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airwindows/plugins/MacAU/Chamber2/Chamber2.cpp
Christopher Johnson 78241bb382 Verbity2
2023-02-04 15:41:29 -05:00

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/*
* File: Chamber2.cpp
*
* Version: 1.0
*
* Created: 2/1/23
*
* Copyright: Copyright © 2023 Airwindows, Airwindows uses the MIT license
*
* 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
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* notice and the following text and disclaimers in all such redistributions of the Apple Software.
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* endorse or promote products derived from the Apple Software without specific prior written
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*
* 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.
*
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* UNDER THEORY OF CONTRACT, TORT (INCLUDING NEGLIGENCE), STRICT LIABILITY OR OTHERWISE, EVEN
* IF APPLE HAS BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*
*/
/*=============================================================================
Chamber2.cpp
=============================================================================*/
#include "Chamber2.h"
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
COMPONENT_ENTRY(Chamber2)
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// Chamber2::Chamber2
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
Chamber2::Chamber2(AudioUnit component)
: AUEffectBase(component)
{
CreateElements();
Globals()->UseIndexedParameters(kNumberOfParameters);
SetParameter(kParam_One, kDefaultValue_ParamOne );
SetParameter(kParam_Two, kDefaultValue_ParamTwo );
SetParameter(kParam_Three, kDefaultValue_ParamThree );
SetParameter(kParam_Four, kDefaultValue_ParamFour );
#if AU_DEBUG_DISPATCHER
mDebugDispatcher = new AUDebugDispatcher (this);
#endif
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// Chamber2::GetParameterValueStrings
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult Chamber2::GetParameterValueStrings(AudioUnitScope inScope,
AudioUnitParameterID inParameterID,
CFArrayRef * outStrings)
{
return kAudioUnitErr_InvalidProperty;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// Chamber2::GetParameterInfo
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult Chamber2::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;
case kParam_Two:
AUBase::FillInParameterName (outParameterInfo, kParameterTwoName, false);
outParameterInfo.unit = kAudioUnitParameterUnit_Generic;
outParameterInfo.minValue = 0.0;
outParameterInfo.maxValue = 1.0;
outParameterInfo.defaultValue = kDefaultValue_ParamTwo;
break;
case kParam_Three:
AUBase::FillInParameterName (outParameterInfo, kParameterThreeName, false);
outParameterInfo.unit = kAudioUnitParameterUnit_Generic;
outParameterInfo.minValue = 0.0;
outParameterInfo.maxValue = 1.0;
outParameterInfo.defaultValue = kDefaultValue_ParamThree;
break;
case kParam_Four:
AUBase::FillInParameterName (outParameterInfo, kParameterFourName, false);
outParameterInfo.unit = kAudioUnitParameterUnit_Generic;
outParameterInfo.minValue = 0.0;
outParameterInfo.maxValue = 1.0;
outParameterInfo.defaultValue = kDefaultValue_ParamFour;
break;
default:
result = kAudioUnitErr_InvalidParameter;
break;
}
} else {
result = kAudioUnitErr_InvalidParameter;
}
return result;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// Chamber2::GetPropertyInfo
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult Chamber2::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 Chamber2::SupportedNumChannels(const AUChannelInfo ** outInfo)
