FFmpeg  4.2.2
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 
60 {
63  static const enum AVSampleFormat sample_fmts[] = {
69  };
70  int ret;
71 
72  layouts = ff_all_channel_counts();
73  if (!layouts)
74  return AVERROR(ENOMEM);
75  ret = ff_set_common_channel_layouts(ctx, layouts);
76  if (ret < 0)
77  return ret;
78 
79  formats = ff_make_format_list(sample_fmts);
80  if (!formats)
81  return AVERROR(ENOMEM);
82  ret = ff_set_common_formats(ctx, formats);
83  if (ret < 0)
84  return ret;
85 
86  formats = ff_all_samplerates();
87  if (!formats)
88  return AVERROR(ENOMEM);
89  return ff_set_common_samplerates(ctx, formats);
90 }
91 
92 static double fade_gain(int curve, int64_t index, int64_t range)
93 {
94 #define CUBE(a) ((a)*(a)*(a))
95  double gain;
96 
97  gain = av_clipd(1.0 * index / range, 0, 1.0);
98 
99  switch (curve) {
100  case QSIN:
101  gain = sin(gain * M_PI / 2.0);
102  break;
103  case IQSIN:
104  /* 0.6... = 2 / M_PI */
105  gain = 0.6366197723675814 * asin(gain);
106  break;
107  case ESIN:
108  gain = 1.0 - cos(M_PI / 4.0 * (CUBE(2.0*gain - 1) + 1));
109  break;
110  case HSIN:
111  gain = (1.0 - cos(gain * M_PI)) / 2.0;
112  break;
113  case IHSIN:
114  /* 0.3... = 1 / M_PI */
115  gain = 0.3183098861837907 * acos(1 - 2 * gain);
116  break;
117  case EXP:
118  /* -11.5... = 5*ln(0.1) */
119  gain = exp(-11.512925464970227 * (1 - gain));
120  break;
121  case LOG:
122  gain = av_clipd(1 + 0.2 * log10(gain), 0, 1.0);
123  break;
124  case PAR:
125  gain = 1 - sqrt(1 - gain);
126  break;
127  case IPAR:
128  gain = (1 - (1 - gain) * (1 - gain));
129  break;
130  case QUA:
131  gain *= gain;
132  break;
133  case CUB:
134  gain = CUBE(gain);
135  break;
136  case SQU:
137  gain = sqrt(gain);
138  break;
139  case CBR:
140  gain = cbrt(gain);
141  break;
142  case DESE:
143  gain = gain <= 0.5 ? cbrt(2 * gain) / 2: 1 - cbrt(2 * (1 - gain)) / 2;
144  break;
145  case DESI:
146  gain = gain <= 0.5 ? CUBE(2 * gain) / 2: 1 - CUBE(2 * (1 - gain)) / 2;
147  break;
148  case LOSI: {
149  const double a = 1. / (1. - 0.787) - 1;
150  double A = 1. / (1.0 + exp(0 -((gain-0.5) * a * 2.0)));
151  double B = 1. / (1.0 + exp(a));
152  double C = 1. / (1.0 + exp(0-a));
153  gain = (A - B) / (C - B);
154  }
155  break;
156  case NONE:
157  gain = 1.0;
158  break;
159  }
160 
161  return gain;
162 }
163 
164 #define FADE_PLANAR(name, type) \
165 static void fade_samples_## name ##p(uint8_t **dst, uint8_t * const *src, \
166  int nb_samples, int channels, int dir, \
167  int64_t start, int64_t range, int curve) \
168 { \
169  int i, c; \
170  \
171  for (i = 0; i < nb_samples; i++) { \
172  double gain = fade_gain(curve, start + i * dir, range); \
173  for (c = 0; c < channels; c++) { \
174  type *d = (type *)dst[c]; \
175  const type *s = (type *)src[c]; \
176  \
177  d[i] = s[i] * gain; \
178  } \
179  } \
180 }
181 
182 #define FADE(name, type) \
183 static void fade_samples_## name (uint8_t **dst, uint8_t * const *src, \
184  int nb_samples, int channels, int dir, \
185  int64_t start, int64_t range, int curve) \
