/*
*	File:		ConsoleLABuss.cpp
*	
*	Version:	1.0
* 
*	Created:	12/4/23
*	
*	Copyright:  Copyright © 2023 Airwindows, Airwindows uses the MIT license
* 
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/*=============================================================================
	ConsoleLABuss.cpp
	
=============================================================================*/
#include "ConsoleLABuss.h"


//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

AUDIOCOMPONENT_ENTRY(AUBaseFactory, ConsoleLABuss)


//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
//	ConsoleLABuss::ConsoleLABuss
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ConsoleLABuss::ConsoleLABuss(AudioUnit component)
	: AUEffectBase(component)
{
	CreateElements();
	Globals()->UseIndexedParameters(kNumberOfParameters);
	SetParameter(kParam_One, kDefaultValue_ParamOne );
         
#if AU_DEBUG_DISPATCHER
	mDebugDispatcher = new AUDebugDispatcher (this);
#endif
	
}


//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
//	ConsoleLABuss::GetParameterValueStrings
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult			ConsoleLABuss::GetParameterValueStrings(AudioUnitScope		inScope,
                                                                AudioUnitParameterID	inParameterID,
                                                                CFArrayRef *		outStrings)
{
        
    return kAudioUnitErr_InvalidProperty;
}



//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
//	ConsoleLABuss::GetParameterInfo
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult			ConsoleLABuss::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_Generic;
                outParameterInfo.minValue = 0.0;
                outParameterInfo.maxValue = 1.0;
                outParameterInfo.defaultValue = kDefaultValue_ParamOne;
                break;
           default:
                result = kAudioUnitErr_InvalidParameter;
                break;
            }
	} else {
        result = kAudioUnitErr_InvalidParameter;
    }
    


	return result;
}

//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
//	ConsoleLABuss::GetPropertyInfo
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult			ConsoleLABuss::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 ConsoleLABuss::SupportedNumChannels(const AUChannelInfo ** outInfo)
{
	if (outInfo != NULL)
	{
		static AUChannelInfo info;
		info.inChannels = 2;
		info.outChannels = 2;
		*outInfo = &info;
	}

	return 1;
}

//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
//	ConsoleLABuss::GetProperty
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult			ConsoleLABuss::GetProperty(	AudioUnitPropertyID inID,
                                                        AudioUnitScope 		inScope,
                                                        AudioUnitElement 	inElement,
                                                        void *			outData )
{
	return AUEffectBase::GetProperty (inID, inScope, inElement, outData);
}

//	ConsoleLABuss::Initialize
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult ConsoleLABuss::Initialize()
{
    ComponentResult result = AUEffectBase::Initialize();
    if (result == noErr)
        Reset(kAudioUnitScope_Global, 0);
    return result;
}

#pragma mark ____ConsoleLABussEffectKernel



//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
//	ConsoleLABuss::ConsoleLABussKernel::Reset()
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult		ConsoleLABuss::Reset(AudioUnitScope inScope, AudioUnitElement inElement)
{
	lastSinewL = lastSinewR = 0.0;
	subAL = subAR = subBL = subBR = subCL = subCR = 0.0;
	gainA = gainB = 1.0;
	fpdL = 1.0; while (fpdL < 16386) fpdL = rand()*UINT32_MAX;
	fpdR = 1.0; while (fpdR < 16386) fpdR = rand()*UINT32_MAX;
	return noErr;
}

//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
//	ConsoleLABuss::ProcessBufferLists
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
OSStatus		ConsoleLABuss::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;
	double overallscale = 1.0;
	overallscale /= 44100.0;
	overallscale *= GetSampleRate();
	
	gainA = gainB;
	gainB = sqrt(GetParameter( kParam_One )); //smoothed master fader from Z2 filters
	//this will be applied three times: this is to make the various tone alterations
	//hit differently at different master fader drive levels.
	//in particular, backing off the master fader tightens the super lows
	//but opens up the modified Sinew, because more of the attentuation happens before
	//you even get to slew clipping :) and if the fader is not active, it bypasses completely.
	
	double threshSinew = 0.718/overallscale;
	double subTrim = 0.0011 / overallscale;
	
	while (nSampleFrames-- > 0) {
		double inputSampleL = *inputL;
		double inputSampleR = *inputR;
		if (fabs(inputSampleL)<1.18e-23) inputSampleL = fpdL * 1.18e-17;
		if (fabs(inputSampleR)<1.18e-23) inputSampleR = fpdR * 1.18e-17;
		
		double temp = (double)nSampleFrames/inFramesToProcess;
		double gain = (gainA*temp)+(gainB*(1.0-temp));
		//setting up smoothed master fader
		
