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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  double min_diff, max_diff;
37  double diff1_sum;
38  uint64_t mask;
39  uint64_t min_count, max_count;
40  uint64_t nb_samples;
41 } ChannelStats;
42 
43 typedef struct {
44  const AVClass *class;
47  uint64_t tc_samples;
48  double time_constant;
49  double mult;
50  int metadata;
52  int nb_frames;
54 
55 #define OFFSET(x) offsetof(AudioStatsContext, x)
56 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
57 
58 static const AVOption astats_options[] = {
59  { "length", "set the window length", OFFSET(time_constant), AV_OPT_TYPE_DOUBLE, {.dbl=.05}, .01, 10, FLAGS },
60  { "metadata", "inject metadata in the filtergraph", OFFSET(metadata), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, FLAGS },
61  { "reset", "recalculate stats after this many frames", OFFSET(reset_count), AV_OPT_TYPE_INT, {.i64=0}, 0, INT_MAX, FLAGS },
62  { NULL }
63 };
64 
65 AVFILTER_DEFINE_CLASS(astats);
66 
68 {
71  static const enum AVSampleFormat sample_fmts[] = {
74  };
75  int ret;
76 
77  layouts = ff_all_channel_counts();
78  if (!layouts)
79  return AVERROR(ENOMEM);
80  ret = ff_set_common_channel_layouts(ctx, layouts);
81  if (ret < 0)
82  return ret;
83 
84  formats = ff_make_format_list(sample_fmts);
85  if (!formats)
86  return AVERROR(ENOMEM);
87  ret = ff_set_common_formats(ctx, formats);
88  if (ret < 0)
89  return ret;
90 
91  formats = ff_all_samplerates();
92  if (!formats)
93  return AVERROR(ENOMEM);
94  return ff_set_common_samplerates(ctx, formats);
95 }
96 
98 {
99  int c;
100 
101  for (c = 0; c < s->nb_channels; c++) {
102  ChannelStats *p = &s->chstats[c];
103 
104  p->min = p->min_sigma_x2 = DBL_MAX;
105  p->max = p->max_sigma_x2 = DBL_MIN;
106  p->min_diff = DBL_MAX;
107  p->max_diff = DBL_MIN;
108  p->sigma_x = 0;
109  p->sigma_x2 = 0;
110  p->avg_sigma_x2 = 0;
111  p->min_sigma_x2 = 0;
112  p->max_sigma_x2 = 0;
113  p->min_run = 0;
114  p->max_run = 0;
115  p->min_runs = 0;
116  p->max_runs = 0;
117  p->diff1_sum = 0;
118  p->mask = 0;
119  p->min_count = 0;
120  p->max_count = 0;
121  p->nb_samples = 0;
122  }
123 }
124 
125 static int config_output(AVFilterLink *outlink)
126 {
127  AudioStatsContext *s = outlink->src->priv;
128 
129  s->chstats = av_calloc(sizeof(*s->chstats), outlink->channels);
130  if (!s->chstats)
131  return AVERROR(ENOMEM);
132  s->nb_channels = outlink->channels;
133  s->mult = exp((-1 / s->time_constant / outlink->sample_rate));
134  s->tc_samples = 5 * s->time_constant * outlink->sample_rate + .5;
135  s->nb_frames = 0;
136 
137  reset_stats(s);
138 
139  return 0;
140 }
141 
142 static unsigned bit_depth(uint64_t mask)
143 {
144  unsigned result = 64;
145 
146  for (; result && !(mask & 1); --result, mask >>= 1);
147 
148  return result;
149 }
150 
151 static inline void update_stat(AudioStatsContext *s, ChannelStats *p, double d)
152 {
153  if (d < p->min) {
154  p->min = d;
155  p->min_run = 1;
156  p->min_runs = 0;
157  p->min_count = 1;
158  } else if (d == p->min) {
159  p->min_count++;
160  p->min_run = d == p->last ? p->min_run + 1 : 1;
