/* * File: VoiceOfTheStarship.cpp * * Version: 1.0 * * Created: 10/12/10 * * 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. * */ /*============================================================================= VoiceOfTheStarship.h =============================================================================*/ #include "VoiceOfTheStarship.h" //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ COMPONENT_ENTRY(VoiceOfTheStarship) //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // VoiceOfTheStarship::VoiceOfTheStarship //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ VoiceOfTheStarship::VoiceOfTheStarship(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 } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // VoiceOfTheStarship::GetParameterValueStrings //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult VoiceOfTheStarship::GetParameterValueStrings(AudioUnitScope inScope, AudioUnitParameterID inParameterID, CFArrayRef * outStrings) { return kAudioUnitErr_InvalidProperty; } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // VoiceOfTheStarship::GetParameterInfo //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult VoiceOfTheStarship::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_Indexed; outParameterInfo.minValue = 0; outParameterInfo.maxValue = 16; outParameterInfo.defaultValue = kDefaultValue_ParamTwo; break; default: result = kAudioUnitErr_InvalidParameter; break; } } else { result = kAudioUnitErr_InvalidParameter; } return result; } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // VoiceOfTheStarship::GetPropertyInfo //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult VoiceOfTheStarship::GetPropertyInfo (AudioUnitPropertyID inID, AudioUnitScope inScope, AudioUnitElement inElement, UInt32 & outDataSize, Boolean & outWritable) { return AUEffectBase::GetPropertyInfo (inID, inScope, inElement, outDataSize, outWritable); } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // VoiceOfTheStarship::GetProperty //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult VoiceOfTheStarship::GetProperty( AudioUnitPropertyID inID, AudioUnitScope inScope, AudioUnitElement inElement, void * outData ) { return AUEffectBase::GetProperty (inID, inScope, inElement, outData); } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // VoiceOfTheStarship::Initialize //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult VoiceOfTheStarship::Initialize() { ComponentResult result = AUEffectBase::Initialize(); if (result == noErr) Reset(kAudioUnitScope_Global, 0); return result; } #pragma mark ____VoiceOfTheStarshipEffectKernel //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // VoiceOfTheStarship::VoiceOfTheStarshipKernel::Reset() //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ void VoiceOfTheStarship::VoiceOfTheStarshipKernel::Reset() { position = 99999999; quadratic = 0; noiseA = 0.0; noiseB = 0.0; noiseC = 0.0; flip = false; filterflip = false; for(int count = 0; count < 11; count++) {b[count] = 0.0; f[count] = 0.0;} lastAlgorithm = 0; fpNShape = 0.0; } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // VoiceOfTheStarship::VoiceOfTheStarshipKernel::Process //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ void VoiceOfTheStarship::VoiceOfTheStarshipKernel::Process( const Float32 *inSourceP, Float32 *inDestP, UInt32 inFramesToProcess, UInt32 inNumChannels, bool &ioSilence ) { //This code will pass-thru the audio data. //This is where you want to process data to produce an effect. UInt32 nSampleFrames = inFramesToProcess; const Float32 *sourceP = inSourceP; Float32 *destP = inDestP; Float64 cutoff = pow((GetParameter( kParam_One )*0.89)+0.1,3); if (cutoff > 1.0) cutoff = 1.0; Float64 invcutoff = 1.0 - cutoff; //this is the lowpass Float64 overallscale = ((1.0-GetParameter( kParam_One ))*9.0)+1.0; Float64 gain = overallscale; if (gain > 1.0) {f[0] = 1.0; gain -= 1.0;} else {f[0] = gain; gain = 0.0;} if (gain > 1.0) {f[1] = 1.0; gain -= 1.0;} else {f[1] = gain; gain = 0.0;} if (gain > 