FFmpeg coverage

Directory: ../../../ffmpeg/
File: src/libavfilter/af_alimiter.c
Date: 2024-04-25 15:36:26
Exec Total Coverage
Lines: 148 203 72.9%
Functions: 6 6 100.0%
Branches: 66 124 53.2%
Line Branch Exec Source
1 /*
2 * Copyright (C) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen and others
3 * Copyright (c) 2015 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
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 * Lookahead limiter filter
25 */
26
27 #include "libavutil/channel_layout.h"
28 #include "libavutil/common.h"
29 #include "libavutil/fifo.h"
30 #include "libavutil/mem.h"
31 #include "libavutil/opt.h"
32
33 #include "audio.h"
34 #include "avfilter.h"
35 #include "internal.h"
36
37 typedef struct MetaItem {
38 int64_t pts;
39 int nb_samples;
40 } MetaItem;
41
42 typedef struct AudioLimiterContext {
43 const AVClass *class;
44
45 double limit;
46 double attack;
47 double release;
48 double att;
49 double level_in;
50 double level_out;
51 int auto_release;
52 int auto_level;
53 double asc;
54 int asc_c;
55 int asc_pos;
56 double asc_coeff;
57
58 double *buffer;
59 int buffer_size;
60 int pos;
61 int *nextpos;
62 double *nextdelta;
63
64 int in_trim;
65 int out_pad;
66 int64_t next_in_pts;
67 int64_t next_out_pts;
68 int latency;
69
70 AVFifo *fifo;
71
72 double delta;
73 int nextiter;
74 int nextlen;
75 int asc_changed;
76 } AudioLimiterContext;
77
78 #define OFFSET(x) offsetof(AudioLimiterContext, x)
79 #define AF AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM | AV_OPT_FLAG_RUNTIME_PARAM
80
81 static const AVOption alimiter_options[] = {
82 { "level_in", "set input level", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625, 64, AF },
83 { "level_out", "set output level", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625, 64, AF },
84 { "limit", "set limit", OFFSET(limit), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.0625, 1, AF },
85 { "attack", "set attack", OFFSET(attack), AV_OPT_TYPE_DOUBLE, {.dbl=5}, 0.1, 80, AF },
86 { "release", "set release", OFFSET(release), AV_OPT_TYPE_DOUBLE, {.dbl=50}, 1, 8000, AF },
87 { "asc", "enable asc", OFFSET(auto_release), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, AF },
88 { "asc_level", "set asc level", OFFSET(asc_coeff), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 1, AF },
89 { "level", "auto level", OFFSET(auto_level), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, AF },
90 { "latency", "compensate delay", OFFSET(latency), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, AF },
91 { NULL }
92 };
93
94 AVFILTER_DEFINE_CLASS(alimiter);
95
96 2 static av_cold int init(AVFilterContext *ctx)
97 {
98 2 AudioLimiterContext *s = ctx->priv;
99
100 2 s->attack /= 1000.;
101 2 s->release /= 1000.;
102 2 s->att = 1.;
103 2 s->asc_pos = -1;
104 2 s->asc_coeff = pow(0.5, s->asc_coeff - 0.5) * 2 * -1;
105
106 2 return 0;
107 }
108
109 138816 static double get_rdelta(AudioLimiterContext *s, double release, int sample_rate,
110 double peak, double limit, double patt, int asc)
111 {
112 138816 double rdelta = (1.0 - patt) / (sample_rate * release);
113
114
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 138816 if (asc && s->auto_release && s->asc_c > 0) {
115 ✗ double a_att = limit / (s->asc_coeff * s->asc) * (double)s->asc_c;
116
117 ✗ if (a_att > patt) {
118 ✗ double delta = FFMAX((a_att - patt) / (sample_rate * release), rdelta / 10);
119
120 ✗ if (delta < rdelta)
121 ✗ rdelta = delta;
122 }
123 }
124
125 138816 return rdelta;
126 }
127
128 65 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
129 {
130 65 AVFilterContext *ctx = inlink->dst;
131 65 AudioLimiterContext *s = ctx->priv;
132 65 AVFilterLink *outlink = ctx->outputs[0];
133 65 const double *src = (const double *)in->data[0];
134 65 const int channels = inlink->ch_layout.nb_channels;
