FFmpeg coverage


Directory: ../../../ffmpeg/
File: src/libavfilter/af_aiir.c
Date: 2024-07-24 19:24:46
Exec Total Coverage
Lines: 0 752 0.0%
Functions: 0 47 0.0%
Branches: 0 561 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 "formats.h"
31 #include "internal.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(AVFilterContext *ctx)
82 {
83 AudioIIRContext *s = ctx->priv;
84 AVFilterFormats *formats;
85 enum AVSampleFormat sample_fmts[] = {
86 AV_SAMPLE_FMT_DBLP,
87 AV_SAMPLE_FMT_NONE
88 };
89 static const enum AVPixelFormat pix_fmts[] = {
90 AV_PIX_FMT_RGB0,
91 AV_PIX_FMT_NONE
92 };
93 int ret;
94
95 if (s->response) {
96 AVFilterLink *videolink = ctx->outputs[1];
97
98 formats = ff_make_format_list(pix_fmts);
99 if ((ret = ff_formats_ref(formats, &videolink->incfg.formats)) < 0)
100 return ret;
101 }
102
103 ret = ff_set_common_all_channel_counts(ctx);
104 if (ret < 0)
105 return ret;
106
107 sample_fmts[0] = s->sample_format;
108 ret = ff_set_common_formats_from_list(ctx, sample_fmts);
109 if (ret < 0)
110 return ret;
111
112 return ff_set_common_all_samplerates(ctx);
113 }
114
115 #define IIR_CH(name, type, min, max, need_clipping) \
116 static int iir_ch_## name(AVFilterContext *ctx, void *arg, int ch, int nb_jobs) \
117 { \
118 AudioIIRContext *s = ctx->priv; \
119 const double ig = s->dry_gain; \
120 const double og = s->wet_gain; \
121 const double mix = s->mix; \
122 ThreadData *td = arg; \
123 AVFrame *in = td->in, *out = td->out; \
124 const type *src = (const type *)in->extended_data[ch]; \
125 double *oc = (double *)s->iir[ch].cache[0]; \
126 double *ic = (double *)s->iir[ch].cache[1]; \
127 const int nb_a = s->iir[ch].nb_ab[0]; \
128 const int nb_b = s->iir[ch].nb_ab[1]; \
129 const double *a = s->iir[ch].ab[0]; \
130 const double *b = s->iir[ch].ab[1]; \
131 const double g = s->iir[ch].g; \
132 int *clippings = &s->iir[ch].clippings; \
133 type *dst = (type *)out->extended_data[ch]; \
134 int n; \
135 \
136 for (n = 0; n < in->nb_samples; n++) { \
137 double sample = 0.; \
138 int x; \
139 \
140 memmove(&ic[1], &ic[0], (nb_b - 1) * sizeof(*ic)); \
141 memmove(&oc[1], &oc[0], (nb_a - 1) * sizeof(*oc)); \
142 ic[0] = src[n] * ig; \
143 for (x = 0; x < nb_b; x++) \
144 sample += b[x] * ic[x]; \
145 \
146 for (x = 1; x < nb_a; x++) \
147 sample -= a[x] * oc[x]; \
148 \
149 oc[0] = sample; \
150 sample *= og * g; \
151 sample = sample * mix + ic[0] * (1. - mix); \
152 if (need_clipping && sample < min) { \
153 (*clippings)++; \
154 dst[n] = min; \
155 } else if (need_clipping && sample > max) { \
156 (*clippings)++; \
157 dst[n] = max; \
158 } else { \
159 dst[n] = sample; \
160 } \
161 } \
162 \
163 return 0; \
164 }
165
166 IIR_CH(s16p, int16_t, INT16_MIN, INT16_MAX, 1)
167 IIR_CH(s32p, int32_t, INT32_MIN, INT32_MAX, 1)
168 IIR_CH(fltp, float, -1., 1., 0)
169 IIR_CH(dblp, double, -1., 1., 0)
170
171 #define SERIAL_IIR_CH(name, type, min, max, need_clipping) \
172 static int iir_ch_serial_## name(AVFilterContext *ctx, void *arg, \
173 int ch, int nb_jobs) \
174 { \
175 AudioIIRContext *s = ctx->priv; \
176 const double ig = s->dry_gain; \
177 const double og = s->wet_gain; \
178 const double mix = s->mix; \
179 const double imix = 1. - mix; \
180 ThreadData *td = arg; \
181 AVFrame *in = td->in, *out = td->out; \
182 const type *src = (const type *)in->extended_data[ch]; \
183 type *dst = (type *)out->extended_data[ch]; \
184 IIRChannel *iir = &s->iir[ch]; \
185 const double g = iir->g; \
186 int *clippings = &iir->clippings; \
187 int nb_biquads = (FFMAX(iir->nb_ab[0], iir->nb_ab[1]) + 1) / 2; \
188 int n, i; \
189 \
190 for (i = nb_biquads - 1; i >= 0; i--) { \
191 const double a1 = -iir->biquads[i].a[1]; \
192 const double a2 = -iir->biquads[i].a[2]; \
193 const double b0 = iir->biquads[i].b[0]; \
194 const double b1 = iir->biquads[i].b[1]; \
195 const double b2 = iir->biquads[i].b[2]; \
196 double w1 = iir->biquads[i].w1; \
197 double w2 = iir->biquads[i].w2; \
198 \
199 for (n = 0; n < in->nb_samples; n++) { \
200 double i0 = ig * (i ? dst[n] : src[n]); \
201 double o0 = i0 * b0 + w1; \
202 \
203 w1 = b1 * i0 + w2 + a1 * o0; \
204 w2 = b2 * i0 + a2 * o0; \
205 o0 *= og * g; \
206 \
207 o0 = o0 * mix + imix * i0; \
208 if (need_clipping && o0 < min) { \
209 (*clippings)++; \
210 dst[n] = min; \
211 } else if (need_clipping && o0 > max) { \
212 (*clippings)++; \
213 dst[n] = max; \
214 } else { \
215 dst[n] = o0; \
216 } \
217 } \
218 iir->biquads[i].w1 = w1; \
219 iir->biquads[i].w2 = w2; \
220 } \
221 \
222 return 0; \
223 }
224
225 SERIAL_IIR_CH(s16p, int16_t, INT16_MIN, INT16_MAX, 1)
226 SERIAL_IIR_CH(s32p, int32_t, INT32_MIN, INT32_MAX, 1)
227 SERIAL_IIR_CH(fltp, float, -1., 1., 0)
228 SERIAL_IIR_CH(dblp, double, -1., 1., 0)
229
230 #define PARALLEL_IIR_CH(name, type, min, max, need_clipping) \
231 static int iir_ch_parallel_## name(AVFilterContext *ctx, void *arg, \
232 int ch, int nb_jobs) \
233 { \
234 AudioIIRContext *s = ctx->priv; \
235 const double ig = s->dry_gain; \
236 const double og = s->wet_gain; \
237 const double mix = s->mix; \
238 const double imix = 1. - mix; \
239 ThreadData *td = arg; \
240 AVFrame *in = td->in, *out = td->out; \
241 const type *src = (const type *)in->extended_data[ch]; \
242 type *dst = (type *)out->extended_data[ch]; \
243 IIRChannel *iir = &s->iir[ch]; \
244 const double g = iir->g; \
245 const double fir = iir->fir; \
246 int *clippings = &iir->clippings; \
247 int nb_biquads = (FFMAX(iir->nb_ab[0], iir->nb_ab[1]) + 1) / 2; \
248 int n, i; \
249 \
250 for (i = 0; i < nb_biquads; i++) { \
251 const double a1 = -iir->biquads[i].a[1]; \
252 const double a2 = -iir->biquads[i].a[2]; \
