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/opt.h"
26 #include "libavutil/xga_font_data.h"
27 #include "audio.h"
28 #include "avfilter.h"
29 #include "internal.h"
30
31 typedef struct ThreadData {
32 AVFrame *in, *out;
33 } ThreadData;
34
35 typedef struct Pair {
36 int a, b;
37 } Pair;
38
39 typedef struct BiquadContext {
40 double a[3];
41 double b[3];
42 double w1, w2;
43 } BiquadContext;
44
45 typedef struct IIRChannel {
46 int nb_ab[2];
47 double *ab[2];
48 double g;
49 double *cache[2];
50 double fir;
51 BiquadContext *biquads;
52 int clippings;
53 } IIRChannel;
54
55 typedef struct AudioIIRContext {
56 const AVClass *class;
57 char *a_str, *b_str, *g_str;
58 double dry_gain, wet_gain;
59 double mix;
60 int normalize;
61 int format;
62 int process;
63 int precision;
64 int response;
65 int w, h;
66 int ir_channel;
67 AVRational rate;
68
69 AVFrame *video;
70
71 IIRChannel *iir;
72 int channels;
73 enum AVSampleFormat sample_format;
74
75 int (*iir_channel)(AVFilterContext *ctx, void *arg, int ch, int nb_jobs);
76 } AudioIIRContext;
77
query_formats(AVFilterContext * ctx)78 static int query_formats(AVFilterContext *ctx)
79 {
80 AudioIIRContext *s = ctx->priv;
81 AVFilterFormats *formats;
82 enum AVSampleFormat sample_fmts[] = {
83 AV_SAMPLE_FMT_DBLP,
84 AV_SAMPLE_FMT_NONE
85 };
86 static const enum AVPixelFormat pix_fmts[] = {
87 AV_PIX_FMT_RGB0,
88 AV_PIX_FMT_NONE
89 };
90 int ret;
91
92 if (s->response) {
93 AVFilterLink *videolink = ctx->outputs[1];
94
95 formats = ff_make_format_list(pix_fmts);
96 if ((ret = ff_formats_ref(formats, &videolink->incfg.formats)) < 0)
97 return ret;
98 }
99
100 ret = ff_set_common_all_channel_counts(ctx);
101 if (ret < 0)
102 return ret;
103
104 sample_fmts[0] = s->sample_format;
105 ret = ff_set_common_formats_from_list(ctx, sample_fmts);
106 if (ret < 0)
107 return ret;
108
109 return ff_set_common_all_samplerates(ctx);
110 }
111
112 #define IIR_CH(name, type, min, max, need_clipping) \
113 static int iir_ch_## name(AVFilterContext *ctx, void *arg, int ch, int nb_jobs) \
114 { \
115 AudioIIRContext *s = ctx->priv; \
116 const double ig = s->dry_gain; \
117 const double og = s->wet_gain; \
118 const double mix = s->mix; \
119 ThreadData *td = arg; \
120 AVFrame *in = td->in, *out = td->out; \
121 const type *src = (const type *)in->extended_data[ch]; \
122 double *oc = (double *)s->iir[ch].cache[0]; \
123 double *ic = (double *)s->iir[ch].cache[1]; \
124 const int nb_a = s->iir[ch].nb_ab[0]; \
125 const int nb_b = s->iir[ch].nb_ab[1]; \
126 const double *a = s->iir[ch].ab[0]; \
127 const double *b = s->iir[ch].ab[1]; \
128 const double g = s->iir[ch].g; \
129 int *clippings = &s->iir[ch].clippings; \
130 type *dst = (type *)out->extended_data[ch]; \
131 int n; \
132 \
133 for (n = 0; n < in->nb_samples; n++) { \
134 double sample = 0.; \
135 int x; \
136 \
137 memmove(&ic[1], &ic[0], (nb_b - 1) * sizeof(*ic)); \
138 memmove(&oc[1], &oc[0], (nb_a - 1) * sizeof(*oc)); \
139 ic[0] = src[n] * ig; \
140 for (x = 0; x < nb_b; x++) \
141 sample += b[x] * ic[x]; \
142 \
143 for (x = 1; x < nb_a; x++) \
144 sample -= a[x] * oc[x]; \
145 \
146 oc[0] = sample; \
147 sample *= og * g; \
148 sample = sample * mix + ic[0] * (1. - mix); \
149 if (need_clipping && sample < min) { \
150 (*clippings)++; \
151 dst[n] = min; \
152 } else if (need_clipping && sample > max) { \
153 (*clippings)++; \
154 dst[n] = max; \
155 } else { \
156 dst[n] = sample; \
157 } \
158 } \
159 \
160 return 0; \
161 }
162
163 IIR_CH(s16p, int16_t, INT16_MIN, INT16_MAX, 1)
164 IIR_CH(s32p, int32_t, INT32_MIN, INT32_MAX, 1)
165 IIR_CH(fltp, float, -1., 1., 0)
166 IIR_CH(dblp, double, -1., 1., 0)
167
168 #define SERIAL_IIR_CH(name, type, min, max, need_clipping) \
169 static int iir_ch_serial_## name(AVFilterContext *ctx, void *arg, \
170 int ch, int nb_jobs) \
171 { \
172 AudioIIRContext *s = ctx->priv; \
173 const double ig = s->dry_gain; \
174 const double og = s->wet_gain; \
175 const double mix = s->mix; \
176 const double imix = 1. - mix; \
177 ThreadData *td = arg; \
178 AVFrame *in = td->in, *out = td->out; \
179 const type *src = (const type *)in->extended_data[ch]; \
180 type *dst = (type *)out->extended_data[ch]; \
181 IIRChannel *iir = &s->iir[ch]; \
182 const double g = iir->g; \
183 int *clippings = &iir->clippings; \
184 int nb_biquads = (FFMAX(iir->nb_ab[0], iir->nb_ab[1]) + 1) / 2; \
185 int n, i; \
186 \
187 for (i = nb_biquads - 1; i >= 0; i--) { \
188 const double a1 = -iir->biquads[i].a[1]; \
189 const double a2 = -iir->biquads[i].a[2]; \
190 const double b0 = iir->biquads[i].b[0]; \
191 const double b1 = iir->biquads[i].b[1]; \
192 const double b2 = iir->biquads[i].b[2]; \
193 double w1 = iir->biquads[i].w1; \
194 double w2 = iir->biquads[i].w2; \
195 \
196 for (n = 0; n < in->nb_samples; n++) { \
197 double i0 = ig * (i ? dst[n] : src[n]); \
198 double o0 = i0 * b0 + w1; \
199 \
200 w1 = b1 * i0 + w2 + a1 * o0; \
201 w2 = b2 * i0 + a2 * o0; \
202 o0 *= og * g; \
203 \
204 o0 = o0 * mix + imix * i0; \
205 if (need_clipping && o0 < min) { \
206 (*clippings)++; \
207 dst[n] = min; \
208 } else if (need_clipping && o0 > max) { \
209 (*clippings)++; \
210 dst[n] = max; \
211 } else { \
212 dst[n] = o0; \
213 } \
214 } \
215 iir->biquads[i].w1 = w1; \
216 iir->biquads[i].w2 = w2; \
217 } \
218 \
219 return 0; \
220 }
221
222 SERIAL_IIR_CH(s16p, int16_t, INT16_MIN, INT16_MAX, 1)
223 SERIAL_IIR_CH(s32p, int32_t, INT32_MIN, INT32_MAX, 1)
224 SERIAL_IIR_CH(fltp, float, -1., 1., 0)
225 SERIAL_IIR_CH(dblp, double, -1., 1., 0)
226
227 #define PARALLEL_IIR_CH(name, type, min, max, need_clipping) \
228 static int iir_ch_parallel_## name(AVFilterContext *ctx, void *arg, \
229 int ch, int nb_jobs) \
230 { \
231 AudioIIRContext *s = ctx->priv; \
232 const double ig = s->dry_gain; \
233 const double og = s->wet_gain; \
234 const double mix = s->mix; \
235 const double imix = 1. - mix; \
236 ThreadData *td = arg; \
237 AVFrame *in = td->in, *out = td->out; \
