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  1. /*
  2.  * Copyright (c) 2009 Rob Sykes <robs@users.sourceforge.net>
  3.  * Copyright (c) 2013 Paul B Mahol
  4.  *
  5.  * This file is part of FFmpeg.
  6.  *
  7.  * FFmpeg is free software; you can redistribute it and/or
  8.  * modify it under the terms of the GNU Lesser General Public
  9.  * License as published by the Free Software Foundation; either
  10.  * version 2.1 of the License, or (at your option) any later version.
  11.  *
  12.  * FFmpeg is distributed in the hope that it will be useful,
  13.  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  14.  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  15.  * Lesser General Public License for more details.
  16.  *
  17.  * You should have received a copy of the GNU Lesser General Public
  18.  * License along with FFmpeg; if not, write to the Free Software
  19.  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  20.  */
  21.  
  22. #include <float.h>
  23.  
  24. #include "libavutil/opt.h"
  25. #include "audio.h"
  26. #include "avfilter.h"
  27. #include "internal.h"
  28.  
  29. typedef struct ChannelStats {
  30.     double last;
  31.     double sigma_x, sigma_x2;
  32.     double avg_sigma_x2, min_sigma_x2, max_sigma_x2;
  33.     double min, max;
  34.     double min_run, max_run;
  35.     double min_runs, max_runs;
  36.     uint64_t min_count, max_count;
  37.     uint64_t nb_samples;
  38. } ChannelStats;
  39.  
  40. typedef struct {
  41.     const AVClass *class;
  42.     ChannelStats *chstats;
  43.     int nb_channels;
  44.     uint64_t tc_samples;
  45.     double time_constant;
  46.     double mult;
  47. } AudioStatsContext;
  48.  
  49. #define OFFSET(x) offsetof(AudioStatsContext, x)
  50. #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
  51.  
  52. static const AVOption astats_options[] = {
  53.     { "length", "set the window length", OFFSET(time_constant), AV_OPT_TYPE_DOUBLE, {.dbl=.05}, .01, 10, FLAGS },
  54.     { NULL }
  55. };
  56.  
  57. AVFILTER_DEFINE_CLASS(astats);
  58.  
  59. static int query_formats(AVFilterContext *ctx)
  60. {
  61.     AVFilterFormats *formats;
  62.     AVFilterChannelLayouts *layouts;
  63.     static const enum AVSampleFormat sample_fmts[] = {
  64.         AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
  65.         AV_SAMPLE_FMT_NONE
  66.     };
  67.  
  68.     layouts = ff_all_channel_layouts();
  69.     if (!layouts)
  70.         return AVERROR(ENOMEM);
  71.     ff_set_common_channel_layouts(ctx, layouts);
  72.  
  73.     formats = ff_make_format_list(sample_fmts);
  74.     if (!formats)
  75.         return AVERROR(ENOMEM);
  76.     ff_set_common_formats(ctx, formats);
  77.  
  78.     formats = ff_all_samplerates();
  79.     if (!formats)
  80.         return AVERROR(ENOMEM);
  81.     ff_set_common_samplerates(ctx, formats);
  82.  
  83.     return 0;
  84. }
  85.  
  86. static int config_output(AVFilterLink *outlink)
  87. {
  88.     AudioStatsContext *s = outlink->src->priv;
  89.     int c;
  90.  
  91.     s->chstats = av_calloc(sizeof(*s->chstats), outlink->channels);
  92.     if (!s->chstats)
  93.         return AVERROR(ENOMEM);
  94.     s->nb_channels = outlink->channels;
  95.     s->mult = exp((-1 / s->time_constant / outlink->sample_rate));
  96.     s->tc_samples = 5 * s->time_constant * outlink->sample_rate + .5;
  97.  
  98.     for (c = 0; c < s->nb_channels; c++) {
  99.         ChannelStats *p = &s->chstats[c];
  100.  
  101.         p->min = p->min_sigma_x2 = DBL_MAX;
  102.         p->max = p->max_sigma_x2 = DBL_MIN;
  103.     }
  104.  
  105.     return 0;
  106. }
  107.  
  108. static inline void update_stat(AudioStatsContext *s, ChannelStats *p, double d)
  109. {
  110.     if (d < p->min) {
  111.         p->min = d;
  112.         p->min_run = 1;
  113.         p->min_runs = 0;
  114.         p->min_count = 1;
  115.     } else if (d == p->min) {
  116.         p->min_count++;
  117.         p->min_run = d == p->last ? p->min_run + 1 : 1;
  118.     } else if (p->last == p->min) {
  119.         p->min_runs += p->min_run * p->min_run;
  120.     }
  121.  
  122.     if (d > p->max) {
  123.         p->max = d;
  124.         p->max_run = 1;
  125.         p->max_runs = 0;
  126.         p->max_count = 1;
  127.     } else if (d == p->max) {
  128.         p->max_count++;
  129.         p->max_run = d == p->last ? p->max_run + 1 : 1;
  130.     } else if (p->last == p->max) {
  131.         p->max_runs += p->max_run * p->max_run;
  132.     }
  133.  
  134.     p->sigma_x += d;
  135.     p->sigma_x2 += d * d;
  136.     p->avg_sigma_x2 = p->avg_sigma_x2 * s->mult + (1.0 - s->mult) * d * d;
  137.     p->last = d;
  138.  
  139.     if (p->nb_samples >= s->tc_samples) {
  140.         p->max_sigma_x2 = FFMAX(p->max_sigma_x2, p->avg_sigma_x2);
  141.         p->min_sigma_x2 = FFMIN(p->min_sigma_x2, p->avg_sigma_x2);
  142.     }
  143.     p->nb_samples++;
  144. }
  145.  
  146. static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
  147. {
  148.     AudioStatsContext *s = inlink->dst->priv;
  149.     const int channels = s->nb_channels;
  150.     const double *src;
  151.     int i, c;
  152.  
  153.     switch (inlink->format) {
  154.     case AV_SAMPLE_FMT_DBLP:
  155.         for (c = 0; c < channels; c++) {
  156.             ChannelStats *p = &s->chstats[c];
  157.             src = (const double *)buf->extended_data[c];
  158.  
  159.             for (i = 0; i < buf->nb_samples; i++, src++)
  160.                 update_stat(s, p, *src);
  161.         }
  162.         break;
  163.     case AV_SAMPLE_FMT_DBL:
  164.         src = (const double *)buf->extended_data[0];
  165.  
  166.         for (i = 0; i < buf->nb_samples; i++) {
  167.             for (c = 0; c < channels; c++, src++)
  168.                 update_stat(s, &s->chstats[c], *src);
  169.         }
  170.         break;
  171.     }
  172.  
  173.     return ff_filter_frame(inlink->dst->outputs[0], buf);
  174. }
  175.  
  176. #define LINEAR_TO_DB(x) (log10(x) * 20)
  177.  
  178. static void print_stats(AVFilterContext *ctx)
  179. {
  180.     AudioStatsContext *s = ctx->priv;
  181.     uint64_t min_count = 0, max_count = 0, nb_samples = 0;
  182.     double min_runs = 0, max_runs = 0,
  183.            min = DBL_MAX, max = DBL_MIN,
  184.            max_sigma_x = 0,
  185.            sigma_x = 0,
  186.            sigma_x2 = 0,
  187.            min_sigma_x2 = DBL_MAX,
  188.            max_sigma_x2 = DBL_MIN;
  189.     int c;
  190.  
  191.     for (c = 0; c < s->nb_channels; c++) {
  192.         ChannelStats *p = &s->chstats[c];
  193.  
  194.         if (p->nb_samples < s->tc_samples)
  195.             p->min_sigma_x2 = p->max_sigma_x2 = p->sigma_x2 / p->nb_samples;
  196.  
  197.         min = FFMIN(min, p->min);
  198.         max = FFMAX(max, p->max);
  199.         min_sigma_x2 = FFMIN(min_sigma_x2, p->min_sigma_x2);
  200.         max_sigma_x2 = FFMAX(max_sigma_x2, p->max_sigma_x2);
  201.         sigma_x += p->sigma_x;
  202.         sigma_x2 += p->sigma_x2;
  203.         min_count += p->min_count;
  204.         max_count += p->max_count;
  205.         min_runs += p->min_runs;
  206.         max_runs += p->max_runs;
  207.         nb_samples += p->nb_samples;
  208.         if (fabs(p->sigma_x) > fabs(max_sigma_x))
  209.             max_sigma_x = p->sigma_x;
  210.  