253 const double b1 = iir->biquads[i].b[1]; \
254 const double b2 = iir->biquads[i].b[2]; \
255 double w1 = iir->biquads[i].w1; \
256 double w2 = iir->biquads[i].w2; \
257 \
258 for (n = 0; n < in->nb_samples; n++) { \
259 double i0 = ig * src[n]; \
260 double o0 = w1; \
261 \
262 w1 = b1 * i0 + w2 + a1 * o0; \
263 w2 = b2 * i0 + a2 * o0; \
264 o0 *= og * g; \
265 o0 += dst[n]; \
266 \
267 if (need_clipping && o0 < min) { \
268 (*clippings)++; \
269 dst[n] = min; \
270 } else if (need_clipping && o0 > max) { \
271 (*clippings)++; \
272 dst[n] = max; \
273 } else { \
274 dst[n] = o0; \
275 } \
276 } \
277 iir->biquads[i].w1 = w1; \
278 iir->biquads[i].w2 = w2; \
279 } \
280 \
281 for (n = 0; n < in->nb_samples; n++) { \
282 dst[n] += fir * src[n]; \
283 dst[n] = dst[n] * mix + imix * src[n]; \
284 } \
285 \
286 return 0; \
287 }
288
289 PARALLEL_IIR_CH(s16p, int16_t, INT16_MIN, INT16_MAX, 1)
290 PARALLEL_IIR_CH(s32p, int32_t, INT32_MIN, INT32_MAX, 1)
291 PARALLEL_IIR_CH(fltp, float, -1., 1., 0)
292 PARALLEL_IIR_CH(dblp, double, -1., 1., 0)
293
294 #define LATTICE_IIR_CH(name, type, min, max, need_clipping) \
295 static int iir_ch_lattice_## name(AVFilterContext *ctx, void *arg, \
296 int ch, int nb_jobs) \
297 { \
298 AudioIIRContext *s = ctx->priv; \
299 const double ig = s->dry_gain; \
300 const double og = s->wet_gain; \
301 const double mix = s->mix; \
302 ThreadData *td = arg; \
303 AVFrame *in = td->in, *out = td->out; \
304 const type *src = (const type *)in->extended_data[ch]; \
305 double n0, n1, p0, *x = (double *)s->iir[ch].cache[0]; \
306 const int nb_stages = s->iir[ch].nb_ab[1]; \
307 const double *v = s->iir[ch].ab[0]; \
308 const double *k = s->iir[ch].ab[1]; \
309 const double g = s->iir[ch].g; \
310 int *clippings = &s->iir[ch].clippings; \
311 type *dst = (type *)out->extended_data[ch]; \
312 int n; \
313 \
314 for (n = 0; n < in->nb_samples; n++) { \
315 const double in = src[n] * ig; \
316 double out = 0.; \
317 \
318 n1 = in; \
319 for (int i = nb_stages - 1; i >= 0; i--) { \
320 n0 = n1 - k[i] * x[i]; \
321 p0 = n0 * k[i] + x[i]; \
322 out += p0 * v[i+1]; \
323 x[i] = p0; \
324 n1 = n0; \
325 } \
326 \
327 out += n1 * v[0]; \
328 memmove(&x[1], &x[0], nb_stages * sizeof(*x)); \
329 x[0] = n1; \
330 out *= og * g; \
331 out = out * mix + in * (1. - mix); \
332 if (need_clipping && out < min) { \
333 (*clippings)++; \
334 dst[n] = min; \
335 } else if (need_clipping && out > max) { \
336 (*clippings)++; \
337 dst[n] = max; \
338 } else { \
339 dst[n] = out; \
340 } \
341 } \
342 \
343 return 0; \
344 }
345
346 LATTICE_IIR_CH(s16p, int16_t, INT16_MIN, INT16_MAX, 1)
347 LATTICE_IIR_CH(s32p, int32_t, INT32_MIN, INT32_MAX, 1)
348 LATTICE_IIR_CH(fltp, float, -1., 1., 0)
349 LATTICE_IIR_CH(dblp, double, -1., 1., 0)
350
351 static void count_coefficients(char *item_str, int *nb_items)
352 {
353 char *p;
354
355 if (!item_str)
356 return;
357
358 *nb_items = 1;
359 for (p = item_str; *p && *p != '|'; p++) {
360 if (*p == ' ')
361 (*nb_items)++;
362 }
363 }
364
365 static int read_gains(AVFilterContext *ctx, char *item_str, int nb_items)
366 {
367 AudioIIRContext *s = ctx->priv;
368 char *p, *arg, *old_str, *prev_arg = NULL, *saveptr = NULL;
369 int i;
370
371 p = old_str = av_strdup(item_str);
372 if (!p)
373 return AVERROR(ENOMEM);
374 for (i = 0; i < nb_items; i++) {
375 if (!(arg = av_strtok(p, "|", &saveptr)))
376 arg = prev_arg;
377
378 if (!arg) {
379 av_freep(&old_str);
380 return AVERROR(EINVAL);
381 }
382
383 p = NULL;
384 if (av_sscanf(arg, "%lf", &s->iir[i].g) != 1) {
385 av_log(ctx, AV_LOG_ERROR, "Invalid gains supplied: %s\n", arg);
386 av_freep(&old_str);
387 return AVERROR(EINVAL);
388 }
389
390 prev_arg = arg;
391 }
392
393 av_freep(&old_str);
394
395 return 0;
396 }
397
398 static int read_tf_coefficients(AVFilterContext *ctx, char *item_str, int nb_items, double *dst)
399 {
400 char *p, *arg, *old_str, *saveptr = NULL;
401 int i;
402
403 p = old_str = av_strdup(item_str);
404 if (!p)
405 return AVERROR(ENOMEM);
406 for (i = 0; i < nb_items; i++) {
407 if (!(arg = av_strtok(p, " ", &saveptr)))
408 break;
409
410 p = NULL;
411 if (av_sscanf(arg, "%lf", &dst[i]) != 1) {
412 av_log(ctx, AV_LOG_ERROR, "Invalid coefficients supplied: %s\n", arg);
413 av_freep(&old_str);
414 return AVERROR(EINVAL);
415 }
416 }
417
418 av_freep(&old_str);
419
420 return 0;
421 }
422
423 static int read_zp_coefficients(AVFilterContext *ctx, char *item_str, int nb_items, double *dst, const char *format)
424 {
425 char *p, *arg, *old_str, *saveptr = NULL;
426 int i;
427
428 p = old_str = av_strdup(item_str);
429 if (!p)
430 return AVERROR(ENOMEM);
431 for (i = 0; i < nb_items; i++) {
432 if (!(arg = av_strtok(p, " ", &saveptr)))
433 break;
434
435 p = NULL;
436 if (av_sscanf(arg, format, &dst[i*2], &dst[i*2+1]) != 2) {
437 av_log(ctx, AV_LOG_ERROR, "Invalid coefficients supplied: %s\n", arg);
438 av_freep(&old_str);
439 return AVERROR(EINVAL);
440 }
441 }
442
443 av_freep(&old_str);
444
445 return 0;
446 }
447
448 static const char *const format[] = { "%lf", "%lf %lfi", "%lf %lfr", "%lf %lfd", "%lf %lfi" };
449
450 static int read_channels(AVFilterContext *ctx, int channels, uint8_t *item_str, int ab)
451 {
452 AudioIIRContext *s = ctx->priv;
453 char *p, *arg, *old_str, *prev_arg = NULL, *saveptr = NULL;
454 int i, ret;
455
456 p = old_str = av_strdup(item_str);
457 if (!p)
458 return AVERROR(ENOMEM);
459 for (i = 0; i < channels; i++) {
460 IIRChannel *iir = &s->iir[i];
461
462 if (!(arg = av_strtok(p, "|", &saveptr)))
463 arg = prev_arg;
464
465 if (!arg) {
466 av_freep(&old_str);
467 return AVERROR(EINVAL);
468 }
469
470 count_coefficients(arg, &iir->nb_ab[ab]);
