Updates (in case my plane crashes)

1 files changed, 462 insertions, 0 deletions

diff --git a/plugins/MacAU/AQuickVoiceClip/AQuickVoiceClip.cpp b/plugins/MacAU/AQuickVoiceClip/AQuickVoiceClip.cpp
new file mode 100755
index 0000000..9fbaade
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+++ b/plugins/MacAU/AQuickVoiceClip/AQuickVoiceClip.cpp
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+/*
+*	File:		AQuickVoiceClip.cpp
+*	
+*	Version:	1.0
+* 
+*	Created:	10/3/13
+*	
+*	Copyright:  Copyright © 2013 Airwindows, All Rights Reserved
+* 
+*	Disclaimer:	IMPORTANT:  This Apple software is supplied to you by Apple Computer, Inc. ("Apple") in 
+*				consideration of your agreement to the following terms, and your use, installation, modification 
+*				or redistribution of this Apple software constitutes acceptance of these terms.  If you do 
+*				not agree with these terms, please do not use, install, modify or redistribute this Apple 
+*				software.
+*
+*				In consideration of your agreement to abide by the following terms, and subject to these terms, 
+*				Apple grants you a personal, non-exclusive license, under Apple's copyrights in this 
+*				original Apple software (the "Apple Software"), to use, reproduce, modify and redistribute the 
+*				Apple Software, with or without modifications, in source and/or binary forms; provided that if you 
+*				redistribute the Apple Software in its entirety and without modifications, you must retain this 
+*				notice and the following text and disclaimers in all such redistributions of the Apple Software. 
+*				Neither the name, trademarks, service marks or logos of Apple Computer, Inc. may be used to 
+*				endorse or promote products derived from the Apple Software without specific prior written 
+*				permission from Apple.  Except as expressly stated in this notice, no other rights or 
+*				licenses, express or implied, are granted by Apple herein, including but not limited to any 
+*				patent rights that may be infringed by your derivative works or by other works in which the 
+*				Apple Software may be incorporated.
+*
+*				The Apple Software is provided by Apple on an "AS IS" basis.  APPLE MAKES NO WARRANTIES, EXPRESS OR 
+*				IMPLIED, INCLUDING WITHOUT LIMITATION THE IMPLIED WARRANTIES OF NON-INFRINGEMENT, MERCHANTABILITY 
+*				AND FITNESS FOR A PARTICULAR PURPOSE, REGARDING THE APPLE SOFTWARE OR ITS USE AND OPERATION ALONE 
+*				OR IN COMBINATION WITH YOUR PRODUCTS.
+*
+*				IN NO EVENT SHALL APPLE BE LIABLE FOR ANY SPECIAL, INDIRECT, INCIDENTAL OR CONSEQUENTIAL 
+*				DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS 
+*				OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) ARISING IN ANY WAY OUT OF THE USE, 
+*				REPRODUCTION, MODIFICATION AND/OR DISTRIBUTION OF THE APPLE SOFTWARE, HOWEVER CAUSED AND WHETHER 
+*				UNDER THEORY OF CONTRACT, TORT (INCLUDING NEGLIGENCE), STRICT LIABILITY OR OTHERWISE, EVEN 
+*				IF APPLE HAS BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*
+*/
+/*=============================================================================
+	AQuickVoiceClip.cpp
+	
+=============================================================================*/
+#include "AQuickVoiceClip.h"
+
+
+//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+
+COMPONENT_ENTRY(AQuickVoiceClip)
+
+
+//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+//	AQuickVoiceClip::AQuickVoiceClip
+//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+AQuickVoiceClip::AQuickVoiceClip(AudioUnit component)
+	: AUEffectBase(component)
+{
+	CreateElements();
+	Globals()->UseIndexedParameters(kNumberOfParameters);
+	SetParameter(kParam_One, kDefaultValue_ParamOne );
+         
+#if AU_DEBUG_DISPATCHER
+	mDebugDispatcher = new AUDebugDispatcher (this);
+#endif
+	
+}
+
+
+//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+//	AQuickVoiceClip::GetParameterValueStrings
+//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ComponentResult			AQuickVoiceClip::GetParameterValueStrings(AudioUnitScope		inScope,
+                                                                AudioUnitParameterID	inParameterID,
+                                                                CFArrayRef *		outStrings)
+{
+        
+    return kAudioUnitErr_InvalidProperty;
+}
+
+
+
+//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+//	AQuickVoiceClip::GetParameterInfo
+//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ComponentResult			AQuickVoiceClip::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 = 30.0;
+                outParameterInfo.maxValue = 3000.0;
