/* * File: ConsoleX2Channel.cpp * * Version: 1.0 * * Created: 10/7/25 * * Copyright: Copyright © 2025 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. * */ /*============================================================================= ConsoleX2Channel.cpp =============================================================================*/ #include "ConsoleX2Channel.h" //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ COMPONENT_ENTRY(ConsoleX2Channel) //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // ConsoleX2Channel::ConsoleX2Channel //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ConsoleX2Channel::ConsoleX2Channel(AudioUnit component) : AUEffectBase(component) { CreateElements(); Globals()->UseIndexedParameters(kNumberOfParameters); SetParameter(kParam_TRM, kDefaultValue_ParamTRM ); SetParameter(kParam_MOR, kDefaultValue_ParamMOR ); SetParameter(kParam_HIG, kDefaultValue_ParamHIG ); SetParameter(kParam_HMG, kDefaultValue_ParamHMG ); SetParameter(kParam_LMG, kDefaultValue_ParamLMG ); SetParameter(kParam_BSG, kDefaultValue_ParamBSG ); SetParameter(kParam_HIF, kDefaultValue_ParamHIF ); SetParameter(kParam_HMF, kDefaultValue_ParamHMF ); SetParameter(kParam_LMF, kDefaultValue_ParamLMF ); SetParameter(kParam_BSF, kDefaultValue_ParamBSF ); SetParameter(kParam_THR, kDefaultValue_ParamTHR ); SetParameter(kParam_ATK, kDefaultValue_ParamATK ); SetParameter(kParam_RLS, kDefaultValue_ParamRLS ); SetParameter(kParam_GAT, kDefaultValue_ParamGAT ); SetParameter(kParam_LOP, kDefaultValue_ParamLOP ); SetParameter(kParam_HIP, kDefaultValue_ParamHIP ); SetParameter(kParam_PAN, kDefaultValue_ParamPAN ); SetParameter(kParam_FAD, kDefaultValue_ParamFAD ); #if AU_DEBUG_DISPATCHER mDebugDispatcher = new AUDebugDispatcher (this); #endif } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // ConsoleX2Channel::GetParameterValueStrings //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult ConsoleX2Channel::GetParameterValueStrings(AudioUnitScope inScope, AudioUnitParameterID inParameterID, CFArrayRef * outStrings) { return kAudioUnitErr_InvalidProperty; } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // ConsoleX2Channel::GetParameterInfo //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult ConsoleX2Channel::GetParameterInfo(AudioUnitScope inScope, AudioUnitParameterID inParameterID, AudioUnitParameterInfo &outParameterInfo ) { ComponentResult result = noErr; outParameterInfo.flags = kAudioUnitParameterFlag_IsWritable | kAudioUnitParameterFlag_IsReadable; if (inScope == kAudioUnitScope_Global) { switch(inParameterID) { case kParam_TRM: AUBase::FillInParameterName (outParameterInfo, kParameterTRMName, false); outParameterInfo.unit = kAudioUnitParameterUnit_Indexed; outParameterInfo.minValue = 0; outParameterInfo.maxValue = 4; outParameterInfo.defaultValue = kDefaultValue_ParamTRM; break; case kParam_MOR: AUBase::FillInParameterName (outParameterInfo, kParameterMORName, false); outParameterInfo.unit = kAudioUnitParameterUnit_Generic; outParameterInfo.minValue = 0.0; outParameterInfo.maxValue = 1.0; outParameterInfo.defaultValue = kDefaultValue_ParamMOR; break; case kParam_HIG: AUBase::FillInParameterName (outParameterInfo, kParameterHIGName, false); outParameterInfo.unit = kAudioUnitParameterUnit_CustomUnit; outParameterInfo.unitName = kParameterHIGUnit; outParameterInfo.minValue = 0.0; outParameterInfo.maxValue = 1.0; outParameterInfo.defaultValue = kDefaultValue_ParamHIG; break; case kParam_HMG: AUBase::FillInParameterName (outParameterInfo, kParameterHMGName, false); outParameterInfo.unit = kAudioUnitParameterUnit_Generic; outParameterInfo.minValue = 0.0; outParameterInfo.maxValue = 1.0; outParameterInfo.defaultValue = kDefaultValue_ParamHMG; break; case kParam_LMG: AUBase::FillInParameterName (outParameterInfo, kParameterLMGName, false); outParameterInfo.unit = kAudioUnitParameterUnit_Generic; outParameterInfo.minValue = 0.0; outParameterInfo.maxValue = 1.0; outParameterInfo.defaultValue = kDefaultValue_ParamLMG; break; case kParam_BSG: AUBase::FillInParameterName (outParameterInfo, kParameterBSGName, false); outParameterInfo.unit = kAudioUnitParameterUnit_Generic; outParameterInfo.minValue = 0.0; outParameterInfo.maxValue = 1.0; outParameterInfo.defaultValue = kDefaultValue_ParamBSG; break; case kParam_HIF: AUBase::FillInParameterName (outParameterInfo, kParameterHIFName, false); outParameterInfo.unit = kAudioUnitParameterUnit_CustomUnit; outParameterInfo.unitName = kParameterHIFUnit; outParameterInfo.minValue = 0.0; outParameterInfo.maxValue = 1.0; outParameterInfo.defaultValue = kDefaultValue_ParamHIF; break; case kParam_HMF: AUBase::FillInParameterName (outParameterInfo, kParameterHMFName, false); outParameterInfo.unit = kAudioUnitParameterUnit_Generic; outParameterInfo.minValue = 0.0; outParameterInfo.maxValue = 1.0; outParameterInfo.defaultValue = kDefaultValue_ParamHMF; break; case kParam_LMF: AUBase::FillInParameterName (outParameterInfo, kParameterLMFName, false); outParameterInfo.unit = kAudioUnitParameterUnit_Generic; outParameterInfo.minValue = 0.0; outParameterInfo.maxValue = 1.0; outParameterInfo.defaultValue = kDefaultValue_ParamLMF; break; case kParam_BSF: AUBase::FillInParameterName (outParameterInfo, kParameterBSFName, false); outParameterInfo.unit = kAudioUnitParameterUnit_Generic; outParameterInfo.minValue = 0.0; outParameterInfo.maxValue = 1.0; outParameterInfo.defaultValue = kDefaultValue_ParamBSF; break; case kParam_THR: AUBase::FillInParameterName (outParameterInfo, kParameterTHRName, false); outParameterInfo.unit = kAudioUnitParameterUnit_CustomUnit; outParameterInfo.unitName = kParameterTHRUnit; outParameterInfo.minValue = 0.0; outParameterInfo.maxValue = 1.0; outParameterInfo.defaultValue = kDefaultValue_ParamTHR; break; case kParam_ATK: AUBase::FillInParameterName (outParameterInfo, kParameterATKName, false); outParameterInfo.unit = kAudioUnitParameterUnit_Generic; outParameterInfo.minValue = 0.0; outParameterInfo.maxValue = 1.0; outParameterInfo.defaultValue = kDefaultValue_ParamATK; break; case kParam_RLS: AUBase::FillInParameterName (outParameterInfo, kParameterRLSName, false); outParameterInfo.unit = kAudioUnitParameterUnit_Generic; outParameterInfo.minValue = 0.0; outParameterInfo.maxValue = 1.0; outParameterInfo.defaultValue = kDefaultValue_ParamRLS; break; case kParam_GAT: AUBase::FillInParameterName (outParameterInfo, kParameterGATName, false); outParameterInfo.unit = kAudioUnitParameterUnit_Generic; outParameterInfo.minValue = 0.0; outParameterInfo.maxValue = 1.0; outParameterInfo.defaultValue = kDefaultValue_ParamGAT; break; case kParam_LOP: AUBase::FillInParameterName (outParameterInfo, kParameterLOPName, false); outParameterInfo.unit = kAudioUnitParameterUnit_CustomUnit; outParameterInfo.unitName = kParameterLOPUnit; outParameterInfo.minValue = 0.0; outParameterInfo.maxValue = 1.0; outParameterInfo.defaultValue = kDefaultValue_ParamLOP; break; case kParam_HIP: AUBase::FillInParameterName (outParameterInfo, kParameterHIPName, false); outParameterInfo.unit = kAudioUnitParameterUnit_Generic; outParameterInfo.minValue = 0.0; outParameterInfo.maxValue = 1.0; outParameterInfo.defaultValue = kDefaultValue_ParamHIP; break; case kParam_PAN: AUBase::FillInParameterName (outParameterInfo, kParameterPANName, false); outParameterInfo.unit = kAudioUnitParameterUnit_Generic; outParameterInfo.minValue = 0.0; outParameterInfo.maxValue = 1.0; outParameterInfo.defaultValue = kDefaultValue_ParamPAN; break; case kParam_FAD: AUBase::FillInParameterName (outParameterInfo, kParameterFADName, false); outParameterInfo.unit = kAudioUnitParameterUnit_Generic; outParameterInfo.minValue = 0.0; outParameterInfo.maxValue = 1.0; outParameterInfo.defaultValue = kDefaultValue_ParamFAD; break; default: result = kAudioUnitErr_InvalidParameter; break; } } else { result = kAudioUnitErr_InvalidParameter; } return result; } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // ConsoleX2Channel::GetPropertyInfo //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult ConsoleX2Channel::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 ConsoleX2Channel::SupportedNumChannels(const AUChannelInfo ** outInfo) { if (outInfo != NULL) { static AUChannelInfo info; info.inChannels = 2; info.outChannels = 2; *outInfo = &info; } return 1; } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // ConsoleX2Channel::GetProperty //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult ConsoleX2Channel::GetProperty( AudioUnitPropertyID inID, AudioUnitScope inScope, AudioUnitElement inElement, void * outData ) { return AUEffectBase::GetProperty (inID, inScope, inElement, outData); } // ConsoleX2Channel::Initialize //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult ConsoleX2Channel::Initialize() { ComponentResult result = AUEffectBase::Initialize(); if (result == noErr) Reset(kAudioUnitScope_Global, 0); return result; } #pragma mark ____ConsoleX2ChannelEffectKernel //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // ConsoleX2Channel::ConsoleX2ChannelKernel::Reset() //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult ConsoleX2Channel::Reset(AudioUnitScope inScope, AudioUnitElement inElement) { for (int x = 0; x < biq_total; x++) { highA[x] = 0.0; highB[x] = 0.0; highC[x] = 0.0; midA[x] = 0.0; midB[x] = 0.0; midC[x] = 0.0; lowA[x] = 0.0; lowB[x] = 0.0; lowC[x] = 0.0; } highLIIR = 0.0; highRIIR = 0.0; midLIIR = 0.0; midRIIR = 0.0; lowLIIR = 0.0; lowRIIR = 0.0; //SmoothEQ2 for (int x = 0; x < bez_total; x++) bezComp[x] = 0.0; bezComp[bez_cycle] = 1.0; bezMax = 0.0; bezMin = 0.0; bezGate = 2.0; //Dynamics3 for(int count = 0; count < 22; count++) { iirHPositionL[count] = 0.0; iirHAngleL[count] = 0.0; iirHPositionR[count] = 0.0; iirHAngleR[count] = 0.0; } hBypass = false; for(int count = 0; count < 14; count++) { iirLPositionL[count] = 0.0; iirLAngleL[count] = 0.0; iirLPositionR[count] = 0.0; iirLAngleR[count] = 0.0; } lBypass = false; //Cabs2 for(int count = 0; count < dscBuf+2; count++) { dBaL[count] = 0.0; dBaR[count] = 0.0; } dBaPosL = 0.0; dBaPosR = 0.0; dBaXL = 1; dBaXR = 1; //Discontapeity for (int x = 0; x < 33; x++) {avg32L[x] = 0.0; post32L[x] = 0.0; avg32R[x] = 0.0; post32R[x] = 0.0;} for (int x = 0; x < 17; x++) {avg16L[x] = 0.0; post16L[x] = 0.0; avg16R[x] = 0.0; post16R[x] = 0.0;} for (int x = 0; x < 9; x++) {avg8L[x] = 0.0; post8L[x] = 0.0; avg8R[x] = 0.0; post8R[x] = 0.0;} for (int x = 0; x < 5; x++) {avg4L[x] = 0.0; post4L[x] = 0.0; avg4R[x] = 0.0; post4R[x] = 0.0;} for (int x = 0; x < 3; x++) {avg2L[x] = 0.0; post2L[x] = 0.0; avg2R[x] = 0.0; post2R[x] = 0.0;} avgPos = 0; lastDarkL = 0.0; lastDarkR = 0.0; //preTapeHack lFreqA = 1.0; lFreqB = 1.0; hFreqA = 0.0; hFreqB = 0.0; panA = 0.5; panB = 0.5; inTrimA = 0.5; inTrimB = 0.5; fpdL = 1.0; while (fpdL < 16386) fpdL = rand()*UINT32_MAX; fpdR = 1.0; while (fpdR < 16386) fpdR = rand()*UINT32_MAX; return noErr; } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // ConsoleX2Channel::ProcessBufferLists //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ OSStatus ConsoleX2Channel::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 spacing = floor(overallscale*2.0); if (spacing < 2) spacing = 2; if (spacing > 32) spacing = 32; double moreTapeHack = (GetParameter( kParam_MOR )*2.0)+1.0; bool tapehackOff = (GetParameter( kParam_MOR ) == 0.0); switch ((int)GetParameter( kParam_TRM )){ case 0: moreTapeHack *= 0.5; break; case 1: break; case 2: moreTapeHack *= 2.0; break; case 3: moreTapeHack *= 4.0; break; case 4: moreTapeHack *= 8.0; break; } double moreDiscontinuity = fmax(pow(GetParameter( kParam_MOR )*0.42,3.0)*overallscale,0.00001); //Discontapeity double trebleGain = (GetParameter( kParam_HIG )-0.5)*2.0; trebleGain = 1.0+(trebleGain*fabs(trebleGain)*fabs(trebleGain)); double highmidGain = (GetParameter( kParam_HMG )-0.5)*2.0; highmidGain = 1.0+(highmidGain*fabs(highmidGain)*fabs(highmidGain)); double lowmidGain = (GetParameter( kParam_LMG )-0.5)*2.0; lowmidGain = 1.0+(lowmidGain*fabs(lowmidGain)*fabs(lowmidGain)); double bassGain = (GetParameter( kParam_BSG )-0.5)*2.0; bassGain = 1.0+(bassGain*fabs(bassGain)*fabs(bassGain)); double highCoef = 0.0; double midCoef = 0.0; double lowCoef = 0.0; bool eqOff = (trebleGain == 1.0 && highmidGain == 1.0 && lowmidGain == 1.0 && bassGain == 1.0); //we get to completely bypass EQ if we're truly not using it. The mechanics of it mean that //it cancels out to bit-identical anyhow, but we get to skip the calculation if (!eqOff) { double trebleRef = GetParameter( kParam_HIF )-0.5; double highmidRef = GetParameter( kParam_HMF )-0.5; double lowmidRef = GetParameter( kParam_LMF )-0.5; double bassRef = GetParameter( kParam_BSF )-0.5; double highF = 0.75 + ((trebleRef+trebleRef+trebleRef+highmidRef)*0.125); double bassF = 0.25 + ((lowmidRef+bassRef+bassRef+bassRef)*0.125); double midF = (highF*0.5) + (bassF*0.5) + ((highmidRef+lowmidRef)*0.125); double highQ = fmax(fmin(1.0+(highmidRef-trebleRef),4.0),0.125); double midQ = fmax(fmin(1.0+(lowmidRef-highmidRef),4.0),0.125); double lowQ = fmax(fmin(1.0+(bassRef-lowmidRef),4.0),0.125); highA[biq_freq] = ((pow(highF,3)*20000.0)/GetSampleRate()); highC[biq_freq] = highB[biq_freq] = highA[biq_freq] = fmax(fmin(highA[biq_freq],0.4999),0.00025); double highFreq = pow(highF,3)*20000.0; double omega = 2.0*M_PI*(highFreq/GetSampleRate()); double biqK = 2.0-cos(omega); highCoef = -sqrt((biqK*biqK)-1.0)+biqK; highA[biq_reso] = 2.24697960 * highQ; highB[biq_reso] = 0.80193774 * highQ; highC[biq_reso] = 0.55495813 * highQ; midA[biq_freq] = ((pow(midF,3)*20000.0)/GetSampleRate()); midC[biq_freq] = midB[biq_freq] = midA[biq_freq] = fmax(fmin(midA[biq_freq],0.4999),0.00025); double midFreq = pow(midF,3)*20000.0; omega = 2.0*M_PI*(midFreq/GetSampleRate()); biqK = 2.0-cos(omega); midCoef = -sqrt((biqK*biqK)-1.0)+biqK; midA[biq_reso] = 2.24697960 * midQ; midB[biq_reso] = 0.80193774 * midQ; midC[biq_reso] = 0.55495813 * midQ; lowA[biq_freq] = ((pow(bassF,3)*20000.0)/GetSampleRate()); lowC[biq_freq] = lowB[biq_freq] = lowA[biq_freq] = fmax(fmin(lowA[biq_freq],0.4999),0.00025); double lowFreq = pow(bassF,3)*20000.0; omega = 2.0*M_PI*(lowFreq/GetSampleRate()); biqK = 2.0-cos(omega); lowCoef = -sqrt((biqK*biqK)-1.0)+biqK; lowA[biq_reso] = 2.24697960 * lowQ; lowB[biq_reso] = 0.80193774 * lowQ; lowC[biq_reso] = 0.55495813 * lowQ; biqK = tan(M_PI * highA[biq_freq]); double norm = 1.0 / (1.0 + biqK / highA[biq_reso] + biqK * biqK); highA[biq_a0] = biqK * biqK * norm; highA[biq_a1] = 2.0 * highA[biq_a0]; highA[biq_a2] = highA[biq_a0]; highA[biq_b1] = 2.0 * (biqK * biqK - 1.0) * norm; highA[biq_b2] = (1.0 - biqK / highA[biq_reso] + biqK * biqK) * norm; biqK = tan(M_PI * highB[biq_freq]); norm = 1.0 / (1.0 + biqK / highB[biq_reso] + biqK * biqK); highB[biq_a0] = biqK * biqK * norm; highB[biq_a1] = 2.0 * highB[biq_a0]; highB[biq_a2] = highB[biq_a0]; highB[biq_b1] = 2.0 * (biqK * biqK - 1.0) * norm; highB[biq_b2] = (1.0 - biqK / highB[biq_reso] + biqK * biqK) * norm; biqK = tan(M_PI * highC[biq_freq]); norm = 1.0 / (1.0 + biqK / highC[biq_reso] + biqK * biqK); highC[biq_a0] = biqK * biqK * norm; highC[biq_a1] = 2.0 * highC[biq_a0]; highC[biq_a2] = highC[biq_a0]; highC[biq_b1] = 2.0 * (biqK * biqK - 1.0) * norm; highC[biq_b2] = (1.0 - biqK / highC[biq_reso] + biqK * biqK) * norm; biqK = tan(M_PI * midA[biq_freq]); norm = 1.0 / (1.0 + biqK / midA[biq_reso] + biqK * biqK); midA[biq_a0] = biqK * biqK * norm; midA[biq_a1] = 2.0 * midA[biq_a0]; midA[biq_a2] = midA[biq_a0]; midA[biq_b1] = 2.0 * (biqK * biqK - 1.0) * norm; midA[biq_b2] = (1.0 - biqK / midA[biq_reso] + biqK * biqK) * norm; biqK = tan(M_PI * midB[biq_freq]); norm = 1.0 / (1.0 + biqK / midB[biq_reso] + biqK * biqK); midB[biq_a0] = biqK * biqK * norm; midB[biq_a1] = 2.0 * midB[biq_a0]; midB[biq_a2] = midB[biq_a0]; midB[biq_b1] = 2.0 * (biqK * biqK - 1.0) * norm; midB[biq_b2] = (1.0 - biqK / midB[biq_reso] + biqK * biqK) * norm; biqK = tan(M_PI * midC[biq_freq]); norm = 1.0 / (1.0 + biqK / midC[biq_reso] + biqK * biqK); midC[biq_a0] = biqK * biqK * norm; midC[biq_a1] = 2.0 * midC[biq_a0]; midC[biq_a2] = midC[biq_a0]; midC[biq_b1] = 2.0 * (biqK * biqK - 1.0) * norm; midC[biq_b2] = (1.0 - biqK / midC[biq_reso] + biqK * biqK) * norm; biqK = tan(M_PI * lowA[biq_freq]); norm = 1.0 / (1.0 + biqK / lowA[biq_reso] + biqK * biqK); lowA[biq_a0] = biqK * biqK * norm; lowA[biq_a1] = 2.0 * lowA[biq_a0]; lowA[biq_a2] = lowA[biq_a0]; lowA[biq_b1] = 2.0 * (biqK * biqK - 1.0) * norm; lowA[biq_b2] = (1.0 - biqK / lowA[biq_reso] + biqK * biqK) * norm; biqK = tan(M_PI * lowB[biq_freq]); norm = 1.0 / (1.0 + biqK / lowB[biq_reso] + biqK * biqK); lowB[biq_a0] = biqK * biqK * norm; lowB[biq_a1] = 2.0 * lowB[biq_a0]; lowB[biq_a2] = lowB[biq_a0]; lowB[biq_b1] = 2.0 * (biqK * biqK - 1.0) * norm; lowB[biq_b2] = (1.0 - biqK / lowB[biq_reso] + biqK * biqK) * norm; biqK = tan(M_PI * lowC[biq_freq]); norm = 1.0 / (1.0 + biqK / lowC[biq_reso] + biqK * biqK); lowC[biq_a0] = biqK * biqK * norm; lowC[biq_a1] = 2.0 * lowC[biq_a0]; lowC[biq_a2] = lowC[biq_a0]; lowC[biq_b1] = 2.0 * (biqK * biqK - 1.0) * norm; lowC[biq_b2] = (1.0 - biqK / lowC[biq_reso] + biqK * biqK) * norm; } //SmoothEQ2 double bezThresh = pow(1.0-GetParameter( kParam_THR ), 4.0) * 8.0; double bezRez = pow(1.0-GetParameter( kParam_ATK ), 4.0) / overallscale; double sloRez = pow(1.0-GetParameter( kParam_RLS ), 4.0) / overallscale; double gate = pow(GetParameter( kParam_GAT ),4.0); bezRez = fmin(fmax(bezRez,0.0001),1.0); sloRez = fmin(fmax(sloRez,0.0001),1.0); //Dynamics3 lFreqA = lFreqB; lFreqB = pow(fmax(GetParameter( kParam_LOP ),0.002),overallscale); //the lowpass hFreqA = hFreqB; hFreqB = pow(GetParameter( kParam_HIP ),overallscale+2.0); //the highpass //Cabs2 panA = panB; panB = GetParameter( kParam_PAN )*1.57079633; inTrimA = inTrimB; inTrimB = GetParameter( kParam_FAD )*2.0; //Console 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; inputSampleL *= moreTapeHack; inputSampleR *= moreTapeHack; //trim control gets to work even when MORE is off if (!tapehackOff) { double darkSampleL = inputSampleL; double darkSampleR = inputSampleR; if (avgPos > 31) avgPos = 0; if (spacing > 31) { avg32L[avgPos] = darkSampleL; avg32R[avgPos] = darkSampleR; darkSampleL = 0.0; darkSampleR = 0.0; for (int x = 0; x < 32; x++) {darkSampleL += avg32L[x]; darkSampleR += avg32R[x];} darkSampleL /= 32.0; darkSampleR /= 32.0; } if (spacing > 15) { avg16L[avgPos%16] = darkSampleL; avg16R[avgPos%16] = darkSampleR; darkSampleL = 0.0; darkSampleR = 0.0; for (int x = 0; x < 16; x++) {darkSampleL += avg16L[x]; darkSampleR += avg16R[x];} darkSampleL /= 16.0; darkSampleR /= 16.0; } if (spacing > 7) { avg8L[avgPos%8] = darkSampleL; avg8R[avgPos%8] = darkSampleR; darkSampleL = 0.0; darkSampleR = 0.0; for (int x = 0; x < 8; x++) {darkSampleL += avg8L[x]; darkSampleR += avg8R[x];} darkSampleL /= 8.0; darkSampleR /= 8.0; } if (spacing > 3) { avg4L[avgPos%4] = darkSampleL; avg4R[avgPos%4] = darkSampleR; darkSampleL = 0.0; darkSampleR = 0.0; for (int x = 0; x < 4; x++) {darkSampleL += avg4L[x]; darkSampleR += avg4R[x];} darkSampleL /= 4.0; darkSampleR /= 4.0; } if (spacing > 1) { avg2L[avgPos%2] = darkSampleL; avg2R[avgPos%2] = darkSampleR; darkSampleL = 0.0; darkSampleR = 0.0; for (int x = 0; x < 2; x++) {darkSampleL += avg2L[x]; darkSampleR += avg2R[x];} darkSampleL /= 2.0; darkSampleR /= 2.0; } //only update avgPos after the post-distortion filter stage double avgSlewL = fmin(fabs(lastDarkL-inputSampleL)*0.12*overallscale,1.0); avgSlewL = 1.0-(1.0-avgSlewL*1.0-avgSlewL); inputSampleL = (inputSampleL*(1.0-avgSlewL)) + (darkSampleL*avgSlewL); lastDarkL = darkSampleL; double avgSlewR = fmin(fabs(lastDarkR-inputSampleR)*0.12*overallscale,1.0); avgSlewR = 1.0-(1.0-avgSlewR*1.0-avgSlewR); inputSampleR = (inputSampleR*(1.0-avgSlewR)) + (darkSampleR*avgSlewR); lastDarkR = darkSampleR; //begin Discontinuity section inputSampleL *= moreDiscontinuity; dBaL[dBaXL] = inputSampleL; dBaPosL *= 0.5; dBaPosL += fabs((inputSampleL*((inputSampleL*0.25)-0.5))*0.5); dBaPosL = fmin(dBaPosL,1.0); int dBdly = floor(dBaPosL*dscBuf); double dBi = (dBaPosL*dscBuf)-dBdly; inputSampleL = dBaL[dBaXL-dBdly +((dBaXL-dBdly < 0)?dscBuf:0)]*(1.0-dBi); dBdly++; inputSampleL += dBaL[dBaXL-dBdly +((dBaXL-dBdly < 0)?dscBuf:0)]*dBi; dBaXL++; if (dBaXL < 0 || dBaXL >= dscBuf) dBaXL = 0; inputSampleL /= moreDiscontinuity; //end Discontinuity section, begin TapeHack section inputSampleL = fmax(fmin(inputSampleL,2.305929007734908),-2.305929007734908); double addtwo = inputSampleL * inputSampleL; double empower = inputSampleL * addtwo; // inputSampleL to the third power inputSampleL -= (empower / 6.0); empower *= addtwo; // to the fifth power inputSampleL += (empower / 69.0); empower *= addtwo; //seventh inputSampleL -= (empower / 2530.08); empower *= addtwo; //ninth inputSampleL += (empower / 224985.6); empower *= addtwo; //eleventh inputSampleL -= (empower / 9979200.0f); //this is a degenerate form of a Taylor Series to approximate sin() //end TapeHack section //begin Discontinuity section inputSampleR *= moreDiscontinuity; dBaR[dBaXR] = inputSampleR; dBaPosR *= 0.5; dBaPosR += fabs((inputSampleR*((inputSampleR*0.25)-0.5))*0.5); dBaPosR = fmin(dBaPosR,1.0); dBdly = floor(dBaPosR*dscBuf); dBi = (dBaPosR*dscBuf)-dBdly; inputSampleR = dBaR[dBaXR-dBdly +((dBaXR-dBdly < 0)?dscBuf:0)]*(1.0-dBi); dBdly++; inputSampleR += dBaR[dBaXR-dBdly +((dBaXR-dBdly < 0)?dscBuf:0)]*dBi; dBaXR++; if (dBaXR < 0 || dBaXR >= dscBuf) dBaXR = 0; inputSampleR /= moreDiscontinuity; //end Discontinuity section, begin TapeHack