FFmpeg coverage

Directory: ../../../ffmpeg/
File: src/libavfilter/af_aiir.c
Date: 2024-11-20 23:03:26
Exec Total Coverage
Lines: 0 749 0.0%
Functions: 0 47 0.0%
Branches: 0 559 0.0%
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1 /*
2 * Copyright (c) 2018 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 <float.h>
22
23 #include "libavutil/avstring.h"
24 #include "libavutil/intreadwrite.h"
25 #include "libavutil/mem.h"
26 #include "libavutil/opt.h"
27 #include "libavutil/xga_font_data.h"
28 #include "audio.h"
29 #include "avfilter.h"
30 #include "filters.h"
31 #include "formats.h"
32 #include "video.h"
33
34 typedef struct ThreadData {
35 AVFrame *in, *out;
36 } ThreadData;
37
38 typedef struct Pair {
39 int a, b;
40 } Pair;
41
42 typedef struct BiquadContext {
43 double a[3];
44 double b[3];
45 double w1, w2;
46 } BiquadContext;
47
48 typedef struct IIRChannel {
49 int nb_ab[2];
50 double *ab[2];
51 double g;
52 double *cache[2];
53 double fir;
54 BiquadContext *biquads;
55 int clippings;
56 } IIRChannel;
57
58 typedef struct AudioIIRContext {
59 const AVClass *class;
60 char *a_str, *b_str, *g_str;
61 double dry_gain, wet_gain;
62 double mix;
63 int normalize;
64 int format;
65 int process;
66 int precision;
67 int response;
68 int w, h;
69 int ir_channel;
70 AVRational rate;
71
72 AVFrame *video;
73
74 IIRChannel *iir;
75 int channels;
76 enum AVSampleFormat sample_format;
77
78 int (*iir_channel)(AVFilterContext *ctx, void *arg, int ch, int nb_jobs);
79 } AudioIIRContext;
80
81 ✗ static int query_formats(const AVFilterContext *ctx,
82 AVFilterFormatsConfig **cfg_in,
83 AVFilterFormatsConfig **cfg_out)
84 {
85 ✗ const AudioIIRContext *s = ctx->priv;
86 AVFilterFormats *formats;
87 ✗ enum AVSampleFormat sample_fmts[] = {
88 AV_SAMPLE_FMT_DBLP,
89 AV_SAMPLE_FMT_NONE
90 };
91 static const enum AVPixelFormat pix_fmts[] = {
92 AV_PIX_FMT_RGB0,
93 AV_PIX_FMT_NONE
94 };
95 int ret;
96
97 ✗ if (s->response) {
98 ✗ formats = ff_make_format_list(pix_fmts);
99 ✗ if ((ret = ff_formats_ref(formats, &cfg_out[1]->formats)) < 0)
100 ✗ return ret;
101 }
102
103 ✗ sample_fmts[0] = s->sample_format;
104 ✗ ret = ff_set_common_formats_from_list2(ctx, cfg_in, cfg_out, sample_fmts);
105 ✗ if (ret < 0)
106 ✗ return ret;
107
108 ✗ return 0;
109 }
110
111 #define IIR_CH(name, type, min, max, need_clipping) \
112 static int iir_ch_## name(AVFilterContext *ctx, void *arg, int ch, int nb_jobs) \
113 { \
114 AudioIIRContext *s = ctx->priv; \
115 const double ig = s->dry_gain; \
116 const double og = s->wet_gain; \
117 const double mix = s->mix; \
118 ThreadData *td = arg; \
119 AVFrame *in = td->in, *out = td->out; \
120 const type *src = (const type *)in->extended_data[ch]; \
121 double *oc = (double *)s->iir[ch].cache[0]; \
122 double *ic = (double *)s->iir[ch].cache[1]; \
123 const int nb_a = s->iir[ch].nb_ab[0]; \
124 const int nb_b = s->iir[ch].nb_ab[1]; \
125 const double *a = s->iir[ch].ab[0]; \
126 const double *b = s->iir[ch].ab[1]; \
127 const double g = s->iir[ch].g; \
128 int *clippings = &s->iir[ch].clippings; \
129 type *dst = (type *)out->extended_data[ch]; \
130 int n; \
131 \
132 for (n = 0; n < in->nb_samples; n++) { \
133 double sample = 0.; \
134 int x; \
135 \
136 memmove(&ic[1], &ic[0], (nb_b - 1) * sizeof(*ic)); \
137 memmove(&oc[1], &oc[0], (nb_a - 1) * sizeof(*oc)); \
138 ic[0] = src[n] * ig; \
139 for (x = 0; x < nb_b; x++) \
140 sample += b[x] * ic[x]; \
141 \
142 for (x = 1; x < nb_a; x++) \
143 sample -= a[x] * oc[x]; \
144 \
145 oc[0] = sample; \
146 sample *= og * g; \
147 sample = sample * mix + ic[0] * (1. - mix); \
148 if (need_clipping && sample < min) { \
149 (*clippings)++; \
150 dst[n] = min; \
151 } else if (need_clipping && sample > max) { \
152 (*clippings)++; \
153 dst[n] = max; \
154 } else { \
155 dst[n] = sample; \
156 } \
157 } \
158 \
159 return 0; \
160 }
161
162 ✗ IIR_CH(s16p, int16_t, INT16_MIN, INT16_MAX, 1)
163 ✗ IIR_CH(s32p, int32_t, INT32_MIN, INT32_MAX, 1)
164 ✗ IIR_CH(fltp, float, -1., 1., 0)
165 ✗ IIR_CH(dblp, double, -1., 1., 0)
166
167 #define SERIAL_IIR_CH(name, type, min, max, need_clipping) \
168 static int iir_ch_serial_## name(AVFilterContext *ctx, void *arg, \
169 int ch, int nb_jobs) \
170 { \
171 AudioIIRContext *s = ctx->priv; \
172 const double ig = s->dry_gain; \
173 const double og = s->wet_gain; \
174 const double mix = s->mix; \
175 const double imix = 1. - mix; \
176 ThreadData *td = arg; \
177 AVFrame *in = td->in, *out = td->out; \
178 const type *src = (const type *)in->extended_data[ch]; \
179 type *dst = (type *)out->extended_data[ch]; \
180 IIRChannel *iir = &s->iir[ch]; \
181 const double g = iir->g; \
182 int *clippings = &iir->clippings; \
183 int nb_biquads = (FFMAX(iir->nb_ab[0], iir->nb_ab[1]) + 1) / 2; \
184 int n, i; \
185 \
186 for (i = nb_biquads - 1; i >= 0; i--) { \
187 const double a1 = -iir->biquads[i].a[1]; \
188 const double a2 = -iir->biquads[i].a[2]; \
189 const double b0 = iir->biquads[i].b[0]; \
190 const double b1 = iir->biquads[i].b[1]; \
191 const double b2 = iir->biquads[i].b[2]; \
192 double w1 = iir->biquads[i].w1; \
193 double w2 = iir->biquads[i].w2; \
194 \
195 for (n = 0; n < in->nb_samples; n++) { \
196 double i0 = ig * (i ? dst[n] : src[n]); \
197 double o0 = i0 * b0 + w1; \
198 \
199 w1 = b1 * i0 + w2 + a1 * o0; \
200 w2 = b2 * i0 + a2 * o0; \
201 o0 *= og * g; \
202 \
203 o0 = o0 * mix + imix * i0; \
204 if (need_clipping && o0 < min) { \
205 (*clippings)++; \
206 dst[n] = min; \
207 } else if (need_clipping && o0 > max) { \
208 (*clippings)++; \
209 dst[n] = max; \
210 } else { \
211 dst[n] = o0; \
212 } \
213 } \
214 iir->biquads[i].w1 = w1; \
215 iir->biquads[i].w2 = w2; \
216 } \
217 \
218 return 0; \
219 }
220
221 ✗ SERIAL_IIR_CH(s16p, int16_t, INT16_MIN, INT16_MAX, 1)
222 ✗ SERIAL_IIR_CH(s32p, int32_t, INT32_MIN, INT32_MAX, 1)
223 ✗ SERIAL_IIR_CH(fltp, float, -1., 1., 0)
224 ✗ SERIAL_IIR_CH(dblp, double, -1., 1., 0)
225
226 #define PARALLEL_IIR_CH(name, type, min, max, need_clipping) \
