/*
  * Copyright (c) 2012 Nicolas George
  *
  * 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 "libavutil/channel_layout.h"

 #include "libavutil/avassert.h"
 #include "audio.h"
 #include "avfilter.h"
 #include "internal.h"
 

 typedef struct VolDetectContext {

     /**
      * Number of samples at each PCM value.
      * histogram[0x8000 + i] is the number of samples at value i.
      * The extra element is there for symmetry.
      */
     uint64_t histogram[0x10001];
 } VolDetectContext;
 
 static int query_formats(AVFilterContext *ctx)
 {

     static const enum AVSampleFormat sample_fmts[] = {

         AV_SAMPLE_FMT_S16,
         AV_SAMPLE_FMT_S16P,
         AV_SAMPLE_FMT_NONE
     };
     AVFilterFormats *formats;

     AVFilterChannelLayouts *layouts;
     int ret;

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

 
     layouts = ff_all_channel_counts();
     if (!layouts)
         return AVERROR(ENOMEM);
     ret = ff_set_common_channel_layouts(ctx, layouts);
     if (ret < 0)
         return ret;
 

     return ff_set_common_formats(ctx, formats);

 }
 

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

 {
     AVFilterContext *ctx = inlink->dst;
     VolDetectContext *vd = ctx->priv;

     int nb_samples  = samples->nb_samples;

     int nb_channels = samples->channels;

     int nb_planes   = nb_channels;

     int plane, i;
     int16_t *pcm;
 
     if (!av_sample_fmt_is_planar(samples->format)) {
         nb_samples *= nb_channels;
         nb_planes = 1;
     }
     for (plane = 0; plane < nb_planes; plane++) {
         pcm = (int16_t *)samples->extended_data[plane];
         for (i = 0; i < nb_samples; i++)
             vd->histogram[pcm[i] + 0x8000]++;
     }
 

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

 }
 
 #define MAX_DB 91
 
 static inline double logdb(uint64_t v)
 {
     double d = v / (double)(0x8000 * 0x8000);
     if (!v)
         return MAX_DB;

     return -log10(d) * 10;

 }
 
 static void print_stats(AVFilterContext *ctx)
 {
     VolDetectContext *vd = ctx->priv;
     int i, max_volume, shift;
     uint64_t nb_samples = 0, power = 0, nb_samples_shift = 0, sum = 0;
     uint64_t histdb[MAX_DB + 1] = { 0 };
 
     for (i = 0; i < 0x10000; i++)
         nb_samples += vd->histogram[i];
     av_log(ctx, AV_LOG_INFO, "n_samples: %"PRId64"\n", nb_samples);
     if (!nb_samples)
         return;
 
     /* If nb_samples > 1<<34, there is a risk of overflow in the
        multiplication or the sum: shift all histogram values to avoid that.
        The total number of samples must be recomputed to avoid rounding
        errors. */
     shift = av_log2(nb_samples >> 33);
     for (i = 0; i < 0x10000; i++) {
         nb_samples_shift += vd->histogram[i] >> shift;
         power += (i - 0x8000) * (i - 0x8000) * (vd->histogram[i] >> shift);
     }
     if (!nb_samples_shift)
         return;
     power = (power + nb_samples_shift / 2) / nb_samples_shift;
     av_assert0(power <= 0x8000 * 0x8000);
     av_log(ctx, AV_LOG_INFO, "mean_volume: %.1f dB\n", -logdb(power));
 
     max_volume = 0x8000;
     while (max_volume > 0 && !vd->histogram[0x8000 + max_volume] &&
                              !vd->histogram[0x8000 - max_volume])
         max_volume--;
     av_log(ctx, AV_LOG_INFO, "max_volume: %.1f dB\n", -logdb(max_volume * max_volume));
 
     for (i = 0; i < 0x10000; i++)
         histdb[(int)logdb((i - 0x8000) * (i - 0x8000))] += vd->histogram[i];
     for (i = 0; i <= MAX_DB && !histdb[i]; i++);
     for (; i <= MAX_DB && sum < nb_samples / 1000; i++) {
         av_log(ctx, AV_LOG_INFO, "histogram_%ddb: %"PRId64"\n", i, histdb[i]);
         sum += histdb[i];
     }
 }
 

 static av_cold void uninit(AVFilterContext *ctx)

 {

     print_stats(ctx);

 }
 

 static const AVFilterPad volumedetect_inputs[] = {
     {

         .name         = "default",
         .type         = AVMEDIA_TYPE_AUDIO,
         .filter_frame = filter_frame,

     },
     { NULL }
 };
 
 static const AVFilterPad volumedetect_outputs[] = {
     {
         .name = "default",
         .type = AVMEDIA_TYPE_AUDIO,
     },
     { NULL }
 };
 

 AVFilter ff_af_volumedetect = {

     .name          = "volumedetect",
     .description   = NULL_IF_CONFIG_SMALL("Detect audio volume."),
     .priv_size     = sizeof(VolDetectContext),
     .query_formats = query_formats,

     .uninit        = uninit,

     .inputs        = volumedetect_inputs,
     .outputs       = volumedetect_outputs,

 };