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airwindows/plugins/MacSignedAU/DubPlate2/DubPlate2.cpp
Christopher Johnson 21901acae0 DubPlate2
2025-03-30 15:34:42 -04:00

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/*
* File: DubPlate2.cpp
*
* Version: 1.0
*
* Created: 3/21/25
*
* Copyright: Copyright © 2025 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
* original Apple software (the "Apple Software"), to use, reproduce, modify and redistribute the
* Apple Software, with or without modifications, in source and/or binary forms; provided that if you
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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.
*
* IN NO EVENT SHALL APPLE BE LIABLE FOR ANY SPECIAL, INDIRECT, INCIDENTAL OR CONSEQUENTIAL
* DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS
* OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) ARISING IN ANY WAY OUT OF THE USE,
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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.
*
*/
/*=============================================================================
DubPlate2.cpp
=============================================================================*/
#include "DubPlate2.h"
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
AUDIOCOMPONENT_ENTRY(AUBaseFactory, DubPlate2)
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// DubPlate2::DubPlate2
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
DubPlate2::DubPlate2(AudioUnit component)
: AUEffectBase(component)
{
CreateElements();
Globals()->UseIndexedParameters(kNumberOfParameters);
SetParameter(kParam_A, kDefaultValue_ParamA );
SetParameter(kParam_B, kDefaultValue_ParamB );
SetParameter(kParam_C, kDefaultValue_ParamC );
#if AU_DEBUG_DISPATCHER
mDebugDispatcher = new AUDebugDispatcher (this);
#endif
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// DubPlate2::GetParameterValueStrings
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult DubPlate2::GetParameterValueStrings(AudioUnitScope inScope,
AudioUnitParameterID inParameterID,
CFArrayRef * outStrings)
{
return kAudioUnitErr_InvalidProperty;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// DubPlate2::GetParameterInfo
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult DubPlate2::GetParameterInfo(AudioUnitScope inScope,
AudioUnitParameterID inParameterID,
AudioUnitParameterInfo &outParameterInfo )
{
ComponentResult result = noErr;
outParameterInfo.flags = kAudioUnitParameterFlag_IsWritable
| kAudioUnitParameterFlag_IsReadable;
if (inScope == kAudioUnitScope_Global) {
switch(inParameterID)
{
case kParam_A:
AUBase::FillInParameterName (outParameterInfo, kParameterAName, false);
outParameterInfo.unit = kAudioUnitParameterUnit_Generic;
outParameterInfo.minValue = 0.0;
outParameterInfo.maxValue = 1.0;
outParameterInfo.defaultValue = kDefaultValue_ParamA;
break;
case kParam_B:
AUBase::FillInParameterName (outParameterInfo, kParameterBName, false);
outParameterInfo.unit = kAudioUnitParameterUnit_Generic;
outParameterInfo.minValue = 0.0;
outParameterInfo.maxValue = 1.0;
outParameterInfo.defaultValue = kDefaultValue_ParamB;
break;
case kParam_C:
AUBase::FillInParameterName (outParameterInfo, kParameterCName, false);
outParameterInfo.unit = kAudioUnitParameterUnit_Generic;
outParameterInfo.minValue = 0.0;
outParameterInfo.maxValue = 1.0;
outParameterInfo.defaultValue = kDefaultValue_ParamC;
break;
default:
result = kAudioUnitErr_InvalidParameter;
break;
}
} else {
result = kAudioUnitErr_InvalidParameter;
}
return result;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// DubPlate2::GetPropertyInfo
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult DubPlate2::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 DubPlate2::SupportedNumChannels(const AUChannelInfo ** outInfo)
{
if (outInfo != NULL)
{
static AUChannelInfo info;
info.inChannels = 2;
info.outChannels = 2;
*outInfo = &info;
}
return 1;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// DubPlate2::GetProperty
