/* * File: ConsoleLAChannel.cpp * * Version: 1.0 * * Created: 12/4/23 * * Copyright: Copyright © 2023 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. 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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. * */ /*============================================================================= ConsoleLAChannel.cpp =============================================================================*/ #include "ConsoleLAChannel.h" //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ AUDIOCOMPONENT_ENTRY(AUBaseFactory, ConsoleLAChannel) //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // ConsoleLAChannel::ConsoleLAChannel //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ConsoleLAChannel::ConsoleLAChannel(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 ); SetParameter(kParam_Five, kDefaultValue_ParamFive ); #if AU_DEBUG_DISPATCHER mDebugDispatcher = new AUDebugDispatcher (this); #endif } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // ConsoleLAChannel::GetParameterValueStrings //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult ConsoleLAChannel::GetParameterValueStrings(AudioUnitScope inScope, AudioUnitParameterID inParameterID, CFArrayRef * outStrings) { return kAudioUnitErr_InvalidProperty; } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // ConsoleLAChannel::GetParameterInfo //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult ConsoleLAChannel::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; case kParam_Four: AUBase::FillInParameterName (outParameterInfo, kParameterFourName, false); outParameterInfo.unit = kAudioUnitParameterUnit_Generic; outParameterInfo.minValue = 0.0; outParameterInfo.maxValue = 1.0; outParameterInfo.defaultValue = kDefaultValue_ParamFour; break; case kParam_Five: AUBase::FillInParameterName (outParameterInfo, kParameterFiveName, false); outParameterInfo.unit = kAudioUnitParameterUnit_Generic; outParameterInfo.minValue = 0.0; outParameterInfo.maxValue = 1.0; outParameterInfo.defaultValue = kDefaultValue_ParamFive; break; default: result = kAudioUnitErr_InvalidParameter; break; } } else { result = kAudioUnitErr_InvalidParameter; } return result; } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // ConsoleLAChannel::GetPropertyInfo //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult ConsoleLAChannel::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 ConsoleLAChannel::SupportedNumChannels(const AUChannelInfo ** outInfo) { if (outInfo != NULL) { static AUChannelInfo info; info.inChannels = 2; info.outChannels = 2; *outInfo = &info; } return 1; } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // ConsoleLAChannel::GetProperty //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult ConsoleLAChannel::GetProperty( AudioUnitPropertyID inID, AudioUnitScope inScope, AudioUnitElement inElement, void * outData ) { return AUEffectBase::GetProperty (inID, inScope, inElement, outData); } // ConsoleLAChannel::Initialize //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult ConsoleLAChannel::Initialize() { ComponentResult result = AUEffectBase::Initialize(); if (result == noErr) Reset(kAudioUnitScope_Global, 0); return result; } #pragma mark ____ConsoleLAChannelEffectKernel //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // ConsoleLAChannel::ConsoleLAChannelKernel::Reset() //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult ConsoleLAChannel::Reset(AudioUnitScope inScope, AudioUnitElement inElement) { for(int count = 0; count < 222; count++) {hullL[count] = 0.0; hullR[count] = 0.0;} hullp = 1; for (int x = 0; x < 21; x++) pearB[x] = 0.0; subAL = subAR = subBL = subBR = subCL = subCR = 0.0; midA = midB = 0.0; bassA = bassB = 0.0; gainA = gainB = 1.0; fpdL = 1.0; while (fpdL < 16386) fpdL = rand()*UINT32_MAX; fpdR = 1.0; while (fpdR < 16386) fpdR = rand()*UINT32_MAX; return noErr; } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // ConsoleLAChannel::ProcessBufferLists //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ OSStatus ConsoleLAChannel::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(); //will be over 1.0848 when over 48k int cycleEnd = floor(overallscale); if (cycleEnd < 1) cycleEnd = 1; if (cycleEnd > 4) cycleEnd = 4; int limit = 4*cycleEnd; double divisor = 2.0/limit; double treble = (GetParameter( kParam_One )*6.0)-3.0; midA = midB; midB = (GetParameter( kParam_Two )*6.0)-3.0; bassA = bassB; bassB = (GetParameter( kParam_Three )*6.0)-3.0; //these should stack to go up to -3.0 to 3.0 if (treble < 0.0) treble /= 3.0; if (midB < 0.0) midB /= 3.0; if (bassB < 0.0) bassB /= 3.0; //and then become -1.0 to 3.0; //there will be successive sin/cos stages w. dry/wet in these double freqMid = 0.024; switch (cycleEnd) { case 1: //base sample rate, no change break; case 2: //96k tier freqMid = 0.012; break; case 3: //192k tier freqMid = 0.006; break; } int bitshiftL = 0; int bitshiftR = 0; double panControl = (GetParameter( kParam_Four )*2.0)-1.0; //-1.0 to 1.0 double panAttenuation = (1.0-fabs(panControl)); int panBits = 20; //start centered if (panAttenuation > 0.0) panBits = floor(1.0 / panAttenuation); if (panControl > 0.25) bitshiftL += panBits; if (panControl < -0.25) bitshiftR += panBits; if (bitshiftL < 0) bitshiftL = 0; if (bitshiftL > 17) bitshiftL = 17; if (bitshiftR < 0) bitshiftR = 0; if (bitshiftR > 17) bitshiftR = 17; double gainL = 1.0; double gainR = 1.0; switch (bitshiftL) { case 17: gainL = 0.0; break; case 16: gainL = 0.0000152587890625; break; case 15: gainL = 0.000030517578125; break; case 14: gainL = 0.00006103515625; break; case 13: gainL = 0.0001220703125; break; case 12: gainL = 0.000244140625; break; case 11: gainL = 0.00048828125; break; case 10: gainL = 0.0009765625; break; case 9: gainL = 0.001953125; break; case 8: gainL = 0.00390625; break; case 7: gainL = 0.0078125; break; case 6: gainL = 0.015625; break; case 5: gainL = 0.03125; break; case 4: gainL = 0.0625; break; case 3: gainL = 0.125; break; case 2: gainL = 0.25; break; case 1: gainL = 0.5; break; case 0: break; } switch (bitshiftR) { case 17: gainR = 0.0; break; case 16: gainR = 0.0000152587890625; break; case 15: gainR = 0.000030517578125; break; case 14: gainR = 0.00006103515625; break; case 13: gainR = 0.0001220703125; break; case 12: gainR = 0.000244140625; break; case 11: gainR = 0.00048828125; break; case 10: gainR = 0.0009765625; break; case 9: gainR = 0.001953125; break; case 8: gainR = 0.00390625; break; case 7: gainR = 0.0078125; break; case 6: gainR = 0.015625; break; case 5: gainR = 0.03125; break; case 4: gainR = 0.0625; break; case 3: gainR = 0.125; break; case 2: gainR = 0.25; break; case 1: gainR = 0.5; break; case 0: break; } gainA = gainB; gainB = GetParameter( kParam_Five )*2.0; //smoothed master fader from Z2 filters //BitShiftGain pre gain trim goes here double subTrim = 0.0011 / overallscale; 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; double temp = (double)nSampleFrames/inFramesToProcess; double gain = (gainA*temp)+(gainB*(1.0-temp)); double mid = (midA*temp)+(midB*(1.0-temp)); double bass = (bassA*temp)+(bassB*(1.0-temp)); //begin Hull2 Treble hullp--; if (hullp < 0) hullp += 60; hullL[hullp] = hullL[hullp+60] = inputSampleL; hullR[hullp] = hullR[hullp+60] = inputSampleR; int x = hullp; double bassL = 0.0; double bassR = 0.0; while (x < hullp+(limit/2)) { bassL += hullL[x] * divisor; bassR += hullR[x] * divisor; x++; } bassL += bassL * 0.125; bassR += bassR * 0.125; while (x < hullp+limit) { bassL -= hullL[x] * 0.125 * divisor; bassR -= hullR[x] * 0.125 * divisor; x++; } hullL[hullp+20] = hullL[hullp+80] = bassL; hullR[hullp+20] = hullR[hullp+80] = bassR; x = hullp+20; bassL = bassR = 0.0; while (x < hullp+20+(limit/2)) { bassL += hullL[x] * divisor; bassR += hullR[x] * divisor; x++; } bassL += bassL * 0.125; bassR += bassR * 0.125; while (x < hullp+20+limit) { bassL -= hullL[x] * 0.125 * divisor; bassR -= hullR[x] * 0.125 * divisor; x++; } hullL[hullp+40] = hullL[hullp+100] = bassL; hullR[hullp+40] = hullR[hullp+100] = bassR; x = hullp+40; bassL = bassR = 0.0; while (x < hullp+40+(limit/2)) { bassL += hullL[x] * divisor; bassR += hullR[x] * divisor; x++; } bassL += bassL * 0.125; bassR += bassR * 0.125; while (x < hullp+40+limit) { bassL -= hullL[x] * 0.125 * divisor; bassR -= hullR[x] * 0.125 * divisor; x++; } double trebleL = inputSampleL - bassL; inputSampleL = bassL; double trebleR = inputSampleR - bassR; inputSampleR = bassR; //end Hull2 treble //begin Pear filter stages //at this point 'bass' is actually still mid and bass double slew = ((bassL - pearB[0]) + pearB[1])*freqMid*0.5; pearB[0] = bassL = (freqMid * bassL) + ((1.0-freqMid) * (pearB[0] + pearB[1])); pearB[1] = slew; slew = ((bassR - pearB[2]) + pearB[3])*freqMid*0.5; pearB[2] = bassR = (freqMid * bassR) + ((1.0-freqMid) * (pearB[2] + pearB[3])); pearB[3] = slew; slew = ((bassL - pearB[4]) + pearB[5])*freqMid*0.5; pearB[4] = bassL = (freqMid * bassL) + ((1.0-freqMid) * (pearB[4] + pearB[5])); pearB[5] = slew; slew = ((bassR - pearB[6]) + pearB[7])*freqMid*0.5; pearB[6] = bassR = (freqMid * bassR) + ((1.0-freqMid) * (pearB[6] + pearB[7])); pearB[7] = slew; slew = ((bassL - pearB[8]) + pearB[9])*freqMid*0.5; pearB[8] = bassL = (freqMid * bassL) + ((1.0-freqMid) * (pearB[8] + pearB[9])); pearB[9] = slew; slew = ((bassR - pearB[10]) + pearB[11])*freqMid*0.5; pearB[10] = bassR = (freqMid * bassR) + ((1.0-freqMid) * (pearB[10] + pearB[11])); pearB[11] = slew; slew = ((bassL - pearB[12]) + pearB[13])*freqMid*0.5; pearB[12] = bassL = (freqMid * bassL) + ((1.0-freqMid) * (pearB[12] + pearB[13])); pearB[13] = slew; slew = ((bassR - pearB[14]) + pearB[15])*freqMid*0.5; pearB[14] = bassR = (freqMid * bassR) + ((1.0-freqMid) * (pearB[14] + pearB[15])); pearB[15] = slew; slew = ((bassL - pearB[16]) + pearB[17])*freqMid*0.5; pearB[16] = bassL = (freqMid * bassL) + ((1.0-freqMid) * (pearB[16] + pearB[17])); pearB[17] = slew; slew = ((bassR - pearB[18]) + pearB[19])*freqMid*0.5; pearB[18] = bassR = (freqMid * bassR) + ((1.0-freqMid) * (pearB[18] + pearB[19])); pearB[19] = slew; double midL = inputSampleL - bassL; double midR = inputSampleR - bassR; //we now have three bands out of hull and pear filters double w = 0.0; //filter into bands, apply the sin/cos to each band if (treble > 0.0) { w = treble; if (w > 1.0) w = 1.0; trebleL = (trebleL*(1.0-w)) + (sin(trebleL*M_PI_2)*treble); trebleR = (trebleR*(1.0-w)) + (sin(trebleR*M_PI_2)*treble); } if (treble < 0.0) { if (trebleL > 1.0) trebleL = 1.0; if (trebleL < -1.0) trebleL = -1.0; if (trebleR > 1.0) trebleR = 1.0; if (trebleR < -1.0) trebleR = -1.0; w = -treble; if (w > 1.0) w = 1.0; if (trebleL > 0) trebleL = (trebleL*(1.0-w))+((1.0-cos(trebleL*w))*(1.0-w)); else trebleL = (trebleL*(1.0-w))+((-1.0+cos(-trebleL*w))*(1.0-w)); if (trebleR > 0) trebleR = (trebleR*(1.0-w))+((1.0-cos(trebleR*w))*(1.0-w)); else trebleR = (trebleR*(1.0-w))+((-1.0+cos(-trebleR*w))*(1.0-w)); } //cosine stages for EQ or expansion if (midL > 1.0) midL = 