FFmpeg
af_afade.c
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1 /*
2  * Copyright (c) 2013-2015 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 /**
22  * @file
23  * fade audio filter
24  */
25 
26 #include "libavutil/opt.h"
27 #include "audio.h"
28 #include "avfilter.h"
29 #include "filters.h"
30 #include "internal.h"
31 
32 typedef struct AudioFadeContext {
33  const AVClass *class;
34  int type;
35  int curve, curve2;
36  int64_t nb_samples;
37  int64_t start_sample;
38  int64_t duration;
39  int64_t start_time;
40  int overlap;
41  int cf0_eof;
43  int64_t pts;
44 
45  void (*fade_samples)(uint8_t **dst, uint8_t * const *src,
46  int nb_samples, int channels, int direction,
47  int64_t start, int64_t range, int curve);
48  void (*crossfade_samples)(uint8_t **dst, uint8_t * const *cf0,
49  uint8_t * const *cf1,
50  int nb_samples, int channels,
51  int curve0, int curve1);
53 
55 
56 #define OFFSET(x) offsetof(AudioFadeContext, x)
57 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
58 #define TFLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
59 
61 {
64  static const enum AVSampleFormat sample_fmts[] = {
70  };
71  int ret;
72 
73  layouts = ff_all_channel_counts();
74  if (!layouts)
75  return AVERROR(ENOMEM);
76  ret = ff_set_common_channel_layouts(ctx, layouts);
77  if (ret < 0)
78  return ret;
79 
80  formats = ff_make_format_list(sample_fmts);
81  if (!formats)
82  return AVERROR(ENOMEM);
83  ret = ff_set_common_formats(ctx, formats);
84  if (ret < 0)
85  return ret;
86 
87  formats = ff_all_samplerates();
88  if (!formats)
89  return AVERROR(ENOMEM);
90  return ff_set_common_samplerates(ctx, formats);
91 }
92 
93 static double fade_gain(int curve, int64_t index, int64_t range)
94 {
95 #define CUBE(a) ((a)*(a)*(a))
96  double gain;
97 
98  gain = av_clipd(1.0 * index / range, 0, 1.0);
99 
100  switch (curve) {
101  case QSIN:
102  gain = sin(gain * M_PI / 2.0);
103  break;
104  case IQSIN:
105  /* 0.6... = 2 / M_PI */
106  gain = 0.6366197723675814 * asin(gain);
107  break;
108  case ESIN:
109  gain = 1.0 - cos(M_PI / 4.0 * (CUBE(2.0*gain - 1) + 1));
110  break;
111  case HSIN:
112  gain = (1.0 - cos(gain * M_PI)) / 2.0;
113  break;
114  case IHSIN:
115  /* 0.3... = 1 / M_PI */
116  gain = 0.3183098861837907 * acos(1 - 2 * gain);
117  break;
118  case EXP:
119  /* -11.5... = 5*ln(0.1) */
120  gain = exp(-11.512925464970227 * (1 - gain));
121  break;
122  case LOG:
123  gain = av_clipd(1 + 0.2 * log10(gain), 0, 1.0);
124  break;
125  case PAR:
126  gain = 1 - sqrt(1 - gain);
127  break;
128  case IPAR:
129  gain = (1 - (1 - gain) * (1 - gain));
130  break;
131  case QUA:
132  gain *= gain;
133  break;
134  case CUB:
135  gain = CUBE(gain);
136  break;
137  case SQU:
138  gain = sqrt(gain);
139  break;
140  case CBR:
141  gain = cbrt(gain);
142  break;
143  case DESE:
144  gain = gain <= 0.5 ? cbrt(2 * gain) / 2: 1 - cbrt(2 * (1 - gain)) / 2;
145  break;
146  case DESI:
147  gain = gain <= 0.5 ? CUBE(2 * gain) / 2: 1 - CUBE(2 * (1 - gain)) / 2;
148  break;
149  case LOSI: {
150  const double a = 1. / (1. - 0.787) - 1;
151  double A = 1. / (1.0 + exp(0 -((gain-0.5) * a * 2.0)));
152  double B = 1. / (1.0 + exp(a));
153  double C = 1. / (1.0 + exp(0-a));
154  gain = (A - B) / (C - B);
155  }
156  break;
157  case SINC:
158  gain = gain >= 1.0 ? 1.0 : sin(M_PI * (1.0 - gain)) / (M_PI * (1.0 - gain));
159  break;
160  case ISINC:
161  gain = gain <= 0.0 ? 0.0 : 1.0 - sin(M_PI * gain) / (M_PI * gain);
162  break;
163  case NONE:
164  gain = 1.0;
165  break;
166  }
167 
168  return gain;
169 }
170 
171 #define FADE_PLANAR(name, type) \
172 static void fade_samples_## name ##p(uint8_t **dst, uint8_t * const *src, \
173  int nb_samples, int channels, int dir, \
174  int64_t start, int64_t range, int curve) \
175 { \
176  int i, c; \
177  \
178  for (i = 0; i < nb_samples; i++) { \
179  double gain = fade_gain(curve, start + i * dir, range); \
