[FFmpeg-cvslog] avfilter/af_agate: implement mode option
Paul B Mahol
git at videolan.org
Thu Apr 18 00:43:03 EEST 2019
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Wed Apr 17 23:33:51 2019 +0200| [8a3ed5a3136461bc0460de8d0de48f3093719de6] | committer: Paul B Mahol
avfilter/af_agate: implement mode option
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=8a3ed5a3136461bc0460de8d0de48f3093719de6
---
doc/filters.texi | 14 +++++++++++++
libavfilter/af_agate.c | 57 ++++++++++++++++++++++++++++++--------------------
2 files changed, 48 insertions(+), 23 deletions(-)
diff --git a/doc/filters.texi b/doc/filters.texi
index 05a9ca3684..4dd1a5de85 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -1308,9 +1308,16 @@ Shorter signals than the chosen attack time will be left untouched.
Set input level before filtering.
Default is 1. Allowed range is from 0.015625 to 64.
+ at item mode
+Set the mode of operation. Can be @code{upward} or @code{downward}.
+Default is @code{downward}. If set to @code{upward} mode, higher parts of signal
+will be amplified, expanding dynamic range in upward direction.
+Otherwise, in case of @code{downward} lower parts of signal will be reduced.
+
@item range
Set the level of gain reduction when the signal is below the threshold.
Default is 0.06125. Allowed range is from 0 to 1.
+Setting this to 0 disables reduction and then filter behaves like expander.
@item threshold
If a signal rises above this level the gain reduction is released.
@@ -4331,9 +4338,16 @@ The filter accepts the following options:
Set input level before filtering.
Default is 1. Allowed range is from 0.015625 to 64.
+ at item mode
+Set the mode of operation. Can be @code{upward} or @code{downward}.
+Default is @code{downward}. If set to @code{upward} mode, higher parts of signal
+will be amplified, expanding dynamic range in upward direction.
+Otherwise, in case of @code{downward} lower parts of signal will be reduced.
+
@item range
Set the level of gain reduction when the signal is below the threshold.
Default is 0.06125. Allowed range is from 0 to 1.
+Setting this to 0 disables reduction and then filter behaves like expander.
@item threshold
If a signal rises above this level the gain reduction is released.
diff --git a/libavfilter/af_agate.c b/libavfilter/af_agate.c
index ba96863a68..0609dc222e 100644
--- a/libavfilter/af_agate.c
+++ b/libavfilter/af_agate.c
@@ -47,11 +47,13 @@ typedef struct AudioGateContext {
double range;
int link;
int detection;
+ int mode;
double thres;
double knee_start;
- double lin_knee_stop;
double knee_stop;
+ double lin_knee_start;
+ double lin_knee_stop;
double lin_slope;
double attack_coeff;
double release_coeff;
@@ -65,6 +67,9 @@ typedef struct AudioGateContext {
static const AVOption options[] = {
{ "level_in", "set input level", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
+ { "mode", "set mode", OFFSET(mode), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A, "mode" },
+ { "downward",0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "mode" },
+ { "upward", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "mode" },
{ "range", "set max gain reduction", OFFSET(range), AV_OPT_TYPE_DOUBLE, {.dbl=0.06125}, 0, 1, A },
{ "threshold", "set threshold", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=0.125}, 0, 1, A },
{ "ratio", "set ratio", OFFSET(ratio), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 1, 9000, A },
@@ -88,7 +93,6 @@ static int agate_config_input(AVFilterLink *inlink)
AudioGateContext *s = ctx->priv;
double lin_threshold = s->threshold;
double lin_knee_sqrt = sqrt(s->knee);
- double lin_knee_start;
if (s->detection)
lin_threshold *= lin_threshold;
@@ -96,9 +100,9 @@ static int agate_config_input(AVFilterLink *inlink)
s->attack_coeff = FFMIN(1., 1. / (s->attack * inlink->sample_rate / 4000.));
s->release_coeff = FFMIN(1., 1. / (s->release * inlink->sample_rate / 4000.));
s->lin_knee_stop = lin_threshold * lin_knee_sqrt;
- lin_knee_start = lin_threshold / lin_knee_sqrt;
+ s->lin_knee_start = lin_threshold / lin_knee_sqrt;
s->thres = log(lin_threshold);
- s->knee_start = log(lin_knee_start);
+ s->knee_start = log(s->lin_knee_start);
s->knee_stop = log(s->lin_knee_stop);
return 0;
@@ -112,26 +116,26 @@ static int agate_config_input(AVFilterLink *inlink)
static double output_gain(double lin_slope, double ratio, double thres,
double knee, double knee_start, double knee_stop,
- double lin_knee_stop, double range)
+ double range, int mode)
{
- if (lin_slope < lin_knee_stop) {
- double slope = log(lin_slope);
- double tratio = ratio;
- double gain = 0.;
- double delta = 0.;
-
- if (IS_FAKE_INFINITY(ratio))
- tratio = 1000.;
- gain = (slope - thres) * tratio + thres;
- delta = tratio;
-
- if (knee > 1. && slope > knee_start) {
+ double slope = log(lin_slope);
+ double tratio = ratio;
+ double gain = 0.;
+ double delta = 0.;
+
+ if (IS_FAKE_INFINITY(ratio))
+ tratio = 1000.;
+ gain = (slope - thres) * tratio + thres;
+ delta = tratio;
+
+ if (mode) {
+ if (knee > 1. && slope < knee_stop)
+ gain = hermite_interpolation(slope, knee_stop, knee_start, ((knee_stop - thres) * tratio + thres), knee_start, delta, 1.);
+ } else {
+ if (knee > 1. && slope > knee_start)
gain = hermite_interpolation(slope, knee_start, knee_stop, ((knee_start - thres) * tratio + thres), knee_stop, delta, 1.);
- }
- return FFMAX(range, exp(gain - slope));
}
-
- return 1.;
+ return FFMAX(range, exp(gain - slope));
}
static void gate(AudioGateContext *s,
@@ -146,6 +150,7 @@ static void gate(AudioGateContext *s,
for (n = 0; n < nb_samples; n++, src += inlink->channels, dst += inlink->channels, scsrc += sclink->channels) {
double abs_sample = fabs(scsrc[0] * level_sc), gain = 1.0;
+ int detected;
if (s->link == 1) {
for (c = 1; c < sclink->channels; c++)
@@ -161,10 +166,16 @@ static void gate(AudioGateContext *s,
abs_sample *= abs_sample;
s->lin_slope += (abs_sample - s->lin_slope) * (abs_sample > s->lin_slope ? attack_coeff : release_coeff);
- if (s->lin_slope > 0.0)
+
+ if (s->mode)
+ detected = s->lin_slope > s->lin_knee_start;
+ else
+ detected = s->lin_slope < s->lin_knee_stop;
+
+ if (s->lin_slope > 0.0 && detected)
gain = output_gain(s->lin_slope, s->ratio, s->thres,
s->knee, s->knee_start, s->knee_stop,
- s->lin_knee_stop, s->range);
+ s->range, s->mode);
for (c = 0; c < inlink->channels; c++)
dst[c] = src[c] * level_in * gain * makeup;
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