GCC Code Coverage Report
Directory: ../../../ffmpeg/ Exec Total Coverage
File: src/libavfilter/af_alimiter.c Lines: 130 170 76.5 %
Date: 2020-08-11 16:46:18 Branches: 57 104 54.8 %

Line Branch Exec Source
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/*
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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/avassert.h"
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#include "libavutil/channel_layout.h"
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#include "libavutil/common.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 "formats.h"
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#include "internal.h"
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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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    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 A AV_OPT_FLAG_AUDIO_PARAM
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#define F AV_OPT_FLAG_FILTERING_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, A|F },
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    { "level_out", "set output level", OFFSET(level_out),    AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625,   64, A|F },
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    { "limit",     "set limit",        OFFSET(limit),        AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.0625,    1, A|F },
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    { "attack",    "set attack",       OFFSET(attack),       AV_OPT_TYPE_DOUBLE, {.dbl=5},    0.1,   80, A|F },
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    { "release",   "set release",      OFFSET(release),      AV_OPT_TYPE_DOUBLE, {.dbl=50},     1, 8000, A|F },
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    { "asc",       "enable asc",       OFFSET(auto_release), AV_OPT_TYPE_BOOL,   {.i64=0},      0,    1, A|F },
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    { "asc_level", "set asc level",    OFFSET(asc_coeff),    AV_OPT_TYPE_DOUBLE, {.dbl=0.5},    0,    1, A|F },
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    { "level",     "auto level",       OFFSET(auto_level),   AV_OPT_TYPE_BOOL,   {.i64=1},      0,    1, A|F },
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    { NULL }
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};
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AVFILTER_DEFINE_CLASS(alimiter);
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1
static av_cold int init(AVFilterContext *ctx)
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{
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1
    AudioLimiterContext *s = ctx->priv;
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1
    s->attack   /= 1000.;
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1
    s->release  /= 1000.;
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1
    s->att       = 1.;
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1
    s->asc_pos   = -1;
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1
    s->asc_coeff = pow(0.5, s->asc_coeff - 0.5) * 2 * -1;
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1
    return 0;
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}
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138816
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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138816
    double rdelta = (1.0 - patt) / (sample_rate * release);
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138816
    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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138816
    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->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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259
    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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    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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264859
    for (n = 0; n < in->nb_samples; n++) {
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264600
        double peak = 0;
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793800
        for (c = 0; c < channels; c++) {
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529200
            double sample = src[c] * level_in;
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153
529200
            buffer[s->pos + c] = sample;
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529200
            peak = FFMAX(peak, fabs(sample));
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        }
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264600
        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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264600
        if (peak > limit) {
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138816
            double patt = FFMIN(limit / peak, 1.);
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138816
            double rdelta = get_rdelta(s, release, inlink->sample_rate,
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                                       peak, limit, patt, 0);
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138816
            double delta = (limit / peak - s->att) / buffer_size * channels;
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138816
            int found = 0;
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138816
            if (delta < s->delta) {
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1669
                s->delta = delta;
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1669
                nextpos[0] = s->pos;
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                nextpos[1] = -1;
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                nextdelta[0] = rdelta;
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1669
                s->nextlen = 1;
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1669
                s->nextiter= 0;
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            } else {
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564375
                for (i = s->nextiter; i < s->nextiter + s->nextlen; i++) {
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439864
                    int j = i % buffer_size;
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                    double ppeak, pdelta;
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879728
                    ppeak = fabs(buffer[nextpos[j]]) > fabs(buffer[nextpos[j] + 1]) ?
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439864
                            fabs(buffer[nextpos[j]]) : fabs(buffer[nextpos[j] + 1]);
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439864
                    pdelta = (limit / peak - limit / ppeak) / (((buffer_size - nextpos[j] + s->pos) % buffer_size) / channels);
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439864
                    if (pdelta < nextdelta[j]) {
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12636
                        nextdelta[j] = pdelta;
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12636
                        found = 1;
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12636
                        break;
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                    }
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                }
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137147
                if (found) {
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12636
                    s->nextlen = i - s->nextiter + 1;
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12636
                    nextpos[(s->nextiter + s->nextlen) % buffer_size] = s->pos;
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12636
                    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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12636
                    s->nextlen++;
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                }
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            }
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        }
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264600
        buf = &s->buffer[(s->pos + channels) % buffer_size];
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264600
        peak = 0;
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793800
        for (c = 0; c < channels; c++) {
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529200
            double sample = buf[c];
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529200
            peak = FFMAX(peak, fabs(sample));
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        }
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264600
        if (s->pos == s->asc_pos && !s->asc_changed)
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            s->asc_pos = -1;
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264600
        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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264600
        s->att += s->delta;
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793800
        for (c = 0; c < channels; c++)
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529200
            dst[c] = buf[c] * s->att;
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264600
        if ((s->pos + channels) % buffer_size == nextpos[s->nextiter]) {
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2019
            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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                    int pnextpos = nextpos[(s->nextiter + 1) % buffer_size];
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                    double ppeak = fabs(buffer[pnextpos]) > fabs(buffer[pnextpos + 1]) ?
