mirror of
https://github.com/FFmpeg/FFmpeg.git
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532 lines
19 KiB
C
532 lines
19 KiB
C
/*
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* Copyright (c) 2009 Rob Sykes <robs@users.sourceforge.net>
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* Copyright (c) 2013 Paul B Mahol
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include <float.h>
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#include "libavutil/opt.h"
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#include "audio.h"
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#include "avfilter.h"
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#include "internal.h"
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typedef struct ChannelStats {
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double last;
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double sigma_x, sigma_x2;
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double avg_sigma_x2, min_sigma_x2, max_sigma_x2;
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double min, max;
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double nmin, nmax;
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double min_run, max_run;
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double min_runs, max_runs;
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double min_diff, max_diff;
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double diff1_sum;
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uint64_t mask, imask;
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uint64_t min_count, max_count;
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uint64_t nb_samples;
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} ChannelStats;
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typedef struct {
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const AVClass *class;
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ChannelStats *chstats;
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int nb_channels;
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uint64_t tc_samples;
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double time_constant;
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double mult;
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int metadata;
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int reset_count;
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int nb_frames;
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int maxbitdepth;
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} AudioStatsContext;
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#define OFFSET(x) offsetof(AudioStatsContext, x)
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#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption astats_options[] = {
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{ "length", "set the window length", OFFSET(time_constant), AV_OPT_TYPE_DOUBLE, {.dbl=.05}, .01, 10, FLAGS },
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{ "metadata", "inject metadata in the filtergraph", OFFSET(metadata), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, FLAGS },
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{ "reset", "recalculate stats after this many frames", OFFSET(reset_count), AV_OPT_TYPE_INT, {.i64=0}, 0, INT_MAX, FLAGS },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(astats);
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static int query_formats(AVFilterContext *ctx)
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{
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AVFilterFormats *formats;
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AVFilterChannelLayouts *layouts;
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static const enum AVSampleFormat sample_fmts[] = {
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AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P,
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AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32P,
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AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64P,
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AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
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AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
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AV_SAMPLE_FMT_NONE
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};
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int ret;
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layouts = ff_all_channel_counts();
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if (!layouts)
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return AVERROR(ENOMEM);
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ret = ff_set_common_channel_layouts(ctx, layouts);
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if (ret < 0)
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return ret;
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formats = ff_make_format_list(sample_fmts);
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if (!formats)
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return AVERROR(ENOMEM);
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ret = ff_set_common_formats(ctx, formats);
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if (ret < 0)
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return ret;
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formats = ff_all_samplerates();
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if (!formats)
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return AVERROR(ENOMEM);
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return ff_set_common_samplerates(ctx, formats);
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}
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static void reset_stats(AudioStatsContext *s)
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{
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int c;
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for (c = 0; c < s->nb_channels; c++) {
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ChannelStats *p = &s->chstats[c];
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p->min = p->nmin = p->min_sigma_x2 = DBL_MAX;
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p->max = p->nmax = p->max_sigma_x2 = DBL_MIN;
