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Diffstat (limited to 'plugins/MacVST/Chorus/source/ChorusProc.cpp')
| -rwxr-xr-x | plugins/MacVST/Chorus/source/ChorusProc.cpp | 324 |
1 files changed, 324 insertions, 0 deletions
diff --git a/plugins/MacVST/Chorus/source/ChorusProc.cpp b/plugins/MacVST/Chorus/source/ChorusProc.cpp new file mode 100755 index 0000000..aed2bc0 --- /dev/null +++ b/plugins/MacVST/Chorus/source/ChorusProc.cpp @@ -0,0 +1,324 @@ +/* ======================================== + * Chorus - Chorus.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __Chorus_H +#include "Chorus.h" +#endif + +void Chorus::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 speed = pow(A,4) * 0.001; + speed *= overallscale; + int loopLimit = (int)(totalsamples * 0.499); + int count; + double range = pow(B,4) * loopLimit * 0.499; + double wet = C; + double modulation = range*wet; + double dry = 1.0 - wet; + double tupi = 3.141592653589793238 * 2.0; + double offset; + //this is a double buffer so we will be splitting it in two + + float fpTemp; + long double fpOld = 0.618033988749894848204586; //golden ratio! + long double fpNew = 1.0 - fpOld; + + long double inputSampleL; + long double inputSampleR; + double drySampleL; + 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; + + airFactorL = airPrevL - inputSampleL; + if (fpFlip) {airEvenL += airFactorL; airOddL -= airFactorL; airFactorL = airEvenL;} + else {airOddL += airFactorL; airEvenL -= airFactorL; airFactorL = airOddL;} + airOddL = (airOddL - ((airOddL - airEvenL)/256.0)) / 1.0001; + airEvenL = (airEvenL - ((airEvenL - airOddL)/256.0)) / 1.0001; + airPrevL = inputSampleL; + inputSampleL += (airFactorL*wet); + //air, compensates for loss of highs in flanger's interpolation + + airFactorR = airPrevR - inputSampleR; + if (fpFlip) {airEvenR += airFactorR; airOddR -= airFactorR; airFactorR = airEvenR;} + else {airOddR += airFactorR; airEvenR -= airFactorR; airFactorR = airOddR;} + airOddR = (airOddR - ((airOddR - airEvenR)/256.0)) / 1.0001; + airEvenR = (airEvenR - ((airEvenR - airOddR)/256.0)) / 1.0001; + airPrevR = inputSampleR; + inputSampleR += (airFactorR*wet); + //air, compensates for loss of highs in flanger's interpolation + + if (gcount < 1 || gcount > loopLimit) {gcount = loopLimit;} + count = gcount; + dL[count+loopLimit] = dL[count] = inputSampleL; + dR[count+loopLimit] = dR[count] = inputSampleR; + gcount--; + //double buffer + + offset = range + (modulation * sin(sweep)); + count += (int)floor(offset); + + inputSampleL = dL[count] * (1-(offset-floor(offset))); //less as value moves away from .0 + inputSampleL += dL[count+1]; //we can assume always using this in one way or another? + inputSampleL += (dL[count+2] * (offset-floor(offset))); //greater as value moves away from .0 + inputSampleL -= (((dL[count]-dL[count+1])-(dL[count+1]-dL[count+2]))/50); //interpolation hacks 'r us + + inputSampleR = dR[count] * (1-(offset-floor(offset))); //less as value moves away from .0 + inputSampleR += dR[count+1]; //we can assume always using this in one way or another? + inputSampleR += (dR[count+2] * (offset-floor(offset))); //greater as value moves away from .0 + inputSampleR -= (((dR[count]-dR[count+1])-(dR[count+1]-dR[count+2]))/50); //interpolation hacks 'r us + + inputSampleL *= 0.5; //to get a comparable level + inputSampleR *= 0.5; //to get a comparable level + //sliding + + sweep += speed; + if (sweep > tupi){sweep -= tupi;} + //still scrolling through the samples, remember + + if (wet !=1.0) { + inputSampleL = (inputSampleL * wet) + (drySampleL * dry); + inputSampleR = (inputSampleR * wet) + (drySampleR * dry); + } + + //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 Chorus::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 speed = pow(A,4) * 0.001; + speed *= overallscale; + int loopLimit = (int)(totalsamples * 0.499); + int count; + double range = pow(B,4) * loopLimit * 0.499; + double wet = C; + double modulation = range*wet; + double dry = 1.0 - wet; + double tupi = 3.141592653589793238 * 2.0; + double offset; + //this is a double buffer so we will be splitting it in two + + 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; + double drySampleL; + 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; + + airFactorL = airPrevL - inputSampleL; + if (fpFlip) {airEvenL += airFactorL; airOddL -= airFactorL; airFactorL = airEvenL;} + else {airOddL += airFactorL; airEvenL -= airFactorL; airFactorL = airOddL;} + airOddL = (airOddL - ((airOddL - airEvenL)/256.0)) / 1.0001; + airEvenL = (airEvenL - ((airEvenL - airOddL)/256.0)) / 1.0001; + airPrevL = inputSampleL; + inputSampleL += (airFactorL*wet); + //air, compensates for loss of highs in flanger's interpolation + + airFactorR = airPrevR - inputSampleR; + if (fpFlip) {airEvenR += airFactorR; airOddR -= airFactorR; airFactorR = airEvenR;} + else {airOddR += airFactorR; airEvenR -= airFactorR; airFactorR = airOddR;} + airOddR = (airOddR - ((airOddR - airEvenR)/256.0)) / 1.0001; + airEvenR = (airEvenR - ((airEvenR - airOddR)/256.0)) / 1.0001; + airPrevR = inputSampleR; + inputSampleR += (airFactorR*wet); + //air, compensates for loss of highs in flanger's interpolation + + if (gcount < 1 || gcount > loopLimit) {gcount = loopLimit;} + count = gcount; + dL[count+loopLimit] = dL[count] = inputSampleL; + dR[count+loopLimit] = dR[count] = inputSampleR; + gcount--; + //double buffer + + offset = range + (modulation * sin(sweep)); + count += (int)floor(offset); + + inputSampleL = dL[count] * (1-(offset-floor(offset))); //less as value moves away from .0 + inputSampleL += dL[count+1]; //we can assume always using this in one way or another? + inputSampleL += (dL[count+2] * (offset-floor(offset))); //greater as value moves away from .0 + inputSampleL -= (((dL[count]-dL[count+1])-(dL[count+1]-dL[count+2]))/50); //interpolation hacks 'r us + + inputSampleR = dR[count] * (1-(offset-floor(offset))); //less as value moves away from .0 + inputSampleR += dR[count+1]; //we can assume always using this in one way or another? + inputSampleR += (dR[count+2] * (offset-floor(offset))); //greater as value moves away from .0 + inputSampleR -= (((dR[count]-dR[count+1])-(dR[count+1]-dR[count+2]))/50); //interpolation hacks 'r us + + inputSampleL *= 0.5; //to get a comparable level + inputSampleR *= 0.5; //to get a comparable level + //sliding + + sweep += speed; + if (sweep > tupi){sweep -= tupi;} + //still scrolling through the samples, remember + + if (wet !=1.0) { + inputSampleL = (inputSampleL * wet) + (drySampleL * dry); + inputSampleR = (inputSampleR * wet) + (drySampleR * dry); + } + + //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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