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author | Chris Johnson <jinx6568@sover.net> | 2018-07-08 19:30:08 -0400 |
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committer | Chris Johnson <jinx6568@sover.net> | 2018-07-08 19:30:08 -0400 |
commit | 6dd0cc75eef5294133c324ca225275247923cccd (patch) | |
tree | 36adae747e33f75862f594ea675f94608a6ae787 /plugins/MacVST/DrumSlam/source | |
parent | 31d06ef1a29836dbc357a004cb422563c698c88e (diff) | |
download | airwindows-lv2-port-6dd0cc75eef5294133c324ca225275247923cccd.tar.gz airwindows-lv2-port-6dd0cc75eef5294133c324ca225275247923cccd.tar.bz2 airwindows-lv2-port-6dd0cc75eef5294133c324ca225275247923cccd.zip |
DrumSlam
Diffstat (limited to 'plugins/MacVST/DrumSlam/source')
-rwxr-xr-x | plugins/MacVST/DrumSlam/source/DrumSlam.cpp | 159 | ||||
-rwxr-xr-x | plugins/MacVST/DrumSlam/source/DrumSlam.h | 92 | ||||
-rwxr-xr-x | plugins/MacVST/DrumSlam/source/DrumSlamProc.cpp | 494 |
3 files changed, 745 insertions, 0 deletions
diff --git a/plugins/MacVST/DrumSlam/source/DrumSlam.cpp b/plugins/MacVST/DrumSlam/source/DrumSlam.cpp new file mode 100755 index 0000000..8a6ae7b --- /dev/null +++ b/plugins/MacVST/DrumSlam/source/DrumSlam.cpp @@ -0,0 +1,159 @@ +/* ======================================== + * DrumSlam - DrumSlam.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __DrumSlam_H +#include "DrumSlam.h" +#endif + +AudioEffect* createEffectInstance(audioMasterCallback audioMaster) {return new DrumSlam(audioMaster);} + +DrumSlam::DrumSlam(audioMasterCallback audioMaster) : + AudioEffectX(audioMaster, kNumPrograms, kNumParameters) +{ + A = 0.0; + B = 1.0; + C = 1.0; + + iirSampleAL = 0.0; + iirSampleBL = 0.0; + iirSampleCL = 0.0; + iirSampleDL = 0.0; + iirSampleEL = 0.0; + iirSampleFL = 0.0; + iirSampleGL = 0.0; + iirSampleHL = 0.0; + lastSampleL = 0.0; + + iirSampleAR = 0.0; + iirSampleBR = 0.0; + iirSampleCR = 0.0; + iirSampleDR = 0.0; + iirSampleER = 0.0; + iirSampleFR = 0.0; + iirSampleGR = 0.0; + iirSampleHR = 0.0; + lastSampleR = 0.0; + + fpNShapeLA = 0.0; + fpNShapeLB = 0.0; + fpNShapeRA = 0.0; + fpNShapeRB = 0.0; + fpFlip = true; + //this is reset: values being initialized only once. Startup values, whatever they are. + + _canDo.insert("plugAsChannelInsert"); // plug-in can be used as a channel insert effect. + _canDo.insert("plugAsSend"); // plug-in can be used as a send effect. + _canDo.insert("x2in2out"); + setNumInputs(kNumInputs); + setNumOutputs(kNumOutputs); + setUniqueID(kUniqueId); + canProcessReplacing(); // supports output replacing + canDoubleReplacing(); // supports double precision processing + programsAreChunks(true); + vst_strncpy (_programName, "Default", kVstMaxProgNameLen); // default program name +} + +DrumSlam::~DrumSlam() {} +VstInt32 DrumSlam::getVendorVersion () {return 1000;} +void DrumSlam::setProgramName(char *name) {vst_strncpy (_programName, name, kVstMaxProgNameLen);} +void DrumSlam::getProgramName(char *name) {vst_strncpy (name, _programName, kVstMaxProgNameLen);} +//airwindows likes to ignore this stuff. Make your own programs, and make a different plugin rather than +//trying to do versioning and preventing people from using older versions. Maybe they like the old one! + +static float pinParameter(float data) +{ + if (data < 0.0f) return 0.0f; + if (data > 1.0f) return 1.0f; + return data; +} + +VstInt32 DrumSlam::getChunk (void** data, bool isPreset) +{ + float *chunkData = (float *)calloc(kNumParameters, sizeof(float)); + chunkData[0] = A; + chunkData[1] = B; + chunkData[2] = C; + /* Note: The way this is set up, it will break if you manage to save settings on an Intel + machine and load them on a PPC Mac. However, it's fine if you stick to the machine you + started with. */ + + *data = chunkData; + return kNumParameters * sizeof(float); +} + +VstInt32 DrumSlam::setChunk (void* data, VstInt32 byteSize, bool isPreset) +{ + float *chunkData = (float *)data; + A = pinParameter(chunkData[0]); + B = pinParameter(chunkData[1]); + C = pinParameter(chunkData[2]); + /* We're ignoring byteSize as we found it to be a filthy liar */ + + /* calculate any other fields you need here - you could copy in + code from setParameter() here. */ + return 0; +} + +void DrumSlam::setParameter(VstInt32 index, float value) { + switch (index) { + case kParamA: A = value; break; + case kParamB: B = value; break; + case kParamC: C = value; break; + default: throw; // unknown parameter, shouldn't happen! + } +} + +float DrumSlam::getParameter(VstInt32 index) { + switch (index) { + case kParamA: return A; break; + case kParamB: return B; break; + case kParamC: return C; break; + default: break; // unknown parameter, shouldn't happen! + } return 0.0; //we only need to update the relevant name, this is simple to manage +} + +void DrumSlam::getParameterName(VstInt32 index, char *text) { + switch (index) { + case kParamA: vst_strncpy (text, "Drive", kVstMaxParamStrLen); break; + case kParamB: vst_strncpy (text, "Output", kVstMaxParamStrLen); break; + case kParamC: vst_strncpy (text, "Dry/Wet", kVstMaxParamStrLen); break; + default: break; // unknown parameter, shouldn't happen! + } //this is our labels for displaying in the VST host +} + +void DrumSlam::getParameterDisplay(VstInt32 index, char *text) { + switch (index) { + case kParamA: float2string ((A*3.0)+1.0, text, kVstMaxParamStrLen); break; + case kParamB: float2string (B, text, kVstMaxParamStrLen); break; + case kParamC: float2string (C, text, kVstMaxParamStrLen); break; + default: break; // unknown parameter, shouldn't happen! + } //this displays the values and handles 'popups' where it's discrete choices +} + +void DrumSlam::getParameterLabel(VstInt32 index, char *text) { + switch (index) { + case kParamA: vst_strncpy (text, "", kVstMaxParamStrLen); break; + case kParamB: vst_strncpy (text, "", kVstMaxParamStrLen); break; + case kParamC: vst_strncpy (text, "", kVstMaxParamStrLen); break; + default: break; // unknown parameter, shouldn't happen! + } +} + +VstInt32 DrumSlam::canDo(char *text) +{ return (_canDo.find(text) == _canDo.end()) ? -1: 1; } // 1 = yes, -1 = no, 0 = don't know + +bool DrumSlam::getEffectName(char* name) { + vst_strncpy(name, "DrumSlam", kVstMaxProductStrLen); return true; +} + +VstPlugCategory DrumSlam::getPlugCategory() {return kPlugCategEffect;} + +bool DrumSlam::getProductString(char* text) { + vst_strncpy (text, "airwindows DrumSlam", kVstMaxProductStrLen); return true; +} + +bool DrumSlam::getVendorString(char* text) { + vst_strncpy (text, "airwindows", kVstMaxVendorStrLen); return true; +} diff --git a/plugins/MacVST/DrumSlam/source/DrumSlam.h b/plugins/MacVST/DrumSlam/source/DrumSlam.h new file mode 100755 index 0000000..5ec2fda --- /dev/null +++ b/plugins/MacVST/DrumSlam/source/DrumSlam.h @@ -0,0 +1,92 @@ +/* ======================================== + * DrumSlam - DrumSlam.h + * Created 8/12/11 by SPIAdmin + * Copyright (c) 2011 __MyCompanyName__, All rights reserved + * ======================================== */ + +#ifndef __DrumSlam_H +#define __DrumSlam_H + +#ifndef __audioeffect__ +#include "audioeffectx.h" +#endif + +#include <set> +#include <string> +#include <math.h> + +enum { + kParamA = 0, + kParamB = 1, + kParamC = 2, + kNumParameters = 3 +}; // + +const int kNumPrograms = 0; +const int kNumInputs = 2; +const int kNumOutputs = 2; +const unsigned long kUniqueId = 'drsl'; //Change this to what the AU identity is! + +class DrumSlam : + public AudioEffectX +{ +public: + DrumSlam(audioMasterCallback audioMaster); + ~DrumSlam(); + virtual bool getEffectName(char* name); // The plug-in name + virtual VstPlugCategory getPlugCategory(); // The general category for the plug-in + virtual bool getProductString(char* text); // This is a unique plug-in string provided by Steinberg + virtual bool getVendorString(char* text); // Vendor info + virtual VstInt32 getVendorVersion(); // Version number + virtual void processReplacing (float** inputs, float** outputs, VstInt32 sampleFrames); + virtual void processDoubleReplacing (double** inputs, double** outputs, VstInt32 sampleFrames); + virtual void getProgramName(char *name); // read the name from the host + virtual void setProgramName(char *name); // changes the name of the preset displayed in the host + virtual VstInt32 getChunk (void** data, bool isPreset); + virtual VstInt32 setChunk (void* data, VstInt32 byteSize, bool isPreset); + virtual float getParameter(VstInt32 index); // get the parameter value at the specified index + virtual void setParameter(VstInt32 index, float value); // set the parameter at index to value + virtual void getParameterLabel(VstInt32 index, char *text); // label for the parameter (eg dB) + virtual void getParameterName(VstInt32 index, char *text); // name of the parameter + virtual void getParameterDisplay(VstInt32 index, char *text); // text description of the current value + virtual VstInt32 canDo(char *text); +private: + char _programName[kVstMaxProgNameLen + 1]; + std::set< std::string > _canDo; + + long double fpNShapeLA; + long double fpNShapeLB; + long double fpNShapeRA; + long double fpNShapeRB; + bool fpFlip; + //default stuff + + double iirSampleAL; + double iirSampleBL; + double iirSampleCL; + double iirSampleDL; + double iirSampleEL; + double iirSampleFL; + double iirSampleGL; + double iirSampleHL; + double lastSampleL; + + double iirSampleAR; + double iirSampleBR; + double iirSampleCR; + double iirSampleDR; + double iirSampleER; + double iirSampleFR; + double iirSampleGR; + double iirSampleHR; + double lastSampleR; + + float A; + float B; + float C; + float D; + float E; //parameters. Always 0-1, and we scale/alter them elsewhere. + +}; + +#endif diff --git a/plugins/MacVST/DrumSlam/source/DrumSlamProc.cpp b/plugins/MacVST/DrumSlam/source/DrumSlamProc.cpp new file mode 100755 index 0000000..171b353 --- /dev/null +++ b/plugins/MacVST/DrumSlam/source/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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