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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 #define FF_INTERNAL_FIELDS 1
27 #include "framequeue.h"
28 
29 #include "libavutil/audio_fifo.h"
30 #include "libavutil/opt.h"
31 #include "audio.h"
32 #include "avfilter.h"
33 #include "filters.h"
34 #include "internal.h"
35 
36 typedef struct AudioFadeContext {
37  const AVClass *class;
38  int type;
39  int curve, curve2;
40  int64_t nb_samples;
41  int64_t start_sample;
42  int64_t duration;
43  int64_t start_time;
44  int overlap;
45  int cf0_eof;
46  int prev_size;
47  int crossfade_is_over;
48  AVAudioFifo *fifo[2];
49  int64_t pts;
50 
51  void (*fade_samples)(uint8_t **dst, uint8_t * const *src,
52  int nb_samples, int channels, int direction,
53  int64_t start, int64_t range, int curve);
54  void (*crossfade_samples)(uint8_t **dst, uint8_t * const *cf0,
55  uint8_t * const *cf1,
56  int nb_samples, int channels,
57  int curve0, int curve1);
58 } AudioFadeContext;
59 
60 enum CurveType { TRI, QSIN, ESIN, HSIN, LOG, IPAR, QUA, CUB, SQU, CBR, PAR, EXP, IQSIN, IHSIN, DESE, DESI, LOSI, NB_CURVES };
61 
62 #define OFFSET(x) offsetof(AudioFadeContext, x)
63 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
64 
65 static int query_formats(AVFilterContext *ctx)
66 {
67  AVFilterFormats *formats;
68  AVFilterChannelLayouts *layouts;
69  static const enum AVSampleFormat sample_fmts[] = {
70  AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P,
71  AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32P,
72  AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
73  AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
74  AV_SAMPLE_FMT_NONE
75  };
76  int ret;
77 
78  layouts = ff_all_channel_counts();
79  if (!layouts)
80  return AVERROR(ENOMEM);
81  ret = ff_set_common_channel_layouts(ctx, layouts);
82  if (ret < 0)
83  return ret;
84 
85  formats = ff_make_format_list(sample_fmts);
86  if (!formats)
87  return AVERROR(ENOMEM);
88  ret = ff_set_common_formats(ctx, formats);
89  if (ret < 0)
90  return ret;
91 
92  formats = ff_all_samplerates();
93  if (!formats)
94  return AVERROR(ENOMEM);
95  return ff_set_common_samplerates(ctx, formats);
96 }
97 
98 static double fade_gain(int curve, int64_t index, int64_t range)
99 {
100 #define CUBE(a) ((a)*(a)*(a))
101  double gain;
102 
103  gain = av_clipd(1.0 * index / range, 0, 1.0);
104 
105  switch (curve) {
106  case QSIN:
107  gain = sin(gain * M_PI / 2.0);
108  break;
109  case IQSIN:
110  /* 0.6... = 2 / M_PI */
111  gain = 0.6366197723675814 * asin(gain);
112  break;
113  case ESIN:
114  gain = 1.0 - cos(M_PI / 4.0 * (CUBE(2.0*gain - 1) + 1));
115  break;
116  case HSIN:
117  gain = (1.0 - cos(gain * M_PI)) / 2.0;
118  break;
119  case IHSIN:
120  /* 0.3... = 1 / M_PI */
121  gain = 0.3183098861837907 * acos(1 - 2 * gain);
122  break;
123  case EXP:
124  /* -11.5... = 5*ln(0.1) */
125  gain = exp(-11.512925464970227 * (1 - gain));
126  break;
127  case LOG:
128  gain = av_clipd(1 + 0.2 * log10(gain), 0, 1.0);
129  break;
130  case PAR:
131  gain = 1 - sqrt(1 - gain);
132  break;
133  case IPAR:
134  gain = (1 - (1 - gain) * (1 - gain));
135  break;
136  case QUA:
137  gain *= gain;
138  break;
139  case CUB:
140  gain = CUBE(gain);
141  break;
142  case SQU:
143  gain = sqrt(gain);
144  break;
145  case CBR:
146  gain = cbrt(gain);
147  break;
148  case DESE:
149  gain = gain <= 0.5 ? cbrt(2 * gain) / 2: 1 - cbrt(2 * (1 - gain)) / 2;
150  break;
151  case DESI:
152  gain = gain <= 0.5 ? CUBE(2 * gain) / 2: 1 - CUBE(2 * (1 - gain)) / 2;
153  break;
154  case LOSI: {
155  const double a = 1. / (1. - 0.787) - 1;
156  double A = 1. / (1.0 + exp(0 -((gain-0.5) * a * 2.0)));
157  double B = 1. / (1.0 + exp(a));
158  double C = 1. / (1.0 + exp(0-a));
159  gain = (A - B) / (C - B);
160  }
161  break;
162  }
163 
164  return gain;
165 }
166 
167 #define FADE_PLANAR(name, type) \
