FFmpeg coverage

Directory: ../../../ffmpeg/
File: src/libavfilter/af_asoftclip.c
Date: 2026-09-27 22:20:00
Exec Total Coverage
Lines: 0 253 0.0%
Functions: 0 9 0.0%
Branches: 0 135 0.0%
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1 /*
2 * Copyright (c) 2019 The FFmpeg Project
3 *
4 * This file is part of FFmpeg.
5 *
6 * FFmpeg is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Lesser General Public
8 * License as published by the Free Software Foundation; either
9 * version 2.1 of the License, or (at your option) any later version.
10 *
11 * FFmpeg is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Lesser General Public License for more details.
15 *
16 * You should have received a copy of the GNU Lesser General Public
17 * License along with FFmpeg; if not, write to the Free Software
18 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19 */
20
21 #include "libavutil/avassert.h"
22 #include "libavutil/channel_layout.h"
23 #include "libavutil/opt.h"
24 #include "avfilter.h"
25 #include "audio.h"
26 #include "filters.h"
27
28 #define MAX_OVERSAMPLE 64
29
30 enum ASoftClipTypes {
31 ASC_HARD = -1,
32 ASC_TANH,
33 ASC_ATAN,
34 ASC_CUBIC,
35 ASC_EXP,
36 ASC_ALG,
37 ASC_QUINTIC,
38 ASC_SIN,
39 ASC_ERF,
40 NB_TYPES,
41 };
42
43 typedef struct Lowpass {
44 float fb0, fb1, fb2;
45 float fa0, fa1, fa2;
46
47 double db0, db1, db2;
48 double da0, da1, da2;
49 } Lowpass;
50
51 typedef struct ASoftClipContext {
52 const AVClass *class;
53
54 int type;
55 int oversample;
56 int64_t delay;
57 double threshold;
58 double output;
59 double param;
60
61 Lowpass lowpass[MAX_OVERSAMPLE];
62 AVFrame *frame[2];
63
64 void (*filter)(struct ASoftClipContext *s, void **dst, const void **src,
65 int nb_samples, int channels, int start, int end);
66 } ASoftClipContext;
67
68 #define OFFSET(x) offsetof(ASoftClipContext, x)
69 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
70
71 static const AVOption asoftclip_options[] = {
72 { "type", "set softclip type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=0}, -1, NB_TYPES-1, A, .unit = "types" },
73 { "hard", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_HARD}, 0, 0, A, .unit = "types" },
74 { "tanh", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_TANH}, 0, 0, A, .unit = "types" },
75 { "atan", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_ATAN}, 0, 0, A, .unit = "types" },
76 { "cubic", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_CUBIC}, 0, 0, A, .unit = "types" },
77 { "exp", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_EXP}, 0, 0, A, .unit = "types" },
78 { "alg", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_ALG}, 0, 0, A, .unit = "types" },
79 { "quintic", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_QUINTIC},0, 0, A, .unit = "types" },
80 { "sin", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_SIN}, 0, 0, A, .unit = "types" },
