airwindows/plugins/MacSignedAU/Chamber/Chamber.cpp
2022-11-21 09:20:21 -05:00

431 lines
18 KiB
C++

/*
* File: Chamber.cpp
*
* Version: 1.0
*
* Created: 6/21/21
*
* Copyright: Copyright © 2021 Airwindows, Airwindows uses the MIT license
*
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/*=============================================================================
Chamber.cpp
=============================================================================*/
#include "Chamber.h"
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
AUDIOCOMPONENT_ENTRY(AUBaseFactory, Chamber)
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// Chamber::Chamber
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
Chamber::Chamber(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 );
SetParameter(kParam_Five, kDefaultValue_ParamFive );
#if AU_DEBUG_DISPATCHER
mDebugDispatcher = new AUDebugDispatcher (this);
#endif
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// Chamber::GetParameterValueStrings
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult Chamber::GetParameterValueStrings(AudioUnitScope inScope,
AudioUnitParameterID inParameterID,
CFArrayRef * outStrings)
{
return kAudioUnitErr_InvalidProperty;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// Chamber::GetParameterInfo
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult Chamber::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;
case kParam_Five:
AUBase::FillInParameterName (outParameterInfo, kParameterFiveName, false);
outParameterInfo.unit = kAudioUnitParameterUnit_Generic;
outParameterInfo.minValue = 0.0;
outParameterInfo.maxValue = 1.0;
outParameterInfo.defaultValue = kDefaultValue_ParamFive;
break;
default:
result = kAudioUnitErr_InvalidParameter;
break;
}
} else {
result = kAudioUnitErr_InvalidParameter;
}
return result;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// Chamber::GetPropertyInfo
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult Chamber::GetPropertyInfo (AudioUnitPropertyID inID,
AudioUnitScope inScope,
AudioUnitElement inElement,
UInt32 & outDataSize,
Boolean & outWritable)
{
return AUEffectBase::GetPropertyInfo (inID, inScope, inElement, outDataSize, outWritable);
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// Chamber::GetProperty
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult Chamber::GetProperty( AudioUnitPropertyID inID,
AudioUnitScope inScope,
AudioUnitElement inElement,
void * outData )
{
return AUEffectBase::GetProperty (inID, inScope, inElement, outData);
}
// Chamber::Initialize
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult Chamber::Initialize()
{
ComponentResult result = AUEffectBase::Initialize();
if (result == noErr)
Reset(kAudioUnitScope_Global, 0);
return result;
}
#pragma mark ____ChamberEffectKernel
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// Chamber::ChamberKernel::Reset()
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
void Chamber::ChamberKernel::Reset()
{
iirA = 0.0;
iirB = 0.0;
iirC = 0.0;
for(int count = 0; count < 19999; count++) {aE[count] = 0.0;}
for(int count = 0; count < 12360; count++) {aF[count] = 0.0;}
for(int count = 0; count < 7639; count++) {aG[count] = 0.0;}
for(int count = 0; count < 4721; count++) {aH[count] = 0.0;}
for(int count = 0; count < 2915; count++) {aA[count] = 0.0;}
for(int count = 0; count < 1803; count++) {aB[count] = 0.0;}
for(int count = 0; count < 1114; count++) {aC[count] = 0.0;}
for(int count = 0; count < 688; count++) {aD[count] = 0.0;}
for(int count = 0; count < 425; count++) {aI[count] = 0.0;}
for(int count = 0; count < 263; count++) {aJ[count] = 0.0;}
for(int count = 0; count < 162; count++) {aK[count] = 0.0;}
for(int count = 0; count < 100; count++) {aL[count] = 0.0;}
feedbackA = 0.0;
feedbackB = 0.0;
feedbackC = 0.0;
feedbackD = 0.0;
previousA = 0.0;
previousB = 0.0;
previousC = 0.0;
previousD = 0.0;
for(int count = 0; count < 9; count++) {lastRef[count] = 0.0;}
cycle = 0;
countI = 1;
countJ = 1;
countK = 1;
countL = 1;
countA = 1;
countB = 1;
countC = 1;
countD = 1;
countE = 1;
countF = 1;
countG = 1;
countH = 1;
fpd = 1.0; while (fpd < 16386) fpd = rand()*UINT32_MAX;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// Chamber::ChamberKernel::Process
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
void Chamber::ChamberKernel::Process( const Float32 *inSourceP,
Float32 *inDestP,
UInt32 inFramesToProcess,
UInt32 inNumChannels,
bool &ioSilence )
{
UInt32 nSampleFrames = inFramesToProcess;
const Float32 *sourceP = inSourceP;
Float32 *destP = inDestP;
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 = (pow(GetParameter( kParam_One ),2)*0.9)+0.1;
Float64 regen = (1.0-(pow(1.0-GetParameter( kParam_Two ),6)))*0.123;
Float64 highpass = (pow(GetParameter( kParam_Three ),2.0))/sqrt(overallscale);
Float64 lowpass = (1.0-pow(GetParameter( kParam_Four ),2.0))/sqrt(overallscale);
Float64 interpolate = size*0.381966011250105;
Float64 wet = GetParameter( kParam_Five )*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.
