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avfilter: add audio psychoacoustic clipper

Paul B Mahol committed 5 years ago

Commit eeab62a1 parent 966fc3c6 files · +719 −1Browse files

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6 files
  1. Changelog +1 −0
  2. doc/filters.texi +36 −0
  3. libavfilter/Makefile +1 −0
  4. libavfilter/af_apsyclip.c +679 −0
  5. libavfilter/allfilters.c +1 −0
  6. libavfilter/version.h +1 −1
Changelog+1−0
@@ -19,6 +19,7 @@ version <next>: · 18 unmodified lines
19- swscale slice threading19- swscale slice threading
20- MSN Siren decoder20- MSN Siren decoder
21- scharr video filter21- scharr video filter
22- apsyclip audio filter
22 23
23 24
24version 4.4:25version 4.4:
doc/filters.texi+36−0
@@ -2363,6 +2363,42 @@ Default value is 8. · 2362 unmodified lines
2363 2363
2364This filter supports the all above options as @ref{commands}.2364This filter supports the all above options as @ref{commands}.
2365 2365
2366@section apsyclip
2367Apply Psychoacoustic clipper to input audio stream.
2368
2369The filter accepts the following options:
2370
2371@table @option
2372@item level_in
2373Set input gain. By default it is 1. Range is [0.015625 - 64].
2374
2375@item level_out
2376Set output gain. By default it is 1. Range is [0.015625 - 64].
2377
2378@item clip
2379Set the clipping start value. Default value is 0dBFS or 1.
2380
2381@item diff
2382Output only difference samples, useful to hear introduced distortions.
2383By default is disabled.
2384
2385@item adaptive
2386Set strenght of adaptive distortion applied. Default value is 0.5.
2387Allowed range is from 0 to 1.
2388
2389@item iterations
2390Set number of iterations of psychoacoustic clipper.
2391Allowed range is from 1 to 20. Default value is 10.
2392
2393@item level
2394Auto level output signal. Default is disabled.
2395This normalizes audio back to 0dBFS if enabled.
2396@end table
2397
2398@subsection Commands
2399
2400This filter supports the all above options as @ref{commands}.
2401
2366@section apulsator2402@section apulsator
2367 2403
2368Audio pulsator is something between an autopanner and a tremolo.2404Audio pulsator is something between an autopanner and a tremolo.
libavfilter/Makefile+1−0
@@ -74,6 +74,7 @@ OBJS-$(CONFIG_APAD_FILTER) += af_apad.o · 73 unmodified lines
74OBJS-$(CONFIG_APERMS_FILTER) += f_perms.o74OBJS-$(CONFIG_APERMS_FILTER) += f_perms.o
75OBJS-$(CONFIG_APHASER_FILTER) += af_aphaser.o generate_wave_table.o75OBJS-$(CONFIG_APHASER_FILTER) += af_aphaser.o generate_wave_table.o
76OBJS-$(CONFIG_APHASESHIFT_FILTER) += af_afreqshift.o76OBJS-$(CONFIG_APHASESHIFT_FILTER) += af_afreqshift.o
77OBJS-$(CONFIG_APSYCLIP_FILTER) += af_apsyclip.o
77OBJS-$(CONFIG_APULSATOR_FILTER) += af_apulsator.o78OBJS-$(CONFIG_APULSATOR_FILTER) += af_apulsator.o
78OBJS-$(CONFIG_AREALTIME_FILTER) += f_realtime.o79OBJS-$(CONFIG_AREALTIME_FILTER) += f_realtime.o
79OBJS-$(CONFIG_ARESAMPLE_FILTER) += af_aresample.o80OBJS-$(CONFIG_ARESAMPLE_FILTER) += af_aresample.o
libavfilter/af_apsyclip.c+679−0
@@ -0,0 +1,679 @@
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/opt.h"
23#include "libavutil/tx.h"
24#include "audio.h"
25#include "avfilter.h"
26#include "filters.h"
27#include "internal.h"
28
29typedef struct AudioPsyClipContext {
30 const AVClass *class;
31
32 double level_in;
33 double level_out;
34 double clip_level;
35 double adaptive;
