/*
  * 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 min_non_zero;

     double sigma_x, sigma_x2;
     double avg_sigma_x2, min_sigma_x2, max_sigma_x2;
     double min, max;

     double nmin, nmax;

     double min_run, max_run;
     double min_runs, max_runs;

     double min_diff, max_diff;
     double diff1_sum;

     double diff1_sum_x2;

     uint64_t mask, imask;

     uint64_t min_count, max_count;
     uint64_t nb_samples;
 } ChannelStats;
 

 typedef struct AudioStatsContext {

     const AVClass *class;
     ChannelStats *chstats;
     int nb_channels;
     uint64_t tc_samples;
     double time_constant;
     double mult;

     int metadata;

     int reset_count;
     int nb_frames;

     int maxbitdepth;

 } 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 },

     { "metadata", "inject metadata in the filtergraph", OFFSET(metadata), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, FLAGS },

     { "reset", "recalculate stats after this many frames", OFFSET(reset_count), AV_OPT_TYPE_INT, {.i64=0}, 0, INT_MAX, 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_S16, AV_SAMPLE_FMT_S16P,
         AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32P,

         AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64P,

         AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,

         AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
         AV_SAMPLE_FMT_NONE
     };

     int ret;

     layouts = ff_all_channel_counts();

     if (!layouts)
         return AVERROR(ENOMEM);

     ret = ff_set_common_channel_layouts(ctx, layouts);
     if (ret < 0)
         return ret;

 
     formats = ff_make_format_list(sample_fmts);
     if (!formats)
         return AVERROR(ENOMEM);

     ret = ff_set_common_formats(ctx, formats);
     if (ret < 0)
         return ret;

 
     formats = ff_all_samplerates();
     if (!formats)
         return AVERROR(ENOMEM);

     return ff_set_common_samplerates(ctx, formats);

 }
 

 static void reset_stats(AudioStatsContext *s)
 {
     int c;
 
     for (c = 0; c < s->nb_channels; c++) {
         ChannelStats *p = &s->chstats[c];
 

         p->min = p->nmin = p->min_sigma_x2 = DBL_MAX;
         p->max = p->nmax = p->max_sigma_x2 = DBL_MIN;

         p->min_non_zero = DBL_MAX;

         p->min_diff = DBL_MAX;
         p->max_diff = DBL_MIN;
         p->sigma_x = 0;
         p->sigma_x2 = 0;
         p->avg_sigma_x2 = 0;
         p->min_run = 0;
         p->max_run = 0;
         p->min_runs = 0;
         p->max_runs = 0;
         p->diff1_sum = 0;

         p->diff1_sum_x2 = 0;

         p->mask = 0;

         p->imask = 0xFFFFFFFFFFFFFFFF;

         p->min_count = 0;
         p->max_count = 0;
         p->nb_samples = 0;

     }
 }
 

 static int config_output(AVFilterLink *outlink)
 {
     AudioStatsContext *s = outlink->src->priv;
 
     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;

     s->nb_frames = 0;

     s->maxbitdepth = av_get_bytes_per_sample(outlink->format) * 8;

     reset_stats(s);

 
     return 0;
 }
 

 static void bit_depth(AudioStatsContext *s, uint64_t mask, uint64_t imask, AVRational *depth)

 {

     unsigned result = s->maxbitdepth;

     mask = mask & (~imask);
 

     for (; result && !(mask & 1); --result, mask >>= 1);
 

     depth->den = result;
     depth->num = 0;
 
     for (; result; --result, mask >>= 1)
         if (mask & 1)
             depth->num++;

 }
 

 static inline void update_stat(AudioStatsContext *s, ChannelStats *p, double d, double nd, int64_t i)

 {
     if (d < p->min) {
         p->min = d;

         p->nmin = nd;

         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 != 0 && FFABS(d) < p->min_non_zero)
         p->min_non_zero = FFABS(d);
 

     if (d > p->max) {
         p->max = d;

         p->nmax = nd;

