text: /*
  *	File:		Monitoring2.cpp
  *	
  *	Version:	1.0
  * 
  *	Created:	9/13/21
  *	
  *	Copyright:  Copyright � 2021 Airwindows, All Rights Reserved
  * 
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  /*=============================================================================
  	Monitoring2.cpp
  	
  =============================================================================*/
  #include "Monitoring2.h"
  
  
  //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
  
  AUDIOCOMPONENT_ENTRY(AUBaseFactory, Monitoring2)
  
  
  //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
  //	Monitoring2::Monitoring2
  //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
  Monitoring2::Monitoring2(AudioUnit component)
  	: AUEffectBase(component)
  {
  	CreateElements();
  	Globals()->UseIndexedParameters(kNumberOfParameters);
  	SetParameter(kParam_One, kDefaultValue_ParamOne );
           
  #if AU_DEBUG_DISPATCHER
  	mDebugDispatcher = new AUDebugDispatcher (this);
  #endif
  	
  }
  
  
  //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
  //	Monitoring2::GetParameterValueStrings
  //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
  ComponentResult			Monitoring2::GetParameterValueStrings(AudioUnitScope		inScope,
                                                                  AudioUnitParameterID	inParameterID,
                                                                  CFArrayRef *		outStrings)
  {
  	if ((inScope == kAudioUnitScope_Global) && (inParameterID == kParam_One)) //ID must be actual name of parameter identifier, not number
  	{
  		if (outStrings == NULL) return noErr;
  		CFStringRef strings [] =
  		{
  			kMenuItem_DKAD,
  			kMenuItem_DKCD,
  			kMenuItem_PEAK,
  			kMenuItem_SLEW,
  			kMenuItem_SUBS,
  			kMenuItem_MONO,
  			kMenuItem_SIDE,
  			kMenuItem_VINYL,
  			kMenuItem_AURAT,
  			kMenuItem_MONORAT,
  			kMenuItem_MONOLAT,
  			kMenuItem_PHONE,
  			kMenuItem_CANSA,
  			kMenuItem_CANSB,
  			kMenuItem_CANSC,
  			kMenuItem_CANSD,
  			kMenuItem_TRICK
  		};
  		*outStrings = CFArrayCreate (
  									 NULL,
  									 (const void **) strings,
  									 (sizeof (strings) / sizeof (strings [0])),
  									 NULL
  									 );
  		return noErr;
  	}
      return kAudioUnitErr_InvalidProperty;
  }
  
  
  
  //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
  //	Monitoring2::GetParameterInfo
  //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
  ComponentResult			Monitoring2::GetParameterInfo(AudioUnitScope		inScope,
                                                          AudioUnitParameterID	inParameterID,
                                                          AudioUnitParameterInfo	&outParameterInfo )
  {
  	ComponentResult result = noErr;
  
  	outParameterInfo.flags = 	kAudioUnitParameterFlag_IsWritable
  						|		kAudioUnitParameterFlag_IsReadable;
      
      if (inScope == kAudioUnitScope_Global) {
          switch(inParameterID)
          {
  			case kParam_One:
                  AUBase::FillInParameterName (outParameterInfo, kParameterOneName, false);
  				outParameterInfo.unit = kAudioUnitParameterUnit_Indexed;
                  outParameterInfo.minValue = kDKAD;
                  outParameterInfo.maxValue = kTRICK;
                  outParameterInfo.defaultValue = kDefaultValue_ParamOne;
                  break;
  			default:
                  result = kAudioUnitErr_InvalidParameter;
                  break;
              }
  	} else {
          result = kAudioUnitErr_InvalidParameter;
      }
      
  
  
  	return result;
  }
  
  //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
  //	Monitoring2::GetPropertyInfo
  //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
  ComponentResult			Monitoring2::GetPropertyInfo (AudioUnitPropertyID	inID,
                                                          AudioUnitScope		inScope,
                                                          AudioUnitElement	inElement,
                                                          UInt32 &		outDataSize,
                                                          Boolean &		outWritable)
  {
  	return AUEffectBase::GetPropertyInfo (inID, inScope, inElement, outDataSize, outWritable);
  }
  
