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/* ========================================
* Crystal - Crystal.h
* Copyright (c) 2016 airwindows, All rights reserved
* ======================================== */
#ifndef __Crystal_H
#include "Crystal.h"
#endif
void Crystal::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames)
{
float* in1 = inputs[0];
float* in2 = inputs[1];
float* out1 = outputs[0];
float* out2 = outputs[1];
double threshold = A;
double hardness;
double breakup = (1.0-(threshold/2.0))*3.14159265358979;
double bridgerectifier;
double sqdrive = B;
if (sqdrive > 1.0) sqdrive *= sqdrive;
sqdrive = sqrt(sqdrive);
double indrive = C*3.0;
if (indrive > 1.0) indrive *= indrive;
indrive *= (1.0-(0.1695*sqdrive));
//no gain loss of convolution for APIcolypse
//calibrate this to match noise level with character at 1.0
//you get for instance 0.819 and 1.0-0.819 is 0.181
double randy;
double outlevel = D;
if (threshold < 1) hardness = 1.0 / (1.0-threshold);
else hardness = 999999999999999999999.0;
//set up hardness to exactly fill gap between threshold and 0db
//if threshold is literally 1 then hardness is infinite, so we make it very big
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 again. We want a 'air' hiss
inputSampleL *= indrive;
inputSampleR *= indrive;
//calibrated to match gain through convolution and -0.3 correction
if (sqdrive > 0.0){
bL[23] = bL[22]; bL[22] = bL[21]; bL[21] = bL[20]; bL[20] = bL[19]; bL[19] = bL[18]; bL[18] = bL[17]; bL[17] = bL[16]; bL[16] = bL[15];
bL[15] = bL[14]; bL[14] = bL[13]; bL[13] = bL[12]; bL[12] = bL[11]; bL[11] = bL[10]; bL[10] = bL[9]; bL[9] = bL[8]; bL[8] = bL[7];
bL[7] = bL[6]; bL[6] = bL[5]; bL[5] = bL[4]; bL[4] = bL[3]; bL[3] = bL[2]; bL[2] = bL[1]; bL[1] = bL[0]; bL[0] = inputSampleL * sqdrive;
inputSampleL += (bL[1] * (0.38856694371895023 + (0.14001177830115491*fabs(bL[1]))));
inputSampleL -= (bL[2] * (0.17469488984546111 + (0.05204541941091459*fabs(bL[2]))));
inputSampleL += (bL[3] * (0.11643521461774288 - (0.01193121216518472*fabs(bL[3]))));
inputSampleL -= (bL[4] * (0.08874416268268183 - (0.05867502375036486*fabs(bL[4]))));
inputSampleL += (bL[5] * (0.07222999223073785 - (0.08519974113692971*fabs(bL[5]))));
inputSampleL -= (bL[6] * (0.06103207678880003 - (0.09230674983449150*fabs(bL[6]))));
inputSampleL += (bL[7] * (0.05277389277465404 - (0.08487342372497046*fabs(bL[7]))));
inputSampleL -= (bL[8] * (0.04631144388636078 - (0.06976851898821038*fabs(bL[8]))));
inputSampleL += (bL[9] * (0.04102721072495113 - (0.05337974329110802*fabs(bL[9]))));
inputSampleL -= (bL[10] * (0.03656047655964371 - (0.03990914278458497*fabs(bL[10]))));
inputSampleL += (bL[11] * (0.03268677450573373 - (0.03090433934018759*fabs(bL[11]))));
inputSampleL -= (bL[12] * (0.02926012259262895 - (0.02585223214266682*fabs(bL[12]))));
inputSampleL += (bL[13] * (0.02618257163789973 - (0.02326667039588473*fabs(bL[13]))));
inputSampleL -= (bL[14] * (0.02338568277879992 - (0.02167067760829789*fabs(bL[14]))));
inputSampleL += (bL[15] * (0.02082142324645262 - (0.02013392273267951*fabs(bL[15]))));
inputSampleL -= (bL[16] * (0.01845525966656259 - (0.01833038930966512*fabs(bL[16]))));
inputSampleL += (bL[17] * (0.01626113504980445 - (0.01631893218593511*fabs(bL[17]))));
inputSampleL -= (bL[18] * (0.01422084088669267 - (0.01427828125219885*fabs(bL[18]))));
inputSampleL += (bL[19] * (0.01231993595709338 - (0.01233991521342998*fabs(bL[19]))));
inputSampleL -= (bL[20] * (0.01054774630451013 - (0.01054774630542346*fabs(bL[20]))));
inputSampleL += (bL[21] * (0.00889548162355088 - (0.00889548162263755*fabs(bL[21]))));
inputSampleL -= (bL[22] * (0.00735749099304526 - (0.00735749099395860*fabs(bL[22]))));
inputSampleL += (bL[23] * (0.00592812350468000 - (0.00592812350376666*fabs(bL[23]))));
} //the Character plugins as individual processors did this. BussColors applies an averaging factor to produce
// more of a consistent variation between soft and loud convolutions. For years I thought this code was a
//mistake and did nothing, but in fact what it's doing is producing slightly different curves for every single
//convolution kernel location: this will be true of the Character individual plugins as well.
//calibrated to match gain through convolution and -0.3 correction
if (sqdrive > 0.0){
bR[23] = bR[22]; bR[22] = bR[21]; bR[21] = bR[20]; bR[20] = bR[19]; bR[19] = bR[18]; bR[18] = bR[17]; bR[17] = bR[16]; bR[16] = bR[15];
bR[15] = bR[14]; bR[14] = bR[13]; bR[13] = bR[12]; bR[12] = bR[11]; bR[11] = bR[10]; bR[10] = bR[9]; bR[9] = bR[8]; bR[8] = bR[7];
bR[7] = bR[6]; bR[6] = bR[5]; bR[5] = bR[4]; bR[4] = bR[3]; bR[3] = bR[2]; bR[2] = bR[1]; bR[1] = bR[0]; bR[0] = inputSampleR * sqdrive;
inputSampleR += (bR[1] * (0.38856694371895023 + (0.14001177830115491*fabs(bR[1]))));
inputSampleR -= (bR[2] * (0.17469488984546111 + (0.05204541941091459*fabs(bR[2]))));
inputSampleR += (bR[3] * (0.11643521461774288 - (0.01193121216518472*fabs(bR[3]))));
inputSampleR -= (bR[4] * (0.08874416268268183 - (0.05867502375036486*fabs(bR[4]))));
inputSampleR += (bR[5] * (0.07222999223073785 - (0.08519974113692971*fabs(bR[5]))));
inputSampleR -= (bR[6] * (0.06103207678880003 - (0.09230674983449150*fabs(bR[6]))));
inputSampleR += (bR[7] * (0.05277389277465404 - (0.08487342372497046*fabs(bR[7]))));
inputSampleR -= (bR[8] * (0.04631144388636078 - (0.06976851898821038*fabs(bR[8]))));
inputSampleR += (bR[9] * (0.04102721072495113 - (0.05337974329110802*fabs(bR[9]))));
inputSampleR -= (bR[10] * (0.03656047655964371 - (0.03990914278458497*fabs(bR[10]))));
inputSampleR += (bR[11] * (0.03268677450573373 - (0.03090433934018759*fabs(bR[11]))));
inputSampleR -= (bR[12] * (0.02926012259262895 - (0.02585223214266682*fabs(bR[12]))));
inputSampleR += (bR[13] * (0.02618257163789973 - (0.02326667039588473*fabs(bR[13]))));
inputSampleR -= (bR[14] * (0.02338568277879992 - (0.02167067760829789*fabs(bR[14]))));
inputSampleR += (bR[15] * (0.02082142324645262 - (0.02013392273267951*fabs(bR[15]))));
inputSampleR -= (bR[16] * (0.01845525966656259 - (0.01833038930966512*fabs(bR[16]))));
inputSampleR += (bR[17] * (0.01626113504980445 - (0.01631893218593511*fabs(bR[17]))));
inputSampleR -= (bR[18] * (0.01422084088669267 - (0.01427828125219885*fabs(bR[18]))));
inputSampleR += (bR[19] * (0.01231993595709338 - (0.01233991521342998*fabs(bR[19]))));
inputSampleR -= (bR[20] * (0.01054774630451013 - (0.01054774630542346*fabs(bR[20]))));