227 static int iir_ch_parallel_## name(AVFilterContext *ctx, void *arg, \
228 int ch, int nb_jobs) \
229 { \
230 AudioIIRContext *s = ctx->priv; \
231 const double ig = s->dry_gain; \
232 const double og = s->wet_gain; \
233 const double mix = s->mix; \
234 const double imix = 1. - mix; \
235 ThreadData *td = arg; \
236 AVFrame *in = td->in, *out = td->out; \
237 const type *src = (const type *)in->extended_data[ch]; \
238 type *dst = (type *)out->extended_data[ch]; \
239 IIRChannel *iir = &s->iir[ch]; \
240 const double g = iir->g; \
241 const double fir = iir->fir; \
242 int *clippings = &iir->clippings; \
243 int nb_biquads = (FFMAX(iir->nb_ab[0], iir->nb_ab[1]) + 1) / 2; \
244 int n, i; \
245 \
246 for (i = 0; i < nb_biquads; i++) { \
247 const double a1 = -iir->biquads[i].a[1]; \
248 const double a2 = -iir->biquads[i].a[2]; \
249 const double b1 = iir->biquads[i].b[1]; \
250 const double b2 = iir->biquads[i].b[2]; \
251 double w1 = iir->biquads[i].w1; \
252 double w2 = iir->biquads[i].w2; \
253 \
254 for (n = 0; n < in->nb_samples; n++) { \
255 double i0 = ig * src[n]; \
256 double o0 = w1; \
257 \
258 w1 = b1 * i0 + w2 + a1 * o0; \
259 w2 = b2 * i0 + a2 * o0; \
260 o0 *= og * g; \
261 o0 += dst[n]; \
262 \
263 if (need_clipping && o0 < min) { \
264 (*clippings)++; \
265 dst[n] = min; \
266 } else if (need_clipping && o0 > max) { \
267 (*clippings)++; \
268 dst[n] = max; \
269 } else { \
270 dst[n] = o0; \
271 } \
272 } \
273 iir->biquads[i].w1 = w1; \
274 iir->biquads[i].w2 = w2; \
275 } \
276 \
277 for (n = 0; n < in->nb_samples; n++) { \
278 dst[n] += fir * src[n]; \
279 dst[n] = dst[n] * mix + imix * src[n]; \
280 } \
281 \
282 return 0; \
283 }
284
285 ✗ PARALLEL_IIR_CH(s16p, int16_t, INT16_MIN, INT16_MAX, 1)
286 ✗ PARALLEL_IIR_CH(s32p, int32_t, INT32_MIN, INT32_MAX, 1)
287 ✗ PARALLEL_IIR_CH(fltp, float, -1., 1., 0)
288 ✗ PARALLEL_IIR_CH(dblp, double, -1., 1., 0)
289
290 #define LATTICE_IIR_CH(name, type, min, max, need_clipping) \
291 static int iir_ch_lattice_## name(AVFilterContext *ctx, void *arg, \
292 int ch, int nb_jobs) \
293 { \
294 AudioIIRContext *s = ctx->priv; \
295 const double ig = s->dry_gain; \
296 const double og = s->wet_gain; \
297 const double mix = s->mix; \
298 ThreadData *td = arg; \
299 AVFrame *in = td->in, *out = td->out; \
300 const type *src = (const type *)in->extended_data[ch]; \
301 double n0, n1, p0, *x = (double *)s->iir[ch].cache[0]; \
302 const int nb_stages = s->iir[ch].nb_ab[1]; \
303 const double *v = s->iir[ch].ab[0]; \
304 const double *k = s->iir[ch].ab[1]; \
305 const double g = s->iir[ch].g; \
306 int *clippings = &s->iir[ch].clippings; \
307 type *dst = (type *)out->extended_data[ch]; \
308 int n; \
309 \
310 for (n = 0; n < in->nb_samples; n++) { \
311 const double in = src[n] * ig; \
312 double out = 0.; \
313 \
314 n1 = in; \
315 for (int i = nb_stages - 1; i >= 0; i--) { \
316 n0 = n1 - k[i] * x[i]; \
317 p0 = n0 * k[i] + x[i]; \
318 out += p0 * v[i+1]; \
319 x[i] = p0; \
320 n1 = n0; \
321 } \
322 \
323 out += n1 * v[0]; \
324 memmove(&x[1], &x[0], nb_stages * sizeof(*x)); \
325 x[0] = n1; \
326 out *= og * g; \
327 out = out * mix + in * (1. - mix); \
328 if (need_clipping && out < min) { \
329 (*clippings)++; \
330 dst[n] = min; \
331 } else if (need_clipping && out > max) { \
332 (*clippings)++; \
333 dst[n] = max; \
334 } else { \
335 dst[n] = out; \
336 } \
337 } \
338 \
339 return 0; \
340 }
341
342 ✗ LATTICE_IIR_CH(s16p, int16_t, INT16_MIN, INT16_MAX, 1)
343 ✗ LATTICE_IIR_CH(s32p, int32_t, INT32_MIN, INT32_MAX, 1)
344 ✗ LATTICE_IIR_CH(fltp, float, -1., 1., 0)
345 ✗ LATTICE_IIR_CH(dblp, double, -1., 1., 0)
346
347 ✗ static void count_coefficients(char *item_str, int *nb_items)
348 {
349 char *p;
350
351 ✗ if (!item_str)
352 ✗ return;
353
354 ✗ *nb_items = 1;
355 ✗ for (p = item_str; *p && *p != '|'; p++) {
356 ✗ if (*p == ' ')
357 ✗ (*nb_items)++;
358 }
359 }
360
361 ✗ static int read_gains(AVFilterContext *ctx, char *item_str, int nb_items)
362 {
363 ✗ AudioIIRContext *s = ctx->priv;
364 ✗ char *p, *arg, *old_str, *prev_arg = NULL, *saveptr = NULL;
365 int i;
366
367 ✗ p = old_str = av_strdup(item_str);
368 ✗ if (!p)
369 ✗ return AVERROR(ENOMEM);
370 ✗ for (i = 0; i < nb_items; i++) {
371 ✗ if (!(arg = av_strtok(p, "|", &saveptr)))
372 ✗ arg = prev_arg;
373
374 ✗ if (!arg) {
375 ✗ av_freep(&old_str);
376 ✗ return AVERROR(EINVAL);
377 }
378
379 ✗ p = NULL;
380 ✗ if (av_sscanf(arg, "%lf", &s->iir[i].g) != 1) {
381 ✗ av_log(ctx, AV_LOG_ERROR, "Invalid gains supplied: %s\n", arg);
382 ✗ av_freep(&old_str);
383 ✗ return AVERROR(EINVAL);
384 }
385
386 ✗ prev_arg = arg;
387 }
388
389 ✗ av_freep(&old_str);
390
391 ✗ return 0;
392 }
393
394 ✗ static int read_tf_coefficients(AVFilterContext *ctx, char *item_str, int nb_items, double *dst)
395 {
396 ✗ char *p, *arg, *old_str, *saveptr = NULL;
397 int i;
398
399 ✗ p = old_str = av_strdup(item_str);
400 ✗ if (!p)
401 ✗ return AVERROR(ENOMEM);
402 ✗ for (i = 0; i < nb_items; i++) {
403 ✗ if (!(arg = av_strtok(p, " ", &saveptr)))
404 ✗ break;
405
406 ✗ p = NULL;
407 ✗ if (av_sscanf(arg, "%lf", &dst[i]) != 1) {
408 ✗ av_log(ctx, AV_LOG_ERROR, "Invalid coefficients supplied: %s\n", arg);
409 ✗ av_freep(&old_str);
410 ✗ return AVERROR(EINVAL);
411 }
412 }
413
414 ✗ av_freep(&old_str);
415
416 ✗ return 0;
417 }
418
419 ✗ static int read_zp_coefficients(AVFilterContext *ctx, char *item_str, int nb_items, double *dst, const char *format)
420 {
421 ✗ char *p, *arg, *old_str, *saveptr = NULL;
422 int i;
423
424 ✗ p = old_str = av_strdup(item_str);
425 ✗ if (!p)
426 ✗ return AVERROR(ENOMEM);
427 ✗ for (i = 0; i < nb_items; i++) {
428 ✗ if (!(arg = av_strtok(p, " ", &saveptr)))
429 ✗ break;
430
431 ✗ p = NULL;
432 ✗ if (av_sscanf(arg, format, &dst[i*2], &dst[i*2+1]) != 2) {
433 ✗ av_log(ctx, AV_LOG_ERROR, "Invalid coefficients supplied: %s\n", arg);
434 ✗ av_freep(&old_str);
435 ✗ return AVERROR(EINVAL);
436 }
437 }
438
439 ✗ av_freep(&old_str);
440
441 ✗ return 0;
442 }
443
444 static const char *const format[] = { "%lf", "%lf %lfi", "%lf %lfr", "%lf %lfd", "%lf %lfi" };
