Pédale Vite — User manual


  1. Introduction
  2. Usage
    1. Hardware description
    2. Startup
    3. Calibration
    4. First steps
  3. Reference
    1. Concepts
    2. LEDS
    3. Main screens
    4. Editing a program
    5. Program banks
    6. Pedalboard configuration
    7. Volume and levels
    8. Other settings
    9. Tuner
    10. Restart
  4. Available effects
    1. Color Me — Vocal filter
    2. CompEx — Dynamics compressor/expander
    3. Convert To Stereo
    4. Delay — Stereo delay
    5. Distortion Tone Stage
    6. Double Distortion
    7. Envelope Follower
    8. Filter Squeezer — Distorted resonant filter
    9. FlanCho — Chorus and flanger
    10. Freeverb — Reverberation
    11. Frequency Shifter
    12. Harmonic Tremolo — Two-band tremolo
    13. Input Impedance Fix
    14. Noise Bleach — Noise gate
    15. Noise Chlorine — Noise gate
    16. Onset detector
    17. Parametric Equalizer
    18. Phaser AP
    19. Phaser HT
    20. Pitch Detector
    21. Pulse Width Modulation Distortion
    22. SiemensGirlz — Creative delay
    23. Simple Distortion
    24. Speaker Emulator
    25. Tremolo
    26. Wah-wah
    27. Wah-wah (CryBaby)
    28. Control ramp — Parameter transition
    29. LFO — Low frequency oscillator
  5. Troubleshooting
  6. Change log

I. Introduction

Pédale Vite is a DIY multi-effect pedalboard for guitar. This manual explains how to use the pedalboard once assembled and functional.

II. Usage

Hardware description

Back panel

  1. Rasberry Pi 3 USB and Ethernet ports
  2. Jacks for expression pedals
  3. Power switch
  4. 220–240V power lead
  5. Audio output to amplifier or mixing desk
  6. Headphone plug, stereo jack
  7. Guitar audio input

Access to the Pi is not required during normal operation and is for maintenance only.

The first expression pedal is on the left hand side of the panel, and therefore on the right hand side when facing the pedalboard.

All jacks are unbalanced ¼” (mono signal), except for the headphone jack which is stereo.

Les canaux gauches et droits des entrées et des sorties sont inversés afin qu’ils soient dans le bon ordre quand on se place devant le pédalier. Les entrées font environ 500 kΩ d’impédance. Utiliser le canal gauche pour brancher la guitare ou tout autre signal mono. Quand le pédalier est configuré en mono, les deux sorties diffusent le même signal. The left and right channels of the inputs and outputs are inverted so that they are in the right order when standing in front of the pedalboard. The input impedance is about 500kΩ. Use the left channel to connect the guitar or any other mono signal. When the pedalboard is set to mono, both outputs send the same signal.

Front panel

  1. Effect control knob
  2. Display
  3. Navigation knobs
  4. Navigation buttons
  5. Control LED, their use depends on the context.
  6. Two rows of six footswitches to change the effects.

The navigation buttons are arranged as follows. Top row : Select/validate (Select), top arrow (), cancel/escape (Esc). Bottom row : left arrow (), down arrow (), right arrow ().

Navigation knobs are usually redundant with navigation buttons. The left one allows to move vertically, and the right one to move horizontally or to change a value.


  1. Connect the power lead (4).
  2. Plug the guitar on the left input (7).
  3. Plug the left output (5) on an amplifier.
  4. Switch on the pedalboard (3).

After less than ten seconds, the display should show the current program, called “Bypass”.

  1. Switch on the amplifier.
  2. Ply.

Sound is too loud? Too quiet? You don’t hear anything? Please report to the calibration section.

To stop, perform operations in reverse order: switch off the amplifier, switch off the pedalboard and unplug it.

Note: it is not recommended to have the amplifier running while the pedalboard is switched on or off.


In order for the pedalboard to give optimum results, it must be calibrated, especially the input and output levels. The goal is to have as strong an input level as possible without the incoming guitar signal saturating at that point. Then, if you play on an amp, you have to go out at a level similar to the guitar plugged in directly.

Unfortunately, there is no practical adjustment for calibration outside the enclosure. It is therefore necessary to venture inside the machine.

Turn off the pedalboard and turn it over. Remove the 8 screws holding the base with a Phillips screwdriver. Lift up the base, the sound card is attached to it. Be careful not to unplug or pull out any cables. Place the base close to the housing.

The audio interface is attached on the inner side. Plug and turn on the pedalboard, still open. Avoid putting fingers on the power cables. It is up to you to find an effective method of handling the case, as you will have to manipulate the knobs and check the front panel to adjust the audio interface. Placing and securing the enclosure on the edge should allow this without too much hassle.

Press the Select button, scroll down to the Volume & Levels menu, and press Select again.

This takes you to the level control screen. The upper meter shows the input levels, the middle one the output levels. The levels to be considered (peaks) are given by the small strokes moving to the right of the solid bars (RMS).

Input level

Play as hard as possible. Input level should never exceed 0dB, and rarely go above −6dB. However it should be as high as possible.

Input level is set at the minimum by default, which is fine for an electric guitar plugged directly to the pedalboard, given the audio interface. Anyway, it may be required to adjust it.

Turn the input 1 knob (left) on the audio interface until the level looks correct. Then set the same level on the input 2.

Niveau de sortie

Once we are done with the input level, we can set the output level. Make sure the “Bypass” program is activated, so the guitar input goes directly to the output without any modification.

The output volume should be set to 0dB in the Volume and levels screen. If it is not the case, go to the Output volume settings with the vertical arrows. Change the volume with the horizontal arrows.

Then set the output level on the audio interface in order to get the same sound as with a guitar plugged directly to the amplifier. An AB/Y box is a great help for this. If you don’t have one, you can plug alternatively the guitar on the pedalboard and on the amplifier to compare the volumes.

Finally, you can set the headphone volume, which doesn’t require anything else than your taste and ears.

Now, switch off the pedalboard and put back the base. Make sure that wires don’t suffer from the operation and that everything fits together without forcing on the enclosure content.

