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/*

 * Copyright (c) 2009 Rob Sykes 

 * Copyright (c) 2013 Paul B Mahol

 *

 * This file is part of FFmpeg.

 *

 * FFmpeg is free software; you can redistribute it and/or

 * modify it under the terms of the GNU Lesser General Public

 * License as published by the Free Software Foundation; either

 * version 2.1 of the License, or (at your option) any later version.

 *

 * FFmpeg is distributed in the hope that it will be useful,

 * but WITHOUT ANY WARRANTY; without even the implied warranty of

 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU

 * Lesser General Public License for more details.

 *

 * You should have received a copy of the GNU Lesser General Public

 * License along with FFmpeg; if not, write to the Free Software

 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA

 */

#include 

#include "libavutil/opt.h"

#include "audio.h"

#include "avfilter.h"

#include "internal.h"

typedef struct ChannelStats {

    double last;

    double sigma_x, sigma_x2;

    double avg_sigma_x2, min_sigma_x2, max_sigma_x2;

    double min, max;

    double min_run, max_run;

    double min_runs, max_runs;

    uint64_t min_count, max_count;

    uint64_t nb_samples;

} ChannelStats;

typedef struct {

    const AVClass *class;

    ChannelStats *chstats;

    int nb_channels;

    uint64_t tc_samples;

    double time_constant;

    double mult;

} AudioStatsContext;

#define OFFSET(x) offsetof(AudioStatsContext, x)

#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM

static const AVOption astats_options[] = {

    { "length", "set the window length", OFFSET(time_constant), AV_OPT_TYPE_DOUBLE, {.dbl=.05}, .01, 10, FLAGS },

    { NULL }

};

AVFILTER_DEFINE_CLASS(astats);

static int query_formats(AVFilterContext *ctx)

{

    AVFilterFormats *formats;

    AVFilterChannelLayouts *layouts;

    static const enum AVSampleFormat sample_fmts[] = {

        AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,

        AV_SAMPLE_FMT_NONE

    };

    layouts = ff_all_channel_layouts();

    if (!layouts)

        return AVERROR(ENOMEM);

    ff_set_common_channel_layouts(ctx, layouts);

    formats = ff_make_format_list(sample_fmts);

    if (!formats)

        return AVERROR(ENOMEM);

    ff_set_common_formats(ctx, formats);

    formats = ff_all_samplerates();

    if (!formats)

        return AVERROR(ENOMEM);

    ff_set_common_samplerates(ctx, formats);

    return 0;

}

static int config_output(AVFilterLink *outlink)

{

    AudioStatsContext *s = outlink->src->priv;

    int c;

    s->chstats = av_calloc(sizeof(*s->chstats), outlink->channels);

    if (!s->chstats)

        return AVERROR(ENOMEM);

    s->nb_channels = outlink->channels;

    s->mult = exp((-1 / s->time_constant / outlink->sample_rate));

    s->tc_samples = 5 * s->time_constant * outlink->sample_rate + .5;

    for (c = 0; c < s->nb_channels; c++) {

        ChannelStats *p = &s->chstats[c];

        p->min = p->min_sigma_x2 = DBL_MAX;

        p->max = p->max_sigma_x2 = DBL_MIN;

    }

    return 0;

}

static inline void update_stat(AudioStatsContext *s, ChannelStats *p, double d)

{

    if (d < p->min) {

        p->min = d;

        p->min_run = 1;

        p->min_runs = 0;

        p->min_count = 1;

    } else if (d == p->min) {

        p->min_count++;

        p->min_run = d == p->last ? p->min_run + 1 : 1;

    } else if (p->last == p->min) {

        p->min_runs += p->min_run * p->min_run;

    }

    if (d > p->max) {

        p->max = d;

        p->max_run = 1;

        p->max_runs = 0;

        p->max_count = 1;

    } else if (d == p->max) {

        p->max_count++;

        p->max_run = d == p->last ? p->max_run + 1 : 1;

    } else if (p->last == p->max) {

        p->max_runs += p->max_run * p->max_run;

    }

    p->sigma_x += d;

    p->sigma_x2 += d * d;

    p->avg_sigma_x2 = p->avg_sigma_x2 * s->mult + (1.0 - s->mult) * d * d;

    p->last = d;

    if (p->nb_samples >= s->tc_samples) {

        p->max_sigma_x2 = FFMAX(p->max_sigma_x2, p->avg_sigma_x2);

        p->min_sigma_x2 = FFMIN(p->min_sigma_x2, p->avg_sigma_x2);

    }

    p->nb_samples++;

}

static int filter_frame(AVFilterLink *inlink, AVFrame *buf)

{

    AudioStatsContext *s = inlink->dst->priv;

    const int channels = s->nb_channels;

    const double *src;

    int i, c;

    switch (inlink->format) {

    case AV_SAMPLE_FMT_DBLP:

        for (c = 0; c < channels; c++) {

            ChannelStats *p = &s->chstats[c];

            src = (const double *)buf->extended_data[c];

            for (i = 0; i < buf->nb_samples; i++, src++)

                update_stat(s, p, *src);

        }

        break;

    case AV_SAMPLE_FMT_DBL:

        src = (const double *)buf->extended_data[0];

        for (i = 0; i < buf->nb_samples; i++) {

            for (c = 0; c < channels; c++, src++)

                update_stat(s, &s->chstats[c], *src);

        }

        break;

    }

    return ff_filter_frame(inlink->dst->outputs[0], buf);

