/* * 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 update_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++) update_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++) update_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: %"PRId64"\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: %"PRId64"\n", nb_samples / s->nb_channels); } static av_cold 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 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, };