GCC Code Coverage Report
Directory: ../../../ffmpeg/ Exec Total Coverage
File: src/libavfilter/af_acrossover.c Lines: 0 170 0.0 %
Date: 2020-10-23 17:01:47 Branches: 0 72 0.0 %

Line Branch Exec Source
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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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 * Crossover filter
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 *
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 * Split an audio stream into several bands.
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 */
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#include "libavutil/attributes.h"
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#include "libavutil/avstring.h"
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#include "libavutil/channel_layout.h"
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#include "libavutil/eval.h"
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#include "libavutil/internal.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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#define MAX_SPLITS 16
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#define MAX_BANDS MAX_SPLITS + 1
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typedef struct BiquadContext {
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    double a0, a1, a2;
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    double b1, b2;
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    double i1, i2;
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    double o1, o2;
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} BiquadContext;
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typedef struct CrossoverChannel {
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    BiquadContext lp[MAX_BANDS][4];
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    BiquadContext hp[MAX_BANDS][4];
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} CrossoverChannel;
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typedef struct AudioCrossoverContext {
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    const AVClass *class;
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    char *splits_str;
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    int order;
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    int filter_count;
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    int nb_splits;
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    float *splits;
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    CrossoverChannel *xover;
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    AVFrame *input_frame;
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    AVFrame *frames[MAX_BANDS];
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} AudioCrossoverContext;
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#define OFFSET(x) offsetof(AudioCrossoverContext, x)
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#define AF AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption acrossover_options[] = {
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    { "split", "set split frequencies", OFFSET(splits_str), AV_OPT_TYPE_STRING, {.str="500"}, 0, 0, AF },
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    { "order", "set order",             OFFSET(order),      AV_OPT_TYPE_INT,    {.i64=1},     0, 2, AF, "m" },
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    { "2nd",   "2nd order",             0,                  AV_OPT_TYPE_CONST,  {.i64=0},     0, 0, AF, "m" },
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    { "4th",   "4th order",             0,                  AV_OPT_TYPE_CONST,  {.i64=1},     0, 0, AF, "m" },
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    { "8th",   "8th order",             0,                  AV_OPT_TYPE_CONST,  {.i64=2},     0, 0, AF, "m" },
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    { NULL }
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};
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AVFILTER_DEFINE_CLASS(acrossover);
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static av_cold int init(AVFilterContext *ctx)
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{
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    AudioCrossoverContext *s = ctx->priv;
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    char *p, *arg, *saveptr = NULL;
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    int i, ret = 0;
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    s->splits = av_calloc(MAX_SPLITS, sizeof(*s->splits));
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    if (!s->splits)
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        return AVERROR(ENOMEM);
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    p = s->splits_str;
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    for (i = 0; i < MAX_SPLITS; i++) {
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        float freq;
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        if (!(arg = av_strtok(p, " |", &saveptr)))
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            break;
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        p = NULL;
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        if (av_sscanf(arg, "%f", &freq) != 1) {
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            av_log(ctx, AV_LOG_ERROR, "Invalid syntax for frequency[%d].\n", i);
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            return AVERROR(EINVAL);
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        }
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        if (freq <= 0) {
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            av_log(ctx, AV_LOG_ERROR, "Frequency %f must be positive number.\n", freq);
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            return AVERROR(EINVAL);
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        }
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        if (i > 0 && freq <= s->splits[i-1]) {
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            av_log(ctx, AV_LOG_ERROR, "Frequency %f must be in increasing order.\n", freq);
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            return AVERROR(EINVAL);
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        }
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        s->splits[i] = freq;
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    }
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    s->nb_splits = i;
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    for (i = 0; i <= s->nb_splits; i++) {
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        AVFilterPad pad  = { 0 };
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        char *name;
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        pad.type = AVMEDIA_TYPE_AUDIO;
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        name = av_asprintf("out%d", ctx->nb_outputs);
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        if (!name)
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            return AVERROR(ENOMEM);
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        pad.name = name;
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        if ((ret = ff_insert_outpad(ctx, i, &pad)) < 0) {
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            av_freep(&pad.name);
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            return ret;
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        }
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    }
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    return ret;
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}
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static void set_lp(BiquadContext *b, double fc, double q, double sr)
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{
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    double omega = 2.0 * M_PI * fc / sr;
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    double sn = sin(omega);
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    double cs = cos(omega);
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    double alpha = sn / (2. * q);
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    double inv = 1.0 / (1.0 + alpha);
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    b->a0 = (1. - cs) * 0.5 * inv;
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    b->a1 = (1. - cs) * inv;
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    b->a2 = b->a0;
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    b->b1 = -2. * cs * inv;
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    b->b2 = (1. - alpha) * inv;
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}
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static void set_hp(BiquadContext *b, double fc, double q, double sr)
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{
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    double omega = 2 * M_PI * fc / sr;
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    double sn = sin(omega);
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    double cs = cos(omega);
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    double alpha = sn / (2 * q);
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    double inv = 1.0 / (1.0 + alpha);
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    b->a0 = inv * (1. + cs) / 2.;
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    b->a1 = -2. * b->a0;
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    b->a2 = b->a0;
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    b->b1 = -2. * cs * inv;
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    b->b2 = (1. - alpha) * inv;
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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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    AudioCrossoverContext *s = ctx->priv;
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    int ch, band, sample_rate = inlink->sample_rate;
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    double q;
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    s->xover = av_calloc(inlink->channels, sizeof(*s->xover));
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    if (!s->xover)
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        return AVERROR(ENOMEM);
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    switch (s->order) {
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    case 0:
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        q = 0.5;
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        s->filter_count = 1;
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        break;
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    case 1:
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        q = M_SQRT1_2;
