->iterations - s->iterations / 3 && diff_achieved > 0) {
470		✗	float diff_needed = orig_peak - 1.f;
471		✗	float diff_ratio = diff_needed / diff_achieved;
472			// If a good amount of peak reduction was already achieved,
473			// don't shift the mask_curve by the full peak value
474			// On the other hand, if only a little peak reduction was achieved,
475			// don't shift the mask_curve by the enormous diff_ratio.
476		✗	diff_ratio = FFMIN(diff_ratio, peak);
477		✗	mask_curve_shift = FFMAX(mask_curve_shift, diff_ratio);
478			} else {
479			// If the peak got higher than the input or we are in the last 1/3 rounds,
480			// go back to the heavy-handed peak heuristic.
481		✗	mask_curve_shift = FFMAX(mask_curve_shift, peak);
482			}
483			}
484
485		✗	mask_curve_shift = 1.f + (mask_curve_shift - 1.f) * s->adaptive;
486
487			// Be less strict in the next iteration.
488			// This helps with peak control.
489		✗	for (int i = 0; i < s->fft_size / 2 + 1; i++)
490		✗	mask_curve[i] *= mask_curve_shift;
491			}
492
493			// do overlap & add
494		✗	apply_window(s, clipping_delta, out_dist_frame, 1);
495
496		✗	for (int i = 0; i < s->overlap; i++) {
497			// 4 times overlap with squared hanning window results in 1.5 time increase in amplitude
498		✗	if (!ctx->is_disabled) {
499		✗	out_samples[i] = out_dist_frame[i] / 1.5f;
500		✗	if (!diff_only)
501		✗	out_samples[i] += in_frame[i];
502		✗	if (s->auto_level)
503		✗	out_samples[i] *= clip_level_inv;
504		✗	out_samples[i] *= level_out;
505			} else {
506		✗	out_samples[i] = in_frame[i];
507			}
508			}
509		✗	}
510
511		✗	static int psy_channel(AVFilterContext *ctx, AVFrame *in, AVFrame *out, int ch)
512			{
513		✗	AudioPsyClipContext *s = ctx->priv;
514		✗	const float *src = (const float *)in->extended_data[ch];
515		✗	float *in_buffer = (float *)s->in_buffer->extended_data[ch];
516		✗	float *dst = (float *)out->extended_data[ch];
517
518		✗	for (int n = 0; n < s->overlap; n++)
519		✗	in_buffer[n] = src[n] * s->level_in;
520
521		✗	feed(ctx, ch, in_buffer, dst, s->diff_only,
522		✗	(float *)(s->in_frame->extended_data[ch]),
523		✗	(float *)(s->out_dist_frame->extended_data[ch]),
524		✗	(float *)(s->windowed_frame->extended_data[ch]),
525		✗	(float *)(s->clipping_delta->extended_data[ch]),
526		✗	(float *)(s->spectrum_buf->extended_data[ch]),
527		✗	(float *)(s->mask_curve->extended_data[ch]));
528
529		✗	return 0;
530			}
531
532		✗	static int psy_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
533			{
534		✗	AudioPsyClipContext *s = ctx->priv;
535		✗	AVFrame *out = arg;
536		✗	const int start = (out->ch_layout.nb_channels * jobnr) / nb_jobs;
537		✗	const int end = (out->ch_layout.nb_channels * (jobnr+1)) / nb_jobs;
538
539		✗	for (int ch = start; ch < end; ch++)
540		✗	psy_channel(ctx, s->in, out, ch);
541
542		✗	return 0;
543			}
544
545		✗	static int filter_frame(AVFilterLink *inlink, AVFrame *in)
546			{
547		✗	AVFilterContext *ctx = inlink->dst;
548		✗	AVFilterLink *outlink = ctx->outputs[0];
549		✗	AudioPsyClipContext *s = ctx->priv;
550			AVFrame *out;
551			int ret;
552