168 static void fade_samples_## name ##p(uint8_t **dst, uint8_t * const *src, \
169  int nb_samples, int channels, int dir, \
170  int64_t start, int64_t range, int curve) \
171 { \
172  int i, c; \
173  \
174  for (i = 0; i < nb_samples; i++) { \
175  double gain = fade_gain(curve, start + i * dir, range); \
176  for (c = 0; c < channels; c++) { \
177  type *d = (type *)dst[c]; \
178  const type *s = (type *)src[c]; \
179  \
180  d[i] = s[i] * gain; \
181  } \
182  } \
183 }
184 
185 #define FADE(name, type) \
186 static void fade_samples_## name (uint8_t **dst, uint8_t * const *src, \
187  int nb_samples, int channels, int dir, \
188  int64_t start, int64_t range, int curve) \
189 { \
190  type *d = (type *)dst[0]; \
191  const type *s = (type *)src[0]; \
192  int i, c, k = 0; \
193  \
194  for (i = 0; i < nb_samples; i++) { \
195  double gain = fade_gain(curve, start + i * dir, range); \
196  for (c = 0; c < channels; c++, k++) \
197  d[k] = s[k] * gain; \
198  } \
199 }
200 
201 FADE_PLANAR(dbl, double)
202 FADE_PLANAR(flt, float)
203 FADE_PLANAR(s16, int16_t)
204 FADE_PLANAR(s32, int32_t)
205 
206 FADE(dbl, double)
207 FADE(flt, float)
208 FADE(s16, int16_t)
209 FADE(s32, int32_t)
210 
211 static int config_output(AVFilterLink *outlink)
212 {
213  AVFilterContext *ctx = outlink->src;
214  AudioFadeContext *s = ctx->priv;
215 
216  switch (outlink->format) {
217  case AV_SAMPLE_FMT_DBL: s->fade_samples = fade_samples_dbl; break;
218  case AV_SAMPLE_FMT_DBLP: s->fade_samples = fade_samples_dblp; break;
219  case AV_SAMPLE_FMT_FLT: s->fade_samples = fade_samples_flt; break;
220  case AV_SAMPLE_FMT_FLTP: s->fade_samples = fade_samples_fltp; break;
221  case AV_SAMPLE_FMT_S16: s->fade_samples = fade_samples_s16; break;
222  case AV_SAMPLE_FMT_S16P: s->fade_samples = fade_samples_s16p; break;
223  case AV_SAMPLE_FMT_S32: s->fade_samples = fade_samples_s32; break;
224  case AV_SAMPLE_FMT_S32P: s->fade_samples = fade_samples_s32p; break;
225  }
226 
227  if (s->duration)
228  s->nb_samples = av_rescale(s->duration, outlink->sample_rate, AV_TIME_BASE);
229  if (s->start_time)
230  s->start_sample = av_rescale(s->start_time, outlink->sample_rate, AV_TIME_BASE);
231 
232  return 0;
233 }
234 
235 #if CONFIG_AFADE_FILTER
236 
237 static const AVOption afade_options[] = {
238  { "type", "set the fade direction", OFFSET(type), AV_OPT_TYPE_INT, {.i64 = 0 }, 0, 1, FLAGS, "type" },
239  { "t", "set the fade direction", OFFSET(type), AV_OPT_TYPE_INT, {.i64 = 0 }, 0, 1, FLAGS, "type" },
240  { "in", "fade-in", 0, AV_OPT_TYPE_CONST, {.i64 = 0 }, 0, 0, FLAGS, "type" },
241  { "out", "fade-out", 0, AV_OPT_TYPE_CONST, {.i64 = 1 }, 0, 0, FLAGS, "type" },
242  { "start_sample", "set number of first sample to start fading", OFFSET(start_sample), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT64_MAX, FLAGS },
243  { "ss", "set number of first sample to start fading", OFFSET(start_sample), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT64_MAX, FLAGS },
244  { "nb_samples", "set number of samples for fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT64, {.i64 = 44100}, 1, INT64_MAX, FLAGS },
245  { "ns", "set number of samples for fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT64, {.i64 = 44100}, 1, INT64_MAX, FLAGS },
246  { "start_time", "set time to start fading", OFFSET(start_time), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, INT64_MAX, FLAGS },
247  { "st", "set time to start fading", OFFSET(start_time), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, INT64_MAX, FLAGS },
248  { "duration", "set fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, INT32_MAX, FLAGS },
249  { "d", "set fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, INT32_MAX, FLAGS },
250  { "curve", "set fade curve type", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, 0, NB_CURVES - 1, FLAGS, "curve" },
