avfilter: add agate filter

Signed-off-by: Paul B Mahol <onemda@gmail.com>

8 files changed, 333 insertions, 24 deletions

diff --git a/Changelog b/Changelog
index 12fe77c708..6200eebf71 100644
--- a/Changelog
+++ b/Changelog
@@ -10,6 +10,7 @@ version <next>:
 - stereotools filter
 - rubberband filter
 - tremolo filter
+- agate filter
 
 
 version 2.8:
diff --git a/doc/filters.texi b/doc/filters.texi
index 5bbbaf0b78..5b00eca19f 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -641,6 +641,57 @@ Force the output to either unsigned 8-bit or signed 16-bit stereo
 aformat=sample_fmts=u8|s16:channel_layouts=stereo
 @end example
 
+@section agate
+
+A gate is mainly used to reduce lower parts of a signal. This kind of signal
+processing reduces disturbing noise between useful signals.
+
+Gating is done by detecting the volume below a chosen level @var{threshold}
+and divide it by the factor set with @var{ratio}. The bottom of the noise
+floor is set via @var{range}. Because an exact manipulation of the signal
+would cause distortion of the waveform the reduction can be levelled over
+time. This is done by setting @var{attack} and @var{release}.
+
+@var{attack} determines how long the signal has to fall below the threshold
+before any reduction will occur and @var{release} sets the time the signal
+has to raise above the threshold to reduce the reduction again.
+Shorter signals than the chosen attack time will be left untouched.
+
+@table @option
+@item level_in
+Set input level before filtering.
+
+@item range
+Set the level of gain reduction when the signal is below the threshold.
+
+@item threshold
+If a signal rises above this level the gain reduction is released.
+
+@item ratio
+Set a ratio about which the signal is reduced.
+
+@item attack
+Amount of milliseconds the signal has to rise above the threshold before gain
+reduction stops.
+
+@item release
+Amount of milliseconds the signal has to fall below the threshold before the
+reduction is increased again.
+
+@item makeup
+Set amount of amplification of signal after processing.
+
+@item knee
+Curve the sharp knee around the threshold to enter gain reduction more softly.
+
+@item detection
+Choose if exact signal should be taken for detection or an RMS like one.
+
+@item link
+Choose if the average level between all channels or the louder channel affects
+the reduction.
+@end table
+
 @section alimiter
 
