@@ -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:

@@ -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.

@@ -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

new file mode 100644

@@ -0,0 +1,274 @@

+/*

+ * 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,

+};

@@ -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);

@@ -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, \