mirror of
https://github.com/FFmpeg/FFmpeg.git
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246 lines
8.3 KiB
C
246 lines
8.3 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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#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 "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 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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double thres;
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double knee_start;
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double lin_knee_stop;
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double 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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} 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
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static const AVOption agate_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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{ "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=0}, 0, 1, A, "detection" },
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{ "peak", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "detection" },
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{ "rms", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "detection" },
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{ "link", "set link", OFFSET(link), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A, "link" },
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{ "average", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "link" },
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{ "maximum", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "link" },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(agate);
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static int query_formats(AVFilterContext *ctx)
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{
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AVFilterFormats *formats = NULL;
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AVFilterChannelLayouts *layouts;
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int ret;
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if ((ret = ff_add_format(&formats, AV_SAMPLE_FMT_DBL)) < 0)
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return ret;
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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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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_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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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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double lin_knee_start;
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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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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(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 lin_knee_stop, double range)
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{
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if (lin_slope < lin_knee_stop) {
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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 (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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return 1.;
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}
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static void gate(AudioGateContext *s, const double *src, double *dst, const double *scsrc,
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int nb_samples, AVFilterLink *inlink, AVFilterLink *sclink)
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{
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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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const double level_in = s->level_in;
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int n, c;
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for (n = 0; n < nb_samples; n++, src += inlink->channels, dst += inlink->channels, scsrc += sclink->channels) {
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double abs_sample = fabs(scsrc[0]), gain = 1.0;
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for (c = 0; c < inlink->channels; c++)
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dst[c] = src[c] * level_in;
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if (s->link == 1) {
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for (c = 1; c < sclink->channels; c++)
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abs_sample = FFMAX(fabs(scsrc[c]), abs_sample);
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} else {
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for (c = 1; c < sclink->channels; c++)
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abs_sample += fabs(scsrc[c]);
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abs_sample /= sclink->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->lin_slope > 0.0)
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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->lin_knee_stop, s->range);
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for (c = 0; c < inlink->channels; c++)
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dst[c] *= gain * makeup;
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}
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}
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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(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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gate(s, src, dst, src, in->nb_samples, 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 = config_input,
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},
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{ NULL }
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};
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static const AVFilterPad 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_agate = {
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.name = "agate",
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.description = NULL_IF_CONFIG_SMALL("Audio gate."),
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.query_formats = query_formats,
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.priv_size = sizeof(AudioGateContext),
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.priv_class = &agate_class,
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.inputs = inputs,
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.outputs = outputs,
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};
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