diff options
Diffstat (limited to 'plugins/WinVST/Spiral2/Spiral2Proc.cpp')
| -rwxr-xr-x | plugins/WinVST/Spiral2/Spiral2Proc.cpp | 289 |
1 files changed, 289 insertions, 0 deletions
diff --git a/plugins/WinVST/Spiral2/Spiral2Proc.cpp b/plugins/WinVST/Spiral2/Spiral2Proc.cpp new file mode 100755 index 0000000..d309448 --- /dev/null +++ b/plugins/WinVST/Spiral2/Spiral2Proc.cpp @@ -0,0 +1,289 @@ +/* ======================================== + * 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. +} |