FFmpeg coverage

Directory: ../../../ffmpeg/
File: src/libavfilter/af_adelay.c
Date: 2025-10-27 14:11:36
Exec Total Coverage
Lines: 110 196 56.1%
Functions: 6 16 37.5%
Branches: 60 212 28.3%
Line Branch Exec Source
1 /*
2 * Copyright (c) 2013 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 #include "libavutil/avstring.h"
22 #include "libavutil/mem.h"
23 #include "libavutil/opt.h"
24 #include "libavutil/samplefmt.h"
25 #include "avfilter.h"
26 #include "audio.h"
27 #include "filters.h"
28
29 typedef struct ChanDelay {
30 int64_t delay;
31 size_t delay_index;
32 size_t index;
33 unsigned int samples_size;
34 uint8_t *samples;
35 } ChanDelay;
36
37 typedef struct AudioDelayContext {
38 const AVClass *class;
39 int all;
40 char *delays;
41 ChanDelay *chandelay;
42 int nb_delays;
43 int block_align;
44 int64_t padding;
45 int64_t max_delay;
46 int64_t offset;
47 int64_t next_pts;
48 int eof;
49
50 AVFrame *input;
51
52 void (*delay_channel)(ChanDelay *d, int nb_samples,
53 const uint8_t *src, uint8_t *dst);
54 int (*resize_channel_samples)(ChanDelay *d, int64_t new_delay);
55 } AudioDelayContext;
56
57 #define OFFSET(x) offsetof(AudioDelayContext, x)
58 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
59
60 static const AVOption adelay_options[] = {
61 { "delays", "set list of delays for each channel", OFFSET(delays), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A | AV_OPT_FLAG_RUNTIME_PARAM },
62 { "all", "use last available delay for remained channels", OFFSET(all), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
63 { NULL }
64 };
65
66 AVFILTER_DEFINE_CLASS(adelay);
67
68 #define DELAY(name, type, fill) \
69 static void delay_channel_## name ##p(ChanDelay *d, int nb_samples, \
70 const uint8_t *ssrc, uint8_t *ddst) \
71 { \
72 const type *src = (type *)ssrc; \
73 type *dst = (type *)ddst; \
74 type *samples = (type *)d->samples; \
75 \
76 while (nb_samples) { \
77 if (d->delay_index < d->delay) { \
78 const int len = FFMIN(nb_samples, d->delay - d->delay_index); \
79 \
80 memcpy(&samples[d->delay_index], src, len * sizeof(type)); \
81 memset(dst, fill, len * sizeof(type)); \
82 d->delay_index += len; \
83 src += len; \
84 dst += len; \
85 nb_samples -= len; \
86 } else { \
87 *dst = samples[d->index]; \
88 samples[d->index] = *src; \
89 nb_samples--; \
90 d->index++; \
91 src++, dst++; \
92 d->index = d->index >= d->delay ? 0 : d->index; \
93 } \
94 } \
95 }
96
97 ✗ DELAY(u8, uint8_t, 0x80)
98
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 264667 DELAY(s16, int16_t, 0)
99 ✗ DELAY(s32, int32_t, 0)
100 ✗ DELAY(flt, float, 0)
101 ✗ DELAY(dbl, double, 0)
102
103 #define CHANGE_DELAY(name, type, fill) \
104 static int resize_samples_## name ##p(ChanDelay *d, int64_t new_delay) \
105 { \
106 type *samples; \
107 \
108 if (new_delay == d->delay) { \
109 return 0; \
110 } \
111 \
112 if (new_delay == 0) { \
113 av_freep(&d->samples); \
114 d->samples_size = 0; \
115 d->delay = 0; \
116 d->index = 0; \
117 d->delay_index = 0; \
118 return 0; \
119 } \
120 \
121 samples = (type *) av_fast_realloc(d->samples, &d->samples_size, new_delay * sizeof(type)); \
122 if (!samples) { \
123 return AVERROR(ENOMEM); \
124 } \
125 \
126 if (new_delay < d->delay) { \
127 if (d->index > new_delay) { \
128 d->index -= new_delay; \
129 memmove(samples, &samples[new_delay], d->index * sizeof(type)); \
130 d->delay_index = new_delay; \
131 } else if (d->delay_index > d->index) { \
132 memmove(&samples[d->index], &samples[d->index+(d->delay-new_delay)], \
133 (new_delay - d->index) * sizeof(type)); \
134 d->delay_index -= d->delay - new_delay; \
135 } \
136 } else { \
137 size_t block_size; \
138 if (d->delay_index >= d->delay) { \
139 block_size = (d->delay - d->index) * sizeof(type); \
140 memmove(&samples[d->index+(new_delay - d->delay)], &samples[d->index], block_size); \
141 d->delay_index = new_delay; \
142 } else { \
