[FFmpeg-cvslog] avfilter/af_asoftclip: add two more useful options for finer filtering
Paul B Mahol
git at videolan.org
Fri Dec 18 14:12:57 EET 2020
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Fri Dec 18 13:09:44 2020 +0100| [7e3f20c43c42c588730b6f9ab7cdf6b325c7ea8d] | committer: Paul B Mahol
avfilter/af_asoftclip: add two more useful options for finer filtering
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=7e3f20c43c42c588730b6f9ab7cdf6b325c7ea8d
---
doc/filters.texi | 6 +++
libavfilter/af_asoftclip.c | 124 +++++++++++++++++++++++++++++++--------------
2 files changed, 92 insertions(+), 38 deletions(-)
diff --git a/doc/filters.texi b/doc/filters.texi
index 8c291746bb..d634fa25fd 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -2454,6 +2454,12 @@ It accepts the following values:
@item erf
@end table
+ at item threshold
+Set threshold from where to start clipping. Default value is 0dB or 1.
+
+ at item output
+Set gain applied to output. Default value is 0dB or 1.
+
@item param
Set additional parameter which controls sigmoid function.
diff --git a/libavfilter/af_asoftclip.c b/libavfilter/af_asoftclip.c
index aaae3c6d4b..4900cb6ddf 100644
--- a/libavfilter/af_asoftclip.c
+++ b/libavfilter/af_asoftclip.c
@@ -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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