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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_INT, {.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_layouts();
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  memset(s->chstats, 0, sizeof(*s->chstats));
102 
103  for (c = 0; c < s->nb_channels; c++) {
104  ChannelStats *p = &s->chstats[c];
105 
106  p->min = p->min_sigma_x2 = DBL_MAX;
107  p->max = p->max_sigma_x2 = DBL_MIN;
108  p->min_diff = p->max_diff = -1;
109  }
110 }
111 
112 static int config_output(AVFilterLink *outlink)
113 {
114  AudioStatsContext *s = outlink->src->priv;
115 
116  s->chstats = av_calloc(sizeof(*s->chstats), outlink->channels);
117  if (!s->chstats)
118  return AVERROR(ENOMEM);
119  s->nb_channels = outlink->channels;
120  s->mult = exp((-1 / s->time_constant / outlink->sample_rate));
121  s->tc_samples = 5 * s->time_constant * outlink->sample_rate + .5;
122 
123  reset_stats(s);
124 
125  return 0;
126 }
127 
128 static unsigned bit_depth(uint64_t mask)
129 {
130  unsigned result = 64;
131 
132  for (; result && !(mask & 1); --result, mask >>= 1);
133 
134  return result;
135 }
136 
137 static inline void update_stat(AudioStatsContext *s, ChannelStats *p, double d)
138 {
139  if (d < p->min) {
140  p->min = d;
141  p->min_run = 1;
142  p->min_runs = 0;
143  p->min_count = 1;
144  } else if (d == p->min) {
145  p->min_count++;
146  p->min_run = d == p->last ? p->min_run + 1 : 1;
147  } else if (p->last == p->min) {
148  p->min_runs += p->min_run * p->min_run;
149  }
150 
151  if (d > p->max) {
152  p->max = d;
153  p->max_run = 1;
154  p->max_runs = 0;
155  p->max_count = 1;
156  } else if (d == p->max) {
157  p->max_count++;
158  p->max_run = d == p->last ? p->max_run + 1 : 1;
159  } else if (p->last == p->max) {
160  p->max_runs += p->max_run * p->max_run;
161  }
162 
163  p->sigma_x += d;
164  p->sigma_x2 += d * d;
165  p->avg_sigma_x2 = p->avg_sigma_x2 * s->mult + (1.0 - s->mult) * d * d;
166  p->min_diff = FFMIN(p->min_diff == -1 ? DBL_MAX : p->min_diff, FFABS(d - (p->min_diff == -1 ? DBL_MAX : p->last)));
167  p->max_diff = FFMAX(p->max_diff, FFABS(d - (p->max_diff == -1 ? d : p->last)));
168  p->diff1_sum += FFABS(d - p->last);
169  p->last = d;
170  p->mask |= llrint(d * (UINT64_C(1) << 63));
171 
172  if (p->nb_samples >= s->tc_samples) {
175  }
176  p->nb_samples++;
177 }
178 
179 static void set_meta(AVDictionary **metadata, int chan, const char *key,
180  const char *fmt, double val)
181 {
182  uint8_t value[128];
183  uint8_t key2[128];
184 
185  snprintf(value, sizeof(value), fmt, val);
186  if (chan)
187  snprintf(key2, sizeof(key2), "lavfi.astats.%d.%s", chan, key);
188  else
189  snprintf(key2, sizeof(key2), "lavfi.astats.%s", key);
190  av_dict_set(metadata, key2, value, 0);
191 }
192 
193 #define LINEAR_TO_DB(x) (log10(x) * 20)
194 
195 static void set_metadata(AudioStatsContext *s, AVDictionary **metadata)
196 {
197  uint64_t mask = 0, min_count = 0, max_count = 0, nb_samples = 0;
198  double min_runs = 0, max_runs = 0,
199  min = DBL_MAX, max = DBL_MIN, min_diff = DBL_MAX, max_diff = 0,
200  max_sigma_x = 0,
