File: filter.c

package info (click to toggle)
bristol 0.60.10-3
  • links: PTS, VCS
  • area: main
  • in suites: wheezy
  • size: 15,652 kB
  • sloc: ansic: 124,457; sh: 10,579; makefile: 111
file content (1026 lines) | stat: -rw-r--r-- 28,810 bytes parent folder | download | duplicates (3)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
397
398
399
400
401
402
403
404
405
406
407
408
409
410
411
412
413
414
415
416
417
418
419
420
421
422
423
424
425
426
427
428
429
430
431
432
433
434
435
436
437
438
439
440
441
442
443
444
445
446
447
448
449
450
451
452
453
454
455
456
457
458
459
460
461
462
463
464
465
466
467
468
469
470
471
472
473
474
475
476
477
478
479
480
481
482
483
484
485
486
487
488
489
490
491
492
493
494
495
496
497
498
499
500
501
502
503
504
505
506
507
508
509
510
511
512
513
514
515
516
517
518
519
520
521
522
523
524
525
526
527
528
529
530
531
532
533
534
535
536
537
538
539
540
541
542
543
544
545
546
547
548
549
550
551
552
553
554
555
556
557
558
559
560
561
562
563
564
565
566
567
568
569
570
571
572
573
574
575
576
577
578
579
580
581
582
583
584
585
586
587
588
589
590
591
592
593
594
595
596
597
598
599
600
601
602
603
604
605
606
607
608
609
610
611
612
613
614
615
616
617
618
619
620
621
622
623
624
625
626
627
628
629
630
631
632
633
634
635
636
637
638
639
640
641
642
643
644
645
646
647
648
649
650
651
652
653
654
655
656
657
658
659
660
661
662
663
664
665
666
667
668
669
670
671
672
673
674
675
676
677
678
679
680
681
682
683
684
685
686
687
688
689
690
691
692
693
694
695
696
697
698
699
700
701
702
703
704
705
706
707
708
709
710
711
712
713
714
715
716
717
718
719
720
721
722
723
724
725
726
727
728
729
730
731
732
733
734
735
736
737
738
739
740
741
742
743
744
745
746
747
748
749
750
751
752
753
754
755
756
757
758
759
760
761
762
763
764
765
766
767
768
769
770
771
772
773
774
775
776
777
778
779
780
781
782
783
784
785
786
787
788
789
790
791
792
793
794
795
796
797
798
799
800
801
802
803
804
805
806
807
808
809
810
811
812
813
814
815
816
817
818
819
820
821
822
823
824
825
826
827
828
829
830
831
832
833
834
835
836
837
838
839
840
841
842
843
844
845
846
847
848
849
850
851
852
853
854
855
856
857
858
859
860
861
862
863
864
865
866
867
868
869
870
871
872
873
874
875
876
877
878
879
880
881
882
883
884
885
886
887
888
889
890
891
892
893
894
895
896
897
898
899
900
901
902
903
904
905
906
907
908
909
910
911
912
913
914
915
916
917
918
919
920
921
922
923
924
925
926
927
928
929
930
931
932
933
934
935
936
937
938
939
940
941
942
943
944
945
946
947
948
949
950
951
952
953
954
955
956
957
958
959
960
961
962
963
964
965
966
967
968
969
970
971
972
973
974
975
976
977
978
979
980
981
982
983
984
985
986
987
988
989
990
991
992
993
994
995
996
997
998
999
1000
1001
1002
1003
1004
1005
1006
1007
1008
1009
1010
1011
1012
1013
1014
1015
1016
1017
1018
1019
1020
1021
1022
1023
1024
1025
1026

/*
 *  Diverse Bristol audio routines.
 *  Copyright (c) by Nick Copeland <nickycopeland@hotmail.com> 1996,2012
 *
 *
 *   This program is free software; you can redistribute it and/or modify
 *   it under the terms of the GNU General Public License as published by
 *   the Free Software Foundation; either version 3 of the License, or
 *   (at your option) any later version.
 *
 *   This program is distributed in the hope that it will be useful,
 *   but WITHOUT ANY WARRANTY; without even the implied warranty of
 *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 *   GNU General Public License for more details.
 *
 *   You should have received a copy of the GNU General Public License
 *   along with this program; if not, see <http://www.gnu.org/licenses/>.
 *
 */

/*#define BRISTOL_DBG */
/*
 * Need to have basic template for an operator. Will consist of
 *
 *	filterinit()
 *	operate()
 *	reset()
 *	destroy()
 *
 *	destroy() is in the library.
 *
 * Operate will be called when all the inputs have been loaded, and the result
 * will be an output buffer written to the next operator.
 *
 * This file includes the Colin Fletcher fkc factorisation optimisation, his
 * v2 factorisation and some other optimisation removing x, x1, x1, tanh1,
 * tanh2, kf and exp_out variables.
 */
/*
 * This implements three filter algorithms used by the different bristol
 * emulations. These are a butterworth used by the leslie, a chamberlain 
 * used generally, and a rooney used for some of the filter layering and
 * eventually the mixing.
 */
#include <math.h>