inputSampleR += (bR[21] * (0.00889548162355088 - (0.00889548162263755*fabs(bR[21]))));
inputSampleR -= (bR[22] * (0.00735749099304526 - (0.00735749099395860*fabs(bR[22]))));
inputSampleR += (bR[23] * (0.00592812350468000 - (0.00592812350376666*fabs(bR[23]))));
} //the Character plugins as individual processors did this. BussColors applies an averaging factor to produce
// more of a consistent variation between soft and loud convolutions. For years I thought this code was a
//mistake and did nothing, but in fact what it's doing is producing slightly different curves for every single
//convolution kernel location: this will be true of the Character individual plugins as well.
if (fabs(inputSampleL) > threshold)
{
bridgerectifier = (fabs(inputSampleL)-threshold)*hardness;
//skip flat area if any, scale to distortion limit
if (bridgerectifier > breakup) bridgerectifier = breakup;
//max value for sine function, 'breakup' modeling for trashed console tone
//more hardness = more solidness behind breakup modeling. more softness, more 'grunge' and sag
bridgerectifier = sin(bridgerectifier)/hardness;
//do the sine factor, scale back to proper amount
if (inputSampleL > 0) inputSampleL = bridgerectifier+threshold;
else inputSampleL = -(bridgerectifier+threshold);
} //otherwise we leave it untouched by the overdrive stuff
//this is the notorious New Channel Density algorithm. It's much less popular than the original Density,
//because it introduces a point where the saturation 'curve' changes from straight to curved.
//People don't like these discontinuities, but you can use them for effect or to grit up the sound.
if (fabs(inputSampleR) > threshold)
{
bridgerectifier = (fabs(inputSampleR)-threshold)*hardness;
//skip flat area if any, scale to distortion limit
if (bridgerectifier > breakup) bridgerectifier = breakup;
//max value for sine function, 'breakup' modeling for trashed console tone
//more hardness = more solidness behind breakup modeling. more softness, more 'grunge' and sag
bridgerectifier = sin(bridgerectifier)/hardness;
//do the sine factor, scale back to proper amount
if (inputSampleR > 0) inputSampleR = bridgerectifier+threshold;
else inputSampleR = -(bridgerectifier+threshold);
} //otherwise we leave it untouched by the overdrive stuff
//this is the notorious New Channel Density algorithm. It's much less popular than the original Density,
//because it introduces a point where the saturation 'curve' changes from straight to curved.
//People don't like these discontinuities, but you can use them for effect or to grit up the sound.
randy = ((rand()/(double)RAND_MAX)*0.022);
bridgerectifier = ((inputSampleL*(1-randy))+(lastSampleL*randy)) * outlevel;
lastSampleL = inputSampleL;
inputSampleL = bridgerectifier; //applies a tiny 'fuzz' to highs: from original Crystal.
randy = ((rand()/(double)RAND_MAX)*0.022);
bridgerectifier = ((inputSampleR*(1-randy))+(lastSampleR*randy)) * outlevel;
lastSampleR = inputSampleR;
inputSampleR = bridgerectifier; //applies a tiny 'fuzz' to highs: from original Crystal.
//This is akin to the old Chrome Oxide plugin, applying a fuzz to only the slews. The noise only appears
//when current and old samples are different from each other, otherwise you can't tell it's there.
//This is not only during silence but the tops of low frequency waves: it scales down to affect lows more gently.
//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 Crystal::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames)
{
double* in1 = inputs[0];
double* in2 = inputs[1];
double* out1 = outputs[0];
double* out2 = outputs[1];
double threshold = A;
double hardness;
double breakup = (1.0-(threshold/2.0))*3.14159265358979;
double bridgerectifier;
double sqdrive = B;
if (sqdrive > 1.0) sqdrive *= sqdrive;
sqdrive = sqrt(sqdrive);
double indrive = C*3.0;
if (indrive > 1.0) indrive *= indrive;
indrive *= (1.0-(0.1695*sqdrive));
//no gain loss of convolution for APIcolypse
//calibrate this to match noise level with character at 1.0
//you get for instance 0.819 and 1.0-0.819 is 0.181
double randy;
double outlevel = D;
if (threshold < 1) hardness = 1.0 / (1.0-threshold);
else hardness = 999999999999999999999.0;
//set up hardness to exactly fill gap between threshold and 0db
//if threshold is literally 1 then hardness is infinite, so we make it very big
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 again. We want a 'air' hiss
inputSampleL *= indrive;
inputSampleR *= indrive;
//calibrated to match gain through convolution and -0.3 correction
if (sqdrive > 0.0){
bL[23] = bL[22]; bL[22] = bL[21]; bL[21] = bL[20]; bL[20] = bL[19]; bL[19] = bL[18]; bL[18] = bL[17]; bL[17] = bL[16]; bL[16] = bL[15];
bL[15] = bL[14]; bL[14] = bL[13]; bL[13] = bL[12]; bL[12] = bL[11]; bL[11] = bL[10]; bL[10] = bL[9]; bL[9] = bL[8]; bL[8] = bL[7];
bL[7] = bL[6]; bL[6] = bL[5]; bL[5] = bL[4]; bL[4] = bL[3]; bL[3] = bL[2]; bL[2] = bL[1]; bL[1] = bL[0]; bL[0] = inputSampleL * sqdrive;
inputSampleL += (bL[1] * (0.38856694371895023 + (0.14001177830115491*fabs(bL[1]))));
inputSampleL -= (bL[2] * (0.17469488984546111 + (0.05204541941091459*fabs(bL[2]))));
inputSampleL += (bL[3] * (0.11643521461774288 - (0.01193121216518472*fabs(bL[3]))));
inputSampleL -= (bL[4] * (0.08874416268268183 - (0.05867502375036486*fabs(bL[4]))));
inputSampleL += (bL[5] * (0.07222999223073785 - (0.08519974113692971*fabs(bL[5]))));
inputSampleL -= (bL[6] * (0.06103207678880003 - (0.09230674983449150*fabs(bL[6]))));
inputSampleL += (bL[7] * (0.05277389277465404 - (0.08487342372497046*fabs(bL[7]))));
inputSampleL -= (bL[8] * (0.04631144388636078 - (0.06976851898821038*fabs(bL[8]))));
inputSampleL += (bL[9] * (0.04102721072495113 - (0.05337974329110802*fabs(bL[9]))));
inputSampleL -= (bL[10] * (0.03656047655964371 - (0.03990914278458497*fabs(bL[10]))));
inputSampleL += (bL[11] * (0.03268677450573373 - (0.03090433934018759*fabs(bL[11]))));
inputSampleL -= (bL[12] * (0.02926012259262895 - (0.02585223214266682*fabs(bL[12]))));
inputSampleL += (bL[13] * (0.02618257163789973 - (0.02326667039588473*fabs(bL[13]))));
inputSampleL -= (bL[14] * (0.02338568277879992 - (0.02167067760829789*fabs(bL[14]))));
inputSampleL += (bL[15] * (0.02082142324645262 - (0.02013392273267951*fabs(bL[15]))));
inputSampleL -= (bL[16] * (0.01845525966656259 - (0.01833038930966512*fabs(bL[16]))));
inputSampleL += (bL[17] * (0.01626113504980445 - (0.01631893218593511*fabs(bL[17]))));
inputSampleL -= (bL[18] * (0.01422084088669267 - (0.01427828125219885*fabs(bL[18]))));
inputSampleL += (bL[19] * (0.01231993595709338 - (0.01233991521342998*fabs(bL[19]))));
inputSampleL -= (bL[20] * (0.01054774630451013 - (0.01054774630542346*fabs(bL[20]))));
inputSampleL += (bL[21] * (0.00889548162355088 - (0.00889548162263755*fabs(bL[21]))));
inputSampleL -= (bL[22] * (0.00735749099304526 - (0.00735749099395860*fabs(bL[22]))));
inputSampleL += (bL[23] * (0.00592812350468000 - (0.00592812350376666*fabs(bL[23]))));
} //the Character plugins as individual processors did this. BussColors applies an averaging factor to produce
// more of a consistent variation between soft and loud convolutions. For years I thought this code was a
//mistake and did nothing, but in fact what it's doing is producing slightly different curves for every single
//convolution kernel location: this will be true of the Character individual plugins as well.
//calibrated to match gain through convolution and -0.3 correction
if (sqdrive > 0.0){
bR[23] = bR[22]; bR[22] = bR[21]; bR[21] = bR[20]; bR[20] = bR[19]; bR[19] = bR[18]; bR[18] = bR[17]; bR[17] = bR[16]; bR[16] = bR[15];
bR[15] = bR[14]; bR[14] = bR[13]; bR[13] = bR[12]; bR[12] = bR[11]; bR[11] = bR[10]; bR[10] = bR[9]; bR[9] = bR[8]; bR[8] = bR[7];
bR[7] = bR[6]; bR[6] = bR[5]; bR[5] = bR[4]; bR[4] = bR[3]; bR[3] = bR[2]; bR[2] = bR[1]; bR[1] = bR[0]; bR[0] = inputSampleR * sqdrive;
inputSampleR += (bR[1] * (0.38856694371895023 + (0.14001177830115491*fabs(bR[1]))));
inputSampleR -= (bR[2] * (0.17469488984546111 + (0.05204541941091459*fabs(bR[2]))));
inputSampleR += (bR[3] * (0.11643521461774288 - (0.01193121216518472*fabs(bR[3]))));
inputSampleR -= (bR[4] * (0.08874416268268183 - (0.05867502375036486*fabs(bR[4]))));
inputSampleR += (bR[5] * (0.07222999223073785 - (0.08519974113692971*fabs(bR[5]))));
inputSampleR -= (bR[6] * (0.06103207678880003 - (0.09230674983449150*fabs(bR[6]))));
inputSampleR += (bR[7] * (0.05277389277465404 - (0.08487342372497046*fabs(bR[7]))));
inputSampleR -= (bR[8] * (0.04631144388636078 - (0.06976851898821038*fabs(bR[8]))));
inputSampleR += (bR[9] * (0.04102721072495113 - (0.05337974329110802*fabs(bR[9]))));
inputSampleR -= (bR[10] * (0.03656047655964371 - (0.03990914278458497*fabs(bR[10]))));
inputSampleR += (bR[11] * (0.03268677450573373 - (0.03090433934018759*fabs(bR[11]))));
inputSampleR -= (bR[12] * (0.02926012259262895 - (0.02585223214266682*fabs(bR[12]))));
inputSampleR += (bR[13] * (0.02618257163789973 - (0.02326667039588473*fabs(bR[13]))));
inputSampleR -= (bR[14] * (0.02338568277879992 - (0.02167067760829789*fabs(bR[14]))));
inputSampleR += (bR[15] * (0.02082142324645262 - (0.02013392273267951*fabs(bR[15]))));
inputSampleR -= (bR[16] * (0.01845525966656259 - (0.01833038930966512*fabs(bR[16]))));
inputSampleR += (bR[17] * (0.01626113504980445 - (0.01631893218593511*fabs(bR[17]))));
inputSampleR -= (bR[18] * (0.01422084088669267 - (0.01427828125219885*fabs(bR[18]))));
inputSampleR += (bR[19] * (0.01231993595709338 - (0.01233991521342998*fabs(bR[19]))));
inputSampleR -= (bR[20] * (0.01054774630451013 - (0.01054774630542346*fabs(bR[20]))));
inputSampleR += (bR[21] * (0.00889548162355088 - (0.00889548162263755*fabs(bR[21]))));
inputSampleR -= (bR[22] * (0.00735749099304526 - (0.00735749099395860*fabs(bR[22]))));
inputSampleR += (bR[23] * (0.00592812350468000 - (0.00592812350376666*fabs(bR[23]))));
} //the Character plugins as individual processors did this. BussColors applies an averaging factor to produce
// more of a consistent variation between soft and loud convolutions. For years I thought this code was a
//mistake and did nothing, but in fact what it's doing is producing slightly different curves for every single
//convolution kernel location: this will be true of the Character individual plugins as well.
