avfilter: add audio dynamic smooth filter

3 files changed, 144 insertions, 0 deletions

diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 551d13aadc..c8082c4a2f 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -44,6 +44,7 @@ OBJS-$(CONFIG_ADECORRELATE_FILTER)           += af_adecorrelate.o
 OBJS-$(CONFIG_ADELAY_FILTER)                 += af_adelay.o
 OBJS-$(CONFIG_ADENORM_FILTER)                += af_adenorm.o
 OBJS-$(CONFIG_ADERIVATIVE_FILTER)            += af_aderivative.o
+OBJS-$(CONFIG_ADYNAMICSMOOTH_FILTER)         += af_adynamicsmooth.o
 OBJS-$(CONFIG_AECHO_FILTER)                  += af_aecho.o
 OBJS-$(CONFIG_AEMPHASIS_FILTER)              += af_aemphasis.o
 OBJS-$(CONFIG_AEVAL_FILTER)                  += aeval.o
diff --git a/libavfilter/af_adynamicsmooth.c b/libavfilter/af_adynamicsmooth.c
new file mode 100644
index 0000000000..4e00fecc6a
--- /dev/null
+++ b/libavfilter/af_adynamicsmooth.c
@@ -0,0 +1,142 @@
+/*
+ * 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/ffmath.h"
+#include "libavutil/opt.h"
+#include "avfilter.h"
+#include "audio.h"
+#include "formats.h"
+
+typedef struct AudioDynamicSmoothContext {
+    const AVClass *class;
+
+    double sensitivity;
+    double basefreq;
+
+    AVFrame *coeffs;
+} AudioDynamicSmoothContext;
+
+static int config_input(AVFilterLink *inlink)
+{
+    AVFilterContext *ctx = inlink->dst;
+    AudioDynamicSmoothContext *s = ctx->priv;
+
+    s->coeffs = ff_get_audio_buffer(inlink, 3);
+    if (!s->coeffs)
+        return AVERROR(ENOMEM);
+
+    return 0;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+    AVFilterContext *ctx = inlink->dst;
+    AVFilterLink *outlink = ctx->outputs[0];
+    AudioDynamicSmoothContext *s = ctx->priv;
+    const double sensitivity = s->sensitivity;
+    const double wc = s->basefreq / in->sample_rate;
+    AVFrame *out;
+
+    if (av_frame_is_writable(in)) {
+        out = in;
+    } else {
+        out = ff_get_audio_buffer(outlink, in->nb_samples);
+        if (!out) {
+            av_frame_free(&in);
+            return AVERROR(ENOMEM);
+        }
+        av_frame_copy_props(out, in);
+    }
+
+    for (int ch = 0; ch < out->channels; ch++) {
+        const double *src = (const double *)in->extended_data[ch];
+        double *dst = (double *)out->extended_data[ch];
+        double *coeffs = (double *)s->coeffs->extended_data[ch];
+        double low1 = coeffs[0];
+        double low2 = coeffs[1];
+        double inz  = coeffs[2];
+
+        for (int n = 0; n < out->nb_samples; n++) {
+            double low1z = low1;
+            double low2z = low2;
+            double bandz = low2z - low1z;
+            double wd = wc + sensitivity * fabs(bandz);
+            double g = fmin(1., wd * (5.9948827 + wd * (-11.969296 + wd * 15.959062)));
+
+            low1 = low1z + g * (0.5 * (src[n] + inz)   - low1z);
+            low2 = low2z + g * (0.5 * (low1   + low1z) - low2z);
+            inz = src[n];
+            dst[n] = ctx->is_disabled ? src[n] : low2;
+        }
+
+        coeffs[0] = low1;
+        coeffs[1] = low2;
+        coeffs[2] = inz;
+    }
+
+    if (out != in)
+        av_frame_free(&in);
+    return ff_filter_frame(outlink, out);
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+    AudioDynamicSmoothContext *s = ctx->priv;
+
+    av_frame_free(&s->coeffs);
+}
+
+#define OFFSET(x) offsetof(AudioDynamicSmoothContext, x)
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
+
+static const AVOption adynamicsmooth_options[] = {
+    { "sensitivity",  "set smooth sensitivity",  OFFSET(sensitivity),  AV_OPT_TYPE_DOUBLE, {.dbl=2},     0, 1000000, FLAGS },
+    { "basefreq",     "set base frequency",      OFFSET(basefreq),     AV_OPT_TYPE_DOUBLE, {.dbl=22050}, 2, 1000000, FLAGS },
+    { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(adynamicsmooth);
+
+static const AVFilterPad inputs[] = {
+    {
+        .name         = "default",
+        .type         = AVMEDIA_TYPE_AUDIO,
+        .filter_frame = filter_frame,
+        .config_props = config_input,
+    },
+};
+
+static const AVFilterPad outputs[] = {
+    {
+        .name = "default",
+        .type = AVMEDIA_TYPE_AUDIO,
+    },
+};
+
+const AVFilter ff_af_adynamicsmooth = {
+    .name            = "adynamicsmooth",
+    .description     = NULL_IF_CONFIG_SMALL("Apply Dynamic Smoothing of input audio."),
+    .priv_size       = sizeof(AudioDynamicSmoothContext),
+    .priv_class      = &adynamicsmooth_class,
+    .uninit          = uninit,
+    FILTER_INPUTS(inputs),
+    FILTER_OUTPUTS(outputs),
+    FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBLP),
+    .flags           = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
+    .process_command = ff_filter_process_command,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 00c36c3f63..c5c0e9b28b 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -37,6 +37,7 @@ extern const AVFilter ff_af_adecorrelate;
 extern const AVFilter ff_af_adelay;
 extern const AVFilter ff_af_adenorm;
 extern const AVFilter ff_af_aderivative;
+extern const AVFilter ff_af_adynamicsmooth;
 extern const AVFilter ff_af_aecho;
 extern const AVFilter ff_af_aemphasis;
 extern const AVFilter ff_af_aeval;