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Code Issues 33 Projects Packages Wiki Activity Actions
airwindows/plugins/MacAU/Monitoring/Monitoring.cpp
Christopher Johnson f06250c082 Clarifying Licensing in boilerplate code generation
2022-11-21 09:20:21 -05:00

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/*
* File: Monitoring.cpp
*
* Version: 1.0
*
* Created: 9/2/19
*
* Copyright: Copyright © 2019 Airwindows, Airwindows uses the MIT license
*
* Disclaimer: IMPORTANT: This Apple software is supplied to you by Apple Computer, Inc. ("Apple") in
* consideration of your agreement to the following terms, and your use, installation, modification
* or redistribution of this Apple software constitutes acceptance of these terms. If you do
* not agree with these terms, please do not use, install, modify or redistribute this Apple
* software.
*
* In consideration of your agreement to abide by the following terms, and subject to these terms,
* Apple grants you a personal, non-exclusive license, under Apple's copyrights in this
* original Apple software (the "Apple Software"), to use, reproduce, modify and redistribute the
* Apple Software, with or without modifications, in source and/or binary forms; provided that if you
* redistribute the Apple Software in its entirety and without modifications, you must retain this
* notice and the following text and disclaimers in all such redistributions of the Apple Software.
* Neither the name, trademarks, service marks or logos of Apple Computer, Inc. may be used to
* endorse or promote products derived from the Apple Software without specific prior written
* permission from Apple. Except as expressly stated in this notice, no other rights or
* licenses, express or implied, are granted by Apple herein, including but not limited to any
* patent rights that may be infringed by your derivative works or by other works in which the
* Apple Software may be incorporated.
*
* The Apple Software is provided by Apple on an "AS IS" basis. APPLE MAKES NO WARRANTIES, EXPRESS OR
* IMPLIED, INCLUDING WITHOUT LIMITATION THE IMPLIED WARRANTIES OF NON-INFRINGEMENT, MERCHANTABILITY
* AND FITNESS FOR A PARTICULAR PURPOSE, REGARDING THE APPLE SOFTWARE OR ITS USE AND OPERATION ALONE
* OR IN COMBINATION WITH YOUR PRODUCTS.
*
* IN NO EVENT SHALL APPLE BE LIABLE FOR ANY SPECIAL, INDIRECT, INCIDENTAL OR CONSEQUENTIAL
* DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS
* OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) ARISING IN ANY WAY OUT OF THE USE,
* REPRODUCTION, MODIFICATION AND/OR DISTRIBUTION OF THE APPLE SOFTWARE, HOWEVER CAUSED AND WHETHER
* UNDER THEORY OF CONTRACT, TORT (INCLUDING NEGLIGENCE), STRICT LIABILITY OR OTHERWISE, EVEN
* IF APPLE HAS BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*
*/
/*=============================================================================
Monitoring.cpp
=============================================================================*/
#include "Monitoring.h"
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
COMPONENT_ENTRY(Monitoring)
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// Monitoring::Monitoring
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
Monitoring::Monitoring(AudioUnit component)
: AUEffectBase(component)
{
CreateElements();
Globals()->UseIndexedParameters(kNumberOfParameters);
SetParameter(kParam_One, kDefaultValue_ParamOne );
#if AU_DEBUG_DISPATCHER
mDebugDispatcher = new AUDebugDispatcher (this);
#endif
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// Monitoring::GetParameterValueStrings
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult Monitoring::GetParameterValueStrings(AudioUnitScope inScope,
AudioUnitParameterID inParameterID,
CFArrayRef * outStrings)
{
if ((inScope == kAudioUnitScope_Global) && (inParameterID == kParam_One)) //ID must be actual name of parameter identifier, not number
{
if (outStrings == NULL) return noErr;
CFStringRef strings [] =
{
kMenuItem_NJAD,
kMenuItem_NJCD,
kMenuItem_PEAK,
kMenuItem_SLEW,
