393 lines
14 KiB
C
393 lines
14 KiB
C
/*
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* Copyright (C) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen, Damien Zammit
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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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* Audio (Sidechain) Gate filter
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*/
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#include "config_components.h"
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#include "libavutil/audio_fifo.h"
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#include "libavutil/channel_layout.h"
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#include "libavutil/opt.h"
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#include "avfilter.h"
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#include "audio.h"
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#include "filters.h"
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#include "formats.h"
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#include "hermite.h"
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typedef struct AudioGateContext {
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const AVClass *class;
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double level_in;
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double level_sc;
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double attack;
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double release;
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double threshold;
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double ratio;
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double knee;
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double makeup;
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double range;
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int link;
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int detection;
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int mode;
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double thres;
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double knee_start;
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double knee_stop;
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double lin_knee_start;
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double lin_knee_stop;
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double lin_slope;
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double attack_coeff;
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double release_coeff;
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AVAudioFifo *fifo[2];
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int64_t pts;
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} AudioGateContext;
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#define OFFSET(x) offsetof(AudioGateContext, x)
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#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
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static const AVOption options[] = {
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{ "level_in", "set input level", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
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{ "mode", "set mode", OFFSET(mode), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A, .unit = "mode" },
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{ "downward",0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, .unit = "mode" },
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{ "upward", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, .unit = "mode" },
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{ "range", "set max gain reduction", OFFSET(range), AV_OPT_TYPE_DOUBLE, {.dbl=0.06125}, 0, 1, A },
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{ "threshold", "set threshold", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=0.125}, 0, 1, A },
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{ "ratio", "set ratio", OFFSET(ratio), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 1, 9000, A },
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{ "attack", "set attack", OFFSET(attack), AV_OPT_TYPE_DOUBLE, {.dbl=20}, 0.01, 9000, A },
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{ "release", "set release", OFFSET(release), AV_OPT_TYPE_DOUBLE, {.dbl=250}, 0.01, 9000, A },
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{ "makeup", "set makeup gain", OFFSET(makeup), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 1, 64, A },
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{ "knee", "set knee", OFFSET(knee), AV_OPT_TYPE_DOUBLE, {.dbl=2.828427125}, 1, 8, A },
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{ "detection", "set detection", OFFSET(detection), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, A, .unit = "detection" },
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{ "peak", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, .unit = "detection" },
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{ "rms", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, .unit = "detection" },
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{ "link", "set link", OFFSET(link), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A, .unit = "link" },
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{ "average", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, .unit = "link" },
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{ "maximum", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, .unit = "link" },
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{ "level_sc", "set sidechain gain", OFFSET(level_sc), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS_EXT(agate_sidechaingate, "agate/sidechaingate", options);
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static int agate_config_input(AVFilterLink *inlink)
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{
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AVFilterContext *ctx = inlink->dst;
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AudioGateContext *s = ctx->priv;
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double lin_threshold = s->threshold;
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double lin_knee_sqrt = sqrt(s->knee);
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if (s->detection)
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lin_threshold *= lin_threshold;
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s->attack_coeff = FFMIN(1., 1. / (s->attack * inlink->sample_rate / 4000.));
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s->release_coeff = FFMIN(1., 1. / (s->release * inlink->sample_rate / 4000.));
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s->lin_knee_stop = lin_threshold * lin_knee_sqrt;
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s->lin_knee_start = lin_threshold / lin_knee_sqrt;
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s->thres = log(lin_threshold);
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s->knee_start = log(s->lin_knee_start);
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s->knee_stop = log(s->lin_knee_stop);
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return 0;
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}
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// A fake infinity value (because real infinity may break some hosts)
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#define FAKE_INFINITY (65536.0 * 65536.0)
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// Check for infinity (with appropriate-ish tolerance)
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#define IS_FAKE_INFINITY(value) (fabs(value-FAKE_INFINITY) < 1.0)
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static double output_gain(double lin_slope, double ratio, double thres,
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double knee, double knee_start, double knee_stop,
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double range, int mode)
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{
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double slope = log(lin_slope);
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double tratio = ratio;
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double gain = 0.;
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double delta = 0.;
