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af_acrossover.c
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1/*
2 * This file is part of FFmpeg.
3 *
4 * FFmpeg is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Lesser General Public
6 * License as published by the Free Software Foundation; either
7 * version 2.1 of the License, or (at your option) any later version.
8 *
9 * FFmpeg is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Lesser General Public License for more details.
13 *
14 * You should have received a copy of the GNU Lesser General Public
15 * License along with FFmpeg; if not, write to the Free Software
16 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
17 */
18
19/**
20 * @file
21 * Crossover filter
22 *
23 * Split an audio stream into several bands.
24 */
25
26#include "libavutil/attributes.h"
27#include "libavutil/avstring.h"
28#include "libavutil/channel_layout.h"
29#include "libavutil/float_dsp.h"
30#include "libavutil/internal.h"
31#include "libavutil/mem.h"
32#include "libavutil/opt.h"
33
34#include "audio.h"
35#include "avfilter.h"
36#include "filters.h"
37#include "formats.h"
38
39#define MAX_SPLITS 16
40#define MAX_BANDS MAX_SPLITS + 1
41
42#define B0 0
43#define B1 1
44#define B2 2
45#define A1 3
46#define A2 4
47
48typedef struct BiquadCoeffs {
49 double cd[5];
50 float cf[5];
51} BiquadCoeffs;
52
83
84#define OFFSET(x) offsetof(AudioCrossoverContext, x)
85#define AF AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM
86
87static const AVOption acrossover_options[] = {
88 { "split", "set split frequencies", OFFSET(splits_str), AV_OPT_TYPE_STRING, {.str="500"}, 0, 0, AF },
89 { "order", "set filter order", OFFSET(order_opt), AV_OPT_TYPE_INT, {.i64=1}, 0, 9, AF, .unit = "m" },
90 { "2nd", "2nd order (12 dB/8ve)", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, .unit = "m" },
91 { "4th", "4th order (24 dB/8ve)", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, .unit = "m" },
92 { "6th", "6th order (36 dB/8ve)", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, .unit = "m" },
93 { "8th", "8th order (48 dB/8ve)", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, .unit = "m" },
94 { "10th", "10th order (60 dB/8ve)",0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, AF, .unit = "m" },
95 { "12th", "12th order (72 dB/8ve)",0, AV_OPT_TYPE_CONST, {.i64=5}, 0, 0, AF, .unit = "m" },
96 { "14th", "14th order (84 dB/8ve)",0, AV_OPT_TYPE_CONST, {.i64=6}, 0, 0, AF, .unit = "m" },
97 { "16th", "16th order (96 dB/8ve)",0, AV_OPT_TYPE_CONST, {.i64=7}, 0, 0, AF, .unit = "m" },
98 { "18th", "18th order (108 dB/8ve)",0, AV_OPT_TYPE_CONST, {.i64=8}, 0, 0, AF, .unit = "m" },
99 { "20th", "20th order (120 dB/8ve)",0, AV_OPT_TYPE_CONST, {.i64=9}, 0, 0, AF, .unit = "m" },
100 { "level", "set input gain", OFFSET(level_in), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
101 { "gain", "set output bands gain", OFFSET(gains_str), AV_OPT_TYPE_STRING, {.str="1.f"}, 0, 0, AF },
102 { "precision", "set processing precision", OFFSET(precision), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, AF, .unit = "precision" },
103 { "auto", "set auto processing precision", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, .unit = "precision" },
104 { "float", "set single-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, .unit = "precision" },
105 { "double","set double-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, .unit = "precision" },
106 { NULL }
107};
108
109AVFILTER_DEFINE_CLASS(acrossover);
110
111static int query_formats(const AVFilterContext *ctx,
112 AVFilterFormatsConfig **cfg_in,
113 AVFilterFormatsConfig **cfg_out)
114{
115 const AudioCrossoverContext *s = ctx->priv;
116 static const enum AVSampleFormat auto_sample_fmts[] = {
117 AV_SAMPLE_FMT_FLTP,
118 AV_SAMPLE_FMT_DBLP,
119 AV_SAMPLE_FMT_NONE
120 };
121 enum AVSampleFormat sample_fmts[] = {
122 AV_SAMPLE_FMT_FLTP,
123 AV_SAMPLE_FMT_NONE
124 };
125 const enum AVSampleFormat *sample_fmts_list = sample_fmts;
126 int ret;
127
128 switch (s->precision) {
129 case 0:
130 sample_fmts_list = auto_sample_fmts;
131 break;
132 case 1:
133 sample_fmts[0] = AV_SAMPLE_FMT_FLTP;
134 break;
135 case 2:
136 sample_fmts[0] = AV_SAMPLE_FMT_DBLP;
137 break;
138 default:
139 break;
140 }
141 ret = ff_set_sample_formats_from_list2(ctx, cfg_in, cfg_out, sample_fmts_list);
142 if (ret < 0)
143 return ret;
144
145 return 0;
146}
147
148static int parse_gains(AVFilterContext *ctx)
149{
150 AudioCrossoverContext *s = ctx->priv;
151 char *p, *arg, *saveptr = NULL;
152 int i, ret = 0;
153
154 saveptr = NULL;
155 p = s->gains_str;
156 for (i = 0; i < MAX_BANDS; i++) {
157 float gain;
158 char c[3] = { 0 };
159
160 if (!(arg = av_strtok(p, " |", &saveptr)))
161 break;
162
163 p = NULL;
164
165 if (av_sscanf(arg, "%f%2s", &gain, c) < 1) {
166 av_log(ctx, AV_LOG_ERROR, "Invalid syntax for gain[%d].\n", i);
167 ret = AVERROR(EINVAL);
168 break;
169 }
170
171 if (c[0] == 'd' && c[1] == 'B')
172 s->gains[i] = expf(gain * M_LN10 / 20.f);
173 else
174 s->gains[i] = gain;
175 if (!isfinite(s->gains[i])) {
176 av_log(ctx, AV_LOG_ERROR, "Gain %f must be finite.\n", gain);
177 return AVERROR(EINVAL);
178 }
179 }
180
181 for (; i < MAX_BANDS; i++)
182 s->gains[i] = 1.f;
183
184 return ret;
185}
186
187static av_cold int init(AVFilterContext *ctx)
188{
189 AudioCrossoverContext *s = ctx->priv;
190 char *p, *arg, *saveptr = NULL;
191 int i, ret = 0;
192
193 s->fdsp = avpriv_float_dsp_alloc(0);
194 if (!s->fdsp)
195 return AVERROR(ENOMEM);
196
197 p = s->splits_str;
198 for (i = 0; i < MAX_SPLITS; i++) {
199 float freq;
200
201 if (!(arg = av_strtok(p, " |", &saveptr)))
202 break;
203
204 p = NULL;
205
206 if (av_sscanf(arg, "%f", &freq) != 1) {
207 av_log(ctx, AV_LOG_ERROR, "Invalid syntax for frequency[%d].\n", i);
208 return AVERROR(EINVAL);
209 }
210 if (!isfinite(freq) || freq <= 0) {
211 av_log(ctx, AV_LOG_ERROR, "Frequency %f must be a positive finite number.\n", freq);
212 return AVERROR(EINVAL);
213 }
214
215 if (i > 0 && freq <= s->splits[i-1]) {
216 av_log(ctx, AV_LOG_ERROR, "Frequency %f must be in increasing order.\n", freq);
217 return AVERROR(EINVAL);
218 }
219
220 s->splits[i] = freq;
221 }
222
223 s->nb_splits = i;
224
225 ret = parse_gains(ctx);
226 if (ret < 0)
227 return ret;
228
229 for (i = 0; i <= s->nb_splits; i++) {
230 AVFilterPad pad = { 0 };
231 char *name;
232
233 pad.type = AVMEDIA_TYPE_AUDIO;
234 name = av_asprintf("out%d", ctx->nb_outputs);
235 if (!name)
236 return AVERROR(ENOMEM);
237 pad.name = name;
238
239 if ((ret = ff_append_outpad_free_name(ctx, &pad)) < 0)
240 return ret;
241 }
242