135 65 const int buffer_size = s->buffer_size;
136 65 double *dst, *buffer = s->buffer;
137 65 const double release = s->release;
138 65 const double limit = s->limit;
139 65 double *nextdelta = s->nextdelta;
140
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 65 double level = s->auto_level ? 1 / limit : 1;
141 65 const double level_out = s->level_out;
142 65 const double level_in = s->level_in;
143 65 int *nextpos = s->nextpos;
144 AVFrame *out;
145 double *buf;
146 int n, c, i;
147 int new_out_samples;
148 int64_t out_duration;
149 int64_t in_duration;
150 int64_t in_pts;
151 MetaItem meta;
152
153
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 65 if (av_frame_is_writable(in)) {
154 65 out = in;
155 } else {
156 ✗ out = ff_get_audio_buffer(outlink, in->nb_samples);
157 ✗ if (!out) {
158 ✗ av_frame_free(&in);
159 ✗ return AVERROR(ENOMEM);
160 }
161 ✗ av_frame_copy_props(out, in);
162 }
163 65 dst = (double *)out->data[0];
164
165
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 264665 for (n = 0; n < in->nb_samples; n++) {
166 264600 double peak = 0;
167
168
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 793800 for (c = 0; c < channels; c++) {
169 529200 double sample = src[c] * level_in;
170
171 529200 buffer[s->pos + c] = sample;
172
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 529200 peak = FFMAX(peak, fabs(sample));
173 }
174
175
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 264600 if (s->auto_release && peak > limit) {
176 ✗ s->asc += peak;
177 ✗ s->asc_c++;
178 }
179
180
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 264600 if (peak > limit) {
181
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 138816 double patt = FFMIN(limit / peak, 1.);
182 138816 double rdelta = get_rdelta(s, release, inlink->sample_rate,
183 peak, limit, patt, 0);
184 138816 double delta = (limit / peak - s->att) / buffer_size * channels;
185 138816 int found = 0;
186
187
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 138816 if (delta < s->delta) {
188 1669 s->delta = delta;
189 1669 nextpos[0] = s->pos;
190 1669 nextpos[1] = -1;
191 1669 nextdelta[0] = rdelta;
192 1669 s->nextlen = 1;
193 1669 s->nextiter= 0;
194 } else {
195
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 564375 for (i = s->nextiter; i < s->nextiter + s->nextlen; i++) {
196 439864 int j = i % buffer_size;
197 439864 double ppeak = 0, pdelta;
198
199
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 439864 if (nextpos[j] >= 0)
200
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 1319592 for (c = 0; c < channels; c++) {
201
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 879728 ppeak = FFMAX(ppeak, fabs(buffer[nextpos[j] + c]));
202 }
203 439864 pdelta = (limit / peak - limit / ppeak) / (((buffer_size - nextpos[j] + s->pos) % buffer_size) / channels);
204
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 439864 if (pdelta < nextdelta[j]) {
205 12636 nextdelta[j] = pdelta;
206 12636 found = 1;
207 12636 break;
208 }
209 }
210
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 137147 if (found) {
211 12636 s->nextlen = i - s->nextiter + 1;
212 12636 nextpos[(s->nextiter + s->nextlen) % buffer_size] = s->pos;
213 12636 nextdelta[(s->nextiter + s->nextlen) % buffer_size] = rdelta;
214 12636 nextpos[(s->nextiter + s->nextlen + 1) % buffer_size] = -1;
215 12636 s->nextlen++;
216 }
217 }
218 }
219
220 264600 buf = &s->buffer[(s->pos + channels) % buffer_size];
221 264600 peak = 0;
222
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 793800 for (c = 0; c < channels; c++) {
223 529200 double sample = buf[c];
224
225
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 529200 peak = FFMAX(peak, fabs(sample));
226 }
227
228
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 264600 if (s->pos == s->asc_pos && !s->asc_changed)
