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mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-23 12:43:46 +02:00
FFmpeg/libavfilter/af_agate.c
Anton Khirnov f143723095 lavfi/af_agate: convert to query_func2()
Drop redundant ff_set_common_all_channel_counts() /
ff_set_common_all_samplerates() calls, since those happen implicitly in
generic code.
2024-09-05 10:03:33 +02:00

395 lines
14 KiB
C

/*
* Copyright (C) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen, Damien Zammit
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* Audio (Sidechain) Gate filter
*/
#include "config_components.h"
#include "libavutil/audio_fifo.h"
#include "libavutil/channel_layout.h"
#include "libavutil/opt.h"
#include "avfilter.h"
#include "audio.h"
#include "filters.h"
#include "formats.h"
#include "hermite.h"
typedef struct AudioGateContext {
const AVClass *class;
double level_in;
double level_sc;
double attack;
double release;
double threshold;
double ratio;
double knee;
double makeup;
double range;
int link;
int detection;
int mode;
double thres;
double knee_start;
double knee_stop;
double lin_knee_start;
double lin_knee_stop;
double lin_slope;
double attack_coeff;
double release_coeff;
AVAudioFifo *fifo[2];
int64_t pts;
} AudioGateContext;
#define OFFSET(x) offsetof(AudioGateContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
static const AVOption options[] = {
{ "level_in", "set input level", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
{ "mode", "set mode", OFFSET(mode), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A, .unit = "mode" },
{ "downward",0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, .unit = "mode" },
{ "upward", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, .unit = "mode" },
{ "range", "set max gain reduction", OFFSET(range), AV_OPT_TYPE_DOUBLE, {.dbl=0.06125}, 0, 1, A },
{ "threshold", "set threshold", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=0.125}, 0, 1, A },
{ "ratio", "set ratio", OFFSET(ratio), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 1, 9000, A },
{ "attack", "set attack", OFFSET(attack), AV_OPT_TYPE_DOUBLE, {.dbl=20}, 0.01, 9000, A },
{ "release", "set release", OFFSET(release), AV_OPT_TYPE_DOUBLE, {.dbl=250}, 0.01, 9000, A },
{ "makeup", "set makeup gain", OFFSET(makeup), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 1, 64, A },
{ "knee", "set knee", OFFSET(knee), AV_OPT_TYPE_DOUBLE, {.dbl=2.828427125}, 1, 8, A },
{ "detection", "set detection", OFFSET(detection), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, A, .unit = "detection" },
{ "peak", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, .unit = "detection" },
{ "rms", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, .unit = "detection" },
{ "link", "set link", OFFSET(link), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A, .unit = "link" },
{ "average", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, .unit = "link" },
{ "maximum", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, .unit = "link" },
{ "level_sc", "set sidechain gain", OFFSET(level_sc), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
{ NULL }
};
AVFILTER_DEFINE_CLASS_EXT(agate_sidechaingate, "agate/sidechaingate", options);
static int agate_config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
AudioGateContext *s = ctx->priv;
double lin_threshold = s->threshold;
double lin_knee_sqrt = sqrt(s->knee);
if (s->detection)
lin_threshold *= lin_threshold;
s->attack_coeff = FFMIN(1., 1. / (s->attack * inlink->sample_rate / 4000.));
s->release_coeff = FFMIN(1., 1. / (s->release * inlink->sample_rate / 4000.));
s->lin_knee_stop = lin_threshold * lin_knee_sqrt;
s->lin_knee_start = lin_threshold / lin_knee_sqrt;
s->thres = log(lin_threshold);
s->knee_start = log(s->lin_knee_start);
s->knee_stop = log(s->lin_knee_stop);
return 0;
}
// A fake infinity value (because real infinity may break some hosts)
#define FAKE_INFINITY (65536.0 * 65536.0)
// Check for infinity (with appropriate-ish tolerance)
#define IS_FAKE_INFINITY(value) (fabs(value-FAKE_INFINITY) < 1.0)
static double output_gain(double lin_slope, double ratio, double thres,
double knee, double knee_start, double knee_stop,
double range, int mode)
{
double slope = log(lin_slope);
double tratio = ratio;
double gain = 0.;
double delta = 0.;
if (IS_FAKE_INFINITY(ratio))
tratio = 1000.;
gain = (slope - thres) * tratio + thres;
delta = tratio;
if (mode) {
if (knee > 1. && slope < knee_stop)
gain = hermite_interpolation(slope, knee_stop, knee_start, ((knee_stop - thres) * tratio + thres), knee_start, delta, 1.);
} else {
if (knee > 1. && slope > knee_start)
gain = hermite_interpolation(slope, knee_start, knee_stop, ((knee_start - thres) * tratio + thres), knee_stop, delta, 1.);
}
return FFMAX(range, exp(gain - slope));
}
static void gate(AudioGateContext *s,
const double *src, double *dst, const double *scsrc,
int nb_samples, double level_in, double level_sc,
AVFilterLink *inlink, AVFilterLink *sclink)
{
AVFilterContext *ctx = inlink->dst;
const double makeup = s->makeup;
const double attack_coeff = s->attack_coeff;
const double release_coeff = s->release_coeff;
int n, c;
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) {
double abs_sample = fabs(scsrc[0] * level_sc), gain = 1.0;
double factor;
int detected;
if (s->link == 1) {
for (c = 1; c < sclink->ch_layout.nb_channels; c++)
abs_sample = FFMAX(fabs(scsrc[c] * level_sc), abs_sample);
} else {
for (c = 1; c < sclink->ch_layout.nb_channels; c++)
abs_sample += fabs(scsrc[c] * level_sc);
abs_sample /= sclink->ch_layout.nb_channels;
}
if (s->detection)
abs_sample *= abs_sample;
s->lin_slope += (abs_sample - s->lin_slope) * (abs_sample > s->lin_slope ? attack_coeff : release_coeff);
if (s->mode)
detected = s->lin_slope > s->lin_knee_start;
else
detected = s->lin_slope < s->lin_knee_stop;
if (s->lin_slope > 0.0 && detected)
gain = output_gain(s->lin_slope, s->ratio, s->thres,
s->knee, s->knee_start, s->knee_stop,
s->range, s->mode);
factor = ctx->is_disabled ? 1.f : level_in * gain * makeup;
for (c = 0; c < inlink->ch_layout.nb_channels; c++)
dst[c] = src[c] * factor;
}
}
#if CONFIG_AGATE_FILTER
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
const double *src = (const double *)in->data[0];
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
AudioGateContext *s = ctx->priv;
AVFrame *out;
double *dst;
if (av_frame_is_writable(in)) {
out = in;
} else {
out = ff_get_audio_buffer(outlink, in->nb_samples);
if (!out) {
av_frame_free(&in);
return AVERROR(ENOMEM);
}
av_frame_copy_props(out, in);
}
dst = (double *)out->data[0];
gate(s, src, dst, src, in->nb_samples,
s->level_in, s->level_in, inlink, inlink);
if (out != in)
av_frame_free(&in);
return ff_filter_frame(outlink, out);
}
static const AVFilterPad inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
.config_props = agate_config_input,
},
};
const AVFilter ff_af_agate = {
.name = "agate",
.description = NULL_IF_CONFIG_SMALL("Audio gate."),
.priv_class = &agate_sidechaingate_class,
.priv_size = sizeof(AudioGateContext),
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(ff_audio_default_filterpad),
FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBL),
.process_command = ff_filter_process_command,
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC,
};
#endif /* CONFIG_AGATE_FILTER */
#if CONFIG_SIDECHAINGATE_FILTER
static int activate(AVFilterContext *ctx)
{
AudioGateContext *s = ctx->priv;