201  diff1_sum = 0,
202  sigma_x = 0,
203  sigma_x2 = 0,
204  min_sigma_x2 = DBL_MAX,
205  max_sigma_x2 = DBL_MIN;
206  int c;
207 
208  for (c = 0; c < s->nb_channels; c++) {
209  ChannelStats *p = &s->chstats[c];
210 
211  if (p->nb_samples < s->tc_samples)
212  p->min_sigma_x2 = p->max_sigma_x2 = p->sigma_x2 / p->nb_samples;
213 
214  min = FFMIN(min, p->min);
215  max = FFMAX(max, p->max);
216  min_diff = FFMIN(min_diff, p->min_diff);
217  max_diff = FFMAX(max_diff, p->max_diff);
218  diff1_sum += p->diff1_sum,
219  min_sigma_x2 = FFMIN(min_sigma_x2, p->min_sigma_x2);
220  max_sigma_x2 = FFMAX(max_sigma_x2, p->max_sigma_x2);
221  sigma_x += p->sigma_x;
222  sigma_x2 += p->sigma_x2;
223  min_count += p->min_count;
224  max_count += p->max_count;
225  min_runs += p->min_runs;
226  max_runs += p->max_runs;
227  mask |= p->mask;
228  nb_samples += p->nb_samples;
229  if (fabs(p->sigma_x) > fabs(max_sigma_x))
230  max_sigma_x = p->sigma_x;
231 
232  set_meta(metadata, c + 1, "DC_offset", "%f", p->sigma_x / p->nb_samples);
233  set_meta(metadata, c + 1, "Min_level", "%f", p->min);
234  set_meta(metadata, c + 1, "Max_level", "%f", p->max);
235  set_meta(metadata, c + 1, "Min_difference", "%f", p->min_diff);
236  set_meta(metadata, c + 1, "Max_difference", "%f", p->max_diff);
237  set_meta(metadata, c + 1, "Mean_difference", "%f", p->diff1_sum / (p->nb_samples - 1));
238  set_meta(metadata, c + 1, "Peak_level", "%f", LINEAR_TO_DB(FFMAX(-p->min, p->max)));
239  set_meta(metadata, c + 1, "RMS_level", "%f", LINEAR_TO_DB(sqrt(p->sigma_x2 / p->nb_samples)));
240  set_meta(metadata, c + 1, "RMS_peak", "%f", LINEAR_TO_DB(sqrt(p->max_sigma_x2)));
241  set_meta(metadata, c + 1, "RMS_trough", "%f", LINEAR_TO_DB(sqrt(p->min_sigma_x2)));
242  set_meta(metadata, c + 1, "Crest_factor", "%f", p->sigma_x2 ? FFMAX(-p->min, p->max) / sqrt(p->sigma_x2 / p->nb_samples) : 1);
243  set_meta(metadata, c + 1, "Flat_factor", "%f", LINEAR_TO_DB((p->min_runs + p->max_runs) / (p->min_count + p->max_count)));
244  set_meta(metadata, c + 1, "Peak_count", "%f", (float)(p->min_count + p->max_count));
245  set_meta(metadata, c + 1, "Bit_depth", "%f", bit_depth(p->mask));
246  }
247 
248  set_meta(metadata, 0, "Overall.DC_offset", "%f", max_sigma_x / (nb_samples / s->nb_channels));
249  set_meta(metadata, 0, "Overall.Min_level", "%f", min);
250  set_meta(metadata, 0, "Overall.Max_level", "%f", max);
251  set_meta(metadata, 0, "Overall.Min_difference", "%f", min_diff);
252  set_meta(metadata, 0, "Overall.Max_difference", "%f", max_diff);
253  set_meta(metadata, 0, "Overall.Mean_difference", "%f", diff1_sum / (nb_samples - s->nb_channels));
254  set_meta(metadata, 0, "Overall.Peak_level", "%f", LINEAR_TO_DB(FFMAX(-min, max)));
255  set_meta(metadata, 0, "Overall.RMS_level", "%f", LINEAR_TO_DB(sqrt(sigma_x2 / nb_samples)));
256  set_meta(metadata, 0, "Overall.RMS_peak", "%f", LINEAR_TO_DB(sqrt(max_sigma_x2)));
257  set_meta(metadata, 0, "Overall.RMS_trough", "%f", LINEAR_TO_DB(sqrt(min_sigma_x2)));
258  set_meta(metadata, 0, "Overall.Flat_factor", "%f", LINEAR_TO_DB((min_runs + max_runs) / (min_count + max_count)));