  //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
  // state that plugin supports only stereo-in/stereo-out processing
  UInt32 Monitoring2::SupportedNumChannels(const AUChannelInfo ** outInfo)
  {
  	if (outInfo != NULL)
  	{
  		static AUChannelInfo info;
  		info.inChannels = 2;
  		info.outChannels = 2;
  		*outInfo = &info;
  	}
  
  	return 1;
  }
  
  //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
  //	Monitoring2::GetProperty
  //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
  ComponentResult			Monitoring2::GetProperty(	AudioUnitPropertyID inID,
                                                          AudioUnitScope 		inScope,
                                                          AudioUnitElement 	inElement,
                                                          void *			outData )
  {
  	return AUEffectBase::GetProperty (inID, inScope, inElement, outData);
  }
  
  //	Monitoring2::Initialize
  //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
  ComponentResult Monitoring2::Initialize()
  {
      ComponentResult result = AUEffectBase::Initialize();
      if (result == noErr)
          Reset(kAudioUnitScope_Global, 0);
      return result;
  }
  
  #pragma mark ____Monitoring2EffectKernel
  
  
  
  //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
  //	Monitoring2::Monitoring2Kernel::Reset()
  //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
  ComponentResult		Monitoring2::Reset(AudioUnitScope inScope, AudioUnitElement inElement)
  {	
  	for(int count = 0; count < 99; count++) {
  		darkSampleL[count] = 0;
  		darkSampleR[count] = 0;
  	}
  	
  	for(int count = 0; count < 1502; count++) {
  		aL[count] = 0.0; bL[count] = 0.0; cL[count] = 0.0; dL[count] = 0.0;
  		aR[count] = 0.0; bR[count] = 0.0; cR[count] = 0.0; dR[count] = 0.0;
  	}
  	ax = 1; bx = 1; cx = 1; dx = 1;
  	//PeaksOnly
  	lastSampleL = 0.0; lastSampleR = 0.0;
  	//SlewOnly
  	iirSampleAL = 0.0; iirSampleBL = 0.0; iirSampleCL = 0.0; iirSampleDL = 0.0; iirSampleEL = 0.0; iirSampleFL = 0.0; iirSampleGL = 0.0;
  	iirSampleHL = 0.0; iirSampleIL = 0.0; iirSampleJL = 0.0; iirSampleKL = 0.0; iirSampleLL = 0.0; iirSampleML = 0.0; iirSampleNL = 0.0; iirSampleOL = 0.0; iirSamplePL = 0.0;
  	iirSampleQL = 0.0; iirSampleRL = 0.0; iirSampleSL = 0.0;
  	iirSampleTL = 0.0; iirSampleUL = 0.0; iirSampleVL = 0.0;
  	iirSampleWL = 0.0; iirSampleXL = 0.0; iirSampleYL = 0.0; iirSampleZL = 0.0;
  	
  	iirSampleAR = 0.0; iirSampleBR = 0.0; iirSampleCR = 0.0; iirSampleDR = 0.0; iirSampleER = 0.0; iirSampleFR = 0.0; iirSampleGR = 0.0;
  	iirSampleHR = 0.0; iirSampleIR = 0.0; iirSampleJR = 0.0; iirSampleKR = 0.0; iirSampleLR = 0.0; iirSampleMR = 0.0; iirSampleNR = 0.0; iirSampleOR = 0.0; iirSamplePR = 0.0;
  	iirSampleQR = 0.0; iirSampleRR = 0.0; iirSampleSR = 0.0;
  	iirSampleTR = 0.0; iirSampleUR = 0.0; iirSampleVR = 0.0;
  	iirSampleWR = 0.0; iirSampleXR = 0.0; iirSampleYR = 0.0; iirSampleZR = 0.0; // o/`	
  	//SubsOnly
  	for (int x = 0; x < fix_total; x++) {biquad[x] = 0.0;}
  	//Bandpasses
  	fpdL = 1.0; while (fpdL < 16386) fpdL = rand()*UINT32_MAX;
  	fpdR = 1.0; while (fpdR < 16386) fpdR = rand()*UINT32_MAX;
  	return noErr;
  }
  