238 const type *src = (const type *)in->extended_data[ch]; \
239 type *dst = (type *)out->extended_data[ch]; \
240 IIRChannel *iir = &s->iir[ch]; \
241 const double g = iir->g; \
242 const double fir = iir->fir; \
243 int *clippings = &iir->clippings; \
244 int nb_biquads = (FFMAX(iir->nb_ab[0], iir->nb_ab[1]) + 1) / 2; \
245 int n, i; \
246 \
247 for (i = 0; i < nb_biquads; i++) { \
248 const double a1 = -iir->biquads[i].a[1]; \
249 const double a2 = -iir->biquads[i].a[2]; \
250 const double b1 = iir->biquads[i].b[1]; \
251 const double b2 = iir->biquads[i].b[2]; \
252 double w1 = iir->biquads[i].w1; \
253 double w2 = iir->biquads[i].w2; \
254 \
255 for (n = 0; n < in->nb_samples; n++) { \
256 double i0 = ig * src[n]; \
257 double o0 = w1; \
258 \
259 w1 = b1 * i0 + w2 + a1 * o0; \
260 w2 = b2 * i0 + a2 * o0; \
261 o0 *= og * g; \
262 o0 += dst[n]; \
263 \
264 if (need_clipping && o0 < min) { \
265 (*clippings)++; \
266 dst[n] = min; \
267 } else if (need_clipping && o0 > max) { \
268 (*clippings)++; \
269 dst[n] = max; \
270 } else { \
271 dst[n] = o0; \
272 } \
273 } \
274 iir->biquads[i].w1 = w1; \
275 iir->biquads[i].w2 = w2; \
276 } \
277 \
278 for (n = 0; n < in->nb_samples; n++) { \
279 dst[n] += fir * src[n]; \
280 dst[n] = dst[n] * mix + imix * src[n]; \
281 } \
282 \
283 return 0; \
284 }
285
286 PARALLEL_IIR_CH(s16p, int16_t, INT16_MIN, INT16_MAX, 1)
287 PARALLEL_IIR_CH(s32p, int32_t, INT32_MIN, INT32_MAX, 1)
288 PARALLEL_IIR_CH(fltp, float, -1., 1., 0)
289 PARALLEL_IIR_CH(dblp, double, -1., 1., 0)
290
291 #define LATTICE_IIR_CH(name, type, min, max, need_clipping) \
292 static int iir_ch_lattice_## name(AVFilterContext *ctx, void *arg, \
293 int ch, int nb_jobs) \
294 { \
295 AudioIIRContext *s = ctx->priv; \
296 const double ig = s->dry_gain; \
297 const double og = s->wet_gain; \
298 const double mix = s->mix; \
299 ThreadData *td = arg; \
300 AVFrame *in = td->in, *out = td->out; \
301 const type *src = (const type *)in->extended_data[ch]; \
302 double n0, n1, p0, *x = (double *)s->iir[ch].cache[0]; \
303 const int nb_stages = s->iir[ch].nb_ab[1]; \
304 const double *v = s->iir[ch].ab[0]; \
305 const double *k = s->iir[ch].ab[1]; \
306 const double g = s->iir[ch].g; \
307 int *clippings = &s->iir[ch].clippings; \
308 type *dst = (type *)out->extended_data[ch]; \
309 int n; \
310 \
311 for (n = 0; n < in->nb_samples; n++) { \
312 const double in = src[n] * ig; \
313 double out = 0.; \
314 \
315 n1 = in; \
316 for (int i = nb_stages - 1; i >= 0; i--) { \
317 n0 = n1 - k[i] * x[i]; \
318 p0 = n0 * k[i] + x[i]; \
319 out += p0 * v[i+1]; \
320 x[i] = p0; \
321 n1 = n0; \
322 } \
323 \
324 out += n1 * v[0]; \
325 memmove(&x[1], &x[0], nb_stages * sizeof(*x)); \
326 x[0] = n1; \
327 out *= og * g; \
328 out = out * mix + in * (1. - mix); \
329 if (need_clipping && out < min) { \
330 (*clippings)++; \
331 dst[n] = min; \
332 } else if (need_clipping && out > max) { \
333 (*clippings)++; \
334 dst[n] = max; \
335 } else { \
336 dst[n] = out; \
337 } \
338 } \
339 \
340 return 0; \
341 }
342
343 LATTICE_IIR_CH(s16p, int16_t, INT16_MIN, INT16_MAX, 1)
344 LATTICE_IIR_CH(s32p, int32_t, INT32_MIN, INT32_MAX, 1)
345 LATTICE_IIR_CH(fltp, float, -1., 1., 0)
346 LATTICE_IIR_CH(dblp, double, -1., 1., 0)
347
count_coefficients(char * item_str,int * nb_items)348 static void count_coefficients(char *item_str, int *nb_items)
349 {
350 char *p;
351
352 if (!item_str)
353 return;
354
355 *nb_items = 1;
356 for (p = item_str; *p && *p != '|'; p++) {
357 if (*p == ' ')
358 (*nb_items)++;
359 }
360 }
361
read_gains(AVFilterContext * ctx,char * item_str,int nb_items)362 static int read_gains(AVFilterContext *ctx, char *item_str, int nb_items)
363 {
364 AudioIIRContext *s = ctx->priv;
365 char *p, *arg, *old_str, *prev_arg = NULL, *saveptr = NULL;
366 int i;
367
368 p = old_str = av_strdup(item_str);
369 if (!p)
370 return AVERROR(ENOMEM);
371 for (i = 0; i < nb_items; i++) {
372 if (!(arg = av_strtok(p, "|", &saveptr)))
373 arg = prev_arg;
374
375 if (!arg) {
376 av_freep(&old_str);
377 return AVERROR(EINVAL);
378 }
379
380 p = NULL;
381 if (av_sscanf(arg, "%lf", &s->iir[i].g) != 1) {
382 av_log(ctx, AV_LOG_ERROR, "Invalid gains supplied: %s\n", arg);
383 av_freep(&old_str);
384 return AVERROR(EINVAL);
385 }
386
387 prev_arg = arg;
388 }
389
390 av_freep(&old_str);
391
392 return 0;
393 }
394
read_tf_coefficients(AVFilterContext * ctx,char * item_str,int nb_items,double * dst)395 static int read_tf_coefficients(AVFilterContext *ctx, char *item_str, int nb_items, double *dst)
396 {
397 char *p, *arg, *old_str, *saveptr = NULL;
398 int i;
399
400 p = old_str = av_strdup(item_str);
401 if (!p)
402 return AVERROR(ENOMEM);
403 for (i = 0; i < nb_items; i++) {
404 if (!(arg = av_strtok(p, " ", &saveptr)))
405 break;
406
407 p = NULL;
408 if (av_sscanf(arg, "%lf", &dst[i]) != 1) {
409 av_log(ctx, AV_LOG_ERROR, "Invalid coefficients supplied: %s\n", arg);
410 av_freep(&old_str);
411 return AVERROR(EINVAL);
412 }
413 }
414
415 av_freep(&old_str);
416
417 return 0;
418 }
419
read_zp_coefficients(AVFilterContext * ctx,char * item_str,int nb_items,double * dst,const char * format)420 static int read_zp_coefficients(AVFilterContext *ctx, char *item_str, int nb_items, double *dst, const char *format)
421 {
422 char *p, *arg, *old_str, *saveptr = NULL;
423 int i;
424
425 p = old_str = av_strdup(item_str);
426 if (!p)
427 return AVERROR(ENOMEM);
428 for (i = 0; i < nb_items; i++) {
429 if (!(arg = av_strtok(p, " ", &saveptr)))
430 break;
431
432 p = NULL;
433 if (av_sscanf(arg, format, &dst[i*2], &dst[i*2+1]) != 2) {
434 av_log(ctx, AV_LOG_ERROR, "Invalid coefficients supplied: %s\n", arg);
435 av_freep(&old_str);
436 return AVERROR(EINVAL);
437 }
438 }
439
440 av_freep(&old_str);
441
442 return 0;
443 }
444
445 static const char *const format[] = { "%lf", "%lf %lfi", "%lf %lfr", "%lf %lfd", "%lf %lfi" };
446
