mirror of
https://github.com/FFmpeg/FFmpeg.git
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275 lines
8.8 KiB
C
275 lines
8.8 KiB
C
/*
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* Copyright (c) 2009 Rob Sykes <robs@users.sourceforge.net>
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* Copyright (c) 2013 Paul B Mahol
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include <float.h>
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#include "libavutil/opt.h"
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#include "audio.h"
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#include "avfilter.h"
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#include "internal.h"
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typedef struct ChannelStats {
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double last;
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double sigma_x, sigma_x2;
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double avg_sigma_x2, min_sigma_x2, max_sigma_x2;
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double min, max;
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double min_run, max_run;
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double min_runs, max_runs;
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uint64_t min_count, max_count;
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uint64_t nb_samples;
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} ChannelStats;
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typedef struct {
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const AVClass *class;
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ChannelStats *chstats;
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int nb_channels;
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uint64_t tc_samples;
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double time_constant;
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double mult;
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} AudioStatsContext;
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#define OFFSET(x) offsetof(AudioStatsContext, x)
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#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption astats_options[] = {
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{ "length", "set the window length", OFFSET(time_constant), AV_OPT_TYPE_DOUBLE, {.dbl=.05}, .01, 10, FLAGS },
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{NULL},
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};
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AVFILTER_DEFINE_CLASS(astats);
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static int query_formats(AVFilterContext *ctx)
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{
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AVFilterFormats *formats;
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AVFilterChannelLayouts *layouts;
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static const enum AVSampleFormat sample_fmts[] = {
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AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
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AV_SAMPLE_FMT_NONE
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};
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layouts = ff_all_channel_layouts();
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if (!layouts)
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return AVERROR(ENOMEM);
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ff_set_common_channel_layouts(ctx, layouts);
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formats = ff_make_format_list(sample_fmts);
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if (!formats)
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return AVERROR(ENOMEM);
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ff_set_common_formats(ctx, formats);
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formats = ff_all_samplerates();
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if (!formats)
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return AVERROR(ENOMEM);
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ff_set_common_samplerates(ctx, formats);
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return 0;
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}
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static int config_output(AVFilterLink *outlink)
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{
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AudioStatsContext *s = outlink->src->priv;
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int c;
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s->chstats = av_calloc(sizeof(*s->chstats), outlink->channels);
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if (!s->chstats)
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return AVERROR(ENOMEM);
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s->nb_channels = outlink->channels;
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s->mult = exp((-1 / s->time_constant / outlink->sample_rate));
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s->tc_samples = 5 * s->time_constant * outlink->sample_rate + .5;
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for (c = 0; c < s->nb_channels; c++) {
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ChannelStats *p = &s->chstats[c];
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p->min = p->min_sigma_x2 = DBL_MAX;
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p->max = p->max_sigma_x2 = DBL_MIN;
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}
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return 0;
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}
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static inline void update_stat(AudioStatsContext *s, ChannelStats *p, double d)
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{
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if (d < p->min) {
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p->min = d;
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p->min_run = 1;
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p->min_runs = 0;
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p->min_count = 1;
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} else if (d == p->min) {
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p->min_count++;
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p->min_run = d == p->last ? p->min_run + 1 : 1;
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} else if (p->last == p->min) {
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p->min_runs += p->min_run * p->min_run;
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}
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if (d > p->max) {
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p->max = d;
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p->max_run = 1;
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p->max_runs = 0;
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p->max_count = 1;
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} else if (d == p->max) {
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p->max_count++;
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p->max_run = d == p->last ? p->max_run + 1 : 1;
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} else if (p->last == p->max) {
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p->max_runs += p->max_run * p->max_run;
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}
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p->sigma_x += d;
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p->sigma_x2 += d * d;
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p->avg_sigma_x2 = p->avg_sigma_x2 * s->mult + (1.0 - s->mult) * d * d;
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p->last = d;
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if (p->nb_samples >= s->tc_samples) {
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p->max_sigma_x2 = FFMAX(p->max_sigma_x2, p->avg_sigma_x2);
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p->min_sigma_x2 = FFMIN(p->min_sigma_x2, p->avg_sigma_x2);
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}
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p->nb_samples++;
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}
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static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
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{
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AudioStatsContext *s = inlink->dst->priv;
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const int channels = s->nb_channels;
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const double *src;
