From 633be2e22c6648c901f08f3b4cd4e8e14ea86443 Mon Sep 17 00:00:00 2001 From: Chris Johnson Date: Mon, 22 Oct 2018 18:04:06 -0400 Subject: Updates (in case my plane crashes) --- .../WinVST/ADClip7/.vs/Console4Channel64/v14/.suo | Bin 0 -> 32768 bytes plugins/WinVST/ADClip7/.vs/VSTProject/v14/.suo | Bin 0 -> 22528 bytes plugins/WinVST/ADClip7/ADClip7.cpp | 165 ++++ plugins/WinVST/ADClip7/ADClip7.h | 85 ++ plugins/WinVST/ADClip7/ADClip7Proc.cpp | 953 +++++++++++++++++++++ plugins/WinVST/ADClip7/VSTProject.sln | 28 + plugins/WinVST/ADClip7/VSTProject.vcxproj | 183 ++++ plugins/WinVST/ADClip7/VSTProject.vcxproj.filters | 48 ++ plugins/WinVST/ADClip7/VSTProject.vcxproj.user | 19 + plugins/WinVST/ADClip7/vstplug.def | 3 + 10 files changed, 1484 insertions(+) create mode 100755 plugins/WinVST/ADClip7/.vs/Console4Channel64/v14/.suo create mode 100755 plugins/WinVST/ADClip7/.vs/VSTProject/v14/.suo create mode 100755 plugins/WinVST/ADClip7/ADClip7.cpp create mode 100755 plugins/WinVST/ADClip7/ADClip7.h create mode 100755 plugins/WinVST/ADClip7/ADClip7Proc.cpp create mode 100755 plugins/WinVST/ADClip7/VSTProject.sln create mode 100755 plugins/WinVST/ADClip7/VSTProject.vcxproj create mode 100755 plugins/WinVST/ADClip7/VSTProject.vcxproj.filters create mode 100755 plugins/WinVST/ADClip7/VSTProject.vcxproj.user create mode 100755 plugins/WinVST/ADClip7/vstplug.def (limited to 'plugins/WinVST/ADClip7') diff --git a/plugins/WinVST/ADClip7/.vs/Console4Channel64/v14/.suo b/plugins/WinVST/ADClip7/.vs/Console4Channel64/v14/.suo new file mode 100755 index 0000000..777b846 Binary files /dev/null and b/plugins/WinVST/ADClip7/.vs/Console4Channel64/v14/.suo differ diff --git a/plugins/WinVST/ADClip7/.vs/VSTProject/v14/.suo b/plugins/WinVST/ADClip7/.vs/VSTProject/v14/.suo new file mode 100755 index 0000000..41b2402 Binary files /dev/null and b/plugins/WinVST/ADClip7/.vs/VSTProject/v14/.suo differ diff --git a/plugins/WinVST/ADClip7/ADClip7.cpp b/plugins/WinVST/ADClip7/ADClip7.cpp new file mode 100755 index 0000000..fb6d6c6 --- /dev/null +++ b/plugins/WinVST/ADClip7/ADClip7.cpp @@ -0,0 +1,165 @@ +/* ======================================== + * ADClip7 - ADClip7.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __ADClip7_H +#include "ADClip7.h" +#endif + +AudioEffect* createEffectInstance(audioMasterCallback audioMaster) {return new ADClip7(audioMaster);} + +ADClip7::ADClip7(audioMasterCallback audioMaster) : + AudioEffectX(audioMaster, kNumPrograms, kNumParameters) +{ + A = 0.0; + B = 0.5; + C = 0.5; + D = 0.0; + + lastSampleL = 0.0; + lastSampleR = 0.0; + for(int count = 0; count < 22199; count++) {bL[count] = 0; bR[count] = 0;} + gcount = 0; + lowsL = 0; + lowsR = 0; + refclipL = 0.99; + refclipR = 0.99; + iirLowsAL = 0.0; + iirLowsAR = 0.0; + iirLowsBL = 0.0; + iirLowsBR = 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 +} + +ADClip7::~ADClip7() {} +VstInt32 ADClip7::getVendorVersion () {return 1000;} +void ADClip7::setProgramName(char *name) {vst_strncpy (_programName, name, kVstMaxProgNameLen);} +void ADClip7::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 ADClip7::getChunk (void** data, bool isPreset) +{ + float *chunkData = (float *)calloc(kNumParameters, sizeof(float)); + chunkData[0] = A; + chunkData[1] = B; + chunkData[2] = C; + chunkData[3] = D; + /* 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 ADClip7::setChunk (void* data, VstInt32 byteSize, bool isPreset) +{ + float *chunkData = (float *)data; + A = pinParameter(chunkData[0]); + B = pinParameter(chunkData[1]); + C = pinParameter(chunkData[2]); + D = pinParameter(chunkData[3]); + /* 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 ADClip7::setParameter(VstInt32 index, float value) { + switch (index) { + case kParamA: A = value; break; + case kParamB: B = value; break; + case kParamC: C = value; break; + case kParamD: D = value; break; + default: throw; // unknown parameter, shouldn't happen! + } +} + +float ADClip7::getParameter(VstInt32 index) { + switch (index) { + case kParamA: return A; break; + case kParamB: return B; break; + case kParamC: return C; break; + case kParamD: return D; 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 ADClip7::getParameterName(VstInt32 index, char *text) { + switch (index) { + case kParamA: vst_strncpy (text, "Boost", kVstMaxParamStrLen); break; + case kParamB: vst_strncpy (text, "Soften", kVstMaxParamStrLen); break; + case kParamC: vst_strncpy (text, "Enhance", kVstMaxParamStrLen); break; + case kParamD: vst_strncpy (text, "Mode", kVstMaxParamStrLen); break; + default: break; // unknown parameter, shouldn't happen! + } //this is our labels for displaying in the VST host +} + +void ADClip7::getParameterDisplay(VstInt32 index, char *text) { + switch (index) { + case kParamA: float2string (A*18.0, text, kVstMaxParamStrLen); break; + case kParamB: float2string (B, text, kVstMaxParamStrLen); break; + case kParamC: float2string (C, text, kVstMaxParamStrLen); break; + case kParamD: switch((VstInt32)( D * 2.999 )) //0 to almost edge of # of params + {case 0: vst_strncpy (text, "Normal", kVstMaxParamStrLen); break; + case 