kMenuItem_SUBS,
kMenuItem_MONO,
kMenuItem_SIDE,
kMenuItem_VINYL,
kMenuItem_AURAT,
kMenuItem_MONORAT,
kMenuItem_MONOLAT,
kMenuItem_PHONE,
kMenuItem_CANSA,
kMenuItem_CANSB,
kMenuItem_CANSC,
kMenuItem_CANSD,
kMenuItem_TRICK
};
*outStrings = CFArrayCreate (
NULL,
(const void **) strings,
(sizeof (strings) / sizeof (strings [0])),
NULL
);
return noErr;
}
return kAudioUnitErr_InvalidProperty;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// Monitoring::GetParameterInfo
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult Monitoring::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_Indexed;
outParameterInfo.minValue = kNJAD;
outParameterInfo.maxValue = kTRICK;
outParameterInfo.defaultValue = kDefaultValue_ParamOne;
break;
default:
result = kAudioUnitErr_InvalidParameter;
break;
}
} else {
result = kAudioUnitErr_InvalidParameter;
}
return result;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// Monitoring::GetPropertyInfo
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult Monitoring::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 Monitoring::SupportedNumChannels(const AUChannelInfo ** outInfo)
{
if (outInfo != NULL)
{
static AUChannelInfo info;
info.inChannels = 2;
info.outChannels = 2;
*outInfo = &info;
}
return 1;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// Monitoring::GetProperty
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult Monitoring::GetProperty( AudioUnitPropertyID inID,
AudioUnitScope inScope,
AudioUnitElement inElement,
void * outData )
{
return AUEffectBase::GetProperty (inID, inScope, inElement, outData);
}
// Monitoring::Initialize
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult Monitoring::Initialize()
{
ComponentResult result = AUEffectBase::Initialize();
if (result == noErr)
Reset(kAudioUnitScope_Global, 0);
return result;
}
#pragma mark ____MonitoringEffectKernel
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// Monitoring::MonitoringKernel::Reset()
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult Monitoring::Reset(AudioUnitScope inScope, AudioUnitElement inElement)
{
bynL[0] = 1000.0;
bynL[1] = 301.0;
bynL[2] = 176.0;
bynL[3] = 125.0;
bynL[4] = 97.0;
bynL[5] = 79.0;
bynL[6] = 67.0;
bynL[7] = 58.0;
bynL[8] = 51.0;
bynL[9] = 46.0;
bynL[10] = 1000.0;
noiseShapingL = 0.0;
bynR[0] = 1000.0;
bynR[1] = 301.0;
bynR[2] = 176.0;
bynR[3] = 125.0;
bynR[4] = 97.0;
bynR[5] = 79.0;
bynR[6] = 67.0;
bynR[7] = 58.0;
bynR[8] = 51.0;
bynR[9] = 46.0;
bynR[10] = 1000.0;
noiseShapingR = 0.0;
//end NJAD
for(int count = 0; count < 1502; count++) {
aL[count] = 0.0; bL[count] = 0.0; cL[count] = 0.0; dL[count] = 0.0;
aR[count] = 0.0; bR[count] = 0.0; cR[count] = 0.0; dR[count] = 0.0;
}
ax = 1; bx = 1; cx = 1; dx = 1;
//PeaksOnly
lastSampleL = 0.0; lastSampleR = 0.0;
//SlewOnly
iirSampleAL = 0.0; iirSampleBL = 0.0; iirSampleCL = 0.0; iirSampleDL = 0.0; iirSampleEL = 0.0; iirSampleFL = 0.0; iirSampleGL = 0.0;
iirSampleHL = 0.0; iirSampleIL = 0.0; iirSampleJL = 0.0; iirSampleKL = 0.0; iirSampleLL = 0.0; iirSampleML = 0.0; iirSampleNL = 0.0; iirSampleOL = 0.0; iirSamplePL = 0.0;
iirSampleQL = 0.0; iirSampleRL = 0.0; iirSampleSL = 0.0;
iirSampleTL = 0.0; iirSampleUL = 0.0; iirSampleVL = 0.0;
iirSampleWL = 0.0; iirSampleXL = 0.0; iirSampleYL = 0.0; iirSampleZL = 0.0;
iirSampleAR = 0.0; iirSampleBR = 0.0; iirSampleCR = 0.0; iirSampleDR = 0.0; iirSampleER = 0.0; iirSampleFR = 0.0; iirSampleGR = 0.0;
iirSampleHR = 0.0; iirSampleIR = 0.0; iirSampleJR = 0.0; iirSampleKR = 0.0; iirSampleLR = 0.0; iirSampleMR = 0.0; iirSampleNR = 0.0; iirSampleOR = 0.0; iirSamplePR = 0.0;
iirSampleQR = 0.0; iirSampleRR = 0.0; iirSampleSR = 0.0;
iirSampleTR = 0.0; iirSampleUR = 0.0; iirSampleVR = 0.0;
iirSampleWR = 0.0; iirSampleXR = 0.0; iirSampleYR = 0.0; iirSampleZR = 0.0; // o/`
//SubsOnly
for (int x = 0; x < 11; x++) {biquad[x] = 0.0;}
//Bandpasses
fpdL = 1.0; while (fpdL < 16386) fpdL = rand()*UINT32_MAX;
fpdR = 1.0; while (fpdR < 16386) fpdR = rand()*UINT32_MAX;
return noErr;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// Monitoring::ProcessBufferLists
