FFmpeg coverage

Directory: ../../../ffmpeg/
File: src/libavfilter/af_acrossover.c
Date: 2024-11-20 23:03:26
Exec Total Coverage
Lines: 0 250 0.0%
Functions: 0 16 0.0%
Branches: 0 157 0.0%
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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
48 typedef struct BiquadCoeffs {
49 double cd[5];
50 float cf[5];
51 } BiquadCoeffs;
52
53 typedef struct AudioCrossoverContext {
54 const AVClass *class;
55
56 char *splits_str;
57 char *gains_str;
58 int order_opt;
59 float level_in;
60 int precision;
61
62 int order;
63 int filter_count;
64 int first_order;
65 int ap_filter_count;
66 int nb_splits;
67 float splits[MAX_SPLITS];
68
69 float gains[MAX_BANDS];
70
71 BiquadCoeffs lp[MAX_BANDS][20];
72 BiquadCoeffs hp[MAX_BANDS][20];
73 BiquadCoeffs ap[MAX_BANDS][20];
74
75 AVFrame *xover;
76
77 AVFrame *frames[MAX_BANDS];
78
79 int (*filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs);
80
81 AVFloatDSPContext *fdsp;
82 } AudioCrossoverContext;
83
84 #define OFFSET(x) offsetof(AudioCrossoverContext, x)
85 #define AF AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM
86
87 static 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
109 AVFILTER_DEFINE_CLASS(acrossover);
110
111 ✗ static 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_common_formats_from_list2(ctx, cfg_in, cfg_out, sample_fmts_list);
142 ✗ if (ret < 0)
143 ✗ return ret;
144
145 ✗ return 0;
146 }
147
148 ✗ static 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 }
176
177 ✗ for (; i < MAX_BANDS; i++)
178 ✗ s->gains[i] = 1.f;
179
180 ✗ return ret;
181 }
182
183 ✗ static av_cold int init(AVFilterContext *ctx)
184 {
185 ✗ AudioCrossoverContext *s = ctx->priv;
186 ✗ char *p, *arg, *saveptr = NULL;
187 ✗ int i, ret = 0;
188
189 ✗ s->fdsp = avpriv_float_dsp_alloc(0);
190 ✗ if (!s->fdsp)
191 ✗ return AVERROR(ENOMEM);
192
193 ✗ p = s->splits_str;
194 ✗ for (i = 0; i < MAX_SPLITS; i++) {
195 float freq;
196
197 ✗ if (!(arg = av_strtok(p, " |", &saveptr)))
198 ✗ break;
199
200 ✗ p = NULL;
201
202 ✗ if (av_sscanf(arg, "%f", &freq) != 1) {
203 ✗ av_log(ctx, AV_LOG_ERROR, "Invalid syntax for frequency[%d].\n", i);
204 ✗ return AVERROR(EINVAL);
205 }
206 ✗ if (freq <= 0) {
207 ✗ av_log(ctx, AV_LOG_ERROR, "Frequency %f must be positive number.\n", freq);
208 ✗ return AVERROR(EINVAL);
209 }
210
211 ✗ if (i > 0 && freq <= s->splits[i-1]) {
212 ✗ av_log(ctx, AV_LOG_ERROR, "Frequency %f must be in increasing order.\n", freq);
213 ✗ return AVERROR(EINVAL);
214 }
215
216 ✗ s->splits[i] = freq;
217 }
218
219 ✗ s->nb_splits = i;
220
221 ✗ ret = parse_gains(ctx);
222 ✗ if (ret < 0)
223 ✗ return ret;
224
225 ✗ for (i = 0; i <= s->nb_splits; i++) {
226 ✗ AVFilterPad pad = { 0 };
227 char *name;
228
229 ✗ pad.type = AVMEDIA_TYPE_AUDIO;
230 ✗ name = av_asprintf("out%d", ctx->nb_outputs);
231 ✗ if (!name)
232 ✗ return AVERROR(ENOMEM);
233 ✗ pad.name = name;
234
