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  3. SignalProcessing

SignalProcessing

API > API/Runtime

   
Name SignalProcessing
Type Runtime
Location /Engine/Source/Runtime/SignalProcessing/
Module Build Rules SignalProcessing.Build.cs

Classes

Name Remarks
FADEnvelope Sample accurate attack-decay style envelope generator
FAlignedBlockBuffer First In First Out Buffer designed for audio buffers.
FAllPassFractionalDelay Performs a fractional delay utilzing a single order all pass filter.
FAmp Class which manages scaling audio input and performing panning operations.
FAsyncSpectrumAnalyzer SpectrumAnalyzer for computing spectrum in async task.
FAsyncSpectrumAnalyzerScopeLock  
FAttackRelease Conversion between attack/release time and attack/release sample counts.
FAttackReleaseSmoother Smooths signals using attack and release settings.
FAudioFileWriter  
FBaseChannelFormatConverter FBaseChannelFormatConverter implements channel conversion using a simple mixing matrix.
FBiquadFilter Biquad filter class which wraps a biquad filter struct Handles multi-channel audio to avoid calculating filter coefficients for multiple channels of audio.
FBitCrusher Bit crushing effect https://en.wikipedia.org/wiki/Bitcrusher
FBlockCorrelator Correlation calculator which utilizes FFT to perform fast correlation calculations.
FBufferLinearEase Class which handles a vectorized interpolation of an entire buffer to the values of a target buffer
FBufferOnePoleLPF Simple 1-pole lowpass filter.
FCheckedAudioBuffer Opaque wrapper around FAlignedFloatBuffer.
FChorus  
FContiguousSparse2DKernelTransform FContiguousSparse2DKernelTransform
FConvolutionFactory FConvolutionFactory
FDelay Circular Buffer Delay Line.
FDelayAPF Implementation of a delay line with a feedback/feedforward gain coefficient APF filters pass all frequencies but changes phase relationships of frequencies
FDelayStereo  
FDynamicDelayAPF All Pass Filter with a long fractional delay which can be set per a sample.
FDynamicsProcessor Dynamic range compressor https://en.wikipedia.org/wiki/Dynamic_range_compression
FDynamicStateVariableFilter Combination of two state-variable filters, the first filtering a key signal, which drives gain in the second
FEarlyReflections Basic implementation of a 4x4 Feedback Delay Network.
FEarlyReflectionsFast Basic implementation of early reflections using a predelay, low pass filter and feedback delay network (FDN).
FEnvelope Envelope class generates ADSR style envelope.
FEnvelopeFollower A simple utility that returns a smoothed value given audio input using an RC circuit.
FEqualizer Equalizer filter An equalizer is a cascaded (serial) band of parametric EQs This filter allows for setting each band with variable Bandwidth/Q, Frequency, and Gain
FEventQuantizer Class which handles the details of event quantization.
FExponentialEase Simple exponential easing class. Useful for cheaply and smoothly interpolating parameters.
FFeedbackDelayNetwork 4 channel feedback delay network (FDN) for artificial reverberation.
FFFTFactory FFFTFactory
FFlanger  
FFoldbackDistortion Foldback distortion effect https://en.wikipedia.org/wiki/Foldback_(power_supply_design)
FGrain Class representing a grain of audio.
FGrainDelay  
FGrainEnvelope Simple class that generates an envelope and lets you retrieve interpolated values at any given fraction.
FGranularSynth A stereo granulator.
FInlineEnvelopeFollower FInlineEnvelopeFollower is useful for low sample rate use cases and where samples are only available one at a time.
FIntegerDelay An adjustable delay line. Delays values are limited to integer values.
FInterpolatedHPF  
FInterpolatedLPF  
FInterpolatedMultiTapDelay Delay line supporting multiple interpolated tap reads before advancing InBufferSizeSamples must be >= the size of the output buffer passed into Read If the delay length can be less than the output buffer size, Advance should be called before Read
FLadderFilter  
FLateReflectionsFast FLateReflections generates the long tail reverb of an input audio signal using a relatively fast algorithm using all pass filter delay lines.
FLateReflectionsPlate Single plate channel for plate reverberation.
FLFO Low frequency oscillator.
FLinearCurve A simple curve class that takes FVector2D points Required to decouple from Engine module
FLinearEase Simple easing function used to help interpolate params.
FLinearInterpFractionalDelay Fractional delay using linear interpolation.
FLinearPitchShifter  
FLinkwitzRileyBandSplitter Helper for Multi-Band processing to generate Linwitz-Riley filtered outputs from input https://en.wikipedia.org/wiki/Linkwitz%E2%80%93Riley_filter
FLongDelayAPF All Pass Filter with a long delay.
FMeanSquaredFIR Compute mean squared using FIR method.
FMeanSquaredIIR Compute mean squared using IIR method.
FMidiNoteQuantizer  
FModulationMatrix  
FMovingAverager This class buffers audio while maintaining a running average of the underlying buffer.
FMovingVectorAverager Vectorized version of FMovingAverager.
FMultichannelLinearResampler Linear resampler working on multichannel buffers
FOnePoleFilter A virtual analog one-pole filter. Defaults to a low-pass mode, but can be switched to high-pass
FOnePoleLPF Simple 1-pole lowpass filter.
FOnePoleLPFBank One pole LPF filter for multiple channels
FOpusEncoder  
FOsc Pitched oscillator.
FParam  
FPatchInput Handle to a patch. Should only be used by a single thread.
FPatchMixer This class is used for retrieving and mixing down audio from multiple threads.
FPatchMixerSplitter This class is used to mix multiple inputs from disparate threads to a single mixdown and deliver that mixdown to multiple outputs.
FPatchSplitter This class is used to post audio from one source to multiple threads.
FPhaser  
FPinkNoise Pink noise generator 1/Frequency noise spectrum
FPlateReverb  
FPlateReverbFast The Plate Reverb emulates the interactions between a sound, the listener and the space they share.
FRingModulation Ring modulation effect https://en.wikipedia.org/wiki/Ring_modulation
FRuntimeResampler Linear interpolating resampler which supports any number of channels of audio, interleaved or deinterleaved.
FSampleBufferReader  
FSilenceDetection This object will return buffered audio while the input signal is louder than the specified threshold, and buffer audio when the input signal otherwise.
FSineOsc FOsc Direct-form sinusoid oscillator.
FSineWaveTableOsc A sinusoidal wave table oscillator class.
FSinOsc2DRotation Generates a sine wave in the given buffer with the given frequency.
FSlowAdaptiveGainControl This object accepts an input buffer and current amplitude estimate of that input buffer, Then applies a computed gain target.
FSpectrumAnalysisAsyncWorker  
FSpectrumAnalyzer Class built to be a rolling spectrum analyzer for arbitrary, monaural audio data.
FSpectrumAnalyzerBuffer This class locks an input buffer (for writing) and an output buffer (for reading).
FSpectrumAnalyzerScopeLock  
FSphericalHarmonicCalculator Azimuth angle is measured CCW from front.
FStateVariableFilter  
FTapDelayPitchShifter Class which reads doppler-shifted tap from a delay buffer.
FVariablePoleFilter  
FVolumeFader Control-rate fader for managing volume fades of various standard shapes.
FWavEncoder  
FWaveShaper A digital wave shaping effect to cause audio distortion https://en.wikipedia.org/wiki/Waveshaper
FWaveTableOsc A wave table oscillator class.
FWhiteNoise White noise generator Flat spectrum
FWindow Class used to generate, contain and apply a DSP window of a given type.
TAutoDeinterleaveView TAutoDeinterleaveView
TAutoSlidingWindow TAutoSlidingWindow
TCircularAudioBuffer Basic implementation of a circular buffer built for pushing and popping arbitrary amounts of data at once.
TDeinterleaveView TDeinterleaveView
TParams Simple parameter object which uses critical section to write to and read from data.
TSample TSample Variant type to simplify converting and performing operations on fixed precision and floating point samples.
TSampleBuffer TSampleBuffer
TSampleRef TSampleRef Ref version of TSample.
TScopedSlidingWindow TScopedSlidingWindow
TSlidingBuffer TSlidingBuffer
TSlidingWindow Forward delcaration.

Structs

Name Remarks
DisjointedArrayView  
FAudioBufferDistanceAttenuationSettings Settings for audio buffer distance attenuation.
FBlockCorrelatorSettings Settings for block correlator
FChannelMapParams Parameters for creating 2D channel maps.
FCheckedBufferState  
FConvertDeinterleaveParams Parameters for creating a IConvertDeinterleave object.
FConvolutionSettings FConvolutionSettings
FDirectedGraphAlgo  
FEarlyReflectionsFastSettings  
FEarlyReflectionsSettings  
FEnvelopeFollowerInitParams  
FEventQuantizationSettings  
FFDNCoefficients Filter coefficients of 4 channel feedback delay network.
FFDNDelaySettings Delay settings of 4 channel feedback delay network.
FFFTSettings FFFTSettings
FGrainData  
FInlineEnvelopeFollowerInitParams  
FLateReflectionsFastSettings Settings for controlling the FLateReflections.
FLateReflectionsPlateDelays Delay line settings for reverb plate delay Tap locations are determined by analyzing these delay values.
FLateReflectionsPlateOutputs Structure to hold the various delay line tap outputs produced from a plate.
FLinkwitzRileyBandFilter  
FMelSpectrumKernelSettings Settings for a mel kernel which transforms an linearly space spectrum (e.g. FFT Magnitude) to a mel spectrum
FMultibandBuffer  
FOscFrequencyMod Struct which wraps all factors which contribute to pitch of the oscillator.
FPassiveFilterParams  
FPatch  
FPatchDestination  
FPatchOutput This class can be thought of as an output for a single constructed instance of FPatchInput.
FPatchSource  
FPlateReverbFastSettings Settings for plate reverb.
FPlateReverbSettings  
FPseudoConstantQ  
FPseudoConstantQBandSettings Settings for a single constant q band.
FPseudoConstantQKernelSettings Settings for Pseudo Constant Q Kernel generation.
FSpectrumAnalyzerSettings  
FSpectrumBandExtractorSettings Settings for band extractor.
FSpectrumBandExtractorSpectrumSettings Settings describing the spectrum used for in the band extractor.
FVectorLinearResampler FVectorLinearResampler
FViterbi  
FWetDry  
IConvertDeinterleave IConvertDeinterleave is an interface for transforming multichannel interleaved audio samples into multichannel deinterleaved samples.
ScaleDegreeSet  
TGetPower This allows us to write a compile time exponent of a number.

Interfaces

Name Remarks
IAudioEncoder IAudioEncoder Interface used to implement a runtime audio encoder.
IChannelFormatConverter Inteface for Channel Format Converters which process deinterleaved audio.
IConvolutionAlgorithm IConvolutionAlgorithm
IConvolutionAlgorithmFactory IConvolutionAlgorithmFactory
IFFTAlgorithm IFFTAlgorithmInterface for FFT algorithm.
IFFTAlgorithmFactory IFFTAlgorithmFactory
IFilter Base class for filters usable in synthesis.
IOscBase Oscillator base class.
IQuantizedEventListener Event listener interface.
ISampleRateConverter  
ISpectrumBandExtractor Interface for spectrum band extractors.
IViterbiInitialProbability Interface class for viterbi initial log probabilities.
IViterbiObservations Interface class for viterbi observations.
IViterbiTransitionProbability Interface class for viterbi transition log probabilities.

Typedefs

Name Type Remarks Include Path
AlignedByteBuffer TArray< uint8, FAudioBufferAlignedAllocator > Deprecated in favor of versions above. DSP/AlignedBuffer.h
AlignedFloatBuffer TArray< float, FAudioBufferAlignedAllocator >   DSP/AlignedBuffer.h
AlignedInt32Buffer TArray< int32, FAudioBufferAlignedAllocator >   DSP/AlignedBuffer.h
DefaultUSoundWaveSampleType int16   SampleBuffer.h
FAlignedByteBuffer TArray< uint8, FAudioBufferAlignedAllocator >   DSP/AlignedBuffer.h
FAlignedFloatBuffer TArray< float, FAudioBufferAlignedAllocator >   DSP/AlignedBuffer.h
FAlignedInt32Buffer TArray< int32, FAudioBufferAlignedAllocator >   DSP/AlignedBuffer.h
FAudioBufferAlignedAllocator TAlignedHeapAllocator< 16 > Aligned allocator used for fast operations. DSP/AlignedBuffer.h
FDirectedEdge TTuple< int32, int32 > A pair of int32s represent a directed edge. DSP/DirectedGraphAlgo.h
FDirectedTreeElement UE::MathCore::Graph::FDirectedTreeElement An element in a directed tree with references to children of a vertex. DSP/DirectedGraphAlgo.h
FEnvelope TArray< float >   DSP/GrainEnvelope.h
FMultichannelBuffer TArray< Audio::FAlignedFloatBuffer > A deinterleaved multichannel buffer DSP/MultichannelBuffer.h
FMultichannelBufferView TArray< TArrayView< float > > A deinterleaved multichannel buffer view. DSP/MultichannelBuffer.h
FMultichannelCircularBuffer TArray< Audio::TCircularAudioBuffer< float > > A deinterleaved multichannel circular buffer DSP/MultichannelBuffer.h
FPatchOutputStrongPtr TSharedPtr< FPatchOutput, ESPMode::ThreadSafe > Patch outputs are owned by the FPatchMixer, and are pinned by the FPatchInput. DSP/MultithreadedPatching.h
FPatchOutputWeakPtr TWeakPtr< FPatchOutput, ESPMode::ThreadSafe >   DSP/MultithreadedPatching.h
FSampleBuffer TSampleBuffer FSampleBuffer is a strictly defined TSampleBuffer that uses the same sample format we use for USoundWaves. SampleBuffer.h
FSpectrumAnalyzerTask FAsyncTask< FSpectrumAnalysisAsyncWorker >   DSP/SpectrumAnalyzer.h
TStackArrayOfPointers TArray< T *, TInlineAllocator< MaxStackMultiMonoChannels > > Stack array of pointers alias. DSP/MultiMono.h

