1
0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-07 11:13:41 +02:00
FFmpeg/libavfilter/af_deesser.c
Andreas Rheinhardt 790f793844 avutil/common: Don't auto-include mem.h
There are lots of files that don't need it: The number of object
files that actually need it went down from 2011 to 884 here.

Keep it for external users in order to not cause breakages.

Also improve the other headers a bit while just at it.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2024-03-31 00:08:43 +01:00

207 lines
7.0 KiB
C

/*
* Copyright (c) 2018 Chris Johnson
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in all
* copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
* SOFTWARE.
*/
#include "libavutil/channel_layout.h"
#include "libavutil/mem.h"
#include "libavutil/opt.h"
#include "avfilter.h"
#include "audio.h"
typedef struct DeesserChannel {
double s1, s2, s3;
double m1, m2;
double ratioA, ratioB;
double iirSampleA, iirSampleB;
int flip;
} DeesserChannel;
typedef struct DeesserContext {
const AVClass *class;
double intensity;
double max;
double frequency;
int mode;
DeesserChannel *chan;
} DeesserContext;
enum OutModes {
IN_MODE,
OUT_MODE,
ESS_MODE,
NB_MODES
};
#define OFFSET(x) offsetof(DeesserContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
static const AVOption deesser_options[] = {
{ "i", "set intensity", OFFSET(intensity), AV_OPT_TYPE_DOUBLE, {.dbl=0.0}, 0.0, 1.0, A },
{ "m", "set max deessing", OFFSET(max), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0.0, 1.0, A },
{ "f", "set frequency", OFFSET(frequency), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0.0, 1.0, A },
{ "s", "set output mode", OFFSET(mode), AV_OPT_TYPE_INT, {.i64=OUT_MODE}, 0, NB_MODES-1, A, .unit = "mode" },
{ "i", "input", 0, AV_OPT_TYPE_CONST, {.i64=IN_MODE}, 0, 0, A, .unit = "mode" },
{ "o", "output", 0, AV_OPT_TYPE_CONST, {.i64=OUT_MODE}, 0, 0, A, .unit = "mode" },
{ "e", "ess", 0, AV_OPT_TYPE_CONST, {.i64=ESS_MODE}, 0, 0, A, .unit = "mode" },
{ NULL }
};
AVFILTER_DEFINE_CLASS(deesser);
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
DeesserContext *s = ctx->priv;
s->chan = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->chan));
if (!s->chan)
return AVERROR(ENOMEM);
for (int i = 0; i < inlink->ch_layout.nb_channels; i++) {
DeesserChannel *chan = &s->chan[i];
chan->ratioA = chan->ratioB = 1.0;
}
return 0;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
DeesserContext *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);
}
for (int ch = 0; ch < inlink->ch_layout.nb_channels; ch++) {
DeesserChannel *dec = &s->chan[ch];
double *src = (double *)in->extended_data[ch];
double *dst = (double *)out->extended_data[ch];
double overallscale = inlink->sample_rate < 44100 ? 44100.0 / inlink->sample_rate : inlink->sample_rate / 44100.0;
double intensity = pow(s->intensity, 5) * (8192 / overallscale);
double maxdess = 1.0 / pow(10.0, ((s->max - 1.0) * 48.0) / 20);
double iirAmount = pow(s->frequency, 2) / overallscale;
double offset;
double sense;
double recovery;
double attackspeed;
for (int i = 0; i < in->nb_samples; i++) {
double sample = src[i];
dec->s3 = dec->s2;
dec->s2 = dec->s1;
dec->s1 = sample;
dec->m1 = (dec->s1 - dec->s2) * ((dec->s1 - dec->s2) / 1.3);
dec->m2 = (dec->s2 - dec->s3) * ((dec->s1 - dec->s2) / 1.3);
sense = (dec->m1 - dec->m2) * ((dec->m1 - dec->m2) / 1.3);
attackspeed = 7.0 + sense * 1024;
sense = 1.0 + intensity * intensity * sense;
sense = FFMIN(sense, intensity);
recovery = 1.0 + (0.01 / sense);
offset = 1.0 - fabs(sample);
if (dec->flip) {
dec->iirSampleA = (dec->iirSampleA * (1.0 - (offset * iirAmount))) +
(sample * (offset * iirAmount));
if (dec->ratioA < sense) {
dec->ratioA = ((dec->ratioA * attackspeed) + sense) / (attackspeed + 1.0);
} else {
dec->ratioA = 1.0 + ((dec->ratioA - 1.0) / recovery);
}
dec->ratioA = FFMIN(dec->ratioA, maxdess);
sample = dec->iirSampleA + ((sample - dec->iirSampleA) / dec->ratioA);
} else {
dec->iirSampleB = (dec->iirSampleB * (1.0 - (offset * iirAmount))) +
(sample * (offset * iirAmount));
if (dec->ratioB < sense) {
dec->ratioB = ((dec->ratioB * attackspeed) + sense) / (attackspeed + 1.0);
} else {
dec->ratioB = 1.0 + ((dec->ratioB - 1.0) / recovery);
}
dec->ratioB = FFMIN(dec->ratioB, maxdess);
sample = dec->iirSampleB + ((sample - dec->iirSampleB) / dec->ratioB);
}
dec->flip = !dec->flip;
if (ctx->is_disabled)
sample = src[i];
switch (s->mode) {
case IN_MODE: dst[i] = src[i]; break;
case OUT_MODE: dst[i] = sample; break;
case ESS_MODE: dst[i] = src[i] - sample; break;
}
}
}
if (out != in)
av_frame_free(&in);
return ff_filter_frame(outlink, out);
}
static av_cold void uninit(AVFilterContext *ctx)
{
DeesserContext *s = ctx->priv;
av_freep(&s->chan);
}
static const AVFilterPad inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
.config_props = config_input,
},
};
const AVFilter ff_af_deesser = {
.name = "deesser",
.description = NULL_IF_CONFIG_SMALL("Apply de-essing to the audio."),
.priv_size = sizeof(DeesserContext),
.priv_class = &deesser_class,
.uninit = uninit,
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(ff_audio_default_filterpad),
FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBLP),
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
};