235 ✗ if ((ret = ff_append_outpad_free_name(ctx, &pad)) < 0)
236 ✗ return ret;
237 }
238
239 ✗ return ret;
240 }
241
242 ✗ static void set_lp(BiquadCoeffs *b, double fc, double q, double sr)
243 {
244 ✗ double omega = 2. * M_PI * fc / sr;
245 ✗ double cosine = cos(omega);
246 ✗ double alpha = sin(omega) / (2. * q);
247
248 ✗ double b0 = (1. - cosine) / 2.;
249 ✗ double b1 = 1. - cosine;
250 ✗ double b2 = (1. - cosine) / 2.;
251 ✗ double a0 = 1. + alpha;
252 ✗ double a1 = -2. * cosine;
253 ✗ double a2 = 1. - alpha;
254
255 ✗ b->cd[B0] = b0 / a0;
256 ✗ b->cd[B1] = b1 / a0;
257 ✗ b->cd[B2] = b2 / a0;
258 ✗ b->cd[A1] = -a1 / a0;
259 ✗ b->cd[A2] = -a2 / a0;
260
261 ✗ b->cf[B0] = b->cd[B0];
262 ✗ b->cf[B1] = b->cd[B1];
263 ✗ b->cf[B2] = b->cd[B2];
264 ✗ b->cf[A1] = b->cd[A1];
265 ✗ b->cf[A2] = b->cd[A2];
266 ✗ }
267
268 ✗ static void set_hp(BiquadCoeffs *b, double fc, double q, double sr)
269 {
270 ✗ double omega = 2. * M_PI * fc / sr;
271 ✗ double cosine = cos(omega);
272 ✗ double alpha = sin(omega) / (2. * q);
273
274 ✗ double b0 = (1. + cosine) / 2.;
275 ✗ double b1 = -1. - cosine;
276 ✗ double b2 = (1. + cosine) / 2.;
277 ✗ double a0 = 1. + alpha;
278 ✗ double a1 = -2. * cosine;
279 ✗ double a2 = 1. - alpha;
280
281 ✗ b->cd[B0] = b0 / a0;
282 ✗ b->cd[B1] = b1 / a0;
283 ✗ b->cd[B2] = b2 / a0;
284 ✗ b->cd[A1] = -a1 / a0;
285 ✗ b->cd[A2] = -a2 / a0;
286
287 ✗ b->cf[B0] = b->cd[B0];
288 ✗ b->cf[B1] = b->cd[B1];
289 ✗ b->cf[B2] = b->cd[B2];
290 ✗ b->cf[A1] = b->cd[A1];
291 ✗ b->cf[A2] = b->cd[A2];
292 ✗ }
293
294 ✗ static void set_ap(BiquadCoeffs *b, double fc, double q, double sr)
295 {
296 ✗ double omega = 2. * M_PI * fc / sr;
297 ✗ double cosine = cos(omega);
298 ✗ double alpha = sin(omega) / (2. * q);
299
300 ✗ double a0 = 1. + alpha;
301 ✗ double a1 = -2. * cosine;
302 ✗ double a2 = 1. - alpha;
303 ✗ double b0 = a2;
304 ✗ double b1 = a1;
305 ✗ double b2 = a0;
306
307 ✗ b->cd[B0] = b0 / a0;
308 ✗ b->cd[B1] = b1 / a0;
309 ✗ b->cd[B2] = b2 / a0;
310 ✗ b->cd[A1] = -a1 / a0;
311 ✗ b->cd[A2] = -a2 / a0;
312
313 ✗ b->cf[B0] = b->cd[B0];
314 ✗ b->cf[B1] = b->cd[B1];
315 ✗ b->cf[B2] = b->cd[B2];
316 ✗ b->cf[A1] = b->cd[A1];
317 ✗ b->cf[A2] = b->cd[A2];
318 ✗ }
319
320 ✗ static void set_ap1(BiquadCoeffs *b, double fc, double sr)
321 {
322 ✗ double omega = 2. * M_PI * fc / sr;
323
324 ✗ b->cd[A1] = exp(-omega);
325 ✗ b->cd[A2] = 0.;
326 ✗ b->cd[B0] = -b->cd[A1];
327 ✗ b->cd[B1] = 1.;
328 ✗ b->cd[B2] = 0.;
329
330 ✗ b->cf[B0] = b->cd[B0];
331 ✗ b->cf[B1] = b->cd[B1];
332 ✗ b->cf[B2] = b->cd[B2];
333 ✗ b->cf[A1] = b->cd[A1];
334 ✗ b->cf[A2] = b->cd[A2];
335 ✗ }
336
337 ✗ static void calc_q_factors(int order, double *q)
338 {
339 ✗ double n = order / 2.;
340
341 ✗ for (int i = 0; i < n / 2; i++)
342 ✗ q[i] = 1. / (-2. * cos(M_PI * (2. * (i + 1) + n - 1.) / (2. * n)));
343 ✗ }
344
345 #define BIQUAD_PROCESS(name, type) \
346 static void biquad_process_## name(const type *const c, \
