1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
397
398
399
400
401
402
403
404
405
406
407
408
409
410
411
412
413
414
415
416
417
418
419
420
421
422
423
424
425
426
427
428
429
430
431
432
433
434
435
436
437
438
439
440
441
442
443
444
445
446
447
448
449
450
451
452
453
454
455
456
457
458
459
460
461
462
463
464
465
466
467
468
469
470
471
472
473
474
475
476
477
478
479
480
481
482
483
484
485
486
487
488
489
490
491
492
493
494
495
496
497
498
499
500
501
502
503
504
505
506
507
508
509
510
511
512
513
514
515
516
517
518
519
520
521
522
523
524
525
526
527
528
529
530
531
532
533
534
535
536
537
538
539
540
541
542
543
544
545
546
547
548
549
550
551
552
553
554
555
556
557
558
559
560
561
562
563
564
565
566
567
568
569
570
571
572
573
574
575
576
577
578
579
580
581
582
583
584
585
586
587
588
589
590
591
592
593
594
595
596
597
598
599
600
601
602
603
604
605
606
607
608
609
610
611
612
613
614
615
616
617
618
619
620
621
622
623
624
625
626
627
628
629
630
631
632
633
634
635
636
637
638
639
640
641
642
643
644
645
646
647
648
649
650
651
652
653
654
655
656
657
658
659
660
661
662
663
664
665
666
667
668
669
670
671
672
673
674
675
676
677
678
679
680
681
682
683
684
685
686
687
688
689
690
691
692
693
694
695
696
697
698
699
700
701
702
703
704
705
706
707
708
709
710
711
712
713
714
715
716
717
718
719
720
721
722
723
724
725
726
727
728
729
730
731
732
733
734
735
736
737
738
739
740
741
742
743
744
745
746
747
748
749
750
751
752
753
754
755
756
757
758
759
760
761
762
763
764
765
766
767
768
769
770
771
772
773
774
775
776
777
778
779
780
781
782
783
784
785
786
787
788
789
790
791
792
793
794
795
796
797
798
799
800
801
802
803
804
805
806
807
808
809
810
811
812
813
814
815
816
817
818
819
820
821
822
823
824
825
|
/*
* Copyright (c) 2013-2015 Paul B Mahol
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* fade audio filter
*/
#include "config_components.h"
#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/opt.h"
#include "audio.h"
#include "avfilter.h"
#include "filters.h"
typedef struct AudioFadeContext {
const AVClass *class;
int nb_inputs;
int type;
int curve, curve2;
int64_t nb_samples;
int64_t start_sample;
int64_t duration;
int64_t start_time;
double silence;
double unity;
int overlap;
int64_t pts;
int xfade_idx;
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,
double silence, double unity);
void (*scale_samples)(uint8_t **dst, uint8_t * const *src,
int nb_samples, int channels, double unity);
void (*crossfade_samples)(uint8_t **dst, uint8_t * const *cf0,
uint8_t * const *cf1,
int nb_samples, int channels,
int curve0, int curve1);
} AudioFadeContext;
enum CurveType { NONE = -1, TRI, QSIN, ESIN, HSIN, LOG, IPAR, QUA, CUB, SQU, CBR, PAR, EXP, IQSIN, IHSIN, DESE, DESI, LOSI, SINC, ISINC, QUAT, QUATR, QSIN2, HSIN2, NB_CURVES };
#define OFFSET(x) offsetof(AudioFadeContext, x)
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
#define TFLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32P,
AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
AV_SAMPLE_FMT_NONE
};
static double fade_gain(int curve, int64_t index, int64_t range, double silence, double unity)
{
#define CUBE(a) ((a)*(a)*(a))
double gain;
gain = av_clipd(1.0 * index / range, 0, 1.0);
switch (curve) {
case QSIN:
gain = sin(gain * M_PI / 2.0);
break;
case IQSIN:
/* 0.6... = 2 / M_PI */
gain = 0.6366197723675814 * asin(gain);
break;
case ESIN:
gain = 1.0 - cos(M_PI / 4.0 * (CUBE(2.0*gain - 1) + 1));
break;
case HSIN:
gain = (1.0 - cos(gain * M_PI)) / 2.0;
break;
case IHSIN:
/* 0.3... = 1 / M_PI */
gain = 0.3183098861837907 * acos(1 - 2 * gain);
break;
case EXP:
/* -11.5... = 5*ln(0.1) */
