/*
* File: NCSeventeen.cpp
*
* Version: 1.0
*
* Created: 9/27/10
*
* Copyright: Copyright � 2010 Airwindows, All Rights Reserved
*
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/*=============================================================================
NCSeventeen.h
=============================================================================*/
#include "NCSeventeen.h"
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
COMPONENT_ENTRY(NCSeventeen)
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// NCSeventeen::NCSeventeen
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
NCSeventeen::NCSeventeen(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
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// NCSeventeen::GetParameterValueStrings
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult NCSeventeen::GetParameterValueStrings(AudioUnitScope inScope,
AudioUnitParameterID inParameterID,
CFArrayRef * outStrings)
{
return kAudioUnitErr_InvalidProperty;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// NCSeventeen::GetParameterInfo
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult NCSeventeen::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_Decibels;
outParameterInfo.minValue = 0.0;
outParameterInfo.maxValue = 24.0;
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;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// NCSeventeen::GetPropertyInfo
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult NCSeventeen::GetPropertyInfo (AudioUnitPropertyID inID,
AudioUnitScope inScope,
AudioUnitElement inElement,
UInt32 & outDataSize,
Boolean & outWritable)
{
return AUEffectBase::GetPropertyInfo (inID, inScope, inElement, outDataSize, outWritable);
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// NCSeventeen::GetProperty
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult NCSeventeen::GetProperty( AudioUnitPropertyID inID,
AudioUnitScope inScope,
AudioUnitElement inElement,
void * outData )
{
return AUEffectBase::GetProperty (inID, inScope, inElement, outData);
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// NCSeventeen::Initialize
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult NCSeventeen::Initialize()
{
ComponentResult result = AUEffectBase::Initialize();
if (result == noErr)
Reset(kAudioUnitScope_Global, 0);
return result;
}
#pragma mark ____NCSeventeenEffectKernel
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// NCSeventeen::NCSeventeenKernel::Reset()
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
void NCSeventeen::NCSeventeenKernel::Reset()
{
lastSample = 0.0;
iirSampleA = 0.0;
iirSampleB = 0.0;
flip = false;
basslev = 0.0;
treblev = 0.0;
cheblev = 0.0;
fpNShape = 0.0;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// NCSeventeen::NCSeventeenKernel::Process
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
void NCSeventeen::NCSeventeenKernel::Process( const Float32 *inSourceP,
Float32 *inDestP,
UInt32 inFramesToProcess,
UInt32 inNumChannels,
bool &ioSilence )
{
UInt32 nSampleFrames = inFramesToProcess;
const Float32 *sourceP = inSourceP;
Float32 *destP = inDestP;
Float64 inP2;
Float64 chebyshev;
Float64 overallscale = 1.0;
overallscale /= 44100.0;
overallscale *= GetSampleRate();
Float64 IIRscaleback = 0.0004716;
Float64 bassScaleback = 0.0002364;
Float64 trebleScaleback = 0.0005484;
Float64 addBassBuss = 0.000243;
Float64 addTrebBuss = 0.000407;
Float64 addShortBuss = 0.000326;
IIRscaleback /= overallscale;
bassScaleback /= overallscale;
trebleScaleback /= overallscale;
addBassBuss /= overallscale;
addTrebBuss /= overallscale;
addShortBuss /= overallscale;
Float64 limitingBass = 0.39;
Float64 limitingTreb = 0.6;
Float64 limiting = 0.36;
Float64 maxfeedBass = 0.972;
Float64 maxfeedTreb = 0.972;
Float64 maxfeed = 0.975;
Float64 bridgerectifier;
long double inputSample;
Float64 lowSample;
Float64 highSample;
Float64 distSample;
Float64 minusSample;
Float64 plusSample;
Float64 gain = pow(10.0,GetParameter( kParam_One )/20);
Float64 outgain = GetParameter( kParam_Two );
while (nSampleFrames-- > 0) {
inputSample = *sourceP;
if (inputSample<1.2e-38 && -inputSample<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;
inputSample = 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.