section inputSampleR = fmax(fmin(inputSampleR,2.305929007734908),-2.305929007734908); addtwo = inputSampleR * inputSampleR; empower = inputSampleR * addtwo; // inputSampleR to the third power inputSampleR -= (empower / 6.0); empower *= addtwo; // to the fifth power inputSampleR += (empower / 69.0); empower *= addtwo; //seventh inputSampleR -= (empower / 2530.08); empower *= addtwo; //ninth inputSampleR += (empower / 224985.6); empower *= addtwo; //eleventh inputSampleR -= (empower / 9979200.0f); //this is a degenerate form of a Taylor Series to approximate sin() //end TapeHack section //Discontapeity darkSampleL = inputSampleL; darkSampleR = inputSampleR; if (avgPos > 31) avgPos = 0; if (spacing > 31) { post32L[avgPos] = darkSampleL; post32R[avgPos] = darkSampleR; darkSampleL = 0.0; darkSampleR = 0.0; for (int x = 0; x < 32; x++) {darkSampleL += post32L[x]; darkSampleR += post32R[x];} darkSampleL /= 32.0; darkSampleR /= 32.0; } if (spacing > 15) { post16L[avgPos%16] = darkSampleL; post16R[avgPos%16] = darkSampleR; darkSampleL = 0.0; darkSampleR = 0.0; for (int x = 0; x < 16; x++) {darkSampleL += post16L[x]; darkSampleR += post16R[x];} darkSampleL /= 16.0; darkSampleR /= 16.0; } if (spacing > 7) { post8L[avgPos%8] = darkSampleL; post8R[avgPos%8] = darkSampleR; darkSampleL = 0.0; darkSampleR = 0.0; for (int x = 0; x < 8; x++) {darkSampleL += post8L[x]; darkSampleR += post8R[x];} darkSampleL /= 8.0; darkSampleR /= 8.0; } if (spacing > 3) { post4L[avgPos%4] = darkSampleL; post4R[avgPos%4] = darkSampleR; darkSampleL = 0.0; darkSampleR = 0.0; for (int x = 0; x < 4; x++) {darkSampleL += post4L[x]; darkSampleR += post4R[x];} darkSampleL /= 4.0; darkSampleR /= 4.0; } if (spacing > 1) { post2L[avgPos%2] = darkSampleL; post2R[avgPos%2] = darkSampleR; darkSampleL = 0.0; darkSampleR = 0.0; for (int x = 0; x < 2; x++) {darkSampleL += post2L[x]; darkSampleR += post2R[x];} darkSampleL /= 2.0; darkSampleR /= 2.0; } avgPos++; inputSampleL = (inputSampleL*(1.0-avgSlewL)) + (darkSampleL*avgSlewL); inputSampleR = (inputSampleR*(1.0-avgSlewR)) + (darkSampleR*avgSlewR); //use the previously calculated depth of the filter } if (!eqOff) { double trebleL = inputSampleL; double outSample = (trebleL * highA[biq_a0]) + highA[biq_sL1]; highA[biq_sL1] = (trebleL * highA[biq_a1]) - (outSample * highA[biq_b1]) + highA[biq_sL2]; highA[biq_sL2] = (trebleL * highA[biq_a2]) - (outSample * highA[biq_b2]); double highmidL = outSample; trebleL -= highmidL; outSample = (highmidL * midA[biq_a0]) + midA[biq_sL1]; midA[biq_sL1] = (highmidL * midA[biq_a1]) - (outSample * midA[biq_b1]) + midA[biq_sL2]; midA[biq_sL2] = (highmidL * midA[biq_a2]) - (outSample * midA[biq_b2]); double lowmidL = outSample; highmidL -= lowmidL; outSample = (lowmidL * lowA[biq_a0]) + lowA[biq_sL1]; lowA[biq_sL1] = (lowmidL * lowA[biq_a1]) - (outSample * lowA[biq_b1]) + lowA[biq_sL2]; lowA[biq_sL2] = (lowmidL * lowA[biq_a2]) - (outSample * lowA[biq_b2]); double bassL = outSample; lowmidL -= bassL; trebleL = (bassL*bassGain) + (lowmidL*lowmidGain) + (highmidL*highmidGain) + (trebleL*trebleGain); //first stage of three crossovers outSample = (trebleL * highB[biq_a0]) + highB[biq_sL1]; highB[biq_sL1] = (trebleL * highB[biq_a1]) - (outSample * highB[biq_b1]) + highB[biq_sL2]; highB[biq_sL2] = (trebleL * highB[biq_a2]) - (outSample * highB[biq_b2]); highmidL = outSample; trebleL -= highmidL; outSample = (highmidL * midB[biq_a0]) + midB[biq_sL1]; midB[biq_sL1] = (highmidL * midB[biq_a1]) - (outSample * midB[biq_b1]) + midB[biq_sL2]; midB[biq_sL2] = (highmidL * midB[biq_a2]) - (outSample * midB[biq_b2]); lowmidL = outSample; highmidL -= lowmidL; outSample = (lowmidL * lowB[biq_a0]) + lowB[biq_sL1]; lowB[biq_sL1] = (lowmidL * lowB[biq_a1]) - (outSample * lowB[biq_b1]) + lowB[biq_sL2]; lowB[biq_sL2] = (lowmidL * lowB[biq_a2]) - (outSample * lowB[biq_b2]); bassL = outSample; lowmidL -= bassL; trebleL = (bassL*bassGain) + (lowmidL*lowmidGain) + (highmidL*highmidGain) + (trebleL*trebleGain); //second stage of three crossovers outSample = (trebleL * highC[biq_a0]) + highC[biq_sL1]; highC[biq_sL1] = (trebleL * highC[biq_a1]) - (outSample * highC[biq_b1]) + highC[biq_sL2]; highC[biq_sL2] = (trebleL * highC[biq_a2]) - (outSample * highC[biq_b2]); highmidL = outSample; trebleL -= highmidL; outSample = (highmidL * midC[biq_a0]) + midC[biq_sL1]; midC[biq_sL1] = (highmidL * midC[biq_a1]) - (outSample * midC[biq_b1]) + midC[biq_sL2]; midC[biq_sL2] = (highmidL * midC[biq_a2]) - (outSample * midC[biq_b2]); lowmidL = outSample; highmidL -= lowmidL; outSample = (lowmidL * lowC[biq_a0]) + lowC[biq_sL1]; lowC[biq_sL1] = (lowmidL * lowC[biq_a1]) - (outSample * lowC[biq_b1]) + lowC[biq_sL2]; lowC[biq_sL2] = (lowmidL * lowC[biq_a2]) - (outSample * lowC[biq_b2]); bassL = outSample; lowmidL -= bassL; trebleL = (bassL*bassGain) + (lowmidL*lowmidGain) + (highmidL*highmidGain) + (trebleL*trebleGain); //third stage of three crossovers highLIIR = (highLIIR*highCoef) + (trebleL*(1.0-highCoef)); highmidL = highLIIR; trebleL -= highmidL; midLIIR = (midLIIR*midCoef) + (highmidL*(1.0-midCoef)); lowmidL = midLIIR; highmidL -= lowmidL; lowLIIR = (lowLIIR*lowCoef) + (lowmidL*(1.0-lowCoef)); bassL = lowLIIR; lowmidL -= bassL; inputSampleL = (bassL*bassGain) + (lowmidL*lowmidGain) + (highmidL*highmidGain) + (trebleL*trebleGain); //fourth stage of three crossovers is the exponential filters double trebleR = inputSampleR; outSample = (trebleR * highA[biq_a0]) + highA[biq_sR1]; highA[biq_sR1] = (trebleR * highA[biq_a1]) - (outSample * highA[biq_b1]) + highA[biq_sR2]; highA[biq_sR2] = (trebleR * highA[biq_a2]) - (outSample * highA[biq_b2]); double highmidR = outSample; trebleR -= highmidR; outSample = (highmidR * midA[biq_a0]) + midA[biq_sR1]; midA[biq_sR1] = (highmidR * midA[biq_a1]) - (outSample * midA[biq_b1]) + midA[biq_sR2]; midA[biq_sR2] = (highmidR * midA[biq_a2]) - (outSample * midA[biq_b2]); double lowmidR = outSample; highmidR -= lowmidR; outSample = (lowmidR * lowA[biq_a0]) + lowA[biq_sR1]; lowA[biq_sR1] = (lowmidR * lowA[biq_a1]) - (outSample * lowA[biq_b1]) + lowA[biq_sR2]; lowA[biq_sR2] = (lowmidR * lowA[biq_a2]) - (outSample * lowA[biq_b2]); double bassR = outSample; lowmidR -= bassR; trebleR = (bassR*bassGain) + (lowmidR*lowmidGain) + (highmidR*highmidGain) + (trebleR*trebleGain); //first stage of three crossovers outSample = (trebleR * highB[biq_a0]) + highB[biq_sR1]; highB[biq_sR1] = (trebleR * highB[biq_a1]) - (outSample * highB[biq_b1]) + highB[biq_sR2]; highB[biq_sR2] = (trebleR * highB[biq_a2]) - (outSample * highB[biq_b2]); highmidR = outSample; trebleR -= highmidR; outSample = (highmidR * midB[biq_a0]) + midB[biq_sR1]; midB[biq_sR1] = (highmidR * midB[biq_a1]) - (outSample * midB[biq_b1]) + midB[biq_sR2]; midB[biq_sR2] = (highmidR * midB[biq_a2]) - (outSample * midB[biq_b2]); lowmidR = outSample; highmidR -= lowmidR; outSample = (lowmidR * lowB[biq_a0]) + lowB[biq_sR1]; lowB[biq_sR1] = (lowmidR * lowB[biq_a1]) - (outSample * lowB[biq_b1]) + lowB[biq_sR2]; lowB[biq_sR2] = (lowmidR * lowB[biq_a2]) - (outSample * lowB[biq_b2]); bassR = outSample; lowmidR -= bassR; trebleR = (bassR*bassGain) + (lowmidR*lowmidGain) + (highmidR*highmidGain) + (trebleR*trebleGain); //second stage of three crossovers outSample = (trebleR * highC[biq_a0]) + highC[biq_sR1]; highC[biq_sR1] = (trebleR * highC[biq_a1]) - (outSample * highC[biq_b1]) + highC[biq_sR2]; highC[biq_sR2] = (trebleR * highC[biq_a2]) - (outSample * highC[biq_b2]); highmidR = outSample; trebleR -= highmidR; outSample = (highmidR * midC[biq_a0]) + midC[biq_sR1]; midC[biq_sR1] = (highmidR * midC[biq_a1]) - (outSample * midC[biq_b1]) + midC[biq_sR2]; midC[biq_sR2] = (highmidR * midC[biq_a2]) - (outSample * midC[biq_b2]); lowmidR = outSample; highmidR -= lowmidR; outSample = (lowmidR * lowC[biq_a0]) + lowC[biq_sR1]; lowC[biq_sR1] = (lowmidR * lowC[biq_a1]) - (outSample * lowC[biq_b1]) + lowC[biq_sR2]; lowC[biq_sR2] = (lowmidR * lowC[biq_a2]) - (outSample * lowC[biq_b2]); bassR = outSample; lowmidR -= bassR; trebleR = (bassR*bassGain) + (lowmidR*lowmidGain) + (highmidR*highmidGain) + (trebleR*trebleGain); //third stage of three crossovers highRIIR = (highRIIR*highCoef) + (trebleR*(1.0-highCoef)); highmidR = highRIIR; trebleR -= highmidR; midRIIR = (midRIIR*midCoef) + (highmidR*(1.0-midCoef)); lowmidR = midRIIR; highmidR -= lowmidR; lowRIIR = (lowRIIR*lowCoef) + (lowmidR*(1.0-lowCoef)); bassR = lowRIIR; lowmidR -= bassR; inputSampleR = (bassR*bassGain) + (lowmidR*lowmidGain) + (highmidR*highmidGain) + (trebleR*trebleGain); //fourth stage of three crossovers is the exponential filters } //SmoothEQ2 if (bezThresh > 0.0) { if (fmax(fabs(inputSampleL),fabs(inputSampleR)) > gate) bezGate = overallscale/fmin(bezRez,sloRez); else bezGate = bezGate = fmax(0.000001, bezGate-fmin(bezRez,sloRez)); inputSampleL *= (bezThresh+1.0); inputSampleR *= (bezThresh+1.0); double ctrl = fmax(fabs(inputSampleL),fabs(inputSampleR)); bezMax = fmax(bezMax,ctrl); bezMin = fmax(bezMin-sloRez,ctrl); bezComp[bez_cycle] += bezRez; bezComp[bez_Ctrl] += (bezMin * bezRez); if (bezComp[bez_cycle] > 1.0) { if (bezGate < 1.0) bezComp[bez_Ctrl] /= bezGate; bezComp[bez_cycle] -= 1.0; bezComp[bez_C] = bezComp[bez_B]; bezComp[bez_B] = bezComp[bez_A]; bezComp[bez_A] = bezComp[bez_Ctrl]; bezComp[bez_Ctrl] = 0.0; bezMax = 0.0; } double CB = (bezComp[bez_C]*(1.0-bezComp[bez_cycle]))+(bezComp[bez_B]*bezComp[bez_cycle]); double BA = (bezComp[bez_B]*(1.0-bezComp[bez_cycle]))+(bezComp[bez_A]*bezComp[bez_cycle]); double CBA = (bezComp[bez_B]+(CB*(1.0-bezComp[bez_cycle]))+(BA*bezComp[bez_cycle]))*0.5; inputSampleL *= 1.0-(fmin(CBA*bezThresh,1.0)); inputSampleR *= 1.0-(fmin(CBA*bezThresh,1.0)); } else bezComp[bez_Ctrl] = 0.0; //Dynamics3 const double temp = (double)nSampleFrames/inFramesToProcess; const double hFreq = (hFreqA*temp)+(hFreqB*(1.0-temp)); if (hFreq > 0.0) { double lowSampleL = inputSampleL; double lowSampleR = inputSampleR; for(int count = 0; count < 21; count++) { iirHAngleL[count] = (iirHAngleL[count]*(1.0-hFreq))+((lowSampleL-iirHPositionL[count])*hFreq); lowSampleL = ((iirHPositionL[count]+(iirHAngleL[count]*hFreq))*(1.0-hFreq))+(lowSampleL*hFreq); iirHPositionL[count] = ((iirHPositionL[count]+(iirHAngleL[count]*hFreq))*(1.0-hFreq))+(lowSampleL*hFreq); inputSampleL -= (lowSampleL * (1.0/21.0));//left iirHAngleR[count] = (iirHAngleR[count]*(1.0-hFreq))+((lowSampleR-iirHPositionR[count])*hFreq); lowSampleR = ((iirHPositionR[count]+(iirHAngleR[count]*hFreq))*(1.0-hFreq))+(lowSampleR*hFreq); iirHPositionR[count] = ((iirHPositionR[count]+(iirHAngleR[count]*hFreq))*(1.0-hFreq))+(lowSampleR*hFreq); inputSampleR -= (lowSampleR * (1.0/21.0));//right } //the highpass hBypass = false; } else { if (!hBypass) { hBypass = true; for(int count = 0; count < 22; count++) { iirHPositionL[count] = 0.0; iirHAngleL[count] = 0.0; iirHPositionR[count] = 0.0; iirHAngleR[count] = 0.0; } } //blank out highpass if jut switched off } const double lFreq = (lFreqA*temp)+(lFreqB*(1.0-temp)); if (lFreq < 1.0) { for(int count = 0; count < 13; count++) { iirLAngleL[count] = (iirLAngleL[count]*(1.0-lFreq))+((inputSampleL-iirLPositionL[count])*lFreq); inputSampleL = ((iirLPositionL[count]+(iirLAngleL[count]*lFreq))*(1.0-lFreq))+(inputSampleL*lFreq); iirLPositionL[count] = ((iirLPositionL[count]+(iirLAngleL[count]*lFreq))*(1.0-lFreq))+(inputSampleL*lFreq);//left iirLAngleR[count] = (iirLAngleR[count]*(1.0-lFreq))+((inputSampleR-iirLPositionR[count])*lFreq); inputSampleR = ((iirLPositionR[count]+(iirLAngleR[count]*lFreq))*(1.0-lFreq))+(inputSampleR*lFreq); iirLPositionR[count] = ((iirLPositionR[count]+(iirLAngleR[count]*lFreq))*(1.0-lFreq))+(inputSampleR*lFreq);//right } //the lowpass lBypass = false; } else { if (!lBypass) { lBypass = true; for(int count = 0; count < 14; count++) { iirLPositionL[count] = 0.0; iirLAngleL[count] = 0.0; iirLPositionR[count] = 0.0; iirLAngleR[count] = 0.0; } } //blank out lowpass if just switched off } //Cabs2 double gainR = (panA*temp)+(panB*(1.0-temp)); double gainL = 1.57079633-gainR; gainR = sin(gainR); gainL = sin(gainL); double gain = (inTrimA*temp)+(inTrimB*(1.0-temp)); if (gain > 1.0) gain *= gain; if (gain < 1.0) gain = 1.0-pow(1.0-gain,2); inputSampleL = inputSampleL * gainL * gain; inputSampleR = inputSampleR * gainR * gain; //applies pan section, and smoothed fader gain if (inputSampleL > 1.0) inputSampleL = 1.0; else if (inputSampleL > 0.0) inputSampleL = -expm1((log1p(-inputSampleL) * 1.618033988749895)); if (inputSampleL < -1.0) inputSampleL = -1.0; else if (inputSampleL < 0.0) inputSampleL = expm1((log1p(inputSampleL) * 1.618033988749895)); if (inputSampleR > 1.0) inputSampleR = 1.0; else if (inputSampleR > 0.0) inputSampleR = -expm1((log1p(-inputSampleR) * 1.618033988749895)); if (inputSampleR < -1.0) inputSampleR = -1.0; else if (inputSampleR < 0.0) inputSampleR = expm1((log1p(inputSampleR) * 1.618033988749895)); //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; }