186 { \
187  type *d = (type *)dst[0]; \
188  const type *s = (type *)src[0]; \
189  int i, c, k = 0; \
190  \
191  for (i = 0; i < nb_samples; i++) { \
192  double gain = fade_gain(curve, start + i * dir, range); \
193  for (c = 0; c < channels; c++, k++) \
194  d[k] = s[k] * gain; \
195  } \
196 }
197 
198 FADE_PLANAR(dbl, double)
199 FADE_PLANAR(flt, float)
200 FADE_PLANAR(s16, int16_t)
201 FADE_PLANAR(s32, int32_t)
202 
203 FADE(dbl, double)
204 FADE(flt, float)
205 FADE(s16, int16_t)
206 FADE(s32, int32_t)
207 
208 static int config_output(AVFilterLink *outlink)
209 {
210  AVFilterContext *ctx = outlink->src;
211  AudioFadeContext *s = ctx->priv;
212 
213  switch (outlink->format) {
214  case AV_SAMPLE_FMT_DBL: s->fade_samples = fade_samples_dbl; break;
215  case AV_SAMPLE_FMT_DBLP: s->fade_samples = fade_samples_dblp; break;
216  case AV_SAMPLE_FMT_FLT: s->fade_samples = fade_samples_flt; break;
217  case AV_SAMPLE_FMT_FLTP: s->fade_samples = fade_samples_fltp; break;
218  case AV_SAMPLE_FMT_S16: s->fade_samples = fade_samples_s16; break;
219  case AV_SAMPLE_FMT_S16P: s->fade_samples = fade_samples_s16p; break;
220  case AV_SAMPLE_FMT_S32: s->fade_samples = fade_samples_s32; break;
221  case AV_SAMPLE_FMT_S32P: s->fade_samples = fade_samples_s32p; break;
222  }
223 
224  if (s->duration)
225  s->nb_samples = av_rescale(s->duration, outlink->sample_rate, AV_TIME_BASE);
226  if (s->start_time)
227  s->start_sample = av_rescale(s->start_time, outlink->sample_rate, AV_TIME_BASE);
228 
229  return 0;
230 }
231 
232 #if CONFIG_AFADE_FILTER
233 
234 static const AVOption afade_options[] = {
235  { "type", "set the fade direction", OFFSET(type), AV_OPT_TYPE_INT, {.i64 = 0 }, 0, 1, FLAGS, "type" },
236  { "t", "set the fade direction", OFFSET(type), AV_OPT_TYPE_INT, {.i64 = 0 }, 0, 1, FLAGS, "type" },
237  { "in", "fade-in", 0, AV_OPT_TYPE_CONST, {.i64 = 0 }, 0, 0, FLAGS, "type" },
238  { "out", "fade-out", 0, AV_OPT_TYPE_CONST, {.i64 = 1 }, 0, 0, FLAGS, "type" },
239  { "start_sample", "set number of first sample to start fading", OFFSET(start_sample), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT64_MAX, FLAGS },
240  { "ss", "set number of first sample to start fading", OFFSET(start_sample), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT64_MAX, FLAGS },
241  { "nb_samples", "set number of samples for fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT64, {.i64 = 44100}, 1, INT64_MAX, FLAGS },
242  { "ns", "set number of samples for fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT64, {.i64 = 44100}, 1, INT64_MAX, FLAGS },
243  { "start_time", "set time to start fading", OFFSET(start_time), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, INT64_MAX, FLAGS },
244  { "st", "set time to start fading", OFFSET(start_time), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, INT64_MAX, FLAGS },
245  { "duration", "set fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, INT64_MAX, FLAGS },
246  { "d", "set fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, INT64_MAX, FLAGS },