445
446 ✗ static int read_channels(AVFilterContext *ctx, int channels, uint8_t *item_str, int ab)
447 {
448 ✗ AudioIIRContext *s = ctx->priv;
449 ✗ char *p, *arg, *old_str, *prev_arg = NULL, *saveptr = NULL;
450 int i, ret;
451
452 ✗ p = old_str = av_strdup(item_str);
453 ✗ if (!p)
454 ✗ return AVERROR(ENOMEM);
455 ✗ for (i = 0; i < channels; i++) {
456 ✗ IIRChannel *iir = &s->iir[i];
457
458 ✗ if (!(arg = av_strtok(p, "|", &saveptr)))
459 ✗ arg = prev_arg;
460
461 ✗ if (!arg) {
462 ✗ av_freep(&old_str);
463 ✗ return AVERROR(EINVAL);
464 }
465
466 ✗ count_coefficients(arg, &iir->nb_ab[ab]);
467
468 ✗ p = NULL;
469 ✗ iir->cache[ab] = av_calloc(iir->nb_ab[ab] + 1, sizeof(double));
470 ✗ iir->ab[ab] = av_calloc(iir->nb_ab[ab] * (!!s->format + 1), sizeof(double));
471 ✗ if (!iir->ab[ab] || !iir->cache[ab]) {
472 ✗ av_freep(&old_str);
473 ✗ return AVERROR(ENOMEM);
474 }
475
476 ✗ if (s->format > 0) {
477 ✗ ret = read_zp_coefficients(ctx, arg, iir->nb_ab[ab], iir->ab[ab], format[s->format]);
478 } else {
479 ✗ ret = read_tf_coefficients(ctx, arg, iir->nb_ab[ab], iir->ab[ab]);
480 }
481 ✗ if (ret < 0) {
482 ✗ av_freep(&old_str);
483 ✗ return ret;
484 }
485 ✗ prev_arg = arg;
486 }
487
488 ✗ av_freep(&old_str);
489
490 ✗ return 0;
491 }
492
493 ✗ static void cmul(double re, double im, double re2, double im2, double *RE, double *IM)
494 {
495 ✗ *RE = re * re2 - im * im2;
496 ✗ *IM = re * im2 + re2 * im;
497 ✗ }
498
499 ✗ static int expand(AVFilterContext *ctx, double *pz, int n, double *coefs)
500 {
501 ✗ coefs[2 * n] = 1.0;
502
503 ✗ for (int i = 1; i <= n; i++) {
504 ✗ for (int j = n - i; j < n; j++) {
505 double re, im;
506
507 ✗ cmul(coefs[2 * (j + 1)], coefs[2 * (j + 1) + 1],
508 ✗ pz[2 * (i - 1)], pz[2 * (i - 1) + 1], &re, &im);
509
510 ✗ coefs[2 * j] -= re;
511 ✗ coefs[2 * j + 1] -= im;
512 }
513 }
514
515 ✗ for (int i = 0; i < n + 1; i++) {
516 ✗ if (fabs(coefs[2 * i + 1]) > FLT_EPSILON) {
517 ✗ av_log(ctx, AV_LOG_ERROR, "coefs: %f of z^%d is not real; poles/zeros are not complex conjugates.\n",
518 ✗ coefs[2 * i + 1], i);
519 ✗ return AVERROR(EINVAL);
520 }
521 }
522
523 ✗ return 0;
524 }
525
526 ✗ static void normalize_coeffs(AVFilterContext *ctx, int ch)
527 {
528 ✗ AudioIIRContext *s = ctx->priv;
529 ✗ IIRChannel *iir = &s->iir[ch];
530 ✗ double sum_den = 0.;
531
532 ✗ if (!s->normalize)
533 ✗ return;
534
535 ✗ for (int i = 0; i < iir->nb_ab[1]; i++) {
536 ✗ sum_den += iir->ab[1][i];
537 }
538
539 ✗ if (sum_den > 1e-6) {
540 ✗ double factor, sum_num = 0.;
541
542 ✗ for (int i = 0; i < iir->nb_ab[0]; i++) {
543 ✗ sum_num += iir->ab[0][i];
544 }
545
546 ✗ factor = sum_num / sum_den;
547
548 ✗ for (int i = 0; i < iir->nb_ab[1]; i++) {
549 ✗ iir->ab[1][i] *= factor;
550 }
551 }
552 }
553
554 ✗ static int convert_zp2tf(AVFilterContext *ctx, int channels)
555 {
556 ✗ AudioIIRContext *s = ctx->priv;
557 ✗ int ch, i, j, ret = 0;
558
559 ✗ for (ch = 0; ch < channels; ch++) {
560 ✗ IIRChannel *iir = &s->iir[ch];
561 double *topc, *botc;
562
563 ✗ topc = av_calloc((iir->nb_ab[1] + 1) * 2, sizeof(*topc));
564 ✗ botc = av_calloc((iir->nb_ab[0] + 1) * 2, sizeof(*botc));
565 ✗ if (!topc || !botc) {
566 ✗ ret = AVERROR(ENOMEM);
567 ✗ goto fail;
568 }
569
570 ✗ ret = expand(ctx, iir->ab[0], iir->nb_ab[0], botc);
571 ✗ if (ret < 0) {
572 ✗ goto fail;
573 }
574
575 ✗ ret = expand(ctx, iir->ab[1], iir->nb_ab[1], topc);
576 ✗ if (ret < 0) {
577 ✗ goto fail;
578 }
579
580 ✗ for (j = 0, i = iir->nb_ab[1]; i >= 0; j++, i--) {
581 ✗ iir->ab[1][j] = topc[2 * i];
582 }
583 ✗ iir->nb_ab[1]++;
584
585 ✗ for (j = 0, i = iir->nb_ab[0]; i >= 0; j++, i--) {
586 ✗ iir->ab[0][j] = botc[2 * i];
587 }
588 ✗ iir->nb_ab[0]++;
589
590 ✗ normalize_coeffs(ctx, ch);
591
592 ✗ fail:
593 ✗ av_free(topc);
594 ✗ av_free(botc);
595 ✗ if (ret < 0)
596 ✗ break;
597 }
598
599 ✗ return ret;
600 }
601
602 ✗ static int decompose_zp2biquads(AVFilterContext *ctx, int channels)
603 {
604 ✗ AudioIIRContext *s = ctx->priv;
605 int ch, ret;
606
607 ✗ for (ch = 0; ch < channels; ch++) {
608 ✗ IIRChannel *iir = &s->iir[ch];
609 ✗ int nb_biquads = (FFMAX(iir->nb_ab[0], iir->nb_ab[1]) + 1) / 2;
610 ✗ int current_biquad = 0;
611
612 ✗ iir->biquads = av_calloc(nb_biquads, sizeof(BiquadContext));
613 ✗ if (!iir->biquads)
614 ✗ return AVERROR(ENOMEM);
615
616 ✗ while (nb_biquads--) {
617 ✗ Pair outmost_pole = { -1, -1 };
618 ✗ Pair nearest_zero = { -1, -1 };
619 ✗ double zeros[4] = { 0 };
620 ✗ double poles[4] = { 0 };
621 ✗ double b[6] = { 0 };
622 ✗ double a[6] = { 0 };
623 ✗ double min_distance = DBL_MAX;
624 ✗ double max_mag = 0;
625 double factor;
626 int i;
627
628 ✗ for (i = 0; i < iir->nb_ab[0]; i++) {
629 double mag;
630
631 ✗ if (isnan(iir->ab[0][2 * i]) || isnan(iir->ab[0][2 * i + 1]))
632 ✗ continue;
633 ✗ mag = hypot(iir->ab[0][2 * i], iir->ab[0][2 * i + 1]);
634
635 ✗ if (mag > max_mag) {
636 ✗ max_mag = mag;
637 ✗ outmost_pole.a = i;
638 }
639 }
640
641 ✗ for (i = 0; i < iir->nb_ab[0]; i++) {
642 ✗ if (isnan(iir->ab[0][2 * i]) || isnan(iir->ab[0][2 * i + 1]))
643 ✗ continue;
644
645 ✗ if (iir->ab[0][2 * i ] == iir->ab[0][2 * outmost_pole.a ] &&
646 ✗ iir->ab[0][2 * i + 1] == -iir->ab[0][2 * outmost_pole.a + 1]) {
647 ✗ outmost_pole.b = i;
648 ✗ break;
649 }
650 }
651
652 ✗ av_log(ctx, AV_LOG_VERBOSE, "outmost_pole is %d.%d\n", outmost_pole.a, outmost_pole.b);
653
654 ✗ if (outmost_pole.a < 0 || outmost_pole.b < 0)
655 ✗ return AVERROR(EINVAL);
656
657 ✗ for (i = 0; i < iir->nb_ab[1]; i++) {
658 double distance;
659
660 ✗ if (isnan(iir->ab[1][2 * i]) || isnan(iir->ab[1][2 * i + 1]))
661 ✗ continue;
662 ✗ distance = hypot(iir->ab[0][2 * outmost_pole.a ] - iir->ab[1][2 * i ],
663 ✗ iir->ab[0][2 * outmost_pole.a + 1] - iir->ab[1][2 * i + 1]);
664
665 ✗ if (distance < min_distance) {
666 ✗ min_distance = distance;
667 ✗ nearest_zero.a = i;
668 }
669 }
670
671 ✗ for (i = 0; i < iir->nb_ab[1]; i++) {
672 ✗ if (isnan(iir->ab[1][2 * i]) || isnan(iir->ab[1][2 * i + 1]))
673 ✗ continue;
674
675 ✗ if (iir->ab[1][2 * i ] == iir->ab[1][2 * nearest_zero.a ] &&
