/* * File: Righteous4.cpp * * Version: 1.0 * * Created: 4/8/18 * * Copyright: Copyright © 2018 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. 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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. * */ /*============================================================================= Righteous4.cpp =============================================================================*/ #include "Righteous4.h" //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ COMPONENT_ENTRY(Righteous4) //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // Righteous4::Righteous4 //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ Righteous4::Righteous4(AudioUnit component) : AUEffectBase(component) { CreateElements(); Globals()->UseIndexedParameters(kNumberOfParameters); SetParameter(kParam_One, kDefaultValue_ParamOne ); SetParameter(kParam_Two, kDefaultValue_ParamTwo ); #if AU_DEBUG_DISPATCHER mDebugDispatcher = new AUDebugDispatcher (this); #endif } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // Righteous4::GetParameterValueStrings //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult Righteous4::GetParameterValueStrings(AudioUnitScope inScope, AudioUnitParameterID inParameterID, CFArrayRef * outStrings) { if ((inScope == kAudioUnitScope_Global) && (inParameterID == kParam_Two)) //ID must be actual name of parameter identifier, not number { if (outStrings == NULL) return noErr; CFStringRef strings [] = { kMenuItem_16bit, kMenuItem_24bit, kMenuItem_32bit, }; *outStrings = CFArrayCreate ( NULL, (const void **) strings, (sizeof (strings) / sizeof (strings [0])), NULL ); return noErr; } return kAudioUnitErr_InvalidProperty; } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // Righteous4::GetParameterInfo //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult Righteous4::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 = -28.0; outParameterInfo.maxValue = -4.0; outParameterInfo.defaultValue = kDefaultValue_ParamOne; break; case kParam_Two: AUBase::FillInParameterName (outParameterInfo, kParameterTwoName, false); outParameterInfo.unit = kAudioUnitParameterUnit_Indexed; outParameterInfo.minValue = k16bit; outParameterInfo.maxValue = k32bit; outParameterInfo.defaultValue = kDefaultValue_ParamTwo; break; default: result = kAudioUnitErr_InvalidParameter; break; } } else { result = kAudioUnitErr_InvalidParameter; } return result; } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // Righteous4::GetPropertyInfo //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult Righteous4::GetPropertyInfo (AudioUnitPropertyID inID, AudioUnitScope inScope, AudioUnitElement inElement, UInt32 & outDataSize, Boolean & outWritable) { return AUEffectBase::GetPropertyInfo (inID, inScope, inElement, outDataSize, outWritable); } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // Righteous4::GetProperty //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult Righteous4::GetProperty( AudioUnitPropertyID inID, AudioUnitScope inScope, AudioUnitElement inElement, void * outData ) { return AUEffectBase::GetProperty (inID, inScope, inElement, outData); } // Righteous4::Initialize //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult Righteous4::Initialize() { ComponentResult result = AUEffectBase::Initialize(); if (result == noErr) Reset(kAudioUnitScope_Global, 0); return result; } #pragma mark ____Righteous4EffectKernel //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // Righteous4::Righteous4Kernel::Reset() //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ void Righteous4::Righteous4Kernel::Reset() { midSampleA = 0.0; midSampleB = 0.0; midSampleC = 0.0; midSampleD = 0.0; midSampleE = 0.0; midSampleF = 0.0; midSampleG = 0.0; midSampleH = 0.0; midSampleI = 0.0; midSampleJ = 0.0; midSampleK = 0.0; midSampleL = 0.0; midSampleM = 0.0; midSampleN = 0.0; midSampleO = 0.0; midSampleP = 0.0; midSampleQ = 0.0; midSampleR = 0.0; midSampleS = 0.0; midSampleT = 0.0; midSampleU = 0.0; midSampleV = 0.0; midSampleW = 0.0; midSampleX = 0.0; midSampleY = 0.0; midSampleZ = 0.0; byn[0] = 1000; byn[1] = 301; byn[2] = 176; byn[3] = 125; byn[4] = 97; byn[5] = 79; byn[6] = 67; byn[7] = 58; byn[8] = 51; byn[9] = 46; byn[10] = 1000; noiseShaping = 0.0; lastSample = 0.0; IIRsample = 0.0; gwPrev = 0.0; gwA = 0.0; gwB = 0.0; fpNShape = 0.0; } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // Righteous4::Righteous4Kernel::Process //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ void Righteous4::Righteous4Kernel::Process( const Float32 *inSourceP, Float32 *inDestP, UInt32 inFramesToProcess, UInt32 inNumChannels, bool &ioSilence ) { UInt32 nSampleFrames = inFramesToProcess; const Float32 *sourceP = inSourceP; Float32 *destP = inDestP; long double fpOld = 0.618033988749894848204586; //golden ratio! long double fpNew = 1.0 - fpOld; Float64 overallscale = 1.0; overallscale /= 44100.0; overallscale *= GetSampleRate(); Float64 IIRscaleback = 0.0002597;//scaleback of harmonic avg IIRscaleback /= overallscale; IIRscaleback = 1.0 - IIRscaleback; Float64 target = GetParameter( kParam_One ); target += 17; //gives us scaled distortion factor based on test conditions target = pow(10.0,target/20.0); //we will multiply and divide by this //ShortBuss section if (target == 0) target = 1; //insanity check int bitDepth = (int) GetParameter( kParam_Two ); // +1 for Reaper bug workaround Float64 fusswithscale = 149940.0; //corrected Float64 cutofffreq = 20; //was 46/2.0 Float64 midAmount = (cutofffreq)/fusswithscale; midAmount /= overallscale; Float64 midaltAmount = 1.0 - midAmount; Float64 gwAfactor = 0.718; gwAfactor -= (overallscale*0.05); //0.2 at 176K, 0.1 at 88.2K, 0.05 at 44.1K //reduce slightly to not less than 0.5 to increase effect Float64 gwBfactor = 1.0 - gwAfactor; Float64 softness = 0.2135; Float64 hardness = 1.0 - softness; Float64 refclip = pow(10.0,-0.0058888); while (nSampleFrames-- > 0) { long double 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. } Float64 drySample = inputSample; //begin the whole distortion dealiebop inputSample /= target; //running shortbuss on direct sample IIRsample *= IIRscaleback; Float64 secondharmonic = sin((2 * inputSample * inputSample) * IIRsample); //secondharmonic is calculated before IIRsample is updated, to delay reaction long double bridgerectifier = inputSample; if (bridgerectifier > 1.2533141373155) bridgerectifier = 1.2533141373155; if (bridgerectifier < -1.2533141373155) bridgerectifier = -1.2533141373155; //clip to 1.2533141373155 to reach maximum output bridgerectifier = sin(bridgerectifier * fabs(bridgerectifier)) / ((bridgerectifier == 0.0) ?1:fabs(bridgerectifier)); if (inputSample > bridgerectifier) IIRsample += ((inputSample - bridgerectifier)*0.0009); if (inputSample < -bridgerectifier) IIRsample += ((inputSample + bridgerectifier)*0.0009); //manipulate IIRSample inputSample = bridgerectifier; //apply the distortion transform for reals. Has been converted back to -1/1 //apply resonant highpass Float64 tempSample = inputSample; midSampleA = (midSampleA * midaltAmount) + (tempSample * midAmount); tempSample -= midSampleA; Float64 correction = midSampleA; midSampleB = (midSampleB * midaltAmount) + (tempSample * midAmount); tempSample -= midSampleB; correction += midSampleB; midSampleC = (midSampleC * midaltAmount) + (tempSample * midAmount); tempSample -= midSampleC; correction += midSampleC; midSampleD = (midSampleD * midaltAmount) + (tempSample * midAmount); tempSample -= midSampleD; correction += midSampleD; midSampleE = (midSampleE * midaltAmount) + (tempSample * midAmount); tempSample -= midSampleE; correction += midSampleE; midSampleF = (midSampleF * midaltAmount) + (tempSample * midAmount); tempSample -= midSampleF; correction += midSampleF; midSampleG = (midSampleG * midaltAmount) + (tempSample * midAmount); tempSample -= midSampleG; correction += midSampleG; midSampleH = (midSampleH * midaltAmount) + (tempSample * midAmount); tempSample -= midSampleH; correction += midSampleH; midSampleI = (midSampleI * midaltAmount) + (tempSample * midAmount); tempSample -= midSampleI; correction += midSampleI; midSampleJ = (midSampleJ * midaltAmount) + (tempSample * midAmount); tempSample -= midSampleJ; correction += midSampleJ; midSampleK = (midSampleK * midaltAmount) + (tempSample * midAmount); tempSample -= midSampleK; correction += midSampleK; midSampleL = (midSampleL * midaltAmount) + (tempSample * midAmount); tempSample -= midSampleL; correction += midSampleL; midSampleM = (midSampleM * midaltAmount) + (tempSample * midAmount); tempSample -= midSampleM; correction += midSampleM; midSampleN = (midSampleN * midaltAmount) + (tempSample * midAmount); tempSample -= midSampleN; correction += midSampleN; midSampleO = (midSampleO * midaltAmount) + (tempSample * midAmount); tempSample -= midSampleO; correction += midSampleO; midSampleP = (midSampleP * midaltAmount) + (tempSample * midAmount); tempSample -= midSampleP; correction += midSampleP; midSampleQ = (midSampleQ * midaltAmount) + (tempSample * midAmount); tempSample -= midSampleQ; correction += midSampleQ; midSampleR = (midSampleR * midaltAmount) + (tempSample * midAmount); tempSample -= midSampleR; correction += midSampleR; midSampleS = (midSampleS * midaltAmount) + (tempSample * midAmount); tempSample -= midSampleS; correction += midSampleS; midSampleT = (midSampleT * midaltAmount) + (tempSample * midAmount); tempSample -= midSampleT; correction += midSampleT; midSampleU = (midSampleU * midaltAmount) + (tempSample * midAmount); tempSample -= midSampleU; correction += midSampleU; midSampleV = (midSampleV * midaltAmount) + (tempSample * midAmount); tempSample -= midSampleV; correction += midSampleV; midSampleW = (midSampleW * midaltAmount) + (tempSample * midAmount); tempSample -= midSampleW; correction += midSampleW; midSampleX = (midSampleX * midaltAmount) + (tempSample * midAmount); tempSample -= midSampleX; correction += midSampleX; midSampleY = (midSampleY * midaltAmount) + (tempSample * midAmount); tempSample -= midSampleY; correction += midSampleY; midSampleZ = (midSampleZ * midaltAmount) + (tempSample * midAmount); tempSample -= midSampleZ; correction += midSampleZ; correction *= fabs(secondharmonic); //scale it directly by second harmonic: DC block is now adding harmonics too correction -= secondharmonic*fpOld; //apply the shortbuss processing to output DCblock by subtracting it //we are not a peak limiter! not using it to clip or nothin' //adding it inversely, it's the same as adding to inputsample only we are accumulating 'stuff' in 'correction' inputSample -= correction; if (inputSample < 0) inputSample = (inputSample * fpNew) - (sin(-inputSample)*fpOld); //lastly, class A clipping on the negative to combat the one-sidedness //uses bloom/antibloom to dial in previous unconstrained behavior //end the whole distortion dealiebop inputSample *= target; //begin simplified Groove Wear, outside the scaling //varies depending on what sample rate you're at: //high sample rate makes it more airy gwB = gwA; gwA = tempSample = (inputSample-gwPrev); tempSample *= gwAfactor; tempSample += (gwB * gwBfactor); correction = (inputSample-gwPrev) - tempSample; gwPrev = inputSample; inputSample -= correction; //simplified Groove Wear. //begin simplified ADClip drySample = inputSample; 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 = drySample; //output dither section if (bitDepth == 3) { //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 } else { //entire Naturalize section used when not on 32 bit out inputSample -= noiseShaping; if (bitDepth == 2) inputSample *= 8388608.0; //go to dither at 24 bit if (bitDepth == 1) inputSample *= 32768.0; //go to dither at 16 bit Float64 benfordize = floor(inputSample); while (benfordize >= 1.0) {benfordize /= 10;} if (benfordize < 1.0) {benfordize *= 10;} if (benfordize < 1.0) {benfordize *= 10;} if (benfordize < 1.0) {benfordize *= 10;} if (benfordize < 1.0) {benfordize *= 10;} if (benfordize < 1.0) {benfordize *= 10;} int hotbinA = floor(benfordize); //hotbin becomes the Benford bin value for this number floored Float64 totalA = 0; if ((hotbinA > 0) && (hotbinA < 10)) { byn[hotbinA] += 1; totalA += (301-byn[1]); totalA += (176-byn[2]); totalA += (125-byn[3]); totalA += (97-byn[4]); totalA += (79-byn[5]); totalA += (67-byn[6]); totalA += (58-byn[7]); totalA += (51-byn[8]); totalA += (46-byn[9]); byn[hotbinA] -= 1; } else {hotbinA = 10;} //produce total number- smaller is closer to Benford real benfordize = ceil(inputSample); while (benfordize >= 1.0) {benfordize /= 10;} if (benfordize < 1.0) {benfordize *= 10;} if (benfordize < 1.0) {benfordize *= 10;} if (benfordize < 1.0) {benfordize *= 10;} if (benfordize < 1.0) {benfordize *= 10;} if (benfordize < 1.0) {benfordize *= 10;} int hotbinB = floor(benfordize); //hotbin becomes the Benford bin value for this number ceiled Float64 totalB = 0; if ((hotbinB > 0) && (hotbinB < 10)) { byn[hotbinB] += 1; totalB += (301-byn[1]); totalB += (176-byn[2]); totalB += (125-byn[3]); totalB += (97-byn[4]); totalB += (79-byn[5]); totalB += (67-byn[6]); totalB += (58-byn[7]); totalB += (51-byn[8]); totalB += (46-byn[9]); byn[hotbinB] -= 1; } else {hotbinB = 10;} //produce total number- smaller is closer to Benford real if (totalA < totalB) { byn[hotbinA] += 1; inputSample = floor(inputSample); } else { byn[hotbinB] += 1; inputSample = ceil(inputSample); } //assign the relevant one to the delay line //and floor/ceil signal accordingly totalA = byn[1] + byn[2] + byn[3] + byn[4] + byn[5] + byn[6] + byn[7] + byn[8] + byn[9]; totalA /= 1000; if (totalA = 0) totalA = 1; byn[1] /= totalA; byn[2] /= totalA; byn[3] /= totalA; byn[4] /= totalA; byn[5] /= totalA; byn[6] /= totalA; byn[7] /= totalA; byn[8] /= totalA; byn[9] /= totalA; byn[10] /= 2; //catchall for garbage data if (bitDepth == 2) inputSample /= 8388608.0; if (bitDepth == 1) inputSample /= 32768.0; noiseShaping += inputSample - drySample; } if (inputSample > refclip) inputSample = refclip; if (inputSample < -refclip) inputSample = -refclip; //iron bar prohibits any overs *destP = inputSample; sourceP += inNumChannels; destP += inNumChannels; } }