/*
* File: ChorusEnsemble.cpp
*
* Version: 1.0
*
* Created: 2/12/07
*
* Copyright: Copyright � 2007 Airwindows, All Rights Reserved
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/*=============================================================================
ChorusEnsemble.h
=============================================================================*/
#include "ChorusEnsemble.h"
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
COMPONENT_ENTRY(ChorusEnsemble)
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// ChorusEnsemble::ChorusEnsemble
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ChorusEnsemble::ChorusEnsemble(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
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// ChorusEnsemble::GetParameterValueStrings
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult ChorusEnsemble::GetParameterValueStrings(AudioUnitScope inScope,
AudioUnitParameterID inParameterID,
CFArrayRef * outStrings)
{
return kAudioUnitErr_InvalidProperty;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// ChorusEnsemble::GetParameterInfo
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult ChorusEnsemble::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;
case kParam_Three:
AUBase::FillInParameterName (outParameterInfo, kParameterThreeName, false);
outParameterInfo.unit = kAudioUnitParameterUnit_Generic;
outParameterInfo.minValue = 0.0;
outParameterInfo.maxValue = 1.0;
outParameterInfo.defaultValue = kDefaultValue_ParamThree;
break;
default:
result = kAudioUnitErr_InvalidParameter;
break;
}
} else {
result = kAudioUnitErr_InvalidParameter;
}
return result;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// ChorusEnsemble::GetPropertyInfo
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult ChorusEnsemble::GetPropertyInfo (AudioUnitPropertyID inID,
AudioUnitScope inScope,
AudioUnitElement inElement,
UInt32 & outDataSize,
Boolean & outWritable)
{
return AUEffectBase::GetPropertyInfo (inID, inScope, inElement, outDataSize, outWritable);
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// ChorusEnsemble::GetProperty
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult ChorusEnsemble::GetProperty( AudioUnitPropertyID inID,
AudioUnitScope inScope,
AudioUnitElement inElement,
void * outData )
{
return AUEffectBase::GetProperty (inID, inScope, inElement, outData);
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// ChorusEnsemble::Initialize
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult ChorusEnsemble::Initialize()
{
ComponentResult result = AUEffectBase::Initialize();
if (result == noErr)
Reset(kAudioUnitScope_Global, 0);
return result;
}
#pragma mark ____ChorusEnsembleEffectKernel
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// ChorusEnsemble::ChorusEnsembleKernel::Reset()
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
void ChorusEnsemble::ChorusEnsembleKernel::Reset()
{
for(int count = 0; count < totalsamples-1; count++) {d[count] = 0;}
sweep = 3.141592653589793238 / 2.0;
gcount = 0;
airPrev = 0.0;
airEven = 0.0;
airOdd = 0.0;
airFactor = 0.0;
fpNShape = 0.0;
fpFlip = false;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// ChorusEnsemble::ChorusEnsembleKernel::Process
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
void ChorusEnsemble::ChorusEnsembleKernel::Process( const Float32 *inSourceP,
Float32 *inDestP,
UInt32 inFramesToProcess,
UInt32 inNumChannels, // for version 2 AudioUnits inNumChannels is always 1
bool &ioSilence )
{
UInt32 nSampleFrames = inFramesToProcess;
const Float32 *sourceP = inSourceP;
Float32 *destP = inDestP;
Float64 overallscale = 1.0;
overallscale /= 44100.0;
overallscale *= GetSampleRate();
Float64 speed = pow(GetParameter( kParam_One ),3) * 0.001;
speed *= overallscale;
int loopLimit = (int)(totalsamples * 0.499);
int count;
Float64 range = pow(GetParameter( kParam_Two ),3) * loopLimit * 0.12;
Float64 wet = GetParameter( kParam_Three );
Float64 modulation = range*wet;
Float64 dry = 1.0 - wet;
Float64 tupi = 3.141592653589793238 * 2.0;
Float64 offset;
Float64 start[4];
long double inputSample;
Float64 drySample;
//now we'll precalculate some stuff that needn't be in every sample
start[0] = range;
start[1] = range * 2;
start[2] = range * 3;
start[3] = range * 4;
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.
}
drySample = inputSample;
airFactor = airPrev - inputSample;
if (fpFlip) {airEven += airFactor; airOdd -= airFactor; airFactor = airEven;}
else {airOdd += airFactor; airEven -= airFactor; airFactor = airOdd;}
airOdd = (airOdd - ((airOdd - airEven)/256.0)) / 1.0001;
airEven = (airEven - ((airEven - airOdd)/256.0)) / 1.0001;
airPrev = inputSample;
inputSample += (airFactor * wet * 0.64);
//air, compensates for loss of highs in flanger's interpolation
if (gcount < 1 || gcount > loopLimit) {gcount = loopLimit;}
count = gcount;
d[count+loopLimit] = d[count] = inputSample;
//double buffer, inverted so we get some cancellation at small range values
offset = start[0] + (modulation * sin(sweep));
count = gcount + (int)floor(offset);
inputSample = d[count] * (1-(offset-floor(offset))); //less as value moves away from .0
inputSample += d[count+1]; //we can assume always using this in one way or another?
inputSample += (d[count+2] * (offset-floor(offset))); //greater as value moves away from .0
inputSample -= (((d[count]-d[count+1])-(d[count+1]-d[count+2]))/50); //interpolation hacks 'r us
offset = start[1] + (modulation * sin(sweep + 1.0));
count = gcount + (int)floor(offset);
inputSample += d[count] * (1-(offset-floor(offset))); //less as value moves away from .0
inputSample += d[count+1]; //we can assume always using this in one way or another?
inputSample += (d[count+2] * (offset-floor(offset))); //greater as value moves away from .0
inputSample -= (((d[count]-d[count+1])-(d[count+1]-d[count+2]))/50); //interpolation hacks 'r us
offset = start[2] + (modulation * sin(sweep + 2.0));
count = gcount + (int)floor(offset);
inputSample += d[count] * (1-(offset-floor(offset))); //less as value moves away from .0
inputSample += d[count+1]; //we can assume always using this in one way or another?
inputSample += (d[count+2] * (offset-floor(offset))); //greater as value moves away from .0
inputSample -= (((d[count]-d[count+1])-(d[count+1]-d[count+2]))/50); //interpolation hacks 'r us
offset = start[3] + (modulation * sin(sweep + 3.0));
count = gcount + (int)floor(offset);
inputSample += d[count] * (1-(offset-floor(offset))); //less as value moves away from .0
inputSample += d[count+1]; //we can assume always using this in one way or another?
inputSample += (d[count+2] * (offset-floor(offset))); //greater as value moves away from .0
inputSample -= (((d[count]-d[count+1])-(d[count+1]-d[count+2]))/50); //interpolation hacks 'r us
inputSample *= 0.125; //to get a comparable level
gcount--;
//sliding
sweep += speed;
if (sweep > tupi){sweep -= tupi;}
//still scrolling through the samples, remember
if (wet != 1.0) {
inputSample = (inputSample * wet) + (drySample * dry);
}
//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;
destP += inNumChannels;
sourceP += inNumChannels;
}
}
