mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-12-28 20:53:54 +02:00
371 lines
11 KiB
C
371 lines
11 KiB
C
/*
|
|
* Copyright (C) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen and others
|
|
* Copyright (c) 2015 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
|
|
*/
|
|
|
|
/**
|
|
* @file
|
|
* Lookahead limiter filter
|
|
*/
|
|
|
|
#include "libavutil/avassert.h"
|
|
#include "libavutil/channel_layout.h"
|
|
#include "libavutil/common.h"
|
|
#include "libavutil/opt.h"
|
|
|
|
#include "audio.h"
|
|
#include "avfilter.h"
|
|
#include "formats.h"
|
|
#include "internal.h"
|
|
|
|
typedef struct AudioLimiterContext {
|
|
const AVClass *class;
|
|
|
|
double limit;
|
|
double attack;
|
|
double release;
|
|
double att;
|
|
double level_in;
|
|
double level_out;
|
|
int auto_release;
|
|
int auto_level;
|
|
double asc;
|
|
int asc_c;
|
|
int asc_pos;
|
|
double asc_coeff;
|
|
|
|
double *buffer;
|
|
int buffer_size;
|
|
int pos;
|
|
int *nextpos;
|
|
double *nextdelta;
|
|
|
|
double delta;
|
|
int nextiter;
|
|
int nextlen;
|
|
int asc_changed;
|
|
} AudioLimiterContext;
|
|
|
|
#define OFFSET(x) offsetof(AudioLimiterContext, x)
|
|
#define A AV_OPT_FLAG_AUDIO_PARAM
|
|
#define F AV_OPT_FLAG_FILTERING_PARAM
|
|
|
|
static const AVOption alimiter_options[] = {
|
|
{ "level_in", "set input level", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625, 64, A|F },
|
|
{ "level_out", "set output level", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625, 64, A|F },
|
|
{ "limit", "set limit", OFFSET(limit), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.0625, 1, A|F },
|
|
{ "attack", "set attack", OFFSET(attack), AV_OPT_TYPE_DOUBLE, {.dbl=5}, 0.1, 80, A|F },
|
|
{ "release", "set release", OFFSET(release), AV_OPT_TYPE_DOUBLE, {.dbl=50}, 1, 8000, A|F },
|
|
{ "asc", "enable asc", OFFSET(auto_release), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A|F },
|
|
{ "asc_level", "set asc level", OFFSET(asc_coeff), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 1, A|F },
|
|
{ "level", "auto level", OFFSET(auto_level), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, A|F },
|
|
{ NULL }
|
|
};
|
|
|
|
AVFILTER_DEFINE_CLASS(alimiter);
|
|
|
|
static av_cold int init(AVFilterContext *ctx)
|
|
{
|
|
AudioLimiterContext *s = ctx->priv;
|
|
|
|
s->attack /= 1000.;
|
|
s->release /= 1000.;
|
|
s->att = 1.;
|
|
s->asc_pos = -1;
|
|
s->asc_coeff = pow(0.5, s->asc_coeff - 0.5) * 2 * -1;
|
|
|
|
return 0;
|
|
}
|
|
|
|
static double get_rdelta(AudioLimiterContext *s, double release, int sample_rate,
|
|
double peak, double limit, double patt, int asc)
|
|
{
|
|
double rdelta = (1.0 - patt) / (sample_rate * release);
|
|
|
|
if (asc && s->auto_release && s->asc_c > 0) {
|
|
double a_att = limit / (s->asc_coeff * s->asc) * (double)s->asc_c;
|
|
|
|
if (a_att > patt) {
|
|
double delta = FFMAX((a_att - patt) / (sample_rate * release), rdelta / 10);
|
|
|
|
if (delta < rdelta)
|
|
rdelta = delta;
|
|
}
|
|
}
|
|
|
|
return rdelta;
|
|
}
|
|
|
|
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