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult DubPlate2::GetProperty( AudioUnitPropertyID inID,
AudioUnitScope inScope,
AudioUnitElement inElement,
void * outData )
{
return AUEffectBase::GetProperty (inID, inScope, inElement, outData);
}
// DubPlate2::Initialize
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult DubPlate2::Initialize()
{
ComponentResult result = AUEffectBase::Initialize();
if (result == noErr)
Reset(kAudioUnitScope_Global, 0);
return result;
}
#pragma mark ____DubPlate2EffectKernel
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// DubPlate2::DubPlate2Kernel::Reset()
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult DubPlate2::Reset(AudioUnitScope inScope, AudioUnitElement inElement)
{
for (int x = 0; x < bax_total; x++) {baxH[x] = 0.0;baxL[x] = 0.0;}
flip = false;
iirA = 0.0;
iirB = 0.0;
iirC = 0.0;
iirD = 0.0;
lastSinewAL = 0.0;
lastSinewAR = 0.0;
lastSinewBL = 0.0;
lastSinewBR = 0.0;
fpdL = 1.0; while (fpdL < 16386) fpdL = rand()*UINT32_MAX;
fpdR = 1.0; while (fpdR < 16386) fpdR = rand()*UINT32_MAX;
return noErr;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// DubPlate2::ProcessBufferLists
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
OSStatus DubPlate2::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();
double inputGain = pow(GetParameter( kParam_A )*2.0,2.0);
double trebleGain = pow(GetParameter( kParam_B )*2.0,2.0);
double trebleFreq = ((2000.0*trebleGain)+200.0)/GetSampleRate();
if (trebleFreq > 0.45) trebleFreq = 0.45;
baxH[bax_freq] = trebleFreq;
baxH[bax_reso] = 0.57735026919; //bessel second order
double K = tan(M_PI * baxH[bax_freq]); //lowpass
double norm = 1.0 / (1.0 + K / baxH[bax_reso] + K * K);
baxH[bax_a0] = K * K * norm;
baxH[bax_a1] = 2.0 * baxH[bax_a0];
baxH[bax_a2] = baxH[bax_a0];
baxH[bax_b1] = 2.0 * (K * K - 1.0) * norm;
baxH[bax_b2] = (1.0 - K / baxH[bax_reso] + K * K) * norm;
//end bax highpass
double bassGain = pow(GetParameter( kParam_C )*2.0,2.0);
double bassFreq = pow((1.0-GetParameter( kParam_C ))*2.0,2.0);
bassFreq = ((2000.0*bassFreq)+200.0)/GetSampleRate();
if (bassFreq > 0.45) bassFreq = 0.45;
baxL[bax_freq] = bassFreq;
baxL[bax_reso] = 0.57735026919; //bessel second order
K = tan(M_PI * baxL[bax_freq]); //lowpass
norm = 1.0 / (1.0 + K / baxL[bax_reso] + K * K);
baxL[bax_a0] = K * K * norm;
baxL[bax_a1] = 2.0 * baxL[bax_a0];
baxL[bax_a2] = baxL[bax_a0];
baxL[bax_b1] = 2.0 * (K * K - 1.0) * norm;
baxL[bax_b2] = (1.0 - K / baxL[bax_reso] + K * K) * norm;
//end bax lowpass
double iirSide = 0.01862 / overallscale;
double iirMid = 0.01102 / overallscale;
double threshSinewA = 0.1442 / overallscale;
double threshSinewB = 0.0274 / overallscale;
while (nSampleFrames-- > 0) {
double inputSampleL = *inputL;
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;
inputSampleL = sin(fmax(fmin(inputSampleL*inputGain,M_PI_2),-M_PI_2));
inputSampleR = sin(fmax(fmin(inputSampleR*inputGain,M_PI_2),-M_PI_2));
//encode Console5: good cleanness
double trebleSampleL;
double trebleSampleR;
double bassSampleL;
double bassSampleR;
if (flip) {
trebleSampleL = (inputSampleL * baxH[bax_a0]) + baxH[bax_sLA1];
baxH[bax_sLA1] = (inputSampleL * baxH[bax_a1]) - (trebleSampleL * baxH[bax_b1]) + baxH[bax_sLA2];
baxH[bax_sLA2] = (inputSampleL * baxH[bax_a2]) - (trebleSampleL * baxH[bax_b2]);
trebleSampleL = inputSampleL - trebleSampleL;
trebleSampleR = (inputSampleR * baxH[bax_a0]) + baxH[bax_sRA1];
baxH[bax_sRA1] = (inputSampleR * baxH[bax_a1]) - (trebleSampleR * baxH[bax_b1]) + baxH[bax_sRA2];
baxH[bax_sRA2] = (inputSampleR * baxH[bax_a2]) - (trebleSampleR * baxH[bax_b2]);
trebleSampleR = inputSampleR - trebleSampleR;
bassSampleL = (inputSampleL * baxL[bax_a0]) + baxL[bax_sLA1];
baxL[bax_sLA1] = (inputSampleL * baxL[bax_a1]) - (bassSampleL * baxL[bax_b1]) + baxL[bax_sLA2];
baxL[bax_sLA2] = (inputSampleL * baxL[bax_a2]) - (bassSampleL * baxL[bax_b2]);
bassSampleR = (inputSampleR * baxL[bax_a0]) + baxL[bax_sRA1];