180  for (c = 0; c < channels; c++) { \
181  type *d = (type *)dst[c]; \
182  const type *s = (type *)src[c]; \
183  \
184  d[i] = s[i] * gain; \
185  } \
186  } \
187 }
188 
189 #define FADE(name, type) \
190 static void fade_samples_## name (uint8_t **dst, uint8_t * const *src, \
191  int nb_samples, int channels, int dir, \
192  int64_t start, int64_t range, int curve) \
193 { \
194  type *d = (type *)dst[0]; \
195  const type *s = (type *)src[0]; \
196  int i, c, k = 0; \
197  \
198  for (i = 0; i < nb_samples; i++) { \
199  double gain = fade_gain(curve, start + i * dir, range); \
200  for (c = 0; c < channels; c++, k++) \
201  d[k] = s[k] * gain; \
202  } \
203 }
204 
205 FADE_PLANAR(dbl, double)
206 FADE_PLANAR(flt, float)
207 FADE_PLANAR(s16, int16_t)
208 FADE_PLANAR(s32, int32_t)
209 
210 FADE(dbl, double)
211 FADE(flt, float)
212 FADE(s16, int16_t)
213 FADE(s32, int32_t)
214 
215 static int config_output(AVFilterLink *outlink)
216 {
217  AVFilterContext *ctx = outlink->src;
218  AudioFadeContext *s = ctx->priv;
219 
220  switch (outlink->format) {
221  case AV_SAMPLE_FMT_DBL: s->fade_samples = fade_samples_dbl; break;
222  case AV_SAMPLE_FMT_DBLP: s->fade_samples = fade_samples_dblp; break;
223  case AV_SAMPLE_FMT_FLT: s->fade_samples = fade_samples_flt; break;
224  case AV_SAMPLE_FMT_FLTP: s->fade_samples = fade_samples_fltp; break;
225  case AV_SAMPLE_FMT_S16: s->fade_samples = fade_samples_s16; break;
226  case AV_SAMPLE_FMT_S16P: s->fade_samples = fade_samples_s16p; break;
227  case AV_SAMPLE_FMT_S32: s->fade_samples = fade_samples_s32; break;
228  case AV_SAMPLE_FMT_S32P: s->fade_samples = fade_samples_s32p; break;
229  }
230 
231  if (s->duration)
232  s->nb_samples = av_rescale(s->duration, outlink->sample_rate, AV_TIME_BASE);
233  s->duration = 0;
234  if (s->start_time)
235  s->start_sample = av_rescale(s->start_time, outlink->sample_rate, AV_TIME_BASE);
236  s->start_time = 0;
237 
238  return 0;
239 }
240 
241 #if CONFIG_AFADE_FILTER
242 
243 static const AVOption afade_options[] = {
244  { "type", "set the fade direction", OFFSET(type), AV_OPT_TYPE_INT, {.i64 = 0 }, 0, 1, TFLAGS, "type" },
245  { "t", "set the fade direction", OFFSET(type), AV_OPT_TYPE_INT, {.i64 = 0 }, 0, 1, TFLAGS, "type" },
246  { "in", "fade-in", 0, AV_OPT_TYPE_CONST, {.i64 = 0 }, 0, 0, TFLAGS, "type" },
247  { "out", "fade-out", 0, AV_OPT_TYPE_CONST, {.i64 = 1 }, 0, 0, TFLAGS, "type" },
248  { "start_sample", "set number of first sample to start fading", OFFSET(start_sample), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT64_MAX, TFLAGS },
249  { "ss", "set number of first sample to start fading", OFFSET(start_sample), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT64_MAX, TFLAGS },
250  { "nb_samples", "set number of samples for fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT64, {.i64 = 44100}, 1, INT64_MAX, TFLAGS },
251  { "ns", "set number of samples for fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT64, {.i64 = 44100}, 1, INT64_MAX, TFLAGS },
252  { "start_time", "set time to start fading", OFFSET(start_time), AV_OPT_TYPE_DURATION, {.i64 = 0 }, 0, INT64_MAX, TFLAGS },
253  { "st", "set time to start fading", OFFSET(start_time), AV_OPT_TYPE_DURATION, {.i64 = 0 }, 0, INT64_MAX, TFLAGS },
254  { "duration", "set fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0 }, 0, INT64_MAX, TFLAGS },
255  { "d", "set fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0 }, 0, INT64_MAX, TFLAGS },
256  { "curve", "set fade curve type", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, NONE, NB_CURVES - 1, TFLAGS, "curve" },
257  { "c", "set fade curve type", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, NONE, NB_CURVES - 1, TFLAGS, "curve" },
258  { "nofade", "no fade; keep audio as-is", 0, AV_OPT_TYPE_CONST, {.i64 = NONE }, 0, 0, TFLAGS, "curve" },
259  { "tri", "linear slope", 0, AV_OPT_TYPE_CONST, {.i64 = TRI }, 0, 0, TFLAGS, "curve" },