gain = exp(-11.512925464970227 * (1 - gain));
break;
case LOG:
gain = av_clipd(1 + 0.2 * log10(gain), 0, 1.0);
break;
case PAR:
gain = 1 - sqrt(1 - gain);
break;
case IPAR:
gain = (1 - (1 - gain) * (1 - gain));
break;
case QUA:
gain *= gain;
break;
case CUB:
gain = CUBE(gain);
break;
case SQU:
gain = sqrt(gain);
break;
case CBR:
gain = cbrt(gain);
break;
case DESE:
gain = gain <= 0.5 ? cbrt(2 * gain) / 2: 1 - cbrt(2 * (1 - gain)) / 2;
break;
case DESI:
gain = gain <= 0.5 ? CUBE(2 * gain) / 2: 1 - CUBE(2 * (1 - gain)) / 2;
break;
case LOSI: {
const double a = 1. / (1. - 0.787) - 1;
double A = 1. / (1.0 + exp(0 -((gain-0.5) * a * 2.0)));
double B = 1. / (1.0 + exp(a));
double C = 1. / (1.0 + exp(0-a));
gain = (A - B) / (C - B);
}
break;
case SINC:
gain = gain >= 1.0 ? 1.0 : sin(M_PI * (1.0 - gain)) / (M_PI * (1.0 - gain));
break;
case ISINC:
gain = gain <= 0.0 ? 0.0 : 1.0 - sin(M_PI * gain) / (M_PI * gain);
break;
case QUAT:
gain = gain * gain * gain * gain;
break;
case QUATR:
gain = pow(gain, 0.25);
break;
case QSIN2:
gain = sin(gain * M_PI / 2.0) * sin(gain * M_PI / 2.0);
break;
case HSIN2:
gain = pow((1.0 - cos(gain * M_PI)) / 2.0, 2.0);
break;
case NONE:
gain = 1.0;
break;
}
return silence + (unity - silence) * gain;
}
#define FADE_PLANAR(name, type) \
static void fade_samples_## name ##p(uint8_t **dst, uint8_t * const *src, \
int nb_samples, int channels, int dir, \
int64_t start, int64_t range,int curve,\
double silence, double unity) \
{ \
int i, c; \
\
for (i = 0; i < nb_samples; i++) { \
double gain = fade_gain(curve, start + i * dir,range,silence,unity);\
for (c = 0; c < channels; c++) { \
type *d = (type *)dst[c]; \
const type *s = (type *)src[c]; \
\
d[i] = s[i] * gain; \
} \
} \
}
#define FADE(name, type) \
static void fade_samples_## name (uint8_t **dst, uint8_t * const *src, \
int nb_samples, int channels, int dir, \
int64_t start, int64_t range, int curve, \
double silence, double unity) \
{ \
type *d = (type *)dst[0]; \
const type *s = (type *)src[0]; \
int i, c, k = 0; \
\
for (i = 0; i < nb_samples; i++) { \
double gain = fade_gain(curve, start + i * dir,range,silence,unity);\
for (c = 0; c < channels; c++, k++) \
d[k] = s[k] * gain; \
} \
}
FADE_PLANAR(dbl, double)
FADE_PLANAR(flt, float)
FADE_PLANAR(s16, int16_t)
FADE_PLANAR(s32, int32_t)
FADE(dbl, double)
FADE(flt, float)
FADE(s16, int16_t)
FADE(s32, int32_t)
#define SCALE_PLANAR(name, type) \
static void scale_samples_## name ##p(uint8_t **dst, uint8_t * const *src, \
int nb_samples, int channels, \
double gain) \
{ \
int i, c; \
\
for (i = 0; i < nb_samples; i++) { \
for (c = 0; c < channels; c++) { \
type *d = (type *)dst[c]; \
const type *s = (type *)src[c]; \
\
d[i] = s[i] * gain; \
} \
} \
}
#define SCALE(name, type) \
static void scale_samples_## name (uint8_t **dst, uint8_t * const *src, \
int nb_samples, int channels, double gain)\
{ \
type *d = (type *)dst[0]; \
const type *s = (type *)src[0]; \
int i, c, k = 0; \
\
for (i = 0; i < nb_samples; i++) { \
for (c = 0; c < channels; c++, k++) \
d[k] = s[k] * gain; \
} \
}
SCALE_PLANAR(dbl, double)
SCALE_PLANAR(flt, float)
SCALE_PLANAR(s16, int16_t)
SCALE_PLANAR(s32, int32_t)
SCALE(dbl, double)
SCALE(flt, float)
SCALE(s16, int16_t)
SCALE(s32, int32_t)
static int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AudioFadeContext *s = ctx->priv;
switch (outlink->format) {
case AV_SAMPLE_FMT_DBL: s->fade_samples = fade_samples_dbl;
s->scale_samples = scale_samples_dbl;
break;
case AV_SAMPLE_FMT_DBLP: s->fade_samples = fade_samples_dblp;
s->scale_samples = scale_samples_dblp;
break;
case AV_SAMPLE_FMT_FLT: s->fade_samples = fade_samples_flt;
s->scale_samples = scale_samples_flt;
break;
case AV_SAMPLE_FMT_FLTP: s->fade_samples = fade_samples_fltp;
s->scale_samples = scale_samples_fltp;
break;
case AV_SAMPLE_FMT_S16: s->fade_samples = fade_samples_s16;
s->scale_samples = scale_samples_s16;
break;
case AV_SAMPLE_FMT_S16P: s->fade_samples = fade_samples_s16p;
s->scale_samples = scale_samples_s16p;
break;
case AV_SAMPLE_FMT_S32: s->fade_samples = fade_samples_s32;
s->scale_samples = scale_samples_s32;
break;
case AV_SAMPLE_FMT_S32P: s->fade_samples = fade_samples_s32p;
s->scale_samples = scale_samples_s32p;
break;
}
if (s->duration)
s->nb_samples = av_rescale(s->duration, outlink->sample_rate, AV_TIME_BASE);
s->duration = 0;
if (s->start_time)
s->start_sample = av_rescale(s->start_time, outlink->sample_rate, AV_TIME_BASE);
s->start_time = 0;
return 0;
}
#if CONFIG_AFADE_FILTER
static const AVOption afade_options[] = {
{ "type", "set the fade direction", OFFSET(type), AV_OPT_TYPE_INT, {.i64 = 0 }, 0, 1, TFLAGS, .unit = "type" },
{ "t", "set the fade direction", OFFSET(type), AV_OPT_TYPE_INT, {.i64 = 0 }, 0, 1, TFLAGS, .unit = "type" },
{ "in", "fade-in", 0, AV_OPT_TYPE_CONST, {.i64 = 0 }, 0, 0, TFLAGS, .unit = "type" },
{ "out", "fade-out", 0, AV_OPT_TYPE_CONST, {.i64 = 1 }, 0, 0, TFLAGS, .unit = "type" },
{ "start_sample", "set number of first sample to start fading", OFFSET(start_sample), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT64_MAX, TFLAGS },
{ "ss", "set number of first sample to start fading", OFFSET(start_sample), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT64_MAX, TFLAGS },
{ "nb_samples", "set number of samples for fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT64, {.i64 = 44100}, 1, INT64_MAX, TFLAGS },
{ "ns", "set number of samples for fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT64, {.i64 = 44100}, 1, INT64_MAX, TFLAGS },
{ "start_time", "set time to start fading", OFFSET(start_time), AV_OPT_TYPE_DURATION, {.i64 = 0 }, 0, INT64_MAX, TFLAGS },
{ "st", "set time to start fading", OFFSET(start_time), AV_OPT_TYPE_DURATION, {.i64 = 0 }, 0, INT64_MAX, TFLAGS },
{ "duration", "set fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0 }, 0, INT64_MAX, TFLAGS },
{ "d", "set fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0 }, 0, INT64_MAX, TFLAGS },
{ "curve", "set fade curve type", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, NONE, NB_CURVES - 1, TFLAGS, .unit = "curve" },
{ "c", "set fade curve type", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, NONE, NB_CURVES - 1, TFLAGS, .unit = "curve" },
{ "nofade", "no fade; keep audio as-is", 0, AV_OPT_TYPE_CONST, {.i64 = NONE }, 0, 0, TFLAGS, .unit = "curve" },
{ "tri", "linear slope", 0, AV_OPT_TYPE_CONST, {.i64 = TRI }, 0, 0, TFLAGS, .unit = "curve" },
{ "qsin", "quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = QSIN }, 0, 0, TFLAGS, .unit = "curve" },
{ "esin", "exponential sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = ESIN }, 0, 0, TFLAGS, .unit = "curve" },
{ "hsin", "half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = HSIN }, 0, 0, TFLAGS, .unit = "curve" },
{ "log", "logarithmic", 0, AV_OPT_TYPE_CONST, {.i64 = LOG }, 0, 0, TFLAGS, .unit = "curve" },
{ "ipar", "inverted parabola", 0, AV_OPT_TYPE_CONST, {.i64 = IPAR }, 0, 0, TFLAGS, .unit = "curve" },
{ "qua", "quadratic", 0, AV_OPT_TYPE_CONST, {.i64 = QUA }, 0, 0, TFLAGS, .unit = "curve" },
{ "cub", "cubic", 0, AV_OPT_TYPE_CONST, {.i64 = CUB }, 0, 0, TFLAGS, .unit = "curve" },
{ "squ", "square root", 0, AV_OPT_TYPE_CONST, {.i64 = SQU }, 0, 0, TFLAGS, .unit = "curve" },
{ "cbr", "cubic root", 0, AV_OPT_TYPE_CONST, {.i64 = CBR }, 0, 0, TFLAGS, .unit = "curve" },
{ "par", "parabola", 0, AV_OPT_TYPE_CONST, {.i64 = PAR }, 0, 0, TFLAGS, .unit = "curve" },
{ "exp", "exponential", 0, AV_OPT_TYPE_CONST, {.i64 = EXP }, 0, 0, TFLAGS, .unit = "curve" },
{ "iqsin", "inverted quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IQSIN}, 0, 0, TFLAGS, .unit = "curve" },
{ "ihsin", "inverted half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IHSIN}, 0, 0, TFLAGS, .unit = "curve" },
{ "dese", "double-exponential seat", 0, AV_OPT_TYPE_CONST, {.i64 = DESE }, 0, 0, TFLAGS, .unit = "curve" },
{ "desi", "double-exponential sigmoid", 0, AV_OPT_TYPE_CONST, {.i64 = DESI }, 0, 0, TFLAGS, .unit = "curve" },
{ "losi", "logistic sigmoid", 0, AV_OPT_TYPE_CONST, {.i64 = LOSI }, 0, 0, TFLAGS, .unit = "curve" },
{ "sinc", "sine cardinal function", 0, AV_OPT_TYPE_CONST, {.i64 = SINC }, 0, 0, TFLAGS, .unit = "curve" },
{ "isinc", "inverted sine cardinal function", 0, AV_OPT_TYPE_CONST, {.i64 = ISINC}, 0, 0, TFLAGS, .unit = "curve" },
{ "quat", "quartic", 0, AV_OPT_TYPE_CONST, {.i64 = QUAT }, 0, 0, TFLAGS, .unit = "curve" },
{ "quatr", "quartic root", 0, AV_OPT_TYPE_CONST, {.i64 = QUATR}, 0, 0, TFLAGS, .unit = "curve" },
{ "qsin2", "squared quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = QSIN2}, 0, 0, TFLAGS, .unit = "curve" },
{ "hsin2", "squared half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = HSIN2}, 0, 0, TFLAGS, .unit = "curve" },
{ "silence", "set the silence gain", OFFSET(silence), AV_OPT_TYPE_DOUBLE, {.dbl = 0 }, 0, 1, TFLAGS },
{ "unity", "set the unity gain", OFFSET(unity), AV_OPT_TYPE_DOUBLE, {.dbl = 1 }, 0, 1, TFLAGS },
{ NULL }
};
AVFILTER_DEFINE_CLASS(afade);
static av_cold int init(AVFilterContext *ctx)
{
AudioFadeContext *s = ctx->priv;
if (INT64_MAX - s->nb_samples < s->start_sample)
return AVERROR(EINVAL);
return 0;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
{
AudioFadeContext *s = inlink->dst->priv;
AVFilterLink *outlink = inlink->dst->outputs[0];
int nb_samples = buf->nb_samples;
AVFrame *out_buf;
int64_t cur_sample = av_rescale_q(buf->pts, inlink->time_base, (AVRational){1, inlink->sample_rate});
if (s->unity == 1.0 &&
((!s->type && (s->start_sample + s->nb_samples < cur_sample)) ||
( s->type && (cur_sample + nb_samples < s->start_sample))))
return ff_filter_frame(outlink, buf);
if (av_frame_is_writable(buf)) {
out_buf = buf;
} else {
out_buf = ff_get_audio_buffer(outlink, nb_samples);
if (!out_buf)
return AVERROR(ENOMEM);
av_frame_copy_props(out_buf, buf);
}
if ((!s->type && (cur_sample + nb_samples < s->start_sample)) ||
( s->type && (s->start_sample + s->nb_samples < cur_sample))) {
if (s->silence == 0.) {
av_samples_set_silence(out_buf->extended_data, 0, nb_samples,
out_buf->ch_layout.nb_channels, out_buf->format);
} else {
s->scale_samples(out_buf->extended_data, buf->extended_data,
nb_samples, buf->ch_layout.nb_channels,
s->silence);
}
} else if (( s->type && (cur_sample + nb_samples < s->start_sample)) ||
(!s->type && (s->start_sample + s->nb_samples < cur_sample))) {
s->scale_samples(out_buf->extended_data, buf->extended_data,
nb_samples, buf->ch_layout.nb_channels,
s->unity);
} else {
int64_t start;
if (!s->type)
start = cur_sample - s->start_sample;
else
start = s->start_sample + s->nb_samples - cur_sample;
s->fade_samples(out_buf->extended_data, buf->extended_data,
nb_samples, buf->ch_layout.nb_channels,
s->type ? -1 : 1, start,
s->nb_samples, s->curve, s->silence, s->unity);
}
if (buf != out_buf)
av_frame_free(&buf);
return ff_filter_frame(outlink, out_buf);
}
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
char *res, int res_len, int flags)
{
int ret;
ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
if (ret < 0)
return ret;
return config_output(ctx->outputs[0]);
}
static const AVFilterPad avfilter_af_afade_inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
},
};
static const AVFilterPad avfilter_af_afade_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
},
};
const FFFilter ff_af_afade = {
.p.name = "afade",
.p.description = NULL_IF_CONFIG_SMALL("Fade in/out input audio."),
.p.priv_class = &afade_class,
.p.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC,
.priv_size = sizeof(AudioFadeContext),
.init = init,
FILTER_INPUTS(avfilter_af_afade_inputs),
FILTER_OUTPUTS(avfilter_af_afade_outputs),
FILTER_SAMPLEFMTS_ARRAY(sample_fmts),
.process_command = process_command,
};
#endif /* CONFIG_AFADE_FILTER */
#if CONFIG_ACROSSFADE_FILTER
static const AVOption acrossfade_options[] = {
{ "inputs", "set number of input files to cross fade", OFFSET(nb_inputs), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT32_MAX, FLAGS },
{ "n", "set number of input files to cross fade", OFFSET(nb_inputs), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT32_MAX, FLAGS },
{ "nb_samples", "set number of samples for cross fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT64, {.i64 = 44100}, 1, INT32_MAX/10, FLAGS },
{ "ns", "set number of samples for cross fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT64, {.i64 = 44100}, 1, INT32_MAX/10, FLAGS },
{ "duration", "set cross fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0 }, 0, 60000000, FLAGS },
{ "d", "set cross fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0 }, 0, 60000000, FLAGS },
{ "overlap", "overlap 1st stream end with 2nd stream start", OFFSET(overlap), AV_OPT_TYPE_BOOL, {.i64 = 1 }, 0, 1, FLAGS },
{ "o", "overlap 1st stream end with 2nd stream start", OFFSET(overlap), AV_OPT_TYPE_BOOL, {.i64 = 1 }, 0, 1, FLAGS },
{ "curve1", "set fade curve type for 1st stream", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, NONE, NB_CURVES - 1, FLAGS, .unit = "curve" },
{ "c1", "set fade curve type for 1st stream", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, NONE, NB_CURVES - 1, FLAGS, .unit = "curve" },
{ "nofade", "no fade; keep audio as-is", 0, AV_OPT_TYPE_CONST, {.i64 = NONE }, 0, 0, FLAGS, .unit = "curve" },
{ "tri", "linear slope", 0, AV_OPT_TYPE_CONST, {.i64 = TRI }, 0, 0, FLAGS, .unit = "curve" },
{ "qsin", "quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = QSIN }, 0, 0, FLAGS, .unit = "curve" },
{ "esin", "exponential sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = ESIN }, 0, 0, FLAGS, .unit = "curve" },
{ "hsin", "half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = HSIN }, 0, 0, FLAGS, .unit = "curve" },
{ "log", "logarithmic", 0, AV_OPT_TYPE_CONST, {.i64 = LOG }, 0, 0, FLAGS, .unit = "curve" },
{ "ipar", "inverted parabola", 0, AV_OPT_TYPE_CONST, {.i64 = IPAR }, 0, 0, FLAGS, .unit = "curve" },
{ "qua", "quadratic", 0, AV_OPT_TYPE_CONST, {.i64 = QUA }, 0, 0, FLAGS, .unit = "curve" },
{ "cub", "cubic", 0, AV_OPT_TYPE_CONST, {.i64 = CUB }, 0, 0, FLAGS, .unit = "curve" },
{ "squ", "square root", 0, AV_OPT_TYPE_CONST, {.i64 = SQU }, 0, 0, FLAGS, .unit = "curve" },
{ "cbr", "cubic root", 0, AV_OPT_TYPE_CONST, {.i64 = CBR }, 0, 0, FLAGS, .unit = "curve" },
{ "par", "parabola", 0, AV_OPT_TYPE_CONST, {.i64 = PAR }, 0, 0, FLAGS, .unit = "curve" },
{ "exp", "exponential", 0, AV_OPT_TYPE_CONST, {.i64 = EXP }, 0, 0, FLAGS, .unit = "curve" },
{ "iqsin", "inverted quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IQSIN}, 0, 0, FLAGS, .unit = "curve" },
{ "ihsin", "inverted half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IHSIN}, 0, 0, FLAGS, .unit = "curve" },