161  } else if (p->last == p->min) {
162  p->min_runs += p->min_run * p->min_run;
163  }
164 
165  if (d > p->max) {
166  p->max = d;
167  p->max_run = 1;
168  p->max_runs = 0;
169  p->max_count = 1;
170  } else if (d == p->max) {
171  p->max_count++;
172  p->max_run = d == p->last ? p->max_run + 1 : 1;
173  } else if (p->last == p->max) {
174  p->max_runs += p->max_run * p->max_run;
175  }
176 
177  p->sigma_x += d;
178  p->sigma_x2 += d * d;
179  p->avg_sigma_x2 = p->avg_sigma_x2 * s->mult + (1.0 - s->mult) * d * d;
180  p->min_diff = FFMIN(p->min_diff, fabs(d - p->last));
181  p->max_diff = FFMAX(p->max_diff, fabs(d - p->last));
182  p->diff1_sum += fabs(d - p->last);
183  p->last = d;
184  p->mask |= llrint(d * (UINT64_C(1) << 63));
185 
186  if (p->nb_samples >= s->tc_samples) {
189  }
190  p->nb_samples++;
191 }
192 
193 static void set_meta(AVDictionary **metadata, int chan, const char *key,
194  const char *fmt, double val)
195 {
196  uint8_t value[128];
197  uint8_t key2[128];
198 
199  snprintf(value, sizeof(value), fmt, val);
200  if (chan)
201  snprintf(key2, sizeof(key2), "lavfi.astats.%d.%s", chan, key);
202  else
203  snprintf(key2, sizeof(key2), "lavfi.astats.%s", key);
204  av_dict_set(metadata, key2, value, 0);
205 }
206 
207 #define LINEAR_TO_DB(x) (log10(x) * 20)
208 
209 static void set_metadata(AudioStatsContext *s, AVDictionary **metadata)
210 {
211  uint64_t mask = 0, min_count = 0, max_count = 0, nb_samples = 0;
212  double min_runs = 0, max_runs = 0,
213  min = DBL_MAX, max = DBL_MIN, min_diff = DBL_MAX, max_diff = 0,
214  max_sigma_x = 0,
215  diff1_sum = 0,
216  sigma_x = 0,
217  sigma_x2 = 0,
218  min_sigma_x2 = DBL_MAX,
219  max_sigma_x2 = DBL_MIN;
220  int c;
221 
222  for (c = 0; c < s->nb_channels; c++) {
223  ChannelStats *p = &s->chstats[c];
224 
225  if (p->nb_samples < s->tc_samples)
226  p->min_sigma_x2 = p->max_sigma_x2 = p->sigma_x2 / p->nb_samples;
227 
228  min = FFMIN(min, p->min);
229  max = FFMAX(max, p->max);
230  min_diff = FFMIN(min_diff, p->min_diff);
231  max_diff = FFMAX(max_diff, p->max_diff);
232  diff1_sum += p->diff1_sum,
233  min_sigma_x2 = FFMIN(min_sigma_x2, p->min_sigma_x2);
234  max_sigma_x2 = FFMAX(max_sigma_x2, p->max_sigma_x2);
235  sigma_x += p->sigma_x;
236  sigma_x2 += p->sigma_x2;
237  min_count += p->min_count;
238  max_count += p->max_count;
239  min_runs += p->min_runs;
240  max_runs += p->max_runs;
241  mask |= p->mask;
242  nb_samples += p->nb_samples;
243  if (fabs(p->sigma_x) > fabs(max_sigma_x))
244  max_sigma_x = p->sigma_x;
245 
246  set_meta(metadata, c + 1, "DC_offset", "%f", p->sigma_x / p->nb_samples);
247  set_meta(metadata, c + 1, "Min_level", "%f", p->min);
248  set_meta(metadata, c + 1, "Max_level", "%f", p->max);
249  set_meta(metadata, c + 1, "Min_difference", "%f", p->min_diff);
250  set_meta(metadata, c + 1, "Max_difference", "%f", p->max_diff);
251  set_meta(metadata, c + 1, "Mean_difference", "%f", p->diff1_sum / (p->nb_samples - 1));
252  set_meta(metadata, c + 1, "Peak_level", "%f", LINEAR_TO_DB(FFMAX(-p->min, p->max)));