  //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
  //	Monitoring2::ProcessBufferLists
  //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
  OSStatus		Monitoring2::ProcessBufferLists(AudioUnitRenderActionFlags & ioActionFlags,
  													const AudioBufferList & inBuffer,
                                                      AudioBufferList & outBuffer,
                                                      UInt32 			inFramesToProcess)
  {
  	Float32 * inputL = (Float32*)(inBuffer.mBuffers[0].mData);
  	Float32 * inputR = (Float32*)(inBuffer.mBuffers[1].mData);
  	Float32 * outputL = (Float32*)(outBuffer.mBuffers[0].mData);
  	Float32 * outputR = (Float32*)(outBuffer.mBuffers[1].mData);
  	UInt32 nSampleFrames = inFramesToProcess;
  	long double overallscale = 1.0;
  	overallscale /= 44100.0;
  	overallscale *= GetSampleRate();
  	
  	int depth = (int)(17.0*overallscale);
  	if (depth < 3) depth = 3;
  	if (depth > 98) depth = 98; //for Dark
  	
  	int processing = (int) GetParameter( kParam_One );
  	int am = (int)149.0 * overallscale;
  	int bm = (int)179.0 * overallscale;
  	int cm = (int)191.0 * overallscale;
  	int dm = (int)223.0 * overallscale; //these are 'good' primes, spacing out the allpasses
  	int allpasstemp;
  	//for PeaksOnly
  	biquad[fix_freq] = 0.0375/overallscale; biquad[fix_reso] = 0.1575; //define as AURAT, MONORAT, MONOLAT unless overridden
  	if (processing == kVINYL) {biquad[fix_freq] = 0.0385/overallscale; biquad[fix_reso] = 0.0825;}
  	if (processing == kPHONE) {biquad[fix_freq] = 0.1245/overallscale; biquad[fix_reso] = 0.46;}	
  	double K = tan(M_PI * biquad[fix_freq]);
  	double norm = 1.0 / (1.0 + K / biquad[fix_reso] + K * K);
  	biquad[fix_a0] = K / biquad[fix_reso] * norm;
  	biquad[fix_a2] = -biquad[fix_a0]; //for bandpass, ignore [fix_a1] = 0.0
  	biquad[fix_b1] = 2.0 * (K * K - 1.0) * norm;
  	biquad[fix_b2] = (1.0 - K / biquad[fix_reso] + K * K) * norm;
  	//for Bandpasses
  	
  	while (nSampleFrames-- > 0) {
  		long double inputSampleL = *inputL;
  		long double inputSampleR = *inputR;
  		if (fabs(inputSampleL)<1.18e-23) inputSampleL = fpdL * 1.18e-17;
  		fpdL ^= fpdL << 13; fpdL ^= fpdL >> 17; fpdL ^= fpdL << 5;
  		if (fabs(inputSampleR)<1.18e-23) inputSampleR = fpdR * 1.18e-17;
  		fpdR ^= fpdR << 13; fpdR ^= fpdR >> 17; fpdR ^= fpdR << 5;
  		//we need to make our dither run up here, there's no spot on the end to do it
  		