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
OSStatus Monitoring::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;
double overallscale = 1.0;
overallscale /= 44100.0;
overallscale *= GetSampleRate();
int processing = (int) GetParameter( kParam_One );
int am = (int)149.0 * overallscale;
int bm = (int)179.0 * overallscale;
int cm = (int)191.0 * overallscale;
int dm = (int)223.0 * overallscale; //these are 'good' primes, spacing out the allpasses
int allpasstemp;
//for PeaksOnly
biquad[0] = 0.0375/overallscale; biquad[1] = 0.1575; //define as AURAT, MONORAT, MONOLAT unless overridden
if (processing == kVINYL) {biquad[0] = 0.0385/overallscale; biquad[1] = 0.0825;}
if (processing == kPHONE) {biquad[0] = 0.1245/overallscale; biquad[1] = 0.46;}
double K = tan(M_PI * biquad[0]);
double norm = 1.0 / (1.0 + K / biquad[1] + K * K);
biquad[2] = K / biquad[1] * norm;
biquad[4] = -biquad[2]; //for bandpass, ignore [3] = 0.0
biquad[5] = 2.0 * (K * K - 1.0) * norm;
biquad[6] = (1.0 - K / biquad[1] + K * K) * norm;
//for Bandpasses
while (nSampleFrames-- > 0) {
double inputSampleL = *inputL;
double inputSampleR = *inputR;
if (fabs(inputSampleL)<1.18e-23) inputSampleL = fpdL * 1.18e-17;
if (fabs(inputSampleR)<1.18e-23) inputSampleR = fpdR * 1.18e-17;
switch (processing)
{
case 0:
case 1:
break;
case 2:
if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; inputSampleL = asin(inputSampleL);
if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; inputSampleR = asin(inputSampleR);
//amplitude aspect
allpasstemp = ax - 1; if (allpasstemp < 0 || allpasstemp > am) allpasstemp = am;
inputSampleL -= aL[allpasstemp]*0.5; aL[ax] = inputSampleL; inputSampleL *= 0.5;
inputSampleR -= aR[allpasstemp]*0.5; aR[ax] = inputSampleR; inputSampleR *= 0.5;
ax--; if (ax < 0 || ax > am) {ax = am;}
inputSampleL += (aL[ax]);
inputSampleR += (aR[ax]);
//a single Midiverb-style allpass
if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; inputSampleL = asin(inputSampleL);
if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; inputSampleR = asin(inputSampleR);
//amplitude aspect
allpasstemp = bx - 1; if (allpasstemp < 0 || allpasstemp > bm) allpasstemp = bm;
inputSampleL -= bL[allpasstemp]*0.5; bL[bx] = inputSampleL; inputSampleL *= 0.5;
inputSampleR -= bR[allpasstemp]*0.5; bR[bx] = inputSampleR; inputSampleR *= 0.5;
bx--; if (bx < 0 || bx > bm) {bx = bm;}
inputSampleL += (bL[bx]);
inputSampleR += (bR[bx]);
//a single Midiverb-style allpass
if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; inputSampleL = asin(inputSampleL);
if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; inputSampleR = asin(inputSampleR);
//amplitude aspect
allpasstemp = cx - 1; if (allpasstemp < 0 || allpasstemp > cm) allpasstemp = cm;
inputSampleL -= cL[allpasstemp]*0.5; cL[cx] = inputSampleL; inputSampleL *= 0.5;
inputSampleR -= cR[allpasstemp]*0.5; cR[cx] = inputSampleR; inputSampleR *= 0.5;
cx--; if (cx < 0 || cx > cm) {cx = cm;}
inputSampleL += (cL[cx]);
inputSampleR += (cR[cx]);
//a single Midiverb-style allpass
if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; inputSampleL = asin(inputSampleL);
if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; inputSampleR = asin(inputSampleR);
//amplitude aspect
allpasstemp = dx - 1; if (allpasstemp < 0 || allpasstemp > dm) allpasstemp = dm;
inputSampleL -= dL[allpasstemp]*0.5; dL[dx] = inputSampleL; inputSampleL *= 0.5;
inputSampleR -= dR[allpasstemp]*0.5; dR[dx] = inputSampleR; inputSampleR *= 0.5;
dx--; if (dx < 0 || dx > dm) {dx = dm;}
inputSampleL += (dL[dx]);
inputSampleR += (dR[dx]);
//a single Midiverb-style allpass
if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; inputSampleL = asin(inputSampleL);
if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; inputSampleR = asin(inputSampleR);
//amplitude aspect
inputSampleL *= 0.63679; inputSampleR *= 0.63679; //scale it to 0dB output at full blast
//PeaksOnly
break;
case 3:
Float64 trim;
trim = 2.302585092994045684017991; //natural logarithm of 10
double slewSample; slewSample = (inputSampleL - lastSampleL)*trim;
lastSampleL = inputSampleL;
if (slewSample > 1.0) slewSample = 1.0; if (slewSample < -1.0) slewSample = -1.0;
inputSampleL = slewSample;
slewSample = (inputSampleR - lastSampleR)*trim;
lastSampleR = inputSampleR;
if (slewSample > 1.0) slewSample = 1.0; if (slewSample < -1.0) slewSample = -1.0;
inputSampleR = slewSample;
//SlewOnly
break;
case 4:
Float64 iirAmount; iirAmount = (2250/44100.0) / overallscale;
Float64 gain; gain = 1.42;
inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
iirSampleAL = (iirSampleAL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleAL;
iirSampleAR = (iirSampleAR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleAR;
inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
iirSampleBL = (iirSampleBL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleBL;
iirSampleBR = (iirSampleBR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleBR;
inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
iirSampleCL = (iirSampleCL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleCL;
iirSampleCR = (iirSampleCR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleCR;
inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
iirSampleDL = (iirSampleDL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleDL;
iirSampleDR = (iirSampleDR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleDR;
inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
iirSampleEL = (iirSampleEL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleEL;
iirSampleER = (iirSampleER * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleER;
inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
iirSampleFL = (iirSampleFL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleFL;
iirSampleFR = (iirSampleFR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleFR;
inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
iirSampleGL = (iirSampleGL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleGL;
iirSampleGR = (iirSampleGR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleGR;
inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
iirSampleHL = (iirSampleHL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleHL;
iirSampleHR = (iirSampleHR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleHR;
inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
iirSampleIL = (iirSampleIL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleIL;
iirSampleIR = (iirSampleIR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleIR;
inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
iirSampleJL = (iirSampleJL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleJL;
iirSampleJR = (iirSampleJR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleJR;
inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
iirSampleKL = (iirSampleKL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleKL;
iirSampleKR = (iirSampleKR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleKR;
inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
iirSampleLL = (iirSampleLL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleLL;
iirSampleLR = (iirSampleLR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleLR;
inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
iirSampleML = (iirSampleML * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleML;
iirSampleMR = (iirSampleMR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleMR;
inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
iirSampleNL = (iirSampleNL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleNL;
iirSampleNR = (iirSampleNR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleNR;
inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
iirSampleOL = (iirSampleOL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleOL;
iirSampleOR = (iirSampleOR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleOR;
inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
iirSamplePL = (iirSamplePL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSamplePL;
