FFmpeg  4.3.8
af_axcorrelate.c
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1 /*
2  * Copyright (c) 2019 Paul B Mahol
3  *
4  * This file is part of FFmpeg.
5  *
6  * FFmpeg is free software; you can redistribute it and/or
7  * modify it under the terms of the GNU Lesser General Public
8  * License as published by the Free Software Foundation; either
9  * version 2.1 of the License, or (at your option) any later version.
10  *
11  * FFmpeg is distributed in the hope that it will be useful,
12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14  * Lesser General Public License for more details.
15  *
16  * You should have received a copy of the GNU Lesser General Public
17  * License along with FFmpeg; if not, write to the Free Software
18  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19  */
20 
21 #include "libavutil/avassert.h"
22 #include "libavutil/audio_fifo.h"
24 #include "libavutil/common.h"
25 #include "libavutil/opt.h"
26 
27 #include "audio.h"
28 #include "avfilter.h"
29 #include "formats.h"
30 #include "filters.h"
31 #include "internal.h"
32 
33 typedef struct AudioXCorrelateContext {
34  const AVClass *class;
35 
36  int size;
37  int algo;
39 
45  int used;
46 
49 
51 {
54  static const enum AVSampleFormat sample_fmts[] = {
57  };
58  int ret;
59 
60  layouts = ff_all_channel_counts();
61  if (!layouts)
62  return AVERROR(ENOMEM);
63  ret = ff_set_common_channel_layouts(ctx, layouts);
64  if (ret < 0)
65  return ret;
66 
67  formats = ff_make_format_list(sample_fmts);
68  if (!formats)
69  return AVERROR(ENOMEM);
70  ret = ff_set_common_formats(ctx, formats);
71  if (ret < 0)
72  return ret;
73 
74  formats = ff_all_samplerates();
75  if (!formats)
76  return AVERROR(ENOMEM);
77  return ff_set_common_samplerates(ctx, formats);
78 }
79 
80 static float mean_sum(const float *in, int size)
81 {
82  float mean_sum = 0.f;
83 
84  for (int i = 0; i < size; i++)
85  mean_sum += in[i];
86 
87  return mean_sum;
88 }
89 
90 static float square_sum(const float *x, const float *y, int size)
91 {
92  float square_sum = 0.f;
93 
94  for (int i = 0; i < size; i++)
95  square_sum += x[i] * y[i];
96 
97  return square_sum;
98 }
99 
100 static float xcorrelate(const float *x, const float *y, float sumx, float sumy, int size)
101 {
102  const float xm = sumx / size, ym = sumy / size;
103  float num = 0.f, den, den0 = 0.f, den1 = 0.f;
104 
105  for (int i = 0; i < size; i++) {
106  float xd = x[i] - xm;
107  float yd = y[i] - ym;
108 
109  num += xd * yd;
110  den0 += xd * xd;
111  den1 += yd * yd;
112  }
113 
114  num /= size;
115  den = sqrtf((den0 * den1) / (size * size));
116 
117  return den <= 1e-6f ? 0.f : num / den;
118 }
119 
121 {
122  AudioXCorrelateContext *s = ctx->priv;
123  const int size = s->size;
124  int used;
125 
126  for (int ch = 0; ch < out->channels; ch++) {
127  const float *x = (const float *)s->cache[0]->extended_data[ch];
128  const float *y = (const float *)s->cache[1]->extended_data[ch];
129  float *sumx = (float *)s->mean_sum[0]->extended_data[ch];
130  float *sumy = (float *)s->mean_sum[1]->extended_data[ch];
131  float *dst = (float *)out->extended_data[ch];
132 
133  used = s->used;
134  if (!used) {
135  sumx[0] = mean_sum(x, size);
136  sumy[0] = mean_sum(y, size);
137  used = 1;
138  }
139 
140  for (int n = 0; n < out->nb_samples; n++) {
141  dst[n] = xcorrelate(x + n, y + n, sumx[0], sumy[0], size);
142 
143  sumx[0] -= x[n];
144  sumx[0] += x[n + size];
145  sumy[0] -= y[n];
146  sumy[0] += y[n + size];
147  }
148  }
149 
150  return used;
151 }
152 
154 {
155  AudioXCorrelateContext *s = ctx->priv;
156  const int size = s->size;
157  int used;
158 
159  for (int ch = 0; ch < out->channels; ch++) {