471
472 p = NULL;
473 iir->cache[ab] = av_calloc(iir->nb_ab[ab] + 1, sizeof(double));
474 iir->ab[ab] = av_calloc(iir->nb_ab[ab] * (!!s->format + 1), sizeof(double));
475 if (!iir->ab[ab] || !iir->cache[ab]) {
476 av_freep(&old_str);
477 return AVERROR(ENOMEM);
478 }
479
480 if (s->format > 0) {
481 ret = read_zp_coefficients(ctx, arg, iir->nb_ab[ab], iir->ab[ab], format[s->format]);
482 } else {
483 ret = read_tf_coefficients(ctx, arg, iir->nb_ab[ab], iir->ab[ab]);
484 }
485 if (ret < 0) {
486 av_freep(&old_str);
487 return ret;
488 }
489 prev_arg = arg;
490 }
491
492 av_freep(&old_str);
493
494 return 0;
495 }
496
497 static void cmul(double re, double im, double re2, double im2, double *RE, double *IM)
498 {
499 *RE = re * re2 - im * im2;
500 *IM = re * im2 + re2 * im;
501 }
502
503 static int expand(AVFilterContext *ctx, double *pz, int n, double *coefs)
504 {
505 coefs[2 * n] = 1.0;
506
507 for (int i = 1; i <= n; i++) {
508 for (int j = n - i; j < n; j++) {
509 double re, im;
510
511 cmul(coefs[2 * (j + 1)], coefs[2 * (j + 1) + 1],
512 pz[2 * (i - 1)], pz[2 * (i - 1) + 1], &re, &im);
513
514 coefs[2 * j] -= re;
515 coefs[2 * j + 1] -= im;
516 }
517 }
518
519 for (int i = 0; i < n + 1; i++) {
520 if (fabs(coefs[2 * i + 1]) > FLT_EPSILON) {
521 av_log(ctx, AV_LOG_ERROR, "coefs: %f of z^%d is not real; poles/zeros are not complex conjugates.\n",
522 coefs[2 * i + 1], i);
523 return AVERROR(EINVAL);
524 }
525 }
526
527 return 0;
528 }
529
530 static void normalize_coeffs(AVFilterContext *ctx, int ch)
531 {
532 AudioIIRContext *s = ctx->priv;
533 IIRChannel *iir = &s->iir[ch];
534 double sum_den = 0.;
535
536 if (!s->normalize)
537 return;
538
539 for (int i = 0; i < iir->nb_ab[1]; i++) {
540 sum_den += iir->ab[1][i];
541 }
542
543 if (sum_den > 1e-6) {
544 double factor, sum_num = 0.;
545
546 for (int i = 0; i < iir->nb_ab[0]; i++) {
547 sum_num += iir->ab[0][i];
548 }
549
550 factor = sum_num / sum_den;
551
552 for (int i = 0; i < iir->nb_ab[1]; i++) {
553 iir->ab[1][i] *= factor;
554 }
555 }
556 }
557
558 static int convert_zp2tf(AVFilterContext *ctx, int channels)
559 {
560 AudioIIRContext *s = ctx->priv;
561 int ch, i, j, ret = 0;
562
563 for (ch = 0; ch < channels; ch++) {
564 IIRChannel *iir = &s->iir[ch];
565 double *topc, *botc;
566
567 topc = av_calloc((iir->nb_ab[1] + 1) * 2, sizeof(*topc));
568 botc = av_calloc((iir->nb_ab[0] + 1) * 2, sizeof(*botc));
569 if (!topc || !botc) {
570 ret = AVERROR(ENOMEM);
571 goto fail;
572 }
573
574 ret = expand(ctx, iir->ab[0], iir->nb_ab[0], botc);
575 if (ret < 0) {
576 goto fail;
577 }
578
579 ret = expand(ctx, iir->ab[1], iir->nb_ab[1], topc);
580 if (ret < 0) {
581 goto fail;
582 }
583
584 for (j = 0, i = iir->nb_ab[1]; i >= 0; j++, i--) {
585 iir->ab[1][j] = topc[2 * i];
586 }
587 iir->nb_ab[1]++;
588
589 for (j = 0, i = iir->nb_ab[0]; i >= 0; j++, i--) {
590 iir->ab[0][j] = botc[2 * i];
591 }
592 iir->nb_ab[0]++;
593
594 normalize_coeffs(ctx, ch);
595
596 fail:
597 av_free(topc);
598 av_free(botc);
599 if (ret < 0)
600 break;
601 }
602
603 return ret;
604 }
605
606 static int decompose_zp2biquads(AVFilterContext *ctx, int channels)
607 {
608 AudioIIRContext *s = ctx->priv;
609 int ch, ret;
610
611 for (ch = 0; ch < channels; ch++) {
612 IIRChannel *iir = &s->iir[ch];
613 int nb_biquads = (FFMAX(iir->nb_ab[0], iir->nb_ab[1]) + 1) / 2;
614 int current_biquad = 0;
615
616 iir->biquads = av_calloc(nb_biquads, sizeof(BiquadContext));
617 if (!iir->biquads)
618 return AVERROR(ENOMEM);
619
620 while (nb_biquads--) {
621 Pair outmost_pole = { -1, -1 };
622 Pair nearest_zero = { -1, -1 };
623 double zeros[4] = { 0 };
624 double poles[4] = { 0 };
625 double b[6] = { 0 };
626 double a[6] = { 0 };
627 double min_distance = DBL_MAX;
628 double max_mag = 0;
629 double factor;
630 int i;
631
632 for (i = 0; i < iir->nb_ab[0]; i++) {
633 double mag;
634
635 if (isnan(iir->ab[0][2 * i]) || isnan(iir->ab[0][2 * i + 1]))
636 continue;
637 mag = hypot(iir->ab[0][2 * i], iir->ab[0][2 * i + 1]);
638
639 if (mag > max_mag) {
640 max_mag = mag;
641 outmost_pole.a = i;
642 }
643 }
644
645 for (i = 0; i < iir->nb_ab[0]; i++) {
646 if (isnan(iir->ab[0][2 * i]) || isnan(iir->ab[0][2 * i + 1]))
647 continue;
648
649 if (iir->ab[0][2 * i ] == iir->ab[0][2 * outmost_pole.a ] &&
650 iir->ab[0][2 * i + 1] == -iir->ab[0][2 * outmost_pole.a + 1]) {
651 outmost_pole.b = i;
652 break;
653 }
654 }
655
656 av_log(ctx, AV_LOG_VERBOSE, "outmost_pole is %d.%d\n", outmost_pole.a, outmost_pole.b);
657
658 if (outmost_pole.a < 0 || outmost_pole.b < 0)
659 return AVERROR(EINVAL);
660
661 for (i = 0; i < iir->nb_ab[1]; i++) {
662 double distance;
663
664 if (isnan(iir->ab[1][2 * i]) || isnan(iir->ab[1][2 * i + 1]))
665 continue;
666 distance = hypot(iir->ab[0][2 * outmost_pole.a ] - iir->ab[1][2 * i ],
667 iir->ab[0][2 * outmost_pole.a + 1] - iir->ab[1][2 * i + 1]);
668
669 if (distance < min_distance) {
670 min_distance = distance;
671 nearest_zero.a = i;
672 }
673 }
674
675 for (i = 0; i < iir->nb_ab[1]; i++) {
676 if (isnan(iir->ab[1][2 * i]) || isnan(iir->ab[1][2 * i + 1]))
677 continue;
678
679 if (iir->ab[1][2 * i ] == iir->ab[1][2 * nearest_zero.a ] &&
680 iir->ab[1][2 * i + 1] == -iir->ab[1][2 * nearest_zero.a + 1]) {
681 nearest_zero.b = i;
682 break;
683 }
684 }
685
686 av_log(ctx, AV_LOG_VERBOSE, "nearest_zero is %d.%d\n", nearest_zero.a, nearest_zero.b);
687
688 if (nearest_zero.a < 0 || nearest_zero.b < 0)
689 return AVERROR(EINVAL);
690
691 poles[0] = iir->ab[0][2 * outmost_pole.a ];
692 poles[1] = iir->ab[0][2 * outmost_pole.a + 1];
693
694 zeros[0] = iir->ab[1][2 * nearest_zero.a ];
695 zeros[1] = iir->ab[1][2 * nearest_zero.a + 1];
