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
|
/* ========================================
* Spiral2 - Spiral2.h
* Copyright (c) 2016 airwindows, All rights reserved
* ======================================== */
#ifndef __Spiral2_H
#include "Spiral2.h"
#endif
void Spiral2::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames)
{
float* in1 = inputs[0];
float* in2 = inputs[1];
float* out1 = outputs[0];
float* out2 = outputs[1];
double overallscale = 1.0;
overallscale /= 44100.0;
overallscale *= getSampleRate();
double gain = pow(A*2.0,2.0);
double iirAmount = pow(B,3.0)/overallscale;
double presence = C;
double output = D;
double wet = E;
while (--sampleFrames >= 0)
{
long double inputSampleL = *in1;
long double inputSampleR = *in2;
static int noisesourceL = 0;
static int noisesourceR = 850010;
int residue;
double applyresidue;
noisesourceL = noisesourceL % 1700021; noisesourceL++;
residue = noisesourceL * noisesourceL;
residue = residue % 170003; residue *= residue;
residue = residue % 17011; residue *= residue;
residue = residue % 1709; residue *= residue;
residue = residue % 173; residue *= residue;
residue = residue % 17;
applyresidue = residue;
applyresidue *= 0.00000001;
applyresidue *= 0.00000001;
inputSampleL += applyresidue;
if (inputSampleL<1.2e-38 && -inputSampleL<1.2e-38) {
inputSampleL -= applyresidue;
}
noisesourceR = noisesourceR % 1700021; noisesourceR++;
residue = noisesourceR * noisesourceR;
residue = residue % 170003; residue *= residue;
residue = residue % 17011; residue *= residue;
residue = residue % 1709; residue *= residue;
residue = residue % 173; residue *= residue;
residue = residue % 17;
applyresidue = residue;
applyresidue *= 0.00000001;
applyresidue *= 0.00000001;
inputSampleR += applyresidue;
if (inputSampleR<1.2e-38 && -inputSampleR<1.2e-38) {
inputSampleR -= applyresidue;
}
//for live air, we always apply the dither noise. Then, if our result is
//effectively digital black, we'll subtract it aSpiral2. We want a 'air' hiss
long double drySampleL = inputSampleL;
long double drySampleR = inputSampleR;
if (gain != 1.0) {
inputSampleL *= gain;
inputSampleR *= gain;
prevSampleL *= gain;
prevSampleR *= gain;
}
if (flip)
{
iirSampleAL = (iirSampleAL * (1 - iirAmount)) + (inputSampleL * iirAmount);
iirSampleAR = (iirSampleAR * (1 - iirAmount)) + (inputSampleR * iirAmount);
inputSampleL -= iirSampleAL;
inputSampleR -= iirSampleAR;
}
else
{
iirSampleBL = (iirSampleBL * (1 - iirAmount)) + (inputSampleL * iirAmount);
iirSampleBR = (iirSampleBR * (1 - iirAmount)) + (inputSampleR * iirAmount);
inputSampleL -= iirSampleBL;
inputSampleR -= iirSampleBR;
}
//highpass section
long double presenceSampleL = sin(inputSampleL * fabs(prevSampleL)) / ((prevSampleL == 0.0) ?1:fabs(prevSampleL));
long double presenceSampleR = sin(inputSampleR * fabs(prevSampleR)) / ((prevSampleR == 0.0) ?1:fabs(prevSampleR));
//change from first Spiral: delay of one sample on the scaling factor.
inputSampleL = sin(inputSampleL * fabs(inputSampleL)) / ((inputSampleL == 0.0) ?1:fabs(inputSampleL));
inputSampleR = sin(inputSampleR * fabs(inputSampleR)) / ((inputSampleR == 0.0) ?1:fabs(inputSampleR));
if (output < 1.0) {
inputSampleL *= output;
inputSampleR *= output;
presenceSampleL *= output;
presenceSampleR *= output;
}
if (presence > 0.0) {
inputSampleL = (inputSampleL * (1.0-presence)) + (presenceSampleL * presence);
inputSampleR = (inputSampleR * (1.0-presence)) + (presenceSampleR * presence);
}
if (wet < 1.0) {
inputSampleL = (drySampleL * (1.0-wet)) + (inputSampleL * wet);
inputSampleR = (drySampleR * (1.0-wet)) + (inputSampleR * wet);
}
//nice little output stage template: if we have another scale of floating point
//number, we really don't want to meaninglessly multiply that by 1.0.
prevSampleL = drySampleL;
prevSampleR = drySampleR;
flip = !flip;
//noise shaping to 32-bit floating point
float fpTemp = inputSampleL;
fpNShapeL += (inputSampleL-fpTemp);
inputSampleL += fpNShapeL;
//if this confuses you look at the wordlength for fpTemp :)
fpTemp = inputSampleR;
fpNShapeR += (inputSampleR-fpTemp);
inputSampleR += fpNShapeR;
//for deeper space and warmth, we try a non-oscillating noise shaping
//that is kind of ruthless: it will forever retain the rounding errors
//except we'll dial it back a hair at the end of every buffer processed
//end noise shaping on 32 bit output
*out1 = inputSampleL;
*out2 = inputSampleR;
*in1++;
*in2++;
*out1++;
*out2++;
}
fpNShapeL *= 0.999999;
fpNShapeR *= 0.999999;
//we will just delicately dial back the FP noise shaping, not even every sample
//this is a good place to put subtle 'no runaway' calculations, though bear in mind
//that it will be called more often when you use shorter sample buffers in the DAW.
//So, very low latency operation will call these calculations more often.
}
void Spiral2::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames)
{
double* in1 = inputs[0];
double* in2 = inputs[1];
double* out1 = outputs[0];
double* out2 = outputs[1];
double overallscale = 1.0;
overallscale /= 44100.0;
overallscale *= getSampleRate();
double gain = pow(A*2.0,2.0);
double iirAmount = pow(B,3.0)/overallscale;
double presence = C;
double output = D;
double wet = E;
while (--sampleFrames >= 0)
{
long double inputSampleL = *in1;
long double inputSampleR = *in2;
static int noisesourceL = 0;
static int noisesourceR = 850010;
int residue;
double applyresidue;
noisesourceL = noisesourceL % 1700021; noisesourceL++;
residue = noisesourceL * noisesourceL;
residue = residue % 170003; residue *= residue;
residue = residue % 17011; residue *= residue;
residue = residue % 1709; residue *= residue;
residue = residue % 173; residue *= residue;
residue = residue % 17;
applyresidue = residue;
applyresidue *= 0.00000001;
applyresidue *= 0.00000001;
inputSampleL += applyresidue;
if (inputSampleL<1.2e-38 && -inputSampleL<1.2e-38) {
inputSampleL -= applyresidue;
}
noisesourceR = noisesourceR % 1700021; noisesourceR++;
residue = noisesourceR * noisesourceR;
residue = residue % 170003; residue *= residue;
residue = residue % 17011; residue *= residue;
residue = residue % 1709; residue *= residue;
residue = residue % 173; residue *= residue;
residue = residue % 17;
applyresidue = residue;
applyresidue *= 0.00000001;
applyresidue *= 0.00000001;
inputSampleR += applyresidue;
if (inputSampleR<1.2e-38 && -inputSampleR<1.2e-38) {
inputSampleR -= applyresidue;
}
//for live air, we always apply the dither noise. Then, if our result is
//effectively digital black, we'll subtract it aSpiral2. We want a 'air' hiss
long double drySampleL = inputSampleL;
long double drySampleR = inputSampleR;
if (gain != 1.0) {
inputSampleL *= gain;
inputSampleR *= gain;
prevSampleL *= gain;
prevSampleR *= gain;
}
if (flip)
{
iirSampleAL = (iirSampleAL * (1 - iirAmount)) + (inputSampleL * iirAmount);
iirSampleAR = (iirSampleAR * (1 - iirAmount)) + (inputSampleR * iirAmount);
inputSampleL -= iirSampleAL;
inputSampleR -= iirSampleAR;
}
else
{
iirSampleBL = (iirSampleBL * (1 - iirAmount)) + (inputSampleL * iirAmount);
iirSampleBR = (iirSampleBR * (1 - iirAmount)) + (inputSampleR * iirAmount);
inputSampleL -= iirSampleBL;
inputSampleR -= iirSampleBR;
}
//highpass section
long double presenceSampleL = sin(inputSampleL * fabs(prevSampleL)) / ((prevSampleL == 0.0) ?1:fabs(prevSampleL));
long double presenceSampleR = sin(inputSampleR * fabs(prevSampleR)) / ((prevSampleR == 0.0) ?1:fabs(prevSampleR));
//change from first Spiral: delay of one sample on the scaling factor.
inputSampleL = sin(inputSampleL * fabs(inputSampleL)) / ((inputSampleL == 0.0) ?1:fabs(inputSampleL));
inputSampleR = sin(inputSampleR * fabs(inputSampleR)) / ((inputSampleR == 0.0) ?1:fabs(inputSampleR));
if (output < 1.0) {
inputSampleL *= output;
inputSampleR *= output;
presenceSampleL *= output;
presenceSampleR *= output;
}
if (presence > 0.0) {
inputSampleL = (inputSampleL * (1.0-presence)) + (presenceSampleL * presence);
inputSampleR = (inputSampleR * (1.0-presence)) + (presenceSampleR * presence);
}
if (wet < 1.0) {
inputSampleL = (drySampleL * (1.0-wet)) + (inputSampleL * wet);
inputSampleR = (drySampleR * (1.0-wet)) + (inputSampleR * wet);
}
//nice little output stage template: if we have another scale of floating point
//number, we really don't want to meaninglessly multiply that by 1.0.
prevSampleL = drySampleL;
prevSampleR = drySampleR;
flip = !flip;
//noise shaping to 64-bit floating point
double fpTemp = inputSampleL;
fpNShapeL += (inputSampleL-fpTemp);
inputSampleL += fpNShapeL;
//if this confuses you look at the wordlength for fpTemp :)
fpTemp = inputSampleR;
fpNShapeR += (inputSampleR-fpTemp);
inputSampleR += fpNShapeR;
//for deeper space and warmth, we try a non-oscillating noise shaping
//that is kind of ruthless: it will forever retain the rounding errors
//except we'll dial it back a hair at the end of every buffer processed
//end noise shaping on 64 bit output
*out1 = inputSampleL;
*out2 = inputSampleR;
*in1++;
*in2++;
*out1++;
*out2++;
}
fpNShapeL *= 0.999999;
fpNShapeR *= 0.999999;
//we will just delicately dial back the FP noise shaping, not even every sample
//this is a good place to put subtle 'no runaway' calculations, though bear in mind
//that it will be called more often when you use shorter sample buffers in the DAW.
//So, very low latency operation will call these calculations more often.
}
|