1.0) {f[2] = 1.0; gain -= 1.0;} else {f[2] = gain; gain = 0.0;} if (gain > 1.0) {f[3] = 1.0; gain -= 1.0;} else {f[3] = gain; gain = 0.0;} if (gain > 1.0) {f[4] = 1.0; gain -= 1.0;} else {f[4] = gain; gain = 0.0;} if (gain > 1.0) {f[5] = 1.0; gain -= 1.0;} else {f[5] = gain; gain = 0.0;} if (gain > 1.0) {f[6] = 1.0; gain -= 1.0;} else {f[6] = gain; gain = 0.0;} if (gain > 1.0) {f[7] = 1.0; gain -= 1.0;} else {f[7] = gain; gain = 0.0;} if (gain > 1.0) {f[8] = 1.0; gain -= 1.0;} else {f[8] = gain; gain = 0.0;} if (gain > 1.0) {f[9] = 1.0; gain -= 1.0;} else {f[9] = gain; gain = 0.0;} //this is the moving average with remainders if (overallscale < 1.0) overallscale = 1.0; f[0] /= overallscale; f[1] /= overallscale; f[2] /= overallscale; f[3] /= overallscale; f[4] /= overallscale; f[5] /= overallscale; f[6] /= overallscale; f[7] /= overallscale; f[8] /= overallscale; f[9] /= overallscale; //and now it's neatly scaled, too int lowcut = GetParameter( kParam_Two ); if (lastAlgorithm != lowcut) { noiseA = 0.0; noiseB = 0.0; noiseC = 0.0; for(int count = 0; count < 11; count++) {b[count] = 0.0;} lastAlgorithm = lowcut; } //cuts the noise back to 0 if we are changing algorithms, //because that also changes gains and can make loud pops. //We still get pops, but they'd be even worse int dcut; if (lowcut > 15) {lowcut = 1151; dcut= 11517;} if (lowcut == 15) {lowcut = 113; dcut= 1151;} if (lowcut == 14) {lowcut = 71; dcut= 719;} if (lowcut == 13) {lowcut = 53; dcut= 541;} if (lowcut == 12) {lowcut = 31; dcut= 311;} if (lowcut == 11) {lowcut = 23; dcut= 233;} if (lowcut == 10) {lowcut = 19; dcut= 191;} if (lowcut == 9) {lowcut = 17; dcut= 173;} if (lowcut == 8) {lowcut = 13; dcut= 131;} if (lowcut == 7) {lowcut = 11; dcut= 113;} if (lowcut == 6) {lowcut = 7; dcut= 79;} if (lowcut == 5) {lowcut = 6; dcut= 67;} if (lowcut == 4) {lowcut = 5; dcut= 59;} if (lowcut == 3) {lowcut = 4; dcut= 43;} if (lowcut == 2) {lowcut = 3; dcut= 37;} if (lowcut == 1) {lowcut = 2; dcut= 23;} if (lowcut < 1) {lowcut = 1; dcut= 11;} //this is the mechanism for cutting back subs without filtering Float64 rumbletrim = sqrt(lowcut); //this among other things is just to give volume compensation Float64 inputSample; while (nSampleFrames-- > 0) { inputSample = *sourceP; //we then ignore this! quadratic -= 1; if (quadratic < 0) { position += 1; quadratic = position * position; quadratic = quadratic % 170003; //% is C++ mod operator quadratic *= quadratic; quadratic = quadratic % 17011; //% is C++ mod operator quadratic *= quadratic; quadratic = quadratic % 1709; //% is C++ mod operator quadratic *= quadratic; quadratic = quadratic % dcut; //% is C++ mod operator quadratic *= quadratic; quadratic = quadratic % lowcut; //sets density of the centering force if (noiseA < 0) {flip = true;} else {flip = false;} //every time we come here, we force the random walk to be //toward the center of the waveform. Without this, //it's a pure random walk that will generate DC. } if (flip) noiseA += (rand()/(double)RAND_MAX); else noiseA -= (rand()/(double)RAND_MAX); //here's the guts of the random walk if (filterflip) {noiseB *= invcutoff; noiseB += (noiseA*cutoff); inputSample = noiseB;} else {noiseC *= invcutoff; noiseC += (noiseA*cutoff); inputSample = noiseC;} //now we have the output of the filter as inputSample. //this filter is shallower than a straight IIR: it's interleaved b[9] = b[8]; b[8] = b[7]; b[7] = b[6]; b[6] = b[5]; b[5] = b[4]; b[4] = b[3]; b[3] = b[2]; b[2] = b[1]; b[1] = b[0]; b[0] = inputSample; inputSample *= f[0]; inputSample += (b[1] * f[1]); inputSample += (b[2] * f[2]); inputSample += (b[3] * f[3]); inputSample += (b[4] * f[4]); inputSample += (b[5] * f[5]); inputSample += (b[6] * f[6]); inputSample += (b[7] * f[7]); inputSample += (b[8] * f[8]); inputSample += (b[9] * f[9]); inputSample *= 0.1; inputSample *= invcutoff; inputSample /= rumbletrim; flip = !flip; filterflip = !filterflip; //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; sourceP += inNumChannels; destP += inNumChannels; } }