/*
* File: ADClip7.cpp
*
* Version: 1.0
*
* Created: 8/2/17
*
* Copyright: Copyright � 2017 Airwindows, All Rights Reserved
*
* 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.
*
*/
/*=============================================================================
ADClip7.cpp
=============================================================================*/
#include "ADClip7.h"
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
COMPONENT_ENTRY(ADClip7)
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// ADClip7::ADClip7
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ADClip7::ADClip7(AudioUnit component)
: AUEffectBase(component)
{
CreateElements();
Globals()->UseIndexedParameters(kNumberOfParameters);
SetParameter(kParam_One, kDefaultValue_ParamOne );
SetParameter(kParam_Two, kDefaultValue_ParamTwo );
SetParameter(kParam_Three, kDefaultValue_ParamThree );
SetParameter(kParam_Four, kDefaultValue_ParamFour );
#if AU_DEBUG_DISPATCHER
mDebugDispatcher = new AUDebugDispatcher (this);
#endif
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// ADClip7::GetParameterValueStrings
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult ADClip7::GetParameterValueStrings(AudioUnitScope inScope,
AudioUnitParameterID inParameterID,
CFArrayRef * outStrings)
{
if ((inScope == kAudioUnitScope_Global) && (inParameterID == kParam_Four)) //ID must be actual name of parameter identifier, not number
{
if (outStrings == NULL) return noErr;
CFStringRef strings [] =
{
kMenuItem_Normal,
kMenuItem_Gain,
kMenuItem_Clip,
};
*outStrings = CFArrayCreate (
NULL,
(const void **) strings,
(sizeof (strings) / sizeof (strings [0])),
NULL
);
return noErr;
}
return kAudioUnitErr_InvalidProperty;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// ADClip7::GetParameterInfo
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult ADClip7::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 = 18.0;
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;
case kParam_Four:
AUBase::FillInParameterName (outParameterInfo, kParameterFourName, false);
outParameterInfo.unit = kAudioUnitParameterUnit_Indexed;
outParameterInfo.minValue = kNormal;
outParameterInfo.maxValue = kClip;
outParameterInfo.defaultValue = kDefaultValue_ParamFour;
break;
default:
result = kAudioUnitErr_InvalidParameter;
break;
}
} else {
result = kAudioUnitErr_InvalidParameter;
}
return result;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// ADClip7::GetPropertyInfo
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult ADClip7::GetPropertyInfo (AudioUnitPropertyID inID,
AudioUnitScope inScope,
AudioUnitElement inElement,
UInt32 & outDataSize,
Boolean & outWritable)
{
return AUEffectBase::GetPropertyInfo (inID, inScope, inElement, outDataSize, outWritable);
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// ADClip7::GetProperty
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult ADClip7::GetProperty( AudioUnitPropertyID inID,
AudioUnitScope inScope,
AudioUnitElement inElement,
void * outData )
{
return AUEffectBase::GetProperty (inID, inScope, inElement, outData);
}
// ADClip7::Initialize
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ComponentResult ADClip7::Initialize()
{
ComponentResult result = AUEffectBase::Initialize();
if (result == noErr)
Reset(kAudioUnitScope_Global, 0);
return result;
}
#pragma mark ____ADClip7EffectKernel
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// ADClip7::ADClip7Kernel::Reset()
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
void ADClip7::ADClip7Kernel::Reset()
{
lastSample = 0.0;
for(int count = 0; count < 22199; count++) {b[count] = 0;}
gcount = 0;
lows = 0;
refclip = 0.99;
iirLowsA = 0.0;
iirLowsB = 0.0;
fpNShape = 0.0;
}
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// ADClip7::ADClip7Kernel::Process
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
void ADClip7::ADClip7Kernel::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 fpOld = 0.618033988749894848204586; //golden ratio!