#include "bristol.h"
#include "bristolblo.h"
#include "filter.h"

/*
 * The name of this operator, IO count, and IO names.
 */
#define OPNAME "DCF"
#define OPDESCRIPTION "Digital Filter One"
#define PCOUNT 7
#define IOCOUNT 3

#define FILTER_IN_IND 0
#define FILTER_MOD_IND 1
#define FILTER_OUT_IND 2

static float srate;

/*
 * When this uses cfreq we get very good keyboard tracking as cfreq is a true
 * frequency mapping. This filter is supposed to be used with the cFreq tables
 * which do not have that same mapping - they are buffer strides rather than
 * frequency tables and this needs to be corrected here as the use of cfreq
 * causes cutoff jumps when glide is applied.
 *
 * cFreq = 1024 * cfreq / srate
 * cfreq = cFreq * srate / 1024
 * cfreq = cFreq * srate / 1024
 */
//#define getcoff(c, k) ((c*c)*(1.0f-k)*20000 + k*4*(voice->cFreq * srate / 1024)) / srate
//#define getcoff(c, k) ((c*c)*(1.0f-k)*20000.0f + voice->cfreq*k*srate*0.00390625f) / srate
//#define getcoff(c, k) ((c*c)*(1.0f-k)*20000.0f + k*4*voice->cfreq) / srate
#define getcoff(c, k) ((c*c)*(1.0f-k)*20000.0f + k*k*20000.0) / srate

//#define TANHF(x) btanhf(voice->baudio->mixflags, x)
#define TANHF(x) btanhf(mode, x)
//#define TANHF(v) (v + 1e-10f) / sqrtf(1 + v * v)
//#define TANHF(x) mode == 0? tanhf(x):0
#define TANHFEED(x) btanhfeed(mode, x)
//#define TANHFEED tanhf
//#define TANHFEED(v) (v + 1e-10f) / sqrtf(1 + v * v)
#define _f_lim (20000.0 / srate)

#define V2 40000.0
#define OV2 0.000025
#define F_RESAMPLE 88000

static float
btanhf(int mode, float v)
{
	/*
	 * This should be 4 modes as we have two bit flags. Mode 1 are the real
	 * lightweight chamberlain. That is more work on flag checking.
	 */
	switch (mode) {
		case 0: return(v);
		default:
		case 2: return((v + 1e-10f) / sqrtf(1 + v * v));
		case 3: return(tanhf(v));
	}
}

static float
btanhfeed(int mode, float v)
{
	/*
	 * This should be 4 modes as we have two bit flags. Mode 1 are the real
	 * lightweight chamberlain. That is more work on flag checking.
	 */
	switch (mode) {
		default:
		case 0:
		case 2: return((v + 1e-10f) / sqrtf(1 + v * v));
		case 3: return(tanhf(v));
	}
}

/*
 * Reset any local memory information.
 */
static int destroy(bristolOP *operator)
{
#ifdef BRISTOL_DBG
	printf("destroy(%x)\n", operator);
#endif

	/*
	 * Unmalloc anything we added to this structure
	 */
	bristolfree(operator->specs);

	/*
	 * Free any local memory. We should also free ourselves, since we did the
	 * initial allocation.
	 */
	cleanup(operator);
	return(0);
}

#define ROOT2 1.4142135623730950488
double pidsr;

/*
 * Reset any local memory information.
 */
static int reset(bristolOP *operator, bristolOPParams *param)
{
	float *a, c, d;

#ifdef BRISTOL_DBG
	printf("reset(%x)\n", operator);
#endif

	param->param[0].float_val = 0.5;
	param->param[1].float_val = 0.5;
	param->param[2].float_val = 1.0;
	param->param[3].int_val = 0;
	param->param[4].int_val = 0;
	param->param[5].float_val = 1.0;

	a = (float *) bristolmalloc(8 * sizeof(float));
	param->param[0].mem = a;

	pidsr = M_PI / sqrt(M_E);

	/*
	 * Configure a lowpass butterworth
	 */
	c = 1.0 / (float) tan((double) pidsr * 0.5);

	a[1] = 1.0 / (1.0 + ROOT2 * c + c * c);
	a[2] = 2 * a[1];
	a[3] = a[1];
	a[4] = 2.0 * (1.0 - c*c) * a[1];
	a[5] = (1.0 - ROOT2 * c + c * c) * a[1];

	a[6] = 0;
	a[7] = 0;

