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329 lines
15 KiB
C++
Executable file
329 lines
15 KiB
C++
Executable file
/*
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* File: PurestEcho.cpp
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*
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* Version: 1.0
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*
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* Created: 6/2/17
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*
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* Copyright: Copyright © 2017 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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PurestEcho.cpp
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=============================================================================*/
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#include "PurestEcho.h"
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//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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AUDIOCOMPONENT_ENTRY(AUBaseFactory, PurestEcho)
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//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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// PurestEcho::PurestEcho
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//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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PurestEcho::PurestEcho(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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// PurestEcho::GetParameterValueStrings
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//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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ComponentResult PurestEcho::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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// PurestEcho::GetParameterInfo
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//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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ComponentResult PurestEcho::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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// PurestEcho::GetPropertyInfo
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//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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ComponentResult PurestEcho::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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// PurestEcho::GetProperty
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//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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ComponentResult PurestEcho::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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// PurestEcho::Initialize
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//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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ComponentResult PurestEcho::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 ____PurestEchoEffectKernel
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//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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// PurestEcho::PurestEchoKernel::Reset()
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//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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void PurestEcho::PurestEchoKernel::Reset()
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{
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//totalsamples comes from the .h file: it's a const static number that defines
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//the whole delay buffer. We still have a hardcoded delay buffer, but some might like
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//to use this to define the buffer in terms of seconds: samples as a factor of GetSampleRate()
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//The danger there, of course, is having a user start up the plugin at 384K and smashing their memory
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for(int count = 0; count < totalsamples-1; count++) {d[count] = 0;}
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gcount = 0;
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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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// PurestEcho::PurestEchoKernel::Process
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//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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void PurestEcho::PurestEchoKernel::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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int loopLimit = (int)(totalsamples * 0.499);
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//this is a double buffer so we will be splitting it in two
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Float64 time = pow(GetParameter( kParam_One ),2) * 0.999;
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Float64 tap1 = GetParameter( kParam_Two );
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Float64 tap2 = GetParameter( kParam_Three );
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Float64 tap3 = GetParameter( kParam_Four );
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Float64 tap4 = GetParameter( kParam_Five );
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Float64 gainTrim = 1.0 / (1.0 + tap1 + tap2 + tap3 + tap4);
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//this becomes our equal-loudness mechanic. 0.2 to 1.0 gain on all things.
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Float64 tapsTrim = gainTrim * 0.5;
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//the taps interpolate and require half that gain: 0.1 to 0.5 on all taps.
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int position1 = (int)(loopLimit * time * 0.25);
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int position2 = (int)(loopLimit * time * 0.5);
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int position3 = (int)(loopLimit * time * 0.75);
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int position4 = (int)(loopLimit * time);
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//basic echo information: we're taking four equally spaced echoes and setting their levels as desired.
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//position4 is what you'd have for 'just set a delay time'
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Float64 volAfter1 = (loopLimit * time * 0.25) - position1;
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Float64 volAfter2 = (loopLimit * time * 0.5) - position2;
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Float64 volAfter3 = (loopLimit * time * 0.75) - position3;
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Float64 volAfter4 = (loopLimit * time) - position4;
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//these are 0-1: casting to an (int) truncates fractional numbers towards zero (and is faster than floor() )
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//so, when we take the integer number (all above zero) and subtract it from the real value, we get 0-1
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Float64 volBefore1 = (1.0 - volAfter1) * tap1;
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Float64 volBefore2 = (1.0 - volAfter2) * tap2;
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Float64 volBefore3 = (1.0 - volAfter3) * tap3;
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Float64 volBefore4 = (1.0 - volAfter4) * tap4;
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//and if we are including a bit of the previous/next sample to interpolate, then if the sample position is 1.0001
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//we'll be leaning most heavily on the 'before' sample which is nearer to us, and the 'after' sample is almost not used.
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//if the sample position is 1.9999, the 'after' sample is strong and 'before' is almost not used.
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volAfter1 *= tap1;
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volAfter2 *= tap2;
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volAfter3 *= tap3;
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volAfter4 *= tap4;
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//and like with volBefore, we also want to scale this 'interpolate' to the loudness of this tap.
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//We do it here because we can do it only once per audio buffer, not on every sample. This assumes we're
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//not moving the tap every sample: if so we'd have to do this every sample as well.
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int oneBefore1 = position1 - 1;
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int oneBefore2 = position2 - 1;
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int oneBefore3 = position3 - 1;
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int oneBefore4 = position4 - 1;
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if (oneBefore1 < 0) oneBefore1 = 0;
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if (oneBefore2 < 0) oneBefore2 = 0;
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if (oneBefore3 < 0) oneBefore3 = 0;
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if (oneBefore4 < 0) oneBefore4 = 0;
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int oneAfter1 = position1 + 1;
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int oneAfter2 = position2 + 1;
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int oneAfter3 = position3 + 1;
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int oneAfter4 = position4 + 1;
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//this is setting up the way we interpolate samples: we're doing an echo-darkening thing
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//to make it sound better. Pretty much no acoustic delay in human-breathable air will give
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//you zero attenuation at 22 kilohertz: forget this at your peril ;)
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Float64 delaysBuffer;
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double inputSample;
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while (nSampleFrames-- > 0) {
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inputSample = *sourceP;
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if (fabs(inputSample)<1.18e-23) inputSample = fpd * 1.18e-17;
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if (gcount < 0 || gcount > loopLimit) gcount = loopLimit;
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d[gcount+loopLimit] = d[gcount] = inputSample * tapsTrim; //this is how the double buffer works:
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//we can look for delay taps without ever having to 'wrap around' within our calculation.
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//As long as the delay tap is less than our loop limit we can always just add it to where we're
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//at, and get a valid sample back right away, no matter where we are in the buffer.
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//The 0.5 is taking into account the interpolation, by padding down the whole buffer.
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delaysBuffer = (d[gcount+oneBefore4]*volBefore4);
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delaysBuffer += (d[gcount+oneAfter4]*volAfter4);
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delaysBuffer += (d[gcount+oneBefore3]*volBefore3);
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delaysBuffer += (d[gcount+oneAfter3]*volAfter3);
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delaysBuffer += (d[gcount+oneBefore2]*volBefore2);
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delaysBuffer += (d[gcount+oneAfter2]*volAfter2);
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delaysBuffer += (d[gcount+oneBefore1]*volBefore1);
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delaysBuffer += (d[gcount+oneAfter1]*volAfter1);
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//These are the interpolated samples. We're adding them first, because we know they're smaller
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//and while the value of delaysBuffer is small we'll add similarly small values to it. Note the order.
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delaysBuffer += (d[gcount+position4]*tap4);
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delaysBuffer += (d[gcount+position3]*tap3);
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delaysBuffer += (d[gcount+position2]*tap2);
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delaysBuffer += (d[gcount+position1]*tap1);
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//These are the primary samples for the echo, and we're adding them last. As before we're starting with the
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//most delayed echoes, and ending with what we think might be the loudest echo. We're building this delaybuffer
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//from the faintest noises to the loudest, to avoid adding a bunch of teeny values at the end.
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//You can of course put the last echo as loudest, but with diminishing echo volumes this is optimal.
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//This technique is also present in other plugins such as Iron Oxide.
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inputSample = (inputSample * gainTrim) + delaysBuffer;
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//this could be just inputSample += d[gcount+position1];
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//for literally a single, full volume echo combined with dry.
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//What I'm doing is making the echoes more interesting.
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gcount--;
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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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