260  { "qsin", "quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = QSIN }, 0, 0, TFLAGS, "curve" },
261  { "esin", "exponential sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = ESIN }, 0, 0, TFLAGS, "curve" },
262  { "hsin", "half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = HSIN }, 0, 0, TFLAGS, "curve" },
263  { "log", "logarithmic", 0, AV_OPT_TYPE_CONST, {.i64 = LOG }, 0, 0, TFLAGS, "curve" },
264  { "ipar", "inverted parabola", 0, AV_OPT_TYPE_CONST, {.i64 = IPAR }, 0, 0, TFLAGS, "curve" },
265  { "qua", "quadratic", 0, AV_OPT_TYPE_CONST, {.i64 = QUA }, 0, 0, TFLAGS, "curve" },
266  { "cub", "cubic", 0, AV_OPT_TYPE_CONST, {.i64 = CUB }, 0, 0, TFLAGS, "curve" },
267  { "squ", "square root", 0, AV_OPT_TYPE_CONST, {.i64 = SQU }, 0, 0, TFLAGS, "curve" },
268  { "cbr", "cubic root", 0, AV_OPT_TYPE_CONST, {.i64 = CBR }, 0, 0, TFLAGS, "curve" },
269  { "par", "parabola", 0, AV_OPT_TYPE_CONST, {.i64 = PAR }, 0, 0, TFLAGS, "curve" },
270  { "exp", "exponential", 0, AV_OPT_TYPE_CONST, {.i64 = EXP }, 0, 0, TFLAGS, "curve" },
271  { "iqsin", "inverted quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IQSIN}, 0, 0, TFLAGS, "curve" },
272  { "ihsin", "inverted half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IHSIN}, 0, 0, TFLAGS, "curve" },
273  { "dese", "double-exponential seat", 0, AV_OPT_TYPE_CONST, {.i64 = DESE }, 0, 0, TFLAGS, "curve" },
274  { "desi", "double-exponential sigmoid", 0, AV_OPT_TYPE_CONST, {.i64 = DESI }, 0, 0, TFLAGS, "curve" },
275  { "losi", "logistic sigmoid", 0, AV_OPT_TYPE_CONST, {.i64 = LOSI }, 0, 0, TFLAGS, "curve" },
276  { "sinc", "sine cardinal function", 0, AV_OPT_TYPE_CONST, {.i64 = SINC }, 0, 0, TFLAGS, "curve" },
277  { "isinc", "inverted sine cardinal function", 0, AV_OPT_TYPE_CONST, {.i64 = ISINC}, 0, 0, TFLAGS, "curve" },
278  { NULL }
279 };
280 
281 AVFILTER_DEFINE_CLASS(afade);
282 
283 static av_cold int init(AVFilterContext *ctx)
284 {
285  AudioFadeContext *s = ctx->priv;
286 
287  if (INT64_MAX - s->nb_samples < s->start_sample)
288  return AVERROR(EINVAL);
289 
290  return 0;
291 }
292 
293 static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
294 {
295  AudioFadeContext *s = inlink->dst->priv;
296  AVFilterLink *outlink = inlink->dst->outputs[0];
297  int nb_samples = buf->nb_samples;
298  AVFrame *out_buf;
299  int64_t cur_sample = av_rescale_q(buf->pts, inlink->time_base, (AVRational){1, inlink->sample_rate});
300 
301  if ((!s->type && (s->start_sample + s->nb_samples < cur_sample)) ||
302  ( s->type && (cur_sample + nb_samples < s->start_sample)))
303  return ff_filter_frame(outlink, buf);
304 
305  if (av_frame_is_writable(buf)) {
306  out_buf = buf;
307  } else {
308  out_buf = ff_get_audio_buffer(outlink, nb_samples);
309  if (!out_buf)
310  return AVERROR(ENOMEM);
311  av_frame_copy_props(out_buf, buf);
312  }
313 
314  if ((!s->type && (cur_sample + nb_samples < s->start_sample)) ||
315  ( s->type && (s->start_sample + s->nb_samples < cur_sample))) {
316  av_samples_set_silence(out_buf->extended_data, 0, nb_samples,
317  out_buf->channels, out_buf->format);
318  } else {
319  int64_t start;
320 
321  if (!s->type)
322  start = cur_sample - s->start_sample;
323  else
324  start = s->start_sample + s->nb_samples - cur_sample;
325 
326  s->fade_samples(out_buf->extended_data, buf->extended_data,
327  nb_samples, buf->channels,
328  s->type ? -1 : 1, start,
329  s->nb_samples, s->curve);
330  }
331 
332  if (buf != out_buf)
333  av_frame_free(&buf);
334 
335  return ff_filter_frame(outlink, out_buf);
336 }
337 
338 static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
339  char *res, int res_len, int flags)
340 {
341  int ret;
342 
343  ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
344  if (ret < 0)
345  return ret;
346 
347  return config_output(ctx->outputs[0]);
348 }
349 
350 static const AVFilterPad avfilter_af_afade_inputs[] = {
351  {
352  .name = "default",
353  .type = AVMEDIA_TYPE_AUDIO,
354  .filter_frame = filter_frame,
355  },
356  { NULL }
357 };
358 
359 static const AVFilterPad avfilter_af_afade_outputs[] = {
360  {
361  .name = "default",
362  .type = AVMEDIA_TYPE_AUDIO,
363  .config_props = config_output,
364  },
365  { NULL }
366 };
367 
369  .name = "afade",
370  .description = NULL_IF_CONFIG_SMALL("Fade in/out input audio."),
371  .query_formats = query_formats,
372  .priv_size = sizeof(AudioFadeContext),
373  .init = init,
374  .inputs = avfilter_af_afade_inputs,
375  .outputs = avfilter_af_afade_outputs,
376  .priv_class = &afade_class,
379 };
380 
381 #endif /* CONFIG_AFADE_FILTER */
382 
383 #if CONFIG_ACROSSFADE_FILTER
384 
385 static const AVOption acrossfade_options[] = {
386  { "nb_samples", "set number of samples for cross fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 44100}, 1, INT32_MAX/10, FLAGS },
387  { "ns", "set number of samples for cross fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 44100}, 1, INT32_MAX/10, FLAGS },
388  { "duration", "set cross fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0 }, 0, 60000000, FLAGS },
389  { "d", "set cross fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0 }, 0, 60000000, FLAGS },
390  { "overlap", "overlap 1st stream end with 2nd stream start", OFFSET(overlap), AV_OPT_TYPE_BOOL, {.i64 = 1 }, 0, 1, FLAGS },
391  { "o", "overlap 1st stream end with 2nd stream start", OFFSET(overlap), AV_OPT_TYPE_BOOL, {.i64 = 1 }, 0, 1, FLAGS },
392  { "curve1", "set fade curve type for 1st stream", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, NONE, NB_CURVES - 1, FLAGS, "curve" },
393  { "c1", "set fade curve type for 1st stream", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, NONE, NB_CURVES - 1, FLAGS, "curve" },
394  { "nofade", "no fade; keep audio as-is", 0, AV_OPT_TYPE_CONST, {.i64 = NONE }, 0, 0, FLAGS, "curve" },
395  { "tri", "linear slope", 0, AV_OPT_TYPE_CONST, {.i64 = TRI }, 0, 0, FLAGS, "curve" },
396  { "qsin", "quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = QSIN }, 0, 0, FLAGS, "curve" },
397  { "esin", "exponential sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = ESIN }, 0, 0, FLAGS, "curve" },
398  { "hsin", "half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = HSIN }, 0, 0, FLAGS, "curve" },
399  { "log", "logarithmic", 0, AV_OPT_TYPE_CONST, {.i64 = LOG }, 0, 0, FLAGS, "curve" },
400  { "ipar", "inverted parabola", 0, AV_OPT_TYPE_CONST, {.i64 = IPAR }, 0, 0, FLAGS, "curve" },
401  { "qua", "quadratic", 0, AV_OPT_TYPE_CONST, {.i64 = QUA }, 0, 0, FLAGS, "curve" },
402  { "cub", "cubic", 0, AV_OPT_TYPE_CONST, {.i64 = CUB }, 0, 0, FLAGS, "curve" },
403  { "squ", "square root", 0, AV_OPT_TYPE_CONST, {.i64 = SQU }, 0, 0, FLAGS, "curve" },
404  { "cbr", "cubic root", 0, AV_OPT_TYPE_CONST, {.i64 = CBR }, 0, 0, FLAGS, "curve" },
405  { "par", "parabola", 0, AV_OPT_TYPE_CONST, {.i64 = PAR }, 0, 0, FLAGS, "curve" },
406  { "exp", "exponential", 0, AV_OPT_TYPE_CONST, {.i64 = EXP }, 0, 0, FLAGS, "curve" },
407  { "iqsin", "inverted quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IQSIN}, 0, 0, FLAGS, "curve" },
408  { "ihsin", "inverted half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IHSIN}, 0, 0, FLAGS, "curve" },
409  { "dese", "double-exponential seat", 0, AV_OPT_TYPE_CONST, {.i64 = DESE }, 0, 0, FLAGS, "curve" },
410  { "desi", "double-exponential sigmoid", 0, AV_OPT_TYPE_CONST, {.i64 = DESI }, 0, 0, FLAGS, "curve" },
411  { "losi", "logistic sigmoid", 0, AV_OPT_TYPE_CONST, {.i64 = LOSI }, 0, 0, FLAGS, "curve" },
412  { "sinc", "sine cardinal function", 0, AV_OPT_TYPE_CONST, {.i64 = SINC }, 0, 0, FLAGS, "curve" },
413  { "isinc", "inverted sine cardinal function", 0, AV_OPT_TYPE_CONST, {.i64 = ISINC}, 0, 0, FLAGS, "curve" },
414  { "curve2", "set fade curve type for 2nd stream", OFFSET(curve2), AV_OPT_TYPE_INT, {.i64 = TRI }, NONE, NB_CURVES - 1, FLAGS, "curve" },
