2 * Copyright (C) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen and others
3 * Copyright (c) 2015 Paul B Mahol
5 * This file is part of FFmpeg.
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.
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.
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
24 * Lookahead limiter filter
27 #include "libavutil/avassert.h"
28 #include "libavutil/channel_layout.h"
29 #include "libavutil/common.h"
30 #include "libavutil/opt.h"
37 typedef struct AudioLimiterContext {
63 } AudioLimiterContext;
65 #define OFFSET(x) offsetof(AudioLimiterContext, x)
66 #define A AV_OPT_FLAG_AUDIO_PARAM
67 #define F AV_OPT_FLAG_FILTERING_PARAM
69 static 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 },
81 AVFILTER_DEFINE_CLASS(alimiter);
83 static av_cold int init(AVFilterContext *ctx)
85 AudioLimiterContext *s = ctx->priv;
91 s->asc_coeff = pow(0.5, s->asc_coeff - 0.5) * 2 * -1;
96 static double get_rdelta(AudioLimiterContext *s, double release, int sample_rate,
97 double peak, double limit, double patt, int asc)
99 double rdelta = (1.0 - patt) / (sample_rate * release);
101 if (asc && s->auto_release && s->asc_c > 0) {
102 double a_att = limit / (s->asc_coeff * s->asc) * (double)s->asc_c;
105 double delta = FFMAX((a_att - patt) / (sample_rate * release), rdelta / 10);
115 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
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;
135 if (av_frame_is_writable(in)) {
138 out = ff_get_audio_buffer(outlink, in->nb_samples);
141 return AVERROR(ENOMEM);
143 av_frame_copy_props(out, in);
145 dst = (double *)out->data[0];
147 for (n = 0; n < in->nb_samples; n++) {
150 for (c = 0; c < channels; c++) {
151 double sample = src[c] * level_in;
153 buffer[s->pos + c] = sample;
154 peak = FFMAX(peak, fabs(sample));
157 if (s->auto_release && 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;
169 if (delta < s->delta) {
173 nextdelta[0] = rdelta;
177 for (i = s->nextiter; i < s->nextiter + s->nextlen; i++) {
178 int j = i % buffer_size;
179 double ppeak, pdelta;
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;
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;
200 buf = &s->buffer[(s->pos + channels) % buffer_size];
202 for (c = 0; c < channels; c++) {
203 double sample = buf[c];
205 peak = FFMAX(peak, fabs(sample));
208 if (s->pos == s->asc_pos && !s->asc_changed)
211 if (s->auto_release && s->asc_pos == -1 && peak > limit) {
218 for (c = 0; c < channels; c++)
219 dst[c] = buf[c] * s->att;
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)
238 s->delta = nextdelta[s->nextiter];
239 s->att = limit / peak;
243 nextpos[s->nextiter] = -1;
244 s->nextiter = (s->nextiter + 1) % buffer_size;
256 s->att = 0.0000000000001;
257 s->delta = (1.0 - s->att) / (inlink->sample_rate * release);
260 if (s->att != 1. && (1. - s->att) < 0.0000000000001)
263 if (s->delta != 0. && fabs(s->delta) < 0.00000000000001)
266 for (c = 0; c < channels; c++)
267 dst[c] = av_clipd(dst[c], -limit, limit) * level * level_out;
269 s->pos = (s->pos + channels) % buffer_size;
277 return ff_filter_frame(outlink, out);
280 static int query_formats(AVFilterContext *ctx)
282 AVFilterFormats *formats;
283 AVFilterChannelLayouts *layouts;
284 static const enum AVSampleFormat sample_fmts[] = {
290 layouts = ff_all_channel_counts();
292 return AVERROR(ENOMEM);
293 ret = ff_set_common_channel_layouts(ctx, layouts);
297 formats = ff_make_format_list(sample_fmts);
299 return AVERROR(ENOMEM);
300 ret = ff_set_common_formats(ctx, formats);
304 formats = ff_all_samplerates();
306 return AVERROR(ENOMEM);
307 return ff_set_common_samplerates(ctx, formats);
310 static int config_input(AVFilterLink *inlink)
312 AVFilterContext *ctx = inlink->dst;
313 AudioLimiterContext *s = ctx->priv;
316 obuffer_size = inlink->sample_rate * inlink->channels * 100 / 1000. + inlink->channels;
317 if (obuffer_size < inlink->channels)
318 return AVERROR(EINVAL);
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);
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;
330 if (s->buffer_size <= 0) {
331 av_log(ctx, AV_LOG_ERROR, "Attack is too small.\n");
332 return AVERROR(EINVAL);
338 static av_cold void uninit(AVFilterContext *ctx)
340 AudioLimiterContext *s = ctx->priv;
342 av_freep(&s->buffer);
343 av_freep(&s->nextdelta);
344 av_freep(&s->nextpos);
347 static const AVFilterPad alimiter_inputs[] = {
350 .type = AVMEDIA_TYPE_AUDIO,
351 .filter_frame = filter_frame,
352 .config_props = config_input,
357 static const AVFilterPad alimiter_outputs[] = {
360 .type = AVMEDIA_TYPE_AUDIO,
365 AVFilter ff_af_alimiter = {
367 .description = NULL_IF_CONFIG_SMALL("Audio lookahead limiter."),
368 .priv_size = sizeof(AudioLimiterContext),
369 .priv_class = &alimiter_class,
372 .query_formats = query_formats,
373 .inputs = alimiter_inputs,
374 .outputs = alimiter_outputs,