delayE = 19900*size;
delayF = delayE*0.618033988749894848204586;
delayG = delayF*0.618033988749894848204586;
delayH = delayG*0.618033988749894848204586;
delayA = delayH*0.618033988749894848204586;
delayB = delayA*0.618033988749894848204586;
delayC = delayB*0.618033988749894848204586;
delayD = delayC*0.618033988749894848204586;
delayI = delayD*0.618033988749894848204586;
delayJ = delayI*0.618033988749894848204586;
delayK = delayJ*0.618033988749894848204586;
delayL = delayK*0.618033988749894848204586;
//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) {
double inputSample = *sourceP;
if (fabs(inputSample)<1.18e-23) inputSample = fpd * 1.18e-17;
double drySample = inputSample;
iirC = (iirC*(1.0-highpass))+(inputSample*highpass); inputSample -= iirC;
//initial highpass
iirA = (iirA*(1.0-lowpass))+(inputSample*lowpass); inputSample = iirA;
//initial filter
cycle++;
if (cycle == cycleEnd) { //hit the end point and we do a reverb sample
feedbackA = (feedbackA*(1.0-interpolate))+(previousA*interpolate); previousA = feedbackA;
feedbackB = (feedbackB*(1.0-interpolate))+(previousB*interpolate); previousB = feedbackB;
feedbackC = (feedbackC*(1.0-interpolate))+(previousC*interpolate); previousC = feedbackC;
feedbackD = (feedbackD*(1.0-interpolate))+(previousD*interpolate); previousD = feedbackD;
aI[countI] = inputSample + (feedbackA * regen);
aJ[countJ] = inputSample + (feedbackB * regen);
aK[countK] = inputSample + (feedbackC * regen);
aL[countL] = inputSample + (feedbackD * 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;
Float64 outI = aI[countI-((countI > delayI)?delayI+1:0)];
Float64 outJ = aJ[countJ-((countJ > delayJ)?delayJ+1:0)];
Float64 outK = aK[countK-((countK > delayK)?delayK+1:0)];
Float64 outL = aL[countL-((countL > delayL)?delayL+1:0)];
//first block: now we have four outputs
aA[countA] = (outI - (outJ + outK + outL));
aB[countB] = (outJ - (outI + outK + outL));
aC[countC] = (outK - (outI + outJ + outL));
aD[countD] = (outL - (outI + outJ + outK));
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;
Float64 outA = aA[countA-((countA > delayA)?delayA+1:0)];
Float64 outB = aB[countB-((countB > delayB)?delayB+1:0)];
Float64 outC = aC[countC-((countC > delayC)?delayC+1:0)];
Float64 outD = aD[countD-((countD > delayD)?delayD+1:0)];
//second block: four more outputs
aE[countE] = (outA - (outB + outC + outD));
aF[countF] = (outB - (outA + outC + outD));
aG[countG] = (outC - (outA + outB + outD));
aH[countH] = (outD - (outA + outB + outC));
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;
Float64 outE = aE[countE-((countE > delayE)?delayE+1:0)];
Float64 outF = aF[countF-((countF > delayF)?delayF+1:0)];
Float64 outG = aG[countG-((countG > delayG)?delayG+1:0)];
Float64 outH = aH[countH-((countH > delayH)?delayH+1:0)];
//third block: final outputs
feedbackA = (outE - (outF + outG + outH));
feedbackB = (outF - (outE + outG + outH));
feedbackC = (outG - (outE + outF + outH));
feedbackD = (outH - (outE + outF + outG));
//which we need to feed back into the input again, a bit
inputSample = (outE + outF + outG + outH)/8.0;
//and take the final combined sum of outputs
if (cycleEnd == 4) {
lastRef[0] = lastRef[4]; //start from previous last
lastRef[2] = (lastRef[0] + inputSample)/2; //half
lastRef[1] = (lastRef[0] + lastRef[2])/2; //one quarter
lastRef[3] = (lastRef[2] + inputSample)/2; //three quarters
lastRef[4] = inputSample; //full
}
if (cycleEnd == 3) {
lastRef[0] = lastRef[3]; //start from previous last
lastRef[2] = (lastRef[0]+lastRef[0]+inputSample)/3; //third
lastRef[1] = (lastRef[0]+inputSample+inputSample)/3; //two thirds
lastRef[3] = inputSample; //full
}
if (cycleEnd == 2) {
lastRef[0] = lastRef[2]; //start from previous last
lastRef[1] = (lastRef[0] + inputSample)/2; //half
lastRef[2] = inputSample; //full
}
if (cycleEnd == 1) lastRef[0] = inputSample;
cycle = 0; //reset
inputSample = lastRef[cycle];
} else {
inputSample = lastRef[cycle];
//we are going through our references now
}
switch (cycleEnd) //multi-pole average using lastRef[] variables
{
case 4:
lastRef[8] = inputSample; inputSample = (inputSample+lastRef[7])*0.5;
lastRef[7] = lastRef[8]; //continue, do not break
case 3:
lastRef[8] = inputSample; inputSample = (inputSample+lastRef[6])*0.5;
lastRef[6] = lastRef[8]; //continue, do not break
case 2:
lastRef[8] = inputSample; inputSample = (inputSample+lastRef[5])*0.5;
lastRef[5] = lastRef[8]; //continue, do not break
case 1:
break; //no further averaging
}
iirB = (iirB*(1.0-lowpass))+(inputSample*lowpass); inputSample = iirB;
//end filter
if (wet < 1.0) inputSample *= wet;
if (dry < 1.0) drySample *= dry;
inputSample += drySample;
//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 floating point dither
int expon; frexpf((float)inputSample, &expon);
fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5;
inputSample += ((double(fpd)-uint32_t(0x7fffffff)) * 5.5e-36l * pow(2,expon+62));
//end 32 bit floating point dither
*destP = inputSample;
sourceP += inNumChannels; destP += inNumChannels;
}
}