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

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/*
* File: MultiBandDistortion.cpp
*
* Version: 1.0
*
* Created: 4/24/11
*
* Copyright: Copyright © 2011 Airwindows, Airwindows uses the MIT license
*
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/*=============================================================================
MultiBandDistortion.h
=============================================================================*/
#include "MultiBandDistortion.h"
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
COMPONENT_ENTRY(MultiBandDistortion)
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// MultiBandDistortion::MultiBandDistortion
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
MultiBandDistortion::MultiBandDistortion(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 );
SetParameter(kParam_Seven, kDefaultValue_ParamSeven );
SetParameter(kParam_Eight, kDefaultValue_ParamEight );
SetParameter(kParam_Nine, kDefaultValue_ParamNine );
#if AU_DEBUG_DISPATCHER
mDebugDispatcher = new AUDebugDispatcher (this);
#endif
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// MultiBandDistortion::GetParameterValueStrings
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult MultiBandDistortion::GetParameterValueStrings(AudioUnitScope inScope,
AudioUnitParameterID inParameterID,
CFArrayRef * outStrings)
{
return kAudioUnitErr_InvalidProperty;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// MultiBandDistortion::GetParameterInfo
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult MultiBandDistortion::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_Decibels;
outParameterInfo.minValue = 0.0;
outParameterInfo.maxValue = 48.0;
outParameterInfo.defaultValue = kDefaultValue_ParamTwo;
break;
case kParam_Three:
AUBase::FillInParameterName (outParameterInfo, kParameterThreeName, false);
outParameterInfo.unit = kAudioUnitParameterUnit_Decibels;
outParameterInfo.minValue = 0.0;
outParameterInfo.maxValue = 48.0;
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;
case kParam_Seven:
AUBase::FillInParameterName (outParameterInfo, kParameterSevenName, false);
outParameterInfo.unit = kAudioUnitParameterUnit_Generic;
outParameterInfo.minValue = 0.0;
outParameterInfo.maxValue = 1.0;
outParameterInfo.defaultValue = kDefaultValue_ParamSeven;
break;
case kParam_Eight:
AUBase::FillInParameterName (outParameterInfo, kParameterEightName, false);
outParameterInfo.unit = kAudioUnitParameterUnit_Generic;
outParameterInfo.minValue = 0.0;
outParameterInfo.maxValue = 1.0;
outParameterInfo.defaultValue = kDefaultValue_ParamEight;
break;
case kParam_Nine:
AUBase::FillInParameterName (outParameterInfo, kParameterNineName, false);
outParameterInfo.unit = kAudioUnitParameterUnit_Decibels;
outParameterInfo.minValue = -48.0;
outParameterInfo.maxValue = 0.0;
outParameterInfo.defaultValue = kDefaultValue_ParamNine;
break;
default:
result = kAudioUnitErr_InvalidParameter;
break;
}
} else {
result = kAudioUnitErr_InvalidParameter;
}
return result;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// MultiBandDistortion::GetPropertyInfo
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult MultiBandDistortion::GetPropertyInfo (AudioUnitPropertyID inID,
AudioUnitScope inScope,
AudioUnitElement inElement,
UInt32 & outDataSize,
Boolean & outWritable)
{
return AUEffectBase::GetPropertyInfo (inID, inScope, inElement, outDataSize, outWritable);
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// MultiBandDistortion::GetProperty
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult MultiBandDistortion::GetProperty( AudioUnitPropertyID inID,
AudioUnitScope inScope,
AudioUnitElement inElement,
void * outData )
{
return AUEffectBase::GetProperty (inID, inScope, inElement, outData);