read_channels(AVFilterContext * ctx,int channels,uint8_t * item_str,int ab)447 static int read_channels(AVFilterContext *ctx, int channels, uint8_t *item_str, int ab)
448 {
449 AudioIIRContext *s = ctx->priv;
450 char *p, *arg, *old_str, *prev_arg = NULL, *saveptr = NULL;
451 int i, ret;
452
453 p = old_str = av_strdup(item_str);
454 if (!p)
455 return AVERROR(ENOMEM);
456 for (i = 0; i < channels; i++) {
457 IIRChannel *iir = &s->iir[i];
458
459 if (!(arg = av_strtok(p, "|", &saveptr)))
460 arg = prev_arg;
461
462 if (!arg) {
463 av_freep(&old_str);
464 return AVERROR(EINVAL);
465 }
466
467 count_coefficients(arg, &iir->nb_ab[ab]);
468
469 p = NULL;
470 iir->cache[ab] = av_calloc(iir->nb_ab[ab] + 1, sizeof(double));
471 iir->ab[ab] = av_calloc(iir->nb_ab[ab] * (!!s->format + 1), sizeof(double));
472 if (!iir->ab[ab] || !iir->cache[ab]) {
473 av_freep(&old_str);
474 return AVERROR(ENOMEM);
475 }
476
477 if (s->format > 0) {
478 ret = read_zp_coefficients(ctx, arg, iir->nb_ab[ab], iir->ab[ab], format[s->format]);
479 } else {
480 ret = read_tf_coefficients(ctx, arg, iir->nb_ab[ab], iir->ab[ab]);
481 }
482 if (ret < 0) {
483 av_freep(&old_str);
484 return ret;
485 }
486 prev_arg = arg;
487 }
488
489 av_freep(&old_str);
490
491 return 0;
492 }
493
cmul(double re,double im,double re2,double im2,double * RE,double * IM)494 static void cmul(double re, double im, double re2, double im2, double *RE, double *IM)
495 {
496 *RE = re * re2 - im * im2;
497 *IM = re * im2 + re2 * im;
498 }
499
expand(AVFilterContext * ctx,double * pz,int n,double * coefs)500 static int expand(AVFilterContext *ctx, double *pz, int n, double *coefs)
501 {
502 coefs[2 * n] = 1.0;
503
504 for (int i = 1; i <= n; i++) {
505 for (int j = n - i; j < n; j++) {
506 double re, im;
507
508 cmul(coefs[2 * (j + 1)], coefs[2 * (j + 1) + 1],
509 pz[2 * (i - 1)], pz[2 * (i - 1) + 1], &re, &im);
510
511 coefs[2 * j] -= re;
512 coefs[2 * j + 1] -= im;
513 }
514 }
515
516 for (int i = 0; i < n + 1; i++) {
517 if (fabs(coefs[2 * i + 1]) > FLT_EPSILON) {
518 av_log(ctx, AV_LOG_ERROR, "coefs: %f of z^%d is not real; poles/zeros are not complex conjugates.\n",
519 coefs[2 * i + 1], i);
520 return AVERROR(EINVAL);
521 }
522 }
523
524 return 0;
525 }
526
normalize_coeffs(AVFilterContext * ctx,int ch)527 static void normalize_coeffs(AVFilterContext *ctx, int ch)
528 {
529 AudioIIRContext *s = ctx->priv;
530 IIRChannel *iir = &s->iir[ch];
531 double sum_den = 0.;
532
533 if (!s->normalize)
534 return;
535
536 for (int i = 0; i < iir->nb_ab[1]; i++) {
537 sum_den += iir->ab[1][i];
538 }
539
540 if (sum_den > 1e-6) {
541 double factor, sum_num = 0.;
542
543 for (int i = 0; i < iir->nb_ab[0]; i++) {
544 sum_num += iir->ab[0][i];
545 }
546
547 factor = sum_num / sum_den;
548
549 for (int i = 0; i < iir->nb_ab[1]; i++) {
550 iir->ab[1][i] *= factor;
551 }
552 }
553 }
554
convert_zp2tf(AVFilterContext * ctx,int channels)555 static int convert_zp2tf(AVFilterContext *ctx, int channels)
556 {
557 AudioIIRContext *s = ctx->priv;
558 int ch, i, j, ret = 0;
559
560 for (ch = 0; ch < channels; ch++) {
561 IIRChannel *iir = &s->iir[ch];
562 double *topc, *botc;
563
564 topc = av_calloc((iir->nb_ab[1] + 1) * 2, sizeof(*topc));
565 botc = av_calloc((iir->nb_ab[0] + 1) * 2, sizeof(*botc));
566 if (!topc || !botc) {
567 ret = AVERROR(ENOMEM);
568 goto fail;
569 }
570
571 ret = expand(ctx, iir->ab[0], iir->nb_ab[0], botc);
572 if (ret < 0) {
573 goto fail;
574 }
575
576 ret = expand(ctx, iir->ab[1], iir->nb_ab[1], topc);
577 if (ret < 0) {
578 goto fail;
579 }
580
581 for (j = 0, i = iir->nb_ab[1]; i >= 0; j++, i--) {
582 iir->ab[1][j] = topc[2 * i];
583 }
584 iir->nb_ab[1]++;
585
586 for (j = 0, i = iir->nb_ab[0]; i >= 0; j++, i--) {
587 iir->ab[0][j] = botc[2 * i];
588 }
589 iir->nb_ab[0]++;
590
591 normalize_coeffs(ctx, ch);
592
593 fail:
594 av_free(topc);
595 av_free(botc);
596 if (ret < 0)
597 break;
598 }
599
600 return ret;
601 }
602
decompose_zp2biquads(AVFilterContext * ctx,int channels)603 static int decompose_zp2biquads(AVFilterContext *ctx, int channels)
604 {
605 AudioIIRContext *s = ctx->priv;
606 int ch, ret;
607
608 for (ch = 0; ch < channels; ch++) {
609 IIRChannel *iir = &s->iir[ch];
610 int nb_biquads = (FFMAX(iir->nb_ab[0], iir->nb_ab[1]) + 1) / 2;
611 int current_biquad = 0;
612
613 iir->biquads = av_calloc(nb_biquads, sizeof(BiquadContext));
614 if (!iir->biquads)
615 return AVERROR(ENOMEM);
616
617 while (nb_biquads--) {
618 Pair outmost_pole = { -1, -1 };
619 Pair nearest_zero = { -1, -1 };
620 double zeros[4] = { 0 };
621 double poles[4] = { 0 };
622 double b[6] = { 0 };
623 double a[6] = { 0 };
624 double min_distance = DBL_MAX;
625 double max_mag = 0;
626 double factor;
627 int i;
628
629 for (i = 0; i < iir->nb_ab[0]; i++) {
630 double mag;
631
632 if (isnan(iir->ab[0][2 * i]) || isnan(iir->ab[0][2 * i + 1]))
633 continue;
634 mag = hypot(iir->ab[0][2 * i], iir->ab[0][2 * i + 1]);
635
636 if (mag > max_mag) {
637 max_mag = mag;
638 outmost_pole.a = i;
639 }
640 }
641
642 for (i = 0; i < iir->nb_ab[0]; i++) {
643 if (isnan(iir->ab[0][2 * i]) || isnan(iir->ab[0][2 * i + 1]))
644 continue;
645
646 if (iir->ab[0][2 * i ] == iir->ab[0][2 * outmost_pole.a ] &&
647 iir->ab[0][2 * i + 1] == -iir->ab[0][2 * outmost_pole.a + 1]) {
648 outmost_pole.b = i;
649 break;
650 }
651 }
652
653 av_log(ctx, AV_LOG_VERBOSE, "outmost_pole is %d.%d\n", outmost_pole.a, outmost_pole.b);
654
655 if (outmost_pole.a < 0 || outmost_pole.b < 0)
656 return AVERROR(EINVAL);
657
658 for (i = 0; i < iir->nb_ab[1]; i++) {
659 double distance;
660
661 if (isnan(iir->ab[1][2 * i]) || isnan(iir->ab[1][2 * i + 1]))
662 continue;
663 distance = hypot(iir->ab[0][2 * outmost_pole.a ] - iir->ab[1][2 * i ],
664 iir->ab[0][2 * outmost_pole.a + 1] - iir->ab[1][2 * i + 1]);
665
666 if (distance < min_distance) {
667 min_distance = distance;
668 nearest_zero.a = i;
669 }
670 }
671
672 for (i = 0; i < iir->nb_ab[1]; i++) {
673 if (isnan(iir->ab[1][2 * i]) || isnan(iir->ab[1][2 * i + 1]))
674 continue;
675