676 ✗ iir->ab[1][2 * i + 1] == -iir->ab[1][2 * nearest_zero.a + 1]) {
677 ✗ nearest_zero.b = i;
678 ✗ break;
679 }
680 }
681
682 ✗ av_log(ctx, AV_LOG_VERBOSE, "nearest_zero is %d.%d\n", nearest_zero.a, nearest_zero.b);
683
684 ✗ if (nearest_zero.a < 0 || nearest_zero.b < 0)
685 ✗ return AVERROR(EINVAL);
686
687 ✗ poles[0] = iir->ab[0][2 * outmost_pole.a ];
688 ✗ poles[1] = iir->ab[0][2 * outmost_pole.a + 1];
689
690 ✗ zeros[0] = iir->ab[1][2 * nearest_zero.a ];
691 ✗ zeros[1] = iir->ab[1][2 * nearest_zero.a + 1];
692
693 ✗ if (nearest_zero.a == nearest_zero.b && outmost_pole.a == outmost_pole.b) {
694 ✗ zeros[2] = 0;
695 ✗ zeros[3] = 0;
696
697 ✗ poles[2] = 0;
698 ✗ poles[3] = 0;
699 } else {
700 ✗ poles[2] = iir->ab[0][2 * outmost_pole.b ];
701 ✗ poles[3] = iir->ab[0][2 * outmost_pole.b + 1];
702
703 ✗ zeros[2] = iir->ab[1][2 * nearest_zero.b ];
704 ✗ zeros[3] = iir->ab[1][2 * nearest_zero.b + 1];
705 }
706
707 ✗ ret = expand(ctx, zeros, 2, b);
708 ✗ if (ret < 0)
709 ✗ return ret;
710
711 ✗ ret = expand(ctx, poles, 2, a);
712 ✗ if (ret < 0)
713 ✗ return ret;
714
715 ✗ iir->ab[0][2 * outmost_pole.a] = iir->ab[0][2 * outmost_pole.a + 1] = NAN;
716 ✗ iir->ab[0][2 * outmost_pole.b] = iir->ab[0][2 * outmost_pole.b + 1] = NAN;
717 ✗ iir->ab[1][2 * nearest_zero.a] = iir->ab[1][2 * nearest_zero.a + 1] = NAN;
718 ✗ iir->ab[1][2 * nearest_zero.b] = iir->ab[1][2 * nearest_zero.b + 1] = NAN;
719
720 ✗ iir->biquads[current_biquad].a[0] = 1.;
721 ✗ iir->biquads[current_biquad].a[1] = a[2] / a[4];
722 ✗ iir->biquads[current_biquad].a[2] = a[0] / a[4];
723 ✗ iir->biquads[current_biquad].b[0] = b[4] / a[4];
724 ✗ iir->biquads[current_biquad].b[1] = b[2] / a[4];
725 ✗ iir->biquads[current_biquad].b[2] = b[0] / a[4];
726
727 ✗ if (s->normalize &&
728 ✗ fabs(iir->biquads[current_biquad].b[0] +
729 ✗ iir->biquads[current_biquad].b[1] +
730 ✗ iir->biquads[current_biquad].b[2]) > 1e-6) {
731 ✗ factor = (iir->biquads[current_biquad].a[0] +
732 ✗ iir->biquads[current_biquad].a[1] +
733 ✗ iir->biquads[current_biquad].a[2]) /
734 ✗ (iir->biquads[current_biquad].b[0] +
735 ✗ iir->biquads[current_biquad].b[1] +
736 ✗ iir->biquads[current_biquad].b[2]);
737
738 ✗ av_log(ctx, AV_LOG_VERBOSE, "factor=%f\n", factor);
739
740 ✗ iir->biquads[current_biquad].b[0] *= factor;
741 ✗ iir->biquads[current_biquad].b[1] *= factor;
742 ✗ iir->biquads[current_biquad].b[2] *= factor;
743 }
744
745 ✗ iir->biquads[current_biquad].b[0] *= (current_biquad ? 1.0 : iir->g);
746 ✗ iir->biquads[current_biquad].b[1] *= (current_biquad ? 1.0 : iir->g);
747 ✗ iir->biquads[current_biquad].b[2] *= (current_biquad ? 1.0 : iir->g);
748
749 ✗ av_log(ctx, AV_LOG_VERBOSE, "a=%f %f %f:b=%f %f %f\n",
750 ✗ iir->biquads[current_biquad].a[0],
751 ✗ iir->biquads[current_biquad].a[1],
752 ✗ iir->biquads[current_biquad].a[2],
753 ✗ iir->biquads[current_biquad].b[0],
754 ✗ iir->biquads[current_biquad].b[1],
755 ✗ iir->biquads[current_biquad].b[2]);
756
757 ✗ current_biquad++;
758 }
759 }
760
761 ✗ return 0;
762 }
763
764 ✗ static void biquad_process(double *x, double *y, int length,
765 double b0, double b1, double b2,
766 double a1, double a2)
767 {
768 ✗ double w1 = 0., w2 = 0.;
769
770 ✗ a1 = -a1;
771 ✗ a2 = -a2;
772
773 ✗ for (int n = 0; n < length; n++) {
774 ✗ double out, in = x[n];
775
776 ✗ y[n] = out = in * b0 + w1;
777 ✗ w1 = b1 * in + w2 + a1 * out;
778 ✗ w2 = b2 * in + a2 * out;
779 }
780 ✗ }
781
782 ✗ static void solve(double *matrix, double *vector, int n, double *y, double *x, double *lu)
783 {
784 ✗ double sum = 0.;
785
786 ✗ for (int i = 0; i < n; i++) {
787 ✗ for (int j = i; j < n; j++) {
788 ✗ sum = 0.;
789 ✗ for (int k = 0; k < i; k++)
790 ✗ sum += lu[i * n + k] * lu[k * n + j];
791 ✗ lu[i * n + j] = matrix[j * n + i] - sum;
792 }
793 ✗ for (int j = i + 1; j < n; j++) {
794 ✗ sum = 0.;
795 ✗ for (int k = 0; k < i; k++)
796 ✗ sum += lu[j * n + k] * lu[k * n + i];
797 ✗ lu[j * n + i] = (1. / lu[i * n + i]) * (matrix[i * n + j] - sum);
798 }
799 }
800
801 ✗ for (int i = 0; i < n; i++) {
802 ✗ sum = 0.;
803 ✗ for (int k = 0; k < i; k++)
804 ✗ sum += lu[i * n + k] * y[k];
805 ✗ y[i] = vector[i] - sum;
806 }
807
808 ✗ for (int i = n - 1; i >= 0; i--) {
809 ✗ sum = 0.;
810 ✗ for (int k = i + 1; k < n; k++)
811 ✗ sum += lu[i * n + k] * x[k];
812 ✗ x[i] = (1 / lu[i * n + i]) * (y[i] - sum);
813 }
814 ✗ }
815
816 ✗ static int convert_serial2parallel(AVFilterContext *ctx, int channels)
817 {
818 ✗ AudioIIRContext *s = ctx->priv;
819
820 ✗ for (int ch = 0; ch < channels; ch++) {
821 ✗ IIRChannel *iir = &s->iir[ch];
822 ✗ int nb_biquads = (FFMAX(iir->nb_ab[0], iir->nb_ab[1]) + 1) / 2;
823 ✗ int length = nb_biquads * 2 + 1;
824 ✗ double *impulse = av_calloc(length, sizeof(*impulse));
825 ✗ double *y = av_calloc(length, sizeof(*y));
826 ✗ double *resp = av_calloc(length, sizeof(*resp));
827 ✗ double *M = av_calloc((length - 1) * nb_biquads, 2 * 2 * sizeof(*M));
828 double *W;
829
830 ✗ if (!impulse || !y || !resp || !M) {
831 ✗ av_free(impulse);
832 ✗ av_free(y);
833 ✗ av_free(resp);
834 ✗ av_free(M);
835 ✗ return AVERROR(ENOMEM);
836 }
837 ✗ W = M + (length - 1) * 2 * nb_biquads;
838
839 ✗ impulse[0] = 1.;
840
841 ✗ for (int n = 0; n < nb_biquads; n++) {
842 ✗ BiquadContext *biquad = &iir->biquads[n];
843
844 ✗ biquad_process(n ? y : impulse, y, length,
845 biquad->b[0], biquad->b[1], biquad->b[2],
846 biquad->a[1], biquad->a[2]);
847 }
848
849 ✗ for (int n = 0; n < nb_biquads; n++) {
850 ✗ BiquadContext *biquad = &iir->biquads[n];
851
852 ✗ biquad_process(impulse, resp, length - 1,
853 1., 0., 0., biquad->a[1], biquad->a[2]);
854
855 ✗ memcpy(M + n * 2 * (length - 1), resp, sizeof(*resp) * (length - 1));
856 ✗ memcpy(M + n * 2 * (length - 1) + length, resp, sizeof(*resp) * (length - 2));
857 ✗ memset(resp, 0, length * sizeof(*resp));
858 }
859
860 ✗ solve(M, &y[1], length - 1, &impulse[1], resp, W);
861
862 ✗ iir->fir = y[0];
863
864 ✗ for (int n = 0; n < nb_biquads; n++) {
865 ✗ BiquadContext *biquad = &iir->biquads[n];
866
867 ✗ biquad->b[0] = 0.;
868 ✗ biquad->b[1] = resp[n * 2 + 0];
