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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#include "config_components.h"
27
28#include "libavutil/avassert.h"
29#include "libavutil/avstring.h"
30#include "libavutil/opt.h"
31#include "audio.h"
32#include "avfilter.h"
33#include "filters.h"
34
35typedef struct AudioFadeContext {
36 const AVClass *class;
37 int nb_inputs;
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 double silence;
45 double unity;
46 int overlap;
47 int64_t pts;
48 int xfade_idx;
49
50 void (*fade_samples)(uint8_t **dst, uint8_t * const *src,
51 int nb_samples, int channels, int direction,
52 int64_t start, int64_t range, int curve,
53 double silence, double unity);
54 void (*scale_samples)(uint8_t **dst, uint8_t * const *src,
55 int nb_samples, int channels, double unity);
56 void (*crossfade_samples)(uint8_t **dst, uint8_t * const *cf0,
57 uint8_t * const *cf1,
58 int nb_samples, int channels,
59 int curve0, int curve1);
60} AudioFadeContext;
61
62enum 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 };
63
64#define OFFSET(x) offsetof(AudioFadeContext, x)
65#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
66#define TFLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
67
75
76static double fade_gain(int curve, int64_t index, int64_t range, double silence, double unity)
77{
78#define CUBE(a) ((a)*(a)*(a))
79 double gain;
80
81 gain = av_clipd(1.0 * index / range, 0, 1.0);
82
83 switch (curve) {
84 case QSIN:
85 gain = sin(gain * M_PI / 2.0);
86 break;
87 case IQSIN:
88 /* 0.6... = 2 / M_PI */
89 gain = 0.6366197723675814 * asin(gain);
90 break;
91 case ESIN:
92 gain = 1.0 - cos(M_PI / 4.0 * (CUBE(2.0*gain - 1) + 1));
93 break;
94 case HSIN:
95 gain = (1.0 - cos(gain * M_PI)) / 2.0;
96 break;
97 case IHSIN:
98 /* 0.3... = 1 / M_PI */
99 gain = 0.3183098861837907 * acos(1 - 2 * gain);
100 break;
101 case EXP:
102 /* -11.5... = 5*ln(0.1) */
103 gain = exp(-11.512925464970227 * (1 - gain));
104 break;
105 case LOG:
106 gain = av_clipd(1 + 0.2 * log10(gain), 0, 1.0);
107 break;
108 case PAR:
109 gain = 1 - sqrt(1 - gain);
110 break;
111 case IPAR:
112 gain = (1 - (1 - gain) * (1 - gain));
113 break;
114 case QUA:
115 gain *= gain;
116 break;
117 case CUB:
118 gain = CUBE(gain);
119 break;
120 case SQU:
121 gain = sqrt(gain);
122 break;
123 case CBR:
124 gain = cbrt(gain);
125 break;
126 case DESE:
127 gain = gain <= 0.5 ? cbrt(2 * gain) / 2: 1 - cbrt(2 * (1 - gain)) / 2;
128 break;
129 case DESI:
130 gain = gain <= 0.5 ? CUBE(2 * gain) / 2: 1 - CUBE(2 * (1 - gain)) / 2;
131 break;
132 case LOSI: {
133 const double a = 1. / (1. - 0.787) - 1;
134 double A = 1. / (1.0 + exp(0 -((gain-0.5) * a * 2.0)));
135 double B = 1. / (1.0 + exp(a));
136 double C = 1. / (1.0 + exp(0-a));
137 gain = (A - B) / (C - B);
138 }
139 break;
140 case SINC:
141 gain = gain >= 1.0 ? 1.0 : sin(M_PI * (1.0 - gain)) / (M_PI * (1.0 - gain));
142 break;
143 case ISINC:
144 gain = gain <= 0.0 ? 0.0 : 1.0 - sin(M_PI * gain) / (M_PI * gain);
145 break;
146 case QUAT:
147 gain = gain * gain * gain * gain;
148 break;
149 case QUATR:
150 gain = pow(gain, 0.25);
151 break;
152 case QSIN2:
153 gain = sin(gain * M_PI / 2.0) * sin(gain * M_PI / 2.0);
154 break;
155 case HSIN2:
156 gain = pow((1.0 - cos(gain * M_PI)) / 2.0, 2.0);
157 break;
158 case NONE:
159 gain = 1.0;
160 break;
161 }
162
163 return silence + (unity - silence) * gain;
164}
165
166#define FADE_PLANAR(name, type) \
167static void fade_samples_## name ##p(uint8_t **dst, uint8_t * const *src, \
168 int nb_samples, int channels, int dir, \
169 int64_t start, int64_t range,int curve,\
170 double silence, double unity) \
171{ \
172 int i, c; \
173 \
174 for (i = 0; i < nb_samples; i++) { \
175 double gain = fade_gain(curve, start + i * dir,range,silence,unity);\
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) \
186static 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 double silence, double unity) \
190{ \
191 type *d = (type *)dst[0]; \
192 const type *s = (type *)src[0]; \
193 int i, c, k = 0; \
194 \
195 for (i = 0; i < nb_samples; i++) { \
196 double gain = fade_gain(curve, start + i * dir,range,silence,unity);\
197 for (c = 0; c < channels; c++, k++) \
198 d[k] = s[k] * gain; \
199 } \
200}
201
202FADE_PLANAR(dbl, double)
203FADE_PLANAR(flt, float)
204FADE_PLANAR(s16, int16_t)
205FADE_PLANAR(s32, int32_t)
206
207FADE(dbl, double)
208FADE(flt, float)
209FADE(s16, int16_t)
210FADE(s32, int32_t)
211
212#define SCALE_PLANAR(name, type) \
213static void scale_samples_## name ##p(uint8_t **dst, uint8_t * const *src, \
214 int nb_samples, int channels, \
215 double gain) \
216{ \
217 int i, c; \
218 \
219 for (i = 0; i < nb_samples; i++) { \
220 for (c = 0; c < channels; c++) { \
221 type *d = (type *)dst[c]; \
222 const type *s = (type *)src[c]; \
223 \
224 d[i] = s[i] * gain; \
225 } \
226 } \
227}
228
229#define SCALE(name, type) \
230static void scale_samples_## name (uint8_t **dst, uint8_t * const *src, \
231 int nb_samples, int channels, double gain)\
232{ \
233 type *d = (type *)dst[0]; \
234 const type *s = (type *)src[0]; \
235 int i, c, k = 0; \
236 \
237 for (i = 0; i < nb_samples; i++) { \
238 for (c = 0; c < channels; c++, k++) \
239 d[k] = s[k] * gain; \
240 } \
241}
242
243SCALE_PLANAR(dbl, double)
244SCALE_PLANAR(flt, float)
245SCALE_PLANAR(s16, int16_t)
246SCALE_PLANAR(s32, int32_t)
247
248SCALE(dbl, double)
249SCALE(flt, float)
250SCALE(s16, int16_t)
251SCALE(s32, int32_t)
252
253static int config_output(AVFilterLink *outlink)
254{
255 AVFilterContext *ctx = outlink->src;
256 AudioFadeContext *s = ctx->priv;
257
258 switch (outlink->format) {
259 case AV_SAMPLE_FMT_DBL: s->fade_samples = fade_samples_dbl;
260 s->scale_samples = scale_samples_dbl;
261 break;
262 case AV_SAMPLE_FMT_DBLP: s->fade_samples = fade_samples_dblp;
263 s->scale_samples = scale_samples_dblp;
264 break;
265 case AV_SAMPLE_FMT_FLT: s->fade_samples = fade_samples_flt;
266 s->scale_samples = scale_samples_flt;
267 break;
268 case AV_SAMPLE_FMT_FLTP: s->fade_samples = fade_samples_fltp;
269 s->scale_samples = scale_samples_fltp;
270 break;
271 case AV_SAMPLE_FMT_S16: s->fade_samples = fade_samples_s16;
272 s->scale_samples = scale_samples_s16;
273 break;
274 case AV_SAMPLE_FMT_S16P: s->fade_samples = fade_samples_s16p;
275 s->scale_samples = scale_samples_s16p;
276 break;
277 case AV_SAMPLE_FMT_S32: s->fade_samples = fade_samples_s32;
278 s->scale_samples = scale_samples_s32;
279 break;
280 case AV_SAMPLE_FMT_S32P: s->fade_samples = fade_samples_s32p;
281 s->scale_samples = scale_samples_s32p;
282 break;
283 }
284
285 if (s->duration)
286 s->nb_samples = av_rescale(s->duration, outlink->sample_rate, AV_TIME_BASE);
287 s->duration = 0;
288 if (s->start_time)
289 s->start_sample = av_rescale(s->start_time, outlink->sample_rate, AV_TIME_BASE);
290 s->start_time = 0;
291
292 return 0;
293}
294
295#if CONFIG_AFADE_FILTER
296
297static const AVOption afade_options[] = {
298 { "type", "set the fade direction", OFFSET(type), AV_OPT_TYPE_INT, {.i64 = 0 }, 0, 1, TFLAGS, .unit = "type" },
299 { "t", "set the fade direction", OFFSET(type), AV_OPT_TYPE_INT, {.i64 = 0 }, 0, 1, TFLAGS, .unit = "type" },
300 { "in", "fade-in", 0, AV_OPT_TYPE_CONST, {.i64 = 0 }, 0, 0, TFLAGS, .unit = "type" },
301 { "out", "fade-out", 0, AV_OPT_TYPE_CONST, {.i64 = 1 }, 0, 0, TFLAGS, .unit = "type" },
302 { "start_sample", "set number of first sample to start fading", OFFSET(start_sample), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT64_MAX, TFLAGS },
303 { "ss", "set number of first sample to start fading", OFFSET(start_sample), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT64_MAX, TFLAGS },
304 { "nb_samples", "set number of samples for fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT64, {.i64 = 44100}, 1, INT64_MAX, TFLAGS },
305 { "ns", "set number of samples for fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT64, {.i64 = 44100}, 1, INT64_MAX, TFLAGS },
306 { "start_time", "set time to start fading", OFFSET(start_time), AV_OPT_TYPE_DURATION, {.i64 = 0 }, 0, INT64_MAX, TFLAGS },
307 { "st", "set time to start fading", OFFSET(start_time), AV_OPT_TYPE_DURATION, {.i64 = 0 }, 0, INT64_MAX, TFLAGS },
308 { "duration", "set fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0 }, 0, INT64_MAX, TFLAGS },
309 { "d", "set fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0 }, 0, INT64_MAX, TFLAGS },
310 { "curve", "set fade curve type", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, NONE, NB_CURVES - 1, TFLAGS, .unit = "curve" },
311 { "c", "set fade curve type", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, NONE, NB_CURVES - 1, TFLAGS, .unit = "curve" },
312 { "nofade", "no fade; keep audio as-is", 0, AV_OPT_TYPE_CONST, {.i64 = NONE }, 0, 0, TFLAGS, .unit = "curve" },
313 { "tri", "linear slope", 0, AV_OPT_TYPE_CONST, {.i64 = TRI }, 0, 0, TFLAGS, .unit = "curve" },
314 { "qsin", "quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = QSIN }, 0, 0, TFLAGS, .unit = "curve" },
315 { "esin", "exponential sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = ESIN }, 0, 0, TFLAGS, .unit = "curve" },
316 { "hsin", "half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = HSIN }, 0, 0, TFLAGS, .unit = "curve" },
317 { "log", "logarithmic", 0, AV_OPT_TYPE_CONST, {.i64 = LOG }, 0, 0, TFLAGS, .unit = "curve" },
318 { "ipar", "inverted parabola", 0, AV_OPT_TYPE_CONST, {.i64 = IPAR }, 0, 0, TFLAGS, .unit = "curve" },
319 { "qua", "quadratic", 0, AV_OPT_TYPE_CONST, {.i64 = QUA }, 0, 0, TFLAGS, .unit = "curve" },
320 { "cub", "cubic", 0, AV_OPT_TYPE_CONST, {.i64 = CUB }, 0, 0, TFLAGS, .unit = "curve" },
321 { "squ", "square root", 0, AV_OPT_TYPE_CONST, {.i64 = SQU }, 0, 0, TFLAGS, .unit = "curve" },
322 { "cbr", "cubic root", 0, AV_OPT_TYPE_CONST, {.i64 = CBR }, 0, 0, TFLAGS, .unit = "curve" },
323 { "par", "parabola", 0, AV_OPT_TYPE_CONST, {.i64 = PAR }, 0, 0, TFLAGS, .unit = "curve" },
324 { "exp", "exponential", 0, AV_OPT_TYPE_CONST, {.i64 = EXP }, 0, 0, TFLAGS, .unit = "curve" },
325 { "iqsin", "inverted quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IQSIN}, 0, 0, TFLAGS, .unit = "curve" },
326 { "ihsin", "inverted half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IHSIN}, 0, 0, TFLAGS, .unit = "curve" },
327 { "dese", "double-exponential seat", 0, AV_OPT_TYPE_CONST, {.i64 = DESE }, 0, 0, TFLAGS, .unit = "curve" },
328 { "desi", "double-exponential sigmoid", 0, AV_OPT_TYPE_CONST, {.i64 = DESI }, 0, 0, TFLAGS, .unit = "curve" },
329 { "losi", "logistic sigmoid", 0, AV_OPT_TYPE_CONST, {.i64 = LOSI }, 0, 0, TFLAGS, .unit = "curve" },
330 { "sinc", "sine cardinal function", 0, AV_OPT_TYPE_CONST, {.i64 = SINC }, 0, 0, TFLAGS, .unit = "curve" },
331 { "isinc", "inverted sine cardinal function", 0, AV_OPT_TYPE_CONST, {.i64 = ISINC}, 0, 0, TFLAGS, .unit = "curve" },
332 { "quat", "quartic", 0, AV_OPT_TYPE_CONST, {.i64 = QUAT }, 0, 0, TFLAGS, .unit = "curve" },
333 { "quatr", "quartic root", 0, AV_OPT_TYPE_CONST, {.i64 = QUATR}, 0, 0, TFLAGS, .unit = "curve" },
334 { "qsin2", "squared quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = QSIN2}, 0, 0, TFLAGS, .unit = "curve" },
335 { "hsin2", "squared half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = HSIN2}, 0, 0, TFLAGS, .unit = "curve" },
336 { "silence", "set the silence gain", OFFSET(silence), AV_OPT_TYPE_DOUBLE, {.dbl = 0 }, 0, 1, TFLAGS },
337 { "unity", "set the unity gain", OFFSET(unity), AV_OPT_TYPE_DOUBLE, {.dbl = 1 }, 0, 1, TFLAGS },
338 { NULL }
339};
340
341AVFILTER_DEFINE_CLASS(afade);
342
343static av_cold int init(AVFilterContext *ctx)
344{
345 AudioFadeContext *s = ctx->priv;
346
347 if (INT64_MAX - s->nb_samples < s->start_sample)
348 return AVERROR(EINVAL);
349
350 return 0;
351}
352
353static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
354{
355 AudioFadeContext *s = inlink->dst->priv;
356 AVFilterLink *outlink = inlink->dst->outputs[0];
357 int nb_samples = buf->nb_samples;
358 AVFrame *out_buf;
359 int64_t cur_sample = av_rescale_q(buf->pts, inlink->time_base, (AVRational){1, inlink->sample_rate});
360
361 if (s->unity == 1.0 &&
362 ((!s->type && (s->start_sample + s->nb_samples < cur_sample)) ||
363 ( s->type && (cur_sample + nb_samples < s->start_sample))))
364 return ff_filter_frame(outlink, buf);
365
366 if (av_frame_is_writable(buf)) {
367 out_buf = buf;
368 } else {
369 out_buf = ff_get_audio_buffer(outlink, nb_samples);
370 if (!out_buf)
371 return AVERROR(ENOMEM);
372 av_frame_copy_props(out_buf, buf);
373 }
374
375 if ((!s->type && (cur_sample + nb_samples < s->start_sample)) ||
376 ( s->type && (s->start_sample + s->nb_samples < cur_sample))) {
377 if (s->silence == 0.) {
378 av_samples_set_silence(out_buf->extended_data, 0, nb_samples,
379 out_buf->ch_layout.nb_channels, out_buf->format);
380 } else {
381 s->scale_samples(out_buf->extended_data, buf->extended_data,
382 nb_samples, buf->ch_layout.nb_channels,
383 s->silence);
384 }
385 } else if (( s->type && (cur_sample + nb_samples < s->start_sample)) ||
386 (!s->type && (s->start_sample + s->nb_samples < cur_sample))) {
387 s->scale_samples(out_buf->extended_data, buf->extended_data,
388 nb_samples, buf->ch_layout.nb_channels,
389 s->unity);
390 } else {
391 int64_t start;
392
393 if (!s->type)
394 start = cur_sample - s->start_sample;
395 else
396 start = s->start_sample + s->nb_samples - cur_sample;
397
398 s->fade_samples(out_buf->extended_data, buf->extended_data,
399 nb_samples, buf->ch_layout.nb_channels,
400 s->type ? -1 : 1, start,
401 s->nb_samples, s->curve, s->silence, s->unity);
402 }
403
404 if (buf != out_buf)
405 av_frame_free(&buf);
406
407 return ff_filter_frame(outlink, out_buf);
408}
409
410static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
411 char *res, int res_len, int flags)
412{
413 int ret;
414
415 ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
416 if (ret < 0)
417 return ret;
418
419 return config_output(ctx->outputs[0]);
420}
421
422static const AVFilterPad avfilter_af_afade_inputs[] = {
423 {
424 .name = "default",
425 .type = AVMEDIA_TYPE_AUDIO,
426 .filter_frame = filter_frame,
427 },
428};
429
430static const AVFilterPad avfilter_af_afade_outputs[] = {
431 {
432 .name = "default",
433 .type = AVMEDIA_TYPE_AUDIO,
434 .config_props = config_output,
435 },
436};
437
438const FFFilter ff_af_afade = {
439 .p.name = "afade",
440 .p.description = NULL_IF_CONFIG_SMALL("Fade in/out input audio."),
441 .p.priv_class = &afade_class,
442 .p.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC,
443 .priv_size = sizeof(AudioFadeContext),
444 .init = init,
445 FILTER_INPUTS(avfilter_af_afade_inputs),
446 FILTER_OUTPUTS(avfilter_af_afade_outputs),
447 FILTER_SAMPLEFMTS_ARRAY(sample_fmts),
448 .process_command = process_command,
449};
450
451#endif /* CONFIG_AFADE_FILTER */
452
453#if CONFIG_ACROSSFADE_FILTER
454
455static const AVOption acrossfade_options[] = {
456 { "inputs", "set number of input files to cross fade", OFFSET(nb_inputs), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT32_MAX, FLAGS },
457 { "n", "set number of input files to cross fade", OFFSET(nb_inputs), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT32_MAX, FLAGS },
458 { "nb_samples", "set number of samples for cross fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT64, {.i64 = 44100}, 1, INT32_MAX/10, FLAGS },
459 { "ns", "set number of samples for cross fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT64, {.i64 = 44100}, 1, INT32_MAX/10, FLAGS },
460 { "duration", "set cross fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0 }, 0, 60000000, FLAGS },
461 { "d", "set cross fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0 }, 0, 60000000, FLAGS },
462 { "overlap", "overlap 1st stream end with 2nd stream start", OFFSET(overlap), AV_OPT_TYPE_BOOL, {.i64 = 1 }, 0, 1, FLAGS },
463 { "o", "overlap 1st stream end with 2nd stream start", OFFSET(overlap), AV_OPT_TYPE_BOOL, {.i64 = 1 }, 0, 1, FLAGS },
464 { "curve1", "set fade curve type for 1st stream", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, NONE, NB_CURVES - 1, FLAGS, .unit = "curve" },
465 { "c1", "set fade curve type for 1st stream", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, NONE, NB_CURVES - 1, FLAGS, .unit = "curve" },
466 { "nofade", "no fade; keep audio as-is", 0, AV_OPT_TYPE_CONST, {.i64 = NONE }, 0, 0, FLAGS, .unit = "curve" },
467 { "tri", "linear slope", 0, AV_OPT_TYPE_CONST, {.i64 = TRI }, 0, 0, FLAGS, .unit = "curve" },
468 { "qsin", "quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = QSIN }, 0, 0, FLAGS, .unit = "curve" },
469 { "esin", "exponential sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = ESIN }, 0, 0, FLAGS, .unit = "curve" },
470 { "hsin", "half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = HSIN }, 0, 0, FLAGS, .unit = "curve" },
471 { "log", "logarithmic", 0, AV_OPT_TYPE_CONST, {.i64 = LOG }, 0, 0, FLAGS, .unit = "curve" },
472 { "ipar", "inverted parabola", 0, AV_OPT_TYPE_CONST, {.i64 = IPAR }, 0, 0, FLAGS, .unit = "curve" },
473 { "qua", "quadratic", 0, AV_OPT_TYPE_CONST, {.i64 = QUA }, 0, 0, FLAGS, .unit = "curve" },
474 { "cub", "cubic", 0, AV_OPT_TYPE_CONST, {.i64 = CUB }, 0, 0, FLAGS, .unit = "curve" },
475 { "squ", "square root", 0, AV_OPT_TYPE_CONST, {.i64 = SQU }, 0, 0, FLAGS, .unit = "curve" },
476 { "cbr", "cubic root", 0, AV_OPT_TYPE_CONST, {.i64 = CBR }, 0, 0, FLAGS, .unit = "curve" },
477 { "par", "parabola", 0, AV_OPT_TYPE_CONST, {.i64 = PAR }, 0, 0, FLAGS, .unit = "curve" },
478 { "exp", "exponential", 0, AV_OPT_TYPE_CONST, {.i64 = EXP }, 0, 0, FLAGS, .unit = "curve" },
479 { "iqsin", "inverted quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IQSIN}, 0, 0, FLAGS, .unit = "curve" },
480 { "ihsin", "inverted half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IHSIN}, 0, 0, FLAGS, .unit = "curve" },
481 { "dese", "double-exponential seat", 0, AV_OPT_TYPE_CONST, {.i64 = DESE }, 0, 0, FLAGS, .unit = "curve" },
482 { "desi", "double-exponential sigmoid", 0, AV_OPT_TYPE_CONST, {.i64 = DESI }, 0, 0, FLAGS, .unit = "curve" },
483 { "losi", "logistic sigmoid", 0, AV_OPT_TYPE_CONST, {.i64 = LOSI }, 0, 0, FLAGS, .unit = "curve" },
484 { "sinc", "sine cardinal function", 0, AV_OPT_TYPE_CONST, {.i64 = SINC }, 0, 0, FLAGS, .unit = "curve" },
485 { "isinc", "inverted sine cardinal function", 0, AV_OPT_TYPE_CONST, {.i64 = ISINC}, 0, 0, FLAGS, .unit = "curve" },
486 { "quat", "quartic", 0, AV_OPT_TYPE_CONST, {.i64 = QUAT }, 0, 0, FLAGS, .unit = "curve" },
487 { "quatr", "quartic root", 0, AV_OPT_TYPE_CONST, {.i64 = QUATR}, 0, 0, FLAGS, .unit = "curve" },
488 { "qsin2", "squared quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = QSIN2}, 0, 0, FLAGS, .unit = "curve" },
489 { "hsin2", "squared half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = HSIN2}, 0, 0, FLAGS, .unit = "curve" },
490 { "curve2", "set fade curve type for 2nd stream", OFFSET(curve2), AV_OPT_TYPE_INT, {.i64 = TRI }, NONE, NB_CURVES - 1, FLAGS, .unit = "curve" },
491 { "c2", "set fade curve type for 2nd stream", OFFSET(curve2), AV_OPT_TYPE_INT, {.i64 = TRI }, NONE, NB_CURVES - 1, FLAGS, .unit = "curve" },
492 { NULL }
493};
494
495AVFILTER_DEFINE_CLASS(acrossfade);
496
497#define CROSSFADE_PLANAR(name, type) \
498static void crossfade_samples_## name ##p(uint8_t **dst, uint8_t * const *cf0, \
499 uint8_t * const *cf1, \
500 int nb_samples, int channels, \
501 int curve0, int curve1) \
502{ \
503 int i, c; \
504 \
505 for (i = 0; i < nb_samples; i++) { \
506 double gain0 = fade_gain(curve0, nb_samples - 1 - i, nb_samples,0.,1.);\
507 double gain1 = fade_gain(curve1, i, nb_samples, 0., 1.); \
508 for (c = 0; c < channels; c++) { \
509 type *d = (type *)dst[c]; \
510 const type *s0 = (type *)cf0[c]; \
511 const type *s1 = (type *)cf1[c]; \
512 \
513 d[i] = s0[i] * gain0 + s1[i] * gain1; \
514 } \
515 } \
516}
517
518#define CROSSFADE(name, type) \
519static void crossfade_samples_## name (uint8_t **dst, uint8_t * const *cf0, \
520 uint8_t * const *cf1, \
521 int nb_samples, int channels, \
522 int curve0, int curve1) \
523{ \
524 type *d = (type *)dst[0]; \
525 const type *s0 = (type *)cf0[0]; \
526 const type *s1 = (type *)cf1[0]; \
527 int i, c, k = 0; \
528 \
529 for (i = 0; i < nb_samples; i++) { \
530 double gain0 = fade_gain(curve0, nb_samples - 1-i,nb_samples,0.,1.);\
531 double gain1 = fade_gain(curve1, i, nb_samples, 0., 1.); \
532 for (c = 0; c < channels; c++, k++) \
533 d[k] = s0[k] * gain0 + s1[k] * gain1; \
534 } \
535}
536
537CROSSFADE_PLANAR(dbl, double)
538CROSSFADE_PLANAR(flt, float)
539CROSSFADE_PLANAR(s16, int16_t)
540CROSSFADE_PLANAR(s32, int32_t)
541
542CROSSFADE(dbl, double)
543CROSSFADE(flt, float)
544CROSSFADE(s16, int16_t)
545CROSSFADE(s32, int32_t)
546
547static int pass_frame(AVFilterLink *inlink, AVFilterLink *outlink, int64_t *pts)
548{
549 AVFrame *in;
550 int ret = ff_inlink_consume_frame(inlink, &in);
551 if (ret < 0)
552 return ret;
553 av_assert1(ret);
554 in->pts = *pts;
555 *pts += av_rescale_q(in->nb_samples,
556 (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
557 return ff_filter_frame(outlink, in);
558}
559
560static int pass_samples(AVFilterLink *inlink, AVFilterLink *outlink, unsigned nb_samples, int64_t *pts)
561{
562 AVFrame *in;
563 int ret = ff_inlink_consume_samples(inlink, nb_samples, nb_samples, &in);
564 if (ret < 0)
565 return ret;
566 av_assert1(ret);
567 in->pts = *pts;
568 *pts += av_rescale_q(in->nb_samples,
569 (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
570 return ff_filter_frame(outlink, in);
571}
572
573static int pass_crossfade(AVFilterContext *ctx, const int idx0, const int idx1)
574{
575 AudioFadeContext *s = ctx->priv;
576 AVFilterLink *outlink = ctx->outputs[0];
577 AVFrame *out, *cf[2] = { NULL };
578 int ret;
579
580 AVFilterLink *in0 = ctx->inputs[idx0];
581 AVFilterLink *in1 = ctx->inputs[idx1];
582 int queued_samples0 = ff_inlink_queued_samples(in0);
583 int queued_samples1 = ff_inlink_queued_samples(in1);
584
585 /* Limit to the relevant region */
586 av_assert1(queued_samples0 <= s->nb_samples);
587 if (ff_outlink_get_status(in1) && idx1 < s->nb_inputs - 1)
588 queued_samples1 /= 2; /* reserve second half for next fade-out */
589 queued_samples1 = FFMIN(queued_samples1, s->nb_samples);
590
591 if (s->overlap) {
592 int nb_samples = FFMIN(queued_samples0, queued_samples1);
593 if (nb_samples < s->nb_samples) {
594 av_log(ctx, AV_LOG_WARNING, "Input %d duration (%d samples) "
595 "is shorter than crossfade duration (%"PRId64" samples), "
596 "crossfade will be shorter by %"PRId64" samples.\n",
597 queued_samples0 <= queued_samples1 ? idx0 : idx1,
598 nb_samples, s->nb_samples, s->nb_samples - nb_samples);
599
600 if (queued_samples0 > nb_samples) {
601 ret = pass_samples(in0, outlink, queued_samples0 - nb_samples, &s->pts);
602 if (ret < 0)
603 return ret;
604 }
605
606 if (!nb_samples)
607 return 0; /* either input was completely empty */
608 }
609
610 av_assert1(nb_samples > 0);
611 out = ff_get_audio_buffer(outlink, nb_samples);
612 if (!out)
613 return AVERROR(ENOMEM);
614
615 ret = ff_inlink_consume_samples(in0, nb_samples, nb_samples, &cf[0]);
616 if (ret < 0) {
617 av_frame_free(&out);
618 return ret;
619 }
620
621 ret = ff_inlink_consume_samples(in1, nb_samples, nb_samples, &cf[1]);
622 if (ret < 0) {
623 av_frame_free(&cf[0]);
624 av_frame_free(&out);
625 return ret;
626 }
627
628 s->crossfade_samples(out->extended_data, cf[0]->extended_data,
629 cf[1]->extended_data, nb_samples,
630 out->ch_layout.nb_channels, s->curve, s->curve2);
631 out->pts = s->pts;
632 s->pts += av_rescale_q(nb_samples,
633 (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
634 av_frame_free(&cf[0]);
635 av_frame_free(&cf[1]);
636 return ff_filter_frame(outlink, out);
637 } else {
638 if (queued_samples0 < s->nb_samples) {
639 av_log(ctx, AV_LOG_WARNING, "Input %d duration (%d samples) "
640 "is shorter than crossfade duration (%"PRId64" samples), "
641 "fade-out will be shorter by %"PRId64" samples.\n",
642 idx0, queued_samples0, s->nb_samples,
643 s->nb_samples - queued_samples0);
644 if (!queued_samples0)
645 goto fade_in;
646 }
647
648 out = ff_get_audio_buffer(outlink, queued_samples0);
649 if (!out)
650 return AVERROR(ENOMEM);
651
652 ret = ff_inlink_consume_samples(in0, queued_samples0, queued_samples0, &cf[0]);
653 if (ret < 0) {
654 av_frame_free(&out);
655 return ret;
656 }
657
658 s->fade_samples(out->extended_data, cf[0]->extended_data, cf[0]->nb_samples,
659 outlink->ch_layout.nb_channels, -1, cf[0]->nb_samples - 1, cf[0]->nb_samples, s->curve, 0., 1.);
660 out->pts = s->pts;
661 s->pts += av_rescale_q(cf[0]->nb_samples,
662 (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
663 av_frame_free(&cf[0]);
664 ret = ff_filter_frame(outlink, out);
665 if (ret < 0)
666 return ret;
667
668 fade_in:
669 if (queued_samples1 < s->nb_samples) {
670 av_log(ctx, AV_LOG_WARNING, "Input %d duration (%d samples) "
671 "is shorter than crossfade duration (%"PRId64" samples), "
672 "fade-in will be shorter by %"PRId64" samples.\n",
673 idx1, ff_inlink_queued_samples(in1), s->nb_samples,
674 s->nb_samples - queued_samples1);
675 if (!queued_samples1)
676 return 0;
677 }
678
679 out = ff_get_audio_buffer(outlink, queued_samples1);
680 if (!out)
681 return AVERROR(ENOMEM);
682
683 ret = ff_inlink_consume_samples(in1, queued_samples1, queued_samples1, &cf[1]);
684 if (ret < 0) {
685 av_frame_free(&out);
686 return ret;
687 }
688
689 s->fade_samples(out->extended_data, cf[1]->extended_data, cf[1]->nb_samples,
690 outlink->ch_layout.nb_channels, 1, 0, cf[1]->nb_samples, s->curve2, 0., 1.);
691 out->pts = s->pts;
692 s->pts += av_rescale_q(cf[1]->nb_samples,
693 (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
694 av_frame_free(&cf[1]);
695 return ff_filter_frame(outlink, out);
696 }
697}
698
699static int activate(AVFilterContext *ctx)
700{
701 AudioFadeContext *s = ctx->priv;
702 const int idx0 = s->xfade_idx;
703 const int idx1 = s->xfade_idx + 1;
704 AVFilterLink *outlink = ctx->outputs[0];
705 AVFilterLink *in0 = ctx->inputs[idx0];
706
707 FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, ctx);
708
709 if (idx0 == s->nb_inputs - 1) {
710 /* Last active input, read until EOF */
711 if (ff_inlink_queued_frames(in0))
712 return pass_frame(in0, outlink, &s->pts);
713 FF_FILTER_FORWARD_STATUS(in0, outlink);
714 FF_FILTER_FORWARD_WANTED(outlink, in0);
715 return FFERROR_NOT_READY;
716 }
717
718 AVFilterLink *in1 = ctx->inputs[idx1];
719 int queued_samples0 = ff_inlink_queued_samples(in0);
720 if (queued_samples0 > s->nb_samples) {
721 AVFrame *frame = ff_inlink_peek_frame(in0, 0);
722 if (queued_samples0 - s->nb_samples >= frame->nb_samples)
723 return pass_frame(in0, outlink, &s->pts);
724 }
725
726 /* Continue reading until EOF */
727 if (ff_outlink_get_status(in0)) {
728 if (queued_samples0 > s->nb_samples)
729 return pass_samples(in0, outlink, queued_samples0 - s->nb_samples, &s->pts);
730 } else {
731 FF_FILTER_FORWARD_WANTED(outlink, in0);
732 return FFERROR_NOT_READY;
733 }
734
735 /* At this point, in0 has reached EOF with no more samples remaining
736 * except those that we want to crossfade */
737 av_assert0(queued_samples0 <= s->nb_samples);
738 int queued_samples1 = ff_inlink_queued_samples(in1);
739
740 /* If this clip is sandwiched between two other clips, buffer at least
741 * twice the total crossfade duration to ensure that we won't reach EOF
742 * during the second fade (in which case we would shorten the fade) */
743 int needed_samples = s->nb_samples;
744 if (idx1 < s->nb_inputs - 1)
745 needed_samples *= 2;
746
747 if (queued_samples1 >= needed_samples || ff_outlink_get_status(in1)) {
748 /* The first filter may EOF before delivering any samples, in which
749 * case it's possible for pass_crossfade() to be a no-op. Just ensure
750 * the activate() function runs again after incrementing the index to
751 * ensure we correctly move on to the next input in that case. */
752 s->xfade_idx++;
753 ff_filter_set_ready(ctx, 10);
754 return pass_crossfade(ctx, idx0, idx1);
755 } else {
756 FF_FILTER_FORWARD_WANTED(outlink, in1);
757 return FFERROR_NOT_READY;
758 }
759}
760
761static av_cold int acrossfade_init(AVFilterContext *ctx)
762{
763 AudioFadeContext *s = ctx->priv;
764 int ret;
765
766 for (int i = 0; i < s->nb_inputs; i++) {
767 AVFilterPad pad = {
768 .name = av_asprintf("crossfade%d", i),
769 .type = AVMEDIA_TYPE_AUDIO,
770 };
771 if (!pad.name)
772 return AVERROR(ENOMEM);
773
774 ret = ff_append_inpad_free_name(ctx, &pad);
775 if (ret < 0)
776 return ret;
777 }
778
779 return 0;
780}
781
782static int acrossfade_config_output(AVFilterLink *outlink)
783{
784 AVFilterContext *ctx = outlink->src;
785 AudioFadeContext *s = ctx->priv;
786
787 outlink->time_base = ctx->inputs[0]->time_base;
788
789 switch (outlink->format) {
790 case AV_SAMPLE_FMT_DBL: s->crossfade_samples = crossfade_samples_dbl; break;
791 case AV_SAMPLE_FMT_DBLP: s->crossfade_samples = crossfade_samples_dblp; break;
792 case AV_SAMPLE_FMT_FLT: s->crossfade_samples = crossfade_samples_flt; break;
793 case AV_SAMPLE_FMT_FLTP: s->crossfade_samples = crossfade_samples_fltp; break;
794 case AV_SAMPLE_FMT_S16: s->crossfade_samples = crossfade_samples_s16; break;
795 case AV_SAMPLE_FMT_S16P: s->crossfade_samples = crossfade_samples_s16p; break;
796 case AV_SAMPLE_FMT_S32: s->crossfade_samples = crossfade_samples_s32; break;
797 case AV_SAMPLE_FMT_S32P: s->crossfade_samples = crossfade_samples_s32p; break;
798 }
799
800 config_output(outlink);
801
802 return 0;
803}
804
805static const AVFilterPad avfilter_af_acrossfade_outputs[] = {
806 {
807 .name = "default",
808 .type = AVMEDIA_TYPE_AUDIO,
809 .config_props = acrossfade_config_output,
810 },
811};
812
813const FFFilter ff_af_acrossfade = {
814 .p.name = "acrossfade",
815 .p.description = NULL_IF_CONFIG_SMALL("Cross fade two input audio streams."),
816 .p.priv_class = &acrossfade_class,
817 .p.flags = AVFILTER_FLAG_DYNAMIC_INPUTS,
818 .priv_size = sizeof(AudioFadeContext),
819 .init = acrossfade_init,
820 .activate = activate,
821 FILTER_OUTPUTS(avfilter_af_acrossfade_outputs),
822 FILTER_SAMPLEFMTS_ARRAY(sample_fmts),
823};
824
825#endif /* CONFIG_ACROSSFADE_FILTER */
dst
uint8_t ptrdiff_t const uint8_t ptrdiff_t int intptr_t intptr_t int int16_t * dst