676 if (iir->ab[1][2 * i ] == iir->ab[1][2 * nearest_zero.a ] &&
677 iir->ab[1][2 * i + 1] == -iir->ab[1][2 * nearest_zero.a + 1]) {
678 nearest_zero.b = i;
679 break;
680 }
681 }
682
683 av_log(ctx, AV_LOG_VERBOSE, "nearest_zero is %d.%d\n", nearest_zero.a, nearest_zero.b);
684
685 if (nearest_zero.a < 0 || nearest_zero.b < 0)
686 return AVERROR(EINVAL);
687
688 poles[0] = iir->ab[0][2 * outmost_pole.a ];
689 poles[1] = iir->ab[0][2 * outmost_pole.a + 1];
690
691 zeros[0] = iir->ab[1][2 * nearest_zero.a ];
692 zeros[1] = iir->ab[1][2 * nearest_zero.a + 1];
693
694 if (nearest_zero.a == nearest_zero.b && outmost_pole.a == outmost_pole.b) {
695 zeros[2] = 0;
696 zeros[3] = 0;
697
698 poles[2] = 0;
699 poles[3] = 0;
700 } else {
701 poles[2] = iir->ab[0][2 * outmost_pole.b ];
702 poles[3] = iir->ab[0][2 * outmost_pole.b + 1];
703
704 zeros[2] = iir->ab[1][2 * nearest_zero.b ];
705 zeros[3] = iir->ab[1][2 * nearest_zero.b + 1];
706 }
707
708 ret = expand(ctx, zeros, 2, b);
709 if (ret < 0)
710 return ret;
711
712 ret = expand(ctx, poles, 2, a);
713 if (ret < 0)
714 return ret;
715
716 iir->ab[0][2 * outmost_pole.a] = iir->ab[0][2 * outmost_pole.a + 1] = NAN;
717 iir->ab[0][2 * outmost_pole.b] = iir->ab[0][2 * outmost_pole.b + 1] = NAN;
718 iir->ab[1][2 * nearest_zero.a] = iir->ab[1][2 * nearest_zero.a + 1] = NAN;
719 iir->ab[1][2 * nearest_zero.b] = iir->ab[1][2 * nearest_zero.b + 1] = NAN;
720
721 iir->biquads[current_biquad].a[0] = 1.;
722 iir->biquads[current_biquad].a[1] = a[2] / a[4];
723 iir->biquads[current_biquad].a[2] = a[0] / a[4];
724 iir->biquads[current_biquad].b[0] = b[4] / a[4];
725 iir->biquads[current_biquad].b[1] = b[2] / a[4];
726 iir->biquads[current_biquad].b[2] = b[0] / a[4];
727
728 if (s->normalize &&
729 fabs(iir->biquads[current_biquad].b[0] +
730 iir->biquads[current_biquad].b[1] +
731 iir->biquads[current_biquad].b[2]) > 1e-6) {
732 factor = (iir->biquads[current_biquad].a[0] +
733 iir->biquads[current_biquad].a[1] +
734 iir->biquads[current_biquad].a[2]) /
735 (iir->biquads[current_biquad].b[0] +
736 iir->biquads[current_biquad].b[1] +
737 iir->biquads[current_biquad].b[2]);
738
739 av_log(ctx, AV_LOG_VERBOSE, "factor=%f\n", factor);
740
741 iir->biquads[current_biquad].b[0] *= factor;
742 iir->biquads[current_biquad].b[1] *= factor;
743 iir->biquads[current_biquad].b[2] *= factor;
744 }
745
746 iir->biquads[current_biquad].b[0] *= (current_biquad ? 1.0 : iir->g);
747 iir->biquads[current_biquad].b[1] *= (current_biquad ? 1.0 : iir->g);
748 iir->biquads[current_biquad].b[2] *= (current_biquad ? 1.0 : iir->g);
749
750 av_log(ctx, AV_LOG_VERBOSE, "a=%f %f %f:b=%f %f %f\n",
751 iir->biquads[current_biquad].a[0],
752 iir->biquads[current_biquad].a[1],
753 iir->biquads[current_biquad].a[2],
754 iir->biquads[current_biquad].b[0],
755 iir->biquads[current_biquad].b[1],
756 iir->biquads[current_biquad].b[2]);
757
758 current_biquad++;
759 }
760 }
761
762 return 0;
763 }
764
biquad_process(double * x,double * y,int length,double b0,double b1,double b2,double a1,double a2)765 static void biquad_process(double *x, double *y, int length,
766 double b0, double b1, double b2,
767 double a1, double a2)
768 {
769 double w1 = 0., w2 = 0.;
770
771 a1 = -a1;
772 a2 = -a2;
773
774 for (int n = 0; n < length; n++) {
775 double out, in = x[n];
776
777 y[n] = out = in * b0 + w1;
778 w1 = b1 * in + w2 + a1 * out;
779 w2 = b2 * in + a2 * out;
780 }
781 }
782
solve(double * matrix,double * vector,int n,double * y,double * x,double * lu)783 static void solve(double *matrix, double *vector, int n, double *y, double *x, double *lu)
784 {
785 double sum = 0.;
786
787 for (int i = 0; i < n; i++) {
788 for (int j = i; j < n; j++) {
789 sum = 0.;
790 for (int k = 0; k < i; k++)
791 sum += lu[i * n + k] * lu[k * n + j];
792 lu[i * n + j] = matrix[j * n + i] - sum;
793 }
794 for (int j = i + 1; j < n; j++) {
795 sum = 0.;
796 for (int k = 0; k < i; k++)
797 sum += lu[j * n + k] * lu[k * n + i];
798 lu[j * n + i] = (1. / lu[i * n + i]) * (matrix[i * n + j] - sum);
799 }
800 }
801
802 for (int i = 0; i < n; i++) {
803 sum = 0.;
804 for (int k = 0; k < i; k++)
805 sum += lu[i * n + k] * y[k];
806 y[i] = vector[i] - sum;
807 }
808
809 for (int i = n - 1; i >= 0; i--) {
810 sum = 0.;
811 for (int k = i + 1; k < n; k++)
812 sum += lu[i * n + k] * x[k];
813 x[i] = (1 / lu[i * n + i]) * (y[i] - sum);
814 }
815 }
816
convert_serial2parallel(AVFilterContext * ctx,int channels)817 static int convert_serial2parallel(AVFilterContext *ctx, int channels)
818 {
819 AudioIIRContext *s = ctx->priv;
820 int ret = 0;
821
822 for (int ch = 0; ch < channels; ch++) {
823 IIRChannel *iir = &s->iir[ch];
824 int nb_biquads = (FFMAX(iir->nb_ab[0], iir->nb_ab[1]) + 1) / 2;
825 int length = nb_biquads * 2 + 1;
826 double *impulse = av_calloc(length, sizeof(*impulse));
827 double *y = av_calloc(length, sizeof(*y));
828 double *resp = av_calloc(length, sizeof(*resp));
829 double *M = av_calloc((length - 1) * 2 * nb_biquads, sizeof(*M));
830 double *W = av_calloc((length - 1) * 2 * nb_biquads, sizeof(*W));
831
832 if (!impulse || !y || !resp || !M) {
833 av_free(impulse);
834 av_free(y);
835 av_free(resp);
836 av_free(M);
837 av_free(W);
838 return AVERROR(ENOMEM);
839 }
840
841 impulse[0] = 1.;
842
843 for (int n = 0; n < nb_biquads; n++) {
844 BiquadContext *biquad = &iir->biquads[n];
845
846 biquad_process(n ? y : impulse, y, length,
847 biquad->b[0], biquad->b[1], biquad->b[2],
848 biquad->a[1], biquad->a[2]);
849 }
850
851 for (int n = 0; n < nb_biquads; n++) {
852 BiquadContext *biquad = &iir->biquads[n];
853
854 biquad_process(impulse, resp, length - 1,
855 1., 0., 0., biquad->a[1], biquad->a[2]);
856
857 memcpy(M + n * 2 * (length - 1), resp, sizeof(*resp) * (length - 1));
858 memcpy(M + n * 2 * (length - 1) + length, resp, sizeof(*resp) * (length - 2));
859 memset(resp, 0, length * sizeof(*resp));
860 }
861
862 solve(M, &y[1], length - 1, &impulse[1], resp, W);
863
864 iir->fir = y[0];
