/* * File: PurestEcho.cpp * * Version: 1.0 * * Created: 6/2/17 * * Copyright: Copyright © 2017 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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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 ); #if AU_DEBUG_DISPATCHER mDebugDispatcher = new AUDebugDispatcher (this); #endif } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // PurestEcho::GetParameterValueStrings //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult PurestEcho::GetParameterValueStrings(AudioUnitScope inScope, AudioUnitParameterID inParameterID, CFArrayRef * outStrings) { return kAudioUnitErr_InvalidProperty; } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // PurestEcho::GetParameterInfo //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult PurestEcho::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; case kParam_Three: AUBase::FillInParameterName (outParameterInfo, kParameterThreeName, false); outParameterInfo.unit = kAudioUnitParameterUnit_Generic; outParameterInfo.minValue = 0.0; outParameterInfo.maxValue = 1.0; 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; default: result = kAudioUnitErr_InvalidParameter; break; } } else { result = kAudioUnitErr_InvalidParameter; } return result; } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // PurestEcho::GetPropertyInfo //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult PurestEcho::GetPropertyInfo (AudioUnitPropertyID inID, AudioUnitScope inScope, AudioUnitElement inElement, UInt32 & outDataSize, Boolean & outWritable) { return AUEffectBase::GetPropertyInfo (inID, inScope, inElement, outDataSize, outWritable); } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // PurestEcho::GetProperty //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult PurestEcho::GetProperty( AudioUnitPropertyID inID, AudioUnitScope inScope, AudioUnitElement inElement, void * outData ) { return AUEffectBase::GetProperty (inID, inScope, inElement, outData); } // PurestEcho::Initialize //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult PurestEcho::Initialize() { ComponentResult result = AUEffectBase::Initialize(); if (result == noErr) Reset(kAudioUnitScope_Global, 0); return result; } #pragma mark ____PurestEchoEffectKernel //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // PurestEcho::PurestEchoKernel::Reset() //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ void PurestEcho::PurestEchoKernel::Reset() { //totalsamples comes from the .h file: it's a const static number that defines //the whole delay buffer. We still have a hardcoded delay buffer, but some might like //to use this to define the buffer in terms of seconds: samples as a factor of GetSampleRate() //The danger there, of course, is having a user start up the plugin at 384K and smashing their memory for(int count = 0; count < totalsamples-1; count++) {d[count] = 0;} gcount = 0; fpNShape = 0.0; } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // PurestEcho::PurestEchoKernel::Process //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ void PurestEcho::PurestEchoKernel::Process( const Float32 *inSourceP, Float32 *inDestP, UInt32 inFramesToProcess, UInt32 inNumChannels, bool &ioSilence ) { UInt32 nSampleFrames = inFramesToProcess; const Float32 *sourceP = inSourceP; Float32 *destP = inDestP; int loopLimit = (int)(totalsamples * 0.499); //this is a double buffer so we will be splitting it in two Float64 time = pow(GetParameter( kParam_One ),2) * 0.999; Float64 tap1 = GetParameter( kParam_Two ); Float64 tap2 = GetParameter( kParam_Three ); Float64 tap3 = GetParameter( kParam_Four ); Float64 tap4 = GetParameter( kParam_Five ); Float64 gainTrim = 1.0 / (1.0 + tap1 + tap2 + tap3 + tap4); //this becomes our equal-loudness mechanic. 0.2 to 1.0 gain on all things. Float64 tapsTrim = gainTrim * 0.5; //the taps interpolate and require half that gain: 0.1 to 0.5 on all taps. int position1 = (int)(loopLimit * time * 0.25); int position2 = (int)(loopLimit * time * 0.5); int position3 = (int)(loopLimit * time * 0.75); int position4 = (int)(loopLimit * time); //basic echo information: we're taking four equally spaced echoes and setting their levels as desired. //position4 is what you'd have for 'just set a delay time' Float64 volAfter1 = (loopLimit * time * 0.25) - position1; Float64 volAfter2 = (loopLimit * time * 0.5) - position2; Float64 volAfter3 = (loopLimit * time * 0.75) - position3; Float64 volAfter4 = (loopLimit * time) - position4; //these are 0-1: casting to an (int) truncates fractional numbers towards zero (and is faster than floor() ) //so, when we take the integer number (all above zero) and subtract it from the real value, we get 0-1 Float64 volBefore1 = (1.0 - volAfter1) * tap1; Float64 volBefore2 = (1.0 - volAfter2) * tap2; Float64 volBefore3 = (1.0 - volAfter3) * tap3; Float64 volBefore4 = (1.0 - volAfter4) * tap4; //and if we are including a bit of the previous/next sample to interpolate, then if the sample position is 1.0001 //we'll be leaning most heavily on the 'before' sample which is nearer to us, and the 'after' sample is almost not used. //if the sample position is 1.9999, the 'after' sample is strong and 'before' is almost not used. volAfter1 *= tap1; volAfter2 *= tap2; volAfter3 *= tap3; volAfter4 *= tap4; //and like with volBefore, we also want to scale this 'interpolate' to the loudness of this tap. //We do it here because we can do it only once per audio buffer, not on every sample. This assumes we're //not moving the tap every sample: if so we'd have to do this every sample as well. int oneBefore1 = position1 - 1; int oneBefore2 = position2 - 1; int oneBefore3 = position3 - 1; int oneBefore4 = position4 - 1; if (oneBefore1 < 0) oneBefore1 = 0; if (oneBefore2 < 0) oneBefore2 = 0; if (oneBefore3 < 0) oneBefore3 = 0; if (oneBefore4 < 0) oneBefore4 = 0; int oneAfter1 = position1 + 1; int oneAfter2 = position2 + 1; int oneAfter3 = position3 + 1; int oneAfter4 = position4 + 1; //this is setting up the way we interpolate samples: we're doing an echo-darkening thing //to make it sound better. Pretty much no acoustic delay in human-breathable air will give //you zero attenuation at 22 kilohertz: forget this at your peril ;) Float64 delaysBuffer; long double inputSample; 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. } if (gcount < 0 || gcount > loopLimit) gcount = loopLimit; d[gcount+loopLimit] = d[gcount] = inputSample * tapsTrim; //this is how the double buffer works: //we can look for delay taps without ever having to 'wrap around' within our calculation. //As long as the delay tap is less than our loop limit we can always just add it to where we're //at, and get a valid sample back right away, no matter where we are in the buffer. //The 0.5 is taking into account the interpolation, by padding down the whole buffer. delaysBuffer = (d[gcount+oneBefore4]*volBefore4); delaysBuffer += (d[gcount+oneAfter4]*volAfter4); delaysBuffer += (d[gcount+oneBefore3]*volBefore3); delaysBuffer += (d[gcount+oneAfter3]*volAfter3); delaysBuffer += (d[gcount+oneBefore2]*volBefore2); delaysBuffer += (d[gcount+oneAfter2]*volAfter2); delaysBuffer += (d[gcount+oneBefore1]*volBefore1); delaysBuffer += (d[gcount+oneAfter1]*volAfter1); //These are the interpolated samples. We're adding them first, because we know they're smaller //and while the value of delaysBuffer is small we'll add similarly small values to it. Note the order. delaysBuffer += (d[gcount+position4]*tap4); delaysBuffer += (d[gcount+position3]*tap3); delaysBuffer += (d[gcount+position2]*tap2); delaysBuffer += (d[gcount+position1]*tap1); //These are the primary samples for the echo, and we're adding them last. As before we're starting with the //most delayed echoes, and ending with what we think might be the loudest echo. We're building this delaybuffer //from the faintest noises to the loudest, to avoid adding a bunch of teeny values at the end. //You can of course put the last echo as loudest, but with diminishing echo volumes this is optimal. //This technique is also present in other plugins such as Iron Oxide. inputSample = (inputSample * gainTrim) + delaysBuffer; //this could be just inputSample += d[gcount+position1]; //for literally a single, full volume echo combined with dry. //What I'm doing is making the echoes more interesting. gcount--; //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; } }