baxL[bax_sRA1] = (inputSampleR * baxL[bax_a1]) - (bassSampleR * baxL[bax_b1]) + baxL[bax_sRA2];
baxL[bax_sRA2] = (inputSampleR * baxL[bax_a2]) - (bassSampleR * baxL[bax_b2]);
} else {
trebleSampleL = (inputSampleL * baxH[bax_a0]) + baxH[bax_sLB1];
baxH[bax_sLB1] = (inputSampleL * baxH[bax_a1]) - (trebleSampleL * baxH[bax_b1]) + baxH[bax_sLB2];
baxH[bax_sLB2] = (inputSampleL * baxH[bax_a2]) - (trebleSampleL * baxH[bax_b2]);
trebleSampleL = inputSampleL - trebleSampleL;
trebleSampleR = (inputSampleR * baxH[bax_a0]) + baxH[bax_sRB1];
baxH[bax_sRB1] = (inputSampleR * baxH[bax_a1]) - (trebleSampleR * baxH[bax_b1]) + baxH[bax_sRB2];
baxH[bax_sRB2] = (inputSampleR * baxH[bax_a2]) - (trebleSampleR * baxH[bax_b2]);
trebleSampleR = inputSampleR - trebleSampleR;
bassSampleL = (inputSampleL * baxL[bax_a0]) + baxL[bax_sLB1];
baxL[bax_sLB1] = (inputSampleL * baxL[bax_a1]) - (bassSampleL * baxL[bax_b1]) + baxL[bax_sLB2];
baxL[bax_sLB2] = (inputSampleL * baxL[bax_a2]) - (bassSampleL * baxL[bax_b2]);
bassSampleR = (inputSampleR * baxL[bax_a0]) + baxL[bax_sRB1];
baxL[bax_sRB1] = (inputSampleR * baxL[bax_a1]) - (bassSampleR * baxL[bax_b1]) + baxL[bax_sRB2];
baxL[bax_sRB2] = (inputSampleR * baxL[bax_a2]) - (bassSampleR * baxL[bax_b2]);
}
trebleSampleL *= trebleGain;
trebleSampleR *= trebleGain;
bassSampleL *= bassGain;
bassSampleR *= bassGain;
inputSampleL = bassSampleL + trebleSampleL; //interleaved biquad
inputSampleR = bassSampleR + trebleSampleR; //interleaved biquad
inputSampleL = asin(fmax(fmin(inputSampleL,0.99999),-0.99999));
inputSampleR = asin(fmax(fmin(inputSampleR,0.99999),-0.99999));
//amplitude aspect
double mid = inputSampleL + inputSampleR;
double side = inputSampleL - inputSampleR;
//assign mid and side.Between these sections, you can do mid/side processing
double temp = side;
double correction;
if (flip) {
iirA = (iirA * (1.0 - iirSide)) + (temp * iirSide); temp -= iirA; correction = iirA;
} else {
iirB = (iirB * (1.0 - iirSide)) + (temp * iirSide); temp -= iirB; correction = iirB;
}
iirC = (iirC * (1.0 - iirSide)) + (temp * iirSide); temp -= iirC; correction += (iirC * 0.162);
side -= sin(correction);
iirD = (iirD * (1.0 - iirMid)) + (mid * iirMid);
mid -= sin(iirD);
//gonna cut those lows a hair
inputSampleL = (mid+side)/2.0;
inputSampleR = (mid-side)/2.0;
//unassign mid and side
temp = inputSampleL;
double sinew = threshSinewA * cos(lastSinewAL*lastSinewAL);
if (inputSampleL - lastSinewAL > sinew) temp = lastSinewAL + sinew;
if (-(inputSampleL - lastSinewAL) > sinew) temp = lastSinewAL - sinew;
lastSinewAL = temp;
if (lastSinewAL > 1.0) lastSinewAL = 1.0;
if (lastSinewAL < -1.0) lastSinewAL = -1.0;
inputSampleL = (inputSampleL * 0.5)+(lastSinewAL * 0.5);
sinew = threshSinewB * cos(lastSinewBL*lastSinewBL);
if (inputSampleL - lastSinewBL > sinew) temp = lastSinewBL + sinew;
if (-(inputSampleL - lastSinewBL) > sinew) temp = lastSinewBL - sinew;
lastSinewBL = temp;
if (lastSinewBL > 1.0) lastSinewBL = 1.0;
if (lastSinewBL < -1.0) lastSinewBL = -1.0;
inputSampleL = (inputSampleL * 0.5)+(lastSinewBL * 0.5);
temp = inputSampleR;
sinew = threshSinewA * cos(lastSinewAR*lastSinewAR);
if (inputSampleR - lastSinewAR > sinew) temp = lastSinewAR + sinew;
if (-(inputSampleR - lastSinewAR) > sinew) temp = lastSinewAR - sinew;
lastSinewAR = temp;
if (lastSinewAR > 1.0) lastSinewAR = 1.0;
if (lastSinewAR < -1.0) lastSinewAR = -1.0;
inputSampleR = (inputSampleR * 0.5)+(lastSinewAR * 0.5);
sinew = threshSinewB * cos(lastSinewBR*lastSinewBR);
if (inputSampleR - lastSinewBR > sinew) temp = lastSinewBR + sinew;
if (-(inputSampleR - lastSinewBR) > sinew) temp = lastSinewBR - sinew;
lastSinewBR = temp;
if (lastSinewBR > 1.0) lastSinewBR = 1.0;
if (lastSinewBR < -1.0) lastSinewBR = -1.0;
inputSampleR = (inputSampleR * 0.5)+(lastSinewBR * 0.5);
//run Sinew to stop excess slews, two layers to make it more audible
flip = !flip; //both for Baxandall3 and DubPlate highpass
//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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