347 type *b, \
348 type *dst, const type *src, \
349 int nb_samples) \
350 { \
351 const type b0 = c[B0]; \
352 const type b1 = c[B1]; \
353 const type b2 = c[B2]; \
354 const type a1 = c[A1]; \
355 const type a2 = c[A2]; \
356 type z1 = b[0]; \
357 type z2 = b[1]; \
358 \
359 for (int n = 0; n + 1 < nb_samples; n++) { \
360 type in = src[n]; \
361 type out; \
362 \
363 out = in * b0 + z1; \
364 z1 = b1 * in + z2 + a1 * out; \
365 z2 = b2 * in + a2 * out; \
366 dst[n] = out; \
367 \
368 n++; \
369 in = src[n]; \
370 out = in * b0 + z1; \
371 z1 = b1 * in + z2 + a1 * out; \
372 z2 = b2 * in + a2 * out; \
373 dst[n] = out; \
374 } \
375 \
376 if (nb_samples & 1) { \
377 const int n = nb_samples - 1; \
378 const type in = src[n]; \
379 type out; \
380 \
381 out = in * b0 + z1; \
382 z1 = b1 * in + z2 + a1 * out; \
383 z2 = b2 * in + a2 * out; \
384 dst[n] = out; \
385 } \
386 \
387 b[0] = z1; \
388 b[1] = z2; \
389 }
390
391 ✗ BIQUAD_PROCESS(fltp, float)
392 ✗ BIQUAD_PROCESS(dblp, double)
393
394 #define XOVER_PROCESS(name, type, one, ff) \
395 static int filter_channels_## name(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) \
396 { \
397 AudioCrossoverContext *s = ctx->priv; \
398 AVFrame *in = arg; \
399 AVFrame **frames = s->frames; \
400 const int start = (in->ch_layout.nb_channels * jobnr) / nb_jobs; \
401 const int end = (in->ch_layout.nb_channels * (jobnr+1)) / nb_jobs; \
402 const int nb_samples = in->nb_samples; \
403 const int nb_outs = ctx->nb_outputs; \
404 const int first_order = s->first_order; \
405 \
406 for (int ch = start; ch < end; ch++) { \
407 const type *src = (const type *)in->extended_data[ch]; \
408 type *xover = (type *)s->xover->extended_data[ch]; \
409 \
410 s->fdsp->vector_## ff ##mul_scalar((type *)frames[0]->extended_data[ch], src, \
411 s->level_in, FFALIGN(nb_samples, sizeof(type))); \
412 \
413 for (int band = 0; band < nb_outs; band++) { \
414 for (int f = 0; band + 1 < nb_outs && f < s->filter_count; f++) { \
415 const type *prv = (const type *)frames[band]->extended_data[ch]; \
416 type *dst = (type *)frames[band + 1]->extended_data[ch]; \
417 const type *hsrc = f == 0 ? prv : dst; \
418 type *hp = xover + nb_outs * 20 + band * 20 + f * 2; \
419 const type *const hpc = (type *)&s->hp[band][f].c ## ff; \
420 \
421 biquad_process_## name(hpc, hp, dst, hsrc, nb_samples); \
422 } \
423 \
424 for (int f = 0; band + 1 < nb_outs && f < s->filter_count; f++) { \
425 type *dst = (type *)frames[band]->extended_data[ch]; \
426 const type *lsrc = dst; \
427 type *lp = xover + band * 20 + f * 2; \
428 const type *const lpc = (type *)&s->lp[band][f].c ## ff; \
429 \
430 biquad_process_## name(lpc, lp, dst, lsrc, nb_samples); \
431 } \
432 \
433 for (int aband = band + 1; aband + 1 < nb_outs; aband++) { \
434 if (first_order) { \
435 const type *asrc = (const type *)frames[band]->extended_data[ch]; \