247  { "curve", "set fade curve type", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, 0, NB_CURVES - 1, FLAGS, "curve" },
248  { "c", "set fade curve type", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, 0, NB_CURVES - 1, FLAGS, "curve" },
249  { "tri", "linear slope", 0, AV_OPT_TYPE_CONST, {.i64 = TRI }, 0, 0, FLAGS, "curve" },
250  { "qsin", "quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = QSIN }, 0, 0, FLAGS, "curve" },
251  { "esin", "exponential sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = ESIN }, 0, 0, FLAGS, "curve" },
252  { "hsin", "half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = HSIN }, 0, 0, FLAGS, "curve" },
253  { "log", "logarithmic", 0, AV_OPT_TYPE_CONST, {.i64 = LOG }, 0, 0, FLAGS, "curve" },
254  { "ipar", "inverted parabola", 0, AV_OPT_TYPE_CONST, {.i64 = IPAR }, 0, 0, FLAGS, "curve" },
255  { "qua", "quadratic", 0, AV_OPT_TYPE_CONST, {.i64 = QUA }, 0, 0, FLAGS, "curve" },
256  { "cub", "cubic", 0, AV_OPT_TYPE_CONST, {.i64 = CUB }, 0, 0, FLAGS, "curve" },
257  { "squ", "square root", 0, AV_OPT_TYPE_CONST, {.i64 = SQU }, 0, 0, FLAGS, "curve" },
258  { "cbr", "cubic root", 0, AV_OPT_TYPE_CONST, {.i64 = CBR }, 0, 0, FLAGS, "curve" },
259  { "par", "parabola", 0, AV_OPT_TYPE_CONST, {.i64 = PAR }, 0, 0, FLAGS, "curve" },
260  { "exp", "exponential", 0, AV_OPT_TYPE_CONST, {.i64 = EXP }, 0, 0, FLAGS, "curve" },
261  { "iqsin", "inverted quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IQSIN}, 0, 0, FLAGS, "curve" },
262  { "ihsin", "inverted half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IHSIN}, 0, 0, FLAGS, "curve" },
263  { "dese", "double-exponential seat", 0, AV_OPT_TYPE_CONST, {.i64 = DESE }, 0, 0, FLAGS, "curve" },
264  { "desi", "double-exponential sigmoid", 0, AV_OPT_TYPE_CONST, {.i64 = DESI }, 0, 0, FLAGS, "curve" },
265  { "losi", "logistic sigmoid", 0, AV_OPT_TYPE_CONST, {.i64 = LOSI }, 0, 0, FLAGS, "curve" },
266  { "nofade", "no fade; keep audio as-is", 0, AV_OPT_TYPE_CONST, {.i64 = NONE }, 0, 0, FLAGS, "curve" },
267  { NULL }
268 };
269 
270 AVFILTER_DEFINE_CLASS(afade);
271 
272 static av_cold int init(AVFilterContext *ctx)
273 {
274  AudioFadeContext *s = ctx->priv;
275 
276  if (INT64_MAX - s->nb_samples < s->start_sample)
277  return AVERROR(EINVAL);
278 
279  return 0;
280 }
281 
282 static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
283 {
284  AudioFadeContext *s = inlink->dst->priv;
285  AVFilterLink *outlink = inlink->dst->outputs[0];
286  int nb_samples = buf->nb_samples;
287  AVFrame *out_buf;
288  int64_t cur_sample = av_rescale_q(buf->pts, inlink->time_base, (AVRational){1, inlink->sample_rate});
289 
290  if ((!s->type && (s->start_sample + s->nb_samples < cur_sample)) ||
291  ( s->type && (cur_sample + nb_samples < s->start_sample)))
292  return ff_filter_frame(outlink, buf);
293 
294  if (av_frame_is_writable(buf)) {
295  out_buf = buf;
296  } else {
297  out_buf = ff_get_audio_buffer(outlink, nb_samples);
298  if (!out_buf)
299  return AVERROR(ENOMEM);
300  av_frame_copy_props(out_buf, buf);
301  }
302 