 The limiter prevents input signal from raising over a desired threshold.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 8db5da9bc6..be177db4e1 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -29,6 +29,7 @@ OBJS-$(CONFIG_AECHO_FILTER)                  += af_aecho.o
 OBJS-$(CONFIG_AEVAL_FILTER)                  += aeval.o
 OBJS-$(CONFIG_AFADE_FILTER)                  += af_afade.o
 OBJS-$(CONFIG_AFORMAT_FILTER)                += af_aformat.o
+OBJS-$(CONFIG_AGATE_FILTER)                  += af_agate.o
 OBJS-$(CONFIG_AINTERLEAVE_FILTER)            += f_interleave.o
 OBJS-$(CONFIG_ALIMITER_FILTER)               += af_alimiter.o
 OBJS-$(CONFIG_ALLPASS_FILTER)                += af_biquads.o
diff --git a/libavfilter/af_agate.c b/libavfilter/af_agate.c
new file mode 100644
index 0000000000..46ee226235
--- /dev/null
+++ b/libavfilter/af_agate.c
@@ -0,0 +1,237 @@
+/*
+ * Copyright (C) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen, Damien Zammit
+ *
+ * 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
+ */
+
+#include "libavutil/channel_layout.h"
+#include "libavutil/opt.h"
+#include "avfilter.h"
+#include "audio.h"
+#include "formats.h"
+#include "hermite.h"
+
+typedef struct AudioGateContext {
+    const AVClass *class;
+
+    double level_in;
+    double attack;
+    double release;
+    double threshold;
+    double ratio;
+    double knee;
+    double makeup;
+    double range;
+    int link;
+    int detection;
+
+    double thres;
+    double knee_start;
+    double lin_knee_stop;
+    double knee_stop;
+    double lin_slope;
+    double attack_coeff;
+    double release_coeff;
+} AudioGateContext;
+
+#define OFFSET(x) offsetof(AudioGateContext, x)
+#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption agate_options[] = {
+    { "level_in",  "set input level",        OFFSET(level_in),  AV_OPT_TYPE_DOUBLE, {.dbl=1},           0.015625,   64, A },
+    { "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 },
+    { "attack",    "set attack",             OFFSET(attack),    AV_OPT_TYPE_DOUBLE, {.dbl=20},          0.01, 9000, A },
+    { "release",   "set release",            OFFSET(release),   AV_OPT_TYPE_DOUBLE, {.dbl=250},         0.01, 9000, A },
+    { "makeup",    "set makeup gain",        OFFSET(makeup),    AV_OPT_TYPE_DOUBLE, {.dbl=1},           1,   64, A },
+    { "knee",      "set knee",               OFFSET(knee),      AV_OPT_TYPE_DOUBLE, {.dbl=2.828427125}, 1,    8, A },
+    { "detection", "set detection",          OFFSET(detection), AV_OPT_TYPE_INT,    {.i64=0},           0,    1, A, "detection" },
+    {   "peak",    0,                        0,                 AV_OPT_TYPE_CONST,  {.i64=0},           0,    0, A, "detection" },
+    {   "rms",     0,                        0,                 AV_OPT_TYPE_CONST,  {.i64=1},           0,    0, A, "detection" },
+    { "link",      "set link",               OFFSET(link),      AV_OPT_TYPE_INT,    {.i64=0},           0,    1, A, "link" },
+    {   "average", 0,                        0,                 AV_OPT_TYPE_CONST,  {.i64=0},           0,    0, A, "link" },
+    {   "maximum", 0,                        0,                 AV_OPT_TYPE_CONST,  {.i64=1},           0,    0, A, "link" },
+    { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(agate);
+
+static int query_formats(AVFilterContext *ctx)
+{
+    AVFilterFormats *formats = NULL;
+    AVFilterChannelLayouts *layouts;
+    int ret;
+
+    ff_add_format(&formats, AV_SAMPLE_FMT_DBL);
+    ret = ff_set_common_formats(ctx, formats);
+    if (ret < 0)
+        return ret;
+
+    layouts = ff_all_channel_counts();
+    if (!layouts)
+        return AVERROR(ENOMEM);
+    ret = ff_set_common_channel_layouts(ctx, layouts);
+    if (ret < 0)
+        return ret;
+
+    formats = ff_all_samplerates();
+    if (!formats)
+        return AVERROR(ENOMEM);
+
+    return ff_set_common_samplerates(ctx, formats);
+}
+
+static int config_input(AVFilterLink *inlink)
+{
+    AVFilterContext *ctx = inlink->dst;
+    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;
+
+    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->thres = log(lin_threshold);
+    s->knee_start = log(lin_knee_start);
+    s->knee_stop = log(s->lin_knee_stop);
+
+    return 0;
+}
+
+// A fake infinity value (because real infinity may break some hosts)
+#define FAKE_INFINITY (65536.0 * 65536.0)
+
+// Check for infinity (with appropriate-ish tolerance)
+#define IS_FAKE_INFINITY(value) (fabs(value-FAKE_INFINITY) < 1.0)
+
+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)
+{
+    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) {
+            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.;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+    AVFilterContext *ctx = inlink->dst;
+    AVFilterLink *outlink = ctx->outputs[0];
+    AudioGateContext *s = ctx->priv;
+    const double *src = (const double *)in->data[0];
+    const double makeup = s->makeup;
+    const double attack_coeff = s->attack_coeff;
+    const double release_coeff = s->release_coeff;
+    const double level_in = s->level_in;
+    AVFrame *out = NULL;
+    double *dst;
+    int n, c;
+
+    if (av_frame_is_writable(in)) {
+        out = in;
+    } else {
+        AVFrame *out = ff_get_audio_buffer(inlink, in->nb_samples);
+        if (!out) {
+            av_frame_free(&in);
+            return AVERROR(ENOMEM);
+        }
+        av_frame_copy_props(out, in);
+    }
+    dst = (double *)out->data[0];
+
+    for (n = 0; n < in->nb_samples; n++, src += inlink->channels, dst += inlink->channels) {
+        double abs_sample = FFABS(src[0]), gain = 1.0;
+
+        for (c = 0; c < inlink->channels; c++)
+            dst[c] = src[c] * level_in;
+
+        if (s->link == 1) {
+            for (c = 1; c < inlink->channels; c++)
+                abs_sample = FFMAX(FFABS(src[c]), abs_sample);
+        } else {
+            for (c = 1; c < inlink->channels; c++)
+                abs_sample += FFABS(src[c]);
+
+            abs_sample /= inlink->channels;
+        }
+
+        if (s->detection)
+            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)
+            gain = output_gain(s->lin_slope, s->ratio, s->thres,
+                               s->knee, s->knee_start, s->knee_stop,
+                               s->lin_knee_stop, s->range);
+
+        for (c = 0; c < inlink->channels; c++)
+            dst[c] *= gain * makeup;
+    }
+
+    if (out != in)
+        av_frame_free(&in);
+    return ff_filter_frame(outlink, out);
+}
+
+static const AVFilterPad inputs[] = {
+    {
+        .name         = "default",
+        .type         = AVMEDIA_TYPE_AUDIO,
+        .filter_frame = filter_frame,
+        .config_props = config_input,
+    },
+    { NULL }
+};
+
+static const AVFilterPad outputs[] = {
+    {
+        .name = "default",
+        .type = AVMEDIA_TYPE_AUDIO,
+    },
+    { NULL }
+};
+
+AVFilter ff_af_agate = {
+    .name           = "agate",
+    .description    = NULL_IF_CONFIG_SMALL("Audio gate."),
+    .query_formats  = query_formats,
+    .priv_size      = sizeof(AudioGateContext),
+    .priv_class     = &agate_class,
+    .inputs         = inputs,
+    .outputs        = outputs,
+};
diff --git a/libavfilter/af_sidechaincompress.c b/libavfilter/af_sidechaincompress.c
index 40ffca6dc3..b8a81fc0bb 100644
--- a/libavfilter/af_sidechaincompress.c
+++ b/libavfilter/af_sidechaincompress.c
@@ -32,6 +32,7 @@
 #include "audio.h"
 #include "avfilter.h"
 #include "formats.h"
+#include "hermite.h"
 #include "internal.h"
 