AVFrame *out = NULL, *in[2] = { NULL };
int ret, i, nb_samples;
double *dst;
FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
if ((ret = ff_inlink_consume_frame(ctx->inputs[0], &in[0])) > 0) {
av_audio_fifo_write(s->fifo[0], (void **)in[0]->extended_data,
in[0]->nb_samples);
av_frame_free(&in[0]);
}
if (ret < 0)
return ret;
if ((ret = ff_inlink_consume_frame(ctx->inputs[1], &in[1])) > 0) {
av_audio_fifo_write(s->fifo[1], (void **)in[1]->extended_data,
in[1]->nb_samples);
av_frame_free(&in[1]);
}
if (ret < 0)
return ret;
nb_samples = FFMIN(av_audio_fifo_size(s->fifo[0]), av_audio_fifo_size(s->fifo[1]));
if (nb_samples) {
out = ff_get_audio_buffer(ctx->outputs[0], nb_samples);
if (!out)
return AVERROR(ENOMEM);
for (i = 0; i < 2; i++) {
in[i] = ff_get_audio_buffer(ctx->inputs[i], nb_samples);
if (!in[i]) {
av_frame_free(&in[0]);
av_frame_free(&in[1]);
av_frame_free(&out);
return AVERROR(ENOMEM);
}
av_audio_fifo_read(s->fifo[i], (void **)in[i]->data, nb_samples);
}
dst = (double *)out->data[0];
out->pts = s->pts;
s->pts += av_rescale_q(nb_samples, (AVRational){1, ctx->outputs[0]->sample_rate}, ctx->outputs[0]->time_base);
gate(s, (double *)in[0]->data[0], dst,
(double *)in[1]->data[0], nb_samples,
s->level_in, s->level_sc,
ctx->inputs[0], ctx->inputs[1]);
av_frame_free(&in[0]);
av_frame_free(&in[1]);
ret = ff_filter_frame(ctx->outputs[0], out);
if (ret < 0)
return ret;
}
FF_FILTER_FORWARD_STATUS(ctx->inputs[0], ctx->outputs[0]);
FF_FILTER_FORWARD_STATUS(ctx->inputs[1], ctx->outputs[0]);
if (ff_outlink_frame_wanted(ctx->outputs[0])) {
if (!av_audio_fifo_size(s->fifo[0]))
ff_inlink_request_frame(ctx->inputs[0]);
if (!av_audio_fifo_size(s->fifo[1]))
ff_inlink_request_frame(ctx->inputs[1]);
}
return 0;
}
static int scquery_formats(const AVFilterContext *ctx,
AVFilterFormatsConfig **cfg_in,
AVFilterFormatsConfig **cfg_out)
{
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_DBL,
AV_SAMPLE_FMT_NONE
};
int ret;
/* Generic code will link the channel properties of the main input and the
* output; it won't touch the second input as its channel_layouts is already
* set. */
ret = ff_channel_layouts_ref(ff_all_channel_counts(),
&cfg_in[1]->channel_layouts);
if (ret < 0)
return ret;
if ((ret = ff_set_common_formats_from_list2(ctx, cfg_in, cfg_out, sample_fmts)) < 0)
return ret;
return 0;
}
static int scconfig_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AudioGateContext *s = ctx->priv;
outlink->time_base = ctx->inputs[0]->time_base;
s->fifo[0] = av_audio_fifo_alloc(ctx->inputs[0]->format, ctx->inputs[0]->ch_layout.nb_channels, 1024);
s->fifo[1] = av_audio_fifo_alloc(ctx->inputs[1]->format, ctx->inputs[1]->ch_layout.nb_channels, 1024);
if (!s->fifo[0] || !s->fifo[1])
return AVERROR(ENOMEM);
agate_config_input(ctx->inputs[0]);
return 0;
}
static av_cold void uninit(AVFilterContext *ctx)
{
AudioGateContext *s = ctx->priv;
av_audio_fifo_free(s->fifo[0]);
av_audio_fifo_free(s->fifo[1]);
}
static const AVFilterPad sidechaingate_inputs[] = {
{
.name = "main",
.type = AVMEDIA_TYPE_AUDIO,
},{
.name = "sidechain",
.type = AVMEDIA_TYPE_AUDIO,
},
};
static const AVFilterPad sidechaingate_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = scconfig_output,
},
};
const AVFilter ff_af_sidechaingate = {
.name = "sidechaingate",
.description = NULL_IF_CONFIG_SMALL("Audio sidechain gate."),
.priv_class = &agate_sidechaingate_class,
.priv_size = sizeof(AudioGateContext),
.activate = activate,
.uninit = uninit,
FILTER_INPUTS(sidechaingate_inputs),
FILTER_OUTPUTS(sidechaingate_outputs),
FILTER_QUERY_FUNC2(scquery_formats),
.process_command = ff_filter_process_command,
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
};
#endif /* CONFIG_SIDECHAINGATE_FILTER */