GCC Code Coverage Report
Directory: ../../../ffmpeg/ Exec Total Coverage
File: src/libavfilter/af_chorus.c Lines: 139 165 84.2 %
Date: 2020-08-14 10:39:37 Branches: 60 100 60.0 %

Line Branch Exec Source
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/*
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 * Copyright (c) 1998 Juergen Mueller And Sundry Contributors
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 * This source code is freely redistributable and may be used for
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 * any purpose.  This copyright notice must be maintained.
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 * Juergen Mueller And Sundry Contributors are not responsible for
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 * the consequences of using this software.
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 *
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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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 * chorus audio filter
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 */
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#include "libavutil/avstring.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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#include "generate_wave_table.h"
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typedef struct ChorusContext {
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    const AVClass *class;
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    float in_gain, out_gain;
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    char *delays_str;
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    char *decays_str;
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    char *speeds_str;
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    char *depths_str;
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    float *delays;
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    float *decays;
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    float *speeds;
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    float *depths;
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    uint8_t **chorusbuf;
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    int **phase;
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    int *length;
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    int32_t **lookup_table;
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    int *counter;
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    int num_chorus;
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    int max_samples;
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    int channels;
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    int modulation;
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    int fade_out;
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    int64_t next_pts;
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} ChorusContext;
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#define OFFSET(x) offsetof(ChorusContext, x)
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#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption chorus_options[] = {
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    { "in_gain",  "set input gain",  OFFSET(in_gain),    AV_OPT_TYPE_FLOAT,  {.dbl=.4}, 0, 1, A },
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    { "out_gain", "set output gain", OFFSET(out_gain),   AV_OPT_TYPE_FLOAT,  {.dbl=.4}, 0, 1, A },
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    { "delays",   "set delays",      OFFSET(delays_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
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    { "decays",   "set decays",      OFFSET(decays_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
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    { "speeds",   "set speeds",      OFFSET(speeds_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
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    { "depths",   "set depths",      OFFSET(depths_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
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    { NULL }
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};
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AVFILTER_DEFINE_CLASS(chorus);
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4
static void count_items(char *item_str, int *nb_items)
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{
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    char *p;
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4
    *nb_items = 1;
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29
    for (p = item_str; *p; p++) {
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25
        if (*p == '|')
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            (*nb_items)++;
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    }
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4
}
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4
static void fill_items(char *item_str, int *nb_items, float *items)
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{
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4
    char *p, *saveptr = NULL;
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4
    int i, new_nb_items = 0;
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4
    p = item_str;
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8
    for (i = 0; i < *nb_items; i++) {
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4
        char *tstr = av_strtok(p, "|", &saveptr);
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4
        p = NULL;
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4
        if (tstr)
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4
            new_nb_items += sscanf(tstr, "%f", &items[new_nb_items]) == 1;
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    }
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4
    *nb_items = new_nb_items;
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4
}
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1
static av_cold int init(AVFilterContext *ctx)
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{
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1
    ChorusContext *s = ctx->priv;
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    int nb_delays, nb_decays, nb_speeds, nb_depths;
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    if (!s->delays_str || !s->decays_str || !s->speeds_str || !s->depths_str) {
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        av_log(ctx, AV_LOG_ERROR, "Both delays & decays & speeds & depths must be set.\n");
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        return AVERROR(EINVAL);
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    }
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1
    count_items(s->delays_str, &nb_delays);
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1
    count_items(s->decays_str, &nb_decays);
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1
    count_items(s->speeds_str, &nb_speeds);
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1
    count_items(s->depths_str, &nb_depths);
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1
    s->delays = av_realloc_f(s->delays, nb_delays, sizeof(*s->delays));
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1
