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431 lines
18 KiB
C++
431 lines
18 KiB
C++
/*
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* File: Chamber.cpp
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*
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* Version: 1.0
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*
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* Created: 6/21/21
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*
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* Copyright: Copyright © 2021 Airwindows, Airwindows uses the MIT license
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*
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* Disclaimer: IMPORTANT: This Apple software is supplied to you by Apple Computer, Inc. ("Apple") in
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* consideration of your agreement to the following terms, and your use, installation, modification
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* or redistribution of this Apple software constitutes acceptance of these terms. If you do
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* not agree with these terms, please do not use, install, modify or redistribute this Apple
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* software.
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*
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* In consideration of your agreement to abide by the following terms, and subject to these terms,
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* Apple grants you a personal, non-exclusive license, under Apple's copyrights in this
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* original Apple software (the "Apple Software"), to use, reproduce, modify and redistribute the
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* Apple Software, with or without modifications, in source and/or binary forms; provided that if you
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* redistribute the Apple Software in its entirety and without modifications, you must retain this
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* notice and the following text and disclaimers in all such redistributions of the Apple Software.
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* Neither the name, trademarks, service marks or logos of Apple Computer, Inc. may be used to
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* endorse or promote products derived from the Apple Software without specific prior written
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* permission from Apple. Except as expressly stated in this notice, no other rights or
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* licenses, express or implied, are granted by Apple herein, including but not limited to any
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* patent rights that may be infringed by your derivative works or by other works in which the
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* Apple Software may be incorporated.
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*
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* The Apple Software is provided by Apple on an "AS IS" basis. APPLE MAKES NO WARRANTIES, EXPRESS OR
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* IMPLIED, INCLUDING WITHOUT LIMITATION THE IMPLIED WARRANTIES OF NON-INFRINGEMENT, MERCHANTABILITY
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* AND FITNESS FOR A PARTICULAR PURPOSE, REGARDING THE APPLE SOFTWARE OR ITS USE AND OPERATION ALONE
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* OR IN COMBINATION WITH YOUR PRODUCTS.
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*
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* IN NO EVENT SHALL APPLE BE LIABLE FOR ANY SPECIAL, INDIRECT, INCIDENTAL OR CONSEQUENTIAL
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* DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS
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* OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) ARISING IN ANY WAY OUT OF THE USE,
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* REPRODUCTION, MODIFICATION AND/OR DISTRIBUTION OF THE APPLE SOFTWARE, HOWEVER CAUSED AND WHETHER
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* UNDER THEORY OF CONTRACT, TORT (INCLUDING NEGLIGENCE), STRICT LIABILITY OR OTHERWISE, EVEN
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* IF APPLE HAS BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*
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*/
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/*=============================================================================
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Chamber.cpp
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=============================================================================*/
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#include "Chamber.h"
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//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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AUDIOCOMPONENT_ENTRY(AUBaseFactory, Chamber)
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//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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// Chamber::Chamber
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//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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Chamber::Chamber(AudioUnit component)
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: AUEffectBase(component)
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{
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CreateElements();
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Globals()->UseIndexedParameters(kNumberOfParameters);
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SetParameter(kParam_One, kDefaultValue_ParamOne );
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SetParameter(kParam_Two, kDefaultValue_ParamTwo );
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SetParameter(kParam_Three, kDefaultValue_ParamThree );
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SetParameter(kParam_Four, kDefaultValue_ParamFour );
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SetParameter(kParam_Five, kDefaultValue_ParamFive );
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#if AU_DEBUG_DISPATCHER
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mDebugDispatcher = new AUDebugDispatcher (this);
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#endif
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}
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//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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// Chamber::GetParameterValueStrings
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//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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ComponentResult Chamber::GetParameterValueStrings(AudioUnitScope inScope,
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AudioUnitParameterID inParameterID,
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CFArrayRef * outStrings)
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{
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return kAudioUnitErr_InvalidProperty;
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}
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//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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// Chamber::GetParameterInfo
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//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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ComponentResult Chamber::GetParameterInfo(AudioUnitScope inScope,
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AudioUnitParameterID inParameterID,
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AudioUnitParameterInfo &outParameterInfo )
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{
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ComponentResult result = noErr;
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outParameterInfo.flags = kAudioUnitParameterFlag_IsWritable
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| kAudioUnitParameterFlag_IsReadable;
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if (inScope == kAudioUnitScope_Global) {
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switch(inParameterID)
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{
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case kParam_One:
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AUBase::FillInParameterName (outParameterInfo, kParameterOneName, false);
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outParameterInfo.unit = kAudioUnitParameterUnit_Generic;
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outParameterInfo.minValue = 0.0;
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outParameterInfo.maxValue = 1.0;
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outParameterInfo.defaultValue = kDefaultValue_ParamOne;
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break;
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case kParam_Two:
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AUBase::FillInParameterName (outParameterInfo, kParameterTwoName, false);
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outParameterInfo.unit = kAudioUnitParameterUnit_Generic;
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outParameterInfo.minValue = 0.0;
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outParameterInfo.maxValue = 1.0;
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outParameterInfo.defaultValue = kDefaultValue_ParamTwo;
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break;
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case kParam_Three:
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AUBase::FillInParameterName (outParameterInfo, kParameterThreeName, false);
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outParameterInfo.unit = kAudioUnitParameterUnit_Generic;
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outParameterInfo.minValue = 0.0;
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outParameterInfo.maxValue = 1.0;
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outParameterInfo.defaultValue = kDefaultValue_ParamThree;
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break;
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case kParam_Four:
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AUBase::FillInParameterName (outParameterInfo, kParameterFourName, false);
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outParameterInfo.unit = kAudioUnitParameterUnit_Generic;
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outParameterInfo.minValue = 0.0;
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outParameterInfo.maxValue = 1.0;
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outParameterInfo.defaultValue = kDefaultValue_ParamFour;
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break;
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case kParam_Five:
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AUBase::FillInParameterName (outParameterInfo, kParameterFiveName, false);
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outParameterInfo.unit = kAudioUnitParameterUnit_Generic;
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outParameterInfo.minValue = 0.0;
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outParameterInfo.maxValue = 1.0;
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outParameterInfo.defaultValue = kDefaultValue_ParamFive;
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break;
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default:
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result = kAudioUnitErr_InvalidParameter;
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break;
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}
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} else {
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result = kAudioUnitErr_InvalidParameter;
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}
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return result;
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}
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//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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// Chamber::GetPropertyInfo
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//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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ComponentResult Chamber::GetPropertyInfo (AudioUnitPropertyID inID,
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AudioUnitScope inScope,
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AudioUnitElement inElement,
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UInt32 & outDataSize,
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Boolean & outWritable)
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{
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return AUEffectBase::GetPropertyInfo (inID, inScope, inElement, outDataSize, outWritable);
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}
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//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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// Chamber::GetProperty
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//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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ComponentResult Chamber::GetProperty( AudioUnitPropertyID inID,
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AudioUnitScope inScope,
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AudioUnitElement inElement,
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void * outData )
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{
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return AUEffectBase::GetProperty (inID, inScope, inElement, outData);
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}
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// Chamber::Initialize
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//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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ComponentResult Chamber::Initialize()
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{
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ComponentResult result = AUEffectBase::Initialize();
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if (result == noErr)
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Reset(kAudioUnitScope_Global, 0);
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return result;
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}
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#pragma mark ____ChamberEffectKernel
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//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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// Chamber::ChamberKernel::Reset()
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//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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void Chamber::ChamberKernel::Reset()
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{
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iirA = 0.0;
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iirB = 0.0;
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iirC = 0.0;
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for(int count = 0; count < 19999; count++) {aE[count] = 0.0;}
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for(int count = 0; count < 12360; count++) {aF[count] = 0.0;}
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for(int count = 0; count < 7639; count++) {aG[count] = 0.0;}
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for(int count = 0; count < 4721; count++) {aH[count] = 0.0;}
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for(int count = 0; count < 2915; count++) {aA[count] = 0.0;}
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for(int count = 0; count < 1803; count++) {aB[count] = 0.0;}
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for(int count = 0; count < 1114; count++) {aC[count] = 0.0;}
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for(int count = 0; count < 688; count++) {aD[count] = 0.0;}
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for(int count = 0; count < 425; count++) {aI[count] = 0.0;}
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for(int count = 0; count < 263; count++) {aJ[count] = 0.0;}
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for(int count = 0; count < 162; count++) {aK[count] = 0.0;}
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for(int count = 0; count < 100; count++) {aL[count] = 0.0;}
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feedbackA = 0.0;
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feedbackB = 0.0;
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feedbackC = 0.0;
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feedbackD = 0.0;
