/* * File: Thunder.cpp * * Version: 1.0 * * Created: 9/19/16 * * Copyright: Copyright © 2016 Airwindows, All Rights Reserved * * Disclaimer: IMPORTANT: This Apple software is supplied to you by Apple Computer, Inc. ("Apple") in * consideration of your agreement to the following terms, and your use, installation, modification * or redistribution of this Apple software constitutes acceptance of these terms. 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APPLE MAKES NO WARRANTIES, EXPRESS OR * IMPLIED, INCLUDING WITHOUT LIMITATION THE IMPLIED WARRANTIES OF NON-INFRINGEMENT, MERCHANTABILITY * AND FITNESS FOR A PARTICULAR PURPOSE, REGARDING THE APPLE SOFTWARE OR ITS USE AND OPERATION ALONE * OR IN COMBINATION WITH YOUR PRODUCTS. * * IN NO EVENT SHALL APPLE BE LIABLE FOR ANY SPECIAL, INDIRECT, INCIDENTAL OR CONSEQUENTIAL * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS * OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) ARISING IN ANY WAY OUT OF THE USE, * REPRODUCTION, MODIFICATION AND/OR DISTRIBUTION OF THE APPLE SOFTWARE, HOWEVER CAUSED AND WHETHER * UNDER THEORY OF CONTRACT, TORT (INCLUDING NEGLIGENCE), STRICT LIABILITY OR OTHERWISE, EVEN * IF APPLE HAS BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. * */ /*============================================================================= Thunder.cpp =============================================================================*/ #include "Thunder.h" //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ COMPONENT_ENTRY(Thunder) //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // Thunder::Thunder //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ Thunder::Thunder(AudioUnit component) : AUEffectBase(component) { CreateElements(); Globals()->UseIndexedParameters(kNumberOfParameters); SetParameter(kParam_One, kDefaultValue_ParamOne ); SetParameter(kParam_Two, kDefaultValue_ParamTwo ); #if AU_DEBUG_DISPATCHER mDebugDispatcher = new AUDebugDispatcher (this); #endif } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // Thunder::GetParameterValueStrings //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult Thunder::GetParameterValueStrings(AudioUnitScope inScope, AudioUnitParameterID inParameterID, CFArrayRef * outStrings) { return kAudioUnitErr_InvalidProperty; } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // Thunder::GetParameterInfo //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult Thunder::GetParameterInfo(AudioUnitScope inScope, AudioUnitParameterID inParameterID, AudioUnitParameterInfo &outParameterInfo ) { ComponentResult result = noErr; outParameterInfo.flags = kAudioUnitParameterFlag_IsWritable | kAudioUnitParameterFlag_IsReadable; if (inScope == kAudioUnitScope_Global) { switch(inParameterID) { case kParam_One: AUBase::FillInParameterName (outParameterInfo, kParameterOneName, false); outParameterInfo.unit = kAudioUnitParameterUnit_Generic; outParameterInfo.minValue = 0.0; outParameterInfo.maxValue = 1.0; outParameterInfo.defaultValue = kDefaultValue_ParamOne; break; case kParam_Two: AUBase::FillInParameterName (outParameterInfo, kParameterTwoName, false); outParameterInfo.unit = kAudioUnitParameterUnit_Generic; outParameterInfo.minValue = 0.0; outParameterInfo.maxValue = 1.0; outParameterInfo.defaultValue = kDefaultValue_ParamTwo; break; default: result = kAudioUnitErr_InvalidParameter; break; } } else { result = kAudioUnitErr_InvalidParameter; } return result; } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // Thunder::GetPropertyInfo //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult Thunder::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 Thunder::SupportedNumChannels(const AUChannelInfo ** outInfo) { if (outInfo != NULL) { static AUChannelInfo info; info.inChannels = 2; info.outChannels = 2; *outInfo = &info; } return 1; } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // Thunder::GetProperty //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult Thunder::GetProperty( AudioUnitPropertyID inID, AudioUnitScope inScope, AudioUnitElement inElement, void * outData ) { return