         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 += nd;
     p->sigma_x2 += nd * nd;
     p->avg_sigma_x2 = p->avg_sigma_x2 * s->mult + (1.0 - s->mult) * nd * nd;

     p->min_diff = FFMIN(p->min_diff, fabs(d - p->last));
     p->max_diff = FFMAX(p->max_diff, fabs(d - p->last));

     p->diff1_sum += fabs(d - p->last);

     p->diff1_sum_x2 += (d - p->last) * (d - p->last);

     p->last = d;

     p->mask |= i;
     p->imask &= i;

 
     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 void set_meta(AVDictionary **metadata, int chan, const char *key,
                      const char *fmt, double val)
 {
     uint8_t value[128];
     uint8_t key2[128];
 
     snprintf(value, sizeof(value), fmt, val);
     if (chan)
         snprintf(key2, sizeof(key2), "lavfi.astats.%d.%s", chan, key);
     else
         snprintf(key2, sizeof(key2), "lavfi.astats.%s", key);
     av_dict_set(metadata, key2, value, 0);
 }
 
 #define LINEAR_TO_DB(x) (log10(x) * 20)
 
 static void set_metadata(AudioStatsContext *s, AVDictionary **metadata)
 {

     uint64_t mask = 0, imask = 0xFFFFFFFFFFFFFFFF, min_count = 0, max_count = 0, nb_samples = 0;

     double min_runs = 0, max_runs = 0,

            min = DBL_MAX, max = DBL_MIN, min_diff = DBL_MAX, max_diff = 0,

            nmin = DBL_MAX, nmax = DBL_MIN,

            max_sigma_x = 0,

            diff1_sum = 0,

            diff1_sum_x2 = 0,

            sigma_x = 0,
            sigma_x2 = 0,
            min_sigma_x2 = DBL_MAX,
            max_sigma_x2 = DBL_MIN;

     AVRational depth;

     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);

         nmin = FFMIN(nmin, p->nmin);
         nmax = FFMAX(nmax, p->nmax);

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

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

         diff1_sum += p->diff1_sum;
         diff1_sum_x2 += p->diff1_sum_x2;

         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;

         mask |= p->mask;

         imask &= p->imask;

         nb_samples += p->nb_samples;
         if (fabs(p->sigma_x) > fabs(max_sigma_x))
             max_sigma_x = p->sigma_x;
 
         set_meta(metadata, c + 1, "DC_offset", "%f", p->sigma_x / p->nb_samples);
         set_meta(metadata, c + 1, "Min_level", "%f", p->min);
         set_meta(metadata, c + 1, "Max_level", "%f", p->max);

         set_meta(metadata, c + 1, "Min_difference", "%f", p->min_diff);

         set_meta(metadata, c + 1, "Max_difference", "%f", p->max_diff);

         set_meta(metadata, c + 1, "Mean_difference", "%f", p->diff1_sum / (p->nb_samples - 1));

         set_meta(metadata, c + 1, "RMS_difference", "%f", sqrt(p->diff1_sum_x2 / (p->nb_samples - 1)));

         set_meta(metadata, c + 1, "Peak_level", "%f", LINEAR_TO_DB(FFMAX(-p->nmin, p->nmax)));

         set_meta(metadata, c + 1, "RMS_level", "%f", LINEAR_TO_DB(sqrt(p->sigma_x2 / p->nb_samples)));
         set_meta(metadata, c + 1, "RMS_peak", "%f", LINEAR_TO_DB(sqrt(p->max_sigma_x2)));
         set_meta(metadata, c + 1, "RMS_trough", "%f", LINEAR_TO_DB(sqrt(p->min_sigma_x2)));
         set_meta(metadata, c + 1, "Crest_factor", "%f", p->sigma_x2 ? FFMAX(-p->min, p->max) / sqrt(p->sigma_x2 / p->nb_samples) : 1);
         set_meta(metadata, c + 1, "Flat_factor", "%f", LINEAR_TO_DB((p->min_runs + p->max_runs) / (p->min_count + p->max_count)));
         set_meta(metadata, c + 1, "Peak_count", "%f", (float)(p->min_count + p->max_count));

         bit_depth(s, p->mask, p->imask, &depth);

         set_meta(metadata, c + 1, "Bit_depth", "%f", depth.num);
         set_meta(metadata, c + 1, "Bit_depth2", "%f", depth.den);

         set_meta(metadata, c + 1, "Dynamic_range", "%f", LINEAR_TO_DB(2 * FFMAX(FFABS(p->min), FFABS(p->max))/ p->min_non_zero));

