362 lines
11 KiB
C
362 lines
11 KiB
C
/*
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* Copyright (C) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen and others
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* Copyright (c) 2015 Paul B Mahol
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* Lookahead limiter filter
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*/
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#include "libavutil/avassert.h"
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#include "libavutil/channel_layout.h"
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#include "libavutil/common.h"
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#include "libavutil/opt.h"
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#include "audio.h"
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#include "avfilter.h"
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#include "formats.h"
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#include "internal.h"
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typedef struct AudioLimiterContext {
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const AVClass *class;
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double limit;
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double attack;
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double release;
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double att;
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int auto_release;
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double asc;
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int asc_c;
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int asc_pos;
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double asc_coeff;
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double *buffer;
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int buffer_size;
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int pos;
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int *nextpos;
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double *nextdelta;
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double delta;
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int nextiter;
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int nextlen;
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int asc_changed;
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} AudioLimiterContext;
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#define OFFSET(x) offsetof(AudioLimiterContext, x)
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#define A AV_OPT_FLAG_AUDIO_PARAM
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#define F AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption alimiter_options[] = {
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{ "limit", "set limit", OFFSET(limit), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.0625, 1, A|F },
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{ "attack", "set attack", OFFSET(attack), AV_OPT_TYPE_DOUBLE, {.dbl=5}, 0.1, 80, A|F },
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{ "release", "set release", OFFSET(release), AV_OPT_TYPE_DOUBLE, {.dbl=50}, 1, 8000, A|F },
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{ "asc", "enable asc", OFFSET(auto_release), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A|F },
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{ "asc_level", "set asc level", OFFSET(asc_coeff), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 1, A|F },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(alimiter);
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static av_cold int init(AVFilterContext *ctx)
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{
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AudioLimiterContext *s = ctx->priv;
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s->attack /= 1000.;
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s->release /= 1000.;
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s->att = 1.;
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s->asc_pos = -1;
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s->asc_coeff = pow(0.5, s->asc_coeff - 0.5) * 2 * -1;
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return 0;
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}
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static double get_rdelta(AudioLimiterContext *s, double release, int sample_rate,
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double peak, double limit, double patt, int asc)
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{
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double rdelta = (1.0 - patt) / (sample_rate * release);
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if (asc && s->auto_release && s->asc_c > 0) {
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double a_att = limit / (s->asc_coeff * s->asc) * (double)s->asc_c;
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if (a_att > patt) {
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double delta = FFMAX((a_att - patt) / (sample_rate * release), rdelta / 10);
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if (delta < rdelta)
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rdelta = delta;
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}
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}
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return rdelta;
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}
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static int filter_frame(AVFilterLink *inlink, AVFrame *in)
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{
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AVFilterContext *ctx = inlink->dst;
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AudioLimiterContext *s = ctx->priv;
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AVFilterLink *outlink = ctx->outputs[0];
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const double *src = (const double *)in->data[0];
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const int channels = inlink->channels;
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const int buffer_size = s->buffer_size;
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double *dst, *buffer = s->buffer;
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const double release = s->release;
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const double limit = s->limit;
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double *nextdelta = s->nextdelta;
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int *nextpos = s->nextpos;
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AVFrame *out;
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double *buf;
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int n, c, i;
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if (av_frame_is_writable(in)) {
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out = in;
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} else {
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out = ff_get_audio_buffer(inlink, in->nb_samples);
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if (!out) {
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av_frame_free(&in);
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return AVERROR(ENOMEM);
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}
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av_frame_copy_props(out, in);
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}
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dst = (double *)out->data[0];
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for (n = 0; n < in->nb_samples; n++) {
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double peak = 0;
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for (c = 0; c < channels; c++) {
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double sample = src[c];
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buffer[s->pos + c] = sample;
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peak = FFMAX(peak, fabs(sample));
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}
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if (s->auto_release && peak > limit) {
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s->asc += peak;
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s->asc_c++;
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}
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if (peak > limit) {
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double patt = FFMIN(limit / peak, 1.);
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double rdelta = get_rdelta(s, release, inlink->sample_rate,
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peak, limit, patt, 0);
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double delta = (limit / peak - s->att) / buffer_size * channels;
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int found = 0;
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if (delta < s->delta) {
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s->delta = delta;
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nextpos[0] = s->pos;
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nextpos[1] = -1;
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nextdelta[0] = rdelta;
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s->nextlen = 1;
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s->nextiter= 0;
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} else {
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for (i = s->nextiter; i < s->nextiter + s->nextlen; i++) {
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int j = i % buffer_size;
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double ppeak, pdelta;
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ppeak = fabs(buffer[nextpos[j]]) > fabs(buffer[nextpos[j] + 1]) ?