  		switch (processing)
  		{
  			case kDKAD:
  			case kDKCD:
  				break;
  			case kPEAK:
  				if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; inputSampleL = asin(inputSampleL);
  				if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; inputSampleR = asin(inputSampleR);
  				//amplitude aspect
  				allpasstemp = ax - 1; if (allpasstemp < 0 || allpasstemp > am) allpasstemp = am;
  				inputSampleL -= aL[allpasstemp]*0.5; aL[ax] = inputSampleL; inputSampleL *= 0.5;
  				inputSampleR -= aR[allpasstemp]*0.5; aR[ax] = inputSampleR; inputSampleR *= 0.5;
  				ax--; if (ax < 0 || ax > am) {ax = am;}
  				inputSampleL += (aL[ax]);
  				inputSampleR += (aR[ax]);
  				//a single Midiverb-style allpass
  				
  				if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; inputSampleL = asin(inputSampleL);
  				if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; inputSampleR = asin(inputSampleR);
  				//amplitude aspect
  				
  				allpasstemp = bx - 1; if (allpasstemp < 0 || allpasstemp > bm) allpasstemp = bm;
  				inputSampleL -= bL[allpasstemp]*0.5; bL[bx] = inputSampleL; inputSampleL *= 0.5;
  				inputSampleR -= bR[allpasstemp]*0.5; bR[bx] = inputSampleR; inputSampleR *= 0.5;
  				bx--; if (bx < 0 || bx > bm) {bx = bm;}
  				inputSampleL += (bL[bx]);
  				inputSampleR += (bR[bx]);
  				//a single Midiverb-style allpass
  				
  				if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; inputSampleL = asin(inputSampleL);
  				if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; inputSampleR = asin(inputSampleR);
  				//amplitude aspect
  				
  				allpasstemp = cx - 1; if (allpasstemp < 0 || allpasstemp > cm) allpasstemp = cm;
  				inputSampleL -= cL[allpasstemp]*0.5; cL[cx] = inputSampleL; inputSampleL *= 0.5;
  				inputSampleR -= cR[allpasstemp]*0.5; cR[cx] = inputSampleR; inputSampleR *= 0.5;
  				cx--; if (cx < 0 || cx > cm) {cx = cm;}
  				inputSampleL += (cL[cx]);
  				inputSampleR += (cR[cx]);
  				//a single Midiverb-style allpass
  				
  				if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; inputSampleL = asin(inputSampleL);
  				if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; inputSampleR = asin(inputSampleR);
  				//amplitude aspect
  				
  				allpasstemp = dx - 1; if (allpasstemp < 0 || allpasstemp > dm) allpasstemp = dm;
  				inputSampleL -= dL[allpasstemp]*0.5; dL[dx] = inputSampleL; inputSampleL *= 0.5;
  				inputSampleR -= dR[allpasstemp]*0.5; dR[dx] = inputSampleR; inputSampleR *= 0.5;
  				dx--; if (dx < 0 || dx > dm) {dx = dm;}
  				inputSampleL += (dL[dx]);
  				inputSampleR += (dR[dx]);
  				//a single Midiverb-style allpass
  				
  				if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; inputSampleL = asin(inputSampleL);
  				if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; inputSampleR = asin(inputSampleR);
  				//amplitude aspect
  				
  				inputSampleL *= 0.63679; inputSampleR *= 0.63679; //scale it to 0dB output at full blast
  				//PeaksOnly
  				break;
  			case kSLEW:
  				Float64 trim;
  				trim = 2.302585092994045684017991; //natural logarithm of 10
  				long double slewSample; slewSample = (inputSampleL - lastSampleL)*trim;
  				lastSampleL = inputSampleL;
  				if (slewSample > 1.0) slewSample = 1.0; if (slewSample < -1.0) slewSample = -1.0;
  				inputSampleL = slewSample;
  				slewSample = (inputSampleR - lastSampleR)*trim;
  				lastSampleR = inputSampleR;
  				if (slewSample > 1.0) slewSample = 1.0; if (slewSample < -1.0) slewSample = -1.0;
  				inputSampleR = slewSample;
  				//SlewOnly
  				break;
  			case kSUBS:
  				Float64 iirAmount; iirAmount = (2250/44100.0) / overallscale;
  				Float64 gain; gain = 1.42;
  				inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
  				