869 ✗ biquad->b[2] = resp[n * 2 + 1];
870 }
871
872 ✗ av_free(impulse);
873 ✗ av_free(y);
874 ✗ av_free(resp);
875 ✗ av_free(M);
876 }
877
878 ✗ return 0;
879 }
880
881 ✗ static void convert_pr2zp(AVFilterContext *ctx, int channels)
882 {
883 ✗ AudioIIRContext *s = ctx->priv;
884 int ch;
885
886 ✗ for (ch = 0; ch < channels; ch++) {
887 ✗ IIRChannel *iir = &s->iir[ch];
888 int n;
889
890 ✗ for (n = 0; n < iir->nb_ab[0]; n++) {
891 ✗ double r = iir->ab[0][2*n];
892 ✗ double angle = iir->ab[0][2*n+1];
893
894 ✗ iir->ab[0][2*n] = r * cos(angle);
895 ✗ iir->ab[0][2*n+1] = r * sin(angle);
896 }
897
898 ✗ for (n = 0; n < iir->nb_ab[1]; n++) {
899 ✗ double r = iir->ab[1][2*n];
900 ✗ double angle = iir->ab[1][2*n+1];
901
902 ✗ iir->ab[1][2*n] = r * cos(angle);
903 ✗ iir->ab[1][2*n+1] = r * sin(angle);
904 }
905 }
906 ✗ }
907
908 ✗ static void convert_sp2zp(AVFilterContext *ctx, int channels)
909 {
910 ✗ AudioIIRContext *s = ctx->priv;
911 int ch;
912
913 ✗ for (ch = 0; ch < channels; ch++) {
914 ✗ IIRChannel *iir = &s->iir[ch];
915 int n;
916
917 ✗ for (n = 0; n < iir->nb_ab[0]; n++) {
918 ✗ double sr = iir->ab[0][2*n];
919 ✗ double si = iir->ab[0][2*n+1];
920
921 ✗ iir->ab[0][2*n] = exp(sr) * cos(si);
922 ✗ iir->ab[0][2*n+1] = exp(sr) * sin(si);
923 }
924
925 ✗ for (n = 0; n < iir->nb_ab[1]; n++) {
926 ✗ double sr = iir->ab[1][2*n];
927 ✗ double si = iir->ab[1][2*n+1];
928
929 ✗ iir->ab[1][2*n] = exp(sr) * cos(si);
930 ✗ iir->ab[1][2*n+1] = exp(sr) * sin(si);
931 }
932 }
933 ✗ }
934
935 ✗ static double fact(double i)
936 {
937 ✗ if (i <= 0.)
938 ✗ return 1.;
939 ✗ return i * fact(i - 1.);
940 }
941
942 ✗ static double coef_sf2zf(double *a, int N, int n)
943 {
944 ✗ double z = 0.;
945
946 ✗ for (int i = 0; i <= N; i++) {
947 ✗ double acc = 0.;
948
949 ✗ for (int k = FFMAX(n - N + i, 0); k <= FFMIN(i, n); k++) {
950 ✗ acc += ((fact(i) * fact(N - i)) /
951 ✗ (fact(k) * fact(i - k) * fact(n - k) * fact(N - i - n + k))) *
952 ✗ ((k & 1) ? -1. : 1.);
953 }
954
955 ✗ z += a[i] * pow(2., i) * acc;
956 }
957
958 ✗ return z;
959 }
960
961 ✗ static void convert_sf2tf(AVFilterContext *ctx, int channels)
962 {
963 ✗ AudioIIRContext *s = ctx->priv;
964 int ch;
965
966 ✗ for (ch = 0; ch < channels; ch++) {
967 ✗ IIRChannel *iir = &s->iir[ch];
968 ✗ double *temp0 = av_calloc(iir->nb_ab[0], sizeof(*temp0));
969 ✗ double *temp1 = av_calloc(iir->nb_ab[1], sizeof(*temp1));
970
971 ✗ if (!temp0 || !temp1)
972 ✗ goto next;
973
974 ✗ memcpy(temp0, iir->ab[0], iir->nb_ab[0] * sizeof(*temp0));
975 ✗ memcpy(temp1, iir->ab[1], iir->nb_ab[1] * sizeof(*temp1));
976
977 ✗ for (int n = 0; n < iir->nb_ab[0]; n++)
978 ✗ iir->ab[0][n] = coef_sf2zf(temp0, iir->nb_ab[0] - 1, n);
979
980 ✗ for (int n = 0; n < iir->nb_ab[1]; n++)
981 ✗ iir->ab[1][n] = coef_sf2zf(temp1, iir->nb_ab[1] - 1, n);
982
983 ✗ next:
984 ✗ av_free(temp0);
985 ✗ av_free(temp1);
986 }
987 ✗ }
988
989 ✗ static void convert_pd2zp(AVFilterContext *ctx, int channels)
990 {
991 ✗ AudioIIRContext *s = ctx->priv;
992 int ch;
993
994 ✗ for (ch = 0; ch < channels; ch++) {
995 ✗ IIRChannel *iir = &s->iir[ch];
996 int n;
997
998 ✗ for (n = 0; n < iir->nb_ab[0]; n++) {
999 ✗ double r = iir->ab[0][2*n];
1000 ✗ double angle = M_PI*iir->ab[0][2*n+1]/180.;
1001
1002 ✗ iir->ab[0][2*n] = r * cos(angle);
1003 ✗ iir->ab[0][2*n+1] = r * sin(angle);
1004 }
1005
1006 ✗ for (n = 0; n < iir->nb_ab[1]; n++) {
1007 ✗ double r = iir->ab[1][2*n];
1008 ✗ double angle = M_PI*iir->ab[1][2*n+1]/180.;
1009
1010 ✗ iir->ab[1][2*n] = r * cos(angle);
1011 ✗ iir->ab[1][2*n+1] = r * sin(angle);
1012 }
1013 }
1014 ✗ }
1015
1016 ✗ static void check_stability(AVFilterContext *ctx, int channels)
1017 {
1018 ✗ AudioIIRContext *s = ctx->priv;
1019 int ch;
1020
1021 ✗ for (ch = 0; ch < channels; ch++) {
1022 ✗ IIRChannel *iir = &s->iir[ch];
1023
1024 ✗ for (int n = 0; n < iir->nb_ab[0]; n++) {
1025 ✗ double pr = hypot(iir->ab[0][2*n], iir->ab[0][2*n+1]);
1026
1027 ✗ if (pr >= 1.) {
1028 ✗ av_log(ctx, AV_LOG_WARNING, "pole %d at channel %d is unstable\n", n, ch);
1029 ✗ break;
1030 }
1031 }
1032 }
1033 ✗ }
1034
1035 ✗ static void drawtext(AVFrame *pic, int x, int y, const char *txt, uint32_t color)
1036 {
1037 const uint8_t *font;
1038 int font_height;
1039 int i;
1040
1041 ✗ font = avpriv_cga_font, font_height = 8;
1042
1043 ✗ for (i = 0; txt[i]; i++) {
1044 int char_y, mask;
1045
1046 ✗ uint8_t *p = pic->data[0] + y * pic->linesize[0] + (x + i * 8) * 4;
1047 ✗ for (char_y = 0; char_y < font_height; char_y++) {
1048 ✗ for (mask = 0x80; mask; mask >>= 1) {
1049 ✗ if (font[txt[i] * font_height + char_y] & mask)
1050 ✗ AV_WL32(p, color);
1051 ✗ p += 4;
1052 }
1053 ✗ p += pic->linesize[0] - 8 * 4;
1054 }
1055 }
1056 ✗ }
1057
1058 ✗ static void draw_line(AVFrame *out, int x0, int y0, int x1, int y1, uint32_t color)
1059 {
1060 ✗ int dx = FFABS(x1-x0);
1061 ✗ int dy = FFABS(y1-y0), sy = y0 < y1 ? 1 : -1;
1062 ✗ int err = (dx>dy ? dx : -dy) / 2, e2;
1063
1064 for (;;) {
1065 ✗ AV_WL32(out->data[0] + y0 * out->linesize[0] + x0 * 4, color);
1066
1067 ✗ if (x0 == x1 && y0 == y1)
1068 ✗ break;
1069
1070 ✗ e2 = err;
1071
1072 ✗ if (e2 >-dx) {
1073 ✗ err -= dy;
1074 ✗ x0--;
1075 }
1076
1077 ✗ if (e2 < dy) {
1078 ✗ err += dx;
1079 ✗ y0 += sy;
1080 }
1081 }
1082 ✗ }
1083
1084 ✗ static double distance(double x0, double x1, double y0, double y1)
1085 {
1086 ✗ return hypot(x0 - x1, y0 - y1);
1087 }
1088
1089 ✗ static void get_response(int channel, int format, double w,
1090 const double *b, const double *a,
1091 int nb_b, int nb_a, double *magnitude, double *phase)
1092 {
1093 double realz, realp;
1094 double imagz, imagp;
1095 double real, imag;
1096 double div;
1097
1098 ✗ if (format == 0) {
1099 ✗ realz = 0., realp = 0.;
1100 ✗ imagz = 0., imagp = 0.;
1101 ✗ for (int x = 0; x < nb_a; x++) {
1102 ✗ realz += cos(-x * w) * a[x];
1103 ✗ imagz += sin(-x * w) * a[x];
1104 }
1105
1106 ✗ for (int x = 0; x < nb_b; x++) {
1107 ✗ realp += cos(-x * w) * b[x];
1108 ✗ imagp += sin(-x * w) * b[x];
1109 }
1110
1111 ✗ div = realp * realp + imagp * imagp;
1112 ✗ real = (realz * realp + imagz * imagp) / div;
