airwindows/plugins/MacAU/TapeDelay2/TapeDelay2.cpp
2022-11-21 09:20:21 -05:00

362 lines
15 KiB
C++

/*
* File: TapeDelay2.cpp
*
* Version: 1.0
*
* Created: 7/19/21
*
* Copyright: Copyright © 2021 Airwindows, Airwindows uses the MIT license
*
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/*=============================================================================
TapeDelay2.cpp
=============================================================================*/
#include "TapeDelay2.h"
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
COMPONENT_ENTRY(TapeDelay2)
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// TapeDelay2::TapeDelay2
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
TapeDelay2::TapeDelay2(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 );
SetParameter(kParam_Six, kDefaultValue_ParamSix );
#if AU_DEBUG_DISPATCHER
mDebugDispatcher = new AUDebugDispatcher (this);
#endif
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// TapeDelay2::GetParameterValueStrings
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult TapeDelay2::GetParameterValueStrings(AudioUnitScope inScope,
AudioUnitParameterID inParameterID,
CFArrayRef * outStrings)
{
return kAudioUnitErr_InvalidProperty;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// TapeDelay2::GetParameterInfo
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult TapeDelay2::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;
case kParam_Six:
AUBase::FillInParameterName (outParameterInfo, kParameterSixName, false);
outParameterInfo.unit = kAudioUnitParameterUnit_Generic;
outParameterInfo.minValue = 0.0;
outParameterInfo.maxValue = 1.0;
outParameterInfo.defaultValue = kDefaultValue_ParamSix;
break;
default:
result = kAudioUnitErr_InvalidParameter;
break;
}
} else {
result = kAudioUnitErr_InvalidParameter;
}
return result;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// TapeDelay2::GetPropertyInfo
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult TapeDelay2::GetPropertyInfo (AudioUnitPropertyID inID,
AudioUnitScope inScope,
AudioUnitElement inElement,
UInt32 & outDataSize,
Boolean & outWritable)
{
return AUEffectBase::GetPropertyInfo (inID, inScope, inElement, outDataSize, outWritable);
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// TapeDelay2::GetProperty
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult TapeDelay2::GetProperty( AudioUnitPropertyID inID,
AudioUnitScope inScope,
AudioUnitElement inElement,
void * outData )
{
return AUEffectBase::GetProperty (inID, inScope, inElement, outData);
}
// TapeDelay2::Initialize
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult TapeDelay2::Initialize()
{
ComponentResult result = AUEffectBase::Initialize();
if (result == noErr)
Reset(kAudioUnitScope_Global, 0);
return result;
}
#pragma mark ____TapeDelay2EffectKernel
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// TapeDelay2::TapeDelay2Kernel::Reset()
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
void TapeDelay2::TapeDelay2Kernel::Reset()
{
for(int x = 0; x < 88210; x++) {d[x] = 0.0;}
prevSample = 0.0;
delay = 0.0;
sweep = 0.0;
for (int x = 0; x < 9; x++) {regenFilter[x] = 0.0; outFilter[x] = 0.0;}
for (int x = 0; x < 9; x++) {lastRef[x] = 0.0;}
cycle = 0;
fpd = 1.0; while (fpd < 16386) fpd = rand()*UINT32_MAX;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// TapeDelay2::TapeDelay2Kernel::Process
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
void TapeDelay2::TapeDelay2Kernel::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 baseSpeed = (pow(GetParameter( kParam_One ),4)*25.0)+1.0;
Float64 feedback = pow(GetParameter( kParam_Two ),2);
//[0] is frequency: 0.000001 to 0.499999 is near-zero to near-Nyquist
//[1] is resonance, 0.7071 is Butterworth. Also can't be zero
regenFilter[0] = ((pow(GetParameter( kParam_Three ),3)*0.4)+0.0001);
regenFilter[1] = pow(GetParameter( kParam_Four ),2)+0.01; //resonance
double K = tan(M_PI * regenFilter[0]);
double norm = 1.0 / (1.0 + K / regenFilter[1] + K * K);
regenFilter[2] = K / regenFilter[1] * norm;
regenFilter[4] = -regenFilter[2];
regenFilter[5] = 2.0 * (K * K - 1.0) * norm;
regenFilter[6] = (1.0 - K / regenFilter[1] + K * K) * norm;
//[0] is frequency: 0.000001 to 0.499999 is near-zero to near-Nyquist
//[1] is resonance, 0.7071 is Butterworth. Also can't be zero
outFilter[0] = regenFilter[0];
outFilter[1] = regenFilter[1] * 1.618033988749894848204586; //resonance
K = tan(M_PI * outFilter[0]);
norm = 1.0 / (1.0 + K / outFilter[1] + K * K);
outFilter[2] = K / outFilter[1] * norm;
outFilter[4] = -outFilter[2];
outFilter[5] = 2.0 * (K * K - 1.0) * norm;
outFilter[6] = (1.0 - K / outFilter[1] + K * K) * norm;
Float64 vibSpeed = pow(GetParameter( kParam_Five ),5) * baseSpeed * ((regenFilter[0]*0.09)+0.025);//0.05
Float64 wet = GetParameter( kParam_Six )*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 echo 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 echo loudness.
while (nSampleFrames-- > 0) {
double inputSample = *sourceP;
if (fabs(inputSample)<1.18e-23) inputSample = fpd * 1.18e-17;
double drySample = inputSample;
cycle++;
if (cycle == cycleEnd) {
Float64 speed = baseSpeed + (vibSpeed * (sin(sweep)+1.0));
sweep += (0.05*inputSample*inputSample); if (sweep > 6.283185307179586) sweep -= 6.283185307179586;
int pos = floor(delay);
Float64 newSample = inputSample + d[pos]*feedback;
double tempSample = (newSample * regenFilter[2]) + regenFilter[7];
regenFilter[7] = -(tempSample * regenFilter[5]) + regenFilter[8];
regenFilter[8] = (newSample * regenFilter[4]) - (tempSample * regenFilter[6]);
newSample = tempSample;
delay -= speed; if (delay < 0) delay += 88200.0;
Float64 increment = (newSample - prevSample) / speed;
d[pos] = prevSample;
while (pos != floor(delay)) {
d[pos] = prevSample;
prevSample += increment;
pos--; if (pos < 0) pos += 88200;
}
prevSample = newSample;
pos = floor(delay); inputSample = d[pos];
tempSample = (inputSample * outFilter[2]) + outFilter[7];
outFilter[7] = -(tempSample * outFilter[5]) + outFilter[8];
outFilter[8] = (inputSample * outFilter[4]) - (tempSample * outFilter[6]);
inputSample = tempSample;
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
}
if (wet < 1.0) inputSample *= wet;
if (dry < 1.0) drySample *= dry;
inputSample += drySample;
//this is our submix echo dry/wet: 0.5 is BOTH at FULL VOLUME
//purpose is that, if you're adding echo, 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;
}
}