|
|
{
|
|
AVFilterContext *ctx = inlink->dst;
|
|
AudioLimiterContext *s = ctx->priv;
|
|
AVFilterLink *outlink = ctx->outputs[0];
|
|
const double *src = (const double *)in->data[0];
|
|
const int channels = inlink->channels;
|
|
const int buffer_size = s->buffer_size;
|
|
double *dst, *buffer = s->buffer;
|
|
const double release = s->release;
|
|
const double limit = s->limit;
|
|
double *nextdelta = s->nextdelta;
|
|
double level = s->auto_level ? 1 / limit : 1;
|
|
const double level_out = s->level_out;
|
|
const double level_in = s->level_in;
|
|
int *nextpos = s->nextpos;
|
|
AVFrame *out;
|
|
double *buf;
|
|
int n, c, i;
|
|
|
|
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);
|
|
}
|
|
dst = (double *)out->data[0];
|
|
|
|
for (n = 0; n < in->nb_samples; n++) {
|
|
double peak = 0;
|
|
|
|
for (c = 0; c < channels; c++) {
|
|
double sample = src[c] * level_in;
|
|
|
|
buffer[s->pos + c] = sample;
|
|
peak = FFMAX(peak, fabs(sample));
|
|
}
|
|
|
|
if (s->auto_release && peak > limit) {
|
|
s->asc += peak;
|
|
s->asc_c++;
|
|
}
|
|
|
|
if (peak > limit) {
|
|
double patt = FFMIN(limit / peak, 1.);
|
|
double rdelta = get_rdelta(s, release, inlink->sample_rate,
|
|
peak, limit, patt, 0);
|
|
double delta = (limit / peak - s->att) / buffer_size * channels;
|
|
int found = 0;
|
|
|
|
if (delta < s->delta) {
|
|
s->delta = delta;
|
|
nextpos[0] = s->pos;
|
|
nextpos[1] = -1;
|
|
nextdelta[0] = rdelta;
|
|
s->nextlen = 1;
|
|
s->nextiter= 0;
|
|
} else {
|
|
for (i = s->nextiter; i < s->nextiter + s->nextlen; i++) {
|
|
int j = i % buffer_size;
|
|
double ppeak, pdelta;
|
|
|
|
ppeak = fabs(buffer[nextpos[j]]) > fabs(buffer[nextpos[j] + 1]) ?
|
|
fabs(buffer[nextpos[j]]) : fabs(buffer[nextpos[j] + 1]);
|
|
pdelta = (limit / peak - limit / ppeak) / (((buffer_size - nextpos[j] + s->pos) % buffer_size) / channels);
|
|
if (pdelta < nextdelta[j]) {
|
|
nextdelta[j] = pdelta;
|
|
found = 1;
|
|
break;
|
|
}
|
|
}
|
|
if (found) {
|
|
s->nextlen = i - s->nextiter + 1;
|
|
nextpos[(s->nextiter + s->nextlen) % buffer_size] = s->pos;
|
|
nextdelta[(s->nextiter + s->nextlen) % buffer_size] = rdelta;
|
|
nextpos[(s->nextiter + s->nextlen + 1) % buffer_size] = -1;
|
|
s->nextlen++;
|
|
}
|
|
}
|
|
}
|
|
|
|
buf = &s->buffer[(s->pos + channels) % buffer_size];
|
|
peak = 0;
|
|
for (c = 0; c < channels; c++) {
|
|
double sample = buf[c];
|
|
|
|
peak = FFMAX(peak, fabs(sample));
|
|
}
|
|
|
|
if (s->pos == s->asc_pos && !s->asc_changed)
|
|
s->asc_pos = -1;
|
|
|
|
if (s->auto_release && s->asc_pos == -1 && peak > limit) {
|
|
s->asc -= peak;
|
|
s->asc_c--;
|
|
}
|
|
|
|
s->att += s->delta;
|
|
|
|
for (c = 0; c < channels; c++)
|
|
dst[c] = buf[c] * s->att;
|
|
|
|
if ((s->pos + channels) % buffer_size == nextpos[s->nextiter]) {
|
|
if (s->auto_release) {
|
|
s->delta = get_rdelta(s, release, inlink->sample_rate,
|
|
peak, limit, s->att, 1);
|
|
if (s->nextlen > 1) {
|
|
int pnextpos = nextpos[(s->nextiter + 1) % buffer_size];
|
|
double ppeak = fabs(buffer[pnextpos]) > fabs(buffer[pnextpos + 1]) ?