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p->min_diff = DBL_MAX;
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p->max_diff = DBL_MIN;
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p->sigma_x = 0;
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p->sigma_x2 = 0;
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p->avg_sigma_x2 = 0;
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p->min_sigma_x2 = 0;
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p->max_sigma_x2 = 0;
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p->min_run = 0;
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p->max_run = 0;
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p->min_runs = 0;
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p->max_runs = 0;
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p->diff1_sum = 0;
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p->mask = 0;
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p->imask = 0xFFFFFFFFFFFFFFFF;
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p->min_count = 0;
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p->max_count = 0;
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p->nb_samples = 0;
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}
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}
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static int config_output(AVFilterLink *outlink)
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{
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AudioStatsContext *s = outlink->src->priv;
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s->chstats = av_calloc(sizeof(*s->chstats), outlink->channels);
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if (!s->chstats)
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return AVERROR(ENOMEM);
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s->nb_channels = outlink->channels;
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s->mult = exp((-1 / s->time_constant / outlink->sample_rate));
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s->tc_samples = 5 * s->time_constant * outlink->sample_rate + .5;
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s->nb_frames = 0;
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s->maxbitdepth = av_get_bytes_per_sample(outlink->format) * 8;
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reset_stats(s);
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return 0;
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}
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static void bit_depth(AudioStatsContext *s, uint64_t mask, uint64_t imask, AVRational *depth)
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{
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unsigned result = s->maxbitdepth;
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mask = mask & (~imask);
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for (; result && !(mask & 1); --result, mask >>= 1);
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depth->den = result;
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depth->num = 0;
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for (; result; --result, mask >>= 1)
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if (mask & 1)
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depth->num++;
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}
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static inline void update_stat(AudioStatsContext *s, ChannelStats *p, double d, double nd, int64_t i)
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{
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if (d < p->min) {
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p->min = d;
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p->nmin = nd;
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p->min_run = 1;
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p->min_runs = 0;
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p->min_count = 1;
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} else if (d == p->min) {
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p->min_count++;
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p->min_run = d == p->last ? p->min_run + 1 : 1;
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} else if (p->last == p->min) {
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p->min_runs += p->min_run * p->min_run;
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}
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if (d > p->max) {
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p->max = d;
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p->nmax = nd;
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p->max_run = 1;
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p->max_runs = 0;
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p->max_count = 1;
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} else if (d == p->max) {
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p->max_count++;
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p->max_run = d == p->last ? p->max_run + 1 : 1;
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} else if (p->last == p->max) {
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p->max_runs += p->max_run * p->max_run;
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}
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p->sigma_x += nd;
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p->sigma_x2 += nd * nd;
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p->avg_sigma_x2 = p->avg_sigma_x2 * s->mult + (1.0 - s->mult) * nd * nd;
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p->min_diff = FFMIN(p->min_diff, fabs(d - p->last));
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p->max_diff = FFMAX(p->max_diff, fabs(d - p->last));
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p->diff1_sum += fabs(d - p->last);
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p->last = d;
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p->mask |= i;
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p->imask &= i;
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if (p->nb_samples >= s->tc_samples) {
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p->max_sigma_x2 = FFMAX(p->max_sigma_x2, p->avg_sigma_x2);