Plug the pedalboard and switch it on. Check that everything is still OK.

First steps

III. Reference


Effects on the pedalboard are organized in programs. A program is made of an effect chain and some settings to control the effects from footswitches, pedals and knobs, the controllers.

Programs are grouped in banks. Each bank is made of 16 programs, and the pedalboard contains 64 banks. The pedalboard makes easy the navigation between the programs of a single bank.

A program is made of an effect chain. For each effect, it is possible to set its volume, the mix between the input signal and the output signal, and to bypass the processing.


There are three LEDs on the pedalboard, a green one surrounded by two red ones. Their meaning depends on the context, on the activated function. Usually they should be off, excepted when using the tuner.

Normal operations

The leftmost red LED indicates that the output sound clips. You should lower the main volume or the volume of one of the activated effects.

The green LED indicates that the pedalboard is performing a write operation on the internal drive (the SD card). Such an operation is usually quick, anyway make sure not to switch off the pedalboard at this moment. The memory card is delicate and could be damaged.

The rightmost red LED indicates that the processing unit is overloaded. There are too much simultaneous effects, or they have too complex settings, and the sound cannot be produced on time. It is suitable to recall a simpler program, or to change the effect parameters to reduce the load. It can also be a temporary overload, in which case there is nothing special to do. The sound should come back and the LED turn off.

However, if the overload lasts, the audio engine gives up its recovery attempt. This state is indicated with all the three LEDs lit. In this case, a simpler program should be activated and the audio engine restarted manually. This can be obtained by pressing Esc three times on the main screen.


In tuner mode, the LEDs indicates the rightness of the played note. The leftmost LED indicates that the note is too low, and the rightmost one that the note is too high.

When the played note is right, the green LED is lit.

Main screens

Current program and information

This is the startup screen. This is also the one on the top of the menu hierarchy. If you get lost, just press Esc one or more times to get back here.

Several important information are displayed here: On the top, the current program and bank. The last modified parameter on the middle. On the bottom, the Raspberry Pi IP address, if it is available. Indeed it is required to connect on the machine with SSH for maintenance operations.

The Esc key in this screen activates some other functions. They depends how many times the key is pressed:

  1. Updates the display, particularly the IP address.
  2. Resets the display hardware.
  3. Resets the audio engine.

Main menu

This screen is reached by pressing Select in the current program screen. It can be exited with Esc. Navigation is done with and . Select opens the page highlighted with the cursor.

Edit programEditing a program
BanksAccess to Program Banks
Pedal layoutPedalboard Configuration
Volume & levelsAccess to Volumes and Levels
Other settingsOther Settings
RestartRestart options

Editing a program

Effect chain

This is the main page to edit a program. Navigations is done with the and keys. Select enters the highlighted object et Esc gets back to the current program screen.

The first line is the program name, which can be edited.

Controllers… gives access to the player/program interactions.

Save to… saves the currently edited program.

The following list is the effect chain, in processing order. Each line is a distinct effet. <Empty> indicates that there is no effect on the line and <End> terminates the list. A line written in bold indicates that at least one controller applies on the effect.

The effect on the highlighted line (including <Empty> and <End>) can be changed with and . These keys sweep all the available effects. If <End> is replaced with an effect, a new empty line is automatically inserted after. In return, it is possible to remove the effect from the last line. Select gives access to the effect configuration.

Finally, the last list shows the effects that don’t need audio to work, like the LFOs for example.

Configurations and parameters are memorized for each kind of effect. If the effect type is changed by mistake, settings are kept and applied back when the initial effect is restored. It’s also possible to try several effects by adjusting their parameters and comparing them without hasle.

Also, an effect of a new kind generally appears with the same configuration it had during its previous instantiation. To restart from default settings, one can use Reset from the Effect Configuration.

Save Program

When a program is saved, the program list from the current bank is displayed.

The very first line indicates the bank where the program will be saved. It is possible to change the bank with the horizontal arrows.

The program slot where the settings shall be overwritten is chosen with Select. The bold line indicates the active program. It is the default selection. Esc cancels the operation and gets back to the effect chain.

Once the saving location is validated, one can type the the program name.

The screen title reminds the storage location. The name correspounding to active settings is pre-filled. One can modify it by navigating in the list of characters. Select validates the highlighted character or action. Esc cancels the operation and gets back to the effect chain.

OKValidates the choice and does the data saving. The green LED lights up shortly during this time.
CANCELCancels the operation.
SPCInsert a space at the carret location.
DELDelete the character at the left of the carret.
and Move the carret.

Parameter list

The first line FX setup… jumps to the Effect Configuration screen. For some effects, there can be a second line Graphic editing… to edit the parameters using a graphic representation. It’s the case for PEq for example.

The following three lines are common to all the effects and deal with the mixing parameters. The value is on the right of the parameter. The following lines are specific to the effect. A parameter written in bold indicates that it is modulated by at least one controller.

One can change the selected parameter with the horizontal arrows, and enter the Parameter Edit screen (more accurate values, controllers…) with Select.


Deactivate the effect when set to On.

Effect mix

Mixes the effect output with its input. The parameters indicates the output ratio, in percent.


Adjustment output volume, in dB. This parameters helps to homogenize the volume, within the effect chain and between the different programs. This volume is applied before the input/output mix.

Parameter Edit

The name, the value and possibly the unit of the parameter are indicated on the top.

One can change the parameter value using four scale factors (the dots) and the horizontal arrows.

The Controllers… line allows to chose the direct links and modulations for the parameters by pressing Select. Bold face indicates that at least one controller is attached to the parameter. The last line is about time parameters. It indicates (if relevant) if the parameters follows the tempo or if it is independant from it.

Controller Choice

Two types of controllers can be selected here:

One can navigate using the vertical arrows and go to the Controller Edit screen with Select. A line <Empty> indicates that there is no controller attached here. It is always there at the end of the modulation list to allow adding a new one. One can go back to the Parameter Edit screen with the Esc key.

Controller Edit

One can setup a controller (direct link or modulation) on this screen. A direct link controller requires two parameter values: a value for the controller’s low position, and another for the high position. The parameter will take all the values between these bounds. By default, the minimum and maximum parameter values are used.

A modulation controller requires only one setting, its depth, in percent of the parameter range.

The first line Src indicates the controller source. One can chose it by cycling within the list using the horizontal arrows. The available sources are the following ones:

<Empty/Delete>Only for editing. Indicates that there is no controller or it is to be deleted.
Expression0–2Expression pedals.
Knob0–4Rotary encoders.

There can be other modulation sources, depending on the program configuration: LFOs, envelope followers… These sources appear first on the list.

Step is only for rotary encoders. It selects its sensitivity. The fractional value indicates the fraction of the parameter range between two consecutive steps. It can be set with the horizontal arrows.

Below, there are the bounding values. They depend on the controller type. They are changed exactly like in the Parameter Edit screen.

Final indicates the final, modulated parameter value. If the parameter is modulated by multiple source, the final value reflects the contribution of all the modulations, not the edited one only.

Curve selects the curve to apply on the source. It changes the controller sensitivity depending on the value ranges. It can be set with the horizontal arrows.

LinearNeutral curve, default.
SquareParabolic curve. Accuracy in the low range, sensitivity in the high range.
CubicLike Square, but more pronounced.
Sq invSaturated curve. It is the opposite of Square: high range is accurate, and low range is sensitive.
Cb invLike the previous one, but more pronounced.
S 1Accuracy in the bottom and top ranges, sensitivity in the mid range.
S 2Same as S 1, but more pronounced.
Flat 1Mid range is accurate, and bottom and top ranges are sensitive.
Flat 2Like Flat 1, but more pronounced.
Prog 1Several curves with more or less progressive slopes.
Prog 2
Prog 3
Prog 4
Sat 1Several curves with more or less saturated slopes.
Sat 2
Sat 3
Sat 4

The curve graphic can be shown with Select.

Range is used only on modulation with some specific sources only. It can convert a unipolar signal (posivite values only) into a bipolar signal (positive and negative values). The signal shape is preserved, the value range is just stretched down to the negative range. This allows putting the neutral modulation value in the middle of the controller course. One can select Bipolar or Unipolar with the horizontal arrows.

Clip restricts the source range. The main purpose is to assign their own variation range to different parameters controlled by a unique source, thus achieving threshold effects. Clip is processed after the Range function. Src← and Src→ clip the modulation source range. These bounds are mapped to new vanues given by Dest↓ and Dest↑. Input values in this raneg are linearly interpolated. Modulations out of this range are clipped to Dest↓ or Dest↑.

Esc exits the screen. Modification are immediately taken into account in real time, there is no specific validation to do.

Effect Configuration

The first line indicates the effect type, which can be changed with the horizontal arrows.

Insert before adds an empty block in the chain before the selected effect. This empty block becomes the current effect and is immediately editable.

Move… moves the effect within the chain. After having pressed on Select, the chain appears and it is possible to move the effect with the vertical arrows. Changes are immediately effective. Select or Esc to exit.

Presets… is still a work in progress…

Reset restores the default parameters.

Chan is useful for effects that can convert a mono signal to stereo (chorus, delay…) Auto selects the stereo, which will be effective only with a global stereo output. prefer mono keeps the mono signal in all the cases. If the input signal is already stereo, it keeps stereo, whatever the selected option. Use Select to change the settings.

State indicates if the effect memory must be cleaned (fresh) when the program is activated, or kept from the previous use (keep). Cleaning the memory may have an impact on the response time but should be light.

Name is the name of the effect instance. It can be used to identify it with global-range pedals and controllers, not depending on the program. For example, you can have a pedalboard-range footswitch to change the bypass state of any effect called “distortion”.

Program banks

Organize banks and programs

Move banks

Move programs

Pedalboard configuration

This part is used to adjust the action of each footswitch at the global level. It is also possible to adjust these settings at the bank and program level, the menus are identical and the configured actions overlap.

The list above allows you to see at a glance what each footswitch does. The twelve footswitches are numbered in two rows: first the top (farthest), from 1 to 6 from left to right, then the bottom (the closest) from 7 to 12. Use the vertical arrows to navigate and Select to edit a footswitch.

In this menu, Full edit… allows editing the content of the footswitch and Clear/empty to reset it. The other functions are not implemented yet.

The menu above gives access to the three footswitch trigger modes and indicates whether or not this mode is used. The modes are the following:

PressImmediate action when pressed
HoldAction after having hold it pressed for about 2 seconds
ReleaseAction when the footswitch is released

The three modes can be used simultaneously to obtain different functions from the same footswitch. However when Hold is used, it is recommended to move the Press content to Release in order to avoid unwanted aliasing.

There is no automatic repeat like on a computer keyboard. A new action is triggered only after the footswitch state is actually changed.

Each triggering mode is linked to a cycle made of one or more Steps. When the footswitch is triggerred, a cycle step is activated, and the next trigger will activate the next step in the cycle, until the cycle is finished and starts again.

In a majority of cases, cycles only have one steps. On the other hand, the Hold and Release modes accept only one step.

The menu above allows you to modify a step. A step consists of one or more actions that will be carried out simultaneously when the step is activated.

Volume and levels

This screen shows and controls the levels. The bars indicate the input and output volumes. When a bar is split, it means that the signal is stereo. This is always the case for the input, even if only the left channel is taken into account in the effect chain. The small The small line indicates the level of the peaks with a stationary time of 2s, and the solid bar the RMS volume.

The DSP bar indicates the CPU resources eaten by the audio processing. The hatched part is the average load, and the solid part the peaks. They should stay as low as possible and never touch the right part of the gauge, otherwise clicks and sound loss may occur.

It is possible to set the Output volume with the horizontal arrows.

The symbols at the left of the DSP gauge indicate if the input and output are mono (one circle) or stereo (two interlaced circles).

When the cursor is on the DSP, the audio engine can be restarted by pressing on Select.

Other settings


Several tuning types can be selected with the horizontal arrows:

GuitarE A110 D G B E, default setting
BassE A55 D G
ChromaticAll the notes of the chromatic scale

The closest detected note in the selected scale is displayed. The LEDs show if the tuning is right or not. The error in cent is also given on the display.