Definition dsp.h:87
sample_fmts
static enum AVSampleFormat sample_fmts[]
Definition adpcmenc.c:933
process_command
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, char *res, int res_len, int flags)
Definition af_acrusher.c:307
CurveType
CurveType
Definition af_afade.c:62
QUAT
@ QUAT
Definition af_afade.c:62
QSIN
@ QSIN
Definition af_afade.c:62
IQSIN
@ IQSIN
Definition af_afade.c:62
IHSIN
@ IHSIN
Definition af_afade.c:62
QSIN2
@ QSIN2
Definition af_afade.c:62
DESI
@ DESI
Definition af_afade.c:62
ISINC
@ ISINC
Definition af_afade.c:62
LOSI
@ LOSI
Definition af_afade.c:62
NB_CURVES
@ NB_CURVES
Definition af_afade.c:62
HSIN2
@ HSIN2
Definition af_afade.c:62
QUATR
@ QUATR
Definition af_afade.c:62
QUA
@ QUA
Definition af_afade.c:62
DESE
@ DESE
Definition af_afade.c:62
SQU
@ SQU
Definition af_afade.c:62
TRI
@ TRI
Definition af_afade.c:62
CBR
@ CBR
Definition af_afade.c:62
CUB
@ CUB
Definition af_afade.c:62
IPAR
@ IPAR
Definition af_afade.c:62
ESIN
@ ESIN
Definition af_afade.c:62
HSIN
@ HSIN
Definition af_afade.c:62
FADE_PLANAR
#define FADE_PLANAR(name, type)
Definition af_afade.c:166
FADE
#define FADE(name, type)
Definition af_afade.c:185
CUBE
#define CUBE(a)
TFLAGS
#define TFLAGS
Definition af_afade.c:66
SCALE_PLANAR
#define SCALE_PLANAR(name, type)
Definition af_afade.c:212
fade_gain
static double fade_gain(int curve, int64_t index, int64_t range, double silence, double unity)
Definition af_afade.c:76
SCALE
#define SCALE(name, type)
Definition af_afade.c:229
OFFSET
#define OFFSET(x)
Definition af_afade.c:64
config_output
static int config_output(AVFilterLink *outlink)
Definition af_afade.c:253
ff_af_afade
const FFFilter ff_af_afade
ff_af_acrossfade
const FFFilter ff_af_acrossfade
PAR
#define PAR
Definition apedec.c:1740
out
static FILE * out
Definition api-movenc-test.c:55
ctx
static AVFormatContext * ctx
Definition api-movenc-test.c:49
channels
channels
Definition aptx.h:31
A
#define A(x)
Definition vpx_arith.h:28
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:74
int32_t
int32_t
Definition audioconvert.c:56
avassert.h
simple assert() macros that are a bit more flexible than ISO C assert().
av_assert1
#define av_assert1(cond)
assert() equivalent, that does not lie in speed critical code.
Definition avassert.h:58
av_assert0
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition avassert.h:42
ff_append_inpad_free_name
int ff_append_inpad_free_name(AVFilterContext *f, AVFilterPad *p)
Definition avfilter.c:132
ff_outlink_get_status
int ff_outlink_get_status(AVFilterLink *link)
Get the status on an output link.
Definition avfilter.c:1648
ff_filter_frame
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition avfilter.c:1068
ff_filter_process_command
int ff_filter_process_command(AVFilterContext *ctx, const char *cmd, const char *arg, char *res, int res_len, int flags)
Generic processing of user supplied commands that are set in the same way as the filter options.
Definition avfilter.c:906
ff_inlink_queued_frames
size_t ff_inlink_queued_frames(AVFilterLink *link)
Get the number of frames available on the link.
Definition avfilter.c:1483
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:1540
ff_inlink_peek_frame
AVFrame * ff_inlink_peek_frame(AVFilterLink *link, size_t idx)
Access a frame in the link fifo without consuming it.
Definition avfilter.c:1561
ff_filter_set_ready
void ff_filter_set_ready(AVFilterContext *filter, unsigned priority)
Mark a filter ready and schedule it for activation.
Definition avfilter.c:229
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:1520
ff_inlink_queued_samples
int ff_inlink_queued_samples(AVFilterLink *link)
Definition avfilter.c:1495
avfilter.h
Main libavfilter public API header.
av_asprintf
char * av_asprintf(const char *fmt,...)
Definition avstring.c:115
flags
#define flags(name, subs,...)
Definition cbs_h264.c:74
i
#define i(width, name, range_min, range_max)
Definition cbs_h264.c:63
s
#define s(width, name)
Definition cbs_vp9.c:198
FLAGS
#define FLAGS
Definition cmdutils.c:598
av_clipd
#define av_clipd
Definition common.h:148
NULL
#define NULL
Definition coverity.c:32
int64_t
long long int64_t
Definition coverity.c:34
frame
static AVFrame * frame
Definition demux_decode.c:54
filter_frame
static int filter_frame(DBEDecodeContext *s, AVFrame *frame)
Definition dolby_e.c:1067
init
int(* init)(AVBSFContext *ctx)
Definition dts2pts.c:608
exp
int8_t exp
Definition eval.c:76
duration
static int64_t duration
Definition ffplay.c:330
start_time
static int64_t start_time
Definition ffplay.c:329
AV_OPT_TYPE_CONST
@ AV_OPT_TYPE_CONST
Special option type for declaring named constants.
Definition opt.h:298
AV_OPT_TYPE_DURATION
@ AV_OPT_TYPE_DURATION
Underlying C type is int64_t.
Definition opt.h:318
AV_OPT_TYPE_INT64
@ AV_OPT_TYPE_INT64
Underlying C type is int64_t.
Definition opt.h:262
AV_OPT_TYPE_INT
@ AV_OPT_TYPE_INT
Underlying C type is int.
Definition opt.h:258
AV_OPT_TYPE_DOUBLE
@ AV_OPT_TYPE_DOUBLE
Underlying C type is double.
Definition opt.h:266
AV_OPT_TYPE_BOOL
@ AV_OPT_TYPE_BOOL
Underlying C type is int.