36 int auto_level;
37 int diff_only;
38 int iterations;
39 char *protections_str;
40 double *protections;
41
42 int num_psy_bins;
43 int fft_size;
44 int overlap;
45 int channels;
46
47 int spread_table_rows;
48 int *spread_table_index;
49 int (*spread_table_range)[2];
50 float *window, *inv_window, *spread_table, *margin_curve;
51
52 AVFrame *in;
53 AVFrame *in_buffer;
54 AVFrame *in_frame;
55 AVFrame *out_dist_frame;
56 AVFrame *windowed_frame;
57 AVFrame *clipping_delta;
58 AVFrame *spectrum_buf;
59 AVFrame *mask_curve;
60
61 AVTXContext **tx_ctx;
62 av_tx_fn tx_fn;
63 AVTXContext **itx_ctx;
64 av_tx_fn itx_fn;
65} AudioPsyClipContext;
66
67#define OFFSET(x) offsetof(AudioPsyClipContext, x)
68#define FLAGS AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM | AV_OPT_FLAG_RUNTIME_PARAM
69
70static const AVOption apsyclip_options[] = {
71 { "level_in", "set input level", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625, 64, FLAGS },
72 { "level_out", "set output level", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625, 64, FLAGS },
73 { "clip", "set clip level", OFFSET(clip_level), AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625, 1, FLAGS },
74 { "diff", "enable difference", OFFSET(diff_only), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, FLAGS },
75 { "adaptive", "set adaptive distortion", OFFSET(adaptive), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 1, FLAGS },
76 { "iterations", "set iterations", OFFSET(iterations), AV_OPT_TYPE_INT, {.i64=10}, 1, 20, FLAGS },
77 { "level", "set auto level", OFFSET(auto_level), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, FLAGS },
78 {NULL}
79};
80
81AVFILTER_DEFINE_CLASS(apsyclip);
82
83static int query_formats(AVFilterContext *ctx)
84{
85 static const enum AVSampleFormat sample_fmts[] = {
86 AV_SAMPLE_FMT_FLTP,
87 AV_SAMPLE_FMT_NONE
88 };
89 int ret;
90
91 ret = ff_set_common_all_channel_counts(ctx);
92 if (ret < 0)
93 return ret;
94
95 ret = ff_set_common_formats_from_list(ctx, sample_fmts);
96 if (ret < 0)
97 return ret;
98
99 return ff_set_common_all_samplerates(ctx);
100}
101
102static void generate_hann_window(float *window, float *inv_window, int size)
103{
104 for (int i = 0; i < size; i++) {
105 float value = 0.5f * (1.f - cosf(2.f * M_PI * i / size));
106
107 window[i] = value;
108 // 1/window to calculate unwindowed peak.
109 inv_window[i] = value > 0.01f ? 1.f / value : 0.f;
110 }
111}
112
113static void set_margin_curve(AudioPsyClipContext *s,
114 const int (*points)[2], int num_points, int sample_rate)
115{
116 int j = 0;
117
118 s->margin_curve[0] = points[0][1];
119
120 for (int i = 0; i < num_points - 1; i++) {
121 while (j < s->fft_size / 2 + 1 && j * sample_rate / s->fft_size < points[i + 1][0]) {
122 // linearly interpolate between points
123 int binHz = j * sample_rate / s->fft_size;
124 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]);
125 j++;
126 }
127 }
128 // handle bins after the last point
129 while (j < s->fft_size / 2 + 1) {
130 s->margin_curve[j] = points[num_points - 1][1];
131 j++;
132 }
133
134 // convert margin curve to linear amplitude scale
135 for (j = 0; j < s->fft_size / 2 + 1; j++)
136 s->margin_curve[j] = powf(10.f, s->margin_curve[j] / 20.f);
137}
138
139static void generate_spread_table(AudioPsyClipContext *s)
140{
141 // Calculate tent-shape function in log-log scale.
