[FFmpeg-devel] [PATCH] astats filter

Stefano Sabatini stefasab at gmail.com
Wed Apr 24 19:51:05 CEST 2013


On date Tuesday 2013-04-23 12:59:08 +0000, Paul B Mahol encoded:
> Signed-off-by: Paul B Mahol <onemda at gmail.com>
> ---
>  doc/filters.texi         |  44 ++++++++
>  libavfilter/Makefile     |   1 +
>  libavfilter/af_astats.c  | 287 +++++++++++++++++++++++++++++++++++++++++++++++
>  libavfilter/allfilters.c |   1 +
>  4 files changed, 333 insertions(+)
>  create mode 100644 libavfilter/af_astats.c
> 
> diff --git a/doc/filters.texi b/doc/filters.texi
> index d5fda03..1e2363d 100644
> --- a/doc/filters.texi
> +++ b/doc/filters.texi
> @@ -990,6 +990,50 @@ the data is treated as if all the planes were concatenated.
>  A list of Adler-32 checksums for each data plane.
>  @end table
>  
> + at section astats
> +
> +Display time domain statistical information about the audio channels.
> +Statistics are calculated and displayed for each audio channel and,
> +where applicable, an overall figure is also given.
> +
> +The filter accepts the following option:
> + at table @option
> + at item length
> +Short window length. Default is 50ms.

nit: specify the unit (I think it is seconds), and range.

Also it is not clear what this "short window" refers to.

> + at end table
> +
> +A description of each shown parameter follows:
> +
> + at table @option
> + at item DC offset
> +Mean amplitude displacement from zero.
> +
> + at item Min level
> +Minimal sample level.
> +
> + at item Max level
> +Maximal sample level.
> +
> + at item Peak level dB
> + at item RMS level dB
> +Standard peak and RMS level measured in dBFS.
> +
> + at item RMS peak dB

> + at item RMS through dB
> +Peak and trough values for RMS level measured over a short window.

trough or through?

> +
> + at item Crest factor
> +Standard ratio of peak to RMS level (note: not in dB).
> +
> + at item Flat factor
> +Flatness (i.e. consecutive samples with the same value) of the signal at its peak levels
> +(i.e. either @var{Min level} or @var{Max level}).
> +
> + at item Peak count
> +Number of occasions (not the number of samples) that the signal attained either
> + at var{Min level} or @var{Max level}.
> + at end table
> +
>  @section astreamsync
>  
>  Forward two audio streams and control the order the buffers are forwarded.
> diff --git a/libavfilter/Makefile b/libavfilter/Makefile
> index 4fce503..2b2adcb 100644
> --- a/libavfilter/Makefile
> +++ b/libavfilter/Makefile
> @@ -69,6 +69,7 @@ OBJS-$(CONFIG_ASETRATE_FILTER)               += af_asetrate.o
>  OBJS-$(CONFIG_ASETTB_FILTER)                 += f_settb.o
>  OBJS-$(CONFIG_ASHOWINFO_FILTER)              += af_ashowinfo.o
>  OBJS-$(CONFIG_ASPLIT_FILTER)                 += split.o
> +OBJS-$(CONFIG_ASTATS_FILTER)                 += af_astats.o
>  OBJS-$(CONFIG_ASTREAMSYNC_FILTER)            += af_astreamsync.o
>  OBJS-$(CONFIG_ASYNCTS_FILTER)                += af_asyncts.o
>  OBJS-$(CONFIG_ATEMPO_FILTER)                 += af_atempo.o
> diff --git a/libavfilter/af_astats.c b/libavfilter/af_astats.c
> new file mode 100644
> index 0000000..547cfc2
> --- /dev/null
> +++ b/libavfilter/af_astats.c
> @@ -0,0 +1,287 @@
> +/*
> + * Copyright (c) 2009 Rob Sykes <robs at users.sourceforge.net>
> + * 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;

better names / doxyes?

> +} 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 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;
> +    int i, c;
> +
> +    switch (inlink->format) {

> +    case AV_SAMPLE_FMT_DBLP:
> +        for (c = 0; c < channels; c++) {
> +            ChannelStats *p = &s->chstats[c];
> +            const double *src = (const double *)buf->extended_data[c];
> +
> +            for (i = 0; i < buf->nb_samples; i++, src++)
> +                stat(s, p, *src);
> +        }
> +        break;
> +    case AV_SAMPLE_FMT_DBL: {
> +        const double *src = (const double *)buf->extended_data[0];
> +
> +        for (i = 0; i < buf->nb_samples; i++) {
> +            for (c = 0; c < channels; c++, src++) {
> +                ChannelStats *p = &s->chstats[c];
> +
> +                stat(s, p, *src);

you can directly use s->chstats[c]

> +            }
> +        }
> +        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;
> +    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 through 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: %lld\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", sigma_x / nb_samples);
> +    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 through 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);

I wonder what would be a sane way to propagate this information to the
outside world (through a log, or maybe using metadata?).

[...]
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FFmpeg = Fostering Fast Mystic Proud Efficient Gadget


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