diff options
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/WinVST/DeEss | |
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/WinVST/DeEss')
-rwxr-xr-x | plugins/WinVST/DeEss/.vs/Console4Channel64/v14/.suo | bin | 0 -> 32768 bytes | |||
-rwxr-xr-x | plugins/WinVST/DeEss/.vs/VSTProject/v14/.suo | bin | 0 -> 22528 bytes | |||
-rwxr-xr-x | plugins/WinVST/DeEss/DeEss.cpp | 152 | ||||
-rwxr-xr-x | plugins/WinVST/DeEss/DeEss.h | 115 | ||||
-rwxr-xr-x | plugins/WinVST/DeEss/DeEssProc.cpp | 348 | ||||
-rwxr-xr-x | plugins/WinVST/DeEss/VSTProject.sln | 28 | ||||
-rwxr-xr-x | plugins/WinVST/DeEss/VSTProject.vcxproj | 183 | ||||
-rwxr-xr-x | plugins/WinVST/DeEss/VSTProject.vcxproj.filters | 48 | ||||
-rwxr-xr-x | plugins/WinVST/DeEss/VSTProject.vcxproj.user | 19 | ||||
-rwxr-xr-x | plugins/WinVST/DeEss/vstplug.def | 3 |
10 files changed, 896 insertions, 0 deletions
diff --git a/plugins/WinVST/DeEss/.vs/Console4Channel64/v14/.suo b/plugins/WinVST/DeEss/.vs/Console4Channel64/v14/.suo Binary files differnew file mode 100755 index 0000000..777b846 --- /dev/null +++ b/plugins/WinVST/DeEss/.vs/Console4Channel64/v14/.suo diff --git a/plugins/WinVST/DeEss/.vs/VSTProject/v14/.suo b/plugins/WinVST/DeEss/.vs/VSTProject/v14/.suo Binary files differnew file mode 100755 index 0000000..bf3a84e --- /dev/null +++ b/plugins/WinVST/DeEss/.vs/VSTProject/v14/.suo diff --git a/plugins/WinVST/DeEss/DeEss.cpp b/plugins/WinVST/DeEss/DeEss.cpp new file mode 100755 index 0000000..64a9503 --- /dev/null +++ b/plugins/WinVST/DeEss/DeEss.cpp @@ -0,0 +1,152 @@ +/* ======================================== + * DeEss - DeEss.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __DeEss_H +#include "DeEss.h" +#endif + +AudioEffect* createEffectInstance(audioMasterCallback audioMaster) {return new DeEss(audioMaster);} + +DeEss::DeEss(audioMasterCallback audioMaster) : + AudioEffectX(audioMaster, kNumPrograms, kNumParameters) +{ + A = 0.0; + B = 0.5; //-48.0 to 0.0 + C = 0.5; + + s1L = s2L = s3L = s4L = s5L = s6L= s7L = 0.0; + m1L = m2L = m3L = m4L = m5L = m6L = 0.0; + c1L = c2L = c3L = c4L = c5L = 0.0; + ratioAL = ratioBL = 1.0; + iirSampleAL = 0.0; + iirSampleBL = 0.0; + + s1R = s2R = s3R = s4R = s5R = s6R = s7R = 0.0; + m1R = m2R = m3R = m4R = m5R = m6R = 0.0; + c1R = c2R = c3R = c4R = c5R = 0.0; + ratioAR = ratioBR = 1.0; + iirSampleAR = 0.0; + iirSampleBR = 0.0; + + flip = false; + + fpNShapeL = 0.0; + fpNShapeR = 0.0; + //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 +} + +DeEss::~DeEss() {} +VstInt32 DeEss::getVendorVersion () {return 1000;} +void DeEss::setProgramName(char *name) {vst_strncpy (_programName, name, kVstMaxProgNameLen);} +void DeEss::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 DeEss::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 DeEss::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 DeEss::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 DeEss::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 DeEss::getParameterName(VstInt32 index, char *text) { + switch (index) { + case kParamA: vst_strncpy (text, "Intense", kVstMaxParamStrLen); break; + case kParamB: vst_strncpy (text, "Max DS", kVstMaxParamStrLen); break; + case kParamC: vst_strncpy (text, "Freq", kVstMaxParamStrLen); break; + default: break; // unknown parameter, shouldn't happen! + } //this is our labels for displaying in the VST host +} + +void DeEss::getParameterDisplay(VstInt32 index, char *text) { + switch (index) { + case kParamA: float2string (A, text, kVstMaxParamStrLen); break; + case kParamB: float2string ((B-1.0)*48.0, 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 DeEss::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 DeEss::canDo(char *text) +{ return (_canDo.find(text) == _canDo.end()) ? -1: 1; } // 1 = yes, -1 = no, 0 = don't know + +bool DeEss::getEffectName(char* name) { + vst_strncpy(name, "DeEss", kVstMaxProductStrLen); return true; +} + +VstPlugCategory DeEss::getPlugCategory() {return kPlugCategEffect;} + +bool DeEss::getProductString(char* text) { + vst_strncpy (text, "airwindows DeEss", kVstMaxProductStrLen); return true; +} + +bool DeEss::getVendorString(char* text) { + vst_strncpy (text, "airwindows", kVstMaxVendorStrLen); return true; +} diff --git a/plugins/WinVST/DeEss/DeEss.h b/plugins/WinVST/DeEss/DeEss.h new file mode 100755 index 0000000..dcab1e6 --- /dev/null +++ b/plugins/WinVST/DeEss/DeEss.h @@ -0,0 +1,115 @@ +/* ======================================== + * DeEss - DeEss.h + * Created 8/12/11 by SPIAdmin + * Copyright (c) 2011 __MyCompanyName__, All rights reserved + * ======================================== */ + +#ifndef __DeEss_H +#define __DeEss_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 = 'dess'; //Change this to what the AU identity is! + +class DeEss : + public AudioEffectX +{ +public: + DeEss(audioMasterCallback audioMaster); + ~DeEss(); + 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 s1L; + double s2L; + double s3L; + double s4L; + double s5L; + double s6L; + double s7L; + double m1L; + double m2L; + double m3L; + double m4L; + double m5L; + double m6L; + double c1L; + double c2L; + double c3L; + double c4L; + double c5L; + double ratioAL; + double ratioBL; + double iirSampleAL; + double iirSampleBL; + + double s1R; + double s2R; + double s3R; + double s4R; + double s5R; + double s6R; + double s7R; + double m1R; + double m2R; + double m3R; + double m4R; + double m5R; + double m6R; + double c1R; + double c2R; + double c3R; + double c4R; + double c5R; + double ratioAR; + double ratioBR; + double iirSampleAR; + double iirSampleBR; + + + bool flip; + + long double fpNShapeL; + long double fpNShapeR; + //default stuff + + float A; + float B; + float C; +}; + +#endif diff --git a/plugins/WinVST/DeEss/DeEssProc.cpp b/plugins/WinVST/DeEss/DeEssProc.cpp new file mode 100755 index 0000000..9a8e14b --- /dev/null +++ b/plugins/WinVST/DeEss/DeEssProc.cpp @@ -0,0 +1,348 @@ +/* ======================================== + * DeEss - DeEss.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __DeEss_H +#include "DeEss.h" +#endif + +void DeEss::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 intensity = pow(A,5)*(8192/overallscale); + double maxdess = 1.0 / pow(10.0,((B-1.0)*48.0)/20); + double iirAmount = pow(C,2)/overallscale; + double offset; + double sense; + double recovery; + double attackspeed; + + while (--sampleFrames >= 0) + { + long double inputSampleL = *in1; + long double inputSampleR = *in2; + + static int noisesourceL = 0; + static int noisesourceR = 850010; + int residue; + double applyresidue; + + noisesourceL = noisesourceL % 1700021; noisesourceL++; + residue = noisesourceL * noisesourceL; + residue = residue % 170003; residue *= residue; + residue = residue % 17011; residue *= residue; + residue = residue % 1709; residue *= residue; + residue = residue % 173; residue *= residue; + residue = residue % 17; + applyresidue = residue; + applyresidue *= 0.00000001; + applyresidue *= 0.00000001; + inputSampleL += applyresidue; + if (inputSampleL<1.2e-38 && -inputSampleL<1.2e-38) { + inputSampleL -= applyresidue; + } + + noisesourceR = noisesourceR % 1700021; noisesourceR++; + residue = noisesourceR * noisesourceR; + residue = residue % 170003; residue *= residue; + residue = residue % 17011; residue *= residue; + residue = residue % 1709; residue *= residue; + residue = residue % 173; residue *= residue; + residue = residue % 17; + applyresidue = residue; + applyresidue *= 0.00000001; + applyresidue *= 0.00000001; + inputSampleR += applyresidue; + if (inputSampleR<1.2e-38 && -inputSampleR<1.2e-38) { + inputSampleR -= applyresidue; + } + //for live air, we always apply the dither noise. Then, if our result is + //effectively digital black, we'll subtract it aDeEss. We want a 'air' hiss + + s3L = s2L; + s2L = s1L; + s1L = inputSampleL; + m1L = (s1L-s2L)*((s1L-s2L)/1.3); + m2L = (s2L-s3L)*((s1L-s2L)/1.3); + sense = fabs((m1L-m2L)*((m1L-m2L)/1.3)); + //this will be 0 for smooth, high for SSS + attackspeed = 7.0+(sense*1024); + //this does not vary with intensity, but it does react to onset transients + + sense = 1.0+(intensity*intensity*sense); + if (sense > intensity) {sense = intensity;} + //this will be 1 for smooth, 'intensity' for SSS + recovery = 1.0+(0.01/sense); + //this will be 1.1 for smooth, 1.0000000...1 for SSS + + offset = 1.0-fabs(inputSampleL); + + if (flip) { + iirSampleAL = (iirSampleAL * (1 - (offset * iirAmount))) + (inputSampleL * (offset * iirAmount)); + if (ratioAL < sense) + {ratioAL = ((ratioAL*attackspeed)+sense)/(attackspeed+1.0);} + else + {ratioAL = 1.0+((ratioAL-1.0)/recovery);} + //returny to 1/1 code + if (ratioAL > maxdess){ratioAL = maxdess;} + inputSampleL = iirSampleAL+((inputSampleL-iirSampleAL)/ratioAL); + } + else { + iirSampleBL = (iirSampleBL * (1 - (offset * iirAmount))) + (inputSampleL * (offset * iirAmount)); + if (ratioBL < sense) + {ratioBL = ((ratioBL*attackspeed)+sense)/(attackspeed+1.0);} + else + {ratioBL = 1.0+((ratioBL-1.0)/recovery);} + //returny to 1/1 code + if (ratioBL > maxdess){ratioBL = maxdess;} + inputSampleL = iirSampleBL+((inputSampleL-iirSampleBL)/ratioBL); + } //have the ratio chase Sense + + s3R = s2R; + s2R = s1R; + s1R = inputSampleR; + m1R = (s1R-s2R)*((s1R-s2R)/1.3); + m2R = (s2R-s3R)*((s1R-s2R)/1.3); + sense = fabs((m1R-m2R)*((m1R-m2R)/1.3)); + //this will be 0 for smooth, high for SSS + attackspeed = 7.0+(sense*1024); + //this does not vary with intensity, but it does react to onset transients + + sense = 1.0+(intensity*intensity*sense); + if (sense > intensity) {sense = intensity;} + //this will be 1 for smooth, 'intensity' for SSS + recovery = 1.0+(0.01/sense); + //this will be 1.1 for smooth, 1.0000000...1 for SSS + + offset = 1.0-fabs(inputSampleR); + + if (flip) { + iirSampleAR = (iirSampleAR * (1 - (offset * iirAmount))) + (inputSampleR * (offset * iirAmount)); + if (ratioAR < sense) + {ratioAR = ((ratioAR*attackspeed)+sense)/(attackspeed+1.0);} + else + {ratioAR = 1.0+((ratioAR-1.0)/recovery);} + //returny to 1/1 code + if (ratioAR > maxdess){ratioAR = maxdess;} + inputSampleR = iirSampleAR+((inputSampleR-iirSampleAR)/ratioAR); + } + else { + iirSampleBR = (iirSampleBR * (1 - (offset * iirAmount))) + (inputSampleR * (offset * iirAmount)); + if (ratioBR < sense) + {ratioBR = ((ratioBR*attackspeed)+sense)/(attackspeed+1.0);} + else + {ratioBR = 1.0+((ratioBR-1.0)/recovery);} + //returny to 1/1 code + if (ratioBR > maxdess){ratioBR = maxdess;} + inputSampleR = iirSampleBR+((inputSampleR-iirSampleBR)/ratioBR); + } //have the ratio chase Sense + + flip = !flip; + + //noise shaping to 32-bit floating point + float fpTemp = inputSampleL; + fpNShapeL += (inputSampleL-fpTemp); + inputSampleL += fpNShapeL; + //if this confuses you look at the wordlength for fpTemp :) + fpTemp = inputSampleR; + fpNShapeR += (inputSampleR-fpTemp); + inputSampleR += fpNShapeR; + //for deeper space and warmth, we try a non-oscillating noise shaping + //that is kind of ruthless: it will forever retain the rounding errors + //except we'll dial it back a hair at the end of every buffer processed + //end noise shaping on 32 bit output + + *out1 = inputSampleL; + *out2 = inputSampleR; + + *in1++; + *in2++; + *out1++; + *out2++; + } + fpNShapeL *= 0.999999; + fpNShapeR *= 0.999999; + //we will just delicately dial back the FP noise shaping, not even every sample + //this is a good place to put subtle 'no runaway' calculations, though bear in mind + //that it will be called more often when you use shorter sample buffers in the DAW. + //So, very low latency operation will call these calculations more often. +} + +void DeEss::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 intensity = pow(A,5)*(8192/overallscale); + double maxdess = 1.0 / pow(10.0,((B-1.0)*48.0)/20); + double iirAmount = pow(C,2)/overallscale; + double offset; + double sense; + double recovery; + double attackspeed; + + while (--sampleFrames >= 0) + { + long double inputSampleL = *in1; + long double inputSampleR = *in2; + + static int noisesourceL = 0; + static int noisesourceR = 850010; + int residue; + double applyresidue; + + noisesourceL = noisesourceL % 1700021; noisesourceL++; + residue = noisesourceL * noisesourceL; + residue = residue % 170003; residue *= residue; + residue = residue % 17011; residue *= residue; + residue = residue % 1709; residue *= residue; + residue = residue % 173; residue *= residue; + residue = residue % 17; + applyresidue = residue; + applyresidue *= 0.00000001; + applyresidue *= 0.00000001; + inputSampleL += applyresidue; + if (inputSampleL<1.2e-38 && -inputSampleL<1.2e-38) { + inputSampleL -= applyresidue; + } + + noisesourceR = noisesourceR % 1700021; noisesourceR++; + residue = noisesourceR * noisesourceR; + residue = residue % 170003; residue *= residue; + residue = residue % 17011; residue *= residue; + residue = residue % 1709; residue *= residue; + residue = residue % 173; residue *= residue; + residue = residue % 17; + applyresidue = residue; + applyresidue *= 0.00000001; + applyresidue *= 0.00000001; + inputSampleR += applyresidue; + if (inputSampleR<1.2e-38 && -inputSampleR<1.2e-38) { + inputSampleR -= applyresidue; + } + //for live air, we always apply the dither noise. Then, if our result is + //effectively digital black, we'll subtract it aDeEss. We want a 'air' hiss + + s3L = s2L; + s2L = s1L; + s1L = inputSampleL; + m1L = (s1L-s2L)*((s1L-s2L)/1.3); + m2L = (s2L-s3L)*((s1L-s2L)/1.3); + sense = fabs((m1L-m2L)*((m1L-m2L)/1.3)); + //this will be 0 for smooth, high for SSS + attackspeed = 7.0+(sense*1024); + //this does not vary with intensity, but it does react to onset transients + + sense = 1.0+(intensity*intensity*sense); + if (sense > intensity) {sense = intensity;} + //this will be 1 for smooth, 'intensity' for SSS + recovery = 1.0+(0.01/sense); + //this will be 1.1 for smooth, 1.0000000...1 for SSS + + offset = 1.0-fabs(inputSampleL); + + if (flip) { + iirSampleAL = (iirSampleAL * (1 - (offset * iirAmount))) + (inputSampleL * (offset * iirAmount)); + if (ratioAL < sense) + {ratioAL = ((ratioAL*attackspeed)+sense)/(attackspeed+1.0);} + else + {ratioAL = 1.0+((ratioAL-1.0)/recovery);} + //returny to 1/1 code + if (ratioAL > maxdess){ratioAL = maxdess;} + inputSampleL = iirSampleAL+((inputSampleL-iirSampleAL)/ratioAL); + } + else { + iirSampleBL = (iirSampleBL * (1 - (offset * iirAmount))) + (inputSampleL * (offset * iirAmount)); + if (ratioBL < sense) + {ratioBL = ((ratioBL*attackspeed)+sense)/(attackspeed+1.0);} + else + {ratioBL = 1.0+((ratioBL-1.0)/recovery);} + //returny to 1/1 code + if (ratioBL > maxdess){ratioBL = maxdess;} + inputSampleL = iirSampleBL+((inputSampleL-iirSampleBL)/ratioBL); + } //have the ratio chase Sense + + s3R = s2R; + s2R = s1R; + s1R = inputSampleR; + m1R = (s1R-s2R)*((s1R-s2R)/1.3); + m2R = (s2R-s3R)*((s1R-s2R)/1.3); + sense = fabs((m1R-m2R)*((m1R-m2R)/1.3)); + //this will be 0 for smooth, high for SSS + attackspeed = 7.0+(sense*1024); + //this does not vary with intensity, but it does react to onset transients + + sense = 1.0+(intensity*intensity*sense); + if (sense > intensity) {sense = intensity;} + //this will be 1 for smooth, 'intensity' for SSS + recovery = 1.0+(0.01/sense); + //this will be 1.1 for smooth, 1.0000000...1 for SSS + + offset = 1.0-fabs(inputSampleR); + + if (flip) { + iirSampleAR = (iirSampleAR * (1 - (offset * iirAmount))) + (inputSampleR * (offset * iirAmount)); + if (ratioAR < sense) + {ratioAR = ((ratioAR*attackspeed)+sense)/(attackspeed+1.0);} + else + {ratioAR = 1.0+((ratioAR-1.0)/recovery);} + //returny to 1/1 code + if (ratioAR > maxdess){ratioAR = maxdess;} + inputSampleR = iirSampleAR+((inputSampleR-iirSampleAR)/ratioAR); + } + else { + iirSampleBR = (iirSampleBR * (1 - (offset * iirAmount))) + (inputSampleR * (offset * iirAmount)); + if (ratioBR < sense) + {ratioBR = ((ratioBR*attackspeed)+sense)/(attackspeed+1.0);} + else + {ratioBR = 1.0+((ratioBR-1.0)/recovery);} + //returny to 1/1 code + if (ratioBR > maxdess){ratioBR = maxdess;} + inputSampleR = iirSampleBR+((inputSampleR-iirSampleBR)/ratioBR); + } //have the ratio chase Sense + + flip = !flip; + + //noise shaping to 64-bit floating point + double fpTemp = inputSampleL; + fpNShapeL += (inputSampleL-fpTemp); + inputSampleL += fpNShapeL; + //if this confuses you look at the wordlength for fpTemp :) + fpTemp = inputSampleR; + fpNShapeR += (inputSampleR-fpTemp); + inputSampleR += fpNShapeR; + //for deeper space and warmth, we try a non-oscillating noise shaping + //that is kind of ruthless: it will forever retain the rounding errors + //except we'll dial it back a hair at the end of every buffer processed + //end noise shaping on 64 bit output + + *out1 = inputSampleL; + *out2 = inputSampleR; + + *in1++; + *in2++; + *out1++; + *out2++; + } + fpNShapeL *= 0.999999; + fpNShapeR *= 0.999999; + //we will just delicately dial back the FP noise shaping, not even every sample + //this is a good place to put subtle 'no runaway' calculations, though bear in mind + //that it will be called more often when you use shorter sample buffers in the DAW. + //So, very low latency operation will call these calculations more often. +} diff --git a/plugins/WinVST/DeEss/VSTProject.sln b/plugins/WinVST/DeEss/VSTProject.sln new file mode 100755 index 0000000..694b424 --- /dev/null +++ b/plugins/WinVST/DeEss/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
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+ <Link>
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\ No newline at end of file diff --git a/plugins/WinVST/DeEss/VSTProject.vcxproj.filters b/plugins/WinVST/DeEss/VSTProject.vcxproj.filters new file mode 100755 index 0000000..6c43e45 --- /dev/null +++ b/plugins/WinVST/DeEss/VSTProject.vcxproj.filters @@ -0,0 +1,48 @@ +<?xml version="1.0" encoding="utf-8"?>
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\ No newline at end of file |