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int i, c;
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switch (inlink->format) {
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case AV_SAMPLE_FMT_DBLP:
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for (c = 0; c < channels; c++) {
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ChannelStats *p = &s->chstats[c];
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src = (const double *)buf->extended_data[c];
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for (i = 0; i < buf->nb_samples; i++, src++)
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update_stat(s, p, *src);
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}
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break;
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case AV_SAMPLE_FMT_DBL:
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src = (const double *)buf->extended_data[0];
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for (i = 0; i < buf->nb_samples; i++) {
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for (c = 0; c < channels; c++, src++)
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update_stat(s, &s->chstats[c], *src);
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}
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break;
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}
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return ff_filter_frame(inlink->dst->outputs[0], buf);
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}
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#define LINEAR_TO_DB(x) (log10(x) * 20)
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static void print_stats(AVFilterContext *ctx)
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{
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AudioStatsContext *s = ctx->priv;
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uint64_t min_count = 0, max_count = 0, nb_samples = 0;
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double min_runs = 0, max_runs = 0,
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min = DBL_MAX, max = DBL_MIN,
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max_sigma_x = 0,
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sigma_x = 0,
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sigma_x2 = 0,
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min_sigma_x2 = DBL_MAX,
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max_sigma_x2 = DBL_MIN;
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int c;
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for (c = 0; c < s->nb_channels; c++) {
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ChannelStats *p = &s->chstats[c];
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if (p->nb_samples < s->tc_samples)
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p->min_sigma_x2 = p->max_sigma_x2 = p->sigma_x2 / p->nb_samples;
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min = FFMIN(min, p->min);
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max = FFMAX(max, p->max);
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min_sigma_x2 = FFMIN(min_sigma_x2, p->min_sigma_x2);
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max_sigma_x2 = FFMAX(max_sigma_x2, p->max_sigma_x2);
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sigma_x += p->sigma_x;
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sigma_x2 += p->sigma_x2;
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min_count += p->min_count;
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max_count += p->max_count;
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min_runs += p->min_runs;
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max_runs += p->max_runs;
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nb_samples += p->nb_samples;
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if (fabs(p->sigma_x) > fabs(max_sigma_x))
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max_sigma_x = p->sigma_x;
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av_log(ctx, AV_LOG_INFO, "Channel: %d\n", c + 1);
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av_log(ctx, AV_LOG_INFO, "DC offset: %f\n", p->sigma_x / p->nb_samples);
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av_log(ctx, AV_LOG_INFO, "Min level: %f\n", p->min);
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av_log(ctx, AV_LOG_INFO, "Max level: %f\n", p->max);
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av_log(ctx, AV_LOG_INFO, "Peak level dB: %f\n", LINEAR_TO_DB(FFMAX(-p->min, p->max)));
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av_log(ctx, AV_LOG_INFO, "RMS level dB: %f\n", LINEAR_TO_DB(sqrt(p->sigma_x2 / p->nb_samples)));
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av_log(ctx, AV_LOG_INFO, "RMS peak dB: %f\n", LINEAR_TO_DB(sqrt(p->max_sigma_x2)));
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if (p->min_sigma_x2 != 1)
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av_log(ctx, AV_LOG_INFO, "RMS trough dB: %f\n",LINEAR_TO_DB(sqrt(p->min_sigma_x2)));
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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);
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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)));
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av_log(ctx, AV_LOG_INFO, "Peak count: %"PRId64"\n", p->min_count + p->max_count);
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}
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av_log(ctx, AV_LOG_INFO, "Overall\n");
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av_log(ctx, AV_LOG_INFO, "DC offset: %f\n", max_sigma_x / (nb_samples / s->nb_channels));
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av_log(ctx, AV_LOG_INFO, "Min level: %f\n", min);
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av_log(ctx, AV_LOG_INFO, "Max level: %f\n", max);
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av_log(ctx, AV_LOG_INFO, "Peak level dB: %f\n", LINEAR_TO_DB(FFMAX(-min, max)));
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av_log(ctx, AV_LOG_INFO, "RMS level dB: %f\n", LINEAR_TO_DB(sqrt(sigma_x2 / nb_samples)));
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av_log(ctx, AV_LOG_INFO, "RMS peak dB: %f\n", LINEAR_TO_DB(sqrt(max_sigma_x2)));
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if (min_sigma_x2 != 1)
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av_log(ctx, AV_LOG_INFO, "RMS trough dB: %f\n", LINEAR_TO_DB(sqrt(min_sigma_x2)));
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av_log(ctx, AV_LOG_INFO, "Flat factor: %f\n", LINEAR_TO_DB((min_runs + max_runs) / (min_count + max_count)));
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av_log(ctx, AV_LOG_INFO, "Peak count: %f\n", (min_count + max_count) / (double)s->nb_channels);
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av_log(ctx, AV_LOG_INFO, "Number of samples: %"PRId64"\n", nb_samples / s->nb_channels);
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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AudioStatsContext *s = ctx->priv;
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print_stats(ctx);
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av_freep(&s->chstats);
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}
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static const AVFilterPad astats_inputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_frame = filter_frame,
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},
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{ NULL }
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};
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static const AVFilterPad astats_outputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.config_props = config_output,
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},
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{ NULL }
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};
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AVFilter avfilter_af_astats = {
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.name = "astats",
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.description = NULL_IF_CONFIG_SMALL("Show time domain statistics about audio frames."),
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.query_formats = query_formats,
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.priv_size = sizeof(AudioStatsContext),
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.priv_class = &astats_class,
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.uninit = uninit,
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.inputs = astats_inputs,
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.outputs = astats_outputs,
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};
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