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if (IS_FAKE_INFINITY(ratio))
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tratio = 1000.;
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gain = (slope - thres) * tratio + thres;
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delta = tratio;
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if (mode) {
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if (knee > 1. && slope < knee_stop)
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gain = hermite_interpolation(slope, knee_stop, knee_start, ((knee_stop - thres) * tratio + thres), knee_start, delta, 1.);
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} else {
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if (knee > 1. && slope > knee_start)
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gain = hermite_interpolation(slope, knee_start, knee_stop, ((knee_start - thres) * tratio + thres), knee_stop, delta, 1.);
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}
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return FFMAX(range, exp(gain - slope));
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}
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static void gate(AudioGateContext *s,
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const double *src, double *dst, const double *scsrc,
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int nb_samples, double level_in, double level_sc,
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AVFilterLink *inlink, AVFilterLink *sclink)
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{
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AVFilterContext *ctx = inlink->dst;
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const double makeup = s->makeup;
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const double attack_coeff = s->attack_coeff;
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const double release_coeff = s->release_coeff;
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int n, c;
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for (n = 0; n < nb_samples; n++, src += inlink->ch_layout.nb_channels, dst += inlink->ch_layout.nb_channels, scsrc += sclink->ch_layout.nb_channels) {
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double abs_sample = fabs(scsrc[0] * level_sc), gain = 1.0;
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double factor;
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int detected;
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if (s->link == 1) {
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for (c = 1; c < sclink->ch_layout.nb_channels; c++)
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abs_sample = FFMAX(fabs(scsrc[c] * level_sc), abs_sample);
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} else {
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for (c = 1; c < sclink->ch_layout.nb_channels; c++)
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abs_sample += fabs(scsrc[c] * level_sc);
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abs_sample /= sclink->ch_layout.nb_channels;
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}
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if (s->detection)
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abs_sample *= abs_sample;
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s->lin_slope += (abs_sample - s->lin_slope) * (abs_sample > s->lin_slope ? attack_coeff : release_coeff);
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if (s->mode)
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detected = s->lin_slope > s->lin_knee_start;
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else
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detected = s->lin_slope < s->lin_knee_stop;
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if (s->lin_slope > 0.0 && detected)
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gain = output_gain(s->lin_slope, s->ratio, s->thres,
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s->knee, s->knee_start, s->knee_stop,
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s->range, s->mode);
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factor = ctx->is_disabled ? 1.f : level_in * gain * makeup;
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for (c = 0; c < inlink->ch_layout.nb_channels; c++)
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dst[c] = src[c] * factor;
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}
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}
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#if CONFIG_AGATE_FILTER
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static int filter_frame(AVFilterLink *inlink, AVFrame *in)
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{
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const double *src = (const double *)in->data[0];
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AVFilterContext *ctx = inlink->dst;
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AVFilterLink *outlink = ctx->outputs[0];
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AudioGateContext *s = ctx->priv;
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AVFrame *out;
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double *dst;
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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(outlink, 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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gate(s, src, dst, src, in->nb_samples,
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s->level_in, s->level_in, inlink, inlink);
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if (out != in)
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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 const AVFilterPad inputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_frame = filter_frame,
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.config_props = agate_config_input,
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},
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};
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const AVFilter ff_af_agate = {
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.name = "agate",
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.description = NULL_IF_CONFIG_SMALL("Audio gate."),
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.priv_class = &agate_sidechaingate_class,
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.priv_size = sizeof(AudioGateContext),
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FILTER_INPUTS(inputs),
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FILTER_OUTPUTS(ff_audio_default_filterpad),
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FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBL),
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.process_command = ff_filter_process_command,
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.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC,
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};
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#endif /* CONFIG_AGATE_FILTER */
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#if CONFIG_SIDECHAINGATE_FILTER
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static int activate(AVFilterContext *ctx)
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{
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AudioGateContext *s = ctx->priv;
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AVFrame *out = NULL, *in[2] = { NULL };
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int ret, i, nb_samples;
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double *dst;
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FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
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if ((ret = ff_inlink_consume_frame(ctx->inputs[0], &in[0])) > 0) {
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av_audio_fifo_write(s->fifo[0], (void **)in[0]->extended_data,
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in[0]->nb_samples);
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av_frame_free(&in[0]);
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}
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if (ret < 0)
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return ret;
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if ((ret = ff_inlink_consume_frame(ctx->inputs[1], &in[1])) > 0) {