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        s->filter_count = 2;
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        break;
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    case 2:
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        q = 0.54;
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        s->filter_count = 4;
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        break;
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    }
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    for (ch = 0; ch < inlink->channels; ch++) {
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        for (band = 0; band <= s->nb_splits; band++) {
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            set_lp(&s->xover[ch].lp[band][0], s->splits[band], q, sample_rate);
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            set_hp(&s->xover[ch].hp[band][0], s->splits[band], q, sample_rate);
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            if (s->order > 1) {
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                set_lp(&s->xover[ch].lp[band][1], s->splits[band], 1.34, sample_rate);
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                set_hp(&s->xover[ch].hp[band][1], s->splits[band], 1.34, sample_rate);
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                set_lp(&s->xover[ch].lp[band][2], s->splits[band],    q, sample_rate);
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                set_hp(&s->xover[ch].hp[band][2], s->splits[band],    q, sample_rate);
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                set_lp(&s->xover[ch].lp[band][3], s->splits[band], 1.34, sample_rate);
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                set_hp(&s->xover[ch].hp[band][3], s->splits[band], 1.34, sample_rate);
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            } else {
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                set_lp(&s->xover[ch].lp[band][1], s->splits[band], q, sample_rate);
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                set_hp(&s->xover[ch].hp[band][1], s->splits[band], q, sample_rate);
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            }
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        }
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    }
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    return 0;
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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_DBLP,
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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 double biquad_process(BiquadContext *b, double in)
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{
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    double out = in * b->a0 + b->i1 * b->a1 + b->i2 * b->a2 - b->o1 * b->b1 - b->o2 * b->b2;
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    b->i2 = b->i1;
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    b->o2 = b->o1;
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    b->i1 = in;
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    b->o1 = out;
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    return out;
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}
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static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
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{
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    AudioCrossoverContext *s = ctx->priv;
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    AVFrame *in = s->input_frame;
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    AVFrame **frames = s->frames;
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    const int start = (in->channels * jobnr) / nb_jobs;
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    const int end = (in->channels * (jobnr+1)) / nb_jobs;
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    int f, band;
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    for (int ch = start; ch < end; ch++) {
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        const double *src = (const double *)in->extended_data[ch];
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        CrossoverChannel *xover = &s->xover[ch];
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        for (int i = 0; i < in->nb_samples; i++) {
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            double sample = src[i], lo, hi;
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            for (band = 0; band < ctx->nb_outputs; band++) {
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                double *dst = (double *)frames[band]->extended_data[ch];
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                lo = sample;
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                hi = sample;
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                for (f = 0; band + 1 < ctx->nb_outputs && f < s->filter_count; f++) {
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                    BiquadContext *lp = &xover->lp[band][f];
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                    lo = biquad_process(lp, lo);
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                }
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                for (f = 0; band + 1 < ctx->nb_outputs && f < s->filter_count; f++) {
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                    BiquadContext *hp = &xover->hp[band][f];
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                    hi = biquad_process(hp, hi);
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                }
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                dst[i] = lo;
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                sample = hi;
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            }
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        }
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    }
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    return 0;
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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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    AudioCrossoverContext *s = ctx->priv;
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    AVFrame **frames = s->frames;
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    int i, ret = 0;
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    for (i = 0; i < ctx->nb_outputs; i++) {
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        frames[i] = ff_get_audio_buffer(ctx->outputs[i], in->nb_samples);
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        if (!frames[i]) {
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            ret = AVERROR(ENOMEM);
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            break;
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        }
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        frames[i]->pts = in->pts;
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    }
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    if (ret < 0)
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        goto fail;
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    s->input_frame = in;
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    ctx->internal->execute(ctx, filter_channels, NULL, NULL, FFMIN(inlink->channels,
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                                                                   ff_filter_get_nb_threads(ctx)));
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    for (i = 0; i < ctx->nb_outputs; i++) {
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        ret = ff_filter_frame(ctx->outputs[i], frames[i]);
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        frames[i] = NULL;
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        if (ret < 0)
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            break;
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    }
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fail:
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    for (i = 0; i < ctx->nb_outputs; i++)
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        av_frame_free(&frames[i]);
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    av_frame_free(&in);
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    s->input_frame = NULL;
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    return ret;
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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    AudioCrossoverContext *s = ctx->priv;
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    int i;
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    av_freep(&s->splits);
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    av_freep(&s->xover);
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    for (i = 0; i < ctx->nb_outputs; i++)
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        av_freep(&ctx->output_pads[i].name);
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}
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static const AVFilterPad inputs[] = {
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    {
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        .name         = "default",
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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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    { NULL }
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};
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AVFilter ff_af_acrossover = {
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    .name           = "acrossover",
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    .description    = NULL_IF_CONFIG_SMALL("Split audio into per-bands streams."),
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    .priv_size      = sizeof(AudioCrossoverContext),
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    .priv_class     = &acrossover_class,
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    .init           = init,
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    .uninit         = uninit,
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    .query_formats  = query_formats,
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    .inputs         = inputs,
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    .outputs        = NULL,
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    .flags          = AVFILTER_FLAG_DYNAMIC_OUTPUTS |
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                      AVFILTER_FLAG_SLICE_THREADS,
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};