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Diffstat (limited to 'plugins/MacAU/Monitoring/Monitoring.cpp')
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diff --git a/plugins/MacAU/Monitoring/Monitoring.cpp b/plugins/MacAU/Monitoring/Monitoring.cpp new file mode 100755 index 0000000..eadf110 --- /dev/null +++ b/plugins/MacAU/Monitoring/Monitoring.cpp @@ -0,0 +1,727 @@ +/* +* File: Monitoring.cpp +* +* Version: 1.0 +* +* Created: 9/2/19 +* +* Copyright: Copyright © 2019 Airwindows, All Rights Reserved +* +* Disclaimer: IMPORTANT: This Apple software is supplied to you by Apple Computer, Inc. ("Apple") in +* consideration of your agreement to the following terms, and your use, installation, modification +* or redistribution of this Apple software constitutes acceptance of these terms. If you do +* not agree with these terms, please do not use, install, modify or redistribute this Apple +* software. +* +* In consideration of your agreement to abide by the following terms, and subject to these terms, +* Apple grants you a personal, non-exclusive license, under Apple's copyrights in this +* original Apple software (the "Apple Software"), to use, reproduce, modify and redistribute the +* Apple Software, with or without modifications, in source and/or binary forms; provided that if you +* redistribute the Apple Software in its entirety and without modifications, you must retain this +* notice and the following text and disclaimers in all such redistributions of the Apple Software. +* Neither the name, trademarks, service marks or logos of Apple Computer, Inc. may be used to +* endorse or promote products derived from the Apple Software without specific prior written +* permission from Apple. Except as expressly stated in this notice, no other rights or +* licenses, express or implied, are granted by Apple herein, including but not limited to any +* patent rights that may be infringed by your derivative works or by other works in which the +* Apple Software may be incorporated. +* +* The Apple Software is provided by Apple on an "AS IS" basis. APPLE MAKES NO WARRANTIES, EXPRESS OR +* IMPLIED, INCLUDING WITHOUT LIMITATION THE IMPLIED WARRANTIES OF NON-INFRINGEMENT, MERCHANTABILITY +* AND FITNESS FOR A PARTICULAR PURPOSE, REGARDING THE APPLE SOFTWARE OR ITS USE AND OPERATION ALONE +* OR IN COMBINATION WITH YOUR PRODUCTS. +* +* IN NO EVENT SHALL APPLE BE LIABLE FOR ANY SPECIAL, INDIRECT, INCIDENTAL OR CONSEQUENTIAL +* DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS +* OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) ARISING IN ANY WAY OUT OF THE USE, +* REPRODUCTION, MODIFICATION AND/OR DISTRIBUTION OF THE APPLE SOFTWARE, HOWEVER CAUSED AND WHETHER +* UNDER THEORY OF CONTRACT, TORT (INCLUDING NEGLIGENCE), STRICT LIABILITY OR OTHERWISE, EVEN +* IF APPLE HAS BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +* +*/ +/*============================================================================= + Monitoring.cpp + +=============================================================================*/ +#include "Monitoring.h" + + +//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +COMPONENT_ENTRY(Monitoring) + + +//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +// Monitoring::Monitoring +//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +Monitoring::Monitoring(AudioUnit component) + : AUEffectBase(component) +{ + CreateElements(); + Globals()->UseIndexedParameters(kNumberOfParameters); + SetParameter(kParam_One, kDefaultValue_ParamOne ); + +#if AU_DEBUG_DISPATCHER + mDebugDispatcher = new AUDebugDispatcher (this); +#endif + +} + + +//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +// Monitoring::GetParameterValueStrings +//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +ComponentResult