    s->decays = av_realloc_f(s->decays, nb_decays, sizeof(*s->decays));
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1
    s->speeds = av_realloc_f(s->speeds, nb_speeds, sizeof(*s->speeds));
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1
    s->depths = av_realloc_f(s->depths, nb_depths, sizeof(*s->depths));
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1
    if (!s->delays || !s->decays || !s->speeds || !s->depths)
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        return AVERROR(ENOMEM);
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1
    fill_items(s->delays_str, &nb_delays, s->delays);
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1
    fill_items(s->decays_str, &nb_decays, s->decays);
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1
    fill_items(s->speeds_str, &nb_speeds, s->speeds);
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1
    fill_items(s->depths_str, &nb_depths, s->depths);
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    if (nb_delays != nb_decays && nb_delays != nb_speeds && nb_delays != nb_depths) {
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        av_log(ctx, AV_LOG_ERROR, "Number of delays & decays & speeds & depths given must be same.\n");
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        return AVERROR(EINVAL);
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    }
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1
    s->num_chorus = nb_delays;
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1
    if (s->num_chorus < 1) {
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        av_log(ctx, AV_LOG_ERROR, "At least one delay & decay & speed & depth must be set.\n");
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        return AVERROR(EINVAL);
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    }
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1
    s->length = av_calloc(s->num_chorus, sizeof(*s->length));
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1
    s->lookup_table = av_calloc(s->num_chorus, sizeof(*s->lookup_table));
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    if (!s->length || !s->lookup_table)
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        return AVERROR(ENOMEM);
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1
    s->next_pts = AV_NOPTS_VALUE;
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1
    return 0;
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}
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1
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_FLTP, AV_SAMPLE_FMT_NONE
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    };
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    int ret;
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1
    layouts = ff_all_channel_counts();
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1
    if (!layouts)
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        return AVERROR(ENOMEM);
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1
    ret = ff_set_common_channel_layouts(ctx, layouts);
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1
    if (ret < 0)
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        return ret;
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1
    formats = ff_make_format_list(sample_fmts);
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1
    if (!formats)
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        return AVERROR(ENOMEM);
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1
    ret = ff_set_common_formats(ctx, formats);
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1
    if (ret < 0)
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        return ret;
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1
    formats = ff_all_samplerates();
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1
    if (!formats)
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        return AVERROR(ENOMEM);
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1
    return ff_set_common_samplerates(ctx, formats);
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}
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1
static int config_output(AVFilterLink *outlink)
187
{
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1
    AVFilterContext *ctx = outlink->src;
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1
    ChorusContext *s = ctx->priv;
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1
    float sum_in_volume = 1.0;
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    int n;
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1
    s->channels = outlink->channels;
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2
    for (n = 0; n < s->num_chorus; n++) {
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1
        int samples = (int) ((s->delays[n] + s->depths[n]) * outlink->sample_rate / 1000.0);
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1
        int depth_samples = (int) (s->depths[n] * outlink->sample_rate / 1000.0);
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1
        s->length[n] = outlink->sample_rate / s->speeds[n];
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1
        s->lookup_table[n] = av_malloc(sizeof(int32_t) * s->length[n]);
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1
        if (!s->lookup_table[n])
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            return AVERROR(ENOMEM);
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1
        ff_generate_wave_table(WAVE_SIN, AV_SAMPLE_FMT_S32, s->lookup_table[n],
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1
                               s->length[n], 0., depth_samples, 0);
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1
        s->max_samples = FFMAX(s->max_samples, samples);
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    }
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2
    for (n = 0; n < s->num_chorus; n++)
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1
        sum_in_volume += s->decays[n];
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1
    if (s->in_gain * (sum_in_volume) > 1.0 / s->out_gain)
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        av_log(ctx, AV_LOG_WARNING, "output gain can cause saturation or clipping of output\n");
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1
    s->counter = av_calloc(outlink->channels, sizeof(*s->counter));
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1
    if (!s->counter)
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        return AVERROR(ENOMEM);
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1
    s->phase = av_calloc(outlink->channels, sizeof(*s->phase));
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1
    if (!s->phase)
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        return AVERROR(ENOMEM);
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2
    for (n = 0; n < outlink->channels; n++) {
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1
        s->phase[n] = av_calloc(s->num_chorus, sizeof(int));
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1
        if (!s->phase[n])
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            return AVERROR(ENOMEM);
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    }
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1
    s->fade_out = s->max_samples;
