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author | airwindows <jinx6568@sover.net> | 2018-03-04 19:17:09 -0500 |
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committer | airwindows <jinx6568@sover.net> | 2018-03-04 19:17:09 -0500 |
commit | f2951d9baa5c7755f278668e8e79f508ab7c5ac3 (patch) | |
tree | 8d4d9ec4e39fd371906167479b32a80ef43b2ae7 /plugins/WinVST/HardVacuum/HardVacuumProc.cpp | |
parent | bb21995711adcd0ebdc62697480c2f8981b61162 (diff) | |
download | airwindows-lv2-port-f2951d9baa5c7755f278668e8e79f508ab7c5ac3.tar.gz airwindows-lv2-port-f2951d9baa5c7755f278668e8e79f508ab7c5ac3.tar.bz2 airwindows-lv2-port-f2951d9baa5c7755f278668e8e79f508ab7c5ac3.zip |
Hard Vacuum
Diffstat (limited to 'plugins/WinVST/HardVacuum/HardVacuumProc.cpp')
-rwxr-xr-x | plugins/WinVST/HardVacuum/HardVacuumProc.cpp | 407 |
1 files changed, 407 insertions, 0 deletions
diff --git a/plugins/WinVST/HardVacuum/HardVacuumProc.cpp b/plugins/WinVST/HardVacuum/HardVacuumProc.cpp new file mode 100755 index 0000000..f8fe4fd --- /dev/null +++ b/plugins/WinVST/HardVacuum/HardVacuumProc.cpp @@ -0,0 +1,407 @@ +/* ======================================== + * HardVacuum - HardVacuum.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __HardVacuum_H +#include "HardVacuum.h" +#endif + +void HardVacuum::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames) +{ + float* in1 = inputs[0]; + float* in2 = inputs[1]; + float* out1 = outputs[0]; + float* out2 = outputs[1]; + + double multistage = A*2.0; + if (multistage > 1) multistage *= multistage; + //WE MAKE LOUD NOISE! RAWWWK! + double countdown; + double warmth = B; + double invwarmth = 1.0-warmth; + warmth /= 1.57079633; + double aura = C*3.1415926; + double out = D; + double wet = E; + double dry = 1.0-wet; + double drive; + double positive; + double negative; + double bridgerectifierL; + double bridgerectifierR; + double skewL; + double skewR; + + float fpTemp; + long double fpOld = 0.618033988749894848204586; //golden ratio! + long double fpNew = 1.0 - fpOld; + + double drySampleL; + double drySampleR; + long double inputSampleL; + long double inputSampleR; + + 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; + + skewL = (inputSampleL - lastSampleL); + skewR = (inputSampleR - lastSampleR); + lastSampleL = inputSampleL; + lastSampleR = inputSampleR; + //skew will be direction/angle + bridgerectifierL = fabs(skewL); + bridgerectifierR = fabs(skewR); + if (bridgerectifierL > 3.1415926) bridgerectifierL = 3.1415926; + if (bridgerectifierR > 3.1415926) bridgerectifierR = 3.1415926; + //for skew we want it to go to zero effect again, so we use full range of the sine + + bridgerectifierL = sin(bridgerectifierL); + bridgerectifierR = sin(bridgerectifierR); + if (skewL > 0) skewL = bridgerectifierL*aura; + else skewL = -bridgerectifierL*aura; + if (skewR > 0) skewR = bridgerectifierR*aura; + else skewR = -bridgerectifierR*aura; + //skew is now sined and clamped and then re-amplified again + skewL *= inputSampleL; + skewR *= inputSampleR; + //cools off sparkliness and crossover distortion + skewL *= 1.557079633; + skewR *= 1.557079633; + //crank up the gain on this so we can make it sing + //We're doing all this here so skew isn't incremented by each stage + + countdown = multistage; + //begin the torture + + while (countdown > 0) + { + if (countdown > 1.0) drive = 1.557079633; + else drive = countdown * (1.0+(0.557079633*invwarmth)); + //full crank stages followed by the proportional one + //whee. 1 at full warmth to 1.5570etc at no warmth + positive = drive - warmth; + negative = drive + warmth; + //set up things so we can do repeated iterations, assuming that + //wet is always going to be 0-1 as in the previous plug. + bridgerectifierL = fabs(inputSampleL); + bridgerectifierR = fabs(inputSampleR); + bridgerectifierL += skewL; + bridgerectifierR += skewR; + //apply it here so we don't overload + if (bridgerectifierL > 1.57079633) bridgerectifierL = 1.57079633; + if (bridgerectifierR > 1.57079633) bridgerectifierR = 1.57079633; + bridgerectifierL = sin(bridgerectifierL); + bridgerectifierR = sin(bridgerectifierR); + //the distortion section. + bridgerectifierL *= drive; + bridgerectifierR *= drive; + bridgerectifierL += skewL; + bridgerectifierR += skewR; + //again + if (bridgerectifierL > 1.57079633) bridgerectifierL = 1.57079633; + if (bridgerectifierR > 1.57079633) bridgerectifierR = 1.57079633; + bridgerectifierL = sin(bridgerectifierL); + bridgerectifierR = sin(bridgerectifierR); + if (inputSampleL > 0) + { + inputSampleL = (inputSampleL*(1-positive+skewL))+(bridgerectifierL*(positive+skewL)); + } + else + { + inputSampleL = (inputSampleL*(1-negative+skewL))-(bridgerectifierL*(negative+skewL)); + } + if (inputSampleR > 0) + { + inputSampleR = (inputSampleR*(1-positive+skewR))+(bridgerectifierR*(positive+skewR)); + } + else + { + inputSampleR = (inputSampleR*(1-negative+skewR))-(bridgerectifierR*(negative+skewR)); + } + //blend according to positive and negative controls + countdown -= 1.0; + //step down a notch and repeat. + } + + if (out != 1.0) { + inputSampleL *= out; + inputSampleR *= out; + } + + 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 HardVacuum::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames) +{ + double* in1 = inputs[0]; + double* in2 = inputs[1]; + double* out1 = outputs[0]; + double* out2 = outputs[1]; + + double multistage = A*2.0; + if (multistage > 1) multistage *= multistage; + //WE MAKE LOUD NOISE! RAWWWK! + double countdown; + double warmth = B; + double invwarmth = 1.0-warmth; + warmth /= 1.57079633; + double aura = C*3.1415926; + double out = D; + double wet = E; + double dry = 1.0-wet; + double drive; + double positive; + double negative; + double bridgerectifierL; + double bridgerectifierR; + double skewL; + double skewR; + + double fpTemp; + long double fpOld = 0.618033988749894848204586; //golden ratio! + long double fpNew = 1.0 - fpOld; + + double drySampleL; + double drySampleR; + long double inputSampleL; + long double inputSampleR; + + + 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; + + skewL = (inputSampleL - lastSampleL); + skewR = (inputSampleR - lastSampleR); + lastSampleL = inputSampleL; + lastSampleR = inputSampleR; + //skew will be direction/angle + bridgerectifierL = fabs(skewL); + bridgerectifierR = fabs(skewR); + if (bridgerectifierL > 3.1415926) bridgerectifierL = 3.1415926; + if (bridgerectifierR > 3.1415926) bridgerectifierR = 3.1415926; + //for skew we want it to go to zero effect again, so we use full range of the sine + + bridgerectifierL = sin(bridgerectifierL); + bridgerectifierR = sin(bridgerectifierR); + if (skewL > 0) skewL = bridgerectifierL*aura; + else skewL = -bridgerectifierL*aura; + if (skewR > 0) skewR = bridgerectifierR*aura; + else skewR = -bridgerectifierR*aura; + //skew is now sined and clamped and then re-amplified again + skewL *= inputSampleL; + skewR *= inputSampleR; + //cools off sparkliness and crossover distortion + skewL *= 1.557079633; + skewR *= 1.557079633; + //crank up the gain on this so we can make it sing + //We're doing all this here so skew isn't incremented by each stage + + countdown = multistage; + //begin the torture + + while (countdown > 0) + { + if (countdown > 1.0) drive = 1.557079633; + else drive = countdown * (1.0+(0.557079633*invwarmth)); + //full crank stages followed by the proportional one + //whee. 1 at full warmth to 1.5570etc at no warmth + positive = drive - warmth; + negative = drive + warmth; + //set up things so we can do repeated iterations, assuming that + //wet is always going to be 0-1 as in the previous plug. + bridgerectifierL = fabs(inputSampleL); + bridgerectifierR = fabs(inputSampleR); + bridgerectifierL += skewL; + bridgerectifierR += skewR; + //apply it here so we don't overload + if (bridgerectifierL > 1.57079633) bridgerectifierL = 1.57079633; + if (bridgerectifierR > 1.57079633) bridgerectifierR = 1.57079633; + bridgerectifierL = sin(bridgerectifierL); + bridgerectifierR = sin(bridgerectifierR); + //the distortion section. + bridgerectifierL *= drive; + bridgerectifierR *= drive; + bridgerectifierL += skewL; + bridgerectifierR += skewR; + //again + if (bridgerectifierL > 1.57079633) bridgerectifierL = 1.57079633; + if (bridgerectifierR > 1.57079633) bridgerectifierR = 1.57079633; + bridgerectifierL = sin(bridgerectifierL); + bridgerectifierR = sin(bridgerectifierR); + if (inputSampleL > 0) + { + inputSampleL = (inputSampleL*(1-positive+skewL))+(bridgerectifierL*(positive+skewL)); + } + else + { + inputSampleL = (inputSampleL*(1-negative+skewL))-(bridgerectifierL*(negative+skewL)); + } + if (inputSampleR > 0) + { + inputSampleR = (inputSampleR*(1-positive+skewR))+(bridgerectifierR*(positive+skewR)); + } + else + { + inputSampleR = (inputSampleR*(1-negative+skewR))-(bridgerectifierR*(negative+skewR)); + } + //blend according to positive and negative controls + countdown -= 1.0; + //step down a notch and repeat. + } + + if (out != 1.0) { + inputSampleL *= out; + inputSampleR *= out; + } + + 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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