303  if ((!s->type && (cur_sample + nb_samples < s->start_sample)) ||
304  ( s->type && (s->start_sample + s->nb_samples < cur_sample))) {
305  av_samples_set_silence(out_buf->extended_data, 0, nb_samples,
306  out_buf->channels, out_buf->format);
307  } else {
308  int64_t start;
309 
310  if (!s->type)
311  start = cur_sample - s->start_sample;
312  else
313  start = s->start_sample + s->nb_samples - cur_sample;
314 
315  s->fade_samples(out_buf->extended_data, buf->extended_data,
316  nb_samples, buf->channels,
317  s->type ? -1 : 1, start,
318  s->nb_samples, s->curve);
319  }
320 
321  if (buf != out_buf)
322  av_frame_free(&buf);
323 
324  return ff_filter_frame(outlink, out_buf);
325 }
326 
327 static const AVFilterPad avfilter_af_afade_inputs[] = {
328  {
329  .name = "default",
330  .type = AVMEDIA_TYPE_AUDIO,
331  .filter_frame = filter_frame,
332  },
333  { NULL }
334 };
335 
336 static const AVFilterPad avfilter_af_afade_outputs[] = {
337  {
338  .name = "default",
339  .type = AVMEDIA_TYPE_AUDIO,
340  .config_props = config_output,
341  },
342  { NULL }
343 };
344 
346  .name = "afade",
347  .description = NULL_IF_CONFIG_SMALL("Fade in/out input audio."),
348  .query_formats = query_formats,
349  .priv_size = sizeof(AudioFadeContext),
350  .init = init,
351  .inputs = avfilter_af_afade_inputs,
352  .outputs = avfilter_af_afade_outputs,
353  .priv_class = &afade_class,
355 };
356 
357 #endif /* CONFIG_AFADE_FILTER */
358 
359 #if CONFIG_ACROSSFADE_FILTER
360 
361 static const AVOption acrossfade_options[] = {
362  { "nb_samples", "set number of samples for cross fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 44100}, 1, INT32_MAX/10, FLAGS },
363  { "ns", "set number of samples for cross fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 44100}, 1, INT32_MAX/10, FLAGS },
364  { "duration", "set cross fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, 60000000, FLAGS },
365  { "d", "set cross fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, 60000000, FLAGS },
366  { "overlap", "overlap 1st stream end with 2nd stream start", OFFSET(overlap), AV_OPT_TYPE_BOOL, {.i64 = 1 }, 0, 1, FLAGS },
367  { "o", "overlap 1st stream end with 2nd stream start", OFFSET(overlap), AV_OPT_TYPE_BOOL, {.i64 = 1 }, 0, 1, FLAGS },
368  { "curve1", "set fade curve type for 1st stream", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, 0, NB_CURVES - 1, FLAGS, "curve" },
369  { "c1", "set fade curve type for 1st stream", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, 0, NB_CURVES - 1, FLAGS, "curve" },
370  { "tri", "linear slope", 0, AV_OPT_TYPE_CONST, {.i64 = TRI }, 0, 0, FLAGS, "curve" },
371  { "qsin", "quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = QSIN }, 0, 0, FLAGS, "curve" },
372  { "esin", "exponential sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = ESIN }, 0, 0, FLAGS, "curve" },
373  { "hsin", "half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = HSIN }, 0, 0, FLAGS, "curve" },
374  { "log", "logarithmic", 0, AV_OPT_TYPE_CONST, {.i64 = LOG }, 0, 0, FLAGS, "curve" },