865
866 for (int n = 0; n < nb_biquads; n++) {
867 BiquadContext *biquad = &iir->biquads[n];
868
869 biquad->b[0] = 0.;
870 biquad->b[1] = resp[n * 2 + 0];
871 biquad->b[2] = resp[n * 2 + 1];
872 }
873
874 av_free(impulse);
875 av_free(y);
876 av_free(resp);
877 av_free(M);
878 av_free(W);
879
880 if (ret < 0)
881 return ret;
882 }
883
884 return 0;
885 }
886
convert_pr2zp(AVFilterContext * ctx,int channels)887 static void convert_pr2zp(AVFilterContext *ctx, int channels)
888 {
889 AudioIIRContext *s = ctx->priv;
890 int ch;
891
892 for (ch = 0; ch < channels; ch++) {
893 IIRChannel *iir = &s->iir[ch];
894 int n;
895
896 for (n = 0; n < iir->nb_ab[0]; n++) {
897 double r = iir->ab[0][2*n];
898 double angle = iir->ab[0][2*n+1];
899
900 iir->ab[0][2*n] = r * cos(angle);
901 iir->ab[0][2*n+1] = r * sin(angle);
902 }
903
904 for (n = 0; n < iir->nb_ab[1]; n++) {
905 double r = iir->ab[1][2*n];
906 double angle = iir->ab[1][2*n+1];
907
908 iir->ab[1][2*n] = r * cos(angle);
909 iir->ab[1][2*n+1] = r * sin(angle);
910 }
911 }
912 }
913
convert_sp2zp(AVFilterContext * ctx,int channels)914 static void convert_sp2zp(AVFilterContext *ctx, int channels)
915 {
916 AudioIIRContext *s = ctx->priv;
917 int ch;
918
919 for (ch = 0; ch < channels; ch++) {
920 IIRChannel *iir = &s->iir[ch];
921 int n;
922
923 for (n = 0; n < iir->nb_ab[0]; n++) {
924 double sr = iir->ab[0][2*n];
925 double si = iir->ab[0][2*n+1];
926
927 iir->ab[0][2*n] = exp(sr) * cos(si);
928 iir->ab[0][2*n+1] = exp(sr) * sin(si);
929 }
930
931 for (n = 0; n < iir->nb_ab[1]; n++) {
932 double sr = iir->ab[1][2*n];
933 double si = iir->ab[1][2*n+1];
934
935 iir->ab[1][2*n] = exp(sr) * cos(si);
936 iir->ab[1][2*n+1] = exp(sr) * sin(si);
937 }
938 }
939 }
940
fact(double i)941 static double fact(double i)
942 {
943 if (i <= 0.)
944 return 1.;
945 return i * fact(i - 1.);
946 }
947
coef_sf2zf(double * a,int N,int n)948 static double coef_sf2zf(double *a, int N, int n)
949 {
950 double z = 0.;
951
952 for (int i = 0; i <= N; i++) {
953 double acc = 0.;
954
955 for (int k = FFMAX(n - N + i, 0); k <= FFMIN(i, n); k++) {
956 acc += ((fact(i) * fact(N - i)) /
957 (fact(k) * fact(i - k) * fact(n - k) * fact(N - i - n + k))) *
958 ((k & 1) ? -1. : 1.);
959 }
960
961 z += a[i] * pow(2., i) * acc;
962 }
963
964 return z;
965 }
966
convert_sf2tf(AVFilterContext * ctx,int channels)967 static void convert_sf2tf(AVFilterContext *ctx, int channels)
968 {
969 AudioIIRContext *s = ctx->priv;
970 int ch;
971
972 for (ch = 0; ch < channels; ch++) {
973 IIRChannel *iir = &s->iir[ch];
974 double *temp0 = av_calloc(iir->nb_ab[0], sizeof(*temp0));
975 double *temp1 = av_calloc(iir->nb_ab[1], sizeof(*temp1));
976
977 if (!temp0 || !temp1)
978 goto next;
979
980 memcpy(temp0, iir->ab[0], iir->nb_ab[0] * sizeof(*temp0));
981 memcpy(temp1, iir->ab[1], iir->nb_ab[1] * sizeof(*temp1));
982
983 for (int n = 0; n < iir->nb_ab[0]; n++)
984 iir->ab[0][n] = coef_sf2zf(temp0, iir->nb_ab[0] - 1, n);
985
986 for (int n = 0; n < iir->nb_ab[1]; n++)
987 iir->ab[1][n] = coef_sf2zf(temp1, iir->nb_ab[1] - 1, n);
988
989 next:
990 av_free(temp0);
991 av_free(temp1);
992 }
993 }
994
convert_pd2zp(AVFilterContext * ctx,int channels)995 static void convert_pd2zp(AVFilterContext *ctx, int channels)
996 {
997 AudioIIRContext *s = ctx->priv;
998 int ch;
999
1000 for (ch = 0; ch < channels; ch++) {
1001 IIRChannel *iir = &s->iir[ch];
1002 int n;
1003
1004 for (n = 0; n < iir->nb_ab[0]; n++) {
1005 double r = iir->ab[0][2*n];
1006 double angle = M_PI*iir->ab[0][2*n+1]/180.;
1007
1008 iir->ab[0][2*n] = r * cos(angle);
1009 iir->ab[0][2*n+1] = r * sin(angle);
1010 }
1011
1012 for (n = 0; n < iir->nb_ab[1]; n++) {
1013 double r = iir->ab[1][2*n];
1014 double angle = M_PI*iir->ab[1][2*n+1]/180.;
1015
1016 iir->ab[1][2*n] = r * cos(angle);
1017 iir->ab[1][2*n+1] = r * sin(angle);
1018 }
1019 }
1020 }
1021
check_stability(AVFilterContext * ctx,int channels)1022 static void check_stability(AVFilterContext *ctx, int channels)
1023 {
1024 AudioIIRContext *s = ctx->priv;
1025 int ch;
1026
1027 for (ch = 0; ch < channels; ch++) {
1028 IIRChannel *iir = &s->iir[ch];
1029
1030 for (int n = 0; n < iir->nb_ab[0]; n++) {
1031 double pr = hypot(iir->ab[0][2*n], iir->ab[0][2*n+1]);
1032
1033 if (pr >= 1.) {
1034 av_log(ctx, AV_LOG_WARNING, "pole %d at channel %d is unstable\n", n, ch);
1035 break;
1036 }
1037 }
1038 }
1039 }
1040
drawtext(AVFrame * pic,int x,int y,const char * txt,uint32_t color)1041 static void drawtext(AVFrame *pic, int x, int y, const char *txt, uint32_t color)
1042 {
1043 const uint8_t *font;
1044 int font_height;
1045 int i;
1046
1047 font = avpriv_cga_font, font_height = 8;
1048
1049 for (i = 0; txt[i]; i++) {
1050 int char_y, mask;
1051
1052 uint8_t *p = pic->data[0] + y * pic->linesize[0] + (x + i * 8) * 4;
1053 for (char_y = 0; char_y < font_height; char_y++) {
1054 for (mask = 0x80; mask; mask >>= 1) {
1055 if (font[txt[i] * font_height + char_y] & mask)
1056 AV_WL32(p, color);
1057 p += 4;
1058 }
1059 p += pic->linesize[0] - 8 * 4;
1060 }
1061 }
1062 }
1063
draw_line(AVFrame * out,int x0,int y0,int x1,int y1,uint32_t color)1064 static void draw_line(AVFrame *out, int x0, int y0, int x1, int y1, uint32_t color)
1065 {
1066 int dx = FFABS(x1-x0);
1067 int dy = FFABS(y1-y0), sy = y0 < y1 ? 1 : -1;
1068 int err = (dx>dy ? dx : -dy) / 2, e2;
1069
1070 for (;;) {
1071 AV_WL32(out->data[0] + y0 * out->linesize[0] + x0 * 4, color);
1072
1073 if (x0 == x1 && y0 == y1)
1074 break;
1075
1076 e2 = err;
1077
1078 if (e2 >-dx) {
1079 err -= dy;
1080 x0--;
1081 }
1082
1083 if (e2 < dy) {
1084 err += dx;
1085 y0 += sy;
1086 }
1087 }
1088 }
1089
distance(double x0,double x1,double y0,double y1)1090 static double distance(double x0, double x1, double y0, double y1)
1091 {
1092 return hypot(x0 - x1, y0 - y1);
1093 }
1094
get_response(int channel,int format,double w,const double * b,const double * a,int nb_b,int nb_a,double * magnitude,double * phase)1095 static void get_response(int channel, int format, double w,
