/*
* File: AQuickVoiceClip.cpp
*
* Version: 1.0
*
* Created: 10/3/13
*
* Copyright: Copyright � 2013 Airwindows, All Rights Reserved
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/*=============================================================================
AQuickVoiceClip.cpp
=============================================================================*/
#include "AQuickVoiceClip.h"
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
COMPONENT_ENTRY(AQuickVoiceClip)
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// AQuickVoiceClip::AQuickVoiceClip
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
AQuickVoiceClip::AQuickVoiceClip(AudioUnit component)
: AUEffectBase(component)
{
CreateElements();
Globals()->UseIndexedParameters(kNumberOfParameters);
SetParameter(kParam_One, kDefaultValue_ParamOne );
#if AU_DEBUG_DISPATCHER
mDebugDispatcher = new AUDebugDispatcher (this);
#endif
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// AQuickVoiceClip::GetParameterValueStrings
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult AQuickVoiceClip::GetParameterValueStrings(AudioUnitScope inScope,
AudioUnitParameterID inParameterID,
CFArrayRef * outStrings)
{
return kAudioUnitErr_InvalidProperty;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// AQuickVoiceClip::GetParameterInfo
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult AQuickVoiceClip::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 = 30.0;
outParameterInfo.maxValue = 3000.0;
outParameterInfo.defaultValue = kDefaultValue_ParamOne;
break;
default:
result = kAudioUnitErr_InvalidParameter;
break;
}
} else {
result = kAudioUnitErr_InvalidParameter;
}
return result;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// AQuickVoiceClip::GetPropertyInfo
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult AQuickVoiceClip::GetPropertyInfo (AudioUnitPropertyID inID,
AudioUnitScope inScope,
AudioUnitElement inElement,
UInt32 & outDataSize,
Boolean & outWritable)
{
return AUEffectBase::GetPropertyInfo (inID, inScope, inElement, outDataSize, outWritable);
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// AQuickVoiceClip::GetProperty
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult AQuickVoiceClip::GetProperty( AudioUnitPropertyID inID,
AudioUnitScope inScope,
AudioUnitElement inElement,
void * outData )
{
return AUEffectBase::GetProperty (inID, inScope, inElement, outData);
}
// AQuickVoiceClip::Initialize
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult AQuickVoiceClip::Initialize()
{
ComponentResult result = AUEffectBase::Initialize();
if (result == noErr)
Reset(kAudioUnitScope_Global, 0);
return result;
}
#pragma mark ____AQuickVoiceClipEffectKernel
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// AQuickVoiceClip::AQuickVoiceClipKernel::Reset()
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
void AQuickVoiceClip::AQuickVoiceClipKernel::Reset()
{
ataLast6Sample = ataLast5Sample = ataLast4Sample = ataLast3Sample = ataLast2Sample = ataLast1Sample = 0.0;
ataHalfwaySample = ataHalfDrySample = ataHalfDiffSample = 0.0;
ataLastDiffSample = ataDrySample = ataDiffSample = ataPrevDiffSample = 0.0;
ataK1 = -0.646; //first FIR shaping of interpolated sample, brightens
ataK2 = 0.311; //second FIR shaping of interpolated sample, thickens
ataK6 = -0.093; //third FIR shaping of interpolated sample, brings air
ataK7 = 0.057; //fourth FIR shaping of interpolated sample, thickens
ataK8 = -0.023; //fifth FIR shaping of interpolated sample, brings air
ataK3 = 0.114; //add raw to interpolated dry, toughens
ataK4 = 0.886; //remainder of interpolated dry, adds up to 1.0
ataK5 = 0.431; //subtract this much prev. diff sample, brightens. 0.431 becomes flat
lastSample = 0.0;
lastOutSample = 0.0;
lastOut2Sample = 0.0;
lastOut3Sample = 0.0;
lpDepth = 0.0;
overshoot = 0.0;
overall = 0;
iirSampleA = 0.0;
iirSampleB = 0.0;
iirSampleC = 0.0;
iirSampleD = 0.0;
flip = false;
fpNShape = 0.0;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// AQuickVoiceClip::AQuickVoiceClipKernel::Process
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
void AQuickVoiceClip::AQuickVoiceClipKernel::Process( const Float32 *inSourceP,
Float32 *inDestP,
UInt32 inFramesToProcess,
UInt32 inNumChannels,
bool &ioSilence )
{
UInt32 nSampleFrames = inFramesToProcess;
const Float32 *sourceP = inSourceP;
Float32 *destP = inDestP;
Float64 overallscale = 1.0;
overallscale /= 44100.0;
overallscale *= GetSampleRate();
Float64 softness = 0.484416;
Float64 hardness = 1.0 - softness;
Float64 iirAmount = GetParameter( kParam_One )/8000.0;
iirAmount /= overallscale;
Float64 altAmount = (1.0 - iirAmount);
Float64 cancelnew = 0.0682276;
Float64 cancelold = 1.0 - cancelnew;
Float64 maxRecent;
Float64 lpSpeed = 0.0009;
Float64 cliplevel = 0.98;
Float64 refclip = 0.5; //preset to cut out gain quite a lot. 91%? no touchy unless clip
Float64 inputSample;
Float64 passThrough;
Float64 outputSample;
bool clipOnset;
Float64 drySample;
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.
