[FFmpeg-devel] [PATCH] astats filter
Paul B Mahol
onemda at gmail.com
Wed Apr 24 19:58:28 CEST 2013
On 4/24/13, Stefano Sabatini <stefasab at gmail.com> wrote:
> 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?).
There is only per frame metadata.
>
> [...]
> --
> FFmpeg = Fostering Fast Mystic Proud Efficient Gadget
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