1096 const double *b, const double *a,
1097 int nb_b, int nb_a, double *magnitude, double *phase)
1098 {
1099 double realz, realp;
1100 double imagz, imagp;
1101 double real, imag;
1102 double div;
1103
1104 if (format == 0) {
1105 realz = 0., realp = 0.;
1106 imagz = 0., imagp = 0.;
1107 for (int x = 0; x < nb_a; x++) {
1108 realz += cos(-x * w) * a[x];
1109 imagz += sin(-x * w) * a[x];
1110 }
1111
1112 for (int x = 0; x < nb_b; x++) {
1113 realp += cos(-x * w) * b[x];
1114 imagp += sin(-x * w) * b[x];
1115 }
1116
1117 div = realp * realp + imagp * imagp;
1118 real = (realz * realp + imagz * imagp) / div;
1119 imag = (imagz * realp - imagp * realz) / div;
1120
1121 *magnitude = hypot(real, imag);
1122 *phase = atan2(imag, real);
1123 } else {
1124 double p = 1., z = 1.;
1125 double acc = 0.;
1126
1127 for (int x = 0; x < nb_a; x++) {
1128 z *= distance(cos(w), a[2 * x], sin(w), a[2 * x + 1]);
1129 acc += atan2(sin(w) - a[2 * x + 1], cos(w) - a[2 * x]);
1130 }
1131
1132 for (int x = 0; x < nb_b; x++) {
1133 p *= distance(cos(w), b[2 * x], sin(w), b[2 * x + 1]);
1134 acc -= atan2(sin(w) - b[2 * x + 1], cos(w) - b[2 * x]);
1135 }
1136
1137 *magnitude = z / p;
1138 *phase = acc;
1139 }
1140 }
1141
draw_response(AVFilterContext * ctx,AVFrame * out,int sample_rate)1142 static void draw_response(AVFilterContext *ctx, AVFrame *out, int sample_rate)
1143 {
1144 AudioIIRContext *s = ctx->priv;
1145 double *mag, *phase, *temp, *delay, min = DBL_MAX, max = -DBL_MAX;
1146 double min_delay = DBL_MAX, max_delay = -DBL_MAX, min_phase, max_phase;
1147 int prev_ymag = -1, prev_yphase = -1, prev_ydelay = -1;
1148 char text[32];
1149 int ch, i;
1150
1151 memset(out->data[0], 0, s->h * out->linesize[0]);
1152
1153 phase = av_malloc_array(s->w, sizeof(*phase));
1154 temp = av_malloc_array(s->w, sizeof(*temp));
1155 mag = av_malloc_array(s->w, sizeof(*mag));
1156 delay = av_malloc_array(s->w, sizeof(*delay));
1157 if (!mag || !phase || !delay || !temp)
1158 goto end;
1159
1160 ch = av_clip(s->ir_channel, 0, s->channels - 1);
1161 for (i = 0; i < s->w; i++) {
1162 const double *b = s->iir[ch].ab[0];
1163 const double *a = s->iir[ch].ab[1];
1164 const int nb_b = s->iir[ch].nb_ab[0];
1165 const int nb_a = s->iir[ch].nb_ab[1];
1166 double w = i * M_PI / (s->w - 1);
1167 double m, p;
1168
1169 get_response(ch, s->format, w, b, a, nb_b, nb_a, &m, &p);
1170
1171 mag[i] = s->iir[ch].g * m;
1172 phase[i] = p;
1173 min = fmin(min, mag[i]);
1174 max = fmax(max, mag[i]);
1175 }
1176
1177 temp[0] = 0.;
1178 for (i = 0; i < s->w - 1; i++) {
1179 double d = phase[i] - phase[i + 1];
1180 temp[i + 1] = ceil(fabs(d) / (2. * M_PI)) * 2. * M_PI * ((d > M_PI) - (d < -M_PI));
1181 }
1182
1183 min_phase = phase[0];
1184 max_phase = phase[0];
1185 for (i = 1; i < s->w; i++) {
1186 temp[i] += temp[i - 1];
1187 phase[i] += temp[i];
1188 min_phase = fmin(min_phase, phase[i]);
1189 max_phase = fmax(max_phase, phase[i]);
1190 }
1191
1192 for (i = 0; i < s->w - 1; i++) {
1193 double div = s->w / (double)sample_rate;
1194
1195 delay[i + 1] = -(phase[i] - phase[i + 1]) / div;
1196 min_delay = fmin(min_delay, delay[i + 1]);
1197 max_delay = fmax(max_delay, delay[i + 1]);
1198 }
1199 delay[0] = delay[1];
1200
1201 for (i = 0; i < s->w; i++) {
1202 int ymag = mag[i] / max * (s->h - 1);
1203 int ydelay = (delay[i] - min_delay) / (max_delay - min_delay) * (s->h - 1);
1204 int yphase = (phase[i] - min_phase) / (max_phase - min_phase) * (s->h - 1);
1205
1206 ymag = s->h - 1 - av_clip(ymag, 0, s->h - 1);
1207 yphase = s->h - 1 - av_clip(yphase, 0, s->h - 1);
1208 ydelay = s->h - 1 - av_clip(ydelay, 0, s->h - 1);
1209
1210 if (prev_ymag < 0)
1211 prev_ymag = ymag;
1212 if (prev_yphase < 0)
1213 prev_yphase = yphase;
1214 if (prev_ydelay < 0)
1215 prev_ydelay = ydelay;
1216
1217 draw_line(out, i, ymag, FFMAX(i - 1, 0), prev_ymag, 0xFFFF00FF);
1218 draw_line(out, i, yphase, FFMAX(i - 1, 0), prev_yphase, 0xFF00FF00);
1219 draw_line(out, i, ydelay, FFMAX(i - 1, 0), prev_ydelay, 0xFF00FFFF);
1220
1221 prev_ymag = ymag;
1222 prev_yphase = yphase;
1223 prev_ydelay = ydelay;
1224 }
1225
1226 if (s->w > 400 && s->h > 100) {
1227 drawtext(out, 2, 2, "Max Magnitude:", 0xDDDDDDDD);
1228 snprintf(text, sizeof(text), "%.2f", max);
1229 drawtext(out, 15 * 8 + 2, 2, text, 0xDDDDDDDD);
1230
1231 drawtext(out, 2, 12, "Min Magnitude:", 0xDDDDDDDD);
1232 snprintf(text, sizeof(text), "%.2f", min);
1233 drawtext(out, 15 * 8 + 2, 12, text, 0xDDDDDDDD);
1234
1235 drawtext(out, 2, 22, "Max Phase:", 0xDDDDDDDD);
1236 snprintf(text, sizeof(text), "%.2f", max_phase);
1237 drawtext(out, 15 * 8 + 2, 22, text, 0xDDDDDDDD);
1238
1239 drawtext(out, 2, 32, "Min Phase:", 0xDDDDDDDD);
1240 snprintf(text, sizeof(text), "%.2f", min_phase);
1241 drawtext(out, 15 * 8 + 2, 32, text, 0xDDDDDDDD);
1242
1243 drawtext(out, 2, 42, "Max Delay:", 0xDDDDDDDD);
1244 snprintf(text, sizeof(text), "%.2f", max_delay);
1245 drawtext(out, 11 * 8 + 2, 42, text, 0xDDDDDDDD);
1246
1247 drawtext(out, 2, 52, "Min Delay:", 0xDDDDDDDD);
1248 snprintf(text, sizeof(text), "%.2f", min_delay);
1249 drawtext(out, 11 * 8 + 2, 52, text, 0xDDDDDDDD);
1250 }
1251
1252 end:
1253 av_free(delay);
1254 av_free(temp);
1255 av_free(phase);
1256 av_free(mag);
1257 }
1258
config_output(AVFilterLink * outlink)1259 static int config_output(AVFilterLink *outlink)
1260 {
1261 AVFilterContext *ctx = outlink->src;
1262 AudioIIRContext *s = ctx->priv;
1263 AVFilterLink *inlink = ctx->inputs[0];
1264 int ch, ret, i;
1265
1266 s->channels = inlink->ch_layout.nb_channels;
1267 s->iir = av_calloc(s->channels, sizeof(*s->iir));
1268 if (!s->iir)
1269 return AVERROR(ENOMEM);
1270
1271 ret = read_gains(ctx, s->g_str, inlink->ch_layout.nb_channels);
1272 if (ret < 0)
1273 return ret;
1274
1275 ret = read_channels(ctx, inlink->ch_layout.nb_channels, s->a_str, 0);
1276 if (ret < 0)
1277 return ret;
1278
1279 ret = read_channels(ctx, inlink->ch_layout.nb_channels, s->b_str, 1);