696
697 if (nearest_zero.a == nearest_zero.b && outmost_pole.a == outmost_pole.b) {
698 zeros[2] = 0;
699 zeros[3] = 0;
700
701 poles[2] = 0;
702 poles[3] = 0;
703 } else {
704 poles[2] = iir->ab[0][2 * outmost_pole.b ];
705 poles[3] = iir->ab[0][2 * outmost_pole.b + 1];
706
707 zeros[2] = iir->ab[1][2 * nearest_zero.b ];
708 zeros[3] = iir->ab[1][2 * nearest_zero.b + 1];
709 }
710
711 ret = expand(ctx, zeros, 2, b);
712 if (ret < 0)
713 return ret;
714
715 ret = expand(ctx, poles, 2, a);
716 if (ret < 0)
717 return ret;
718
719 iir->ab[0][2 * outmost_pole.a] = iir->ab[0][2 * outmost_pole.a + 1] = NAN;
720 iir->ab[0][2 * outmost_pole.b] = iir->ab[0][2 * outmost_pole.b + 1] = NAN;
721 iir->ab[1][2 * nearest_zero.a] = iir->ab[1][2 * nearest_zero.a + 1] = NAN;
722 iir->ab[1][2 * nearest_zero.b] = iir->ab[1][2 * nearest_zero.b + 1] = NAN;
723
724 iir->biquads[current_biquad].a[0] = 1.;
725 iir->biquads[current_biquad].a[1] = a[2] / a[4];
726 iir->biquads[current_biquad].a[2] = a[0] / a[4];
727 iir->biquads[current_biquad].b[0] = b[4] / a[4];
728 iir->biquads[current_biquad].b[1] = b[2] / a[4];
729 iir->biquads[current_biquad].b[2] = b[0] / a[4];
730
731 if (s->normalize &&
732 fabs(iir->biquads[current_biquad].b[0] +
733 iir->biquads[current_biquad].b[1] +
734 iir->biquads[current_biquad].b[2]) > 1e-6) {
735 factor = (iir->biquads[current_biquad].a[0] +
736 iir->biquads[current_biquad].a[1] +
737 iir->biquads[current_biquad].a[2]) /
738 (iir->biquads[current_biquad].b[0] +
739 iir->biquads[current_biquad].b[1] +
740 iir->biquads[current_biquad].b[2]);
741
742 av_log(ctx, AV_LOG_VERBOSE, "factor=%f\n", factor);
743
744 iir->biquads[current_biquad].b[0] *= factor;
745 iir->biquads[current_biquad].b[1] *= factor;
746 iir->biquads[current_biquad].b[2] *= factor;
747 }
748
749 iir->biquads[current_biquad].b[0] *= (current_biquad ? 1.0 : iir->g);
750 iir->biquads[current_biquad].b[1] *= (current_biquad ? 1.0 : iir->g);
751 iir->biquads[current_biquad].b[2] *= (current_biquad ? 1.0 : iir->g);
752
753 av_log(ctx, AV_LOG_VERBOSE, "a=%f %f %f:b=%f %f %f\n",
754 iir->biquads[current_biquad].a[0],
755 iir->biquads[current_biquad].a[1],
756 iir->biquads[current_biquad].a[2],
757 iir->biquads[current_biquad].b[0],
758 iir->biquads[current_biquad].b[1],
759 iir->biquads[current_biquad].b[2]);
760
761 current_biquad++;
762 }
763 }
764
765 return 0;
766 }
767
768 static void biquad_process(double *x, double *y, int length,
769 double b0, double b1, double b2,
770 double a1, double a2)
771 {
772 double w1 = 0., w2 = 0.;
773
774 a1 = -a1;
775 a2 = -a2;
776
777 for (int n = 0; n < length; n++) {
778 double out, in = x[n];
779
780 y[n] = out = in * b0 + w1;
781 w1 = b1 * in + w2 + a1 * out;
782 w2 = b2 * in + a2 * out;
783 }
784 }
785
786 static void solve(double *matrix, double *vector, int n, double *y, double *x, double *lu)
787 {
788 double sum = 0.;
789
790 for (int i = 0; i < n; i++) {
791 for (int j = i; j < n; j++) {
792 sum = 0.;
793 for (int k = 0; k < i; k++)
794 sum += lu[i * n + k] * lu[k * n + j];
795 lu[i * n + j] = matrix[j * n + i] - sum;
796 }
797 for (int j = i + 1; j < n; j++) {
798 sum = 0.;
799 for (int k = 0; k < i; k++)
800 sum += lu[j * n + k] * lu[k * n + i];
801 lu[j * n + i] = (1. / lu[i * n + i]) * (matrix[i * n + j] - sum);
802 }
803 }
804
805 for (int i = 0; i < n; i++) {
806 sum = 0.;
807 for (int k = 0; k < i; k++)
808 sum += lu[i * n + k] * y[k];
809 y[i] = vector[i] - sum;
810 }
811
812 for (int i = n - 1; i >= 0; i--) {
813 sum = 0.;
814 for (int k = i + 1; k < n; k++)
815 sum += lu[i * n + k] * x[k];
816 x[i] = (1 / lu[i * n + i]) * (y[i] - sum);
817 }
818 }
819
820 static int convert_serial2parallel(AVFilterContext *ctx, int channels)
821 {
822 AudioIIRContext *s = ctx->priv;
823
824 for (int ch = 0; ch < channels; ch++) {
825 IIRChannel *iir = &s->iir[ch];
826 int nb_biquads = (FFMAX(iir->nb_ab[0], iir->nb_ab[1]) + 1) / 2;
827 int length = nb_biquads * 2 + 1;
828 double *impulse = av_calloc(length, sizeof(*impulse));
829 double *y = av_calloc(length, sizeof(*y));
830 double *resp = av_calloc(length, sizeof(*resp));
831 double *M = av_calloc((length - 1) * nb_biquads, 2 * 2 * sizeof(*M));
832 double *W;
833
834 if (!impulse || !y || !resp || !M) {
835 av_free(impulse);
836 av_free(y);
837 av_free(resp);
838 av_free(M);
839 return AVERROR(ENOMEM);
840 }
841 W = M + (length - 1) * 2 * nb_biquads;
842
843 impulse[0] = 1.;
844
845 for (int n = 0; n < nb_biquads; n++) {
846 BiquadContext *biquad = &iir->biquads[n];
847
848 biquad_process(n ? y : impulse, y, length,
849 biquad->b[0], biquad->b[1], biquad->b[2],
850 biquad->a[1], biquad->a[2]);
851 }
852
853 for (int n = 0; n < nb_biquads; n++) {
854 BiquadContext *biquad = &iir->biquads[n];
855
856 biquad_process(impulse, resp, length - 1,
857 1., 0., 0., biquad->a[1], biquad->a[2]);
858
859 memcpy(M + n * 2 * (length - 1), resp, sizeof(*resp) * (length - 1));
860 memcpy(M + n * 2 * (length - 1) + length, resp, sizeof(*resp) * (length - 2));
861 memset(resp, 0, length * sizeof(*resp));
862 }
863
864 solve(M, &y[1], length - 1, &impulse[1], resp, W);
865
866 iir->fir = y[0];
867
868 for (int n = 0; n < nb_biquads; n++) {
869 BiquadContext *biquad = &iir->biquads[n];
870
871 biquad->b[0] = 0.;
872 biquad->b[1] = resp[n * 2 + 0];
873 biquad->b[2] = resp[n * 2 + 1];
874 }
875
876 av_free(impulse);
877 av_free(y);
878 av_free(resp);
879 av_free(M);
880 }
881
882 return 0;
883 }
884
885 static void convert_pr2zp(AVFilterContext *ctx, int channels)
886 {
887 AudioIIRContext *s = ctx->priv;
888 int ch;
889
890 for (ch = 0; ch < channels; ch++) {
891 IIRChannel *iir = &s->iir[ch];
