ffmpeg/libavfilter/af_adynamicequalizer.c

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/*
* 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 <float.h>
#include "libavutil/opt.h"
#include "avfilter.h"
#include "audio.h"
#include "formats.h"
typedef struct AudioDynamicEqualizerContext {
const AVClass *class;
double threshold;
double dfrequency;
double dqfactor;
double tfrequency;
double tqfactor;
double ratio;
double range;
double makeup;
double attack;
double release;
double attack_coef;
double release_coef;
int mode;
int direction;
int detection;
int tftype;
int dftype;
double da[3], dm[3];
AVFrame *state;
} AudioDynamicEqualizerContext;
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
AudioDynamicEqualizerContext *s = ctx->priv;
s->state = ff_get_audio_buffer(inlink, 8);
if (!s->state)
return AVERROR(ENOMEM);
for (int ch = 0; ch < inlink->ch_layout.nb_channels; ch++) {
double *state = (double *)s->state->extended_data[ch];
state[4] = 1.;
}
return 0;
}
static double get_svf(double in, const double *m, const double *a, double *b)
{
const double v0 = in;
const double v3 = v0 - b[1];
const double v1 = a[0] * b[0] + a[1] * v3;
const double v2 = b[1] + a[1] * b[0] + a[2] * v3;
b[0] = 2. * v1 - b[0];
b[1] = 2. * v2 - b[1];
return m[0] * v0 + m[1] * v1 + m[2] * v2;
}
typedef struct ThreadData {
AVFrame *in, *out;
} ThreadData;
static double get_coef(double x, double sr)
{
return exp(-1000. / (x * sr));
}
static int filter_prepare(AVFilterContext *ctx)
{
AudioDynamicEqualizerContext *s = ctx->priv;
const double sample_rate = ctx->inputs[0]->sample_rate;
const double dfrequency = fmin(s->dfrequency, sample_rate * 0.5);
const double dg = tan(M_PI * dfrequency / sample_rate);
const double dqfactor = s->dqfactor;
const int dftype = s->dftype;
double *da = s->da;
double *dm = s->dm;
double k;
s->attack_coef = get_coef(s->attack, sample_rate);
s->release_coef = get_coef(s->release, sample_rate);
switch (dftype) {
case 0:
k = 1. / dqfactor;
da[0] = 1. / (1. + dg * (dg + k));
da[1] = dg * da[0];
da[2] = dg * da[1];
dm[0] = 0.;
dm[1] = k;
dm[2] = 0.;
break;
case 1:
k = 1. / dqfactor;
da[0] = 1. / (1. + dg * (dg + k));
da[1] = dg * da[0];
da[2] = dg * da[1];
dm[0] = 0.;
dm[1] = 0.;
dm[2] = 1.;
break;
case 2:
k = 1. / dqfactor;
da[0] = 1. / (1. + dg * (dg + k));
da[1] = dg * da[0];
da[2] = dg * da[1];
dm[0] = 0.;
dm[1] = -k;
dm[2] = -1.;
break;
case 3:
k = 1. / dqfactor;
da[0] = 1. / (1. + dg * (dg + k));
da[1] = dg * da[0];
da[2] = dg * da[1];
dm[0] = 0.;
dm[1] = -k;
dm[2] = -2.;
break;
}
return 0;
}
static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
{
AudioDynamicEqualizerContext *s = ctx->priv;
ThreadData *td = arg;
AVFrame *in = td->in;
AVFrame *out = td->out;
const double sample_rate = in->sample_rate;
const double makeup = s->makeup;
const double ratio = s->ratio;
const double range = s->range;
const double tfrequency = fmin(s->tfrequency, sample_rate * 0.5);
const double release = s->release_coef;
const double irelease = 1. - release;
const double attack = s->attack_coef;
const double iattack = 1. - attack;
const double tqfactor = s->tqfactor;
const double fg = tan(M_PI * tfrequency / sample_rate);
const int start = (in->ch_layout.nb_channels * jobnr) / nb_jobs;
const int end = (in->ch_layout.nb_channels * (jobnr+1)) / nb_jobs;
const int detection = s->detection;
const int direction = s->direction;
const int tftype = s->tftype;
const int mode = s->mode;
const double *da = s->da;
const double *dm = s->dm;
for (int ch = start; ch < end; ch++) {
const double *src = (const double *)in->extended_data[ch];
double *dst = (double *)out->extended_data[ch];
double *state = (double *)s->state->extended_data[ch];
const double threshold = detection == 0 ? state[5] : s->threshold;
if (detection < 0)
state[5] = threshold;
for (int n = 0; n < out->nb_samples; n++) {
double detect, gain, v, listen;
double fa[3], fm[3];
double k, g;
detect = listen = get_svf(src[n], dm, da, state);
detect = fabs(detect);
if (detection > 0)
state[5] = fmax(state[5], detect);
if (direction == 0) {
if (detect < threshold) {
if (mode == 0)
detect = 1. / av_clipd(1. + makeup + (threshold - detect) * ratio, 1., range);
else
detect = av_clipd(1. + makeup + (threshold - detect) * ratio, 1., range);
} else {
detect = 1.;
}
} else {
if (detect > threshold) {
if (mode == 0)
detect = 1. / av_clipd(1. + makeup + (detect - threshold) * ratio, 1., range);
else
detect = av_clipd(1. + makeup + (detect - threshold) * ratio, 1., range);
} else {
detect = 1.;
}
}
if (direction == 0) {
if (detect > state[4]) {
detect = iattack * detect + attack * state[4];
} else {
detect = irelease * detect + release * state[4];
}
} else {
if (detect < state[4]) {
detect = iattack * detect + attack * state[4];
} else {
detect = irelease * detect + release * state[4];
}
}
if (state[4] != detect || n == 0) {
state[4] = gain = detect;