415  { "c2", "set fade curve type for 2nd stream", OFFSET(curve2), AV_OPT_TYPE_INT, {.i64 = TRI }, NONE, NB_CURVES - 1, FLAGS, "curve" },
416  { NULL }
417 };
418 
419 AVFILTER_DEFINE_CLASS(acrossfade);
420 
421 #define CROSSFADE_PLANAR(name, type) \
422 static void crossfade_samples_## name ##p(uint8_t **dst, uint8_t * const *cf0, \
423  uint8_t * const *cf1, \
424  int nb_samples, int channels, \
425  int curve0, int curve1) \
426 { \
427  int i, c; \
428  \
429  for (i = 0; i < nb_samples; i++) { \
430  double gain0 = fade_gain(curve0, nb_samples - 1 - i, nb_samples); \
431  double gain1 = fade_gain(curve1, i, nb_samples); \
432  for (c = 0; c < channels; c++) { \
433  type *d = (type *)dst[c]; \
434  const type *s0 = (type *)cf0[c]; \
435  const type *s1 = (type *)cf1[c]; \
436  \
437  d[i] = s0[i] * gain0 + s1[i] * gain1; \
438  } \
439  } \
440 }
441 
442 #define CROSSFADE(name, type) \
443 static void crossfade_samples_## name (uint8_t **dst, uint8_t * const *cf0, \
444  uint8_t * const *cf1, \
445  int nb_samples, int channels, \
446  int curve0, int curve1) \
447 { \
448  type *d = (type *)dst[0]; \
449  const type *s0 = (type *)cf0[0]; \
450  const type *s1 = (type *)cf1[0]; \
451  int i, c, k = 0; \
452  \
453  for (i = 0; i < nb_samples; i++) { \
454  double gain0 = fade_gain(curve0, nb_samples - 1 - i, nb_samples); \
455  double gain1 = fade_gain(curve1, i, nb_samples); \
456  for (c = 0; c < channels; c++, k++) \
457  d[k] = s0[k] * gain0 + s1[k] * gain1; \
458  } \
459 }
460 
461 CROSSFADE_PLANAR(dbl, double)
462 CROSSFADE_PLANAR(flt, float)
463 CROSSFADE_PLANAR(s16, int16_t)
464 CROSSFADE_PLANAR(s32, int32_t)
465 
466 CROSSFADE(dbl, double)
467 CROSSFADE(flt, float)
468 CROSSFADE(s16, int16_t)
469 CROSSFADE(s32, int32_t)
470 
471 static int activate(AVFilterContext *ctx)
472 {
473  AudioFadeContext *s = ctx->priv;
474  AVFilterLink *outlink = ctx->outputs[0];
475  AVFrame *in = NULL, *out, *cf[2] = { NULL };
476  int ret = 0, nb_samples, status;
477  int64_t pts;
478 
479  FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, ctx);
480 
481  if (s->crossfade_is_over) {
482  ret = ff_inlink_consume_frame(ctx->inputs[1], &in);
483  if (ret > 0) {
484  in->pts = s->pts;
485  s->pts += av_rescale_q(in->nb_samples,
486  (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
487  return ff_filter_frame(outlink, in);
488  } else if (ret < 0) {
489  return ret;
490  } else if (ff_inlink_acknowledge_status(ctx->inputs[1], &status, &pts)) {
491  ff_outlink_set_status(ctx->outputs[0], status, pts);
492  return 0;
493  } else if (!ret) {
494  if (ff_outlink_frame_wanted(ctx->outputs[0])) {
496  return 0;
497  }
498  }
499  }
500 
501  if (ff_inlink_queued_samples(ctx->inputs[0]) > s->nb_samples) {
503  if (nb_samples > 0) {
505  if (ret < 0) {
506  return ret;
507  }
508  }
509  in->pts = s->pts;
510  s->pts += av_rescale_q(in->nb_samples,
511  (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
512  return ff_filter_frame(outlink, in);
513  } else if (ff_inlink_queued_samples(ctx->inputs[0]) >= s->nb_samples &&
514  ff_inlink_queued_samples(ctx->inputs[1]) >= s->nb_samples && s->cf0_eof) {
515  if (s->overlap) {
516  out = ff_get_audio_buffer(outlink, s->nb_samples);
517  if (!out)
518  return AVERROR(ENOMEM);
519 
520  ret = ff_inlink_consume_samples(ctx->inputs[0], s->nb_samples, s->nb_samples, &cf[0]);
521  if (ret < 0) {
522  av_frame_free(&out);
523  return ret;
524  }
525 
526  ret = ff_inlink_consume_samples(ctx->inputs[1], s->nb_samples, s->nb_samples, &cf[1]);
527  if (ret < 0) {
528  av_frame_free(&out);
529  return ret;
530  }
531 
532  s->crossfade_samples(out->extended_data, cf[0]->extended_data,
533  cf[1]->extended_data,
534  s->nb_samples, out->channels,
535  s->curve, s->curve2);
536  out->pts = s->pts;
537  s->pts += av_rescale_q(s->nb_samples,
538  (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
539  s->crossfade_is_over = 1;
540  av_frame_free(&cf[0]);
541  av_frame_free(&cf[1]);
542  return ff_filter_frame(outlink, out);
