/* ======================================== * 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(); 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 fpFlip = !fpFlip; //stereo 32 bit dither, made small and tidy. int expon; frexpf((float)inputSampleL, &expon); long double dither = (rand()/(RAND_MAX*7.737125245533627e+25))*pow(2,expon+62); inputSampleL += (dither-fpNShapeL); fpNShapeL = dither; frexpf((float)inputSampleR, &expon); dither = (rand()/(RAND_MAX*7.737125245533627e+25))*pow(2,expon+62); inputSampleR += (dither-fpNShapeR); fpNShapeR = dither; //end 32 bit dither *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(); 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 fpFlip = !fpFlip; //stereo 64 bit dither, made small and tidy. int expon; frexp((double)inputSampleL, &expon); long double dither = (rand()/(RAND_MAX*7.737125245533627e+25))*pow(2,expon+62); dither /= 536870912.0; //needs this to scale to 64 bit zone inputSampleL += (dither-fpNShapeL); fpNShapeL = dither; frexp((double)inputSampleR, &expon); dither = (rand()/(RAND_MAX*7.737125245533627e+25))*pow(2,expon+62); dither /= 536870912.0; //needs this to scale to 64 bit zone inputSampleR += (dither-fpNShapeR); fpNShapeR = dither; //end 64 bit dither *out1 = inputSampleL; *out2 = inputSampleR; *in1++; *in2++; *out1++; *out2++; } }