251  { "c", "set fade curve type", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, 0, NB_CURVES - 1, FLAGS, "curve" },
252  { "tri", "linear slope", 0, AV_OPT_TYPE_CONST, {.i64 = TRI }, 0, 0, FLAGS, "curve" },
253  { "qsin", "quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = QSIN }, 0, 0, FLAGS, "curve" },
254  { "esin", "exponential sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = ESIN }, 0, 0, FLAGS, "curve" },
255  { "hsin", "half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = HSIN }, 0, 0, FLAGS, "curve" },
256  { "log", "logarithmic", 0, AV_OPT_TYPE_CONST, {.i64 = LOG }, 0, 0, FLAGS, "curve" },
257  { "ipar", "inverted parabola", 0, AV_OPT_TYPE_CONST, {.i64 = IPAR }, 0, 0, FLAGS, "curve" },
258  { "qua", "quadratic", 0, AV_OPT_TYPE_CONST, {.i64 = QUA }, 0, 0, FLAGS, "curve" },
259  { "cub", "cubic", 0, AV_OPT_TYPE_CONST, {.i64 = CUB }, 0, 0, FLAGS, "curve" },
260  { "squ", "square root", 0, AV_OPT_TYPE_CONST, {.i64 = SQU }, 0, 0, FLAGS, "curve" },
261  { "cbr", "cubic root", 0, AV_OPT_TYPE_CONST, {.i64 = CBR }, 0, 0, FLAGS, "curve" },
262  { "par", "parabola", 0, AV_OPT_TYPE_CONST, {.i64 = PAR }, 0, 0, FLAGS, "curve" },
263  { "exp", "exponential", 0, AV_OPT_TYPE_CONST, {.i64 = EXP }, 0, 0, FLAGS, "curve" },
264  { "iqsin", "inverted quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IQSIN}, 0, 0, FLAGS, "curve" },
265  { "ihsin", "inverted half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IHSIN}, 0, 0, FLAGS, "curve" },
266  { "dese", "double-exponential seat", 0, AV_OPT_TYPE_CONST, {.i64 = DESE }, 0, 0, FLAGS, "curve" },
267  { "desi", "double-exponential sigmoid", 0, AV_OPT_TYPE_CONST, {.i64 = DESI }, 0, 0, FLAGS, "curve" },
268  { "losi", "logistic sigmoid", 0, AV_OPT_TYPE_CONST, {.i64 = LOSI }, 0, 0, FLAGS, "curve" },
269  { NULL }
270 };
271 
272 AVFILTER_DEFINE_CLASS(afade);
273 
274 static av_cold int init(AVFilterContext *ctx)
275 {
276  AudioFadeContext *s = ctx->priv;
277 
278  if (INT64_MAX - s->nb_samples < s->start_sample)
279  return AVERROR(EINVAL);
280 
281  return 0;
282 }
283 
284 static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
285 {
286  AudioFadeContext *s = inlink->dst->priv;
287  AVFilterLink *outlink = inlink->dst->outputs[0];
288  int nb_samples = buf->nb_samples;
289  AVFrame *out_buf;
290  int64_t cur_sample = av_rescale_q(buf->pts, inlink->time_base, (AVRational){1, inlink->sample_rate});
291 
292  if ((!s->type && (s->start_sample + s->nb_samples < cur_sample)) ||
293  ( s->type && (cur_sample + nb_samples < s->start_sample)))
294  return ff_filter_frame(outlink, buf);
295 
296  if (av_frame_is_writable(buf)) {
297  out_buf = buf;
298  } else {
299  out_buf = ff_get_audio_buffer(outlink, nb_samples);
300  if (!out_buf)
301  return AVERROR(ENOMEM);
302  av_frame_copy_props(out_buf, buf);
303  }
304 
305  if ((!s->type && (cur_sample + nb_samples < s->start_sample)) ||
306  ( s->type && (s->start_sample + s->nb_samples < cur_sample))) {
307  av_samples_set_silence(out_buf->extended_data, 0, nb_samples,
308  out_buf->channels, out_buf->format);
309  } else {
310  int64_t start;
311 
312  if (!s->type)
313  start = cur_sample - s->start_sample;
314  else
315  start = s->start_sample + s->nb_samples - cur_sample;
316 
317  s->fade_samples(out_buf->extended_data, buf->extended_data,
318  nb_samples, buf->channels,
319  s->type ? -1 : 1, start,
320  s->nb_samples, s->curve);
321  }
322 
323  if (buf != out_buf)
324  av_frame_free(&buf);
325 
326  return ff_filter_frame(outlink, out_buf);
327 }
328 
329 static const AVFilterPad avfilter_af_afade_inputs[] = {
330  {
331  .name = "default",
332  .type = AVMEDIA_TYPE_AUDIO,
333  .filter_frame = filter_frame,
334  },
335  { NULL }
336 };
337 
338 static const AVFilterPad avfilter_af_afade_outputs[] = {
339  {
340  .name = "default",
341  .type = AVMEDIA_TYPE_AUDIO,
342  .config_props = config_output,
343  },
344  { NULL }
345 };
346 
347 AVFilter ff_af_afade = {
348  .name = "afade",
