/*
* File: Pyewacket.cpp
*
* Version: 1.0
*
* Created: 10/20/16
*
* Copyright: Copyright � 2016 Airwindows, All Rights Reserved
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/*=============================================================================
Pyewacket.cpp
=============================================================================*/
#include "Pyewacket.h"
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
COMPONENT_ENTRY(Pyewacket)
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// Pyewacket::Pyewacket
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
Pyewacket::Pyewacket(AudioUnit component)
: AUEffectBase(component)
{
CreateElements();
Globals()->UseIndexedParameters(kNumberOfParameters);
SetParameter(kParam_One, kDefaultValue_ParamOne );
SetParameter(kParam_Two, kDefaultValue_ParamTwo );
SetParameter(kParam_Three, kDefaultValue_ParamThree );
#if AU_DEBUG_DISPATCHER
mDebugDispatcher = new AUDebugDispatcher (this);
#endif
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// Pyewacket::GetParameterValueStrings
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult Pyewacket::GetParameterValueStrings(AudioUnitScope inScope,
AudioUnitParameterID inParameterID,
CFArrayRef * outStrings)
{
return kAudioUnitErr_InvalidProperty;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// Pyewacket::GetParameterInfo
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult Pyewacket::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 = -12.0;
outParameterInfo.maxValue = 12.0;
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 = -12.0;
outParameterInfo.maxValue = 12.0;
outParameterInfo.defaultValue = kDefaultValue_ParamThree;
break;
default:
result = kAudioUnitErr_InvalidParameter;
break;
}
} else {
result = kAudioUnitErr_InvalidParameter;
}
return result;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// Pyewacket::GetPropertyInfo
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult Pyewacket::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 Pyewacket::SupportedNumChannels(const AUChannelInfo ** outInfo)
{
if (outInfo != NULL)
{
static AUChannelInfo info;
info.inChannels = 2;
info.outChannels = 2;
*outInfo = &info;
}
return 1;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// Pyewacket::GetProperty
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult Pyewacket::GetProperty( AudioUnitPropertyID inID,
AudioUnitScope inScope,
AudioUnitElement inElement,
void * outData )
{
return AUEffectBase::GetProperty (inID, inScope, inElement, outData);
}
// Pyewacket::Initialize
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult Pyewacket::Initialize()
{
ComponentResult result = AUEffectBase::Initialize();
if (result == noErr)
Reset(kAudioUnitScope_Global, 0);
return result;
}
#pragma mark ____PyewacketEffectKernel
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// Pyewacket::PyewacketKernel::Reset()
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult Pyewacket::Reset(AudioUnitScope inScope, AudioUnitElement inElement)
{
fpNShapeL = 0.0;
fpNShapeR = 0.0;
chase = 1.0;
lastrectifierL = 0.0;
lastrectifierR = 0.0;
return noErr;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// Pyewacket::ProcessBufferLists
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
OSStatus Pyewacket::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;
Float64 overallscale = 1.0;
overallscale /= 44100.0;
overallscale *= GetSampleRate();
if (overallscale < 0.1) overallscale = 1.0;
//insanity check
long double fpOld = 0.618033988749894848204586; //golden ratio!
long double fpNew = 1.0 - fpOld;
long double inputSampleL;
long double inputSampleR;
long double drySampleL;
long double drySampleR;
Float64 bridgerectifier;
Float64 temprectifier;
Float64 inputSense;
Float64 inputGain = pow(10.0,(GetParameter( kParam_One ))/20.0);
Float64 attack = ((GetParameter( kParam_Two )+0.5)*0.006)/overallscale;
Float64 decay = ((GetParameter( kParam_Two )+0.01)*0.0004)/overallscale;
Float64 outputGain = pow(10.0,(GetParameter( kParam_Three ))/20.0);
Float64 wet;
Float64 maxblur;
Float64 blurdry;
Float64 out;
Float64 dry;
while (nSampleFrames-- > 0) {
inputSampleL = *inputL;
inputSampleR = *inputR;
if (inputSampleL<1.2e-38 && -inputSampleL<1.2e-38) {
static int noisesource = 0;
//this declares a variable before anything else is compiled. It won't keep assigning
//it to 0 for every sample, it's as if the declaration doesn't exist in this context,
//but it lets me add this denormalization fix in a single place rather than updating
//it in three different locations. The variable isn't thread-safe but this is only
//a random seed and we can share it with whatever.
