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avfilter: add audio soft clip filter

tags/n4.2
Paul B Mahol 6 years ago
parent
bf15dcc5c8
commit
c2f305ca17
6 changed files with 318 additions and 1 deletions
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  1. +1
    -0
      Changelog
  2. +27
    -0
      doc/filters.texi
  3. +1
    -0
      libavfilter/Makefile
  4. +287
    -0
      libavfilter/af_asoftclip.c
  5. +1
    -0
      libavfilter/allfilters.c
  6. +1
    -1
      libavfilter/version.h

+ 1
- 0
Changelog View File

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


version 4.1:


+ 27
- 0
doc/filters.texi View File

@@ -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



+ 1
- 0
libavfilter/Makefile View File

@@ -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


+ 287
- 0
libavfilter/af_asoftclip.c View File

@@ -0,0 +1,287 @@
/*
* 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,
};

+ 1
- 0
libavfilter/allfilters.c View File

@@ -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;


+ 1
- 1
libavfilter/version.h View File

@@ -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, \


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