astats filter

Signed-off-by: Paul B Mahol <onemda@gmail.com>

Changelog

@@ -33,6 +33,7 @@ version <next>:
 - timeline editing with filters
 - vidstabdetect and vidstabtransform filters for video stabilization using
   the vid.stab library
+- astats filter
 
 
 version 1.2:

doc/filters.texi

@@ -990,6 +990,51 @@ the data is treated as if all the planes were concatenated.
 A list of Adler-32 checksums for each data plane.
 @end table
 
+@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:
+@table @option
+@item length
+Short window length in seconds, used for peak and trough RMS measurement.
+Default is @code{0.05} (50 miliseconds). Allowed range is @code{[0.1 - 10]}.
+@end table
+
+A description of each shown parameter follows:
+
+@table @option
+@item DC offset
+Mean amplitude displacement from zero.
+
+@item Min level
+Minimal sample level.
+
+@item Max level
+Maximal sample level.
+
+@item Peak level dB
+@item RMS level dB
+Standard peak and RMS level measured in dBFS.
+
+@item RMS peak dB
+@item RMS trough dB
+Peak and trough values for RMS level measured over a short window.
+
+@item Crest factor
+Standard ratio of peak to RMS level (note: not in dB).
+
+@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}).
+
+@item Peak count
+Number of occasions (not the number of samples) that the signal attained either
+@var{Min level} or @var{Max level}.
+@end table
+
 @section astreamsync
 
 Forward two audio streams and control the order the buffers are forwarded.

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

libavfilter/af_astats.c

+/*
+ * Copyright (c) 2009 Rob Sykes <robs@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;
+} 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;
+    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++)
+                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++)
+                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: %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", 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: %lld\n", nb_samples / s->nb_channels);
+}
+
+static 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 avfilter_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,
+};

libavfilter/allfilters.c

@@ -67,6 +67,7 @@ void avfilter_register_all(void)
     REGISTER_FILTER(ASETTB,         asettb,         af);
     REGISTER_FILTER(ASHOWINFO,      ashowinfo,      af);
     REGISTER_FILTER(ASPLIT,         asplit,         af);
+    REGISTER_FILTER(ASTATS,         astats,         af);
     REGISTER_FILTER(ASTREAMSYNC,    astreamsync,    af);
     REGISTER_FILTER(ASYNCTS,        asyncts,        af);
     REGISTER_FILTER(ATEMPO,         atempo,         af);

libavfilter/version.h

@@ -29,8 +29,8 @@
 #include "libavutil/avutil.h"
 
 #define LIBAVFILTER_VERSION_MAJOR  3
-#define LIBAVFILTER_VERSION_MINOR  60
-#define LIBAVFILTER_VERSION_MICRO 102
+#define LIBAVFILTER_VERSION_MINOR  61
+#define LIBAVFILTER_VERSION_MICRO 100
 
 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
                                                LIBAVFILTER_VERSION_MINOR, \