}
passThrough = ataDrySample = inputSample;
ataHalfDrySample = ataHalfwaySample = (inputSample + ataLast1Sample + (ataLast2Sample*ataK1) + (ataLast3Sample*ataK2) + (ataLast4Sample*ataK6) + (ataLast5Sample*ataK7) + (ataLast6Sample*ataK8)) / 2.0;
ataLast6Sample = ataLast5Sample; ataLast5Sample = ataLast4Sample; ataLast4Sample = ataLast3Sample; ataLast3Sample = ataLast2Sample; ataLast2Sample = ataLast1Sample; ataLast1Sample = inputSample;
//setting up oversampled special antialiasing
clipOnset = false;
maxRecent = fabs( ataLast6Sample );
if (fabs( ataLast5Sample ) > maxRecent ) maxRecent = fabs( ataLast5Sample );
if (fabs( ataLast4Sample ) > maxRecent ) maxRecent = fabs( ataLast4Sample );
if (fabs( ataLast3Sample ) > maxRecent ) maxRecent = fabs( ataLast3Sample );
if (fabs( ataLast2Sample ) > maxRecent ) maxRecent = fabs( ataLast2Sample );
if (fabs( ataLast1Sample ) > maxRecent ) maxRecent = fabs( ataLast1Sample );
if (fabs( inputSample ) > maxRecent ) maxRecent = fabs( inputSample );
//this gives us something that won't cut out in zero crossings, to interpolate with
maxRecent *= 2.0;
//by refclip this is 1.0 and fully into the antialiasing
if (maxRecent > 1.0) maxRecent = 1.0;
//and it tops out at 1. Higher means more antialiasing, lower blends into passThrough without antialiasing
ataHalfwaySample -= overall;
//subtract dist-cancel from input after getting raw input, before doing anything
drySample = ataHalfwaySample;
if (lastSample >= refclip)
{
lpDepth += 0.1;
if (ataHalfwaySample < refclip)
{
lastSample = ((refclip*hardness) + (ataHalfwaySample * softness));
}
else lastSample = refclip;
}
if (lastSample <= -refclip)
{
lpDepth += 0.1;
if (ataHalfwaySample > -refclip)
{
lastSample = ((-refclip*hardness) + (ataHalfwaySample * softness));
}
else lastSample = -refclip;
}
if (ataHalfwaySample > refclip)
{
lpDepth += 0.1;
if (lastSample < refclip)
{
ataHalfwaySample = ((refclip*hardness) + (lastSample * softness));
}
else ataHalfwaySample = refclip;
}
if (ataHalfwaySample < -refclip)
{
lpDepth += 0.1;
if (lastSample > -refclip)
{
ataHalfwaySample = ((-refclip*hardness) + (lastSample * softness));
}
else ataHalfwaySample = -refclip;
}
outputSample = lastSample;
lastSample = ataHalfwaySample;
ataHalfwaySample = outputSample;
//swap around in a circle for one final ADClip,
//this time not tracking overshoot anymore
//end interpolated sample
//begin raw sample- inputSample and ataDrySample handled separately here
inputSample -= overall;
//subtract dist-cancel from input after getting raw input, before doing anything
drySample = inputSample;
if (lastSample >= refclip)
{
lpDepth += 0.1;
if (inputSample < refclip)
{
lastSample = ((refclip*hardness) + (inputSample * softness));
}
else lastSample = refclip;
}
if (lastSample <= -refclip)
{
lpDepth += 0.1;
if (inputSample > -refclip)
{
lastSample = ((-refclip*hardness) + (inputSample * softness));
}
else lastSample = -refclip;
}
if (inputSample > refclip)
{
lpDepth += 0.1;
if (lastSample < refclip)
{
inputSample = ((refclip*hardness) + (lastSample * softness));
}
else inputSample = refclip;
}
if (inputSample < -refclip)
{
lpDepth += 0.1;
if (lastSample > -refclip)
{
inputSample = ((-refclip*hardness) + (lastSample * softness));
}
else inputSample = -refclip;
}
outputSample = lastSample;
lastSample = inputSample;
inputSample = outputSample;
//end raw sample
ataHalfDrySample = (ataDrySample*ataK3)+(ataHalfDrySample*ataK4);
ataHalfDiffSample = (ataHalfwaySample - ataHalfDrySample)/2.0;
ataLastDiffSample = ataDiffSample*ataK5;
ataDiffSample = (inputSample - ataDrySample)/2.0;
ataDiffSample += ataHalfDiffSample;
ataDiffSample -= ataLastDiffSample;
inputSample = ataDrySample;
inputSample += ataDiffSample;
overall = (overall * cancelold) + (ataDiffSample * cancelnew);
//apply all the diffs to a lowpassed IIR
if (flip)
{
iirSampleA = (iirSampleA * altAmount) + (inputSample * iirAmount);
inputSample -= iirSampleA;
iirSampleC = (iirSampleC * altAmount) + (passThrough * iirAmount);
passThrough -= iirSampleC;
}
else
{
iirSampleB = (iirSampleB * altAmount) + (inputSample * iirAmount);
inputSample -= iirSampleB;
iirSampleD = (iirSampleD * altAmount) + (passThrough * iirAmount);
passThrough -= iirSampleD;
}
flip = not flip;
//highpass section
lastOut3Sample = lastOut2Sample;
lastOut2Sample = lastOutSample;
lastOutSample = inputSample;
lpDepth -= lpSpeed;
if (lpDepth > 0.0)
{
if (lpDepth > 1.0) lpDepth = 1.0;
inputSample *= (1.0-lpDepth);
inputSample += (((lastOutSample + lastOut2Sample + lastOut3Sample) / 3.6)*lpDepth);
}
if (lpDepth < 0.0) lpDepth = 0.0;
inputSample *= (1.0-maxRecent);
inputSample += (passThrough * maxRecent);
//there's our raw signal, without antialiasing. Brings up low level stuff and softens more when hot
if (inputSample > cliplevel) inputSample = cliplevel;
if (inputSample < -cliplevel) inputSample = -cliplevel;
//final 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;
sourceP += inNumChannels; destP += inNumChannels;
}
}
