Name |
Description |
|
---|---|---|
|
FAcousticEchoCancellation |
This class uses an adaptive filter to cancel out any rendered audio signal that might be picked up by the mic. |
|
FAdaptiveFilter |
This filter takes a precomputed set of FIR weights in the frequency domain, and linearly converges to it. |
|
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 |
|
|
FAudioFFTAlgorithmFactory |
FFT Algorithm factory for this FFT implementation. |
|
FAudioFileWriter |
|
|
FBaseChannelFormatConverter |
FBaseChannelFormatConverter implements channel conversion using a simple mixing matrix. |
|
FBiquad |
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. |
|
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. |
|
FBlockCorrelatorSettings |
Settings for block correlator |
|
FBufferLinearEase |
Class which handles a vectorized interpolation of an entire buffer to the values of a target buffer |
|
FBufferOnePoleLPF |
Simple 1-pole lowpass filter. |
|
FChorus |
|
|
FContiguousSparse2DKernelTransform |
|
|
FConvolutionFactory |
|
|
FConvolutionSettings |
|
|
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 |
|
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). |
|
FEarlyReflectionsFastSettings |
|
|
FEarlyReflectionsSettings |
|
|
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 |
|
FEventQuantizationSettings |
|
|
FEventQuantizer |
Class which handles the details of event quantization. |
|
FExponentialEase |
Simple exponential easing class. Useful for cheaply and smoothly interpolating parameters. |
|
FFDAPFilterComputer |
This class takes an incoming signal and an outgoing signal, Correlates them, and returns the frequency values of the weight targets to pass to an adaptive filter. |
|
FFDNCoefficients |
Filter coefficients of 4 channel feedback delay network. |
|
FFDNDelaySettings |
Delay settings of 4 channel feedback delay network. |
|
FFeedbackDelayNetwork |
4 channel feedback delay network (FDN) for artificial reverberation. |
|
FFFTConvolver |
|
|
FFFTFactory |
|
|
FFFTSettings |
|
|
FFoldbackDistortion |
Foldback distortion effect https://en.wikipedia.org/wiki/Foldback_(power_supply_design)) |
|
FFTFreqDomainData |
|
|
FFTTimeDomainData |
|
|
FGrain |
Class representing a grain of audio. |
|
FGrainData |
|
|
FGrainEnvelope |
Simple class that generates an envelope and lets you retrieve interpolated values at any given fraction. |
|
FGranularSynth |
A stereo granulator. |
|
FIntegerDelay |
An adjustable delay line. Delays values are limited to integer values. |
|
FInterpolatedHPF |
|
|
FInterpolatedLPF |
|
|
FLadderFilter |
|
|
FLateReflectionsFast |
FLateReflections generates the long tail reverb of an input audio signal using a relatively fast algorithm using all pass filter delay lines. |
|
FLateReflectionsFastSettings |
Settings for controlling the FLateReflections. |
|
FLateReflectionsPlate |
Single plate channel for plate reverberation. |
|
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. |
|
FLFO |
Low frequency oscillator. |
|
FLinearEase |
Simple easing function used to help interpolate params. |
|
FLinearInterpFractionalDelay |
Fractional delay using linear interpolation. |
|
FLinkwitzRileyBandFilter |
|
|
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. |
|
FMelSpectrumKernelSettings |
Settings for a mel kernel which transforms an linearly space spectrum (e.g. FFT Magnitude) to a mel spectrum |
|
FModulationMatrix |
|
|
FMovingAverager |
This class buffers audio while maintaining a running average of the underlying buffer. |
|
FMovingVectorAverager |
Vectorized version of FMovingAverager. |
|
FMultibandBuffer |
|
|
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 |
|
FOsc |
Pitched oscillator. |
|
FOscFrequencyMod |
Struct which wraps all factors which contribute to pitch of the oscillator. |
|
FParam |
|
|
FPassiveFilterParams |
|
|
FPatch |
|
|
FPatchDestination |
|
|
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. |
|
FPatchOutput |
This class can be thought of as an output for a single constructed instance of FPatchInput. |
|
FPatchSource |
|
|
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. |
|
FPlateReverbFastSettings |
Settings for plate reverb. |
|
FPlateReverbSettings |
|
|
FPseudoConstantQ |
|
|
FPseudoConstantQBandSettings |
Settings for a single constant q band. |
|
FPseudoConstantQKernelSettings |
Settings for Pseudo Constant Q Kernel generation. |
|
FrequencyBuffer |
|
|
FRingModulation |
Ring modulation effect https://en.wikipedia.org/wiki/Ring_modulation |
|
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. |
|
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 |
|
|
FSpectrumAnalyzerSettings |
|
|
FSpectrumBandExtractorSettings |
Settings for band extractor. |
|
FSpectrumBandExtractorSpectrumSettings |
Settings describing the spectrum used for in the band extractor. |
|
FStateVariableFilter |
|
|
FVariablePoleFilter |
