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avfilter: add audio limiter filter

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Signed-off-by: Paul B Mahol <onemda@gmail.com>
This commit is contained in:
Paul B Mahol 2015-09-05 19:12:58 +00:00
parent a0a2ca024b
commit 39c61d8459
6 changed files with 400 additions and 1 deletions
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1
Changelog
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@ -5,6 +5,7 @@ version <next>:
- DXV decoding
- extrastereo filter
- ocr filter
- alimiter filter
version 2.8:

35
doc/filters.texi
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@ -641,6 +641,41 @@ Force the output to either unsigned 8-bit or signed 16-bit stereo
aformat=sample_fmts=u8|s16:channel_layouts=stereo
@end example
@section alimiter
The limiter prevents input signal from raising over a desired threshold.
This limiter uses lookahead technology to prevent your signal from distorting.
It means that there is a small delay after signal is processed. Keep in mind
that the delay it produces is the attack time you set.
The filter accepts the following options:
@table @option
@item limit
Don't let signals above this level pass the limiter. The removed amplitude is
added automatically. Default is 1.
@item attack
The limiter will reach its attenuation level in this amount of time in
milliseconds. Default is 5 milliseconds.
@item release
Come back from limiting to attenuation 1.0 in this amount of milliseconds.
Default is 50 milliseconds.
@item asc
When gain reduction is always needed ASC takes care of releasing to an
average reduction level rather than reaching a reduction of 0 in the release
time.
@item asc_level
Select how much the release time is affected by ASC, 0 means nearly no changes
in release time while 1 produces higher release times.
@end table
Depending on picked setting it is recommended to upsample input 2x or 4x times
with @ref{aresample} before applying this filter.
@section allpass
Apply a two-pole all-pass filter with central frequency (in Hz)

1
libavfilter/Makefile
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@ -30,6 +30,7 @@ OBJS-$(CONFIG_AEVAL_FILTER) += aeval.o
OBJS-$(CONFIG_AFADE_FILTER) += af_afade.o
OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o
OBJS-$(CONFIG_AINTERLEAVE_FILTER) += f_interleave.o
OBJS-$(CONFIG_ALIMITER_FILTER) += af_alimiter.o
OBJS-$(CONFIG_ALLPASS_FILTER) += af_biquads.o
OBJS-$(CONFIG_AMERGE_FILTER) += af_amerge.o
OBJS-$(CONFIG_AMIX_FILTER) += af_amix.o

361
libavfilter/af_alimiter.c Normal file
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@ -0,0 +1,361 @@
/*
* 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;
int auto_release;
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[] = {
{ "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 },
{ 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;
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(inlink, 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];
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);
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("Lookahead limiter."),
.priv_size = sizeof(AudioLimiterContext),
.priv_class = &alimiter_class,
.init = init,
.uninit = uninit,
.query_formats = query_formats,
.inputs = alimiter_inputs,
.outputs = alimiter_outputs,
};

1
libavfilter/allfilters.c
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@ -52,6 +52,7 @@ void avfilter_register_all(void)
REGISTER_FILTER(AFADE, afade, af);
REGISTER_FILTER(AFORMAT, aformat, af);
REGISTER_FILTER(AINTERLEAVE, ainterleave, af);
REGISTER_FILTER(ALIMITER, alimiter, af);
REGISTER_FILTER(ALLPASS, allpass, af);
REGISTER_FILTER(AMERGE, amerge, af);
REGISTER_FILTER(AMIX, amix, af);

2
libavfilter/version.h
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@ -30,7 +30,7 @@
#include "libavutil/version.h"
#define LIBAVFILTER_VERSION_MAJOR 6
#define LIBAVFILTER_VERSION_MINOR 2
#define LIBAVFILTER_VERSION_MINOR 3
#define LIBAVFILTER_VERSION_MICRO 100
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \