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FFmpeg/libavfilter/adynamicequalizer_template.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
*/
#undef ftype
#undef SQRT
#undef TAN
#undef ONE
#undef TWO
#undef ZERO
#undef FMAX
#undef FMIN
#undef CLIP
#undef SAMPLE_FORMAT
#undef FABS
#undef FLOG
#undef FEXP
#undef FLOG2
#undef FLOG10
#undef FEXP2
#undef FEXP10
#undef EPSILON
#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 FLOG logf
#define FEXP expf
#define FLOG2 log2f
#define FLOG10 log10f
#define FEXP2 exp2f
#define FEXP10 ff_exp10f
#define EPSILON (1.f / (1 << 23))
#define ftype float
#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 FLOG log
#define FEXP exp
#define FLOG2 log2
#define FLOG10 log10
#define FEXP2 exp2
#define FEXP10 ff_exp10
#define EPSILON (1.0 / (1LL << 53))
#define ftype double
#endif
#define LIN2LOG(x) (20.0 * FLOG10(x))
#define LOG2LIN(x) (FEXP10(x / 20.0))
#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->threshold_log = LIN2LOG(s->threshold);
s->dattack_coef = get_coef(s->dattack, sample_rate);
s->drelease_coef = get_coef(s->drelease, sample_rate);
s->gattack_coef = s->dattack_coef * 0.25;
s->grelease_coef = s->drelease_coef * 0.25;
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;
}
#define PEAKS(empty_value,op,sample, psample)\
if (!empty && psample == ss[front]) { \
ss[front] = empty_value; \
if (back != front) { \
front--; \
if (front < 0) \
front = n - 1; \
} \
empty = front == back; \
} \
\
if (!empty && sample op ss[front]) { \
while (1) { \
ss[front] = empty_value; \
if (back == front) { \
empty = 1; \
break; \
} \
front--; \
if (front < 0) \
front = n - 1; \
} \
} \
\
while (!empty && sample op ss[back]) { \
ss[back] = empty_value; \
if (back == front) { \
empty = 1; \
break; \
} \
back++; \
if (back >= n) \
back = 0; \
} \
\
if (!empty) { \
back--; \
if (back < 0) \
back = n - 1; \
}
static void fn(queue_sample)(ChannelContext *cc,
const ftype x,
const int nb_samples)
{
ftype *ss = cc->dqueue;
ftype *qq = cc->queue;
int front = cc->front;
int back = cc->back;
int empty, n, pos = cc->position;
ftype px = qq[pos];
fn(cc->sum) += x;
fn(cc->log_sum) += FLOG2(x);
if (cc->size >= nb_samples) {
fn(cc->sum) -= px;
fn(cc->log_sum) -= FLOG2(px);
}
qq[pos] = x;
pos++;
if (pos >= nb_samples)
pos = 0;
cc->position = pos;
if (cc->size < nb_samples)
cc->size++;
n = cc->size;
empty = (front == back) && (ss[front] == ZERO);
PEAKS(ZERO, >, x, px)
ss[back] = x;
cc->front = front;
cc->back = back;
}
static ftype fn(get_peak)(ChannelContext *cc, ftype *score)
{
ftype s, *ss = cc->dqueue;
s = FEXP2(fn(cc->log_sum) / cc->size) / (fn(cc->sum) / cc->size);
*score = LIN2LOG(s);
return ss[cc->front];
}
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 int isample_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 int mode = s->mode;
const ftype power = (mode == CUT_BELOW || mode == CUT_ABOVE) ? -ONE : ONE;
const ftype grelease = s->grelease_coef;
const ftype gattack = s->gattack_coef;
const ftype drelease = s->drelease_coef;
const ftype dattack = s->dattack_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 is_disabled = ctx->is_disabled;
const int detection = s->detection;
const int tftype = s->tftype;
const ftype *da = fn(s->da);
const ftype *dm = fn(s->dm);
if (detection == DET_ON) {
for (int ch = start; ch < end; ch++) {
const ftype *src = (const ftype *)in->extended_data[ch];
ChannelContext *cc = &s->cc[ch];
ftype *tstate = fn(cc->tstate);
ftype new_threshold = ZERO;
if (cc->detection != detection) {
cc->detection = detection;
fn(cc->new_threshold_log) = LIN2LOG(EPSILON);
}