253  set_meta(metadata, c + 1, "RMS_level", "%f", LINEAR_TO_DB(sqrt(p->sigma_x2 / p->nb_samples)));
254  set_meta(metadata, c + 1, "RMS_peak", "%f", LINEAR_TO_DB(sqrt(p->max_sigma_x2)));
255  set_meta(metadata, c + 1, "RMS_trough", "%f", LINEAR_TO_DB(sqrt(p->min_sigma_x2)));
256  set_meta(metadata, c + 1, "Crest_factor", "%f", p->sigma_x2 ? FFMAX(-p->min, p->max) / sqrt(p->sigma_x2 / p->nb_samples) : 1);
257  set_meta(metadata, c + 1, "Flat_factor", "%f", LINEAR_TO_DB((p->min_runs + p->max_runs) / (p->min_count + p->max_count)));
258  set_meta(metadata, c + 1, "Peak_count", "%f", (float)(p->min_count + p->max_count));
259  set_meta(metadata, c + 1, "Bit_depth", "%f", bit_depth(p->mask));
260  }
261 
262  set_meta(metadata, 0, "Overall.DC_offset", "%f", max_sigma_x / (nb_samples / s->nb_channels));
263  set_meta(metadata, 0, "Overall.Min_level", "%f", min);
264  set_meta(metadata, 0, "Overall.Max_level", "%f", max);
265  set_meta(metadata, 0, "Overall.Min_difference", "%f", min_diff);
266  set_meta(metadata, 0, "Overall.Max_difference", "%f", max_diff);
267  set_meta(metadata, 0, "Overall.Mean_difference", "%f", diff1_sum / (nb_samples - s->nb_channels));
268  set_meta(metadata, 0, "Overall.Peak_level", "%f", LINEAR_TO_DB(FFMAX(-min, max)));
269  set_meta(metadata, 0, "Overall.RMS_level", "%f", LINEAR_TO_DB(sqrt(sigma_x2 / nb_samples)));
270  set_meta(metadata, 0, "Overall.RMS_peak", "%f", LINEAR_TO_DB(sqrt(max_sigma_x2)));
271  set_meta(metadata, 0, "Overall.RMS_trough", "%f", LINEAR_TO_DB(sqrt(min_sigma_x2)));
272  set_meta(metadata, 0, "Overall.Flat_factor", "%f", LINEAR_TO_DB((min_runs + max_runs) / (min_count + max_count)));
273  set_meta(metadata, 0, "Overall.Peak_count", "%f", (float)(min_count + max_count) / (double)s->nb_channels);
274  set_meta(metadata, 0, "Overall.Bit_depth", "%f", bit_depth(mask));
275  set_meta(metadata, 0, "Overall.Number_of_samples", "%f", nb_samples / s->nb_channels);
276 }
277 
278 static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
279 {
280  AudioStatsContext *s = inlink->dst->priv;
281  AVDictionary **metadata = avpriv_frame_get_metadatap(buf);
282  const int channels = s->nb_channels;
283  const double *src;
284  int i, c;
285 
286  if (s->reset_count > 0) {
287  if (s->nb_frames >= s->reset_count) {
288  reset_stats(s);
289  s->nb_frames = 0;
290  }
291  s->nb_frames++;
292  }
293 
294  switch (inlink->format) {
295  case AV_SAMPLE_FMT_DBLP:
296  for (c = 0; c < channels; c++) {
297  ChannelStats *p = &s->chstats[c];
298  src = (const double *)buf->extended_data[c];
299 
300  for (i = 0; i < buf->nb_samples; i++, src++)
301  update_stat(s, p, *src);
302  }
303  break;
304  case AV_SAMPLE_FMT_DBL:
305  src = (const double *)buf->extended_data[0];
306 
307  for (i = 0; i < buf->nb_samples; i++) {
308  for (c = 0; c < channels; c++, src++)
309  update_stat(s, &s->chstats[c], *src);
310  }
311  break;
312  }
313 
314  if (s->metadata)
315  set_metadata(s, metadata);
316 
317  return ff_filter_frame(inlink->dst->outputs[0], buf);
318 }
319 
321 {
322  AudioStatsContext *s = ctx->priv;
323  uint64_t mask = 0, min_count = 0, max_count = 0, nb_samples = 0;