 typedef struct SidechainCompressContext {
@@ -90,29 +91,6 @@ static av_cold int init(AVFilterContext *ctx)
     return 0;
 }
 
-static inline double hermite_interpolation(double x, double x0, double x1,
-                                           double p0, double p1,
-                                           double m0, double m1)
-{
-    double width = x1 - x0;
-    double t = (x - x0) / width;
-    double t2, t3;
-    double ct0, ct1, ct2, ct3;
-
-    m0 *= width;
-    m1 *= width;
-
-    t2 = t*t;
-    t3 = t2*t;
-    ct0 = p0;
-    ct1 = m0;
-
-    ct2 = -3 * p0 - 2 * m0 + 3 * p1 - m1;
-    ct3 = 2 * p0 + m0  - 2 * p1 + m1;
-
-    return ct3 * t3 + ct2 * t2 + ct1 * t + ct0;
-}
-
 // A fake infinity value (because real infinity may break some hosts)
 #define FAKE_INFINITY (65536.0 * 65536.0)
 
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 292ecde289..7c93c1dcda 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -51,6 +51,7 @@ void avfilter_register_all(void)
     REGISTER_FILTER(AEVAL,          aeval,          af);
     REGISTER_FILTER(AFADE,          afade,          af);
     REGISTER_FILTER(AFORMAT,        aformat,        af);
+    REGISTER_FILTER(AGATE,          agate,          af);
     REGISTER_FILTER(AINTERLEAVE,    ainterleave,    af);
     REGISTER_FILTER(ALIMITER,       alimiter,       af);
     REGISTER_FILTER(ALLPASS,        allpass,        af);
diff --git a/libavfilter/hermite.h b/libavfilter/hermite.h
new file mode 100644
index 0000000000..6142391b2e
--- /dev/null
+++ b/libavfilter/hermite.h
@@ -0,0 +1,40 @@
+/*
+ * 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
+ */
+
+inline double hermite_interpolation(double x, double x0, double x1,
+                                    double p0, double p1,
+                                    double m0, double m1)
+{
+    double width = x1 - x0;
+    double t = (x - x0) / width;
+    double t2, t3;
+    double ct0, ct1, ct2, ct3;
+
+    m0 *= width;
+    m1 *= width;
+
+    t2 = t*t;
+    t3 = t2*t;
+    ct0 = p0;
+    ct1 = m0;
+
+    ct2 = -3 * p0 - 2 * m0 + 3 * p1 - m1;
+    ct3 = 2 * p0 + m0  - 2 * p1 + m1;
+
+    return ct3 * t3 + ct2 * t2 + ct1 * t + ct0;
+}
diff --git a/libavfilter/version.h b/libavfilter/version.h
index 96503770e1..1426cf8100 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -30,7 +30,7 @@
 #include "libavutil/version.h"
 
 #define LIBAVFILTER_VERSION_MAJOR   6
-#define LIBAVFILTER_VERSION_MINOR   7
+#define LIBAVFILTER_VERSION_MINOR   8
 #define LIBAVFILTER_VERSION_MICRO 100
 
 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \