/* * File: AQuickVoiceClip.cpp * * Version: 1.0 * * Created: 10/3/13 * * Copyright: Copyright © 2013 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. * */ /*============================================================================= AQuickVoiceClip.cpp =============================================================================*/ #include "AQuickVoiceClip.h" //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ COMPONENT_ENTRY(AQuickVoiceClip) //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // AQuickVoiceClip::AQuickVoiceClip //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ AQuickVoiceClip::AQuickVoiceClip(AudioUnit component) : AUEffectBase(component) { CreateElements(); Globals()->UseIndexedParameters(kNumberOfParameters); SetParameter(kParam_One, kDefaultValue_ParamOne ); #if AU_DEBUG_DISPATCHER mDebugDispatcher = new AUDebugDispatcher (this); #endif } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // AQuickVoiceClip::GetParameterValueStrings //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult AQuickVoiceClip::GetParameterValueStrings(AudioUnitScope inScope, AudioUnitParameterID inParameterID, CFArrayRef * outStrings) { return kAudioUnitErr_InvalidProperty; } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // AQuickVoiceClip::GetParameterInfo //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult AQuickVoiceClip::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 = 30.0; outParameterInfo.maxValue = 3000.0; outParameterInfo.defaultValue = kDefaultValue_ParamOne; break; default: result = kAudioUnitErr_InvalidParameter; break; } } else { result = kAudioUnitErr_InvalidParameter; } return result; } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // AQuickVoiceClip::GetPropertyInfo //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult AQuickVoiceClip::GetPropertyInfo (AudioUnitPropertyID inID, AudioUnitScope inScope, AudioUnitElement inElement, UInt32 & outDataSize, Boolean & outWritable) { return AUEffectBase::GetPropertyInfo (inID, inScope, inElement, outDataSize, outWritable); } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // AQuickVoiceClip::GetProperty //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult AQuickVoiceClip::GetProperty( AudioUnitPropertyID inID, AudioUnitScope inScope, AudioUnitElement inElement, void * outData ) { return AUEffectBase::GetProperty (inID, inScope, inElement, outData); } // AQuickVoiceClip::Initialize //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ComponentResult AQuickVoiceClip::Initialize() { ComponentResult result = AUEffectBase::Initialize(); if (result == noErr) Reset(kAudioUnitScope_Global, 0); return result; } #pragma mark ____AQuickVoiceClipEffectKernel //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // AQuickVoiceClip::AQuickVoiceClipKernel::Reset() //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ void AQuickVoiceClip::AQuickVoiceClipKernel::Reset() { ataLast6Sample = ataLast5Sample = ataLast4Sample = ataLast3Sample = ataLast2Sample = ataLast1Sample = 0.0; ataHalfwaySample = ataHalfDrySample = ataHalfDiffSample = 0.0; ataLastDiffSample = ataDrySample = ataDiffSample = ataPrevDiffSample = 0.0; ataK1 = -0.646; //first FIR shaping of interpolated sample, brightens ataK2 = 0.311; //second FIR shaping of interpolated sample, thickens ataK6 = -0.093; //third FIR shaping of interpolated sample, brings air ataK7 = 0.057; //fourth FIR shaping of interpolated sample, thickens ataK8 = -0.023; //fifth FIR shaping of interpolated sample, brings air ataK3 = 0.114; //add raw to interpolated dry, toughens ataK4 = 0.886; //remainder of interpolated dry, adds up to 1.0 ataK5 = 0.431; //subtract this much prev. diff sample, brightens. 0.431 becomes flat lastSample = 0.0; lastOutSample = 0.0; lastOut2Sample = 0.0; lastOut3Sample = 0.0; lpDepth = 0.0; overshoot = 0.0; overall = 0; iirSampleA = 0.0; iirSampleB = 0.0; iirSampleC = 0.0; iirSampleD = 0.0; flip = false; fpNShape = 0.0; } //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ // AQuickVoiceClip::AQuickVoiceClipKernel::Process //~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ void AQuickVoiceClip::AQuickVoiceClipKernel::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 softness = 0.484416; Float64 hardness = 1.0 - softness; Float64 iirAmount = GetParameter( kParam_One )/8000.0; iirAmount /= overallscale; Float64 altAmount = (1.0 - iirAmount); Float64 cancelnew = 0.0682276; Float64 cancelold = 1.0 - cancelnew; Float64 maxRecent; Float64 lpSpeed = 0.0009; Float64 cliplevel = 0.98; Float64 refclip = 0.5; //preset to cut out gain quite a lot. 91%? no touchy unless clip Float64 inputSample; Float64 passThrough; Float64 outputSample; bool clipOnset; Float64 drySample; 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. } passThrough = ataDrySample = inputSample; ataHalfDrySample = ataHalfwaySample = (inputSample + ataLast1Sample + (ataLast2Sample*ataK1) + (ataLast3Sample*ataK2) + (ataLast4Sample*ataK6) + (ataLast5Sample*ataK7) + (ataLast6Sample*ataK8)) / 2.0; ataLast6Sample = ataLast5Sample; ataLast5Sample = ataLast4Sample; ataLast4Sample = ataLast3Sample; ataLast3Sample = ataLast2Sample; ataLast2Sample = ataLast1Sample; ataLast1Sample = inputSample; //setting up oversampled special antialiasing clipOnset = false; maxRecent = fabs( ataLast6Sample ); if (fabs( ataLast5Sample ) > maxRecent ) maxRecent = fabs( ataLast5Sample ); if (fabs( ataLast4Sample ) > maxRecent ) maxRecent = fabs( ataLast4Sample ); if (fabs( ataLast3Sample ) > maxRecent ) maxRecent = fabs( ataLast3Sample ); if (fabs( ataLast2Sample ) > maxRecent ) maxRecent = fabs( ataLast2Sample ); if (fabs( ataLast1Sample ) > maxRecent ) maxRecent = fabs( ataLast1Sample ); if (fabs( inputSample ) > maxRecent ) maxRecent = fabs( inputSample ); //this gives us something that won't cut out in zero crossings, to interpolate with maxRecent *= 2.0; //by refclip this is 1.0 and fully into the antialiasing if (maxRecent > 1.0) maxRecent = 1.0; //and it tops out at 1. Higher means more antialiasing, lower blends into passThrough without antialiasing ataHalfwaySample -= overall; //subtract dist-cancel from input after getting raw input, before doing anything drySample = ataHalfwaySample; if (lastSample >= refclip) { lpDepth += 0.1; if (ataHalfwaySample < refclip) { lastSample = ((refclip*hardness) + (ataHalfwaySample * softness)); } else lastSample = refclip; } if (lastSample <= -refclip) { lpDepth += 0.1; if (ataHalfwaySample > -refclip) { lastSample = ((-refclip*hardness) + (ataHalfwaySample * softness)); } else lastSample = -refclip; } if (ataHalfwaySample > refclip) { lpDepth += 0.1; if (lastSample < refclip) { ataHalfwaySample = ((refclip*hardness) + (lastSample * softness)); } else ataHalfwaySample = refclip; } if (ataHalfwaySample < -refclip) { lpDepth += 0.1; if (lastSample > -refclip) { ataHalfwaySample = ((-refclip*hardness) + (lastSample * softness)); } else ataHalfwaySample = -refclip; } outputSample = lastSample; lastSample = ataHalfwaySample; ataHalfwaySample = outputSample; //swap around in a circle for one final ADClip, //this time not tracking overshoot anymore //end interpolated sample //begin raw sample- inputSample and ataDrySample handled separately here inputSample -= overall; //subtract dist-cancel from input after getting raw input, before doing anything drySample = inputSample; if (lastSample >= refclip) { lpDepth += 0.1; if (inputSample < refclip) { lastSample = ((refclip*hardness) + (inputSample * softness)); } else lastSample = refclip; } if (lastSample <= -refclip) { lpDepth += 0.1; if (inputSample > -refclip) { lastSample = ((-refclip*hardness) + (inputSample * softness)); } else lastSample = -refclip; } if (inputSample > refclip) { lpDepth += 0.1; if (lastSample < refclip) { inputSample = ((refclip*hardness) + (lastSample * softness)); } else inputSample = refclip; } if (inputSample < -refclip) { lpDepth += 0.1; if (lastSample > -refclip) { inputSample = ((-refclip*hardness) + (lastSample * softness)); } else inputSample = -refclip; } outputSample = lastSample; lastSample = inputSample; inputSample = outputSample; //end raw sample ataHalfDrySample = (ataDrySample*ataK3)+(ataHalfDrySample*ataK4); ataHalfDiffSample = (ataHalfwaySample - ataHalfDrySample)/2.0; ataLastDiffSample = ataDiffSample*ataK5; ataDiffSample = (inputSample - ataDrySample)/2.0; ataDiffSample += ataHalfDiffSample; ataDiffSample -= ataLastDiffSample; inputSample = ataDrySample; inputSample += ataDiffSample; overall = (overall * cancelold) + (ataDiffSample * cancelnew); //apply all the diffs to a lowpassed IIR if (flip) { iirSampleA = (iirSampleA * altAmount) + (inputSample * iirAmount); inputSample -= iirSampleA; iirSampleC = (iirSampleC * altAmount) + (passThrough * iirAmount); passThrough -= iirSampleC; } else { iirSampleB = (iirSampleB * altAmount) + (inputSample * iirAmount); inputSample -= iirSampleB; iirSampleD = (iirSampleD * altAmount) + (passThrough * iirAmount); passThrough -= iirSampleD; } flip = not flip; //highpass section lastOut3Sample = lastOut2Sample; lastOut2Sample = lastOutSample; lastOutSample = inputSample; lpDepth -= lpSpeed; if (lpDepth > 0.0) { if (lpDepth > 1.0) lpDepth = 1.0; inputSample *= (1.0-lpDepth); inputSample += (((lastOutSample + lastOut2Sample + lastOut3Sample) / 3.6)*lpDepth); } if (lpDepth < 0.0) lpDepth = 0.0; inputSample *= (1.0-maxRecent); inputSample += (passThrough * maxRecent); //there's our raw signal, without antialiasing. Brings up low level stuff and softens more when hot if (inputSample > cliplevel) inputSample = cliplevel; if (inputSample < -cliplevel) inputSample = -cliplevel; //final 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; sourceP += inNumChannels; destP += inNumChannels; } }