1280 if (ret < 0)
1281 return ret;
1282
1283 if (s->format == -1) {
1284 convert_sf2tf(ctx, inlink->ch_layout.nb_channels);
1285 s->format = 0;
1286 } else if (s->format == 2) {
1287 convert_pr2zp(ctx, inlink->ch_layout.nb_channels);
1288 } else if (s->format == 3) {
1289 convert_pd2zp(ctx, inlink->ch_layout.nb_channels);
1290 } else if (s->format == 4) {
1291 convert_sp2zp(ctx, inlink->ch_layout.nb_channels);
1292 }
1293 if (s->format > 0) {
1294 check_stability(ctx, inlink->ch_layout.nb_channels);
1295 }
1296
1297 av_frame_free(&s->video);
1298 if (s->response) {
1299 s->video = ff_get_video_buffer(ctx->outputs[1], s->w, s->h);
1300 if (!s->video)
1301 return AVERROR(ENOMEM);
1302
1303 draw_response(ctx, s->video, inlink->sample_rate);
1304 }
1305
1306 if (s->format == 0)
1307 av_log(ctx, AV_LOG_WARNING, "transfer function coefficients format is not recommended for too high number of zeros/poles.\n");
1308
1309 if (s->format > 0 && s->process == 0) {
1310 av_log(ctx, AV_LOG_WARNING, "Direct processsing is not recommended for zp coefficients format.\n");
1311
1312 ret = convert_zp2tf(ctx, inlink->ch_layout.nb_channels);
1313 if (ret < 0)
1314 return ret;
1315 } else if (s->format == -2 && s->process > 0) {
1316 av_log(ctx, AV_LOG_ERROR, "Only direct processing is implemented for lattice-ladder function.\n");
1317 return AVERROR_PATCHWELCOME;
1318 } else if (s->format <= 0 && s->process == 1) {
1319 av_log(ctx, AV_LOG_ERROR, "Serial processing is not implemented for transfer function.\n");
1320 return AVERROR_PATCHWELCOME;
1321 } else if (s->format <= 0 && s->process == 2) {
1322 av_log(ctx, AV_LOG_ERROR, "Parallel processing is not implemented for transfer function.\n");
1323 return AVERROR_PATCHWELCOME;
1324 } else if (s->format > 0 && s->process == 1) {
1325 ret = decompose_zp2biquads(ctx, inlink->ch_layout.nb_channels);
1326 if (ret < 0)
1327 return ret;
1328 } else if (s->format > 0 && s->process == 2) {
1329 if (s->precision > 1)
1330 av_log(ctx, AV_LOG_WARNING, "Parallel processing is not recommended for fixed-point precisions.\n");
1331 ret = decompose_zp2biquads(ctx, inlink->ch_layout.nb_channels);
1332 if (ret < 0)
1333 return ret;
1334 ret = convert_serial2parallel(ctx, inlink->ch_layout.nb_channels);
1335 if (ret < 0)
1336 return ret;
1337 }
1338
1339 for (ch = 0; s->format == -2 && ch < inlink->ch_layout.nb_channels; ch++) {
1340 IIRChannel *iir = &s->iir[ch];
1341
1342 if (iir->nb_ab[0] != iir->nb_ab[1] + 1) {
1343 av_log(ctx, AV_LOG_ERROR, "Number of ladder coefficients must be one more than number of reflection coefficients.\n");
1344 return AVERROR(EINVAL);
1345 }
1346 }
1347
1348 for (ch = 0; s->format == 0 && ch < inlink->ch_layout.nb_channels; ch++) {
1349 IIRChannel *iir = &s->iir[ch];
1350
1351 for (i = 1; i < iir->nb_ab[0]; i++) {
1352 iir->ab[0][i] /= iir->ab[0][0];
1353 }
1354
1355 iir->ab[0][0] = 1.0;
1356 for (i = 0; i < iir->nb_ab[1]; i++) {
1357 iir->ab[1][i] *= iir->g;
1358 }
1359
1360 normalize_coeffs(ctx, ch);
1361 }
1362
1363 switch (inlink->format) {
1364 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;
1365 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;
1366 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;
1367 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;
1368 }
1369
1370 if (s->format == -2) {
1371 switch (inlink->format) {
1372 case AV_SAMPLE_FMT_DBLP: s->iir_channel = iir_ch_lattice_dblp; break;
1373 case AV_SAMPLE_FMT_FLTP: s->iir_channel = iir_ch_lattice_fltp; break;
1374 case AV_SAMPLE_FMT_S32P: s->iir_channel = iir_ch_lattice_s32p; break;
1375 case AV_SAMPLE_FMT_S16P: s->iir_channel = iir_ch_lattice_s16p; break;
1376 }
1377 }
1378
1379 return 0;
1380 }
1381
filter_frame(AVFilterLink * inlink,AVFrame * in)1382 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
1383 {
1384 AVFilterContext *ctx = inlink->dst;
1385 AudioIIRContext *s = ctx->priv;
1386 AVFilterLink *outlink = ctx->outputs[0];
1387 ThreadData td;
1388 AVFrame *out;
1389 int ch, ret;
1390
1391 if (av_frame_is_writable(in) && s->process != 2) {
1392 out = in;
1393 } else {
1394 out = ff_get_audio_buffer(outlink, in->nb_samples);
1395 if (!out) {
1396 av_frame_free(&in);
1397 return AVERROR(ENOMEM);
1398 }
1399 av_frame_copy_props(out, in);
1400 }
1401
1402 td.in = in;
1403 td.out = out;
1404 ff_filter_execute(ctx, s->iir_channel, &td, NULL, outlink->ch_layout.nb_channels);
1405
1406 for (ch = 0; ch < outlink->ch_layout.nb_channels; ch++) {
1407 if (s->iir[ch].clippings > 0)
1408 av_log(ctx, AV_LOG_WARNING, "Channel %d clipping %d times. Please reduce gain.\n",
1409 ch, s->iir[ch].clippings);
1410 s->iir[ch].clippings = 0;
1411 }
1412
1413 if (in != out)
1414 av_frame_free(&in);
1415
1416 if (s->response) {
1417 AVFilterLink *outlink = ctx->outputs[1];
1418 int64_t old_pts = s->video->pts;
1419 int64_t new_pts = av_rescale_q(out->pts, ctx->inputs[0]->time_base, outlink->time_base);
1420
1421 if (new_pts > old_pts) {
1422 AVFrame *clone;
1423
1424 s->video->pts = new_pts;
1425 clone = av_frame_clone(s->video);
1426 if (!clone)
1427 return AVERROR(ENOMEM);
1428 ret = ff_filter_frame(outlink, clone);
1429 if (ret < 0)
1430 return ret;
1431 }
1432 }
1433
1434 return ff_filter_frame(outlink, out);
1435 }
1436
config_video(AVFilterLink * outlink)1437 static int config_video(AVFilterLink *outlink)
1438 {
1439 AVFilterContext *ctx = outlink->src;
1440 AudioIIRContext *s = ctx->priv;
1441
1442 outlink->sample_aspect_ratio = (AVRational){1,1};
1443 outlink->w = s->w;
1444 outlink->h = s->h;
1445 outlink->frame_rate = s->rate;
1446 outlink->time_base = av_inv_q(outlink->frame_rate);
1447
1448 return 0;
1449 }
1450
init(AVFilterContext * ctx)1451 static av_cold int init(AVFilterContext *ctx)
1452 {
1453 AudioIIRContext *s = ctx->priv;
1454 AVFilterPad pad, vpad;
1455 int ret;
1456
1457 if (!s->a_str || !s->b_str || !s->g_str) {
1458 av_log(ctx, AV_LOG_ERROR, "Valid coefficients are mandatory.\n");
1459 return AVERROR(EINVAL);