1113 ✗ imag = (imagz * realp - imagp * realz) / div;
1114
1115 ✗ *magnitude = hypot(real, imag);
1116 ✗ *phase = atan2(imag, real);
1117 } else {
1118 ✗ double p = 1., z = 1.;
1119 ✗ double acc = 0.;
1120
1121 ✗ for (int x = 0; x < nb_a; x++) {
1122 ✗ z *= distance(cos(w), a[2 * x], sin(w), a[2 * x + 1]);
1123 ✗ acc += atan2(sin(w) - a[2 * x + 1], cos(w) - a[2 * x]);
1124 }
1125
1126 ✗ for (int x = 0; x < nb_b; x++) {
1127 ✗ p *= distance(cos(w), b[2 * x], sin(w), b[2 * x + 1]);
1128 ✗ acc -= atan2(sin(w) - b[2 * x + 1], cos(w) - b[2 * x]);
1129 }
1130
1131 ✗ *magnitude = z / p;
1132 ✗ *phase = acc;
1133 }
1134 ✗ }
1135
1136 ✗ static void draw_response(AVFilterContext *ctx, AVFrame *out, int sample_rate)
1137 {
1138 ✗ AudioIIRContext *s = ctx->priv;
1139 ✗ double *mag, *phase, *temp, *delay, min = DBL_MAX, max = -DBL_MAX;
1140 ✗ double min_delay = DBL_MAX, max_delay = -DBL_MAX, min_phase, max_phase;
1141 ✗ int prev_ymag = -1, prev_yphase = -1, prev_ydelay = -1;
1142 char text[32];
1143 int ch, i;
1144
1145 ✗ memset(out->data[0], 0, s->h * out->linesize[0]);
1146
1147 ✗ phase = av_malloc_array(s->w, sizeof(*phase));
1148 ✗ temp = av_malloc_array(s->w, sizeof(*temp));
1149 ✗ mag = av_malloc_array(s->w, sizeof(*mag));
1150 ✗ delay = av_malloc_array(s->w, sizeof(*delay));
1151 ✗ if (!mag || !phase || !delay || !temp)
1152 ✗ goto end;
1153
1154 ✗ ch = av_clip(s->ir_channel, 0, s->channels - 1);
1155 ✗ for (i = 0; i < s->w; i++) {
1156 ✗ const double *b = s->iir[ch].ab[0];
1157 ✗ const double *a = s->iir[ch].ab[1];
1158 ✗ const int nb_b = s->iir[ch].nb_ab[0];
1159 ✗ const int nb_a = s->iir[ch].nb_ab[1];
1160 ✗ double w = i * M_PI / (s->w - 1);
1161 double m, p;
1162
1163 ✗ get_response(ch, s->format, w, b, a, nb_b, nb_a, &m, &p);
1164
1165 ✗ mag[i] = s->iir[ch].g * m;
1166 ✗ phase[i] = p;
1167 ✗ min = fmin(min, mag[i]);
1168 ✗ max = fmax(max, mag[i]);
1169 }
1170
1171 ✗ temp[0] = 0.;
1172 ✗ for (i = 0; i < s->w - 1; i++) {
1173 ✗ double d = phase[i] - phase[i + 1];
1174 ✗ temp[i + 1] = ceil(fabs(d) / (2. * M_PI)) * 2. * M_PI * ((d > M_PI) - (d < -M_PI));
1175 }
1176
1177 ✗ min_phase = phase[0];
1178 ✗ max_phase = phase[0];
1179 ✗ for (i = 1; i < s->w; i++) {
1180 ✗ temp[i] += temp[i - 1];
1181 ✗ phase[i] += temp[i];
1182 ✗ min_phase = fmin(min_phase, phase[i]);
1183 ✗ max_phase = fmax(max_phase, phase[i]);
1184 }
1185
1186 ✗ for (i = 0; i < s->w - 1; i++) {
1187 ✗ double div = s->w / (double)sample_rate;
1188
1189 ✗ delay[i + 1] = -(phase[i] - phase[i + 1]) / div;
1190 ✗ min_delay = fmin(min_delay, delay[i + 1]);
1191 ✗ max_delay = fmax(max_delay, delay[i + 1]);
1192 }
1193 ✗ delay[0] = delay[1];
1194
1195 ✗ for (i = 0; i < s->w; i++) {
1196 ✗ int ymag = mag[i] / max * (s->h - 1);
1197 ✗ int ydelay = (delay[i] - min_delay) / (max_delay - min_delay) * (s->h - 1);
1198 ✗ int yphase = (phase[i] - min_phase) / (max_phase - min_phase) * (s->h - 1);
1199
1200 ✗ ymag = s->h - 1 - av_clip(ymag, 0, s->h - 1);
1201 ✗ yphase = s->h - 1 - av_clip(yphase, 0, s->h - 1);
1202 ✗ ydelay = s->h - 1 - av_clip(ydelay, 0, s->h - 1);
1203
1204 ✗ if (prev_ymag < 0)
1205 ✗ prev_ymag = ymag;
1206 ✗ if (prev_yphase < 0)
1207 ✗ prev_yphase = yphase;
1208 ✗ if (prev_ydelay < 0)
1209 ✗ prev_ydelay = ydelay;
1210
1211 ✗ draw_line(out, i, ymag, FFMAX(i - 1, 0), prev_ymag, 0xFFFF00FF);
1212 ✗ draw_line(out, i, yphase, FFMAX(i - 1, 0), prev_yphase, 0xFF00FF00);
1213 ✗ draw_line(out, i, ydelay, FFMAX(i - 1, 0), prev_ydelay, 0xFF00FFFF);
1214
1215 ✗ prev_ymag = ymag;
1216 ✗ prev_yphase = yphase;
1217 ✗ prev_ydelay = ydelay;
1218 }
1219
1220 ✗ if (s->w > 400 && s->h > 100) {
1221 ✗ drawtext(out, 2, 2, "Max Magnitude:", 0xDDDDDDDD);
1222 ✗ snprintf(text, sizeof(text), "%.2f", max);
1223 ✗ drawtext(out, 15 * 8 + 2, 2, text, 0xDDDDDDDD);
1224
1225 ✗ drawtext(out, 2, 12, "Min Magnitude:", 0xDDDDDDDD);
1226 ✗ snprintf(text, sizeof(text), "%.2f", min);
1227 ✗ drawtext(out, 15 * 8 + 2, 12, text, 0xDDDDDDDD);
1228
1229 ✗ drawtext(out, 2, 22, "Max Phase:", 0xDDDDDDDD);
1230 ✗ snprintf(text, sizeof(text), "%.2f", max_phase);
1231 ✗ drawtext(out, 15 * 8 + 2, 22, text, 0xDDDDDDDD);
1232
1233 ✗ drawtext(out, 2, 32, "Min Phase:", 0xDDDDDDDD);
1234 ✗ snprintf(text, sizeof(text), "%.2f", min_phase);
1235 ✗ drawtext(out, 15 * 8 + 2, 32, text, 0xDDDDDDDD);
1236
1237 ✗ drawtext(out, 2, 42, "Max Delay:", 0xDDDDDDDD);
1238 ✗ snprintf(text, sizeof(text), "%.2f", max_delay);
1239 ✗ drawtext(out, 11 * 8 + 2, 42, text, 0xDDDDDDDD);
1240
1241 ✗ drawtext(out, 2, 52, "Min Delay:", 0xDDDDDDDD);
1242 ✗ snprintf(text, sizeof(text), "%.2f", min_delay);
1243 ✗ drawtext(out, 11 * 8 + 2, 52, text, 0xDDDDDDDD);
1244 }
1245
1246 ✗ end:
1247 ✗ av_free(delay);
1248 ✗ av_free(temp);
1249 ✗ av_free(phase);
1250 ✗ av_free(mag);
1251 ✗ }
1252
1253 ✗ static int config_output(AVFilterLink *outlink)
1254 {
1255 ✗ AVFilterContext *ctx = outlink->src;
1256 ✗ AudioIIRContext *s = ctx->priv;
1257 ✗ AVFilterLink *inlink = ctx->inputs[0];
1258 int ch, ret, i;
1259
1260 ✗ s->channels = inlink->ch_layout.nb_channels;
1261 ✗ s->iir = av_calloc(s->channels, sizeof(*s->iir));
1262 ✗ if (!s->iir)
1263 ✗ return AVERROR(ENOMEM);
1264
1265 ✗ ret = read_gains(ctx, s->g_str, inlink->ch_layout.nb_channels);
1266 ✗ if (ret < 0)
1267 ✗ return ret;
1268
1269 ✗ ret = read_channels(ctx, inlink->ch_layout.nb_channels, s->a_str, 0);
1270 ✗ if (ret < 0)
1271 ✗ return ret;
1272
1273 ✗ ret = read_channels(ctx, inlink->ch_layout.nb_channels, s->b_str, 1);
1274 ✗ if (ret < 0)
1275 ✗ return ret;
1276
1277 ✗ if (s->format == -1) {
1278 ✗ convert_sf2tf(ctx, inlink->ch_layout.nb_channels);
1279 ✗ s->format = 0;
1280 ✗ } else if (s->format == 2) {
1281 ✗ convert_pr2zp(ctx, inlink->ch_layout.nb_channels);
1282 ✗ } else if (s->format == 3) {
1283 ✗ convert_pd2zp(ctx, inlink->ch_layout.nb_channels);
1284 ✗ } else if (s->format == 4) {
1285 ✗ convert_sp2zp(ctx, inlink->ch_layout.nb_channels);
1286 }
1287 ✗ if (s->format > 0) {
1288 ✗ check_stability(ctx, inlink->ch_layout.nb_channels);
1289 }
1290
1291 ✗ av_frame_free(&s->video);
