/*
* File: Thunder.cpp
*
* Version: 1.0
*
* Created: 9/19/16
*
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/*=============================================================================
Thunder.cpp
=============================================================================*/
#include "Thunder.h"
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
COMPONENT_ENTRY(Thunder)
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// Thunder::Thunder
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
Thunder::Thunder(AudioUnit component)
: AUEffectBase(component)
{
CreateElements();
Globals()->UseIndexedParameters(kNumberOfParameters);
SetParameter(kParam_One, kDefaultValue_ParamOne );
SetParameter(kParam_Two, kDefaultValue_ParamTwo );
#if AU_DEBUG_DISPATCHER
mDebugDispatcher = new AUDebugDispatcher (this);
#endif
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// Thunder::GetParameterValueStrings
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult Thunder::GetParameterValueStrings(AudioUnitScope inScope,
AudioUnitParameterID inParameterID,
CFArrayRef * outStrings)
{
return kAudioUnitErr_InvalidProperty;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// Thunder::GetParameterInfo
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult Thunder::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;
default:
result = kAudioUnitErr_InvalidParameter;
break;
}
} else {
result = kAudioUnitErr_InvalidParameter;
}
return result;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// Thunder::GetPropertyInfo
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult Thunder::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 Thunder::SupportedNumChannels(const AUChannelInfo ** outInfo)
{
if (outInfo != NULL)
{
static AUChannelInfo info;
info.inChannels = 2;
info.outChannels = 2;
*outInfo = &info;
}
return 1;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// Thunder::GetProperty
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult Thunder::GetProperty( AudioUnitPropertyID inID,
AudioUnitScope inScope,
AudioUnitElement inElement,
void * outData )
{
return AUEffectBase::GetProperty (inID, inScope, inElement, outData);
}
// Thunder::Initialize
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult Thunder::Initialize()
{
ComponentResult result = AUEffectBase::Initialize();
if (result == noErr)
Reset(kAudioUnitScope_Global, 0);
return result;
}
#pragma mark ____ThunderEffectKernel
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// Thunder::ThunderKernel::Reset()
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult Thunder::Reset(AudioUnitScope inScope, AudioUnitElement inElement)
{
fpNShapeL = 0.0;
fpNShapeR = 0.0;
muSpeedA = 10000;
muSpeedB = 10000;
muCoefficientA = 1;
muCoefficientB = 1;
muVary = 1;
gateL = 0.0;
gateR = 0.0;
iirSampleAL = 0.0;
iirSampleBL = 0.0;
iirSampleAR = 0.0;
iirSampleBR = 0.0;
iirSampleAM = 0.0;
iirSampleBM = 0.0;
iirSampleCM = 0.0;
flip = false;
return noErr;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// Thunder::ProcessBufferLists
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
OSStatus Thunder::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();
Float64 thunder = GetParameter( kParam_One ) * 0.4;
Float64 threshold = 1.0 - (thunder * 2.0);
if (threshold < 0.01) threshold = 0.01;
Float64 muMakeupGain = 1.0 / threshold;
Float64 release = pow((1.28-thunder),5)*32768.0;
release /= overallscale;
Float64 fastest = sqrt(release);
Float64 EQ = ((0.0275 / GetSampleRate())*32000.0);
Float64 dcblock = EQ / 300.0;
Float64 basstrim = (0.01/EQ)+1.0;
//FF parameters also ride off Speed
Float64 outputGain = GetParameter( kParam_Two );
Float64 coefficient;
Float64 inputSense;
Float64 resultL;
Float64 resultR;
Float64 resultM;
Float64 resultML;
Float64 resultMR;
long double inputSampleL;
long double inputSampleR;
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.