243 return ret;
244}
245
246static void set_lp(BiquadCoeffs *b, double fc, double q, double sr)
247{
248 double omega = 2. * M_PI * fc / sr;
249 double cosine = cos(omega);
250 double alpha = sin(omega) / (2. * q);
251
252 double b0 = (1. - cosine) / 2.;
253 double b1 = 1. - cosine;
254 double b2 = (1. - cosine) / 2.;
255 double a0 = 1. + alpha;
256 double a1 = -2. * cosine;
257 double a2 = 1. - alpha;
258
259 b->cd[B0] = b0 / a0;
260 b->cd[B1] = b1 / a0;
261 b->cd[B2] = b2 / a0;
262 b->cd[A1] = -a1 / a0;
263 b->cd[A2] = -a2 / a0;
264
265 b->cf[B0] = b->cd[B0];
266 b->cf[B1] = b->cd[B1];
267 b->cf[B2] = b->cd[B2];
268 b->cf[A1] = b->cd[A1];
269 b->cf[A2] = b->cd[A2];
270}
271
272static void set_hp(BiquadCoeffs *b, double fc, double q, double sr)
273{
274 double omega = 2. * M_PI * fc / sr;
275 double cosine = cos(omega);
276 double alpha = sin(omega) / (2. * q);
277
278 double b0 = (1. + cosine) / 2.;
279 double b1 = -1. - cosine;
280 double b2 = (1. + cosine) / 2.;
281 double a0 = 1. + alpha;
282 double a1 = -2. * cosine;
283 double a2 = 1. - alpha;
284
285 b->cd[B0] = b0 / a0;
286 b->cd[B1] = b1 / a0;
287 b->cd[B2] = b2 / a0;
288 b->cd[A1] = -a1 / a0;
289 b->cd[A2] = -a2 / a0;
290
291 b->cf[B0] = b->cd[B0];
292 b->cf[B1] = b->cd[B1];
293 b->cf[B2] = b->cd[B2];
294 b->cf[A1] = b->cd[A1];
295 b->cf[A2] = b->cd[A2];
296}
297
298static void set_ap(BiquadCoeffs *b, double fc, double q, double sr)
299{
300 double omega = 2. * M_PI * fc / sr;
301 double cosine = cos(omega);
302 double alpha = sin(omega) / (2. * q);
303
304 double a0 = 1. + alpha;
305 double a1 = -2. * cosine;
306 double a2 = 1. - alpha;
307 double b0 = a2;
308 double b1 = a1;
309 double b2 = a0;
310
311 b->cd[B0] = b0 / a0;
312 b->cd[B1] = b1 / a0;
313 b->cd[B2] = b2 / a0;
314 b->cd[A1] = -a1 / a0;
315 b->cd[A2] = -a2 / a0;
316
317 b->cf[B0] = b->cd[B0];
318 b->cf[B1] = b->cd[B1];
319 b->cf[B2] = b->cd[B2];
320 b->cf[A1] = b->cd[A1];
321 b->cf[A2] = b->cd[A2];
322}
323
324static void set_ap1(BiquadCoeffs *b, double fc, double sr)
325{
326 double omega = 2. * M_PI * fc / sr;
327
328 b->cd[A1] = exp(-omega);
329 b->cd[A2] = 0.;
330 b->cd[B0] = -b->cd[A1];
331 b->cd[B1] = 1.;
332 b->cd[B2] = 0.;
333
334 b->cf[B0] = b->cd[B0];
335 b->cf[B1] = b->cd[B1];
336 b->cf[B2] = b->cd[B2];
337 b->cf[A1] = b->cd[A1];
338 b->cf[A2] = b->cd[A2];
339}
340
341static void calc_q_factors(int order, double *q)
342{
343 double n = order / 2.;
344
345 for (int i = 0; i < n / 2; i++)
346 q[i] = 1. / (-2. * cos(M_PI * (2. * (i + 1) + n - 1.) / (2. * n)));
347}
348
349#define BIQUAD_PROCESS(name, type) \
350static void biquad_process_## name(const type *const c, \
351 type *b, \
352 type *dst, const type *src, \
353 int nb_samples) \
354{ \
355 const type b0 = c[B0]; \
356 const type b1 = c[B1]; \
357 const type b2 = c[B2]; \
358 const type a1 = c[A1]; \
359 const type a2 = c[A2]; \
360 type z1 = b[0]; \
361 type z2 = b[1]; \
362 \
363 for (int n = 0; n + 1 < nb_samples; n++) { \
364 type in = src[n]; \
365 type out; \
366 \
367 out = in * b0 + z1; \
368 z1 = b1 * in + z2 + a1 * out; \
369 z2 = b2 * in + a2 * out; \
370 dst[n] = out; \
371 \
372 n++; \
373 in = src[n]; \
374 out = in * b0 + z1; \
375 z1 = b1 * in + z2 + a1 * out; \
376 z2 = b2 * in + a2 * out; \
377 dst[n] = out; \
378 } \
379 \
380 if (nb_samples & 1) { \