1292 ✗ if (s->response) {
1293 ✗ s->video = ff_get_video_buffer(ctx->outputs[1], s->w, s->h);
1294 ✗ if (!s->video)
1295 ✗ return AVERROR(ENOMEM);
1296
1297 ✗ draw_response(ctx, s->video, inlink->sample_rate);
1298 }
1299
1300 ✗ if (s->format == 0)
1301 ✗ av_log(ctx, AV_LOG_WARNING, "transfer function coefficients format is not recommended for too high number of zeros/poles.\n");
1302
1303 ✗ if (s->format > 0 && s->process == 0) {
1304 ✗ av_log(ctx, AV_LOG_WARNING, "Direct processing is not recommended for zp coefficients format.\n");
1305
1306 ✗ ret = convert_zp2tf(ctx, inlink->ch_layout.nb_channels);
1307 ✗ if (ret < 0)
1308 ✗ return ret;
1309 ✗ } else if (s->format == -2 && s->process > 0) {
1310 ✗ av_log(ctx, AV_LOG_ERROR, "Only direct processing is implemented for lattice-ladder function.\n");
1311 ✗ return AVERROR_PATCHWELCOME;
1312 ✗ } else if (s->format <= 0 && s->process == 1) {
1313 ✗ av_log(ctx, AV_LOG_ERROR, "Serial processing is not implemented for transfer function.\n");
1314 ✗ return AVERROR_PATCHWELCOME;
1315 ✗ } else if (s->format <= 0 && s->process == 2) {
1316 ✗ av_log(ctx, AV_LOG_ERROR, "Parallel processing is not implemented for transfer function.\n");
1317 ✗ return AVERROR_PATCHWELCOME;
1318 ✗ } else if (s->format > 0 && s->process == 1) {
1319 ✗ ret = decompose_zp2biquads(ctx, inlink->ch_layout.nb_channels);
1320 ✗ if (ret < 0)
1321 ✗ return ret;
1322 ✗ } else if (s->format > 0 && s->process == 2) {
1323 ✗ if (s->precision > 1)
1324 ✗ av_log(ctx, AV_LOG_WARNING, "Parallel processing is not recommended for fixed-point precisions.\n");
1325 ✗ ret = decompose_zp2biquads(ctx, inlink->ch_layout.nb_channels);
1326 ✗ if (ret < 0)
1327 ✗ return ret;
1328 ✗ ret = convert_serial2parallel(ctx, inlink->ch_layout.nb_channels);
1329 ✗ if (ret < 0)
1330 ✗ return ret;
1331 }
1332
1333 ✗ for (ch = 0; s->format == -2 && ch < inlink->ch_layout.nb_channels; ch++) {
1334 ✗ IIRChannel *iir = &s->iir[ch];
1335
1336 ✗ if (iir->nb_ab[0] != iir->nb_ab[1] + 1) {
1337 ✗ av_log(ctx, AV_LOG_ERROR, "Number of ladder coefficients must be one more than number of reflection coefficients.\n");
1338 ✗ return AVERROR(EINVAL);
1339 }
1340 }
1341
1342 ✗ for (ch = 0; s->format == 0 && ch < inlink->ch_layout.nb_channels; ch++) {
1343 ✗ IIRChannel *iir = &s->iir[ch];
1344
1345 ✗ for (i = 1; i < iir->nb_ab[0]; i++) {
1346 ✗ iir->ab[0][i] /= iir->ab[0][0];
1347 }
1348
1349 ✗ iir->ab[0][0] = 1.0;
1350 ✗ for (i = 0; i < iir->nb_ab[1]; i++) {
1351 ✗ iir->ab[1][i] *= iir->g;
1352 }
1353
1354 ✗ normalize_coeffs(ctx, ch);
1355 }
1356
1357 ✗ switch (inlink->format) {
1358 ✗ case AV_SAMPLE_FMT_DBLP: s->iir_channel = s->process == 2 ? iir_ch_parallel_dblp : s->process == 1 ? iir_ch_serial_dblp : iir_ch_dblp; break;
1359 ✗ case AV_SAMPLE_FMT_FLTP: s->iir_channel = s->process == 2 ? iir_ch_parallel_fltp : s->process == 1 ? iir_ch_serial_fltp : iir_ch_fltp; break;
1360 ✗ case AV_SAMPLE_FMT_S32P: s->iir_channel = s->process == 2 ? iir_ch_parallel_s32p : s->process == 1 ? iir_ch_serial_s32p : iir_ch_s32p; break;
1361 ✗ case AV_SAMPLE_FMT_S16P: s->iir_channel = s->process == 2 ? iir_ch_parallel_s16p : s->process == 1 ? iir_ch_serial_s16p : iir_ch_s16p; break;
1362 }
1363
1364 ✗ if (s->format == -2) {
1365 ✗ switch (inlink->format) {
1366 ✗ case AV_SAMPLE_FMT_DBLP: s->iir_channel = iir_ch_lattice_dblp; break;
1367 ✗ case AV_SAMPLE_FMT_FLTP: s->iir_channel = iir_ch_lattice_fltp; break;
1368 ✗ case AV_SAMPLE_FMT_S32P: s->iir_channel = iir_ch_lattice_s32p; break;
1369 ✗ case AV_SAMPLE_FMT_S16P: s->iir_channel = iir_ch_lattice_s16p; break;
1370 }
1371 }
1372
1373 ✗ return 0;
1374 }
1375
1376 ✗ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
1377 {
1378 ✗ AVFilterContext *ctx = inlink->dst;
1379 ✗ AudioIIRContext *s = ctx->priv;
1380 ✗ AVFilterLink *outlink = ctx->outputs[0];
1381 ThreadData td;
1382 AVFrame *out;
1383 int ch, ret;
1384
1385 ✗ if (av_frame_is_writable(in) && s->process != 2) {
1386 ✗ out = in;
1387 } else {
1388 ✗ out = ff_get_audio_buffer(outlink, in->nb_samples);
1389 ✗ if (!out) {
1390 ✗ av_frame_free(&in);
1391 ✗ return AVERROR(ENOMEM);
1392 }
1393 ✗ av_frame_copy_props(out, in);
1394 }
1395
1396 ✗ td.in = in;
1397 ✗ td.out = out;
1398 ✗ ff_filter_execute(ctx, s->iir_channel, &td, NULL, outlink->ch_layout.nb_channels);
1399
1400 ✗ for (ch = 0; ch < outlink->ch_layout.nb_channels; ch++) {
1401 ✗ if (s->iir[ch].clippings > 0)
1402 ✗ av_log(ctx, AV_LOG_WARNING, "Channel %d clipping %d times. Please reduce gain.\n",
1403 ✗ ch, s->iir[ch].clippings);
1404 ✗ s->iir[ch].clippings = 0;
1405 }
1406
1407 ✗ if (in != out)
1408 ✗ av_frame_free(&in);
1409
1410 ✗ if (s->response) {
1411 ✗ AVFilterLink *outlink = ctx->outputs[1];
1412 ✗ int64_t old_pts = s->video->pts;
1413 ✗ int64_t new_pts = av_rescale_q(out->pts, ctx->inputs[0]->time_base, outlink->time_base);
1414
1415 ✗ if (new_pts > old_pts) {
1416 AVFrame *clone;
1417
1418 ✗ s->video->pts = new_pts;
1419 ✗ clone = av_frame_clone(s->video);
1420 ✗ if (!clone)
1421 ✗ return AVERROR(ENOMEM);
1422 ✗ ret = ff_filter_frame(outlink, clone);
1423 ✗ if (ret < 0)
1424 ✗ return ret;
1425 }
1426 }
1427
1428 ✗ return ff_filter_frame(outlink, out);
1429 }
1430
1431 ✗ static int config_video(AVFilterLink *outlink)
1432 {
1433 ✗ FilterLink *l = ff_filter_link(outlink);
1434 ✗ AVFilterContext *ctx = outlink->src;
1435 ✗ AudioIIRContext *s = ctx->priv;
1436
1437 ✗ outlink->sample_aspect_ratio = (AVRational){1,1};
1438 ✗ outlink->w = s->w;
1439 ✗ outlink->h = s->h;
1440 ✗ l->frame_rate = s->rate;
1441 ✗ outlink->time_base = av_inv_q(l->frame_rate);
1442
1443 ✗ return 0;
1444 }
1445
1446 ✗ static av_cold int init(AVFilterContext *ctx)
1447 {
1448 ✗ AudioIIRContext *s = ctx->priv;
1449 AVFilterPad pad, vpad;
1450 int ret;
1451
1452 ✗ if (!s->a_str || !s->b_str || !s->g_str) {
1453 ✗ av_log(ctx, AV_LOG_ERROR, "Valid coefficients are mandatory.\n");
1454 ✗ return AVERROR(EINVAL);
1455 }
1456
1457 ✗ switch (s->precision) {
1458 ✗ case 0: s->sample_format = AV_SAMPLE_FMT_DBLP; break;