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// MultiBandDistortion::Initialize
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult MultiBandDistortion::Initialize()
{
ComponentResult result = AUEffectBase::Initialize();
if (result == noErr)
Reset(kAudioUnitScope_Global, 0);
return result;
}
#pragma mark ____MultiBandDistortionEffectKernel
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// MultiBandDistortion::MultiBandDistortionKernel::Reset()
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
void MultiBandDistortion::MultiBandDistortionKernel::Reset()
{
ataLast3Sample = ataLast2Sample = ataLast1Sample = 0.0;
ataHalfwaySample = ataHalfDrySample = ataHalfDiffSample = 0.0;
ataA = ataB = ataC = ataDrySample = ataDiffSample = ataPrevDiffSample = 0.0;
ataUpsampleHighTweak = 0.0414213562373095048801688; //more adds treble to upsampling
ataDecay = 0.915965594177219015; //Catalan's constant, more adds focus and clarity
ataFlip = false; //end reset of antialias parameters
iirSampleA = 0; iirSampleB = 0;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// MultiBandDistortion::MultiBandDistortionKernel::Process
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
void MultiBandDistortion::MultiBandDistortionKernel::Process( const Float32 *inSourceP,
Float32 *inDestP,
UInt32 inFramesToProcess,
UInt32 inNumChannels,
bool &ioSilence )
{
UInt32 nSampleFrames = inFramesToProcess;
const Float32 *sourceP = inSourceP;
Float32 *destP = inDestP;
Float64 overallscale = 1.0;
overallscale /= 44100.0;
overallscale *= GetSampleRate();
Float64 iirAmount = pow(GetParameter( kParam_One ),3)/overallscale;
Float64 gainH = pow(10.0,GetParameter( kParam_Two )/20);
Float64 thresholdH = GetParameter( kParam_Four );
Float64 hardnessH;
if (thresholdH < 1.0) hardnessH = 1.0 / (1.0-thresholdH);
else hardnessH = 999999999999999999999.0;
Float64 gainL = pow(10.0,GetParameter( kParam_Three )/20);
Float64 thresholdL = GetParameter( kParam_Five );
Float64 hardnessL;
if (thresholdL < 1.0) hardnessL = 1.0 / (1.0-thresholdL);
else hardnessL = 999999999999999999999.0;
Float64 breakup = 1.5707963267949;
Float64 bridgerectifier;
Float64 outputH = GetParameter( kParam_Six );
Float64 outputL = GetParameter( kParam_Seven );
Float64 outputD = GetParameter( kParam_Eight )*0.597;
Float64 outtrim = outputH + outputL + outputD;
outputH *= outtrim;
outputL *= outtrim;
outputD *= outtrim;
Float64 outputGlobal = pow(10.0,GetParameter( kParam_Nine )/20);
Float64 inputSample;
Float64 tempH;
Float64 tempL;
while (nSampleFrames-- > 0) {
ataDrySample = inputSample = *sourceP;
ataHalfDrySample = ataHalfwaySample = (inputSample + ataLast1Sample + ((-ataLast2Sample + ataLast3Sample) * ataUpsampleHighTweak)) / 2.0;
ataLast3Sample = ataLast2Sample; ataLast2Sample = ataLast1Sample; ataLast1Sample = inputSample;
//setting up oversampled special antialiasing
//pre-center code on inputSample and halfwaySample in parallel
//begin interpolated sample- change inputSample -> ataHalfwaySample, ataDrySample -> ataHalfDrySample
tempL = iirSampleA = (iirSampleA * (1 - iirAmount)) + (ataHalfwaySample * iirAmount);
tempH = ataHalfwaySample - iirSampleA;
//highpass section
tempH *= gainH;
if (fabs(tempH) > thresholdH)
{
bridgerectifier = (fabs(tempH)-thresholdH)*hardnessH;
//skip flat area if any, scale to distortion limit
if (bridgerectifier > breakup) bridgerectifier = breakup;
//max value for sine function, 'breakup' modeling for trashed console tone
//more hardness = more solidness behind breakup modeling. more softness, more 'grunge' and sag
bridgerectifier = sin(bridgerectifier)/hardnessH;
//do the sine factor, scale back to proper amount
if (tempH > 0) tempH = bridgerectifier+thresholdH;
else tempH = -(bridgerectifier+thresholdH);
}
//ADClip
tempL *= gainL;
if (fabs(tempL) > thresholdL)
{