892 int n;
893
894 for (n = 0; n < iir->nb_ab[0]; n++) {
895 double r = iir->ab[0][2*n];
896 double angle = iir->ab[0][2*n+1];
897
898 iir->ab[0][2*n] = r * cos(angle);
899 iir->ab[0][2*n+1] = r * sin(angle);
900 }
901
902 for (n = 0; n < iir->nb_ab[1]; n++) {
903 double r = iir->ab[1][2*n];
904 double angle = iir->ab[1][2*n+1];
905
906 iir->ab[1][2*n] = r * cos(angle);
907 iir->ab[1][2*n+1] = r * sin(angle);
908 }
909 }
910 }
911
912 static void convert_sp2zp(AVFilterContext *ctx, int channels)
913 {
914 AudioIIRContext *s = ctx->priv;
915 int ch;
916
917 for (ch = 0; ch < channels; ch++) {
918 IIRChannel *iir = &s->iir[ch];
919 int n;
920
921 for (n = 0; n < iir->nb_ab[0]; n++) {
922 double sr = iir->ab[0][2*n];
923 double si = iir->ab[0][2*n+1];
924
925 iir->ab[0][2*n] = exp(sr) * cos(si);
926 iir->ab[0][2*n+1] = exp(sr) * sin(si);
927 }
928
929 for (n = 0; n < iir->nb_ab[1]; n++) {
930 double sr = iir->ab[1][2*n];
931 double si = iir->ab[1][2*n+1];
932
933 iir->ab[1][2*n] = exp(sr) * cos(si);
934 iir->ab[1][2*n+1] = exp(sr) * sin(si);
935 }
936 }
937 }
938
939 static double fact(double i)
940 {
941 if (i <= 0.)
942 return 1.;
943 return i * fact(i - 1.);
944 }
945
946 static double coef_sf2zf(double *a, int N, int n)
947 {
948 double z = 0.;
949
950 for (int i = 0; i <= N; i++) {
951 double acc = 0.;
952
953 for (int k = FFMAX(n - N + i, 0); k <= FFMIN(i, n); k++) {
954 acc += ((fact(i) * fact(N - i)) /
955 (fact(k) * fact(i - k) * fact(n - k) * fact(N - i - n + k))) *
956 ((k & 1) ? -1. : 1.);
957 }
958
959 z += a[i] * pow(2., i) * acc;
960 }
961
962 return z;
963 }
964
965 static void convert_sf2tf(AVFilterContext *ctx, int channels)
966 {
967 AudioIIRContext *s = ctx->priv;
968 int ch;
969
970 for (ch = 0; ch < channels; ch++) {
971 IIRChannel *iir = &s->iir[ch];
972 double *temp0 = av_calloc(iir->nb_ab[0], sizeof(*temp0));
973 double *temp1 = av_calloc(iir->nb_ab[1], sizeof(*temp1));
974
975 if (!temp0 || !temp1)
976 goto next;
977
978 memcpy(temp0, iir->ab[0], iir->nb_ab[0] * sizeof(*temp0));
979 memcpy(temp1, iir->ab[1], iir->nb_ab[1] * sizeof(*temp1));
980
981 for (int n = 0; n < iir->nb_ab[0]; n++)
982 iir->ab[0][n] = coef_sf2zf(temp0, iir->nb_ab[0] - 1, n);
983
984 for (int n = 0; n < iir->nb_ab[1]; n++)
985 iir->ab[1][n] = coef_sf2zf(temp1, iir->nb_ab[1] - 1, n);
986
987 next:
988 av_free(temp0);
989 av_free(temp1);
990 }
991 }
992
993 static void convert_pd2zp(AVFilterContext *ctx, int channels)
994 {
995 AudioIIRContext *s = ctx->priv;
996 int ch;
997
998 for (ch = 0; ch < channels; ch++) {
999 IIRChannel *iir = &s->iir[ch];
1000 int n;
1001
1002 for (n = 0; n < iir->nb_ab[0]; n++) {
1003 double r = iir->ab[0][2*n];
1004 double angle = M_PI*iir->ab[0][2*n+1]/180.;
1005
1006 iir->ab[0][2*n] = r * cos(angle);
1007 iir->ab[0][2*n+1] = r * sin(angle);
1008 }
1009
1010 for (n = 0; n < iir->nb_ab[1]; n++) {
1011 double r = iir->ab[1][2*n];
1012 double angle = M_PI*iir->ab[1][2*n+1]/180.;
1013
1014 iir->ab[1][2*n] = r * cos(angle);
1015 iir->ab[1][2*n+1] = r * sin(angle);
1016 }
1017 }
1018 }
1019
1020 static void check_stability(AVFilterContext *ctx, int channels)
1021 {
1022 AudioIIRContext *s = ctx->priv;
1023 int ch;
1024
1025 for (ch = 0; ch < channels; ch++) {
1026 IIRChannel *iir = &s->iir[ch];
1027
1028 for (int n = 0; n < iir->nb_ab[0]; n++) {
1029 double pr = hypot(iir->ab[0][2*n], iir->ab[0][2*n+1]);
1030
1031 if (pr >= 1.) {
1032 av_log(ctx, AV_LOG_WARNING, "pole %d at channel %d is unstable\n", n, ch);
1033 break;
1034 }
1035 }
1036 }
1037 }
1038
1039 static void drawtext(AVFrame *pic, int x, int y, const char *txt, uint32_t color)
1040 {
1041 const uint8_t *font;
1042 int font_height;
1043 int i;
1044
1045 font = avpriv_cga_font, font_height = 8;
1046
1047 for (i = 0; txt[i]; i++) {
1048 int char_y, mask;
1049
1050 uint8_t *p = pic->data[0] + y * pic->linesize[0] + (x + i * 8) * 4;
1051 for (char_y = 0; char_y < font_height; char_y++) {
1052 for (mask = 0x80; mask; mask >>= 1) {
1053 if (font[txt[i] * font_height + char_y] & mask)
1054 AV_WL32(p, color);
1055 p += 4;
1056 }
1057 p += pic->linesize[0] - 8 * 4;
1058 }
1059 }
1060 }
1061
1062 static void draw_line(AVFrame *out, int x0, int y0, int x1, int y1, uint32_t color)
1063 {
1064 int dx = FFABS(x1-x0);
1065 int dy = FFABS(y1-y0), sy = y0 < y1 ? 1 : -1;
1066 int err = (dx>dy ? dx : -dy) / 2, e2;
1067
1068 for (;;) {
1069 AV_WL32(out->data[0] + y0 * out->linesize[0] + x0 * 4, color);
1070
1071 if (x0 == x1 && y0 == y1)
1072 break;
1073
1074 e2 = err;
1075
1076 if (e2 >-dx) {
1077 err -= dy;
1078 x0--;
1079 }
1080
1081 if (e2 < dy) {
1082 err += dx;
1083 y0 += sy;
1084 }
1085 }
1086 }
1087
1088 static double distance(double x0, double x1, double y0, double y1)
1089 {
1090 return hypot(x0 - x1, y0 - y1);
1091 }
1092
1093 static void get_response(int channel, int format, double w,
1094 const double *b, const double *a,
1095 int nb_b, int nb_a, double *magnitude, double *phase)
1096 {
1097 double realz, realp;
1098 double imagz, imagp;
1099 double real, imag;
1100 double div;
1101
1102 if (format == 0) {
1103 realz = 0., realp = 0.;
1104 imagz = 0., imagp = 0.;
1105 for (int x = 0; x < nb_a; x++) {
1106 realz += cos(-x * w) * a[x];
1107 imagz += sin(-x * w) * a[x];
1108 }
1109
1110 for (int x = 0; x < nb_b; x++) {
1111 realp += cos(-x * w) * b[x];
1112 imagp += sin(-x * w) * b[x];
1113 }
1114
1115 div = realp * realp + imagp * imagp;
1116 real = (realz * realp + imagz * imagp) / div;
1117 imag = (imagz * realp - imagp * realz) / div;
1118
1119 *magnitude = hypot(real, imag);
1120 *phase = atan2(imag, real);
1121 } else {
1122 double p = 1., z = 1.;
1123 double acc = 0.;
1124