81 { "erf", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_ERF}, 0, 0, A, .unit = "types" },
82 { "threshold", "set softclip threshold", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.000001, 1, A },
83 { "output", "set softclip output gain", OFFSET(output), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.000001, 16, A },
84 { "param", "set softclip parameter", OFFSET(param), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.01, 3, A },
85 { "oversample", "set oversample factor", OFFSET(oversample), AV_OPT_TYPE_INT, {.i64=1}, 1, MAX_OVERSAMPLE, A },
86 { NULL }
87 };
88
89 AVFILTER_DEFINE_CLASS(asoftclip);
90
91 ✗ static void get_lowpass(Lowpass *s,
92 double frequency,
93 double sample_rate)
94 {
95 ✗ double w0 = 2 * M_PI * frequency / sample_rate;
96 ✗ double alpha = sin(w0) / (2 * 0.8);
97 double factor;
98
99 ✗ s->da0 = 1 + alpha;
100 ✗ s->da1 = -2 * cos(w0);
101 ✗ s->da2 = 1 - alpha;
102 ✗ s->db0 = (1 - cos(w0)) / 2;
103 ✗ s->db1 = 1 - cos(w0);
104 ✗ s->db2 = (1 - cos(w0)) / 2;
105
106 ✗ s->da1 /= s->da0;
107 ✗ s->da2 /= s->da0;
108 ✗ s->db0 /= s->da0;
109 ✗ s->db1 /= s->da0;
110 ✗ s->db2 /= s->da0;
111 ✗ s->da0 /= s->da0;
112
113 ✗ factor = (s->da0 + s->da1 + s->da2) / (s->db0 + s->db1 + s->db2);
114 ✗ s->db0 *= factor;
115 ✗ s->db1 *= factor;
116 ✗ s->db2 *= factor;
117
118 ✗ s->fa0 = s->da0;
119 ✗ s->fa1 = s->da1;
120 ✗ s->fa2 = s->da2;
121 ✗ s->fb0 = s->db0;
122 ✗ s->fb1 = s->db1;
123 ✗ s->fb2 = s->db2;
124 ✗ }
125
126 ✗ static inline float run_lowpassf(const Lowpass *const s,
127 float src, float *w)
128 {
129 float dst;
130
131 ✗ dst = src * s->fb0 + w[0];
132 ✗ w[0] = s->fb1 * src + w[1] - s->fa1 * dst;
133 ✗ w[1] = s->fb2 * src - s->fa2 * dst;
134
135 ✗ return dst;
136 }
137
138 ✗ static void filter_flt(ASoftClipContext *s,
139 void **dptr, const void **sptr,
140 int nb_samples, int channels,
141 int start, int end)
142 {
143 ✗ const int oversample = s->oversample;
144 ✗ const int nb_osamples = nb_samples * oversample;
145 ✗ const float scale = oversample > 1 ? oversample * 0.5f : 1.f;
146 ✗ float threshold = s->threshold;
147 ✗ float gain = s->output * threshold;
148 ✗ float factor = 1.f / threshold;
149 ✗ float param = s->param;
150
151 ✗ for (int c = start; c < end; c++) {
152 ✗ float *w = (float *)(s->frame[0]->extended_data[c]) + 2 * (oversample - 1);
153 ✗ const float *src = sptr[c];
154 ✗ float *dst = dptr[c];
155
156 ✗ for (int n = 0; n < nb_samples; n++) {
157 ✗ dst[oversample * n] = src[n];
158
159 ✗ for (int m = 1; m < oversample; m++)
160 ✗ dst[oversample * n + m] = 0.f;
161 }
162
163 ✗ for (int n = 0; n < nb_osamples && oversample > 1; n++)
164 ✗ dst[n] = run_lowpassf(&s->lowpass[oversample - 1], dst[n], w);
165
166 ✗ switch (s->type) {
167 ✗ case ASC_HARD:
168 ✗ for (int n = 0; n < nb_osamples; n++) {
169 ✗ dst[n] = av_clipf(dst[n] * factor, -1.f, 1.f);
170 ✗ dst[n] *= gain;
171 }
172 ✗ break;
173 ✗ case ASC_TANH:
174 ✗ for (int n = 0; n < nb_osamples; n++) {
175 ✗ dst[n] = tanhf(dst[n] * factor * param);
176 ✗ dst[n] *= gain;
177 }
178 ✗ break;
179 ✗ case ASC_ATAN:
180 ✗ for (int n = 0; n < nb_osamples; n++) {
181 ✗ dst[n] = 2.f / M_PI * atanf(dst[n] * factor * param);
182 ✗ dst[n] *= gain;
183 }
184 ✗ break;
185 ✗ case ASC_CUBIC:
186 ✗ for (int n = 0; n < nb_osamples; n++) {
187 ✗ float sample = dst[n] * factor;
188
189 ✗ if (FFABS(sample) >= 1.5f)
190 ✗ dst[n] = FFSIGN(sample);
191 else
192 ✗ dst[n] = sample - 0.1481f * powf(sample, 3.f);
193 ✗ dst[n] *= gain;
194 }
195 ✗ break;
196 ✗ case ASC_EXP:
197 ✗ for (int n = 0; n < nb_osamples; n++) {
198 ✗ dst[n] = 2.f / (1.f + expf(-2.f * dst[n] * factor)) - 1.;
199 ✗ dst[n] *= gain;
200 }
201 ✗ break;
202 ✗ case ASC_ALG:
203 ✗ for (int n = 0; n < nb_osamples; n++) {
204 ✗ float sample = dst[n] * factor;
205
206 ✗ dst[n] = sample / (sqrtf(param + sample * sample));
207 ✗ dst[n] *= gain;
208 }
209 ✗ break;
210 ✗ case ASC_QUINTIC:
211 ✗ for (int n = 0; n < nb_osamples; n++) {
212 ✗ float sample = dst[n] * factor;
213
214 ✗ if (FFABS(sample) >= 1.25)
215 ✗ dst[n] = FFSIGN(sample);
216 else
217 ✗ dst[n] = sample - 0.08192f * powf(sample, 5.f);
218 ✗ dst[n] *= gain;
219 }
220 ✗ break;
221 ✗ case ASC_SIN:
222 ✗ for (int n = 0; n < nb_osamples; n++) {
223 ✗ float sample = dst[n] * factor;
224
225 ✗ if (FFABS(sample) >= M_PI_2)
226 ✗ dst[n] = FFSIGN(sample);
227 else
228 ✗ dst[n] = sinf(sample);
229 ✗ dst[n] *= gain;
230 }
231 ✗ break;
232 ✗ case ASC_ERF:
233 ✗ for (int n = 0; n < nb_osamples; n++) {
234 ✗ dst[n] = erff(dst[n] * factor);
235 ✗ dst[n] *= gain;
236 }
237 ✗ break;
238 ✗ default:
239 ✗ av_assert0(0);
240 }
241
242 ✗ w = (float *)(s->frame[1]->extended_data[c]) + 2 * (oversample - 1);
243 ✗ for (int n = 0; n < nb_osamples && oversample > 1; n++)
244 ✗ dst[n] = run_lowpassf(&s->lowpass[oversample - 1], dst[n], w);
245
246 ✗ for (int n = 0; n < nb_samples; n++)
247 ✗ dst[n] = dst[n * oversample] * scale;
248 }
249 ✗ }
250
251 ✗ static inline double run_lowpassd(const Lowpass *const s,
252 double src, double *w)
253 {
254 double dst;
255
256 ✗ dst = src * s->db0 + w[0];
257 ✗ w[0] = s->db1 * src + w[1] - s->da1 * dst;
258 ✗ w[1] = s->db2 * src - s->da2 * dst;
259
260 ✗ return dst;
261 }
262
263 ✗ static void filter_dbl(ASoftClipContext *s,
264 void **dptr, const void **sptr,
265 int nb_samples, int channels,
266 int start, int end)
267 {
268 ✗ const int oversample = s->oversample;
269 ✗ const int nb_osamples = nb_samples * oversample;
270 ✗ const double scale = oversample > 1 ? oversample * 0.5 : 1.;
271 ✗ double threshold = s->threshold;
272 ✗ double gain = s->output * threshold;
273 ✗ double factor = 1. / threshold;
274 ✗ double param = s->param;
275
276 ✗ for (int c = start; c < end; c++) {
277 ✗ double *w = (double *)(s->frame[0]->extended_data[c]) + 2 * (oversample - 1);
278 ✗ const double *src = sptr[c];
279 ✗ double *dst = dptr[c];