259  set_meta(metadata, 0, "Overall.Peak_count", "%f", (float)(min_count + max_count) / (double)s->nb_channels);
260  set_meta(metadata, 0, "Overall.Bit_depth", "%f", bit_depth(mask));
261  set_meta(metadata, 0, "Overall.Number_of_samples", "%f", nb_samples / s->nb_channels);
262 }
263 
264 static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
265 {
266  AudioStatsContext *s = inlink->dst->priv;
267  AVDictionary **metadata = avpriv_frame_get_metadatap(buf);
268  const int channels = s->nb_channels;
269  const double *src;
270  int i, c;
271 
272  switch (inlink->format) {
273  case AV_SAMPLE_FMT_DBLP:
274  for (c = 0; c < channels; c++) {
275  ChannelStats *p = &s->chstats[c];
276  src = (const double *)buf->extended_data[c];
277 
278  for (i = 0; i < buf->nb_samples; i++, src++)
279  update_stat(s, p, *src);
280  }
281  break;
282  case AV_SAMPLE_FMT_DBL:
283  src = (const double *)buf->extended_data[0];
284 
285  for (i = 0; i < buf->nb_samples; i++) {
286  for (c = 0; c < channels; c++, src++)
287  update_stat(s, &s->chstats[c], *src);
288  }
289  break;
290  }
291 
292  if (s->metadata)
293  set_metadata(s, metadata);
294 
295  if (s->reset_count > 0) {
296  s->nb_frames++;
297  if (s->nb_frames >= s->reset_count) {
298  reset_stats(s);
299  s->nb_frames = 0;
300  }
301  }
302 
303  return ff_filter_frame(inlink->dst->outputs[0], buf);
304 }
305 
306 static void print_stats(AVFilterContext *ctx)
307 {
308  AudioStatsContext *s = ctx->priv;
309  uint64_t mask = 0, min_count = 0, max_count = 0, nb_samples = 0;
310  double min_runs = 0, max_runs = 0,
311  min = DBL_MAX, max = DBL_MIN, min_diff = DBL_MAX, max_diff = 0,
312  max_sigma_x = 0,
313  diff1_sum = 0,
314  sigma_x = 0,
315  sigma_x2 = 0,
316  min_sigma_x2 = DBL_MAX,
317  max_sigma_x2 = DBL_MIN;
318  int c;
319 
320  for (c = 0; c < s->nb_channels; c++) {
321  ChannelStats *p = &s->chstats[c];
322 
323  if (p->nb_samples < s->tc_samples)
324  p->min_sigma_x2 = p->max_sigma_x2 = p->sigma_x2 / p->nb_samples;
325 
326  min = FFMIN(min, p->min);
327  max = FFMAX(max, p->max);
328  min_diff = FFMIN(min_diff, p->min_diff);
329  max_diff = FFMAX(max_diff, p->max_diff);
330  diff1_sum += p->diff1_sum,
331  min_sigma_x2 = FFMIN(min_sigma_x2, p->min_sigma_x2);
332  max_sigma_x2 = FFMAX(max_sigma_x2, p->max_sigma_x2);
333  sigma_x += p->sigma_x;
334  sigma_x2 += p->sigma_x2;
335  min_count += p->min_count;
336  max_count += p->max_count;
337  min_runs += p->min_runs;
338  max_runs += p->max_runs;
339  mask |= p->mask;
340  nb_samples += p->nb_samples;
341  if (fabs(p->sigma_x) > fabs(max_sigma_x))
342  max_sigma_x = p->sigma_x;
343 
344  av_log(ctx, AV_LOG_INFO, "Channel: %d\n", c + 1);
345  av_log(ctx, AV_LOG_INFO, "DC offset: %f\n", p->sigma_x / p->nb_samples);
346  av_log(ctx, AV_LOG_INFO, "Min level: %f\n", p->min);
347  av_log(ctx, AV_LOG_INFO, "Max level: %f\n", p->max);
348  av_log(ctx, AV_LOG_INFO, "Min difference: %f\n", p->min_diff);
349  av_log(ctx, AV_LOG_INFO, "Max difference: %f\n", p->max_diff);
350  av_log(ctx, AV_LOG_INFO, "Mean difference: %f\n", p->diff1_sum / (p->nb_samples - 1));
