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FFmpeg/libavfilter/af_alimiter.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

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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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