FFmpeg coverage


Directory: ../../../ffmpeg/
File: src/libavfilter/af_afade.c
Date: 2025-10-11 17:33:36
Exec Total Coverage
Lines: 155 342 45.3%
Functions: 12 35 34.3%
Branches: 74 274 27.0%

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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
35 typedef 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
62 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 };
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
68 static const enum AVSampleFormat sample_fmts[] = {
69 AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P,
70 AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32P,
71 AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
72 AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
73 AV_SAMPLE_FMT_NONE
74 };
75
76 913952 static 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 913952 gain = av_clipd(1.0 * index / range, 0, 1.0);
82
83
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913952 switch (curve) {
84 90112 case QSIN:
85 90112 gain = sin(gain * M_PI / 2.0);
86 90112 break;
87 90112 case IQSIN:
88 /* 0.6... = 2 / M_PI */
89 90112 gain = 0.6366197723675814 * asin(gain);
90 90112 break;
91 90112 case ESIN:
92 90112 gain = 1.0 - cos(M_PI / 4.0 * (CUBE(2.0*gain - 1) + 1));
93 90112 break;
94 90112 case HSIN:
95 90112 gain = (1.0 - cos(gain * M_PI)) / 2.0;
96 90112 break;
97 case IHSIN:
98 /* 0.3... = 1 / M_PI */
99 gain = 0.3183098861837907 * acos(1 - 2 * gain);
100 break;
101 266512 case EXP:
102 /* -11.5... = 5*ln(0.1) */
103 266512 gain = exp(-11.512925464970227 * (1 - gain));
104 266512 break;
105 286992 case LOG:
106 286992 gain = av_clipd(1 + 0.2 * log10(gain), 0, 1.0);
107 286992 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 913952 return silence + (unity - silence) * gain;
164 }
165
166 #define FADE_PLANAR(name, type) \
167 static 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) \
186 static void fade_samples_## name (uint8_t **dst, uint8_t * const *src, \
187 int nb_samples, int channels, int dir, \
188 int64_t start, int64_t range, int curve, \
189 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
202 FADE_PLANAR(dbl, double)
203 FADE_PLANAR(flt, float)
204 FADE_PLANAR(s16, int16_t)
205 FADE_PLANAR(s32, int32_t)
206
207 FADE(dbl, double)
208 FADE(flt, float)
209
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1683593 FADE(s16, int16_t)
210 FADE(s32, int32_t)
211
212 #define SCALE_PLANAR(name, type) \
213 static 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) \
230 static 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
243 SCALE_PLANAR(dbl, double)
244 SCALE_PLANAR(flt, float)
245 SCALE_PLANAR(s16, int16_t)
246 SCALE_PLANAR(s32, int32_t)
247
248 SCALE(dbl, double)
249 SCALE(flt, float)
250 SCALE(s16, int16_t)
251 SCALE(s32, int32_t)
252
253 7 static int config_output(AVFilterLink *outlink)
254 {
255 7 AVFilterContext *ctx = outlink->src;
256 7 AudioFadeContext *s = ctx->priv;
257
258
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7 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 7 case AV_SAMPLE_FMT_S16: s->fade_samples = fade_samples_s16;
272 7 s->scale_samples = scale_samples_s16;
273 7 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
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7 if (s->duration)
286 7 s->nb_samples = av_rescale(s->duration, outlink->sample_rate, AV_TIME_BASE);
287 7 s->duration = 0;
288
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7 if (s->start_time)
289 s->start_sample = av_rescale(s->start_time, outlink->sample_rate, AV_TIME_BASE);
290 7 s->start_time = 0;
291
292 7 return 0;
293 }
294
295 #if CONFIG_AFADE_FILTER
296
297 static 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
341 AVFILTER_DEFINE_CLASS(afade);
342
343 12 static av_cold int init(AVFilterContext *ctx)
344 {
345 12 AudioFadeContext *s = ctx->priv;
346
347
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12 if (INT64_MAX - s->nb_samples < s->start_sample)
348 return AVERROR(EINVAL);
349
350 12 return 0;
351 }
352
353 390 static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
354 {
355 390 AudioFadeContext *s = inlink->dst->priv;
356 390 AVFilterLink *outlink = inlink->dst->outputs[0];
357 390 int nb_samples = buf->nb_samples;
358 AVFrame *out_buf;
359 390 int64_t cur_sample = av_rescale_q(buf->pts, inlink->time_base, (AVRational){1, inlink->sample_rate});
360
361
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390 if (s->unity == 1.0 &&
362
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390 ((!s->type && (s->start_sample + s->nb_samples < cur_sample)) ||
363