1125 for (int x = 0; x < nb_a; x++) {
1126 z *= distance(cos(w), a[2 * x], sin(w), a[2 * x + 1]);
1127 acc += atan2(sin(w) - a[2 * x + 1], cos(w) - a[2 * x]);
1128 }
1129
1130 for (int x = 0; x < nb_b; x++) {
1131 p *= distance(cos(w), b[2 * x], sin(w), b[2 * x + 1]);
1132 acc -= atan2(sin(w) - b[2 * x + 1], cos(w) - b[2 * x]);
1133 }
1134
1135 *magnitude = z / p;
1136 *phase = acc;
1137 }
1138 }
1139
1140 static void draw_response(AVFilterContext *ctx, AVFrame *out, int sample_rate)
1141 {
1142 AudioIIRContext *s = ctx->priv;
1143 double *mag, *phase, *temp, *delay, min = DBL_MAX, max = -DBL_MAX;
1144 double min_delay = DBL_MAX, max_delay = -DBL_MAX, min_phase, max_phase;
1145 int prev_ymag = -1, prev_yphase = -1, prev_ydelay = -1;
1146 char text[32];
1147 int ch, i;
1148
1149 memset(out->data[0], 0, s->h * out->linesize[0]);
1150
1151 phase = av_malloc_array(s->w, sizeof(*phase));
1152 temp = av_malloc_array(s->w, sizeof(*temp));
1153 mag = av_malloc_array(s->w, sizeof(*mag));
1154 delay = av_malloc_array(s->w, sizeof(*delay));
1155 if (!mag || !phase || !delay || !temp)
1156 goto end;
1157
1158 ch = av_clip(s->ir_channel, 0, s->channels - 1);
1159 for (i = 0; i < s->w; i++) {
1160 const double *b = s->iir[ch].ab[0];
1161 const double *a = s->iir[ch].ab[1];
1162 const int nb_b = s->iir[ch].nb_ab[0];
1163 const int nb_a = s->iir[ch].nb_ab[1];
1164 double w = i * M_PI / (s->w - 1);
1165 double m, p;
1166
1167 get_response(ch, s->format, w, b, a, nb_b, nb_a, &m, &p);
1168
1169 mag[i] = s->iir[ch].g * m;
1170 phase[i] = p;
1171 min = fmin(min, mag[i]);
1172 max = fmax(max, mag[i]);
1173 }
1174
1175 temp[0] = 0.;
1176 for (i = 0; i < s->w - 1; i++) {
1177 double d = phase[i] - phase[i + 1];
1178 temp[i + 1] = ceil(fabs(d) / (2. * M_PI)) * 2. * M_PI * ((d > M_PI) - (d < -M_PI));
1179 }
1180
1181 min_phase = phase[0];
1182 max_phase = phase[0];
1183 for (i = 1; i < s->w; i++) {
1184 temp[i] += temp[i - 1];
1185 phase[i] += temp[i];
1186 min_phase = fmin(min_phase, phase[i]);
1187 max_phase = fmax(max_phase, phase[i]);
1188 }
1189
1190 for (i = 0; i < s->w - 1; i++) {
1191 double div = s->w / (double)sample_rate;
1192
1193 delay[i + 1] = -(phase[i] - phase[i + 1]) / div;
1194 min_delay = fmin(min_delay, delay[i + 1]);
1195 max_delay = fmax(max_delay, delay[i + 1]);
1196 }
1197 delay[0] = delay[1];
1198
1199 for (i = 0; i < s->w; i++) {
1200 int ymag = mag[i] / max * (s->h - 1);
1201 int ydelay = (delay[i] - min_delay) / (max_delay - min_delay) * (s->h - 1);
1202 int yphase = (phase[i] - min_phase) / (max_phase - min_phase) * (s->h - 1);
1203
1204 ymag = s->h - 1 - av_clip(ymag, 0, s->h - 1);
1205 yphase = s->h - 1 - av_clip(yphase, 0, s->h - 1);
1206 ydelay = s->h - 1 - av_clip(ydelay, 0, s->h - 1);
1207
1208 if (prev_ymag < 0)
1209 prev_ymag = ymag;
1210 if (prev_yphase < 0)
1211 prev_yphase = yphase;
1212 if (prev_ydelay < 0)
1213 prev_ydelay = ydelay;
1214
1215 draw_line(out, i, ymag, FFMAX(i - 1, 0), prev_ymag, 0xFFFF00FF);
1216 draw_line(out, i, yphase, FFMAX(i - 1, 0), prev_yphase, 0xFF00FF00);
1217 draw_line(out, i, ydelay, FFMAX(i - 1, 0), prev_ydelay, 0xFF00FFFF);
1218
1219 prev_ymag = ymag;
1220 prev_yphase = yphase;
1221 prev_ydelay = ydelay;
1222 }
1223
1224 if (s->w > 400 && s->h > 100) {
1225 drawtext(out, 2, 2, "Max Magnitude:", 0xDDDDDDDD);
1226 snprintf(text, sizeof(text), "%.2f", max);
1227 drawtext(out, 15 * 8 + 2, 2, text, 0xDDDDDDDD);
1228
1229 drawtext(out, 2, 12, "Min Magnitude:", 0xDDDDDDDD);
1230 snprintf(text, sizeof(text), "%.2f", min);
1231 drawtext(out, 15 * 8 + 2, 12, text, 0xDDDDDDDD);
1232
1233 drawtext(out, 2, 22, "Max Phase:", 0xDDDDDDDD);
1234 snprintf(text, sizeof(text), "%.2f", max_phase);
1235 drawtext(out, 15 * 8 + 2, 22, text, 0xDDDDDDDD);
1236
1237 drawtext(out, 2, 32, "Min Phase:", 0xDDDDDDDD);
1238 snprintf(text, sizeof(text), "%.2f", min_phase);
1239 drawtext(out, 15 * 8 + 2, 32, text, 0xDDDDDDDD);
1240
1241 drawtext(out, 2, 42, "Max Delay:", 0xDDDDDDDD);
1242 snprintf(text, sizeof(text), "%.2f", max_delay);
1243 drawtext(out, 11 * 8 + 2, 42, text, 0xDDDDDDDD);
1244
1245 drawtext(out, 2, 52, "Min Delay:", 0xDDDDDDDD);
1246 snprintf(text, sizeof(text), "%.2f", min_delay);
1247 drawtext(out, 11 * 8 + 2, 52, text, 0xDDDDDDDD);
1248 }
1249
1250 end:
1251 av_free(delay);
1252 av_free(temp);
1253 av_free(phase);
1254 av_free(mag);
1255 }
1256
1257 static int config_output(AVFilterLink *outlink)
1258 {
1259 AVFilterContext *ctx = outlink->src;
1260 AudioIIRContext *s = ctx->priv;
1261 AVFilterLink *inlink = ctx->inputs[0];
1262 int ch, ret, i;
1263
1264 s->channels = inlink->ch_layout.nb_channels;
1265 s->iir = av_calloc(s->channels, sizeof(*s->iir));
1266 if (!s->iir)
1267 return AVERROR(ENOMEM);
1268
1269 ret = read_gains(ctx, s->g_str, inlink->ch_layout.nb_channels);
1270 if (ret < 0)
1271 return ret;
1272
1273 ret = read_channels(ctx, inlink->ch_layout.nb_channels, s->a_str, 0);
1274 if (ret < 0)
1275 return ret;
1276
1277 ret = read_channels(ctx, inlink->ch_layout.nb_channels, s->b_str, 1);
1278 if (ret < 0)
1279 return ret;
1280
1281 if (s->format == -1) {
1282 convert_sf2tf(ctx, inlink->ch_layout.nb_channels);
1283 s->format = 0;
1284 } else if (s->format == 2) {
1285 convert_pr2zp(ctx, inlink->ch_layout.nb_channels);
1286 } else if (s->format == 3) {
1287 convert_pd2zp(ctx, inlink->ch_layout.nb_channels);
1288 } else if (s->format == 4) {
1289 convert_sp2zp(ctx, inlink->ch_layout.nb_channels);
1290 }
1291 if (s->format > 0) {
1292 check_stability(ctx, inlink->ch_layout.nb_channels);
1293 }
1294
1295 av_frame_free(&s->video);
1296 if (s->response) {
1297 s->video = ff_get_video_buffer(ctx->outputs[1], s->w, s->h);
