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authorChris Johnson <jinx6568@sover.net>2018-12-02 15:04:58 -0500
committerChris Johnson <jinx6568@sover.net>2018-12-02 15:04:58 -0500
commit7b85f3eb20819c11eb67146d826f9e3d8a8e7d39 (patch)
treecee73e1581c048702a8f2d13fb036d23f0f6f9bd /plugins/LinuxVST
parent633be2e22c6648c901f08f3b4cd4e8e14ea86443 (diff)
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Gatelope
Diffstat (limited to 'plugins/LinuxVST')
-rwxr-xr-xplugins/LinuxVST/CMakeLists.txt1
-rwxr-xr-xplugins/LinuxVST/src/Gatelope/Gatelope.cpp162
-rwxr-xr-xplugins/LinuxVST/src/Gatelope/Gatelope.h82
-rwxr-xr-xplugins/LinuxVST/src/Gatelope/GatelopeProc.cpp436
4 files changed, 681 insertions, 0 deletions
diff --git a/plugins/LinuxVST/CMakeLists.txt b/plugins/LinuxVST/CMakeLists.txt
index aa842bc..44e2607 100755
--- a/plugins/LinuxVST/CMakeLists.txt
+++ b/plugins/LinuxVST/CMakeLists.txt
@@ -65,6 +65,7 @@ add_airwindows_plugin(FathomFive)
add_airwindows_plugin(Floor)
add_airwindows_plugin(Fracture)
add_airwindows_plugin(FromTape)
+add_airwindows_plugin(Gatelope)
add_airwindows_plugin(Golem)
add_airwindows_plugin(GrooveWear)
add_airwindows_plugin(GuitarConditioner)
diff --git a/plugins/LinuxVST/src/Gatelope/Gatelope.cpp b/plugins/LinuxVST/src/Gatelope/Gatelope.cpp
new file mode 100755
index 0000000..41ed2af
--- /dev/null
+++ b/plugins/LinuxVST/src/Gatelope/Gatelope.cpp
@@ -0,0 +1,162 @@
+/* ========================================
+ * Gatelope - Gatelope.h
+ * Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __Gatelope_H
+#include "Gatelope.h"
+#endif
+
+AudioEffect* createEffectInstance(audioMasterCallback audioMaster) {return new Gatelope(audioMaster);}
+
+Gatelope::Gatelope(audioMasterCallback audioMaster) :
+ AudioEffectX(audioMaster, kNumPrograms, kNumParameters)
+{
+ A = 0.0;
+ B = 1.0;
+ C = 0.5;
+ D = 0.0;
+ E = 1.0;
+ iirLowpassAL = 0.0;
+ iirLowpassBL = 0.0;
+ iirHighpassAL = 0.0;
+ iirHighpassBL = 0.0;
+ iirLowpassAR = 0.0;
+ iirLowpassBR = 0.0;
+ iirHighpassAR = 0.0;
+ iirHighpassBR = 0.0;
+ treblefreq = 1.0;
+ bassfreq = 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
+}
+
+Gatelope::~Gatelope() {}
+VstInt32 Gatelope::getVendorVersion () {return 1000;}
+void Gatelope::setProgramName(char *name) {vst_strncpy (_programName, name, kVstMaxProgNameLen);}
+void Gatelope::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 Gatelope::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;
+ /* 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 Gatelope::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]);
+ /* 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 Gatelope::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;
+ default: throw; // unknown parameter, shouldn't happen!
+ }
+}
+
+float Gatelope::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;
+ default: break; // unknown parameter, shouldn't happen!
+ } return 0.0; //we only need to update the relevant name, this is simple to manage
+}
+
+void Gatelope::getParameterName(VstInt32 index, char *text) {
+ switch (index) {
+ case kParamA: vst_strncpy (text, "Thresh", kVstMaxParamStrLen); break;
+ case kParamB: vst_strncpy (text, "TrebSus", kVstMaxParamStrLen); break;
+ case kParamC: vst_strncpy (text, "BassSus", kVstMaxParamStrLen); break;
+ case kParamD: vst_strncpy (text, "AttackS", kVstMaxParamStrLen); break;
+ case kParamE: 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 Gatelope::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;
+ default: break; // unknown parameter, shouldn't happen!
+ } //this displays the values and handles 'popups' where it's discrete choices
+}
+
+void Gatelope::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;
+ default: break; // unknown parameter, shouldn't happen!