349  .description = NULL_IF_CONFIG_SMALL("Fade in/out input audio."),
350  .query_formats = query_formats,
351  .priv_size = sizeof(AudioFadeContext),
352  .init = init,
353  .inputs = avfilter_af_afade_inputs,
354  .outputs = avfilter_af_afade_outputs,
355  .priv_class = &afade_class,
356  .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC,
357 };
358 
359 #endif /* CONFIG_AFADE_FILTER */
360 
361 #if CONFIG_ACROSSFADE_FILTER
362 
363 static const AVOption acrossfade_options[] = {
364  { "nb_samples", "set number of samples for cross fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 44100}, 1, INT32_MAX/10, FLAGS },
365  { "ns", "set number of samples for cross fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 44100}, 1, INT32_MAX/10, FLAGS },
366  { "duration", "set cross fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, 60000000, FLAGS },
367  { "d", "set cross fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, 60000000, FLAGS },
368  { "overlap", "overlap 1st stream end with 2nd stream start", OFFSET(overlap), AV_OPT_TYPE_BOOL, {.i64 = 1 }, 0, 1, FLAGS },
369  { "o", "overlap 1st stream end with 2nd stream start", OFFSET(overlap), AV_OPT_TYPE_BOOL, {.i64 = 1 }, 0, 1, FLAGS },
370  { "curve1", "set fade curve type for 1st stream", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, 0, NB_CURVES - 1, FLAGS, "curve" },
371  { "c1", "set fade curve type for 1st stream", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, 0, NB_CURVES - 1, FLAGS, "curve" },
372  { "tri", "linear slope", 0, AV_OPT_TYPE_CONST, {.i64 = TRI }, 0, 0, FLAGS, "curve" },
373  { "qsin", "quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = QSIN }, 0, 0, FLAGS, "curve" },
374  { "esin", "exponential sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = ESIN }, 0, 0, FLAGS, "curve" },
375  { "hsin", "half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = HSIN }, 0, 0, FLAGS, "curve" },
376  { "log", "logarithmic", 0, AV_OPT_TYPE_CONST, {.i64 = LOG }, 0, 0, FLAGS, "curve" },
377  { "ipar", "inverted parabola", 0, AV_OPT_TYPE_CONST, {.i64 = IPAR }, 0, 0, FLAGS, "curve" },
378  { "qua", "quadratic", 0, AV_OPT_TYPE_CONST, {.i64 = QUA }, 0, 0, FLAGS, "curve" },
379  { "cub", "cubic", 0, AV_OPT_TYPE_CONST, {.i64 = CUB }, 0, 0, FLAGS, "curve" },
380  { "squ", "square root", 0, AV_OPT_TYPE_CONST, {.i64 = SQU }, 0, 0, FLAGS, "curve" },
381  { "cbr", "cubic root", 0, AV_OPT_TYPE_CONST, {.i64 = CBR }, 0, 0, FLAGS, "curve" },
382  { "par", "parabola", 0, AV_OPT_TYPE_CONST, {.i64 = PAR }, 0, 0, FLAGS, "curve" },
383  { "exp", "exponential", 0, AV_OPT_TYPE_CONST, {.i64 = EXP }, 0, 0, FLAGS, "curve" },
384  { "iqsin", "inverted quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IQSIN}, 0, 0, FLAGS, "curve" },
385  { "ihsin", "inverted half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IHSIN}, 0, 0, FLAGS, "curve" },
386  { "dese", "double-exponential seat", 0, AV_OPT_TYPE_CONST, {.i64 = DESE }, 0, 0, FLAGS, "curve" },
387  { "desi", "double-exponential sigmoid", 0, AV_OPT_TYPE_CONST, {.i64 = DESI }, 0, 0, FLAGS, "curve" },
388  { "losi", "logistic sigmoid", 0, AV_OPT_TYPE_CONST, {.i64 = LOSI }, 0, 0, FLAGS, "curve" },
389  { "curve2", "set fade curve type for 2nd stream", OFFSET(curve2), AV_OPT_TYPE_INT, {.i64 = TRI }, 0, NB_CURVES - 1, FLAGS, "curve" },
390  { "c2", "set fade curve type for 2nd stream", OFFSET(curve2), AV_OPT_TYPE_INT, {.i64 = TRI }, 0, NB_CURVES - 1, FLAGS, "curve" },
391  { NULL }
392 };
393 
394 AVFILTER_DEFINE_CLASS(acrossfade);
395 
396 #define CROSSFADE_PLANAR(name, type) \
397 static void crossfade_samples_## name ##p(uint8_t **dst, uint8_t * const *cf0, \
398  uint8_t * const *cf1, \
399  int nb_samples, int channels, \
400  int curve0, int curve1) \
401 { \
402  int i, c; \
403  \
404  for (i = 0; i < nb_samples; i++) { \
405  double gain0 = fade_gain(curve0, nb_samples - 1 - i, nb_samples); \
406  double gain1 = fade_gain(curve1, i, nb_samples); \