324  double min_runs = 0, max_runs = 0,
325  min = DBL_MAX, max = DBL_MIN, min_diff = DBL_MAX, max_diff = 0,
326  max_sigma_x = 0,
327  diff1_sum = 0,
328  sigma_x = 0,
329  sigma_x2 = 0,
330  min_sigma_x2 = DBL_MAX,
331  max_sigma_x2 = DBL_MIN;
332  int c;
333 
334  for (c = 0; c < s->nb_channels; c++) {
335  ChannelStats *p = &s->chstats[c];
336 
337  if (p->nb_samples < s->tc_samples)
338  p->min_sigma_x2 = p->max_sigma_x2 = p->sigma_x2 / p->nb_samples;
339 
340  min = FFMIN(min, p->min);
341  max = FFMAX(max, p->max);
342  min_diff = FFMIN(min_diff, p->min_diff);
343  max_diff = FFMAX(max_diff, p->max_diff);
344  diff1_sum += p->diff1_sum,
345  min_sigma_x2 = FFMIN(min_sigma_x2, p->min_sigma_x2);
346  max_sigma_x2 = FFMAX(max_sigma_x2, p->max_sigma_x2);
347  sigma_x += p->sigma_x;
348  sigma_x2 += p->sigma_x2;
349  min_count += p->min_count;
350  max_count += p->max_count;
351  min_runs += p->min_runs;
352  max_runs += p->max_runs;
353  mask |= p->mask;
354  nb_samples += p->nb_samples;
355  if (fabs(p->sigma_x) > fabs(max_sigma_x))
356  max_sigma_x = p->sigma_x;
357 
358  av_log(ctx, AV_LOG_INFO, "Channel: %d\n", c + 1);
359  av_log(ctx, AV_LOG_INFO, "DC offset: %f\n", p->sigma_x / p->nb_samples);
360  av_log(ctx, AV_LOG_INFO, "Min level: %f\n", p->min);
361  av_log(ctx, AV_LOG_INFO, "Max level: %f\n", p->max);
362  av_log(ctx, AV_LOG_INFO, "Min difference: %f\n", p->min_diff);
363  av_log(ctx, AV_LOG_INFO, "Max difference: %f\n", p->max_diff);
364  av_log(ctx, AV_LOG_INFO, "Mean difference: %f\n", p->diff1_sum / (p->nb_samples - 1));
365  av_log(ctx, AV_LOG_INFO, "Peak level dB: %f\n", LINEAR_TO_DB(FFMAX(-p->min, p->max)));
366  av_log(ctx, AV_LOG_INFO, "RMS level dB: %f\n", LINEAR_TO_DB(sqrt(p->sigma_x2 / p->nb_samples)));
367  av_log(ctx, AV_LOG_INFO, "RMS peak dB: %f\n", LINEAR_TO_DB(sqrt(p->max_sigma_x2)));
368  if (p->min_sigma_x2 != 1)
369  av_log(ctx, AV_LOG_INFO, "RMS trough dB: %f\n",LINEAR_TO_DB(sqrt(p->min_sigma_x2)));
370  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);
371  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)));
372  av_log(ctx, AV_LOG_INFO, "Peak count: %"PRId64"\n", p->min_count + p->max_count);
373  av_log(ctx, AV_LOG_INFO, "Bit depth: %u\n", bit_depth(p->mask));
374  }
375 
376  av_log(ctx, AV_LOG_INFO, "Overall\n");
377  av_log(ctx, AV_LOG_INFO, "DC offset: %f\n", max_sigma_x / (nb_samples / s->nb_channels));
378  av_log(ctx, AV_LOG_INFO, "Min level: %f\n", min);
379  av_log(ctx, AV_LOG_INFO, "Max level: %f\n", max);
380  av_log(ctx, AV_LOG_INFO, "Min difference: %f\n", min_diff);
381  av_log(ctx, AV_LOG_INFO, "Max difference: %f\n", max_diff);
382  av_log(ctx, AV_LOG_INFO, "Mean difference: %f\n", diff1_sum / (nb_samples - s->nb_channels));
383  av_log(ctx, AV_LOG_INFO, "Peak level dB: %f\n", LINEAR_TO_DB(FFMAX(-min, max)));
384  av_log(ctx, AV_LOG_INFO, "RMS level dB: %f\n", LINEAR_TO_DB(sqrt(sigma_x2 / nb_samples)));
385  av_log(ctx, AV_LOG_INFO, "RMS peak dB: %f\n", LINEAR_TO_DB(sqrt(max_sigma_x2)));
386  if (min_sigma_x2 != 1)
387  av_log(ctx, AV_LOG_INFO, "RMS trough dB: %f\n", LINEAR_TO_DB(sqrt(min_sigma_x2)));