Esc exits the screen.

IV. Available effects

Color Me — Vocal filter

Color Me sculpts the sound spectrum with formants (resonant bumps centered around characteristic frequencies), giving a vocal color. A series of two to four vowels is chosen from a dozen. The main parameter is used to make the transition between the selected vowels, giving a sound tending towards one or the other.

Note: with very accentuated settings, it can be judicious to make follow the effect with a compressor in order to limit too strong resonances.

Vowel morphing

This parameter controls the transition between the selected vowels. It is assigned by default to a free controller.

Formant resonance

Each formant is modeled by a bell filter. This parameter is used to adjust the resonance of this filter.

Formant selectivity

Used to control the selectivity (Q) of the formant filter. It is often best to vary it in conjunction with Formant resonance.

Formant transpose

Allows to shift the formants by more or less one octave. This shift reflects the size of the emulated vocal apparatus and influences the timbre of the synthetic voice, giving something from Jabba the Hutt to Mickey Mouse.

Number of formants

Only two formants make it possible to identify a vowel but the resulting sound is somewhat weird. A third formant gives more clarity and realism. This parameter allows you to choose between two or three.

Number of vowels

Number of voyels transformed by the Vowel morphing.

Vowel 1, 2, 3, 4 type

Vowel type. The vowels of the following list are taken from the French language.


Vowel 1, 2, 3, 4 set

There is three sets of vowels, formant frequencies coming from different studies. This parameter selects the set and allows to get some variations on a give vowel.

CallCalliope (Tubach, 1989)
GDGroupe Didactique (Landron, 2011), Paris area speakers
G&AGendrot and Adda-Decker (2005), radio voices

CompEx — Dynamics compressor/expander

CompEx is a relatively simple dynamic processor and can do compression and expansion.

First, set a threshold level. The for each range above or below this level, one can select if the dynamics will be increased or reduced. Generally “above” is more interesting for compression. “Below” can be used for gates.

Ratio High

Dynamic amplification rate above the threshold. If the rate is less than 1, the effect is a compressor. If it is greater than 1, it’s an expander. In this case the settings are sensitive and the result can be difficult to control.

Ratio Low

Dynamic amplification rate below the threshold. Use a value greater than 1 to get a gate.


The threshold level. The lower, the more the Ratio High effect is pronounced.

Attack time

Attack time for the volume envelope detection, in ms. The lower, the quicker the effect reacts to dymanic changes.

Release time

Release time for the volume envelope detection. Generally greater than the attack time by one order of magnitude.

Make-up gain

Output volume. Although volume compensation is automatically based on the Ratio High value, manual adjustments are sometimes necessary.

Knee shape

The threshold level is actually a range where the dynamic processing changes progressively from the Ratio Low to the Ratio High. This parameter indicates the depth of the range.

Convert To Stereo

It’s not really an effect, it’s a device to help changing a mono source to stereo. The setting in the effect configuration is overriding the effect. So please ensure that you are in Auto.

Delay — Stereo delay

A classic stereo echo effect. It can generate a stereo signal from a mono input.

There are actually two independant delay lines, one for each channel. For each line, one can set the delay time, its output level, the feedback level and the feedback filtering (low- or high-pass). By default, both lines are linked together and use the parameters from the left line.

It is also possible to cross the line feedbacks.

Input mix level

Dry level copied to the output.

Delay mix level

Level of the delayed signal.

Time L/R

Delay time for the left and right lines.

Feedback L/R

Level of the output signal fed back into the lines. When the level is 0, there is only one echo. When it is close to 100%, the echo is repeated almost infinitely.

Filter L/R

Filtering of the line feedback. When the rate is negative, the filter is a low-pass one. When it is positive, the filter is a high-pass one. The parameter is directly linked to the cutoff frequency. The higher in absolute value, the stronger is the filtering.

Channel link

Links the right channel to the left one.


Cross-feedback level. At 0%, both lines are independant. At 100%, the lines are completely crossed, giving the illusion that the sound reflects from left to right and from right to left.

Distortion Tone Stage

This is a standard tone stack often found in distortion pedals, like the DS-1 or Big Muff Pi. It completes a pure distortion stage. It is easier to setup than a complete parameteric equalizer. It achieves a balance between the bass and the treble.

This kind of filter generally has a dip in the mids. Here, it has been modified to add a mid level to change the filtering shape. It is also possible to chose the filtering center frequency, which can be used to mimic the settings of most pedals using this tone stack.


Balances between the basses (low values) and trebles (high values). 50% is balanced.

Mid boost

Amplifies (positive values) or cut (negative values) the mid frequencies. At 0, mids are neutral. For the original circuit tone, use a value near −75%.

Mid frequency

Selects the center frequency.

Double Distortion

This is a two-stage distortion. Each stage is made of a low-cut filter, a bias, the distortion itself and finally a high-cut filter.

But first, the sound is cut into two frequency bands. The low band is not modified, and the high band is distorted.

It has also a system to adapt the volumes depending on the dynamics and to modify the gain depending on the transients.

Crossover freq

This is the frequency for the first split. Thus, it is possible to let the lowest components unaltered for more dynamics and clarity.

LPF freq

Frequency for the initial low-pass filter (2nd order).

Attack gain mod

This parameter sets how the attack transients modify the gain. A positive value increases the gain on attacks, and a negative one reduces it.

Sustain gain mod

Sets how a release modifies the gain, to prolongate or reduce the note sustain. A positive value increases the gain when the note fades out.

Stage 1/2 HPF freq

Cutoff frequency of the high-pass filter for stages 1 and 2.

Stage 1/2 Bias

Bias for the stages 1 et 2. When the bias is increased in absolute value, the distortion becomes more asymetric and favor the even harmonics. But beware, when the bias is too high, the signal can be cut, which gives an effect of defective circuit.

Stage 1/2 Type

Type of distortion for each stage.

Stage 1/2 Gain

Distortion gain for each stage.

