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avfilter/af_asoftclip: add two more useful options for finer filtering

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pull/371/head
Paul B Mahol 4 years ago
parent 1eb751955e
commit 7e3f20c43c
2 changed files with 92 additions and 38 deletions
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  1. 6
      doc/filters.texi
  2. 124
      libavfilter/af_asoftclip.c

6
doc/filters.texi
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@ -2454,6 +2454,12 @@ It accepts the following values:
@item erf
@end table
@item threshold
Set threshold from where to start clipping. Default value is 0dB or 1.
@item output
Set gain applied to output. Default value is 0dB or 1.
@item param
Set additional parameter which controls sigmoid function.

124
libavfilter/af_asoftclip.c
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@ -45,6 +45,8 @@ typedef struct ASoftClipContext {
int type;
int oversample;
int64_t delay;
double threshold;
double output;
double param;
SwrContext *up_ctx;
@ -71,6 +73,8 @@ static const AVOption asoftclip_options[] = {
{ "quintic", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_QUINTIC},0, 0, A, "types" },
{ "sin", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_SIN}, 0, 0, A, "types" },
{ "erf", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_ERF}, 0, 0, A, "types" },
{ "threshold", "set softclip threshold", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.000001, 1, A },
{ "output", "set softclip output gain", OFFSET(output), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.000001, 16, A },
{ "param", "set softclip parameter", OFFSET(param), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.01, 3, A },
{ "oversample", "set oversample factor", OFFSET(oversample), AV_OPT_TYPE_INT, {.i64=1}, 1, 32, F },
{ NULL }
@ -108,13 +112,14 @@ static int query_formats(AVFilterContext *ctx)
return ff_set_common_samplerates(ctx, formats);
}
#define SQR(x) ((x) * (x))
static void filter_flt(ASoftClipContext *s,
void **dptr, const void **sptr,
int nb_samples, int channels,
int start, int end)
{
float threshold = s->threshold;
float gain = s->output * threshold;
float factor = 1.f / threshold;
float param = s->param;
for (int c = start; c < end; c++) {
@ -124,53 +129,73 @@ static void filter_flt(ASoftClipContext *s,
switch (s->type) {
case ASC_HARD:
for (int n = 0; n < nb_samples; n++) {
dst[n] = av_clipf(src[n], -1.f, 1.f);
dst[n] = av_clipf(src[n] * factor, -1.f, 1.f);
dst[n] *= gain;
}
break;
case ASC_TANH:
for (int n = 0; n < nb_samples; n++) {
dst[n] = tanhf(src[n] * param);
dst[n] = tanhf(src[n] * factor * param);
dst[n] *= gain;
}
break;
case ASC_ATAN:
for (int n = 0; n < nb_samples; n++)
dst[n] = 2.f / M_PI * atanf(src[n] * param);
for (int n = 0; n < nb_samples; n++) {
dst[n] = 2.f / M_PI * atanf(src[n] * factor * param);
dst[n] *= gain;
}
break;
case ASC_CUBIC:
for (int n = 0; n < nb_samples; n++) {
if (FFABS(src[n]) >= 1.5f)
dst[n] = FFSIGN(src[n]);
float sample = src[n] * factor;
if (FFABS(sample) >= 1.5f)
dst[n] = FFSIGN(sample);
else
dst[n] = src[n] - 0.1481f * powf(src[n], 3.f);
dst[n] = sample - 0.1481f * powf(sample, 3.f);
dst[n] *= gain;
}
break;
case ASC_EXP:
for (int n = 0; n < nb_samples; n++)
dst[n] = 2.f / (1.f + expf(-2.f * src[n])) - 1.;
for (int n = 0; n < nb_samples; n++) {
dst[n] = 2.f / (1.f + expf(-2.f * src[n] * factor)) - 1.;
dst[n] *= gain;