280
281 ✗ for (int n = 0; n < nb_samples; n++) {
282 ✗ dst[oversample * n] = src[n];
283
284 ✗ for (int m = 1; m < oversample; m++)
285 ✗ dst[oversample * n + m] = 0.f;
286 }
287
288 ✗ for (int n = 0; n < nb_osamples && oversample > 1; n++)
289 ✗ dst[n] = run_lowpassd(&s->lowpass[oversample - 1], dst[n], w);
290
291 ✗ switch (s->type) {
292 ✗ case ASC_HARD:
293 ✗ for (int n = 0; n < nb_osamples; n++) {
294 ✗ dst[n] = av_clipd(dst[n] * factor, -1., 1.);
295 ✗ dst[n] *= gain;
296 }
297 ✗ break;
298 ✗ case ASC_TANH:
299 ✗ for (int n = 0; n < nb_osamples; n++) {
300 ✗ dst[n] = tanh(dst[n] * factor * param);
301 ✗ dst[n] *= gain;
302 }
303 ✗ break;
304 ✗ case ASC_ATAN:
305 ✗ for (int n = 0; n < nb_osamples; n++) {
306 ✗ dst[n] = 2. / M_PI * atan(dst[n] * factor * param);
307 ✗ dst[n] *= gain;
308 }
309 ✗ break;
310 ✗ case ASC_CUBIC:
311 ✗ for (int n = 0; n < nb_osamples; n++) {
312 ✗ double sample = dst[n] * factor;
313
314 ✗ if (FFABS(sample) >= 1.5)
315 ✗ dst[n] = FFSIGN(sample);
316 else
317 ✗ dst[n] = sample - 0.1481 * pow(sample, 3.);
318 ✗ dst[n] *= gain;
319 }
320 ✗ break;
321 ✗ case ASC_EXP:
322 ✗ for (int n = 0; n < nb_osamples; n++) {
323 ✗ dst[n] = 2. / (1. + exp(-2. * dst[n] * factor)) - 1.;
324 ✗ dst[n] *= gain;
325 }
326 ✗ break;
327 ✗ case ASC_ALG:
328 ✗ for (int n = 0; n < nb_osamples; n++) {
329 ✗ double sample = dst[n] * factor;
330
331 ✗ dst[n] = sample / (sqrt(param + sample * sample));
332 ✗ dst[n] *= gain;
333 }
334 ✗ break;
335 ✗ case ASC_QUINTIC:
336 ✗ for (int n = 0; n < nb_osamples; n++) {
337 ✗ double sample = dst[n] * factor;
338
339 ✗ if (FFABS(sample) >= 1.25)
340 ✗ dst[n] = FFSIGN(sample);
341 else
342 ✗ dst[n] = sample - 0.08192 * pow(sample, 5.);
343 ✗ dst[n] *= gain;
344 }
345 ✗ break;
346 ✗ case ASC_SIN:
347 ✗ for (int n = 0; n < nb_osamples; n++) {
348 ✗ double sample = dst[n] * factor;
349
350 ✗ if (FFABS(sample) >= M_PI_2)
351 ✗ dst[n] = FFSIGN(sample);
352 else
353 ✗ dst[n] = sin(sample);
354 ✗ dst[n] *= gain;
355 }
356 ✗ break;
357 ✗ case ASC_ERF:
358 ✗ for (int n = 0; n < nb_osamples; n++) {
359 ✗ dst[n] = erf(dst[n] * factor);
360 ✗ dst[n] *= gain;
361 }
362 ✗ break;
363 ✗ default:
364 ✗ av_assert0(0);
365 }
366
367 ✗ w = (double *)(s->frame[1]->extended_data[c]) + 2 * (oversample - 1);
368 ✗ for (int n = 0; n < nb_osamples && oversample > 1; n++)
369 ✗ dst[n] = run_lowpassd(&s->lowpass[oversample - 1], dst[n], w);
370
371 ✗ for (int n = 0; n < nb_samples; n++)
372 ✗ dst[n] = dst[n * oversample] * scale;
373 }
374 ✗ }
375
376 ✗ static int config_input(AVFilterLink *inlink)
377 {
378 ✗ AVFilterContext *ctx = inlink->dst;
379 ✗ ASoftClipContext *s = ctx->priv;
380
381 ✗ switch (inlink->format) {
382 ✗ case AV_SAMPLE_FMT_FLTP: s->filter = filter_flt; break;
383 ✗ case AV_SAMPLE_FMT_DBLP: s->filter = filter_dbl; break;
384 ✗ default: av_assert0(0);
385 }
386
387 ✗ s->frame[0] = ff_get_audio_buffer(inlink, 2 * MAX_OVERSAMPLE);