	/*
	 * Then configure a bandbass butterworth.
	 */
	a = (float *) bristolmalloc(8 * sizeof(float));
	param->param[1].mem = a;

	c = 1.0 / tan((double)(pidsr * 0.5));
	d = 2.0 * cos(2.0 * (double)(pidsr * 0.5));

	a[1] = 1.0 / (1.0 + c);
	a[2] = 0.0;
	a[3] = -a[1];
	a[4] = - c * d * a[1];
	a[5] = (c - 1.0) * a[1];

	a[6] = 0;
	a[7] = 0;

	return(0);
}

/*
 * Alter an internal parameter of an operator.
 */
static int
param(bristolOP *operator, bristolOPParams *param,
	unsigned char index, float value)
{
	int intvalue = value * CONTROLLER_RANGE;

#ifdef BRISTOL_DBG
	printf("param(%x, %f)\n", index, value);
#endif

	if (blo.flags & BRISTOL_LWF)
		param->param[4].int_val = 0;

	switch (index) {
		case 0:
		{
			if (param->param[4].int_val == 3) {
				param->param[index].float_val =
					gainTable[(int) (value * (CONTROLLER_RANGE - 1))].gain;
			} else if (param->param[4].int_val == 4) {
				param->param[index].float_val = value;
			} else if (param->param[4].int_val == 1) {
				/*param->param[index].float_val = */
				/*	gainTable[(int) (value * (CONTROLLER_RANGE - 1))].gain; */
				if ((param->param[index].float_val = value) < (float) 0.000122)
					param->param[index].float_val = (float) 0.000122;
			} else {
					param->param[index].float_val = value / 3;
			}
/*printf("Configuring filter %f\n", param->param[index].float_val); */
			param->param[index].int_val = intvalue;
			break;
		}
		case 1:
			if (param->param[4].int_val == 1)
				param->param[index].float_val =
					gainTable[CONTROLLER_RANGE - 1
						- (int) (value * (CONTROLLER_RANGE - 1))].gain;
			else
				param->param[index].float_val = value;
			param->param[index].int_val = intvalue;
			break;
		case 2:
		case 5: /* Gain */
			param->param[index].float_val = value;
			break;
		case 3:
			if (value > 0)
				param->param[index].int_val = 1;
			else
				param->param[index].int_val = 0;
			param->param[index].float_val = value;
			break;
		case 4:
			/* See if we are trying to avoid Huovilainen filters */
			if ((intvalue == 4) && (blo.flags & BRISTOL_LWF))
				intvalue = 0;
			if (intvalue > 1) {
				param->param[index].int_val = intvalue;
			} else if (value > 0) {
				float floatvalue = ((float) param->param[0].int_val) /
					(CONTROLLER_RANGE - 1);

				param->param[index].int_val = 1;
				/*
				printf("Selected rooney filter\n");
				 * We also need to rework a couple of the parameters when
				 * changing the filter selection
				 */
				if ((param->param[0].float_val = floatvalue)
					< (float) 0.000122)
					param->param[0].float_val = (float) 0.000122;

				floatvalue = ((float) param->param[1].int_val) /
					(CONTROLLER_RANGE - 1);

				param->param[1].float_val =
					gainTable[CONTROLLER_RANGE - 1
						- (int) (floatvalue * (CONTROLLER_RANGE - 1))].gain;
			} else {
				float floatvalue = ((float) param->param[0].int_val) /
					(CONTROLLER_RANGE - 1);

				printf("Selected chamberlain filter\n");
				param->param[index].int_val = 0;
				param->param[0].float_val = value / 3;

				floatvalue = ((float) param->param[1].int_val) /
					(CONTROLLER_RANGE - 1);
				param->param[1].float_val = value;
			}
			break;
		case 6: /* LP/BP/HP */
			param->param[index].int_val = intvalue;
			break;
	}

	return(0);
}

/*
 * Nice filter, but not with dynamic parameters for cutoff or mods. Use by the
 * leslie for the crossover between horns and voice coil.
 */
void butter_filter(register float *in, register float *out, register float *a,
register float gain, register int count) /* Filter loop */
{
    register float t, y;

    do {
      t = *in++ - a[4] * a[6] - a[5] * a[7];
      y = t * a[1] + a[2] * a[6] + a[3] * a[7];
      a[7] = a[6];
      a[6] = t;
      *out++ += y * gain;
    } while (--count);
}

/*
 * This looks odd being global however it is for denormal reduction, we inject
 * a stupidly small amount of noise into the houvilainen filter to give is some
 * constant signal. The noise algorithm will work over multiple voices with no
 * detrimental effect.
 */
static float scale = 0.00000000001f;
static int dngx1 = 0x67452301;
static int dngx2 = 0xefcdab89;

/*
 * filter - takes input signal and filters it according to the mod level.
 */
static int operate(register bristolOP *operator, bristolVoice *voice,
	bristolOPParams *param,
	void *lcl)
{
	bristolFILTERlocal *local = lcl;
	register int count, mode;
	register float *ib, *ob, *mb;
	bristolFILTER *specs;
	register float BLim, Res, Mod, gain;
	register float Bout = local->Bout;
	register float oSource = local->oSource;

	register float velocity, output, ccut, cvdelta, cutoff;

	mode = bfiltertype(voice->baudio->mixflags);