142
143 // As an optimization, only consider bins close to "bin"
144 // This reduces the number of multiplications needed in calculate_mask_curve
145 // The masking contribution at faraway bins is negligeable
146
147 // Another optimization to save memory and speed up the calculation of the
148 // spread table is to calculate and store only 2 spread functions per
149 // octave, and reuse the same spread function for multiple bins.
150 int table_index = 0;
151 int bin = 0;
152 int increment = 1;
153
154 while (bin < s->num_psy_bins) {
155 float sum = 0;
156 int base_idx = table_index * s->num_psy_bins;
157 int start_bin = bin * 3 / 4;
158 int end_bin = FFMIN(s->num_psy_bins, ((bin + 1) * 4 + 2) / 3);
159 int next_bin;
160
161 for (int j = start_bin; j < end_bin; j++) {
162 // add 0.5 so i=0 doesn't get log(0)
163 float rel_idx_log = FFABS(logf((j + 0.5f) / (bin + 0.5f)));
164 float value;
165 if (j >= bin) {
166 // mask up
167 value = expf(-rel_idx_log * 40.f);
168 } else {
169 // mask down
170 value = expf(-rel_idx_log * 80.f);
171 }
172 // the spreading function is centred in the row
173 sum += value;
174 s->spread_table[base_idx + s->num_psy_bins / 2 + j - bin] = value;
175 }
176 // now normalize it
177 for (int j = start_bin; j < end_bin; j++) {
178 s->spread_table[base_idx + s->num_psy_bins / 2 + j - bin] /= sum;
179 }
180
181 s->spread_table_range[table_index][0] = start_bin - bin;
182 s->spread_table_range[table_index][1] = end_bin - bin;
183
184 if (bin <= 1) {
185 next_bin = bin + 1;
186 } else {
187 if ((bin & (bin - 1)) == 0) {
188 // power of 2
189 increment = bin / 2;
190 }
191
192 next_bin = bin + increment;
193 }
194
195 // set bins between "bin" and "next_bin" to use this table_index
196 for (int i = bin; i < next_bin; i++)
197 s->spread_table_index[i] = table_index;
198
199 bin = next_bin;
200 table_index++;
201 }
202}
203
204static int config_input(AVFilterLink *inlink)
205{
206 AVFilterContext *ctx = inlink->dst;
207 AudioPsyClipContext *s = ctx->priv;
208 static const int points[][2] = { {0,14}, {125,14}, {250,16}, {500,18}, {1000,20}, {2000,20}, {4000,20}, {8000,15}, {16000,5}, {20000,-10} };
209 static const int num_points = 10;
210 float scale;
211 int ret;
212
213 s->fft_size = inlink->sample_rate > 100000 ? 1024 : inlink->sample_rate > 50000 ? 512 : 256;
214 s->overlap = s->fft_size / 4;
215
216 // The psy masking calculation is O(n^2),
217 // so skip it for frequencies not covered by base sampling rantes (i.e. 44k)
218 if (inlink->sample_rate <= 50000) {
219 s->num_psy_bins = s->fft_size / 2;
220 } else if (inlink->sample_rate <= 100000) {
221 s->num_psy_bins = s->fft_size / 4;
222 } else {
223 s->num_psy_bins = s->fft_size / 8;
224 }
225
226 s->window = av_calloc(s->fft_size, sizeof(*s->window));
227 s->inv_window = av_calloc(s->fft_size, sizeof(*s->inv_window));
228 if (!s->window || !s->inv_window)
229 return AVERROR(ENOMEM);
230
231 s->in_buffer = ff_get_audio_buffer(inlink, s->fft_size * 2);