143 d->delay_index += new_delay - d->delay; \
144 } \
145 block_size = (new_delay - d->delay) * sizeof(type); \
146 memset(&samples[d->index], fill, block_size); \
147 } \
148 d->delay = new_delay; \
149 d->samples = (void *) samples; \
150 return 0; \
151 }
152
153 ✗ CHANGE_DELAY(u8, uint8_t, 0x80)
154 ✗ CHANGE_DELAY(s16, int16_t, 0)
155 ✗ CHANGE_DELAY(s32, int32_t, 0)
156 ✗ CHANGE_DELAY(flt, float, 0)
157 ✗ CHANGE_DELAY(dbl, double, 0)
158
159 2 static int parse_delays(char *p, char **saveptr, int64_t *result, AVFilterContext *ctx, int sample_rate) {
160 float delay, div;
161 int ret;
162 char *arg;
163 2 char type = 0;
164
165
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 2 if (!(arg = av_strtok(p, "|", saveptr)))
166 1 return 1;
167
168 1 ret = av_sscanf(arg, "%"SCNd64"%c", result, &type);
169
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 1 if (ret != 2 || type != 'S') {
170
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 1 div = type == 's' ? 1.0 : 1000.0;
171
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 1 if (av_sscanf(arg, "%f", &delay) != 1) {
172 ✗ av_log(ctx, AV_LOG_ERROR, "Invalid syntax for delay.\n");
173 ✗ return AVERROR(EINVAL);
174 }
175 1 *result = delay * sample_rate / div;
176 }
177
178
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 1 if (*result < 0) {
179 ✗ av_log(ctx, AV_LOG_ERROR, "Delay must be non negative number.\n");
180 ✗ return AVERROR(EINVAL);
181 }
182 1 return 0;
183 }
184
185 1 static int config_input(AVFilterLink *inlink)
186 {
187 1 AVFilterContext *ctx = inlink->dst;
188 1 AudioDelayContext *s = ctx->priv;
189 1 char *p, *saveptr = NULL;
190 int i;
191
192 1 s->next_pts = AV_NOPTS_VALUE;
193 1 s->chandelay = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->chandelay));
194
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 1 if (!s->chandelay)
195 ✗ return AVERROR(ENOMEM);
196 1 s->nb_delays = inlink->ch_layout.nb_channels;
197 1 s->block_align = av_get_bytes_per_sample(inlink->format);
198
199 1 p = s->delays;
200
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 2 for (i = 0; i < s->nb_delays; i++) {
201 2 ChanDelay *d = &s->chandelay[i];
202 int ret;
203
204 2 ret = parse_delays(p, &saveptr, &d->delay, ctx, inlink->sample_rate);
205
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 2 if (ret == 1)
206 1 break;
207
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 1 else if (ret < 0)
208 ✗ return ret;
209 1 p = NULL;
210 }
211
212
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 1 if (s->all && i) {
213 ✗ for (int j = i; j < s->nb_delays; j++)
214 ✗ s->chandelay[j].delay = s->chandelay[i-1].delay;
215 }
216
217 1 s->padding = s->chandelay[0].delay;
218
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 2 for (i = 1; i < s->nb_delays; i++) {
219 1 ChanDelay *d = &s->chandelay[i];
220
221 1 s->padding = FFMIN(s->padding, d->delay);
222 }
223
224
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 1 if (s->padding) {
225 ✗ for (i = 0; i < s->nb_delays; i++) {
226 ✗ ChanDelay *d = &s->chandelay[i];
227
228 ✗ d->delay -= s->padding;
229 }
230
231 ✗ s->offset = av_rescale_q(s->padding,
232 av_make_q(1, inlink->sample_rate),
233 inlink->time_base);
234 }
235
236
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 3 for (i = 0; i < s->nb_delays; i++) {
237 2 ChanDelay *d = &s->chandelay[i];
238
239
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 2 if (!d->delay)
240 1 continue;
241
242 if (d->delay > SIZE_MAX) {
243 av_log(ctx, AV_LOG_ERROR, "Requested delay is too big.\n");
244 return AVERROR(EINVAL);
245 }
246
247 1 d->samples = av_malloc_array(d->delay, s->block_align);
248
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 1 if (!d->samples)
249 ✗ return AVERROR(ENOMEM);
250 1 d->samples_size = d->delay * s->block_align;
251
252 1 s->max_delay = FFMAX(s->max_delay, d->delay);
253 }
254
255
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 1 switch (inlink->format) {
