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path: root/libavfilter/af_alimiter.c (plain)
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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/avassert.h"
28#include "libavutil/channel_layout.h"
29#include "libavutil/common.h"
30#include "libavutil/opt.h"
31
32#include "audio.h"
33#include "avfilter.h"
34#include "formats.h"
35#include "internal.h"
36
37typedef struct AudioLimiterContext {
38 const AVClass *class;
39
40 double limit;
41 double attack;
42 double release;
43 double att;
44 double level_in;
45 double level_out;
46 int auto_release;
47 int auto_level;
48 double asc;
49 int asc_c;
50 int asc_pos;
51 double asc_coeff;
52
53 double *buffer;
54 int buffer_size;
55 int pos;
56 int *nextpos;
57 double *nextdelta;
58
59 double delta;
60 int nextiter;
61 int nextlen;
62 int asc_changed;
63} AudioLimiterContext;
64
65#define OFFSET(x) offsetof(AudioLimiterContext, x)
66#define A AV_OPT_FLAG_AUDIO_PARAM
67#define F AV_OPT_FLAG_FILTERING_PARAM
68
69static const AVOption alimiter_options[] = {
70 { "level_in", "set input level", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625, 64, A|F },
71 { "level_out", "set output level", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625, 64, A|F },
72 { "limit", "set limit", OFFSET(limit), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.0625, 1, A|F },
73 { "attack", "set attack", OFFSET(attack), AV_OPT_TYPE_DOUBLE, {.dbl=5}, 0.1, 80, A|F },
74 { "release", "set release", OFFSET(release), AV_OPT_TYPE_DOUBLE, {.dbl=50}, 1, 8000, A|F },
75 { "asc", "enable asc", OFFSET(auto_release), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A|F },
76 { "asc_level", "set asc level", OFFSET(asc_coeff), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 1, A|F },
77 { "level", "auto level", OFFSET(auto_level), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, A|F },
78 { NULL }
79};
80
81AVFILTER_DEFINE_CLASS(alimiter);
82
83static av_cold int init(AVFilterContext *ctx)
84{
85 AudioLimiterContext *s = ctx->priv;
86
87 s->attack /= 1000.;
88 s->release /= 1000.;
89 s->att = 1.;
90 s->asc_pos = -1;
91 s->asc_coeff = pow(0.5, s->asc_coeff - 0.5) * 2 * -1;
92
93 return 0;
94}
95
96static double get_rdelta(AudioLimiterContext *s, double release, int sample_rate,
97 double peak, double limit, double patt, int asc)
98{
99 double rdelta = (1.0 - patt) / (sample_rate * release);
100
101 if (asc && s->auto_release && s->asc_c > 0) {
102 double a_att = limit / (s->asc_coeff * s->asc) * (double)s->asc_c;
103
104 if (a_att > patt) {
105 double delta = FFMAX((a_att - patt) / (sample_rate * release), rdelta / 10);
106
107 if (delta < rdelta)
108 rdelta = delta;
109 }
110 }
111
112 return rdelta;
113}
114
115static int filter_frame(AVFilterLink *inlink, AVFrame *in)
116{
117 AVFilterContext *ctx = inlink->dst;
118 AudioLimiterContext *s = ctx->priv;
119 AVFilterLink *outlink = ctx->outputs[0];
120 const double *src = (const double *)in->data[0];
