/* ========================================
* DrumSlam - DrumSlam.h
* Copyright (c) 2016 airwindows, All rights reserved
* ======================================== */
#ifndef __DrumSlam_H
#include "DrumSlam.h"
#endif
void DrumSlam::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 iirAmountL = 0.0819;
iirAmountL /= overallscale;
double iirAmountH = 0.377933067;
iirAmountH /= overallscale;
double drive = (A*3.0)+1.0;
double out = B;
double wet = C;
double dry = 1.0 - wet;
while (--sampleFrames >= 0)
{
long double inputSampleL = *in1;
long double inputSampleR = *in2;
if (inputSampleL<1.2e-38 && -inputSampleL<1.2e-38) {
static int noisesource = 0;
//this declares a variable before anything else is compiled. It won't keep assigning
//it to 0 for every sample, it's as if the declaration doesn't exist in this context,
//but it lets me add this denormalization fix in a single place rather than updating
//it in three different locations. The variable isn't thread-safe but this is only
//a random seed and we can share it with whatever.
noisesource = noisesource % 1700021; noisesource++;
int residue = noisesource * noisesource;
residue = residue % 170003; residue *= residue;
residue = residue % 17011; residue *= residue;
residue = residue % 1709; residue *= residue;
residue = residue % 173; residue *= residue;
residue = residue % 17;
double applyresidue = residue;
applyresidue *= 0.00000001;
applyresidue *= 0.00000001;
inputSampleL = applyresidue;
}
if (inputSampleR<1.2e-38 && -inputSampleR<1.2e-38) {
static int noisesource = 0;
noisesource = noisesource % 1700021; noisesource++;
int residue = noisesource * noisesource;
residue = residue % 170003; residue *= residue;
residue = residue % 17011; residue *= residue;
residue = residue % 1709; residue *= residue;
residue = residue % 173; residue *= residue;
residue = residue % 17;
double applyresidue = residue;
applyresidue *= 0.00000001;
applyresidue *= 0.00000001;
inputSampleR = applyresidue;
//this denormalization routine produces a white noise at -300 dB which the noise
//shaping will interact with to produce a bipolar output, but the noise is actually
//all positive. That should stop any variables from going denormal, and the routine
//only kicks in if digital black is input. As a final touch, if you save to 24-bit
//the silence will return to being digital black again.
}
long double drySampleL = inputSampleL;
long double drySampleR = inputSampleR;
long double lowSampleL;
long double lowSampleR;
long double midSampleL;
long double midSampleR;
long double highSampleL;
long double highSampleR;
inputSampleL *= drive;
inputSampleR *= drive;
if (fpFlip)
{
iirSampleAL = (iirSampleAL * (1 - iirAmountL)) + (inputSampleL * iirAmountL);
iirSampleBL = (iirSampleBL * (1 - iirAmountL)) + (iirSampleAL * iirAmountL);
lowSampleL = iirSampleBL;
iirSampleAR = (iirSampleAR * (1 - iirAmountL)) + (inputSampleR * iirAmountL);
iirSampleBR = (iirSampleBR * (1 - iirAmountL)) + (iirSampleAR * iirAmountL);
lowSampleR = iirSampleBR;
iirSampleEL = (iirSampleEL * (1 - iirAmountH)) + (inputSampleL * iirAmountH);
iirSampleFL = (iirSampleFL * (1 - iirAmountH)) + (iirSampleEL * iirAmountH);
midSampleL = iirSampleFL - iirSampleBL;
iirSampleER = (iirSampleER * (1 - iirAmountH)) + (inputSampleR * iirAmountH);
iirSampleFR = (iirSampleFR * (1 - iirAmountH)) + (iirSampleER * iirAmountH);
midSampleR = iirSampleFR - iirSampleBR;
highSampleL = inputSampleL - iirSampleFL;
highSampleR = inputSampleR - iirSampleFR;
}
else
{
iirSampleCL = (iirSampleCL * (1 - iirAmountL)) + (inputSampleL * iirAmountL);
iirSampleDL = (iirSampleDL * (1 - iirAmountL)) + (iirSampleCL * iirAmountL);
lowSampleL = iirSampleDL;
iirSampleCR = (iirSampleCR * (1 - iirAmountL)) + (inputSampleR * iirAmountL);
iirSampleDR = (iirSampleDR * (1 - iirAmountL)) + (iirSampleCR * iirAmountL);
lowSampleR = iirSampleDR;
iirSampleGL = (iirSampleGL * (1 - iirAmountH)) + (inputSampleL * iirAmountH);
iirSampleHL = (iirSampleHL * (1 - iirAmountH)) + (iirSampleGL * iirAmountH);
midSampleL = iirSampleHL - iirSampleDL;
iirSampleGR = (iirSampleGR * (1 - iirAmountH)) + (inputSampleR * iirAmountH);
iirSampleHR = (iirSampleHR * (1 - iirAmountH)) + (iirSampleGR * iirAmountH);
midSampleR = iirSampleHR - iirSampleDR;
highSampleL = inputSampleL - iirSampleHL;
highSampleR = inputSampleR - iirSampleHR;
}
fpFlip = !fpFlip;
//generate the tone bands we're using
if (lowSampleL > 1.0) {lowSampleL = 1.0;}
if (lowSampleL < -1.0) {lowSampleL = -1.0;}
if (lowSampleR > 1.0) {lowSampleR = 1.0;}
if (lowSampleR < -1.0) {lowSampleR = -1.0;}
lowSampleL -= (lowSampleL * (fabs(lowSampleL) * 0.448) * (fabs(lowSampleL) * 0.448) );
lowSampleR -= (lowSampleR * (fabs(lowSampleR) * 0.448) * (fabs(lowSampleR) * 0.448) );
lowSampleL *= drive;
lowSampleR *= drive;
if (highSampleL > 1.0) {highSampleL = 1.0;}
if (highSampleL < -1.0) {highSampleL = -1.0;}
if (highSampleR > 1.0) {highSampleR = 1.0;}
if (highSampleR < -1.0) {highSampleR = -1.0;}
highSampleL -= (highSampleL * (fabs(highSampleL) * 0.599) * (fabs(highSampleL) * 0.599) );
highSampleR -= (highSampleR * (fabs(highSampleR) * 0.599) * (fabs(highSampleR) * 0.599) );
highSampleL *= drive;
highSampleR *= drive;
midSampleL = midSampleL * drive;
midSampleR = midSampleR * drive;
long double skew = (midSampleL - lastSampleL);
lastSampleL = midSampleL;
//skew will be direction/angle
long double bridgerectifier = fabs(skew);
if (bridgerectifier > 3.1415926) bridgerectifier = 3.1415926;
//for skew we want it to go to zero effect again, so we use full range of the sine
bridgerectifier = sin(bridgerectifier);
if (skew > 0) skew = bridgerectifier*3.1415926;
else skew = -bridgerectifier*3.1415926;
//skew is now sined and clamped and then re-amplified again
skew *= midSampleL;
//cools off sparkliness and crossover distortion
skew *= 1.557079633;
//crank up the gain on this so we can make it sing
bridgerectifier = fabs(midSampleL);
bridgerectifier += skew;
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
bridgerectifier = sin(bridgerectifier);
bridgerectifier *= drive;
bridgerectifier += skew;
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
bridgerectifier = sin(bridgerectifier);
if (midSampleL > 0)
{
midSampleL = bridgerectifier;
}
else
{
midSampleL = -bridgerectifier;
}
//blend according to positive and negative controls, left
skew = (midSampleR - lastSampleR);
lastSampleR = midSampleR;
//skew will be direction/angle
bridgerectifier = fabs(skew);
if (bridgerectifier > 3.1415926) bridgerectifier = 3.1415926;
//for skew we want it to go to zero effect again, so we use full range of the sine
bridgerectifier = sin(bridgerectifier);
if (skew > 0) skew = bridgerectifier*3.1415926;
else skew = -bridgerectifier*3.1415926;
//skew is now sined and clamped and then re-amplified again
skew *= midSampleR;
//cools off sparkliness and crossover distortion
skew *= 1.557079633;
//crank up the gain on this so we can make it sing
bridgerectifier = fabs(midSampleR);
bridgerectifier += skew;
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
bridgerectifier = sin(bridgerectifier);
bridgerectifier *= drive;
bridgerectifier += skew;
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
bridgerectifier = sin(bridgerectifier);
if (midSampleR > 0)
{
midSampleR = bridgerectifier;
}
else
{
midSampleR = -bridgerectifier;
}
//blend according to positive and negative controls, right
inputSampleL = ((lowSampleL + midSampleL + highSampleL)/drive)*out;
inputSampleR = ((lowSampleR + midSampleR + highSampleR)/drive)*out;
if (wet !=1.0) {
inputSampleL = (inputSampleL * wet) + (drySampleL * dry);
inputSampleR = (inputSampleR * wet) + (drySampleR * dry);
}
//stereo 32 bit dither, made small and tidy.
int expon; frexpf((float)inputSampleL, &expon);
long double dither = (rand()/(RAND_MAX*7.737125245533627e+25))*pow(2,expon+62);
inputSampleL += (dither-fpNShapeL); fpNShapeL = dither;
frexpf((float)inputSampleR, &expon);
dither = (rand()/(RAND_MAX*7.737125245533627e+25))*pow(2,expon+62);
inputSampleR += (dither-fpNShapeR); fpNShapeR = dither;
//end 32 bit dither
*out1 = inputSampleL;
*out2 = inputSampleR;
*in1++;
*in2++;
*out1++;
*out2++;
}
}
void DrumSlam::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 iirAmountL = 0.0819;
iirAmountL /= overallscale;
double iirAmountH = 0.377933067;
iirAmountH /= overallscale;
double drive = (A*3.0)+1.0;
double out = B;
double wet = C;
double dry = 1.0 - wet;
while (--sampleFrames >= 0)
{
long double inputSampleL = *in1;
long double inputSampleR = *in2;
if (inputSampleL<1.2e-38 && -inputSampleL<1.2e-38) {
static int noisesource = 0;
//this declares a variable before anything else is compiled. It won't keep assigning
//it to 0 for every sample, it's as if the declaration doesn't exist in this context,
//but it lets me add this denormalization fix in a single place rather than updating
//it in three different locations. The variable isn't thread-safe but this is only
//a random seed and we can share it with whatever.
noisesource = noisesource % 1700021; noisesource++;
int residue = noisesource * noisesource;
residue = residue % 170003; residue *= residue;
residue = residue % 17011; residue *= residue;
residue = residue % 1709; residue *= residue;
residue = residue % 173; residue *= residue;
residue = residue % 17;
double applyresidue = residue;
applyresidue *= 0.00000001;
applyresidue *= 0.00000001;
inputSampleL = applyresidue;
}
if (inputSampleR<1.2e-38 && -inputSampleR<1.2e-38) {
static int noisesource = 0;
noisesource = noisesource % 1700021; noisesource++;
int residue = noisesource * noisesource;
residue = residue % 170003; residue *= residue;
residue = residue % 17011; residue *= residue;
residue = residue % 1709; residue *= residue;
residue = residue % 173; residue *= residue;
residue = residue % 17;
double applyresidue = residue;
applyresidue *= 0.00000001;
applyresidue *= 0.00000001;
inputSampleR = applyresidue;
//this denormalization routine produces a white noise at -300 dB which the noise
//shaping will interact with to produce a bipolar output, but the noise is actually
//all positive. That should stop any variables from going denormal, and the routine
//only kicks in if digital black is input. As a final touch, if you save to 24-bit
//the silence will return to being digital black again.
}
long double drySampleL = inputSampleL;
long double drySampleR = inputSampleR;
long double lowSampleL;
long double lowSampleR;
long double midSampleL;
long double midSampleR;
long double highSampleL;
long double highSampleR;
inputSampleL *= drive;
inputSampleR *= drive;
if (fpFlip)
{
iirSampleAL = (iirSampleAL * (1 - iirAmountL)) + (inputSampleL * iirAmountL);
iirSampleBL = (iirSampleBL * (1 - iirAmountL)) + (iirSampleAL * iirAmountL);
lowSampleL = iirSampleBL;
iirSampleAR = (iirSampleAR * (1 - iirAmountL)) + (inputSampleR * iirAmountL);
iirSampleBR = (iirSampleBR * (1 - iirAmountL)) + (iirSampleAR * iirAmountL);
lowSampleR = iirSampleBR;
iirSampleEL = (iirSampleEL * (1 - iirAmountH)) + (inputSampleL * iirAmountH);
iirSampleFL = (iirSampleFL * (1 - iirAmountH)) + (iirSampleEL * iirAmountH);
midSampleL = iirSampleFL - iirSampleBL;
iirSampleER = (iirSampleER * (1 - iirAmountH)) + (inputSampleR * iirAmountH);
iirSampleFR = (iirSampleFR * (1 - iirAmountH)) + (iirSampleER * iirAmountH);
midSampleR = iirSampleFR - iirSampleBR;
highSampleL = inputSampleL - iirSampleFL;
highSampleR = inputSampleR - iirSampleFR;
}
else
{
iirSampleCL = (iirSampleCL * (1 - iirAmountL)) + (inputSampleL * iirAmountL);
iirSampleDL = (iirSampleDL * (1 - iirAmountL)) + (iirSampleCL * iirAmountL);
lowSampleL = iirSampleDL;
iirSampleCR = (iirSampleCR * (1 - iirAmountL)) + (inputSampleR * iirAmountL);
iirSampleDR = (iirSampleDR * (1 - iirAmountL)) + (iirSampleCR * iirAmountL);
lowSampleR = iirSampleDR;
iirSampleGL = (iirSampleGL * (1 - iirAmountH)) + (inputSampleL * iirAmountH);
iirSampleHL = (iirSampleHL * (1 - iirAmountH)) + (iirSampleGL * iirAmountH);
midSampleL = iirSampleHL - iirSampleDL;
iirSampleGR = (iirSampleGR * (1 - iirAmountH)) + (inputSampleR * iirAmountH);
iirSampleHR = (iirSampleHR * (1 - iirAmountH)) + (iirSampleGR * iirAmountH);
midSampleR = iirSampleHR - iirSampleDR;
highSampleL = inputSampleL - iirSampleHL;
highSampleR = inputSampleR - iirSampleHR;
}
fpFlip = !fpFlip;
//generate the tone bands we're using
if (lowSampleL > 1.0) {lowSampleL = 1.0;}
if (lowSampleL < -1.0) {lowSampleL = -1.0;}
if (lowSampleR > 1.0) {lowSampleR = 1.0;}
if (lowSampleR < -1.0) {lowSampleR = -1.0;}
lowSampleL -= (lowSampleL * (fabs(lowSampleL) * 0.448) * (fabs(lowSampleL) * 0.448) );
lowSampleR -= (lowSampleR * (fabs(lowSampleR) * 0.448) * (fabs(lowSampleR) * 0.448) );
lowSampleL *= drive;
lowSampleR *= drive;
if (highSampleL > 1.0) {highSampleL = 1.0;}
if (highSampleL < -1.0) {highSampleL = -1.0;}
if (highSampleR > 1.0) {highSampleR = 1.0;}
if (highSampleR < -1.0) {highSampleR = -1.0;}
highSampleL -= (highSampleL * (fabs(highSampleL) * 0.599) * (fabs(highSampleL) * 0.599) );
highSampleR -= (highSampleR * (fabs(highSampleR) * 0.599) * (fabs(highSampleR) * 0.599) );
highSampleL *= drive;
highSampleR *= drive;
midSampleL = midSampleL * drive;
midSampleR = midSampleR * drive;
long double skew = (midSampleL - lastSampleL);
lastSampleL = midSampleL;
//skew will be direction/angle
long double bridgerectifier = fabs(skew);
if (bridgerectifier > 3.1415926) bridgerectifier = 3.1415926;
//for skew we want it to go to zero effect again, so we use full range of the sine
bridgerectifier = sin(bridgerectifier);
if (skew > 0) skew = bridgerectifier*3.1415926;
else skew = -bridgerectifier*3.1415926;
//skew is now sined and clamped and then re-amplified again
skew *= midSampleL;
//cools off sparkliness and crossover distortion
skew *= 1.557079633;
//crank up the gain on this so we can make it sing
bridgerectifier = fabs(midSampleL);
bridgerectifier += skew;
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
bridgerectifier = sin(bridgerectifier);
bridgerectifier *= drive;
bridgerectifier += skew;
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
bridgerectifier = sin(bridgerectifier);
if (midSampleL > 0)
{
midSampleL = bridgerectifier;
}
else
{
midSampleL = -bridgerectifier;
}
//blend according to positive and negative controls, left
skew = (midSampleR - lastSampleR);
lastSampleR = midSampleR;
//skew will be direction/angle
bridgerectifier = fabs(skew);
if (bridgerectifier > 3.1415926) bridgerectifier = 3.1415926;
//for skew we want it to go to zero effect again, so we use full range of the sine
bridgerectifier = sin(bridgerectifier);
if (skew > 0) skew = bridgerectifier*3.1415926;
else skew = -bridgerectifier*3.1415926;
//skew is now sined and clamped and then re-amplified again
skew *= midSampleR;
//cools off sparkliness and crossover distortion
skew *= 1.557079633;
//crank up the gain on this so we can make it sing
bridgerectifier = fabs(midSampleR);
bridgerectifier += skew;
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
bridgerectifier = sin(bridgerectifier);
bridgerectifier *= drive;
bridgerectifier += skew;
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
bridgerectifier = sin(bridgerectifier);
if (midSampleR > 0)
{
midSampleR = bridgerectifier;
}
else
{
midSampleR = -bridgerectifier;
}
//blend according to positive and negative controls, right
inputSampleL = ((lowSampleL + midSampleL + highSampleL)/drive)*out;
inputSampleR = ((lowSampleR + midSampleR + highSampleR)/drive)*out;
if (wet !=1.0) {
inputSampleL = (inputSampleL * wet) + (drySampleL * dry);
inputSampleR = (inputSampleR * wet) + (drySampleR * dry);
}
//stereo 64 bit dither, made small and tidy.
int expon; frexp((double)inputSampleL, &expon);
long double dither = (rand()/(RAND_MAX*7.737125245533627e+25))*pow(2,expon+62);
dither /= 536870912.0; //needs this to scale to 64 bit zone
inputSampleL += (dither-fpNShapeL); fpNShapeL = dither;
frexp((double)inputSampleR, &expon);
dither = (rand()/(RAND_MAX*7.737125245533627e+25))*pow(2,expon+62);
dither /= 536870912.0; //needs this to scale to 64 bit zone
inputSampleR += (dither-fpNShapeR); fpNShapeR = dither;
//end 64 bit dither
*out1 = inputSampleL;
*out2 = inputSampleR;
*in1++;
*in2++;
*out1++;
*out2++;
}
}
