diff options
Diffstat (limited to 'plugins/WinVST/DrumSlam/DrumSlamProc.cpp')
| -rwxr-xr-x | plugins/WinVST/DrumSlam/DrumSlamProc.cpp | 494 |
1 files changed, 494 insertions, 0 deletions
diff --git a/plugins/WinVST/DrumSlam/DrumSlamProc.cpp b/plugins/WinVST/DrumSlam/DrumSlamProc.cpp new file mode 100755 index 0000000..171b353 --- /dev/null +++ b/plugins/WinVST/DrumSlam/DrumSlamProc.cpp @@ -0,0 +1,494 @@ +/* ======================================== + * DrumSlam - DrumSlam.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __DrumSlam_H +#include "DrumSlam.h" +#endif + +void DrumSlam::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 iirAmountL = 0.0819; + iirAmountL /= overallscale; + double iirAmountH = 0.377933067; + iirAmountH /= overallscale; + double drive = (A*3.0)+1.0; + double out = B; + double wet = C; + double dry = 1.0 - wet; + long double fpOld = 0.618033988749894848204586; //golden ratio! + long double fpNew = 1.0 - fpOld; + + while (--sampleFrames >= 0) + { + long double inputSampleL = *in1; + long double 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. + } + long double drySampleL = inputSampleL; + long double drySampleR = inputSampleR; + long double lowSampleL; + long double lowSampleR; + long double midSampleL; + long double midSampleR; + long double highSampleL; + long double highSampleR; + + + inputSampleL *= drive; + inputSampleR *= drive; + + if (fpFlip) + { + iirSampleAL = (iirSampleAL * (1 - iirAmountL)) + (inputSampleL * iirAmountL); + iirSampleBL = (iirSampleBL * (1 - iirAmountL)) + (iirSampleAL * iirAmountL); + lowSampleL = iirSampleBL; + + iirSampleAR = (iirSampleAR * (1 - iirAmountL)) + (inputSampleR * iirAmountL); + iirSampleBR = (iirSampleBR * (1 - iirAmountL)) + (iirSampleAR * iirAmountL); + lowSampleR = iirSampleBR; + + iirSampleEL = (iirSampleEL * (1 - iirAmountH)) + (inputSampleL * iirAmountH); + iirSampleFL = (iirSampleFL * (1 - iirAmountH)) + (iirSampleEL * iirAmountH); + midSampleL = iirSampleFL - iirSampleBL; + + iirSampleER = (iirSampleER * (1 - iirAmountH)) + (inputSampleR * iirAmountH); + iirSampleFR = (iirSampleFR * (1 - iirAmountH)) + (iirSampleER * iirAmountH); + midSampleR = iirSampleFR - iirSampleBR; + + highSampleL = inputSampleL - iirSampleFL; + highSampleR = inputSampleR - iirSampleFR; + } + else + { + iirSampleCL = (iirSampleCL * (1 - iirAmountL)) + (inputSampleL * iirAmountL); + iirSampleDL = (iirSampleDL * (1 - iirAmountL)) + (iirSampleCL * iirAmountL); + lowSampleL = iirSampleDL; + + iirSampleCR = (iirSampleCR * (1 - iirAmountL)) + (inputSampleR * iirAmountL); + iirSampleDR = (iirSampleDR * (1 - iirAmountL)) + (iirSampleCR * iirAmountL); + lowSampleR = iirSampleDR; + + iirSampleGL = (iirSampleGL * (1 - iirAmountH)) + (inputSampleL * iirAmountH); + iirSampleHL = (iirSampleHL * (1 - iirAmountH)) + (iirSampleGL * iirAmountH); + midSampleL = iirSampleHL - iirSampleDL; + + iirSampleGR = (iirSampleGR * (1 - iirAmountH)) + (inputSampleR * iirAmountH); + iirSampleHR = (iirSampleHR * (1 - iirAmountH)) + (iirSampleGR * iirAmountH); + midSampleR = iirSampleHR - iirSampleDR; + + highSampleL = inputSampleL - iirSampleHL; + highSampleR = inputSampleR - iirSampleHR; + } + //generate the tone bands we're using + if (lowSampleL > 1.0) {lowSampleL = 1.0;} + if (lowSampleL < -1.0) {lowSampleL = -1.0;} + if (lowSampleR > 1.0) {lowSampleR = 1.0;} + if (lowSampleR < -1.0) {lowSampleR = -1.0;} + lowSampleL -= (lowSampleL * (fabs(lowSampleL) * 0.448) * (fabs(lowSampleL) * 0.448) ); + lowSampleR -= (lowSampleR * (fabs(lowSampleR) * 0.448) * (fabs(lowSampleR) * 0.448) ); + lowSampleL *= drive; + lowSampleR *= drive; + + if (highSampleL > 1.0) {highSampleL = 1.0;} + if (highSampleL < -1.0) {highSampleL = -1.0;} + if (highSampleR > 1.0) {highSampleR = 1.0;} + if (highSampleR < -1.0) {highSampleR = -1.0;} + highSampleL -= (highSampleL * (fabs(highSampleL) * 0.599) * (fabs(highSampleL) * 0.599) ); + highSampleR -= (highSampleR * (fabs(highSampleR) * 0.599) * (fabs(highSampleR) * 0.599) ); + highSampleL *= drive; + highSampleR *= drive; + + midSampleL = midSampleL * drive; + midSampleR = midSampleR * drive; + + long double skew = (midSampleL - lastSampleL); + lastSampleL = midSampleL; + //skew will be direction/angle + long double bridgerectifier = fabs(skew); + if (bridgerectifier > 3.1415926) bridgerectifier = 3.1415926; + //for skew we want it to go to zero effect again, so we use full range of the sine + bridgerectifier = sin(bridgerectifier); + if (skew > 0) skew = bridgerectifier*3.1415926; + else skew = -bridgerectifier*3.1415926; + //skew is now sined and clamped and then re-amplified again + skew *= midSampleL; + //cools off sparkliness and crossover distortion + skew *= 1.557079633; + //crank up the gain on this so we can make it sing + bridgerectifier = fabs(midSampleL); + bridgerectifier += skew; + if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; + bridgerectifier = sin(bridgerectifier); + bridgerectifier *= drive; + bridgerectifier += skew; + if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; + bridgerectifier = sin(bridgerectifier); + if (midSampleL > 0) + { + midSampleL = bridgerectifier; + } + else + { + midSampleL = -bridgerectifier; + } + //blend according to positive and negative controls, left + + skew = (midSampleR - lastSampleR); + lastSampleR = midSampleR; + //skew will be direction/angle + bridgerectifier = fabs(skew); + if (bridgerectifier > 3.1415926) bridgerectifier = 3.1415926; + //for skew we want it to go to zero effect again, so we use full range of the sine + bridgerectifier = sin(bridgerectifier); + if (skew > 0) skew = bridgerectifier*3.1415926; + else skew = -bridgerectifier*3.1415926; + //skew is now sined and clamped and then re-amplified again + skew *= midSampleR; + //cools off sparkliness and crossover distortion + skew *= 1.557079633; + //crank up the gain on this so we can make it sing + bridgerectifier = fabs(midSampleR); + bridgerectifier += skew; + if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; + bridgerectifier = sin(bridgerectifier); + bridgerectifier *= drive; + bridgerectifier += skew; + if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; + bridgerectifier = sin(bridgerectifier); + if (midSampleR > 0) + { + midSampleR = bridgerectifier; + } + else + { + midSampleR = -bridgerectifier; + } + //blend according to positive and negative controls, right + + inputSampleL = ((lowSampleL + midSampleL + highSampleL)/drive)*out; + inputSampleR = ((lowSampleR + midSampleR + highSampleR)/drive)*out; + + if (wet !=1.0) { + inputSampleL = (inputSampleL * wet) + (drySampleL * dry); + inputSampleR = (inputSampleR * wet) + (drySampleR * dry); + } + + //noise shaping to 32-bit floating point + float fpTemp; + 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 DrumSlam::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 iirAmountL = 0.0819; + iirAmountL /= overallscale; + double iirAmountH = 0.377933067; + iirAmountH /= overallscale; + double drive = (A*3.0)+1.0; + double out = B; + double wet = C; + double dry = 1.0 - wet; + long double fpOld = 0.618033988749894848204586; //golden ratio! + long double fpNew = 1.0 - fpOld; + + while (--sampleFrames >= 0) + { + long double inputSampleL = *in1; + long double 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. + } + long double drySampleL = inputSampleL; + long double drySampleR = inputSampleR; + long double lowSampleL; + long double lowSampleR; + long double midSampleL; + long double midSampleR; + long double highSampleL; + long double highSampleR; + + + inputSampleL *= drive; + inputSampleR *= drive; + + if (fpFlip) + { + iirSampleAL = (iirSampleAL * (1 - iirAmountL)) + (inputSampleL * iirAmountL); + iirSampleBL = (iirSampleBL * (1 - iirAmountL)) + (iirSampleAL * iirAmountL); + lowSampleL = iirSampleBL; + + iirSampleAR = (iirSampleAR * (1 - iirAmountL)) + (inputSampleR * iirAmountL); + iirSampleBR = (iirSampleBR * (1 - iirAmountL)) + (iirSampleAR * iirAmountL); + lowSampleR = iirSampleBR; + + iirSampleEL = (iirSampleEL * (1 - iirAmountH)) + (inputSampleL * iirAmountH); + iirSampleFL = (iirSampleFL * (1 - iirAmountH)) + (iirSampleEL * iirAmountH); + midSampleL = iirSampleFL - iirSampleBL; + + iirSampleER = (iirSampleER * (1 - iirAmountH)) + (inputSampleR * iirAmountH); + iirSampleFR = (iirSampleFR * (1 - iirAmountH)) + (iirSampleER * iirAmountH); + midSampleR = iirSampleFR - iirSampleBR; + + highSampleL = inputSampleL - iirSampleFL; + highSampleR = inputSampleR - iirSampleFR; + } + else + { + iirSampleCL = (iirSampleCL * (1 - iirAmountL)) + (inputSampleL * iirAmountL); + iirSampleDL = (iirSampleDL * (1 - iirAmountL)) + (iirSampleCL * iirAmountL); + lowSampleL = iirSampleDL; + + iirSampleCR = (iirSampleCR * (1 - iirAmountL)) + (inputSampleR * iirAmountL); + iirSampleDR = (iirSampleDR * (1 - iirAmountL)) + (iirSampleCR * iirAmountL); + lowSampleR = iirSampleDR; + + iirSampleGL = (iirSampleGL * (1 - iirAmountH)) + (inputSampleL * iirAmountH); + iirSampleHL = (iirSampleHL * (1 - iirAmountH)) + (iirSampleGL * iirAmountH); + midSampleL = iirSampleHL - iirSampleDL; + + iirSampleGR = (iirSampleGR * (1 - iirAmountH)) + (inputSampleR * iirAmountH); + iirSampleHR = (iirSampleHR * (1 - iirAmountH)) + (iirSampleGR * iirAmountH); + midSampleR = iirSampleHR - iirSampleDR; + + highSampleL = inputSampleL - iirSampleHL; + highSampleR = inputSampleR - iirSampleHR; + } + //generate the tone bands we're using + if (lowSampleL > 1.0) {lowSampleL = 1.0;} + if (lowSampleL < -1.0) {lowSampleL = -1.0;} + if (lowSampleR > 1.0) {lowSampleR = 1.0;} + if (lowSampleR < -1.0) {lowSampleR = -1.0;} + lowSampleL -= (lowSampleL * (fabs(lowSampleL) * 0.448) * (fabs(lowSampleL) * 0.448) ); + lowSampleR -= (lowSampleR * (fabs(lowSampleR) * 0.448) * (fabs(lowSampleR) * 0.448) ); + lowSampleL *= drive; + lowSampleR *= drive; + + if (highSampleL > 1.0) {highSampleL = 1.0;} + if (highSampleL < -1.0) {highSampleL = -1.0;} + if (highSampleR > 1.0) {highSampleR = 1.0;} + if (highSampleR < -1.0) {highSampleR = -1.0;} + highSampleL -= (highSampleL * (fabs(highSampleL) * 0.599) * (fabs(highSampleL) * 0.599) ); + highSampleR -= (highSampleR * (fabs(highSampleR) * 0.599) * (fabs(highSampleR) * 0.599) ); + highSampleL *= drive; + highSampleR *= drive; + + midSampleL = midSampleL * drive; + midSampleR = midSampleR * drive; + + long double skew = (midSampleL - lastSampleL); + lastSampleL = midSampleL; + //skew will be direction/angle + long double bridgerectifier = fabs(skew); + if (bridgerectifier > 3.1415926) bridgerectifier = 3.1415926; + //for skew we want it to go to zero effect again, so we use full range of the sine + bridgerectifier = sin(bridgerectifier); + if (skew > 0) skew = bridgerectifier*3.1415926; + else skew = -bridgerectifier*3.1415926; + //skew is now sined and clamped and then re-amplified again + skew *= midSampleL; + //cools off sparkliness and crossover distortion + skew *= 1.557079633; + //crank up the gain on this so we can make it sing + bridgerectifier = fabs(midSampleL); + bridgerectifier += skew; + if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; + bridgerectifier = sin(bridgerectifier); + bridgerectifier *= drive; + bridgerectifier += skew; + if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; + bridgerectifier = sin(bridgerectifier); + if (midSampleL > 0) + { + midSampleL = bridgerectifier; + } + else + { + midSampleL = -bridgerectifier; + } + //blend according to positive and negative controls, left + + skew = (midSampleR - lastSampleR); + lastSampleR = midSampleR; + //skew will be direction/angle + bridgerectifier = fabs(skew); + if (bridgerectifier > 3.1415926) bridgerectifier = 3.1415926; + //for skew we want it to go to zero effect again, so we use full range of the sine + bridgerectifier = sin(bridgerectifier); + if (skew > 0) skew = bridgerectifier*3.1415926; + else skew = -bridgerectifier*3.1415926; + //skew is now sined and clamped and then re-amplified again + skew *= midSampleR; + //cools off sparkliness and crossover distortion + skew *= 1.557079633; + //crank up the gain on this so we can make it sing + bridgerectifier = fabs(midSampleR); + bridgerectifier += skew; + if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; + bridgerectifier = sin(bridgerectifier); + bridgerectifier *= drive; + bridgerectifier += skew; + if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; + bridgerectifier = sin(bridgerectifier); + if (midSampleR > 0) + { + midSampleR = bridgerectifier; + } + else + { + midSampleR = -bridgerectifier; + } + //blend according to positive and negative controls, right + + inputSampleL = ((lowSampleL + midSampleL + highSampleL)/drive)*out; + inputSampleR = ((lowSampleR + midSampleR + highSampleR)/drive)*out; + + if (wet !=1.0) { + inputSampleL = (inputSampleL * wet) + (drySampleL * dry); + inputSampleR = (inputSampleR * wet) + (drySampleR * dry); + } + + //noise shaping to 64-bit floating point + double fpTemp; + 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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