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FFmpeg/libavfilter/af_astats.c

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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 min_non_zero;

  32.     double sigma_x, sigma_x2;

  33.     double avg_sigma_x2, min_sigma_x2, max_sigma_x2;

  34.     double min, max;

  35.     double nmin, nmax;

  36.     double min_run, max_run;

  37.     double min_runs, max_runs;

  38.     double min_diff, max_diff;

  39.     double diff1_sum;

  40.     double diff1_sum_x2;

  41.     uint64_t mask, imask;

  42.     uint64_t min_count, max_count;

  43.     uint64_t nb_samples;

  44. } ChannelStats;

  45. 

  46. typedef struct AudioStatsContext {

  47.     const AVClass *class;

  48.     ChannelStats *chstats;

  49.     int nb_channels;

  50.     uint64_t tc_samples;

  51.     double time_constant;

  52.     double mult;

  53.     int metadata;

  54.     int reset_count;

  55.     int nb_frames;

  56.     int maxbitdepth;

  57. } AudioStatsContext;

  58. 

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

  60. #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM

  61. 

  62. static const AVOption astats_options[] = {

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

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

  65.     { "reset", "recalculate stats after this many frames", OFFSET(reset_count), AV_OPT_TYPE_INT, {.i64=0}, 0, INT_MAX, FLAGS },

  66.     { NULL }

  67. };

  68. 

  69. AVFILTER_DEFINE_CLASS(astats);

  70. 

  71. static int query_formats(AVFilterContext *ctx)

  72. {

  73.     AVFilterFormats *formats;

  74.     AVFilterChannelLayouts *layouts;

  75.     static const enum AVSampleFormat sample_fmts[] = {

  76.         AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P,

  77.         AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32P,

  78.         AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64P,

  79.         AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,

  80.         AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,

  81.         AV_SAMPLE_FMT_NONE

  82.     };

  83.     int ret;

  84. 

  85.     layouts = ff_all_channel_counts();

  86.     if (!layouts)

  87.         return AVERROR(ENOMEM);

  88.     ret = ff_set_common_channel_layouts(ctx, layouts);

  89.     if (ret < 0)

  90.         return ret;

  91. 

  92.     formats = ff_make_format_list(sample_fmts);

  93.     if (!formats)

  94.         return AVERROR(ENOMEM);

  95.     ret = ff_set_common_formats(ctx, formats);

  96.     if (ret < 0)

  97.         return ret;

  98. 

  99.     formats = ff_all_samplerates();

  100.     if (!formats)

  101.         return AVERROR(ENOMEM);

  102.     return ff_set_common_samplerates(ctx, formats);

  103. }

  104. 

  105. static void reset_stats(AudioStatsContext *s)

  106. {

  107.     int c;

  108. 

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

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

  111. 

  112.         p->min = p->nmin = p->min_sigma_x2 = DBL_MAX;

  113.         p->max = p->nmax = p->max_sigma_x2 = DBL_MIN;

  114.         p->min_non_zero = DBL_MAX;

  115.         p->min_diff = DBL_MAX;

  116.         p->max_diff = DBL_MIN;

  117.         p->sigma_x = 0;

  118.         p->sigma_x2 = 0;

  119.         p->avg_sigma_x2 = 0;

  120.         p->min_run = 0;

  121.         p->max_run = 0;

  122.         p->min_runs = 0;

  123.         p->max_runs = 0;

  124.         p->diff1_sum = 0;

  125.         p->diff1_sum_x2 = 0;

  126.         p->mask = 0;

  127.         p->imask = 0xFFFFFFFFFFFFFFFF;

  128.         p->min_count = 0;

  129.         p->max_count = 0;

  130.         p->nb_samples = 0;

  131.     }

  132. }

  133. 

  134. static int config_output(AVFilterLink *outlink)

  135. {

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

  137. 

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

  139.     if (!s->chstats)

  140.         return AVERROR(ENOMEM);

  141.     s->nb_channels = outlink->channels;

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

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

  144.     s->nb_frames = 0;

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

  146. 

  147.     reset_stats(s);

  148. 

  149.     return 0;

  150. }

  151. 

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

  153. {

  154.     unsigned result = s->maxbitdepth;

  155. 

  156.     mask = mask & (~imask);

  157. 

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

  159. 

  160.     depth->den = result;

  161.     depth->num = 0;

  162. 

  163.     for (; result; --result, mask >>= 1)

  164.         if (mask & 1)

  165.             depth->num++;

  166. }

  167. 

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

  169. {

  170.     if (d < p->min) {

  171.         p->min = d;

  172.         p->nmin = nd;

  173.         p->min_run = 1;

  174.         p->min_runs = 0;

  175.         p->min_count = 1;

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

  177.         p->min_count++;

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

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

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

  181.     }

  182. 

  183.     if (d != 0 && FFABS(d) < p->min_non_zero)

  184.         p->min_non_zero = FFABS(d);

  185. 

  186.     if (d > p->max) {

  187.         p->max = d;

  188.         p->nmax = nd;

  189.         p->max_run = 1;

  190.         p->max_runs = 0;

  191.         p->max_count = 1;

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

  193.         p->max_count++;

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

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

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

  197.     }

  198. 

  199.     p->sigma_x += nd;

  200.     p->sigma_x2 += nd * nd;

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

  202.     p->min_diff = FFMIN(p->min_diff, fabs(d - p->last));

  203.     p->max_diff = FFMAX(p->max_diff, fabs(d - p->last));

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

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

  206.     p->last = d;

  207.     p->mask |= i;

  208.     p->imask &= i;

  209. 

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

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

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

  213.     }

  214.     p->nb_samples++;

  215. }

  216. 

  217. static void set_meta(AVDictionary **metadata, int chan, const char *key,

  218.                      const char *fmt, double val)

  219. {

  220.     uint8_t value[128];

  221.     uint8_t key2[128];

  222. 

  223.     snprintf(value, sizeof(value), fmt, val);

  224.     if (chan)

  225.         snprintf(key2, sizeof(key2), "lavfi.astats.%d.%s", chan, key);

  226.     else

  227.         snprintf(key2, sizeof(key2), "lavfi.astats.%s", key);

  228.     av_dict_set(metadata, key2, value, 0);

  229. }

  230. 

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

  232. 

  233. static void set_metadata(AudioStatsContext *s, AVDictionary **metadata)

  234. {

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

  236.     double min_runs = 0, max_runs = 0,

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

  238.            nmin = DBL_MAX, nmax = DBL_MIN,

  239.            max_sigma_x = 0,

  240.            diff1_sum = 0,

  241.            diff1_sum_x2 = 0,

  242.            sigma_x = 0,

  243.            sigma_x2 = 0,

  244.            min_sigma_x2 = DBL_MAX,

  245.            max_sigma_x2 = DBL_MIN;

  246.     AVRational depth;

  247.     int c;

  248. 

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

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

  251. 

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

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

  254. 

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

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

  257.         nmin = FFMIN(nmin, p->nmin);

  258.         nmax = FFMAX(nmax, p->nmax);

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

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

  261.         diff1_sum += p->diff1_sum;

  262.         diff1_sum_x2 += p->diff1_sum_x2;

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

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

  265.         sigma_x += p->sigma_x;

  266.         sigma_x2 += p->sigma_x2;

  267.         min_count += p->min_count;

  268.         max_count += p->max_count;

  269.         min_runs += p->min_runs;

  270.         max_runs += p->max_runs;

  271.         mask |= p->mask;

  272.         imask &= p->imask;

  273.         nb_samples += p->nb_samples;

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

  275.             max_sigma_x = p->sigma_x;

  276. 

  277.         set_meta(metadata, c + 1, "DC_offset", "%f", p->sigma_x / p->nb_samples);

  278.         set_meta(metadata, c + 1, "Min_level", "%f", p->min);

  279.         set_meta(metadata, c + 1, "Max_level", "%f", p->max);

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

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

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

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

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

  285.         set_meta(metadata, c + 1, "RMS_level", "%f", LINEAR_TO_DB(sqrt(p->sigma_x2 / p->nb_samples)));

  286.         set_meta(metadata, c + 1, "RMS_peak", "%f", LINEAR_TO_DB(sqrt(p->max_sigma_x2)));

  287.         set_meta(metadata, c + 1, "RMS_trough", "%f", LINEAR_TO_DB(sqrt(p->min_sigma_x2)));

  288.         set_meta(metadata, c + 1, "Crest_factor", "%f", p->sigma_x2 ? FFMAX(-p->min, p->max) / sqrt(p->sigma_x2 / p->nb_samples) : 1);

  289.         set_meta(metadata, c + 1, "Flat_factor", "%f", LINEAR_TO_DB((p->min_runs + p->max_runs) / (p->min_count + p->max_count)));

  290.         set_meta(metadata, c + 1, "Peak_count", "%f", (float)(p->min_count + p->max_count));

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

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

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

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

  295.     }

  296. 

  297.     set_meta(metadata, 0, "Overall.DC_offset", "%f", max_sigma_x / (nb_samples / s->nb_channels));

  298.     set_meta(metadata, 0, "Overall.Min_level", "%f", min);

  299.     set_meta(metadata, 0, "Overall.Max_level", "%f", max);

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

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

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

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

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

  305.     set_meta(metadata, 0, "Overall.RMS_level", "%f", LINEAR_TO_DB(sqrt(sigma_x2 / nb_samples)));

  306.     set_meta(metadata, 0, "Overall.RMS_peak", "%f", LINEAR_TO_DB(sqrt(max_sigma_x2)));

  307.     set_meta(metadata, 0, "Overall.RMS_trough", "%f", LINEAR_TO_DB(sqrt(min_sigma_x2)));

  308.     set_meta(metadata, 0, "Overall.Flat_factor", "%f", LINEAR_TO_DB((min_runs + max_runs) / (min_count + max_count)));

  309.     set_meta(metadata, 0, "Overall.Peak_count", "%f", (float)(min_count + max_count) / (double)s->nb_channels);

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

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

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

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

  314. }

  315. 

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

  317. {

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

  319.     AVDictionary **metadata = &buf->metadata;

  320.     const int channels = s->nb_channels;

  321.     int i, c;

  322. 

  323.     if (s->reset_count > 0) {

  324.         if (s->nb_frames >= s->reset_count) {

  325.             reset_stats(s);

  326.             s->nb_frames = 0;

  327.         }

  328.         s->nb_frames++;

  329.     }

  330. 

  331.     switch (inlink->format) {

  332.     case AV_SAMPLE_FMT_DBLP:

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

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

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

  336. 

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

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

  339.         }

  340.         break;

  341.     case AV_SAMPLE_FMT_DBL: {

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

  343. 

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

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

  346.                 update_stat(s, &s->chstats[c], *src, *src, llrint(*src * (UINT64_C(1) << 63)));

  347.         }}

  348.         break;

  349.     case AV_SAMPLE_FMT_FLTP:

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

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

  352.             const float *src = (const float *)buf->extended_data[c];

  353. 

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

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

  356.         }

  357.         break;

  358.     case AV_SAMPLE_FMT_FLT: {

  359.         const float *src = (const float *)buf->extended_data[0];

  360. 

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

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

  363.                 update_stat(s, &s->chstats[c], *src, *src, llrint(*src * (UINT64_C(1) << 31)));

  364.         }}

  365.         break;

  366.     case AV_SAMPLE_FMT_S64P:

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

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

  369.             const int64_t *src = (const int64_t *)buf->extended_data[c];

  370. 

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

  372.                 update_stat(s, p, *src, *src / (double)INT64_MAX, *src);

  373.         }

  374.         break;

  375.     case AV_SAMPLE_FMT_S64: {

  376.         const int64_t *src = (const int64_t *)buf->extended_data[0];

  377. 

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

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

  380.                 update_stat(s, &s->chstats[c], *src, *src / (double)INT64_MAX, *src);

  381.         }}

  382.         break;

  383.     case AV_SAMPLE_FMT_S32P:

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

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

  386.             const int32_t *src = (const int32_t *)buf->extended_data[c];

  387. 

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

  389.                 update_stat(s, p, *src, *src / (double)INT32_MAX, *src);

  390.         }

  391.         break;

  392.     case AV_SAMPLE_FMT_S32: {

  393.         const int32_t *src = (const int32_t *)buf->extended_data[0];

  394. 

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

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

  397.                 update_stat(s, &s->chstats[c], *src, *src / (double)INT32_MAX, *src);

  398.         }}

  399.         break;

  400.     case AV_SAMPLE_FMT_S16P:

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

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

  403.             const int16_t *src = (const int16_t *)buf->extended_data[c];

  404. 

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

  406.                 update_stat(s, p, *src, *src / (double)INT16_MAX, *src);

  407.         }

  408.         break;

  409.     case AV_SAMPLE_FMT_S16: {

  410.         const int16_t *src = (const int16_t *)buf->extended_data[0];

  411. 

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

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

  414.                 update_stat(s, &s->chstats[c], *src, *src / (double)INT16_MAX, *src);

  415.         }}

  416.         break;

  417.     }

  418. 

  419.     if (s->metadata)

  420.         set_metadata(s, metadata);

  421. 

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

  423. }

  424. 

  425. static void print_stats(AVFilterContext *ctx)

  426. {

  427.     AudioStatsContext *s = ctx->priv;

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

  429.     double min_runs = 0, max_runs = 0,

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

  431.            nmin = DBL_MAX, nmax = DBL_MIN,

  432.            max_sigma_x = 0,

  433.            diff1_sum_x2 = 0,

  434.            diff1_sum = 0,

  435.            sigma_x = 0,

  436.            sigma_x2 = 0,

  437.            min_sigma_x2 = DBL_MAX,

  438.            max_sigma_x2 = DBL_MIN;

  439.     AVRational depth;

  440.     int c;

  441. 

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

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

  444. 

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

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

  447. 

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

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

  450.         nmin = FFMIN(nmin, p->nmin);

  451.         nmax = FFMAX(nmax, p->nmax);

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

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

  454.         diff1_sum_x2 += p->diff1_sum_x2;

  455.         diff1_sum += p->diff1_sum;

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

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

  458.         sigma_x += p->sigma_x;

  459.         sigma_x2 += p->sigma_x2;

  460.         min_count += p->min_count;

  461.         max_count += p->max_count;

  462.         min_runs += p->min_runs;

  463.         max_runs += p->max_runs;

  464.         mask |= p->mask;

  465.         imask &= p->imask;

  466.         nb_samples += p->nb_samples;

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

  468.             max_sigma_x = p->sigma_x;

  469. 

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

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

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

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

  474.         av_log(ctx, AV_LOG_INFO, "Min difference: %f\n", p->min_diff);

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

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

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

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

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

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

  481.         if (p->min_sigma_x2 != 1)

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

  483.         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);

  484.         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)));

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

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

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

  488.         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));

  489.     }

  490. 

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

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

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

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

  495.     av_log(ctx, AV_LOG_INFO, "Min difference: %f\n", min_diff);

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

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

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

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

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

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

  502.     if (min_sigma_x2 != 1)

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

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

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

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

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

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

  509. }

  510. 

  511. static av_cold void uninit(AVFilterContext *ctx)

  512. {

  513.     AudioStatsContext *s = ctx->priv;

  514. 

  515.     if (s->nb_channels)

  516.         print_stats(ctx);

  517.     av_freep(&s->chstats);

  518. }

  519. 

  520. static const AVFilterPad astats_inputs[] = {

  521.     {

  522.         .name         = "default",

  523.         .type         = AVMEDIA_TYPE_AUDIO,

  524.         .filter_frame = filter_frame,

  525.     },

  526.     { NULL }

  527. };

  528. 

  529. static const AVFilterPad astats_outputs[] = {

  530.     {

  531.         .name         = "default",

  532.         .type         = AVMEDIA_TYPE_AUDIO,

  533.         .config_props = config_output,

  534.     },

  535.     { NULL }

  536. };

  537. 

  538. AVFilter ff_af_astats = {

  539.     .name          = "astats",

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

  541.     .query_formats = query_formats,

  542.     .priv_size     = sizeof(AudioStatsContext),

  543.     .priv_class    = &astats_class,

  544.     .uninit        = uninit,

  545.     .inputs        = astats_inputs,

  546.     .outputs       = astats_outputs,

  547. };