1460 }
1461
1462 switch (s->precision) {
1463 case 0: s->sample_format = AV_SAMPLE_FMT_DBLP; break;
1464 case 1: s->sample_format = AV_SAMPLE_FMT_FLTP; break;
1465 case 2: s->sample_format = AV_SAMPLE_FMT_S32P; break;
1466 case 3: s->sample_format = AV_SAMPLE_FMT_S16P; break;
1467 default: return AVERROR_BUG;
1468 }
1469
1470 pad = (AVFilterPad){
1471 .name = "default",
1472 .type = AVMEDIA_TYPE_AUDIO,
1473 .config_props = config_output,
1474 };
1475
1476 ret = ff_append_outpad(ctx, &pad);
1477 if (ret < 0)
1478 return ret;
1479
1480 if (s->response) {
1481 vpad = (AVFilterPad){
1482 .name = "filter_response",
1483 .type = AVMEDIA_TYPE_VIDEO,
1484 .config_props = config_video,
1485 };
1486
1487 ret = ff_append_outpad(ctx, &vpad);
1488 if (ret < 0)
1489 return ret;
1490 }
1491
1492 return 0;
1493 }
1494
uninit(AVFilterContext * ctx)1495 static av_cold void uninit(AVFilterContext *ctx)
1496 {
1497 AudioIIRContext *s = ctx->priv;
1498 int ch;
1499
1500 if (s->iir) {
1501 for (ch = 0; ch < s->channels; ch++) {
1502 IIRChannel *iir = &s->iir[ch];
1503 av_freep(&iir->ab[0]);
1504 av_freep(&iir->ab[1]);
1505 av_freep(&iir->cache[0]);
1506 av_freep(&iir->cache[1]);
1507 av_freep(&iir->biquads);
1508 }
1509 }
1510 av_freep(&s->iir);
1511
1512 av_frame_free(&s->video);
1513 }
1514
1515 static const AVFilterPad inputs[] = {
1516 {
1517 .name = "default",
1518 .type = AVMEDIA_TYPE_AUDIO,
1519 .filter_frame = filter_frame,
1520 },
1521 };
1522
1523 #define OFFSET(x) offsetof(AudioIIRContext, x)
1524 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
1525 #define VF AV_OPT_FLAG_VIDEO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
1526
1527 static const AVOption aiir_options[] = {
1528 { "zeros", "set B/numerator/zeros/reflection coefficients", OFFSET(b_str), AV_OPT_TYPE_STRING, {.str="1+0i 1-0i"}, 0, 0, AF },
1529 { "z", "set B/numerator/zeros/reflection coefficients", OFFSET(b_str), AV_OPT_TYPE_STRING, {.str="1+0i 1-0i"}, 0, 0, AF },
1530 { "poles", "set A/denominator/poles/ladder coefficients", OFFSET(a_str), AV_OPT_TYPE_STRING, {.str="1+0i 1-0i"}, 0, 0, AF },
1531 { "p", "set A/denominator/poles/ladder coefficients", OFFSET(a_str), AV_OPT_TYPE_STRING, {.str="1+0i 1-0i"}, 0, 0, AF },
1532 { "gains", "set channels gains", OFFSET(g_str), AV_OPT_TYPE_STRING, {.str="1|1"}, 0, 0, AF },
1533 { "k", "set channels gains", OFFSET(g_str), AV_OPT_TYPE_STRING, {.str="1|1"}, 0, 0, AF },
1534 { "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, AF },
1535 { "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, AF },
1536 { "format", "set coefficients format", OFFSET(format), AV_OPT_TYPE_INT, {.i64=1}, -2, 4, AF, "format" },
1537 { "f", "set coefficients format", OFFSET(format), AV_OPT_TYPE_INT, {.i64=1}, -2, 4, AF, "format" },
1538 { "ll", "lattice-ladder function", 0, AV_OPT_TYPE_CONST, {.i64=-2}, 0, 0, AF, "format" },
1539 { "sf", "analog transfer function", 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, AF, "format" },
1540 { "tf", "digital transfer function", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "format" },
1541 { "zp", "Z-plane zeros/poles", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "format" },
1542 { "pr", "Z-plane zeros/poles (polar radians)", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "format" },
1543 { "pd", "Z-plane zeros/poles (polar degrees)", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, "format" },
1544 { "sp", "S-plane zeros/poles", 0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, AF, "format" },
1545 { "process", "set kind of processing", OFFSET(process), AV_OPT_TYPE_INT, {.i64=1}, 0, 2, AF, "process" },
1546 { "r", "set kind of processing", OFFSET(process), AV_OPT_TYPE_INT, {.i64=1}, 0, 2, AF, "process" },
1547 { "d", "direct", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "process" },
1548 { "s", "serial", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "process" },
1549 { "p", "parallel", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "process" },
1550 { "precision", "set filtering precision", OFFSET(precision),AV_OPT_TYPE_INT, {.i64=0}, 0, 3, AF, "precision" },
1551 { "e", "set precision", OFFSET(precision),AV_OPT_TYPE_INT, {.i64=0}, 0, 3, AF, "precision" },
1552 { "dbl", "double-precision floating-point", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "precision" },
1553 { "flt", "single-precision floating-point", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "precision" },
1554 { "i32", "32-bit integers", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "precision" },
1555 { "i16", "16-bit integers", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, "precision" },
1556 { "normalize", "normalize coefficients", OFFSET(normalize),AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, AF },
1557 { "n", "normalize coefficients", OFFSET(normalize),AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, AF },
1558 { "mix", "set mix", OFFSET(mix), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, AF },
1559 { "response", "show IR frequency response", OFFSET(response), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, VF },
1560 { "channel", "set IR channel to display frequency response", OFFSET(ir_channel), AV_OPT_TYPE_INT, {.i64=0}, 0, 1024, VF },
1561 { "size", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str = "hd720"}, 0, 0, VF },
1562 { "rate", "set video rate", OFFSET(rate), AV_OPT_TYPE_VIDEO_RATE, {.str = "25"}, 0, INT32_MAX, VF },
1563 { NULL },
1564 };
1565
1566 AVFILTER_DEFINE_CLASS(aiir);
1567
1568 const AVFilter ff_af_aiir = {
1569 .name = "aiir",
1570 .description = NULL_IF_CONFIG_SMALL("Apply Infinite Impulse Response filter with supplied coefficients."),
1571 .priv_size = sizeof(AudioIIRContext),
1572 .priv_class = &aiir_class,
1573 .init = init,
1574 .uninit = uninit,
1575 FILTER_INPUTS(inputs),
1576 FILTER_QUERY_FUNC(query_formats),
1577 .flags = AVFILTER_FLAG_DYNAMIC_OUTPUTS |
1578 AVFILTER_FLAG_SLICE_THREADS,
1579 };
1580