388  av_log(ctx, AV_LOG_INFO, "Flat factor: %f\n", LINEAR_TO_DB((min_runs + max_runs) / (min_count + max_count)));
389  av_log(ctx, AV_LOG_INFO, "Peak count: %f\n", (min_count + max_count) / (double)s->nb_channels);
390  av_log(ctx, AV_LOG_INFO, "Bit depth: %u\n", bit_depth(mask));
391  av_log(ctx, AV_LOG_INFO, "Number of samples: %"PRId64"\n", nb_samples / s->nb_channels);
392 }
393 
395 {
396  AudioStatsContext *s = ctx->priv;
397 
398  if (s->nb_channels)
399  print_stats(ctx);
400  av_freep(&s->chstats);
401 }
402 
403 static const AVFilterPad astats_inputs[] = {
404  {
405  .name = "default",
406  .type = AVMEDIA_TYPE_AUDIO,
407  .filter_frame = filter_frame,
408  },
409  { NULL }
410 };
411 
412 static const AVFilterPad astats_outputs[] = {
413  {
414  .name = "default",
415  .type = AVMEDIA_TYPE_AUDIO,
416  .config_props = config_output,
417  },
418  { NULL }
419 };
420 
422  .name = "astats",
423  .description = NULL_IF_CONFIG_SMALL("Show time domain statistics about audio frames."),
424  .query_formats = query_formats,
425  .priv_size = sizeof(AudioStatsContext),
426  .priv_class = &astats_class,
427  .uninit = uninit,
428  .inputs = astats_inputs,
429  .outputs = astats_outputs,
430 };
#define NULL
Definition: coverity.c:32
int ff_set_common_channel_layouts(AVFilterContext *ctx, AVFilterChannelLayouts *layouts)
A helper for query_formats() which sets all links to the same list of channel layouts/sample rates...
Definition: formats.c:549
const char const char void * val
Definition: avisynth_c.h:634
const char * s
Definition: avisynth_c.h:631
This structure describes decoded (raw) audio or video data.
Definition: frame.h:184
AVOption.
Definition: opt.h:245
const char * fmt
Definition: avisynth_c.h:632
static int query_formats(AVFilterContext *ctx)
Definition: af_astats.c:67
AVFilter ff_af_astats
Definition: af_astats.c:421
static void update_stat(AudioStatsContext *s, ChannelStats *p, double d)
Definition: af_astats.c:151
Main libavfilter public API header.
#define OFFSET(x)
Definition: af_astats.c:55
double min_run
Definition: af_astats.c:34
double min
Definition: af_astats.c:33
double, planar
Definition: samplefmt.h:70
double max_sigma_x2
Definition: af_astats.c:32
static enum AVSampleFormat formats[]
AVFilterFormats * ff_make_format_list(const int *fmts)
Create a list of supported formats.
Definition: formats.c:283
ChannelStats * chstats
Definition: af_astats.c:45
const char * name
Pad name.
Definition: internal.h:59
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1180
uint8_t
#define av_cold
Definition: attributes.h:82
AVOptions.
static const AVFilterPad astats_inputs[]
Definition: af_astats.c:403
#define LINEAR_TO_DB(x)
Definition: af_astats.c:207
#define av_log(a,...)
A filter pad used for either input or output.
Definition: internal.h:53
static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
Definition: af_astats.c:278
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_astats.c:394
int ff_set_common_formats(AVFilterContext *ctx, AVFilterFormats *formats)
A helper for query_formats() which sets all links to the same list of formats.