	/*
	 * Every operator accesses these variables, the count, and a pointer to
	 * each buffer. We should consider passing them as readymade parameters?
	 *
	 * The Filter now takes normalised inputs, in ranges of 12PO.
	 */
	specs = (bristolFILTER *) operator->specs;
	count = specs->spec.io[FILTER_OUT_IND].samplecount;
	ib = specs->spec.io[FILTER_IN_IND].buf;
	mb = specs->spec.io[FILTER_MOD_IND].buf;
	ob = specs->spec.io[FILTER_OUT_IND].buf;

#ifdef BRISTOL_DBG
	printf("filter(%x, %x, %x)\n", operator, param, local);
#endif

/*param->param[4].int_val = 5; */
/*if (mb == 0) */
/*return;

	if (blo.flags & BRISTOL_LWF)
		voice->baudio->mixflags |= BRISTOL_LW_FILTER;
 */

	/* See if we  are limited to lightweight filters */
	if (param->param[4].int_val != 3)
	{
		if ((blo.flags & BRISTOL_LWF) || (mode == 1))
			param->param[4].int_val = 0;
		else
			param->param[4].int_val = 4;
	}

	if (param->param[4].int_val == 1) {
		/*
		 * This is the code from one of the SLab floating point filter routines.
		 * It has been altered here to have a single lowpass filter, and will be
		 * a starting point for the Bristol filters. There will end up being a
		 * number of filter algorithms.
		 */

		if (param->param[3].int_val) {
			if ((BLim = param->param[0].float_val + voice->dFreq /
				(128 * param->param[3].float_val + 1)) > 1.0)
				BLim = 1.0;
			cutoff = param->param[0].float_val * voice->dFreq /
				(8 * param->param[3].float_val + 1);
			if ((Res = param->param[1].float_val * voice->dFreq / 
				(8 * param->param[3].float_val + 1)) == 0)
				Res = 0.000030;
			if (Res > 1.)
				Res = 1.0;
		} else {
			cutoff = BLim = param->param[0].float_val;
			Res = param->param[1].float_val;
		}
		if (voice->flags & BRISTOL_KEYON)
		{
			/*
			 * Do any relevant note_on initialisation.
			 */
			output = 0;
			velocity = 0;
			oSource = 0.0;
			Bout = 0.0;
		} else {
			output = local->output;
			velocity = local->velocity;
		}

		Mod = param->param[2].float_val;
		/*
		 * subtract a delta from original signal?
		 */
		for (; count > 0; count-=1)
		{
			/*
			 * Evaluate a resonant filter
			 */
			cvdelta = cutoff * Res;
			if ((ccut = (cutoff * (1 - Mod)) + *mb * Mod) < 0.001)
				ccut = 0.001;
			velocity += ((*ib - output) * cvdelta);
			if (velocity > ccut)
				velocity = ccut;
			if (velocity < -ccut)
				velocity = -ccut;
			output += velocity;

			/*
			 * Find out our current BLim. We have a specified limit, the
			 * cuttoff. We have our mod signal, and an amount of that mod
			 * signal. The ADSR produces a value between 0.0 and 1.0, so we
			 * could add this to BLim?
			 *
			 * Take the bass component using a rooney filter.
			 */
			if ((gain = (BLim * (1 - Mod) + (*mb++ * Mod))) > 1.0)
				gain = 1.0;
			else if (gain < 0.0)
				gain = 0.0;
			/*
			 * Cross them over. Rooneys are a lot cleaner.
			 */
			*ob++ = ((Bout += (*ib++ - Bout) * gain) * Res +
				output * (1 - Res) * 4) * (1.25 - BLim) * 4;
		}
		/*
		 * Put back the local variables
		 */
		local->Bout = Bout;
		local->oSource = oSource;
		local->output = output;
		local->velocity = velocity;
	} else if (param->param[4].int_val == 2) {
		/* Moog filter */
		register float fc, res, f, fb,
			in1 = local->delay1,
			in2 = local->delay2, 
			in3 = local->delay3,
			in4 = local->delay4,
			out1 = local->out1,
			out2 = local->out2, 
			out3 = local->out3,
			out4 = local->out4;

		Mod = param->param[2].float_val;

		res = param->param[1].float_val * 3.92f;
		fc = param->param[0].float_val;

		if (voice->flags & BRISTOL_KEYON)
			in1 = in2 = in3 = in4 = out1 = out2 = out3 = out4 = 0;

		for (; count > 0; count-=1)
		{
			if ((f = fc * 1.16 + *mb++ * Mod) > 1.0)
				f = 1.0;
			if (f < 0.0f)
				f = 0.000001;
			fb = res * (1.0 - 0.15 * f * f);

