/* * File: NCSeventeen.cpp * * Version: 1.0 * * Created: 9/27/10 * * Copyright: Copyright © 2010 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. * */ /*============================================================================= NCSeventeen.h =============================================================================*/ #include "NCSeventeen.h" //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ COMPONENT_ENTRY(NCSeventeen) //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // NCSeventeen::NCSeventeen //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ NCSeventeen::NCSeventeen(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 } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // NCSeventeen::GetParameterValueStrings //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult NCSeventeen::GetParameterValueStrings(AudioUnitScope inScope, AudioUnitParameterID inParameterID, CFArrayRef * outStrings) { return kAudioUnitErr_InvalidProperty; } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // NCSeventeen::GetParameterInfo //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult NCSeventeen::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_Decibels; outParameterInfo.minValue = 0.0; outParameterInfo.maxValue = 24.0; 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; } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // NCSeventeen::GetPropertyInfo //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult NCSeventeen::GetPropertyInfo (AudioUnitPropertyID inID, AudioUnitScope inScope, AudioUnitElement inElement, UInt32 & outDataSize, Boolean & outWritable) { return AUEffectBase::GetPropertyInfo (inID, inScope, inElement, outDataSize, outWritable); } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // NCSeventeen::GetProperty //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult NCSeventeen::GetProperty( AudioUnitPropertyID inID, AudioUnitScope inScope, AudioUnitElement inElement, void * outData ) { return AUEffectBase::GetProperty (inID, inScope, inElement, outData); } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // NCSeventeen::Initialize //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult NCSeventeen::Initialize() { ComponentResult result = AUEffectBase::Initialize(); if (result == noErr) Reset(kAudioUnitScope_Global, 0); return result; } #pragma mark ____NCSeventeenEffectKernel //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // NCSeventeen::NCSeventeenKernel::Reset() //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ void NCSeventeen::NCSeventeenKernel::Reset() { lastSample = 0.0; iirSampleA = 0.0; iirSampleB = 0.0; flip = false; basslev = 0.0; treblev = 0.0; cheblev = 0.0; fpNShape = 0.0; } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // NCSeventeen::NCSeventeenKernel::Process //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ void NCSeventeen::NCSeventeenKernel::Process( const Float32 *inSourceP, Float32 *inDestP, UInt32 inFramesToProcess, UInt32 inNumChannels, bool &ioSilence ) { UInt32 nSampleFrames = inFramesToProcess; const Float32 *sourceP = inSourceP; Float32 *destP = inDestP; Float64 inP2; Float64 chebyshev; Float64 overallscale = 1.0; overallscale /= 44100.0; overallscale *= GetSampleRate(); Float64 IIRscaleback = 0.0004716; Float64 bassScaleback = 0.0002364; Float64 trebleScaleback = 0.0005484; Float64 addBassBuss = 0.000243; Float64 addTrebBuss = 0.000407; Float64 addShortBuss = 0.000326; IIRscaleback /= overallscale; bassScaleback /= overallscale; trebleScaleback /= overallscale; addBassBuss /= overallscale; addTrebBuss /= overallscale; addShortBuss /= overallscale; Float64 limitingBass = 0.39; Float64 limitingTreb = 0.6; Float64 limiting = 0.36; Float64 maxfeedBass = 0.972; Float64 maxfeedTreb = 0.972; Float64 maxfeed = 0.975; Float64 bridgerectifier; long double inputSample; Float64 lowSample; Float64 highSample; Float64 distSample; Float64 minusSample; Float64 plusSample; Float64 gain = pow(10.0,GetParameter( kParam_One )/20); Float64 outgain = GetParameter( kParam_Two ); while (nSampleFrames-- > 0) { inputSample = *sourceP; if (inputSample<1.2e-38 && -inputSample<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; inputSample = 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. } inputSample *= gain; if (flip) { iirSampleA = (iirSampleA * 0.9) + (inputSample * 0.1); lowSample = iirSampleA; } else { iirSampleB = (iirSampleB * 0.9) + (inputSample * 0.1); lowSample = iirSampleB; } highSample = inputSample - lowSample; flip = !flip; //we now have two bands and the original source //inputSample = lowSample; inP2 = lowSample * lowSample; if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0; chebyshev = (2 * inP2); chebyshev *= basslev; //second harmonic max +1 if (basslev > 0) basslev -= bassScaleback; if (basslev < 0) basslev += bassScaleback; //this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on bridgerectifier = fabs(lowSample); if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; //max value for sine function bridgerectifier = sin(bridgerectifier); if (lowSample > 0.0) distSample = bridgerectifier; else distSample = -bridgerectifier; minusSample = lowSample - distSample; plusSample = lowSample + distSample; if (minusSample > maxfeedBass) minusSample = maxfeedBass; if (plusSample > maxfeedBass) plusSample = maxfeedBass; if (plusSample < -maxfeedBass) plusSample = -maxfeedBass; if (minusSample < -maxfeedBass) minusSample = -maxfeedBass; if (lowSample > distSample) basslev += (minusSample*addBassBuss); if (lowSample < -distSample) basslev -= (plusSample*addBassBuss); if (basslev > 1.0) basslev = 1.0; if (basslev < -1.0) basslev = -1.0; bridgerectifier = fabs(lowSample); if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; //max value for sine function bridgerectifier = sin(bridgerectifier); if (lowSample > 0.0) lowSample = bridgerectifier; else lowSample = -bridgerectifier; //apply the distortion transform for reals lowSample /= (1.0+fabs(basslev*limitingBass)); lowSample += chebyshev; //apply the correction measures //inputSample = highSample; inP2 = highSample * highSample; if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0; chebyshev = (2 * inP2); chebyshev *= treblev; //second harmonic max +1 if (treblev > 0) treblev -= trebleScaleback; if (treblev < 0) treblev += trebleScaleback; //this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on bridgerectifier = fabs(highSample); if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; //max value for sine function bridgerectifier = sin(bridgerectifier); if (highSample > 0.0) distSample = bridgerectifier; else distSample = -bridgerectifier; minusSample = highSample - distSample; plusSample = highSample + distSample; if (minusSample > maxfeedTreb) minusSample = maxfeedTreb; if (plusSample > maxfeedTreb) plusSample = maxfeedTreb; if (plusSample < -maxfeedTreb) plusSample = -maxfeedTreb; if (minusSample < -maxfeedTreb) minusSample = -maxfeedTreb; if (highSample > distSample) treblev += (minusSample*addTrebBuss); if (highSample < -distSample) treblev -= (plusSample*addTrebBuss); if (treblev > 1.0) treblev = 1.0; if (treblev < -1.0) treblev = -1.0; bridgerectifier = fabs(highSample); if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; //max value for sine function bridgerectifier = sin(bridgerectifier); if (highSample > 0.0) highSample = bridgerectifier; else highSample = -bridgerectifier; //apply the distortion transform for reals highSample /= (1.0+fabs(treblev*limitingTreb)); highSample += chebyshev; //apply the correction measures inputSample = lowSample + highSample; inP2 = inputSample * inputSample; if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0; chebyshev = (2 * inP2); chebyshev *= cheblev; //third harmonic max -1 if (cheblev > 0) cheblev -= IIRscaleback; if (cheblev < 0) cheblev += IIRscaleback; //this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on bridgerectifier = fabs(inputSample); if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; //max value for sine function bridgerectifier = sin(bridgerectifier); if (inputSample > 0.0) distSample = bridgerectifier; else distSample = -bridgerectifier; minusSample = inputSample - distSample; plusSample = inputSample + distSample; if (minusSample > maxfeed) minusSample = maxfeed; if (plusSample > maxfeed) plusSample = maxfeed; if (plusSample < -maxfeed) plusSample = -maxfeed; if (minusSample < -maxfeed) minusSample = -maxfeed; if (inputSample > distSample) cheblev += (minusSample*addShortBuss); if (inputSample < -distSample) cheblev -= (plusSample*addShortBuss); if (cheblev > 1.0) cheblev = 1.0; if (cheblev < -1.0) cheblev = -1.0; bridgerectifier = fabs(inputSample); if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; //max value for sine function bridgerectifier = sin(bridgerectifier); if (inputSample > 0.0) inputSample = bridgerectifier; else inputSample = -bridgerectifier; //apply the distortion transform for reals inputSample /= (1.0+fabs(cheblev*limiting)); inputSample += chebyshev; //apply the correction measures if (outgain < 1.0) { inputSample *= outgain; } if (inputSample > 0.95) inputSample = 0.95; if (inputSample < -0.95) inputSample = -0.95; //iron bar //32 bit dither, made small and tidy. int expon; frexpf((Float32)inputSample, &expon); long double dither = (rand()/(RAND_MAX*7.737125245533627e+25))*pow(2,expon+62); inputSample += (dither-fpNShape); fpNShape = dither; //end 32 bit dither *destP = inputSample; //built in output trim and dry/wet by default sourceP += inNumChannels; destP += inNumChannels; } }