436 type *dst = (type *)frames[band]->extended_data[ch]; \
437 type *ap = xover + nb_outs * 40 + (aband * nb_outs + band) * 20; \
438 const type *const apc = (type *)&s->ap[aband][0].c ## ff; \
439 \
440 biquad_process_## name(apc, ap, dst, asrc, nb_samples); \
441 } \
442 \
443 for (int f = first_order; f < s->ap_filter_count; f++) { \
444 const type *asrc = (const type *)frames[band]->extended_data[ch]; \
445 type *dst = (type *)frames[band]->extended_data[ch]; \
446 type *ap = xover + nb_outs * 40 + (aband * nb_outs + band) * 20 + f * 2;\
447 const type *const apc = (type *)&s->ap[aband][f].c ## ff; \
448 \
449 biquad_process_## name(apc, ap, dst, asrc, nb_samples); \
450 } \
451 } \
452 } \
453 \
454 for (int band = 0; band < nb_outs; band++) { \
455 const type gain = s->gains[band] * ((band & 1 && first_order) ? -one : one); \
456 type *dst = (type *)frames[band]->extended_data[ch]; \
457 \
458 s->fdsp->vector_## ff ##mul_scalar(dst, dst, gain, \
459 FFALIGN(nb_samples, sizeof(type))); \
460 } \
461 } \
462 \
463 return 0; \
464 }
465
466 ✗ XOVER_PROCESS(fltp, float, 1.f, f)
467 ✗ XOVER_PROCESS(dblp, double, 1.0, d)
468
469 ✗ static int config_input(AVFilterLink *inlink)
470 {
471 ✗ AVFilterContext *ctx = inlink->dst;
472 ✗ AudioCrossoverContext *s = ctx->priv;
473 ✗ int sample_rate = inlink->sample_rate;
474 double q[16];
475
476 ✗ s->order = (s->order_opt + 1) * 2;
477 ✗ s->filter_count = s->order / 2;
478 ✗ s->first_order = s->filter_count & 1;
479 ✗ s->ap_filter_count = s->filter_count / 2 + s->first_order;
480 ✗ calc_q_factors(s->order, q);
481
482 ✗ for (int band = 0; band <= s->nb_splits; band++) {
483 ✗ if (s->first_order) {
484 ✗ set_lp(&s->lp[band][0], s->splits[band], 0.5, sample_rate);
485 ✗ set_hp(&s->hp[band][0], s->splits[band], 0.5, sample_rate);
486 }
487
488 ✗ for (int n = s->first_order; n < s->filter_count; n++) {
489 ✗ const int idx = s->filter_count / 2 - ((n + s->first_order) / 2 - s->first_order) - 1;
490
491 ✗ set_lp(&s->lp[band][n], s->splits[band], q[idx], sample_rate);
492 ✗ set_hp(&s->hp[band][n], s->splits[band], q[idx], sample_rate);
493 }
494
495 ✗ if (s->first_order)
496 ✗ set_ap1(&s->ap[band][0], s->splits[band], sample_rate);
497
498 ✗ for (int n = s->first_order; n < s->ap_filter_count; n++) {
499 ✗ const int idx = (s->filter_count / 2 - ((n * 2 + s->first_order) / 2 - s->first_order) - 1);
500
501 ✗ set_ap(&s->ap[band][n], s->splits[band], q[idx], sample_rate);
502 }
503 }
504
505 ✗ switch (inlink->format) {
506 ✗ case AV_SAMPLE_FMT_FLTP: s->filter_channels = filter_channels_fltp; break;
507 ✗ case AV_SAMPLE_FMT_DBLP: s->filter_channels = filter_channels_dblp; break;
508 ✗ default: return AVERROR_BUG;
509 }
510
511 ✗ s->xover = ff_get_audio_buffer(inlink, 2 * (ctx->nb_outputs * 10 + ctx->nb_outputs * 10 +
512 ✗ ctx->nb_outputs * ctx->nb_outputs * 10));
513 ✗ if (!s->xover)
514 ✗ return AVERROR(ENOMEM);
515
516 ✗ return 0;
517 }
518