Definition opt.h:326
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:196
AVFILTER_FLAG_DYNAMIC_INPUTS
#define AVFILTER_FLAG_DYNAMIC_INPUTS
The number of the filter inputs is not determined just by AVFilter.inputs.
Definition avfilter.h:155
AVERROR
#define AVERROR(e)
Definition error.h:45
av_frame_is_writable
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
Definition frame.c:535
av_frame_free
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition frame.c:64
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:599
AV_LOG_WARNING
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition log.h:216
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_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
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition avutil.h:201
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition samplefmt.h:55
AV_SAMPLE_FMT_FLTP
@ AV_SAMPLE_FMT_FLTP
float, planar
Definition samplefmt.h:66
AV_SAMPLE_FMT_S16P
@ AV_SAMPLE_FMT_S16P
signed 16 bits, planar
Definition samplefmt.h:64
AV_SAMPLE_FMT_FLT
@ AV_SAMPLE_FMT_FLT
float
Definition samplefmt.h:60
AV_SAMPLE_FMT_S32P
@ AV_SAMPLE_FMT_S32P
signed 32 bits, planar
Definition samplefmt.h:65
AV_SAMPLE_FMT_S32
@ AV_SAMPLE_FMT_S32
signed 32 bits
Definition samplefmt.h:59
AV_SAMPLE_FMT_NONE
@ AV_SAMPLE_FMT_NONE
Definition samplefmt.h:56
AV_SAMPLE_FMT_DBLP
@ AV_SAMPLE_FMT_DBLP
double, planar
Definition samplefmt.h:67
AV_SAMPLE_FMT_DBL
@ AV_SAMPLE_FMT_DBL
double
Definition samplefmt.h:61
AV_SAMPLE_FMT_S16
@ AV_SAMPLE_FMT_S16
signed 16 bits
Definition samplefmt.h:58
av_samples_set_silence
int av_samples_set_silence(uint8_t *const *audio_data, int offset, int nb_samples, int nb_channels, enum AVSampleFormat sample_fmt)
Fill an audio buffer with silence.
Definition samplefmt.c:246
AV_TIME_BASE
#define AV_TIME_BASE
Internal time base represented as integer.
Definition avutil.h:253
index
int index
Definition gxfenc.c:90
a
int a
Definition h264pred_template.c:416
if
if(svq3)
Definition h264pred_template.c:429
B
#define B
Definition huffyuv.h:42
type
cl_device_type type
Definition hwcontext_opencl.c:213
activate
static int activate(AVBitStreamFilterContext *ctx)
Definition lcevc_merge.c:144
config_output
static int config_output(AVBitStreamFilterLink *outlink)
Definition lcevc_merge.c:111
C
#define C
FILTER_INPUTS
#define FILTER_INPUTS(array)
Definition filters.h:264
FILTER_SAMPLEFMTS_ARRAY
#define FILTER_SAMPLEFMTS_ARRAY(array)
Definition filters.h:247
FILTER_OUTPUTS
#define FILTER_OUTPUTS(array)
Definition filters.h:265
FF_FILTER_FORWARD_WANTED
#define FF_FILTER_FORWARD_WANTED(outlink, inlink)
Forward the frame_wanted_out flag from an output link to an input link.
Definition filters.h:694
FF_FILTER_FORWARD_STATUS
#define FF_FILTER_FORWARD_STATUS(inlink, outlink)
Acknowledge the status on an input link and forward it to an output link.
Definition filters.h:666
FFERROR_NOT_READY
#define FFERROR_NOT_READY
Filters implementation helper functions and internal structures.
Definition filters.h:34
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:652
AVFILTER_DEFINE_CLASS
#define AVFILTER_DEFINE_CLASS(fname)
Definition filters.h:478
av_cold
#define av_cold
Definition attributes.h:117
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:88
FFMIN
#define FFMIN(a, b)
Definition macros.h:49
M_PI
#define M_PI
Definition mathematics.h:67
range
enum AVColorRange range
Definition mediacodec_wrapper.c:2594
EXP
@ EXP
Definition mjpeg.h:77
opt.h
AVOptions.
AVChannelLayout::nb_channels
int nb_channels
Number of channels in this layout.
Definition channel_layout.h:329
AVClass
Describe the class of an AVClass context structure.
Definition log.h:76
AVFilterContext
An instance of a filter.
Definition avfilter.h:273
AVFilterContext::priv
void * priv
private data for use by the filter
Definition avfilter.h:288
AVFilterContext::outputs
AVFilterLink ** outputs
array of pointers to output links
Definition avfilter.h:285
AVFilterPad
A filter pad used for either input or output.
Definition filters.h:40
AVFilterPad::name
const char * name
Pad name.
Definition filters.h:46
AVFrame
This structure describes decoded (raw) audio or video data.
Definition frame.h:472
AVFrame::nb_samples
int nb_samples
number of audio samples (per channel) described by this frame
Definition frame.h:552
AVFrame::pts
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition frame.h:574
AVFrame::ch_layout
AVChannelLayout ch_layout
Channel layout of the audio data.
Definition frame.h:815
AVFrame::format
int format
format of the frame, -1 if unknown or unset Values correspond to enum AVPixelFormat for video frames,...
Definition frame.h:559
AVFrame::extended_data
uint8_t ** extended_data
pointers to the data planes/channels.
Definition frame.h:533
AVOption
AVOption.
Definition opt.h:428
AVRational
Rational number (pair of numerator and denominator).
Definition rational.h:58
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, double silence, double unity)
Definition af_afade.c:50
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:56
AudioFadeContext::nb_samples
int64_t nb_samples
Definition af_afade.c:40
AudioFadeContext::scale_samples
void(* scale_samples)(uint8_t **dst, uint8_t *const *src, int nb_samples, int channels, double unity)
Definition af_afade.c:54
AudioFadeContext::start_time
int64_t start_time
Definition af_afade.c:43
AudioFadeContext::start_sample
int64_t start_sample
Definition af_afade.c:41
AudioFadeContext::duration
int64_t duration
Definition af_afade.c:42
cbrt
#define cbrt
Definition tablegen.h:35
av_log
#define av_log(a,...)
Definition tableprint_vlc.h:27
LOG
#define LOG(...)
Definition internal.h:126
src
#define src
Definition vp8dsp.c:248
pts
static int64_t pts
Definition transcode_aac.c:649
NONE
#define NONE
Definition vf_drawvg.c:262
SINC
#define SINC(x)
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