iirSamplePR = (iirSamplePR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSamplePR;
inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
iirSampleQL = (iirSampleQL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleQL;
iirSampleQR = (iirSampleQR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleQR;
inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
iirSampleRL = (iirSampleRL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleRL;
iirSampleRR = (iirSampleRR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleRR;
inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
iirSampleSL = (iirSampleSL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleSL;
iirSampleSR = (iirSampleSR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleSR;
inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
iirSampleTL = (iirSampleTL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleTL;
iirSampleTR = (iirSampleTR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleTR;
inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
iirSampleUL = (iirSampleUL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleUL;
iirSampleUR = (iirSampleUR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleUR;
inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
iirSampleVL = (iirSampleVL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleVL;
iirSampleVR = (iirSampleVR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleVR;
inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
iirSampleWL = (iirSampleWL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleWL;
iirSampleWR = (iirSampleWR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleWR;
inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
iirSampleXL = (iirSampleXL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleXL;
iirSampleXR = (iirSampleXR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleXR;
inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
iirSampleYL = (iirSampleYL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleYL;
iirSampleYR = (iirSampleYR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleYR;
inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1;
if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
iirSampleZL = (iirSampleZL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleZL;
iirSampleZR = (iirSampleZR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleZR;
if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
//SubsOnly
break;
case 5:
case 6:
double mid; mid = inputSampleL + inputSampleR;
double side; side = inputSampleL - inputSampleR;
if (processing < 6) side = 0.0;
else mid = 0.0; //mono monitoring, or side-only monitoring
inputSampleL = (mid+side)/2.0;
inputSampleR = (mid-side)/2.0;
break;
case 7:
case 8:
case 9:
case 10:
case 11:
//Bandpass: changes in EQ are up in the variable defining, not here
//7 Vinyl, 8 9 10 Aurat, 11 Phone
if (processing == 9) {inputSampleR = (inputSampleL + inputSampleR)*0.5;inputSampleL = 0.0;}
if (processing == 10) {inputSampleL = (inputSampleL + inputSampleR)*0.5;inputSampleR = 0.0;}
if (processing == 11) {double M; M = (inputSampleL + inputSampleR)*0.5; inputSampleL = M;inputSampleR = M;}
inputSampleL = sin(inputSampleL); inputSampleR = sin(inputSampleR);
//encode Console5: good cleanness
double tempSampleL; tempSampleL = (inputSampleL * biquad[2]) + biquad[7];
biquad[7] = (-tempSampleL * biquad[5]) + biquad[8];
biquad[8] = (inputSampleL * biquad[4]) - (tempSampleL * biquad[6]);
inputSampleL = tempSampleL; //like mono AU, 7 and 8 store L channel
double tempSampleR; tempSampleR = (inputSampleR * biquad[2]) + biquad[9];
biquad[9] = (-tempSampleR * biquad[5]) + biquad[10];
biquad[10] = (inputSampleR * biquad[4]) - (tempSampleR * biquad[6]);
inputSampleR = tempSampleR; //note: 9 and 10 store the R channel
if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0;
if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0;
//without this, you can get a NaN condition where it spits out DC offset at full blast!
