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ffmpeg/libavfilter/af_alimiter.c

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/*
* 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/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/fifo.h"
#include "libavutil/mem.h"
#include "libavutil/opt.h"
#include "audio.h"
#include "avfilter.h"
#include "internal.h"
typedef struct MetaItem {
int64_t pts;
int nb_samples;
} MetaItem;
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;
int in_trim;
int out_pad;
int64_t next_in_pts;
int64_t next_out_pts;
int latency;
AVFifo *fifo;
double delta;
int nextiter;
int nextlen;
int asc_changed;
} AudioLimiterContext;
#define OFFSET(x) offsetof(AudioLimiterContext, x)
#define AF AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM | AV_OPT_FLAG_RUNTIME_PARAM
static const AVOption alimiter_options[] = {
{ "level_in", "set input level", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625, 64, AF },
{ "level_out", "set output level", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625, 64, AF },
{ "limit", "set limit", OFFSET(limit), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.0625, 1, AF },
{ "attack", "set attack", OFFSET(attack), AV_OPT_TYPE_DOUBLE, {.dbl=5}, 0.1, 80, AF },
{ "release", "set release", OFFSET(release), AV_OPT_TYPE_DOUBLE, {.dbl=50}, 1, 8000, AF },
{ "asc", "enable asc", OFFSET(auto_release), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, AF },
{ "asc_level", "set asc level", OFFSET(asc_coeff), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 1, AF },
{ "level", "auto level", OFFSET(auto_level), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, AF },
{ "latency", "compensate delay", OFFSET(latency), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, AF },
{ 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->ch_layout.nb_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;
int new_out_samples;
int64_t out_duration;
int64_t in_duration;
int64_t in_pts;
MetaItem meta;
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 = 0, pdelta;
if (nextpos[j] >= 0)
for (c = 0; c < channels; c++) {
ppeak = FFMAX(ppeak, fabs(buffer[nextpos[j] + c]));
}
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) {
double ppeak = 0, pdelta;
int pnextpos = nextpos[(s->nextiter + 1) % buffer_size];
for (c = 0; c < channels; c++) {
ppeak = FFMAX(ppeak, fabs(buffer[pnextpos + c]));
}
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;
}
in_duration = av_rescale_q(in->nb_samples, inlink->time_base, av_make_q(1, in->sample_rate));
in_pts = in->pts;
meta = (MetaItem){ in->pts, in->nb_samples };
av_fifo_write(s->fifo, &meta, 1);
if (in != out)
av_frame_free(&in);
new_out_samples = out->nb_samples;
if (s->in_trim > 0) {
int trim = FFMIN(new_out_samples, s->in_trim);
new_out_samples -= trim;
s->in_trim -= trim;
}
if (new_out_samples <= 0) {
av_frame_free(&out);
return 0;
} else if (new_out_samples < out->nb_samples) {
int offset = out->nb_samples - new_out_samples;
memmove(out->extended_data[0], out->extended_data[0] + sizeof(double) * offset * out->ch_layout.nb_channels,
sizeof(double) * new_out_samples * out->ch_layout.nb_channels);
out->nb_samples = new_out_samples;
s->in_trim = 0;
}
av_fifo_read(s->fifo, &meta, 1);
out_duration = av_rescale_q(out->nb_samples, inlink->time_base, av_make_q(1, out->sample_rate));
in_duration = av_rescale_q(meta.nb_samples, inlink->time_base, av_make_q(1, out->sample_rate));
in_pts = meta.pts;
if (s->next_out_pts != AV_NOPTS_VALUE && out->pts != s->next_out_pts &&
s->next_in_pts != AV_NOPTS_VALUE && in_pts == s->next_in_pts) {
out->pts = s->next_out_pts;
} else {
out->pts = in_pts;
}
s->next_in_pts = in_pts + in_duration;
s->next_out_pts = out->pts + out_duration;
return ff_filter_frame(outlink, out);
}
static int request_frame(AVFilterLink* outlink)
{
AVFilterContext *ctx = outlink->src;
AudioLimiterContext *s = (AudioLimiterContext*)ctx->priv;
int ret;
ret = ff_request_frame(ctx->inputs[0]);
if (ret == AVERROR_EOF && s->out_pad > 0) {
AVFrame *frame = ff_get_audio_buffer(outlink, FFMIN(1024, s->out_pad));
if (!frame)
return AVERROR(ENOMEM);
s->out_pad -= frame->nb_samples;
frame->pts = s->next_in_pts;
return filter_frame(ctx->inputs[0], frame);
}
return ret;
}
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
AudioLimiterContext *s = ctx->priv;
int obuffer_size;
obuffer_size = inlink->sample_rate * inlink->ch_layout.nb_channels * 100 / 1000. + inlink->ch_layout.nb_channels;
if (obuffer_size < inlink->ch_layout.nb_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->ch_layout.nb_channels;
s->buffer_size -= s->buffer_size % inlink->ch_layout.nb_channels;
if (s->latency)
s->in_trim = s->out_pad = s->buffer_size / inlink->ch_layout.nb_channels - 1;
s->next_out_pts = AV_NOPTS_VALUE;
s->next_in_pts = AV_NOPTS_VALUE;
s->fifo = av_fifo_alloc2(8, sizeof(MetaItem), AV_FIFO_FLAG_AUTO_GROW);
if (!s->fifo) {
return AVERROR(ENOMEM);
}
if (s->buffer_size <= 0) {
av_log(ctx, AV_LOG_ERROR, "Attack is too small.\n");
return AVERROR(EINVAL);
}
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);
av_fifo_freep2(&s->fifo);
}
static const AVFilterPad alimiter_inputs[] = {
{
.name = "main",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
.config_props = config_input,
},
};
static const AVFilterPad alimiter_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.request_frame = request_frame,
},
};
const 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,
FILTER_INPUTS(alimiter_inputs),
FILTER_OUTPUTS(alimiter_outputs),
FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBL),
.process_command = ff_filter_process_command,
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC,
};
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