/*
* File: PurestEcho.cpp
*
* Version: 1.0
*
* Created: 6/2/17
*
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/*=============================================================================
PurestEcho.cpp
=============================================================================*/
#include "PurestEcho.h"
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
COMPONENT_ENTRY(PurestEcho)
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
// PurestEcho::PurestEcho
//~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
PurestEcho::PurestEcho(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 );
#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;
}
}