inputSampleL = asin(inputSampleL); inputSampleR = asin(inputSampleR);
//amplitude aspect
break;
case 12:
case 13:
case 14:
case 15:
if (processing == 12) {inputSampleL *= 0.855; inputSampleR *= 0.855;}
if (processing == 13) {inputSampleL *= 0.748; inputSampleR *= 0.748;}
if (processing == 14) {inputSampleL *= 0.713; inputSampleR *= 0.713;}
if (processing == 15) {inputSampleL *= 0.680; inputSampleR *= 0.680;}
//we do a volume compensation immediately to gain stage stuff cleanly
inputSampleL = sin(inputSampleL);
inputSampleR = sin(inputSampleR);
double drySampleL; drySampleL = inputSampleL;
double drySampleR; drySampleR = inputSampleR; //everything runs 'inside' Console
double bass; bass = (processing * processing * 0.00001) / overallscale;
//we are using the iir filters from out of SubsOnly
mid = inputSampleL + inputSampleR; side = inputSampleL - inputSampleR;
iirSampleAL = (iirSampleAL * (1.0 - (bass*0.618))) + (side * bass * 0.618); side = side - iirSampleAL;
inputSampleL = (mid+side)/2.0; inputSampleR = (mid-side)/2.0;
//bass narrowing filter
allpasstemp = ax - 1; if (allpasstemp < 0 || allpasstemp > am) allpasstemp = am;
inputSampleL -= aL[allpasstemp]*0.5; aL[ax] = inputSampleL; inputSampleL *= 0.5;
inputSampleR -= aR[allpasstemp]*0.5; aR[ax] = inputSampleR; inputSampleR *= 0.5;
ax--; if (ax < 0 || ax > am) {ax = am;}
inputSampleL += (aL[ax])*0.5; inputSampleR += (aR[ax])*0.5;
if (ax == am) {inputSampleL += (aL[0])*0.5; inputSampleR += (aR[0])*0.5;}
else {inputSampleL += (aL[ax+1])*0.5; inputSampleR += (aR[ax+1])*0.5;}
//a darkened Midiverb-style allpass
if (processing == 12) {inputSampleL *= 0.125; inputSampleR *= 0.125;}
if (processing == 13) {inputSampleL *= 0.25; inputSampleR *= 0.25;}
if (processing == 14) {inputSampleL *= 0.30; inputSampleR *= 0.30;}
if (processing == 15) {inputSampleL *= 0.35; inputSampleR *= 0.35;}
//Cans A suppresses the crossfeed more, Cans B makes it louder
drySampleL += inputSampleR;
drySampleR += inputSampleL; //the crossfeed
allpasstemp = dx - 1; if (allpasstemp < 0 || allpasstemp > dm) allpasstemp = dm;
inputSampleL -= dL[allpasstemp]*0.5; dL[dx] = inputSampleL; inputSampleL *= 0.5;
inputSampleR -= dR[allpasstemp]*0.5; dR[dx] = inputSampleR; inputSampleR *= 0.5;
dx--; if (dx < 0 || dx > dm) {dx = dm;}
inputSampleL += (dL[dx])*0.5; inputSampleR += (dR[dx])*0.5;
if (dx == dm) {inputSampleL += (dL[0])*0.5; inputSampleR += (dR[0])*0.5;}
else {inputSampleL += (dL[dx+1])*0.5; inputSampleR += (dR[dx+1])*0.5;}
//a darkened Midiverb-style allpass, which is stretching the previous one even more
inputSampleL *= 0.25; inputSampleR *= 0.25;
//for all versions of Cans the second level of bloom is this far down
//and, remains on the opposite speaker rather than crossing again to the original side
drySampleL += inputSampleR;
drySampleR += inputSampleL; //add the crossfeed and very faint extra verbyness
inputSampleL = drySampleL;
inputSampleR = drySampleR; //and output our can-opened headphone feed