1.0; if (midL < -1.0) midL = -1.0; if (midR > 1.0) midR = 1.0; if (midR < -1.0) midR = -1.0; if (mid > 0.0) { w = mid; if (w > 1.0) w = 1.0; midL = (midL*(1.0-w)) + (sin(midL*M_PI_2)*mid); midR = (midR*(1.0-w)) + (sin(midR*M_PI_2)*mid); } if (mid < 0.0) { w = -mid; if (w > 1.0) w = 1.0; if (midL > 0) midL = (midL*(1.0-w))+((1.0-cos(midL*w))*(1.0-w)); else midL = (midL*(1.0-w))+((-1.0+cos(-midL*w))*(1.0-w)); if (midR > 0) midR = (midR*(1.0-w))+((1.0-cos(midR*w))*(1.0-w)); else midR = (midR*(1.0-w))+((-1.0+cos(-midR*w))*(1.0-w)); } //cosine stages for EQ or expansion if (bassL > 1.0) bassL = 1.0; if (bassL < -1.0) bassL = -1.0; if (bassR > 1.0) bassR = 1.0; if (bassR < -1.0) bassR = -1.0; if (bass > 0.0) { w = bass; if (w > 1.0) w = 1.0; bassL = (bassL*(1.0-w)) + (sin(bassL*M_PI_2)*bass); bassR = (bassR*(1.0-w)) + (sin(bassR*M_PI_2)*bass); } if (bass < 0.0) { w = -bass; if (w > 1.0) w = 1.0; if (bassL > 0) bassL = (bassL*(1.0-w))+((1.0-cos(bassL*w))*(1.0-w)); else bassL = (bassL*(1.0-w))+((-1.0+cos(-bassL*w))*(1.0-w)); if (bassR > 0) bassR = (bassR*(1.0-w))+((1.0-cos(bassR*w))*(1.0-w)); else bassR = (bassR*(1.0-w))+((-1.0+cos(-bassR*w))*(1.0-w)); } //cosine stages for EQ or expansion inputSampleL = (bassL + midL + trebleL)*gainL*gain; inputSampleR = (bassR + midR + trebleR)*gainR*gain; //applies BitShiftPan pan section, and smoothed fader gain //begin SubTight section double subSampleL = inputSampleL * subTrim; double subSampleR = inputSampleR * subTrim; double scale = 0.5+fabs(subSampleL*0.5); subSampleL = (subAL+(sin(subAL-subSampleL)*scale)); subAL = subSampleL*scale; scale = 0.5+fabs(subSampleR*0.5); subSampleR = (subAR+(sin(subAR-subSampleR)*scale)); subAR = subSampleR*scale; scale = 0.5+fabs(subSampleL*0.5); subSampleL = (subBL+(sin(subBL-subSampleL)*scale)); subBL = subSampleL*scale; scale = 0.5+fabs(subSampleR*0.5); subSampleR = (subBR+(sin(subBR-subSampleR)*scale)); subBR = subSampleR*scale; scale = 0.5+fabs(subSampleL*0.5); subSampleL = (subCL+(sin(subCL-subSampleL)*scale)); subCL = subSampleL*scale; scale = 0.5+fabs(subSampleR*0.5); subSampleR = (subCR+(sin(subCR-subSampleR)*scale)); subCR = subSampleR*scale; if (subSampleL > 0.25) subSampleL = 0.25; if (subSampleL < -0.25) subSampleL = -0.25; if (subSampleR > 0.25) subSampleR = 0.25; if (subSampleR < -0.25) subSampleR = -0.25; inputSampleL += (subSampleL*16.0); inputSampleR += (subSampleR*16.0); //end SubTight section //begin Console7 Channel processing if (inputSampleL > 1.097) inputSampleL = 1.097; if (inputSampleL < -1.097) inputSampleL = -1.097; if (inputSampleR > 1.097) inputSampleR = 1.097; if (inputSampleR < -1.097) inputSampleR = -1.097; inputSampleL = ((sin(inputSampleL*fabs(inputSampleL))/((fabs(inputSampleL) == 0.0) ?1:fabs(inputSampleL)))*0.8)+(sin(inputSampleL)*0.2); inputSampleR = ((sin(inputSampleR*fabs(inputSampleR))/((fabs(inputSampleR) == 0.0) ?1:fabs(inputSampleR)))*0.8)+(sin(inputSampleR)*0.2); //this is a version of Spiral blended 80/20 with regular Density. //It's blending between two different harmonics in the overtones of the algorithm //begin 32 bit stereo floating point dither int expon; frexpf((float)inputSampleL, &expon); fpdL ^= fpdL << 13; fpdL ^= fpdL >> 17; fpdL ^= fpdL << 5; inputSampleL += ((double(fpdL)-uint32_t(0x7fffffff)) * 5.5e-36l * pow(2,expon+62)); frexpf((float)inputSampleR, &expon); fpdR ^= fpdR << 13; fpdR ^= fpdR >> 17; fpdR ^= fpdR << 5; inputSampleR += ((double(fpdR)-uint32_t(0x7fffffff)) * 5.5e-36l * pow(2,expon+62)); //end 32 bit stereo floating point dither *outputL = inputSampleL; *outputR = inputSampleR; //direct stereo out inputL += 1; inputR += 1; outputL += 1; outputR += 1; } return noErr; }