160  const float *x = (const float *)s->cache[0]->extended_data[ch];
161  const float *y = (const float *)s->cache[1]->extended_data[ch];
162  float *num_sum = (float *)s->num_sum->extended_data[ch];
163  float *den_sumx = (float *)s->den_sum[0]->extended_data[ch];
164  float *den_sumy = (float *)s->den_sum[1]->extended_data[ch];
165  float *dst = (float *)out->extended_data[ch];
166 
167  used = s->used;
168  if (!used) {
169  num_sum[0] = square_sum(x, y, size);
170  den_sumx[0] = square_sum(x, x, size);
171  den_sumy[0] = square_sum(y, y, size);
172  used = 1;
173  }
174 
175  for (int n = 0; n < out->nb_samples; n++) {
176  float num, den;
177 
178  num = num_sum[0] / size;
179  den = sqrtf((den_sumx[0] * den_sumy[0]) / (size * size));
180 
181  dst[n] = den <= 1e-6f ? 0.f : num / den;
182 
183  num_sum[0] -= x[n] * y[n];
184  num_sum[0] += x[n + size] * y[n + size];
185  den_sumx[0] -= x[n] * x[n];
186  den_sumx[0] = FFMAX(den_sumx[0], 0.f);
187  den_sumx[0] += x[n + size] * x[n + size];
188  den_sumy[0] -= y[n] * y[n];
189  den_sumy[0] = FFMAX(den_sumy[0], 0.f);
190  den_sumy[0] += y[n + size] * y[n + size];
191  }
192  }
193 
194  return used;
195 }
196 
198 {
199  AudioXCorrelateContext *s = ctx->priv;
200  AVFrame *frame = NULL;
201  int ret, status;
202  int available;
203  int64_t pts;
204 
206 
207  for (int i = 0; i < 2; i++) {
208  ret = ff_inlink_consume_frame(ctx->inputs[i], &frame);
209  if (ret > 0) {
210  if (s->pts == AV_NOPTS_VALUE)
211  s->pts = frame->pts;
212  ret = av_audio_fifo_write(s->fifo[i], (void **)frame->extended_data,
213  frame->nb_samples);
214  av_frame_free(&frame);
215  if (ret < 0)
216  return ret;
217  }
218  }
219 
220  available = FFMIN(av_audio_fifo_size(s->fifo[0]), av_audio_fifo_size(s->fifo[1]));
221  if (available > s->size) {
222  const int out_samples = available - s->size;
223  AVFrame *out;
224 
225  if (!s->cache[0] || s->cache[0]->nb_samples < available) {
226  av_frame_free(&s->cache[0]);
227  s->cache[0] = ff_get_audio_buffer(ctx->outputs[0], available);
228  if (!s->cache[0])
229  return AVERROR(ENOMEM);
230  }
231 
232  if (!s->cache[1] || s->cache[1]->nb_samples < available) {
233  av_frame_free(&s->cache[1]);
234  s->cache[1] = ff_get_audio_buffer(ctx->outputs[0], available);
235  if (!s->cache[1])
236  return AVERROR(ENOMEM);
237  }
238 
239  ret = av_audio_fifo_peek(s->fifo[0], (void **)s->cache[0]->extended_data, available);
240  if (ret < 0)
241  return ret;
242 
243  ret = av_audio_fifo_peek(s->fifo[1], (void **)s->cache[1]->extended_data, available);
244  if (ret < 0)
245  return ret;
246 
247  out = ff_get_audio_buffer(ctx->outputs[0], out_samples);
248  if (!out)
249  return AVERROR(ENOMEM);
250 
251  s->used = s->xcorrelate(ctx, out);
252 
253  out->pts = s->pts;
254  s->pts += out_samples;
255 
256  av_audio_fifo_drain(s->fifo[0], out_samples);
257  av_audio_fifo_drain(s->fifo[1], out_samples);
258 
259  return ff_filter_frame(ctx->outputs[0], out);
260  }
261 
262  if (av_audio_fifo_size(s->fifo[0]) > s->size &&
263  av_audio_fifo_size(s->fifo[1]) > s->size) {
264  ff_filter_set_ready(ctx, 10);
265  return 0;
266  }
267 
268  for (int i = 0; i < 2; i++) {
269  if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) {
270  ff_outlink_set_status(ctx->outputs[0], status, pts);
271  return 0;
272  }
273  }
274 
275  if (ff_outlink_frame_wanted(ctx->outputs[0])) {
276  for (int i = 0; i < 2; i++) {
277  if (av_audio_fifo_size(s->fifo[i]) > s->size)
278  continue;
280  return 0;
281  }
282  }
283 
284  return FFERROR_NOT_READY;
285 }
286 
287 static int config_output(AVFilterLink *outlink)
288 {
289  AVFilterContext *ctx = outlink->src;
290  AVFilterLink *inlink = ctx->inputs[0];