Monitoring::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_NJAD, + kMenuItem_NJCD, + kMenuItem_PEAK, + kMenuItem_SLEW, + kMenuItem_SUBS, + kMenuItem_MONO, + kMenuItem_SIDE, + kMenuItem_VINYL, + kMenuItem_AURAT, + kMenuItem_PHONE, + kMenuItem_CANSA, + kMenuItem_CANSB + }; + *outStrings = CFArrayCreate ( + NULL, + (const void **) strings, + (sizeof (strings) / sizeof (strings [0])), + NULL + ); + return noErr; + } + return kAudioUnitErr_InvalidProperty; +} + + + +//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +// Monitoring::GetParameterInfo +//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +ComponentResult Monitoring::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 = kNJAD; + outParameterInfo.maxValue = kCANSB; + outParameterInfo.defaultValue = kDefaultValue_ParamOne; + break; + default: + result = kAudioUnitErr_InvalidParameter; + break; + } + } else { + result = kAudioUnitErr_InvalidParameter; + } + + + + return result; +} + +//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +// Monitoring::GetPropertyInfo +//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +ComponentResult Monitoring::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 Monitoring::SupportedNumChannels(const AUChannelInfo ** outInfo) +{ + if (outInfo != NULL) + { + static AUChannelInfo info; + info.inChannels = 2; + info.outChannels = 2; + *outInfo = &info; + } + + return 1; +} + +//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +// Monitoring::GetProperty +//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +ComponentResult Monitoring::GetProperty( AudioUnitPropertyID inID, + AudioUnitScope inScope, + AudioUnitElement inElement, + void * outData ) +{ + return AUEffectBase::GetProperty (inID, inScope, inElement, outData); +} + +// Monitoring::Initialize +//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +ComponentResult Monitoring::Initialize() +{ + ComponentResult result = AUEffectBase::Initialize(); + if (result == noErr) + Reset(kAudioUnitScope_Global, 0); + return result; +} + +#pragma mark ____MonitoringEffectKernel + + + +//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +// Monitoring::MonitoringKernel::Reset() +//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +ComponentResult Monitoring::Reset(AudioUnitScope inScope, AudioUnitElement inElement) +{ + bynL[0] = 1000.0; + bynL[1] = 301.0; + bynL[2] = 176.0; + bynL[3] = 125.0; + bynL[4] = 97.0; + bynL[5] = 79.0; + bynL[6] = 67.0; + bynL[7] = 58.0; + bynL[8] = 51.0; + bynL[9] = 46.0; + bynL[10] = 1000.0; + noiseShapingL = 0.0; + bynR[0] = 1000.0; + bynR[1] = 301.0; + bynR[2] = 176.0; + bynR[3] = 125.0; + bynR[4] = 97.0; + bynR[5] = 79.0; + bynR[6] = 67.0; + bynR[7] = 58.0; + bynR[8] = 51.0; + bynR[9] = 46.0; + bynR[10] = 1000.0; + noiseShapingR = 0.0; + //end NJAD + 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 < 11; x++) {biquad[x] = 0.0;} + //Bandpasses + fpd = 17; + return noErr; +} + +//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +// Monitoring::ProcessBufferLists +//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +OSStatus Monitoring::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; + long double overallscale = 1.0; + overallscale /= 44100.0; + overallscale *= GetSampleRate(); + 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[0] = 0.0385/overallscale; biquad[1] = 0.0825; //define as VINYL unless overridden + if (processing == kAURAT) {biquad[0] = 0.0375/overallscale; biquad[1] = 0.1575;} + if (processing == kPHONE) {biquad[0] = 0.1245/overallscale; biquad[1] = 0.46;} + double K = tan(M_PI * biquad[0]); + double norm = 1.0 / (1.0 + K / biquad[1] + K * K); + biquad[2] = K / biquad[1] * norm; + biquad[4] = -biquad[2]; //for bandpass, ignore [3] = 0.0 + biquad[5] = 2.0 * (K * K - 1.0) * norm; + biquad[6] = (1.0 - K / biquad[1] + K * K) * norm; + //for Bandpasses + + while (nSampleFrames-- > 0) { + long double inputSampleL = *inputL; + long double inputSampleR = *inputR; + if (fabs(inputSampleL)<1.18e-37) inputSampleL = fpd * 1.18e-37; + if (fabs(inputSampleR)<1.18e-37) inputSampleR = fpd * 1.18e-37; + + switch (processing) + { + case 0: + case 1: + break; + case 2: + 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 3: + Float64 trim; + trim = 2.302585092994045684017991; //natural logarithm of 10 + long 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 4: + 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 5: + case 6: + long double mid; mid = inputSampleL + inputSampleR; + long double side; side = inputSampleL - inputSampleR; + if (processing < 6) 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 7: + case 8: + case 9: + //Bandpass: changes in EQ are up in the variable defining, not here + inputSampleL = sin(inputSampleL); inputSampleR = sin(inputSampleR); + //encode Console5: good cleanness + + long double tempSampleL; tempSampleL = (inputSampleL * biquad[2]) + biquad[7]; + biquad[7] = (-tempSampleL * biquad[5]) + biquad[8]; + biquad[8] = (inputSampleL * biquad[4]) - (tempSampleL * biquad[6]); + inputSampleL = tempSampleL; //like mono AU, 7 and 8 store L channel + + long double tempSampleR; tempSampleR = (inputSampleR * biquad[2]) + biquad[9]; + biquad[9] = (-tempSampleR * biquad[5]) + biquad[10]; + biquad[10] = (inputSampleR * biquad[4]) - (tempSampleR * biquad[6]); + 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 10: + case 11: + inputSampleL = sin(inputSampleL); + inputSampleR = sin(inputSampleR); + long double drySampleL; drySampleL = inputSampleL; + long double drySampleR; drySampleR = inputSampleR; //everything runs 'inside' Console + + 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 (processing == 10) {inputSampleL *= 0.125; inputSampleR *= 0.125;} + else {inputSampleL *= 0.25; inputSampleR *= 0.25;} + //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]); + inputSampleR += (dR[dx]); + //a single Midiverb-style allpass, which is stretching the previous one even more + + if (processing == 10) {inputSampleL *= 0.5; inputSampleR *= 0.5;} + else {inputSampleL *= 0.25; inputSampleR *= 0.25;} + //Cans A already had crossfeeds down, bloom is louder. Cans B sits on bloom more + + drySampleL += inputSampleL; + drySampleR += inputSampleR; //add the crossfeed and very faint extra verbyness + + inputSampleL = drySampleL; + inputSampleR = drySampleR; //and output our can-opened headphone feed + + 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; + } + + + //begin Not Just Another Dither + if (processing == 1) { + inputSampleL = inputSampleL * 32768.0; //or 16 bit option + inputSampleR = inputSampleR * 32768.0; //or 16 bit option + } else { + inputSampleL = inputSampleL * 8388608.0; //for literally everything else + inputSampleR = inputSampleR * 8388608.0; //we will apply the 24 bit NJAD + } //on the not unreasonable assumption that we are very likely playing back on 24 bit DAC + //if we're not, then all we did was apply a Benford Realness function at 24 bits down. + + bool cutbinsL; cutbinsL = false; + bool cutbinsR; cutbinsR = false; + long double drySampleL; drySampleL = inputSampleL; + long double drySampleR; drySampleR = inputSampleR; + inputSampleL -= noiseShapingL; + inputSampleR -= noiseShapingR; + //NJAD L + long double benfordize; benfordize = floor(inputSampleL); + while (benfordize >= 1.0) benfordize /= 10; + while (benfordize < 1.0 && benfordize > 0.0000001) benfordize *= 10; + int hotbinA; hotbinA = floor(benfordize); + //hotbin becomes the Benford bin value for this number floored + long double totalA; totalA = 0; + if ((hotbinA > 0) && (hotbinA < 10)) + { + bynL[hotbinA] += 1; if (bynL[hotbinA] > 982) cutbinsL = true; + totalA += (301-bynL[1]); totalA += (176-bynL[2]); totalA += (125-bynL[3]); + totalA += (97-bynL[4]); totalA += (79-bynL[5]); totalA += (67-bynL[6]); + totalA += (58-bynL[7]); totalA += (51-bynL[8]); totalA += (46-bynL[9]); bynL[hotbinA] -= 1; + } else hotbinA = 10; + //produce total number- smaller is closer to Benford real + benfordize = ceil(inputSampleL); + while (benfordize >= 1.0) benfordize /= 10; + while (benfordize < 1.0 && benfordize > 0.0000001) benfordize *= 10; + int hotbinB; hotbinB = floor(benfordize); + //hotbin becomes the Benford bin value for this number ceiled + long double totalB; totalB = 0; + if ((hotbinB > 0) && (hotbinB < 10)) + { + bynL[hotbinB] += 1; if (bynL[hotbinB] > 982) cutbinsL = true; + totalB += (301-bynL[1]); totalB += (176-bynL[2]); totalB += (125-bynL[3]); + totalB += (97-bynL[4]); totalB += (79-bynL[5]); totalB += (67-bynL[6]); + totalB += (58-bynL[7]); totalB += (51-bynL[8]); totalB += (46-bynL[9]); bynL[hotbinB] -= 1; + } else hotbinB = 10; + //produce total number- smaller is closer to Benford real + long double outputSample; + if (totalA < totalB) {bynL[hotbinA] += 1; outputSample = floor(inputSampleL);} + else {bynL[hotbinB] += 1; outputSample = floor(inputSampleL+1);} + //assign the relevant one to the delay line + //and floor/ceil signal accordingly + if (cutbinsL) { + bynL[1] *= 0.99; bynL[2] *= 0.99; bynL[3] *= 0.99; bynL[4] *= 0.99; bynL[5] *= 0.99; + bynL[6] *= 0.99; bynL[7] *= 0.99; bynL[8] *= 0.99; bynL[9] *= 0.99; bynL[10] *= 0.99; + } + noiseShapingL += outputSample - drySampleL; + if (noiseShapingL > fabs(inputSampleL)) noiseShapingL = fabs(inputSampleL); + if (noiseShapingL < -fabs(inputSampleL)) noiseShapingL = -fabs(inputSampleL); + if (processing == 1) inputSampleL = outputSample / 32768.0; + else inputSampleL = outputSample / 8388608.0; + if (inputSampleL > 1.0) inputSampleL = 1.0; + if (inputSampleL < -1.0) inputSampleL = -1.0; + //finished NJAD L + + //NJAD R + benfordize = floor(inputSampleR); + while (benfordize >= 1.0) benfordize /= 10; + while (benfordize < 1.0 && benfordize > 0.0000001) benfordize *= 10; + hotbinA = floor(benfordize); + //hotbin becomes the Benford bin value for this number floored + totalA = 0; + if ((hotbinA > 0) && (hotbinA < 10)) + { + bynR[hotbinA] += 1; if (bynR[hotbinA] > 982) cutbinsR = true; + totalA += (301-bynR[1]); totalA += (176-bynR[2]); totalA += (125-bynR[3]); + totalA += (97-bynR[4]); totalA += (79-bynR[5]); totalA += (67-bynR[6]); + totalA += (58-bynR[7]); totalA += (51-bynR[8]); totalA += (46-bynR[9]); bynR[hotbinA] -= 1; + } else hotbinA = 10; + //produce total number- smaller is closer to Benford real + benfordize = ceil(inputSampleR); + while (benfordize >= 1.0) benfordize /= 10; + while (benfordize < 1.0 && benfordize > 0.0000001) benfordize *= 10; + hotbinB = floor(benfordize); + //hotbin becomes the Benford bin value for this number ceiled + totalB = 0; + if ((hotbinB > 0) && (hotbinB < 10)) + { + bynR[hotbinB] += 1; if (bynR[hotbinB] > 982) cutbinsR = true; + totalB += (301-bynR[1]); totalB += (176-bynR[2]); totalB += (125-bynR[3]); + totalB += (97-bynR[4]); totalB += (79-bynR[5]); totalB += (67-bynR[6]); + totalB += (58-bynR[7]); totalB += (51-bynR[8]); totalB += (46-bynR[9]); bynR[hotbinB] -= 1; + } else hotbinB = 10; + //produce total number- smaller is closer to Benford real + if (totalA < totalB) {bynR[hotbinA] += 1; outputSample = floor(inputSampleR);} + else {bynR[hotbinB] += 1; outputSample = floor(inputSampleR+1);} + //assign the relevant one to the delay line + //and floor/ceil signal accordingly + if (cutbinsR) { + bynR[1] *= 0.99; bynR[2] *= 0.99; bynR[3] *= 0.99; bynR[4] *= 0.99; bynR[5] *= 0.99; + bynR[6] *= 0.99; bynR[7] *= 0.99; bynR[8] *= 0.99; bynR[9] *= 0.99; bynR[10] *= 0.99; + } + noiseShapingR += outputSample - drySampleR; + if (noiseShapingR > fabs(inputSampleR)) noiseShapingR = fabs(inputSampleR); + if (noiseShapingR < -fabs(inputSampleR)) noiseShapingR = -fabs(inputSampleR); + if (processing == 1) inputSampleR = outputSample / 32768.0; + else inputSampleR = outputSample / 8388608.0; + if (inputSampleR > 1.0) inputSampleR = 1.0; + if (inputSampleR < -1.0) inputSampleR = -1.0; + //finished NJAD R + + //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; +} + |