diff options
Diffstat (limited to 'plugins/WinVST/GuitarConditioner/GuitarConditionerProc.cpp')
| -rwxr-xr-x | plugins/WinVST/GuitarConditioner/GuitarConditionerProc.cpp | 385 |
1 files changed, 385 insertions, 0 deletions
diff --git a/plugins/WinVST/GuitarConditioner/GuitarConditionerProc.cpp b/plugins/WinVST/GuitarConditioner/GuitarConditionerProc.cpp new file mode 100755 index 0000000..83970f2 --- /dev/null +++ b/plugins/WinVST/GuitarConditioner/GuitarConditionerProc.cpp @@ -0,0 +1,385 @@ +/* ======================================== + * GuitarConditioner - GuitarConditioner.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __GuitarConditioner_H +#include "GuitarConditioner.h" +#endif + +void GuitarConditioner::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(); + float fpTemp; + long double fpOld = 0.618033988749894848204586; //golden ratio! + long double fpNew = 1.0 - fpOld; + + long double inputSampleL; + long double inputSampleR; + long double trebleL; + long double bassL; + long double trebleR; + long double bassR; + + double iirTreble = 0.287496/overallscale; //tight is -1 + double iirBass = 0.085184/overallscale; //tight is 1 + iirTreble += iirTreble; + iirBass += iirBass; //simple double when tight is -1 or 1 + double tightBass = 0.6666666666; + double tightTreble = -0.3333333333; + double offset; + double clamp; + double threshTreble = 0.0081/overallscale; + double threshBass = 0.0256/overallscale; + + 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. + } + + trebleL = bassL = inputSampleL; + trebleR = bassR = inputSampleR; + trebleL += trebleL; //+3dB on treble as the highpass is higher + trebleR += trebleR; //+3dB on treble as the highpass is higher + + offset = (1 + tightTreble) + ((1-fabs(trebleL))*tightTreble); //treble HP + if (offset < 0) offset = 0; + if (offset > 1) offset = 1; //made offset for HP + if (fpFlip) { + iirSampleTAL = (iirSampleTAL * (1 - (offset * iirTreble))) + (trebleL * (offset * iirTreble)); + trebleL = trebleL - iirSampleTAL; + } else { + iirSampleTBL = (iirSampleTBL * (1 - (offset * iirTreble))) + (trebleL * (offset * iirTreble)); + trebleL = trebleL - iirSampleTBL; + } //done trebleL HP + + offset = (1 + tightTreble) + ((1-fabs(trebleR))*tightTreble); //treble HP + if (offset < 0) offset = 0; + if (offset > 1) offset = 1; //made offset for HP + if (fpFlip) { + iirSampleTAR = (iirSampleTAR * (1 - (offset * iirTreble))) + (trebleR * (offset * iirTreble)); + trebleR = trebleR - iirSampleTAR; + } else { + iirSampleTBR = (iirSampleTBR * (1 - (offset * iirTreble))) + (trebleR * (offset * iirTreble)); + trebleR = trebleR - iirSampleTBR; + } //done trebleR HP + + offset = (1 - tightBass) + (fabs(bassL)*tightBass); //bass HP + if (offset < 0) offset = 0; + if (offset > 1) offset = 1; + if (fpFlip) { + iirSampleBAL = (iirSampleBAL * (1 - (offset * iirBass))) + (bassL * (offset * iirBass)); + bassL = bassL - iirSampleBAL; + } else { + iirSampleBBL = (iirSampleBBL * (1 - (offset * iirBass))) + (bassL * (offset * iirBass)); + bassL = bassL - iirSampleBBL; + } //done bassL HP + + offset = (1 - tightBass) + (fabs(bassR)*tightBass); //bass HP + if (offset < 0) offset = 0; + if (offset > 1) offset = 1; + if (fpFlip) { + iirSampleBAR = (iirSampleBAR * (1 - (offset * iirBass))) + (bassR * (offset * iirBass)); + bassR = bassR - iirSampleBAR; + } else { + iirSampleBBR = (iirSampleBBR * (1 - (offset * iirBass))) + (bassR * (offset * iirBass)); + bassR = bassR - iirSampleBBR; + } //done bassR HP + + inputSampleL = trebleL; + clamp = inputSampleL - lastSampleTL; + if (clamp > threshTreble) + trebleL = lastSampleTL + threshTreble; + if (-clamp > threshTreble) + trebleL = lastSampleTL - threshTreble; + lastSampleTL = trebleL; //trebleL slew + + inputSampleR = trebleR; + clamp = inputSampleR - lastSampleTR; + if (clamp > threshTreble) + trebleR = lastSampleTR + threshTreble; + if (-clamp > threshTreble) + trebleR = lastSampleTR - threshTreble; + lastSampleTR = trebleR; //trebleR slew + + inputSampleL = bassL; + clamp = inputSampleL - lastSampleBL; + if (clamp > threshBass) + bassL = lastSampleBL + threshBass; + if (-clamp > threshBass) + bassL = lastSampleBL - threshBass; + lastSampleBL = bassL; //bassL