{
if (outInfo != NULL)
{
static AUChannelInfo info;
info.inChannels = 2;
info.outChannels = 2;
*outInfo = &info;
}
return 1;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// Chamber2::GetProperty
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult Chamber2::GetProperty( AudioUnitPropertyID inID,
AudioUnitScope inScope,
AudioUnitElement inElement,
void * outData )
{
return AUEffectBase::GetProperty (inID, inScope, inElement, outData);
}
// Chamber2::Initialize
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult Chamber2::Initialize()
{
ComponentResult result = AUEffectBase::Initialize();
if (result == noErr)
Reset(kAudioUnitScope_Global, 0);
return result;
}
#pragma mark ____Chamber2EffectKernel
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// Chamber2::Chamber2Kernel::Reset()
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult Chamber2::Reset(AudioUnitScope inScope, AudioUnitElement inElement)
{
for(int count = 0; count < 9999; count++) {aEL[count] = 0.0;aER[count] = 0.0;}
for(int count = 0; count < 9999; count++) {aFL[count] = 0.0;aFR[count] = 0.0;}
for(int count = 0; count < 9999; count++) {aGL[count] = 0.0;aGR[count] = 0.0;}
for(int count = 0; count < 9999; count++) {aHL[count] = 0.0;aHR[count] = 0.0;}
for(int count = 0; count < 9999; count++) {aAL[count] = 0.0;aAR[count] = 0.0;}
for(int count = 0; count < 9999; count++) {aBL[count] = 0.0;aBR[count] = 0.0;}
for(int count = 0; count < 9999; count++) {aCL[count] = 0.0;aCR[count] = 0.0;}
for(int count = 0; count < 9999; count++) {aDL[count] = 0.0;aDR[count] = 0.0;}
for(int count = 0; count < 9999; count++) {aIL[count] = 0.0;aIR[count] = 0.0;}
for(int count = 0; count < 9999; count++) {aJL[count] = 0.0;aJR[count] = 0.0;}
for(int count = 0; count < 9999; count++) {aKL[count] = 0.0;aKR[count] = 0.0;}
for(int count = 0; count < 9999; count++) {aLL[count] = 0.0;aLR[count] = 0.0;}
for(int count = 0; count < 9999; count++) {aML[count] = 0.0;aMR[count] = 0.0;}
feedbackAL = 0.0; feedbackAR = 0.0;
feedbackBL = 0.0; feedbackBR = 0.0;
feedbackCL = 0.0; feedbackCR = 0.0;
feedbackDL = 0.0; feedbackDR = 0.0;
previousAL = 0.0; previousAR = 0.0;
previousBL = 0.0; previousBR = 0.0;
previousCL = 0.0; previousCR = 0.0;
previousDL = 0.0; previousDR = 0.0;
for(int count = 0; count < 9; count++) {lastRefL[count] = 0.0;lastRefR[count] = 0.0;}
countI = 1;
countJ = 1;
countK = 1;
countL = 1;
countM = 1;
countA = 1;
countB = 1;
countC = 1;
countD = 1;
countE = 1;
countF = 1;
countG = 1;
countH = 1;
cycle = 0;
fpdL = 1.0; while (fpdL < 16386) fpdL = rand()*UINT32_MAX;
fpdR = 1.0; while (fpdR < 16386) fpdR = rand()*UINT32_MAX;
return noErr;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// Chamber2::ProcessBufferLists
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
OSStatus Chamber2::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();
int cycleEnd = floor(overallscale);
if (cycleEnd < 1) cycleEnd = 1;
if (cycleEnd > 4) cycleEnd = 4;
//this is going to be 2 for 88.1 or 96k, 3 for silly people, 4 for 176 or 192k
if (cycle > cycleEnd-1) cycle = cycleEnd-1; //sanity check
Float64 size = (GetParameter( kParam_One )*0.9)+0.1;
Float64 regen = (1.0-(pow(1.0-GetParameter( kParam_Two ),2)))*0.123;
Float64 echoScale = 1.0-GetParameter( kParam_Three );
Float64 echo = 0.618033988749894848204586+((1.0-0.618033988749894848204586)*echoScale);
Float64 interpolate = (1.0-echo)*0.381966011250105;
//this now goes from Chamber, to all the reverb delays being exactly the same
//much larger usage of RAM due to the larger reverb delays everywhere, but
//ability to go to an unusual variation on blurred delay.