407  for (c = 0; c < channels; c++) { \
408  type *d = (type *)dst[c]; \
409  const type *s0 = (type *)cf0[c]; \
410  const type *s1 = (type *)cf1[c]; \
411  \
412  d[i] = s0[i] * gain0 + s1[i] * gain1; \
413  } \
414  } \
415 }
416 
417 #define CROSSFADE(name, type) \
418 static void crossfade_samples_## name (uint8_t **dst, uint8_t * const *cf0, \
419  uint8_t * const *cf1, \
420  int nb_samples, int channels, \
421  int curve0, int curve1) \
422 { \
423  type *d = (type *)dst[0]; \
424  const type *s0 = (type *)cf0[0]; \
425  const type *s1 = (type *)cf1[0]; \
426  int i, c, k = 0; \
427  \
428  for (i = 0; i < nb_samples; i++) { \
429  double gain0 = fade_gain(curve0, nb_samples - 1 - i, nb_samples); \
430  double gain1 = fade_gain(curve1, i, nb_samples); \
431  for (c = 0; c < channels; c++, k++) \
432  d[k] = s0[k] * gain0 + s1[k] * gain1; \
433  } \
434 }
435 
436 CROSSFADE_PLANAR(dbl, double)
437 CROSSFADE_PLANAR(flt, float)
438 CROSSFADE_PLANAR(s16, int16_t)
439 CROSSFADE_PLANAR(s32, int32_t)
440 
441 CROSSFADE(dbl, double)
442 CROSSFADE(flt, float)
443 CROSSFADE(s16, int16_t)
444 CROSSFADE(s32, int32_t)
445 
446 static int activate(AVFilterContext *ctx)
447 {
448  AudioFadeContext *s = ctx->priv;
449  AVFilterLink *outlink = ctx->outputs[0];
450  AVFrame *in = NULL, *out, *cf[2] = { NULL };
451  int ret = 0, nb_samples, status;
452  int64_t pts;
453 
454  FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, ctx);
455 
456  if (s->crossfade_is_over) {
457  ret = ff_inlink_consume_frame(ctx->inputs[1], &in);
458  if (ret < 0) {
459  return ret;
460  } else if (ff_inlink_acknowledge_status(ctx->inputs[1], &status, &pts)) {
461  ff_outlink_set_status(ctx->outputs[0], status, pts);
462  return 0;
463  } else {
464  if (ff_outlink_frame_wanted(ctx->outputs[0]) && !in) {
465  ff_inlink_request_frame(ctx->inputs[1]);
466  return 0;
467  }
468  }
469  in->pts = s->pts;
470  s->pts += av_rescale_q(in->nb_samples,
471  (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
472  return ff_filter_frame(outlink, in);
473  }
474 
475  if (ff_framequeue_queued_samples(&ctx->inputs[0]->fifo) > s->nb_samples) {
476  nb_samples = ff_framequeue_queued_samples(&ctx->inputs[0]->fifo) - s->nb_samples;
477  if (nb_samples > 0) {
478  ret = ff_inlink_consume_samples(ctx->inputs[0], nb_samples, nb_samples, &in);
479  if (ret < 0) {
480  return ret;
481  }
482  }
483  in->pts = s->pts;
484  s->pts += av_rescale_q(in->nb_samples,
485  (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
486  return ff_filter_frame(outlink, in);
487  } else if (ff_framequeue_queued_samples(&ctx->inputs[1]->fifo) >= s->nb_samples) {
488  if (s->overlap) {
489  out = ff_get_audio_buffer(outlink, s->nb_samples);
490  if (!out)
491  return AVERROR(ENOMEM);
492 
493  ret = ff_inlink_consume_samples(ctx->inputs[0], s->nb_samples, s->nb_samples, &cf[0]);
494  if (ret < 0) {
495  av_frame_free(&out);
496  return ret;
497  }
498 
499  ret = ff_inlink_consume_samples(ctx->inputs[1], s->nb_samples, s->nb_samples, &cf[1]);
500  if (ret < 0) {
501  av_frame_free(&out);
502  return ret;
503  }
504 
505  s->crossfade_samples(out->extended_data, cf[0]->extended_data,
506  cf[1]->extended_data,
507  s->nb_samples, out->channels,
508  s->curve, s->curve2);
509  out->pts = s->pts;
510  s->pts += av_rescale_q(s->nb_samples,
511  (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
512  s->crossfade_is_over = 1;
513  av_frame_free(&cf[0]);
514  av_frame_free(&cf[1]);
515  return ff_filter_frame(outlink, out);
516  } else {
517  out = ff_get_audio_buffer(outlink, s->nb_samples);
518  if (!out)
519  return AVERROR(ENOMEM);
520 
521  ret = ff_inlink_consume_samples(ctx->inputs[0], s->nb_samples, s->nb_samples, &cf[0]);
522  if (ret < 0) {
523  av_frame_free(&out);
524  return ret;
525  }
526 
527  s->fade_samples(out->extended_data, cf[0]->extended_data, s->nb_samples,