1: vst_strncpy (text, "Atten", kVstMaxParamStrLen); break; + case 2: vst_strncpy (text, "Clips", kVstMaxParamStrLen); break; + default: break; // unknown parameter, shouldn't happen! + } break; + default: break; // unknown parameter, shouldn't happen! + } //this displays the values and handles 'popups' where it's discrete choices +} + +void ADClip7::getParameterLabel(VstInt32 index, char *text) { + switch (index) { + case kParamA: vst_strncpy (text, "dB", kVstMaxParamStrLen); break; + case kParamB: vst_strncpy (text, "", kVstMaxParamStrLen); break; + case kParamC: vst_strncpy (text, "", kVstMaxParamStrLen); break; + case kParamD: vst_strncpy (text, "", kVstMaxParamStrLen); break; + default: break; // unknown parameter, shouldn't happen! + } +} + +VstInt32 ADClip7::canDo(char *text) +{ return (_canDo.find(text) == _canDo.end()) ? -1: 1; } // 1 = yes, -1 = no, 0 = don't know + +bool ADClip7::getEffectName(char* name) { + vst_strncpy(name, "ADClip7", kVstMaxProductStrLen); return true; +} + +VstPlugCategory ADClip7::getPlugCategory() {return kPlugCategEffect;} + +bool ADClip7::getProductString(char* text) { + vst_strncpy (text, "airwindows ADClip7", kVstMaxProductStrLen); return true; +} + +bool ADClip7::getVendorString(char* text) { + vst_strncpy (text, "airwindows", kVstMaxVendorStrLen); return true; +} diff --git a/plugins/WinVST/ADClip7/ADClip7.h b/plugins/WinVST/ADClip7/ADClip7.h new file mode 100755 index 0000000..f20d3fb --- /dev/null +++ b/plugins/WinVST/ADClip7/ADClip7.h @@ -0,0 +1,85 @@ +/* ======================================== + * ADClip7 - ADClip7.h + * Created 8/12/11 by SPIAdmin + * Copyright (c) 2011 __MyCompanyName__, All rights reserved + * ======================================== */ + +#ifndef __ADClip7_H +#define __ADClip7_H + +#ifndef __audioeffect__ +#include "audioeffectx.h" +#endif + +#include +#include +#include + +enum { + kParamA = 0, + kParamB = 1, + kParamC = 2, + kParamD = 3, + kNumParameters = 4 +}; // + +const int kNumPrograms = 0; +const int kNumInputs = 2; +const int kNumOutputs = 2; +const unsigned long kUniqueId = 'adcr'; //Change this to what the AU identity is! + +class ADClip7 : + public AudioEffectX +{ +public: + ADClip7(audioMasterCallback audioMaster); + ~ADClip7(); + 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 + long double lastSampleL; + long double lastSampleR; + float bL[22200]; + float bR[22200]; + int gcount; + double lowsL; + double lowsR; + double iirLowsAL; + double iirLowsAR; + double iirLowsBL; + double iirLowsBR; + long double refclipL; + long double refclipR; + + float A; + float B; + float C; + float D; + +}; + +#endif diff --git a/plugins/WinVST/ADClip7/ADClip7Proc.cpp b/plugins/WinVST/ADClip7/ADClip7Proc.cpp new file mode 100755 index 0000000..2705d61 --- /dev/null +++ b/plugins/WinVST/ADClip7/ADClip7Proc.cpp @@ -0,0 +1,953 @@ +/* ======================================== + * ADClip7 - ADClip7.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __ADClip7_H +#include "ADClip7.h" +#endif + +void ADClip7::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(); + float fpTemp; + long double fpOld = 0.618033988749894848204586; //golden ratio! + long double fpNew = 1.0 - fpOld; + + double inputGain = pow(10.0,(A*18.0)/20.0); + double softness = B * fpNew; + double hardness = 1.0 - softness; + double highslift = 0.307 * C; + double adjust = pow(highslift,3) * 0.416; + double subslift = 0.796 * C; + double calibsubs = subslift/53; + double invcalibsubs = 1.0 - calibsubs; + double subs = 0.81 + (calibsubs*2); + long double bridgerectifier; + int mode = (int) floor(D*2.999)+1; + double overshootL; + double overshootR; + double offsetH1 = 1.84; + offsetH1 *= overallscale; + double offsetH2 = offsetH1 * 1.9; + double offsetH3 = offsetH1 * 2.7; + double offsetL1 = 612; + offsetL1 *= overallscale; + double offsetL2 = offsetL1 * 2.0; + int refH1 = (int)floor(offsetH1); + int refH2 = (int)floor(offsetH2); + int refH3 = (int)floor(offsetH3); + int refL1 = (int)floor(offsetL1); + int refL2 = (int)floor(offsetL2); + int temp; + double fractionH1 = offsetH1 - floor(offsetH1); + double fractionH2 = offsetH2 - floor(offsetH2); + double fractionH3 = offsetH3 - floor(offsetH3); + double minusH1 = 1.0 - fractionH1; + double minusH2 = 1.0 - fractionH2; + double minusH3 = 1.0 - fractionH3; + double highsL = 0.0; + double highsR = 0.0; + int count = 0; + + 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. + } + + + + if (inputGain != 1.0) { + inputSampleL *= inputGain; + inputSampleR *= inputGain; + } + + overshootL = fabs(inputSampleL) - refclipL; + overshootR = fabs(inputSampleR) - refclipR; + if (overshootL < 0.0) overshootL = 0.0; + if (overshootR < 0.0) overshootR = 0.0; + + if (gcount < 0 || gcount > 11020) {gcount = 11020;} + count = gcount; + bL[count+11020] = bL[count] = overshootL; + bR[count+11020] = bR[count] = overshootR; + gcount--; + + if (highslift > 0.0) + { + //we have a big pile of b[] which is overshoots + temp = count+refH3; + highsL = -(bL[temp] * minusH3); //less as value moves away from .0 + highsL -= bL[temp+1]; //we can assume always using this in one way or another? + highsL -= (bL[temp+2] * fractionH3); //greater as value moves away from .0 + highsL += (((bL[temp]-bL[temp+1])-(bL[temp+1]-bL[temp+2]))/50); //interpolation hacks 'r us + highsL *= adjust; //add in the kernel elements backwards saves multiplies + //stage 3 is a negative add + highsR = -(bR[temp] * minusH3); //less as value moves away from .0 + highsR -= bR[temp+1]; //we can assume always using this in one way or another? + highsR -= (bR[temp+2] * fractionH3); //greater as value moves away from .0 + highsR += (((bR[temp]-bR[temp+1])-(bR[temp+1]-bR[temp+2]))/50); //interpolation hacks 'r us + highsR *= adjust; //add in the kernel elements backwards saves multiplies + //stage 3 is a negative add + temp = count+refH2; + highsL += (bL[temp] * minusH2); //less as value moves away from .0 + highsL += bL[temp+1]; //we can assume always using this in one way or another? + highsL += (bL[temp+2] * fractionH2); //greater as value moves away from .0 + highsL -= (((bL[temp]-bL[temp+1])-(bL[temp+1]-bL[temp+2]))/50); //interpolation hacks 'r us + highsL *= adjust; //add in the kernel elements backwards saves multiplies + //stage 2 is a positive feedback of the overshoot + highsR += (bR[temp] * minusH2); //less as value moves away from .0 + highsR += bR[temp+1]; //we can assume always using this in one way or another? + highsR += (bR[temp+2] * fractionH2); //greater as value moves away from .0 + highsR -= (((bR[temp]-bR[temp+1])-(bR[temp+1]-bR[temp+2]))/50); //interpolation hacks 'r us + highsR *= adjust; //add in the kernel elements backwards saves multiplies + //stage 2 is a positive feedback of the overshoot + temp = count+refH1; + highsL -= (bL[temp] * minusH1); //less as value moves away from .0 + highsL -= bL[temp+1]; //we can assume always using this in one way or another? + highsL -= (bL[temp+2] * fractionH1); //greater as value moves away from .0 + highsL += (((bL[temp]-bL[temp+1])-(bL[temp+1]-bL[temp+2]))/50); //interpolation hacks 'r us + highsL *= adjust; //add in the kernel elements backwards saves multiplies + //stage 1 is a negative feedback of the overshoot + highsR -= (bR[temp] * minusH1); //less as value moves away from .0 + highsR -= bR[temp+1]; //we can assume always using this in one way or another? + highsR -= (bR[temp+2] * fractionH1); //greater as value moves away from .0 + highsR += (((bR[temp]-bR[temp+1])-(bR[temp+1]-bR[temp+2]))/50); //interpolation hacks 'r us + highsR *= adjust; //add in the kernel elements backwards saves multiplies + //stage 1 is a negative feedback of the overshoot + //done with interpolated mostly negative feedback of the overshoot + } + + bridgerectifier = sin(fabs(highsL) * hardness); + //this will wrap around and is scaled back by softness + //wrap around is the same principle as Fracture: no top limit to sin() + if (highsL > 0) highsL = bridgerectifier; + else highsL = -bridgerectifier; + + bridgerectifier = sin(fabs(highsR) * hardness); + //this will wrap around and is scaled back by softness + //wrap around is the same principle as Fracture: no top limit to sin() + if (highsR > 0) highsR = bridgerectifier; + else highsR = -bridgerectifier; + + if (subslift > 0.0) + { + lowsL *= subs; + lowsR *= subs; + //going in we'll reel back some of the swing + temp = count+refL1; + + lowsL -= bL[temp+127]; + lowsL -= bL[temp+113]; + lowsL -= bL[temp+109]; + lowsL -= bL[temp+107]; + lowsL -= bL[temp+103]; + lowsL -= bL[temp+101]; + lowsL -= bL[temp+97]; + lowsL -= bL[temp+89]; + lowsL -= bL[temp+83]; + lowsL -= bL[temp+79]; + lowsL -= bL[temp+73]; + lowsL -= bL[temp+71]; + lowsL -= bL[temp+67]; + lowsL -= bL[temp+61]; + lowsL -= bL[temp+59]; + lowsL -= bL[temp+53]; + lowsL -= bL[temp+47]; + lowsL -= bL[temp+43]; + lowsL -= bL[temp+41]; + lowsL -= bL[temp+37]; + lowsL -= bL[temp+31]; + lowsL -= bL[temp+29]; + lowsL -= bL[temp+23]; + lowsL -= bL[temp+19]; + lowsL -= bL[temp+17]; + lowsL -= bL[temp+13]; + lowsL -= bL[temp+11]; + lowsL -= bL[temp+7]; + lowsL -= bL[temp+5]; + lowsL -= bL[temp+3]; + lowsL -= bL[temp+2]; + lowsL -= bL[temp+1]; + //initial negative lobe + + lowsR -= bR[temp+127]; + lowsR -= bR[temp+113]; + lowsR -= bR[temp+109]; + lowsR -= bR[temp+107]; + lowsR -= bR[temp+103]; + lowsR -= bR[temp+101]; + lowsR -= bR[temp+97]; + lowsR -= bR[temp+89]; + lowsR -= bR[temp+83]; + lowsR -= bR[temp+79]; + lowsR -= bR[temp+73]; + lowsR -= bR[temp+71]; + lowsR -= bR[temp+67]; + lowsR -= bR[temp+61]; + lowsR -= bR[temp+59]; + lowsR -= bR[temp+53]; + lowsR -= bR[temp+47]; + lowsR -= bR[temp+43]; + lowsR -= bR[temp+41]; + lowsR -= bR[temp+37]; + lowsR -= bR[temp+31]; + lowsR -= bR[temp+29]; + lowsR -= bR[temp+23]; + lowsR -= bR[temp+19]; + lowsR -= bR[temp+17]; + lowsR -= bR[temp+13]; + lowsR -= bR[temp+11]; + lowsR -= bR[temp+7]; + lowsR -= bR[temp+5]; + lowsR -= bR[temp+3]; + lowsR -= bR[temp+2]; + lowsR -= bR[temp+1]; + //initial negative lobe + + lowsL *= subs; + lowsL *= subs; + lowsR *= subs; + lowsR *= subs; + //twice, to minimize the suckout in low boost situations + temp = count+refL2; + + lowsL += bL[temp+127]; + lowsL += bL[temp+113]; + lowsL += bL[temp+109]; + lowsL += bL[temp+107]; + lowsL += bL[temp+103]; + lowsL += bL[temp+101]; + lowsL += bL[temp+97]; + lowsL += bL[temp+89]; + lowsL += bL[temp+83]; + lowsL += bL[temp+79]; + lowsL += bL[temp+73]; + lowsL += bL[temp+71]; + lowsL += bL[temp+67]; + lowsL += bL[temp+61]; + lowsL += bL[temp+59]; + lowsL += bL[temp+53]; + lowsL += bL[temp+47]; + lowsL += bL[temp+43]; + lowsL += bL[temp+41]; + lowsL += bL[temp+37]; + lowsL += bL[temp+31]; + lowsL += bL[temp+29]; + lowsL += bL[temp+23]; + lowsL += bL[temp+19]; + lowsL += bL[temp+17]; + lowsL += bL[temp+13]; + lowsL += bL[temp+11]; + lowsL += bL[temp+7]; + lowsL += bL[temp+5]; + lowsL += bL[temp+3]; + lowsL += bL[temp+2]; + lowsL += bL[temp+1]; + //followup positive lobe + + lowsR += bR[temp+127]; + lowsR += bR[temp+113]; + lowsR += bR[temp+109]; + lowsR += bR[temp+107]; + lowsR += bR[temp+103]; + lowsR += bR[temp+101]; + lowsR += bR[temp+97]; + lowsR += bR[temp+89]; + lowsR += bR[temp+83]; + lowsR += bR[temp+79]; + lowsR += bR[temp+73]; + lowsR += bR[temp+71]; + lowsR += bR[temp+67]; + lowsR += bR[temp+61]; + lowsR += bR[temp+59]; + lowsR += bR[temp+53]; + lowsR += bR[temp+47]; + lowsR += bR[temp+43]; + lowsR += bR[temp+41]; + lowsR += bR[temp+37]; + lowsR += bR[temp+31]; + lowsR += bR[temp+29]; + lowsR += bR[temp+23]; + lowsR += bR[temp+19]; + lowsR += bR[temp+17]; + lowsR += bR[temp+13]; + lowsR += bR[temp+11]; + lowsR += bR[temp+7]; + lowsR += bR[temp+5]; + lowsR += bR[temp+3]; + lowsR += bR[temp+2]; + lowsR += bR[temp+1]; + //followup positive lobe + + lowsL *= subs; + lowsR *= subs; + //now we have the lows content to use + } + + bridgerectifier = sin(fabs(lowsL) * softness); + //this will wrap around and is scaled back by hardness: hard = less bass push, more treble + //wrap around is the same principle as Fracture: no top limit to sin() + if (lowsL > 0) lowsL = bridgerectifier; + else lowsL = -bridgerectifier; + + bridgerectifier = sin(fabs(lowsR) * softness); + //this will wrap around and is scaled back by hardness: hard = less bass push, more treble + //wrap around is the same principle as Fracture: no top limit to sin() + if (lowsR > 0) lowsR = bridgerectifier; + else lowsR = -bridgerectifier; + + iirLowsAL = (iirLowsAL * invcalibsubs) + (lowsL * calibsubs); + lowsL = iirLowsAL; + bridgerectifier = sin(fabs(lowsL)); + if (lowsL > 0) lowsL = bridgerectifier; + else lowsL = -bridgerectifier; + + iirLowsAR = (iirLowsAR * invcalibsubs) + (lowsR * calibsubs); + lowsR = iirLowsAR; + bridgerectifier = sin(fabs(lowsR)); + if (lowsR > 0) lowsR = bridgerectifier; + else lowsR = -bridgerectifier; + + iirLowsBL = (iirLowsBL * invcalibsubs) + (lowsL * calibsubs); + lowsL = iirLowsBL; + bridgerectifier = sin(fabs(lowsL)) * 2.0; + if (lowsL > 0) lowsL = bridgerectifier; + else lowsL = -bridgerectifier; + + iirLowsBR = (iirLowsBR * invcalibsubs) + (lowsR * calibsubs); + lowsR = iirLowsBR; + bridgerectifier = sin(fabs(lowsR)) * 2.0; + if (lowsR > 0) lowsR = bridgerectifier; + else lowsR = -bridgerectifier; + + if (highslift > 0.0) inputSampleL += (highsL * (1.0-fabs(inputSampleL*hardness))); + if (subslift > 0.0) inputSampleL += (lowsL * (1.0-fabs(inputSampleL*softness))); + + if (highslift > 0.0) inputSampleR += (highsR * (1.0-fabs(inputSampleR*hardness))); + if (subslift > 0.0) inputSampleR += (lowsR * (1.0-fabs(inputSampleR*softness))); + + if (inputSampleL > refclipL && refclipL > 0.9) refclipL -= 0.01; + if (inputSampleL < -refclipL && refclipL > 0.9) refclipL -= 0.01; + if (refclipL < 0.99) refclipL += 0.00001; + //adjust clip level on the fly + + if (inputSampleR > refclipR && refclipR > 0.9) refclipR -= 0.01; + if (inputSampleR < -refclipR && refclipR > 0.9) refclipR -= 0.01; + if (refclipR < 0.99) refclipR += 0.00001; + //adjust clip level on the fly + + if (lastSampleL >= refclipL) + { + if (inputSampleL < refclipL) lastSampleL = ((refclipL*hardness) + (inputSampleL * softness)); + else lastSampleL = refclipL; + } + + if (lastSampleR >= refclipR) + { + if (inputSampleR < refclipR) lastSampleR = ((refclipR*hardness) + (inputSampleR * softness)); + else lastSampleR = refclipR; + } + + if (lastSampleL <= -refclipL) + { + if (inputSampleL > -refclipL) lastSampleL = ((-refclipL*hardness) + (inputSampleL * softness)); + else lastSampleL = -refclipL; + } + + if (lastSampleR <= -refclipR) + { + if (inputSampleR > -refclipR) lastSampleR = ((-refclipR*hardness) + (inputSampleR * softness)); + else lastSampleR = -refclipR; + } + + if (inputSampleL > refclipL) + { + if (lastSampleL < refclipL) inputSampleL = ((refclipL*hardness) + (lastSampleL * softness)); + else inputSampleL = refclipL; + } + + if (inputSampleR > refclipR) + { + if (lastSampleR < refclipR) inputSampleR = ((refclipR*hardness) + (lastSampleR * softness)); + else inputSampleR = refclipR; + } + + if (inputSampleL < -refclipL) + { + if (lastSampleL > -refclipL) inputSampleL = ((-refclipL*hardness) + (lastSampleL * softness)); + else inputSampleL = -refclipL; + } + + if (inputSampleR < -refclipR) + { + if (lastSampleR > -refclipR) inputSampleR = ((-refclipR*hardness) + (lastSampleR * softness)); + else inputSampleR = -refclipR; + } + lastSampleL = inputSampleL; + lastSampleR = inputSampleR; + + switch (mode) + { + case 1: break; //Normal + case 2: inputSampleL /= inputGain; inputSampleR /= inputGain; break; //Gain Match + case 3: inputSampleL = overshootL + highsL + lowsL; inputSampleR = overshootR + highsR + lowsR; break; //Clip Only + } + //this is our output mode switch, showing the effects + + if (inputSampleL > refclipL) inputSampleL = refclipL; + if (inputSampleL < -refclipL) inputSampleL = -refclipL; + if (inputSampleR > refclipR) inputSampleR = refclipR; + if (inputSampleR < -refclipR) inputSampleR = -refclipR; + //final iron bar + + //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 ADClip7::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 fpTemp; + long double fpOld = 0.618033988749894848204586; //golden ratio! + long double fpNew = 1.0 - fpOld; + + double inputGain = pow(10.0,(A*18.0)/20.0); + double softness = B * fpNew; + double hardness = 1.0 - softness; + double highslift = 0.307 * C; + double adjust = pow(highslift,3) * 0.416; + double subslift = 0.796 * C; + double calibsubs = subslift/53; + double invcalibsubs = 1.0 - calibsubs; + double subs = 0.81 + (calibsubs*2); + long double bridgerectifier; + int mode = (int) floor(D*2.999)+1; + double overshootL; + double overshootR; + double offsetH1 = 1.84; + offsetH1 *= overallscale; + double offsetH2 = offsetH1 * 1.9; + double offsetH3 = offsetH1 * 2.7; + double offsetL1 = 612; + offsetL1 *= overallscale; + double offsetL2 = offsetL1 * 2.0; + int refH1 = (int)floor(offsetH1); + int refH2 = (int)floor(offsetH2); + int refH3 = (int)floor(offsetH3); + int refL1 = (int)floor(offsetL1); + int refL2 = (int)floor(offsetL2); + int temp; + double fractionH1 = offsetH1 - floor(offsetH1); + double fractionH2 = offsetH2 - floor(offsetH2); + double fractionH3 = offsetH3 - floor(offsetH3); + double minusH1 = 1.0 - fractionH1; + double minusH2 = 1.0 - fractionH2; + double minusH3 = 1.0 - fractionH3; + double highsL = 0.0; + double highsR = 0.0; + int count = 0; + + 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. + } + + + + if (inputGain != 1.0) { + inputSampleL *= inputGain; + inputSampleR *= inputGain; + } + + overshootL = fabs(inputSampleL) - refclipL; + overshootR = fabs(inputSampleR) - refclipR; + if (overshootL < 0.0) overshootL = 0.0; + if (overshootR < 0.0) overshootR = 0.0; + + if (gcount < 0 || gcount > 11020) {gcount = 11020;} + count = gcount; + bL[count+11020] = bL[count] = overshootL; + bR[count+11020] = bR[count] = overshootR; + gcount--; + + if (highslift > 0.0) + { + //we have a big pile of b[] which is overshoots + temp = count+refH3; + highsL = -(bL[temp] * minusH3); //less as value moves away from .0 + highsL -= bL[temp+1]; //we can assume always using this in one way or another? + highsL -= (bL[temp+2] * fractionH3); //greater as value moves away from .0 + highsL += (((bL[temp]-bL[temp+1])-(bL[temp+1]-bL[temp+2]))/50); //interpolation hacks 'r us + highsL *= adjust; //add in the kernel elements backwards saves multiplies + //stage 3 is a negative add + highsR = -(bR[temp] * minusH3); //less as value moves away from .0 + highsR -= bR[temp+1]; //we can assume always using this in one way or another? + highsR -= (bR[temp+2] * fractionH3); //greater as value moves away from .0 + highsR += (((bR[temp]-bR[temp+1])-(bR[temp+1]-bR[temp+2]))/50); //interpolation hacks 'r us + highsR *= adjust; //add in the kernel elements backwards saves multiplies + //stage 3 is a negative add + temp = count+refH2; + highsL += (bL[temp] * minusH2); //less as value moves away from .0 + highsL += bL[temp+1]; //we can assume always using this in one way or another? + highsL += (bL[temp+2] * fractionH2); //greater as value moves away from .0 + highsL -= (((bL[temp]-bL[temp+1])-(bL[temp+1]-bL[temp+2]))/50); //interpolation hacks 'r us + highsL *= adjust; //add in the kernel elements backwards saves multiplies + //stage 2 is a positive feedback of the overshoot + highsR += (bR[temp] * minusH2); //less as value moves away from .0 + highsR += bR[temp+1]; //we can assume always using this in one way or another? + highsR += (bR[temp+2] * fractionH2); //greater as value moves away from .0 + highsR -= (((bR[temp]-bR[temp+1])-(bR[temp+1]-bR[temp+2]))/50); //interpolation hacks 'r us + highsR *= adjust; //add in the kernel elements backwards saves multiplies + //stage 2 is a positive feedback of the overshoot + temp = count+refH1; + highsL -= (bL[temp] * minusH1); //less as value moves away from .0 + highsL -= bL[temp+1]; //we can assume always using this in one way or another? + highsL -= (bL[temp+2] * fractionH1); //greater as value moves away from .0 + highsL += (((bL[temp]-bL[temp+1])-(bL[temp+1]-bL[temp+2]))/50); //interpolation hacks 'r us + highsL *= adjust; //add in the kernel elements backwards saves multiplies + //stage 1 is a negative feedback of the overshoot + highsR -= (bR[temp] * minusH1); //less as value moves away from .0 + highsR -= bR[temp+1]; //we can assume always using this in one way or another? + highsR -= (bR[temp+2] * fractionH1); //greater as value moves away from .0 + highsR += (((bR[temp]-bR[temp+1])-(bR[temp+1]-bR[temp+2]))/50); //interpolation hacks 'r us + highsR *= adjust; //add in the kernel elements backwards saves multiplies + //stage 1 is a negative feedback of the overshoot + //done with interpolated mostly negative feedback of the overshoot + } + + bridgerectifier = sin(fabs(highsL) * hardness); + //this will wrap around and is scaled back by softness + //wrap around is the same principle as Fracture: no top limit to sin() + if (highsL > 0) highsL = bridgerectifier; + else highsL = -bridgerectifier; + + bridgerectifier = sin(fabs(highsR) * hardness); + //this will wrap around and is scaled back by softness + //wrap around is the same principle as Fracture: no top limit to sin() + if (highsR > 0) highsR = bridgerectifier; + else highsR = -bridgerectifier; + + if (subslift > 0.0) + { + lowsL *= subs; + lowsR *= subs; + //going in we'll reel back some of the swing + temp = count+refL1; + + lowsL -= bL[temp+127]; + lowsL -= bL[temp+113]; + lowsL -= bL[temp+109]; + lowsL -= bL[temp+107]; + lowsL -= bL[temp+103]; + lowsL -= bL[temp+101]; + lowsL -= bL[temp+97]; + lowsL -= bL[temp+89]; + lowsL -= bL[temp+83]; + lowsL -= bL[temp+79]; + lowsL -= bL[temp+73]; + lowsL -= bL[temp+71]; + lowsL -= bL[temp+67]; + lowsL -= bL[temp+61]; + lowsL -= bL[temp+59]; + lowsL -= bL[temp+53]; + lowsL -= bL[temp+47]; + lowsL -= bL[temp+43]; + lowsL -= bL[temp+41]; + lowsL -= bL[temp+37]; + lowsL -= bL[temp+31]; + lowsL -= bL[temp+29]; + lowsL -= bL[temp+23]; + lowsL -= bL[temp+19]; + lowsL -= bL[temp+17]; + lowsL -= bL[temp+13]; + lowsL -= bL[temp+11]; + lowsL -= bL[temp+7]; + lowsL -= bL[temp+5]; + lowsL -= bL[temp+3]; + lowsL -= bL[temp+2]; + lowsL -= bL[temp+1]; + //initial negative lobe + + lowsR -= bR[temp+127]; + lowsR -= bR[temp+113]; + lowsR -= bR[temp+109]; + lowsR -= bR[temp+107]; + lowsR -= bR[temp+103]; + lowsR -= bR[temp+101]; + lowsR -= bR[temp+97]; + lowsR -= bR[temp+89]; + lowsR -= bR[temp+83]; + lowsR -= bR[temp+79]; + lowsR -= bR[temp+73]; + lowsR -= bR[temp+71]; + lowsR -= bR[temp+67]; + lowsR -= bR[temp+61]; + lowsR -= bR[temp+59]; + lowsR -= bR[temp+53]; + lowsR -= bR[temp+47]; + lowsR -= bR[temp+43]; + lowsR -= bR[temp+41]; + lowsR -= bR[temp+37]; + lowsR -= bR[temp+31]; + lowsR -= bR[temp+29]; + lowsR -= bR[temp+23]; + lowsR -= bR[temp+19]; + lowsR -= bR[temp+17]; + lowsR -= bR[temp+13]; + lowsR -= bR[temp+11]; + lowsR -= bR[temp+7]; + lowsR -= bR[temp+5]; + lowsR -= bR[temp+3]; + lowsR -= bR[temp+2]; + lowsR -= bR[temp+1]; + //initial negative lobe + + lowsL *= subs; + lowsL *= subs; + lowsR *= subs; + lowsR *= subs; + //twice, to minimize the suckout in low boost situations + temp = count+refL2; + + lowsL += bL[temp+127]; + lowsL += bL[temp+113]; + lowsL += bL[temp+109]; + lowsL += bL[temp+107]; + lowsL += bL[temp+103]; + lowsL += bL[temp+101]; + lowsL += bL[temp+97]; + lowsL += bL[temp+89]; + lowsL += bL[temp+83]; + lowsL += bL[temp+79]; + lowsL += bL[temp+73]; + lowsL += bL[temp+71]; + lowsL += bL[temp+67]; + lowsL += bL[temp+61]; + lowsL += bL[temp+59]; + lowsL += bL[temp+53]; + lowsL += bL[temp+47]; + lowsL += bL[temp+43]; + lowsL += bL[temp+41]; + lowsL += bL[temp+37]; + lowsL += bL[temp+31]; + lowsL += bL[temp+29]; + lowsL += bL[temp+23]; + lowsL += bL[temp+19]; + lowsL += bL[temp+17]; + lowsL += bL[temp+13]; + lowsL += bL[temp+11]; + lowsL += bL[temp+7]; + lowsL += bL[temp+5]; + lowsL += bL[temp+3]; + lowsL += bL[temp+2]; + lowsL += bL[temp+1]; + //followup positive lobe + + lowsR += bR[temp+127]; + lowsR += bR[temp+113]; + lowsR += bR[temp+109]; + lowsR += bR[temp+107]; + lowsR += bR[temp+103]; + lowsR += bR[temp+101]; + lowsR += bR[temp+97]; + lowsR += bR[temp+89]; + lowsR += bR[temp+83]; + lowsR += bR[temp+79]; + lowsR += bR[temp+73]; + lowsR += bR[temp+71]; + lowsR += bR[temp+67]; + lowsR += bR[temp+61]; + lowsR += bR[temp+59]; + lowsR += bR[temp+53]; + lowsR += bR[temp+47]; + lowsR += bR[temp+43]; + lowsR += bR[temp+41]; + lowsR += bR[temp+37]; + lowsR += bR[temp+31]; + lowsR += bR[temp+29]; + lowsR += bR[temp+23]; + lowsR += bR[temp+19]; + lowsR += bR[temp+17]; + lowsR += bR[temp+13]; + lowsR += bR[temp+11]; + lowsR += bR[temp+7]; + lowsR += bR[temp+5]; + lowsR += bR[temp+3]; + lowsR += bR[temp+2]; + lowsR += bR[temp+1]; + //followup positive lobe + + lowsL *= subs; + lowsR *= subs; + //now we have the lows content to use + } + + bridgerectifier = sin(fabs(lowsL) * softness); + //this will wrap around and is scaled back by hardness: hard = less bass push, more treble + //wrap around is the same principle as Fracture: no top limit to sin() + if (lowsL > 0) lowsL = bridgerectifier; + else lowsL = -bridgerectifier; + + bridgerectifier = sin(fabs(lowsR) * softness); + //this will wrap around and is scaled back by hardness: hard = less bass push, more treble + //wrap around is the same principle as Fracture: no top limit to sin() + if (lowsR > 0) lowsR = bridgerectifier; + else lowsR = -bridgerectifier; + + iirLowsAL = (iirLowsAL * invcalibsubs) + (lowsL * calibsubs); + lowsL = iirLowsAL; + bridgerectifier = sin(fabs(lowsL)); + if (lowsL > 0) lowsL = bridgerectifier; + else