1298 if (!s->video)
1299 return AVERROR(ENOMEM);
1300
1301 draw_response(ctx, s->video, inlink->sample_rate);
1302 }
1303
1304 if (s->format == 0)
1305 av_log(ctx, AV_LOG_WARNING, "transfer function coefficients format is not recommended for too high number of zeros/poles.\n");
1306
1307 if (s->format > 0 && s->process == 0) {
1308 av_log(ctx, AV_LOG_WARNING, "Direct processing is not recommended for zp coefficients format.\n");
1309
1310 ret = convert_zp2tf(ctx, inlink->ch_layout.nb_channels);
1311 if (ret < 0)
1312 return ret;
1313 } else if (s->format == -2 && s->process > 0) {
1314 av_log(ctx, AV_LOG_ERROR, "Only direct processing is implemented for lattice-ladder function.\n");
1315 return AVERROR_PATCHWELCOME;
1316 } else if (s->format <= 0 && s->process == 1) {
1317 av_log(ctx, AV_LOG_ERROR, "Serial processing is not implemented for transfer function.\n");
1318 return AVERROR_PATCHWELCOME;
1319 } else if (s->format <= 0 && s->process == 2) {
1320 av_log(ctx, AV_LOG_ERROR, "Parallel processing is not implemented for transfer function.\n");
1321 return AVERROR_PATCHWELCOME;
1322 } else if (s->format > 0 && s->process == 1) {
1323 ret = decompose_zp2biquads(ctx, inlink->ch_layout.nb_channels);
1324 if (ret < 0)
1325 return ret;
1326 } else if (s->format > 0 && s->process == 2) {
1327 if (s->precision > 1)
1328 av_log(ctx, AV_LOG_WARNING, "Parallel processing is not recommended for fixed-point precisions.\n");
1329 ret = decompose_zp2biquads(ctx, inlink->ch_layout.nb_channels);
1330 if (ret < 0)
1331 return ret;
1332 ret = convert_serial2parallel(ctx, inlink->ch_layout.nb_channels);
1333 if (ret < 0)
1334 return ret;
1335 }
1336
1337 for (ch = 0; s->format == -2 && ch < inlink->ch_layout.nb_channels; ch++) {
1338 IIRChannel *iir = &s->iir[ch];
1339
1340 if (iir->nb_ab[0] != iir->nb_ab[1] + 1) {
1341 av_log(ctx, AV_LOG_ERROR, "Number of ladder coefficients must be one more than number of reflection coefficients.\n");
1342 return AVERROR(EINVAL);
1343 }
1344 }
1345
1346 for (ch = 0; s->format == 0 && ch < inlink->ch_layout.nb_channels; ch++) {
1347 IIRChannel *iir = &s->iir[ch];
1348
1349 for (i = 1; i < iir->nb_ab[0]; i++) {
1350 iir->ab[0][i] /= iir->ab[0][0];
1351 }
1352
1353 iir->ab[0][0] = 1.0;
1354 for (i = 0; i < iir->nb_ab[1]; i++) {
1355 iir->ab[1][i] *= iir->g;
1356 }
1357
1358 normalize_coeffs(ctx, ch);
1359 }
1360
1361 switch (inlink->format) {
1362 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;
1363 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;
1364 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;
1365 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;
1366 }
1367
1368 if (s->format == -2) {
1369 switch (inlink->format) {
1370 case AV_SAMPLE_FMT_DBLP: s->iir_channel = iir_ch_lattice_dblp; break;
1371 case AV_SAMPLE_FMT_FLTP: s->iir_channel = iir_ch_lattice_fltp; break;
1372 case AV_SAMPLE_FMT_S32P: s->iir_channel = iir_ch_lattice_s32p; break;
1373 case AV_SAMPLE_FMT_S16P: s->iir_channel = iir_ch_lattice_s16p; break;
1374 }
1375 }
1376
1377 return 0;
1378 }
1379
1380 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
1381 {
1382 AVFilterContext *ctx = inlink->dst;
1383 AudioIIRContext *s = ctx->priv;
1384 AVFilterLink *outlink = ctx->outputs[0];
1385 ThreadData td;
1386 AVFrame *out;
1387 int ch, ret;
1388
1389 if (av_frame_is_writable(in) && s->process != 2) {
1390 out = in;
1391 } else {
1392 out = ff_get_audio_buffer(outlink, in->nb_samples);
1393 if (!out) {
1394 av_frame_free(&in);
1395 return AVERROR(ENOMEM);
1396 }
1397 av_frame_copy_props(out, in);
1398 }
1399
1400 td.in = in;
1401 td.out = out;
1402 ff_filter_execute(ctx, s->iir_channel, &td, NULL, outlink->ch_layout.nb_channels);
1403
1404 for (ch = 0; ch < outlink->ch_layout.nb_channels; ch++) {
1405 if (s->iir[ch].clippings > 0)
1406 av_log(ctx, AV_LOG_WARNING, "Channel %d clipping %d times. Please reduce gain.\n",
1407 ch, s->iir[ch].clippings);
1408 s->iir[ch].clippings = 0;
1409 }
1410
1411 if (in != out)
1412 av_frame_free(&in);
1413
1414 if (s->response) {
1415 AVFilterLink *outlink = ctx->outputs[1];
1416 int64_t old_pts = s->video->pts;
1417 int64_t new_pts = av_rescale_q(out->pts, ctx->inputs[0]->time_base, outlink->time_base);
1418
1419 if (new_pts > old_pts) {
1420 AVFrame *clone;
1421
1422 s->video->pts = new_pts;
1423 clone = av_frame_clone(s->video);
1424 if (!clone)
1425 return AVERROR(ENOMEM);
1426 ret = ff_filter_frame(outlink, clone);
1427 if (ret < 0)
1428 return ret;
1429 }
1430 }
1431
1432 return ff_filter_frame(outlink, out);
1433 }
1434
1435 static int config_video(AVFilterLink *outlink)
1436 {
1437 AVFilterContext *ctx = outlink->src;
1438 AudioIIRContext *s = ctx->priv;
1439
1440 outlink->sample_aspect_ratio = (AVRational){1,1};
1441 outlink->w = s->w;
1442 outlink->h = s->h;
1443 outlink->frame_rate = s->rate;
1444 outlink->time_base = av_inv_q(outlink->frame_rate);
1445
1446 return 0;
1447 }
1448
1449 static av_cold int init(AVFilterContext *ctx)
1450 {
1451 AudioIIRContext *s = ctx->priv;
1452 AVFilterPad pad, vpad;
1453 int ret;
1454
1455 if (!s->a_str || !s->b_str || !s->g_str) {
1456 av_log(ctx, AV_LOG_ERROR, "Valid coefficients are mandatory.\n");
1457 return AVERROR(EINVAL);
1458 }
1459
1460 switch (s->precision) {
1461 case 0: s->sample_format = AV_SAMPLE_FMT_DBLP; break;
1462 case 1: s->sample_format = AV_SAMPLE_FMT_FLTP; break;
1463 case 2: s->sample_format = AV_SAMPLE_FMT_S32P; break;
1464 case 3: s->sample_format = AV_SAMPLE_FMT_S16P; break;
1465 default: return AVERROR_BUG;
1466 }
1467
1468 pad = (AVFilterPad){
1469 .name = "default",
1470 .type = AVMEDIA_TYPE_AUDIO,
1471 .config_props = config_output,
1472 };