121 const int channels = inlink->channels;
122 const int buffer_size = s->buffer_size;
123 double *dst, *buffer = s->buffer;
124 const double release = s->release;
125 const double limit = s->limit;
126 double *nextdelta = s->nextdelta;
127 double level = s->auto_level ? 1 / limit : 1;
128 const double level_out = s->level_out;
129 const double level_in = s->level_in;
130 int *nextpos = s->nextpos;
131 AVFrame *out;
132 double *buf;
133 int n, c, i;
134
135 if (av_frame_is_writable(in)) {
136 out = in;
137 } else {
138 out = ff_get_audio_buffer(inlink, in->nb_samples);
139 if (!out) {
140 av_frame_free(&in);
141 return AVERROR(ENOMEM);
142 }
143 av_frame_copy_props(out, in);
144 }
145 dst = (double *)out->data[0];
146
147 for (n = 0; n < in->nb_samples; n++) {
148 double peak = 0;
149
150 for (c = 0; c < channels; c++) {
151 double sample = src[c] * level_in;
152
153 buffer[s->pos + c] = sample;
154 peak = FFMAX(peak, fabs(sample));
155 }
156
157 if (s->auto_release && peak > limit) {
158 s->asc += peak;
159 s->asc_c++;
160 }
161
162 if (peak > limit) {
163 double patt = FFMIN(limit / peak, 1.);
164 double rdelta = get_rdelta(s, release, inlink->sample_rate,
165 peak, limit, patt, 0);
166 double delta = (limit / peak - s->att) / buffer_size * channels;
167 int found = 0;
168
169 if (delta < s->delta) {
170 s->delta = delta;
171 nextpos[0] = s->pos;
172 nextpos[1] = -1;
173 nextdelta[0] = rdelta;
174 s->nextlen = 1;
175 s->nextiter= 0;
176 } else {
177 for (i = s->nextiter; i < s->nextiter + s->nextlen; i++) {
178 int j = i % buffer_size;
179 double ppeak, pdelta;
180
181 ppeak = fabs(buffer[nextpos[j]]) > fabs(buffer[nextpos[j] + 1]) ?
182 fabs(buffer[nextpos[j]]) : fabs(buffer[nextpos[j] + 1]);
183 pdelta = (limit / peak - limit / ppeak) / (((buffer_size - nextpos[j] + s->pos) % buffer_size) / channels);
184 if (pdelta < nextdelta[j]) {
185 nextdelta[j] = pdelta;
186 found = 1;
187 break;
188 }
189 }
190 if (found) {
191 s->nextlen = i - s->nextiter + 1;
192 nextpos[(s->nextiter + s->nextlen) % buffer_size] = s->pos;
193 nextdelta[(s->nextiter + s->nextlen) % buffer_size] = rdelta;
194 nextpos[(s->nextiter + s->nextlen + 1) % buffer_size] = -1;
195 s->nextlen++;
196 }
197 }
198 }
199
200 buf = &s->buffer[(s->pos + channels) % buffer_size];
201 peak = 0;
202 for (c = 0; c < channels; c++) {
203 double sample = buf[c];
204
205 peak = FFMAX(peak, fabs(sample));
206 }
207
208 if (s->pos == s->asc_pos && !s->asc_changed)
209 s->asc_pos = -1;
210
211 if (s->auto_release && s->asc_pos == -1 && peak > limit) {
212 s->asc -= peak;
213 s->asc_c--;
214 }
215
216 s->att += s->delta;
217
218 for (c = 0; c < channels; c++)
219 dst[c] = buf[c] * s->att;
220
221 if ((s->pos + channels) % buffer_size == nextpos[s->nextiter]) {
222 if (s->auto_release) {
223 s->delta = get_rdelta(s, release, inlink->sample_rate,
224 peak, limit, s->att, 1);
225 if (s->nextlen > 1) {
226 int pnextpos = nextpos[(s->nextiter + 1) % buffer_size];
227 double ppeak = fabs(buffer[pnextpos]) > fabs(buffer[pnextpos + 1]) ?