switch (tftype) {
case 0:
k = 1. / (tqfactor * gain);
fa[0] = 1. / (1. + fg * (fg + k));
fa[1] = fg * fa[0];
fa[2] = fg * fa[1];
fm[0] = 1.;
fm[1] = k * (gain * gain - 1.);
fm[2] = 0.;
break;
case 1:
k = 1. / tqfactor;
g = fg / sqrt(gain);
fa[0] = 1. / (1. + g * (g + k));
fa[1] = g * fa[0];
fa[2] = g * fa[1];
fm[0] = 1.;
fm[1] = k * (gain - 1.);
fm[2] = gain * gain - 1.;
break;
case 2:
k = 1. / tqfactor;
g = fg / sqrt(gain);
fa[0] = 1. / (1. + g * (g + k));
fa[1] = g * fa[0];
fa[2] = g * fa[1];
fm[0] = gain * gain;
fm[1] = k * (1. - gain) * gain;
fm[2] = 1. - gain * gain;
break;
}
}
v = get_svf(src[n], fm, fa, &state[2]);
v = mode == -1 ? listen : v;
dst[n] = ctx->is_disabled ? src[n] : v;
}
}
return 0;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
ThreadData td;
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);
}
td.in = in;
td.out = out;
filter_prepare(ctx);
ff_filter_execute(ctx, filter_channels, &td, NULL,
FFMIN(outlink->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx)));
if (out != in)
av_frame_free(&in);
return ff_filter_frame(outlink, out);
}
static av_cold void uninit(AVFilterContext *ctx)
{
AudioDynamicEqualizerContext *s = ctx->priv;
av_frame_free(&s->state);
}
#define OFFSET(x) offsetof(AudioDynamicEqualizerContext, x)
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
static const AVOption adynamicequalizer_options[] = {
{ "threshold", "set detection threshold", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 100, FLAGS },
{ "dfrequency", "set detection frequency", OFFSET(dfrequency), AV_OPT_TYPE_DOUBLE, {.dbl=1000}, 2, 1000000, FLAGS },
{ "dqfactor", "set detection Q factor", OFFSET(dqfactor), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.001, 1000, FLAGS },
{ "tfrequency", "set target frequency", OFFSET(tfrequency), AV_OPT_TYPE_DOUBLE, {.dbl=1000}, 2, 1000000, FLAGS },
{ "tqfactor", "set target Q factor", OFFSET(tqfactor), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.001, 1000, FLAGS },
{ "attack", "set attack duration", OFFSET(attack), AV_OPT_TYPE_DOUBLE, {.dbl=20}, 1, 2000, FLAGS },
{ "release", "set release duration", OFFSET(release), AV_OPT_TYPE_DOUBLE, {.dbl=200}, 1, 2000, FLAGS },
{ "ratio", "set ratio factor", OFFSET(ratio), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 30, FLAGS },
{ "makeup", "set makeup gain", OFFSET(makeup), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 100, FLAGS },
{ "range", "set max gain", OFFSET(range), AV_OPT_TYPE_DOUBLE, {.dbl=50}, 1, 200, FLAGS },
{ "mode", "set mode", OFFSET(mode), AV_OPT_TYPE_INT, {.i64=0}, -1, 1, FLAGS, "mode" },
{ "listen", 0, 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, FLAGS, "mode" },
{ "cut", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, "mode" },
{ "boost", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, "mode" },
{ "dftype", "set detection filter type",OFFSET(dftype), AV_OPT_TYPE_INT, {.i64=0}, 0, 3, FLAGS, "dftype" },
{ "bandpass", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, "dftype" },
{ "lowpass", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, "dftype" },
{ "highpass", 0, 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, FLAGS, "dftype" },
{ "peak", 0, 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, FLAGS, "dftype" },
{ "tftype", "set target filter type", OFFSET(tftype), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, FLAGS, "tftype" },
{ "bell", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, "tftype" },
{ "lowshelf", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, "tftype" },
{ "highshelf",0, 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, FLAGS, "tftype" },
{ "direction", "set direction", OFFSET(direction), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, FLAGS, "direction" },
{ "downward", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, "direction" },
{ "upward", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, "direction" },
{ "auto", "set auto threshold", OFFSET(detection), AV_OPT_TYPE_INT, {.i64=-1}, -1, 1, FLAGS, "auto" },
{ "disabled", 0, 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, FLAGS, "auto" },
{ "off", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, "auto" },
{ "on", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, "auto" },
{ NULL }
};
AVFILTER_DEFINE_CLASS(adynamicequalizer);
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_adynamicequalizer = {
.name = "adynamicequalizer",
.description = NULL_IF_CONFIG_SMALL("Apply Dynamic Equalization of input audio."),
.priv_size = sizeof(AudioDynamicEqualizerContext),
.priv_class = &adynamicequalizer_class,
.uninit = uninit,
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(outputs),
FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBLP),
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
AVFILTER_FLAG_SLICE_THREADS,
.process_command = ff_filter_process_command,
};