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previousA = 0.0;
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previousB = 0.0;
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previousC = 0.0;
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previousD = 0.0;
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for(int count = 0; count < 9; count++) {lastRef[count] = 0.0;}
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cycle = 0;
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countI = 1;
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countJ = 1;
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countK = 1;
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countL = 1;
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countA = 1;
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countB = 1;
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countC = 1;
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countD = 1;
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countE = 1;
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countF = 1;
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countG = 1;
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countH = 1;
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fpd = 1.0; while (fpd < 16386) fpd = rand()*UINT32_MAX;
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}
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//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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// Chamber::ChamberKernel::Process
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//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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void Chamber::ChamberKernel::Process( const Float32 *inSourceP,
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Float32 *inDestP,
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UInt32 inFramesToProcess,
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UInt32 inNumChannels,
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bool &ioSilence )
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{
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UInt32 nSampleFrames = inFramesToProcess;
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const Float32 *sourceP = inSourceP;
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Float32 *destP = inDestP;
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double overallscale = 1.0;
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overallscale /= 44100.0;
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overallscale *= GetSampleRate();
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int cycleEnd = floor(overallscale);
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if (cycleEnd < 1) cycleEnd = 1;
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if (cycleEnd > 4) cycleEnd = 4;
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//this is going to be 2 for 88.1 or 96k, 3 for silly people, 4 for 176 or 192k
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if (cycle > cycleEnd-1) cycle = cycleEnd-1; //sanity check
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Float64 size = (pow(GetParameter( kParam_One ),2)*0.9)+0.1;
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Float64 regen = (1.0-(pow(1.0-GetParameter( kParam_Two ),6)))*0.123;
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Float64 highpass = (pow(GetParameter( kParam_Three ),2.0))/sqrt(overallscale);
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Float64 lowpass = (1.0-pow(GetParameter( kParam_Four ),2.0))/sqrt(overallscale);
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Float64 interpolate = size*0.381966011250105;
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Float64 wet = GetParameter( kParam_Five )*2.0;
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Float64 dry = 2.0 - wet;
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if (wet > 1.0) wet = 1.0;
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if (wet < 0.0) wet = 0.0;
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if (dry > 1.0) dry = 1.0;
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if (dry < 0.0) dry = 0.0;
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//this reverb makes 50% full dry AND full wet, not crossfaded.
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//that's so it can be on submixes without cutting back dry channel when adjusted:
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//unless you go super heavy, you are only adjusting the added verb loudness.
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delayE = 19900*size;
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delayF = delayE*0.618033988749894848204586;
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delayG = delayF*0.618033988749894848204586;
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delayH = delayG*0.618033988749894848204586;
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delayA = delayH*0.618033988749894848204586;
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delayB = delayA*0.618033988749894848204586;
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delayC = delayB*0.618033988749894848204586;
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delayD = delayC*0.618033988749894848204586;
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delayI = delayD*0.618033988749894848204586;
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delayJ = delayI*0.618033988749894848204586;
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delayK = delayJ*0.618033988749894848204586;
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delayL = delayK*0.618033988749894848204586;
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//initially designed around the Fibonnaci series, Chamber uses
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//delay coefficients that are all related to the Golden Ratio,
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//Turns out that as you continue to sustain them, it turns from a
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//chunky slapback effect into a smoother reverb tail that can
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//sustain infinitely.
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while (nSampleFrames-- > 0) {
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double inputSample = *sourceP;
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if (fabs(inputSample)<1.18e-23) inputSample = fpd * 1.18e-17;
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double drySample = inputSample;
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iirC = (iirC*(1.0-highpass))+(inputSample*highpass); inputSample -= iirC;
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//initial highpass
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iirA = (iirA*(1.0-lowpass))+(inputSample*lowpass); inputSample = iirA;
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//initial filter
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cycle++;
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if (cycle == cycleEnd) { //hit the end point and we do a reverb sample
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feedbackA = (feedbackA*(1.0-interpolate))+(previousA*interpolate); previousA = feedbackA;
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feedbackB = (feedbackB*(1.0-interpolate))+(previousB*interpolate); previousB = feedbackB;
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feedbackC = (feedbackC*(1.0-interpolate))+(previousC*interpolate); previousC = feedbackC;
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feedbackD = (feedbackD*(1.0-interpolate))+(previousD*interpolate); previousD = feedbackD;
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aI[countI] = inputSample + (feedbackA * regen);
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aJ[countJ] = inputSample + (feedbackB * regen);
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aK[countK] = inputSample + (feedbackC * regen);
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aL[countL] = inputSample + (feedbackD * regen);
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countI++; if (countI < 0 || countI > delayI) countI = 0;
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countJ++; if (countJ < 0 || countJ > delayJ) countJ = 0;
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countK++; if (countK < 0 || countK > delayK) countK = 0;
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countL++; if (countL < 0 || countL > delayL) countL = 0;
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Float64 outI = aI[countI-((countI > delayI)?delayI+1:0)];