Float64 wet = GetParameter( kParam_Four )*2.0;
Float64 dry = 2.0 - wet;
if (wet > 1.0) wet = 1.0;
if (wet < 0.0) wet = 0.0;
if (dry > 1.0) dry = 1.0;
if (dry < 0.0) dry = 0.0;
//this reverb makes 50% full dry AND full wet, not crossfaded.
//that's so it can be on submixes without cutting back dry channel when adjusted:
//unless you go super heavy, you are only adjusting the added verb loudness.
delayM = sqrt(9900*size);
delayE = 9900*size;
delayF = delayE*echo;
delayG = delayF*echo;
delayH = delayG*echo;
delayA = delayH*echo;
delayB = delayA*echo;
delayC = delayB*echo;
delayD = delayC*echo;
delayI = delayD*echo;
delayJ = delayI*echo;
delayK = delayJ*echo;
delayL = delayK*echo;
//initially designed around the Fibonnaci series, Chamber uses
//delay coefficients that are all related to the Golden Ratio,
//Turns out that as you continue to sustain them, it turns from a
//chunky slapback effect into a smoother reverb tail that can
//sustain infinitely.
while (nSampleFrames-- > 0) {
long double inputSampleL = *inputL;
long 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 drySampleL = inputSampleL;
double drySampleR = inputSampleR;
cycle++;
if (cycle == cycleEnd) { //hit the end point and we do a reverb sample
aML[countM] = inputSampleL;
aMR[countM] = inputSampleR;
countM++; if (countM < 0 || countM > delayM) countM = 0;
inputSampleL = aML[countM-((countM > delayM)?delayM+1:0)];
inputSampleR = aMR[countM-((countM > delayM)?delayM+1:0)];
//predelay to make the first echo still be an echo even when blurred
feedbackAL = (feedbackAL*(1.0-interpolate))+(previousAL*interpolate); previousAL = feedbackAL;
feedbackBL = (feedbackBL*(1.0-interpolate))+(previousBL*interpolate); previousBL = feedbackBL;
feedbackCL = (feedbackCL*(1.0-interpolate))+(previousCL*interpolate); previousCL = feedbackCL;
feedbackDL = (feedbackDL*(1.0-interpolate))+(previousDL*interpolate); previousDL = feedbackDL;
feedbackAR = (feedbackAR*(1.0-interpolate))+(previousAR*interpolate); previousAR = feedbackAR;
feedbackBR = (feedbackBR*(1.0-interpolate))+(previousBR*interpolate); previousBR = feedbackBR;
feedbackCR = (feedbackCR*(1.0-interpolate))+(previousCR*interpolate); previousCR = feedbackCR;
feedbackDR = (feedbackDR*(1.0-interpolate))+(previousDR*interpolate); previousDR = feedbackDR;
aIL[countI] = inputSampleL + (feedbackAL * regen);
aJL[countJ] = inputSampleL + (feedbackBL * regen);
aKL[countK] = inputSampleL + (feedbackCL * regen);
aLL[countL] = inputSampleL + (feedbackDL * regen);
aIR[countI] = inputSampleR + (feedbackAR * regen);
aJR[countJ] = inputSampleR + (feedbackBR * regen);
aKR[countK] = inputSampleR + (feedbackCR * regen);
aLR[countL] = inputSampleR + (feedbackDR * regen);
countI++; if (countI < 0 || countI > delayI) countI = 0;
countJ++; if (countJ < 0 || countJ > delayJ) countJ = 0;
countK++; if (countK < 0 || countK > delayK) countK = 0;
countL++; if (countL < 0 || countL > delayL) countL = 0;
double outIL = aIL[countI-((countI > delayI)?delayI+1:0)];
double outJL = aJL[countJ-((countJ > delayJ)?delayJ+1:0)];