Stage 1/2 LPF freq

Cutoff frequency of the low-pass filter applied after the distortion of each stage. This filter attenuates the harsh frequencies generated by the signal shaping.

Stage 1–2 mix

Mixes both stage outputs in the output signal. The percentage is related to the second stage.

Low band mix

Ratio fo the clean low band in the output.


Adjusts the output volume to the input volume. A null value totally compensates the volume, and 100% keeps the distortion untouched. A low value gives more percussive and less natural sounds.


Threshold below which the volume is not reduced by the Density parameter.

Envelope Follower

This effect is not an audio effect but a controller working by analysing the incoming sound. It can be inserted at any point in the processing chain, it will not modify the sound.

The envelope follower detects the instantaneous sound volume and turns it into a control signal.

Attack time

Detection attack time. This is the speed at which the envelope follows volume increases.

Release time

Detection release time. This is the speed at which the envelope follows volume decreases.

Hold time

Time during which the detector holds its value after an attack, before releasing.


Multiplier applied to the control signal.


Threshold below which the output of the detector keeps quiet. In logarithmic mode, the threshold is used as a floor value, with a minimum of −60dB.


Selects how the volume is turned into a control signal. In linear mode, the raw volume value is directly used. In logarithmic mode, this is a decibel value. The range is set by the Threshold parameter.

Clip envelope

Indicates that the control signal is limited to a maximum of 1.

Clip source

Sets the level below which the source signal is clipped. This parameters helps to neutralize the strongest attacks when the release time is significant, limiting the induced delay between the actual peak end and the envelope release. When the parameter is set to the maximum, the clipping is deactivated.

Low-cut frequency

Cutoff frequency for the low-cut filter. This filter decreases the influence of the bass frequencies in the detection, giving an envelope more reactive to guitar strums. When set to the minimum, the filter is deactivated.

Filter Squeezer — Distorted resonant filter

This is a very specific filter. It is based on the famous Moog 4-pole low-pass filter, and heavily modified.

It gives a sound ranging from something warm and vintage to something very fuzzy via a defective radio or a feedback interference. Some sounds are comparable to those obtained with the Fuzz Factory.


Filter cutoff frequency. This parameter is automatically assigned to the expression pedal.


Resonance of the filter, occurring slightly below the cutoff frequency. The highest values put the filter in self-oscillation.


The color setting influences the destructive and fuzz aspect of the filter.


Adjusts the input level. The filter being very sensitive to the input signal volume, one can obtain very varied sonority by changing this parameter. The inverse gain is applied at output, limited between −12 and 0dB. It can also be wise to place in front of a compressor with strong settings to obtain something somewhat predictable and reproducible.


Activates different types of internal distortion.

FlanCho — Chorus and flanger

It is a relatively classical chorus/flanger. It can generate a stereo signal from a mono input.

The effect works by adding a slightly delayed version of itself to the main signal. The delay continuously varies which creates tone fluctuations.

The chorus is characterized by a significant delay time (around 20ms) and a low depth. For a flanger, a high depth, a short delay time and a minimum feedback are preferable.


Speed of the oscillations of the delay time.


Amplitude of the oscillations of the delay time. At most, these vary between 0 or almost Delay×2. The tonal change is then very pronounced.


Average delay time.


Feedback ratio of the delayed signal. Values close to 100% can alter significantly the volume. Negative values give a more metallic sound than the positive ones.

Waveform type

Shape of the oscillations for the delay time.

Inv.paraInverse parabola
Ramp up
Ramp down
RandomSine with random variations

Waveform shape

How the waveforme is deformed. At 0%, it keeps the orignal shape. Otherwise, it is saturated by the bottom or by the top. It emphasise the longest or shortest time delays.

Number of voices

It is possible to have 1 to 4 chorus/flanger voices simultaneously, for a more or less thick sound. The delay oscillations of each voice are not synchronized with the others.

Phase set

Allows the phase of oscillations to be set manually, in degrees. This parameter allows on the one hand to act on the tone of the effect by associating for example a pedal (expression or not). On the other hand, it provides reproducible sound. Indeed, when the program is activated, the phase is automatically set to the value given by this parameter.

Dry input

Indicates that the input signal is mixed with the output signal. If we want a ratio different of 50–50%, one can put this parameter on Off and use the Effect mix parameter. When the original signal is totally missing and there is no feedback, the obtained effect is a vibrato.

Mix mode

Select whether the delayed signal is added to or subtracted from the original signal. Add causes an amplification of the low frequencies with a very short delay time, whereas Sub cuts them.

Oversampling rate

Oversampling rate for the processing. When set to ×4, shortest delay time can be achieved (a tenth of microseconds) when Depth is close to 100%, which can be useful for devastating flangers. However, it eats more CPU resource.

Freeverb — Reverberation

This effect implements Freeverb, a Schroeder reverberator which coefficients have been carefully tweaked. It has been designed by Jezar at Dreampoint and released in the public domain. The effect is able to generate a stereo signal.


This parameter is the reverberation time. The higher, the more reflective are the simulated walls, and the longer the reverberation. This time is also dependent on the Damping.


Specifies at which point the high frequencies are attenuated during the reverberation.

Level wet

Level of the reverberated signal in the effect output.

Level dry

Level of the original signal in the effect outpout.


Stereo spread of the effect. At 0%, the reverberation is completely mono, and completely stereo at 100%. Note: both channels are always processed independantly, like two mono reverberations. The parameter only controls the output mixing.


When the mode Freeze is activated, the input of the reverberation is cut and the reverberated sound is frozen. It can be interresting to link this parameter to a footswitch.

Low cut

Cutoff frequency of the low-cut filter on the reverberation output.

High cut

Cutoff frequency of the high-cut filter on the reverberation output.

Frequency Shifter

This is a pretty simple effect that shifts all the frequencies of the sound. As the frequencies are added and not multiplied, the harmonic ratios are lost and the sound becomes atonal or inharmonic, approaching a bell sound. However, when the shift frequency falls on a specific note, some linked notes (octave, fifth, quarte...) start to sound right again and take a very special color.

The frequency translator looks like the ring modulator, but it does not reflect the spectral image as is the case for the latter.


