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author | Chris Johnson <jinx6568@sover.net> | 2018-10-22 18:04:06 -0400 |
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committer | Chris Johnson <jinx6568@sover.net> | 2018-10-22 18:04:06 -0400 |
commit | 633be2e22c6648c901f08f3b4cd4e8e14ea86443 (patch) | |
tree | 1e272c3d2b5bd29636b9f9f521af62734e4df012 /plugins/WinVST/Floor/FloorProc.cpp | |
parent | 057757aa8eb0a463caf0cdfdb5894ac5f723ff3f (diff) | |
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Updates (in case my plane crashes)
Diffstat (limited to 'plugins/WinVST/Floor/FloorProc.cpp')
-rwxr-xr-x | plugins/WinVST/Floor/FloorProc.cpp | 546 |
1 files changed, 546 insertions, 0 deletions
diff --git a/plugins/WinVST/Floor/FloorProc.cpp b/plugins/WinVST/Floor/FloorProc.cpp new file mode 100755 index 0000000..69a2ccb --- /dev/null +++ b/plugins/WinVST/Floor/FloorProc.cpp @@ -0,0 +1,546 @@ +/* ======================================== + * Floor - Floor.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __Floor_H +#include "Floor.h" +#endif + +void Floor::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 setting = pow(A,2); + double iirAmount = (setting/4.0)/overallscale; + double tight = -1.0; + double gaintrim = 1.0 + (setting/4.0); + double offset; + double lows; + double density = B; + double bridgerectifier; + double temp; + iirAmount += (iirAmount * tight * tight); + tight /= 3.0; + if (iirAmount <= 0.0) iirAmount = 0.0; + if (iirAmount > 1.0) iirAmount = 1.0; + double wet = C; + double dry = 1.0-wet; + + 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 + long double drySampleL = inputSampleL; + long double drySampleR = inputSampleR; + + //begin left channel + if (tight > 0) offset = (1 - tight) + (fabs(inputSampleL)*tight); + else offset = (1 + tight) + ((1-fabs(inputSampleL))*tight); + if (offset < 0) offset = 0; + if (offset > 1) offset = 1; + iirSample1AL = (iirSample1AL * (1 - (offset * iirAmount))) + (inputSampleL * (offset * iirAmount)); + lows = iirSample1AL; + inputSampleL -= lows; + temp = lows; + if (lows < 0) {lows = -sin(-lows*1.5707963267949);} + if (lows > 0) {lows = sin(lows*1.5707963267949);} + lows -= temp; + inputSampleL += lows; + inputSampleL *= gaintrim; + + if (tight > 0) offset = (1 - tight) + (fabs(inputSampleL)*tight); + else offset = (1 + tight) + ((1-fabs(inputSampleL))*tight); + if (offset < 0) offset = 0; + if (offset > 1) offset = 1; + iirSample1BL = (iirSample1BL * (1 - (offset * iirAmount))) + (inputSampleL * (offset * iirAmount)); + lows = iirSample1BL; + inputSampleL -= lows; + temp = lows; + if (lows < 0) {lows = -sin(-lows*1.5707963267949);} + if (lows > 0) {lows = sin(lows*1.5707963267949);} + lows -= temp; + inputSampleL += lows; + inputSampleL *= gaintrim; + + if (tight > 0) offset = (1 - tight) + (fabs(inputSampleL)*tight); + else offset = (1 + tight) + ((1-fabs(inputSampleL))*tight); + if (offset < 0) offset = 0; + if (offset > 1) offset = 1; + iirSample1CL = (iirSample1CL * (1 - (offset * iirAmount))) + (inputSampleL * (offset * iirAmount)); + lows = iirSample1CL; + inputSampleL -= lows; + temp = lows; + if (lows < 0) {lows = -sin(-lows*1.5707963267949);} + if (lows > 0) {lows = sin(lows*1.5707963267949);} + lows -= temp; + inputSampleL += lows; + inputSampleL *= gaintrim; + + if (tight > 0) offset = (1 - tight) + (fabs(inputSampleL)*tight); + else offset = (1 + tight) + ((1-fabs(inputSampleL))*tight); + if (offset < 0) offset = 0; + if (offset > 1) offset = 1; + iirSample1DL = (iirSample1DL * (1 - (offset * iirAmount))) + (inputSampleL * (offset * iirAmount)); + lows = iirSample1DL; + inputSampleL -= lows; + temp = lows; + if (lows < 0) {lows = -sin(-lows*1.5707963267949);} + if (lows > 0) {lows = sin(lows*1.5707963267949);} + lows -= temp; + inputSampleL += lows; + inputSampleL *= gaintrim; + + if (tight > 0) offset = (1 - tight) + (fabs(inputSampleL)*tight); + else offset = (1 + tight) + ((1-fabs(inputSampleL))*tight); + if (offset < 0) offset = 0; + if (offset > 1) offset = 1; + iirSample1EL = (iirSample1EL * (1 - (offset * iirAmount))) + (inputSampleL * (offset * iirAmount)); + lows = iirSample1EL; + inputSampleL -= lows; + temp = lows; + if (lows < 0) {lows = -sin(-lows*1.5707963267949);} + if (lows > 0) {lows = sin(lows*1.5707963267949);} + lows -= temp; + inputSampleL += lows; + inputSampleL *= gaintrim; + //end left channel + + //begin right channel + if (tight > 0) offset = (1 - tight) + (fabs(inputSampleR)*tight); + else offset = (1 + tight) + ((1-fabs(inputSampleR))*tight); + if (offset < 0) offset = 0; + if (offset > 1) offset = 1; + iirSample1AR = (iirSample1AR * (1 - (offset * iirAmount))) + (inputSampleR * (offset * iirAmount)); + lows = iirSample1AR; + inputSampleR -= lows; + temp = lows; + if (lows < 0) {lows = -sin(-lows*1.5707963267949);} + if (lows > 0) {lows = sin(lows*1.5707963267949);} + lows -= temp; + inputSampleR += lows; + inputSampleR *= gaintrim; + + if (tight > 0) offset = (1 - tight) + (fabs(inputSampleR)*tight); + else offset = (1 + tight) + ((1-fabs(inputSampleR))*tight); + if (offset < 0) offset = 0; + if (offset > 1) offset = 1; + iirSample1BR = (iirSample1BR * (1 - (offset * iirAmount))) + (inputSampleR * (offset * iirAmount)); + lows = iirSample1BR; + inputSampleR -= lows; + temp = lows; + if (lows < 0) {lows = -sin(-lows*1.5707963267949);} + if (lows > 0) {lows = sin(lows*1.5707963267949);} + lows -= temp; + inputSampleR += lows; + inputSampleR *= gaintrim; + + if (tight > 0) offset = (1 - tight) + (fabs(inputSampleR)*tight); + else offset = (1 + tight) + ((1-fabs(inputSampleR))*tight); + if (offset < 0) offset = 0; + if (offset > 1) offset = 1; + iirSample1CR = (iirSample1CR * (1 - (offset * iirAmount))) + (inputSampleR * (offset * iirAmount)); + lows = iirSample1CR; + inputSampleR -= lows; + temp = lows; + if (lows < 0) {lows = -sin(-lows*1.5707963267949);} + if (lows > 0) {lows = sin(lows*1.5707963267949);} + lows -= temp; + inputSampleR += lows; + inputSampleR *= gaintrim; + + if (tight > 0) offset = (1 - tight) + (fabs(inputSampleR)*tight); + else offset = (1 + tight) + ((1-fabs(inputSampleR))*tight); + if (offset < 0) offset = 0; + if (offset > 1) offset = 1; + iirSample1DR = (iirSample1DR * (1 - (offset * iirAmount))) + (inputSampleR * (offset * iirAmount)); + lows = iirSample1DR; + inputSampleR -= lows; + temp = lows; + if (lows < 0) {lows = -sin(-lows*1.5707963267949);} + if (lows > 0) {lows = sin(lows*1.5707963267949);} + lows -= temp; + inputSampleR += lows; + inputSampleR *= gaintrim; + + if (tight > 0) offset = (1 - tight) + (fabs(inputSampleR)*tight); + else offset = (1 + tight) + ((1-fabs(inputSampleR))*tight); + if (offset < 0) offset = 0; + if (offset > 1) offset = 1; + iirSample1ER = (iirSample1ER * (1 - (offset * iirAmount))) + (inputSampleR * (offset * iirAmount)); + lows = iirSample1ER; + inputSampleR -= lows; + temp = lows; + if (lows < 0) {lows = -sin(-lows*1.5707963267949);} + if (lows > 0) {lows = sin(lows*1.5707963267949);} + lows -= temp; + inputSampleR += lows; + inputSampleR *= gaintrim; + //end right channel + + if (inputSampleL > 1.0) inputSampleL = 1.0; + if (inputSampleL < -1.0) inputSampleL = -1.0; + bridgerectifier = fabs(inputSampleL)*1.57079633; + bridgerectifier = sin(bridgerectifier)*1.57079633; + bridgerectifier = (fabs(inputSampleL)*(1-density))+(bridgerectifier*density); + bridgerectifier = sin(bridgerectifier); + if (inputSampleL > 0) inputSampleL = (inputSampleL*(1-density))+(bridgerectifier*density); + else inputSampleL = (inputSampleL*(1-density))-(bridgerectifier*density); + //drive section, left + + if (inputSampleR > 1.0) inputSampleR = 1.0; + if (inputSampleR < -1.0) inputSampleR = -1.0; + bridgerectifier = fabs(inputSampleR)*1.57079633; + bridgerectifier = sin(bridgerectifier)*1.57079633; + bridgerectifier = (fabs(inputSampleR)*(1-density))+(bridgerectifier*density); + bridgerectifier = sin(bridgerectifier); + if (inputSampleR > 0) inputSampleR = (inputSampleR*(1-density))+(bridgerectifier*density); + else inputSampleR = (inputSampleR*(1-density))-(bridgerectifier*density); + //drive section, right + + if (wet !=1.0) { + inputSampleL = (inputSampleL * wet) + (drySampleL * dry); + inputSampleR = (inputSampleR * wet) + (drySampleR * dry); + } + + //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 Floor::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 setting = pow(A,2); + double iirAmount = (setting/4.0)/overallscale; + double tight = -1.0; + double gaintrim = 1.0 + (setting/4.0); + double offset; + double lows; + double density = B; + double bridgerectifier; + double temp; + iirAmount += (iirAmount * tight * tight); + tight /= 3.0; + if (iirAmount <= 0.0) iirAmount = 0.0; + if (iirAmount > 1.0) iirAmount = 1.0; + double wet = C; + double dry = 1.0-wet; + + 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 + long double drySampleL = inputSampleL; + long double drySampleR = inputSampleR; + + //begin left channel + if (tight > 0) offset = (1 - tight) + (fabs(inputSampleL)*tight); + else offset = (1 + tight) + ((1-fabs(inputSampleL))*tight); + if (offset < 0) offset = 0; + if (offset > 1) offset = 1; + iirSample1AL = (iirSample1AL * (1 - (offset * iirAmount))) + (inputSampleL * (offset * iirAmount)); + lows = iirSample1AL; + inputSampleL -= lows; + temp = lows; + if (lows < 0) {lows = -sin(-lows*1.5707963267949);} + if (lows > 0) {lows = sin(lows*1.5707963267949);} + lows -= temp; + inputSampleL += lows; + inputSampleL *= gaintrim; + + if (tight > 0) offset = (1 - tight) + (fabs(inputSampleL)*tight); + else offset = (1 + tight) + ((1-fabs(inputSampleL))*tight); + if (offset < 0) offset = 0; + if (offset > 1) offset = 1; + iirSample1BL = (iirSample1BL * (1 - (offset * iirAmount))) + (inputSampleL * (offset * iirAmount)); + lows = iirSample1BL; + inputSampleL -= lows; + temp = lows; + if (lows < 0) {lows = -sin(-lows*1.5707963267949);} + if (lows > 0) {lows = sin(lows*1.5707963267949);} + lows -= temp; + inputSampleL += lows; + inputSampleL *= gaintrim; + + if (tight > 0) offset = (1 - tight) + (fabs(inputSampleL)*tight); + else offset = (1 + tight) + ((1-fabs(inputSampleL))*tight); + if (offset < 0) offset = 0; + if (offset > 1) offset = 1; + iirSample1CL = (iirSample1CL * (1 - (offset * iirAmount))) + (inputSampleL * (offset * iirAmount)); + lows = iirSample1CL; + inputSampleL -= lows; + temp = lows; + if (lows < 0) {lows = -sin(-lows*1.5707963267949);} + if (lows > 0) {lows = sin(lows*1.5707963267949);} + lows -= temp; + inputSampleL += lows; + inputSampleL *= gaintrim; + + if (tight > 0) offset = (1 - tight) + (fabs(inputSampleL)*tight); + else offset = (1 + tight) + ((1-fabs(inputSampleL))*tight); + if (offset < 0) offset = 0; + if (offset > 1) offset = 1; + iirSample1DL = (iirSample1DL * (1 - (offset * iirAmount))) + (inputSampleL * (offset * iirAmount)); + lows = iirSample1DL; + inputSampleL -= lows; + temp = lows; + if (lows < 0) {lows = -sin(-lows*1.5707963267949);} + if (lows > 0) {lows = sin(lows*1.5707963267949);} + lows -= temp; + inputSampleL += lows; + inputSampleL *= gaintrim; + + if (tight > 0) offset = (1 - tight) + (fabs(inputSampleL)*tight); + else offset = (1 + tight) + ((1-fabs(inputSampleL))*tight); + if (offset < 0) offset = 0; + if (offset > 1) offset = 1; + iirSample1EL = (iirSample1EL * (1 - (offset * iirAmount))) + (inputSampleL * (offset * iirAmount)); + lows = iirSample1EL; + inputSampleL -= lows; + temp = lows; + if (lows < 0) {lows = -sin(-lows*1.5707963267949);} + if (lows > 0) {lows = sin(lows*1.5707963267949);} + lows -= temp; + inputSampleL += lows; + inputSampleL *= gaintrim; + //end left channel + + //begin right channel + if (tight > 0) offset = (1 - tight) + (fabs(inputSampleR)*tight); + else offset = (1 + tight) + ((1-fabs(inputSampleR))*tight); + if (offset < 0) offset = 0; + if (offset > 1) offset = 1; + iirSample1AR = (iirSample1AR * (1 - (offset * iirAmount))) + (inputSampleR * (offset * iirAmount)); + lows = iirSample1AR; + inputSampleR -= lows; + temp = lows; + if (lows < 0) {lows = -sin(-lows*1.5707963267949);} + if (lows > 0) {lows = sin(lows*1.5707963267949);} + lows -= temp; + inputSampleR += lows; + inputSampleR *= gaintrim; + + if (tight > 0) offset = (1 - tight) + (fabs(inputSampleR)*tight); + else offset = (1 + tight) + ((1-fabs(inputSampleR))*tight); + if (offset < 0) offset = 0; + if (offset > 1) offset = 1; + iirSample1BR = (iirSample1BR * (1 - (offset * iirAmount))) + (inputSampleR * (offset * iirAmount)); + lows = iirSample1BR; + inputSampleR -= lows; + temp = lows; + if (lows < 0) {lows = -sin(-lows*1.5707963267949);} + if (lows > 0) {lows = sin(lows*1.5707963267949);} + lows -= temp; + inputSampleR += lows; + inputSampleR *= gaintrim; + + if (tight > 0) offset = (1 - tight) + (fabs(inputSampleR)*tight); + else offset = (1 + tight) + ((1-fabs(inputSampleR))*tight); + if (offset < 0) offset = 0; + if (offset > 1) offset = 1; + iirSample1CR = (iirSample1CR * (1 - (offset * iirAmount))) + (inputSampleR * (offset * iirAmount)); + lows = iirSample1CR; + inputSampleR -= lows; + temp = lows; + if (lows < 0) {lows = -sin(-lows*1.5707963267949);} + if (lows > 0) {lows = sin(lows*1.5707963267949);} + lows -= temp; + inputSampleR += lows; + inputSampleR *= gaintrim; + + if (tight > 0) offset = (1 - tight) + (fabs(inputSampleR)*tight); + else offset = (1 + tight) + ((1-fabs(inputSampleR))*tight); + if (offset < 0) offset = 0; + if (offset > 1) offset = 1; + iirSample1DR = (iirSample1DR * (1 - (offset * iirAmount))) + (inputSampleR * (offset * iirAmount)); + lows = iirSample1DR; + inputSampleR -= lows; + temp = lows; + if (lows < 0) {lows = -sin(-lows*1.5707963267949);} + if (lows > 0) {lows = sin(lows*1.5707963267949);} + lows -= temp; + inputSampleR += lows; + inputSampleR *= gaintrim; + + if (tight > 0) offset = (1 - tight) + (fabs(inputSampleR)*tight); + else offset = (1 + tight) + ((1-fabs(inputSampleR))*tight); + if (offset < 0) offset = 0; + if (offset > 1) offset = 1; + iirSample1ER = (iirSample1ER * (1 - (offset * iirAmount))) + (inputSampleR * (offset * iirAmount)); + lows = iirSample1ER; + inputSampleR -= lows; + temp = lows; + if (lows < 0) {lows = -sin(-lows*1.5707963267949);} + if (lows > 0) {lows = sin(lows*1.5707963267949);} + lows -= temp; + inputSampleR += lows; + inputSampleR *= gaintrim; + //end right channel + + if (inputSampleL > 1.0) inputSampleL = 1.0; + if (inputSampleL < -1.0) inputSampleL = -1.0; + bridgerectifier = fabs(inputSampleL)*1.57079633; + bridgerectifier = sin(bridgerectifier)*1.57079633; + bridgerectifier = (fabs(inputSampleL)*(1-density))+(bridgerectifier*density); + bridgerectifier = sin(bridgerectifier); + if (inputSampleL > 0) inputSampleL = (inputSampleL*(1-density))+(bridgerectifier*density); + else inputSampleL = (inputSampleL*(1-density))-(bridgerectifier*density); + //drive section, left + + if (inputSampleR > 1.0) inputSampleR = 1.0; + if (inputSampleR < -1.0) inputSampleR = -1.0; + bridgerectifier = fabs(inputSampleR)*1.57079633; + bridgerectifier = sin(bridgerectifier)*1.57079633; + bridgerectifier = (fabs(inputSampleR)*(1-density))+(bridgerectifier*density); + bridgerectifier = sin(bridgerectifier); + if (inputSampleR > 0) inputSampleR = (inputSampleR*(1-density))+(bridgerectifier*density); + else inputSampleR = (inputSampleR*(1-density))-(bridgerectifier*density); + //drive section, right + + if (wet !=1.0) { + inputSampleL = (inputSampleL * wet) + (drySampleL * dry); + inputSampleR = (inputSampleR * wet) + (drySampleR * dry); + } + + //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. +} |