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FFmpeg/libavfilter/af_aexciter.c
Andreas Rheinhardt 8a4b76e9ee avfilter/af_aexciter: Store format in filter, remove query function
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2021-10-05 18:01:02 +02:00

286 lines
8.3 KiB
C

/*
* Copyright (c) Markus Schmidt and Christian Holschuh
*
* 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/opt.h"
#include "avfilter.h"
#include "internal.h"
#include "audio.h"
typedef struct ChannelParams {
double blend_old, drive_old;
double rdrive, rbdr, kpa, kpb, kna, knb, ap,
an, imr, kc, srct, sq, pwrq;
double prev_med, prev_out;
double hp[5], lp[5];
double hw[4][2], lw[2][2];
} ChannelParams;
typedef struct AExciterContext {
const AVClass *class;
double level_in;
double level_out;
double amount;
double drive;
double blend;
double freq;
double ceil;
int listen;
ChannelParams *cp;
} AExciterContext;
#define OFFSET(x) offsetof(AExciterContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
static const AVOption aexciter_options[] = {
{ "level_in", "set level in", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 64, A },
{ "level_out", "set level out", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 64, A },
{ "amount", "set amount", OFFSET(amount), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 64, A },
{ "drive", "set harmonics", OFFSET(drive), AV_OPT_TYPE_DOUBLE, {.dbl=8.5}, 0.1, 10, A },
{ "blend", "set blend harmonics", OFFSET(blend), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -10, 10, A },
{ "freq", "set scope", OFFSET(freq), AV_OPT_TYPE_DOUBLE, {.dbl=7500}, 2000, 12000, A },
{ "ceil", "set ceiling", OFFSET(ceil), AV_OPT_TYPE_DOUBLE, {.dbl=9999}, 9999, 20000, A },
{ "listen", "enable listen mode", OFFSET(listen), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
{ NULL }
};
AVFILTER_DEFINE_CLASS(aexciter);
static inline double M(double x)
{
return (fabs(x) > 0.00000001) ? x : 0.0;
}
static inline double D(double x)
{
x = fabs(x);
return (x > 0.00000001) ? sqrt(x) : 0.0;
}
static void set_params(ChannelParams *p,
double blend, double drive,
double srate, double freq,
double ceil)
{
double a0, a1, a2, b0, b1, b2, w0, alpha;
p->rdrive = 12.0 / drive;
p->rbdr = p->rdrive / (10.5 - blend) * 780.0 / 33.0;
p->kpa = D(2.0 * (p->rdrive*p->rdrive) - 1.0) + 1.0;
p->kpb = (2.0 - p->kpa) / 2.0;
p->ap = ((p->rdrive*p->rdrive) - p->kpa + 1.0) / 2.0;
p->kc = p->kpa / D(2.0 * D(2.0 * (p->rdrive*p->rdrive) - 1.0) - 2.0 * p->rdrive*p->rdrive);
p->srct = (0.1 * srate) / (0.1 * srate + 1.0);
p->sq = p->kc*p->kc + 1.0;
p->knb = -1.0 * p->rbdr / D(p->sq);
p->kna = 2.0 * p->kc * p->rbdr / D(p->sq);
p->an = p->rbdr*p->rbdr / p->sq;
p->imr = 2.0 * p->knb + D(2.0 * p->kna + 4.0 * p->an - 1.0);
p->pwrq = 2.0 / (p->imr + 1.0);
w0 = 2 * M_PI * freq / srate;
alpha = sin(w0) / (2. * 0.707);
a0 = 1 + alpha;
a1 = -2 * cos(w0);
a2 = 1 - alpha;
b0 = (1 + cos(w0)) / 2;
b1 = -(1 + cos(w0));
b2 = (1 + cos(w0)) / 2;
p->hp[0] =-a1 / a0;
p->hp[1] =-a2 / a0;
p->hp[2] = b0 / a0;
p->hp[3] = b1 / a0;
p->hp[4] = b2 / a0;
w0 = 2 * M_PI * ceil / srate;
alpha = sin(w0) / (2. * 0.707);
a0 = 1 + alpha;
a1 = -2 * cos(w0);
a2 = 1 - alpha;
b0 = (1 - cos(w0)) / 2;
b1 = 1 - cos(w0);
b2 = (1 - cos(w0)) / 2;
p->lp[0] =-a1 / a0;
p->lp[1] =-a2 / a0;
p->lp[2] = b0 / a0;
p->lp[3] = b1 / a0;
p->lp[4] = b2 / a0;