AUEffectBase::GetProperty (inID, inScope, inElement, outData); } // Thunder::Initialize //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult Thunder::Initialize() { ComponentResult result = AUEffectBase::Initialize(); if (result == noErr) Reset(kAudioUnitScope_Global, 0); return result; } #pragma mark ____ThunderEffectKernel //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // Thunder::ThunderKernel::Reset() //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult Thunder::Reset(AudioUnitScope inScope, AudioUnitElement inElement) { fpNShapeL = 0.0; fpNShapeR = 0.0; muSpeedA = 10000; muSpeedB = 10000; muCoefficientA = 1; muCoefficientB = 1; muVary = 1; gateL = 0.0; gateR = 0.0; iirSampleAL = 0.0; iirSampleBL = 0.0; iirSampleAR = 0.0; iirSampleBR = 0.0; iirSampleAM = 0.0; iirSampleBM = 0.0; iirSampleCM = 0.0; flip = false; return noErr; } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // Thunder::ProcessBufferLists //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ OSStatus Thunder::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; Float64 overallscale = 1.0; overallscale /= 44100.0; overallscale *= GetSampleRate(); Float64 thunder = GetParameter( kParam_One ) * 0.4; Float64 threshold = 1.0 - (thunder * 2.0); if (threshold < 0.01) threshold = 0.01; Float64 muMakeupGain = 1.0 / threshold; Float64 release = pow((1.28-thunder),5)*32768.0; release /= overallscale; Float64 fastest = sqrt(release); Float64 EQ = ((0.0275 / GetSampleRate())*32000.0); Float64 dcblock = EQ / 300.0; Float64 basstrim = (0.01/EQ)+1.0; //FF parameters also ride off Speed Float64 outputGain = GetParameter( kParam_Two ); Float64 coefficient; Float64 inputSense; Float64 resultL; Float64 resultR; Float64 resultM; Float64 resultML; Float64 resultMR; long double inputSampleL; long double inputSampleR; while (nSampleFrames-- > 0) { inputSampleL = *inputL; inputSampleR = *inputR; if (inputSampleL<1.2e-38 && -inputSampleL<1.2e-38) { static int noisesource = 0; //this declares a variable before anything else is compiled. It won't keep assigning //it to 0 for every sample, it's as if the declaration doesn't exist in this context, //but it lets me add this denormalization fix in a single place rather than updating //it in three different locations. The variable isn't thread-safe but this is only //a random seed and we can share it with whatever. noisesource = noisesource % 1700021; noisesource++; int residue = noisesource * noisesource; residue = residue % 170003; residue *= residue; residue = residue % 17011; residue *= residue; residue = residue % 1709; residue *= residue; residue = residue % 173; residue *= residue; residue = residue % 17; double applyresidue = residue; applyresidue *= 0.00000001; applyresidue *= 0.00000001; inputSampleL = applyresidue; } if (inputSampleR<1.2e-38 && -inputSampleR<1.2e-38) { static int noisesource = 0; noisesource = noisesource % 1700021; noisesource++; int residue = noisesource * noisesource; residue = residue % 170003; residue *= residue; residue = residue % 17011; residue *= residue; residue = residue % 1709; residue *= residue; residue = residue % 173; residue *= residue; residue = residue % 17; double applyresidue = residue; applyresidue *= 0.00000001; applyresidue *= 0.00000001; inputSampleR = applyresidue; //this denormalization routine produces a white noise at -300 dB which the noise //shaping will interact with to produce a bipolar output, but the noise is actually //all positive. That should stop any variables from going denormal, and the routine //only kicks in if digital black is input. As a final touch, if you save to 24-bit //the silence will return to being digital black again. } inputSampleL = inputSampleL * muMakeupGain; inputSampleR = inputSampleR * muMakeupGain; if (gateL < fabs(inputSampleL)) gateL = inputSampleL; else gateL -= dcblock; if (gateR < fabs(inputSampleR)) gateR = inputSampleR; else gateR -= dcblock; //setting up gated DC blocking to control the tendency for rumble and offset //begin three FathomFive stages iirSampleAL += (inputSampleL * EQ * thunder); iirSampleAL -= (iirSampleAL * iirSampleAL * iirSampleAL * EQ); if (iirSampleAL > gateL) iirSampleAL -= dcblock; if (iirSampleAL < -gateL) iirSampleAL += dcblock; resultL = iirSampleAL*basstrim; iirSampleBL = (iirSampleBL * (1 - EQ)) + (resultL * EQ); resultL = iirSampleBL; iirSampleAR += (inputSampleR * EQ * thunder); iirSampleAR -= (iirSampleAR * iirSampleAR * iirSampleAR * EQ); if (iirSampleAR > gateR) iirSampleAR -= dcblock; if (iirSampleAR < -gateR) iirSampleAR += dcblock; resultR = iirSampleAR*basstrim; iirSampleBR = (iirSampleBR * (1 - EQ)) + (resultR * EQ); resultR = iirSampleBR; iirSampleAM += ((inputSampleL + inputSampleR) * EQ * thunder); iirSampleAM -= (iirSampleAM * iirSampleAM * iirSampleAM * EQ); resultM = iirSampleAM*basstrim; iirSampleBM = (iirSampleBM * (1 - EQ)) + (resultM * EQ); resultM = iirSampleBM; iirSampleCM = (iirSampleCM * (1 - EQ)) + (resultM * EQ); resultM = fabs(iirSampleCM); resultML = fabs(resultL); resultMR = fabs(resultR); if (resultM > resultML) resultML = resultM; if (resultM > resultMR) resultMR = resultM; //trying to restrict the buzziness if (resultML > 1.0) resultML = 1.0; if (resultMR > 1.0) resultMR = 1.0; //now we have result L, R and M the trigger modulator which must be 0-1 //begin compressor section inputSampleL -= (iirSampleBL * thunder); inputSampleR -= (iirSampleBR * thunder); //highpass the comp section by sneaking out what will be the reinforcement inputSense = fabs(inputSampleL); if (fabs(inputSampleR) > inputSense) inputSense = fabs(inputSampleR); //we will take the greater of either channel and just use that, then apply the result //to both stereo channels. if (flip) { if (inputSense > threshold) { muVary = threshold / inputSense; muAttack = sqrt(fabs(muSpeedA)); muCoefficientA = muCoefficientA * (muAttack-1.0); if (muVary < threshold) { muCoefficientA = muCoefficientA + threshold; } else { muCoefficientA = muCoefficientA + muVary; } muCoefficientA = muCoefficientA / muAttack; } else { muCoefficientA = muCoefficientA * ((muSpeedA * muSpeedA)-1.0); muCoefficientA = muCoefficientA + 1.0; muCoefficientA = muCoefficientA / (muSpeedA * muSpeedA); } muNewSpeed = muSpeedA * (muSpeedA-1); muNewSpeed = muNewSpeed + fabs(inputSense*release)+fastest; muSpeedA = muNewSpeed / muSpeedA; } else { if (inputSense > threshold) { muVary = threshold / inputSense; muAttack = sqrt(fabs(muSpeedB)); muCoefficientB = muCoefficientB * (muAttack-1); if (muVary < threshold) { muCoefficientB = muCoefficientB + threshold; } else { muCoefficientB = muCoefficientB + muVary; } muCoefficientB = muCoefficientB / muAttack; } else { muCoefficientB = muCoefficientB * ((muSpeedB * muSpeedB)-1.0); muCoefficientB = muCoefficientB + 1.0; muCoefficientB = muCoefficientB / (muSpeedB * muSpeedB); } muNewSpeed = muSpeedB * (muSpeedB-1); muNewSpeed = muNewSpeed + fabs(inputSense*release)+fastest; muSpeedB = muNewSpeed / muSpeedB; } //got coefficients, adjusted speeds if (flip) { coefficient = pow(muCoefficientA,2); inputSampleL *= coefficient; inputSampleR *= coefficient; } else { coefficient = pow(muCoefficientB,2); inputSampleL *= coefficient; inputSampleR *= coefficient; } //applied compression with vari-vari-µ-µ-µ-µ-µ-µ-is-the-kitten-song o/~ //applied gain correction to control output level- tends to constrain sound rather than inflate it inputSampleL += (resultL * resultM); inputSampleR += (resultR * resultM); //combine the two by adding the summed channnel of lows if (outputGain != 1.0) { inputSampleL *= outputGain; inputSampleR *= outputGain; } flip = !flip; //stereo 32 bit dither, made small and tidy. int expon; frexpf((Float32)inputSampleL, &expon); long double dither = (rand()/(RAND_MAX*7.737125245533627e+25))*pow(2,expon+62); inputSampleL += (dither-fpNShapeL); fpNShapeL = dither; frexpf((Float32)inputSampleR, &expon); dither = (rand()/(RAND_MAX*7.737125245533627e+25))*pow(2,expon+62); inputSampleR += (dither-fpNShapeR); fpNShapeR = dither; //end 32 bit dither *outputL = inputSampleL; *outputR = inputSampleR; //don't know why we're getting a volume boost, cursed thing inputL += 1; inputR += 1; outputL += 1; outputR += 1; } return noErr; }