381 const int n = nb_samples - 1; \
382 const type in = src[n]; \
383 type out; \
384 \
385 out = in * b0 + z1; \
386 z1 = b1 * in + z2 + a1 * out; \
387 z2 = b2 * in + a2 * out; \
388 dst[n] = out; \
389 } \
390 \
391 b[0] = z1; \
392 b[1] = z2; \
393}
394
395BIQUAD_PROCESS(fltp, float)
396BIQUAD_PROCESS(dblp, double)
397
398#define XOVER_PROCESS(name, type, one, ff) \
399static int filter_channels_## name(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) \
400{ \
401 AudioCrossoverContext *s = ctx->priv; \
402 AVFrame *in = arg; \
403 AVFrame **frames = s->frames; \
404 const int start = ff_slice_pos(in->ch_layout.nb_channels, jobnr, nb_jobs); \
405 const int end = ff_slice_pos(in->ch_layout.nb_channels, jobnr + 1, nb_jobs); \
406 const int nb_samples = in->nb_samples; \
407 const int nb_outs = ctx->nb_outputs; \
408 const int first_order = s->first_order; \
409 \
410 for (int ch = start; ch < end; ch++) { \
411 const type *src = (const type *)in->extended_data[ch]; \
412 type *xover = (type *)s->xover->extended_data[ch]; \
413 \
414 s->fdsp->vector_## ff ##mul_scalar((type *)frames[0]->extended_data[ch], src, \
415 s->level_in, FFALIGN(nb_samples, sizeof(type))); \
416 \
417 for (int band = 0; band < nb_outs; band++) { \
418 for (int f = 0; band + 1 < nb_outs && f < s->filter_count; f++) { \
419 const type *prv = (const type *)frames[band]->extended_data[ch]; \
420 type *dst = (type *)frames[band + 1]->extended_data[ch]; \
421 const type *hsrc = f == 0 ? prv : dst; \
422 type *hp = xover + nb_outs * 20 + band * 20 + f * 2; \
423 const type *const hpc = (type *)&s->hp[band][f].c ## ff; \
424 \
425 biquad_process_## name(hpc, hp, dst, hsrc, nb_samples); \
426 } \
427 \
428 for (int f = 0; band + 1 < nb_outs && f < s->filter_count; f++) { \
429 type *dst = (type *)frames[band]->extended_data[ch]; \
430 const type *lsrc = dst; \
431 type *lp = xover + band * 20 + f * 2; \
432 const type *const lpc = (type *)&s->lp[band][f].c ## ff; \
433 \
434 biquad_process_## name(lpc, lp, dst, lsrc, nb_samples); \
435 } \
436 \
437 for (int aband = band + 1; aband + 1 < nb_outs; aband++) { \
438 if (first_order) { \
439 const type *asrc = (const type *)frames[band]->extended_data[ch]; \
440 type *dst = (type *)frames[band]->extended_data[ch]; \
441 type *ap = xover + nb_outs * 40 + (aband * nb_outs + band) * 20; \
442 const type *const apc = (type *)&s->ap[aband][0].c ## ff; \
443 \
444 biquad_process_## name(apc, ap, dst, asrc, nb_samples); \
445 } \
446 \
447 for (int f = first_order; f < s->ap_filter_count; f++) { \
448 const type *asrc = (const type *)frames[band]->extended_data[ch]; \
449 type *dst = (type *)frames[band]->extended_data[ch]; \
450 type *ap = xover + nb_outs * 40 + (aband * nb_outs + band) * 20 + f * 2;\
451 const type *const apc = (type *)&s->ap[aband][f].c ## ff; \
452 \
453 biquad_process_## name(apc, ap, dst, asrc, nb_samples); \
454 } \
455 } \
456 } \
457 \
458 for (int band = 0; band < nb_outs; band++) { \
459 const type gain = s->gains[band] * ((band & 1 && first_order) ? -one : one); \
460 type *dst = (type *)frames[band]->extended_data[ch]; \
461 \
462 s->fdsp->vector_## ff ##mul_scalar(dst, dst, gain, \
463 FFALIGN(nb_samples, sizeof(type))); \
464 } \
465 } \
466 \
467 return 0; \
468}
469
470XOVER_PROCESS(fltp, float, 1.f, f)