			*ib /= 12;

			*ib -= out4 * fb;
			*ib *= 0.35013 * (f*f)*(f*f);
			out1 = *ib + 0.3 * in1 + (1 - f) * out1; /* Pole 1 */
			in1  = *ib++;
			out2 = out1 + 0.3 * in2 + (1 - f) * out2;  /* Pole 2 */
			in2  = out1;
			out3 = out2 + 0.3 * in3 + (1 - f) * out3;  /* Pole 3 */
			in3  = out2;
			out4 = out3 + 0.3 * in4 + (1 - f) * out4;  /* Pole 4 */
			in4  = out3;

			*ob++ = out4 * 12;
		}

		local->delay1 = in1;
		local->delay2 = in2;
		local->delay3 = in3;
		local->delay4 = in4;
		local->out1 = out1;
		local->out2 = out2;
		local->out3 = out3;
		local->out4 = out4;
	} else if (param->param[4].int_val == 3) {
		/*
		 * Simplified rooney filter.
		 */
		BLim = param->param[0].float_val;
/*		BLim = gainTable[(int) (BLim * 3 * (CONTROLLER_RANGE - 1))].gain; */

		if (voice->flags & BRISTOL_KEYON)
			/*
			 * Do any relevant note_on initialisation.
			 */
			Bout = 0.0;

		/*
		 * subtract a delta from original signal?
		 */
		for (; count > 0; count-=4)
		{
			*ob++ = (Bout += ((*ib++ - Bout) * BLim));
			*ob++ = (Bout += ((*ib++ - Bout) * BLim));
			*ob++ = (Bout += ((*ib++ - Bout) * BLim));
			*ob++ = (Bout += ((*ib++ - Bout) * BLim));
		}

		/*
		 * Put back the local variables
		 */
		local->Bout = Bout;
	} else if ((param->param[4].int_val == 4) && (srate >= F_RESAMPLE)) {
		/*
		 * This is an implementation of Antti Huovilainen's non-linear Moog
		 * emulation, tweaked just slightly to align with the sometimes rather
		 * dubious bristol signal levels (its not that they are bad, just that
		 * historically they have used 1.0 per octave rather than +/-1.0).
		 *
		 * This one does not oversample - if the cutoff does not approach
		 * nyquist then we do not have to correct for its inaccuracies.
		 */
		float az1 = local->az1;
		float az2 = local->az2;
		float az3 = local->az3;
		float az4 = local->az4;
		float az5 = local->az5;
		float ay1 = local->ay1;
		float ay2 = local->ay2;
		float ay3 = local->ay3;
		float ay4 = local->ay4;
		float amf = local->amf;

//		float ov2 = 0.000025;// twice the 'thermal voltage of a transistor'

		float kfc;
		float kfcr;
		float kacr;
		float k2vg;
		float coff;

		float resonance = param->param[1].float_val;
		Mod = param->param[2].float_val * param->param[2].float_val * 0.03;

		/*
		 * Cutoff is a power curve for better control at lower frequencies and
		 * the key is used for tracking purposes, it should be possible to make
		 * it reasonably linear at somewhere under unity
		 *
		 * Cutoff goes from 0 to 1.0 = nyquist. Key tracking should be reviewed,
		 * the value should be quite linear for the filter, 0..Nyquist
		 *
		 * If we take midi key 0 = 8Hz and 127 = 12658.22 Hz then using
		 * our samplerate we should be able to fix some tuning:
		 *
		 * (voice->key.key * 12650.22 + 8) / (127 * srate)
		 *
		 * We want to position param[3] such that it tunes at 0.5 and can then
		 * be notched in the GUI, hence srate/4 rathern than /2.
		 *
		 * Needed some changes to key tracking, it was a bit bipolar.
		coff = (param->param[0].float_val * param->param[0].float_val)
			* (1.0f - param->param[3].float_val)
			* 20000 / srate
				+
			param->param[3].float_val * 4 * voice->cfreq / srate;
		coff = getcoff(param->param[0].float_val, param->param[3].float_val);
		 */
		if (param->param[3].float_val == 0)
			coff = param->param[0].float_val * param->param[0].float_val
				* 20000 / srate;
		else
			coff = param->param[3].float_val * param->param[0].float_val
				* 4 * voice->cfreq / srate;

		 /*
		if (param->param[3].int_val == 0)
			coff = param->param[0].float_val * param->param[0].float_val
				* 20000 / srate;
		else // Fraction of the current frequency by tracking by cutoff.
			coff = param->param[3].float_val * param->param[0].float_val * 4 *
				voice->cfreq / srate;
		 */