232 s->in_frame = ff_get_audio_buffer(inlink, s->fft_size * 2);
233 s->out_dist_frame = ff_get_audio_buffer(inlink, s->fft_size * 2);
234 s->windowed_frame = ff_get_audio_buffer(inlink, s->fft_size * 2);
235 s->clipping_delta = ff_get_audio_buffer(inlink, s->fft_size * 2);
236 s->spectrum_buf = ff_get_audio_buffer(inlink, s->fft_size * 2);
237 s->mask_curve = ff_get_audio_buffer(inlink, s->fft_size / 2 + 1);
238 if (!s->in_buffer || !s->in_frame ||
239 !s->out_dist_frame || !s->windowed_frame ||
240 !s->clipping_delta || !s->spectrum_buf || !s->mask_curve)
241 return AVERROR(ENOMEM);
242
243 generate_hann_window(s->window, s->inv_window, s->fft_size);
244
245 s->margin_curve = av_calloc(s->fft_size / 2 + 1, sizeof(*s->margin_curve));
246 if (!s->margin_curve)
247 return AVERROR(ENOMEM);
248
249 s->spread_table_rows = av_log2(s->num_psy_bins) * 2;
250 s->spread_table = av_calloc(s->spread_table_rows * s->num_psy_bins, sizeof(*s->spread_table));
251 if (!s->spread_table)
252 return AVERROR(ENOMEM);
253
254 s->spread_table_range = av_calloc(s->spread_table_rows * 2, sizeof(*s->spread_table_range));
255 if (!s->spread_table_range)
256 return AVERROR(ENOMEM);
257
258 s->spread_table_index = av_calloc(s->num_psy_bins, sizeof(*s->spread_table_index));
259 if (!s->spread_table_index)
260 return AVERROR(ENOMEM);
261
262 set_margin_curve(s, points, num_points, inlink->sample_rate);
263
264 generate_spread_table(s);
265
266 s->channels = inlink->channels;
267
268 s->tx_ctx = av_calloc(s->channels, sizeof(*s->tx_ctx));
269 s->itx_ctx = av_calloc(s->channels, sizeof(*s->itx_ctx));
270 if (!s->tx_ctx || !s->itx_ctx)
271 return AVERROR(ENOMEM);
272
273 for (int ch = 0; ch < s->channels; ch++) {
274 ret = av_tx_init(&s->tx_ctx[ch], &s->tx_fn, AV_TX_FLOAT_FFT, 0, s->fft_size, &scale, 0);
275 if (ret < 0)
276 return ret;
277
278 ret = av_tx_init(&s->itx_ctx[ch], &s->itx_fn, AV_TX_FLOAT_FFT, 1, s->fft_size, &scale, 0);
279 if (ret < 0)
280 return ret;
281 }
282
283 return 0;
284}
285
286static void apply_window(AudioPsyClipContext *s,
287 const float *in_frame, float *out_frame, const int add_to_out_frame)
288{
289 const float *window = s->window;
290
291 for (int i = 0; i < s->fft_size; i++) {
292 if (add_to_out_frame) {
293 out_frame[i] += in_frame[i] * window[i];
294 } else {
295 out_frame[i] = in_frame[i] * window[i];
296 }
297 }
298}
299
300static void calculate_mask_curve(AudioPsyClipContext *s,
301 const float *spectrum, float *mask_curve)
302{
303 for (int i = 0; i < s->fft_size / 2 + 1; i++)
304 mask_curve[i] = 0;
305
306 for (int i = 0; i < s->num_psy_bins; i++) {
307 float magnitude;
308 int table_idx;
309 int range[2];
310
311 if (i == 0) {
312 magnitude = FFABS(spectrum[0]);
313 } else if (i == s->fft_size / 2) {
314 magnitude = FFABS(spectrum[1]);
315 } else {
316 // although the negative frequencies are omitted because they are redundant,
317 // the magnitude of the positive frequencies are not doubled.
318 // Multiply the magnitude by 2 to simulate adding up the + and - frequencies.