471XOVER_PROCESS(dblp, double, 1.0, d)
472
473static int config_input(AVFilterLink *inlink)
474{
475 AVFilterContext *ctx = inlink->dst;
476 AudioCrossoverContext *s = ctx->priv;
477 int sample_rate = inlink->sample_rate;
478 double q[16];
479
480 s->order = (s->order_opt + 1) * 2;
481 s->filter_count = s->order / 2;
482 s->first_order = s->filter_count & 1;
483 s->ap_filter_count = s->filter_count / 2 + s->first_order;
484 calc_q_factors(s->order, q);
485
486 for (int band = 0; band <= s->nb_splits; band++) {
487 if (s->first_order) {
488 set_lp(&s->lp[band][0], s->splits[band], 0.5, sample_rate);
489 set_hp(&s->hp[band][0], s->splits[band], 0.5, sample_rate);
490 }
491
492 for (int n = s->first_order; n < s->filter_count; n++) {
493 const int idx = s->filter_count / 2 - ((n + s->first_order) / 2 - s->first_order) - 1;
494
495 set_lp(&s->lp[band][n], s->splits[band], q[idx], sample_rate);
496 set_hp(&s->hp[band][n], s->splits[band], q[idx], sample_rate);
497 }
498
499 if (s->first_order)
500 set_ap1(&s->ap[band][0], s->splits[band], sample_rate);
501
502 for (int n = s->first_order; n < s->ap_filter_count; n++) {
503 const int idx = (s->filter_count / 2 - ((n * 2 + s->first_order) / 2 - s->first_order) - 1);
504
505 set_ap(&s->ap[band][n], s->splits[band], q[idx], sample_rate);
506 }
507 }
508
509 switch (inlink->format) {
510 case AV_SAMPLE_FMT_FLTP: s->filter_channels = filter_channels_fltp; break;
511 case AV_SAMPLE_FMT_DBLP: s->filter_channels = filter_channels_dblp; break;
512 default: return AVERROR_BUG;
513 }
514
515 s->xover = ff_get_audio_buffer(inlink, 2 * (ctx->nb_outputs * 10 + ctx->nb_outputs * 10 +
516 ctx->nb_outputs * ctx->nb_outputs * 10));
517 if (!s->xover)
518 return AVERROR(ENOMEM);
519
520 return 0;
521}
522
523static int filter_frame(AVFilterLink *inlink, AVFrame *in)
524{
525 AVFilterContext *ctx = inlink->dst;
526 AudioCrossoverContext *s = ctx->priv;
527 AVFrame **frames = s->frames;
528 int ret = 0;
529
530 for (int i = 0; i < ctx->nb_outputs; i++) {
531 frames[i] = ff_get_audio_buffer(ctx->outputs[i], in->nb_samples);
532 if (!frames[i]) {
533 ret = AVERROR(ENOMEM);
534 break;
535 }
536
537 frames[i]->pts = in->pts;
538 }
539
540 if (ret < 0)
541 goto fail;
542
543 ff_filter_execute(ctx, s->filter_channels, in, NULL,
544 FFMIN(inlink->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx)));
545
546 for (int i = 0; i < ctx->nb_outputs; i++) {
547 if (ff_outlink_get_status(ctx->outputs[i])) {
548 av_frame_free(&frames[i]);
549 continue;
550 }
551
552 ret = ff_filter_frame(ctx->outputs[i], frames[i]);
553 frames[i] = NULL;
554 if (ret < 0)
555 break;
556 }
557
558fail:
559 for (int i = 0; i < ctx->nb_outputs; i++)
560 av_frame_free(&frames[i]);
561
562 return ret;
563}
564
565static int activate(AVFilterContext *ctx)
566{
567 AVFilterLink *inlink = ctx->inputs[0];
568 int status, ret;
569 AVFrame *in;
570 int64_t pts;
571
572 for (int i = 0; i < ctx->nb_outputs; i++) {
573 FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[i], ctx);
574 }
575
576 ret = ff_inlink_consume_frame(inlink, &in);
577 if (ret < 0)
578 return ret;
579 if (ret > 0) {
580 ret = filter_frame(inlink, in);
581 av_frame_free(&in);
582 if (ret < 0)
583 return ret;
584 }
585
586 if (ff_inlink_acknowledge_status(inlink, &status, &pts)) {
