[FFmpeg-devel] [PATCH] astats filter
Paul B Mahol
onemda at gmail.com
Tue Apr 23 14:59:08 CEST 2013
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.
+ 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.
+
+ 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;
+} 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);
+ }
+ }
+ 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);
+
+}
+
+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 statistics about audio frames."),
+ .query_formats = query_formats,
+ .priv_size = sizeof(AudioStatsContext),
+ .priv_class = &astats_class,
+ .uninit = uninit,
+ .inputs = astats_inputs,
+ .outputs = astats_outputs,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index eb5f657..5e8e7d8 100644
--- a/libavfilter/allfilters.c
+++ b/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);
--
1.7.11.2
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