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FFmpeg/libavfilter/af_acontrast.c
Anton Khirnov 1afe42852b lavfi/internal: move functions used by filters to filters.h
internal.h currently mixes interfaces intended to be used by filters
with those that should be limited to generic filter- or graph-level
code.
2024-08-19 21:45:25 +02:00

182 lines
4.9 KiB
C

/*
* Copyright (c) 2008 Rob Sykes
* Copyright (c) 2017 Paul B Mahol
*
* 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 "filters.h"
typedef struct AudioContrastContext {
const AVClass *class;
float contrast;
void (*filter)(void **dst, const void **src,
int nb_samples, int channels, float contrast);
} AudioContrastContext;
#define OFFSET(x) offsetof(AudioContrastContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
static const AVOption acontrast_options[] = {
{ "contrast", "set contrast", OFFSET(contrast), AV_OPT_TYPE_FLOAT, {.dbl=33}, 0, 100, A },
{ NULL }
};
AVFILTER_DEFINE_CLASS(acontrast);
static void filter_flt(void **d, const void **s,
int nb_samples, int channels,
float contrast)
{
const float *src = s[0];
float *dst = d[0];
int n, c;
for (n = 0; n < nb_samples; n++) {
for (c = 0; c < channels; c++) {
float d = src[c] * M_PI_2;
dst[c] = sinf(d + contrast * sinf(d * 4));
}
dst += c;
src += c;
}
}
static void filter_dbl(void **d, const void **s,
int nb_samples, int channels,
float contrast)
{
const double *src = s[0];
double *dst = d[0];
int n, c;
for (n = 0; n < nb_samples; n++) {
for (c = 0; c < channels; c++) {
double d = src[c] * M_PI_2;
dst[c] = sin(d + contrast * sin(d * 4));
}
dst += c;
src += c;
}
}
static void filter_fltp(void **d, const void **s,
int nb_samples, int channels,
float contrast)
{
int n, c;
for (c = 0; c < channels; c++) {
const float *src = s[c];
float *dst = d[c];
for (n = 0; n < nb_samples; n++) {
float d = src[n] * M_PI_2;
dst[n] = sinf(d + contrast * sinf(d * 4));
}
}
}
static void filter_dblp(void **d, const void **s,
int nb_samples, int channels,
float contrast)
{
int n, c;
for (c = 0; c < channels; c++) {
const double *src = s[c];
double *dst = d[c];
for (n = 0; n < nb_samples; n++) {
double d = src[n] * M_PI_2;
dst[n] = sin(d + contrast * sin(d * 4));
}
}
}
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
AudioContrastContext *s = ctx->priv;
switch (inlink->format) {
case AV_SAMPLE_FMT_FLT: s->filter = filter_flt; break;
case AV_SAMPLE_FMT_DBL: s->filter = filter_dbl; break;
case AV_SAMPLE_FMT_FLTP: s->filter = filter_fltp; break;
case AV_SAMPLE_FMT_DBLP: s->filter = filter_dblp; break;
}
return 0;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
AudioContrastContext *s = ctx->priv;
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);
}
s->filter((void **)out->extended_data, (const void **)in->extended_data,
in->nb_samples, in->ch_layout.nb_channels, s->contrast / 750);
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,
},
};
const AVFilter ff_af_acontrast = {
.name = "acontrast",
.description = NULL_IF_CONFIG_SMALL("Simple audio dynamic range compression/expansion filter."),
.priv_size = sizeof(AudioContrastContext),
.priv_class = &acontrast_class,
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(ff_audio_default_filterpad),
FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP),
};