  211.         av_log(ctx, AV_LOG_INFO, "Channel: %d\n", c + 1);
  212.         av_log(ctx, AV_LOG_INFO, "DC offset: %f\n", p->sigma_x / p->nb_samples);
  213.         av_log(ctx, AV_LOG_INFO, "Min level: %f\n", p->min);
  214.         av_log(ctx, AV_LOG_INFO, "Max level: %f\n", p->max);
  215.         av_log(ctx, AV_LOG_INFO, "Peak level dB: %f\n", LINEAR_TO_DB(FFMAX(-p->min, p->max)));
  216.         av_log(ctx, AV_LOG_INFO, "RMS level dB: %f\n", LINEAR_TO_DB(sqrt(p->sigma_x2 / p->nb_samples)));
  217.         av_log(ctx, AV_LOG_INFO, "RMS peak dB: %f\n", LINEAR_TO_DB(sqrt(p->max_sigma_x2)));
  218.         if (p->min_sigma_x2 != 1)
  219.             av_log(ctx, AV_LOG_INFO, "RMS trough dB: %f\n",LINEAR_TO_DB(sqrt(p->min_sigma_x2)));
  220.         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);
  221.         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)));
  222.         av_log(ctx, AV_LOG_INFO, "Peak count: %"PRId64"\n", p->min_count + p->max_count);
  223.     }
  224.  
  225.     av_log(ctx, AV_LOG_INFO, "Overall\n");
  226.     av_log(ctx, AV_LOG_INFO, "DC offset: %f\n", max_sigma_x / (nb_samples / s->nb_channels));
  227.     av_log(ctx, AV_LOG_INFO, "Min level: %f\n", min);
  228.     av_log(ctx, AV_LOG_INFO, "Max level: %f\n", max);
  229.     av_log(ctx, AV_LOG_INFO, "Peak level dB: %f\n", LINEAR_TO_DB(FFMAX(-min, max)));
  230.     av_log(ctx, AV_LOG_INFO, "RMS level dB: %f\n", LINEAR_TO_DB(sqrt(sigma_x2 / nb_samples)));
  231.     av_log(ctx, AV_LOG_INFO, "RMS peak dB: %f\n", LINEAR_TO_DB(sqrt(max_sigma_x2)));
  232.     if (min_sigma_x2 != 1)
  233.         av_log(ctx, AV_LOG_INFO, "RMS trough dB: %f\n", LINEAR_TO_DB(sqrt(min_sigma_x2)));
  234.     av_log(ctx, AV_LOG_INFO, "Flat factor: %f\n", LINEAR_TO_DB((min_runs + max_runs) / (min_count + max_count)));
  235.     av_log(ctx, AV_LOG_INFO, "Peak count: %f\n", (min_count + max_count) / (double)s->nb_channels);
  236.     av_log(ctx, AV_LOG_INFO, "Number of samples: %"PRId64"\n", nb_samples / s->nb_channels);
  237. }
  238.  
  239. static av_cold void uninit(AVFilterContext *ctx)
  240. {
  241.     AudioStatsContext *s = ctx->priv;
  242.  
  243.     print_stats(ctx);
  244.     av_freep(&s->chstats);
  245. }
  246.  
  247. static const AVFilterPad astats_inputs[] = {
  248.     {
  249.         .name         = "default",
  250.         .type         = AVMEDIA_TYPE_AUDIO,
  251.         .filter_frame = filter_frame,
  252.     },
  253.     { NULL }
  254. };
  255.  
  256. static const AVFilterPad astats_outputs[] = {
  257.     {
  258.         .name         = "default",
  259.         .type         = AVMEDIA_TYPE_AUDIO,
  260.         .config_props = config_output,
  261.     },
  262.     { NULL }
  263. };
  264.  
  265. AVFilter avfilter_af_astats = {
  266.     .name          = "astats",
  267.     .description   = NULL_IF_CONFIG_SMALL("Show time domain statistics about audio frames."),
  268.     .query_formats = query_formats,
  269.     .priv_size     = sizeof(AudioStatsContext),
  270.     .priv_class    = &astats_class,
  271.     .uninit        = uninit,
  272.     .inputs        = astats_inputs,
  273.     .outputs       = astats_outputs,
  274. };
  275.