		//begin SubTight section
		double subSampleL = inputSampleL * subTrim;
		double subSampleR = inputSampleR * subTrim;
		
		double scale = 0.5+fabs(subSampleL*0.5);
		subSampleL = (subAL+(sin(subAL-subSampleL)*scale));
		subAL = subSampleL*scale;
		scale = 0.5+fabs(subSampleR*0.5);
		subSampleR = (subAR+(sin(subAR-subSampleR)*scale));
		subAR = subSampleR*scale;
		scale = 0.5+fabs(subSampleL*0.5);
		subSampleL = (subBL+(sin(subBL-subSampleL)*scale));
		subBL = subSampleL*scale;
		scale = 0.5+fabs(subSampleR*0.5);
		subSampleR = (subBR+(sin(subBR-subSampleR)*scale));
		subBR = subSampleR*scale;
		scale = 0.5+fabs(subSampleL*0.5);
		subSampleL = (subCL+(sin(subCL-subSampleL)*scale));
		subCL = subSampleL*scale;
		scale = 0.5+fabs(subSampleR*0.5);
		subSampleR = (subCR+(sin(subCR-subSampleR)*scale));
		subCR = subSampleR*scale;
		if (subSampleL > 0.25) subSampleL = 0.25;
		if (subSampleL < -0.25) subSampleL = -0.25;
		if (subSampleR > 0.25) subSampleR = 0.25;
		if (subSampleR < -0.25) subSampleR = -0.25;
		inputSampleL -= (subSampleL*16.0);
		inputSampleR -= (subSampleR*16.0);
		//end SubTight section
		
		if (gain < 1.0) {
			inputSampleL *= gain;
			inputSampleR *= gain;
		} //if using the master fader, we are going to attenuate three places.
		//subtight is always fully engaged: tighten response when restraining full console
		
		//begin ConsoleZeroBuss which is the one we choose for ConsoleLA
		if (inputSampleL > 2.8) inputSampleL = 2.8;
		if (inputSampleL < -2.8) inputSampleL = -2.8;
		if (inputSampleL > 0.0) inputSampleL = (inputSampleL*2.0)/(3.0-inputSampleL);
		else inputSampleL = -(inputSampleL*-2.0)/(3.0+inputSampleL);
		
		if (inputSampleR > 2.8) inputSampleR = 2.8;
		if (inputSampleR < -2.8) inputSampleR = -2.8;
		if (inputSampleR > 0.0) inputSampleR = (inputSampleR*2.0)/(3.0-inputSampleR);
		else inputSampleR = -(inputSampleR*-2.0)/(3.0+inputSampleR);
		//ConsoleZero Buss
		
		if (gain < 1.0) {
			inputSampleL *= gain;
			inputSampleR *= gain;
		} //if using the master fader, we are going to attenuate three places.
		//after C0Buss but before EverySlew: allow highs to come out a bit more
		//when pulling back master fader. Less drive equals more open
		
		temp = inputSampleL;
		double clamp = inputSampleL - lastSinewL;
		double sinew = threshSinew * cos(lastSinewL);
		if (clamp > sinew) temp = lastSinewL + sinew;
		if (-clamp > sinew) temp = lastSinewL - sinew;
		inputSampleL = lastSinewL = temp;
		if (lastSinewL > 1.0) lastSinewL = 1.0;
		if (lastSinewL < -1.0) lastSinewL = -1.0;
		temp = inputSampleR;
		clamp = inputSampleR - lastSinewR;
		sinew = threshSinew * cos(lastSinewR);
		if (clamp > sinew) temp = lastSinewR + sinew;
		if (-clamp > sinew) temp = lastSinewR - sinew;
		inputSampleR = lastSinewR = temp;
		if (lastSinewR > 1.0) lastSinewR = 1.0;
		if (lastSinewR < -1.0) lastSinewR = -1.0;
		
		if (gain < 1.0) {
			inputSampleL *= gain;
			inputSampleR *= gain;
		} //if using the master fader, we are going to attenuate three places.
		//after EverySlew fades the total output sound: least change in tone here.
		
		//begin 32 bit stereo floating point dither
		int expon; frexpf((float)inputSampleL, &expon);
		fpdL ^= fpdL << 13; fpdL ^= fpdL >> 17; fpdL ^= fpdL << 5;
		inputSampleL += ((double(fpdL)-uint32_t(0x7fffffff)) * 5.5e-36l * pow(2,expon+62));
		frexpf((float)inputSampleR, &expon);
		fpdR ^= fpdR << 13; fpdR ^= fpdR >> 17; fpdR ^= fpdR << 5;
		inputSampleR += ((double(fpdR)-uint32_t(0x7fffffff)) * 5.5e-36l * pow(2,expon+62));
		//end 32 bit stereo floating point dither
		
		*outputL = inputSampleL;
		*outputR = inputSampleR;
		//direct stereo out
		
		inputL += 1;
		inputR += 1;
		outputL += 1;
		outputR += 1;
	}
	return noErr;
}