528  outlink->channels, -1, s->nb_samples - 1, s->nb_samples, s->curve);
529  out->pts = s->pts;
530  s->pts += av_rescale_q(s->nb_samples,
531  (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
532  av_frame_free(&cf[0]);
533  ret = ff_filter_frame(outlink, out);
534  if (ret < 0)
535  return ret;
536 
537  out = ff_get_audio_buffer(outlink, s->nb_samples);
538  if (!out)
539  return AVERROR(ENOMEM);
540 
541  ret = ff_inlink_consume_samples(ctx->inputs[1], s->nb_samples, s->nb_samples, &cf[1]);
542  if (ret < 0) {
543  av_frame_free(&out);
544  return ret;
545  }
546 
547  s->fade_samples(out->extended_data, cf[1]->extended_data, s->nb_samples,
548  outlink->channels, 1, 0, s->nb_samples, s->curve2);
549  out->pts = s->pts;
550  s->pts += av_rescale_q(s->nb_samples,
551  (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
552  s->crossfade_is_over = 1;
553  av_frame_free(&cf[1]);
554  return ff_filter_frame(outlink, out);
555  }
556  } else if (ff_outlink_frame_wanted(ctx->outputs[0])) {
557  if (!s->cf0_eof && ctx->inputs[0]->status_in) {
558  s->cf0_eof = 1;
559  }
560  if (ctx->inputs[1]->status_in) {
561  ff_outlink_set_status(ctx->outputs[0], AVERROR_EOF, AV_NOPTS_VALUE);
562  return 0;
563  }
564  if (!s->cf0_eof)
565  ff_inlink_request_frame(ctx->inputs[0]);
566  else
567  ff_inlink_request_frame(ctx->inputs[1]);
568  return 0;
569  }
570 
571  return ret;
572 }
573 
574 static int acrossfade_config_output(AVFilterLink *outlink)
575 {
576  AVFilterContext *ctx = outlink->src;
577  AudioFadeContext *s = ctx->priv;
578 
579  if (ctx->inputs[0]->sample_rate != ctx->inputs[1]->sample_rate) {
580  av_log(ctx, AV_LOG_ERROR,
581  "Inputs must have the same sample rate "
582  "%d for in0 vs %d for in1\n",
583  ctx->inputs[0]->sample_rate, ctx->inputs[1]->sample_rate);
584  return AVERROR(EINVAL);
585  }
586 
587  outlink->sample_rate = ctx->inputs[0]->sample_rate;
588  outlink->time_base = ctx->inputs[0]->time_base;
589  outlink->channel_layout = ctx->inputs[0]->channel_layout;
590  outlink->channels = ctx->inputs[0]->channels;
591 
592  switch (outlink->format) {
593  case AV_SAMPLE_FMT_DBL: s->crossfade_samples = crossfade_samples_dbl; break;
594  case AV_SAMPLE_FMT_DBLP: s->crossfade_samples = crossfade_samples_dblp; break;
595  case AV_SAMPLE_FMT_FLT: s->crossfade_samples = crossfade_samples_flt; break;
596  case AV_SAMPLE_FMT_FLTP: s->crossfade_samples = crossfade_samples_fltp; break;
597  case AV_SAMPLE_FMT_S16: s->crossfade_samples = crossfade_samples_s16; break;
598  case AV_SAMPLE_FMT_S16P: s->crossfade_samples = crossfade_samples_s16p; break;
599  case AV_SAMPLE_FMT_S32: s->crossfade_samples = crossfade_samples_s32; break;
600  case AV_SAMPLE_FMT_S32P: s->crossfade_samples = crossfade_samples_s32p; break;
601  }
602 
603  config_output(outlink);
604 
605  return 0;
606 }
607 
608 static const AVFilterPad avfilter_af_acrossfade_inputs[] = {
609  {
610  .name = "crossfade0",
611  .type = AVMEDIA_TYPE_AUDIO,
612  },
613  {
614  .name = "crossfade1",
615  .type = AVMEDIA_TYPE_AUDIO,
616  },
617  { NULL }
618 };
619 
620 static const AVFilterPad avfilter_af_acrossfade_outputs[] = {
621  {
622  .name = "default",
623  .type = AVMEDIA_TYPE_AUDIO,
624  .config_props = acrossfade_config_output,
625  },
626  { NULL }
627 };
628 
629 AVFilter ff_af_acrossfade = {
630  .name = "acrossfade",
631  .description = NULL_IF_CONFIG_SMALL("Cross fade two input audio streams."),
632  .query_formats = query_formats,
633  .priv_size = sizeof(AudioFadeContext),
634  .activate = activate,
635  .priv_class = &acrossfade_class,
636  .inputs = avfilter_af_acrossfade_inputs,
637  .outputs = avfilter_af_acrossfade_outputs,
638 };
639 
640 #endif /* CONFIG_ACROSSFADE_FILTER */
AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:69
ff_inlink_consume_frame
int ff_inlink_consume_frame(AVFilterLink *link, AVFrame **rframe)
Take a frame from the link's FIFO and update the link's stats.