double outKL = aKL[countK-((countK > delayK)?delayK+1:0)];
double outLL = aLL[countL-((countL > delayL)?delayL+1:0)];
double outIR = aIR[countI-((countI > delayI)?delayI+1:0)];
double outJR = aJR[countJ-((countJ > delayJ)?delayJ+1:0)];
double outKR = aKR[countK-((countK > delayK)?delayK+1:0)];
double outLR = aLR[countL-((countL > delayL)?delayL+1:0)];
//first block: now we have four outputs
aAL[countA] = (outIL - (outJL + outKL + outLL));
aBL[countB] = (outJL - (outIL + outKL + outLL));
aCL[countC] = (outKL - (outIL + outJL + outLL));
aDL[countD] = (outLL - (outIL + outJL + outKL));
aAR[countA] = (outIR - (outJR + outKR + outLR));
aBR[countB] = (outJR - (outIR + outKR + outLR));
aCR[countC] = (outKR - (outIR + outJR + outLR));
aDR[countD] = (outLR - (outIR + outJR + outKR));
countA++; if (countA < 0 || countA > delayA) countA = 0;
countB++; if (countB < 0 || countB > delayB) countB = 0;
countC++; if (countC < 0 || countC > delayC) countC = 0;
countD++; if (countD < 0 || countD > delayD) countD = 0;
double outAL = aAL[countA-((countA > delayA)?delayA+1:0)];
double outBL = aBL[countB-((countB > delayB)?delayB+1:0)];
double outCL = aCL[countC-((countC > delayC)?delayC+1:0)];
double outDL = aDL[countD-((countD > delayD)?delayD+1:0)];
double outAR = aAR[countA-((countA > delayA)?delayA+1:0)];
double outBR = aBR[countB-((countB > delayB)?delayB+1:0)];
double outCR = aCR[countC-((countC > delayC)?delayC+1:0)];
double outDR = aDR[countD-((countD > delayD)?delayD+1:0)];
//second block: four more outputs
aEL[countE] = (outAL - (outBL + outCL + outDL));
aFL[countF] = (outBL - (outAL + outCL + outDL));
aGL[countG] = (outCL - (outAL + outBL + outDL));
aHL[countH] = (outDL - (outAL + outBL + outCL));
aER[countE] = (outAR - (outBR + outCR + outDR));
aFR[countF] = (outBR - (outAR + outCR + outDR));
aGR[countG] = (outCR - (outAR + outBR + outDR));
aHR[countH] = (outDR - (outAR + outBR + outCR));
countE++; if (countE < 0 || countE > delayE) countE = 0;
countF++; if (countF < 0 || countF > delayF) countF = 0;
countG++; if (countG < 0 || countG > delayG) countG = 0;
countH++; if (countH < 0 || countH > delayH) countH = 0;
double outEL = aEL[countE-((countE > delayE)?delayE+1:0)];
double outFL = aFL[countF-((countF > delayF)?delayF+1:0)];
double outGL = aGL[countG-((countG > delayG)?delayG+1:0)];
double outHL = aHL[countH-((countH > delayH)?delayH+1:0)];
double outER = aER[countE-((countE > delayE)?delayE+1:0)];
double outFR = aFR[countF-((countF > delayF)?delayF+1:0)];
double outGR = aGR[countG-((countG > delayG)?delayG+1:0)];
double outHR = aHR[countH-((countH > delayH)?delayH+1:0)];
//third block: final outputs
feedbackAR = (outEL - (outFL + outGL + outHL));
feedbackBL = (outFL - (outEL + outGL + outHL));
feedbackCR = (outGL - (outEL + outFL + outHL));
feedbackDL = (outHL - (outEL + outFL + outGL));
feedbackAL = (outER - (outFR + outGR + outHR));
feedbackBR = (outFR - (outER + outGR + outHR));
feedbackCL = (outGR - (outER + outFR + outHR));
feedbackDR = (outHR - (outER + outFR + outGR));
//which we need to feed back into the input again, a bit