229 ✗ s->asc_pos = -1;
230
231
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 264600 if (s->auto_release && s->asc_pos == -1 && peak > limit) {
232 ✗ s->asc -= peak;
233 ✗ s->asc_c--;
234 }
235
236 264600 s->att += s->delta;
237
238
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 793800 for (c = 0; c < channels; c++)
239 529200 dst[c] = buf[c] * s->att;
240
241
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 264600 if ((s->pos + channels) % buffer_size == nextpos[s->nextiter]) {
242
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 2019 if (s->auto_release) {
243 ✗ s->delta = get_rdelta(s, release, inlink->sample_rate,
244 peak, limit, s->att, 1);
245 ✗ if (s->nextlen > 1) {
246 ✗ double ppeak = 0, pdelta;
247 ✗ int pnextpos = nextpos[(s->nextiter + 1) % buffer_size];
248
249 ✗ for (c = 0; c < channels; c++) {
250 ✗ ppeak = FFMAX(ppeak, fabs(buffer[pnextpos + c]));
251 }
252 ✗ pdelta = (limit / ppeak - s->att) /
253 ✗ (((buffer_size + pnextpos -
254 ✗ ((s->pos + channels) % buffer_size)) %
255 ✗ buffer_size) / channels);
256 ✗ if (pdelta < s->delta)
257 ✗ s->delta = pdelta;
258 }
259 } else {
260 2019 s->delta = nextdelta[s->nextiter];
261 2019 s->att = limit / peak;
262 }
263
264 2019 s->nextlen -= 1;
265 2019 nextpos[s->nextiter] = -1;
266 2019 s->nextiter = (s->nextiter + 1) % buffer_size;
267 }
268
269
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 264600 if (s->att > 1.) {
270 11 s->att = 1.;
271 11 s->delta = 0.;
272 11 s->nextiter = 0;
273 11 s->nextlen = 0;
274 11 nextpos[0] = -1;
275 }
276
277
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 264600 if (s->att <= 0.) {
278 ✗ s->att = 0.0000000000001;
279 ✗ s->delta = (1.0 - s->att) / (inlink->sample_rate * release);
280 }
281
282
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 264600 if (s->att != 1. && (1. - s->att) < 0.0000000000001)
283 5 s->att = 1.;
284
285
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 264600 if (s->delta != 0. && fabs(s->delta) < 0.00000000000001)
286 ✗ s->delta = 0.;
287
288
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 793800 for (c = 0; c < channels; c++)
289 529200 dst[c] = av_clipd(dst[c], -limit, limit) * level * level_out;
290
291 264600 s->pos = (s->pos + channels) % buffer_size;
292 264600 src += channels;
293 264600 dst += channels;
294 }
295
296 65 in_duration = av_rescale_q(in->nb_samples, inlink->time_base, av_make_q(1, in->sample_rate));
297 65 in_pts = in->pts;
298 65 meta = (MetaItem){ in->pts, in->nb_samples };
299 65 av_fifo_write(s->fifo, &meta, 1);
300
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 65 if (in != out)
301 ✗ av_frame_free(&in);
302
303 65 new_out_samples = out->nb_samples;
304
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 65 if (s->in_trim > 0) {
305 ✗ int trim = FFMIN(new_out_samples, s->in_trim);
306 ✗ new_out_samples -= trim;
307 ✗ s->in_trim -= trim;
308 }
309
310
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 65 if (new_out_samples <= 0) {
311 ✗ av_frame_free(&out);
312 ✗ return 0;
313
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 65 } else if (new_out_samples < out->nb_samples) {
314 ✗ int offset = out->nb_samples - new_out_samples;
315 ✗ memmove(out->extended_data[0], out->extended_data[0] + sizeof(double) * offset * out->ch_layout.nb_channels,
316 ✗ sizeof(double) * new_out_samples * out->ch_layout.nb_channels);
317 ✗ out->nb_samples = new_out_samples;
318 ✗ s->in_trim = 0;
319 }
320
321 65 av_fifo_read(s->fifo, &meta, 1);
322
323 65 out_duration = av_rescale_q(out->nb_samples, inlink->time_base, av_make_q(1, out->sample_rate));
324 65 in_duration = av_rescale_q(meta.nb_samples, inlink->time_base, av_make_q(1, out->sample_rate));
325 65 in_pts = meta.pts;
326
327