228 fabs(buffer[pnextpos]) :
229 fabs(buffer[pnextpos + 1]);
230 double pdelta = (limit / ppeak - s->att) /
231 (((buffer_size + pnextpos -
232 ((s->pos + channels) % buffer_size)) %
233 buffer_size) / channels);
234 if (pdelta < s->delta)
235 s->delta = pdelta;
236 }
237 } else {
238 s->delta = nextdelta[s->nextiter];
239 s->att = limit / peak;
240 }
241
242 s->nextlen -= 1;
243 nextpos[s->nextiter] = -1;
244 s->nextiter = (s->nextiter + 1) % buffer_size;
245 }
246
247 if (s->att > 1.) {
248 s->att = 1.;
249 s->delta = 0.;
250 s->nextiter = 0;
251 s->nextlen = 0;
252 nextpos[0] = -1;
253 }
254
255 if (s->att <= 0.) {
256 s->att = 0.0000000000001;
257 s->delta = (1.0 - s->att) / (inlink->sample_rate * release);
258 }
259
260 if (s->att != 1. && (1. - s->att) < 0.0000000000001)
261 s->att = 1.;
262
263 if (s->delta != 0. && fabs(s->delta) < 0.00000000000001)
264 s->delta = 0.;
265
266 for (c = 0; c < channels; c++)
267 dst[c] = av_clipd(dst[c], -limit, limit) * level * level_out;
268
269 s->pos = (s->pos + channels) % buffer_size;
270 src += channels;
271 dst += channels;
272 }
273
274 if (in != out)
275 av_frame_free(&in);
276
277 return ff_filter_frame(outlink, out);
278}
279
280static int query_formats(AVFilterContext *ctx)
281{
282 AVFilterFormats *formats;
283 AVFilterChannelLayouts *layouts;
284 static const enum AVSampleFormat sample_fmts[] = {
285 AV_SAMPLE_FMT_DBL,
286 AV_SAMPLE_FMT_NONE
287 };
288 int ret;
289
290 layouts = ff_all_channel_counts();
291 if (!layouts)
292 return AVERROR(ENOMEM);
293 ret = ff_set_common_channel_layouts(ctx, layouts);
294 if (ret < 0)
295 return ret;
296
297 formats = ff_make_format_list(sample_fmts);
298 if (!formats)
299 return AVERROR(ENOMEM);
300 ret = ff_set_common_formats(ctx, formats);
301 if (ret < 0)
302 return ret;
303
304 formats = ff_all_samplerates();
305 if (!formats)
306 return AVERROR(ENOMEM);
307 return ff_set_common_samplerates(ctx, formats);
308}
309
310static int config_input(AVFilterLink *inlink)
311{
312 AVFilterContext *ctx = inlink->dst;
313 AudioLimiterContext *s = ctx->priv;
314 int obuffer_size;
315
316 obuffer_size = inlink->sample_rate * inlink->channels * 100 / 1000. + inlink->channels;
317 if (obuffer_size < inlink->channels)
318 return AVERROR(EINVAL);
319
320 s->buffer = av_calloc(obuffer_size, sizeof(*s->buffer));
321 s->nextdelta = av_calloc(obuffer_size, sizeof(*s->nextdelta));
322 s->nextpos = av_malloc_array(obuffer_size, sizeof(*s->nextpos));
323 if (!s->buffer || !s->nextdelta || !s->nextpos)
324 return AVERROR(ENOMEM);
325
326 memset(s->nextpos, -1, obuffer_size * sizeof(*s->nextpos));
327 s->buffer_size = inlink->sample_rate * s->attack * inlink->channels;
328 s->buffer_size -= s->buffer_size % inlink->channels;
329
330 return 0;
331}
332
333static av_cold void uninit(AVFilterContext *ctx)
334{
335 AudioLimiterContext *s = ctx->priv;
336
337 av_freep(&s->buffer);
338 av_freep(&s->nextdelta);
339 av_freep(&s->nextpos);
340}
341
342static const AVFilterPad alimiter_inputs[] = {
343 {
344 .name = "main",
345 .type = AVMEDIA_TYPE_AUDIO,
346 .filter_frame = filter_frame,
347 .config_props = config_input,
348 },
349 { NULL }
350};
351
352static const AVFilterPad alimiter_outputs[] = {
353 {
354 .name = "default",
355 .type = AVMEDIA_TYPE_AUDIO,
356 },
357 { NULL }
358};
359
360AVFilter ff_af_alimiter = {
361 .name = "alimiter",
362 .description = NULL_IF_CONFIG_SMALL("Audio lookahead limiter."),
363 .priv_size = sizeof(AudioLimiterContext),
364 .priv_class = &alimiter_class,
365 .init = init,
366 .uninit = uninit,
367 .query_formats = query_formats,
368 .inputs = alimiter_inputs,
369 .outputs = alimiter_outputs,
370};
371
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