|
|
FViterbi |
|
|
FVolumeFader |
Control-rate fader for managing volume fades of various standard shapes. |
|
FWaveShaper |
A digital wave shaping effect to cause audio distortion https://en.wikipedia.org/wiki/Waveshaper |
|
FWaveTableOsc |
A wave table oscillator class. |
|
FWetDry |
|
|
FWhiteNoise |
White noise generator Flat spectrum |
|
FWindow |
Class used to generate, contain and apply a DSP window of a given type. |
|
IChannelFormatConverter |
Inteface for Channel Format Converters which process deinterleaved audio. |
|
IConvolutionAlgorithm |
|
|
IConvolutionAlgorithmFactory |
|
|
IFFTAlgorithm |
Interface for FFT algorithm. |
|
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. |
|
TAutoDeinterleaveView |
|
|
TAutoSlidingWindow |
|
|
TCircularAudioBuffer |
Basic implementation of a circular buffer built for pushing and popping arbitrary amounts of data at once. |
|
TDeinterleaveView |
|
|
TGetPower |
This allows us to write a compile time exponent of a number. |
|
TParams |
Simple parameter object which uses critical section to write to and read from data. |
|
TSample |
TSample |
|
TSampleRef |
TSampleRef |
|
TScopedSlidingWindow |
|
|
TSlidingBuffer |
|
|
TSlidingWindow |
Forward delcaration. |
Name |
Description |
---|---|
Audio::MaxFilterChannels |
Name |
Description |
---|---|
AlignedByteBuffer |
Deprecated in favor of versions above. |
AlignedFloatBuffer |
|
AlignedInt32Buffer |
|
FAlignedByteBuffer |
|
FAlignedFloatBuffer |
|
FAlignedInt32Buffer |
|
FAudioBufferAlignedAllocator |
Aligned allocator used for fast operations. |
FPatchOutputStrongPtr |
Patch outputs are owned by the FPatchMixer, and are pinned by the FPatchInput. |
FPatchOutputWeakPtr |
|
FSpectrumAnalyzerTask |
|
FStackSampleBuffer |
|
PassiveFilterParams |
Name | Description | ||
---|---|---|---|
|
Audio::ArrayComplexToPower ( |
Compute power of complex data. |
|
|
Audio::BufferComplexToMagnitudeFast ( |
Compute magnitude of complex data. |
|
|
Audio::BufferComplexToMagnitudeFast ( |
Compute magnitude of complex data. |
|
|
Audio::CheckSample ( |
Utility to check for sample clipping. |
|
|
float |
Audio::ConvertToDecibels ( |
Function converts linear scale volume to decibels. |
|
float |
Audio::ConvertToLinear ( |
Function converts decibel to linear scale. |
|
Audio::DecodeMidSide ( |
This function decodes a stereo Mid/Side signal into a stereo Left/Right signal. |
|
|
Audio::EncodeMidSide ( |
This function encodes a stereo Left/Right signal into a stereo Mid/Side signal. |
|
|
float |
Audio::EvaluateChebyshevPolynomial ( |
|
|
float |
Audio::FastSin ( |
Low precision, high performance approximation of sine using parabolic polynomial approx Valid on interval [-PI, PI] |
|
float |
Audio::FastSin2 ( |
Slightly higher precision, high performance approximation of sine using parabolic polynomial approx. |
|
float |
Audio::FastSin3 ( |
Sine approximation using Bhaskara I technique discovered in 7th century. |
|
float |
Audio::FastTan ( |
Based on sin parabolic approximation. |
|
float |
Audio::FastTanh ( |
Fast tanh based on pade approximation. |
|
Audio::Filter ( |
This can be called on any TArrayView |
|
|
Audio::Filter ( |
Static function for applying a filter to any time series. |
|
|
float |
Audio::GetBandwidthFromQ ( |
Helper function to get bandwidth from Q. |
|
float |
Audio::GetBipolar ( |
Gets polar value from unipolar. |
|
float |
Audio::GetFrequencyFromMidi ( |
Using midi tuning standard, compute frequency in hz from midi value. |
|
float |
Audio::GetFrequencyMultiplier ( |
Returns the frequency multiplier to scale a base frequency given the input semitones. |
|
float |
Audio::GetGainForFrequency ( |
|
|
float |
Audio::GetGainFromVelocity ( |
Given a velocity value [0,127], return the linear gain. |
|
float |
Audio::GetLinearFrequencyClamped |
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. |
|
float |
Audio::GetLogFrequencyClamped |
Returns the log frequency of the input value. Maps linear domain and range values to log output (good for linear slider controlling frequency) |
|
float |
Audio::GetMidiFromFrequency ( |
Using midi tuning standard, compute midi from frequency in hz. |
|
float |
Audio::GetPitchScaleFromMIDINote |
Return a pitch scale factor based on the difference between a base midi. |
|
float |
Audio::GetQFromBandwidth ( |
Helper function get Q from bandwidth. |
|
Audio::GetStereoPan ( |
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 |
|
|
float |
Audio::GetUnipolar ( |
Converts bipolar value to unipolar. |
|
float |
Audio::LagrangianInterpolation |
Polynomial interpolation using lagrange polynomials. |
|
Audio::QuadraticPeakInterpolation ( |
Given three values, determine peak location and value of quadratic fitted to the data. |
|
|
float |
Audio::UnderflowClamp ( |
Clamps floats to 0 if they are in sub-normal range. |
|
BreakWhenAudible ( |
||
|
BreakWhenTooLoud ( |