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137 ( s->type && (cur_sample + nb_samples < s->start_sample))))
364 253 return ff_filter_frame(outlink, buf);
365
366
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137 if (av_frame_is_writable(buf)) {
367 137 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
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137 if ((!s->type && (cur_sample + nb_samples < s->start_sample)) ||
376
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137 ( 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
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137 } else if (( s->type && (cur_sample + nb_samples < s->start_sample)) ||
386
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137 (!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
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137 if (!s->type)
394 137 start = cur_sample - s->start_sample;
395 else
396 start = s->start_sample + s->nb_samples - cur_sample;
397
398 137 s->fade_samples(out_buf->extended_data, buf->extended_data,
399 137 nb_samples, buf->ch_layout.nb_channels,
400
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137 s->type ? -1 : 1, start,
401 s->nb_samples, s->curve, s->silence, s->unity);
402 }
403
404
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137 if (buf != out_buf)
405 av_frame_free(&buf);
406
407 137 return ff_filter_frame(outlink, out_buf);
408 }
409
410 static 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
422 static const AVFilterPad avfilter_af_afade_inputs[] = {
423 {
424 .name = "default",
425 .type = AVMEDIA_TYPE_AUDIO,
426 .filter_frame = filter_frame,
427 },
428 };
429
430 static const AVFilterPad avfilter_af_afade_outputs[] = {
431 {
432 .name = "default",
433 .type = AVMEDIA_TYPE_AUDIO,
434 .config_props = config_output,
435 },
436 };
437
438 const 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
455 static 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
495 AVFILTER_DEFINE_CLASS(acrossfade);
496
497 #define CROSSFADE_PLANAR(name, type) \
498 static 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) \
519 static 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
537 CROSSFADE_PLANAR(dbl, double)
538 CROSSFADE_PLANAR(flt, float)
539 CROSSFADE_PLANAR(s16, int16_t)
540 CROSSFADE_PLANAR(s32, int32_t)
541
542 CROSSFADE(dbl, double)
543 CROSSFADE(flt, float)
544
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529202 CROSSFADE(s16, int16_t)
545 CROSSFADE(s32, int32_t)
546
547 104 static int pass_frame(AVFilterLink *inlink, AVFilterLink *outlink, int64_t *pts)
548 {
549 AVFrame *in;
550 104 int ret = ff_inlink_consume_frame(inlink, &in);
551
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104 if (ret < 0)
552 return ret;
553 av_assert1(ret);
554 104 in->pts = *pts;
555 104 *pts += av_rescale_q(in->nb_samples,
556 104 (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
557 104 return ff_filter_frame(outlink, in);
558 }
559
560 2 static int pass_samples(AVFilterLink *inlink, AVFilterLink *outlink, unsigned nb_samples, int64_t *pts)
561 {
562 AVFrame *in;
563 2 int ret = ff_inlink_consume_samples(inlink, nb_samples, nb_samples, &in);
564
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2 if (ret < 0)
565 return ret;
566 av_assert1(ret);
567 2 in->pts = *pts;
568 2 *pts += av_rescale_q(in->nb_samples,
569 2 (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
570 2 return ff_filter_frame(outlink, in);
571 }
572
573 2 static int pass_crossfade(AVFilterContext *ctx, const int idx0, const int idx1)
574 {
575 2 AudioFadeContext *s = ctx->priv;
576 2 AVFilterLink *outlink = ctx->outputs[0];
577 2 AVFrame *out, *cf[2] = { NULL };
578 int ret;
579
580 2 AVFilterLink *in0 = ctx->inputs[idx0];
581 2 AVFilterLink *in1 = ctx->inputs[idx1];
582 2 int queued_samples0 = ff_inlink_queued_samples(in0);
583 2 int queued_samples1 = ff_inlink_queued_samples(in1);
584
585 /* Limit to the relevant region */
586 av_assert1(queued_samples0 <= s->nb_samples);
587
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2 if (ff_outlink_get_status(in1) && idx1 < s->nb_inputs - 1)
588 queued_samples1 /= 2; /* reserve second half for next fade-out */
589 2 queued_samples1 = FFMIN(queued_samples1, s->nb_samples);
590
591
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2 if (s->overlap) {
592 2 int nb_samples = FFMIN(queued_samples0, queued_samples1);
593
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2 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 2 out = ff_get_audio_buffer(outlink, nb_samples);
612
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2 if (!out)
613 return AVERROR(ENOMEM);
614
615 2 ret = ff_inlink_consume_samples(in0, nb_samples, nb_samples, &cf[0]);