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fabs(buffer[nextpos[j]]) : fabs(buffer[nextpos[j] + 1]);
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pdelta = (limit / peak - limit / ppeak) / (((buffer_size - nextpos[j] + s->pos) % buffer_size) / channels);
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if (pdelta < nextdelta[j]) {
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nextdelta[j] = pdelta;
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found = 1;
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break;
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}
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}
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if (found) {
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s->nextlen = i - s->nextiter + 1;
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nextpos[(s->nextiter + s->nextlen) % buffer_size] = s->pos;
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nextdelta[(s->nextiter + s->nextlen) % buffer_size] = rdelta;
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nextpos[(s->nextiter + s->nextlen + 1) % buffer_size] = -1;
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s->nextlen++;
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}
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}
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}
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buf = &s->buffer[(s->pos + channels) % buffer_size];
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peak = 0;
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for (c = 0; c < channels; c++) {
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double sample = buf[c];
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peak = FFMAX(peak, fabs(sample));
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}
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if (s->pos == s->asc_pos && !s->asc_changed)
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s->asc_pos = -1;
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if (s->auto_release && s->asc_pos == -1 && peak > limit) {
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s->asc -= peak;
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s->asc_c--;
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}
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s->att += s->delta;
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for (c = 0; c < channels; c++)
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dst[c] = buf[c] * s->att;
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if ((s->pos + channels) % buffer_size == nextpos[s->nextiter]) {
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if (s->auto_release) {
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s->delta = get_rdelta(s, release, inlink->sample_rate,
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peak, limit, s->att, 1);
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if (s->nextlen > 1) {
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int pnextpos = nextpos[(s->nextiter + 1) % buffer_size];
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double ppeak = fabs(buffer[pnextpos]) > fabs(buffer[pnextpos + 1]) ?
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fabs(buffer[pnextpos]) :
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fabs(buffer[pnextpos + 1]);
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double pdelta = (limit / ppeak - s->att) /
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(((buffer_size + pnextpos -
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((s->pos + channels) % buffer_size)) %
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buffer_size) / channels);
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if (pdelta < s->delta)
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s->delta = pdelta;
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}
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} else {
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s->delta = nextdelta[s->nextiter];
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s->att = limit / peak;
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}
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s->nextlen -= 1;
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nextpos[s->nextiter] = -1;
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s->nextiter = (s->nextiter + 1) % buffer_size;
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}
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if (s->att > 1.) {
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s->att = 1.;
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s->delta = 0.;
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s->nextiter = 0;
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s->nextlen = 0;
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nextpos[0] = -1;
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}
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if (s->att <= 0.) {
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s->att = 0.0000000000001;
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s->delta = (1.0 - s->att) / (inlink->sample_rate * release);
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}
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if (s->att != 1. && (1. - s->att) < 0.0000000000001)
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s->att = 1.;
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if (s->delta != 0. && fabs(s->delta) < 0.00000000000001)
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s->delta = 0.;
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for (c = 0; c < channels; c++)
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dst[c] = av_clipd(dst[c], -limit, limit);
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s->pos = (s->pos + channels) % buffer_size;
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src += channels;
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dst += channels;
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}
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if (in != out)
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av_frame_free(&in);
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return ff_filter_frame(outlink, out);
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}
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static int query_formats(AVFilterContext *ctx)
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{
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AVFilterFormats *formats;
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AVFilterChannelLayouts *layouts;
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static const enum AVSampleFormat sample_fmts[] = {
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AV_SAMPLE_FMT_DBL,
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AV_SAMPLE_FMT_NONE
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};
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int ret;
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layouts = ff_all_channel_counts();
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if (!layouts)
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return AVERROR(ENOMEM);
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ret = ff_set_common_channel_layouts(ctx, layouts);
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if (ret < 0)
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return ret;
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formats = ff_make_format_list(sample_fmts);
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if (!formats)
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return AVERROR(ENOMEM);
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ret = ff_set_common_formats(ctx, formats);
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if (ret < 0)
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return ret;
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formats = ff_all_samplerates();
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if (!formats)
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return AVERROR(ENOMEM);
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return ff_set_common_samplerates(ctx, formats);
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}
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static int config_input(AVFilterLink *inlink)
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{
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AVFilterContext *ctx = inlink->dst;
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AudioLimiterContext *s = ctx->priv;
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int obuffer_size;
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obuffer_size = inlink->sample_rate * inlink->channels * 100 / 1000. + inlink->channels;
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if (obuffer_size < inlink->channels)
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return AVERROR(EINVAL);
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s->buffer = av_calloc(obuffer_size, sizeof(*s->buffer));
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s->nextdelta = av_calloc(obuffer_size, sizeof(*s->nextdelta));
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s->nextpos = av_malloc_array(obuffer_size, sizeof(*s->nextpos));
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if (!s->buffer || !s->nextdelta || !s->nextpos)
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return AVERROR(ENOMEM);
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memset(s->nextpos, -1, obuffer_size * sizeof(*s->nextpos));
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s->buffer_size = inlink->sample_rate * s->attack * inlink->channels;
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s->buffer_size -= s->buffer_size % inlink->channels;
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return 0;
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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AudioLimiterContext *s = ctx->priv;
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av_freep(&s->buffer);
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av_freep(&s->nextdelta);
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av_freep(&s->nextpos);
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}
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static const AVFilterPad alimiter_inputs[] = {
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{
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.name = "main",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_frame = filter_frame,
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.config_props = config_input,
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},
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{ NULL }
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};
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static const AVFilterPad alimiter_outputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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},
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{ NULL }
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};
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AVFilter ff_af_alimiter = {
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.name = "alimiter",
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.description = NULL_IF_CONFIG_SMALL("Lookahead limiter."),
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.priv_size = sizeof(AudioLimiterContext),
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.priv_class = &alimiter_class,
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.init = init,
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.uninit = uninit,
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.query_formats = query_formats,
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.inputs = alimiter_inputs,
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.outputs = alimiter_outputs,
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};
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