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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
*/
#undef ftype
#undef SQRT
#undef TAN
#undef ONE
#undef TWO
#undef ZERO
#undef FMAX
#undef FMIN
#undef CLIP
#undef SAMPLE_FORMAT
#undef EPSILON
#undef FABS
#if DEPTH == 32
#define SAMPLE_FORMAT float
#define SQRT sqrtf
#define TAN tanf
#define ONE 1.f
#define TWO 2.f
#define ZERO 0.f
#define FMIN fminf
#define FMAX fmaxf
#define CLIP av_clipf
#define FABS fabsf
#define ftype float
#define EPSILON (1.f / (1 << 22))
#else
#define SAMPLE_FORMAT double
#define SQRT sqrt
#define TAN tan
#define ONE 1.0
#define TWO 2.0
#define ZERO 0.0
#define FMIN fmin
#define FMAX fmax
#define CLIP av_clipd
#define FABS fabs
#define ftype double
#define EPSILON (1.0 / (1LL << 51))
#endif
#define fn3(a,b) a##_##b
#define fn2(a,b) fn3(a,b)
#define fn(a) fn2(a, SAMPLE_FORMAT)
static ftype fn(get_svf)(ftype in, const ftype *m, const ftype *a, ftype *b)
{
const ftype v0 = in;
const ftype v3 = v0 - b[1];
const ftype v1 = a[0] * b[0] + a[1] * v3;
const ftype v2 = b[1] + a[1] * b[0] + a[2] * v3;
b[0] = TWO * v1 - b[0];
b[1] = TWO * v2 - b[1];
return m[0] * v0 + m[1] * v1 + m[2] * v2;
}
static int fn(filter_prepare)(AVFilterContext *ctx)
{
AudioDynamicEqualizerContext *s = ctx->priv;
const ftype sample_rate = ctx->inputs[0]->sample_rate;
const ftype dfrequency = FMIN(s->dfrequency, sample_rate * 0.5);
const ftype dg = TAN(M_PI * dfrequency / sample_rate);
const ftype dqfactor = s->dqfactor;
const int dftype = s->dftype;
ftype *da = fn(s->da);
ftype *dm = fn(s->dm);
ftype k;
s->attack_coef = get_coef(s->attack, sample_rate);
s->release_coef = get_coef(s->release, sample_rate);
switch (dftype) {
case 0:
k = ONE / dqfactor;
da[0] = ONE / (ONE + dg * (dg + k));
da[1] = dg * da[0];
da[2] = dg * da[1];
dm[0] = ZERO;
dm[1] = k;
dm[2] = ZERO;
break;
case 1:
k = ONE / dqfactor;
da[0] = ONE / (ONE + dg * (dg + k));
da[1] = dg * da[0];
da[2] = dg * da[1];
dm[0] = ZERO;
dm[1] = ZERO;
dm[2] = ONE;
break;
case 2:
k = ONE / dqfactor;
da[0] = ONE / (ONE + dg * (dg + k));
da[1] = dg * da[0];
da[2] = dg * da[1];
dm[0] = ZERO;
dm[1] = -k;
dm[2] = -ONE;
break;
case 3:
k = ONE / dqfactor;
da[0] = ONE / (ONE + dg * (dg + k));
da[1] = dg * da[0];
da[2] = dg * da[1];
dm[0] = ONE;
dm[1] = -k;
dm[2] = -TWO;
break;
}
return 0;
}
static int fn(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 ftype sample_rate = in->sample_rate;
const ftype makeup = s->makeup;
const ftype ratio = s->ratio;
const ftype range = s->range;
const ftype tfrequency = FMIN(s->tfrequency, sample_rate * 0.5);
const ftype release = s->release_coef;
const ftype attack = s->attack_coef;
const ftype tqfactor = s->tqfactor;
const ftype itqfactor = ONE / tqfactor;
const ftype 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 ftype *da = fn(s->da);
const ftype *dm = fn(s->dm);
for (int ch = start; ch < end; ch++) {
const ftype *src = (const ftype *)in->extended_data[ch];
ftype *dst = (ftype *)out->extended_data[ch];
ftype *state = (ftype *)s->state->extended_data[ch];
const ftype threshold = detection == 0 ? state[5] : s->threshold;
ftype fa[3], fm[3];
if (detection < 0)
state[5] = threshold;
memcpy(fa, state + 8, sizeof(fa));
memcpy(fm, state + 11, sizeof(fm));
for (int n = 0; n < out->nb_samples; n++) {
ftype detect, gain, v, listen;
ftype k, g;
detect = listen = fn(get_svf)(src[n], dm, da, state);
detect = FABS(detect);
if (detection > 0)
state[5] = FMAX(state[5], detect);
if (mode >= 0) {
if (direction == 0 && detect < threshold) {
detect = CLIP(ONE + makeup + (threshold - detect) * ratio, ONE, range);
if (!mode)
detect = ONE / detect;
} else if (direction == 1 && detect > threshold) {
detect = CLIP(ONE + makeup + (detect - threshold) * ratio, ONE, range);
if (!mode)
detect = ONE / detect;
} else {
detect = ONE;
}
{
ftype delta = detect - state[4];
if (delta > EPSILON)
detect = state[4] + attack * delta;
else if (delta < -EPSILON)
detect = state[4] + release * delta;
}
}
if (state[4] != detect) {
state[4] = gain = detect;
switch (tftype) {
case 0:
k = itqfactor / gain;
fa[0] = ONE / (ONE + fg * (fg + k));
fa[1] = fg * fa[0];
fa[2] = fg * fa[1];
fm[0] = ONE;
fm[1] = k * (gain * gain - ONE);
fm[2] = ZERO;
break;
case 1:
k = itqfactor;
g = fg / SQRT(gain);
fa[0] = ONE / (ONE + g * (g + k));
fa[1] = g * fa[0];
fa[2] = g * fa[1];
fm[0] = ONE;
fm[1] = k * (gain - ONE);
fm[2] = gain * gain - ONE;
break;
case 2:
k = itqfactor;
g = fg * SQRT(gain);
fa[0] = ONE / (ONE + g * (g + k));
fa[1] = g * fa[0];
fa[2] = g * fa[1];
fm[0] = gain * gain;
fm[1] = k * (ONE - gain) * gain;
fm[2] = ONE - gain * gain;
break;
}
}
v = fn(get_svf)(src[n], fm, fa, &state[2]);
v = mode == -1 ? listen : v;
dst[n] = ctx->is_disabled ? src[n] : v;
}
memcpy(state + 8, fa, sizeof(fa));
memcpy(state + 11, fm, sizeof(fm));
}
return 0;
}
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