587 for (int i = 0; i < ctx->nb_outputs; i++) {
588 if (ff_outlink_get_status(ctx->outputs[i]))
589 continue;
590 ff_outlink_set_status(ctx->outputs[i], status, pts);
591 }
592 return 0;
593 }
594
595 FF_FILTER_FORWARD_WANTED_ANY(ctx, inlink);
596
597 return FFERROR_NOT_READY;
598}
599
600static av_cold void uninit(AVFilterContext *ctx)
601{
602 AudioCrossoverContext *s = ctx->priv;
603
604 av_freep(&s->fdsp);
605 av_frame_free(&s->xover);
606}
607
608static const AVFilterPad inputs[] = {
609 {
610 .name = "default",
611 .type = AVMEDIA_TYPE_AUDIO,
612 .config_props = config_input,
613 },
614};
615
616const FFFilter ff_af_acrossover = {
617 .p.name = "acrossover",
618 .p.description = NULL_IF_CONFIG_SMALL("Split audio into per-bands streams."),
619 .p.priv_class = &acrossover_class,
620 .p.outputs = NULL,
621 .p.flags = AVFILTER_FLAG_DYNAMIC_OUTPUTS |
622 AVFILTER_FLAG_SLICE_THREADS,
623 .priv_size = sizeof(AudioCrossoverContext),
624 .init = init,
625 .activate = activate,
626 .uninit = uninit,
627 FILTER_INPUTS(inputs),
628 FILTER_QUERY_FUNC2(query_formats),
629};
sample_fmts
static enum AVSampleFormat sample_fmts[]
Definition adpcmenc.c:933
query_formats
static int query_formats(const AVFilterContext *ctx, AVFilterFormatsConfig **cfg_in, AVFilterFormatsConfig **cfg_out)
Definition aeval.c:246
inputs
static const AVFilterPad inputs[]
Definition af_aap.c:299
config_input
static int config_input(AVFilterLink *inlink)
Definition af_acontrast.c:121
MAX_BANDS
#define MAX_BANDS
Definition af_acrossover.c:40
MAX_SPLITS
#define MAX_SPLITS
Definition af_acrossover.c:39
set_hp
static void set_hp(BiquadCoeffs *b, double fc, double q, double sr)
Definition af_acrossover.c:272
set_lp
static void set_lp(BiquadCoeffs *b, double fc, double q, double sr)
Definition af_acrossover.c:246
set_ap
static void set_ap(BiquadCoeffs *b, double fc, double q, double sr)
Definition af_acrossover.c:298
AF
#define AF
Definition af_acrossover.c:85
filter_frame
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
Definition af_acrossover.c:523
acrossover_options
static const AVOption acrossover_options[]
Definition af_acrossover.c:87
set_ap1
static void set_ap1(BiquadCoeffs *b, double fc, double sr)
Definition af_acrossover.c:324
query_formats
static int query_formats(const AVFilterContext *ctx, AVFilterFormatsConfig **cfg_in, AVFilterFormatsConfig **cfg_out)
Definition af_acrossover.c:111
activate
static int activate(AVFilterContext *ctx)
Definition af_acrossover.c:565
uninit
static av_cold void uninit(AVFilterContext *ctx)
Definition af_acrossover.c:600
ff_af_acrossover
const FFFilter ff_af_acrossover
Definition af_acrossover.c:616
XOVER_PROCESS
#define XOVER_PROCESS(name, type, one, ff)
Definition af_acrossover.c:398
OFFSET
#define OFFSET(x)
Definition af_acrossover.c:84
BIQUAD_PROCESS
#define BIQUAD_PROCESS(name, type)
Definition af_acrossover.c:349
calc_q_factors
static void calc_q_factors(int order, double *q)
Definition af_acrossover.c:341
parse_gains
static int parse_gains(AVFilterContext *ctx)
Definition af_acrossover.c:148
frames
static int frames
Definition api-movenc-test.c:67
ctx
static AVFormatContext * ctx
Definition api-movenc-test.c:49
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
ff_outlink_get_status
int ff_outlink_get_status(AVFilterLink *link)
Get the status on an output link.