Enums

Public

Name Remarks
Audio::EBiquadFilter::Type Enumeration of filter types.
Audio::EBufferCheckBehavior Copyright Epic Games, Inc. All Rights Reserved.
Audio::EChannelMapMonoUpmixMethod Denotes the upmix method for mixing a mono input into the front left and front right speakers.
Audio::EChannelMapOrder Denotes the order of gain coefficients for a channel map.
Audio::ECheckBufferFlags Copyright Epic Games, Inc. All Rights Reserved.
Audio::EChorusDelays::Type  
Audio::EDynamicFilterType Copyright Epic Games, Inc. All Rights Reserved.
Audio::EDynamicsProcessingMode::Type What mode the compressor is in.
Audio::EDynamicsProcessorChannelLinkMode Copyright Epic Games, Inc. All Rights Reserved.
Audio::EEventQuantization Copyright Epic Games, Inc. All Rights Reserved.
Audio::EFaderCurve Type of fade to use when adjusting a fader's volume over time.
Audio::EFFTScaling EFFTScaling
Audio::EFilter::Type  
Audio::EFilterOrder Copyright Epic Games, Inc. All Rights Reserved.
Audio::EGrainEnvelopeType Copyright Epic Games, Inc. All Rights Reserved.
Audio::EGranularSynthMode Copyright Epic Games, Inc. All Rights Reserved.
Audio::ELFO::Type  
Audio::ELFOMode::Type  
Audio::EMelNormalization Normalization methods for a mel transform.
Audio::EMusicalScale::Scale Any additions to this Enum should also be added to MetasoundMidiNoteQuantizerNode.cpp in the "#pragma region Metasound Enums" area to show up as options in that Node
Audio::EOsc::Type  
Audio::EPeakMode::Type Different modes for the envelope follower.
Audio::EPseudoConstantQNormalization Band normalization schemes
Audio::ESeekType::Type  
Audio::ESpectrumType Copyright Epic Games, Inc. All Rights Reserved.
Audio::EStereoDelayMode::Type The different stereo delay modes.
Audio::EWaveShaperType Copyright Epic Games, Inc. All Rights Reserved.
Audio::EWaveTable::Type Wavetable oscillator types.
Audio::EWindowType Copyright Epic Games, Inc. All Rights Reserved.
Audio::Grain::EEnvelope Grain envelope types.
EOpusFrameSizes Possible frame sizes to use for the encoder.
EOpusMode  

Constants

Name Type Remarks Include Path
Audio::ChannelMapMaxNumChannels uint32 Maximum number of channels supported by channel maps (Corresponds to 7.1 format) DSP/ChannelMap.h
Audio::ChannelMapVorbisNumChannels uint32 Number of channels in a vorbis input (Corresponds to 5.1 with vorbis channel ordering). DSP/ChannelMap.h
Audio::MaxFilterChannels const int32   DSP/Filter.h

Variables

Public

Name Type Remarks Include Path Unreal Specifiers
A1 float Simple biquad filter structure handling a biquad formulation See: https://en.wikipedia.org/wiki/Digital_biquad_filter Calculations of coefficients are handled outside this class. DSP/BiQuadFilter.h  
A2 float   DSP/BiQuadFilter.h  
B1 float   DSP/BiQuadFilter.h  
B2 float   DSP/BiQuadFilter.h  
Edges PRAGMA_DISABLE_DEPRECATION_WARNINGS TArray< FDirectedEdge > A strongly connected component contains a subgraph of strongly connected vertices and their corresponding edges. DSP/DirectedGraphAlgo.h  
FDirectedTree PRAGMA_DISABLE_DEPRECATION_WARNINGS typedef TMap< int32, FDirectedTreeElement > A directed tree graph represenation. DSP/DirectedGraphAlgo.h  
MaxStackMultiMonoChannels int32   DSP/MultiMono.h  
PRAGMA_ENABLE_DEPRECATION_WARNINGS     DSP/DirectedGraphAlgo.h  