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p->min_sigma_x2 = FFMIN(p->min_sigma_x2, p->avg_sigma_x2);
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}
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p->nb_samples++;
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}
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static void set_meta(AVDictionary **metadata, int chan, const char *key,
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const char *fmt, double val)
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{
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uint8_t value[128];
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uint8_t key2[128];
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snprintf(value, sizeof(value), fmt, val);
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if (chan)
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snprintf(key2, sizeof(key2), "lavfi.astats.%d.%s", chan, key);
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else
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snprintf(key2, sizeof(key2), "lavfi.astats.%s", key);
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av_dict_set(metadata, key2, value, 0);
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}
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#define LINEAR_TO_DB(x) (log10(x) * 20)
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static void set_metadata(AudioStatsContext *s, AVDictionary **metadata)
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{
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uint64_t mask = 0, imask = 0xFFFFFFFFFFFFFFFF, min_count = 0, max_count = 0, nb_samples = 0;
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double min_runs = 0, max_runs = 0,
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min = DBL_MAX, max = DBL_MIN, min_diff = DBL_MAX, max_diff = 0,
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nmin = DBL_MAX, nmax = DBL_MIN,
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max_sigma_x = 0,
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diff1_sum = 0,
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sigma_x = 0,
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sigma_x2 = 0,
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min_sigma_x2 = DBL_MAX,
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max_sigma_x2 = DBL_MIN;
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AVRational depth;
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int c;
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for (c = 0; c < s->nb_channels; c++) {
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ChannelStats *p = &s->chstats[c];
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if (p->nb_samples < s->tc_samples)
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p->min_sigma_x2 = p->max_sigma_x2 = p->sigma_x2 / p->nb_samples;
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min = FFMIN(min, p->min);
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max = FFMAX(max, p->max);
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nmin = FFMIN(nmin, p->nmin);
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nmax = FFMAX(nmax, p->nmax);
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min_diff = FFMIN(min_diff, p->min_diff);
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max_diff = FFMAX(max_diff, p->max_diff);
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diff1_sum += p->diff1_sum,
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min_sigma_x2 = FFMIN(min_sigma_x2, p->min_sigma_x2);
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max_sigma_x2 = FFMAX(max_sigma_x2, p->max_sigma_x2);
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sigma_x += p->sigma_x;
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sigma_x2 += p->sigma_x2;
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min_count += p->min_count;
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max_count += p->max_count;
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min_runs += p->min_runs;
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max_runs += p->max_runs;
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mask |= p->mask;
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imask &= p->imask;
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nb_samples += p->nb_samples;
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if (fabs(p->sigma_x) > fabs(max_sigma_x))
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max_sigma_x = p->sigma_x;
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set_meta(metadata, c + 1, "DC_offset", "%f", p->sigma_x / p->nb_samples);
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set_meta(metadata, c + 1, "Min_level", "%f", p->min);
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set_meta(metadata, c + 1, "Max_level", "%f", p->max);
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set_meta(metadata, c + 1, "Min_difference", "%f", p->min_diff);
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set_meta(metadata, c + 1, "Max_difference", "%f", p->max_diff);
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set_meta(metadata, c + 1, "Mean_difference", "%f", p->diff1_sum / (p->nb_samples - 1));
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set_meta(metadata, c + 1, "Peak_level", "%f", LINEAR_TO_DB(FFMAX(-p->nmin, p->nmax)));
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set_meta(metadata, c + 1, "RMS_level", "%f", LINEAR_TO_DB(sqrt(p->sigma_x2 / p->nb_samples)));
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set_meta(metadata, c + 1, "RMS_peak", "%f", LINEAR_TO_DB(sqrt(p->max_sigma_x2)));
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set_meta(metadata, c + 1, "RMS_trough", "%f", LINEAR_TO_DB(sqrt(p->min_sigma_x2)));
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set_meta(metadata, c + 1, "Crest_factor", "%f", p->sigma_x2 ? FFMAX(-p->min, p->max) / sqrt(p->sigma_x2 / p->nb_samples) : 1);
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set_meta(metadata, c + 1, "Flat_factor", "%f", LINEAR_TO_DB((p->min_runs + p->max_runs) / (p->min_count + p->max_count)));
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set_meta(metadata, c + 1, "Peak_count", "%f", (float)(p->min_count + p->max_count));
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bit_depth(s, p->mask, p->imask, &depth);
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set_meta(metadata, c + 1, "Bit_depth", "%f", depth.num);