291  AudioXCorrelateContext *s = ctx->priv;
292 
293  s->pts = AV_NOPTS_VALUE;
294 
295  outlink->format = inlink->format;
296  outlink->channels = inlink->channels;
297  s->fifo[0] = av_audio_fifo_alloc(outlink->format, outlink->channels, s->size);
298  s->fifo[1] = av_audio_fifo_alloc(outlink->format, outlink->channels, s->size);
299  if (!s->fifo[0] || !s->fifo[1])
300  return AVERROR(ENOMEM);
301 
302  s->mean_sum[0] = ff_get_audio_buffer(outlink, 1);
303  s->mean_sum[1] = ff_get_audio_buffer(outlink, 1);
304  s->num_sum = ff_get_audio_buffer(outlink, 1);
305  s->den_sum[0] = ff_get_audio_buffer(outlink, 1);
306  s->den_sum[1] = ff_get_audio_buffer(outlink, 1);
307  if (!s->mean_sum[0] || !s->mean_sum[1] || !s->num_sum ||
308  !s->den_sum[0] || !s->den_sum[1])
309  return AVERROR(ENOMEM);
310 
311  switch (s->algo) {
312  case 0: s->xcorrelate = xcorrelate_slow; break;
313  case 1: s->xcorrelate = xcorrelate_fast; break;
314  }
315 
316  return 0;
317 }
318 
320 {
321  AudioXCorrelateContext *s = ctx->priv;
322 
323  av_audio_fifo_free(s->fifo[0]);
324  av_audio_fifo_free(s->fifo[1]);
325  av_frame_free(&s->cache[0]);
326  av_frame_free(&s->cache[1]);
327  av_frame_free(&s->mean_sum[0]);
328  av_frame_free(&s->mean_sum[1]);
329  av_frame_free(&s->num_sum);
330  av_frame_free(&s->den_sum[0]);
331  av_frame_free(&s->den_sum[1]);
332 }
333 
334 static const AVFilterPad inputs[] = {
335  {
336  .name = "axcorrelate0",
337  .type = AVMEDIA_TYPE_AUDIO,
338  },
339  {
340  .name = "axcorrelate1",
341  .type = AVMEDIA_TYPE_AUDIO,
342  },
343  { NULL }
344 };
345 
346 static const AVFilterPad outputs[] = {
347  {
348  .name = "default",
349  .type = AVMEDIA_TYPE_AUDIO,
350  .config_props = config_output,
351  },
352  { NULL }
353 };
354 
355 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
356 #define OFFSET(x) offsetof(AudioXCorrelateContext, x)
357 
358 static const AVOption axcorrelate_options[] = {
359  { "size", "set segment size", OFFSET(size), AV_OPT_TYPE_INT, {.i64=256}, 2, 131072, AF },
360  { "algo", "set alghorithm", OFFSET(algo), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, AF, "algo" },
361  { "slow", "slow algorithm", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "algo" },
362  { "fast", "fast algorithm", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "algo" },
363  { NULL }
364 };
365 
366 AVFILTER_DEFINE_CLASS(axcorrelate);
367 
369  .name = "axcorrelate",
370  .description = NULL_IF_CONFIG_SMALL("Cross-correlate two audio streams."),
371  .priv_size = sizeof(AudioXCorrelateContext),
372  .priv_class = &axcorrelate_class,
374  .activate = activate,
375  .uninit = uninit,
376  .inputs = inputs,
377  .outputs = outputs,
378 };
float, planar
Definition: samplefmt.h:69
int ff_inlink_consume_frame(AVFilterLink *link, AVFrame **rframe)
Take a frame from the link&#39;s FIFO and update the link&#39;s stats.
Definition: avfilter.c:1476
#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:586
AVAudioFifo * av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels, int nb_samples)
Allocate an AVAudioFifo.
Definition: audio_fifo.c:59
static int xcorrelate_slow(AVFilterContext *ctx, AVFrame *out)
This structure describes decoded (raw) audio or video data.
Definition: frame.h:300
AVOption.
Definition: opt.h:246
static const AVFilterPad inputs[]
#define FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, filter)
Forward the status on an output link to all input links.
Definition: filters.h:212
Main libavfilter public API header.
void av_audio_fifo_free(AVAudioFifo *af)
Free an AVAudioFifo.
Definition: audio_fifo.c:45
AVAudioFifo * fifo[2]
#define FFERROR_NOT_READY
Filters implementation helper functions.