519 ✗ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
520 {
521 ✗ AVFilterContext *ctx = inlink->dst;
522 ✗ AudioCrossoverContext *s = ctx->priv;
523 ✗ AVFrame **frames = s->frames;
524 ✗ int ret = 0;
525
526 ✗ for (int i = 0; i < ctx->nb_outputs; i++) {
527 ✗ frames[i] = ff_get_audio_buffer(ctx->outputs[i], in->nb_samples);
528 ✗ if (!frames[i]) {
529 ✗ ret = AVERROR(ENOMEM);
530 ✗ break;
531 }
532
533 ✗ frames[i]->pts = in->pts;
534 }
535
536 ✗ if (ret < 0)
537 ✗ goto fail;
538
539 ✗ ff_filter_execute(ctx, s->filter_channels, in, NULL,
540 ✗ FFMIN(inlink->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx)));
541
542 ✗ for (int i = 0; i < ctx->nb_outputs; i++) {
543 ✗ if (ff_outlink_get_status(ctx->outputs[i])) {
544 ✗ av_frame_free(&frames[i]);
545 ✗ continue;
546 }
547
548 ✗ ret = ff_filter_frame(ctx->outputs[i], frames[i]);
549 ✗ frames[i] = NULL;
550 ✗ if (ret < 0)
551 ✗ break;
552 }
553
554 ✗ fail:
555 ✗ for (int i = 0; i < ctx->nb_outputs; i++)
556 ✗ av_frame_free(&frames[i]);
557
558 ✗ return ret;
559 }
560
561 ✗ static int activate(AVFilterContext *ctx)
562 {
563 ✗ AVFilterLink *inlink = ctx->inputs[0];
564 int status, ret;
565 AVFrame *in;
566 int64_t pts;
567
568 ✗ for (int i = 0; i < ctx->nb_outputs; i++) {
569 ✗ FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[i], ctx);
570 }
571
572 ✗ ret = ff_inlink_consume_frame(inlink, &in);
573 ✗ if (ret < 0)
574 ✗ return ret;
575 ✗ if (ret > 0) {
576 ✗ ret = filter_frame(inlink, in);
577 ✗ av_frame_free(&in);
578 ✗ if (ret < 0)
579 ✗ return ret;
580 }
581
582 ✗ if (ff_inlink_acknowledge_status(inlink, &status, &pts)) {
583 ✗ for (int i = 0; i < ctx->nb_outputs; i++) {
584 ✗ if (ff_outlink_get_status(ctx->outputs[i]))
585 ✗ continue;
586 ✗ ff_outlink_set_status(ctx->outputs[i], status, pts);
587 }
588 ✗ return 0;
589 }
590
591 ✗ for (int i = 0; i < ctx->nb_outputs; i++) {
592 ✗ if (ff_outlink_get_status(ctx->outputs[i]))
593 ✗ continue;
594
595 ✗ if (ff_outlink_frame_wanted(ctx->outputs[i])) {
596 ✗ ff_inlink_request_frame(inlink);
597 ✗ return 0;
598 }
599 }
600
601 ✗ return FFERROR_NOT_READY;
602 }
603
604 ✗ static av_cold void uninit(AVFilterContext *ctx)
605 {
606 ✗ AudioCrossoverContext *s = ctx->priv;
607
608 ✗ av_freep(&s->fdsp);
609 ✗ av_frame_free(&s->xover);
610 ✗ }
611
612 static const AVFilterPad inputs[] = {
613 {
614 .name = "default",
615 .type = AVMEDIA_TYPE_AUDIO,
616 .config_props = config_input,
617 },
618 };
619
620 const AVFilter ff_af_acrossover = {
621 .name = "acrossover",
622 .description = NULL_IF_CONFIG_SMALL("Split audio into per-bands streams."),
623 .priv_size = sizeof(AudioCrossoverContext),
624 .priv_class = &acrossover_class,
625 .init = init,
626 .activate = activate,
627 .uninit = uninit,
628 FILTER_INPUTS(inputs),
629 .outputs = NULL,
630 FILTER_QUERY_FUNC2(query_formats),
631 .flags = AVFILTER_FLAG_DYNAMIC_OUTPUTS |
632 AVFILTER_FLAG_SLICE_THREADS,
633 };
634