inputSampleL = (outEL + outFL + outGL + outHL)/8.0;
inputSampleR = (outER + outFR + outGR + outHR)/8.0;
//and take the final combined sum of outputs
if (cycleEnd == 4) {
lastRefL[0] = lastRefL[4]; //start from previous last
lastRefL[2] = (lastRefL[0] + inputSampleL)/2; //half
lastRefL[1] = (lastRefL[0] + lastRefL[2])/2; //one quarter
lastRefL[3] = (lastRefL[2] + inputSampleL)/2; //three quarters
lastRefL[4] = inputSampleL; //full
lastRefR[0] = lastRefR[4]; //start from previous last
lastRefR[2] = (lastRefR[0] + inputSampleR)/2; //half
lastRefR[1] = (lastRefR[0] + lastRefR[2])/2; //one quarter
lastRefR[3] = (lastRefR[2] + inputSampleR)/2; //three quarters
lastRefR[4] = inputSampleR; //full
}
if (cycleEnd == 3) {
lastRefL[0] = lastRefL[3]; //start from previous last
lastRefL[2] = (lastRefL[0]+lastRefL[0]+inputSampleL)/3; //third
lastRefL[1] = (lastRefL[0]+inputSampleL+inputSampleL)/3; //two thirds
lastRefL[3] = inputSampleL; //full
lastRefR[0] = lastRefR[3]; //start from previous last
lastRefR[2] = (lastRefR[0]+lastRefR[0]+inputSampleR)/3; //third
lastRefR[1] = (lastRefR[0]+inputSampleR+inputSampleR)/3; //two thirds
lastRefR[3] = inputSampleR; //full
}
if (cycleEnd == 2) {
lastRefL[0] = lastRefL[2]; //start from previous last
lastRefL[1] = (lastRefL[0] + inputSampleL)/2; //half
lastRefL[2] = inputSampleL; //full
lastRefR[0] = lastRefR[2]; //start from previous last
lastRefR[1] = (lastRefR[0] + inputSampleR)/2; //half
lastRefR[2] = inputSampleR; //full
}
if (cycleEnd == 1) {
lastRefL[0] = inputSampleL;
lastRefR[0] = inputSampleR;
}
cycle = 0; //reset
inputSampleL = lastRefL[cycle];
inputSampleR = lastRefR[cycle];
} else {
inputSampleL = lastRefL[cycle];
inputSampleR = lastRefR[cycle];
//we are going through our references now
}
switch (cycleEnd) //multi-pole average using lastRef[] variables
{
case 4:
lastRefL[8] = inputSampleL; inputSampleL = (inputSampleL+lastRefL[7])*0.5;
lastRefL[7] = lastRefL[8]; //continue, do not break
lastRefR[8] = inputSampleR; inputSampleR = (inputSampleR+lastRefR[7])*0.5;
lastRefR[7] = lastRefR[8]; //continue, do not break
case 3:
lastRefL[8] = inputSampleL; inputSampleL = (inputSampleL+lastRefL[6])*0.5;
lastRefL[6] = lastRefL[8]; //continue, do not break
lastRefR[8] = inputSampleR; inputSampleR = (inputSampleR+lastRefR[6])*0.5;
lastRefR[6] = lastRefR[8]; //continue, do not break
case 2:
lastRefL[8] = inputSampleL; inputSampleL = (inputSampleL+lastRefL[5])*0.5;
lastRefL[5] = lastRefL[8]; //continue, do not break
lastRefR[8] = inputSampleR; inputSampleR = (inputSampleR+lastRefR[5])*0.5;
lastRefR[5] = lastRefR[8]; //continue, do not break
case 1:
break; //no further averaging
}
if (wet < 1.0) {inputSampleL *= wet; inputSampleR *= wet;}
if (dry < 1.0) {drySampleL *= dry; drySampleR *= dry;}
inputSampleL += drySampleL;
inputSampleR += drySampleR;
//this is our submix verb dry/wet: 0.5 is BOTH at FULL VOLUME
//purpose is that, if you're adding verb, you're not altering other balances
//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;
}
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