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FFmpeg/libavfilter/af_adynamicequalizer.c

348 lines
12 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 <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 knee;
double slew;
double attack;
double release;
double attack_coef;
double release_coef;
int mode;
int type;
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);
return 0;
}
static double get_svf(double in, double *m, 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;
}
static inline double from_dB(double x)
{
return exp(0.05 * x * M_LN10);
}
static inline double to_dB(double x)
{
return 20. * log10(x);
}
static inline double sqr(double x)
{
return x * x;
}
static double get_gain(double in, double srate, double makeup,
double aattack, double iratio, double knee, double range,
double thresdb, double slewfactor, double *state,
double attack_coeff, double release_coeff, double nc)
{
double width = (6. * knee) + 0.01;
double cdb = 0.;
double Lgain = 1.;
double Lxg, Lxl, Lyg, Lyl, Ly1;
double checkwidth = 0.;
double slewwidth = 1.8;
int attslew = 0;
Lyg = 0.;
Lxg = to_dB(fabs(in) + DBL_EPSILON);
Lyg = Lxg + (iratio - 1.) * sqr(Lxg - thresdb + width * .5) / (2. * width);
checkwidth = 2. * fabs(Lxg - thresdb);
if (2. * (Lxg - thresdb) < -width) {
Lyg = Lxg;
} else if (checkwidth <= width) {
Lyg = thresdb + (Lxg - thresdb) * iratio;
if (checkwidth <= slewwidth) {
if (Lyg >= state[2])
attslew = 1;
}
} else if (2. * (Lxg - thresdb) > width) {
Lyg = thresdb + (Lxg - thresdb) * iratio;
}
attack_coeff = attslew ? aattack : attack_coeff;
Lxl = Lxg - Lyg;
Ly1 = fmax(Lxl, release_coeff * state[1] +(1. - release_coeff) * Lxl);
Lyl = attack_coeff * state[0] + (1. - attack_coeff) * Ly1;
cdb = -Lyl;
Lgain = from_dB(nc * fmin(cdb - makeup, range));
state[0] = Lyl;
state[1] = Ly1;
state[2] = Lyg;
return Lgain;
}
typedef struct ThreadData {
AVFrame *in, *out;
} ThreadData;
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 iratio = 1. / s->ratio;
const double range = s->range;
const double dfrequency = fmin(s->dfrequency, sample_rate * 0.5);
const double tfrequency = fmin(s->tfrequency, sample_rate * 0.5);
const double threshold = to_dB(s->threshold + DBL_EPSILON);
const double release = s->release_coef;
const double attack = s->attack_coef;
const double dqfactor = s->dqfactor;
const double tqfactor = s->tqfactor;
const double fg = tan(M_PI * tfrequency / sample_rate);
const double dg = tan(M_PI * dfrequency / 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 mode = s->mode;
const int type = s->type;
const double knee = s->knee;
const double slew = s->slew;
const double aattack = exp(-1000. / ((s->attack + 2.0 * (slew - 1.)) * sample_rate));
const double nc = mode == 0 ? 1. : -1.;
double da[3], dm[3];
{
double 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] = 1.;
dm[2] = 0.;
}
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];
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);
gain = get_gain(detect, sample_rate, makeup,
aattack, iratio, knee, range, threshold, slew,
&state[4], attack, release, nc);
switch (type) {
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 double get_coef(double x, double sr)
{
return exp(-1000. / (x * sr));
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
AudioDynamicEqualizerContext *s = ctx->priv;
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);
}
s->attack_coef = get_coef(s->attack, in->sample_rate);
s->release_coef = get_coef(s->release, in->sample_rate);
td.in = in;
td.out = out;
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 },
{ "knee", "set knee factor", OFFSET(knee), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 8, FLAGS },
{ "ratio", "set ratio factor", OFFSET(ratio), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 1, 20, FLAGS },
{ "makeup", "set makeup gain", OFFSET(makeup), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 30, FLAGS },
{ "range", "set max gain", OFFSET(range), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 200, FLAGS },
{ "slew", "set slew factor", OFFSET(slew), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 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" },
{ "tftype", "set target filter type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, FLAGS, "type" },
{ "bell", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, "type" },
{ "lowshelf", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, "type" },
{ "highshelf",0, 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, FLAGS, "type" },
{ 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,
};