543  } else {
544  out = ff_get_audio_buffer(outlink, s->nb_samples);
545  if (!out)
546  return AVERROR(ENOMEM);
547 
548  ret = ff_inlink_consume_samples(ctx->inputs[0], s->nb_samples, s->nb_samples, &cf[0]);
549  if (ret < 0) {
550  av_frame_free(&out);
551  return ret;
552  }
553 
554  s->fade_samples(out->extended_data, cf[0]->extended_data, s->nb_samples,
555  outlink->channels, -1, s->nb_samples - 1, s->nb_samples, s->curve);
556  out->pts = s->pts;
557  s->pts += av_rescale_q(s->nb_samples,
558  (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
559  av_frame_free(&cf[0]);
560  ret = ff_filter_frame(outlink, out);
561  if (ret < 0)
562  return ret;
563 
564  out = ff_get_audio_buffer(outlink, s->nb_samples);
565  if (!out)
566  return AVERROR(ENOMEM);
567 
568  ret = ff_inlink_consume_samples(ctx->inputs[1], s->nb_samples, s->nb_samples, &cf[1]);
569  if (ret < 0) {
570  av_frame_free(&out);
571  return ret;
572  }
573 
574  s->fade_samples(out->extended_data, cf[1]->extended_data, s->nb_samples,
575  outlink->channels, 1, 0, s->nb_samples, s->curve2);
576  out->pts = s->pts;
577  s->pts += av_rescale_q(s->nb_samples,
578  (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
579  s->crossfade_is_over = 1;
580  av_frame_free(&cf[1]);
581  return ff_filter_frame(outlink, out);
582  }
583  } else if (ff_outlink_frame_wanted(ctx->outputs[0])) {
584  if (!s->cf0_eof && ff_outlink_get_status(ctx->inputs[0])) {
585  s->cf0_eof = 1;
586  }
587  if (ff_outlink_get_status(ctx->inputs[1])) {
589  return 0;
590  }
591  if (!s->cf0_eof)
593  else
595  return 0;
596  }
597 
598  return ret;
599 }
600 
601 static int acrossfade_config_output(AVFilterLink *outlink)
602 {
603  AVFilterContext *ctx = outlink->src;
604  AudioFadeContext *s = ctx->priv;
605 
606  if (ctx->inputs[0]->sample_rate != ctx->inputs[1]->sample_rate) {
607  av_log(ctx, AV_LOG_ERROR,
608  "Inputs must have the same sample rate "
609  "%d for in0 vs %d for in1\n",
610  ctx->inputs[0]->sample_rate, ctx->inputs[1]->sample_rate);
611  return AVERROR(EINVAL);
612  }
613 
614  outlink->sample_rate = ctx->inputs[0]->sample_rate;
615  outlink->time_base = ctx->inputs[0]->time_base;
616  outlink->channel_layout = ctx->inputs[0]->channel_layout;
617  outlink->channels = ctx->inputs[0]->channels;
618 
619  switch (outlink->format) {
620  case AV_SAMPLE_FMT_DBL: s->crossfade_samples = crossfade_samples_dbl; break;
621  case AV_SAMPLE_FMT_DBLP: s->crossfade_samples = crossfade_samples_dblp; break;
622  case AV_SAMPLE_FMT_FLT: s->crossfade_samples = crossfade_samples_flt; break;
623  case AV_SAMPLE_FMT_FLTP: s->crossfade_samples = crossfade_samples_fltp; break;
624  case AV_SAMPLE_FMT_S16: s->crossfade_samples = crossfade_samples_s16; break;
625  case AV_SAMPLE_FMT_S16P: s->crossfade_samples = crossfade_samples_s16p; break;
626  case AV_SAMPLE_FMT_S32: s->crossfade_samples = crossfade_samples_s32; break;
627  case AV_SAMPLE_FMT_S32P: s->crossfade_samples = crossfade_samples_s32p; break;
628  }
629 
630  config_output(outlink);
631 
632  return 0;
633 }
634 
635 static const AVFilterPad avfilter_af_acrossfade_inputs[] = {
636  {
637  .name = "crossfade0",
638  .type = AVMEDIA_TYPE_AUDIO,
639  },
640  {
641  .name = "crossfade1",
642  .type = AVMEDIA_TYPE_AUDIO,
643  },
644  { NULL }
645 };
646 
647 static const AVFilterPad avfilter_af_acrossfade_outputs[] = {
648  {
649  .name = "default",
650  .type = AVMEDIA_TYPE_AUDIO,
651  .config_props = acrossfade_config_output,
652  },
653  { NULL }
654 };
655 
657  .name = "acrossfade",
658  .description = NULL_IF_CONFIG_SMALL("Cross fade two input audio streams."),
659  .query_formats = query_formats,
660  .priv_size = sizeof(AudioFadeContext),
661  .activate = activate,
662  .priv_class = &acrossfade_class,
663  .inputs = avfilter_af_acrossfade_inputs,
664  .outputs = avfilter_af_acrossfade_outputs,
665 };
666 
667 #endif /* CONFIG_ACROSSFADE_FILTER */