Protected

Name Type Remarks Include Path Unreal Specifiers
X_Z1 float   DSP/BiQuadFilter.h  
X_Z2 float   DSP/BiQuadFilter.h  
Y_Z1 float   DSP/BiQuadFilter.h  
Y_Z2 float   DSP/BiQuadFilter.h  

Functions

Public

Name Remarks Include Path Unreal Specifiers

void Audio::ArrayInterpolate ( const float*RESTRICT InBuffer,
float*RESTRICT OutBuffer,
const int32 NumInSamples,
const int32 NumOutSamples
)

 Interpolates a Mono audio buffer. DSP/FloatArrayMath.h  

void Audio::AssignPcm24Value ( int8* OutPtr,
const int32 InValue,
const bool bIsOutputLittleEndian
)

 Copies 24 bits of a 32 bit integer (assumes data has been clamped to 24-bits) DSP/FloatArrayMath.h  

void Audio::CheckSample ( float InSample,
float Threshold
)

 Utility to check for sample clipping. DSP/Dsp.h  

void Audio::ConvertBipolarBufferToUnipolar ( float* InAlignedBuffer,
int32 NumSamples
)

 Converts in place a buffer into Unipolar (0..1) This is a Vector op version of the GetUnipolar function. DSP/Dsp.h  

float Audio::ConvertToDecibels ( const float InLinear,
const float InFloor
)

 Function converts linear scale volume to decibels. DSP/Dsp.h  

float Audio::ConvertToLinear ( const float InDecibels
)

 Function converts decibel to linear scale. DSP/Dsp.h  

void Audio::DecodeMidSide ( float& MidChannel,
float& SideChannel
)

 Decodes a stereo Mid/Side signal into a stereo Left/Right signal. DSP/Dsp.h  

void Audio::EncodeMidSide ( float& LeftChannel,
float& RightChannel
)

 Encodes a stereo Left/Right signal into a stereo Mid/Side signal. DSP/Dsp.h  

float Audio::FastSin ( const float X
)

 Low precision, high performance approximation of sine using parabolic polynomial approx Valid on interval [-PI, PI] DSP/Dsp.h  

float Audio::FastSin2 ( const float X
)

 Slightly higher precision, high performance approximation of sine using parabolic polynomial approx. DSP/Dsp.h  

float Audio::FastSin3 ( const float X
)

 Valid on interval [-PI, PI] Sine approximation using Bhaskara I technique discovered in 7th century. DSP/Dsp.h  

float Audio::FastTan ( float X
)

 Based on sin parabolic approximation. DSP/Dsp.h  

float Audio::FastTanh ( float X
)

 Fast tanh based on pade approximation. DSP/Dsp.h  

float Audio::GetBandwidthFromQ ( const float InQ
)

 Helper function to get bandwidth from Q. DSP/Dsp.h  

float Audio::GetBipolar ( const float X
)

 Gets polar value from unipolar. DSP/Dsp.h  

float Audio::GetFrequencyFromMidi ( const float InMidiNote
)

 Using midi tuning standard, compute frequency in hz from midi value. DSP/Dsp.h  

float Audio::GetFrequencyMultiplier ( const float InPitchSemitones
)

 Returns the frequency multiplier to scale a base frequency given the input semitones. DSP/Dsp.h  

float Audio::GetGainFromVelocity ( const float InVelocity
)

 Given a velocity value [0,127], return the linear gain. DSP/Dsp.h  

float Audio::GetLinearFrequencyClamped ( const float InFrequencyValue,
const FVector2D& Domain,
const FVector2D& Range
)

 Returns the linear frequency of the input value. Maps log domain and range values to linear output (good for linear slider representation/visualization of log frequency). Reverse of GetLogFrequencyClamped. DSP/Dsp.h  

float Audio::GetLogFrequencyClamped ( const float InValue,
const FVector2D& Domain,
const FVector2D& Range
)

 Returns the log frequency of the input value. Maps linear domain and range values to log output (good for linear slider controlling frequency) DSP/Dsp.h  

float Audio::GetMidiFromFrequency ( const float InFrequency
)