1473
1474 ret = ff_append_outpad(ctx, &pad);
1475 if (ret < 0)
1476 return ret;
1477
1478 if (s->response) {
1479 vpad = (AVFilterPad){
1480 .name = "filter_response",
1481 .type = AVMEDIA_TYPE_VIDEO,
1482 .config_props = config_video,
1483 };
1484
1485 ret = ff_append_outpad(ctx, &vpad);
1486 if (ret < 0)
1487 return ret;
1488 }
1489
1490 return 0;
1491 }
1492
1493 static av_cold void uninit(AVFilterContext *ctx)
1494 {
1495 AudioIIRContext *s = ctx->priv;
1496 int ch;
1497
1498 if (s->iir) {
1499 for (ch = 0; ch < s->channels; ch++) {
1500 IIRChannel *iir = &s->iir[ch];
1501 av_freep(&iir->ab[0]);
1502 av_freep(&iir->ab[1]);
1503 av_freep(&iir->cache[0]);
1504 av_freep(&iir->cache[1]);
1505 av_freep(&iir->biquads);
1506 }
1507 }
1508 av_freep(&s->iir);
1509
1510 av_frame_free(&s->video);
1511 }
1512
1513 static const AVFilterPad inputs[] = {
1514 {
1515 .name = "default",
1516 .type = AVMEDIA_TYPE_AUDIO,
1517 .filter_frame = filter_frame,
1518 },
1519 };
1520
1521 #define OFFSET(x) offsetof(AudioIIRContext, x)
1522 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
1523 #define VF AV_OPT_FLAG_VIDEO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
1524
1525 static const AVOption aiir_options[] = {
1526 { "zeros", "set B/numerator/zeros/reflection coefficients", OFFSET(b_str), AV_OPT_TYPE_STRING, {.str="1+0i 1-0i"}, 0, 0, AF },
1527 { "z", "set B/numerator/zeros/reflection coefficients", OFFSET(b_str), AV_OPT_TYPE_STRING, {.str="1+0i 1-0i"}, 0, 0, AF },
1528 { "poles", "set A/denominator/poles/ladder coefficients", OFFSET(a_str), AV_OPT_TYPE_STRING, {.str="1+0i 1-0i"}, 0, 0, AF },
1529 { "p", "set A/denominator/poles/ladder coefficients", OFFSET(a_str), AV_OPT_TYPE_STRING, {.str="1+0i 1-0i"}, 0, 0, AF },
1530 { "gains", "set channels gains", OFFSET(g_str), AV_OPT_TYPE_STRING, {.str="1|1"}, 0, 0, AF },
1531 { "k", "set channels gains", OFFSET(g_str), AV_OPT_TYPE_STRING, {.str="1|1"}, 0, 0, AF },
1532 { "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, AF },
1533 { "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, AF },
1534 { "format", "set coefficients format", OFFSET(format), AV_OPT_TYPE_INT, {.i64=1}, -2, 4, AF, .unit = "format" },
1535 { "f", "set coefficients format", OFFSET(format), AV_OPT_TYPE_INT, {.i64=1}, -2, 4, AF, .unit = "format" },
1536 { "ll", "lattice-ladder function", 0, AV_OPT_TYPE_CONST, {.i64=-2}, 0, 0, AF, .unit = "format" },
1537 { "sf", "analog transfer function", 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, AF, .unit = "format" },
1538 { "tf", "digital transfer function", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, .unit = "format" },
1539 { "zp", "Z-plane zeros/poles", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, .unit = "format" },
1540 { "pr", "Z-plane zeros/poles (polar radians)", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, .unit = "format" },
1541 { "pd", "Z-plane zeros/poles (polar degrees)", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, .unit = "format" },
1542 { "sp", "S-plane zeros/poles", 0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, AF, .unit = "format" },
1543 { "process", "set kind of processing", OFFSET(process), AV_OPT_TYPE_INT, {.i64=1}, 0, 2, AF, .unit = "process" },
1544 { "r", "set kind of processing", OFFSET(process), AV_OPT_TYPE_INT, {.i64=1}, 0, 2, AF, .unit = "process" },
1545 { "d", "direct", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, .unit = "process" },
1546 { "s", "serial", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, .unit = "process" },
1547 { "p", "parallel", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, .unit = "process" },
1548 { "precision", "set filtering precision", OFFSET(precision),AV_OPT_TYPE_INT, {.i64=0}, 0, 3, AF, .unit = "precision" },
1549 { "e", "set precision", OFFSET(precision),AV_OPT_TYPE_INT, {.i64=0}, 0, 3, AF, .unit = "precision" },
1550 { "dbl", "double-precision floating-point", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, .unit = "precision" },
1551 { "flt", "single-precision floating-point", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, .unit = "precision" },
1552 { "i32", "32-bit integers", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, .unit = "precision" },
1553 { "i16", "16-bit integers", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, .unit = "precision" },
1554 { "normalize", "normalize coefficients", OFFSET(normalize),AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, AF },
1555 { "n", "normalize coefficients", OFFSET(normalize),AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, AF },
1556 { "mix", "set mix", OFFSET(mix), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, AF },
1557 { "response", "show IR frequency response", OFFSET(response), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, VF },
1558 { "channel", "set IR channel to display frequency response", OFFSET(ir_channel), AV_OPT_TYPE_INT, {.i64=0}, 0, 1024, VF },
1559 { "size", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str = "hd720"}, 0, 0, VF },
1560 { "rate", "set video rate", OFFSET(rate), AV_OPT_TYPE_VIDEO_RATE, {.str = "25"}, 0, INT32_MAX, VF },
1561 { NULL },
1562 };
1563
1564 AVFILTER_DEFINE_CLASS(aiir);
1565
1566 const AVFilter ff_af_aiir = {
1567 .name = "aiir",
1568 .description = NULL_IF_CONFIG_SMALL("Apply Infinite Impulse Response filter with supplied coefficients."),
1569 .priv_size = sizeof(AudioIIRContext),
1570 .priv_class = &aiir_class,
1571 .init = init,
1572 .uninit = uninit,
1573 FILTER_INPUTS(inputs),
1574 FILTER_QUERY_FUNC(query_formats),
1575 .flags = AVFILTER_FLAG_DYNAMIC_OUTPUTS |
1576 AVFILTER_FLAG_SLICE_THREADS,
1577 };
1578