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:1492
#define NULL
Definition: coverity.c:32
#define CUBE(a)
static int filter_frame(DBEDecodeContext *s, AVFrame *frame)
Definition: dolby_e.c:1049
This structure describes decoded (raw) audio or video data.
Definition: frame.h:318
AVOption.
Definition: opt.h:248
#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.
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:31
static int config_output(AVFilterLink *outlink)
Definition: af_afade.c:215
double, planar
Definition: samplefmt.h:70
The reader does not expect b to be semantically here and if the code is changed by maybe adding a a division or other the signedness will almost certainly be mistaken To avoid this confusion a new type was SUINT is the C unsigned type but it holds a signed int to use the same example SUINT a
Definition: undefined.txt:36
GLint GLenum type
Definition: opengl_enc.c:104
int ff_set_common_channel_layouts(AVFilterContext *ctx, AVFilterChannelLayouts *channel_layouts)
A helper for query_formats() which sets all links to the same list of channel layouts/sample rates...
Definition: formats.c:569
Definition: af_afade.c:54
fg index
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
Definition: af_afade.c:54
void ff_inlink_request_frame(AVFilterLink *link)
Mark that a frame is wanted on the link.
Definition: avfilter.c:1618
AVFilterFormats * ff_make_format_list(const int *fmts)
Create a list of supported formats.
Definition: formats.c:287
#define AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC
Some filters support a generic "enable" expression option that can be used to enable or disable a fil...
Definition: avfilter.h:126
#define TFLAGS
Definition: af_afade.c:58
const char * name
Pad name.
Definition: internal.h:60
AVFilterLink ** inputs
array of pointers to input links
Definition: avfilter.h:349
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1094
uint8_t
#define av_cold
Definition: attributes.h:88
AVOptions.
filter_frame For filters that do not use the activate() callback
static int query_formats(AVFilterContext *ctx)
Definition: af_afade.c:60
#define FADE_PLANAR(name, type)
Definition: af_afade.c:171
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:411
void(* fade_samples)(uint8_t **dst, uint8_t *const *src, int nb_samples, int channels, int direction, int64_t start, int64_t range, int curve)
Definition: af_afade.c:45
Definition: af_afade.c:54
#define AVERROR_EOF
End of file.
Definition: error.h:55
channels
Definition: aptx.h:33
the definition of that something depends on the semantic of the filter The callback must examine the status of the filter s links and proceed accordingly The status of output links is stored in the status_in and status_out fields and tested by the ff_outlink_frame_wanted() function.If this function returns true
signed 32 bits
Definition: samplefmt.h:62
#define A(x)
Definition: vp56_arith.h:28
#define av_log(a,...)
A filter pad used for either input or output.
Definition: internal.h:54
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
Definition: mathematics.c:142
#define src
Definition: vp8dsp.c:255
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:1447
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:194
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:588
int av_samples_set_silence(uint8_t **audio_data, int offset, int nb_samples, int nb_channels, enum AVSampleFormat sample_fmt)
Fill an audio buffer with silence.
Definition: samplefmt.c:237
CurveType
Definition: af_afade.c:54
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
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:204
#define B
Definition: huffyuvdsp.h:32
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:117
int ff_filter_process_command(AVFilterContext *ctx, const char *cmd, const char *arg, char *res, int res_len, int flags)
Generic processing of user supplied commands that are set in the same way as the filter options...