}

#define LINEAR_TO_DB(x) (log10(x) * 20)

static void print_stats(AVFilterContext *ctx)

{

    AudioStatsContext *s = ctx->priv;

    uint64_t min_count = 0, max_count = 0, nb_samples = 0;

    double min_runs = 0, max_runs = 0,

           min = DBL_MAX, max = DBL_MIN,

           max_sigma_x = 0,

           sigma_x = 0,

           sigma_x2 = 0,

           min_sigma_x2 = DBL_MAX,

           max_sigma_x2 = DBL_MIN;

    int c;

    for (c = 0; c < s->nb_channels; c++) {

        ChannelStats *p = &s->chstats[c];

        if (p->nb_samples < s->tc_samples)

            p->min_sigma_x2 = p->max_sigma_x2 = p->sigma_x2 / p->nb_samples;

        min = FFMIN(min, p->min);

        max = FFMAX(max, p->max);

        min_sigma_x2 = FFMIN(min_sigma_x2, p->min_sigma_x2);

        max_sigma_x2 = FFMAX(max_sigma_x2, p->max_sigma_x2);

        sigma_x += p->sigma_x;

        sigma_x2 += p->sigma_x2;

        min_count += p->min_count;

        max_count += p->max_count;

        min_runs += p->min_runs;

        max_runs += p->max_runs;

        nb_samples += p->nb_samples;

        if (fabs(p->sigma_x) > fabs(max_sigma_x))

            max_sigma_x = p->sigma_x;

        av_log(ctx, AV_LOG_INFO, "Channel: %d\n", c + 1);

        av_log(ctx, AV_LOG_INFO, "DC offset: %f\n", p->sigma_x / p->nb_samples);

        av_log(ctx, AV_LOG_INFO, "Min level: %f\n", p->min);

        av_log(ctx, AV_LOG_INFO, "Max level: %f\n", p->max);

        av_log(ctx, AV_LOG_INFO, "Peak level dB: %f\n", LINEAR_TO_DB(FFMAX(-p->min, p->max)));

        av_log(ctx, AV_LOG_INFO, "RMS level dB: %f\n", LINEAR_TO_DB(sqrt(p->sigma_x2 / p->nb_samples)));

        av_log(ctx, AV_LOG_INFO, "RMS peak dB: %f\n", LINEAR_TO_DB(sqrt(p->max_sigma_x2)));

        if (p->min_sigma_x2 != 1)

            av_log(ctx, AV_LOG_INFO, "RMS trough dB: %f\n",LINEAR_TO_DB(sqrt(p->min_sigma_x2)));

        av_log(ctx, AV_LOG_INFO, "Crest factor: %f\n", p->sigma_x2 ? FFMAX(-p->min, p->max) / sqrt(p->sigma_x2 / p->nb_samples) : 1);

        av_log(ctx, AV_LOG_INFO, "Flat factor: %f\n", LINEAR_TO_DB((p->min_runs + p->max_runs) / (p->min_count + p->max_count)));

        av_log(ctx, AV_LOG_INFO, "Peak count: %"PRId64"\n", p->min_count + p->max_count);

    }

    av_log(ctx, AV_LOG_INFO, "Overall\n");

    av_log(ctx, AV_LOG_INFO, "DC offset: %f\n", max_sigma_x / (nb_samples / s->nb_channels));

    av_log(ctx, AV_LOG_INFO, "Min level: %f\n", min);

    av_log(ctx, AV_LOG_INFO, "Max level: %f\n", max);

    av_log(ctx, AV_LOG_INFO, "Peak level dB: %f\n", LINEAR_TO_DB(FFMAX(-min, max)));

    av_log(ctx, AV_LOG_INFO, "RMS level dB: %f\n", LINEAR_TO_DB(sqrt(sigma_x2 / nb_samples)));

    av_log(ctx, AV_LOG_INFO, "RMS peak dB: %f\n", LINEAR_TO_DB(sqrt(max_sigma_x2)));

    if (min_sigma_x2 != 1)

        av_log(ctx, AV_LOG_INFO, "RMS trough dB: %f\n", LINEAR_TO_DB(sqrt(min_sigma_x2)));

    av_log(ctx, AV_LOG_INFO, "Flat factor: %f\n", LINEAR_TO_DB((min_runs + max_runs) / (min_count + max_count)));

    av_log(ctx, AV_LOG_INFO, "Peak count: %f\n", (min_count + max_count) / (double)s->nb_channels);

    av_log(ctx, AV_LOG_INFO, "Number of samples: %"PRId64"\n", nb_samples / s->nb_channels);

}

static av_cold void uninit(AVFilterContext *ctx)

{

    AudioStatsContext *s = ctx->priv;

    print_stats(ctx);

    av_freep(&s->chstats);

}

static const AVFilterPad astats_inputs[] = {

    {

        .name         = "default",

        .type         = AVMEDIA_TYPE_AUDIO,

        .filter_frame = filter_frame,

    },

    { NULL }

};

static const AVFilterPad astats_outputs[] = {

    {

        .name         = "default",

        .type         = AVMEDIA_TYPE_AUDIO,

        .config_props = config_output,

    },

    { NULL }

};

AVFilter avfilter_af_astats = {

    .name          = "astats",

    .description   = NULL_IF_CONFIG_SMALL("Show time domain statistics about audio frames."),

    .query_formats = query_formats,

    .priv_size     = sizeof(AudioStatsContext),

    .priv_class    = &astats_class,

    .uninit        = uninit,

    .inputs        = astats_inputs,

    .outputs       = astats_outputs,

};