}
inputSample *= gain;
if (flip)
{
iirSampleA = (iirSampleA * 0.9) + (inputSample * 0.1);
lowSample = iirSampleA;
}
else
{
iirSampleB = (iirSampleB * 0.9) + (inputSample * 0.1);
lowSample = iirSampleB;
}
highSample = inputSample - lowSample;
flip = !flip;
//we now have two bands and the original source
//inputSample = lowSample;
inP2 = lowSample * lowSample;
if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0;
chebyshev = (2 * inP2);
chebyshev *= basslev;
//second harmonic max +1
if (basslev > 0) basslev -= bassScaleback;
if (basslev < 0) basslev += bassScaleback;
//this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on
bridgerectifier = fabs(lowSample);
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
//max value for sine function
bridgerectifier = sin(bridgerectifier);
if (lowSample > 0.0) distSample = bridgerectifier;
else distSample = -bridgerectifier;
minusSample = lowSample - distSample;
plusSample = lowSample + distSample;
if (minusSample > maxfeedBass) minusSample = maxfeedBass;
if (plusSample > maxfeedBass) plusSample = maxfeedBass;
if (plusSample < -maxfeedBass) plusSample = -maxfeedBass;
if (minusSample < -maxfeedBass) minusSample = -maxfeedBass;
if (lowSample > distSample) basslev += (minusSample*addBassBuss);
if (lowSample < -distSample) basslev -= (plusSample*addBassBuss);
if (basslev > 1.0) basslev = 1.0;
if (basslev < -1.0) basslev = -1.0;
bridgerectifier = fabs(lowSample);
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
//max value for sine function
bridgerectifier = sin(bridgerectifier);
if (lowSample > 0.0) lowSample = bridgerectifier;
else lowSample = -bridgerectifier;
//apply the distortion transform for reals
lowSample /= (1.0+fabs(basslev*limitingBass));
lowSample += chebyshev;
//apply the correction measures
//inputSample = highSample;
inP2 = highSample * highSample;
if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0;
chebyshev = (2 * inP2);
chebyshev *= treblev;
//second harmonic max +1
if (treblev > 0) treblev -= trebleScaleback;
if (treblev < 0) treblev += trebleScaleback;
//this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on
bridgerectifier = fabs(highSample);
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
//max value for sine function
bridgerectifier = sin(bridgerectifier);
if (highSample > 0.0) distSample = bridgerectifier;
else distSample = -bridgerectifier;
minusSample = highSample - distSample;
plusSample = highSample + distSample;
if (minusSample > maxfeedTreb) minusSample = maxfeedTreb;
if (plusSample > maxfeedTreb) plusSample = maxfeedTreb;
if (plusSample < -maxfeedTreb) plusSample = -maxfeedTreb;
if (minusSample < -maxfeedTreb) minusSample = -maxfeedTreb;
if (highSample > distSample) treblev += (minusSample*addTrebBuss);
if (highSample < -distSample) treblev -= (plusSample*addTrebBuss);
if (treblev > 1.0) treblev = 1.0;
if (treblev < -1.0) treblev = -1.0;
bridgerectifier = fabs(highSample);
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
//max value for sine function
bridgerectifier = sin(bridgerectifier);
if (highSample > 0.0) highSample = bridgerectifier;
else highSample = -bridgerectifier;
//apply the distortion transform for reals
highSample /= (1.0+fabs(treblev*limitingTreb));
highSample += chebyshev;
//apply the correction measures
inputSample = lowSample + highSample;
inP2 = inputSample * inputSample;
if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0;
chebyshev = (2 * inP2);
chebyshev *= cheblev;
//third harmonic max -1
if (cheblev > 0) cheblev -= IIRscaleback;
if (cheblev < 0) cheblev += IIRscaleback;
//this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on
bridgerectifier = fabs(inputSample);
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
//max value for sine function
bridgerectifier = sin(bridgerectifier);
if (inputSample > 0.0) distSample = bridgerectifier;
else distSample = -bridgerectifier;
minusSample = inputSample - distSample;
plusSample = inputSample + distSample;
if (minusSample > maxfeed) minusSample = maxfeed;
if (plusSample > maxfeed) plusSample = maxfeed;
if (plusSample < -maxfeed) plusSample = -maxfeed;
if (minusSample < -maxfeed) minusSample = -maxfeed;
if (inputSample > distSample) cheblev += (minusSample*addShortBuss);
if (inputSample < -distSample) cheblev -= (plusSample*addShortBuss);
if (cheblev > 1.0) cheblev = 1.0;
if (cheblev < -1.0) cheblev = -1.0;
bridgerectifier = fabs(inputSample);
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
//max value for sine function
bridgerectifier = sin(bridgerectifier);
if (inputSample > 0.0) inputSample = bridgerectifier;
else inputSample = -bridgerectifier;
//apply the distortion transform for reals
inputSample /= (1.0+fabs(cheblev*limiting));
inputSample += chebyshev;
//apply the correction measures
if (outgain < 1.0) {
inputSample *= outgain;
}
if (inputSample > 0.95) inputSample = 0.95;
if (inputSample < -0.95) inputSample = -0.95;
//iron bar
//32 bit dither, made small and tidy.
int expon; frexpf((Float32)inputSample, &expon);
long double dither = (rand()/(RAND_MAX*7.737125245533627e+25))*pow(2,expon+62);
inputSample += (dither-fpNShape); fpNShape = dither;
//end 32 bit dither
*destP = inputSample;
//built in output trim and dry/wet by default
sourceP += inNumChannels; destP += inNumChannels;
}
}