+                outParameterInfo.defaultValue = kDefaultValue_ParamOne;
+                break;
+			default:
+                result = kAudioUnitErr_InvalidParameter;
+                break;
+		}
+	} else {
+        result = kAudioUnitErr_InvalidParameter;
+    }
+    
+
+
+	return result;
+}
+
+//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+//	AQuickVoiceClip::GetPropertyInfo
+//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ComponentResult			AQuickVoiceClip::GetPropertyInfo (AudioUnitPropertyID	inID,
+                                                        AudioUnitScope		inScope,
+                                                        AudioUnitElement	inElement,
+                                                        UInt32 &		outDataSize,
+                                                        Boolean &		outWritable)
+{
+	return AUEffectBase::GetPropertyInfo (inID, inScope, inElement, outDataSize, outWritable);
+}
+
+//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+//	AQuickVoiceClip::GetProperty
+//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ComponentResult			AQuickVoiceClip::GetProperty(	AudioUnitPropertyID inID,
+                                                        AudioUnitScope 		inScope,
+                                                        AudioUnitElement 	inElement,
+                                                        void *			outData )
+{
+	return AUEffectBase::GetProperty (inID, inScope, inElement, outData);
+}
+
+//	AQuickVoiceClip::Initialize
+//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ComponentResult AQuickVoiceClip::Initialize()
+{
+    ComponentResult result = AUEffectBase::Initialize();
+    if (result == noErr)
+        Reset(kAudioUnitScope_Global, 0);
+    return result;
+}
+
+#pragma mark ____AQuickVoiceClipEffectKernel
+
+
+
+//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+//	AQuickVoiceClip::AQuickVoiceClipKernel::Reset()
+//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+void		AQuickVoiceClip::AQuickVoiceClipKernel::Reset()
+{
+	ataLast6Sample = ataLast5Sample = ataLast4Sample = ataLast3Sample = ataLast2Sample = ataLast1Sample = 0.0;
+	ataHalfwaySample = ataHalfDrySample = ataHalfDiffSample = 0.0;
+	ataLastDiffSample = ataDrySample = ataDiffSample = ataPrevDiffSample = 0.0;
+	ataK1 = -0.646; //first FIR shaping of interpolated sample, brightens
+	ataK2 = 0.311; //second FIR shaping of interpolated sample, thickens
+	ataK6 = -0.093; //third FIR shaping of interpolated sample, brings air
+	ataK7 = 0.057; //fourth FIR shaping of interpolated sample, thickens
+	ataK8 = -0.023; //fifth FIR shaping of interpolated sample, brings air
+	ataK3 = 0.114; //add raw to interpolated dry, toughens
+	ataK4 = 0.886; //remainder of interpolated dry, adds up to 1.0
+	ataK5 = 0.431; //subtract this much prev. diff sample, brightens.  0.431 becomes flat
+	lastSample = 0.0;
+	lastOutSample = 0.0;
+	lastOut2Sample = 0.0;
+	lastOut3Sample = 0.0;
+	lpDepth = 0.0;
+	overshoot = 0.0;
+	overall = 0;
+	iirSampleA = 0.0;
+	iirSampleB = 0.0;
+	iirSampleC = 0.0;
+	iirSampleD = 0.0;
+	flip = false;
+	demotimer = 0;
+	fpNShapeA = 0.0;
+	fpNShapeB = 0.0;
+	fpFlip = true;
+}
+
+//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+//	AQuickVoiceClip::AQuickVoiceClipKernel::Process
+//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+void		AQuickVoiceClip::AQuickVoiceClipKernel::Process(	const Float32 	*inSourceP,
+                                                    Float32		 	*inDestP,
+                                                    UInt32 			inFramesToProcess,
+                                                    UInt32			inNumChannels, 
+                                                    bool			&ioSilence )
+{
+	UInt32 nSampleFrames = inFramesToProcess;
+	const Float32 *sourceP = inSourceP;
+	Float32 *destP = inDestP;
+	Float64 overallscale = 1.0;
+	overallscale /= 44100.0;
+	overallscale *= GetSampleRate();
+	
+	Float64 softness = 0.484416;
+	Float64 hardness = 1.0 - softness;
+	
+	Float64 iirAmount = GetParameter( kParam_One )/8000.0;
+	
+	iirAmount /= overallscale;
+	Float64 altAmount = (1.0 - iirAmount);
+	
+	Float64 cancelnew = 0.0682276;
+	Float64 cancelold = 1.0 - cancelnew;
+	
+	Float64 maxRecent;
+	
+	Float64 lpSpeed = 0.0009;
+
+	Float64 cliplevel = 0.98;
+	
+	Float64 refclip = 0.5; //preset to cut out gain quite a lot. 91%? no touchy unless clip
+	
+	Float64 inputSample;
+	Float64 passThrough;
+	Float64 outputSample;
+	bool clipOnset;
+	Float64 drySample;
+	Float32 fpTemp;
+	Float64 fpOld = 0.618033988749894848204586; //golden ratio!