+ }
+}
+
+VstInt32 Gatelope::canDo(char *text)
+{ return (_canDo.find(text) == _canDo.end()) ? -1: 1; } // 1 = yes, -1 = no, 0 = don't know
+
+bool Gatelope::getEffectName(char* name) {
+ vst_strncpy(name, "Gatelope", kVstMaxProductStrLen); return true;
+}
+
+VstPlugCategory Gatelope::getPlugCategory() {return kPlugCategEffect;}
+
+bool Gatelope::getProductString(char* text) {
+ vst_strncpy (text, "airwindows Gatelope", kVstMaxProductStrLen); return true;
+}
+
+bool Gatelope::getVendorString(char* text) {
+ vst_strncpy (text, "airwindows", kVstMaxVendorStrLen); return true;
+}
diff --git a/plugins/LinuxVST/src/Gatelope/Gatelope.h b/plugins/LinuxVST/src/Gatelope/Gatelope.h
new file mode 100755
index 0000000..06a0c55
--- /dev/null
+++ b/plugins/LinuxVST/src/Gatelope/Gatelope.h
@@ -0,0 +1,82 @@
+/* ========================================
+ * Gatelope - Gatelope.h
+ * Created 8/12/11 by SPIAdmin
+ * Copyright (c) 2011 __MyCompanyName__, All rights reserved
+ * ======================================== */
+
+#ifndef __Gatelope_H
+#define __Gatelope_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,
+ kNumParameters = 5
+}; //
+
+const int kNumPrograms = 0;
+const int kNumInputs = 2;
+const int kNumOutputs = 2;
+const unsigned long kUniqueId = 'gtlp'; //Change this to what the AU identity is!
+
+class Gatelope :
+ public AudioEffectX
+{
+public:
+ Gatelope(audioMasterCallback audioMaster);
+ ~Gatelope();
+ 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 iirLowpassAR;
+ double iirLowpassBR;
+ double iirHighpassAR;
+ double iirHighpassBR;
+ double iirLowpassAL;
+ double iirLowpassBL;
+ double iirHighpassAL;
+ double iirHighpassBL;
+ double treblefreq;
+ double bassfreq;
+ bool flip;
+ long double fpNShapeL;
+ long double fpNShapeR;
+ //default stuff
+
+ float A;
+ float B;
+ float C;
+ float D;
+ float E; //parameters. Always 0-1, and we scale/alter them elsewhere.
+
+};
+
+#endif
diff --git a/plugins/LinuxVST/src/Gatelope/GatelopeProc.cpp b/plugins/LinuxVST/src/Gatelope/GatelopeProc.cpp
new file mode 100755
index 0000000..dc81def
--- /dev/null
+++ b/plugins/LinuxVST/src/Gatelope/GatelopeProc.cpp
@@ -0,0 +1,436 @@
+/* ========================================
+ * Gatelope - Gatelope.h
+ * Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __Gatelope_H
+#include "Gatelope.h"
+#endif
+
+void Gatelope::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();
+ //speed settings around release
+ double threshold = pow(A,2);
+ //gain settings around threshold
+ double trebledecay = pow(1.0-B,2)/4196.0;
+ double bassdecay = pow(1.0-C,2)/8192.0;
+ double slowAttack = (pow(D,3)*3)+0.003;
+ double wet = E;
+ slowAttack /= overallscale;
+ trebledecay /= overallscale;
+ bassdecay /= overallscale;
+ trebledecay += 1.0;
+ bassdecay += 1.0;
+ double attackSpeed;
+ double highestSample;
+ //this VST version comes from the AU, Gatelinked, because it's stereo.
+ //if used on a mono track it'll act like the mono N to N
+
+ 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 again. We want a 'air' hiss
+ double drySampleL = inputSampleL;
+ double drySampleR = inputSampleR;
+
+ if (fabs(inputSampleL) > fabs(inputSampleR)) {
+ attackSpeed = slowAttack - (fabs(inputSampleL)*slowAttack*0.5);
+ highestSample = fabs(inputSampleL);
+ } else {
+ attackSpeed = slowAttack - (fabs(inputSampleR)*slowAttack*0.5); //we're triggering off the highest amplitude
+ highestSample = fabs(inputSampleR); //and making highestSample the abs() of that amplitude
+ }
+
+ if (attackSpeed < 0.0) attackSpeed = 0.0;
+ //softening onset click depending on how hard we're getting it
+
+ if (flip)
+ {
+ if (highestSample > threshold)
+ {
+ treblefreq += attackSpeed;
+ if (treblefreq > 1.0) treblefreq = 1.0;
+ bassfreq -= attackSpeed;
+ bassfreq -= attackSpeed;
+ if (bassfreq < 0.0) bassfreq = 0.0;
+ iirLowpassAL = iirLowpassBL = inputSampleL;
+ iirHighpassAL = iirHighpassBL = 0.0;
+ iirLowpassAR = iirLowpassBR = inputSampleR;
+ iirHighpassAR = iirHighpassBR = 0.0;
+ }
+ else
+ {
+ treblefreq -= bassfreq;
+ treblefreq /= trebledecay;
+ treblefreq += bassfreq;
+ bassfreq -= treblefreq;
+ bassfreq /= bassdecay;
+ bassfreq += treblefreq;
+ }
+
+ if (treblefreq >= 1.0) {
+ iirLowpassAL = inputSampleL;
+ iirLowpassAR = inputSampleR;
+ } else {
+ iirLowpassAL = (iirLowpassAL * (1.0 - treblefreq)) + (inputSampleL * treblefreq);
+ iirLowpassAR = (iirLowpassAR * (1.0 - treblefreq)) + (inputSampleR * treblefreq);
+ }
+
+ if (bassfreq > 0.0) {
+ iirHighpassAL = (iirHighpassAL * (1.0 - bassfreq)) + (inputSampleL * bassfreq);
+ iirHighpassAR = (iirHighpassAR * (1.0 - bassfreq)) + (inputSampleR * bassfreq);
+ } else {
+ iirHighpassAL = 0.0;
+ iirHighpassAR = 0.0;
+ }
+
+ if (treblefreq > bassfreq) {
+ inputSampleL = (iirLowpassAL - iirHighpassAL);
+ inputSampleR = (iirLowpassAR - iirHighpassAR);
+ } else {
+ inputSampleL = 0.0;
+ inputSampleR = 0.0;
+ }
+ }
+ else
+ {
+ if (highestSample > threshold)
+ {
+ treblefreq += attackSpeed;
+ if (treblefreq > 1.0) treblefreq = 1.0;
+ bassfreq -= attackSpeed;
+ bassfreq -= attackSpeed;
+ if (bassfreq < 0.0) bassfreq = 0.0;
+ iirLowpassAL = iirLowpassBL = inputSampleL;
+ iirHighpassAL = iirHighpassBL = 0.0;
+ iirLowpassAR = iirLowpassBR = inputSampleR;
+ iirHighpassAR = iirHighpassBR = 0.0;
+ }
+ else
+ {
+ treblefreq -= bassfreq;
+ treblefreq /= trebledecay;
+ treblefreq += bassfreq;
+ bassfreq -= treblefreq;
+ bassfreq /= bassdecay;
+ bassfreq += treblefreq;
+ }
+
+ if (treblefreq >= 1.0) {
+ iirLowpassBL = inputSampleL;
+ iirLowpassBR = inputSampleR;
+ } else {
+ iirLowpassBL = (iirLowpassBL * (1.0 - treblefreq)) + (inputSampleL * treblefreq);
+ iirLowpassBR = (iirLowpassBR * (1.0 - treblefreq)) + (inputSampleR * treblefreq);
+ }
+
+ if (bassfreq > 0.0) {
+ iirHighpassBL = (iirHighpassBL * (1.0 - bassfreq)) + (inputSampleL * bassfreq);
+ iirHighpassBR = (iirHighpassBR * (1.0 - bassfreq)) + (inputSampleR * bassfreq);
+ } else {
+ iirHighpassBL = 0.0;
+ iirHighpassBR = 0.0;
+ }
+
+ if (treblefreq > bassfreq) {
+ inputSampleL = (iirLowpassBL - iirHighpassBL);
+ inputSampleR = (iirLowpassBR - iirHighpassBR);
+ } else {
+ inputSampleL = 0.0;
+ inputSampleR = 0.0;
+ }
+ }
+ //done full gated envelope filtered effect
+ inputSampleL = ((1-wet)*drySampleL)+(wet*inputSampleL);
+ inputSampleR = ((1-wet)*drySampleR)+(wet*inputSampleR);
+ //we're going to set up a dry/wet control instead of a min. threshold
+
+ 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 Gatelope::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();
+ //speed settings around release
+ double threshold = pow(A,2);
+ //gain settings around threshold
+ double trebledecay = pow(1.0-B,2)/4196.0;
+ double bassdecay = pow(1.0-C,2)/8192.0;
+ double slowAttack = (pow(D,3)*3)+0.003;
+ double wet = E;
+ slowAttack /= overallscale;
+ trebledecay /= overallscale;
+ bassdecay /= overallscale;
+ trebledecay += 1.0;
+ bassdecay += 1.0;
+ double attackSpeed;
+ double highestSample;
+ //this VST version comes from the AU, Gatelinked, because it's stereo.