     }
 
     set_meta(metadata, 0, "Overall.DC_offset", "%f", max_sigma_x / (nb_samples / s->nb_channels));
     set_meta(metadata, 0, "Overall.Min_level", "%f", min);
     set_meta(metadata, 0, "Overall.Max_level", "%f", max);

     set_meta(metadata, 0, "Overall.Min_difference", "%f", min_diff);

     set_meta(metadata, 0, "Overall.Max_difference", "%f", max_diff);

     set_meta(metadata, 0, "Overall.Mean_difference", "%f", diff1_sum / (nb_samples - s->nb_channels));

     set_meta(metadata, 0, "Overall.RMS_difference", "%f", sqrt(diff1_sum_x2 / (nb_samples - s->nb_channels)));

     set_meta(metadata, 0, "Overall.Peak_level", "%f", LINEAR_TO_DB(FFMAX(-nmin, nmax)));

     set_meta(metadata, 0, "Overall.RMS_level", "%f", LINEAR_TO_DB(sqrt(sigma_x2 / nb_samples)));
     set_meta(metadata, 0, "Overall.RMS_peak", "%f", LINEAR_TO_DB(sqrt(max_sigma_x2)));
     set_meta(metadata, 0, "Overall.RMS_trough", "%f", LINEAR_TO_DB(sqrt(min_sigma_x2)));
     set_meta(metadata, 0, "Overall.Flat_factor", "%f", LINEAR_TO_DB((min_runs + max_runs) / (min_count + max_count)));
     set_meta(metadata, 0, "Overall.Peak_count", "%f", (float)(min_count + max_count) / (double)s->nb_channels);

     bit_depth(s, mask, imask, &depth);

     set_meta(metadata, 0, "Overall.Bit_depth", "%f", depth.num);
     set_meta(metadata, 0, "Overall.Bit_depth2", "%f", depth.den);

     set_meta(metadata, 0, "Overall.Number_of_samples", "%f", nb_samples / s->nb_channels);
 }
 

 static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
 {
     AudioStatsContext *s = inlink->dst->priv;

     AVDictionary **metadata = &buf->metadata;

     const int channels = s->nb_channels;
     int i, c;
 

     if (s->reset_count > 0) {
         if (s->nb_frames >= s->reset_count) {
             reset_stats(s);
             s->nb_frames = 0;
         }
         s->nb_frames++;
     }
 

     switch (inlink->format) {
     case AV_SAMPLE_FMT_DBLP:
         for (c = 0; c < channels; c++) {
             ChannelStats *p = &s->chstats[c];

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

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

                 update_stat(s, p, *src, *src, llrint(*src * (UINT64_C(1) << 63)));

         }
         break;

     case AV_SAMPLE_FMT_DBL: {
         const double *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, *src, llrint(*src * (UINT64_C(1) << 63)));
         }}
         break;
     case AV_SAMPLE_FMT_FLTP:
         for (c = 0; c < channels; c++) {
             ChannelStats *p = &s->chstats[c];
             const float *src = (const float *)buf->extended_data[c];
 
             for (i = 0; i < buf->nb_samples; i++, src++)

                 update_stat(s, p, *src, *src, llrint(*src * (UINT64_C(1) << 31)));

         }
         break;
     case AV_SAMPLE_FMT_FLT: {
         const float *src = (const float *)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, *src, llrint(*src * (UINT64_C(1) << 31)));