  				iirSampleAL = (iirSampleAL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleAL;
  				iirSampleAR = (iirSampleAR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleAR;
  				inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
  				if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
  				if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
  				
  				iirSampleBL = (iirSampleBL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleBL;
  				iirSampleBR = (iirSampleBR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleBR;
  				inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
  				if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
  				if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
  				
  				iirSampleCL = (iirSampleCL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleCL;
  				iirSampleCR = (iirSampleCR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleCR;
  				inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
  				if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
  				if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
  				
  				iirSampleDL = (iirSampleDL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleDL;
  				iirSampleDR = (iirSampleDR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleDR;
  				inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
  				if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
  				if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
  				
  				iirSampleEL = (iirSampleEL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleEL;
  				iirSampleER = (iirSampleER * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleER;
  				inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
  				if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
  				if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
  				
  				iirSampleFL = (iirSampleFL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleFL;
  				iirSampleFR = (iirSampleFR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleFR;
  				inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
  				if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
  				if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
  				
  				iirSampleGL = (iirSampleGL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleGL;
  				iirSampleGR = (iirSampleGR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleGR;
  				inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
  				if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
  				if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
  				
  				iirSampleHL = (iirSampleHL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleHL;
  				iirSampleHR = (iirSampleHR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleHR;
  				inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
  				if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
  				if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
  				
  				iirSampleIL = (iirSampleIL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleIL;
  				iirSampleIR = (iirSampleIR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleIR;
  				inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
  				if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
  				if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
  				
  				iirSampleJL = (iirSampleJL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleJL;
  				iirSampleJR = (iirSampleJR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleJR;
  				inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
  				if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
  				if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
  				
  				iirSampleKL = (iirSampleKL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleKL;
  				iirSampleKR = (iirSampleKR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleKR;
  				inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
  				if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
  				if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
  				
  				iirSampleLL = (iirSampleLL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleLL;
  				iirSampleLR = (iirSampleLR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleLR;
  				inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
  				if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
  				if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
  				
  				iirSampleML = (iirSampleML * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleML;
  				iirSampleMR = (iirSampleMR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleMR;
  				inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
  				if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
  				if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
  				
  				iirSampleNL = (iirSampleNL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleNL;
  				iirSampleNR = (iirSampleNR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleNR;
  				inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
  				if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
  				if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
  				
  				iirSampleOL = (iirSampleOL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleOL;
  				iirSampleOR = (iirSampleOR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleOR;
  				inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
  				if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
  				if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
  				
  				iirSamplePL = (iirSamplePL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSamplePL;
  				iirSamplePR = (iirSamplePR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSamplePR;
  				inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
  				if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
  				if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
  				
  				iirSampleQL = (iirSampleQL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleQL;
  				iirSampleQR = (iirSampleQR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleQR;
  				inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
  				if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
  				if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
  				
  				iirSampleRL = (iirSampleRL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleRL;
  				iirSampleRR = (iirSampleRR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleRR;
  				inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
  				if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
  				if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
  				
  				iirSampleSL = (iirSampleSL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleSL;
  				iirSampleSR = (iirSampleSR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleSR;
  				inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
  				if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
  				if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
  				
  				iirSampleTL = (iirSampleTL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleTL;
  				iirSampleTR = (iirSampleTR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleTR;
  				inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
  				if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
  				if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
  				
  				iirSampleUL = (iirSampleUL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleUL;
  				iirSampleUR = (iirSampleUR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleUR;
  				inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
  				if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
  				if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
  				
  				iirSampleVL = (iirSampleVL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleVL;
  				iirSampleVR = (iirSampleVR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleVR;
  				inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
  				if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
  				if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
  				
  				iirSampleWL = (iirSampleWL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleWL;
  				iirSampleWR = (iirSampleWR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleWR;
  				inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
  				if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
  				if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
  				
