Signed-off-by: Paul B Mahol <one...@gmail.com>
---
With more sample formats support.
---
doc/filters.texi | 27 ++++
libavfilter/Makefile | 1 +
libavfilter/af_asoftclip.c | 289 +++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
4 files changed, 318 insertions(+)
create mode 100644 libavfilter/af_asoftclip.c
diff --git a/doc/filters.texi b/doc/filters.texi
index 4dd1a5de85..465eeb4732 100644
--- a/doc/filters.texi
+++ b/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
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index fef6ec5c55..682df45ef5 100644
--- a/libavfilter/Makefile
+++ b/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
diff --git a/libavfilter/af_asoftclip.c b/libavfilter/af_asoftclip.c
new file mode 100644
index 0000000000..1966613092
--- /dev/null
+++ b/libavfilter/af_asoftclip.c
@@ -0,0 +1,289 @@
+/*
+ * 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;
+ int n, c;
+
+ for (c = 0; c < channels; c++) {
+ const float *src = sptr[c];
+ float *dst = dptr[c];
+
+ switch (s->type) {
+ case ASC_TANH:
+ for (n = 0; n < nb_samples; n++) {
+ dst[n] = tanhf(src[n] * param);
+ }
+ break;
+ case ASC_ATAN:
+ for (n = 0; n < nb_samples; n++)
+ dst[n] = 2.f / M_PI * atanf(src[n] * param);
+ break;
+ case ASC_CUBIC:
+ for (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 (n = 0; n < nb_samples; n++)
+ dst[n] = 2.f / (1.f + expf(-2.f * src[n])) - 1.;
+ break;
+ case ASC_ALG:
+ for (n = 0; n < nb_samples; n++)
+ dst[n] = src[n] / (sqrtf(param + src[n] * src[n]));
+ break;
+ case ASC_QUINTIC:
+ for (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 (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;
+ int n, c;
+
+ for (c = 0; c < channels; c++) {
+ const double *src = sptr[c];
+ double *dst = dptr[c];
+
+ switch (s->type) {
+ case ASC_TANH:
+ for (n = 0; n < nb_samples; n++) {
+ dst[n] = tanh(src[n] * param);
+ }
+ break;
+ case ASC_ATAN:
+ for (n = 0; n < nb_samples; n++)
+ dst[n] = 2. / M_PI * atan(src[n] * param);
+ break;
+ case ASC_CUBIC:
+ for (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 (n = 0; n < nb_samples; n++)
+ dst[n] = 2. / (1. + exp(-2. * src[n])) - 1.;
+ break;
+ case ASC_ALG:
+ for (n = 0; n < nb_samples; n++)
+ dst[n] = src[n] / (sqrt(param + src[n] * src[n]));
+ break;
+ case ASC_QUINTIC:
+ for (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 (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,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index c51ae0f3c7..4d3039d6ba 100644
--- a/libavfilter/allfilters.c
+++ b/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;
--
2.17.1
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