 Using midi tuning standard, compute midi from frequency in hz. DSP/Dsp.h  

float Audio::GetPitchScaleFromMIDINote ( int32 BaseMidiNote,
int32 TargetMidiNote
)

 Return a pitch scale factor based on the difference between a base midi note and a target midi note. Useful for samplers. DSP/Dsp.h  

float Audio::GetQFromBandwidth ( const float InBandwidth
)

 Helper function get Q from bandwidth. DSP/Dsp.h  

float Audio::GetSemitones ( const float InMultiplier
)

 Returns the number of semitones relative to a base frequency given the input frequency multiplier. DSP/Dsp.h  

void Audio::GetStereoPan ( const float InLinearPan,
float& OutLeft,
float& OutRight
)

 Calculates equal power stereo pan using sinusoidal-panning law and cheap approximation for sin InLinear pan is [-1.0, 1.0] so it can be modulated by a bipolar LFO DSP/Dsp.h  

float Audio::GetUnipolar ( const float X
)

 Converts bipolar value to unipolar. DSP/Dsp.h  

bool Audio::IsHostLittleEndian()

 Returns true if host has little endian byte ordering DSP/FloatArrayMath.h  

float Audio::LagrangianInterpolation ( const TArray< FVector2D > Points,
const float Alpha
)

 Polynomial interpolation using lagrange polynomials. DSP/Dsp.h  

TStackArrayOfPointers< T > Audio::MakeMultiMonoPointersFromView ( TArrayView< T > InMultiMono,
const int32 InNumFrames,
const int32 InNumChannels
)

 Helper to create array of pointers from multi-mono. DSP/MultiMono.h  

ECheckBufferFlags Audio::operator& ( const ECheckBufferFlags A,
const ECheckBufferFlags B
)

   DSP/BufferDiagnostics.h  

ECheckBufferFlags Audio::operator| ( const ECheckBufferFlags A,
const ECheckBufferFlags B
)

   DSP/BufferDiagnostics.h  

ECheckBufferFlags & Audio::operator|= ( ECheckBufferFlags& Out,
const ECheckBufferFlags Other
)

   DSP/BufferDiagnostics.h  

bool Audio::QuadraticPeakInterpolation ( const float InValues,
float& OutPeakLoc,
float& OutPeakValue
)

 Given three values, determine peak location and value of quadratic fitted to the data. DSP/Dsp.h  

void Audio::SetMultichannelBufferSize ( int32 InNumFrames,
FMultichannelBuffer& OutBuffer
)

 Set the number of frames for a multichannel buffer. DSP/MultichannelBuffer.h  

float Audio::UnderflowClamp ( const float InValue
)

 Clamps floats to 0 if they are in sub-normal range DSP/Dsp.h  

void BreakWhenAudible ( float* InBuffer,
int32 NumSamples
)

   DSP/AudioDebuggingUtilities.h  

void BreakWhenTooLoud ( float* InBuffer,
int32 NumSamples
)

   DSP/AudioDebuggingUtilities.h  

Static

Name Remarks Include Path Unreal Specifiers

static float Audio::EvaluateChebyshevPolynomial ( float FrequencyRatio,
int32 Order
)

   DSP/PassiveFilter.h  

static void Audio::Filter ( TArrayView< float >& Signal,
const FPassiveFilterParams& InParams
)

 This can be called on any TArrayView whose length is a power of 2. DSP/PassiveFilter.h  

static void Audio::Filter ( TArray< float >& Signal,
const FPassiveFilterParams& InParams
)

 Static function for applying a filter to any time series. DSP/PassiveFilter.h  

static float Audio::GetGainForFrequency ( float NormalizedFreq,
const FPassiveFilterParams& InParams
)

   DSP/PassiveFilter.h  

static void Audio::Grain::GenerateEnvelopeData ( FEnvelope& InData,
const int32 InNumFrames,
const EEnvelope InEnvelopeType
)

 Utility function generates an envelope with the input array for the number of desired frames. DSP/GrainEnvelope.h  

static float Audio::Grain::GetValue ( const FEnvelope& InGrainData,
const float InFraction
)

 Returns the interpolated envelope value given the fraction through the grain. DSP/GrainEnvelope.h  

static bool Audio::IsDenormalized ( const float InValue
)

 IEEE Standard 754 Floating Point Numbers https://www.geeksforgeeks.org/ieee-standard-754-floating-point-numbers/ Denormalized. DSP/BufferDiagnostics.h  

static bool Audio::IsInfinity ( const float InValue
)

 Infinity. DSP/BufferDiagnostics.h  

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