}
static double bprocess(double in, const double *const c,
double *w1, double *w2)
{
double out = c[2] * in + *w1;
*w1 = c[3] * in + *w2 + c[0] * out;
*w2 = c[4] * in + c[1] * out;
return out;
}
static double distortion_process(AExciterContext *s, ChannelParams *p, double in)
{
double proc = in, med;
proc = bprocess(proc, p->hp, &p->hw[0][0], &p->hw[0][1]);
proc = bprocess(proc, p->hp, &p->hw[1][0], &p->hw[1][1]);
if (proc >= 0.0) {
med = (D(p->ap + proc * (p->kpa - proc)) + p->kpb) * p->pwrq;
} else {
med = (D(p->an - proc * (p->kna + proc)) + p->knb) * p->pwrq * -1.0;
}
proc = p->srct * (med - p->prev_med + p->prev_out);
p->prev_med = M(med);
p->prev_out = M(proc);
proc = bprocess(proc, p->hp, &p->hw[2][0], &p->hw[2][1]);
proc = bprocess(proc, p->hp, &p->hw[3][0], &p->hw[3][1]);
if (s->ceil >= 10000.) {
proc = bprocess(proc, p->lp, &p->lw[0][0], &p->lw[0][1]);
proc = bprocess(proc, p->lp, &p->lw[1][0], &p->lw[1][1]);
}
return proc;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
AExciterContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
AVFrame *out;
const double *src = (const double *)in->data[0];
const double level_in = s->level_in;
const double level_out = s->level_out;
const double amount = s->amount;
const double listen = 1.0 - s->listen;
double *dst;
if (av_frame_is_writable(in)) {
out = in;
} else {
out = ff_get_audio_buffer(inlink, in->nb_samples);
if (!out) {
av_frame_free(&in);
return AVERROR(ENOMEM);
}
av_frame_copy_props(out, in);
}
dst = (double *)out->data[0];
for (int n = 0; n < in->nb_samples; n++) {
for (int c = 0; c < inlink->channels; c++) {
double sample = src[c] * level_in;
sample = distortion_process(s, &s->cp[c], sample);
sample = sample * amount + listen * src[c];
sample *= level_out;
if (ctx->is_disabled)
dst[c] = src[c];
else
dst[c] = sample;
}
src += inlink->channels;
dst += inlink->channels;
}
if (in != out)
av_frame_free(&in);
return ff_filter_frame(outlink, out);
}
static av_cold void uninit(AVFilterContext *ctx)
{
AExciterContext *s = ctx->priv;
av_freep(&s->cp);
}
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
AExciterContext *s = ctx->priv;
if (!s->cp)
s->cp = av_calloc(inlink->channels, sizeof(*s->cp));
if (!s->cp)
return AVERROR(ENOMEM);
for (int i = 0; i < inlink->channels; i++)
set_params(&s->cp[i], s->blend, s->drive, inlink->sample_rate,
s->freq, s->ceil);
return 0;
}
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
char *res, int res_len, int flags)
{
AVFilterLink *inlink = ctx->inputs[0];
int ret;
ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
if (ret < 0)
return ret;
return config_input(inlink);
}
static const AVFilterPad avfilter_af_aexciter_inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_input,
.filter_frame = filter_frame,
},
};
static const AVFilterPad avfilter_af_aexciter_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
},
};
const AVFilter ff_af_aexciter = {
.name = "aexciter",
.description = NULL_IF_CONFIG_SMALL("Enhance high frequency part of audio."),
.priv_size = sizeof(AExciterContext),
.priv_class = &aexciter_class,
.uninit = uninit,
FILTER_INPUTS(avfilter_af_aexciter_inputs),
FILTER_OUTPUTS(avfilter_af_aexciter_outputs),
FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBL),
.process_command = process_command,
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
};