//			* voice->cfreq * param->param[3].float_val / srate;
//printf("%f %f %f\n", voice->cfreq, coff, param->param[3].float_val);

		float lim = _f_lim;

		for (; count > 0; count--)
		{
			/*
			 * The 0.83 is arbitrary, it is used to prevent the filter being
			 * overdriven however I think 0.5 would be a better value seeing
			 * as the filter is 2x oversampling?
			 *
			 * We should really interpret coff (the configured frequency) as
			 * a function up to about 20KHz whatever the resampling rate.
			 */
			if ((kfc = coff + *mb++ * Mod) > lim)
				kfc = lim;
			else if (kfc < 1e-10f)
				kfc = 1e-10f;

			// frequency & amplitude correction
			kfcr = kfc * ( kfc * (1.8730 * kfc + 0.4955) - 0.6490) + 0.9988;
			kacr = kfc * (-3.9364 * kfc + 1.8409) + 0.9968;

			k2vg = (1 - expf(-2.0 * M_PI * kfcr * kfc));

			// cascade of 4 1st order sections
			dngx1 ^= dngx2;
			ay1  = az1 + k2vg * (TANHFEED((*ib + dngx2 * scale) * OV2
				- 4*resonance*amf*kacr) - TANHF(az1));
			dngx2 += dngx1;
			az1  = ay1;

			ay2  = az2 + k2vg *(TANHF(ay1) -TANHF(az2));
			az2  = ay2;

			ay3  = az3 + k2vg * (TANHF(ay2) - TANHF(az3));
			az3  = ay3;

			ay4  = az4 + k2vg * (TANHF(ay3) - TANHF(az4));
			az4  = ay4;

			amf  = ay4;

			*ob++ += amf * V2;
			ib++;
		}

		local->az1 = az1;
		local->az2 = az2;
		local->az3 = az3;
		local->az4 = az4;
		local->az5 = az5;
		local->ay1 = ay1;
		local->ay2 = ay2;
		local->ay3 = ay3;
		local->ay4 = ay4;
		local->amf = amf;
	} else if ((param->param[4].int_val == 4) && (srate < F_RESAMPLE)) {
		/*
		 * This is an implementation of Antti Huovilainen's non-linear Moog
		 * emulation, tweaked just slightly to align with the sometimes rather
		 * dubious bristol signal levels (its not that they are bad, just that
		 * historically they have used 1.0 per octave rather than +/-1.0).
		 */
		float az1 = local->az1;
		float az2 = local->az2;
		float az3 = local->az3;
		float az4 = local->az4;
		float az5 = local->az5;
		float ay1 = local->ay1;
		float ay2 = local->ay2;
		float ay3 = local->ay3;
		float ay4 = local->ay4;
		float amf = local->amf;

//		float ov2 = 0.000025;// twice the 'thermal voltage of a transistor'

		float kfc;
		float kfcr;
		float kacr;
		float k2vg;
		float coff;

		float resonance = param->param[1].float_val;
		Mod = param->param[2].float_val * param->param[2].float_val * 0.02;

		/*
		 * Cutoff is a power curve for better control at lower frequencies and
		 * the key is used for tracking purposes, it should be possible to make
		 * it reasonably linear at somewhere under unity
		 *
		 * Cutoff goes from 0 to 1.0 = nyquist. Key tracking should be reviewed,
		 * the value should be quite linear for the filter, 0..Nyquist
		 *
		 * If we take midi key 0 = 8Hz and 127 = 12658.22 Hz then using
		 * our samplerate we should be able to fix some tuning:
		 *
		 * (voice->key.key * 12650.22 + 8) / (127 * srate)
		 *
		 * We want to position param[3] such that it tunes at 0.5 and can then
		 * be notched in the GUI, hence srate/4 rathern than /2.
		 *
		 * Needed some changes to key tracking, it was a bit bipolar.
		coff = (param->param[0].float_val * param->param[0].float_val)
			* (1.0f - param->param[3].float_val)
			* 20000 / srate
				+
			param->param[3].float_val * 4 * voice->cfreq / srate;
		coff = getcoff(param->param[0].float_val, param->param[3].float_val);
		 */
		if (param->param[3].float_val == 0)
			coff = param->param[0].float_val * param->param[0].float_val
				* 20000 / srate;
		else
			coff = param->param[3].float_val * param->param[0].float_val
				* 4 * voice->cfreq / srate;

		 /*
		if (param->param[3].int_val == 0)
			coff = param->param[0].float_val * param->param[0].float_val
				* 20000 / srate;
		else // Fraction of the current frequency by tracking by cutoff.
			coff = param->param[3].float_val * param->param[0].float_val * 4 *
				voice->cfreq / srate;
		 */

//			* voice->cfreq * param->param[3].float_val / srate;
//printf("%f %f %f\n", voice->cfreq, coff, param->param[3].float_val);

		for (; count > 0; count--)
		{
			/*
			 * The 0.83 is arbitrary, it is used to prevent the filter being
			 * overdriven however I think 0.5 would be a better value seeing
			 * as the filter is 2x oversampling?
			 *
			 * We should really interpret coff (the configured frequency) as
			 * a function up to about 20KHz whatever the resampling rate.
			 */
			if ((kfc = coff + *mb++ * Mod) > 0.5)
				kfc = 0.5;
			else if (kfc < 1e-10f)
				kfc = 1e-10f;