Shiftig frequency, in Hz. It can be negative or positive.

Harmonic Tremolo — Two-band tremolo

This is a variation of the tremolo effect found in old Fender Brownface amplifiers. The sound is divided into two bands, bass and treble. The two bands are modulated in amplitude by two opposite signals from the LFO. The sound alternates between treble and bass.


Oscillation speed, in Hz.


Amount of oscillations. At 100%, the modulated sound is silent on the lowest part of the LFO. It is possible to go far beyond this value in order to create a harder effect, sounding like a gate.


Amplitude of the LFO signal used to modulate the bass frequencies. At 0, the sound is not modulated.


Amplitude of the LFO signal used to modulate the treble. At 0, the sound is not modulated. The best results are obtained when the amplitudes of the bass and treble are of opposite signs.

Cutoff frequency

Split frequency between the bass and the treble, in Hz.

Gain saturation

Indicates to what extent the positive parts of the LFO should be saturated in order to avoid exaggerated volume rise when Amount takes high values.


Offset to apply to the values generated by the LFO to shift its curve in the low or high amplitudes. This creates a more or less dense sound, especially when Amount is high. In this case, the tremolo behaves like a gate and this parameter allows to set its opening time.


Allows you to adjust the ratio between bass and treble. Negative values reduce high frequencies and positive values reduce low frequencies.


When the effect output is stereo, this parameter widens the stereo image, especially when the source is mono. Modulation is done with an opposite sign on both channels. Be careful, if the sound goes back to mono later, the effect is lost.

Waveform, Sample and hold, Smoothing, Chaos amount, Phase distortion amount, Phase distortion offset, Sign, Variation 1, Variation 2, Phase set

These parameters are identical to the ones from the LFO.

Input Impedance Fix

This effect allows to correct the input impedance of the sound card, which can be a bit low, resulting in a slight attenuation of the treble. It is a simple first-order linear high-shelf filter.

The effect should be calibrated using an amp with a setting as transparent as possible (clear sound, no equalization). Then we compare the sound of the guitar directly connected with the one coming from the pedalboard with just this effect.

Cutoff frequency

Cutoff frequency.


Treble amplification level.

Noise Bleach — Noise gate

Noise Bleach is a kind of noise gate. It splits the input signal spectrum into adjacent frequency bands covering the entire audible spectrum. The sound on each band passes only if it is above a certain threshold.

To adjust the device, you need the noise without any other “useful” sound. Start by setting the global level to −100dB. Then, for each band, gradually raise its threshold. Stop when sound changes and noise is actually removed. Do not go further, this would be harmful for the audio content.

Repeat on as many bands as necessary. If required, balance the global level with the single band levels.

Global level

This is the overall filter rejection level, to be adapted according to the noise level. All filter levels are scaled according to this parameter. Usual values are in the −100dB range.

Band x-y level

Relative filter threshold for the specified frequency range, in dB. At −∞, the band filter is disabled.

Noise Chlorine — Noise gate

Noise Chlorine is a kind of noise gate, similar to Noise Bleach. It consists of a series of notch filters that reject the target signal only below a certain threshold. For example the 50 or 60Hz from the mains and its harmonics. But it can also be used as a broadband noise filter as a cheap alternative to Noise Bleach.

To adjust the device, you need the noise without any other “useful” sound. Start by setting the global level to −100dB. Then, for each band, select a frequency (intuitively). We can start with a selectivity of 0.5 which is broad enough to start with.

Gradually raise the band threshold. Stop when sound changes and noise is actually removed. Do not go further, this would be harmful for the audio content. Then adjust the frequency and selectivity as needed. If the noise is spread over a large part of the spectrum, the selectivity can be kept low by spacing all bands by one octave.

Repeat on as many bands as necessary. If required, balance the global level with the single band levels.

Global level

This is the overall filter rejection level, to be adapted according to the noise level. All notch filter levels are scaled according to this parameter. Usual values are in the −100dB range.

Notch 1–8 frequency

Notch filter cutoff frequency, in Hz.

Notch 1–8 selectivity (Q)

Filter selectivity. The higher the selectivity, the thinner the filtered band.

Notch 1–8 level

Relative filter threshold for the specified frequency range, in dB. At −∞, the band filter is disabled.

Onset detector

This effect is not an audio effect but a controller working by analysing the incoming sound. It can be inserted at any point in the processing chain, it will not modify the sound.

It can detect the beginning and the end of notes or chords and turn this information into a control signal to modulate other effets. It is actually very simple and can handle only one note or chord at a time. Actually, it’s more a strum and silence detector.

There are two output signals:

  1. Attack signal. Takes a value > 0 if an attack is detected, 0 otherwise. The value is related to the detected velocity.
  2. Release signal. 1 if a release is detected, 0 otherwise.

Note: the detector can exhibit a few milliseconds of latency.

Velocity clipping

Indicates if the detected note velocity should be clipped to a maximum of 1.

Attack threshold

Minimum level for an attack to be detected, in dB. The lower, the more sensitive.

Attack ratio

Volume increase to detect an attack, in percent. The lower, the more sensitive, but can lead to more false triggers.

Release threshold

Maximum level for a release to be detected, in dB. The higher, the more sensitive.

Release ratio

Volume decrease to detect a release, in percent. The lower, the more sensitive, but can lead to unexpected note ends.

Parametric Equalizer

It is a classic 4-, 8- or 16-band parametric equalizer (depending on the selected option). Each band is made of a second-order filter.

Band 1/2/3/4 Type

Filter type for the band:

PeakBell curve. Boosts or cuts the area around the frequency.
Low shelfLow-frequency shelf. The reference frequency is the middle of the transition between the main part and the shelf.
High PassHigh-pass filter
High shelfHigh-frequency shelf
Low PassLow-pass filter

Band 1/2/3/4 Frequency

Band refrence frequency.

Band 1/2/3/4 Q

Band selectivity or quality factor. For the bell filter, the higher Q, the thinner the bell. For shelves, the higher Q, the steeper the slope. But if Q>0.71, a resonance and a notch start to show at both ends of the shelves. Same with the high- and low-pass filters, with a resonance only.