{ "dese", "double-exponential seat", 0, AV_OPT_TYPE_CONST, {.i64 = DESE }, 0, 0, FLAGS, .unit = "curve" },
{ "desi", "double-exponential sigmoid", 0, AV_OPT_TYPE_CONST, {.i64 = DESI }, 0, 0, FLAGS, .unit = "curve" },
{ "losi", "logistic sigmoid", 0, AV_OPT_TYPE_CONST, {.i64 = LOSI }, 0, 0, FLAGS, .unit = "curve" },
{ "sinc", "sine cardinal function", 0, AV_OPT_TYPE_CONST, {.i64 = SINC }, 0, 0, FLAGS, .unit = "curve" },
{ "isinc", "inverted sine cardinal function", 0, AV_OPT_TYPE_CONST, {.i64 = ISINC}, 0, 0, FLAGS, .unit = "curve" },
{ "quat", "quartic", 0, AV_OPT_TYPE_CONST, {.i64 = QUAT }, 0, 0, FLAGS, .unit = "curve" },
{ "quatr", "quartic root", 0, AV_OPT_TYPE_CONST, {.i64 = QUATR}, 0, 0, FLAGS, .unit = "curve" },
{ "qsin2", "squared quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = QSIN2}, 0, 0, FLAGS, .unit = "curve" },
{ "hsin2", "squared half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = HSIN2}, 0, 0, FLAGS, .unit = "curve" },
{ "curve2", "set fade curve type for 2nd stream", OFFSET(curve2), AV_OPT_TYPE_INT, {.i64 = TRI }, NONE, NB_CURVES - 1, FLAGS, .unit = "curve" },
{ "c2", "set fade curve type for 2nd stream", OFFSET(curve2), AV_OPT_TYPE_INT, {.i64 = TRI }, NONE, NB_CURVES - 1, FLAGS, .unit = "curve" },
{ NULL }
};
AVFILTER_DEFINE_CLASS(acrossfade);
#define CROSSFADE_PLANAR(name, type) \
static void crossfade_samples_## name ##p(uint8_t **dst, uint8_t * const *cf0, \
uint8_t * const *cf1, \
int nb_samples, int channels, \
int curve0, int curve1) \
{ \
int i, c; \
\
for (i = 0; i < nb_samples; i++) { \
double gain0 = fade_gain(curve0, nb_samples - 1 - i, nb_samples,0.,1.);\
double gain1 = fade_gain(curve1, i, nb_samples, 0., 1.); \
for (c = 0; c < channels; c++) { \
type *d = (type *)dst[c]; \
const type *s0 = (type *)cf0[c]; \
const type *s1 = (type *)cf1[c]; \
\
d[i] = s0[i] * gain0 + s1[i] * gain1; \
} \
} \
}
#define CROSSFADE(name, type) \
static void crossfade_samples_## name (uint8_t **dst, uint8_t * const *cf0, \
uint8_t * const *cf1, \
int nb_samples, int channels, \
int curve0, int curve1) \
{ \
type *d = (type *)dst[0]; \
const type *s0 = (type *)cf0[0]; \
const type *s1 = (type *)cf1[0]; \
int i, c, k = 0; \
\
for (i = 0; i < nb_samples; i++) { \
double gain0 = fade_gain(curve0, nb_samples - 1-i,nb_samples,0.,1.);\
double gain1 = fade_gain(curve1, i, nb_samples, 0., 1.); \
for (c = 0; c < channels; c++, k++) \
d[k] = s0[k] * gain0 + s1[k] * gain1; \
} \
}
CROSSFADE_PLANAR(dbl, double)
CROSSFADE_PLANAR(flt, float)
CROSSFADE_PLANAR(s16, int16_t)
CROSSFADE_PLANAR(s32, int32_t)
CROSSFADE(dbl, double)
CROSSFADE(flt, float)
CROSSFADE(s16, int16_t)
CROSSFADE(s32, int32_t)
static int pass_frame(AVFilterLink *inlink, AVFilterLink *outlink, int64_t *pts)
{
AVFrame *in;
int ret = ff_inlink_consume_frame(inlink, &in);
if (ret < 0)
return ret;
av_assert1(ret);
in->pts = *pts;
*pts += av_rescale_q(in->nb_samples,
(AVRational){ 1, outlink->sample_rate }, outlink->time_base);
return ff_filter_frame(outlink, in);
}
static int pass_samples(AVFilterLink *inlink, AVFilterLink *outlink, unsigned nb_samples, int64_t *pts)
{
AVFrame *in;
int ret = ff_inlink_consume_samples(inlink, nb_samples, nb_samples, &in);
if (ret < 0)
return ret;
av_assert1(ret);
in->pts = *pts;
*pts += av_rescale_q(in->nb_samples,
(AVRational){ 1, outlink->sample_rate }, outlink->time_base);
return ff_filter_frame(outlink, in);
}
static int pass_crossfade(AVFilterContext *ctx, const int idx0, const int idx1)
{
AudioFadeContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
AVFrame *out, *cf[2] = { NULL };
int ret;
AVFilterLink *in0 = ctx->inputs[idx0];
AVFilterLink *in1 = ctx->inputs[idx1];
int queued_samples0 = ff_inlink_queued_samples(in0);