375  { "ipar", "inverted parabola", 0, AV_OPT_TYPE_CONST, {.i64 = IPAR }, 0, 0, FLAGS, "curve" },
376  { "qua", "quadratic", 0, AV_OPT_TYPE_CONST, {.i64 = QUA }, 0, 0, FLAGS, "curve" },
377  { "cub", "cubic", 0, AV_OPT_TYPE_CONST, {.i64 = CUB }, 0, 0, FLAGS, "curve" },
378  { "squ", "square root", 0, AV_OPT_TYPE_CONST, {.i64 = SQU }, 0, 0, FLAGS, "curve" },
379  { "cbr", "cubic root", 0, AV_OPT_TYPE_CONST, {.i64 = CBR }, 0, 0, FLAGS, "curve" },
380  { "par", "parabola", 0, AV_OPT_TYPE_CONST, {.i64 = PAR }, 0, 0, FLAGS, "curve" },
381  { "exp", "exponential", 0, AV_OPT_TYPE_CONST, {.i64 = EXP }, 0, 0, FLAGS, "curve" },
382  { "iqsin", "inverted quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IQSIN}, 0, 0, FLAGS, "curve" },
383  { "ihsin", "inverted half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IHSIN}, 0, 0, FLAGS, "curve" },
384  { "dese", "double-exponential seat", 0, AV_OPT_TYPE_CONST, {.i64 = DESE }, 0, 0, FLAGS, "curve" },
385  { "desi", "double-exponential sigmoid", 0, AV_OPT_TYPE_CONST, {.i64 = DESI }, 0, 0, FLAGS, "curve" },
386  { "losi", "logistic sigmoid", 0, AV_OPT_TYPE_CONST, {.i64 = LOSI }, 0, 0, FLAGS, "curve" },
387  { "nofade", "no fade; keep audio as-is", 0, AV_OPT_TYPE_CONST, {.i64 = NONE }, 0, 0, FLAGS, "curve" },
388  { "curve2", "set fade curve type for 2nd stream", OFFSET(curve2), AV_OPT_TYPE_INT, {.i64 = TRI }, 0, NB_CURVES - 1, FLAGS, "curve" },
389  { "c2", "set fade curve type for 2nd stream", OFFSET(curve2), AV_OPT_TYPE_INT, {.i64 = TRI }, 0, NB_CURVES - 1, FLAGS, "curve" },
390  { NULL }
391 };
392 
393 AVFILTER_DEFINE_CLASS(acrossfade);
394 
395 #define CROSSFADE_PLANAR(name, type) \
396 static void crossfade_samples_## name ##p(uint8_t **dst, uint8_t * const *cf0, \
397  uint8_t * const *cf1, \
398  int nb_samples, int channels, \
399  int curve0, int curve1) \
400 { \
401  int i, c; \
402  \
403  for (i = 0; i < nb_samples; i++) { \
404  double gain0 = fade_gain(curve0, nb_samples - 1 - i, nb_samples); \
405  double gain1 = fade_gain(curve1, i, nb_samples); \
406  for (c = 0; c < channels; c++) { \
407  type *d = (type *)dst[c]; \
408  const type *s0 = (type *)cf0[c]; \
409  const type *s1 = (type *)cf1[c]; \
410  \
411  d[i] = s0[i] * gain0 + s1[i] * gain1; \
412  } \
413  } \
414 }
415 
416 #define CROSSFADE(name, type) \
417 static void crossfade_samples_## name (uint8_t **dst, uint8_t * const *cf0, \
418  uint8_t * const *cf1, \
419  int nb_samples, int channels, \
420  int curve0, int curve1) \
421 { \
422  type *d = (type *)dst[0]; \
423  const type *s0 = (type *)cf0[0]; \
424  const type *s1 = (type *)cf1[0]; \
425  int i, c, k = 0; \
426  \
427  for (i = 0; i < nb_samples; i++) { \
428  double gain0 = fade_gain(curve0, nb_samples - 1 - i, nb_samples); \
429  double gain1 = fade_gain(curve1, i, nb_samples); \
430  for (c = 0; c < channels; c++, k++) \
431  d[k] = s0[k] * gain0 + s1[k] * gain1; \
432  } \
433 }
434 
435 CROSSFADE_PLANAR(dbl, double)
436 CROSSFADE_PLANAR(flt, float)
437 CROSSFADE_PLANAR(s16, int16_t)
438 CROSSFADE_PLANAR(s32, int32_t)
439 
440 CROSSFADE(dbl, double)
441 CROSSFADE(flt, float)
