GCC Code Coverage Report
Directory: ../../../ffmpeg/ Exec Total Coverage
File: src/libavfilter/af_acrusher.c Lines: 0 138 0.0 %
Date: 2020-08-14 10:39:37 Branches: 0 74 0.0 %

Line Branch Exec Source
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/*
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 * Copyright (c) Markus Schmidt and Christian Holschuh
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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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#include "libavutil/opt.h"
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#include "avfilter.h"
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#include "internal.h"
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#include "audio.h"
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typedef struct LFOContext {
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    double freq;
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    double offset;
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    int srate;
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    double amount;
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    double pwidth;
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    double phase;
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} LFOContext;
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typedef struct SRContext {
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    double target;
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    double real;
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    double samples;
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    double last;
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} SRContext;
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typedef struct ACrusherContext {
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    const AVClass *class;
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    double level_in;
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    double level_out;
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    double bits;
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    double mix;
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    int mode;
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    double dc;
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    double idc;
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    double aa;
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    double samples;
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    int is_lfo;
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    double lforange;
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    double lforate;
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    double sqr;
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    double aa1;
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    double coeff;
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    int    round;
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    double sov;
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    double smin;
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    double sdiff;
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    LFOContext lfo;
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    SRContext *sr;
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} ACrusherContext;
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#define OFFSET(x) offsetof(ACrusherContext, x)
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#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption acrusher_options[] = {
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    { "level_in", "set level in",         OFFSET(level_in),  AV_OPT_TYPE_DOUBLE, {.dbl=1},    0.015625, 64, A },
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    { "level_out","set level out",        OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1},    0.015625, 64, A },
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    { "bits",     "set bit reduction",    OFFSET(bits),      AV_OPT_TYPE_DOUBLE, {.dbl=8},    1,        64, A },
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    { "mix",      "set mix",              OFFSET(mix),       AV_OPT_TYPE_DOUBLE, {.dbl=.5},   0,         1, A },
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    { "mode",     "set mode",             OFFSET(mode),      AV_OPT_TYPE_INT,    {.i64=0},    0,         1, A, "mode" },
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    {   "lin",    "linear",               0,                 AV_OPT_TYPE_CONST,  {.i64=0},    0,         0, A, "mode" },
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    {   "log",    "logarithmic",          0,                 AV_OPT_TYPE_CONST,  {.i64=1},    0,         0, A, "mode" },
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    { "dc",       "set DC",               OFFSET(dc),        AV_OPT_TYPE_DOUBLE, {.dbl=1},  .25,         4, A },
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    { "aa",       "set anti-aliasing",    OFFSET(aa),        AV_OPT_TYPE_DOUBLE, {.dbl=.5},   0,         1, A },
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    { "samples",  "set sample reduction", OFFSET(samples),   AV_OPT_TYPE_DOUBLE, {.dbl=1},    1,       250, A },
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    { "lfo",      "enable LFO",           OFFSET(is_lfo),    AV_OPT_TYPE_BOOL,   {.i64=0},    0,         1, A },
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    { "lforange", "set LFO depth",        OFFSET(lforange),  AV_OPT_TYPE_DOUBLE, {.dbl=20},   1,       250, A },
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    { "lforate",  "set LFO rate",         OFFSET(lforate),   AV_OPT_TYPE_DOUBLE, {.dbl=.3}, .01,       200, A },
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    { NULL }
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};
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AVFILTER_DEFINE_CLASS(acrusher);
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static double samplereduction(ACrusherContext *s, SRContext *sr, double in)
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{
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    sr->samples++;
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    if (sr->samples >= s->round) {
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        sr->target += s->samples;
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        sr->real += s->round;
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        if (sr->target + s->samples >= sr->real + 1) {
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            sr->last = in;
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            sr->target = 0;
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            sr->real   = 0;
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        }
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        sr->samples = 0;
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    }
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    return sr->last;
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}
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static double add_dc(double s, double dc, double idc)
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{
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    return s > 0 ? s * dc : s * idc;
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}
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static double remove_dc(double s, double dc, double idc)
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{
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    return s > 0 ? s * idc : s * dc;
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}
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static inline double factor(double y, double k, double aa1, double aa)
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{
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    return 0.5 * (sin(M_PI * (fabs(y - k) - aa1) / aa - M_PI_2) + 1);
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}
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static double bitreduction(ACrusherContext *s, double in)
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{
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    const double sqr = s->sqr;
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    const double coeff = s->coeff;
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    const double aa = s->aa;
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    const double aa1 = s->aa1;
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    double y, k;
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    // add dc
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    in = add_dc(in, s->dc, s->idc);
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    // main rounding calculation depending on mode
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    // the idea for anti-aliasing:
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    // you need a function f which brings you to the scale, where
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    // you want to round and the function f_b (with f(f_b)=id) which
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    // brings you back to your original scale.
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    //
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    // then you can use the logic below in the following way:
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    // y = f(in) and k = roundf(y)
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    // if (y > k + aa1)
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    //      k = f_b(k) + ( f_b(k+1) - f_b(k) ) * 0.5 * (sin(x - PI/2) + 1)
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    // if (y < k + aa1)
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    //      k = f_b(k) - ( f_b(k+1) - f_b(k) ) * 0.5 * (sin(x - PI/2) + 1)
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    //
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    // whereas x = (fabs(f(in) - k) - aa1) * PI / aa
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    // for both cases.