bridgerectifier = (fabs(tempL)-thresholdL)*hardnessL;
//skip flat area if any, scale to distortion limit
if (bridgerectifier > breakup) bridgerectifier = breakup;
//max value for sine function, 'breakup' modeling for trashed console tone
//more hardness = more solidness behind breakup modeling. more softness, more 'grunge' and sag
bridgerectifier = sin(bridgerectifier)/hardnessL;
//do the sine factor, scale back to proper amount
if (tempL > 0) tempL = bridgerectifier+thresholdL;
else tempL = -(bridgerectifier+thresholdL);
}
//ADClip
ataHalfwaySample = (tempL * outputL) + (tempH * outputH);
//end interpolated sample
//begin raw sample- inputSample and ataDrySample handled separately here
tempL = iirSampleB = (iirSampleB * (1 - iirAmount)) + (inputSample * iirAmount);
tempH = inputSample - iirSampleB;
//highpass section
tempH *= gainH;
if (fabs(tempH) > thresholdH)
{
bridgerectifier = (fabs(tempH)-thresholdH)*hardnessH;
//skip flat area if any, scale to distortion limit
if (bridgerectifier > breakup) bridgerectifier = breakup;
//max value for sine function, 'breakup' modeling for trashed console tone
//more hardness = more solidness behind breakup modeling. more softness, more 'grunge' and sag
bridgerectifier = sin(bridgerectifier)/hardnessH;
//do the sine factor, scale back to proper amount
if (tempH > 0) tempH = bridgerectifier+thresholdH;
else tempH = -(bridgerectifier+thresholdH);
}
//ADClip
tempL *= gainL;
if (fabs(tempL) > thresholdL)
{
bridgerectifier = (fabs(tempL)-thresholdL)*hardnessL;
//skip flat area if any, scale to distortion limit
if (bridgerectifier > breakup) bridgerectifier = breakup;
//max value for sine function, 'breakup' modeling for trashed console tone
//more hardness = more solidness behind breakup modeling. more softness, more 'grunge' and sag
bridgerectifier = sin(bridgerectifier)/hardnessL;
//do the sine factor, scale back to proper amount
if (tempL > 0) tempL = bridgerectifier+thresholdL;
else tempL = -(bridgerectifier+thresholdL);
}
//ADClip
inputSample = (tempL * outputL) + (tempH * outputH);
//end raw sample
//begin center code handling conv stuff tied to 44.1K, or stuff in time domain like delays
//ataHalfwaySample -= inputSample;
//retain only difference with raw signal
//inputSample += convolutionstuff[count];
//ataHalfwaySample += inputSample;
//restore interpolated signal including time domain stuff now
//end center code for handling timedomain/conv stuff
//post-center code on inputSample and halfwaySample in parallel
//begin raw sample- inputSample and ataDrySample handled separately here
//inputSample *= gain;
//end raw sample
//begin interpolated sample- change inputSample -> ataHalfwaySample, ataDrySample -> ataHalfDrySample
//ataHalfwaySample *= gain;
//end interpolated sample
//begin antialiasing section for halfway sample
ataC = ataHalfwaySample - ataHalfDrySample;
if (ataFlip) {ataA *= ataDecay; ataB *= ataDecay; ataA += ataC; ataB -= ataC; ataC = ataA;}
else {ataB *= ataDecay; ataA *= ataDecay; ataB += ataC; ataA -= ataC; ataC = ataB;}
ataHalfDiffSample = (ataC * ataDecay);
//end antialiasing section for halfway sample
//begin antialiasing section for raw sample
ataC = inputSample - ataDrySample;
if (ataFlip) {ataA *= ataDecay; ataB *= ataDecay; ataA += ataC; ataB -= ataC; ataC = ataA;}
else {ataB *= ataDecay; ataA *= ataDecay; ataB += ataC; ataA -= ataC; ataC = ataB;}
ataDiffSample = (ataC * ataDecay);
//end antialiasing section for input sample
ataFlip = !ataFlip;
inputSample = ataDrySample*outputD; inputSample += (ataDiffSample + ataHalfDiffSample);
//apply processing as difference to non-oversampled raw input
//inputSample *= output; *destP = (ataDrySample*dry)+(inputSample*wet);
//built in output trim and dry/wet if desired
*destP = inputSample*outputGlobal;
sourceP += inNumChannels; destP += inNumChannels;
}
}