for (int n = 0; n < in->nb_samples; n++) {
ftype detect = FABS(fn(get_svf)(src[n], dm, da, tstate));
new_threshold = FMAX(new_threshold, detect);
}
fn(cc->new_threshold_log) = FMAX(fn(cc->new_threshold_log), LIN2LOG(new_threshold));
}
} else if (detection == DET_ADAPTIVE) {
for (int ch = start; ch < end; ch++) {
const ftype *src = (const ftype *)in->extended_data[ch];
ChannelContext *cc = &s->cc[ch];
ftype *tstate = fn(cc->tstate);
ftype score, peak;
for (int n = 0; n < in->nb_samples; n++) {
ftype detect = FMAX(FABS(fn(get_svf)(src[n], dm, da, tstate)), EPSILON);
fn(queue_sample)(cc, detect, isample_rate);
}
peak = fn(get_peak)(cc, &score);
if (score >= -3.5) {
fn(cc->threshold_log) = LIN2LOG(peak);
} else if (cc->detection == DET_UNSET) {
fn(cc->threshold_log) = s->threshold_log;
}
cc->detection = detection;
}
} else if (detection == DET_DISABLED) {
for (int ch = start; ch < end; ch++) {
ChannelContext *cc = &s->cc[ch];
fn(cc->threshold_log) = s->threshold_log;
cc->detection = detection;
}
} else if (detection == DET_OFF) {
for (int ch = start; ch < end; ch++) {
ChannelContext *cc = &s->cc[ch];
if (cc->detection == DET_ON)
fn(cc->threshold_log) = fn(cc->new_threshold_log);
else if (cc->detection == DET_UNSET)
fn(cc->threshold_log) = s->threshold_log;
cc->detection = detection;
}
}
for (int ch = start; ch < end; ch++) {
const ftype *src = (const ftype *)in->extended_data[ch];
ftype *dst = (ftype *)out->extended_data[ch];
ChannelContext *cc = &s->cc[ch];
const ftype threshold_log = fn(cc->threshold_log);
ftype *fa = fn(cc->fa), *fm = fn(cc->fm);
ftype *fstate = fn(cc->fstate);
ftype *dstate = fn(cc->dstate);
ftype detect = fn(cc->detect);
ftype lin_gain = fn(cc->lin_gain);
int init = cc->init;
for (int n = 0; n < out->nb_samples; n++) {
ftype new_detect, new_lin_gain = ONE;
ftype f, v, listen, k, g, ld;
listen = fn(get_svf)(src[n], dm, da, dstate);
if (mode > LISTEN) {
new_detect = FABS(listen);
f = (new_detect > detect) * dattack + (new_detect <= detect) * drelease;
detect = f * new_detect + (ONE - f) * detect;
}
switch (mode) {
case LISTEN:
break;
case CUT_BELOW:
case BOOST_BELOW:
ld = LIN2LOG(detect);
if (ld < threshold_log) {
ftype new_log_gain = CLIP(makeup + (threshold_log - ld) * ratio, ZERO, range) * power;
new_lin_gain = LOG2LIN(new_log_gain);
}
break;
case CUT_ABOVE:
case BOOST_ABOVE:
ld = LIN2LOG(detect);
if (ld > threshold_log) {
ftype new_log_gain = CLIP(makeup + (ld - threshold_log) * ratio, ZERO, range) * power;
new_lin_gain = LOG2LIN(new_log_gain);
}
break;
}
f = (new_lin_gain > lin_gain) * gattack + (new_lin_gain <= lin_gain) * grelease;
new_lin_gain = f * new_lin_gain + (ONE - f) * lin_gain;
if (lin_gain != new_lin_gain || !init) {
init = 1;
lin_gain = new_lin_gain;
switch (tftype) {
case 0:
k = itqfactor / lin_gain;
fa[0] = ONE / (ONE + fg * (fg + k));
fa[1] = fg * fa[0];
fa[2] = fg * fa[1];
fm[0] = ONE;
fm[1] = k * (lin_gain * lin_gain - ONE);
fm[2] = ZERO;
break;
case 1:
k = itqfactor;
g = fg / SQRT(lin_gain);
fa[0] = ONE / (ONE + g * (g + k));
fa[1] = g * fa[0];
fa[2] = g * fa[1];
fm[0] = ONE;
fm[1] = k * (lin_gain - ONE);
fm[2] = lin_gain * lin_gain - ONE;
break;
case 2:
k = itqfactor;
g = fg * SQRT(lin_gain);
fa[0] = ONE / (ONE + g * (g + k));
fa[1] = g * fa[0];
fa[2] = g * fa[1];
fm[0] = lin_gain * lin_gain;
fm[1] = k * (ONE - lin_gain) * lin_gain;
fm[2] = ONE - lin_gain * lin_gain;
break;
}
}
v = fn(get_svf)(src[n], fm, fa, fstate);
v = mode == LISTEN ? listen : v;
dst[n] = is_disabled ? src[n] : v;
}
fn(cc->detect) = detect;
fn(cc->lin_gain) = lin_gain;
cc->init = 1;
}
return 0;
}
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