noisesource = noisesource % 1700021; noisesource++;
int residue = noisesource * noisesource;
residue = residue % 170003; residue *= residue;
residue = residue % 17011; residue *= residue;
residue = residue % 1709; residue *= residue;
residue = residue % 173; residue *= residue;
residue = residue % 17;
double applyresidue = residue;
applyresidue *= 0.00000001;
applyresidue *= 0.00000001;
inputSampleL = applyresidue;
}
if (inputSampleR<1.2e-38 && -inputSampleR<1.2e-38) {
static int noisesource = 0;
noisesource = noisesource % 1700021; noisesource++;
int residue = noisesource * noisesource;
residue = residue % 170003; residue *= residue;
residue = residue % 17011; residue *= residue;
residue = residue % 1709; residue *= residue;
residue = residue % 173; residue *= residue;
residue = residue % 17;
double applyresidue = residue;
applyresidue *= 0.00000001;
applyresidue *= 0.00000001;
inputSampleR = applyresidue;
//this denormalization routine produces a white noise at -300 dB which the noise
//shaping will interact with to produce a bipolar output, but the noise is actually
//all positive. That should stop any variables from going denormal, and the routine
//only kicks in if digital black is input. As a final touch, if you save to 24-bit
//the silence will return to being digital black again.
}
if (inputGain != 1.0) {
inputSampleL *= inputGain;
inputSampleR *= inputGain;
}
drySampleL = inputSampleL;
drySampleR = inputSampleR;
inputSense = fabs(inputSampleL);
if (fabs(inputSampleR) > inputSense)
inputSense = fabs(inputSampleR);
//we will take the greater of either channel and just use that, then apply the result
//to both stereo channels.
if (chase < inputSense) chase += attack;
if (chase > 1.0) chase = 1.0;
if (chase > inputSense) chase -= decay;
if (chase < 0.0) chase = 0.0;
//chase will be between 0 and ? (if input is super hot)
out = wet = chase;
if (wet > 1.0) wet = 1.0;
maxblur = wet * fpNew;
blurdry = 1.0 - maxblur;
//scaled back so that blur remains balance of both
if (out > fpOld) out = fpOld - (out - fpOld);
if (out < 0.0) out = 0.0;
dry = 1.0 - wet;
if (inputSampleL > 1.57079633) inputSampleL = 1.57079633;
if (inputSampleL < -1.57079633) inputSampleL = -1.57079633;
if (inputSampleR > 1.57079633) inputSampleR = 1.57079633;
if (inputSampleR < -1.57079633) inputSampleR = -1.57079633;
bridgerectifier = fabs(inputSampleL);
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
temprectifier = 1-cos(bridgerectifier);
bridgerectifier = ((lastrectifierL*maxblur) + (temprectifier*blurdry));
lastrectifierL = temprectifier;
//starved version is also blurred by one sample
if (inputSampleL > 0) inputSampleL = (inputSampleL*dry)+(bridgerectifier*out);
else inputSampleL = (inputSampleL*dry)-(bridgerectifier*out);
bridgerectifier = fabs(inputSampleR);
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
temprectifier = 1-cos(bridgerectifier);
bridgerectifier = ((lastrectifierR*maxblur) + (temprectifier*blurdry));
lastrectifierR = temprectifier;
//starved version is also blurred by one sample
if (inputSampleR > 0) inputSampleR = (inputSampleR*dry)+(bridgerectifier*out);
else inputSampleR = (inputSampleR*dry)-(bridgerectifier*out);
if (outputGain != 1.0) {
inputSampleL *= outputGain;
inputSampleR *= outputGain;
}
//stereo 32 bit dither, made small and tidy.
int expon; frexpf((Float32)inputSampleL, &expon);
long double dither = (rand()/(RAND_MAX*7.737125245533627e+25))*pow(2,expon+62);
inputSampleL += (dither-fpNShapeL); fpNShapeL = dither;
frexpf((Float32)inputSampleR, &expon);
dither = (rand()/(RAND_MAX*7.737125245533627e+25))*pow(2,expon+62);
inputSampleR += (dither-fpNShapeR); fpNShapeR = dither;
//end 32 bit dither
*outputL = inputSampleL;
*outputR = inputSampleR;
inputL += 1;
inputR += 1;
outputL += 1;
outputR += 1;
}
return noErr;
}