256 ✗ case AV_SAMPLE_FMT_U8P : s->delay_channel = delay_channel_u8p ;
257 ✗ s->resize_channel_samples = resize_samples_u8p; break;
258 1 case AV_SAMPLE_FMT_S16P: s->delay_channel = delay_channel_s16p;
259 1 s->resize_channel_samples = resize_samples_s16p; break;
260 ✗ case AV_SAMPLE_FMT_S32P: s->delay_channel = delay_channel_s32p;
261 ✗ s->resize_channel_samples = resize_samples_s32p; break;
262 ✗ case AV_SAMPLE_FMT_FLTP: s->delay_channel = delay_channel_fltp;
263 ✗ s->resize_channel_samples = resize_samples_fltp; break;
264 ✗ case AV_SAMPLE_FMT_DBLP: s->delay_channel = delay_channel_dblp;
265 ✗ s->resize_channel_samples = resize_samples_dblp; break;
266 }
267
268 1 return 0;
269 }
270
271 ✗ static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
272 char *res, int res_len, int flags)
273 {
274 ✗ int ret = AVERROR(ENOSYS);
275 ✗ AVFilterLink *inlink = ctx->inputs[0];
276 ✗ AudioDelayContext *s = ctx->priv;
277
278 ✗ if (!strcmp(cmd, "delays")) {
279 int64_t delay;
280 ✗ char *p, *saveptr = NULL;
281 ✗ int64_t all_delay = -1;
282 ✗ int64_t max_delay = 0;
283 ✗ char *args_cpy = av_strdup(args);
284 ✗ if (args_cpy == NULL) {
285 ✗ return AVERROR(ENOMEM);
286 }
287
288 ✗ ret = 0;
289 ✗ p = args_cpy;
290
291 ✗ if (!strncmp(args, "all:", 4)) {
292 ✗ p = &args_cpy[4];
293 ✗ ret = parse_delays(p, &saveptr, &all_delay, ctx, inlink->sample_rate);
294 ✗ if (ret == 1)
295 ✗ ret = AVERROR(EINVAL);
296 ✗ else if (ret == 0)
297 ✗ delay = all_delay;
298 }
299
300 ✗ if (!ret) {
301 ✗ for (int i = 0; i < s->nb_delays; i++) {
302 ✗ ChanDelay *d = &s->chandelay[i];
303
304 ✗ if (all_delay < 0) {
305 ✗ ret = parse_delays(p, &saveptr, &delay, ctx, inlink->sample_rate);
306 ✗ if (ret != 0) {
307 ✗ ret = 0;
308 ✗ break;
309 }
310 ✗ p = NULL;
311 }
312
313 ✗ ret = s->resize_channel_samples(d, delay);
314 ✗ if (ret)
315 ✗ break;
316 ✗ max_delay = FFMAX(max_delay, d->delay);
317 }
318 ✗ s->max_delay = FFMAX(s->max_delay, max_delay);
319 }
320 ✗ av_freep(&args_cpy);
321 }
322 ✗ return ret;
323 }
324
325 66 static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
326 {
327 66 AVFilterContext *ctx = inlink->dst;
328 66 AVFilterLink *outlink = ctx->outputs[0];
329 66 AudioDelayContext *s = ctx->priv;
330 AVFrame *out_frame;
331 int i;
332
333
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 66 if (ctx->is_disabled || !s->delays) {
334 ✗ s->input = NULL;
335 ✗ return ff_filter_frame(outlink, frame);
336 }
337
338 66 s->next_pts = av_rescale_q(frame->pts, inlink->time_base, outlink->time_base);
339
340 66 out_frame = ff_get_audio_buffer(outlink, frame->nb_samples);
341
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 66 if (!out_frame) {
342 ✗ s->input = NULL;
343 ✗ av_frame_free(&frame);
344 ✗ return AVERROR(ENOMEM);
345 }
346 66 av_frame_copy_props(out_frame, frame);
347
348
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 198 for (i = 0; i < s->nb_delays; i++) {
349 132 ChanDelay *d = &s->chandelay[i];
350 132 const uint8_t *src = frame->extended_data[i];
351 132 uint8_t *dst = out_frame->extended_data[i];
352
353
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 132 if (!d->delay)
354 66 memcpy(dst, src, frame->nb_samples * s->block_align);
355 else
356 66 s->delay_channel(d, frame->nb_samples, src, dst);
357 }
358
359 66 out_frame->pts = s->next_pts + s->offset;
360 66 out_frame->duration = av_rescale_q(out_frame->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
361 66 s->next_pts += out_frame->duration;
362 66 av_frame_free(&frame);
363 66 s->input = NULL;
364 66 return ff_filter_frame(outlink, out_frame);
365 }
366
367 132 static int activate(AVFilterContext *ctx)
368 {
369 132 AVFilterLink *inlink = ctx->inputs[0];
370 132 AVFilterLink *outlink = ctx->outputs[0];
371 132 AudioDelayContext *s = ctx->priv;
372 132 AVFrame *frame = NULL;
373 int ret, status;
374 int64_t pts;
375
376
