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FFmpeg/libavfilter/af_adynamicsmooth.c
2021-12-02 09:39:27 +01:00

143 lines
4.3 KiB
C

/*
* 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
*/
#include "libavutil/ffmath.h"
#include "libavutil/opt.h"
#include "avfilter.h"
#include "audio.h"
#include "formats.h"
typedef struct AudioDynamicSmoothContext {
const AVClass *class;
double sensitivity;
double basefreq;
AVFrame *coeffs;
} AudioDynamicSmoothContext;
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
AudioDynamicSmoothContext *s = ctx->priv;
s->coeffs = ff_get_audio_buffer(inlink, 3);
if (!s->coeffs)
return AVERROR(ENOMEM);
return 0;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
AudioDynamicSmoothContext *s = ctx->priv;
const double sensitivity = s->sensitivity;
const double wc = s->basefreq / in->sample_rate;
AVFrame *out;
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);
}
for (int ch = 0; ch < out->channels; ch++) {
const double *src = (const double *)in->extended_data[ch];
double *dst = (double *)out->extended_data[ch];
double *coeffs = (double *)s->coeffs->extended_data[ch];
double low1 = coeffs[0];
double low2 = coeffs[1];
double inz = coeffs[2];
for (int n = 0; n < out->nb_samples; n++) {
double low1z = low1;
double low2z = low2;
double bandz = low2z - low1z;
double wd = wc + sensitivity * fabs(bandz);
double g = fmin(1., wd * (5.9948827 + wd * (-11.969296 + wd * 15.959062)));
low1 = low1z + g * (0.5 * (src[n] + inz) - low1z);
low2 = low2z + g * (0.5 * (low1 + low1z) - low2z);
inz = src[n];
dst[n] = ctx->is_disabled ? src[n] : low2;
}
coeffs[0] = low1;
coeffs[1] = low2;
coeffs[2] = inz;
}
if (out != in)
av_frame_free(&in);
return ff_filter_frame(outlink, out);
}
static av_cold void uninit(AVFilterContext *ctx)
{
AudioDynamicSmoothContext *s = ctx->priv;
av_frame_free(&s->coeffs);
}
#define OFFSET(x) offsetof(AudioDynamicSmoothContext, x)
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
static const AVOption adynamicsmooth_options[] = {
{ "sensitivity", "set smooth sensitivity", OFFSET(sensitivity), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 0, 1000000, FLAGS },
{ "basefreq", "set base frequency", OFFSET(basefreq), AV_OPT_TYPE_DOUBLE, {.dbl=22050}, 2, 1000000, FLAGS },
{ NULL }
};
AVFILTER_DEFINE_CLASS(adynamicsmooth);
static const AVFilterPad inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
.config_props = config_input,
},
};
static const AVFilterPad outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
},
};
const AVFilter ff_af_adynamicsmooth = {
.name = "adynamicsmooth",
.description = NULL_IF_CONFIG_SMALL("Apply Dynamic Smoothing of input audio."),
.priv_size = sizeof(AudioDynamicSmoothContext),
.priv_class = &adynamicsmooth_class,
.uninit = uninit,
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(outputs),
FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBLP),
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
.process_command = ff_filter_process_command,
};