Definition: avfilter.c:1481
NULL
#define NULL
Definition: coverity.c:32
ff_set_common_channel_layouts
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
AV_OPT_TYPE_INT64
Definition: opt.h:224
CUBE
#define CUBE(a)
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:226
activate
static int activate(AVFilterContext *ctx)
Definition: af_adelay.c:237
AVOption
AVOption.
Definition: opt.h:246
C
#define C
FF_FILTER_FORWARD_STATUS_BACK_ALL
#define FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, filter)
Forward the status on an output link to all input links.
Definition: filters.h:212
avfilter.h
Main libavfilter public API header.
init
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
AV_OPT_TYPE_INT
Definition: opt.h:223
channels
channels
Definition: aptx.c:30
config_output
static int config_output(AVFilterLink *outlink)
Definition: af_afade.c:211
AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:59
AV_SAMPLE_FMT_DBLP
double, planar
Definition: samplefmt.h:70
AV_SAMPLE_FMT_FLT
float
Definition: samplefmt.h:63
IHSIN
Definition: af_afade.c:60
src
#define src
Definition: vp8dsp.c:254
ff_outlink_set_status
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
ESIN
Definition: af_afade.c:60
ff_inlink_request_frame
void ff_inlink_request_frame(AVFilterLink *link)
Mark that a frame is wanted on the link.
Definition: avfilter.c:1607
ff_outlink_frame_wanted
static int ff_outlink_frame_wanted(AVFilterLink *link)
Test if a frame is wanted on an output link.
Definition: filters.h:172
ff_make_format_list
AVFilterFormats * ff_make_format_list(const int *fmts)
Create a list of supported formats.
Definition: formats.c:283
AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC
#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:125
AVFilterPad::name
const char * name
Pad name.
Definition: internal.h:60
AVFilterContext::inputs
AVFilterLink ** inputs
array of pointers to input links
Definition: avfilter.h:346
AV_OPT_TYPE_CONST
Definition: opt.h:232
AudioFadeContext::fade_samples
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:51
ff_filter_frame
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1080
start_time
static int64_t start_time
Definition: ffplay.c:330
uint8_t
uint8_t
Definition: audio_convert.c:194
av_cold
#define av_cold
Definition: attributes.h:82
opt.h
AVOptions.
AudioFadeContext::crossfade_samples
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:54
query_formats
static int query_formats(AVFilterContext *ctx)
Definition: af_afade.c:65
FADE_PLANAR
#define FADE_PLANAR(name, type)
Definition: af_afade.c:167
AVFrame::pts
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:319
duration
int64_t duration
Definition: movenc.c:63
LOSI
Definition: af_afade.c:60
AVERROR_EOF
#define AVERROR_EOF
End of file.
Definition: error.h:55
AV_SAMPLE_FMT_S32
signed 32 bits
Definition: samplefmt.h:62
A
#define A(x)
Definition: vp56_arith.h:28
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:28
AVFilterPad
A filter pad used for either input or output.
Definition: internal.h:54
av_rescale_q
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
ff_inlink_acknowledge_status
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:1436
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
ff_set_common_formats
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
av_samples_set_silence
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
CurveType
Definition: af_afade.c:60
ff_get_audio_buffer
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
AVERROR
#define AVERROR(e)
Definition: error.h:43
av_frame_free
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:202
B
#define B
Definition: huffyuvdsp.h:32
NULL_IF_CONFIG_SMALL
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:186
AVFilterContext::priv
void * priv
private data for use by the filter
Definition: avfilter.h:353
cbrt
#define cbrt
Definition: tablegen.h:35
AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
exp
int8_t exp
Definition: eval.c:72
AVAudioFifo
Context for an Audio FIFO Buffer.
Definition: audio_fifo.c:34
DESI
Definition: af_afade.c:60
AVFrame::channels
int channels
number of audio channels, only used for audio.
Definition: frame.h:531
av_rescale
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
AV_TIME_BASE
#define AV_TIME_BASE
Internal time base represented as integer.
Definition: avutil.h:254
AV_SAMPLE_FMT_S32P
signed 32 bits, planar
Definition: samplefmt.h:68
AudioFadeContext::start_sample
int64_t start_sample
Definition: af_afade.c:41
void
typedef void(APIENTRY *FF_PFNGLACTIVETEXTUREPROC)(GLenum texture)
HSIN
Definition: af_afade.c:60
int32_t
int32_t
Definition: audio_convert.c:194
ctx
AVFormatContext * ctx
Definition: movenc.c:48
QUA
Definition: af_afade.c:60
AV_OPT_TYPE_BOOL
Definition: opt.h:240
s
#define s(width, name)
Definition: cbs_vp9.c:257
AudioFadeContext::pts
int64_t pts
Definition: af_afade.c:49
AV_SAMPLE_FMT_DBL
double
Definition: samplefmt.h:64
AudioFadeContext::crossfade_is_over
int crossfade_is_over
Definition: af_afade.c:47
inputs
static const AVFilterPad inputs[]
Definition: af_acontrast.c:193
AudioFadeContext::duration
int64_t duration
Definition: af_afade.c:42
outputs
static const AVFilterPad outputs[]
Definition: af_acontrast.c:203
AVFilterChannelLayouts
A list of supported channel layouts.
