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path: root/plugins/WinVST/Coils/CoilsProc.cpp
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/* ========================================
 *  Coils - Coils.h
 *  Copyright (c) 2016 airwindows, All rights reserved
 * ======================================== */

#ifndef __Coils_H
#include "Coils.h"
#endif

void Coils::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames) 
{
    float* in1  =  inputs[0];
    float* in2  =  inputs[1];
    float* out1 = outputs[0];
    float* out2 = outputs[1];

	//[0] is frequency: 0.000001 to 0.499999 is near-zero to near-Nyquist
	//[1] is resonance, 0.7071 is Butterworth. Also can't be zero
	double boost = 1.0-pow(A,2);
	if (boost < 0.001) boost = 0.001; //there's a divide, we can't have this be zero
	figureL[0] = figureR[0] = 600.0/getSampleRate(); //fixed frequency, 600hz
	figureL[1] = figureR[1] = 0.023; //resonance
	double offset = (B*2.0)-1.0;
	double sinOffset = sin(offset); //we can cache this, it's expensive
	double wet = C;
	double K = tan(M_PI * figureR[0]);
	double norm = 1.0 / (1.0 + K / figureR[1] + K * K);
	figureL[2] = figureR[2] = K / figureR[1] * norm;
	figureL[4] = figureR[4] = -figureR[2];
	figureL[5] = figureR[5] = 2.0 * (K * K - 1.0) * norm;
	figureL[6] = figureR[6] = (1.0 - K / figureR[1] + K * K) * norm;
	
    while (--sampleFrames >= 0)
    {
		long double inputSampleL = *in1;
		long double inputSampleR = *in2;
		if (fabs(inputSampleL)<1.18e-37) inputSampleL = fpd * 1.18e-37;
		if (fabs(inputSampleR)<1.18e-37) inputSampleR = fpd * 1.18e-37;
		long double drySampleL = inputSampleL;
		long double drySampleR = inputSampleR;
		
		//long double tempSample = (inputSample * figure[2]) + figure[7];
		//figure[7] = -(tempSample * figure[5]) + figure[8];
		//figure[8] = (inputSample * figure[4]) - (tempSample * figure[6]);
		//inputSample = tempSample + sin(drySample-tempSample);
		//or
		//inputSample = tempSample + ((sin(((drySample-tempSample)/boost)+offset)-sinOffset)*boost);
		//
		//given a bandlimited inputSample, freq 600hz and Q of 0.023, this restores a lot of
		//the full frequencies but distorts like a real transformer. Purest case, and since
		//we are not using a high Q we can remove the extra sin/asin on the biquad.
		
				
		long double tempSample = (inputSampleL * figureL[2]) + figureL[7];
		figureL[7] = -(tempSample * figureL[5]) + figureL[8];
		figureL[8] = (inputSampleL * figureL[4]) - (tempSample * figureL[6]);
		inputSampleL = tempSample + ((sin(((drySampleL-tempSample)/boost)+offset)-sinOffset)*boost);
		//given a bandlimited inputSample, freq 600hz and Q of 0.023, this restores a lot of
		//the full frequencies but distorts like a real transformer. Since
		//we are not using a high Q we can remove the extra sin/asin on the biquad.
		
		tempSample = (inputSampleR * figureR[2]) + figureR[7];
		figureR[7] = -(tempSample * figureR[5]) + figureR[8];
		figureR[8] = (inputSampleR * figureR[4]) - (tempSample * figureR[6]);
		inputSampleR = tempSample + ((sin(((drySampleR-tempSample)/boost)+offset)-sinOffset)*boost);
		//given a bandlimited inputSample, freq 600hz and Q of 0.023, this restores a lot of
		//the full frequencies but distorts like a real transformer. Since
		//we are not using a high Q we can remove the extra sin/asin on the biquad.
		
		if (wet !=1.0) {
			inputSampleL = (inputSampleL * wet) + (drySampleL * (1.0-wet));
			inputSampleR = (inputSampleR * wet) + (drySampleR * (1.0-wet));
		}
		
		//begin 32 bit stereo floating point dither
		int expon; frexpf((float)inputSampleL, &expon);
		fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5;
		inputSampleL += ((double(fpd)-uint32_t(0x7fffffff)) * 5.5e-36l * pow(2,expon+62));
		frexpf((float)inputSampleR, &expon);
		fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5;
		inputSampleR += ((double(fpd)-uint32_t(0x7fffffff)) * 5.5e-36l * pow(2,expon+62));
		//end 32 bit stereo floating point dither
		
		*out1 = inputSampleL;
		*out2 = inputSampleR;

		*in1++;
		*in2++;
		*out1++;
		*out2++;
    }
}

void Coils::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames) 
{
    double* in1  =  inputs[0];
    double* in2  =  inputs[1];
    double* out1 = outputs[0];
    double* out2 = outputs[1];
	
	//[0] is frequency: 0.000001 to 0.499999 is near-zero to near-Nyquist
	//[1] is resonance, 0.7071 is Butterworth. Also can't be zero
	double boost = 1.0-pow(A,2);
	if (boost < 0.001) boost = 0.001; //there's a divide, we can't have this be zero
	figureL[0] = figureR[0] = 600.0/getSampleRate(); //fixed frequency, 600hz
	figureL[1] = figureR[1] = 0.023; //resonance
	double offset = (B*2.0)-1.0;
	double sinOffset = sin(offset); //we can cache this, it's expensive
	double wet = C;
	double K = tan(M_PI * figureR[0]);
	double norm = 1.0 / (1.0 + K / figureR[1] + K * K);
	figureL[2] = figureR[2] = K / figureR[1] * norm;
	figureL[4] = figureR[4] = -figureR[2];
	figureL[5] = figureR[5] = 2.0 * (K * K - 1.0) * norm;
	figureL[6] = figureR[6] = (1.0 - K / figureR[1] + K * K) * norm;
	
    while (--sampleFrames >= 0)
    {
		long double inputSampleL = *in1;
		long double inputSampleR = *in2;
		if (fabs(inputSampleL)<1.18e-43) inputSampleL = fpd * 1.18e-43;
		if (fabs(inputSampleR)<1.18e-43) inputSampleR = fpd * 1.18e-43;
		long double drySampleL = inputSampleL;
		long double drySampleR = inputSampleR;
		
		//long double tempSample = (inputSample * figure[2]) + figure[7];
		//figure[7] = -(tempSample * figure[5]) + figure[8];
		//figure[8] = (inputSample * figure[4]) - (tempSample * figure[6]);
		//inputSample = tempSample + sin(drySample-tempSample);
		//or
		//inputSample = tempSample + ((sin(((drySample-tempSample)/boost)+offset)-sinOffset)*boost);
		//
		//given a bandlimited inputSample, freq 600hz and Q of 0.023, this restores a lot of
		//the full frequencies but distorts like a real transformer. Purest case, and since
		//we are not using a high Q we can remove the extra sin/asin on the biquad.
		
		
		long double tempSample = (inputSampleL * figureL[2]) + figureL[7];
		figureL[7] = -(tempSample * figureL[5]) + figureL[8];
		figureL[8] = (inputSampleL * figureL[4]) - (tempSample * figureL[6]);
		inputSampleL = tempSample + ((sin(((drySampleL-tempSample)/boost)+offset)-sinOffset)*boost);
		//given a bandlimited inputSample, freq 600hz and Q of 0.023, this restores a lot of
		//the full frequencies but distorts like a real transformer. Since
		//we are not using a high Q we can remove the extra sin/asin on the biquad.
		
		tempSample = (inputSampleR * figureR[2]) + figureR[7];
		figureR[7] = -(tempSample * figureR[5]) + figureR[8];
		figureR[8] = (inputSampleR * figureR[4]) - (tempSample * figureR[6]);
		inputSampleR = tempSample + ((sin(((drySampleR-tempSample)/boost)+offset)-sinOffset)*boost);
		//given a bandlimited inputSample, freq 600hz and Q of 0.023, this restores a lot of
		//the full frequencies but distorts like a real transformer. Since
		//we are not using a high Q we can remove the extra sin/asin on the biquad.
		
		if (wet !=1.0) {
			inputSampleL = (inputSampleL * wet) + (drySampleL * (1.0-wet));
			inputSampleR = (inputSampleR * wet) + (drySampleR * (1.0-wet));
		}
		
		//begin 64 bit stereo floating point dither
		int expon; frexp((double)inputSampleL, &expon);
		fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5;
		inputSampleL += ((double(fpd)-uint32_t(0x7fffffff)) * 1.1e-44l * pow(2,expon+62));
		frexp((double)inputSampleR, &expon);
		fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5;
		inputSampleR += ((double(fpd)-uint32_t(0x7fffffff)) * 1.1e-44l * pow(2,expon+62));
		//end 64 bit stereo floating point dither
		
		*out1 = inputSampleL;
		*out2 = inputSampleR;

		*in1++;
		*in2++;
		*out1++;
		*out2++;
    }
}