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1
    return av_samples_alloc_array_and_samples(&s->chorusbuf, NULL,
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                                              outlink->channels,
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                                              s->max_samples,
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1
                                              outlink->format, 0);
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}
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#define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
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static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
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{
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    AVFilterContext *ctx = inlink->dst;
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    ChorusContext *s = ctx->priv;
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    AVFrame *out_frame;
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    int c, i, n;
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    if (av_frame_is_writable(frame)) {
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        out_frame = frame;
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    } else {
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        out_frame = ff_get_audio_buffer(ctx->outputs[0], frame->nb_samples);
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        if (!out_frame) {
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            av_frame_free(&frame);
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            return AVERROR(ENOMEM);
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        }
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        av_frame_copy_props(out_frame, frame);
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    }
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    for (c = 0; c < inlink->channels; c++) {
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        const float *src = (const float *)frame->extended_data[c];
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        float *dst = (float *)out_frame->extended_data[c];
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        float *chorusbuf = (float *)s->chorusbuf[c];
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        int *phase = s->phase[c];
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        for (i = 0; i < frame->nb_samples; i++) {
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21935
            float out, in = src[i];
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21935
            out = in * s->in_gain;
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43870
            for (n = 0; n < s->num_chorus; n++) {
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21935
                out += chorusbuf[MOD(s->max_samples + s->counter[c] -
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                                     s->lookup_table[n][phase[n]],
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21935
                                     s->max_samples)] * s->decays[n];
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21935
                phase[n] = MOD(phase[n] + 1, s->length[n]);
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            }
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21935
            out *= s->out_gain;
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21935
            dst[i] = out;
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21935
            chorusbuf[s->counter[c]] = in;
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21935
            s->counter[c] = MOD(s->counter[c] + 1, s->max_samples);
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        }
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    }
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285
11
    s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);
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287
11
    if (frame != out_frame)
288
        av_frame_free(&frame);
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11
    return ff_filter_frame(ctx->outputs[0], out_frame);
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}
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static int request_frame(AVFilterLink *outlink)
294
{
295
10
    AVFilterContext *ctx = outlink->src;
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10
    ChorusContext *s = ctx->priv;
297
    int ret;
298
299
10
    ret = ff_request_frame(ctx->inputs[0]);
300
301

10
    if (ret == AVERROR_EOF && !ctx->is_disabled && s->fade_out) {
302
1
        int nb_samples = FFMIN(s->fade_out, 2048);
303
        AVFrame *frame;
304
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1
        frame = ff_get_audio_buffer(outlink, nb_samples);
306
1
        if (!frame)
307
            return AVERROR(ENOMEM);
308
1
        s->fade_out -= nb_samples;
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1
        av_samples_set_silence(frame->extended_data, 0,
311
                               frame->nb_samples,
312
                               outlink->channels,
313
1
                               frame->format);
314
315
1
        frame->pts = s->next_pts;
316
1
        if (s->next_pts != AV_NOPTS_VALUE)
317
1
            s->next_pts += av_rescale_q(nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
318
319
1
        ret = filter_frame(ctx->inputs[0], frame);
320
    }
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322
10
    return ret;
323
}
324
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1
static av_cold void uninit(AVFilterContext *ctx)
326
{
327
1
    ChorusContext *s = ctx->priv;
328
    int n;
329
330
1
    av_freep(&s->delays);
331
1
    av_freep(&s->decays);
332
1
    av_freep(&s->speeds);
333
1
    av_freep(&s->depths);
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335
1
    if (s->chorusbuf)
336
1
        av_freep(&s->chorusbuf[0]);
337
1
    av_freep(&s->chorusbuf);
338
339
1
    if (s->phase)
340
2
        for (n = 0; n < s->channels; n++)
341
1
            av_freep(&s->phase[n]);
342
1
    av_freep(&s->phase);
343
344
1
    av_freep(&s->counter);
345
1
    av_freep(&s->length);
346
347
1
    if (s->lookup_table)
348
2
        for (n = 0; n < s->num_chorus; n++)
349
1
            av_freep(&s->lookup_table[n]);
350
1
    av_freep(&s->lookup_table);
351
1
}
352
353
static const AVFilterPad chorus_inputs[] = {
354
    {
355
        .name         = "default",
356
        .type         = AVMEDIA_TYPE_AUDIO,
357
        .filter_frame = filter_frame,
358
    },
359
    { NULL }
360
};
361
362
static const AVFilterPad chorus_outputs[] = {
363
    {
364
        .name          = "default",
365
        .type          = AVMEDIA_TYPE_AUDIO,
366
        .request_frame = request_frame,
367
        .config_props  = config_output,
368
    },
369
    { NULL }
370
};
371
372
AVFilter ff_af_chorus = {
373
    .name          = "chorus",
374
    .description   = NULL_IF_CONFIG_SMALL("Add a chorus effect to the audio."),
375
    .query_formats = query_formats,
376
    .priv_size     = sizeof(ChorusContext),
377
    .priv_class    = &chorus_class,
378
    .init          = init,
379
    .uninit        = uninit,
380
    .inputs        = chorus_inputs,
381
    .outputs       = chorus_outputs,
382
};