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author | Chris Johnson <jinx6568@sover.net> | 2018-10-22 18:04:06 -0400 |
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committer | Chris Johnson <jinx6568@sover.net> | 2018-10-22 18:04:06 -0400 |
commit | 633be2e22c6648c901f08f3b4cd4e8e14ea86443 (patch) | |
tree | 1e272c3d2b5bd29636b9f9f521af62734e4df012 /plugins/LinuxVST/src/Noise | |
parent | 057757aa8eb0a463caf0cdfdb5894ac5f723ff3f (diff) | |
download | airwindows-lv2-port-633be2e22c6648c901f08f3b4cd4e8e14ea86443.tar.gz airwindows-lv2-port-633be2e22c6648c901f08f3b4cd4e8e14ea86443.tar.bz2 airwindows-lv2-port-633be2e22c6648c901f08f3b4cd4e8e14ea86443.zip |
Updates (in case my plane crashes)
Diffstat (limited to 'plugins/LinuxVST/src/Noise')
-rwxr-xr-x | plugins/LinuxVST/src/Noise/Noise.cpp | 180 | ||||
-rwxr-xr-x | plugins/LinuxVST/src/Noise/Noise.h | 100 | ||||
-rwxr-xr-x | plugins/LinuxVST/src/Noise/NoiseProc.cpp | 650 |
3 files changed, 930 insertions, 0 deletions
diff --git a/plugins/LinuxVST/src/Noise/Noise.cpp b/plugins/LinuxVST/src/Noise/Noise.cpp new file mode 100755 index 0000000..76d00c8 --- /dev/null +++ b/plugins/LinuxVST/src/Noise/Noise.cpp @@ -0,0 +1,180 @@ +/* ======================================== + * Noise - Noise.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __Noise_H +#include "Noise.h" +#endif + +AudioEffect* createEffectInstance(audioMasterCallback audioMaster) {return new Noise(audioMaster);} + +Noise::Noise(audioMasterCallback audioMaster) : + AudioEffectX(audioMaster, kNumPrograms, kNumParameters) +{ + A = 0.5; + B = 0.5; + C = 0.5; + D = 1.0; + E = 0.0; + F = 1.0; + position = 99999999; + quadratic = 0; + noiseAL = 0.0; + noiseBL = 0.0; + noiseCL = 0.0; + rumbleAL = 0.0; + rumbleBL = 0.0; + surgeL = 0.0; + noiseAR = 0.0; + noiseBR = 0.0; + noiseCR = 0.0; + rumbleAR = 0.0; + rumbleBR = 0.0; + surgeR = 0.0; + flipL = false; + flipR = false; + filterflip = false; + for(int count = 0; count < 11; count++) {bL[count] = 0.0; bR[count] = 0.0; f[count] = 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 +} + +Noise::~Noise() {} +VstInt32 Noise::getVendorVersion () {return 1000;} +void Noise::setProgramName(char *name) {vst_strncpy (_programName, name, kVstMaxProgNameLen);} +void Noise::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 Noise::getChunk (void** data, bool isPreset) +{ + float *chunkData = (float *)calloc(kNumParameters, sizeof(float)); + chunkData[0] = A; + chunkData[1] = B; + chunkData[2] = C; + chunkData[3] = D; + chunkData[4] = E; + chunkData[5] = F; + /* 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 Noise::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]); + E = pinParameter(chunkData[4]); + F = pinParameter(chunkData[5]); + /* 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 Noise::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; + case kParamE: E = value; break; + case kParamF: F = value; break; + default: throw; // unknown parameter, shouldn't happen! + } +} + +float Noise::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; + case kParamE: return E; break; + case kParamF: return F; 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 Noise::getParameterName(VstInt32 index, char *text) { + switch (index) { + case kParamA: vst_strncpy (text, "HighCut", kVstMaxParamStrLen); break; + case kParamB: vst_strncpy (text, "LowCut", kVstMaxParamStrLen); break; + case kParamC: vst_strncpy (text, "LShape", kVstMaxParamStrLen); break; + case kParamD: vst_strncpy (text, "Decay", kVstMaxParamStrLen); break; + case