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    switch (s->mode) {
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    case 0:
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    default:
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        // linear
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        y = in * coeff;
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        k = roundf(y);
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        if (k - aa1 <= y && y <= k + aa1) {
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            k /= coeff;
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        } else if (y > k + aa1) {
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            k = k / coeff + ((k + 1) / coeff - k / coeff) *
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                factor(y, k, aa1, aa);
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        } else {
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            k = k / coeff - (k / coeff - (k - 1) / coeff) *
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                factor(y, k, aa1, aa);
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        }
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        break;
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    case 1:
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        // logarithmic
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        y = sqr * log(fabs(in)) + sqr * sqr;
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        k = roundf(y);
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        if(!in) {
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            k = 0;
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        } else if (k - aa1 <= y && y <= k + aa1) {
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            k = in / fabs(in) * exp(k / sqr - sqr);
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        } else if (y > k + aa1) {
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            double x = exp(k / sqr - sqr);
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            k = FFSIGN(in) * (x + (exp((k + 1) / sqr - sqr) - x) *
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                factor(y, k, aa1, aa));
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        } else {
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            double x = exp(k / sqr - sqr);
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            k = in / fabs(in) * (x - (x - exp((k - 1) / sqr - sqr)) *
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                factor(y, k, aa1, aa));
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        }
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        break;
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    }
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    // mix between dry and wet signal
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    k += (in - k) * s->mix;
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    // remove dc
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    k = remove_dc(k, s->dc, s->idc);
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    return k;
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}
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static double lfo_get(LFOContext *lfo)
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{
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    double phs = FFMIN(100., lfo->phase / FFMIN(1.99, FFMAX(0.01, lfo->pwidth)) + lfo->offset);
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    double val;
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    if (phs > 1)
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        phs = fmod(phs, 1.);
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    val = sin((phs * 360.) * M_PI / 180);
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    return val * lfo->amount;
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}
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static void lfo_advance(LFOContext *lfo, unsigned count)
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{
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    lfo->phase = fabs(lfo->phase + count * lfo->freq * (1. / lfo->srate));
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    if (lfo->phase >= 1.)
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        lfo->phase = fmod(lfo->phase, 1.);
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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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    ACrusherContext *s = ctx->priv;
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    AVFilterLink *outlink = ctx->outputs[0];
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    AVFrame *out;
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    const double *src = (const double *)in->data[0];
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    double *dst;
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    const double level_in = s->level_in;
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    const double level_out = s->level_out;
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    const double mix = s->mix;
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    int n, c;
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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(inlink, 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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        if (s->is_lfo) {
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            s->samples = s->smin + s->sdiff * (lfo_get(&s->lfo) + 0.5);
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            s->round = round(s->samples);
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        }
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        for (c = 0; c < inlink->channels; c++) {
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            double sample = src[c] * level_in;
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            sample = mix * samplereduction(s, &s->sr[c], sample) + src[c] * (1. - mix) * level_in;
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            dst[c] = bitreduction(s, sample) * level_out;
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        }
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        src += c;
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        dst += c;
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        if (s->is_lfo)
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            lfo_advance(&s->lfo, 1);
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    }
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    if (in != out)
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        av_frame_free(&in);
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    return ff_filter_frame(outlink, out);
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}
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static int query_formats(AVFilterContext *ctx)
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{
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    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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    layouts = ff_all_channel_counts();
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    if (!layouts)
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        return AVERROR(ENOMEM);
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    ret = ff_set_common_channel_layouts(ctx, layouts);
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    if (ret < 0)
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        return ret;
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    formats = ff_make_format_list(sample_fmts);
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    if (!formats)
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        return AVERROR(ENOMEM);
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    ret = ff_set_common_formats(ctx, formats);
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    if (ret < 0)
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        return ret;
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    formats = ff_all_samplerates();
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    if (!formats)
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        return AVERROR(ENOMEM);
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    return ff_set_common_samplerates(ctx, formats);
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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    ACrusherContext *s = ctx->priv;
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    av_freep(&s->sr);
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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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    ACrusherContext *s = ctx->priv;
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    double rad, sunder, smax, sover;
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    s->idc = 1. / s->dc;
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    s->coeff = exp2(s->bits) - 1;
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    s->sqr = sqrt(s->coeff / 2);
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    s->aa1 = (1. - s->aa) / 2.;
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    s->round = round(s->samples);
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    rad = s->lforange / 2.;
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    s->smin = FFMAX(s->samples - rad, 1.);
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    sunder   = s->samples - rad - s->smin;
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    smax = FFMIN(s->samples + rad, 250.);
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    sover    = s->samples + rad - smax;
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    smax    -= sunder;
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    s->smin -= sover;
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    s->sdiff = smax - s->smin;
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    s->lfo.freq = s->lforate;
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    s->lfo.pwidth = 1.;
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    s->lfo.srate = inlink->sample_rate;
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    s->lfo.amount = .5;
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    s->sr = av_calloc(inlink->channels, sizeof(*s->sr));
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    if (!s->sr)
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        return AVERROR(ENOMEM);
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    return 0;
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}
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static const AVFilterPad avfilter_af_acrusher_inputs[] = {
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    {
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        .name         = "default",
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        .type         = AVMEDIA_TYPE_AUDIO,
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        .config_props = config_input,
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        .filter_frame = filter_frame,
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    },
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    { NULL }
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};
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static const AVFilterPad avfilter_af_acrusher_outputs[] = {
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    {
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        .name = "default",
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        .type = AVMEDIA_TYPE_AUDIO,
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    },
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    { NULL }
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};
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AVFilter ff_af_acrusher = {
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    .name          = "acrusher",
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    .description   = NULL_IF_CONFIG_SMALL("Reduce audio bit resolution."),
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    .priv_size     = sizeof(ACrusherContext),
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    .priv_class    = &acrusher_class,
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    .uninit        = uninit,
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    .query_formats = query_formats,
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    .inputs        = avfilter_af_acrusher_inputs,
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    .outputs       = avfilter_af_acrusher_outputs,
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};