388 ✗ s->frame[1] = ff_get_audio_buffer(inlink, 2 * MAX_OVERSAMPLE);
389 ✗ if (!s->frame[0] || !s->frame[1])
390 ✗ return AVERROR(ENOMEM);
391
392 ✗ for (int i = 0; i < MAX_OVERSAMPLE; i++) {
393 ✗ get_lowpass(&s->lowpass[i], inlink->sample_rate / 2, inlink->sample_rate * (i + 1));
394 }
395
396 ✗ return 0;
397 }
398
399 typedef struct ThreadData {
400 AVFrame *in, *out;
401 int nb_samples;
402 int channels;
403 } ThreadData;
404
405 ✗ static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
406 {
407 ✗ ASoftClipContext *s = ctx->priv;
408 ✗ ThreadData *td = arg;
409 ✗ AVFrame *out = td->out;
410 ✗ AVFrame *in = td->in;
411 ✗ const int channels = td->channels;
412 ✗ const int nb_samples = td->nb_samples;
413 ✗ const int start = ff_slice_pos(channels, jobnr, nb_jobs);
414 ✗ const int end = ff_slice_pos(channels, jobnr + 1, nb_jobs);
415
416 ✗ s->filter(s, (void **)out->extended_data, (const void **)in->extended_data,
417 nb_samples, channels, start, end);
418
419 ✗ return 0;
420 }
421
422 ✗ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
423 {
424 ✗ AVFilterContext *ctx = inlink->dst;
425 ✗ ASoftClipContext *s = ctx->priv;
426 ✗ AVFilterLink *outlink = ctx->outputs[0];
427 int nb_samples, channels;
428 ThreadData td;
429 AVFrame *out;
430
431 ✗ if (av_frame_is_writable(in) && s->oversample == 1) {
432 ✗ out = in;
433 } else {
434 ✗ out = ff_get_audio_buffer(outlink, in->nb_samples * s->oversample);
435 ✗ if (!out) {
436 ✗ av_frame_free(&in);
437 ✗ return AVERROR(ENOMEM);
438 }
439 ✗ av_frame_copy_props(out, in);
440 }
441
442 ✗ nb_samples = in->nb_samples;
443 ✗ channels = in->ch_layout.nb_channels;
444
445 ✗ td.in = in;
446 ✗ td.out = out;
447 ✗ td.nb_samples = nb_samples;
448 ✗ td.channels = channels;
449 ✗ ff_filter_execute(ctx, filter_channels, &td, NULL,
450 ✗ FFMIN(channels, ff_filter_get_nb_threads(ctx)));
451
452 ✗ if (out != in)
453 ✗ av_frame_free(&in);
454
455 ✗ out->nb_samples /= s->oversample;
456 ✗ return ff_filter_frame(outlink, out);
457 }
458
459 ✗ static av_cold void uninit(AVFilterContext *ctx)
460 {
461 ✗ ASoftClipContext *s = ctx->priv;
462
463 ✗ av_frame_free(&s->frame[0]);
464 ✗ av_frame_free(&s->frame[1]);
465 ✗ }
466
467 static const AVFilterPad inputs[] = {
468 {
469 .name = "default",
470 .type = AVMEDIA_TYPE_AUDIO,
471 .filter_frame = filter_frame,
472 .config_props = config_input,
473 },
474 };
475
476 const FFFilter ff_af_asoftclip = {
477 .p.name = "asoftclip",
478 .p.description = NULL_IF_CONFIG_SMALL("Audio Soft Clipper."),
479 .p.priv_class = &asoftclip_class,
480 .p.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC |
481 AVFILTER_FLAG_SLICE_THREADS,
482 .priv_size = sizeof(ASoftClipContext),
483 FILTER_INPUTS(inputs),
484 FILTER_OUTPUTS(ff_audio_default_filterpad),
485 FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP),
486 .uninit = uninit,
487 .process_command = ff_filter_process_command,
488 };
489