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Float64 outJ = aJ[countJ-((countJ > delayJ)?delayJ+1:0)];
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Float64 outK = aK[countK-((countK > delayK)?delayK+1:0)];
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Float64 outL = aL[countL-((countL > delayL)?delayL+1:0)];
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//first block: now we have four outputs
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aA[countA] = (outI - (outJ + outK + outL));
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aB[countB] = (outJ - (outI + outK + outL));
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aC[countC] = (outK - (outI + outJ + outL));
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aD[countD] = (outL - (outI + outJ + outK));
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countA++; if (countA < 0 || countA > delayA) countA = 0;
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countB++; if (countB < 0 || countB > delayB) countB = 0;
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countC++; if (countC < 0 || countC > delayC) countC = 0;
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countD++; if (countD < 0 || countD > delayD) countD = 0;
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Float64 outA = aA[countA-((countA > delayA)?delayA+1:0)];
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Float64 outB = aB[countB-((countB > delayB)?delayB+1:0)];
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Float64 outC = aC[countC-((countC > delayC)?delayC+1:0)];
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Float64 outD = aD[countD-((countD > delayD)?delayD+1:0)];
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//second block: four more outputs
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aE[countE] = (outA - (outB + outC + outD));
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aF[countF] = (outB - (outA + outC + outD));
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aG[countG] = (outC - (outA + outB + outD));
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aH[countH] = (outD - (outA + outB + outC));
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countE++; if (countE < 0 || countE > delayE) countE = 0;
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countF++; if (countF < 0 || countF > delayF) countF = 0;
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countG++; if (countG < 0 || countG > delayG) countG = 0;
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countH++; if (countH < 0 || countH > delayH) countH = 0;
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Float64 outE = aE[countE-((countE > delayE)?delayE+1:0)];
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Float64 outF = aF[countF-((countF > delayF)?delayF+1:0)];
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Float64 outG = aG[countG-((countG > delayG)?delayG+1:0)];
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Float64 outH = aH[countH-((countH > delayH)?delayH+1:0)];
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//third block: final outputs
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feedbackA = (outE - (outF + outG + outH));
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feedbackB = (outF - (outE + outG + outH));
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feedbackC = (outG - (outE + outF + outH));
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feedbackD = (outH - (outE + outF + outG));
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//which we need to feed back into the input again, a bit
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inputSample = (outE + outF + outG + outH)/8.0;
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//and take the final combined sum of outputs
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if (cycleEnd == 4) {
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lastRef[0] = lastRef[4]; //start from previous last
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lastRef[2] = (lastRef[0] + inputSample)/2; //half
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lastRef[1] = (lastRef[0] + lastRef[2])/2; //one quarter
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lastRef[3] = (lastRef[2] + inputSample)/2; //three quarters
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lastRef[4] = inputSample; //full
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}
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if (cycleEnd == 3) {
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lastRef[0] = lastRef[3]; //start from previous last
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lastRef[2] = (lastRef[0]+lastRef[0]+inputSample)/3; //third
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lastRef[1] = (lastRef[0]+inputSample+inputSample)/3; //two thirds
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lastRef[3] = inputSample; //full
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}
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if (cycleEnd == 2) {
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lastRef[0] = lastRef[2]; //start from previous last
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lastRef[1] = (lastRef[0] + inputSample)/2; //half
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lastRef[2] = inputSample; //full
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}
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if (cycleEnd == 1) lastRef[0] = inputSample;
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cycle = 0; //reset
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inputSample = lastRef[cycle];
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} else {
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inputSample = lastRef[cycle];
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//we are going through our references now
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}
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switch (cycleEnd) //multi-pole average using lastRef[] variables
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{
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case 4:
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lastRef[8] = inputSample; inputSample = (inputSample+lastRef[7])*0.5;
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lastRef[7] = lastRef[8]; //continue, do not break
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case 3:
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lastRef[8] = inputSample; inputSample = (inputSample+lastRef[6])*0.5;
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lastRef[6] = lastRef[8]; //continue, do not break
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case 2:
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lastRef[8] = inputSample; inputSample = (inputSample+lastRef[5])*0.5;
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lastRef[5] = lastRef[8]; //continue, do not break
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case 1:
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break; //no further averaging
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}
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iirB = (iirB*(1.0-lowpass))+(inputSample*lowpass); inputSample = iirB;
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//end filter
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if (wet < 1.0) inputSample *= wet;
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if (dry < 1.0) drySample *= dry;
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inputSample += drySample;
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//this is our submix verb dry/wet: 0.5 is BOTH at FULL VOLUME
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//purpose is that, if you're adding verb, you're not altering other balances
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//begin 32 bit floating point dither
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int expon; frexpf((float)inputSample, &expon);
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fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5;
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inputSample += ((double(fpd)-uint32_t(0x7fffffff)) * 5.5e-36l * pow(2,expon+62));
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//end 32 bit floating point dither
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*destP = inputSample;
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sourceP += inNumChannels; destP += inNumChannels;
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}
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}
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