mid = inputSampleL + inputSampleR; side = inputSampleL - inputSampleR;
iirSampleAR = (iirSampleAR * (1.0 - bass)) + (side * bass); side = side - iirSampleAR;
inputSampleL = (mid+side)/2.0; inputSampleR = (mid-side)/2.0;
//bass narrowing filter
if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; inputSampleL = asin(inputSampleL);
if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; inputSampleR = asin(inputSampleR);
//ConsoleBuss processing
break;
case 16:
double inputSample = (inputSampleL + inputSampleR) * 0.5;
inputSampleL = -inputSample;
inputSampleR = inputSample;
break;
}
//begin Not Just Another Dither
if (processing == 1) {
inputSampleL = inputSampleL * 32768.0; //or 16 bit option
inputSampleR = inputSampleR * 32768.0; //or 16 bit option
} else {
inputSampleL = inputSampleL * 8388608.0; //for literally everything else
inputSampleR = inputSampleR * 8388608.0; //we will apply the 24 bit NJAD
} //on the not unreasonable assumption that we are very likely playing back on 24 bit DAC
//if we're not, then all we did was apply a Benford Realness function at 24 bits down.
bool cutbinsL; cutbinsL = false;
bool cutbinsR; cutbinsR = false;
double drySampleL; drySampleL = inputSampleL;
double drySampleR; drySampleR = inputSampleR;
inputSampleL -= noiseShapingL;
inputSampleR -= noiseShapingR;
//NJAD L
double benfordize; benfordize = floor(inputSampleL);
while (benfordize >= 1.0) benfordize /= 10;
while (benfordize < 1.0 && benfordize > 0.0000001) benfordize *= 10;
int hotbinA; hotbinA = floor(benfordize);
//hotbin becomes the Benford bin value for this number floored
double totalA; totalA = 0;
if ((hotbinA > 0) && (hotbinA < 10))
{
bynL[hotbinA] += 1; if (bynL[hotbinA] > 982) cutbinsL = true;
totalA += (301-bynL[1]); totalA += (176-bynL[2]); totalA += (125-bynL[3]);
totalA += (97-bynL[4]); totalA += (79-bynL[5]); totalA += (67-bynL[6]);
totalA += (58-bynL[7]); totalA += (51-bynL[8]); totalA += (46-bynL[9]); bynL[hotbinA] -= 1;
} else hotbinA = 10;
//produce total number- smaller is closer to Benford real
benfordize = ceil(inputSampleL);
while (benfordize >= 1.0) benfordize /= 10;
while (benfordize < 1.0 && benfordize > 0.0000001) benfordize *= 10;
int hotbinB; hotbinB = floor(benfordize);
//hotbin becomes the Benford bin value for this number ceiled
double totalB; totalB = 0;
if ((hotbinB > 0) && (hotbinB < 10))
{
bynL[hotbinB] += 1; if (bynL[hotbinB] > 982) cutbinsL = true;
totalB += (301-bynL[1]); totalB += (176-bynL[2]); totalB += (125-bynL[3]);
totalB += (97-bynL[4]); totalB += (79-bynL[5]); totalB += (67-bynL[6]);
totalB += (58-bynL[7]); totalB += (51-bynL[8]); totalB += (46-bynL[9]); bynL[hotbinB] -= 1;
} else hotbinB = 10;
//produce total number- smaller is closer to Benford real
double outputSample;
if (totalA < totalB) {bynL[hotbinA] += 1; outputSample = floor(inputSampleL);}
else {bynL[hotbinB] += 1; outputSample = floor(inputSampleL+1);}
//assign the relevant one to the delay line
//and floor/ceil signal accordingly
if (cutbinsL) {