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 132 FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
377
378
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 132 if (!s->input) {
379 132 ret = ff_inlink_consume_frame(inlink, &s->input);
380
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 132 if (ret < 0)
381 ✗ return ret;
382 }
383
384
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 132 if (ff_inlink_acknowledge_status(inlink, &status, &pts)) {
385
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 2 if (status == AVERROR_EOF)
386 2 s->eof = 1;
387 }
388
389
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 132 if (s->next_pts == AV_NOPTS_VALUE && pts != AV_NOPTS_VALUE)
390 1 s->next_pts = av_rescale_q(pts, inlink->time_base, outlink->time_base);
391
392
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 132 if (s->padding) {
393 ✗ int nb_samples = FFMIN(s->padding, 2048);
394
395 ✗ frame = ff_get_audio_buffer(outlink, nb_samples);
396 ✗ if (!frame)
397 ✗ return AVERROR(ENOMEM);
398 ✗ s->padding -= nb_samples;
399
400 ✗ av_samples_set_silence(frame->extended_data, 0,
401 frame->nb_samples,
402 outlink->ch_layout.nb_channels,
403 ✗ frame->format);
404
405 ✗ frame->duration = av_rescale_q(frame->nb_samples,
406 ✗ (AVRational){1, outlink->sample_rate},
407 outlink->time_base);
408 ✗ frame->pts = s->next_pts;
409 ✗ s->next_pts += frame->duration;
410
411 ✗ return ff_filter_frame(outlink, frame);
412 }
413
414
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 132 if (s->input)
415 65 return filter_frame(inlink, s->input);
416
417
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 67 if (s->eof && s->max_delay) {
418 1 int nb_samples = FFMIN(s->max_delay, 2048);
419
420 1 frame = ff_get_audio_buffer(outlink, nb_samples);
421
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 1 if (!frame)
422 ✗ return AVERROR(ENOMEM);
423 1 s->max_delay -= nb_samples;
424
425 1 av_samples_set_silence(frame->extended_data, 0,
426 frame->nb_samples,
427 outlink->ch_layout.nb_channels,
428 1 frame->format);
429
430 1 frame->duration = av_rescale_q(frame->nb_samples,
431 1 (AVRational){1, outlink->sample_rate},
432 outlink->time_base);
433 1 frame->pts = s->next_pts;
434 1 s->next_pts += frame->duration;
435 1 return filter_frame(inlink, frame);
436 }
437
438
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 66 if (s->eof && s->max_delay == 0) {
439 1 ff_outlink_set_status(outlink, AVERROR_EOF, s->next_pts);
440 1 return 0;
441 }
442
443
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 65 if (!s->eof)
444
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 65 FF_FILTER_FORWARD_WANTED(outlink, inlink);
445
446 ✗ return FFERROR_NOT_READY;
447 }
448
449 2 static av_cold void uninit(AVFilterContext *ctx)
450 {
451 2 AudioDelayContext *s = ctx->priv;
452
453
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 2 if (s->chandelay) {
454
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 3 for (int i = 0; i < s->nb_delays; i++)
455 2 av_freep(&s->chandelay[i].samples);
456 }
457 2 av_freep(&s->chandelay);
458 2 }
459
460 static const AVFilterPad adelay_inputs[] = {
461 {
462 .name = "default",
463 .type = AVMEDIA_TYPE_AUDIO,
464 .config_props = config_input,
465 },
466 };
467
468 const FFFilter ff_af_adelay = {
469 .p.name = "adelay",
470 .p.description = NULL_IF_CONFIG_SMALL("Delay one or more audio channels."),
471 .p.priv_class = &adelay_class,
472 .p.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
473 .priv_size = sizeof(AudioDelayContext),
474 .activate = activate,
475 .uninit = uninit,
476 FILTER_INPUTS(adelay_inputs),
477 FILTER_OUTPUTS(ff_audio_default_filterpad),
478 FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_U8P, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S32P,
479 AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP),
480 .process_command = process_command,
481 };
482