int queued_samples1 = ff_inlink_queued_samples(in1);
/* Limit to the relevant region */
av_assert1(queued_samples0 <= s->nb_samples);
if (ff_outlink_get_status(in1) && idx1 < s->nb_inputs - 1)
queued_samples1 /= 2; /* reserve second half for next fade-out */
queued_samples1 = FFMIN(queued_samples1, s->nb_samples);
if (s->overlap) {
int nb_samples = FFMIN(queued_samples0, queued_samples1);
if (nb_samples < s->nb_samples) {
av_log(ctx, AV_LOG_WARNING, "Input %d duration (%d samples) "
"is shorter than crossfade duration (%"PRId64" samples), "
"crossfade will be shorter by %"PRId64" samples.\n",
queued_samples0 <= queued_samples1 ? idx0 : idx1,
nb_samples, s->nb_samples, s->nb_samples - nb_samples);
if (queued_samples0 > nb_samples) {
ret = pass_samples(in0, outlink, queued_samples0 - nb_samples, &s->pts);
if (ret < 0)
return ret;
}
if (!nb_samples)
return 0; /* either input was completely empty */
}
av_assert1(nb_samples > 0);
out = ff_get_audio_buffer(outlink, nb_samples);
if (!out)
return AVERROR(ENOMEM);
ret = ff_inlink_consume_samples(in0, nb_samples, nb_samples, &cf[0]);
if (ret < 0) {
av_frame_free(&out);
return ret;
}
ret = ff_inlink_consume_samples(in1, nb_samples, nb_samples, &cf[1]);
if (ret < 0) {
av_frame_free(&cf[0]);
av_frame_free(&out);
return ret;
}
s->crossfade_samples(out->extended_data, cf[0]->extended_data,
cf[1]->extended_data, nb_samples,
out->ch_layout.nb_channels, s->curve, s->curve2);
out->pts = s->pts;
s->pts += av_rescale_q(nb_samples,
(AVRational){ 1, outlink->sample_rate }, outlink->time_base);
av_frame_free(&cf[0]);
av_frame_free(&cf[1]);
return ff_filter_frame(outlink, out);
} else {
if (queued_samples0 < s->nb_samples) {
av_log(ctx, AV_LOG_WARNING, "Input %d duration (%d samples) "
"is shorter than crossfade duration (%"PRId64" samples), "
"fade-out will be shorter by %"PRId64" samples.\n",
idx0, queued_samples0, s->nb_samples,
s->nb_samples - queued_samples0);
if (!queued_samples0)
goto fade_in;
}
out = ff_get_audio_buffer(outlink, queued_samples0);
if (!out)
return AVERROR(ENOMEM);
ret = ff_inlink_consume_samples(in0, queued_samples0, queued_samples0, &cf[0]);
if (ret < 0) {
av_frame_free(&out);
return ret;
}
s->fade_samples(out->extended_data, cf[0]->extended_data, cf[0]->nb_samples,
outlink->ch_layout.nb_channels, -1, cf[0]->nb_samples - 1, cf[0]->nb_samples, s->curve, 0., 1.);
out->pts = s->pts;
s->pts += av_rescale_q(cf[0]->nb_samples,
(AVRational){ 1, outlink->sample_rate }, outlink->time_base);
av_frame_free(&cf[0]);
ret = ff_filter_frame(outlink, out);
if (ret < 0)
return ret;
fade_in:
if (queued_samples1 < s->nb_samples) {
av_log(ctx, AV_LOG_WARNING, "Input %d duration (%d samples) "
"is shorter than crossfade duration (%"PRId64" samples), "
"fade-in will be shorter by %"PRId64" samples.\n",
idx1, ff_inlink_queued_samples(in1), s->nb_samples,
s->nb_samples - queued_samples1);
if (!queued_samples1)
return 0;
}
out = ff_get_audio_buffer(outlink, queued_samples1);
if (!out)
return AVERROR(ENOMEM);
ret = ff_inlink_consume_samples(in1, queued_samples1, queued_samples1, &cf[1]);
if (ret < 0) {
av_frame_free(&out);
return ret;
}
s->fade_samples(out->extended_data, cf[1]->extended_data, cf[1]->nb_samples,
outlink->ch_layout.nb_channels, 1, 0, cf[1]->nb_samples, s->curve2, 0., 1.);
out->pts = s->pts;
s->pts += av_rescale_q(cf[1]->nb_samples,
(AVRational){ 1, outlink->sample_rate }, outlink->time_base);
av_frame_free(&cf[1]);
return ff_filter_frame(outlink, out);
}
}
static int activate(AVFilterContext *ctx)
{
AudioFadeContext *s = ctx->priv;
const int idx0 = s->xfade_idx;
const int idx1 = s->xfade_idx + 1;
AVFilterLink *outlink = ctx->outputs[0];