+ //if used on a mono track it'll act like the mono N to N
+
+ 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 again. We want a 'air' hiss
+ double drySampleL = inputSampleL;
+ double drySampleR = inputSampleR;
+
+ if (fabs(inputSampleL) > fabs(inputSampleR)) {
+ attackSpeed = slowAttack - (fabs(inputSampleL)*slowAttack*0.5);
+ highestSample = fabs(inputSampleL);
+ } else {
+ attackSpeed = slowAttack - (fabs(inputSampleR)*slowAttack*0.5); //we're triggering off the highest amplitude
+ highestSample = fabs(inputSampleR); //and making highestSample the abs() of that amplitude
+ }
+
+ if (attackSpeed < 0.0) attackSpeed = 0.0;
+ //softening onset click depending on how hard we're getting it
+
+ if (flip)
+ {
+ if (highestSample > threshold)
+ {
+ treblefreq += attackSpeed;
+ if (treblefreq > 1.0) treblefreq = 1.0;
+ bassfreq -= attackSpeed;
+ bassfreq -= attackSpeed;
+ if (bassfreq < 0.0) bassfreq = 0.0;
+ iirLowpassAL = iirLowpassBL = inputSampleL;
+ iirHighpassAL = iirHighpassBL = 0.0;
+ iirLowpassAR = iirLowpassBR = inputSampleR;
+ iirHighpassAR = iirHighpassBR = 0.0;
+ }
+ else
+ {
+ treblefreq -= bassfreq;
+ treblefreq /= trebledecay;
+ treblefreq += bassfreq;
+ bassfreq -= treblefreq;
+ bassfreq /= bassdecay;
+ bassfreq += treblefreq;
+ }
+
+ if (treblefreq >= 1.0) {
+ iirLowpassAL = inputSampleL;
+ iirLowpassAR = inputSampleR;
+ } else {
+ iirLowpassAL = (iirLowpassAL * (1.0 - treblefreq)) + (inputSampleL * treblefreq);
+ iirLowpassAR = (iirLowpassAR * (1.0 - treblefreq)) + (inputSampleR * treblefreq);
+ }
+
+ if (bassfreq > 0.0) {
+ iirHighpassAL = (iirHighpassAL * (1.0 - bassfreq)) + (inputSampleL * bassfreq);
+ iirHighpassAR = (iirHighpassAR * (1.0 - bassfreq)) + (inputSampleR * bassfreq);
+ } else {
+ iirHighpassAL = 0.0;
+ iirHighpassAR = 0.0;
+ }
+
+ if (treblefreq > bassfreq) {
+ inputSampleL = (iirLowpassAL - iirHighpassAL);
+ inputSampleR = (iirLowpassAR - iirHighpassAR);
+ } else {
+ inputSampleL = 0.0;
+ inputSampleR = 0.0;
+ }
+ }
+ else
+ {
+ if (highestSample > threshold)
+ {
+ treblefreq += attackSpeed;
+ if (treblefreq > 1.0) treblefreq = 1.0;
+ bassfreq -= attackSpeed;
+ bassfreq -= attackSpeed;
+ if (bassfreq < 0.0) bassfreq = 0.0;
+ iirLowpassAL = iirLowpassBL = inputSampleL;
+ iirHighpassAL = iirHighpassBL = 0.0;
+ iirLowpassAR = iirLowpassBR = inputSampleR;
+ iirHighpassAR = iirHighpassBR = 0.0;
+ }
+ else
+ {
+ treblefreq -= bassfreq;
+ treblefreq /= trebledecay;
+ treblefreq += bassfreq;
+ bassfreq -= treblefreq;
+ bassfreq /= bassdecay;
+ bassfreq += treblefreq;
+ }
+
+ if (treblefreq >= 1.0) {
+ iirLowpassBL = inputSampleL;
+ iirLowpassBR = inputSampleR;
+ } else {
+ iirLowpassBL = (iirLowpassBL * (1.0 - treblefreq)) + (inputSampleL * treblefreq);
+ iirLowpassBR = (iirLowpassBR * (1.0 - treblefreq)) + (inputSampleR * treblefreq);
+ }
+
+ if (bassfreq > 0.0) {
+ iirHighpassBL = (iirHighpassBL * (1.0 - bassfreq)) + (inputSampleL * bassfreq);
+ iirHighpassBR = (iirHighpassBR * (1.0 - bassfreq)) + (inputSampleR * bassfreq);
+ } else {
+ iirHighpassBL = 0.0;
+ iirHighpassBR = 0.0;
+ }
+
+ if (treblefreq > bassfreq) {
+ inputSampleL = (iirLowpassBL - iirHighpassBL);
+ inputSampleR = (iirLowpassBR - iirHighpassBR);
+ } else {
+ inputSampleL = 0.0;
+ inputSampleR = 0.0;
+ }
+ }
+ //done full gated envelope filtered effect
+ inputSampleL = ((1-wet)*drySampleL)+(wet*inputSampleL);
+ inputSampleR = ((1-wet)*drySampleR)+(wet*inputSampleR);
+ //we're going to set up a dry/wet control instead of a min. threshold
+
+ 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.
+}