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Diffstat (limited to 'plugins/WinVST/Channel6/Channel6Proc.cpp')
| -rwxr-xr-x | plugins/WinVST/Channel6/Channel6Proc.cpp | 294 |
1 files changed, 294 insertions, 0 deletions
diff --git a/plugins/WinVST/Channel6/Channel6Proc.cpp b/plugins/WinVST/Channel6/Channel6Proc.cpp new file mode 100755 index 0000000..d206851 --- /dev/null +++ b/plugins/WinVST/Channel6/Channel6Proc.cpp @@ -0,0 +1,294 @@ +/* ======================================== + * Channel6 - Channel6.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __Channel6_H +#include "Channel6.h" +#endif + +void Channel6::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(); + const double localiirAmount = iirAmount / overallscale; + const double localthreshold = threshold / overallscale; + const double density = pow(drive,2); //this doesn't relate to the plugins Density and Drive much + + 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 again. We want a 'air' hiss + + if (flip) + { + iirSampleLA = (iirSampleLA * (1 - localiirAmount)) + (inputSampleL * localiirAmount); + inputSampleL = inputSampleL - iirSampleLA; + iirSampleRA = (iirSampleRA * (1 - localiirAmount)) + (inputSampleR * localiirAmount); + inputSampleR = inputSampleR - iirSampleRA; + } + else + { + iirSampleLB = (iirSampleLB * (1 - localiirAmount)) + (inputSampleL * localiirAmount); + inputSampleL = inputSampleL - iirSampleLB; + iirSampleRB = (iirSampleRB * (1 - localiirAmount)) + (inputSampleR * localiirAmount); + inputSampleR = inputSampleR - iirSampleRB; + } + //highpass section + long double drySampleL = inputSampleL; + long double drySampleR = inputSampleR; + + + if (inputSampleL > 1.0) inputSampleL = 1.0; + if (inputSampleL < -1.0) inputSampleL = -1.0; + inputSampleL *= 1.2533141373155; + //clip to 1.2533141373155 to reach maximum output + + long double distSampleL = sin(inputSampleL * fabs(inputSampleL)) / ((inputSampleL == 0.0) ?1:fabs(inputSampleL)); + inputSampleL = (drySampleL*(1-density))+(distSampleL*density); + //drive section + + if (inputSampleR > 1.0) inputSampleR = 1.0; + if (inputSampleR < -1.0) inputSampleR = -1.0; + inputSampleR *= 1.2533141373155; + //clip to 1.2533141373155 to reach maximum output + + long double distSampleR = sin(inputSampleR * fabs(inputSampleR)) / ((inputSampleR == 0.0) ?1:fabs(inputSampleR)); + inputSampleR = (drySampleR*(1-density))+(distSampleR*density); + //drive section + + double clamp = inputSampleL - lastSampleL; + if (clamp > localthreshold) + inputSampleL = lastSampleL + localthreshold; + if (-clamp > localthreshold) + inputSampleL = lastSampleL - localthreshold; + lastSampleL = inputSampleL; + + clamp = inputSampleR - lastSampleR; + if (clamp > localthreshold) + inputSampleR = lastSampleR + localthreshold; + if (-clamp > localthreshold) + inputSampleR = lastSampleR - localthreshold; + lastSampleR = inputSampleR; + //slew section + flip = !flip; + + if (output < 1.0) { + inputSampleL *= output; + inputSampleR *= output; + } + + //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 Channel6::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(); + const double localiirAmount = iirAmount / overallscale; + const double localthreshold = threshold / overallscale; + const double density = pow(drive,2); //this doesn't relate to the plugins Density and Drive much + + 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 again. We want a 'air' hiss + + if (flip) + { + iirSampleLA = (iirSampleLA * (1 - localiirAmount)) + (inputSampleL * localiirAmount); + inputSampleL = inputSampleL - iirSampleLA; + iirSampleRA = (iirSampleRA * (1 - localiirAmount)) + (inputSampleR * localiirAmount); + inputSampleR = inputSampleR - iirSampleRA; + } + else + { + iirSampleLB = (iirSampleLB * (1 - localiirAmount)) + (inputSampleL * localiirAmount); + inputSampleL = inputSampleL - iirSampleLB; + iirSampleRB = (iirSampleRB * (1 - localiirAmount)) + (inputSampleR * localiirAmount); + inputSampleR = inputSampleR - iirSampleRB; + } + //highpass section + long double drySampleL = inputSampleL; + long double drySampleR = inputSampleR; + + + if (inputSampleL > 1.0) inputSampleL = 1.0; + if (inputSampleL < -1.0) inputSampleL = -1.0; + inputSampleL *= 1.2533141373155; + //clip to 1.2533141373155 to reach maximum output + + long double distSampleL = sin(inputSampleL * fabs(inputSampleL)) / ((inputSampleL == 0.0) ?1:fabs(inputSampleL)); + inputSampleL = (drySampleL*(1-density))+(distSampleL*density); + //drive section + + if (inputSampleR > 1.0) inputSampleR = 1.0; + if (inputSampleR < -1.0) inputSampleR = -1.0; + inputSampleR *= 1.2533141373155; + //clip to 1.2533141373155 to reach maximum output + + long double distSampleR = sin(inputSampleR * fabs(inputSampleR)) / ((inputSampleR == 0.0) ?1:fabs(inputSampleR)); + inputSampleR = (drySampleR*(1-density))+(distSampleR*density); + //drive section + + double clamp = inputSampleL - lastSampleL; + if (clamp > localthreshold) + inputSampleL = lastSampleL + localthreshold; + if (-clamp > localthreshold) + inputSampleL = lastSampleL - localthreshold; + lastSampleL = inputSampleL; + + clamp = inputSampleR - lastSampleR; + if (clamp > localthreshold) + inputSampleR = lastSampleR + localthreshold; + if (-clamp > localthreshold) + inputSampleR = lastSampleR - localthreshold; + lastSampleR = inputSampleR; + //slew section + flip = !flip; + + if (output < 1.0) { + inputSampleL *= output; + inputSampleR *= output; + } + + //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. +} |