319 magnitude = hypotf(spectrum[2 * i], spectrum[2 * i + 1]) * 2;
320 }
321
322 table_idx = s->spread_table_index[i];
323 range[0] = s->spread_table_range[table_idx][0];
324 range[1] = s->spread_table_range[table_idx][1];
325 int base_idx = table_idx * s->num_psy_bins;
326 int start_bin = FFMAX(0, i + range[0]);
327 int end_bin = FFMIN(s->num_psy_bins, i + range[1]);
328
329 for (int j = start_bin; j < end_bin; j++)
330 mask_curve[j] += s->spread_table[base_idx + s->num_psy_bins / 2 + j - i] * magnitude;
331 }
332
333 // for ultrasonic frequencies, skip the O(n^2) spread calculation and just copy the magnitude
334 for (int i = s->num_psy_bins; i < s->fft_size / 2 + 1; i++) {
335 float magnitude;
336 if (i == s->fft_size / 2) {
337 magnitude = FFABS(spectrum[1]);
338 } else {
339 // although the negative frequencies are omitted because they are redundant,
340 // the magnitude of the positive frequencies are not doubled.
341 // Multiply the magnitude by 2 to simulate adding up the + and - frequencies.
342 magnitude = hypotf(spectrum[2 * i], spectrum[2 * i + 1]) * 2;
343 }
344
345 mask_curve[i] = magnitude;
346 }
347
348 for (int i = 0; i < s->fft_size / 2 + 1; i++)
349 mask_curve[i] = mask_curve[i] / s->margin_curve[i];
350}
351
352static void clip_to_window(AudioPsyClipContext *s,
353 const float *windowed_frame, float *clipping_delta, float delta_boost)
354{
355 const float *window = s->window;
356
357 for (int i = 0; i < s->fft_size; i++) {
358 const float limit = s->clip_level * window[i];
359 const float effective_value = windowed_frame[i] + clipping_delta[i];
360
361 if (effective_value > limit) {
362 clipping_delta[i] += (limit - effective_value) * delta_boost;
363 } else if (effective_value < -limit) {
364 clipping_delta[i] += (-limit - effective_value) * delta_boost;
365 }
366 }
367}
368
369static void limit_clip_spectrum(AudioPsyClipContext *s,
370 float *clip_spectrum, const float *mask_curve)
371{
372 // bin 0
373 float relative_distortion_level = FFABS(clip_spectrum[0]) / mask_curve[0];
374
375 if (relative_distortion_level > 1.f)
376 clip_spectrum[0] /= relative_distortion_level;
377
378 // bin 1..N/2-1
379 for (int i = 1; i < s->fft_size / 2; i++) {
380 float real = clip_spectrum[i * 2];
381 float imag = clip_spectrum[i * 2 + 1];
382 // although the negative frequencies are omitted because they are redundant,
383 // the magnitude of the positive frequencies are not doubled.
384 // Multiply the magnitude by 2 to simulate adding up the + and - frequencies.
385 relative_distortion_level = hypotf(real, imag) * 2 / mask_curve[i];
386 if (relative_distortion_level > 1.0) {
387 clip_spectrum[i * 2] /= relative_distortion_level;
388 clip_spectrum[i * 2 + 1] /= relative_distortion_level;
389 }
390 }
391 // bin N/2
392 relative_distortion_level = FFABS(clip_spectrum[1]) / mask_curve[s->fft_size / 2];
393 if (relative_distortion_level > 1.f)
394 clip_spectrum[1] /= relative_distortion_level;
395}
396
397static void r2c(float *buffer, int size)
398{
399 for (int i = size - 1; i >= 0; i--)
400 buffer[2 * i] = buffer[i];
401
402 for (int i = size - 1; i >= 0; i--)
403 buffer[2 * i + 1] = 0.f;
404}
405
406static void c2r(float *buffer, int size)
407{
408 for (int i = 0; i < size; i++)
409 buffer[i] = buffer[2 * i];
410
411 for (int i = 0; i < size; i++)
412 buffer[i + size] = 0.f;
413}
414
415static void feed(AVFilterContext *ctx, int ch,