442 CROSSFADE(s16, int16_t)
443 CROSSFADE(s32, int32_t)
444 
445 static int activate(AVFilterContext *ctx)
446 {
447  AudioFadeContext *s = ctx->priv;
448  AVFilterLink *outlink = ctx->outputs[0];
449  AVFrame *in = NULL, *out, *cf[2] = { NULL };
450  int ret = 0, nb_samples, status;
451  int64_t pts;
452 
453  FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, ctx);
454 
455  if (s->crossfade_is_over) {
456  ret = ff_inlink_consume_frame(ctx->inputs[1], &in);
457  if (ret < 0) {
458  return ret;
459  } else if (ff_inlink_acknowledge_status(ctx->inputs[1], &status, &pts)) {
460  ff_outlink_set_status(ctx->outputs[0], status, pts);
461  return 0;
462  } else {
463  if (ff_outlink_frame_wanted(ctx->outputs[0]) && !in) {
465  return 0;
466  }
467  }
468  in->pts = s->pts;
469  s->pts += av_rescale_q(in->nb_samples,
470  (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
471  return ff_filter_frame(outlink, in);
472  }
473 
474  if (ff_inlink_queued_samples(ctx->inputs[0]) > s->nb_samples) {
476  if (nb_samples > 0) {
478  if (ret < 0) {
479  return ret;
480  }
481  }
482  in->pts = s->pts;
483  s->pts += av_rescale_q(in->nb_samples,
484  (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
485  return ff_filter_frame(outlink, in);
486  } else if (ff_inlink_queued_samples(ctx->inputs[1]) >= s->nb_samples) {
487  if (s->overlap) {
488  out = ff_get_audio_buffer(outlink, s->nb_samples);
489  if (!out)
490  return AVERROR(ENOMEM);
491 
492  ret = ff_inlink_consume_samples(ctx->inputs[0], s->nb_samples, s->nb_samples, &cf[0]);
493  if (ret < 0) {
494  av_frame_free(&out);
495  return ret;
496  }
497 
498  ret = ff_inlink_consume_samples(ctx->inputs[1], s->nb_samples, s->nb_samples, &cf[1]);
499  if (ret < 0) {
500  av_frame_free(&out);
501  return ret;
502  }
503 
504  s->crossfade_samples(out->extended_data, cf[0]->extended_data,
505  cf[1]->extended_data,
506  s->nb_samples, out->channels,
507  s->curve, s->curve2);
508  out->pts = s->pts;
509  s->pts += av_rescale_q(s->nb_samples,
510  (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
511  s->crossfade_is_over = 1;
512  av_frame_free(&cf[0]);
513  av_frame_free(&cf[1]);
514  return ff_filter_frame(outlink, out);
515  } else {
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  s->fade_samples(out->extended_data, cf[0]->extended_data, s->nb_samples,
527  outlink->channels, -1, s->nb_samples - 1, s->nb_samples, s->curve);
528  out->pts = s->pts;
529  s->pts += av_rescale_q(s->nb_samples,
530  (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
531  av_frame_free(&cf[0]);
532  ret = ff_filter_frame(outlink, out);
533  if (ret < 0)
534  return ret;
535 
536  out = ff_get_audio_buffer(outlink, s->nb_samples);
537  if (!out)
538  return AVERROR(ENOMEM);
539 
540  ret = ff_inlink_consume_samples(ctx->inputs[1], s->nb_samples, s->nb_samples, &cf[1]);
541  if (ret < 0) {
542  av_frame_free(&out);
543  return ret;
544  }
545 
546  s->fade_samples(out->extended_data, cf[1]->extended_data, s->nb_samples,
547  outlink->channels, 1, 0, s->nb_samples, s->curve2);