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 65 if (s->next_out_pts != AV_NOPTS_VALUE && out->pts != s->next_out_pts &&
328 ✗ s->next_in_pts != AV_NOPTS_VALUE && in_pts == s->next_in_pts) {
329 ✗ out->pts = s->next_out_pts;
330 } else {
331 65 out->pts = in_pts;
332 }
333 65 s->next_in_pts = in_pts + in_duration;
334 65 s->next_out_pts = out->pts + out_duration;
335
336 65 return ff_filter_frame(outlink, out);
337 }
338
339 66 static int request_frame(AVFilterLink* outlink)
340 {
341 66 AVFilterContext *ctx = outlink->src;
342 66 AudioLimiterContext *s = (AudioLimiterContext*)ctx->priv;
343 int ret;
344
345 66 ret = ff_request_frame(ctx->inputs[0]);
346
347
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 66 if (ret == AVERROR_EOF && s->out_pad > 0) {
348 ✗ AVFrame *frame = ff_get_audio_buffer(outlink, FFMIN(1024, s->out_pad));
349 ✗ if (!frame)
350 ✗ return AVERROR(ENOMEM);
351
352 ✗ s->out_pad -= frame->nb_samples;
353 ✗ frame->pts = s->next_in_pts;
354 ✗ return filter_frame(ctx->inputs[0], frame);
355 }
356 66 return ret;
357 }
358
359 1 static int config_input(AVFilterLink *inlink)
360 {
361 1 AVFilterContext *ctx = inlink->dst;
362 1 AudioLimiterContext *s = ctx->priv;
363 int obuffer_size;
364
365 1 obuffer_size = inlink->sample_rate * inlink->ch_layout.nb_channels * 100 / 1000. + inlink->ch_layout.nb_channels;
366
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 1 if (obuffer_size < inlink->ch_layout.nb_channels)
367 ✗ return AVERROR(EINVAL);
368
369 1 s->buffer = av_calloc(obuffer_size, sizeof(*s->buffer));
370 1 s->nextdelta = av_calloc(obuffer_size, sizeof(*s->nextdelta));
371 1 s->nextpos = av_malloc_array(obuffer_size, sizeof(*s->nextpos));
372
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 1 if (!s->buffer || !s->nextdelta || !s->nextpos)
373 ✗ return AVERROR(ENOMEM);
374
375 1 memset(s->nextpos, -1, obuffer_size * sizeof(*s->nextpos));
376 1 s->buffer_size = inlink->sample_rate * s->attack * inlink->ch_layout.nb_channels;
377 1 s->buffer_size -= s->buffer_size % inlink->ch_layout.nb_channels;
378
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 1 if (s->latency)
379 ✗ s->in_trim = s->out_pad = s->buffer_size / inlink->ch_layout.nb_channels - 1;
380 1 s->next_out_pts = AV_NOPTS_VALUE;
381 1 s->next_in_pts = AV_NOPTS_VALUE;
382
383 1 s->fifo = av_fifo_alloc2(8, sizeof(MetaItem), AV_FIFO_FLAG_AUTO_GROW);
384
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 1 if (!s->fifo) {
385 ✗ return AVERROR(ENOMEM);
386 }
387
388
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 1 if (s->buffer_size <= 0) {
389 ✗ av_log(ctx, AV_LOG_ERROR, "Attack is too small.\n");
390 ✗ return AVERROR(EINVAL);
391 }
392
393 1 return 0;
394 }
395
396 2 static av_cold void uninit(AVFilterContext *ctx)
397 {
398 2 AudioLimiterContext *s = ctx->priv;
399
400 2 av_freep(&s->buffer);
401 2 av_freep(&s->nextdelta);
402 2 av_freep(&s->nextpos);
403
404 2 av_fifo_freep2(&s->fifo);
405 2 }
406
407 static const AVFilterPad alimiter_inputs[] = {
408 {
409 .name = "main",
410 .type = AVMEDIA_TYPE_AUDIO,
411 .filter_frame = filter_frame,
412 .config_props = config_input,
413 },
414 };
415
416 static const AVFilterPad alimiter_outputs[] = {
417 {
418 .name = "default",
419 .type = AVMEDIA_TYPE_AUDIO,
420 .request_frame = request_frame,
421 },
422 };
423
424 const AVFilter ff_af_alimiter = {
425 .name = "alimiter",
426 .description = NULL_IF_CONFIG_SMALL("Audio lookahead limiter."),
427 .priv_size = sizeof(AudioLimiterContext),
428 .priv_class = &alimiter_class,
429 .init = init,
430 .uninit = uninit,
431 FILTER_INPUTS(alimiter_inputs),
432 FILTER_OUTPUTS(alimiter_outputs),
433 FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBL),
434 .process_command = ff_filter_process_command,
435 .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC,
436 };
437