1/* 2 * Copyright (c) 2009 Rob Sykes <robs@users.sourceforge.net> 3 * Copyright (c) 2013 Paul B Mahol 4 * 5 * This file is part of FFmpeg. 6 * 7 * FFmpeg is free software; you can redistribute it and/or 8 * modify it under the terms of the GNU Lesser General Public 9 * License as published by the Free Software Foundation; either 10 * version 2.1 of the License, or (at your option) any later version. 11 * 12 * FFmpeg is distributed in the hope that it will be useful, 13 * but WITHOUT ANY WARRANTY; without even the implied warranty of 14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU 15 * Lesser General Public License for more details. 16 * 17 * You should have received a copy of the GNU Lesser General Public 18 * License along with FFmpeg; if not, write to the Free Software 19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA 20 */ 21 22#include <float.h> 23 24#include "libavutil/opt.h" 25#include "audio.h" 26#include "avfilter.h" 27#include "internal.h" 28 29typedef struct ChannelStats { 30 double last; 31 double sigma_x, sigma_x2; 32 double avg_sigma_x2, min_sigma_x2, max_sigma_x2; 33 double min, max; 34 double min_run, max_run; 35 double min_runs, max_runs; 36 uint64_t min_count, max_count; 37 uint64_t nb_samples; 38} ChannelStats; 39 40typedef struct { 41 const AVClass *class; 42 ChannelStats *chstats; 43 int nb_channels; 44 uint64_t tc_samples; 45 double time_constant; 46 double mult; 47} AudioStatsContext; 48 49#define OFFSET(x) offsetof(AudioStatsContext, x) 50#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM 51 52static const AVOption astats_options[] = { 53 { "length", "set the window length", OFFSET(time_constant), AV_OPT_TYPE_DOUBLE, {.dbl=.05}, .01, 10, FLAGS }, 54 { NULL } 55}; 56 57AVFILTER_DEFINE_CLASS(astats); 58 59static int query_formats(AVFilterContext *ctx) 60{ 61 AVFilterFormats *formats; 62 AVFilterChannelLayouts *layouts; 63 static const enum AVSampleFormat sample_fmts[] = { 64 AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP, 65 AV_SAMPLE_FMT_NONE 66 }; 67 68 layouts = ff_all_channel_layouts(); 69 if (!layouts) 70 return AVERROR(ENOMEM); 71 ff_set_common_channel_layouts(ctx, layouts); 72 73 formats = ff_make_format_list(sample_fmts); 74 if (!formats) 75 return AVERROR(ENOMEM); 76 ff_set_common_formats(ctx, formats); 77 78 formats = ff_all_samplerates(); 79 if (!formats) 80 return AVERROR(ENOMEM); 81 ff_set_common_samplerates(ctx, formats); 82 83 return 0; 84} 85 86static int config_output(AVFilterLink *outlink) 87{ 88 AudioStatsContext *s = outlink->src->priv; 89 int c; 90 91 s->chstats = av_calloc(sizeof(*s->chstats), outlink->channels); 92 if (!s->chstats) 93 return AVERROR(ENOMEM); 94 s->nb_channels = outlink->channels; 95 s->mult = exp((-1 / s->time_constant / outlink->sample_rate)); 96 s->tc_samples = 5 * s->time_constant * outlink->sample_rate + .5; 97 98 for (c = 0; c < s->nb_channels; c++) { 99 ChannelStats *p = &s->chstats[c]; 100 101 p->min = p->min_sigma_x2 = DBL_MAX; 102 p->max = p->max_sigma_x2 = DBL_MIN; 103 } 104 105 return 0; 106} 107 108static inline void update_stat(AudioStatsContext *s, ChannelStats *p, double d) 109{ 110 if (d < p->min) { 111 p->min = d; 112 p->min_run = 1; 113 p->min_runs = 0; 114 p->min_count = 1; 115 } else if (d == p->min) { 116 p->min_count++; 117 p->min_run = d == p->last ? p->min_run + 1 : 1; 118 } else if (p->last == p->min) { 119 p->min_runs += p->min_run * p->min_run; 120 } 121 122 if (d > p->max) { 123 p->max = d; 124 p->max_run = 1; 125 p->max_runs = 0; 126 p->max_count = 1; 127 } else if (d == p->max) { 128 p->max_count++; 129 p->max_run = d == p->last ? p->max_run + 1 : 1; 130 } else if (p->last == p->max) { 131 p->max_runs += p->max_run * p->max_run; 132 } 133 134 p->sigma_x += d; 135 p->sigma_x2 += d * d; 136 p->avg_sigma_x2 = p->avg_sigma_x2 * s->mult + (1.0 - s->mult) * d * d; 137 p->last = d; 138 139 if (p->nb_samples >= s->tc_samples) { 140 p->max_sigma_x2 = FFMAX(p->max_sigma_x2, p->avg_sigma_x2); 141 p->min_sigma_x2 = FFMIN(p->min_sigma_x2, p->avg_sigma_x2); 142 } 143 p->nb_samples++; 144} 145 146static int filter_frame(AVFilterLink *inlink, AVFrame *buf) 147{ 148 AudioStatsContext *s = inlink->dst->priv; 149 const int channels = s->nb_channels; 150 const double *src; 151 int i, c; 152 153 switch (inlink->format) { 154 case AV_SAMPLE_FMT_DBLP: 155 for (c = 0; c < channels; c++) { 156 ChannelStats *p = &s->chstats[c]; 157 src = (const double *)buf->extended_data[c]; 158 159 for (i = 0; i < buf->nb_samples; i++, src++) 160 update_stat(s, p, *src); 161 } 162 break; 163 case AV_SAMPLE_FMT_DBL: 164 src = (const double *)buf->extended_data[0]; 165 166 for (i = 0; i < buf->nb_samples; i++) { 167 for (c = 0; c < channels; c++, src++) 168 update_stat(s, &s->chstats[c], *src); 169 } 170 break; 171 } 172 173 return ff_filter_frame(inlink->dst->outputs[0], buf); 174} 175 176#define LINEAR_TO_DB(x) (log10(x) * 20) 177 178static void print_stats(AVFilterContext *ctx) 179{ 180 AudioStatsContext *s = ctx->priv; 181 uint64_t min_count = 0, max_count = 0, nb_samples = 0; 182 double min_runs = 0, max_runs = 0, 183 min = DBL_MAX, max = DBL_MIN, 184 max_sigma_x = 0, 185 sigma_x = 0, 186 sigma_x2 = 0, 187 min_sigma_x2 = DBL_MAX, 188 max_sigma_x2 = DBL_MIN; 189 int c; 190 191 for (c = 0; c < s->nb_channels; c++) { 192 ChannelStats *p = &s->chstats[c]; 193 194 if (p->nb_samples < s->tc_samples) 195 p->min_sigma_x2 = p->max_sigma_x2 = p->sigma_x2 / p->nb_samples; 196 197 min = FFMIN(min, p->min); 198 max = FFMAX(max, p->max); 199 min_sigma_x2 = FFMIN(min_sigma_x2, p->min_sigma_x2); 200 max_sigma_x2 = FFMAX(max_sigma_x2, p->max_sigma_x2); 201 sigma_x += p->sigma_x; 202 sigma_x2 += p->sigma_x2; 203 min_count += p->min_count; 204 max_count += p->max_count; 205 min_runs += p->min_runs; 206 max_runs += p->max_runs; 207 nb_samples += p->nb_samples; 208 if (fabs(p->sigma_x) > fabs(max_sigma_x)) 209 max_sigma_x = p->sigma_x; 210 211 av_log(ctx, AV_LOG_INFO, "Channel: %d\n", c + 1); 212 av_log(ctx, AV_LOG_INFO, "DC offset: %f\n", p->sigma_x / p->nb_samples); 213 av_log(ctx, AV_LOG_INFO, "Min level: %f\n", p->min); 214 av_log(ctx, AV_LOG_INFO, "Max level: %f\n", p->max); 215 av_log(ctx, AV_LOG_INFO, "Peak level dB: %f\n", LINEAR_TO_DB(FFMAX(-p->min, p->max))); 216 av_log(ctx, AV_LOG_INFO, "RMS level dB: %f\n", LINEAR_TO_DB(sqrt(p->sigma_x2 / p->nb_samples))); 217 av_log(ctx, AV_LOG_INFO, "RMS peak dB: %f\n", LINEAR_TO_DB(sqrt(p->max_sigma_x2))); 218 if (p->min_sigma_x2 != 1) 219 av_log(ctx, AV_LOG_INFO, "RMS trough dB: %f\n",LINEAR_TO_DB(sqrt(p->min_sigma_x2))); 220 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); 221 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))); 222 av_log(ctx, AV_LOG_INFO, "Peak count: %"PRId64"\n", p->min_count + p->max_count); 223 } 224 225 av_log(ctx, AV_LOG_INFO, "Overall\n"); 226 av_log(ctx, AV_LOG_INFO, "DC offset: %f\n", max_sigma_x / (nb_samples / s->nb_channels)); 227 av_log(ctx, AV_LOG_INFO, "Min level: %f\n", min); 228 av_log(ctx, AV_LOG_INFO, "Max level: %f\n", max); 229 av_log(ctx, AV_LOG_INFO, "Peak level dB: %f\n", LINEAR_TO_DB(FFMAX(-min, max))); 230 av_log(ctx, AV_LOG_INFO, "RMS level dB: %f\n", LINEAR_TO_DB(sqrt(sigma_x2 / nb_samples))); 231 av_log(ctx, AV_LOG_INFO, "RMS peak dB: %f\n", LINEAR_TO_DB(sqrt(max_sigma_x2))); 232 if (min_sigma_x2 != 1) 233 av_log(ctx, AV_LOG_INFO, "RMS trough dB: %f\n", LINEAR_TO_DB(sqrt(min_sigma_x2))); 234 av_log(ctx, AV_LOG_INFO, "Flat factor: %f\n", LINEAR_TO_DB((min_runs + max_runs) / (min_count + max_count))); 235 av_log(ctx, AV_LOG_INFO, "Peak count: %f\n", (min_count + max_count) / (double)s->nb_channels); 236 av_log(ctx, AV_LOG_INFO, "Number of samples: %"PRId64"\n", nb_samples / s->nb_channels); 237} 238 239static av_cold void uninit(AVFilterContext *ctx) 240{ 241 AudioStatsContext *s = ctx->priv; 242 243 print_stats(ctx); 244 av_freep(&s->chstats); 245} 246 247static const AVFilterPad astats_inputs[] = { 248 { 249 .name = "default", 250 .type = AVMEDIA_TYPE_AUDIO, 251 .filter_frame = filter_frame, 252 }, 253 { NULL } 254}; 255 256static const AVFilterPad astats_outputs[] = { 257 { 258 .name = "default", 259 .type = AVMEDIA_TYPE_AUDIO, 260 .config_props = config_output, 261 }, 262 { NULL } 263}; 264 265AVFilter ff_af_astats = { 266 .name = "astats", 267 .description = NULL_IF_CONFIG_SMALL("Show time domain statistics about audio frames."), 268 .query_formats = query_formats, 269 .priv_size = sizeof(AudioStatsContext), 270 .priv_class = &astats_class, 271 .uninit = uninit, 272 .inputs = astats_inputs, 273 .outputs = astats_outputs, 274}; 275