Definition avfilter.c:1648
ff_inlink_acknowledge_status
int ff_inlink_acknowledge_status(AVFilterLink *link, int *rstatus, int64_t *rpts)
Test and acknowledge the change of status on the link.
Definition avfilter.c:1467
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_execute
int ff_filter_execute(AVFilterContext *ctx, avfilter_action_func *func, void *arg, int *ret, int nb_jobs)
Definition avfilter.c:1696
ff_append_outpad_free_name
int ff_append_outpad_free_name(AVFilterContext *f, AVFilterPad *p)
Definition avfilter.c:143
ff_filter_get_nb_threads
int ff_filter_get_nb_threads(AVFilterContext *ctx)
Get number of threads for current filter instance.
Definition avfilter.c:846
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
avfilter.h
Main libavfilter public API header.
av_sscanf
int av_sscanf(const char *string, const char *format,...)
Definition avsscanf.c:962
av_asprintf
char * av_asprintf(const char *fmt,...)
Definition avstring.c:115
A2
#define A2
Definition binkdsp.c:31
A1
#define A1
Definition binkdsp.c:30
i
#define i(width, name, range_min, range_max)
Definition cbs_h264.c:63
f
#define f(width, name)
Definition cbs_vp8.c:236
s
#define s(width, name)
Definition cbs_vp9.c:198
channel_layout.h
Public libavutil channel layout APIs header.
NULL
#define NULL
Definition coverity.c:32
int64_t
long long int64_t
Definition coverity.c:34
fc
static const uint16_t fc[]
Definition dcaenc.h:43
init
int(* init)(AVBSFContext *ctx)
Definition dts2pts.c:608
exp
int8_t exp
Definition eval.c:76
B2
#define B2
Definition faandct.c:43
B1
#define B1
Definition faandct.c:42
B0
#define B0
Definition faandct.c:41
cosine
static av_unused double cosine(double x, const double *params)
Definition filters.c:330
float_dsp.h
ff_set_sample_formats_from_list2
int ff_set_sample_formats_from_list2(const AVFilterContext *ctx, AVFilterFormatsConfig **cfg_in, AVFilterFormatsConfig **cfg_out, const enum AVSampleFormat *fmts)
Definition formats.c:1154
fail
#define fail
Definition test.h:479
AV_OPT_TYPE_CONST
@ AV_OPT_TYPE_CONST
Special option type for declaring named constants.
Definition opt.h:298
AV_OPT_TYPE_INT
@ AV_OPT_TYPE_INT
Underlying C type is int.
Definition opt.h:258
AV_OPT_TYPE_FLOAT
@ AV_OPT_TYPE_FLOAT
Underlying C type is float.
Definition opt.h:270
AV_OPT_TYPE_STRING
@ AV_OPT_TYPE_STRING
Underlying C type is a uint8_t* that is either NULL or points to a C string allocated with the av_mal...
Definition opt.h:275
AVFILTER_FLAG_DYNAMIC_OUTPUTS
#define AVFILTER_FLAG_DYNAMIC_OUTPUTS
The number of the filter outputs is not determined just by AVFilter.outputs.
Definition avfilter.h:161
AVFILTER_FLAG_SLICE_THREADS
#define AVFILTER_FLAG_SLICE_THREADS
The filter supports multithreading by splitting frames into multiple parts and processing them concur...
Definition avfilter.h:166
AVERROR_BUG
#define AVERROR_BUG
Internal bug, also see AVERROR_BUG2.
Definition error.h:52
AVERROR
#define AVERROR(e)
Definition error.h:45
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_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition log.h:210
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_NONE
@ AV_SAMPLE_FMT_NONE
Definition samplefmt.h:56
AV_SAMPLE_FMT_DBLP
@ AV_SAMPLE_FMT_DBLP
double, planar
Definition samplefmt.h:67
av_strtok
char * av_strtok(char *s, const char *delim, char **saveptr)
Split the string into several tokens which can be accessed by successive calls to av_strtok().