553		✗	out = ff_get_audio_buffer(outlink, s->overlap);
554		✗	if (!out) {
555		✗	ret = AVERROR(ENOMEM);
556		✗	goto fail;
557			}
558
559		✗	s->in = in;
560		✗	av_frame_copy_props(out, in);
561		✗	ff_filter_execute(ctx, psy_channels, out, NULL,
562		✗	FFMIN(outlink->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx)));
563
564		✗	out->pts = in->pts;
565		✗	out->nb_samples = in->nb_samples;
566		✗	ret = ff_filter_frame(outlink, out);
567		✗	fail:
568		✗	av_frame_free(&in);
569		✗	s->in = NULL;
570		✗	return ret < 0 ? ret : 0;
571			}
572
573		✗	static int activate(AVFilterContext *ctx)
574			{
575		✗	AVFilterLink *inlink = ctx->inputs[0];
576		✗	AVFilterLink *outlink = ctx->outputs[0];
577		✗	AudioPsyClipContext *s = ctx->priv;
578		✗	AVFrame *in = NULL;
579		✗	int ret = 0, status;
580			int64_t pts;
581
582		✗	FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
583
584		✗	ret = ff_inlink_consume_samples(inlink, s->overlap, s->overlap, &in);
585		✗	if (ret < 0)
586		✗	return ret;
587
588		✗	if (ret > 0) {
589		✗	return filter_frame(inlink, in);
590		✗	} else if (ff_inlink_acknowledge_status(inlink, &status, &pts)) {
591		✗	ff_outlink_set_status(outlink, status, pts);
592		✗	return 0;
593			} else {
594		✗	if (ff_inlink_queued_samples(inlink) >= s->overlap) {
595		✗	ff_filter_set_ready(ctx, 10);
596		✗	} else if (ff_outlink_frame_wanted(outlink)) {
597		✗	ff_inlink_request_frame(inlink);
598			}
599		✗	return 0;
600			}
601			}
602
603		✗	static av_cold void uninit(AVFilterContext *ctx)
604			{
605		✗	AudioPsyClipContext *s = ctx->priv;
606
607		✗	av_freep(&s->window);
608		✗	av_freep(&s->inv_window);
609		✗	av_freep(&s->spread_table);
610		✗	av_freep(&s->spread_table_range);
611		✗	av_freep(&s->spread_table_index);
612		✗	av_freep(&s->margin_curve);
613
614		✗	av_frame_free(&s->in_buffer);
615		✗	av_frame_free(&s->in_frame);
616		✗	av_frame_free(&s->out_dist_frame);
617		✗	av_frame_free(&s->windowed_frame);
618		✗	av_frame_free(&s->clipping_delta);
619		✗	av_frame_free(&s->spectrum_buf);
620		✗	av_frame_free(&s->mask_curve);
621
622		✗	for (int ch = 0; ch < s->channels; ch++) {
623		✗	if (s->tx_ctx)
624		✗	av_tx_uninit(&s->tx_ctx[ch]);
625		✗	if (s->itx_ctx)
626		✗	av_tx_uninit(&s->itx_ctx[ch]);
627			}
628
629		✗	av_freep(&s->tx_ctx);
630		✗	av_freep(&s->itx_ctx);
631		✗	}
632
633			static const AVFilterPad inputs[] = {
634			{
635			.name = "default",
636			.type = AVMEDIA_TYPE_AUDIO,
637			.config_props = config_input,
638			},
639			};
640
641			const AVFilter ff_af_apsyclip = {
642			.name = "apsyclip",
643			.description = NULL_IF_CONFIG_SMALL("Audio Psychoacoustic Clipper."),
644			.priv_size = sizeof(AudioPsyClipContext),
645			.priv_class = &apsyclip_class,
646			.uninit = uninit,
647			FILTER_INPUTS(inputs),
648			FILTER_OUTPUTS(ff_audio_default_filterpad),
649			FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_FLTP),
650			.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
651			AVFILTER_FLAG_SLICE_THREADS,
652			.activate = activate,
653			.process_command = ff_filter_process_command,
654			};
655