416 const float *in_samples, float *out_samples, int diff_only,
417 float *in_frame, float *out_dist_frame,
418 float *windowed_frame, float *clipping_delta,
419 float *spectrum_buf, float *mask_curve)
420{
421 AudioPsyClipContext *s = ctx->priv;
422 const float clip_level_inv = 1.f / s->clip_level;
423 const float level_out = s->level_out;
424 float orig_peak = 0;
425 float peak;
426
427 // shift in/out buffers
428 for (int i = 0; i < s->fft_size - s->overlap; i++) {
429 in_frame[i] = in_frame[i + s->overlap];
430 out_dist_frame[i] = out_dist_frame[i + s->overlap];
431 }
432
433 for (int i = 0; i < s->overlap; i++) {
434 in_frame[i + s->fft_size - s->overlap] = in_samples[i];
435 out_dist_frame[i + s->fft_size - s->overlap] = 0.f;
436 }
437
438 apply_window(s, in_frame, windowed_frame, 0);
439 r2c(windowed_frame, s->fft_size);
440 s->tx_fn(s->tx_ctx[ch], spectrum_buf, windowed_frame, sizeof(float));
441 c2r(windowed_frame, s->fft_size);
442 calculate_mask_curve(s, spectrum_buf, mask_curve);
443
444 // It would be easier to calculate the peak from the unwindowed input.
445 // This is just for consistency with the clipped peak calculateion
446 // because the inv_window zeros out samples on the edge of the window.
447 for (int i = 0; i < s->fft_size; i++)
448 orig_peak = FFMAX(orig_peak, FFABS(windowed_frame[i] * s->inv_window[i]));
449 orig_peak *= clip_level_inv;
450 peak = orig_peak;
451
452 // clear clipping_delta
453 for (int i = 0; i < s->fft_size * 2; i++)
454 clipping_delta[i] = 0.f;
455
456 // repeat clipping-filtering process a few times to control both the peaks and the spectrum
457 for (int i = 0; i < s->iterations; i++) {
458 float mask_curve_shift = 1.122f; // 1.122 is 1dB
459 // The last 1/3 of rounds have boosted delta to help reach the peak target faster
460 float delta_boost = 1.f;
461 if (i >= s->iterations - s->iterations / 3) {
462 // boosting the delta when largs peaks are still present is dangerous
463 if (peak < 2.f)
464 delta_boost = 2.f;
465 }
466
467 clip_to_window(s, windowed_frame, clipping_delta, delta_boost);
468
469 r2c(clipping_delta, s->fft_size);
470 s->tx_fn(s->tx_ctx[ch], spectrum_buf, clipping_delta, sizeof(float));
471
472 limit_clip_spectrum(s, spectrum_buf, mask_curve);
473
474 s->itx_fn(s->itx_ctx[ch], clipping_delta, spectrum_buf, sizeof(float));
475 c2r(clipping_delta, s->fft_size);
476
477 for (int i = 0; i < s->fft_size; i++)
478 clipping_delta[i] /= s->fft_size;
479
480 peak = 0;
481 for (int i = 0; i < s->fft_size; i++)
482 peak = FFMAX(peak, FFABS((windowed_frame[i] + clipping_delta[i]) * s->inv_window[i]));
483 peak *= clip_level_inv;
484
485 // Automatically adjust mask_curve as necessary to reach peak target
486 if (orig_peak > 1.f && peak > 1.f) {
487 float diff_achieved = orig_peak - peak;
488 if (i + 1 < s->iterations - s->iterations / 3 && diff_achieved > 0) {
489 float diff_needed = orig_peak - 1.f;
490 float diff_ratio = diff_needed / diff_achieved;
491 // If a good amount of peak reduction was already achieved,
492 // don't shift the mask_curve by the full peak value
493 // On the other hand, if only a little peak reduction was achieved,
494 // don't shift the mask_curve by the enormous diff_ratio.
495 diff_ratio = FFMIN(diff_ratio, peak);
496 mask_curve_shift = FFMAX(mask_curve_shift, diff_ratio);
497 } else {
498 // If the peak got higher than the input or we are in the last 1/3 rounds,
499 // go back to the heavy-handed peak heuristic.
