/*
* File: StereoFX.cpp
*
* Version: 1.0
*
* Created: 5/21/07
*
* Copyright: Copyright � 2007 Airwindows, All Rights Reserved
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/*=============================================================================
StereoFX.h
=============================================================================*/
#include "StereoFX.h"
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
COMPONENT_ENTRY(StereoFX)
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// StereoFX::StereoFX
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
StereoFX::StereoFX(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
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// StereoFX::GetParameterValueStrings
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult StereoFX::GetParameterValueStrings(AudioUnitScope inScope,
AudioUnitParameterID inParameterID,
CFArrayRef * outStrings)
{
return kAudioUnitErr_InvalidProperty;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// StereoFX::GetParameterInfo
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult StereoFX::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;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// StereoFX::GetPropertyInfo
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult StereoFX::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 StereoFX::SupportedNumChannels(const AUChannelInfo ** outInfo)
{
if (outInfo != NULL)
{
static AUChannelInfo info;
info.inChannels = 2;
info.outChannels = 2;
*outInfo = &info;
}
return 1;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// StereoFX::GetProperty
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult StereoFX::GetProperty( AudioUnitPropertyID inID,
AudioUnitScope inScope,
AudioUnitElement inElement,
void * outData )
{
return AUEffectBase::GetProperty (inID, inScope, inElement, outData);
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// StereoFX::Initialize
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult StereoFX::Initialize()
{
ComponentResult result = AUEffectBase::Initialize();
if (result == noErr)
Reset(kAudioUnitScope_Global, 0);
return result;
}
#pragma mark ____StereoFXEffectKernel
//-----------------------------------------------------------------------------------------
// this is called the reset the DSP state (clear buffers, reset counters, etc.)
ComponentResult StereoFX::Reset(AudioUnitScope inScope, AudioUnitElement inElement)
{
iirSampleA = 0.0;
iirSampleB = 0.0;
flip = false;
fpNShapeL = 0.0;
fpNShapeR = 0.0;
return noErr;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// StereoFX::ProcessBufferLists
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
OSStatus StereoFX::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();
long double inputSampleL;
long double inputSampleR;
long double mid;
long double side;
//High Impact section
Float64 stereowide = GetParameter( kParam_One );
Float64 centersquish = GetParameter( kParam_Three );
Float64 density = stereowide * 2.4;
Float64 sustain = 1.0 - (1.0/(1.0 + (density/7.0)));
//this way, enhance increases up to 50% and then mid falls off beyond that
Float64 bridgerectifier;
Float64 count;
//Highpass section
Float64 iirAmount = pow(GetParameter( kParam_Two ),3)/overallscale;
Float64 tight = -0.33333333333333;
Float64 offset;
//we are setting it up so that to either extreme we can get an audible sound,
//but sort of scaled so small adjustments don't shift the cutoff frequency yet.
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.
}
//assign working variables
mid = inputSampleL + inputSampleR;
side = inputSampleL - inputSampleR;
//assign mid and side. Now, High Impact code
count = density;
while (count > 1.0)
{
bridgerectifier = fabs(side)*1.57079633;
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
//max value for sine function
bridgerectifier = sin(bridgerectifier);
if (side > 0.0) side = bridgerectifier;
else side = -bridgerectifier;
count = count - 1.0;
}
//we have now accounted for any really high density settings.
bridgerectifier = fabs(side)*1.57079633;
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
//max value for sine function
bridgerectifier = sin(bridgerectifier);
if (side > 0) side = (side*(1-count))+(bridgerectifier*count);
else side = (side*(1-count))-(bridgerectifier*count);
//blend according to density control
//done first density. Next, sustain-reducer
bridgerectifier = fabs(side)*1.57079633;
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
bridgerectifier = (1-cos(bridgerectifier))*3.141592653589793;
if (side > 0) side = (side*(1-sustain))+(bridgerectifier*sustain);
else side = (side*(1-sustain))-(bridgerectifier*sustain);
//done with High Impact code
//now, Highpass code
offset = 0.666666666666666 + ((1-fabs(side))*tight);
if (offset < 0) offset = 0;
if (offset > 1) offset = 1;
if (flip)
{
iirSampleA = (iirSampleA * (1 - (offset * iirAmount))) + (side * (offset * iirAmount));
side = side - iirSampleA;
}
else
{
iirSampleB = (iirSampleB * (1 - (offset * iirAmount))) + (side * (offset * iirAmount));
side = side - iirSampleB;
}
//done with Highpass code
bridgerectifier = fabs(mid)/1.273239544735162;
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
bridgerectifier = sin(bridgerectifier)*1.273239544735162;
if (mid > 0) mid = (mid*(1-centersquish))+(bridgerectifier*centersquish);
else mid = (mid*(1-centersquish))-(bridgerectifier*centersquish);
//done with the mid saturating section.
inputSampleL = (mid+side)/2.0;
inputSampleR = (mid-side)/2.0;
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;
}
