/*
  * Copyright (c) 2009 Rob Sykes <robs@users.sourceforge.net>
  * 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 <float.h>
 
 #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 ff_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,
 };