/* * File: MultiBandDistortion.cpp * * Version: 1.0 * * Created: 4/24/11 * * Copyright: Copyright © 2011 Airwindows, Airwindows uses the MIT license * * Disclaimer: IMPORTANT: This Apple software is supplied to you by Apple Computer, Inc. ("Apple") in * consideration of your agreement to the following terms, and your use, installation, modification * or redistribution of this Apple software constitutes acceptance of these terms. 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APPLE MAKES NO WARRANTIES, EXPRESS OR * IMPLIED, INCLUDING WITHOUT LIMITATION THE IMPLIED WARRANTIES OF NON-INFRINGEMENT, MERCHANTABILITY * AND FITNESS FOR A PARTICULAR PURPOSE, REGARDING THE APPLE SOFTWARE OR ITS USE AND OPERATION ALONE * OR IN COMBINATION WITH YOUR PRODUCTS. * * IN NO EVENT SHALL APPLE BE LIABLE FOR ANY SPECIAL, INDIRECT, INCIDENTAL OR CONSEQUENTIAL * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS * OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) ARISING IN ANY WAY OUT OF THE USE, * REPRODUCTION, MODIFICATION AND/OR DISTRIBUTION OF THE APPLE SOFTWARE, HOWEVER CAUSED AND WHETHER * UNDER THEORY OF CONTRACT, TORT (INCLUDING NEGLIGENCE), STRICT LIABILITY OR OTHERWISE, EVEN * IF APPLE HAS BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. * */ /*============================================================================= MultiBandDistortion.h =============================================================================*/ #include "MultiBandDistortion.h" //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ AUDIOCOMPONENT_ENTRY(AUBaseFactory, MultiBandDistortion) //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // MultiBandDistortion::MultiBandDistortion //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ MultiBandDistortion::MultiBandDistortion(AudioUnit component) : AUEffectBase(component) { CreateElements(); Globals()->UseIndexedParameters(kNumberOfParameters); SetParameter(kParam_One, kDefaultValue_ParamOne ); SetParameter(kParam_Two, kDefaultValue_ParamTwo ); SetParameter(kParam_Three, kDefaultValue_ParamThree ); SetParameter(kParam_Four, kDefaultValue_ParamFour ); SetParameter(kParam_Five, kDefaultValue_ParamFive ); SetParameter(kParam_Six, kDefaultValue_ParamSix ); SetParameter(kParam_Seven, kDefaultValue_ParamSeven ); SetParameter(kParam_Eight, kDefaultValue_ParamEight ); SetParameter(kParam_Nine, kDefaultValue_ParamNine ); #if AU_DEBUG_DISPATCHER mDebugDispatcher = new AUDebugDispatcher (this); #endif } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // MultiBandDistortion::GetParameterValueStrings //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult MultiBandDistortion::GetParameterValueStrings(AudioUnitScope inScope, AudioUnitParameterID inParameterID, CFArrayRef * outStrings) { return kAudioUnitErr_InvalidProperty; } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // MultiBandDistortion::GetParameterInfo //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult MultiBandDistortion::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_Decibels; outParameterInfo.minValue = 0.0; outParameterInfo.maxValue = 48.0; outParameterInfo.defaultValue = kDefaultValue_ParamTwo; break; case kParam_Three: AUBase::FillInParameterName (outParameterInfo, kParameterThreeName, false); outParameterInfo.unit = kAudioUnitParameterUnit_Decibels; outParameterInfo.minValue = 0.0; outParameterInfo.maxValue = 48.0; outParameterInfo.defaultValue = kDefaultValue_ParamThree; break; case kParam_Four: AUBase::FillInParameterName (outParameterInfo, kParameterFourName, false); outParameterInfo.unit = kAudioUnitParameterUnit_Generic; outParameterInfo.minValue = 0.0; outParameterInfo.maxValue = 1.0; outParameterInfo.defaultValue = kDefaultValue_ParamFour; break; case kParam_Five: AUBase::FillInParameterName (outParameterInfo, kParameterFiveName, false); outParameterInfo.unit = kAudioUnitParameterUnit_Generic; outParameterInfo.minValue = 0.0; outParameterInfo.maxValue = 1.0; outParameterInfo.defaultValue = kDefaultValue_ParamFive; break; case kParam_Six: AUBase::FillInParameterName (outParameterInfo, kParameterSixName, false); outParameterInfo.unit = kAudioUnitParameterUnit_Generic; outParameterInfo.minValue = 0.0; outParameterInfo.maxValue = 1.0; outParameterInfo.defaultValue = kDefaultValue_ParamSix; break; case kParam_Seven: AUBase::FillInParameterName (outParameterInfo, kParameterSevenName, false); outParameterInfo.unit = kAudioUnitParameterUnit_Generic; outParameterInfo.minValue = 0.0; outParameterInfo.maxValue = 1.0; outParameterInfo.defaultValue = kDefaultValue_ParamSeven; break; case kParam_Eight: AUBase::FillInParameterName (outParameterInfo, kParameterEightName, false); outParameterInfo.unit = kAudioUnitParameterUnit_Generic; outParameterInfo.minValue = 0.0; outParameterInfo.maxValue = 1.0; outParameterInfo.defaultValue = kDefaultValue_ParamEight; break; case kParam_Nine: AUBase::FillInParameterName (outParameterInfo, kParameterNineName, false); outParameterInfo.unit = kAudioUnitParameterUnit_Decibels; outParameterInfo.minValue = -48.0; outParameterInfo.maxValue = 0.0; outParameterInfo.defaultValue = kDefaultValue_ParamNine; break; default: result = kAudioUnitErr_InvalidParameter; break; } } else { result = kAudioUnitErr_InvalidParameter; } return result; } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // MultiBandDistortion::GetPropertyInfo //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult MultiBandDistortion::GetPropertyInfo (AudioUnitPropertyID inID, AudioUnitScope inScope, AudioUnitElement inElement, UInt32 & outDataSize, Boolean & outWritable) { return AUEffectBase::GetPropertyInfo (inID, inScope, inElement, outDataSize, outWritable); } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // MultiBandDistortion::GetProperty //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult MultiBandDistortion::GetProperty( AudioUnitPropertyID inID, AudioUnitScope inScope, AudioUnitElement inElement, void * outData ) { return AUEffectBase::GetProperty (inID, inScope, inElement, outData); } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // MultiBandDistortion::Initialize //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult MultiBandDistortion::Initialize() { ComponentResult result = AUEffectBase::Initialize(); if (result == noErr) Reset(kAudioUnitScope_Global, 0); return result; } #pragma mark ____MultiBandDistortionEffectKernel //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // MultiBandDistortion::MultiBandDistortionKernel::Reset() //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ void MultiBandDistortion::MultiBandDistortionKernel::Reset() { ataLast3Sample = ataLast2Sample = ataLast1Sample = 0.0; ataHalfwaySample = ataHalfDrySample = ataHalfDiffSample = 0.0; ataA = ataB = ataC = ataDrySample = ataDiffSample = ataPrevDiffSample = 0.0; ataUpsampleHighTweak = 0.0414213562373095048801688; //more adds treble to upsampling ataDecay = 0.915965594177219015; //Catalan's constant, more adds focus and clarity ataFlip = false; //end reset of antialias parameters iirSampleA = 0; iirSampleB = 0; } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // MultiBandDistortion::MultiBandDistortionKernel::Process //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ void MultiBandDistortion::MultiBandDistortionKernel::Process( const Float32 *inSourceP, Float32 *inDestP, UInt32 inFramesToProcess, UInt32 inNumChannels, bool &ioSilence ) { UInt32 nSampleFrames = inFramesToProcess; const Float32 *sourceP = inSourceP; Float32 *destP = inDestP; Float64 overallscale = 1.0; overallscale /= 44100.0; overallscale *= GetSampleRate(); Float64 iirAmount = pow(GetParameter( kParam_One ),3)/overallscale; Float64 gainH = pow(10.0,GetParameter( kParam_Two )/20); Float64 thresholdH = GetParameter( kParam_Four ); Float64 hardnessH; if (thresholdH < 1.0) hardnessH = 1.0 / (1.0-thresholdH); else hardnessH = 999999999999999999999.0; Float64 gainL = pow(10.0,GetParameter( kParam_Three )/20); Float64 thresholdL = GetParameter( kParam_Five ); Float64 hardnessL; if (thresholdL < 1.0) hardnessL = 1.0 / (1.0-thresholdL); else hardnessL = 999999999999999999999.0; Float64 breakup = 1.5707963267949; Float64 bridgerectifier; Float64 outputH = GetParameter( kParam_Six ); Float64 outputL = GetParameter( kParam_Seven ); Float64 outputD = GetParameter( kParam_Eight )*0.597; Float64 outtrim = outputH + outputL + outputD; outputH *= outtrim; outputL *= outtrim; outputD *= outtrim; Float64 outputGlobal = pow(10.0,GetParameter( kParam_Nine )/20); Float64 inputSample; Float64 tempH; Float64 tempL; while (nSampleFrames-- > 0) { ataDrySample = inputSample = *sourceP; ataHalfDrySample = ataHalfwaySample = (inputSample + ataLast1Sample + ((-ataLast2Sample + ataLast3Sample) * ataUpsampleHighTweak)) / 2.0; ataLast3Sample = ataLast2Sample; ataLast2Sample = ataLast1Sample; ataLast1Sample = inputSample; //setting up oversampled special antialiasing //pre-center code on inputSample and halfwaySample in parallel //begin interpolated sample- change inputSample -> ataHalfwaySample, ataDrySample -> ataHalfDrySample tempL = iirSampleA = (iirSampleA * (1 - iirAmount)) + (ataHalfwaySample * iirAmount); tempH = ataHalfwaySample - iirSampleA; //highpass section tempH *= gainH; if (fabs(tempH) > thresholdH) { bridgerectifier = (fabs(tempH)-thresholdH)*hardnessH; //skip flat area if any, scale to distortion limit if (bridgerectifier > breakup) bridgerectifier = breakup; //max value for sine function, 'breakup' modeling for trashed console tone //more hardness = more solidness behind breakup modeling. more softness, more 'grunge' and sag bridgerectifier = sin(bridgerectifier)/hardnessH; //do the sine factor, scale back to proper amount if (tempH > 0) tempH = bridgerectifier+thresholdH; else tempH = -(bridgerectifier+thresholdH); } //ADClip tempL *= gainL; if (fabs(tempL) > thresholdL) { bridgerectifier = (fabs(tempL)-thresholdL)*hardnessL; //skip flat area if any, scale to distortion limit if (bridgerectifier > breakup) bridgerectifier = breakup; //max value for sine function, 'breakup' modeling for trashed console tone //more hardness = more solidness behind breakup modeling. more softness, more 'grunge' and sag bridgerectifier = sin(bridgerectifier)/hardnessL; //do the sine factor, scale back to proper amount if (tempL > 0) tempL = bridgerectifier+thresholdL; else tempL = -(bridgerectifier+thresholdL); } //ADClip ataHalfwaySample = (tempL * outputL) + (tempH * outputH); //end interpolated sample //begin raw sample- inputSample and ataDrySample handled separately here tempL = iirSampleB = (iirSampleB * (1 - iirAmount)) + (inputSample * iirAmount); tempH = inputSample - iirSampleB; //highpass section tempH *= gainH; if (fabs(tempH) > thresholdH) { bridgerectifier = (fabs(tempH)-thresholdH)*hardnessH; //skip flat area if any, scale to distortion limit if (bridgerectifier > breakup) bridgerectifier = breakup; //max value for sine function, 'breakup' modeling for trashed console tone //more hardness = more solidness behind breakup modeling. more softness, more 'grunge' and sag bridgerectifier = sin(bridgerectifier)/hardnessH; //do the sine factor, scale back to proper amount if (tempH > 0) tempH = bridgerectifier+thresholdH; else tempH = -(bridgerectifier+thresholdH); } //ADClip tempL *= gainL; if (fabs(tempL) > thresholdL) { bridgerectifier = (fabs(tempL)-thresholdL)*hardnessL; //skip flat area if any, scale to distortion limit if (bridgerectifier > breakup) bridgerectifier = breakup; //max value for sine function, 'breakup' modeling for trashed console tone //more hardness = more solidness behind breakup modeling. more softness, more 'grunge' and sag bridgerectifier = sin(bridgerectifier)/hardnessL; //do the sine factor, scale back to proper amount if (tempL > 0) tempL = bridgerectifier+thresholdL; else tempL = -(bridgerectifier+thresholdL); } //ADClip inputSample = (tempL * outputL) + (tempH * outputH); //end raw sample //begin center code handling conv stuff tied to 44.1K, or stuff in time domain like delays //ataHalfwaySample -= inputSample; //retain only difference with raw signal //inputSample += convolutionstuff[count]; //ataHalfwaySample += inputSample; //restore interpolated signal including time domain stuff now //end center code for handling timedomain/conv stuff //post-center code on inputSample and halfwaySample in parallel //begin raw sample- inputSample and ataDrySample handled separately here //inputSample *= gain; //end raw sample //begin interpolated sample- change inputSample -> ataHalfwaySample, ataDrySample -> ataHalfDrySample //ataHalfwaySample *= gain; //end interpolated sample //begin antialiasing section for halfway sample ataC = ataHalfwaySample - ataHalfDrySample; if (ataFlip) {ataA *= ataDecay; ataB *= ataDecay; ataA += ataC; ataB -= ataC; ataC = ataA;} else {ataB *= ataDecay; ataA *= ataDecay; ataB += ataC; ataA -= ataC; ataC = ataB;} ataHalfDiffSample = (ataC * ataDecay); //end antialiasing section for halfway sample //begin antialiasing section for raw sample ataC = inputSample - ataDrySample; if (ataFlip) {ataA *= ataDecay; ataB *= ataDecay; ataA += ataC; ataB -= ataC; ataC = ataA;} else {ataB *= ataDecay; ataA *= ataDecay; ataB += ataC; ataA -= ataC; ataC = ataB;} ataDiffSample = (ataC * ataDecay); //end antialiasing section for input sample ataFlip = !ataFlip; inputSample = ataDrySample*outputD; inputSample += (ataDiffSample + ataHalfDiffSample); //apply processing as difference to non-oversampled raw input //inputSample *= output; *destP = (ataDrySample*dry)+(inputSample*wet); //built in output trim and dry/wet if desired *destP = inputSample*outputGlobal; sourceP += inNumChannels; destP += inNumChannels; } }