/* * File: Monitoring3.cpp * * Version: 1.0 * * Created: 8/18/22 * * Copyright: Copyright © 2022 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. * */ /*============================================================================= Monitoring3.cpp =============================================================================*/ #include "Monitoring3.h" //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ AUDIOCOMPONENT_ENTRY(AUBaseFactory, Monitoring3) //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // Monitoring3::Monitoring3 //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ Monitoring3::Monitoring3(AudioUnit component) : AUEffectBase(component) { CreateElements(); Globals()->UseIndexedParameters(kNumberOfParameters); SetParameter(kParam_One, kDefaultValue_ParamOne ); #if AU_DEBUG_DISPATCHER mDebugDispatcher = new AUDebugDispatcher (this); #endif } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // Monitoring3::GetParameterValueStrings //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult Monitoring3::GetParameterValueStrings(AudioUnitScope inScope, AudioUnitParameterID inParameterID, CFArrayRef * outStrings) { if ((inScope == kAudioUnitScope_Global) && (inParameterID == kParam_One)) //ID must be actual name of parameter identifier, not number { if (outStrings == NULL) return noErr; CFStringRef strings [] = { kMenuItem_DKAD, kMenuItem_DKCD, kMenuItem_PEAK, kMenuItem_SLEW, kMenuItem_SUBS, kMenuItem_MONO, kMenuItem_SIDE, kMenuItem_VINYL, kMenuItem_AURAT, kMenuItem_MONORAT, kMenuItem_MONOLAT, kMenuItem_PHONE, kMenuItem_CANSA, kMenuItem_CANSB, kMenuItem_CANSC, kMenuItem_CANSD, kMenuItem_TRICK }; *outStrings = CFArrayCreate ( NULL, (const void **) strings, (sizeof (strings) / sizeof (strings [0])), NULL ); return noErr; } return kAudioUnitErr_InvalidProperty; } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // Monitoring3::GetParameterInfo //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult Monitoring3::GetParameterInfo(AudioUnitScope inScope, AudioUnitParameterID inParameterID, AudioUnitParameterInfo &outParameterInfo ) { ComponentResult result = noErr; outParameterInfo.flags = kAudioUnitParameterFlag_IsWritable | kAudioUnitParameterFlag_IsReadable; if (inScope == kAudioUnitScope_Global) { switch(inParameterID) { case kParam_One: AUBase::FillInParameterName (outParameterInfo, kParameterOneName, false); outParameterInfo.unit = kAudioUnitParameterUnit_Indexed; outParameterInfo.minValue = kDKAD; outParameterInfo.maxValue = kTRICK; outParameterInfo.defaultValue = kDefaultValue_ParamOne; break; default: result = kAudioUnitErr_InvalidParameter; break; } } else { result = kAudioUnitErr_InvalidParameter; } return result; } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // Monitoring3::GetPropertyInfo //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult Monitoring3::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 Monitoring3::SupportedNumChannels(const AUChannelInfo ** outInfo) { if (outInfo != NULL) { static AUChannelInfo info; info.inChannels = 2; info.outChannels = 2; *outInfo = &info; } return 1; } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // Monitoring3::GetProperty //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult Monitoring3::GetProperty( AudioUnitPropertyID inID, AudioUnitScope inScope, AudioUnitElement inElement, void * outData ) { return AUEffectBase::GetProperty (inID, inScope, inElement, outData); } // Monitoring3::Initialize //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult Monitoring3::Initialize() { ComponentResult result = AUEffectBase::Initialize(); if (result == noErr) Reset(kAudioUnitScope_Global, 0); return result; } #pragma mark ____Monitoring3EffectKernel //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // Monitoring3::Monitoring3Kernel::Reset() //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult Monitoring3::Reset(AudioUnitScope inScope, AudioUnitElement inElement) { NSOddL = 0.0; NSEvenL = 0.0; prevShapeL = 0.0; NSOddR = 0.0; NSEvenR = 0.0; prevShapeR = 0.0; flip = true; //Ten Nines for(int count = 0; count < 99; count++) { darkSampleL[count] = 0; darkSampleR[count] = 0; } for(int count = 0; count < 1502; count++) { aL[count] = 0.0; bL[count] = 0.0; cL[count] = 0.0; dL[count] = 0.0; aR[count] = 0.0; bR[count] = 0.0; cR[count] = 0.0; dR[count] = 0.0; } ax = 1; bx = 1; cx = 1; dx = 1; //PeaksOnly lastSampleL = 0.0; lastSampleR = 0.0; //SlewOnly iirSampleAL = 0.0; iirSampleBL = 0.0; iirSampleCL = 0.0; iirSampleDL = 0.0; iirSampleEL = 0.0; iirSampleFL = 0.0; iirSampleGL = 0.0; iirSampleHL = 0.0; iirSampleIL = 0.0; iirSampleJL = 0.0; iirSampleKL = 0.0; iirSampleLL = 0.0; iirSampleML = 0.0; iirSampleNL = 0.0; iirSampleOL = 0.0; iirSamplePL = 0.0; iirSampleQL = 0.0; iirSampleRL = 0.0; iirSampleSL = 0.0; iirSampleTL = 0.0; iirSampleUL = 0.0; iirSampleVL = 0.0; iirSampleWL = 0.0; iirSampleXL = 0.0; iirSampleYL = 0.0; iirSampleZL = 0.0; iirSampleAR = 0.0; iirSampleBR = 0.0; iirSampleCR = 0.0; iirSampleDR = 0.0; iirSampleER = 0.0; iirSampleFR = 0.0; iirSampleGR = 0.0; iirSampleHR = 0.0; iirSampleIR = 0.0; iirSampleJR = 0.0; iirSampleKR = 0.0; iirSampleLR = 0.0; iirSampleMR = 0.0; iirSampleNR = 0.0; iirSampleOR = 0.0; iirSamplePR = 0.0; iirSampleQR = 0.0; iirSampleRR = 0.0; iirSampleSR = 0.0; iirSampleTR = 0.0; iirSampleUR = 0.0; iirSampleVR = 0.0; iirSampleWR = 0.0; iirSampleXR = 0.0; iirSampleYR = 0.0; iirSampleZR = 0.0; // o/` //SubsOnly for (int x = 0; x < fix_total; x++) {biquad[x] = 0.0;} //Bandpasses fpdL = 1.0; while (fpdL < 16386) fpdL = rand()*UINT32_MAX; fpdR = 1.0; while (fpdR < 16386) fpdR = rand()*UINT32_MAX; return noErr; } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // Monitoring3::ProcessBufferLists //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ OSStatus Monitoring3::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(); depth = (int)(17.0*overallscale); if (depth < 3) depth = 3; if (depth > 98) depth = 98; //Dark int depth = (int)(17.0*overallscale); if (depth < 3) depth = 3; if (depth > 98) depth = 98; //for Dark int processing = (int) GetParameter( kParam_One ); int am = (int)149.0 * overallscale; int bm = (int)179.0 * overallscale; int cm = (int)191.0 * overallscale; int dm = (int)223.0 * overallscale; //these are 'good' primes, spacing out the allpasses int allpasstemp; //for PeaksOnly biquad[fix_freq] = 0.0375/overallscale; biquad[fix_reso] = 0.1575; //define as AURAT, MONORAT, MONOLAT unless overridden if (processing == kVINYL) {biquad[fix_freq] = 0.0385/overallscale; biquad[fix_reso] = 0.0825;} if (processing == kPHONE) {biquad[fix_freq] = 0.1245/overallscale; biquad[fix_reso] = 0.46;} double K = tan(M_PI * biquad[fix_freq]); double norm = 1.0 / (1.0 + K / biquad[fix_reso] + K * K); biquad[fix_a0] = K / biquad[fix_reso] * norm; biquad[fix_a2] = -biquad[fix_a0]; //for bandpass, ignore [fix_a1] = 0.0 biquad[fix_b1] = 2.0 * (K * K - 1.0) * norm; biquad[fix_b2] = (1.0 - K / biquad[fix_reso] + K * K) * norm; //for Bandpasses while (nSampleFrames-- > 0) { double inputSampleL = *inputL; double inputSampleR = *inputR; if (fabs(inputSampleL)<1.18e-23) inputSampleL = fpdL * 1.18e-17; fpdL ^= fpdL << 13; fpdL ^= fpdL >> 17; fpdL ^= fpdL << 5; if (fabs(inputSampleR)<1.18e-23) inputSampleR = fpdR * 1.18e-17; fpdR ^= fpdR << 13; fpdR ^= fpdR >> 17; fpdR ^= fpdR << 5; //we need to make our dither run up here, there's no spot on the end to do it switch (processing) { case kDKAD: case kDKCD: break; case kPEAK: if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; inputSampleL = asin(inputSampleL); if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; inputSampleR = asin(inputSampleR); //amplitude aspect allpasstemp = ax - 1; if (allpasstemp < 0 || allpasstemp > am) allpasstemp = am; inputSampleL -= aL[allpasstemp]*0.5; aL[ax] = inputSampleL; inputSampleL *= 0.5; inputSampleR -= aR[allpasstemp]*0.5; aR[ax] = inputSampleR; inputSampleR *= 0.5; ax--; if (ax < 0 || ax > am) {ax = am;} inputSampleL += (aL[ax]); inputSampleR += (aR[ax]); //a single Midiverb-style allpass if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; inputSampleL = asin(inputSampleL); if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; inputSampleR = asin(inputSampleR); //amplitude aspect allpasstemp = bx - 1; if (allpasstemp < 0 || allpasstemp > bm) allpasstemp = bm; inputSampleL -= bL[allpasstemp]*0.5; bL[bx] = inputSampleL; inputSampleL *= 0.5; inputSampleR -= bR[allpasstemp]*0.5; bR[bx] = inputSampleR; inputSampleR *= 0.5; bx--; if (bx < 0 || bx > bm) {bx = bm;} inputSampleL += (bL[bx]); inputSampleR += (bR[bx]); //a single Midiverb-style allpass if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; inputSampleL = asin(inputSampleL); if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; inputSampleR = asin(inputSampleR); //amplitude aspect allpasstemp = cx - 1; if (allpasstemp < 0 || allpasstemp > cm) allpasstemp = cm; inputSampleL -= cL[allpasstemp]*0.5; cL[cx] = inputSampleL; inputSampleL *= 0.5; inputSampleR -= cR[allpasstemp]*0.5; cR[cx] = inputSampleR; inputSampleR *= 0.5; cx--; if (cx < 0 || cx > cm) {cx = cm;} inputSampleL += (cL[cx]); inputSampleR += (cR[cx]); //a single Midiverb-style allpass if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; inputSampleL = asin(inputSampleL); if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; inputSampleR = asin(inputSampleR); //amplitude aspect allpasstemp = dx - 1; if (allpasstemp < 0 || allpasstemp > dm) allpasstemp = dm; inputSampleL -= dL[allpasstemp]*0.5; dL[dx] = inputSampleL; inputSampleL *= 0.5; inputSampleR -= dR[allpasstemp]*0.5; dR[dx] = inputSampleR; inputSampleR *= 0.5; dx--; if (dx < 0 || dx > dm) {dx = dm;} inputSampleL += (dL[dx]); inputSampleR += (dR[dx]); //a single Midiverb-style allpass if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; inputSampleL = asin(inputSampleL); if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; inputSampleR = asin(inputSampleR); //amplitude aspect inputSampleL *= 0.63679; inputSampleR *= 0.63679; //scale it to 0dB output at full blast //PeaksOnly break; case kSLEW: Float64 trim; trim = 2.302585092994045684017991; //natural logarithm of 10 double slewSample; slewSample = (inputSampleL - lastSampleL)*trim; lastSampleL = inputSampleL; if (slewSample > 1.0) slewSample = 1.0; if (slewSample < -1.0) slewSample = -1.0; inputSampleL = slewSample; slewSample = (inputSampleR - lastSampleR)*trim; lastSampleR = inputSampleR; if (slewSample > 1.0) slewSample = 1.0; if (slewSample < -1.0) slewSample = -1.0; inputSampleR = slewSample; //SlewOnly break; case kSUBS: Float64 iirAmount; iirAmount = (2250/44100.0) / overallscale; Float64 gain; gain = 1.42; inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1; iirSampleAL = (iirSampleAL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleAL; iirSampleAR = (iirSampleAR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleAR; inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1; if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; iirSampleBL = (iirSampleBL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleBL; iirSampleBR = (iirSampleBR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleBR; inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1; if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; iirSampleCL = (iirSampleCL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleCL; iirSampleCR = (iirSampleCR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleCR; inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1; if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; iirSampleDL = (iirSampleDL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleDL; iirSampleDR = (iirSampleDR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleDR; inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1; if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; iirSampleEL = (iirSampleEL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleEL; iirSampleER = (iirSampleER * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleER; inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1; if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; iirSampleFL = (iirSampleFL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleFL; iirSampleFR = (iirSampleFR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleFR; inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1; if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; iirSampleGL = (iirSampleGL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleGL; iirSampleGR = (iirSampleGR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleGR; inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1; if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; iirSampleHL = (iirSampleHL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleHL; iirSampleHR = (iirSampleHR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleHR; inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1; if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; iirSampleIL = (iirSampleIL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleIL; iirSampleIR = (iirSampleIR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleIR; inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1; if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; iirSampleJL = (iirSampleJL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleJL; iirSampleJR = (iirSampleJR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleJR; inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1; if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; iirSampleKL = (iirSampleKL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleKL; iirSampleKR = (iirSampleKR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleKR; inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1; if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; iirSampleLL = (iirSampleLL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleLL; iirSampleLR = (iirSampleLR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleLR; inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1; if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; iirSampleML = (iirSampleML * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleML; iirSampleMR = (iirSampleMR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleMR; inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1; if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; iirSampleNL = (iirSampleNL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleNL; iirSampleNR = (iirSampleNR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleNR; inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1; if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; iirSampleOL = (iirSampleOL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleOL; iirSampleOR = (iirSampleOR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleOR; inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1; if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; iirSamplePL = (iirSamplePL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSamplePL; iirSamplePR = (iirSamplePR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSamplePR; inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1; if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; iirSampleQL = (iirSampleQL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleQL; iirSampleQR = (iirSampleQR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleQR; inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1; if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; iirSampleRL = (iirSampleRL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleRL; iirSampleRR = (iirSampleRR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleRR; inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1; if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; iirSampleSL = (iirSampleSL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleSL; iirSampleSR = (iirSampleSR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleSR; inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1; if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; iirSampleTL = (iirSampleTL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleTL; iirSampleTR = (iirSampleTR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleTR; inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1; if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; iirSampleUL = (iirSampleUL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleUL; iirSampleUR = (iirSampleUR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleUR; inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1; if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; iirSampleVL = (iirSampleVL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleVL; iirSampleVR = (iirSampleVR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleVR; inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1; if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; iirSampleWL = (iirSampleWL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleWL; iirSampleWR = (iirSampleWR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleWR; inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1; if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; iirSampleXL = (iirSampleXL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleXL; iirSampleXR = (iirSampleXR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleXR; inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1; if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; iirSampleYL = (iirSampleYL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleYL; iirSampleYR = (iirSampleYR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleYR; inputSampleL *= gain; inputSampleR *= gain; gain = ((gain-1)*0.75)+1; if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; iirSampleZL = (iirSampleZL * (1.0-iirAmount)) + (inputSampleL * iirAmount); inputSampleL = iirSampleZL; iirSampleZR = (iirSampleZR * (1.0-iirAmount)) + (inputSampleR * iirAmount); inputSampleR = iirSampleZR; if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; //SubsOnly break; case kMONO: case kSIDE: double mid; mid = inputSampleL + inputSampleR; double side; side = inputSampleL - inputSampleR; if (processing < kSIDE) side = 0.0; else mid = 0.0; //mono monitoring, or side-only monitoring inputSampleL = (mid+side)/2.0; inputSampleR = (mid-side)/2.0; break; case kVINYL: case kAURAT: case kMONORAT: case kMONOLAT: case kPHONE: //Bandpass: changes in EQ are up in the variable defining, not here //7 Vinyl, 8 9 10 Aurat, 11 Phone if (processing == kMONORAT) {inputSampleR = (inputSampleL + inputSampleR)*0.5;inputSampleL = 0.0;} if (processing == kMONOLAT) {inputSampleL = (inputSampleL + inputSampleR)*0.5;inputSampleR = 0.0;} if (processing == kPHONE) {double M; M = (inputSampleL + inputSampleR)*0.5; inputSampleL = M;inputSampleR = M;} inputSampleL = sin(inputSampleL); inputSampleR = sin(inputSampleR); //encode Console5: good cleanness double tempSampleL; tempSampleL = (inputSampleL * biquad[fix_a0]) + biquad[fix_sL1]; biquad[fix_sL1] = (-tempSampleL * biquad[fix_b1]) + biquad[fix_sL2]; biquad[fix_sL2] = (inputSampleL * biquad[fix_a2]) - (tempSampleL * biquad[fix_b2]); inputSampleL = tempSampleL; //like mono AU, 7 and 8 store L channel double tempSampleR; tempSampleR = (inputSampleR * biquad[fix_a0]) + biquad[fix_sR1]; biquad[fix_sR1] = (-tempSampleR * biquad[fix_b1]) + biquad[fix_sR2]; biquad[fix_sR2] = (inputSampleR * biquad[fix_a2]) - (tempSampleR * biquad[fix_b2]); inputSampleR = tempSampleR; //note: 9 and 10 store the R channel if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; //without this, you can get a NaN condition where it spits out DC offset at full blast! inputSampleL = asin(inputSampleL); inputSampleR = asin(inputSampleR); //amplitude aspect break; case kCANSA: case kCANSB: case kCANSC: case kCANSD: if (processing == kCANSA) {inputSampleL *= 0.855; inputSampleR *= 0.855;} if (processing == kCANSB) {inputSampleL *= 0.748; inputSampleR *= 0.748;} if (processing == kCANSC) {inputSampleL *= 0.713; inputSampleR *= 0.713;} if (processing == kCANSD) {inputSampleL *= 0.680; inputSampleR *= 0.680;} //we do a volume compensation immediately to gain stage stuff cleanly inputSampleL = sin(inputSampleL); inputSampleR = sin(inputSampleR); double drySampleL; drySampleL = inputSampleL; double drySampleR; drySampleR = inputSampleR; //everything runs 'inside' Console double bass; bass = (processing * processing * 0.00001) / overallscale; //we are using the iir filters from out of SubsOnly mid = inputSampleL + inputSampleR; side = inputSampleL - inputSampleR; iirSampleAL = (iirSampleAL * (1.0 - (bass*0.618))) + (side * bass * 0.618); side = side - iirSampleAL; inputSampleL = (mid+side)/2.0; inputSampleR = (mid-side)/2.0; //bass narrowing filter allpasstemp = ax - 1; if (allpasstemp < 0 || allpasstemp > am) allpasstemp = am; inputSampleL -= aL[allpasstemp]*0.5; aL[ax] = inputSampleL; inputSampleL *= 0.5; inputSampleR -= aR[allpasstemp]*0.5; aR[ax] = inputSampleR; inputSampleR *= 0.5; ax--; if (ax < 0 || ax > am) {ax = am;} inputSampleL += (aL[ax])*0.5; inputSampleR += (aR[ax])*0.5; if (ax == am) {inputSampleL += (aL[0])*0.5; inputSampleR += (aR[0])*0.5;} else {inputSampleL += (aL[ax+1])*0.5; inputSampleR += (aR[ax+1])*0.5;} //a darkened Midiverb-style allpass if (processing == kCANSA) {inputSampleL *= 0.125; inputSampleR *= 0.125;} if (processing == kCANSB) {inputSampleL *= 0.25; inputSampleR *= 0.25;} if (processing == kCANSC) {inputSampleL *= 0.30; inputSampleR *= 0.30;} if (processing == kCANSD) {inputSampleL *= 0.35; inputSampleR *= 0.35;} //Cans A suppresses the crossfeed more, Cans B makes it louder drySampleL += inputSampleR; drySampleR += inputSampleL; //the crossfeed allpasstemp = dx - 1; if (allpasstemp < 0 || allpasstemp > dm) allpasstemp = dm; inputSampleL -= dL[allpasstemp]*0.5; dL[dx] = inputSampleL; inputSampleL *= 0.5; inputSampleR -= dR[allpasstemp]*0.5; dR[dx] = inputSampleR; inputSampleR *= 0.5; dx--; if (dx < 0 || dx > dm) {dx = dm;} inputSampleL += (dL[dx])*0.5; inputSampleR += (dR[dx])*0.5; if (dx == dm) {inputSampleL += (dL[0])*0.5; inputSampleR += (dR[0])*0.5;} else {inputSampleL += (dL[dx+1])*0.5; inputSampleR += (dR[dx+1])*0.5;} //a darkened Midiverb-style allpass, which is stretching the previous one even more inputSampleL *= 0.25; inputSampleR *= 0.25; //for all versions of Cans the second level of bloom is this far down //and, remains on the opposite speaker rather than crossing again to the original side drySampleL += inputSampleR; drySampleR += inputSampleL; //add the crossfeed and very faint extra verbyness inputSampleL = drySampleL; inputSampleR = drySampleR; //and output our can-opened headphone feed mid = inputSampleL + inputSampleR; side = inputSampleL - inputSampleR; iirSampleAR = (iirSampleAR * (1.0 - bass)) + (side * bass); side = side - iirSampleAR; inputSampleL = (mid+side)/2.0; inputSampleR = (mid-side)/2.0; //bass narrowing filter if (inputSampleL > 1.0) inputSampleL = 1.0; if (inputSampleL < -1.0) inputSampleL = -1.0; inputSampleL = asin(inputSampleL); if (inputSampleR > 1.0) inputSampleR = 1.0; if (inputSampleR < -1.0) inputSampleR = -1.0; inputSampleR = asin(inputSampleR); //ConsoleBuss processing break; case kTRICK: double inputSample = (inputSampleL + inputSampleR) * 0.5; inputSampleL = -inputSample; inputSampleR = inputSample; break; } //begin Dark if (processing == kDKCD) { inputSampleL *= 32768.0; //or 16 bit option inputSampleR *= 32768.0; //or 16 bit option } else { inputSampleL *= 8388608.0; //for literally everything else inputSampleR *= 8388608.0; //we will apply the 24 bit Dark } //on the not unreasonable assumption that we are very likely playing back on 24 bit DAC //We are doing it first Left, then Right, because the loops may run faster if //they aren't too jammed full of variables. This means re-running code. //begin left double correction = 0; if (flip) { NSOddL = (NSOddL * 0.9999999999) + prevShapeL; NSEvenL = (NSEvenL * 0.9999999999) - prevShapeL; correction = NSOddL; } else { NSOddL = (NSOddL * 0.9999999999) - prevShapeL; NSEvenL = (NSEvenL * 0.9999999999) + prevShapeL; correction = NSEvenL; } double shapedSampleL = inputSampleL+correction; //end Ten Nines //begin Dark int quantA = floor(shapedSampleL); int quantB = floor(shapedSampleL+1.0); //to do this style of dither, we quantize in either direction and then //do a reconstruction of what the result will be for each choice. //We then evaluate which one we like, and keep a history of what we previously had float expectedSlew = 0; for(int x = 0; x < depth; x++) { expectedSlew += (darkSampleL[x+1] - darkSampleL[x]); } expectedSlew /= depth; //we have an average of all recent slews //we are doing that to voice the thing down into the upper mids a bit //it mustn't just soften the brightest treble, it must smooth high mids too float testA = fabs((darkSampleL[0] - quantA) - expectedSlew); float testB = fabs((darkSampleL[0] - quantB) - expectedSlew); if (testA < testB) inputSampleL = quantA; else inputSampleL = quantB; //select whichever one departs LEAST from the vector of averaged //reconstructed previous final samples. This will force a kind of dithering //as it'll make the output end up as smooth as possible for(int x = depth; x >=0; x--) { darkSampleL[x+1] = darkSampleL[x]; } darkSampleL[0] = inputSampleL; //end Dark left prevShapeL = (floor(shapedSampleL) - inputSampleL)*0.9999999999; //end Ten Nines left //begin right correction = 0; if (flip) { NSOddR = (NSOddR * 0.9999999999) + prevShapeR; NSEvenR = (NSEvenR * 0.9999999999) - prevShapeR; correction = NSOddR; } else { NSOddR = (NSOddR * 0.9999999999) - prevShapeR; NSEvenR = (NSEvenR * 0.9999999999) + prevShapeR; correction = NSEvenR; } double shapedSampleR = inputSampleR+correction; //end Ten Nines //begin Dark quantA = floor(shapedSampleR); quantB = floor(shapedSampleR+1.0); //to do this style of dither, we quantize in either direction and then //do a reconstruction of what the result will be for each choice. //We then evaluate which one we like, and keep a history of what we previously had expectedSlew = 0; for(int x = 0; x < depth; x++) { expectedSlew += (darkSampleR[x+1] - darkSampleR[x]); } expectedSlew /= depth; //we have an average of all recent slews //we are doing that to voice the thing down into the upper mids a bit //it mustn't just soften the brightest treble, it must smooth high mids too testA = fabs((darkSampleR[0] - quantA) - expectedSlew); testB = fabs((darkSampleR[0] - quantB) - expectedSlew); if (testA < testB) inputSampleR = quantA; else inputSampleR = quantB; //select whichever one departs LEAST from the vector of averaged //reconstructed previous final samples. This will force a kind of dithering //as it'll make the output end up as smooth as possible for(int x = depth; x >=0; x--) { darkSampleR[x+1] = darkSampleR[x]; } darkSampleR[0] = inputSampleR; //end Dark right prevShapeR = (floor(shapedSampleR) - inputSampleR)*0.9999999999; //end Ten Nines flip = !flip; if (processing == kDKCD) { inputSampleL /= 32768.0; inputSampleR /= 32768.0; } else { inputSampleL /= 8388608.0; inputSampleR /= 8388608.0; } //does not use 32 bit stereo floating point dither *outputL = inputSampleL; *outputR = inputSampleR; //direct stereo out inputL += 1; inputR += 1; outputL += 1; outputR += 1; } return noErr; }