Band 1/2/3/4 Gain

Gain of the bell or shelf No effect on the high- and low-pass filters.

Band 1/2/3/4 Bypass

Deactivate a single band.

Édition graphique

You can also adjust the equalizer settings with an immediate graphical preview.

The parameters can be scrolled as usual, they are displayed one by one at the bottom left. The very first parameter R gives the display vertical scale, in dB.

The dots at the bottom right give the precision of the changes obtained with the and keys. This precision can be changed with Select. Esc to go back to the main parameter screen.

Phaser AP

Classic phaser, made of a cascade of all-pass filters with feedback.


Oscillation speed.


Ratio of phased signal in the final mix.


Amount of feedback from the phased signal. The higher the feedback, the more pronounced the effect.

Number of stages

Order for the phasing filter.

Minimum frequency

Minimum cutoff frequency for the phasing filter, in Hz. The oscillator make the filter cutoff frequency vary between the minimum and maximum frequencies.

Maximum frequency

Maximum cutoff frequency for the phasing filter, in Hz. By putting the maximum frequency at the same value as the minimum frequency, it is possible to stop the oscillation and to use the phaser as fixed filter for coloring the sound.

Feedback source

Number of the stage from which the feedback signal is taken.

Waveform, Sample and hold, Smoothing, Chaos amount, Phase distortion amount, Phase distortion offset, Sign, Variation 1, Variation 2, Phase set

These parameters are identical to the ones from the LFO.

Phaser HT

This modulation effect works a little differently from phasers but uses the same principles. However, it has a very typical sound. It can generate a stereo signal from a mono input.

An optional band-pass filter allows the selection of a range of frequencies to be kept from the phased sound. This essentially makes it possible to limit the low frequencies when the settings are a bit strong.

It is also possible to obtain a “barberpole phaser” effect, which gives the impression of having tones that go up or down endlessly.


Oscillation speed.


This is the number of stage of the phaser, amongst 4, 8, 16 et 32. The number of stages changes the general sound color. The larger, the greater the number of notches.

Feedback level

Amount of signal fed back to the phaser input. This feedback is saturated, so it may be interesting to push the input signal level a little to enrich the result.

Feedback color

Changes the feedback tone color.

Phase mix

Amount of phase-shifted signal added to the original signal. Adding both is what produces the phasing effect. This effect is maximum at 50%.

Phase offset

This parameter allows the phase of the oscillations to be moved relative to its natural cycle.

Phase set

Manually sets the phase of oscillations, in degrees. It is interesting to modulate this parameter with an external LFO in frozen mode. It gives an oscillating effect closer to a conventional phaser.


Freezes the phaser oscillations.

BPF freq

Middle band-pass filter frequency, for the phased signal.

BPF selectivity

Selects more or less accurately a band of frequencies with the band-pass filter. When this parameter is minimum, the filtering is deactivated.


Direction of the oscillations on the left channel. Down makes the notches go down, and Up makes them go up.

Mono phase mix

Specifies how the phase-shifted signal should be mixed with the main signal when the output is mono. Left uses only one of both signals, leading to a tone going always in the same direction indicated by Direction. Mixed mixes both signals going up and down, producing a kind of oscillation.

Stereo phase mix

Indicates how the phased signal should be processed when the effect output is stereo.

Spat mixInput channels are mixed before being phased. The spectral notches in the left and right channels sweep in opposite directions, which gives a strong stereo image, including with a mono input signal.
Spat sepLike Spat mix, but the input channels are processed independently.
Bi-monoLeft and right channels are processed independantly and use the Mono phase mix setting to specify the direction of the notch sweeps.

All-pass delay

This parameter allows to introduce a kind of complementary micro-delay in the phase-shifted signal. This increases the number of sweeping notches in the spectrum. The spectral distribution of these notches is guided by the All-pass coefficient value.

All-pass coefficient

Coefficient of the phase-shift filters used to achieve a micro-delay. At 0, the filter behaves like a pure delay. A negative value groups the notches on the bottom of the spectrum, and a positive value on the top. When the coefficient is −0.5, with a delay of about 700µs and a low value for Depth, the notch distribution is almost harmonic, which makes it possible to obtain an audio illusion similar to the Sherpard-Risset.

Pitch Detector

This effect is not an audio effect but a controller working by analysing the incoming sound. It can be inserted at any point in the processing chain, it will not modify the sound.

The pitch detector can retrieve the pitch of the played note and turn it into a control signal to modulate other effects. It work only on monophonic, pitched sounds. Chords and non-pitched sounds are not supported.

Minimum frequency

Minimum detected frequency, in Hz. Any note below this frequency will be ignored. This parameter impacts the processing power taken by the effect.

Maximum frequency

Fréquence maximum de détection, en Hz. Any note above this frequency will be ignored.

Output type

Indicates how the detected note is formated.

PitchThe note is given in octave. 0 is the Middle C and −10 indicates that the detection has failed.
FreqThe note is given in kHz (kilohertz). When the detection fails, it outputs the frequency of the last known note. If there isn’t any last known note, the signal is 0.

Pulse Width Modulation Distortion

This is an almost infinite-gain distortion, the output signal is rectangular. Only the rising wave fronts are taken from the input signal. Downward wave fronts are generated automatically after a fixed period of time, which gives the sound a “PWM oscillator” color, like with a synthesizer.

Pulse width

Pulse duration, in milliseconds. The shorter the duration, the more filled the sound spectrum is, especially in high frequencies, and the lower the apparent volume.

SiemensGirlz — Écho créatif

SiemensGirlz is a relatively complex creative delay effect for obtaining a wide variety of sounds, especially when the parameters are modulated by LFOs.

The delay line effects are as follows, in this order:

In addition, a ducking system attenuates or mutes the output of the feedback lines according to the volume of the input, for example to improve the clarity of the original sound while keeping the delay effect when the input is silent.

General parameters

Tap gain in

Amount of source signal that passes through all taps, in percent. Once automated, this parameter allows to cut or reactivate the input of the taps.

Tap global volume

General volume of all taps in the final mix, in dB.