616
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2 if (ret < 0) {
617 av_frame_free(&out);
618 return ret;
619 }
620
621 2 ret = ff_inlink_consume_samples(in1, nb_samples, nb_samples, &cf[1]);
622
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2 if (ret < 0) {
623 av_frame_free(&cf[0]);
624 av_frame_free(&out);
625 return ret;
626 }
627
628 2 s->crossfade_samples(out->extended_data, cf[0]->extended_data,
629 2 cf[1]->extended_data, nb_samples,
630 2 out->ch_layout.nb_channels, s->curve, s->curve2);
631 2 out->pts = s->pts;
632 2 s->pts += av_rescale_q(nb_samples,
633 2 (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
634 2 av_frame_free(&cf[0]);
635 2 av_frame_free(&cf[1]);
636 2 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
699 382 static int activate(AVFilterContext *ctx)
700 {
701 382 AudioFadeContext *s = ctx->priv;
702 382 const int idx0 = s->xfade_idx;
703 382 const int idx1 = s->xfade_idx + 1;
704 382 AVFilterLink *outlink = ctx->outputs[0];
705 382 AVFilterLink *in0 = ctx->inputs[idx0];
706
707
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385 FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, ctx);
708
709
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381 if (idx0 == s->nb_inputs - 1) {
710 /* Last active input, read until EOF */
711
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5 if (ff_inlink_queued_frames(in0))
712 2 return pass_frame(in0, outlink, &s->pts);
713
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3 FF_FILTER_FORWARD_STATUS(in0, outlink);
714
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2 FF_FILTER_FORWARD_WANTED(outlink, in0);
715 return FFERROR_NOT_READY;
716 }
717
718 376 AVFilterLink *in1 = ctx->inputs[idx1];
719 376 int queued_samples0 = ff_inlink_queued_samples(in0);
720
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376 if (queued_samples0 > s->nb_samples) {
721 210 AVFrame *frame = ff_inlink_peek_frame(in0, 0);
722
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210 if (queued_samples0 - s->nb_samples >= frame->nb_samples)
723 102 return pass_frame(in0, outlink, &s->pts);
724 }
725
726 /* Continue reading until EOF */
727
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274 if (ff_outlink_get_status(in0)) {
728
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122 if (queued_samples0 > s->nb_samples)
729 2 return pass_samples(in0, outlink, queued_samples0 - s->nb_samples, &s->pts);
730 } else {
731
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152 FF_FILTER_FORWARD_WANTED(outlink, in0);
732 1 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
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120 av_assert0(queued_samples0 <= s->nb_samples);
738 120 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) */
743 120 int needed_samples = s->nb_samples;
744
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120 if (idx1 < s->nb_inputs - 1)
745 82 needed_samples *= 2;
746
747
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120 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 2 s->xfade_idx++;
753 2 ff_filter_set_ready(ctx, 10);
754 2 return pass_crossfade(ctx, idx0, idx1);
755 } else {
756
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118 FF_FILTER_FORWARD_WANTED(outlink, in1);
757 return FFERROR_NOT_READY;
758 }
759 }
760
761 2 static av_cold int acrossfade_init(AVFilterContext *ctx)
762 {
763 2 AudioFadeContext *s = ctx->priv;
764 int ret;
765
766
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8 for (int i = 0; i < s->nb_inputs; i++) {
767 12 AVFilterPad pad = {
768 6 .name = av_asprintf("crossfade%d", i),
769 .type = AVMEDIA_TYPE_AUDIO,
770 };
771
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6 if (!pad.name)
772 return AVERROR(ENOMEM);
773
774 6 ret = ff_append_inpad_free_name(ctx, &pad);
775
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6 if (ret < 0)
776 return ret;
777 }
778
779 2 return 0;
780 }
781
782 1 static int acrossfade_config_output(AVFilterLink *outlink)
783 {
784 1 AVFilterContext *ctx = outlink->src;
785 1 AudioFadeContext *s = ctx->priv;
786
787 1 outlink->time_base = ctx->inputs[0]->time_base;
788
789
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1 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 1 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 1 config_output(outlink);
801
802 1 return 0;
803 }
804
805 static const AVFilterPad avfilter_af_acrossfade_outputs[] = {
806 {
807 .name = "default",
808 .type = AVMEDIA_TYPE_AUDIO,
809 .config_props = acrossfade_config_output,
810 },
811 };
812
813 const 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 */
826