Definition avstring.c:179
alpha
static const int16_t alpha[]
Definition ilbcdata.h:55
b
#define b
Definition input.c:43
uninit
static av_cold void uninit(AVBitStreamFilterContext *ctx)
Definition lcevc_merge.c:135
activate
static int activate(AVBitStreamFilterContext *ctx)
Definition lcevc_merge.c:144
arg
const char * arg
Definition jacosubdec.c:65
FILTER_INPUTS
#define FILTER_INPUTS(array)
Definition filters.h:264
FF_FILTER_FORWARD_WANTED_ANY
#define FF_FILTER_FORWARD_WANTED_ANY(filter, inlink)
Forward the frame_wanted_out flag from any of the output links to an input link.
Definition filters.h:705
ff_outlink_set_status
static void ff_outlink_set_status(AVFilterLink *link, int status, int64_t pts)
Set the status field of a link from the source filter.
Definition filters.h:629
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
FILTER_QUERY_FUNC2
#define FILTER_QUERY_FUNC2(func)
Definition filters.h:241
attributes.h
Macro definitions for various function/variable attributes.
av_cold
#define av_cold
Definition attributes.h:117
avpriv_float_dsp_alloc
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition float_dsp.c:135
internal.h
common internal API header
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
isfinite
#define isfinite(x)
Definition libm.h:361
expf
#define expf(x)
Definition libm.h:285
FFMIN
#define FFMIN(a, b)
Definition macros.h:49
M_LN10
#define M_LN10
Definition mathematics.h:49
M_PI
#define M_PI
Definition mathematics.h:67
mem.h
Memory handling functions.
opt.h
AVOptions.
name
const char * name
Definition qsvenc.c:142
AVChannelLayout::nb_channels
int nb_channels
Number of channels in this layout.
Definition channel_layout.h:338
AVClass
Describe the class of an AVClass context structure.
Definition log.h:76
AVFilterContext
An instance of a filter.
Definition avfilter.h:273
AVFilterFormatsConfig
Lists of formats / etc.
Definition avfilter.h:120
AVFilterPad
A filter pad used for either input or output.
Definition filters.h:40
AVFilterPad::type
enum AVMediaType type
AVFilterPad type.
Definition filters.h:51
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
AVOption
AVOption.
Definition opt.h:428
AudioCrossoverContext::lp
BiquadCoeffs lp[MAX_BANDS][20]
Definition af_acrossover.c:71
AudioCrossoverContext::hp
BiquadCoeffs hp[MAX_BANDS][20]
Definition af_acrossover.c:72
AudioCrossoverContext::filter_channels
int(* filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
Definition af_acrossover.c:79
AudioCrossoverContext::gains
float gains[MAX_BANDS]
Definition af_acrossover.c:69
AudioCrossoverContext::ap
BiquadCoeffs ap[MAX_BANDS][20]
Definition af_acrossover.c:73
AudioCrossoverContext::splits
float splits[MAX_SPLITS]
Definition af_acrossover.c:67
AudioCrossoverContext::frames
AVFrame * frames[MAX_BANDS]
Definition af_acrossover.c:77
AudioCrossoverContext::fdsp
AVFloatDSPContext * fdsp
Definition af_acrossover.c:81
BiquadCoeffs::cd
double cd[5]
Definition af_acrossover.c:49
av_freep
#define av_freep(p)
Definition tableprint_vlc.h:35
av_log
#define av_log(a,...)
Definition tableprint_vlc.h:27
pts
static int64_t pts
Definition transcode_aac.c:649
b1
static double b1(void *priv, double x, double y)
Definition vf_xfade.c:2034
a0
static double a0(void *priv, double x, double y)
Definition vf_xfade.c:2028
b2
static double b2(void *priv, double x, double y)
Definition vf_xfade.c:2035
b0
static double b0(void *priv, double x, double y)
Definition vf_xfade.c:2033
a2
static double a2(void *priv, double x, double y)
Definition vf_xfade.c:2030
a1
static double a1(void *priv, double x, double y)
Definition vf_xfade.c:2029
c
static double c[64]
Definition vsrc_mptestsrc.c:89
MAX_BANDS
#define MAX_BANDS
max number of scale factor bands
Definition wmalosslessdec.c:43
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