The following is an overview of all available DSP effect classes that can be used in the Filter node. Assigned DSP effect classes will affect the input ports available to the Filter node.

DSP Nodes

Base

Abstract type that all DSP effects inherit from.

No attributes.

Biquad Filter

Applies a biquad filter^[1].

Attribute Name	Description	Value Range	Input Port
Biquad Type	Type of filter	LowPass HighPass BandPass Notch PeakingEQ LowShelf HighShelf AllPass
Fc	Cutoff/center frequency of the filter	[20, 16000]
Q	Factor that controls the slope of the filter shape	[0.1, 20]
Gain	Gain [dB] of the filter Available for "PeakingEQ", "LowShelf", "HighShelf" and "AllPass" Biquad Types.	[-18, 18]

Bitcrusher

Introduces distortion by reducing the bit depth of the audio input to 8-bit^[2].

Attribute Name	Description	Value Range	Input Port
Bit Mask	An 8-digit string of '0's or '1's that specifies which of the 8 bits of the bit-crushed signal should be "enabled" (1) or "disabled" (0)	String

Compressor

Implements dynamic range compression; Reduces the volume of the input signal if the specified threshold value is exceeded^[3].

Attribute Name	Description	Value Range	Input Port
Threshold	Threshold value of the input level above which the gain begins to be reduced	[-50, 0] dB
Ratio	Specifies the value of n in the ratio n:1. This value controls the amount of gain reduction (e.g. a ratio of 2:1 tells us that if the input signal is 2 dB over the value of Threshold, the resulting output level is reduced by 1 dB).	[1, 50]
Attack	Specifies the amount of time [ms] that the output level takes to reach the value defined by Ratio	[1, 1000] ms
Release	Specifies the amount of time [ms] that the compressor takes to "undo" gain reduction in the case that the signal level has decreased	[1, 1000] ms
Knee Width	Controls the sharpness of the compressor's "knee"	[0, 20]

Distortion

Introduces distortion using non-linear transfer functions.

Attribute Name	Description	Value Range	Input Port
Distortion Type	Defines which model of distortion will be applied	Reciprocal Inspired by the function: y = 1 / (abs(x) + 1) SoftClipper Two-stage quadratic clipping function HardClipper Limits all samples above threshold to the threshold value BufferNormalize All samples in the buffer (10ms) are normalized to their peak value. Introduces heavy compression with crackling Foldback Waveform, instead of being clipped off, is reflected back like a mirror so that the top of the waveform is inverted.
Drive	In the case of the Reciprocal type, it boosts the input signal. For other types, this value is used with an inverted sign as a threshold. On the output, there is gain compensation applied, so the output loudness is not affected by the Drive value too much	[0, 60] dB

Dynamic Equalizer

Implements dynamic range compression for a specified part of the frequency spectrum.

Attribute Name	Description	Value Range	Input Port
Biquad Type	Type of filter, compression will be applied to affected frequency bands	PeakingEQ Compression will be applied around an area of which Fc represents the center frequency. LowShelf Compression will be applied below Fc HighShelf Compression will be applied above Fc
Fc	Cutoff/center frequency of the filter	[20, 16000]
Q	Factor that controls the slope of the filter shape	[0.1, 20]
Ratio	Specifies the value of n in the ratio n:1. This value controls the amount of gain reduction (e.g. a ratio of 2:1 tells us that if the input signal is 2 dB over the value of Threshold, the resulting output level is reduced by 1 dB).	[1, 50]
Threshold	Threshold value of the input level above which the gain begins to be reduced	[-50, 0] dB
Attack	Specifies the amount of time [ms] that the output level takes to reach the value defined by Ratio	[1, 1000] ms
Release	Specifies the amount of time [ms] that the compressor takes to "undo" gain reduction in the case that the signal level has decreased	[1, 1000] ms

Equalizer4

Implements a 4-band equalizer

Attribute Name	Description	Value Range	Input Port
Gain1	Gain [dB] of the first frequency band	[-18, 18]
Gain2	Gain [dB] of the second frequency band	[-18, 18]
Gain3	Gain [dB] of the third frequency band	[-18, 18]
Gain4	Gain [dB] of the fourth frequency band	[-18, 18]
Fc1	Cutoff frequency of the lowshelf filter	[50, 800]
Fc2	Center frequency of the first peak filter	[200, 3000]
Fc3	Center frequency of the second peak filter	[1000, 8000]
Fc4	Cutoff frequency of the highshelf filter	[4000, 16000]
Qfactor2	Factor that controls the slope of the first peak filter's shape	[0.1, 20]
Qfactor3	Factor that controls the slope of the second peak filter's shape	[0.1, 20]

Flanger

The algorithm mixes two identical signals together, one signal delayed by a small and gradually changing period.

