avfilter: add audio soft clip filter

Changelog

@@ -25,6 +25,7 @@ version <next>:
 - AV1 frame split bitstream filter
 - lscr decoder
 - lagfun filter
+- asoftclip filter
 
 
 version 4.1:

doc/filters.texi

@@ -2104,6 +2104,33 @@ audio, the data is treated as if all the planes were concatenated.
 A list of Adler-32 checksums for each data plane.
 @end table
 
+@section asoftclip
+Apply audio soft clipping.
+
+Soft clipping is a type of distortion effect where the amplitude of a signal is saturated
+along a smooth curve, rather than the abrupt shape of hard-clipping.
+
+This filter accepts the following options:
+
+@table @option
+@item type
+Set type of soft-clipping.
+
+It accepts the following values:
+@table @option
+@item tanh
+@item atan
+@item cubic
+@item exp
+@item alg
+@item quintic
+@item sin
+@end table
+
+@item param
+Set additional parameter which controls sigmoid function.
+@end table
+
 @anchor{astats}
 @section astats
 

libavfilter/Makefile

@@ -80,6 +80,7 @@ OBJS-$(CONFIG_ASETRATE_FILTER)               += af_asetrate.o
 OBJS-$(CONFIG_ASETTB_FILTER)                 += settb.o
 OBJS-$(CONFIG_ASHOWINFO_FILTER)              += af_ashowinfo.o
 OBJS-$(CONFIG_ASIDEDATA_FILTER)              += f_sidedata.o
+OBJS-$(CONFIG_ASOFTCLIP_FILTER)              += af_asoftclip.o
 OBJS-$(CONFIG_ASPLIT_FILTER)                 += split.o
 OBJS-$(CONFIG_ASTATS_FILTER)                 += af_astats.o
 OBJS-$(CONFIG_ASTREAMSELECT_FILTER)          += f_streamselect.o framesync.o