  				iirSampleXL = (iirSampleXL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleXL;
  				iirSampleXR = (iirSampleXR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleXR;
  				inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
  				if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
  				if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
  				
  				iirSampleYL = (iirSampleYL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleYL;
  				iirSampleYR = (iirSampleYR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleYR;
  				inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
  				if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
  				if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
  				
  				iirSampleZL = (iirSampleZL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleZL;
  				iirSampleZR = (iirSampleZR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleZR;
  				if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
  				if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;				
  				//SubsOnly
  				break;
  			case kMONO:
  			case kSIDE:
  				long double mid; mid = inputSampleL + inputSampleR;
  				long double side; side = inputSampleL - inputSampleR;
  				if (processing < 6) side = 0.0;
  				else mid = 0.0; //mono monitoring, or side-only monitoring
  				inputSampleL = (mid+side)/2.0;
  				inputSampleR = (mid-side)/2.0; 
  				break;
  			case kVINYL:
  			case kAURAT:
  			case kMONORAT:
  			case kMONOLAT:
  			case kPHONE:
  				//Bandpass: changes in EQ are up in the variable defining, not here
  				//7 Vinyl, 8 9 10 Aurat, 11 Phone
  				
  				if (processing == kMONORAT) {inputSampleR = (inputSampleL + inputSampleR)*0.5;inputSampleL = 0.0;}
  				if (processing == kMONOLAT) {inputSampleL = (inputSampleL + inputSampleR)*0.5;inputSampleR = 0.0;}
  				if (processing == kPHONE) {long double M; M = (inputSampleL + inputSampleR)*0.5; inputSampleL = M;inputSampleR = M;}
  				
  				inputSampleL = sin(inputSampleL); inputSampleR = sin(inputSampleR);
  				//encode Console5: good cleanness
  				
  				long double tempSampleL; tempSampleL = (inputSampleL * biquad[fix_a0]) + biquad[fix_sL1];
  				biquad[fix_sL1] = (-tempSampleL * biquad[fix_b1]) + biquad[fix_sL2];
  				biquad[fix_sL2] = (inputSampleL * biquad[fix_a2]) - (tempSampleL * biquad[fix_b2]);
  				inputSampleL = tempSampleL; //like mono AU, 7 and 8 store L channel
  				
  				long double tempSampleR; tempSampleR = (inputSampleR * biquad[fix_a0]) + biquad[fix_sR1];
  				biquad[fix_sR1] = (-tempSampleR * biquad[fix_b1]) + biquad[fix_sR2];
  				biquad[fix_sR2] = (inputSampleR * biquad[fix_a2]) - (tempSampleR * biquad[fix_b2]);
  				inputSampleR = tempSampleR; //note: 9 and 10 store the R channel
  				
  				if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
  				if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
  				//without this, you can get a NaN condition where it spits out DC offset at full blast!
  				inputSampleL = asin(inputSampleL); inputSampleR = asin(inputSampleR);
  				//amplitude aspect
  				break;
  			case kCANSA:
  			case kCANSB:
  			case kCANSC:
  			case kCANSD:
  				if (processing == kCANSA) {inputSampleL *= 0.855; inputSampleR *= 0.855;}
  				if (processing == kCANSB) {inputSampleL *= 0.748; inputSampleR *= 0.748;}
  				if (processing == kCANSC) {inputSampleL *= 0.713; inputSampleR *= 0.713;}
  				if (processing == kCANSD) {inputSampleL *= 0.680; inputSampleR *= 0.680;}
  				//we do a volume compensation immediately to gain stage stuff cleanly
  				inputSampleL = sin(inputSampleL);
  				inputSampleR = sin(inputSampleR);
  				long double drySampleL; drySampleL = inputSampleL;
  				long double drySampleR; drySampleR = inputSampleR; //everything runs 'inside' Console
  				long double bass; bass = (processing * processing * 0.00001) / overallscale;
  				//we are using the iir filters from out of SubsOnly
  				
