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FFmpeg/libavfilter/af_asoftclip.c

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/*
* 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/avassert.h"
#include "libavutil/channel_layout.h"
#include "libavutil/opt.h"
#include "libswresample/swresample.h"
#include "avfilter.h"
#include "audio.h"
#include "formats.h"
enum ASoftClipTypes {
ASC_HARD = -1,
ASC_TANH,
ASC_ATAN,
ASC_CUBIC,
ASC_EXP,
ASC_ALG,
ASC_QUINTIC,
ASC_SIN,
ASC_ERF,
NB_TYPES,
};
typedef struct ASoftClipContext {
const AVClass *class;
int type;
int oversample;
int64_t delay;
double threshold;
double output;
double param;
SwrContext *up_ctx;
SwrContext *down_ctx;
AVFrame *frame;
void (*filter)(struct ASoftClipContext *s, void **dst, const void **src,
int nb_samples, int channels, int start, int end);
} ASoftClipContext;
#define OFFSET(x) offsetof(ASoftClipContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
#define F 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}, -1, NB_TYPES-1, A, "types" },
{ "hard", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_HARD}, 0, 0, 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" },
{ "erf", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_ERF}, 0, 0, A, "types" },
{ "threshold", "set softclip threshold", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.000001, 1, A },
{ "output", "set softclip output gain", OFFSET(output), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.000001, 16, A },
{ "param", "set softclip parameter", OFFSET(param), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.01, 3, A },
{ "oversample", "set oversample factor", OFFSET(oversample), AV_OPT_TYPE_INT, {.i64=1}, 1, 32, F },
{ 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);
}
static void filter_flt(ASoftClipContext *s,
void **dptr, const void **sptr,
int nb_samples, int channels,
int start, int end)
{
float threshold = s->threshold;
float gain = s->output * threshold;
float factor = 1.f / threshold;
float param = s->param;
for (int c = start; c < end; c++) {
const float *src = sptr[c];
float *dst = dptr[c];
switch (s->type) {
case ASC_HARD:
for (int n = 0; n < nb_samples; n++) {
dst[n] = av_clipf(src[n] * factor, -1.f, 1.f);
dst[n] *= gain;
}
break;
case ASC_TANH:
for (int n = 0; n < nb_samples; n++) {
dst[n] = tanhf(src[n] * factor * param);
dst[n] *= gain;
}
break;
case ASC_ATAN:
for (int n = 0; n < nb_samples; n++) {
dst[n] = 2.f / M_PI * atanf(src[n] * factor * param);
dst[n] *= gain;
}
break;
case ASC_CUBIC:
for (int n = 0; n < nb_samples; n++) {
float sample = src[n] * factor;
if (FFABS(sample) >= 1.5f)
dst[n] = FFSIGN(sample);
else
dst[n] = sample - 0.1481f * powf(sample, 3.f);
dst[n] *= gain;
}
break;
case ASC_EXP:
for (int n = 0; n < nb_samples; n++) {
dst[n] = 2.f / (1.f + expf(-2.f * src[n] * factor)) - 1.;
dst[n] *= gain;
}
break;
case ASC_ALG:
for (int n = 0; n < nb_samples; n++) {
float sample = src[n] * factor;
dst[n] = sample / (sqrtf(param + sample * sample));
dst[n] *= gain;
}
break;
case ASC_QUINTIC:
for (int n = 0; n < nb_samples; n++) {
float sample = src[n] * factor;
if (FFABS(sample) >= 1.25)
dst[n] = FFSIGN(sample);
else
dst[n] = sample - 0.08192f * powf(sample, 5.f);
dst[n] *= gain;
}
break;
case ASC_SIN:
for (int n = 0; n < nb_samples; n++) {
float sample = src[n] * factor;
if (FFABS(sample) >= M_PI_2)
dst[n] = FFSIGN(sample);
else
dst[n] = sinf(sample);
dst[n] *= gain;
}
break;
case ASC_ERF:
for (int n = 0; n < nb_samples; n++) {
dst[n] = erff(src[n] * factor);
dst[n] *= gain;
}
break;
default:
av_assert0(0);
}
}
}
static void filter_dbl(ASoftClipContext *s,
void **dptr, const void **sptr,
int nb_samples, int channels,
int start, int end)
{
double threshold = s->threshold;
double gain = s->output * threshold;
double factor = 1. / threshold;
double param = s->param;
for (int c = start; c < end; c++) {
const double *src = sptr[c];
double *dst = dptr[c];
switch (s->type) {
case ASC_HARD:
for (int n = 0; n < nb_samples; n++) {
dst[n] = av_clipd(src[n] * factor, -1., 1.);
dst[n] *= gain;
}
break;
case ASC_TANH:
for (int n = 0; n < nb_samples; n++) {
dst[n] = tanh(src[n] * factor * param);
dst[n] *= gain;
}
break;
case ASC_ATAN:
for (int n = 0; n < nb_samples; n++) {
dst[n] = 2. / M_PI * atan(src[n] * factor * param);
dst[n] *= gain;
}
break;
case ASC_CUBIC:
for (int n = 0; n < nb_samples; n++) {
double sample = src[n] * factor;
if (FFABS(sample) >= 1.5)
dst[n] = FFSIGN(sample);
else
dst[n] = sample - 0.1481 * pow(sample, 3.);
dst[n] *= gain;
}
break;
case ASC_EXP:
for (int n = 0; n < nb_samples; n++) {
dst[n] = 2. / (1. + exp(-2. * src[n] * factor)) - 1.;
dst[n] *= gain;
}
break;
case ASC_ALG:
for (int n = 0; n < nb_samples; n++) {
double sample = src[n] * factor;
dst[n] = sample / (sqrt(param + sample * sample));
dst[n] *= gain;
}
break;
case ASC_QUINTIC:
for (int n = 0; n < nb_samples; n++) {
double sample = src[n] * factor;
if (FFABS(sample) >= 1.25)
dst[n] = FFSIGN(sample);
else
dst[n] = sample - 0.08192 * pow(sample, 5.);
dst[n] *= gain;
}
break;
case ASC_SIN:
for (int n = 0; n < nb_samples; n++) {
double sample = src[n] * factor;
if (FFABS(sample) >= M_PI_2)
dst[n] = FFSIGN(sample);
else
dst[n] = sin(sample);
dst[n] *= gain;
}
break;
case ASC_ERF:
for (int n = 0; n < nb_samples; n++) {
dst[n] = erf(src[n] * factor);
dst[n] *= gain;
}
break;
default:
av_assert0(0);
}
}
}
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
ASoftClipContext *s = ctx->priv;
int ret;
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;
default: av_assert0(0);
}
if (s->oversample <= 1)
return 0;
s->up_ctx = swr_alloc();
s->down_ctx = swr_alloc();
if (!s->up_ctx || !s->down_ctx)
return AVERROR(ENOMEM);
av_opt_set_int(s->up_ctx, "in_channel_layout", inlink->channel_layout, 0);
av_opt_set_int(s->up_ctx, "in_sample_rate", inlink->sample_rate, 0);
av_opt_set_sample_fmt(s->up_ctx, "in_sample_fmt", inlink->format, 0);
av_opt_set_int(s->up_ctx, "out_channel_layout", inlink->channel_layout, 0);
av_opt_set_int(s->up_ctx, "out_sample_rate", inlink->sample_rate * s->oversample, 0);
av_opt_set_sample_fmt(s->up_ctx, "out_sample_fmt", inlink->format, 0);
av_opt_set_int(s->down_ctx, "in_channel_layout", inlink->channel_layout, 0);
av_opt_set_int(s->down_ctx, "in_sample_rate", inlink->sample_rate * s->oversample, 0);
av_opt_set_sample_fmt(s->down_ctx, "in_sample_fmt", inlink->format, 0);
av_opt_set_int(s->down_ctx, "out_channel_layout", inlink->channel_layout, 0);
av_opt_set_int(s->down_ctx, "out_sample_rate", inlink->sample_rate, 0);
av_opt_set_sample_fmt(s->down_ctx, "out_sample_fmt", inlink->format, 0);
ret = swr_init(s->up_ctx);
if (ret < 0)
return ret;
ret = swr_init(s->down_ctx);
if (ret < 0)
return ret;
return 0;
}
typedef struct ThreadData {
AVFrame *in, *out;
int nb_samples;
int channels;
} ThreadData;
static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
{
ASoftClipContext *s = ctx->priv;
ThreadData *td = arg;
AVFrame *out = td->out;
AVFrame *in = td->in;
const int channels = td->channels;
const int nb_samples = td->nb_samples;
const int start = (channels * jobnr) / nb_jobs;
const int end = (channels * (jobnr+1)) / nb_jobs;
s->filter(s, (void **)out->extended_data, (const void **)in->extended_data,
nb_samples, channels, start, end);
return 0;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
ASoftClipContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
int ret, nb_samples, channels;
ThreadData td;
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;
}
if (s->oversample > 1) {
s->frame = ff_get_audio_buffer(outlink, in->nb_samples * s->oversample);
if (!s->frame) {
ret = AVERROR(ENOMEM);
goto fail;
}
ret = swr_convert(s->up_ctx, (uint8_t**)s->frame->extended_data, in->nb_samples * s->oversample,
(const uint8_t **)in->extended_data, in->nb_samples);
if (ret < 0)
goto fail;
td.in = s->frame;
td.out = s->frame;
td.nb_samples = av_sample_fmt_is_planar(in->format) ? ret : ret * in->channels;
td.channels = channels;
ctx->internal->execute(ctx, filter_channels, &td, NULL, FFMIN(channels,
ff_filter_get_nb_threads(ctx)));
ret = swr_convert(s->down_ctx, (uint8_t**)out->extended_data, out->nb_samples,
(const uint8_t **)s->frame->extended_data, ret);
if (ret < 0)
goto fail;
if (out->pts)
out->pts -= s->delay;
s->delay += in->nb_samples - ret;
out->nb_samples = ret;
av_frame_free(&s->frame);
} else {
td.in = in;
td.out = out;
td.nb_samples = nb_samples;
td.channels = channels;
ctx->internal->execute(ctx, filter_channels, &td, NULL, FFMIN(channels,
ff_filter_get_nb_threads(ctx)));
}
if (out != in)
av_frame_free(&in);
return ff_filter_frame(outlink, out);
fail:
if (out != in)
av_frame_free(&out);
av_frame_free(&in);
av_frame_free(&s->frame);
return ret;
}
static av_cold void uninit(AVFilterContext *ctx)
{
ASoftClipContext *s = ctx->priv;
swr_free(&s->up_ctx);
swr_free(&s->down_ctx);
}
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,
.uninit = uninit,
.process_command = ff_filter_process_command,
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC |
AVFILTER_FLAG_SLICE_THREADS,
};
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