Float64 fpNew = 1.0 - fpOld;
Float64 inputGain = pow(10.0,(GetParameter( kParam_One ))/20.0);
Float64 softness = GetParameter( kParam_Two ) * fpNew;
Float64 hardness = 1.0 - softness;
Float64 highslift = 0.307 * GetParameter( kParam_Three );
Float64 adjust = pow(highslift,3) * 0.416;
Float64 subslift = 0.796 * GetParameter( kParam_Three );
Float64 calibsubs = subslift/53;
Float64 invcalibsubs = 1.0 - calibsubs;
Float64 subs = 0.81 + (calibsubs*2);
long double bridgerectifier;
int mode = (int) GetParameter( kParam_Four );
Float64 overshoot;
Float64 offsetH1 = 1.84;
offsetH1 *= overallscale;
Float64 offsetH2 = offsetH1 * 1.9;
Float64 offsetH3 = offsetH1 * 2.7;
Float64 offsetL1 = 612;
offsetL1 *= overallscale;
Float64 offsetL2 = offsetL1 * 2.0;
int refH1 = (int)floor(offsetH1);
int refH2 = (int)floor(offsetH2);
int refH3 = (int)floor(offsetH3);
int refL1 = (int)floor(offsetL1);
int refL2 = (int)floor(offsetL2);
int temp;
Float64 fractionH1 = offsetH1 - floor(offsetH1);
Float64 fractionH2 = offsetH2 - floor(offsetH2);
Float64 fractionH3 = offsetH3 - floor(offsetH3);
Float64 minusH1 = 1.0 - fractionH1;
Float64 minusH2 = 1.0 - fractionH2;
Float64 minusH3 = 1.0 - fractionH3;
Float64 highs = 0.0;
int count = 0;
long double inputSample;
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.
}
if (inputGain != 1.0) {
inputSample *= inputGain;
}
overshoot = fabs(inputSample) - refclip;
if (overshoot < 0.0) overshoot = 0.0;
if (gcount < 0 || gcount > 11020) {gcount = 11020;}
count = gcount;
b[count+11020] = b[count] = overshoot;
gcount--;
if (highslift > 0.0)
{
//we have a big pile of b[] which is overshoots
temp = count+refH3;
highs = -(b[temp] * minusH3); //less as value moves away from .0
highs -= b[temp+1]; //we can assume always using this in one way or another?
highs -= (b[temp+2] * fractionH3); //greater as value moves away from .0
highs += (((b[temp]-b[temp+1])-(b[temp+1]-b[temp+2]))/50); //interpolation hacks 'r us
highs *= adjust; //add in the kernel elements backwards saves multiplies
//stage 3 is a negative add
temp = count+refH2;
highs += (b[temp] * minusH2); //less as value moves away from .0
highs += b[temp+1]; //we can assume always using this in one way or another?
highs += (b[temp+2] * fractionH2); //greater as value moves away from .0
highs -= (((b[temp]-b[temp+1])-(b[temp+1]-b[temp+2]))/50); //interpolation hacks 'r us
highs *= adjust; //add in the kernel elements backwards saves multiplies
//stage 2 is a positive feedback of the overshoot
temp = count+refH1;
highs -= (b[temp] * minusH1); //less as value moves away from .0
highs -= b[temp+1]; //we can assume always using this in one way or another?