Definition: avfilter.c:882
void * priv
private data for use by the filter
Definition: avfilter.h:356
#define cbrt
Definition: tablegen.h:35
int8_t exp
Definition: eval.c:72
void(* crossfade_samples)(uint8_t **dst, uint8_t *const *cf0, uint8_t *const *cf1, int nb_samples, int channels, int curve0, int curve1)
Definition: af_afade.c:48
Definition: af_afade.c:54
int channels
number of audio channels, only used for audio.
Definition: frame.h:624
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several inputs
int64_t av_rescale(int64_t a, int64_t b, int64_t c)
Rescale a 64-bit integer with rounding to nearest.
Definition: mathematics.c:129
#define AV_TIME_BASE
Internal time base represented as integer.
Definition: avutil.h:254
signed 32 bits, planar
Definition: samplefmt.h:68
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, char *res, int res_len, int flags)
Definition: af_acrusher.c:336
Definition: af_afade.c:54
int64_t start_sample
Definition: af_afade.c:37
typedef void(APIENTRY *FF_PFNGLACTIVETEXTUREPROC)(GLenum texture)
int ff_inlink_queued_samples(AVFilterLink *link)
Definition: avfilter.c:1472
Definition: af_afade.c:54
int32_t
AVFormatContext * ctx
Definition: movenc.c:48
Definition: af_afade.c:54
#define s(width, name)
Definition: cbs_vp9.c:257
int64_t pts
Definition: af_afade.c:43
int crossfade_is_over
Definition: af_afade.c:42
int64_t duration
Definition: af_afade.c:38
static const AVFilterPad outputs[]
Definition: af_acontrast.c:203
A list of supported channel layouts.
Definition: formats.h:86
s EdgeDetect Foobar g libavfilter vf_edgedetect c libavfilter vf_foobar c edit libavfilter and add an entry for foobar following the pattern of the other filters edit libavfilter allfilters and add an entry for foobar following the pattern of the other filters configure make j< whatever > ffmpeg ffmpeg i you should get a foobar png with Lena edge detected That s your new playground is ready Some little details about what s going which in turn will define variables for the build system and the C
int format
format of the frame, -1 if unknown or unset Values correspond to enum AVPixelFormat for video frames...
Definition: frame.h:391
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
Definition: frame.c:595
int ff_inlink_consume_samples(AVFilterLink *link, unsigned min, unsigned max, AVFrame **rframe)
Take samples from the link&#39;s FIFO and update the link&#39;s stats.
Definition: avfilter.c:1511
Definition: af_afade.c:54
Definition: af_afade.c:54
int64_t start_time
Definition: af_afade.c:39
Definition: af_afade.c:54
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)\n", 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:145
int ff_outlink_get_status(AVFilterLink *link)
Get the status on an output link.
Definition: avfilter.c:1641
Rational number (pair of numerator and denominator).
Definition: rational.h:58
Definition: af_afade.c:54
they must not be accessed directly The fifo field contains the frames that are queued in the input for processing by the filter The status_in and status_out fields contains the queued status(EOF or error) of the link
#define FLAGS
Definition: af_afade.c:57
const char * name
Filter name.
Definition: avfilter.h:149
Definition: af_afade.c:54
Definition: af_afade.c:54
AVFilterLink ** outputs
array of pointers to output links
Definition: avfilter.h:353
enum MovChannelLayoutTag * layouts
Definition: mov_chan.c:434
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:422
#define flags(name, subs,...)
Definition: cbs_av1.c:561
Definition: af_afade.c:54
The exact code depends on how similar the blocks are and how related they are to the and needs to apply these operations to the correct inlink or outlink if there are several Macros are available to factor that when no extra processing is inlink
Definition: af_afade.c:54
Definition: af_afade.c:54
signed 16 bits
Definition: samplefmt.h:61
#define av_clipd
Definition: common.h:173
static double fade_gain(int curve, int64_t index, int64_t range)
Definition: af_afade.c:93
#define OFFSET(x)
Definition: af_afade.c:56
Definition: af_afade.c:54
#define AVFILTER_DEFINE_CLASS(fname)
Definition: internal.h:288
A list of supported formats for one end of a filter link.
Definition: formats.h:65
int64_t nb_samples
Definition: af_afade.c:36
An instance of a filter.
Definition: avfilter.h:341
AVFilter ff_af_afade
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:941
FILE * out
Definition: movenc.c:54
signed 16 bits, planar
Definition: samplefmt.h:67
#define M_PI
Definition: mathematics.h:52
formats
Definition: signature.h:48
AVFilter ff_af_acrossfade
Definition: af_afade.c:54
Definition: af_afade.c:54
internal API functions
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later.That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another.Frame references ownership and permissions
AVFilterChannelLayouts * ff_all_channel_counts(void)
Construct an AVFilterChannelLayouts coding for any channel layout, with known or unknown disposition...
Definition: formats.c:437
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:365
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:384
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:576
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
Definition: frame.c:659
#define FADE(name, type)
Definition: af_afade.c:189
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248
Definition: af_afade.c:54