slew + + inputSampleR = bassR; + clamp = inputSampleR - lastSampleBR; + if (clamp > threshBass) + bassR = lastSampleBR + threshBass; + if (-clamp > threshBass) + bassR = lastSampleBR - threshBass; + lastSampleBR = bassR; //bassR slew + + inputSampleL = trebleL + bassL; //final merge + inputSampleR = trebleR + bassR; //final merge + //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 GuitarConditioner::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 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 trebleL; + long double bassL; + long double trebleR; + long double bassR; + + double iirTreble = 0.287496/overallscale; //tight is -1 + double iirBass = 0.085184/overallscale; //tight is 1 + iirTreble += iirTreble; + iirBass += iirBass; //simple double when tight is -1 or 1 + double tightBass = 0.6666666666; + double tightTreble = -0.3333333333; + double offset; + double clamp; + double threshTreble = 0.0081/overallscale; + double threshBass = 0.0256/overallscale; + + + 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. + } + + trebleL = bassL = inputSampleL; + trebleR = bassR = inputSampleR; + trebleL += trebleL; //+3dB on treble as the highpass is higher + trebleR += trebleR; //+3dB on treble as the highpass is higher + + offset = (1 + tightTreble) + ((1-fabs(trebleL))*tightTreble); //treble HP + if (offset < 0) offset = 0; + if (offset > 1) offset = 1; //made offset for HP + if (fpFlip) { + iirSampleTAL = (iirSampleTAL * (1 - (offset * iirTreble))) + (trebleL * (offset * iirTreble)); + trebleL = trebleL - iirSampleTAL; + } else { + iirSampleTBL = (iirSampleTBL * (1 - (offset * iirTreble))) + (trebleL * (offset * iirTreble)); + trebleL = trebleL - iirSampleTBL; + } //done trebleL HP + + offset = (1 + tightTreble) + ((1-fabs(trebleR))*tightTreble); //treble HP + if (offset < 0) offset = 0; + if (offset > 1) offset = 1; //made offset for HP + if (fpFlip) { + iirSampleTAR = (iirSampleTAR * (1 - (offset * iirTreble))) + (trebleR * (offset * iirTreble)); + trebleR = trebleR - iirSampleTAR; + } else { + iirSampleTBR = (iirSampleTBR * (1 - (offset * iirTreble))) + (trebleR * (offset * iirTreble)); + trebleR = trebleR - iirSampleTBR; + } //done trebleR HP + + offset = (1 - tightBass) + (fabs(bassL)*tightBass); //bass HP + if (offset < 0) offset = 0; + if (offset > 1) offset = 1; + if (fpFlip) { + iirSampleBAL = (iirSampleBAL * (1 - (offset * iirBass))) + (bassL * (offset * iirBass)); + bassL = bassL - iirSampleBAL; + } else { + iirSampleBBL = (iirSampleBBL * (1 - (offset * iirBass))) + (bassL * (offset * iirBass)); + bassL = bassL - iirSampleBBL; + } //done bassL HP + + offset = (1 - tightBass) + (fabs(bassR)*tightBass); //bass HP + if (offset < 0) offset = 0; + if (offset > 1) offset = 1; + if (fpFlip) { + iirSampleBAR = (iirSampleBAR * (1 - (offset * iirBass))) + (bassR * (offset * iirBass)); + bassR = bassR - iirSampleBAR; + } else { + iirSampleBBR = (iirSampleBBR * (1 - (offset * iirBass))) + (bassR * (offset * iirBass)); + bassR = bassR - iirSampleBBR; + } //done bassR HP + + inputSampleL = trebleL; + clamp = inputSampleL - lastSampleTL; + if (clamp > threshTreble) + trebleL = lastSampleTL + threshTreble; + if (-clamp > threshTreble) + trebleL = lastSampleTL - threshTreble; + lastSampleTL = trebleL; //trebleL slew + + inputSampleR = trebleR; + clamp = inputSampleR - lastSampleTR; + if (clamp > threshTreble) + trebleR = lastSampleTR + threshTreble; + if (-clamp > threshTreble) + trebleR = lastSampleTR - threshTreble; + lastSampleTR = trebleR; //trebleR slew + + inputSampleL = bassL; + clamp = inputSampleL - lastSampleBL; + if (clamp > threshBass) + bassL = lastSampleBL + threshBass; + if (-clamp > threshBass) + bassL = lastSampleBL - threshBass; + lastSampleBL = bassL; //bassL slew + + inputSampleR = bassR; + clamp = inputSampleR - lastSampleBR; + if (clamp > threshBass) + bassR = lastSampleBR + threshBass; + if (-clamp > threshBass) + bassR = lastSampleBR - threshBass; + lastSampleBR = bassR; //bassR slew + + inputSampleL = trebleL + bassL; //final merge + inputSampleR = trebleR + bassR; //final merge + //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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