         }}
         break;

     case AV_SAMPLE_FMT_S64P:
         for (c = 0; c < channels; c++) {
             ChannelStats *p = &s->chstats[c];
             const int64_t *src = (const int64_t *)buf->extended_data[c];
 
             for (i = 0; i < buf->nb_samples; i++, src++)
                 update_stat(s, p, *src, *src / (double)INT64_MAX, *src);
         }
         break;
     case AV_SAMPLE_FMT_S64: {
         const int64_t *src = (const int64_t *)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, *src / (double)INT64_MAX, *src);
         }}
         break;

     case AV_SAMPLE_FMT_S32P:
         for (c = 0; c < channels; c++) {
             ChannelStats *p = &s->chstats[c];
             const int32_t *src = (const int32_t *)buf->extended_data[c];
 
             for (i = 0; i < buf->nb_samples; i++, src++)
                 update_stat(s, p, *src, *src / (double)INT32_MAX, *src);

         }
         break;

     case AV_SAMPLE_FMT_S32: {
         const int32_t *src = (const int32_t *)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, *src / (double)INT32_MAX, *src);
         }}
         break;
     case AV_SAMPLE_FMT_S16P:
         for (c = 0; c < channels; c++) {
             ChannelStats *p = &s->chstats[c];
             const int16_t *src = (const int16_t *)buf->extended_data[c];
 
             for (i = 0; i < buf->nb_samples; i++, src++)
                 update_stat(s, p, *src, *src / (double)INT16_MAX, *src);
         }
         break;
     case AV_SAMPLE_FMT_S16: {
         const int16_t *src = (const int16_t *)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, *src / (double)INT16_MAX, *src);
         }}
         break;

     }
 

     if (s->metadata)
         set_metadata(s, metadata);
 

     return ff_filter_frame(inlink->dst->outputs[0], buf);
 }
 
 static void print_stats(AVFilterContext *ctx)
 {
     AudioStatsContext *s = ctx->priv;

     uint64_t mask = 0, imask = 0xFFFFFFFFFFFFFFFF, min_count = 0, max_count = 0, nb_samples = 0;

     double min_runs = 0, max_runs = 0,

            min = DBL_MAX, max = DBL_MIN, min_diff = DBL_MAX, max_diff = 0,

            nmin = DBL_MAX, nmax = DBL_MIN,

            max_sigma_x = 0,

            diff1_sum_x2 = 0,

            diff1_sum = 0,

            sigma_x = 0,
            sigma_x2 = 0,
            min_sigma_x2 = DBL_MAX,
            max_sigma_x2 = DBL_MIN;

     AVRational depth;

     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);

         nmin = FFMIN(nmin, p->nmin);
         nmax = FFMAX(nmax, p->nmax);

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

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

         diff1_sum_x2 += p->diff1_sum_x2;
         diff1_sum += p->diff1_sum;

         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;

         mask |= p->mask;

         imask &= p->imask;

         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, "Min difference: %f\n", p->min_diff);

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

         av_log(ctx, AV_LOG_INFO, "Mean difference: %f\n", p->diff1_sum / (p->nb_samples - 1));

         av_log(ctx, AV_LOG_INFO, "RMS difference: %f\n", sqrt(p->diff1_sum_x2 / (p->nb_samples - 1)));

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

         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->nmin, p->nmax) / 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);

         bit_depth(s, p->mask, p->imask, &depth);

         av_log(ctx, AV_LOG_INFO, "Bit depth: %u/%u\n", depth.num, depth.den);

         av_log(ctx, AV_LOG_INFO, "Dynamic range: %f\n", LINEAR_TO_DB(2 * FFMAX(FFABS(p->min), FFABS(p->max))/ p->min_non_zero));

     }
 
     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, "Min difference: %f\n", min_diff);

     av_log(ctx, AV_LOG_INFO, "Max difference: %f\n", max_diff);

     av_log(ctx, AV_LOG_INFO, "Mean difference: %f\n", diff1_sum / (nb_samples - s->nb_channels));

     av_log(ctx, AV_LOG_INFO, "RMS difference: %f\n", sqrt(diff1_sum_x2 / (nb_samples - s->nb_channels)));

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

     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);

     bit_depth(s, mask, imask, &depth);

     av_log(ctx, AV_LOG_INFO, "Bit depth: %u/%u\n", depth.num, depth.den);

     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;
 

     if (s->nb_channels)
         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,
 };