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af_astats.c
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af_astats.c
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/*
* Copyright (c) 2009 Rob Sykes <[email protected]>
* 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 sigma_x, sigma_x2;
double avg_sigma_x2, min_sigma_x2, max_sigma_x2;
double min, max;
double min_run, max_run;
double min_runs, max_runs;
uint64_t min_count, max_count;
uint64_t nb_samples;
} ChannelStats;
typedef struct {
const AVClass *class;
ChannelStats *chstats;
int nb_channels;
uint64_t tc_samples;
double time_constant;
double mult;
} 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 },
{ NULL }
};
AVFILTER_DEFINE_CLASS(astats);
static int query_formats(AVFilterContext *ctx)
{
AVFilterFormats *formats;
AVFilterChannelLayouts *layouts;
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
AV_SAMPLE_FMT_NONE
};
layouts = ff_all_channel_layouts();
if (!layouts)
return AVERROR(ENOMEM);
ff_set_common_channel_layouts(ctx, layouts);
formats = ff_make_format_list(sample_fmts);
if (!formats)
return AVERROR(ENOMEM);
ff_set_common_formats(ctx, formats);
formats = ff_all_samplerates();
if (!formats)
return AVERROR(ENOMEM);
ff_set_common_samplerates(ctx, formats);
return 0;
}
static int config_output(AVFilterLink *outlink)
{
AudioStatsContext *s = outlink->src->priv;
int c;
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;
for (c = 0; c < s->nb_channels; c++) {
ChannelStats *p = &s->chstats[c];
p->min = p->min_sigma_x2 = DBL_MAX;
p->max = p->max_sigma_x2 = DBL_MIN;
}
return 0;
}
static inline void update_stat(AudioStatsContext *s, ChannelStats *p, double d)
{
if (d < p->min) {
p->min = d;
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 > p->max) {
p->max = d;
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 += d;
p->sigma_x2 += d * d;
p->avg_sigma_x2 = p->avg_sigma_x2 * s->mult + (1.0 - s->mult) * d * d;
p->last = d;
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 int filter_frame(AVFilterLink *inlink, AVFrame *buf)
{
AudioStatsContext *s = inlink->dst->priv;
const int channels = s->nb_channels;
const double *src;
int i, c;
switch (inlink->format) {
case AV_SAMPLE_FMT_DBLP:
for (c = 0; c < channels; c++) {
ChannelStats *p = &s->chstats[c];
src = (const double *)buf->extended_data[c];
for (i = 0; i < buf->nb_samples; i++, src++)
update_stat(s, p, *src);
}
break;
case AV_SAMPLE_FMT_DBL:
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);
}
break;
}
return ff_filter_frame(inlink->dst->outputs[0], buf);
}
#define LINEAR_TO_DB(x) (log10(x) * 20)
static void print_stats(AVFilterContext *ctx)
{
AudioStatsContext *s = ctx->priv;
uint64_t min_count = 0, max_count = 0, nb_samples = 0;
double min_runs = 0, max_runs = 0,
min = DBL_MAX, max = DBL_MIN,
max_sigma_x = 0,
sigma_x = 0,
sigma_x2 = 0,
min_sigma_x2 = DBL_MAX,
max_sigma_x2 = DBL_MIN;
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);
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;
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, "Peak level dB: %f\n", LINEAR_TO_DB(FFMAX(-p->min, p->max)));
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->min, p->max) / 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);
}
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, "Peak level dB: %f\n", LINEAR_TO_DB(FFMAX(-min, max)));
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);
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;
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,
};