+	Float64 fpNew = 1.0 - fpOld;
+	
+	while (nSampleFrames-- > 0) {
+		inputSample = *sourceP;
+		if (inputSample<1.2e-38 && -inputSample<1.2e-38) {
+			static int noisesource = 0;
+			//this declares a variable before anything else is compiled. It won't keep assigning
+			//it to 0 for every sample, it's as if the declaration doesn't exist in this context,
+			//but it lets me add this denormalization fix in a single place rather than updating
+			//it in three different locations. The variable isn't thread-safe but this is only
+			//a random seed and we can share it with whatever.
+			noisesource = noisesource % 1700021; noisesource++;
+			int residue = noisesource * noisesource;
+			residue = residue % 170003; residue *= residue;
+			residue = residue % 17011; residue *= residue;
+			residue = residue % 1709; residue *= residue;
+			residue = residue % 173; residue *= residue;
+			residue = residue % 17;
+			double applyresidue = residue;
+			applyresidue *= 0.00000001;
+			applyresidue *= 0.00000001;
+			inputSample = applyresidue;
+			//this denormalization routine produces a white noise at -300 dB which the noise
+			//shaping will interact with to produce a bipolar output, but the noise is actually
+			//all positive. That should stop any variables from going denormal, and the routine
+			//only kicks in if digital black is input. As a final touch, if you save to 24-bit
+			//the silence will return to being digital black again.
+		}
+		passThrough = ataDrySample = inputSample;
+		
+		
+		ataHalfDrySample = ataHalfwaySample = (inputSample + ataLast1Sample + (ataLast2Sample*ataK1) + (ataLast3Sample*ataK2) + (ataLast4Sample*ataK6) + (ataLast5Sample*ataK7) + (ataLast6Sample*ataK8)) / 2.0;
+		ataLast6Sample = ataLast5Sample; ataLast5Sample = ataLast4Sample; ataLast4Sample = ataLast3Sample; ataLast3Sample = ataLast2Sample; ataLast2Sample = ataLast1Sample; ataLast1Sample = inputSample;
+		//setting up oversampled special antialiasing
+		clipOnset = false;
+		
+		
+		maxRecent = fabs( ataLast6Sample );
+		if (fabs( ataLast5Sample ) > maxRecent ) maxRecent = fabs( ataLast5Sample );
+		if (fabs( ataLast4Sample ) > maxRecent ) maxRecent = fabs( ataLast4Sample );
+		if (fabs( ataLast3Sample ) > maxRecent ) maxRecent = fabs( ataLast3Sample );
+		if (fabs( ataLast2Sample ) > maxRecent ) maxRecent = fabs( ataLast2Sample );
+		if (fabs( ataLast1Sample ) > maxRecent ) maxRecent = fabs( ataLast1Sample );
+		if (fabs( inputSample ) > maxRecent ) maxRecent = fabs( inputSample );
+		//this gives us something that won't cut out in zero crossings, to interpolate with
+		
+		maxRecent *= 2.0;
+		//by refclip this is 1.0 and fully into the antialiasing
+		if (maxRecent > 1.0) maxRecent = 1.0;
+		//and it tops out at 1. Higher means more antialiasing, lower blends into passThrough without antialiasing
+		
+		ataHalfwaySample -= overall;
+		//subtract dist-cancel from input after getting raw input, before doing anything
+		
+		drySample = ataHalfwaySample;
+		
+
+		if (lastSample >= refclip)
+		{
+			lpDepth += 0.1;
+			if (ataHalfwaySample < refclip)
+			{
+				lastSample = ((refclip*hardness) + (ataHalfwaySample * softness));
+			}
+			else lastSample = refclip;
+		}
+		
+		if (lastSample <= -refclip)
+		{
+			lpDepth += 0.1;
+			if (ataHalfwaySample > -refclip)
+			{
+				lastSample = ((-refclip*hardness) + (ataHalfwaySample * softness));
+			}
+			else lastSample = -refclip;
+		}
+		
+		if (ataHalfwaySample > refclip)
+		{
+			lpDepth += 0.1;
+			if (lastSample < refclip)
+			{
+				ataHalfwaySample = ((refclip*hardness) + (lastSample * softness));
+			}
+			else ataHalfwaySample = refclip;
+		}
+		
+		if (ataHalfwaySample < -refclip)
+		{
+			lpDepth += 0.1;
+			if (lastSample > -refclip)