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av_audio_fifo_write(s->fifo[1], (void **)in[1]->extended_data,
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in[1]->nb_samples);
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av_frame_free(&in[1]);
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}
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if (ret < 0)
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return ret;
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nb_samples = FFMIN(av_audio_fifo_size(s->fifo[0]), av_audio_fifo_size(s->fifo[1]));
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if (nb_samples) {
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out = ff_get_audio_buffer(ctx->outputs[0], nb_samples);
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if (!out)
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return AVERROR(ENOMEM);
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for (i = 0; i < 2; i++) {
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in[i] = ff_get_audio_buffer(ctx->inputs[i], nb_samples);
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if (!in[i]) {
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av_frame_free(&in[0]);
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av_frame_free(&in[1]);
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av_frame_free(&out);
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return AVERROR(ENOMEM);
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}
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av_audio_fifo_read(s->fifo[i], (void **)in[i]->data, nb_samples);
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}
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dst = (double *)out->data[0];
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out->pts = s->pts;
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s->pts += av_rescale_q(nb_samples, (AVRational){1, ctx->outputs[0]->sample_rate}, ctx->outputs[0]->time_base);
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gate(s, (double *)in[0]->data[0], dst,
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(double *)in[1]->data[0], nb_samples,
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s->level_in, s->level_sc,
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ctx->inputs[0], ctx->inputs[1]);
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av_frame_free(&in[0]);
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av_frame_free(&in[1]);
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ret = ff_filter_frame(ctx->outputs[0], out);
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if (ret < 0)
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return ret;
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}
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FF_FILTER_FORWARD_STATUS(ctx->inputs[0], ctx->outputs[0]);
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FF_FILTER_FORWARD_STATUS(ctx->inputs[1], ctx->outputs[0]);
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if (ff_outlink_frame_wanted(ctx->outputs[0])) {
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if (!av_audio_fifo_size(s->fifo[0]))
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ff_inlink_request_frame(ctx->inputs[0]);
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if (!av_audio_fifo_size(s->fifo[1]))
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ff_inlink_request_frame(ctx->inputs[1]);
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}
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return 0;
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}
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static int scquery_formats(AVFilterContext *ctx)
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{
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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 = ff_channel_layouts_ref(ff_all_channel_counts(),
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&ctx->inputs[1]->outcfg.channel_layouts);
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if (ret < 0)
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return ret;
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/* This will link the channel properties of the main input and the output;
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* it won't touch the second input as its channel_layouts is already set. */
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if ((ret = ff_set_common_all_channel_counts(ctx)) < 0)
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return ret;
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if ((ret = ff_set_common_formats_from_list(ctx, sample_fmts)) < 0)
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return ret;
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return ff_set_common_all_samplerates(ctx);
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}
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static int scconfig_output(AVFilterLink *outlink)
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{
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AVFilterContext *ctx = outlink->src;
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AudioGateContext *s = ctx->priv;
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outlink->time_base = ctx->inputs[0]->time_base;
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s->fifo[0] = av_audio_fifo_alloc(ctx->inputs[0]->format, ctx->inputs[0]->ch_layout.nb_channels, 1024);
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s->fifo[1] = av_audio_fifo_alloc(ctx->inputs[1]->format, ctx->inputs[1]->ch_layout.nb_channels, 1024);
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if (!s->fifo[0] || !s->fifo[1])
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return AVERROR(ENOMEM);
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agate_config_input(ctx->inputs[0]);
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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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AudioGateContext *s = ctx->priv;
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av_audio_fifo_free(s->fifo[0]);
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av_audio_fifo_free(s->fifo[1]);
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}
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static const AVFilterPad sidechaingate_inputs[] = {
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{
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.name = "main",
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.type = AVMEDIA_TYPE_AUDIO,
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},{
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.name = "sidechain",
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.type = AVMEDIA_TYPE_AUDIO,
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},
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};
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static const AVFilterPad sidechaingate_outputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.config_props = scconfig_output,
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},
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};
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const AVFilter ff_af_sidechaingate = {
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.name = "sidechaingate",
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.description = NULL_IF_CONFIG_SMALL("Audio sidechain gate."),
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.priv_class = &agate_sidechaingate_class,
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.priv_size = sizeof(AudioGateContext),
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.activate = activate,
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.uninit = uninit,
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FILTER_INPUTS(sidechaingate_inputs),
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FILTER_OUTPUTS(sidechaingate_outputs),
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FILTER_QUERY_FUNC(scquery_formats),
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.process_command = ff_filter_process_command,
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.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
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};
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#endif /* CONFIG_SIDECHAINGATE_FILTER */
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