1459 ✗ case 1: s->sample_format = AV_SAMPLE_FMT_FLTP; break;
1460 ✗ case 2: s->sample_format = AV_SAMPLE_FMT_S32P; break;
1461 ✗ case 3: s->sample_format = AV_SAMPLE_FMT_S16P; break;
1462 ✗ default: return AVERROR_BUG;
1463 }
1464
1465 ✗ pad = (AVFilterPad){
1466 .name = "default",
1467 .type = AVMEDIA_TYPE_AUDIO,
1468 .config_props = config_output,
1469 };
1470
1471 ✗ ret = ff_append_outpad(ctx, &pad);
1472 ✗ if (ret < 0)
1473 ✗ return ret;
1474
1475 ✗ if (s->response) {
1476 ✗ vpad = (AVFilterPad){
1477 .name = "filter_response",
1478 .type = AVMEDIA_TYPE_VIDEO,
1479 .config_props = config_video,
1480 };
1481
1482 ✗ ret = ff_append_outpad(ctx, &vpad);
1483 ✗ if (ret < 0)
1484 ✗ return ret;
1485 }
1486
1487 ✗ return 0;
1488 }
1489
1490 ✗ static av_cold void uninit(AVFilterContext *ctx)
1491 {
1492 ✗ AudioIIRContext *s = ctx->priv;
1493 int ch;
1494
1495 ✗ if (s->iir) {
1496 ✗ for (ch = 0; ch < s->channels; ch++) {
1497 ✗ IIRChannel *iir = &s->iir[ch];
1498 ✗ av_freep(&iir->ab[0]);
1499 ✗ av_freep(&iir->ab[1]);
1500 ✗ av_freep(&iir->cache[0]);
1501 ✗ av_freep(&iir->cache[1]);
1502 ✗ av_freep(&iir->biquads);
1503 }
1504 }
1505 ✗ av_freep(&s->iir);
1506
1507 ✗ av_frame_free(&s->video);
1508 ✗ }
1509
1510 static const AVFilterPad inputs[] = {
1511 {
1512 .name = "default",
1513 .type = AVMEDIA_TYPE_AUDIO,
1514 .filter_frame = filter_frame,
1515 },
1516 };
1517
1518 #define OFFSET(x) offsetof(AudioIIRContext, x)
1519 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
1520 #define VF AV_OPT_FLAG_VIDEO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
1521
1522 static const AVOption aiir_options[] = {
1523 { "zeros", "set B/numerator/zeros/reflection coefficients", OFFSET(b_str), AV_OPT_TYPE_STRING, {.str="1+0i 1-0i"}, 0, 0, AF },
1524 { "z", "set B/numerator/zeros/reflection coefficients", OFFSET(b_str), AV_OPT_TYPE_STRING, {.str="1+0i 1-0i"}, 0, 0, AF },
1525 { "poles", "set A/denominator/poles/ladder coefficients", OFFSET(a_str), AV_OPT_TYPE_STRING, {.str="1+0i 1-0i"}, 0, 0, AF },
1526 { "p", "set A/denominator/poles/ladder coefficients", OFFSET(a_str), AV_OPT_TYPE_STRING, {.str="1+0i 1-0i"}, 0, 0, AF },
1527 { "gains", "set channels gains", OFFSET(g_str), AV_OPT_TYPE_STRING, {.str="1|1"}, 0, 0, AF },
1528 { "k", "set channels gains", OFFSET(g_str), AV_OPT_TYPE_STRING, {.str="1|1"}, 0, 0, AF },
1529 { "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, AF },
1530 { "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, AF },
1531 { "format", "set coefficients format", OFFSET(format), AV_OPT_TYPE_INT, {.i64=1}, -2, 4, AF, .unit = "format" },
1532 { "f", "set coefficients format", OFFSET(format), AV_OPT_TYPE_INT, {.i64=1}, -2, 4, AF, .unit = "format" },
1533 { "ll", "lattice-ladder function", 0, AV_OPT_TYPE_CONST, {.i64=-2}, 0, 0, AF, .unit = "format" },
1534 { "sf", "analog transfer function", 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, AF, .unit = "format" },
1535 { "tf", "digital transfer function", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, .unit = "format" },
1536 { "zp", "Z-plane zeros/poles", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, .unit = "format" },
1537 { "pr", "Z-plane zeros/poles (polar radians)", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, .unit = "format" },
1538 { "pd", "Z-plane zeros/poles (polar degrees)", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, .unit = "format" },
1539 { "sp", "S-plane zeros/poles", 0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, AF, .unit = "format" },
1540 { "process", "set kind of processing", OFFSET(process), AV_OPT_TYPE_INT, {.i64=1}, 0, 2, AF, .unit = "process" },
1541 { "r", "set kind of processing", OFFSET(process), AV_OPT_TYPE_INT, {.i64=1}, 0, 2, AF, .unit = "process" },
1542 { "d", "direct", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, .unit = "process" },
1543 { "s", "serial", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, .unit = "process" },
1544 { "p", "parallel", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, .unit = "process" },
1545 { "precision", "set filtering precision", OFFSET(precision),AV_OPT_TYPE_INT, {.i64=0}, 0, 3, AF, .unit = "precision" },
1546 { "e", "set precision", OFFSET(precision),AV_OPT_TYPE_INT, {.i64=0}, 0, 3, AF, .unit = "precision" },
1547 { "dbl", "double-precision floating-point", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, .unit = "precision" },
1548 { "flt", "single-precision floating-point", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, .unit = "precision" },
1549 { "i32", "32-bit integers", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, .unit = "precision" },
1550 { "i16", "16-bit integers", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, .unit = "precision" },
1551 { "normalize", "normalize coefficients", OFFSET(normalize),AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, AF },
1552 { "n", "normalize coefficients", OFFSET(normalize),AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, AF },
1553 { "mix", "set mix", OFFSET(mix), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, AF },
1554 { "response", "show IR frequency response", OFFSET(response), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, VF },
1555 { "channel", "set IR channel to display frequency response", OFFSET(ir_channel), AV_OPT_TYPE_INT, {.i64=0}, 0, 1024, VF },
1556 { "size", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str = "hd720"}, 0, 0, VF },
1557 { "rate", "set video rate", OFFSET(rate), AV_OPT_TYPE_VIDEO_RATE, {.str = "25"}, 0, INT32_MAX, VF },
1558 { NULL },
1559 };
1560
1561 AVFILTER_DEFINE_CLASS(aiir);
1562
1563 const AVFilter ff_af_aiir = {
1564 .name = "aiir",
1565 .description = NULL_IF_CONFIG_SMALL("Apply Infinite Impulse Response filter with supplied coefficients."),
1566 .priv_size = sizeof(AudioIIRContext),
1567 .priv_class = &aiir_class,
1568 .init = init,
1569 .uninit = uninit,
1570 FILTER_INPUTS(inputs),
1571 FILTER_QUERY_FUNC2(query_formats),
1572 .flags = AVFILTER_FLAG_DYNAMIC_OUTPUTS |
1573 AVFILTER_FLAG_SLICE_THREADS,
1574 };
1575