From cc5c15595930e473d851b211e4daa4ca3b68e458 Mon Sep 17 00:00:00 2001 From: Paul B Mahol Date: Mon, 22 Apr 2013 12:38:24 +0000 Subject: astats filter Signed-off-by: Paul B Mahol --- libavfilter/af_astats.c | 274 ++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 274 insertions(+) create mode 100644 libavfilter/af_astats.c (limited to 'libavfilter/af_astats.c') diff --git a/libavfilter/af_astats.c b/libavfilter/af_astats.c new file mode 100644 index 0000000..3a63d85 --- /dev/null +++ b/libavfilter/af_astats.c @@ -0,0 +1,274 @@ +/* + * Copyright (c) 2009 Rob Sykes + * 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 + +#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, +}; -- cgit v1.1