highs -= (b[temp+2] * fractionH1); //greater as value moves away from .0
highs += (((b[temp]-b[temp+1])-(b[temp+1]-b[temp+2]))/50); //interpolation hacks 'r us
highs *= adjust; //add in the kernel elements backwards saves multiplies
//stage 1 is a negative feedback of the overshoot
//done with interpolated mostly negative feedback of the overshoot
}
bridgerectifier = sin(fabs(highs) * hardness);
//this will wrap around and is scaled back by softness
//wrap around is the same principle as Fracture: no top limit to sin()
if (highs > 0) highs = bridgerectifier;
else highs = -bridgerectifier;
if (subslift > 0.0)
{
lows *= subs;
//going in we'll reel back some of the swing
temp = count+refL1;
lows -= b[temp+127];
lows -= b[temp+113];
lows -= b[temp+109];
lows -= b[temp+107];
lows -= b[temp+103];
lows -= b[temp+101];
lows -= b[temp+97];
lows -= b[temp+89];
lows -= b[temp+83];
lows -= b[temp+79];
lows -= b[temp+73];
lows -= b[temp+71];
lows -= b[temp+67];
lows -= b[temp+61];
lows -= b[temp+59];
lows -= b[temp+53];
lows -= b[temp+47];
lows -= b[temp+43];
lows -= b[temp+41];
lows -= b[temp+37];
lows -= b[temp+31];
lows -= b[temp+29];
lows -= b[temp+23];
lows -= b[temp+19];
lows -= b[temp+17];
lows -= b[temp+13];
lows -= b[temp+11];
lows -= b[temp+7];
lows -= b[temp+5];
lows -= b[temp+3];
lows -= b[temp+2];
lows -= b[temp+1];
//initial negative lobe
lows *= subs;
lows *= subs;
//twice, to minimize the suckout in low boost situations
temp = count+refL2;
lows += b[temp+127];
lows += b[temp+113];
lows += b[temp+109];
lows += b[temp+107];
lows += b[temp+103];
lows += b[temp+101];
lows += b[temp+97];
lows += b[temp+89];
lows += b[temp+83];
lows += b[temp+79];
lows += b[temp+73];
lows += b[temp+71];
lows += b[temp+67];
lows += b[temp+61];
lows += b[temp+59];
lows += b[temp+53];
lows += b[temp+47];
lows += b[temp+43];
lows += b[temp+41];
lows += b[temp+37];
lows += b[temp+31];
lows += b[temp+29];
lows += b[temp+23];
lows += b[temp+19];
lows += b[temp+17];
lows += b[temp+13];
lows += b[temp+11];
lows += b[temp+7];
lows += b[temp+5];
lows += b[temp+3];
lows += b[temp+2];
lows += b[temp+1];
lows *= subs;
//followup positive lobe
//now we have the lows content to use
}
bridgerectifier = sin(fabs(lows) * softness);
//this will wrap around and is scaled back by hardness: hard = less bass push, more treble
//wrap around is the same principle as Fracture: no top limit to sin()
if (lows > 0) lows = bridgerectifier;
else lows = -bridgerectifier;
iirLowsA = (iirLowsA * invcalibsubs) + (lows * calibsubs);
lows = iirLowsA;
bridgerectifier = sin(fabs(lows));
if (lows > 0) lows = bridgerectifier;
else lows = -bridgerectifier;
iirLowsB = (iirLowsB * invcalibsubs) + (lows * calibsubs);
lows = iirLowsB;
bridgerectifier = sin(fabs(lows)) * 2.0;
if (lows > 0) lows = bridgerectifier;
else lows = -bridgerectifier;
if (highslift > 0.0) inputSample += (highs * (1.0-fabs(inputSample*hardness)));
if (subslift > 0.0) inputSample += (lows * (1.0-fabs(inputSample*softness)));
if (inputSample > refclip && refclip > 0.9) refclip -= 0.01;
if (inputSample < -refclip && refclip > 0.9) refclip -= 0.01;
if (refclip < 0.99) refclip += 0.00001;
//adjust clip level on the fly
if (lastSample >= refclip)
{
if (inputSample < refclip) lastSample = ((refclip*hardness) + (inputSample * softness));
else lastSample = refclip;
}
if (lastSample <= -refclip)
{
if (inputSample > -refclip) lastSample = ((-refclip*hardness) + (inputSample * softness));
else lastSample = -refclip;
}
if (inputSample > refclip)
{
if (lastSample < refclip) inputSample = ((refclip*hardness) + (lastSample * softness));
else inputSample = refclip;
}
if (inputSample < -refclip)
{
if (lastSample > -refclip) inputSample = ((-refclip*hardness) + (lastSample * softness));
else inputSample = -refclip;
}
lastSample = inputSample;
switch (mode)
{
case 1: break; //Normal
case 2: inputSample /= inputGain; break; //Gain Match
case 3: inputSample = overshoot + highs + lows; break; //Clip Only
}
//this is our output mode switch, showing the effects
if (inputSample > refclip) inputSample = refclip;
if (inputSample < -refclip) inputSample = -refclip;
//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;
}
}