  				mid = inputSampleL + inputSampleR; side = inputSampleL - inputSampleR;
  				iirSampleAL = (iirSampleAL * (1.0 - (bass*0.618))) + (side * bass * 0.618); side = side - iirSampleAL;
  				inputSampleL = (mid+side)/2.0; inputSampleR = (mid-side)/2.0;
  				//bass narrowing filter				
  				
  				allpasstemp = ax - 1; if (allpasstemp < 0 || allpasstemp > am) allpasstemp = am;
  				inputSampleL -= aL[allpasstemp]*0.5; aL[ax] = inputSampleL; inputSampleL *= 0.5;
  				inputSampleR -= aR[allpasstemp]*0.5; aR[ax] = inputSampleR; inputSampleR *= 0.5;
  				
  				ax--; if (ax < 0 || ax > am) {ax = am;}
  				inputSampleL += (aL[ax])*0.5; inputSampleR += (aR[ax])*0.5;
  				if (ax == am) {inputSampleL += (aL[0])*0.5; inputSampleR += (aR[0])*0.5;}
  				else {inputSampleL += (aL[ax+1])*0.5; inputSampleR += (aR[ax+1])*0.5;}
  				//a darkened Midiverb-style allpass
  				
  				if (processing == kCANSA) {inputSampleL *= 0.125; inputSampleR *= 0.125;}
  				if (processing == kCANSB) {inputSampleL *= 0.25; inputSampleR *= 0.25;}
  				if (processing == kCANSC) {inputSampleL *= 0.30; inputSampleR *= 0.30;}
  				if (processing == kCANSD) {inputSampleL *= 0.35; inputSampleR *= 0.35;}
  				//Cans A suppresses the crossfeed more, Cans B makes it louder
  				
  				drySampleL += inputSampleR;
  				drySampleR += inputSampleL; //the crossfeed
  				
  				allpasstemp = dx - 1; if (allpasstemp < 0 || allpasstemp > dm) allpasstemp = dm;
  				inputSampleL -= dL[allpasstemp]*0.5; dL[dx] = inputSampleL; inputSampleL *= 0.5;
  				inputSampleR -= dR[allpasstemp]*0.5; dR[dx] = inputSampleR; inputSampleR *= 0.5;
  				
  				dx--; if (dx < 0 || dx > dm) {dx = dm;}
  				inputSampleL += (dL[dx])*0.5; inputSampleR += (dR[dx])*0.5;
  				if (dx == dm) {inputSampleL += (dL[0])*0.5; inputSampleR += (dR[0])*0.5;}
  				else {inputSampleL += (dL[dx+1])*0.5; inputSampleR += (dR[dx+1])*0.5;}
  				//a darkened Midiverb-style allpass, which is stretching the previous one even more
  				
  				inputSampleL *= 0.25; inputSampleR *= 0.25;
  				//for all versions of Cans the second level of bloom is this far down
  				//and, remains on the opposite speaker rather than crossing again to the original side
  				
  				drySampleL += inputSampleR;
  				drySampleR += inputSampleL; //add the crossfeed and very faint extra verbyness
  				
  				inputSampleL = drySampleL;
  				inputSampleR = drySampleR; //and output our can-opened headphone feed
  				
  				mid = inputSampleL + inputSampleR; side = inputSampleL - inputSampleR;
  				iirSampleAR = (iirSampleAR * (1.0 - bass)) + (side * bass); side = side - iirSampleAR;
  				inputSampleL = (mid+side)/2.0; inputSampleR = (mid-side)/2.0;
  				//bass narrowing filter
  				
  				if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; inputSampleL = asin(inputSampleL);
  				if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; inputSampleR = asin(inputSampleR);
  				//ConsoleBuss processing
  				break;
  			case kTRICK:
  				long double inputSample = (inputSampleL + inputSampleR) * 0.5;
  				inputSampleL = -inputSample;
  				inputSampleR = inputSample;
  				break;
  		}
  		