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                                                            fabs(buffer[pnextpos]) :
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                                                            fabs(buffer[pnextpos + 1]);
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                    double 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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2019
                s->delta = nextdelta[s->nextiter];
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2019
                s->att = limit / peak;
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            }
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2019
            s->nextlen -= 1;
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2019
            nextpos[s->nextiter] = -1;
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2019
            s->nextiter = (s->nextiter + 1) % buffer_size;
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        }
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264600
        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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264600
        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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264600
        if (s->att != 1. && (1. - s->att) < 0.0000000000001)
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5
            s->att = 1.;
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264600
        if (s->delta != 0. && fabs(s->delta) < 0.00000000000001)
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            s->delta = 0.;
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793800
        for (c = 0; c < channels; c++)
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529200
            dst[c] = av_clipd(dst[c], -limit, limit) * level * level_out;
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264600
        s->pos = (s->pos + channels) % buffer_size;
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264600
        src += channels;
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264600
        dst += channels;
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    }
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259
    if (in != out)
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        av_frame_free(&in);
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259
    return ff_filter_frame(outlink, out);
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}
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1
static int query_formats(AVFilterContext *ctx)
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{
282
    AVFilterFormats *formats;
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    AVFilterChannelLayouts *layouts;
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    static const enum AVSampleFormat sample_fmts[] = {
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        AV_SAMPLE_FMT_DBL,
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        AV_SAMPLE_FMT_NONE
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    };
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    int ret;
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290
1
    layouts = ff_all_channel_counts();
291
1
    if (!layouts)
292
        return AVERROR(ENOMEM);
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1
    ret = ff_set_common_channel_layouts(ctx, layouts);
294
1
    if (ret < 0)
295
        return ret;
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1
    formats = ff_make_format_list(sample_fmts);
298
1
    if (!formats)
299
        return AVERROR(ENOMEM);
300
1
    ret = ff_set_common_formats(ctx, formats);
301
1
    if (ret < 0)
302
        return ret;
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1
    formats = ff_all_samplerates();
305
1
    if (!formats)
306
        return AVERROR(ENOMEM);
307
1
    return ff_set_common_samplerates(ctx, formats);
308
}
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1
static int config_input(AVFilterLink *inlink)
311
{
312
1
    AVFilterContext *ctx = inlink->dst;
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1
    AudioLimiterContext *s = ctx->priv;
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    int obuffer_size;
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316
1
    obuffer_size = inlink->sample_rate * inlink->channels * 100 / 1000. + inlink->channels;
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1
    if (obuffer_size < inlink->channels)
318
        return AVERROR(EINVAL);
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1
    s->buffer = av_calloc(obuffer_size, sizeof(*s->buffer));
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1
    s->nextdelta = av_calloc(obuffer_size, sizeof(*s->nextdelta));
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1
    s->nextpos = av_malloc_array(obuffer_size, sizeof(*s->nextpos));
323

1
    if (!s->buffer || !s->nextdelta || !s->nextpos)
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        return AVERROR(ENOMEM);
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326
1
    memset(s->nextpos, -1, obuffer_size * sizeof(*s->nextpos));
327
1
    s->buffer_size = inlink->sample_rate * s->attack * inlink->channels;
328
1
    s->buffer_size -= s->buffer_size % inlink->channels;
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330
1
    if (s->buffer_size <= 0) {
331
        av_log(ctx, AV_LOG_ERROR, "Attack is too small.\n");
332
        return AVERROR(EINVAL);
333
    }
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335
1
    return 0;
336
}
337
338
1
static av_cold void uninit(AVFilterContext *ctx)
339
{
340
1
    AudioLimiterContext *s = ctx->priv;
341
342
1
    av_freep(&s->buffer);
343
1
    av_freep(&s->nextdelta);
344
1
    av_freep(&s->nextpos);
345
1
}
346
347
static const AVFilterPad alimiter_inputs[] = {
348
    {
349
        .name         = "main",
350
        .type         = AVMEDIA_TYPE_AUDIO,
351
        .filter_frame = filter_frame,
352
        .config_props = config_input,
353
    },
354
    { NULL }
355
};
356
357
static const AVFilterPad alimiter_outputs[] = {
358
    {
359
        .name = "default",
360
        .type = AVMEDIA_TYPE_AUDIO,
361
    },
362
    { NULL }
363
};
364
365
AVFilter ff_af_alimiter = {
366
    .name           = "alimiter",
367
    .description    = NULL_IF_CONFIG_SMALL("Audio lookahead limiter."),
368
    .priv_size      = sizeof(AudioLimiterContext),
369
    .priv_class     = &alimiter_class,
370
    .init           = init,
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    .uninit         = uninit,
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    .query_formats  = query_formats,
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    .inputs         = alimiter_inputs,
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    .outputs        = alimiter_outputs,
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};