+			{
+				ataHalfwaySample = ((-refclip*hardness) + (lastSample * softness));
+			}
+			else ataHalfwaySample = -refclip;
+		}
+				
+        outputSample = lastSample;
+		lastSample = ataHalfwaySample;
+		ataHalfwaySample = outputSample;
+		//swap around in a circle for one final ADClip,
+		//this time not tracking overshoot anymore
+		
+		//end interpolated sample
+		
+		//begin raw sample- inputSample and ataDrySample handled separately here
+		
+		inputSample -= overall;
+		//subtract dist-cancel from input after getting raw input, before doing anything
+		
+		drySample = inputSample;
+		
+		if (lastSample >= refclip)
+		{
+			lpDepth += 0.1;
+			if (inputSample < refclip)
+			{
+				lastSample = ((refclip*hardness) + (inputSample * softness));
+			}
+			else lastSample = refclip;
+		}
+		
+		if (lastSample <= -refclip)
+		{
+			lpDepth += 0.1;
+			if (inputSample > -refclip)
+			{
+				lastSample = ((-refclip*hardness) + (inputSample * softness));
+			}
+			else lastSample = -refclip;
+		}
+		
+		if (inputSample > refclip)
+		{
+			lpDepth += 0.1;
+			if (lastSample < refclip)
+			{
+				inputSample = ((refclip*hardness) + (lastSample * softness));
+			}
+			else inputSample = refclip;
+		}
+		
+		if (inputSample < -refclip)
+		{
+			lpDepth += 0.1;
+			if (lastSample > -refclip)
+			{
+				inputSample = ((-refclip*hardness) + (lastSample * softness));
+			}
+			else inputSample = -refclip;
+		}
+		
+        outputSample = lastSample;
+		lastSample = inputSample;
+		inputSample = outputSample;
+		
+		//end raw sample
+		
+		ataHalfDrySample = (ataDrySample*ataK3)+(ataHalfDrySample*ataK4);
+		ataHalfDiffSample = (ataHalfwaySample - ataHalfDrySample)/2.0;
+		ataLastDiffSample = ataDiffSample*ataK5;
+		ataDiffSample = (inputSample - ataDrySample)/2.0;
+		ataDiffSample += ataHalfDiffSample;
+		ataDiffSample -= ataLastDiffSample;
+		inputSample = ataDrySample;
+		inputSample += ataDiffSample;
+		
+		
+		overall = (overall * cancelold) + (ataDiffSample * cancelnew);
+		//apply all the diffs to a lowpassed IIR
+		
+		if (flip)
+		{
+			iirSampleA = (iirSampleA * altAmount) + (inputSample * iirAmount);
+			inputSample -= iirSampleA;
+			iirSampleC = (iirSampleC * altAmount) + (passThrough * iirAmount);
+			passThrough -= iirSampleC;
+		}
+		else
+		{
+			iirSampleB = (iirSampleB * altAmount) + (inputSample * iirAmount);
+			inputSample -= iirSampleB;
+			iirSampleD = (iirSampleD * altAmount) + (passThrough * iirAmount);
+			passThrough -= iirSampleD;
+		}
+		flip = not flip;
+		//highpass section 
+		
+		
+		lastOut3Sample = lastOut2Sample;
+		lastOut2Sample = lastOutSample;
+		lastOutSample = inputSample;
+		
+		lpDepth -= lpSpeed;
+		if (lpDepth > 0.0)
+		{
+			if (lpDepth > 1.0) lpDepth = 1.0;
+			inputSample *= (1.0-lpDepth);
+			inputSample += (((lastOutSample + lastOut2Sample + lastOut3Sample) / 3.6)*lpDepth);
+		}
+		if (lpDepth < 0.0) lpDepth = 0.0;
+		
+		//noise shaping to 32-bit floating point
+		if (fpFlip) {
+			fpTemp = inputSample;
+			fpNShapeA = (fpNShapeA*fpOld)+((inputSample-fpTemp)*fpNew);
+			inputSample += fpNShapeA;
+		}
+		else {
+			fpTemp = inputSample;
+			fpNShapeB = (fpNShapeB*fpOld)+((inputSample-fpTemp)*fpNew);
+			inputSample += fpNShapeB;
+		}
+		fpFlip = not fpFlip;
+		//end noise shaping on 32 bit output, hasn't been applied yet
+		
+		inputSample *= (1.0-maxRecent);
+		inputSample += (passThrough * maxRecent);
+		//there's our raw signal, without antialiasing. Brings up low level stuff and softens more when hot
+		
+		if (inputSample > cliplevel) inputSample = cliplevel;
+		if (inputSample < -cliplevel) inputSample = -cliplevel;
+		//final iron bar
+		
+		
+		*destP = inputSample;
+		
+		sourceP += inNumChannels; destP += inNumChannels;
+	}
+}
+