libavfilter/af_asoftclip.c

+/*
+ * Copyright (c) 2019 The FFmpeg Project
+ *
+ * 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"
+
+enum ASoftClipTypes {
+    ASC_TANH,
+    ASC_ATAN,
+    ASC_CUBIC,
+    ASC_EXP,
+    ASC_ALG,
+    ASC_QUINTIC,
+    ASC_SIN,
+    NB_TYPES,
+};
+
+typedef struct ASoftClipContext {
+    const AVClass *class;
+
+    int type;
+    double param;
+
+    void (*filter)(struct ASoftClipContext *s, void **dst, const void **src,
+                   int nb_samples, int channels);
+} ASoftClipContext;
+
+#define OFFSET(x) offsetof(ASoftClipContext, x)
+#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption asoftclip_options[] = {
+    { "type", "set softclip type", OFFSET(type), AV_OPT_TYPE_INT,    {.i64=0},          0, NB_TYPES-1, A, "types" },
+    { "tanh",                NULL,            0, AV_OPT_TYPE_CONST,  {.i64=ASC_TANH},   0,          0, A, "types" },
+    { "atan",                NULL,            0, AV_OPT_TYPE_CONST,  {.i64=ASC_ATAN},   0,          0, A, "types" },
+    { "cubic",               NULL,            0, AV_OPT_TYPE_CONST,  {.i64=ASC_CUBIC},  0,          0, A, "types" },
+    { "exp",                 NULL,            0, AV_OPT_TYPE_CONST,  {.i64=ASC_EXP},    0,          0, A, "types" },
+    { "alg",                 NULL,            0, AV_OPT_TYPE_CONST,  {.i64=ASC_ALG},    0,          0, A, "types" },
+    { "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" },
+    { "param", "set softclip parameter", OFFSET(param), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.01,        3, A },
+    { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(asoftclip);
+
+static int query_formats(AVFilterContext *ctx)
+{
+    AVFilterFormats *formats = NULL;
+    AVFilterChannelLayouts *layouts = NULL;
+    static const enum AVSampleFormat sample_fmts[] = {
+        AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
+        AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
+        AV_SAMPLE_FMT_NONE
+    };
+    int ret;
+
+    formats = ff_make_format_list(sample_fmts);
+    if (!formats)
+        return AVERROR(ENOMEM);
+    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();
+    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)
+{
+    float param = s->param;
+
+    for (int c = 0; c < channels; c++) {
+        const float *src = sptr[c];
+        float *dst = dptr[c];
+
+        switch (s->type) {
+        case ASC_TANH:
+            for (int n = 0; n < nb_samples; n++) {
+                dst[n] = tanhf(src[n] * param);
+            }
+            break;
+        case ASC_ATAN:
+            for (int n = 0; n < nb_samples; n++)
+                dst[n] = 2.f / M_PI * atanf(src[n] * param);
+            break;
+        case ASC_CUBIC:
+            for (int n = 0; n < nb_samples; n++) {
+                if (FFABS(src[n]) >= 1.5f)
+                    dst[n] = FFSIGN(src[n]);
+                else
+                    dst[n] = src[n] - 0.1481f * powf(src[n], 3.f);
+            }
+            break;
+        case ASC_EXP:
+            for (int n = 0; n < nb_samples; n++)
+                dst[n] = 2.f / (1.f + expf(-2.f * src[n])) - 1.;
+            break;
+        case ASC_ALG:
+            for (int n = 0; n < nb_samples; n++)
+                dst[n] = src[n] / (sqrtf(param + src[n] * src[n]));
+            break;
+        case ASC_QUINTIC:
+            for (int n = 0; n < nb_samples; n++) {
+                if (FFABS(src[n]) >= 1.25)
+                    dst[n] = FFSIGN(src[n]);
+                else
+                    dst[n] = src[n] - 0.08192f * powf(src[n], 5.f);
+            }
+            break;
+        case ASC_SIN:
+            for (int n = 0; n < nb_samples; n++) {
+                if (FFABS(src[n]) >= M_PI_2)
+                    dst[n] = FFSIGN(src[n]);
+                else
+                    dst[n] = sinf(src[n]);
+            }
+            break;
+        }
+    }
+}
+
+static void filter_dbl(ASoftClipContext *s,
+                       void **dptr, const void **sptr,
+                       int nb_samples, int channels)
+{
+    double param = s->param;
+
+    for (int c = 0; c < channels; c++) {
+        const double *src = sptr[c];
+        double *dst = dptr[c];
+
+        switch (s->type) {
+        case ASC_TANH:
+            for (int n = 0; n < nb_samples; n++) {
+                dst[n] = tanh(src[n] * param);
+            }
+            break;
+        case ASC_ATAN:
+            for (int n = 0; n < nb_samples; n++)
+                dst[n] = 2. / M_PI * atan(src[n] * param);
+            break;
+        case ASC_CUBIC:
+            for (int n = 0; n < nb_samples; n++) {
+                if (FFABS(src[n]) >= 1.5)
+                    dst[n] = FFSIGN(src[n]);
+                else
+                    dst[n] = src[n] - 0.1481 * pow(src[n], 3.);
+            }
+            break;
+        case ASC_EXP:
+            for (int n = 0; n < nb_samples; n++)
+                dst[n] = 2. / (1. + exp(-2. * src[n])) - 1.;
+            break;
+        case ASC_ALG:
+            for (int n = 0; n < nb_samples; n++)
+                dst[n] = src[n] / (sqrt(param + src[n] * src[n]));
+            break;
+        case ASC_QUINTIC:
+            for (int n = 0; n < nb_samples; n++) {
+                if (FFABS(src[n]) >= 1.25)
+                    dst[n] = FFSIGN(src[n]);
+                else
+                    dst[n] = src[n] - 0.08192 * pow(src[n], 5.);
+            }
+            break;
+        case ASC_SIN:
+            for (int n = 0; n < nb_samples; n++) {
+                if (FFABS(src[n]) >= M_PI_2)
+                    dst[n] = FFSIGN(src[n]);
+                else
+                    dst[n] = sin(src[n]);
+            }
+            break;
+        }
+    }
+}
+
+static int config_input(AVFilterLink *inlink)
+{
+    AVFilterContext *ctx = inlink->dst;
+    ASoftClipContext *s = ctx->priv;
+
+    switch (inlink->format) {
+    case AV_SAMPLE_FMT_FLT:
+    case AV_SAMPLE_FMT_FLTP: s->filter = filter_flt; break;
+    case AV_SAMPLE_FMT_DBL:
+    case AV_SAMPLE_FMT_DBLP: s->filter = filter_dbl; break;
+    }
+
+    return 0;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+    AVFilterContext *ctx = inlink->dst;
+    AVFilterLink *outlink = ctx->outputs[0];
+    ASoftClipContext *s = ctx->priv;
+    int nb_samples, channels;
+    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);
+    }
+
+    if (av_sample_fmt_is_planar(in->format)) {
+        nb_samples = in->nb_samples;
+        channels = in->channels;
+    } else {
+        nb_samples = in->channels * in->nb_samples;
+        channels = 1;
+    }
+
+    s->filter(s, (void **)out->extended_data, (const void **)in->extended_data,
+              nb_samples, channels);
+
+    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_asoftclip = {
+    .name           = "asoftclip",
+    .description    = NULL_IF_CONFIG_SMALL("Audio Soft Clipper."),
+    .query_formats  = query_formats,
+    .priv_size      = sizeof(ASoftClipContext),
+    .priv_class     = &asoftclip_class,
+    .inputs         = inputs,
+    .outputs        = outputs,
+};

libavfilter/allfilters.c

@@ -72,6 +72,7 @@ extern AVFilter ff_af_asetrate;
 extern AVFilter ff_af_asettb;
 extern AVFilter ff_af_ashowinfo;
 extern AVFilter ff_af_asidedata;
+extern AVFilter ff_af_asoftclip;
 extern AVFilter ff_af_asplit;
 extern AVFilter ff_af_astats;
 extern AVFilter ff_af_astreamselect;

libavfilter/version.h

@@ -30,7 +30,7 @@
 #include "libavutil/version.h"
 
 #define LIBAVFILTER_VERSION_MAJOR   7
-#define LIBAVFILTER_VERSION_MINOR  49
+#define LIBAVFILTER_VERSION_MINOR  50
 #define LIBAVFILTER_VERSION_MICRO 100
 
 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \