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790f793844
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>
437 lines
14 KiB
C
437 lines
14 KiB
C
/*
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* Copyright (C) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen and others
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* Copyright (c) 2015 Paul B Mahol
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* Lookahead limiter filter
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*/
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#include "libavutil/channel_layout.h"
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#include "libavutil/common.h"
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#include "libavutil/fifo.h"
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#include "libavutil/mem.h"
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#include "libavutil/opt.h"
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#include "audio.h"
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#include "avfilter.h"
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#include "internal.h"
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typedef struct MetaItem {
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int64_t pts;
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int nb_samples;
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} MetaItem;
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typedef struct AudioLimiterContext {
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const AVClass *class;
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double limit;
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double attack;
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double release;
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double att;
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double level_in;
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double level_out;
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int auto_release;
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int auto_level;
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double asc;
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int asc_c;
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int asc_pos;
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double asc_coeff;
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double *buffer;
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int buffer_size;
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int pos;
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int *nextpos;
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double *nextdelta;
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int in_trim;
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int out_pad;
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int64_t next_in_pts;
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int64_t next_out_pts;
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int latency;
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AVFifo *fifo;
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double delta;
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int nextiter;
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int nextlen;
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int asc_changed;
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} AudioLimiterContext;
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#define OFFSET(x) offsetof(AudioLimiterContext, x)
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#define AF AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM | AV_OPT_FLAG_RUNTIME_PARAM
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static const AVOption alimiter_options[] = {
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{ "level_in", "set input level", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625, 64, AF },
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{ "level_out", "set output level", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625, 64, AF },
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{ "limit", "set limit", OFFSET(limit), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.0625, 1, AF },
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{ "attack", "set attack", OFFSET(attack), AV_OPT_TYPE_DOUBLE, {.dbl=5}, 0.1, 80, AF },
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{ "release", "set release", OFFSET(release), AV_OPT_TYPE_DOUBLE, {.dbl=50}, 1, 8000, AF },
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{ "asc", "enable asc", OFFSET(auto_release), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, AF },
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{ "asc_level", "set asc level", OFFSET(asc_coeff), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 1, AF },
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{ "level", "auto level", OFFSET(auto_level), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, AF },
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{ "latency", "compensate delay", OFFSET(latency), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, AF },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(alimiter);
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static av_cold int init(AVFilterContext *ctx)
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{
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AudioLimiterContext *s = ctx->priv;
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s->attack /= 1000.;
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s->release /= 1000.;
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s->att = 1.;
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s->asc_pos = -1;
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s->asc_coeff = pow(0.5, s->asc_coeff - 0.5) * 2 * -1;
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return 0;
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}
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static double get_rdelta(AudioLimiterContext *s, double release, int sample_rate,
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double peak, double limit, double patt, int asc)
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{
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double rdelta = (1.0 - patt) / (sample_rate * release);
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if (asc && s->auto_release && s->asc_c > 0) {
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double a_att = limit / (s->asc_coeff * s->asc) * (double)s->asc_c;
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if (a_att > patt) {
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double delta = FFMAX((a_att - patt) / (sample_rate * release), rdelta / 10);
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if (delta < rdelta)
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rdelta = delta;
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}
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}
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return rdelta;
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}
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static int filter_frame(AVFilterLink *inlink, AVFrame *in)
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{
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AVFilterContext *ctx = inlink->dst;
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AudioLimiterContext *s = ctx->priv;
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AVFilterLink *outlink = ctx->outputs[0];
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const double *src = (const double *)in->data[0];
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const int channels = inlink->ch_layout.nb_channels;
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const int buffer_size = s->buffer_size;
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double *dst, *buffer = s->buffer;
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const double release = s->release;
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const double limit = s->limit;
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double *nextdelta = s->nextdelta;
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double level = s->auto_level ? 1 / limit : 1;
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const double level_out = s->level_out;
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const double level_in = s->level_in;
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int *nextpos = s->nextpos;
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AVFrame *out;
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double *buf;
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int n, c, i;
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int new_out_samples;
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int64_t out_duration;
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int64_t in_duration;
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int64_t in_pts;
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MetaItem meta;
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if (av_frame_is_writable(in)) {
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out = in;
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} else {
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out = ff_get_audio_buffer(outlink, in->nb_samples);
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if (!out) {
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av_frame_free(&in);
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return AVERROR(ENOMEM);
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}
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av_frame_copy_props(out, in);
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}
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dst = (double *)out->data[0];
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for (n = 0; n < in->nb_samples; n++) {
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double peak = 0;
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for (c = 0; c < channels; c++) {
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double sample = src[c] * level_in;
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buffer[s->pos + c] = sample;
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peak = FFMAX(peak, fabs(sample));
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}
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if (s->auto_release && peak > limit) {
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s->asc += peak;
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s->asc_c++;
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}
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if (peak > limit) {
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double patt = FFMIN(limit / peak, 1.);
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double rdelta = get_rdelta(s, release, inlink->sample_rate,
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peak, limit, patt, 0);
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double delta = (limit / peak - s->att) / buffer_size * channels;
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int found = 0;
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if (delta < s->delta) {
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s->delta = delta;
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nextpos[0] = s->pos;
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nextpos[1] = -1;
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nextdelta[0] = rdelta;
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s->nextlen = 1;
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s->nextiter= 0;
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} else {
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for (i = s->nextiter; i < s->nextiter + s->nextlen; i++) {
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int j = i % buffer_size;
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double ppeak = 0, pdelta;
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if (nextpos[j] >= 0)
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for (c = 0; c < channels; c++) {
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ppeak = FFMAX(ppeak, fabs(buffer[nextpos[j] + c]));
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}
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pdelta = (limit / peak - limit / ppeak) / (((buffer_size - nextpos[j] + s->pos) % buffer_size) / channels);
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if (pdelta < nextdelta[j]) {
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nextdelta[j] = pdelta;
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found = 1;
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break;
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}
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}
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if (found) {
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s->nextlen = i - s->nextiter + 1;
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nextpos[(s->nextiter + s->nextlen) % buffer_size] = s->pos;
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nextdelta[(s->nextiter + s->nextlen) % buffer_size] = rdelta;
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nextpos[(s->nextiter + s->nextlen + 1) % buffer_size] = -1;
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s->nextlen++;
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}
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}
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}
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buf = &s->buffer[(s->pos + channels) % buffer_size];
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peak = 0;
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for (c = 0; c < channels; c++) {
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double sample = buf[c];
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peak = FFMAX(peak, fabs(sample));
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}
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if (s->pos == s->asc_pos && !s->asc_changed)
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s->asc_pos = -1;
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if (s->auto_release && s->asc_pos == -1 && peak > limit) {
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s->asc -= peak;
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s->asc_c--;
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}
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s->att += s->delta;
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for (c = 0; c < channels; c++)
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dst[c] = buf[c] * s->att;
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if ((s->pos + channels) % buffer_size == nextpos[s->nextiter]) {
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if (s->auto_release) {
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s->delta = get_rdelta(s, release, inlink->sample_rate,
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peak, limit, s->att, 1);
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if (s->nextlen > 1) {
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double ppeak = 0, pdelta;
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int pnextpos = nextpos[(s->nextiter + 1) % buffer_size];
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for (c = 0; c < channels; c++) {
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ppeak = FFMAX(ppeak, fabs(buffer[pnextpos + c]));
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}
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pdelta = (limit / ppeak - s->att) /
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(((buffer_size + pnextpos -
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((s->pos + channels) % buffer_size)) %
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buffer_size) / channels);
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if (pdelta < s->delta)
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s->delta = pdelta;
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}
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} else {
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s->delta = nextdelta[s->nextiter];
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s->att = limit / peak;
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}
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s->nextlen -= 1;
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nextpos[s->nextiter] = -1;
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s->nextiter = (s->nextiter + 1) % buffer_size;
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}
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if (s->att > 1.) {
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s->att = 1.;
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s->delta = 0.;
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s->nextiter = 0;
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s->nextlen = 0;
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nextpos[0] = -1;
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}
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if (s->att <= 0.) {
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s->att = 0.0000000000001;
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s->delta = (1.0 - s->att) / (inlink->sample_rate * release);
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}
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if (s->att != 1. && (1. - s->att) < 0.0000000000001)
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s->att = 1.;
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if (s->delta != 0. && fabs(s->delta) < 0.00000000000001)
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s->delta = 0.;
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for (c = 0; c < channels; c++)
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dst[c] = av_clipd(dst[c], -limit, limit) * level * level_out;
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s->pos = (s->pos + channels) % buffer_size;
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src += channels;
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dst += channels;
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}
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in_duration = av_rescale_q(in->nb_samples, inlink->time_base, av_make_q(1, in->sample_rate));
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in_pts = in->pts;
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meta = (MetaItem){ in->pts, in->nb_samples };
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av_fifo_write(s->fifo, &meta, 1);
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if (in != out)
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av_frame_free(&in);
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new_out_samples = out->nb_samples;
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if (s->in_trim > 0) {
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int trim = FFMIN(new_out_samples, s->in_trim);
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new_out_samples -= trim;
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s->in_trim -= trim;
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}
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if (new_out_samples <= 0) {
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av_frame_free(&out);
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return 0;
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} else if (new_out_samples < out->nb_samples) {
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int offset = out->nb_samples - new_out_samples;
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memmove(out->extended_data[0], out->extended_data[0] + sizeof(double) * offset * out->ch_layout.nb_channels,
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sizeof(double) * new_out_samples * out->ch_layout.nb_channels);
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out->nb_samples = new_out_samples;
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s->in_trim = 0;
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}
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av_fifo_read(s->fifo, &meta, 1);
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out_duration = av_rescale_q(out->nb_samples, inlink->time_base, av_make_q(1, out->sample_rate));
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in_duration = av_rescale_q(meta.nb_samples, inlink->time_base, av_make_q(1, out->sample_rate));
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in_pts = meta.pts;
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if (s->next_out_pts != AV_NOPTS_VALUE && out->pts != s->next_out_pts &&
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s->next_in_pts != AV_NOPTS_VALUE && in_pts == s->next_in_pts) {
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out->pts = s->next_out_pts;
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} else {
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out->pts = in_pts;
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}
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s->next_in_pts = in_pts + in_duration;
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s->next_out_pts = out->pts + out_duration;
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return ff_filter_frame(outlink, out);
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}
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static int request_frame(AVFilterLink* outlink)
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{
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AVFilterContext *ctx = outlink->src;
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AudioLimiterContext *s = (AudioLimiterContext*)ctx->priv;
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int ret;
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ret = ff_request_frame(ctx->inputs[0]);
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if (ret == AVERROR_EOF && s->out_pad > 0) {
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AVFrame *frame = ff_get_audio_buffer(outlink, FFMIN(1024, s->out_pad));
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if (!frame)
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return AVERROR(ENOMEM);
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s->out_pad -= frame->nb_samples;
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frame->pts = s->next_in_pts;
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return filter_frame(ctx->inputs[0], frame);
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}
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return ret;
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}
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static int config_input(AVFilterLink *inlink)
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{
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AVFilterContext *ctx = inlink->dst;
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AudioLimiterContext *s = ctx->priv;
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int obuffer_size;
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obuffer_size = inlink->sample_rate * inlink->ch_layout.nb_channels * 100 / 1000. + inlink->ch_layout.nb_channels;
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if (obuffer_size < inlink->ch_layout.nb_channels)
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return AVERROR(EINVAL);
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s->buffer = av_calloc(obuffer_size, sizeof(*s->buffer));
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s->nextdelta = av_calloc(obuffer_size, sizeof(*s->nextdelta));
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s->nextpos = av_malloc_array(obuffer_size, sizeof(*s->nextpos));
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if (!s->buffer || !s->nextdelta || !s->nextpos)
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return AVERROR(ENOMEM);
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memset(s->nextpos, -1, obuffer_size * sizeof(*s->nextpos));
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s->buffer_size = inlink->sample_rate * s->attack * inlink->ch_layout.nb_channels;
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s->buffer_size -= s->buffer_size % inlink->ch_layout.nb_channels;
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if (s->latency)
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s->in_trim = s->out_pad = s->buffer_size / inlink->ch_layout.nb_channels - 1;
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s->next_out_pts = AV_NOPTS_VALUE;
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s->next_in_pts = AV_NOPTS_VALUE;
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s->fifo = av_fifo_alloc2(8, sizeof(MetaItem), AV_FIFO_FLAG_AUTO_GROW);
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if (!s->fifo) {
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return AVERROR(ENOMEM);
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}
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if (s->buffer_size <= 0) {
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av_log(ctx, AV_LOG_ERROR, "Attack is too small.\n");
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return AVERROR(EINVAL);
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}
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return 0;
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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AudioLimiterContext *s = ctx->priv;
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av_freep(&s->buffer);
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av_freep(&s->nextdelta);
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av_freep(&s->nextpos);
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av_fifo_freep2(&s->fifo);
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}
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static const AVFilterPad alimiter_inputs[] = {
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{
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.name = "main",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_frame = filter_frame,
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.config_props = config_input,
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},
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};
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static const AVFilterPad alimiter_outputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.request_frame = request_frame,
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},
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};
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const AVFilter ff_af_alimiter = {
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.name = "alimiter",
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.description = NULL_IF_CONFIG_SMALL("Audio lookahead limiter."),
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.priv_size = sizeof(AudioLimiterContext),
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.priv_class = &alimiter_class,
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.init = init,
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.uninit = uninit,
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FILTER_INPUTS(alimiter_inputs),
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FILTER_OUTPUTS(alimiter_outputs),
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FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBL),
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.process_command = ff_filter_process_command,
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.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC,
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};
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