kParamE: vst_strncpy (text, "Distnc", kVstMaxParamStrLen); break; + case kParamF: 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 Noise::getParameterDisplay(VstInt32 index, char *text) { + switch (index) { + case kParamA: float2string (A, text, kVstMaxParamStrLen); break; + case kParamB: float2string (B, text, kVstMaxParamStrLen); break; + case kParamC: float2string (C, text, kVstMaxParamStrLen); break; + case kParamD: float2string (D, text, kVstMaxParamStrLen); break; + case kParamE: float2string (E, text, kVstMaxParamStrLen); break; + case kParamF: float2string (F, text, kVstMaxParamStrLen); break; + default: break; // unknown parameter, shouldn't happen! + } //this displays the values and handles 'popups' where it's discrete choices +} + +void Noise::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; + case kParamD: vst_strncpy (text, "", kVstMaxParamStrLen); break; + case kParamE: vst_strncpy (text, "", kVstMaxParamStrLen); break; + case kParamF: vst_strncpy (text, "", kVstMaxParamStrLen); break; + default: break; // unknown parameter, shouldn't happen! + } +} + +VstInt32 Noise::canDo(char *text) +{ return (_canDo.find(text) == _canDo.end()) ? -1: 1; } // 1 = yes, -1 = no, 0 = don't know + +bool Noise::getEffectName(char* name) { + vst_strncpy(name, "Noise", kVstMaxProductStrLen); return true; +} + +VstPlugCategory Noise::getPlugCategory() {return kPlugCategEffect;} + +bool Noise::getProductString(char* text) { + vst_strncpy (text, "airwindows Noise", kVstMaxProductStrLen); return true; +} + +bool Noise::getVendorString(char* text) { + vst_strncpy (text, "airwindows", kVstMaxVendorStrLen); return true; +} diff --git a/plugins/LinuxVST/src/Noise/Noise.h b/plugins/LinuxVST/src/Noise/Noise.h new file mode 100755 index 0000000..0ee4d76 --- /dev/null +++ b/plugins/LinuxVST/src/Noise/Noise.h @@ -0,0 +1,100 @@ +/* ======================================== + * Noise - Noise.h + * Created 8/12/11 by SPIAdmin + * Copyright (c) 2011 __MyCompanyName__, All rights reserved + * ======================================== */ + +#ifndef __Noise_H +#define __Noise_H + +#ifndef __audioeffect__ +#include "audioeffectx.h" +#endif + +#include <set> +#include <string> +#include <math.h> + +enum { + kParamA = 0, + kParamB = 1, + kParamC = 2, + kParamD = 3, + kParamE = 4, + kParamF = 5, + kNumParameters = 6 +}; // + +const int kNumPrograms = 0; +const int kNumInputs = 2; +const int kNumOutputs = 2; +const unsigned long kUniqueId = 'nois'; //Change this to what the AU identity is! + +class Noise : + public AudioEffectX +{ +public: + Noise(audioMasterCallback audioMaster); + ~Noise(); + 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; + + double noiseAL; + double noiseBL; + double noiseCL; + double rumbleAL; + double rumbleBL; + double surgeL; + double noiseAR; + double noiseBR; + double noiseCR; + double rumbleAR; + double rumbleBR; + double surgeR; + + int position; + int quadratic; + bool flipL; + bool flipR; + bool filterflip; + + double bL[11]; + double bR[11]; + + double f[11]; + + long double fpNShapeLA; + long double fpNShapeLB; + long double fpNShapeRA; + long double fpNShapeRB; + bool fpFlip; + //default stuff + + float A; + float B; + float C; + float D; + float E; + float F; //parameters. Always 0-1, and we scale/alter them elsewhere. + +}; + +#endif diff --git a/plugins/LinuxVST/src/Noise/NoiseProc.cpp b/plugins/LinuxVST/src/Noise/NoiseProc.cpp new file mode 100755 index 0000000..e185120 --- /dev/null +++ b/plugins/LinuxVST/src/Noise/NoiseProc.cpp @@ -0,0 +1,650 @@ +/* ======================================== + * Noise - Noise.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __Noise_H +#include "Noise.h" +#endif + +void Noise::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames) +{ + float* in1 = inputs[0]; + float* in2 = inputs[1]; + float* out1 = outputs[0]; + float* out2 = outputs[1]; + + double cutoffL; + double cutoffR; + double cutofftarget = (A*3.5); + double rumblecutoff = cutofftarget * 0.005; + double invcutoffL; + double invcutoffR; + double drySampleL; + double drySampleR; + long double inputSampleL; + long double inputSampleR; + double highpass = C*38.0; + int lowcut = floor(highpass); + int dcut; + if (lowcut > 37) {dcut= 1151;} + if (lowcut == 37) {dcut= 1091;} + if (lowcut == 36) {dcut= 1087;} + if (lowcut == 35) {dcut= 1031;} + if (lowcut == 34) {dcut= 1013;} + if (lowcut == 33) {dcut= 971;} + if (lowcut == 32) {dcut= 907;} + if (lowcut == 31) {dcut= 839;} + if (lowcut == 30) {dcut= 797;} + if (lowcut == 29) {dcut= 733;} + if (lowcut == 28) {dcut= 719;} + if (lowcut == 27) {dcut= 673;} + if (lowcut == 26) {dcut= 613;} + if (lowcut == 25) {dcut= 593;} + if (lowcut == 24) {dcut= 541;} + if (lowcut == 23) {dcut= 479;} + if (lowcut == 22) {dcut= 431;} + if (lowcut == 21) {dcut= 419;} + if (lowcut == 20) {dcut= 373;} + if (lowcut == 19) {dcut= 311;} + if (lowcut == 18) {dcut= 293;} + if (lowcut == 17) {dcut= 233;} + if (lowcut == 16) {dcut= 191;} + if (lowcut == 15) {dcut= 173;} + if (lowcut == 14) {dcut= 131;} + if (lowcut == 13) {dcut= 113;} + if (lowcut == 12) {dcut= 71;} + if (lowcut == 11) {dcut= 53;} + if (lowcut == 10) {dcut= 31;} + if (lowcut == 9) {dcut= 27;} + if (lowcut == 8) {dcut= 23;} + if (lowcut == 7) {dcut= 19;} + if (lowcut == 6) {dcut= 17;} + if (lowcut == 5) {dcut= 13;} + if (lowcut == 4) {dcut= 11;} + if (lowcut == 3) {dcut= 7;} + if (lowcut == 2) {dcut= 5;} + if (lowcut < 2) {dcut= 3;} + highpass = B * 22.0; + lowcut = floor(highpass)+1; + + double decay = 0.001 - ((1.0-pow(1.0-D,3))*0.001); + if (decay == 0.001) decay = 0.1; + double wet = F; + double dry = 1.0 - wet; + wet *= 0.01; //correct large gain issue + double correctionSample; + double accumulatorSampleL; + double accumulatorSampleR; + double overallscale = (E*9.0)+1.0; + double gain = overallscale; + + if (gain > 1.0) {f[0] = 1.0; gain -= 1.0;} else {f[0] = gain; gain = 0.0;} + if (gain > 1.0) {f[1] = 1.0; gain -= 1.0;} else {f[1] = gain; gain = 0.0;} + if (gain > 1.0) {f[2] = 1.0; gain -= 1.0;} else {f[2] = gain; gain = 0.0;} + if (gain > 1.0) {f[3] = 1.0; gain -= 1.0;} else {f[3] = gain; gain = 0.0;} + if (gain > 1.0) {f[4] = 1.0; gain -= 1.0;} else {f[4] = gain; gain = 0.0;} + if (gain > 1.0) {f[5] = 1.0; gain -= 1.0;} else {f[5] = gain; gain = 0.0;} + if (gain > 1.0) {f[6] = 1.0; gain -= 1.0;} else {f[6] = gain; gain = 0.0;} + if (gain > 1.0) {f[7] = 1.0; gain -= 1.0;} else {f[7] = gain; gain = 0.0;} + if (gain > 1.0) {f[8] = 1.0; gain -= 1.0;} else {f[8] = gain; gain = 0.0;} + if (gain > 1.0) {f[9] = 1.0; gain -= 1.0;} else {f[9] = gain; gain = 0.0;} + //there, now we have a neat little moving average with remainders + + if (overallscale < 1.0) overallscale = 1.0; + f[0] /= overallscale; + f[1] /= overallscale; + f[2] /= overallscale; + f[3] /= overallscale; + f[4] /= overallscale; + f[5] /= overallscale; + f[6] /= overallscale; + f[7] /= overallscale; + f[8] /= overallscale; + f[9] /= overallscale; + //and now it's neatly scaled, too + + float fpTemp; + long double fpOld = 0.618033988749894848204586; //golden ratio! + long double fpNew = 1.0 - fpOld; + + + + while (--sampleFrames >= 0) + { + inputSampleL = *in1; + inputSampleR = *in2; + if (inputSampleL<1.2e-38 && -inputSampleL<1.2e-38) { + static int noisesource = 0; + //this declares a variable before anything else is compiled. It won't keep assigning + //it to 0 for every sample, it's as if the declaration doesn't exist in this context, + //but it lets me add this denormalization fix in a single place rather than updating + //it in three different locations. The variable isn't thread-safe but this is only + //a random seed and we can share it with whatever. + noisesource = noisesource % 1700021; noisesource++; + int residue = noisesource * noisesource; + residue = residue % 170003; residue *= residue; + residue = residue % 17011; residue *= residue; + residue = residue % 1709; residue *= residue; + residue = residue % 173; residue *= residue; + residue = residue % 17; + double applyresidue = residue; + applyresidue *= 0.00000001; + applyresidue *= 0.00000001; + inputSampleL = applyresidue; + } + if (inputSampleR<1.2e-38 && -inputSampleR<1.2e-38) { + static int noisesource = 0; + noisesource = noisesource % 1700021; noisesource++; + int residue = noisesource * noisesource; + residue = residue % 170003; residue *= residue; + residue = residue % 17011; residue *= residue; + residue = residue % 1709; residue *= residue; + residue = residue % 173; residue *= residue; + residue = residue % 17; + double applyresidue = residue; + applyresidue *= 0.00000001; + applyresidue *= 0.00000001; + inputSampleR = applyresidue; + //this denormalization routine produces a white noise at -300 dB which the noise + //shaping will interact with to produce a bipolar output, but the noise is actually + //all positive. That should stop any variables from going denormal, and the routine + //only kicks in if digital black is input. As a final touch, if you save to 24-bit + //the silence will return to being digital black again. + } + drySampleL = inputSampleL; + drySampleR = inputSampleR; + + if (surgeL<fabs(inputSampleL)) + { + surgeL += (rand()/(double)RAND_MAX)*(fabs(inputSampleL)-surgeL); + if (surgeL > 1.0) surgeL = 1.0; + } + else + { + surgeL -= ((rand()/(double)RAND_MAX)*(surgeL-fabs(inputSampleL))*decay); + if (surgeL < 0.0) surgeL = 0.0; + } + + cutoffL = pow((cutofftarget*surgeL),5); + if (cutoffL > 1.0) cutoffL = 1.0; + invcutoffL = 1.0 - cutoffL; + //set up modified cutoff L + + if (surgeR<fabs(inputSampleR)) + { + surgeR += (rand()/(double)RAND_MAX)*(fabs(inputSampleR)-surgeR); + if (surgeR > 1.0) surgeR = 1.0; + } + else + { + surgeR -= ((rand()/(double)RAND_MAX)*(surgeR-fabs(inputSampleR))*decay); + if (surgeR < 0.0) surgeR = 0.0; + } + + cutoffR = pow((cutofftarget*surgeR),5); + if (cutoffR > 1.0) cutoffR = 1.0; + invcutoffR = 1.0 - cutoffR; + //set up modified cutoff R + + flipL = !flipL; + flipR = !flipR; + filterflip = !filterflip; + quadratic -= 1; + if (quadratic < 0) + { + position += 1; + quadratic = position * position; + quadratic = quadratic % 170003; //% is C++ mod operator + quadratic *= quadratic; + quadratic = quadratic % 17011; //% is C++ mod operator + quadratic *= quadratic; + //quadratic = quadratic % 1709; //% is C++ mod operator + //quadratic *= quadratic; + quadratic = quadratic % dcut; //% is C++ mod operator + quadratic *= quadratic; + quadratic = quadratic % lowcut; + //sets density of the centering force + if (noiseAL < 0) {flipL = true;} + else {flipL = false;} + if (noiseAR < 0) {flipR = true;} + else {flipR = false;} + } + + + if (flipL) noiseAL += (rand()/(double)RAND_MAX); + else noiseAL -= (rand()/(double)RAND_MAX); + if (flipR) noiseAR += (rand()/(double)RAND_MAX); + else noiseAR -= (rand()/(double)RAND_MAX); + + if (filterflip) + { + noiseBL *= invcutoffL; noiseBL += (noiseAL*cutoffL); + inputSampleL = noiseBL+noiseCL; + rumbleAL *= (1.0-rumblecutoff); + rumbleAL += (inputSampleL*rumblecutoff); + + noiseBR *= invcutoffR; noiseBR += (noiseAR*cutoffR); + inputSampleR = noiseBR+noiseCR; + rumbleAR *= (1.0-rumblecutoff); + rumbleAR += (inputSampleR*rumblecutoff); + } + else + { + noiseCL *= invcutoffL; noiseCL += (noiseAL*cutoffL); + inputSampleL = noiseBL+noiseCL; + rumbleBL *= (1.0-rumblecutoff); + rumbleBL += (inputSampleL*rumblecutoff); + + noiseCR *= invcutoffR; noiseCR += (noiseAR*cutoffR); + inputSampleR = noiseBR+noiseCR; + rumbleBR *= (1.0-rumblecutoff); + rumbleBR += (inputSampleR*rumblecutoff); + } + + inputSampleL -= (rumbleAL+rumbleBL); + inputSampleL *= (1.0-rumblecutoff); + + inputSampleR -= (rumbleAR+rumbleBR); + inputSampleR *= (1.0-rumblecutoff); + + inputSampleL *= wet; + inputSampleL += (drySampleL * dry); + + inputSampleR *= wet; + inputSampleR += (drySampleR * dry); + //apply the dry to the noise + + bL[9] = bL[8]; bL[8] = bL[7]; bL[7] = bL[6]; bL[6] = bL[5]; + bL[5] = bL[4]; bL[4] = bL[3]; bL[3] = bL[2]; bL[2] = bL[1]; + bL[1] = bL[0]; bL[0] = accumulatorSampleL = inputSampleL; + + bR[9] = bR[8]; bR[8] = bR[7]; bR[7] = bR[6]; bR[6] = bR[5]; + bR[5] = bR[4]; bR[4] = bR[3]; bR[3] = bR[2]; bR[2] = bR[1]; + bR[1] = bR[0]; bR[0] = accumulatorSampleR = inputSampleR; + + accumulatorSampleL *= f[0]; + accumulatorSampleL += (bL[1] * f[1]); + accumulatorSampleL += (bL[2] * f[2]); + accumulatorSampleL += (bL[3] * f[3]); + accumulatorSampleL += (bL[4] * f[4]); + accumulatorSampleL += (bL[5] * f[5]); + accumulatorSampleL += (bL[6] * f[6]); + accumulatorSampleL += (bL[7] * f[7]); + accumulatorSampleL += (bL[8] * f[8]); + accumulatorSampleL += (bL[9] * f[9]); + //we are doing our repetitive calculations on a separate value + accumulatorSampleR *= f[0]; + accumulatorSampleR += (bR[1] * f[1]); + accumulatorSampleR += (bR[2] * f[2]); + accumulatorSampleR += (bR[3] * f[3]); + accumulatorSampleR += (bR[4] * f[4]); + accumulatorSampleR += (bR[5] * f[5]); + accumulatorSampleR += (bR[6] * f[6]); + accumulatorSampleR += (bR[7] * f[7]); + accumulatorSampleR += (bR[8] * f[8]); + accumulatorSampleR += (bR[9] * f[9]); + //we are doing our repetitive calculations on a separate value + + correctionSample = inputSampleL - accumulatorSampleL; + //we're gonna apply the total effect of all these calculations as a single subtract + //(formerly a more complicated algorithm) + inputSampleL -= correctionSample; + //applying the distance calculation to both the dry AND the noise output to blend them + correctionSample = inputSampleR - accumulatorSampleR; + //we're gonna apply the total effect of all these calculations as a single subtract + //(formerly a more complicated algorithm) + inputSampleR -= correctionSample; + //applying the distance calculation to both the dry AND the noise output to blend them + //sometimes I'm really tired and can't do stuff, and I remember trying to simplify this + //and breaking it somehow. So, there ya go, strange obtuse code. + + //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 Noise::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames) +{ + double* in1 = inputs[0]; + double* in2 = inputs[1]; + double* out1 = outputs[0]; + double* out2 = outputs[1]; + + double cutoffL; + double cutoffR; + double cutofftarget = (A*3.5); + double rumblecutoff = cutofftarget * 0.005; + double invcutoffL; + double invcutoffR; + double drySampleL; + double drySampleR; + long double inputSampleL; + long double inputSampleR; + double highpass = C*38.0; + int lowcut = floor(highpass); + int dcut; + if (lowcut > 37) {dcut= 1151;} + if (lowcut == 37) {dcut= 1091;} + if (lowcut == 36) {dcut= 1087;} + if (lowcut == 35) {dcut= 1031;} + if (lowcut == 34) {dcut= 1013;} + if (lowcut == 33) {dcut= 971;} + if (lowcut == 32) {dcut= 907;} + if (lowcut == 31) {dcut= 839;} + if (lowcut == 30) {dcut= 797;} + if (lowcut == 29) {dcut= 733;} + if (lowcut == 28) {dcut= 719;} + if (lowcut == 27) {dcut= 673;} + if (lowcut == 26) {dcut= 613;} + if (lowcut == 25) {dcut= 593;} + if (lowcut == 24) {dcut= 541;} + if (lowcut == 23) {dcut= 479;} + if (lowcut == 22) {dcut= 431;} + if (lowcut == 21) {dcut= 419;} + if (lowcut == 20) {dcut= 373;} + if (lowcut == 19) {dcut= 311;} + if (lowcut == 18) {dcut= 293;} + if (lowcut == 17) {dcut= 233;} + if (lowcut == 16) {dcut= 191;} + if (lowcut == 15) {dcut= 173;} + if (lowcut == 14) {dcut= 131;} + if (lowcut == 13) {dcut= 113;} + if (lowcut == 12) {dcut= 71;} + if (lowcut == 11) {dcut= 53;} + if (lowcut == 10) {dcut= 31;} + if (lowcut == 9) {dcut= 27;} + if (lowcut == 8) {dcut= 23;} + if (lowcut == 7) {dcut= 19;} + if (lowcut == 6) {dcut= 17;} + if (lowcut == 5) {dcut= 13;} + if (lowcut == 4) {dcut= 11;} + if (lowcut == 3) {dcut= 7;} + if (lowcut == 2) {dcut= 5;} + if (lowcut < 2) {dcut= 3;} + highpass = B * 22.0; + lowcut = floor(highpass)+1; + + double decay = 0.001 - ((1.0-pow(1.0-D,3))*0.001); + if (decay == 0.001) decay = 0.1; + double wet = F; + double dry = 1.0 - wet; + wet *= 0.01; //correct large gain issue + double correctionSample; + double accumulatorSampleL; + double accumulatorSampleR; + double overallscale = (E*9.0)+1.0; + double gain = overallscale; + + if (gain > 1.0) {f[0] = 1.0; gain -= 1.0;} else {f[0] = gain; gain = 0.0;} + if (gain > 1.0) {f[1] = 1.0; gain -= 1.0;} else {f[1] = gain; gain = 0.0;} + if (gain > 1.0) {f[2] = 1.0; gain -= 1.0;} else {f[2] = gain; gain = 0.0;} + if (gain > 1.0) {f[3] = 1.0; gain -= 1.0;} else {f[3] = gain; gain = 0.0;} + if (gain > 1.0) {f[4] = 1.0; gain -= 1.0;} else {f[4] = gain; gain = 0.0;} + if (gain > 1.0) {f[5] = 1.0; gain -= 1.0;} else {f[5] = gain; gain = 0.0;} + if (gain > 1.0) {f[6] = 1.0; gain -= 1.0;} else {f[6] = gain; gain = 0.0;} + if (gain > 1.0) {f[7] = 1.0; gain -= 1.0;} else {f[7] = gain; gain = 0.0;} + if (gain > 1.0) {f[8] = 1.0; gain -= 1.0;} else {f[8] = gain; gain = 0.0;} + if (gain > 1.0) {f[9] = 1.0; gain -= 1.0;} else {f[9] = gain; gain = 0.0;} + //there, now we have a neat little moving average with remainders + + if (overallscale < 1.0) overallscale = 1.0; + f[0] /= overallscale; + f[1] /= overallscale; + f[2] /= overallscale; + f[3] /= overallscale; + f[4] /= overallscale; + f[5] /= overallscale; + f[6] /= overallscale; + f[7] /= overallscale; + f[8] /= overallscale; + f[9] /= overallscale; + //and now it's neatly scaled, too + + double fpTemp; + long double fpOld = 0.618033988749894848204586; //golden ratio! + long double fpNew = 1.0 - fpOld; + + while (--sampleFrames >= 0) + { + inputSampleL = *in1; + inputSampleR = *in2; + if (inputSampleL<1.2e-38 && -inputSampleL<1.2e-38) { + static int noisesource = 0; + //this declares a variable before anything else is compiled. It won't keep assigning + //it to 0 for every sample, it's as if the declaration doesn't exist in this context, + //but it lets me add this denormalization fix in a single place rather than updating + //it in three different locations. The variable isn't thread-safe but this is only + //a random seed and we can share it with whatever. + noisesource = noisesource % 1700021; noisesource++; + int residue = noisesource * noisesource; + residue = residue % 170003; residue *= residue; + residue = residue % 17011; residue *= residue; + residue = residue % 1709; residue *= residue; + residue = residue % 173; residue *= residue; + residue = residue % 17; + double applyresidue = residue; + applyresidue *= 0.00000001; + applyresidue *= 0.00000001; + inputSampleL = applyresidue; + } + if (inputSampleR<1.2e-38 && -inputSampleR<1.2e-38) { + static int noisesource = 0; + noisesource = noisesource % 1700021; noisesource++; + int residue = noisesource * noisesource; + residue = residue % 170003; residue *= residue; + residue = residue % 17011; residue *= residue; + residue = residue % 1709; residue *= residue; + residue = residue % 173; residue *= residue; + residue = residue % 17; + double applyresidue = residue; + applyresidue *= 0.00000001; + applyresidue *= 0.00000001; + inputSampleR = applyresidue; + //this denormalization routine produces a white noise at -300 dB which the noise + //shaping will interact with to produce a bipolar output, but the noise is actually + //all positive. That should stop any variables from going denormal, and the routine + //only kicks in if digital black is input. As a final touch, if you save to 24-bit + //the silence will return to being digital black again. + } + drySampleL = inputSampleL; + drySampleR = inputSampleR; + + if (surgeL<fabs(inputSampleL)) + { + surgeL += (rand()/(double)RAND_MAX)*(fabs(inputSampleL)-surgeL); + if (surgeL > 1.0) surgeL = 1.0; + } + else + { + surgeL -= ((rand()/(double)RAND_MAX)*(surgeL-fabs(inputSampleL))*decay); + if (surgeL < 0.0) surgeL = 0.0; + } + + cutoffL = pow((cutofftarget*surgeL),5); + if (cutoffL > 1.0) cutoffL = 1.0; + invcutoffL = 1.0 - cutoffL; + //set up modified cutoff L + + if (surgeR<fabs(inputSampleR)) + { + surgeR += (rand()/(double)RAND_MAX)*(fabs(inputSampleR)-surgeR); + if (surgeR > 1.0) surgeR = 1.0; + } + else + { + surgeR -= ((rand()/(double)RAND_MAX)*(surgeR-fabs(inputSampleR))*decay); + if (surgeR < 0.0) surgeR = 0.0; + } + + cutoffR = pow((cutofftarget*surgeR),5); + if (cutoffR > 1.0) cutoffR = 1.0; + invcutoffR = 1.0 - cutoffR; + //set up modified cutoff R + + flipL = !flipL; + flipR = !flipR; + filterflip = !filterflip; + quadratic -= 1; + if (quadratic < 0) + { + position += 1; + quadratic = position * position; + quadratic = quadratic % 170003; //% is C++ mod operator + quadratic *= quadratic; + quadratic = quadratic % 17011; //% is C++ mod operator + quadratic *= quadratic; + //quadratic = quadratic % 1709; //% is C++ mod operator + //quadratic *= quadratic; + quadratic = quadratic % dcut; //% is C++ mod operator + quadratic *= quadratic; + quadratic = quadratic % lowcut; + //sets density of the centering force + if (noiseAL < 0) {flipL = true;} + else {flipL = false;} + if (noiseAR < 0) {flipR = true;} + else {flipR = false;} + } + + + if (flipL) noiseAL += (rand()/(double)RAND_MAX); + else noiseAL -= (rand()/(double)RAND_MAX); + if (flipR) noiseAR += (rand()/(double)RAND_MAX); + else noiseAR -= (rand()/(double)RAND_MAX); + + if (filterflip) + { + noiseBL *= invcutoffL; noiseBL += (noiseAL*cutoffL); + inputSampleL = noiseBL+noiseCL; + rumbleAL *= (1.0-rumblecutoff); + rumbleAL += (inputSampleL*rumblecutoff); + + noiseBR *= invcutoffR; noiseBR += (noiseAR*cutoffR); + inputSampleR = noiseBR+noiseCR; + rumbleAR *= (1.0-rumblecutoff); + rumbleAR += (inputSampleR*rumblecutoff); + } + else + { + noiseCL *= invcutoffL; noiseCL += (noiseAL*cutoffL); + inputSampleL = noiseBL+noiseCL; + rumbleBL *= (1.0-rumblecutoff); + rumbleBL += (inputSampleL*rumblecutoff); + + noiseCR *= invcutoffR; noiseCR += (noiseAR*cutoffR); + inputSampleR = noiseBR+noiseCR; + rumbleBR *= (1.0-rumblecutoff); + rumbleBR += (inputSampleR*rumblecutoff); + } + + inputSampleL -= (rumbleAL+rumbleBL); + inputSampleL *= (1.0-rumblecutoff); + + inputSampleR -= (rumbleAR+rumbleBR); + inputSampleR *= (1.0-rumblecutoff); + + inputSampleL *= wet; + inputSampleL += (drySampleL * dry); + + inputSampleR *= wet; + inputSampleR += (drySampleR * dry); + //apply the dry to the noise + + bL[9] = bL[8]; bL[8] = bL[7]; bL[7] = bL[6]; bL[6] = bL[5]; + bL[5] = bL[4]; bL[4] = bL[3]; bL[3] = bL[2]; bL[2] = bL[1]; + bL[1] = bL[0]; bL[0] = accumulatorSampleL = inputSampleL; + + bR[9] = bR[8]; bR[8] = bR[7]; bR[7] = bR[6]; bR[6] = bR[5]; + bR[5] = bR[4]; bR[4] = bR[3]; bR[3] = bR[2]; bR[2] = bR[1]; + bR[1] = bR[0]; bR[0] = accumulatorSampleR = inputSampleR; + + accumulatorSampleL *= f[0]; + accumulatorSampleL += (bL[1] * f[1]); + accumulatorSampleL += (bL[2] * f[2]); + accumulatorSampleL += (bL[3] * f[3]); + accumulatorSampleL += (bL[4] * f[4]); + accumulatorSampleL += (bL[5] * f[5]); + accumulatorSampleL += (bL[6] * f[6]); + accumulatorSampleL += (bL[7] * f[7]); + accumulatorSampleL += (bL[8] * f[8]); + accumulatorSampleL += (bL[9] * f[9]); + //we are doing our repetitive calculations on a separate value + accumulatorSampleR *= f[0]; + accumulatorSampleR += (bR[1] * f[1]); + accumulatorSampleR += (bR[2] * f[2]); + accumulatorSampleR += (bR[3] * f[3]); + accumulatorSampleR += (bR[4] * f[4]); + accumulatorSampleR += (bR[5] * f[5]); + accumulatorSampleR += (bR[6] * f[6]); + accumulatorSampleR += (bR[7] * f[7]); + accumulatorSampleR += (bR[8] * f[8]); + accumulatorSampleR += (bR[9] * f[9]); + //we are doing our repetitive calculations on a separate value + + correctionSample = inputSampleL - accumulatorSampleL; + //we're gonna apply the total effect of all these calculations as a single subtract + //(formerly a more complicated algorithm) + inputSampleL -= correctionSample; + //applying the distance calculation to both the dry AND the noise output to blend them + correctionSample = inputSampleR - accumulatorSampleR; + //we're gonna apply the total effect of all these calculations as a single subtract + //(formerly a more complicated algorithm) + inputSampleR -= correctionSample; + //applying the distance calculation to both the dry AND the noise output to blend them + //sometimes I'm really tired and can't do stuff, and I remember trying to simplify this + //and breaking it somehow. So, there ya go, strange obtuse code. + + //noise shaping to 64-bit floating point + if (fpFlip) { + fpTemp = inputSampleL; + fpNShapeLA = (fpNShapeLA*fpOld)+((inputSampleL-fpTemp)*fpNew); + inputSampleL += fpNShapeLA; + fpTemp = inputSampleR; + fpNShapeRA = (fpNShapeRA*fpOld)+((inputSampleR-fpTemp)*fpNew); + inputSampleR += fpNShapeRA; + } + else { + fpTemp = inputSampleL; + fpNShapeLB = (fpNShapeLB*fpOld)+((inputSampleL-fpTemp)*fpNew); + inputSampleL += fpNShapeLB; + fpTemp = inputSampleR; + fpNShapeRB = (fpNShapeRB*fpOld)+((inputSampleR-fpTemp)*fpNew); + inputSampleR += fpNShapeRB; + } + fpFlip = !fpFlip; + //end noise shaping on 64 bit output + + *out1 = inputSampleL; + *out2 = inputSampleR; + + *in1++; + *in2++; + *out1++; + *out2++; + } +}
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