Attribute Name	Description	Value Range	Input Port
Delay Time	Delay time of signal mixed to input signal	[0.1, 20] ms
Feedback	Feedback loop gain. If -1 or 1, DSP will self oscillate.	[-1, 1]
Frequency	Sin shape LFO modulation frequency	[0.1, 10] Hz
Depth	LFO depth	[0, 1]
Damping	Controls Lowpass filter applied on the feedback loop	[20, 20000] Hz
Spread	Defines the LFO offset between the channels	[0, 1]

QuadDelay

Attribute Name	Description	Value Range	Input Port
Delay Time 1	Left front delay line delay time.	[10, 600] ms
Delay Time 2	Left right delay line delay time.	[10, 600] ms
Delay Time 3	Left back delay line delay time.	[10, 600] ms
Delay Time 4	Right back delay line delay time.	[10, 600] ms
Attenuation Factor	Adjusts distance attenuation of delayed sound calculated based on delay time.	[0, 1]
Damping Min	Damping at Delay Time = 10ms. Interpolated between min/max based on delay time.	[0, 1]
Damping Max	Damping at Delay Time = 600ms. Interpolated between min/max based on delay time.	[0, 1]
Feedback Min	Feedback at Delay Time = 10ms. Interpolated between min/max based on delay time.	[0, 1]
Feedback Max	Feedback at Delay Time = 600ms. Interpolated between min/max based on delay time.	[0, 1]
Azimuth	Sets rotation of delay lines.	[-180, 180] deg

LoudnessNormalization

(Tries to) keep a specified output volume by automatically adjusting the gain of the input signal

Attribute Name	Description	Value Range	Input Port
Target Level	Targetted output RMS level [dBFS]	[-60, 0] dBFS
Response Time	Specifies how quickly [s] the normalization reacts to changes in the input signal	[0.1, 10] s

MonoToStereo

Creates a pseudo stereo output from a mono input.

Works only for channels = 2 setup.

Attribute Name	Description	Value Range	Input Port
Delay Time	outL = -delayIn outR = delayIn	[0.1, 20] ms

OnePoleFilter

Implements a one pole filter. It has a 6dB/oct slope and is cheaper to compute than a biquad filter.

Attribute Name	Description	Value Range	Input Port
Filter Type	Type of Filter	LowPass HighPass
Frequency	Cutoff Frequency of the filter	[40, 15000]

PeakLimiter

Implements a limiter that uses a "Lookahead". Meaning a slight delay (5ms) in audio will be introduced in order to react even to quick peaks.

Attribute Name	Description	Value Range	Input Port
Threshold	Threshold value of the input level above which the gain begins to be reduced	[-50, 0] dB
Attack	Specifies the amount of time [ms] that the output level takes to reach the value defined by the limiter's ratio	[1, 1000] ms
Release	Specifies the amount of time [ms] that the compressor takes to "undo" gain reduction in the case that the signal level has decreased	[1, 1000] ms

Phaser

A cascade of 7 first order all-pass filters modulated by a triangle-shaped LFO

Attribute Name	Description	Value Range	Input Port
Frequency	LFO modulation frequency	[0.1, 10] Hz
Feedback	Feedback loop gain. If -1 or 1, DSP will self oscillate.	[-1, 1]
Offset	Works only if channelCount > 1 Can be set via a signal, but is evaluated only at the playback start and can not be changed during playback. Adds start offset to LFO. Offsets are spread evenly between channels. The last channel's LFO is offset by 0.5 * offset / period.	[0, 1]	without effect (bug)
Spread	Affects all pass filter frequency settings spread. If 0, all filters are set to the same frequency.	[0, 1]

Reverb

Implements a reverb, designed to model the reverb of a sound emitted into a room as perceived by a listener inside this room

Attribute Name	Description	Value Range	Input Port
Room Dimension	Dimensions of the modeled room	n/a
Source Position	Position of the sound source within the room	n/a
Mic Position	Position of the listener within the room	n/a
Absorption	Absorption factor, emulating absorption of sound from the room's surfaces.	[0, 1]
Damping	Damping factor of the reverb. Higher value means more damping of higher frequencies	[0, 1]

Reverb2

Implements a reverb.

Attribute Name	Description	Value Range	Input Port
Room Size	Arbitrary factor for the size of the modeled room	[0, 1]
Damping	Damping factor of the reverb. Higher value means more damping of higher frequencies	[0, 1]
Width	Stereo width of the reverb	[0, 1]

SmallRoomReverb

Implements a reverb, designed to model reverb within small rooms

Attribute Name	Description	Value Range	Input Port
Early Reflections Time	Arbitrary time factor influencing how long it takes for early reflections to reach the listener	[0, 1]	without effect (bug)
Late Reflections Time	Arbitrary time factor influencing how long it takes for late reflections to reach the listener	[0, 1]
Late Reflections Density	Density factor of late reflections	[0, 1]
Late Reflections Damping	Damping factor of the late reflections. A higher value means more damping of higher frequencies	[0, 1]
Width	Stereo width of the reverb	[0, 1]

Tremolo

Amplitude modulation using a triangle-shaped LFO.

Attribute Name	Description	Value Range	Input Port
Frequency	Frequency of the LFO	[0.1, 100] Hz
Offset	Works only if channelCount > 1 Can be set via a signal, but is evaluated only at the playback start and can not be changed during playback. Adds start offset to LFO. Offsets are spread evenly between channels. The last channel's LFO is offset by 0.5 * offset / period.	[0, 1]

VariableRolloffLPF

A lowpass filter with variable rolloff

Attribute Name	Description	Value Range	Input Port
Fc	Cutoff frequency of the filter	[100, 400] Hz
Rolloff	Filter rolloff	[0, 30]
Order	Filter order	[2, 3]

NoiseGate

A noise gate is an audio processor that works to eliminate sounds below a given threshold.

Attribute Name	Description	Value Range	Input Port
Threshold	Defines level at which gate opens.	[-60, 0] dB
Attack	Specifies the amount of time [ms] that takes for gate to fully open.	[0, 500] ms
Release	Specifies the amount of time [ms] that the gate to fully close.	[0, 500] ms