lowsL = -bridgerectifier; + + iirLowsAR = (iirLowsAR * invcalibsubs) + (lowsR * calibsubs); + lowsR = iirLowsAR; + bridgerectifier = sin(fabs(lowsR)); + if (lowsR > 0) lowsR = bridgerectifier; + else lowsR = -bridgerectifier; + + iirLowsBL = (iirLowsBL * invcalibsubs) + (lowsL * calibsubs); + lowsL = iirLowsBL; + bridgerectifier = sin(fabs(lowsL)) * 2.0; + if (lowsL > 0) lowsL = bridgerectifier; + else lowsL = -bridgerectifier; + + iirLowsBR = (iirLowsBR * invcalibsubs) + (lowsR * calibsubs); + lowsR = iirLowsBR; + bridgerectifier = sin(fabs(lowsR)) * 2.0; + if (lowsR > 0) lowsR = bridgerectifier; + else lowsR = -bridgerectifier; + + if (highslift > 0.0) inputSampleL += (highsL * (1.0-fabs(inputSampleL*hardness))); + if (subslift > 0.0) inputSampleL += (lowsL * (1.0-fabs(inputSampleL*softness))); + + if (highslift > 0.0) inputSampleR += (highsR * (1.0-fabs(inputSampleR*hardness))); + if (subslift > 0.0) inputSampleR += (lowsR * (1.0-fabs(inputSampleR*softness))); + + if (inputSampleL > refclipL && refclipL > 0.9) refclipL -= 0.01; + if (inputSampleL < -refclipL && refclipL > 0.9) refclipL -= 0.01; + if (refclipL < 0.99) refclipL += 0.00001; + //adjust clip level on the fly + + if (inputSampleR > refclipR && refclipR > 0.9) refclipR -= 0.01; + if (inputSampleR < -refclipR && refclipR > 0.9) refclipR -= 0.01; + if (refclipR < 0.99) refclipR += 0.00001; + //adjust clip level on the fly + + if (lastSampleL >= refclipL) + { + if (inputSampleL < refclipL) lastSampleL = ((refclipL*hardness) + (inputSampleL * softness)); + else lastSampleL = refclipL; + } + + if (lastSampleR >= refclipR) + { + if (inputSampleR < refclipR) lastSampleR = ((refclipR*hardness) + (inputSampleR * softness)); + else lastSampleR = refclipR; + } + + if (lastSampleL <= -refclipL) + { + if (inputSampleL > -refclipL) lastSampleL = ((-refclipL*hardness) + (inputSampleL * softness)); + else lastSampleL = -refclipL; + } + + if (lastSampleR <= -refclipR) + { + if (inputSampleR > -refclipR) lastSampleR = ((-refclipR*hardness) + (inputSampleR * softness)); + else lastSampleR = -refclipR; + } + + if (inputSampleL > refclipL) + { + if (lastSampleL < refclipL) inputSampleL = ((refclipL*hardness) + (lastSampleL * softness)); + else inputSampleL = refclipL; + } + + if (inputSampleR > refclipR) + { + if (lastSampleR < refclipR) inputSampleR = ((refclipR*hardness) + (lastSampleR * softness)); + else inputSampleR = refclipR; + } + + if (inputSampleL < -refclipL) + { + if (lastSampleL > -refclipL) inputSampleL = ((-refclipL*hardness) + (lastSampleL * softness)); + else inputSampleL = -refclipL; + } + + if (inputSampleR < -refclipR) + { + if (lastSampleR > -refclipR) inputSampleR = ((-refclipR*hardness) + (lastSampleR * softness)); + else inputSampleR = -refclipR; + } + lastSampleL = inputSampleL; + lastSampleR = inputSampleR; + + switch (mode) + { + case 1: break; //Normal + case 2: inputSampleL /= inputGain; inputSampleR /= inputGain; break; //Gain Match + case 3: inputSampleL = overshootL + highsL + lowsL; inputSampleR = overshootR + highsR + lowsR; break; //Clip Only + } + //this is our output mode switch, showing the effects + + if (inputSampleL > refclipL) inputSampleL = refclipL; + if (inputSampleL < -refclipL) inputSampleL = -refclipL; + if (inputSampleR > refclipR) inputSampleR = refclipR; + if (inputSampleR < -refclipR) inputSampleR = -refclipR; + //final iron bar + + //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++; + } +} \ No newline at end of file diff --git a/plugins/WinVST/ADClip7/VSTProject.sln b/plugins/WinVST/ADClip7/VSTProject.sln new file mode 100755 index 0000000..694b424 --- /dev/null +++ b/plugins/WinVST/ADClip7/VSTProject.sln @@ -0,0 +1,28 @@ + +Microsoft Visual Studio Solution File, Format Version 12.00 +# Visual Studio 14 +VisualStudioVersion = 14.0.25420.1 +MinimumVisualStudioVersion = 10.0.40219.1 +Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "VSTProject", "VSTProject.vcxproj", "{16F7AB3C-1AE0-4574-B60C-7B4DED82938C}" +EndProject +Global + GlobalSection(SolutionConfigurationPlatforms) = preSolution + Debug|x64 = Debug|x64 + Debug|x86 = Debug|x86 + Release|x64 = Release|x64 + Release|x86 = Release|x86 + EndGlobalSection + GlobalSection(ProjectConfigurationPlatforms) = postSolution + {16F7AB3C-1AE0-4574-B60C-7B4DED82938C}.Debug|x64.ActiveCfg = Debug|x64 + {16F7AB3C-1AE0-4574-B60C-7B4DED82938C}.Debug|x64.Build.0 = Debug|x64 + {16F7AB3C-1AE0-4574-B60C-7B4DED82938C}.Debug|x86.ActiveCfg = Debug|Win32 + {16F7AB3C-1AE0-4574-B60C-7B4DED82938C}.Debug|x86.Build.0 = Debug|Win32 + {16F7AB3C-1AE0-4574-B60C-7B4DED82938C}.Release|x64.ActiveCfg = Release|x64 + {16F7AB3C-1AE0-4574-B60C-7B4DED82938C}.Release|x64.Build.0 = Release|x64 + {16F7AB3C-1AE0-4574-B60C-7B4DED82938C}.Release|x86.ActiveCfg = Release|Win32 + {16F7AB3C-1AE0-4574-B60C-7B4DED82938C}.Release|x86.Build.0 = Release|Win32 + EndGlobalSection + GlobalSection(SolutionProperties) = preSolution + HideSolutionNode = FALSE + EndGlobalSection +EndGlobal diff --git a/plugins/WinVST/ADClip7/VSTProject.vcxproj b/plugins/WinVST/ADClip7/VSTProject.vcxproj new file mode 100755 index 0000000..9134595 --- /dev/null +++ b/plugins/WinVST/ADClip7/VSTProject.vcxproj @@ -0,0 +1,183 @@ + + + + + Debug + Win32 + + + Release + Win32 + + + Debug + x64 + + + Release + x64 + + + + + + + + + + + + + + + + + {16F7AB3C-1AE0-4574-B60C-7B4DED82938C} + VSTProject + 8.1 + ADClip764 + + + + DynamicLibrary + true + v140 + NotSet + + + DynamicLibrary + false + v140 + false + NotSet + + + DynamicLibrary + true + v140 + NotSet + + + DynamicLibrary + false + v140 + false + NotSet + + + + + + + + + + + + + + + + + + + + + .dll + + + $(SolutionDir)$(Configuration)\ + $(Configuration)\ + $(VC_ExecutablePath_x64);$(WindowsSDK_ExecutablePath);$(VS_ExecutablePath);$(MSBuild_ExecutablePath);$(SystemRoot)\SysWow64;$(FxCopDir);$(PATH) + + + $(SolutionDir)$(Configuration)\ + $(Configuration)\ + $(VC_ExecutablePath_x64);$(WindowsSDK_ExecutablePath);$(VS_ExecutablePath);$(MSBuild_ExecutablePath);$(SystemRoot)\SysWow64;$(FxCopDir);$(PATH) + + + + Level3 + MaxSpeed + true + C:\Users\christopherjohnson\Documents\Visual Studio 2015\Projects\VSTProject\vst2.x;C:\Users\christopherjohnson\Documents\vstsdk2.4;%(AdditionalIncludeDirectories) + WINDOWS;_WINDOWS;WIN32;_USRDLL;_USE_MATH_DEFINES;_CRT_SECURE_NO_DEPRECATE;VST_FORCE_DEPRECATED;%(PreprocessorDefinitions) + MultiThreadedDebug + Speed + false + Default + false + None + + + vstplug.def + libcmt.dll;libcmtd.dll;msvcrt.lib;%(IgnoreSpecificDefaultLibraries) + kernel32.lib;user32.lib;gdi32.lib;winspool.lib;comdlg32.lib;advapi32.lib;shell32.lib;ole32.lib;oleaut32.lib;uuid.lib;odbc32.lib;odbccp32.lib;%(AdditionalDependencies) + + + + + Level3 + MaxSpeed + true + C:\Users\christopherjohnson\Documents\Visual Studio 2015\Projects\VSTProject\vst2.x;C:\Users\christopherjohnson\Documents\vstsdk2.4;%(AdditionalIncludeDirectories) + Speed + WINDOWS;_WINDOWS;WIN32;_USRDLL;_USE_MATH_DEFINES;_CRT_SECURE_NO_DEPRECATE;VST_FORCE_DEPRECATED;%(PreprocessorDefinitions) + false + MultiThreadedDebug + Default + false + None + + + kernel32.lib;user32.lib;gdi32.lib;winspool.lib;comdlg32.lib;advapi32.lib;shell32.lib;ole32.lib;oleaut32.lib;uuid.lib;odbc32.lib;odbccp32.lib;%(AdditionalDependencies) + libcmt.dll;libcmtd.dll;msvcrt.lib;%(IgnoreSpecificDefaultLibraries) + vstplug.def + + + + + Level3 + MaxSpeed + false + false + true + MultiThreaded + C:\Users\christopherjohnson\Documents\Visual Studio 2015\Projects\VSTProject\vst2.x;C:\Users\christopherjohnson\Documents\vstsdk2.4;%(AdditionalIncludeDirectories) + None + Speed + WINDOWS;_WINDOWS;WIN32;_USRDLL;_USE_MATH_DEFINES;_CRT_SECURE_NO_DEPRECATE;VST_FORCE_DEPRECATED;%(PreprocessorDefinitions) + + + true + true + libcmt.dll;libcmtd.dll;msvcrt.lib;libc.lib;libcd.lib;libcmt.lib;msvcrtd.lib;%(IgnoreSpecificDefaultLibraries) + libcmt.lib;uuid.lib;kernel32.lib;user32.lib;gdi32.lib;winspool.lib;comdlg32.lib;advapi32.lib;shell32.lib;ole32.lib;oleaut32.lib;odbc32.lib;odbccp32.lib;%(AdditionalDependencies) + vstplug.def + + + + + Level3 + MaxSpeed + false + false + true + C:\Users\christopherjohnson\Documents\Visual Studio 2015\Projects\VSTProject\vst2.x;C:\Users\christopherjohnson\Documents\vstsdk2.4;%(AdditionalIncludeDirectories) + None + Speed + WINDOWS;_WINDOWS;WIN32;_USRDLL;_USE_MATH_DEFINES;_CRT_SECURE_NO_DEPRECATE;VST_FORCE_DEPRECATED;%(PreprocessorDefinitions) + MultiThreaded + + + true + true + libcmt.dll;libcmtd.dll;msvcrt.lib;libc.lib;libcd.lib;libcmt.lib;msvcrtd.lib;%(IgnoreSpecificDefaultLibraries) + libcmt.lib;uuid.lib;kernel32.lib;user32.lib;gdi32.lib;winspool.lib;comdlg32.lib;advapi32.lib;shell32.lib;ole32.lib;oleaut32.lib;odbc32.lib;odbccp32.lib;%(AdditionalDependencies) + vstplug.def + + + + + + \ No newline at end of file diff --git a/plugins/WinVST/ADClip7/VSTProject.vcxproj.filters b/plugins/WinVST/ADClip7/VSTProject.vcxproj.filters new file mode 100755 index 0000000..9739ed7 --- /dev/null +++ b/plugins/WinVST/ADClip7/VSTProject.vcxproj.filters @@ -0,0 +1,48 @@ + + + + + {4FC737F1-C7A5-4376-A066-2A32D752A2FF} + cpp;c;cc;cxx;def;odl;idl;hpj;bat;asm;asmx + + + {93995380-89BD-4b04-88EB-625FBE52EBFB} + h;hh;hpp;hxx;hm;inl;inc;xsd + + + {67DA6AB6-F800-4c08-8B7A-83BB121AAD01} + rc;ico;cur;bmp;dlg;rc2;rct;bin;rgs;gif;jpg;jpeg;jpe;resx;tiff;tif;png;wav;mfcribbon-ms + + + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + \ No newline at end of file diff --git a/plugins/WinVST/ADClip7/VSTProject.vcxproj.user b/plugins/WinVST/ADClip7/VSTProject.vcxproj.user new file mode 100755 index 0000000..2216267 --- /dev/null +++ b/plugins/WinVST/ADClip7/VSTProject.vcxproj.user @@ -0,0 +1,19 @@ + + + + {ADEFF70D-84BF-47A1-91C3-FF6B0FC71218} + WindowsLocalDebugger + + + {ADEFF70D-84BF-47A1-91C3-FF6B0FC71218} + WindowsLocalDebugger + + + {ADEFF70D-84BF-47A1-91C3-FF6B0FC71218} + WindowsLocalDebugger + + + {ADEFF70D-84BF-47A1-91C3-FF6B0FC71218} + WindowsLocalDebugger + + \ No newline at end of file diff --git a/plugins/WinVST/ADClip7/vstplug.def b/plugins/WinVST/ADClip7/vstplug.def new file mode 100755 index 0000000..5bf499a --- /dev/null +++ b/plugins/WinVST/ADClip7/vstplug.def @@ -0,0 +1,3 @@ +EXPORTS + VSTPluginMain + main=VSTPluginMain \ No newline at end of file -- cgit v1.2.3