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set_meta(metadata, c + 1, "Bit_depth2", "%f", depth.den);
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}
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set_meta(metadata, 0, "Overall.DC_offset", "%f", max_sigma_x / (nb_samples / s->nb_channels));
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set_meta(metadata, 0, "Overall.Min_level", "%f", min);
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set_meta(metadata, 0, "Overall.Max_level", "%f", max);
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set_meta(metadata, 0, "Overall.Min_difference", "%f", min_diff);
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set_meta(metadata, 0, "Overall.Max_difference", "%f", max_diff);
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set_meta(metadata, 0, "Overall.Mean_difference", "%f", diff1_sum / (nb_samples - s->nb_channels));
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set_meta(metadata, 0, "Overall.Peak_level", "%f", LINEAR_TO_DB(FFMAX(-nmin, nmax)));
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set_meta(metadata, 0, "Overall.RMS_level", "%f", LINEAR_TO_DB(sqrt(sigma_x2 / nb_samples)));
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set_meta(metadata, 0, "Overall.RMS_peak", "%f", LINEAR_TO_DB(sqrt(max_sigma_x2)));
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set_meta(metadata, 0, "Overall.RMS_trough", "%f", LINEAR_TO_DB(sqrt(min_sigma_x2)));
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set_meta(metadata, 0, "Overall.Flat_factor", "%f", LINEAR_TO_DB((min_runs + max_runs) / (min_count + max_count)));
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set_meta(metadata, 0, "Overall.Peak_count", "%f", (float)(min_count + max_count) / (double)s->nb_channels);
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bit_depth(s, mask, imask, &depth);
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set_meta(metadata, 0, "Overall.Bit_depth", "%f", depth.num);
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set_meta(metadata, 0, "Overall.Bit_depth2", "%f", depth.den);
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set_meta(metadata, 0, "Overall.Number_of_samples", "%f", nb_samples / s->nb_channels);
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}
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static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
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{
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AudioStatsContext *s = inlink->dst->priv;
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AVDictionary **metadata = &buf->metadata;
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const int channels = s->nb_channels;
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int i, c;
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if (s->reset_count > 0) {
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if (s->nb_frames >= s->reset_count) {
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reset_stats(s);
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s->nb_frames = 0;
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}
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s->nb_frames++;
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}
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switch (inlink->format) {
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case AV_SAMPLE_FMT_DBLP:
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for (c = 0; c < channels; c++) {
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ChannelStats *p = &s->chstats[c];
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const double *src = (const double *)buf->extended_data[c];
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for (i = 0; i < buf->nb_samples; i++, src++)
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update_stat(s, p, *src, *src, llrint(*src * (UINT64_C(1) << 63)));
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}
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break;
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case AV_SAMPLE_FMT_DBL: {
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const double *src = (const double *)buf->extended_data[0];
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for (i = 0; i < buf->nb_samples; i++) {
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for (c = 0; c < channels; c++, src++)
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update_stat(s, &s->chstats[c], *src, *src, llrint(*src * (UINT64_C(1) << 63)));
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}}
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break;
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case AV_SAMPLE_FMT_FLTP:
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for (c = 0; c < channels; c++) {
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ChannelStats *p = &s->chstats[c];
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const float *src = (const float *)buf->extended_data[c];
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for (i = 0; i < buf->nb_samples; i++, src++)
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update_stat(s, p, *src, *src, llrint(*src * (UINT64_C(1) << 31)));
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}
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break;
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case AV_SAMPLE_FMT_FLT: {
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const float *src = (const float *)buf->extended_data[0];
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for (i = 0; i < buf->nb_samples; i++) {
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for (c = 0; c < channels; c++, src++)
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update_stat(s, &s->chstats[c], *src, *src, llrint(*src * (UINT64_C(1) << 31)));
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}}
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break;
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case AV_SAMPLE_FMT_S64P:
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for (c = 0; c < channels; c++) {
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ChannelStats *p = &s->chstats[c];
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const int64_t *src = (const int64_t *)buf->extended_data[c];
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for (i = 0; i < buf->nb_samples; i++, src++)