548  out->pts = s->pts;
549  s->pts += av_rescale_q(s->nb_samples,
550  (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
551  s->crossfade_is_over = 1;
552  av_frame_free(&cf[1]);
553  return ff_filter_frame(outlink, out);
554  }
555  } else if (ff_outlink_frame_wanted(ctx->outputs[0])) {
556  if (!s->cf0_eof && ff_outlink_get_status(ctx->inputs[0])) {
557  s->cf0_eof = 1;
558  }
559  if (ff_outlink_get_status(ctx->inputs[1])) {
561  return 0;
562  }
563  if (!s->cf0_eof)
565  else
567  return 0;
568  }
569 
570  return ret;
571 }
572 
573 static int acrossfade_config_output(AVFilterLink *outlink)
574 {
575  AVFilterContext *ctx = outlink->src;
576  AudioFadeContext *s = ctx->priv;
577 
578  if (ctx->inputs[0]->sample_rate != ctx->inputs[1]->sample_rate) {
579  av_log(ctx, AV_LOG_ERROR,
580  "Inputs must have the same sample rate "
581  "%d for in0 vs %d for in1\n",
582  ctx->inputs[0]->sample_rate, ctx->inputs[1]->sample_rate);
583  return AVERROR(EINVAL);
584  }
585 
586  outlink->sample_rate = ctx->inputs[0]->sample_rate;
587  outlink->time_base = ctx->inputs[0]->time_base;
588  outlink->channel_layout = ctx->inputs[0]->channel_layout;
589  outlink->channels = ctx->inputs[0]->channels;
590 
591  switch (outlink->format) {
592  case AV_SAMPLE_FMT_DBL: s->crossfade_samples = crossfade_samples_dbl; break;
593  case AV_SAMPLE_FMT_DBLP: s->crossfade_samples = crossfade_samples_dblp; break;
594  case AV_SAMPLE_FMT_FLT: s->crossfade_samples = crossfade_samples_flt; break;
595  case AV_SAMPLE_FMT_FLTP: s->crossfade_samples = crossfade_samples_fltp; break;
596  case AV_SAMPLE_FMT_S16: s->crossfade_samples = crossfade_samples_s16; break;
597  case AV_SAMPLE_FMT_S16P: s->crossfade_samples = crossfade_samples_s16p; break;
598  case AV_SAMPLE_FMT_S32: s->crossfade_samples = crossfade_samples_s32; break;
599  case AV_SAMPLE_FMT_S32P: s->crossfade_samples = crossfade_samples_s32p; break;
600  }
601 
602  config_output(outlink);
603 
604  return 0;
605 }
606 
607 static const AVFilterPad avfilter_af_acrossfade_inputs[] = {
608  {
609  .name = "crossfade0",
610  .type = AVMEDIA_TYPE_AUDIO,
611  },
612  {
613  .name = "crossfade1",
614  .type = AVMEDIA_TYPE_AUDIO,
615  },
616  { NULL }
617 };
618 
619 static const AVFilterPad avfilter_af_acrossfade_outputs[] = {
620  {
621  .name = "default",
622  .type = AVMEDIA_TYPE_AUDIO,
623  .config_props = acrossfade_config_output,
624  },
625  { NULL }
626 };
627 
629  .name = "acrossfade",
630  .description = NULL_IF_CONFIG_SMALL("Cross fade two input audio streams."),
631  .query_formats = query_formats,
632  .priv_size = sizeof(AudioFadeContext),
633  .activate = activate,
634  .priv_class = &acrossfade_class,
635  .inputs = avfilter_af_acrossfade_inputs,
636  .outputs = avfilter_af_acrossfade_outputs,
637 };
638 
639 #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:1481
#define NULL
Definition: coverity.c:32
int ff_set_common_channel_layouts(AVFilterContext *ctx, AVFilterChannelLayouts *layouts)
A helper for query_formats() which sets all links to the same list of channel layouts/sample rates...