|
|
fabs(buffer[pnextpos]) :
|
|
fabs(buffer[pnextpos + 1]);
|
|
double pdelta = (limit / ppeak - s->att) /
|
|
(((buffer_size + pnextpos -
|
|
((s->pos + channels) % buffer_size)) %
|
|
buffer_size) / channels);
|
|
if (pdelta < s->delta)
|
|
s->delta = pdelta;
|
|
}
|
|
} else {
|
|
s->delta = nextdelta[s->nextiter];
|
|
s->att = limit / peak;
|
|
}
|
|
|
|
s->nextlen -= 1;
|
|
nextpos[s->nextiter] = -1;
|
|
s->nextiter = (s->nextiter + 1) % buffer_size;
|
|
}
|
|
|
|
if (s->att > 1.) {
|
|
s->att = 1.;
|
|
s->delta = 0.;
|
|
s->nextiter = 0;
|
|
s->nextlen = 0;
|
|
nextpos[0] = -1;
|
|
}
|
|
|
|
if (s->att <= 0.) {
|
|
s->att = 0.0000000000001;
|
|
s->delta = (1.0 - s->att) / (inlink->sample_rate * release);
|
|
}
|
|
|
|
if (s->att != 1. && (1. - s->att) < 0.0000000000001)
|
|
s->att = 1.;
|
|
|
|
if (s->delta != 0. && fabs(s->delta) < 0.00000000000001)
|
|
s->delta = 0.;
|
|
|
|
for (c = 0; c < channels; c++)
|
|
dst[c] = av_clipd(dst[c], -limit, limit) * level * level_out;
|
|
|
|
s->pos = (s->pos + channels) % buffer_size;
|
|
src += channels;
|
|
dst += channels;
|
|
}
|
|
|
|
if (in != out)
|
|
av_frame_free(&in);
|
|
|
|
return ff_filter_frame(outlink, out);
|
|
}
|
|
|
|
static int query_formats(AVFilterContext *ctx)
|
|
{
|
|
AVFilterFormats *formats;
|
|
AVFilterChannelLayouts *layouts;
|
|
static const enum AVSampleFormat sample_fmts[] = {
|
|
AV_SAMPLE_FMT_DBL,
|
|
AV_SAMPLE_FMT_NONE
|
|
};
|
|
int 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_make_format_list(sample_fmts);
|
|
if (!formats)
|
|
return AVERROR(ENOMEM);
|
|
ret = ff_set_common_formats(ctx, formats);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
formats = ff_all_samplerates();
|
|
if (!formats)
|
|
return AVERROR(ENOMEM);
|
|
return ff_set_common_samplerates(ctx, formats);
|
|
}
|
|
|
|
static int config_input(AVFilterLink *inlink)
|
|
{
|
|
AVFilterContext *ctx = inlink->dst;
|
|
AudioLimiterContext *s = ctx->priv;
|
|
int obuffer_size;
|
|
|
|
obuffer_size = inlink->sample_rate * inlink->channels * 100 / 1000. + inlink->channels;
|
|
if (obuffer_size < inlink->channels)
|
|
return AVERROR(EINVAL);
|
|
|
|
s->buffer = av_calloc(obuffer_size, sizeof(*s->buffer));
|
|
s->nextdelta = av_calloc(obuffer_size, sizeof(*s->nextdelta));
|
|
s->nextpos = av_malloc_array(obuffer_size, sizeof(*s->nextpos));
|
|
if (!s->buffer || !s->nextdelta || !s->nextpos)
|
|
return AVERROR(ENOMEM);
|
|
|
|
memset(s->nextpos, -1, obuffer_size * sizeof(*s->nextpos));
|
|
s->buffer_size = inlink->sample_rate * s->attack * inlink->channels;
|
|
s->buffer_size -= s->buffer_size % inlink->channels;
|
|
|
|
return 0;
|
|
}
|
|
|
|
static av_cold void uninit(AVFilterContext *ctx)
|
|
{
|
|
AudioLimiterContext *s = ctx->priv;
|
|
|
|
av_freep(&s->buffer);
|
|
av_freep(&s->nextdelta);
|
|
av_freep(&s->nextpos);
|
|
}
|
|
|
|
static const AVFilterPad alimiter_inputs[] = {
|
|
{
|
|
.name = "main",
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.filter_frame = filter_frame,
|
|
.config_props = config_input,
|
|
},
|
|
{ NULL }
|
|
};
|
|
|
|
static const AVFilterPad alimiter_outputs[] = {
|
|
{
|
|
.name = "default",
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
},
|
|
{ NULL }
|
|
};
|
|
|
|
AVFilter ff_af_alimiter = {
|
|
.name = "alimiter",
|
|
.description = NULL_IF_CONFIG_SMALL("Audio lookahead limiter."),
|
|
.priv_size = sizeof(AudioLimiterContext),
|
|
.priv_class = &alimiter_class,
|
|
.init = init,
|
|
.uninit = uninit,
|
|
.query_formats = query_formats,
|
|
.inputs = alimiter_inputs,
|
|
.outputs = alimiter_outputs,
|
|
};
|