351  av_log(ctx, AV_LOG_INFO, "Peak level dB: %f\n", LINEAR_TO_DB(FFMAX(-p->min, p->max)));
352  av_log(ctx, AV_LOG_INFO, "RMS level dB: %f\n", LINEAR_TO_DB(sqrt(p->sigma_x2 / p->nb_samples)));
353  av_log(ctx, AV_LOG_INFO, "RMS peak dB: %f\n", LINEAR_TO_DB(sqrt(p->max_sigma_x2)));
354  if (p->min_sigma_x2 != 1)
355  av_log(ctx, AV_LOG_INFO, "RMS trough dB: %f\n",LINEAR_TO_DB(sqrt(p->min_sigma_x2)));
356  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);
357  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)));
358  av_log(ctx, AV_LOG_INFO, "Peak count: %"PRId64"\n", p->min_count + p->max_count);
359  av_log(ctx, AV_LOG_INFO, "Bit depth: %u\n", bit_depth(p->mask));
360  }
361 
362  av_log(ctx, AV_LOG_INFO, "Overall\n");
363  av_log(ctx, AV_LOG_INFO, "DC offset: %f\n", max_sigma_x / (nb_samples / s->nb_channels));
364  av_log(ctx, AV_LOG_INFO, "Min level: %f\n", min);
365  av_log(ctx, AV_LOG_INFO, "Max level: %f\n", max);
366  av_log(ctx, AV_LOG_INFO, "Min difference: %f\n", min_diff);
367  av_log(ctx, AV_LOG_INFO, "Max difference: %f\n", max_diff);
368  av_log(ctx, AV_LOG_INFO, "Mean difference: %f\n", diff1_sum / (nb_samples - s->nb_channels));
369  av_log(ctx, AV_LOG_INFO, "Peak level dB: %f\n", LINEAR_TO_DB(FFMAX(-min, max)));
370  av_log(ctx, AV_LOG_INFO, "RMS level dB: %f\n", LINEAR_TO_DB(sqrt(sigma_x2 / nb_samples)));
371  av_log(ctx, AV_LOG_INFO, "RMS peak dB: %f\n", LINEAR_TO_DB(sqrt(max_sigma_x2)));
372  if (min_sigma_x2 != 1)
373  av_log(ctx, AV_LOG_INFO, "RMS trough dB: %f\n", LINEAR_TO_DB(sqrt(min_sigma_x2)));
374  av_log(ctx, AV_LOG_INFO, "Flat factor: %f\n", LINEAR_TO_DB((min_runs + max_runs) / (min_count + max_count)));
375  av_log(ctx, AV_LOG_INFO, "Peak count: %f\n", (min_count + max_count) / (double)s->nb_channels);
376  av_log(ctx, AV_LOG_INFO, "Bit depth: %u\n", bit_depth(mask));
377  av_log(ctx, AV_LOG_INFO, "Number of samples: %"PRId64"\n", nb_samples / s->nb_channels);
378 }
379 
380 static av_cold void uninit(AVFilterContext *ctx)
381 {
382  AudioStatsContext *s = ctx->priv;
383 
384  if (s->nb_channels)
385  print_stats(ctx);
386  av_freep(&s->chstats);
387 }
388 
389 static const AVFilterPad astats_inputs[] = {
390  {
391  .name = "default",
392  .type = AVMEDIA_TYPE_AUDIO,
393  .filter_frame = filter_frame,
394  },
395  { NULL }
396 };
397 
398 static const AVFilterPad astats_outputs[] = {
399  {
400  .name = "default",
401  .type = AVMEDIA_TYPE_AUDIO,
402  .config_props = config_output,
403  },
404  { NULL }
405 };
406 
408  .name = "astats",
409  .description = NULL_IF_CONFIG_SMALL("Show time domain statistics about audio frames."),
410  .query_formats = query_formats,
411  .priv_size = sizeof(AudioStatsContext),
412  .priv_class = &astats_class,
413  .uninit = uninit,
414  .inputs = astats_inputs,
415  .outputs = astats_outputs,
416 };
#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:523
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:171
AVOption.
Definition: opt.h:255
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:407
static const AVFilterPad outputs[]
Definition: af_ashowinfo.c:248
static void update_stat(AudioStatsContext *s, ChannelStats *p, double d)