Definition: formats.c:549
#define CUBE(a)
This structure describes decoded (raw) audio or video data.
Definition: frame.h:295
AVOption.
Definition: opt.h:246
#define C
#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:35
channels
Definition: aptx.c:30
static int config_output(AVFilterLink *outlink)
Definition: af_afade.c:208
double, planar
Definition: samplefmt.h:70
Definition: af_afade.c:54
#define src
Definition: vp8dsp.c:254
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:1607
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:283
#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
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:1080
uint8_t
#define av_cold
Definition: attributes.h:82
AVOptions.
static int query_formats(AVFilterContext *ctx)
Definition: af_afade.c:59
#define FADE_PLANAR(name, type)
Definition: af_afade.c:164
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:388
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
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
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
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
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
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
#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:202
#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:186
void * priv
private data for use by the filter
Definition: avfilter.h:353
#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:601
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
Definition: af_afade.c:54
int64_t start_sample
Definition: af_afade.c:37
int ff_inlink_queued_samples(AVFilterLink *link)
Definition: avfilter.c:1461
Definition: af_afade.c:54
int32_t
AVFormatContext * ctx
Definition: movenc.c:48
Definition: af_afade.c:54
static int activate(AVFilterContext *ctx)
Definition: af_adeclick.c:609
#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
static const AVFilterPad inputs[]
Definition: af_acontrast.c:193
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:85
int format
format of the frame, -1 if unknown or unset Values correspond to enum AVPixelFormat for video frames...
Definition: frame.h:368
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
typedef void(RENAME(mix_any_func_type))
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
Definition: frame.c:594
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:1500
void * buf
Definition: avisynth_c.h:766
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)\", 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
int ff_outlink_get_status(AVFilterLink *link)
Get the status on an output link.
Definition: avfilter.c:1630
int index
Definition: gxfenc.c:89
Rational number (pair of numerator and denominator).
Definition: rational.h:58
Definition: af_afade.c:54
#define FLAGS
Definition: af_afade.c:57
const char * name
Filter name.
Definition: avfilter.h:148
Definition: af_afade.c:54
Definition: af_afade.c:54
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:395
#define flags(name, subs,...)
Definition: cbs_av1.c:561
static int filter_frame(DBEContext *s, AVFrame *frame)
Definition: dolby_e.c:565
Definition: af_afade.c:54
Definition: af_afade.c:54
Definition: af_afade.c:54
signed 16 bits
Definition: samplefmt.h:61
static double fade_gain(int curve, int64_t index, int64_t range)
Definition: af_afade.c:92
#define OFFSET(x)
Definition: af_afade.c:56
Definition: af_afade.c:54
#define AVFILTER_DEFINE_CLASS(fname)
Definition: internal.h:334
A list of supported formats for one end of a filter link.
Definition: formats.h:64
int64_t nb_samples
Definition: af_afade.c:36
An instance of a filter.
Definition: avfilter.h:338
AVFilter ff_af_afade
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:701
FILE * out
Definition: movenc.c:54
void INT64 start
Definition: avisynth_c.h:766
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
internal API functions
AVFilterChannelLayouts * ff_all_channel_counts(void)
Construct an AVFilterChannelLayouts coding for any channel layout, with known or unknown disposition...
Definition: formats.c:410
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:342
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:361
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:556
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
Definition: frame.c:654
#define FADE(name, type)
Definition: af_afade.c:182
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248
Definition: af_afade.c:54