bynL[1] *= 0.99; bynL[2] *= 0.99; bynL[3] *= 0.99; bynL[4] *= 0.99; bynL[5] *= 0.99;
bynL[6] *= 0.99; bynL[7] *= 0.99; bynL[8] *= 0.99; bynL[9] *= 0.99; bynL[10] *= 0.99;
}
noiseShapingL += outputSample - drySampleL;
if (noiseShapingL > fabs(inputSampleL)) noiseShapingL = fabs(inputSampleL);
if (noiseShapingL < -fabs(inputSampleL)) noiseShapingL = -fabs(inputSampleL);
if (processing == 1) inputSampleL = outputSample / 32768.0;
else inputSampleL = outputSample / 8388608.0;
if (inputSampleL > 1.0) inputSampleL = 1.0;
if (inputSampleL < -1.0) inputSampleL = -1.0;
//finished NJAD L
//NJAD R
benfordize = floor(inputSampleR);
while (benfordize >= 1.0) benfordize /= 10;
while (benfordize < 1.0 && benfordize > 0.0000001) benfordize *= 10;
hotbinA = floor(benfordize);
//hotbin becomes the Benford bin value for this number floored
totalA = 0;
if ((hotbinA > 0) && (hotbinA < 10))
{
bynR[hotbinA] += 1; if (bynR[hotbinA] > 982) cutbinsR = true;
totalA += (301-bynR[1]); totalA += (176-bynR[2]); totalA += (125-bynR[3]);
totalA += (97-bynR[4]); totalA += (79-bynR[5]); totalA += (67-bynR[6]);
totalA += (58-bynR[7]); totalA += (51-bynR[8]); totalA += (46-bynR[9]); bynR[hotbinA] -= 1;
} else hotbinA = 10;
//produce total number- smaller is closer to Benford real
benfordize = ceil(inputSampleR);
while (benfordize >= 1.0) benfordize /= 10;
while (benfordize < 1.0 && benfordize > 0.0000001) benfordize *= 10;
hotbinB = floor(benfordize);
//hotbin becomes the Benford bin value for this number ceiled
totalB = 0;
if ((hotbinB > 0) && (hotbinB < 10))
{
bynR[hotbinB] += 1; if (bynR[hotbinB] > 982) cutbinsR = true;
totalB += (301-bynR[1]); totalB += (176-bynR[2]); totalB += (125-bynR[3]);
totalB += (97-bynR[4]); totalB += (79-bynR[5]); totalB += (67-bynR[6]);
totalB += (58-bynR[7]); totalB += (51-bynR[8]); totalB += (46-bynR[9]); bynR[hotbinB] -= 1;
} else hotbinB = 10;
//produce total number- smaller is closer to Benford real
if (totalA < totalB) {bynR[hotbinA] += 1; outputSample = floor(inputSampleR);}
else {bynR[hotbinB] += 1; outputSample = floor(inputSampleR+1);}
//assign the relevant one to the delay line
//and floor/ceil signal accordingly
if (cutbinsR) {
bynR[1] *= 0.99; bynR[2] *= 0.99; bynR[3] *= 0.99; bynR[4] *= 0.99; bynR[5] *= 0.99;
bynR[6] *= 0.99; bynR[7] *= 0.99; bynR[8] *= 0.99; bynR[9] *= 0.99; bynR[10] *= 0.99;
}
noiseShapingR += outputSample - drySampleR;
if (noiseShapingR > fabs(inputSampleR)) noiseShapingR = fabs(inputSampleR);
if (noiseShapingR < -fabs(inputSampleR)) noiseShapingR = -fabs(inputSampleR);
if (processing == 1) inputSampleR = outputSample / 32768.0;
else inputSampleR = outputSample / 8388608.0;
if (inputSampleR > 1.0) inputSampleR = 1.0;
if (inputSampleR < -1.0) inputSampleR = -1.0;
//finished NJAD R
//does not use 32 bit stereo floating point dither
*outputL = inputSampleL;
*outputR = inputSampleR;
//direct stereo out
inputL += 1;
inputR += 1;
outputL += 1;
outputR += 1;
}
return noErr;
}
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