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update_stat(s, p, *src, *src / (double)INT64_MAX, *src);
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}
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break;
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case AV_SAMPLE_FMT_S64: {
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const int64_t *src = (const int64_t *)buf->extended_data[0];
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for (i = 0; i < buf->nb_samples; i++) {
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for (c = 0; c < channels; c++, src++)
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update_stat(s, &s->chstats[c], *src, *src / (double)INT64_MAX, *src);
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}}
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break;
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case AV_SAMPLE_FMT_S32P:
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for (c = 0; c < channels; c++) {
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ChannelStats *p = &s->chstats[c];
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const int32_t *src = (const int32_t *)buf->extended_data[c];
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for (i = 0; i < buf->nb_samples; i++, src++)
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update_stat(s, p, *src, *src / (double)INT32_MAX, *src);
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}
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break;
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case AV_SAMPLE_FMT_S32: {
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const int32_t *src = (const int32_t *)buf->extended_data[0];
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for (i = 0; i < buf->nb_samples; i++) {
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for (c = 0; c < channels; c++, src++)
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update_stat(s, &s->chstats[c], *src, *src / (double)INT32_MAX, *src);
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}}
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break;
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case AV_SAMPLE_FMT_S16P:
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for (c = 0; c < channels; c++) {
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ChannelStats *p = &s->chstats[c];
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const int16_t *src = (const int16_t *)buf->extended_data[c];
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for (i = 0; i < buf->nb_samples; i++, src++)
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update_stat(s, p, *src, *src / (double)INT16_MAX, *src);
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}
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break;
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case AV_SAMPLE_FMT_S16: {
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const int16_t *src = (const int16_t *)buf->extended_data[0];
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for (i = 0; i < buf->nb_samples; i++) {
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for (c = 0; c < channels; c++, src++)
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update_stat(s, &s->chstats[c], *src, *src / (double)INT16_MAX, *src);
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}}
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break;
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}
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if (s->metadata)
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set_metadata(s, metadata);
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return ff_filter_frame(inlink->dst->outputs[0], buf);
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}
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static void print_stats(AVFilterContext *ctx)
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{
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AudioStatsContext *s = ctx->priv;
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uint64_t mask = 0, imask = 0xFFFFFFFFFFFFFFFF, min_count = 0, max_count = 0, nb_samples = 0;
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double min_runs = 0, max_runs = 0,
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min = DBL_MAX, max = DBL_MIN, min_diff = DBL_MAX, max_diff = 0,
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nmin = DBL_MAX, nmax = DBL_MIN,
|
|
max_sigma_x = 0,
|
|
diff1_sum = 0,
|
|
sigma_x = 0,
|
|
sigma_x2 = 0,
|
|
min_sigma_x2 = DBL_MAX,
|
|
max_sigma_x2 = DBL_MIN;
|
|
AVRational depth;
|
|
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);
|
|
nmin = FFMIN(nmin, p->nmin);
|
|
nmax = FFMAX(nmax, p->nmax);
|
|
min_diff = FFMIN(min_diff, p->min_diff);
|
|
max_diff = FFMAX(max_diff, p->max_diff);
|
|
diff1_sum += p->diff1_sum,
|
|
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;
|
|
mask |= p->mask;
|
|
imask &= p->imask;
|
|
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, "Min difference: %f\n", p->min_diff);
|
|
av_log(ctx, AV_LOG_INFO, "Max difference: %f\n", p->max_diff);
|
|
av_log(ctx, AV_LOG_INFO, "Mean difference: %f\n", p->diff1_sum / (p->nb_samples - 1));
|
|
av_log(ctx, AV_LOG_INFO, "Peak level dB: %f\n", LINEAR_TO_DB(FFMAX(-p->nmin, p->nmax)));
|
|
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->nmin, p->nmax) / 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: %"PRId64"\n", p->min_count + p->max_count);
|
|
bit_depth(s, p->mask, p->imask, &depth);
|
|
av_log(ctx, AV_LOG_INFO, "Bit depth: %u/%u\n", depth.num, depth.den);
|
|
}
|
|
|
|
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, "Min difference: %f\n", min_diff);
|
|
av_log(ctx, AV_LOG_INFO, "Max difference: %f\n", max_diff);
|
|
av_log(ctx, AV_LOG_INFO, "Mean difference: %f\n", diff1_sum / (nb_samples - s->nb_channels));
|
|
av_log(ctx, AV_LOG_INFO, "Peak level dB: %f\n", LINEAR_TO_DB(FFMAX(-nmin, nmax)));
|
|
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);
|
|
bit_depth(s, mask, imask, &depth);
|
|
av_log(ctx, AV_LOG_INFO, "Bit depth: %u/%u\n", depth.num, depth.den);
|
|
av_log(ctx, AV_LOG_INFO, "Number of samples: %"PRId64"\n", nb_samples / s->nb_channels);
|
|
}
|
|
|
|
static av_cold void uninit(AVFilterContext *ctx)
|
|
{
|
|
AudioStatsContext *s = ctx->priv;
|
|
|
|
if (s->nb_channels)
|
|
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 ff_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,
|
|
};
|