if (fabs(inputSampleL) > threshold)
{
bridgerectifier = (fabs(inputSampleL)-threshold)*hardness;
//skip flat area if any, scale to distortion limit
if (bridgerectifier > breakup) bridgerectifier = breakup;
//max value for sine function, 'breakup' modeling for trashed console tone
//more hardness = more solidness behind breakup modeling. more softness, more 'grunge' and sag
bridgerectifier = sin(bridgerectifier)/hardness;
//do the sine factor, scale back to proper amount
if (inputSampleL > 0) inputSampleL = bridgerectifier+threshold;
else inputSampleL = -(bridgerectifier+threshold);
} //otherwise we leave it untouched by the overdrive stuff
//this is the notorious New Channel Density algorithm. It's much less popular than the original Density,
//because it introduces a point where the saturation 'curve' changes from straight to curved.
//People don't like these discontinuities, but you can use them for effect or to grit up the sound.
if (fabs(inputSampleR) > threshold)
{
bridgerectifier = (fabs(inputSampleR)-threshold)*hardness;
//skip flat area if any, scale to distortion limit
if (bridgerectifier > breakup) bridgerectifier = breakup;
//max value for sine function, 'breakup' modeling for trashed console tone
//more hardness = more solidness behind breakup modeling. more softness, more 'grunge' and sag
bridgerectifier = sin(bridgerectifier)/hardness;
//do the sine factor, scale back to proper amount
if (inputSampleR > 0) inputSampleR = bridgerectifier+threshold;
else inputSampleR = -(bridgerectifier+threshold);
} //otherwise we leave it untouched by the overdrive stuff
//this is the notorious New Channel Density algorithm. It's much less popular than the original Density,
//because it introduces a point where the saturation 'curve' changes from straight to curved.
//People don't like these discontinuities, but you can use them for effect or to grit up the sound.
randy = ((rand()/(double)RAND_MAX)*0.022);
bridgerectifier = ((inputSampleL*(1-randy))+(lastSampleL*randy)) * outlevel;
lastSampleL = inputSampleL;
inputSampleL = bridgerectifier; //applies a tiny 'fuzz' to highs: from original Crystal.
randy = ((rand()/(double)RAND_MAX)*0.022);
bridgerectifier = ((inputSampleR*(1-randy))+(lastSampleR*randy)) * outlevel;
lastSampleR = inputSampleR;
inputSampleR = bridgerectifier; //applies a tiny 'fuzz' to highs: from original Crystal.
//This is akin to the old Chrome Oxide plugin, applying a fuzz to only the slews. The noise only appears
//when current and old samples are different from each other, otherwise you can't tell it's there.
//This is not only during silence but the tops of low frequency waves: it scales down to affect lows more gently.
//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.
}
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