500 mask_curve_shift = FFMAX(mask_curve_shift, peak);
501 }
502 }
503
504 mask_curve_shift = 1.f + (mask_curve_shift - 1.f) * s->adaptive;
505
506 // Be less strict in the next iteration.
507 // This helps with peak control.
508 for (int i = 0; i < s->fft_size / 2 + 1; i++)
509 mask_curve[i] *= mask_curve_shift;
510 }
511
512 // do overlap & add
513 apply_window(s, clipping_delta, out_dist_frame, 1);
514
515 for (int i = 0; i < s->overlap; i++) {
516 // 4 times overlap with squared hanning window results in 1.5 time increase in amplitude
517 if (!ctx->is_disabled) {
518 out_samples[i] = out_dist_frame[i] / 1.5f;
519 if (!diff_only)
520 out_samples[i] += in_frame[i];
521 if (s->auto_level)
522 out_samples[i] *= clip_level_inv;
523 out_samples[i] *= level_out;
524 } else {
525 out_samples[i] = in_frame[i];
526 }
527 }
528}
529
530static int psy_channel(AVFilterContext *ctx, AVFrame *in, AVFrame *out, int ch)
531{
532 AudioPsyClipContext *s = ctx->priv;
533 const float *src = (const float *)in->extended_data[ch];
534 float *in_buffer = (float *)s->in_buffer->extended_data[ch];
535 float *dst = (float *)out->extended_data[ch];
536
537 for (int n = 0; n < s->overlap; n++)
538 in_buffer[n] = src[n] * s->level_in;
539
540 feed(ctx, ch, in_buffer, dst, s->diff_only,
541 (float *)(s->in_frame->extended_data[ch]),
542 (float *)(s->out_dist_frame->extended_data[ch]),
543 (float *)(s->windowed_frame->extended_data[ch]),
544 (float *)(s->clipping_delta->extended_data[ch]),
545 (float *)(s->spectrum_buf->extended_data[ch]),
546 (float *)(s->mask_curve->extended_data[ch]));
547
548 return 0;
549}
550
551static int psy_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
552{
553 AudioPsyClipContext *s = ctx->priv;
554 AVFrame *out = arg;
555 const int start = (out->channels * jobnr) / nb_jobs;
556 const int end = (out->channels * (jobnr+1)) / nb_jobs;
557
558 for (int ch = start; ch < end; ch++)
559 psy_channel(ctx, s->in, out, ch);
560
561 return 0;
562}
563
564static int filter_frame(AVFilterLink *inlink, AVFrame *in)
565{
566 AVFilterContext *ctx = inlink->dst;
567 AVFilterLink *outlink = ctx->outputs[0];
568 AudioPsyClipContext *s = ctx->priv;
569 AVFrame *out;
570 int ret;
571
572 out = ff_get_audio_buffer(outlink, s->overlap);
573 if (!out) {
574 ret = AVERROR(ENOMEM);
575 goto fail;
576 }
577
578 s->in = in;
579 ff_filter_execute(ctx, psy_channels, out, NULL,
580 FFMIN(outlink->channels, ff_filter_get_nb_threads(ctx)));
581
582 out->pts = in->pts;
583 out->nb_samples = in->nb_samples;
584 ret = ff_filter_frame(outlink, out);
585fail:
586 av_frame_free(&in);
587 s->in = NULL;
588 return ret < 0 ? ret : 0;
589}
590
591static int activate(AVFilterContext *ctx)
592{
593 AVFilterLink *inlink = ctx->inputs[0];
594 AVFilterLink *outlink = ctx->outputs[0];
595 AudioPsyClipContext *s = ctx->priv;
596 AVFrame *in = NULL;
597 int ret = 0, status;
598 int64_t pts;
599
600 FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
601
602 ret = ff_inlink_consume_samples(inlink, s->overlap, s->overlap, &in);
603 if (ret < 0)