}
break;
case ASC_ALG:
for (int n = 0; n < nb_samples; n++)
dst[n] = src[n] / (sqrtf(param + src[n] * src[n]));
for (int n = 0; n < nb_samples; n++) {
float sample = src[n] * factor;
dst[n] = sample / (sqrtf(param + sample * sample));
dst[n] *= gain;
}
break;
case ASC_QUINTIC:
for (int n = 0; n < nb_samples; n++) {
if (FFABS(src[n]) >= 1.25)
dst[n] = FFSIGN(src[n]);
float sample = src[n] * factor;
if (FFABS(sample) >= 1.25)
dst[n] = FFSIGN(sample);
else
dst[n] = src[n] - 0.08192f * powf(src[n], 5.f);
dst[n] = sample - 0.08192f * powf(sample, 5.f);
dst[n] *= gain;
}
break;
case ASC_SIN:
for (int n = 0; n < nb_samples; n++) {
if (FFABS(src[n]) >= M_PI_2)
dst[n] = FFSIGN(src[n]);
float sample = src[n] * factor;
if (FFABS(sample) >= M_PI_2)
dst[n] = FFSIGN(sample);
else
dst[n] = sinf(src[n]);
dst[n] = sinf(sample);
dst[n] *= gain;
}
break;
case ASC_ERF:
for (int n = 0; n < nb_samples; n++) {
dst[n] = erff(src[n]);
dst[n] = erff(src[n] * factor);
dst[n] *= gain;
}
break;
default:
@ -184,6 +209,9 @@ static void filter_dbl(ASoftClipContext *s,
int nb_samples, int channels,
int start, int end)
{
double threshold = s->threshold;
double gain = s->output * threshold;
double factor = 1. / threshold;
double param = s->param;
for (int c = start; c < end; c++) {
@ -193,53 +221,73 @@ static void filter_dbl(ASoftClipContext *s,
switch (s->type) {
case ASC_HARD:
for (int n = 0; n < nb_samples; n++) {
dst[n] = av_clipd(src[n], -1., 1.);
dst[n] = av_clipd(src[n] * factor, -1., 1.);
dst[n] *= gain;
}
break;
case ASC_TANH:
for (int n = 0; n < nb_samples; n++) {
dst[n] = tanh(src[n] * param);
dst[n] = tanh(src[n] * factor * param);
dst[n] *= gain;
}
break;
case ASC_ATAN:
for (int n = 0; n < nb_samples; n++)
dst[n] = 2. / M_PI * atan(src[n] * param);
for (int n = 0; n < nb_samples; n++) {
dst[n] = 2. / M_PI * atan(src[n] * factor * param);
dst[n] *= gain;
}
break;
case ASC_CUBIC:
for (int n = 0; n < nb_samples; n++) {
if (FFABS(src[n]) >= 1.5)
dst[n] = FFSIGN(src[n]);
double sample = src[n] * factor;
if (FFABS(sample) >= 1.5)
dst[n] = FFSIGN(sample);
else
dst[n] = src[n] - 0.1481 * pow(src[n], 3.);
dst[n] = sample - 0.1481 * pow(sample, 3.);
dst[n] *= gain;
}
break;
case ASC_EXP:
for (int n = 0; n < nb_samples; n++)
dst[n] = 2. / (1. + exp(-2. * src[n])) - 1.;
for (int n = 0; n < nb_samples; n++) {
dst[n] = 2. / (1. + exp(-2. * src[n] * factor)) - 1.;
dst[n] *= gain;
}
break;
case ASC_ALG:
for (int n = 0; n < nb_samples; n++)
dst[n] = src[n] / (sqrt(param + src[n] * src[n]));
for (int n = 0; n < nb_samples; n++) {
double sample = src[n] * factor;
dst[n] = sample / (sqrt(param + sample * sample));
dst[n] *= gain;
}
break;
case ASC_QUINTIC:
for (int n = 0; n < nb_samples; n++) {
if (FFABS(src[n]) >= 1.25)
dst[n] = FFSIGN(src[n]);
double sample = src[n] * factor;
if (FFABS(sample) >= 1.25)
dst[n] = FFSIGN(sample);
else
dst[n] = src[n] - 0.08192 * pow(src[n], 5.);
dst[n] = sample - 0.08192 * pow(sample, 5.);
dst[n] *= gain;
}
break;
case ASC_SIN:
for (int n = 0; n < nb_samples; n++) {
if (FFABS(src[n]) >= M_PI_2)
dst[n] = FFSIGN(src[n]);
double sample = src[n] * factor;
if (FFABS(sample) >= M_PI_2)
dst[n] = FFSIGN(sample);
else
dst[n] = sin(src[n]);
dst[n] = sin(sample);
dst[n] *= gain;
}
break;
case ASC_ERF:
for (int n = 0; n < nb_samples; n++) {
dst[n] = erf(src[n]);
dst[n] = erf(src[n] * factor);
dst[n] *= gain;
}
break;
default:

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