Definition: formats.h:85
NB_CURVES
Definition: af_afade.c:60
AVFrame::format
int format
format of the frame, -1 if unknown or unset Values correspond to enum AVPixelFormat for video frames...
Definition: frame.h:299
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
av_frame_is_writable
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
Definition: frame.c:594
ff_inlink_consume_samples
int ff_inlink_consume_samples(AVFilterLink *link, unsigned min, unsigned max, AVFrame **rframe)
Take samples from the link's FIFO and update the link's stats.
Definition: avfilter.c:1500
AudioFadeContext::fifo
AVAudioFifo * fifo[2]
Definition: af_afade.c:48
in
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;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);returnNULL;}returnac;}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;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->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);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> in
Definition: audio_convert.c:194
buf
void * buf
Definition: avisynth_c.h:690
CUB
Definition: af_afade.c:60
AudioFadeContext::start_time
int64_t start_time
Definition: af_afade.c:43
type
GLint GLenum type
Definition: opengl_enc.c:105
SQU
Definition: af_afade.c:60
AVClass
Describe the class of an AVClass context structure.
Definition: log.h:67
AVFilter
Filter definition.
Definition: avfilter.h:144
index
int index
Definition: gxfenc.c:89
AVRational
Rational number (pair of numerator and denominator).
Definition: rational.h:58
PAR
Definition: af_afade.c:60
FLAGS
#define FLAGS
Definition: af_afade.c:63
AVFilter::name
const char * name
Filter name.
Definition: avfilter.h:148
LOG
Definition: af_afade.c:60
DESE
Definition: af_afade.c:60
ff_framequeue_queued_samples
static uint64_t ff_framequeue_queued_samples(const FFFrameQueue *fq)
Get the number of queued samples.
Definition: framequeue.h:154
AVFilterContext::outputs
AVFilterLink ** outputs
array of pointers to output links
Definition: avfilter.h:350
layouts
enum MovChannelLayoutTag * layouts
Definition: mov_chan.c:434
ff_all_samplerates
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:395
pts
static int64_t pts
Definition: transcode_aac.c:647
flags
#define flags(name, subs,...)
Definition: cbs_av1.c:596
filter_frame
static int filter_frame(DBEContext *s, AVFrame *frame)
Definition: dolby_e.c:565
IPAR
Definition: af_afade.c:60
EXP
Definition: af_afade.c:60
AV_OPT_TYPE_DURATION
Definition: opt.h:237
TRI
Definition: af_afade.c:60
AV_SAMPLE_FMT_S16
signed 16 bits
Definition: samplefmt.h:61
fade_gain
static double fade_gain(int curve, int64_t index, int64_t range)
Definition: af_afade.c:98
OFFSET
#define OFFSET(x)
Definition: af_afade.c:62
QSIN
Definition: af_afade.c:60
audio_fifo.h
Audio FIFO Buffer.
AVFILTER_DEFINE_CLASS
#define AVFILTER_DEFINE_CLASS(fname)
Definition: internal.h:334
AVFilterFormats
A list of supported formats for one end of a filter link.
Definition: formats.h:64
AudioFadeContext::nb_samples
int64_t nb_samples
Definition: af_afade.c:40
AVFilterContext
An instance of a filter.
Definition: avfilter.h:338
ff_af_afade
AVFilter ff_af_afade
sample_fmts
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:701
out
FILE * out
Definition: movenc.c:54
start
void INT64 start
Definition: avisynth_c.h:690
AV_SAMPLE_FMT_S16P
signed 16 bits, planar
Definition: samplefmt.h:67
M_PI
#define M_PI
Definition: mathematics.h:52
formats
formats
Definition: signature.h:48
ff_af_acrossfade
AVFilter ff_af_acrossfade
IQSIN
Definition: af_afade.c:60
internal.h
internal API functions
ff_all_channel_counts
AVFilterChannelLayouts * ff_all_channel_counts(void)
Construct an AVFilterChannelLayouts coding for any channel layout, with known or unknown disposition...
Definition: formats.c:410
AVFrame::extended_data
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:273
AVFrame::nb_samples
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:292
ff_set_common_samplerates
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:556
av_frame_copy_props
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
Definition: frame.c:654
FADE
#define FADE(name, type)
Definition: af_afade.c:185
AV_NOPTS_VALUE
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248
CBR
Definition: af_afade.c:60
a
a
Definition: h264pred_template.c:468
Generated on Tue Nov 6 2018 18:11:18 for FFmpeg by   doxygen 1.8.6