			// frequency & amplitude correction
			kfcr = kfc * ( kfc * (1.8730 * kfc + 0.4955) - 0.6490) + 0.9988;
			kacr = kfc * (-3.9364 * kfc + 1.8409) + 0.9968;

			// filter tuning
			k2vg = (1 - expf(-2.0 * M_PI * kfcr * kfc * 0.5));

			// cascade of 4 1st order sections
			ay1  = az1 + k2vg * (TANHFEED((*ib*OV2 - 4*resonance*amf*kacr))
				- TANHF(az1));
			az1  = ay1;

			ay2  = az2 + k2vg * (TANHF(ay1) - TANHF(az2));
			az2  = ay2;

			ay3  = az3 + k2vg * (TANHF(ay2) - TANHF(az3));
			az3  = ay3;

			ay4  = az4 + k2vg * (TANHF(ay3) - TANHF(az4));
			az4  = ay4;

			// 1/2-sample delay for phase compensation
			amf  = (ay4+az5)*0.5;
			az5  = ay4;

			// oversampling (repeat same block) and inject noise for denormals
			dngx1 ^= dngx2;
			ay1  = az1 + k2vg * (TANHFEED((*ib +dngx2 * scale) * OV2
				- 4*resonance*amf*kacr) - TANHF(az1));
			dngx2 += dngx1;
			az1  = ay1;

			ay2  = az2 + k2vg * (TANHF(ay1) - TANHF(az2));
			az2  = ay2;

			ay3  = az3 + k2vg * (TANHF(ay2) - TANHF(az3));
			az3  = ay3;

			ay4  = az4 + k2vg * (TANHF(ay3) - TANHF(az4));
			az4  = ay4;

			// 1/2-sample delay for phase compensation
			amf  = (ay4+az5)*0.5;
			az5  = ay4;

			*ob++ += amf * V2 * 0.5;
			ib++;
		}

		local->az1 = az1;
		local->az2 = az2;
		local->az3 = az3;
		local->az4 = az4;
		local->az5 = az5;
		local->ay1 = ay1;
		local->ay2 = ay2;
		local->ay3 = ay3;
		local->ay4 = ay4;
		local->amf = amf;

	} else {
		/* The chamberlain */
		register float freqcut, highpass, qres,
			delay1 = local->delay1,
			delay2 = local->delay2, 
			delay3 = local->delay3,
			delay4 = local->delay4;
		register int hp = param->param[6].int_val;

		Mod = param->param[2].float_val;

		if (voice->flags & BRISTOL_KEYON)
		{
			delay1 = 0;
			delay2 = 0;
			delay3 = 0;
			delay4 = 0;
		}

		qres = 2.0f - param->param[1].float_val * 1.97f;
		cutoff = param->param[0].float_val;
		gain = param->param[5].float_val * 0.02513;

		/*
		 * The following was for on/off keytracking, needs to be continuous.
		if (param->param[3].int_val)
			cutoff += voice->dFreq / 128.0f;
		 */
		cutoff += param->param[3].float_val * voice->key.key / 512.0f;

		for (; count > 0; count-=1)
		{
			/*
			 * Hal Chamberlin's state variable filter. These are cascaded low
			 * pass (rooney type) filters with feedback. They are not bad - 
			 * they are resonant with a reasonable dB/octave, but the
			 * resonance is not as warm as an analogue equivalent.
			 */
			freqcut = cutoff * 2.0f + *mb++ * Mod / 12.0f;

			if (freqcut > VCF_FREQ_MAX)
				freqcut = VCF_FREQ_MAX;
			else if (freqcut <= 0.000001)
				freqcut = 0.000001;

			/* delay2/4 = lowpass output */
			delay2 = delay2 + freqcut * delay1;

			highpass = *ib++ * 16 - delay2 - qres * delay1;

			/* delay1/3 = bandpass output */
			delay1 = freqcut * highpass + delay1;

			delay4 = delay4 + freqcut * delay3;
			highpass = delay2 - delay4 - qres * delay3;
			delay3 = freqcut * highpass + delay3;

			/* mix filter output into output buffer */
			switch (hp) {
				default: /* LP */
					*ob++ += delay4 * gain;
					break;
				case 1: /* BP */
					*ob++ += delay3 * gain;
					break;
				case 2: /* HP */
					*ob++ += highpass * gain;
					break;
			}
		}

		local->delay1 = delay1;
		local->delay2 = delay2;
		local->delay3 = delay3;
		local->delay4 = delay4;

	}

	return(0);
}

/*
 * Setup any variables in our OP structure, in our IO structures, and malloc
 * any memory we need.
 */
bristolOP *
filterinit(bristolOP **operator, int index, int samplerate, int samplecount)
{
	bristolFILTER *specs;