  		//begin Dark		
  		if (processing == kDKCD) {
  			inputSampleL *= 32768.0; //or 16 bit option
  			inputSampleR *= 32768.0; //or 16 bit option
  		} else {
  			inputSampleL *= 8388608.0; //for literally everything else
  			inputSampleR *= 8388608.0; //we will apply the 24 bit Dark
  		} //on the not unreasonable assumption that we are very likely playing back on 24 bit DAC
  		
  		//We are doing it first Left, then Right, because the loops may run faster if
  		//they aren't too jammed full of variables. This means re-running code.
  		
  		//begin left
  		int quantA = floor(inputSampleL);
  		int quantB = floor(inputSampleL+1.0);
  		//to do this style of dither, we quantize in either direction and then
  		//do a reconstruction of what the result will be for each choice.
  		//We then evaluate which one we like, and keep a history of what we previously had
  		
  		float expectedSlew = 0;
  		for(int x = 0; x < depth; x++) {
  			expectedSlew += (darkSampleL[x+1] - darkSampleL[x]);
  		}
  		expectedSlew /= depth; //we have an average of all recent slews
  		//we are doing that to voice the thing down into the upper mids a bit
  		//it mustn't just soften the brightest treble, it must smooth high mids too
  		
  		float testA = fabs((darkSampleL[0] - quantA) - expectedSlew);
  		float testB = fabs((darkSampleL[0] - quantB) - expectedSlew);
  		
  		if (testA < testB) inputSampleL = quantA;
  		else inputSampleL = quantB;
  		//select whichever one departs LEAST from the vector of averaged
  		//reconstructed previous final samples. This will force a kind of dithering
  		//as it'll make the output end up as smooth as possible
  		
  		for(int x = depth; x >=0; x--) {
  			darkSampleL[x+1] = darkSampleL[x];
  		}
  		darkSampleL[0] = inputSampleL;
  		//end Dark left
  		
  		//begin right
  		quantA = floor(inputSampleR);
  		quantB = floor(inputSampleR+1.0);
  		//to do this style of dither, we quantize in either direction and then
  		//do a reconstruction of what the result will be for each choice.
  		//We then evaluate which one we like, and keep a history of what we previously had
  		
  		expectedSlew = 0;
  		for(int x = 0; x < depth; x++) {
  			expectedSlew += (darkSampleR[x+1] - darkSampleR[x]);
  		}
  		expectedSlew /= depth; //we have an average of all recent slews
  		//we are doing that to voice the thing down into the upper mids a bit
  		//it mustn't just soften the brightest treble, it must smooth high mids too
  		
  		testA = fabs((darkSampleR[0] - quantA) - expectedSlew);
  		testB = fabs((darkSampleR[0] - quantB) - expectedSlew);
  		
  		if (testA < testB) inputSampleR = quantA;
  		else inputSampleR = quantB;
  		//select whichever one departs LEAST from the vector of averaged
  		//reconstructed previous final samples. This will force a kind of dithering
  		//as it'll make the output end up as smooth as possible
  		
  		for(int x = depth; x >=0; x--) {
  			darkSampleR[x+1] = darkSampleR[x];
  		}
  		darkSampleR[0] = inputSampleR;
  		//end Dark right
  		
  		if (processing == kDKCD) {
  			inputSampleL /= 32768.0;
  			inputSampleR /= 32768.0;
  		} else {
  			inputSampleL /= 8388608.0;
  			inputSampleR /= 8388608.0;
  		}
  		//does not use 32 bit stereo floating point dither
  		
  		*outputL = inputSampleL;
  		*outputR = inputSampleR;
  		//direct stereo out
  		
  		inputL += 1;
  		inputR += 1;
  		outputL += 1;
  		outputR += 1;
  	}
  	return noErr;
  }
  
  