Definition: filters.h:34
static void ff_outlink_set_status(AVFilterLink *link, int status, int64_t pts)
Set the status field of a link from the source filter.
Definition: filters.h:189
void ff_inlink_request_frame(AVFilterLink *link)
Mark that a frame is wanted on the link.
Definition: avfilter.c:1602
static int ff_outlink_frame_wanted(AVFilterLink *link)
Test if a frame is wanted on an output link.
Definition: filters.h:172
AVFilterFormats * ff_make_format_list(const int *fmts)
Create a list of supported formats.
Definition: formats.c:300
const char * name
Pad name.
Definition: internal.h:60
AVFilterLink ** inputs
array of pointers to input links
Definition: avfilter.h:346
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1075
#define av_cold
Definition: attributes.h:88
AVOptions.
#define f(width, name)
Definition: cbs_vp9.c:255
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:393
static AVFrame * frame
#define OFFSET(x)
static av_cold void uninit(AVFilterContext *ctx)
A filter pad used for either input or output.
Definition: internal.h:54
static int query_formats(AVFilterContext *ctx)
int ff_inlink_acknowledge_status(AVFilterLink *link, int *rstatus, int64_t *rpts)
Test and acknowledge the change of status on the link.
Definition: avfilter.c:1431
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:269
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:605
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
Definition: audio.c:86
#define AVERROR(e)
Definition: error.h:43
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:203
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:188
void * priv
private data for use by the filter
Definition: avfilter.h:353
simple assert() macros that are a bit more flexible than ISO C assert().
#define FFMAX(a, b)
Definition: common.h:94
Context for an Audio FIFO Buffer.
Definition: audio_fifo.c:34
AVFilter ff_af_axcorrelate
Definition: dct.c:52
int av_audio_fifo_size(AVAudioFifo *af)
Get the current number of samples in the AVAudioFifo available for reading.
Definition: audio_fifo.c:228
int channels
number of audio channels, only used for audio.
Definition: frame.h:606
audio channel layout utility functions
#define FFMIN(a, b)
Definition: common.h:96
#define AF
AVFormatContext * ctx
Definition: movenc.c:48
#define s(width, name)
Definition: cbs_vp9.c:257
int(* xcorrelate)(AVFilterContext *ctx, AVFrame *out)
A list of supported channel layouts.
Definition: formats.h:85
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
static const AVFilterPad outputs[]
long long int64_t
Definition: coverity.c:34
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) #define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac) { } void ff_audio_convert_free(AudioConvert **ac) { if(! *ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);} AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map) { AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method !=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt) > 2) { ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc) { av_free(ac);return NULL;} return ac;} in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar) { ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar ? ac->channels :1;} else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);return ac;} int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in) { int use_generic=1;int len=in->nb_samples;int p;if(ac->dc) { av_log(ac->avr, AV_LOG_TRACE, "%d samples - audio_convert: %s to %s (dithered)\", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
Describe the class of an AVClass context structure.
Definition: log.h:67
Filter definition.
Definition: avfilter.h:144
static int config_output(AVFilterLink *outlink)
const char * name
Filter name.
Definition: avfilter.h:148
AVFilterLink ** outputs
array of pointers to output links
Definition: avfilter.h:350
enum MovChannelLayoutTag * layouts
Definition: mov_chan.c:434
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:439
int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples)
Write data to an AVAudioFifo.
Definition: audio_fifo.c:112
int av_audio_fifo_drain(AVAudioFifo *af, int nb_samples)
Drain data from an AVAudioFifo.
Definition: audio_fifo.c:201
int
common internal and external API header
if(ret< 0)
Definition: vf_mcdeint.c:279
static int activate(AVFilterContext *ctx)
static float square_sum(const float *x, const float *y, int size)
void ff_filter_set_ready(AVFilterContext *filter, unsigned priority)
Mark a filter ready and schedule it for activation.
Definition: avfilter.c:193
static const AVOption axcorrelate_options[]
Audio FIFO Buffer.
A list of supported formats for one end of a filter link.
Definition: formats.h:64
int av_audio_fifo_peek(AVAudioFifo *af, void **data, int nb_samples)
Peek data from an AVAudioFifo.
Definition: audio_fifo.c:138
An instance of a filter.
Definition: avfilter.h:338
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:731
FILE * out
Definition: movenc.c:54
formats
Definition: signature.h:48
AVFILTER_DEFINE_CLASS(axcorrelate)
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:454
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:347
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:366
static int xcorrelate_fast(AVFilterContext *ctx, AVFrame *out)
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:593
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248