}
inputSampleL = inputSampleL * muMakeupGain;
inputSampleR = inputSampleR * muMakeupGain;
if (gateL < fabs(inputSampleL)) gateL = inputSampleL;
else gateL -= dcblock;
if (gateR < fabs(inputSampleR)) gateR = inputSampleR;
else gateR -= dcblock;
//setting up gated DC blocking to control the tendency for rumble and offset
//begin three FathomFive stages
iirSampleAL += (inputSampleL * EQ * thunder);
iirSampleAL -= (iirSampleAL * iirSampleAL * iirSampleAL * EQ);
if (iirSampleAL > gateL) iirSampleAL -= dcblock;
if (iirSampleAL < -gateL) iirSampleAL += dcblock;
resultL = iirSampleAL*basstrim;
iirSampleBL = (iirSampleBL * (1 - EQ)) + (resultL * EQ);
resultL = iirSampleBL;
iirSampleAR += (inputSampleR * EQ * thunder);
iirSampleAR -= (iirSampleAR * iirSampleAR * iirSampleAR * EQ);
if (iirSampleAR > gateR) iirSampleAR -= dcblock;
if (iirSampleAR < -gateR) iirSampleAR += dcblock;
resultR = iirSampleAR*basstrim;
iirSampleBR = (iirSampleBR * (1 - EQ)) + (resultR * EQ);
resultR = iirSampleBR;
iirSampleAM += ((inputSampleL + inputSampleR) * EQ * thunder);
iirSampleAM -= (iirSampleAM * iirSampleAM * iirSampleAM * EQ);
resultM = iirSampleAM*basstrim;
iirSampleBM = (iirSampleBM * (1 - EQ)) + (resultM * EQ);
resultM = iirSampleBM;
iirSampleCM = (iirSampleCM * (1 - EQ)) + (resultM * EQ);
resultM = fabs(iirSampleCM);
resultML = fabs(resultL);
resultMR = fabs(resultR);
if (resultM > resultML) resultML = resultM;
if (resultM > resultMR) resultMR = resultM;
//trying to restrict the buzziness
if (resultML > 1.0) resultML = 1.0;
if (resultMR > 1.0) resultMR = 1.0;
//now we have result L, R and M the trigger modulator which must be 0-1
//begin compressor section
inputSampleL -= (iirSampleBL * thunder);
inputSampleR -= (iirSampleBR * thunder);
//highpass the comp section by sneaking out what will be the reinforcement
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 (flip)
{
if (inputSense > threshold)
{
muVary = threshold / inputSense;
muAttack = sqrt(fabs(muSpeedA));
muCoefficientA = muCoefficientA * (muAttack-1.0);
if (muVary < threshold)
{
muCoefficientA = muCoefficientA + threshold;
}
else
{
muCoefficientA = muCoefficientA + muVary;
}
muCoefficientA = muCoefficientA / muAttack;
}
else
{
muCoefficientA = muCoefficientA * ((muSpeedA * muSpeedA)-1.0);
muCoefficientA = muCoefficientA + 1.0;
muCoefficientA = muCoefficientA / (muSpeedA * muSpeedA);
}
muNewSpeed = muSpeedA * (muSpeedA-1);
muNewSpeed = muNewSpeed + fabs(inputSense*release)+fastest;
muSpeedA = muNewSpeed / muSpeedA;
}
else
{
if (inputSense > threshold)
{
muVary = threshold / inputSense;
muAttack = sqrt(fabs(muSpeedB));
muCoefficientB = muCoefficientB * (muAttack-1);
if (muVary < threshold)
{
muCoefficientB = muCoefficientB + threshold;
}
else
{
muCoefficientB = muCoefficientB + muVary;
}
muCoefficientB = muCoefficientB / muAttack;
}
else
{
muCoefficientB = muCoefficientB * ((muSpeedB * muSpeedB)-1.0);
muCoefficientB = muCoefficientB + 1.0;
muCoefficientB = muCoefficientB / (muSpeedB * muSpeedB);
}
muNewSpeed = muSpeedB * (muSpeedB-1);
muNewSpeed = muNewSpeed + fabs(inputSense*release)+fastest;
muSpeedB = muNewSpeed / muSpeedB;
}
//got coefficients, adjusted speeds
if (flip)
{
coefficient = pow(muCoefficientA,2);
inputSampleL *= coefficient;
inputSampleR *= coefficient;
}
else
{
coefficient = pow(muCoefficientB,2);
inputSampleL *= coefficient;
inputSampleR *= coefficient;
}
//applied compression with vari-vari-�-�-�-�-�-�-is-the-kitten-song o/~
//applied gain correction to control output level- tends to constrain sound rather than inflate it
inputSampleL += (resultL * resultM);
inputSampleR += (resultR * resultM);
//combine the two by adding the summed channnel of lows
if (outputGain != 1.0) {
inputSampleL *= outputGain;
inputSampleR *= outputGain;
}
flip = !flip;
//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;
//don't know why we're getting a volume boost, cursed thing
inputL += 1;
inputR += 1;
outputL += 1;
outputR += 1;
}
return noErr;
}
