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af_astats.c
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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 
29 typedef struct ChannelStats {
30  double last;
31  double sigma_x, 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 
40 typedef struct {
41  const AVClass *class;
44  uint64_t tc_samples;
45  double time_constant;
46  double mult;
48 
49 #define OFFSET(x) offsetof(AudioStatsContext, x)
50 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
51 
52 static 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 
57 AVFILTER_DEFINE_CLASS(astats);
58 
60 {
63  static const enum AVSampleFormat sample_fmts[] = {
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 
86 static 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 
108 static 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) {
142  }
143  p->nb_samples++;
144 }
145 
146 static 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 
178 static 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 
239 static av_cold void uninit(AVFilterContext *ctx)
240 {
241  AudioStatsContext *s = ctx->priv;
242 
243  print_stats(ctx);
244  av_freep(&s->chstats);
245 }
246 
247 static const AVFilterPad astats_inputs[] = {
248  {
249  .name = "default",
250  .type = AVMEDIA_TYPE_AUDIO,
251  .filter_frame = filter_frame,
252  },
253  { NULL }
254 };
255 
256 static const AVFilterPad astats_outputs[] = {
257  {
258  .name = "default",
259  .type = AVMEDIA_TYPE_AUDIO,
260  .config_props = config_output,
261  },
262  { NULL }
263 };
264 
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 };