604 return ret;
605
606 if (ret > 0) {
607 return filter_frame(inlink, in);
608 } else if (ff_inlink_acknowledge_status(inlink, &status, &pts)) {
609 ff_outlink_set_status(outlink, status, pts);
610 return 0;
611 } else {
612 if (ff_inlink_queued_samples(inlink) >= s->overlap) {
613 ff_filter_set_ready(ctx, 10);
614 } else if (ff_outlink_frame_wanted(outlink)) {
615 ff_inlink_request_frame(inlink);
616 }
617 return 0;
618 }
619}
620
621static av_cold void uninit(AVFilterContext *ctx)
622{
623 AudioPsyClipContext *s = ctx->priv;
624
625 av_freep(&s->window);
626 av_freep(&s->inv_window);
627 av_freep(&s->spread_table);
628 av_freep(&s->spread_table_range);
629 av_freep(&s->spread_table_index);
630 av_freep(&s->margin_curve);
631
632 av_frame_free(&s->in_buffer);
633 av_frame_free(&s->in_frame);
634 av_frame_free(&s->out_dist_frame);
635 av_frame_free(&s->windowed_frame);
636 av_frame_free(&s->clipping_delta);
637 av_frame_free(&s->spectrum_buf);
638 av_frame_free(&s->mask_curve);
639
640 for (int ch = 0; ch < s->channels; ch++) {
641 if (s->tx_ctx)
642 av_tx_uninit(&s->tx_ctx[ch]);
643 if (s->itx_ctx)
644 av_tx_uninit(&s->itx_ctx[ch]);
645 }
646
647 av_freep(&s->tx_ctx);
648 av_freep(&s->itx_ctx);
649}
650
651static const AVFilterPad inputs[] = {
652 {
653 .name = "default",
654 .type = AVMEDIA_TYPE_AUDIO,
655 .config_props = config_input,
656 },
657};
658
659static const AVFilterPad outputs[] = {
660 {
661 .name = "default",
662 .type = AVMEDIA_TYPE_AUDIO,
663 },
664};
665
666const AVFilter ff_af_apsyclip = {
667 .name = "apsyclip",
668 .description = NULL_IF_CONFIG_SMALL("Audio Psychoacoustic Clipper."),
669 .query_formats = query_formats,
670 .priv_size = sizeof(AudioPsyClipContext),
671 .priv_class = &apsyclip_class,
672 .uninit = uninit,
673 FILTER_INPUTS(inputs),
674 FILTER_OUTPUTS(outputs),
675 .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
676 AVFILTER_FLAG_SLICE_THREADS,
677 .activate = activate,
678 .process_command = ff_filter_process_command,
679};
libavfilter/allfilters.c+1−0
@@ -67,6 +67,7 @@ extern const AVFilter ff_af_apad; · 66 unmodified lines
67extern const AVFilter ff_af_aperms;67extern const AVFilter ff_af_aperms;
68extern const AVFilter ff_af_aphaser;68extern const AVFilter ff_af_aphaser;
69extern const AVFilter ff_af_aphaseshift;69extern const AVFilter ff_af_aphaseshift;
70extern const AVFilter ff_af_apsyclip;
70extern const AVFilter ff_af_apulsator;71extern const AVFilter ff_af_apulsator;
71extern const AVFilter ff_af_arealtime;72extern const AVFilter ff_af_arealtime;
72extern const AVFilter ff_af_aresample;73extern const AVFilter ff_af_aresample;
libavfilter/version.h+1−1
@@ -30,7 +30,7 @@ · 29 unmodified lines
30#include "libavutil/version.h"30#include "libavutil/version.h"
31 31
32#define LIBAVFILTER_VERSION_MAJOR 832#define LIBAVFILTER_VERSION_MAJOR 8
33#define LIBAVFILTER_VERSION_MINOR 833#define LIBAVFILTER_VERSION_MINOR 9
34#define LIBAVFILTER_VERSION_MICRO 10034#define LIBAVFILTER_VERSION_MICRO 100
35 35
36 36

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