Definition: formats.c:568
static const uint16_t mask[17]
Definition: lzw.c:38
#define AVERROR(e)
Definition: error.h:43
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:176
void * priv
private data for use by the filter
Definition: avfilter.h:320
AVFILTER_DEFINE_CLASS(astats)
uint64_t tc_samples
Definition: af_astats.c:47
double max
Definition: af_astats.c:33
#define FFMAX(a, b)
Definition: common.h:94
int8_t exp
Definition: eval.c:64
double sigma_x2
Definition: af_astats.c:31
static void print_stats(AVFilterContext *ctx)
Definition: af_astats.c:320
#define FFMIN(a, b)
Definition: common.h:96
double min_sigma_x2
Definition: af_astats.c:32
GLsizei GLboolean const GLfloat * value
Definition: opengl_enc.c:109
double max_runs
Definition: af_astats.c:35
AVFormatContext * ctx
Definition: movenc.c:48
static int config_output(AVFilterLink *outlink)
Definition: af_astats.c:125
static void reset_stats(AudioStatsContext *s)
Definition: af_astats.c:97
uint64_t max_count
Definition: af_astats.c:39
static const AVFilterPad outputs[]
Definition: af_afftfilt.c:386
#define src
Definition: vp9dsp.c:530
double sigma_x
Definition: af_astats.c:31
A list of supported channel layouts.
Definition: formats.h:85
static const AVOption astats_options[]
Definition: af_astats.c:58
double min_diff
Definition: af_astats.c:36
#define AV_LOG_INFO
Standard information.
Definition: log.h:187
static const AVFilterPad inputs[]
Definition: af_afftfilt.c:376
uint64_t mask
Definition: af_astats.c:38
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
AVDictionary ** avpriv_frame_get_metadatap(AVFrame *frame)
Definition: frame.c:46
void * buf
Definition: avisynth_c.h:553
#define llrint(x)
Definition: libm.h:394
int av_dict_set(AVDictionary **pm, const char *key, const char *value, int flags)
Set the given entry in *pm, overwriting an existing entry.
Definition: dict.c:69
double avg_sigma_x2
Definition: af_astats.c:32
Describe the class of an AVClass context structure.
Definition: log.h:67
Filter definition.
Definition: avfilter.h:142
double max_run
Definition: af_astats.c:34
double max_diff
Definition: af_astats.c:36
const char * name
Filter name.
Definition: avfilter.h:146
#define snprintf
Definition: snprintf.h:34
#define FLAGS
Definition: af_astats.c:56
double last
Definition: af_astats.c:30
AVFilterLink ** outputs
array of pointers to output links
Definition: avfilter.h:317
enum MovChannelLayoutTag * layouts
Definition: mov_chan.c:434
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:395
void * av_calloc(size_t nmemb, size_t size)
Allocate a block of nmemb * size bytes with alignment suitable for all memory accesses (including vec...
Definition: mem.c:260
double time_constant
Definition: af_astats.c:48
uint64_t nb_samples
Definition: af_astats.c:40
static double c[64]
static unsigned bit_depth(uint64_t mask)
Definition: af_astats.c:142
static void set_meta(AVDictionary **metadata, int chan, const char *key, const char *fmt, double val)
Definition: af_astats.c:193
uint64_t min_count
Definition: af_astats.c:39
double min_runs
Definition: af_astats.c:35
double diff1_sum
Definition: af_astats.c:37
A list of supported formats for one end of a filter link.
Definition: formats.h:64
An instance of a filter.
Definition: avfilter.h:305
static const AVFilterPad astats_outputs[]
Definition: af_astats.c:412
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:701
#define av_freep(p)
internal API functions
AVFilterChannelLayouts * ff_all_channel_counts(void)
Construct an AVFilterChannelLayouts coding for any channel layout, with known or unknown disposition...
Definition: formats.c:410
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:231
float min
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:241
for(j=16;j >0;--j)
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:556
static void set_metadata(AudioStatsContext *s, AVDictionary **metadata)
Definition: af_astats.c:209