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Diffstat (limited to 'plugins/WinVST/Drive/DriveProc.cpp')
| -rwxr-xr-x | plugins/WinVST/Drive/DriveProc.cpp | 312 |
1 files changed, 312 insertions, 0 deletions
diff --git a/plugins/WinVST/Drive/DriveProc.cpp b/plugins/WinVST/Drive/DriveProc.cpp new file mode 100755 index 0000000..3670e85 --- /dev/null +++ b/plugins/WinVST/Drive/DriveProc.cpp @@ -0,0 +1,312 @@ +/* ======================================== + * Drive - Drive.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __Drive_H +#include "Drive.h" +#endif + +void Drive::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 driveone = pow(A*2.0,2); + double iirAmount = pow(B,3)/overallscale; + double output = C; + double wet = D; + double dry = 1.0-wet; + double glitch = 0.60; + double out; + + float fpTemp; + long double fpOld = 0.618033988749894848204586; //golden ratio! + long double fpNew = 1.0 - fpOld; + + long double inputSampleL; + long double inputSampleR; + long double drySampleL; + long double drySampleR; + + while (--sampleFrames >= 0) + { + inputSampleL = *in1; + 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. + } + drySampleL = inputSampleL; + drySampleR = inputSampleR; + + if (fpFlip) + { + iirSampleAL = (iirSampleAL * (1.0 - iirAmount)) + (inputSampleL * iirAmount); + inputSampleL -= iirSampleAL; + iirSampleAR = (iirSampleAR * (1.0 - iirAmount)) + (inputSampleR * iirAmount); + inputSampleR -= iirSampleAR; + } + else + { + iirSampleBL = (iirSampleBL * (1.0 - iirAmount)) + (inputSampleL * iirAmount); + inputSampleL -= iirSampleBL; + iirSampleBR = (iirSampleBR * (1.0 - iirAmount)) + (inputSampleR * iirAmount); + inputSampleR -= iirSampleBR; + } + //highpass section + + + if (inputSampleL > 1.0) inputSampleL = 1.0; + if (inputSampleL < -1.0) inputSampleL = -1.0; + if (inputSampleR > 1.0) inputSampleR = 1.0; + if (inputSampleR < -1.0) inputSampleR = -1.0; + + out = driveone; + while (out > glitch) + { + out -= glitch; + inputSampleL -= (inputSampleL * (fabs(inputSampleL) * glitch) * (fabs(inputSampleL) * glitch) ); + inputSampleL *= (1.0+glitch); + inputSampleR -= (inputSampleR * (fabs(inputSampleR) * glitch) * (fabs(inputSampleR) * glitch) ); + inputSampleR *= (1.0+glitch); + } + //that's taken care of the really high gain stuff + + inputSampleL -= (inputSampleL * (fabs(inputSampleL) * out) * (fabs(inputSampleL) * out) ); + inputSampleL *= (1.0+out); + inputSampleR -= (inputSampleR * (fabs(inputSampleR) * out) * (fabs(inputSampleR) * out) ); + inputSampleR *= (1.0+out); + + if (output < 1.0) { + inputSampleL *= output; + inputSampleR *= output; + } + if (wet < 1.0) { + inputSampleL = (drySampleL * dry)+(inputSampleL * wet); + inputSampleR = (drySampleR * dry)+(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. + + //noise shaping to 32-bit floating point + if (fpFlip) { + fpTemp = inputSampleL; + fpNShapeLA = (fpNShapeLA*fpOld)+((inputSampleL-fpTemp)*fpNew); + inputSampleL += fpNShapeLA; + fpTemp = inputSampleR; + fpNShapeRA = (fpNShapeRA*fpOld)+((inputSampleR-fpTemp)*fpNew); + inputSampleR += fpNShapeRA; + } + else { + fpTemp = inputSampleL; + fpNShapeLB = (fpNShapeLB*fpOld)+((inputSampleL-fpTemp)*fpNew); + inputSampleL += fpNShapeLB; + fpTemp = inputSampleR; + fpNShapeRB = (fpNShapeRB*fpOld)+((inputSampleR-fpTemp)*fpNew); + inputSampleR += fpNShapeRB; + } + fpFlip = !fpFlip; + //end noise shaping on 32 bit output + + *out1 = inputSampleL; + *out2 = inputSampleR; + + *in1++; + *in2++; + *out1++; + *out2++; + } +} + +void Drive::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 driveone = pow(A*2.0,2); + double iirAmount = pow(B,3)/overallscale; + double output = C; + double wet = D; + double dry = 1.0-wet; + double glitch = 0.60; + double out; + + double fpTemp; //this is different from singlereplacing + long double fpOld = 0.618033988749894848204586; //golden ratio! + long double fpNew = 1.0 - fpOld; + + long double inputSampleL; + long double inputSampleR; + long double drySampleL; + long double drySampleR; + + while (--sampleFrames >= 0) + { + inputSampleL = *in1; + 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. + } + drySampleL = inputSampleL; + drySampleR = inputSampleR; + + if (fpFlip) + { + iirSampleAL = (iirSampleAL * (1.0 - iirAmount)) + (inputSampleL * iirAmount); + inputSampleL -= iirSampleAL; + iirSampleAR = (iirSampleAR * (1.0 - iirAmount)) + (inputSampleR * iirAmount); + inputSampleR -= iirSampleAR; + } + else + { + iirSampleBL = (iirSampleBL * (1.0 - iirAmount)) + (inputSampleL * iirAmount); + inputSampleL -= iirSampleBL; + iirSampleBR = (iirSampleBR * (1.0 - iirAmount)) + (inputSampleR * iirAmount); + inputSampleR -= iirSampleBR; + } + //highpass section + + + if (inputSampleL > 1.0) inputSampleL = 1.0; + if (inputSampleL < -1.0) inputSampleL = -1.0; + if (inputSampleR > 1.0) inputSampleR = 1.0; + if (inputSampleR < -1.0) inputSampleR = -1.0; + + out = driveone; + while (out > glitch) + { + out -= glitch; + inputSampleL -= (inputSampleL * (fabs(inputSampleL) * glitch) * (fabs(inputSampleL) * glitch) ); + inputSampleL *= (1.0+glitch); + inputSampleR -= (inputSampleR * (fabs(inputSampleR) * glitch) * (fabs(inputSampleR) * glitch) ); + inputSampleR *= (1.0+glitch); + } + //that's taken care of the really high gain stuff + + inputSampleL -= (inputSampleL * (fabs(inputSampleL) * out) * (fabs(inputSampleL) * out) ); + inputSampleL *= (1.0+out); + inputSampleR -= (inputSampleR * (fabs(inputSampleR) * out) * (fabs(inputSampleR) * out) ); + inputSampleR *= (1.0+out); + + if (output < 1.0) { + inputSampleL *= output; + inputSampleR *= output; + } + if (wet < 1.0) { + inputSampleL = (drySampleL * dry)+(inputSampleL * wet); + inputSampleR = (drySampleR * dry)+(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. + + //noise shaping to 64-bit floating point + if (fpFlip) { + fpTemp = inputSampleL; + fpNShapeLA = (fpNShapeLA*fpOld)+((inputSampleL-fpTemp)*fpNew); + inputSampleL += fpNShapeLA; + fpTemp = inputSampleR; + fpNShapeRA = (fpNShapeRA*fpOld)+((inputSampleR-fpTemp)*fpNew); + inputSampleR += fpNShapeRA; + } + else { + fpTemp = inputSampleL; + fpNShapeLB = (fpNShapeLB*fpOld)+((inputSampleL-fpTemp)*fpNew); + inputSampleL += fpNShapeLB; + fpTemp = inputSampleR; + fpNShapeRB = (fpNShapeRB*fpOld)+((inputSampleR-fpTemp)*fpNew); + inputSampleR += fpNShapeRB; + } + fpFlip = !fpFlip; + //end noise shaping on 64 bit output + + *out1 = inputSampleL; + *out2 = inputSampleR; + + *in1++; + *in2++; + *out1++; + *out2++; + } +}
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