Definition: af_astats.c:137
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:71
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:69
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1158
uint8_t
#define av_cold
Definition: attributes.h:74
AVOptions.
static const AVFilterPad astats_inputs[]
Definition: af_astats.c:389
#define LINEAR_TO_DB(x)
Definition: af_astats.c:193
#define av_log(a,...)
A filter pad used for either input or output.
Definition: internal.h:63
static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
Definition: af_astats.c:264
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_astats.c:380
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:542
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:175
void * priv
private data for use by the filter
Definition: avfilter.h:654
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:79
double sigma_x2
Definition: af_astats.c:31
static void print_stats(AVFilterContext *ctx)
Definition: af_astats.c:306
#define FFMIN(a, b)
Definition: common.h:81
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
static int config_output(AVFilterLink *outlink)
Definition: af_astats.c:112
static void reset_stats(AudioStatsContext *s)
Definition: af_astats.c:97
uint64_t max_count
Definition: af_astats.c:39
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
Definition: common.h:68
double sigma_x
Definition: af_astats.c:31
AVFilterChannelLayouts * ff_all_channel_layouts(void)
Construct an empty AVFilterChannelLayouts/AVFilterFormats struct – representing any channel layout (w...
Definition: formats.c:385
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
uint64_t mask
Definition: af_astats.c:38
AVS_Value src
Definition: avisynth_c.h:482
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:59
AVDictionary ** avpriv_frame_get_metadatap(AVFrame *frame)
Definition: frame.c:47
void * buf
Definition: avisynth_c.h:553
#define llrint(x)
Definition: libm.h:112
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:470
double max_run
Definition: af_astats.c:34
static const AVFilterPad inputs[]
Definition: af_ashowinfo.c:239
double max_diff
Definition: af_astats.c:36
const char * name
Filter name.
Definition: avfilter.h:474
#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:648
enum MovChannelLayoutTag * layouts
Definition: mov_chan.c:434
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:379
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:128
static void set_meta(AVDictionary **metadata, int chan, const char *key, const char *fmt, double val)
Definition: af_astats.c:179
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:633
static const AVFilterPad astats_outputs[]
Definition: af_astats.c:398
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:701
#define av_freep(p)
internal API functions
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:215
float min
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:225
for(j=16;j >0;--j)
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:530
static void set_metadata(AudioStatsContext *s, AVDictionary **metadata)
Definition: af_astats.c:195