AVFilterLink *in0 = ctx->inputs[idx0];
FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, ctx);
if (idx0 == s->nb_inputs - 1) {
/* Last active input, read until EOF */
if (ff_inlink_queued_frames(in0))
return pass_frame(in0, outlink, &s->pts);
FF_FILTER_FORWARD_STATUS(in0, outlink);
FF_FILTER_FORWARD_WANTED(outlink, in0);
return FFERROR_NOT_READY;
}
AVFilterLink *in1 = ctx->inputs[idx1];
int queued_samples0 = ff_inlink_queued_samples(in0);
if (queued_samples0 > s->nb_samples) {
AVFrame *frame = ff_inlink_peek_frame(in0, 0);
if (queued_samples0 - s->nb_samples >= frame->nb_samples)
return pass_frame(in0, outlink, &s->pts);
}
/* Continue reading until EOF */
if (ff_outlink_get_status(in0)) {
if (queued_samples0 > s->nb_samples)
return pass_samples(in0, outlink, queued_samples0 - s->nb_samples, &s->pts);
} else {
FF_FILTER_FORWARD_WANTED(outlink, in0);
return FFERROR_NOT_READY;
}
/* At this point, in0 has reached EOF with no more samples remaining
* except those that we want to crossfade */
av_assert0(queued_samples0 <= s->nb_samples);
int queued_samples1 = ff_inlink_queued_samples(in1);
/* If this clip is sandwiched between two other clips, buffer at least
* twice the total crossfade duration to ensure that we won't reach EOF
* during the second fade (in which case we would shorten the fade) */
int needed_samples = s->nb_samples;
if (idx1 < s->nb_inputs - 1)
needed_samples *= 2;
if (queued_samples1 >= needed_samples || ff_outlink_get_status(in1)) {
/* The first filter may EOF before delivering any samples, in which
* case it's possible for pass_crossfade() to be a no-op. Just ensure
* the activate() function runs again after incrementing the index to
* ensure we correctly move on to the next input in that case. */
s->xfade_idx++;
ff_filter_set_ready(ctx, 10);
return pass_crossfade(ctx, idx0, idx1);
} else {
FF_FILTER_FORWARD_WANTED(outlink, in1);
return FFERROR_NOT_READY;
}
}
static av_cold int acrossfade_init(AVFilterContext *ctx)
{
AudioFadeContext *s = ctx->priv;
int ret;
for (int i = 0; i < s->nb_inputs; i++) {
AVFilterPad pad = {
.name = av_asprintf("crossfade%d", i),
.type = AVMEDIA_TYPE_AUDIO,
};
if (!pad.name)
return AVERROR(ENOMEM);
ret = ff_append_inpad_free_name(ctx, &pad);
if (ret < 0)
return ret;
}
return 0;
}
static int acrossfade_config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AudioFadeContext *s = ctx->priv;
outlink->time_base = ctx->inputs[0]->time_base;
switch (outlink->format) {
case AV_SAMPLE_FMT_DBL: s->crossfade_samples = crossfade_samples_dbl; break;
case AV_SAMPLE_FMT_DBLP: s->crossfade_samples = crossfade_samples_dblp; break;
case AV_SAMPLE_FMT_FLT: s->crossfade_samples = crossfade_samples_flt; break;
case AV_SAMPLE_FMT_FLTP: s->crossfade_samples = crossfade_samples_fltp; break;
case AV_SAMPLE_FMT_S16: s->crossfade_samples = crossfade_samples_s16; break;
case AV_SAMPLE_FMT_S16P: s->crossfade_samples = crossfade_samples_s16p; break;
case AV_SAMPLE_FMT_S32: s->crossfade_samples = crossfade_samples_s32; break;
case AV_SAMPLE_FMT_S32P: s->crossfade_samples = crossfade_samples_s32p; break;
}
config_output(outlink);
return 0;
}
static const AVFilterPad avfilter_af_acrossfade_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = acrossfade_config_output,
},
};
const FFFilter ff_af_acrossfade = {
.p.name = "acrossfade",
.p.description = NULL_IF_CONFIG_SMALL("Cross fade two input audio streams."),
.p.priv_class = &acrossfade_class,
.p.flags = AVFILTER_FLAG_DYNAMIC_INPUTS,
.priv_size = sizeof(AudioFadeContext),
.init = acrossfade_init,
.activate = activate,
FILTER_OUTPUTS(avfilter_af_acrossfade_outputs),
FILTER_SAMPLEFMTS_ARRAY(sample_fmts),
};
#endif /* CONFIG_ACROSSFADE_FILTER */
|