#ifdef BRISTOL_DBG
	printf("filterinit(%x(%x), %i, %i, %i)\n",
		operator, *operator, index, samplerate, samplecount);
#endif

	srate = samplerate;

	*operator = bristolOPinit(operator, index, samplecount);

	/*
	 * Then the local parameters specific to this operator. These will be
	 * the same for each operator, but must be inited in the local code.
	 */
	(*operator)->operate = operate;
	(*operator)->destroy = destroy;
	(*operator)->reset = reset;
	(*operator)->param = param;

	specs = (bristolFILTER *) bristolmalloc0(sizeof(bristolFILTER));
	(*operator)->specs = (bristolOPSpec *) specs;
	(*operator)->size = sizeof(bristolFILTER);

	/*
	 * These are specific to this operator, and will need to be altered for
	 * each operator.
	 */
	specs->spec.opname = OPNAME;
	specs->spec.description = OPDESCRIPTION;
	specs->spec.pcount = PCOUNT;
	specs->spec.iocount = IOCOUNT;
	specs->spec.localsize = sizeof(bristolFILTERlocal);

	/*
	 * Now fill in the control specs for this filter.
	 */
	specs->spec.param[0].pname = "cutoff";
	specs->spec.param[0].description = "Filter cutoff frequency";
	specs->spec.param[0].type = BRISTOL_FLOAT;
	specs->spec.param[0].low = 0;
	specs->spec.param[0].high = 1;
	specs->spec.param[0].flags = BRISTOL_ROTARY|BRISTOL_SLIDER;

	specs->spec.param[1].pname = "resonance";
	specs->spec.param[1].description = "Filter emphasis";
	specs->spec.param[1].type = BRISTOL_FLOAT;
	specs->spec.param[1].low = 0;
	specs->spec.param[1].high = 1;
	specs->spec.param[1].flags = BRISTOL_ROTARY|BRISTOL_SLIDER;

	specs->spec.param[2].pname = "modulation";
	specs->spec.param[2].description = "Depth of modulation control";
	specs->spec.param[2].type = BRISTOL_FLOAT;
	specs->spec.param[2].low = 0;
	specs->spec.param[2].high = 1;
	specs->spec.param[2].flags = BRISTOL_ROTARY|BRISTOL_SLIDER;

	specs->spec.param[3].pname = "keyboard tracking";
	specs->spec.param[3].description = "Cutoff tracks keyboard";
	specs->spec.param[3].type = BRISTOL_TOGGLE;
	specs->spec.param[3].low = 0;
	specs->spec.param[3].high = 1;
	specs->spec.param[3].flags = BRISTOL_BUTTON;

	specs->spec.param[4].pname = "filter type";
	specs->spec.param[4].description = "Chamberlain or Rooney filter";
	specs->spec.param[4].type = BRISTOL_TOGGLE;
	specs->spec.param[4].low = 0;
	specs->spec.param[4].high = 1;
	specs->spec.param[4].flags = BRISTOL_BUTTON;

	specs->spec.param[5].pname = "gain";
	specs->spec.param[5].description = "Filter gain";
	specs->spec.param[5].type = BRISTOL_FLOAT;
	specs->spec.param[5].low = 0;
	specs->spec.param[5].high = 1;
	specs->spec.param[5].flags = BRISTOL_ROTARY|BRISTOL_SLIDER;

	specs->spec.param[6].pname = "LP/BP/HP";
	specs->spec.param[6].description = "Low pass or high pass (type-0 only)";
	specs->spec.param[6].type = BRISTOL_INT;
	specs->spec.param[6].low = 0;
	specs->spec.param[6].high = 2;
	specs->spec.param[6].flags = BRISTOL_ROTARY|BRISTOL_SLIDER;

	/*
	 * Now fill in the dco IO specs.
	 */
	specs->spec.io[0].ioname = "input";
	specs->spec.io[0].description = "Filter Input signal";
	specs->spec.io[0].samplerate = samplerate;
	specs->spec.io[0].samplecount = samplecount;
	specs->spec.io[0].flags = BRISTOL_AC|BRISTOL_INPUT;

	specs->spec.io[1].ioname = "mod";
	specs->spec.io[1].description = "Filter Control Signal";
	specs->spec.io[1].samplerate = samplerate;
	specs->spec.io[1].samplecount = samplecount;
	specs->spec.io[1].flags = BRISTOL_DC|BRISTOL_INPUT|BRISTOL_HIDE;

	specs->spec.io[2].ioname = "output";
	specs->spec.io[2].description = "Filter Output Signal";
	specs->spec.io[2].samplerate = samplerate;
	specs->spec.io[2].samplecount = samplecount;
	specs->spec.io[2].flags = BRISTOL_AC|BRISTOL_OUTPUT;

	return(*operator);
}