Dry volume

Original sound volume in the final mix, in dB.

Dry spread

Distribution of the original sound in the two delay lines. At 0%, the sound goes in the first line exclusively, and at 100%, in the second line. The parameter has no effect when only one line is activated.

Freeze lines

Cuts the input of the lines and increases their feedback level to 100%. This freezes the delay lines, repeating their content infinitely. The sound continues to be gradually altered by the line effects.

Number of feedback lines

Number of feedback delay lines, 0, 1 ou 2.


Cross-feedback level. At 0%, each line is independent. At 100%, the lines swap their feedback signal.

Ducking sensitivity

The ducking reference sensitivity, in decibels. When ducking is activated on a line, the line is switched off when the volume of the input signal exceeds a certain threshold. This parameter sets the latter.

Ducking time

Time constant for the volume analyzer. It indicates the attack time. The release time is automatically deducted.

Tap parameters

Tap input gain

Input tap level, in percent.

Tap spread

Distribution of the tap output in the two delay lines. The parameter works as Dry spread.

Tap delay time, base

Main tap delay time, in milliseconds.

Tap delay time, relative

Multiplier on the main delay time. The mapping of this parameter is linear, which make easier the creation of special effects by automating it.

Tap pitch

Tap pitch shifting, in semi-tones. This is done by internally producing two alternately overlapping resampled taps. Requires a minimum delay on the tap to be effective.

Tap low-cut frequency

Cutoff frequency of the low-cut filter applied between the tap output and the final mix.

Tap high-cut frequency

Cutoff frequency of the high-cut filter applied between the tap output and the final mix.

Tap pan

Panoramic balance for the tap output in the final mix.

Feedback delay line parameters

Line input gain

Line input gain, percent.

Line delay time, base

Main line delay time, milliseconds.

Line delay time, relative

Multiplier on the main delay time. The mapping of this parameter is linear, which make easier the creation of special effects by automating it.

Line speed

Speed of the BBD (bucket brigade device) emulator, in percent. Delay times specified in the previous parameters are valid when the parameter is set to 100%. When the line is faster, delay times become sorter. They become longer when the line goes slower. Change the line speed makes possible to achieve characteristic pitch effects.

Line pitch

Line pitch shifting, in semi-tones.

Line feedback

Line feedback level, percent. The higher, the longer the echoes repeat.

Line low-cut frequency

Cutoff frequency of the low-cut filter applied between the line output and the final mix.

Line high-cut frequency

Cutoff frequency of the high-cut filter applied between the line output and the final mix.

Line pan

Panoramic balance for the line output in the final mix.

Line volume

Line output volume in the final mix.

Line ducking amount

Ducking amount applied to the delay line. The effect depends on the Ducking sensitivity and the Ducking time parameters.

Line filter type

Type of resonant filter used in the feedback processing chain.

PKBell filter
LPLow-pass filter
BPBand-pass filter
HPHigh-pass filter

Line filter frequency

Cutoff frequency of the resonant filter, in Hz.

Line filter resonance

Resonance of the filter, in dB.

Line filter selectivity

Selectivity (Q) of the resonant filter.

Line distortion amount

Distortion amount in the feedback chain, percent.

Line distortion foldback

Shape of the distortion. Changes the harmonic content created by the distortion.

Line high shelf frequency

High-shelf filter cutoff frequency, applied after the distortion.

Line high shelf level

Level of the high shelf, dB. Can damp the harsh frequencies created by the distortion.

Line frequency shifting

Shifting frequency for the frequency shifter, in Hz. 0Hz is neutral (no shifting).

Line reverb mix

Amount of reverberated signal entering the delay line, in precent.

Line reverb decay

Parameter related to the reverberation time. The higher, the longer the reverberation.

Line reverb damp

Damp the high frequencies in the reverberation tail. At 100%, the damping is maximum. This parameter shorten the actual reverberation time.

Simple Distortion

It is a quite simple asymmetric saturation, with a high-pass filter at the input in order to remove DC and to clean up a little the bass.

Distortion gain

Distortion gain in dB.

HPF frequency

Input low-cut filter cutoff frequency, in Hz.


Polarization bias. Increasing the bias allows a more asymmetrical distortion and thus favours even harmonics. Beware, a too high bias tends to cut the signal and gives the impression of a defective circuit.

Speaker Emulator


Selects a type of emulation among those proposed. Mid and Treble tone controls only work for the type 0.

Mid level

Level of the mid-high bump.

Mid frequency

Frequency of the mid-high bump.

Treble level

Level of the treble.

Treble frequency

Frequency separating the treble drop from the mid-high bump.

Comb level

Level of the comb filter. Helps to make the spectral shape a bit more complex.


Indicates on which channels the emulation applies. Can be used to direct a channel on a mixing desk and another one on a guitar amp.


The tremolo effect is a kind of oscillating volume, modulated in amplitude by an LFO.


Oscillation speed, in Hz.


Amount of oscillations. At 100%, the signal is silent on the lowest LFO values. It is possible to go far beyond this value in order to create a harder effect, sounding like a gate.


LFO waveform:

Saw Up
Saw Down

Gain saturation

Indicates to what extent the positive parts of the LFO should be saturated in order to avoid exaggerated volume rise when Amount is high.


Offset to apply to the value generated by the LFO in order to move its curve in the low or high amplitudes. This creates a more or less dense sound, especially when Amount is high. In this case, the tremolo behaves like a gate and this parameter allows to set its opening time.


It is a wah-wah effect more or less modelled on the Cry Baby pedal. The range for the cutoff frequencies has been extended, which makes it possible to obtain a devastating sound in the low frequencies.


Filter cutoff frequency. This parameter is linked by default to the expression pedal. To get the sound of a classic wah pedal, reduce the interval to about 400–1600Hz.


Filter resonance. The higher the value, the more coloured the filter, generating a distinct tone.

Wah-wah (Cry Baby)

This effect is a precise emulation of different Cry Baby wah-wah models. The modeling of these pedals and the resulting algorithm were written by Transmogrifox.


Position of the pedal, which corresponds to the cutoff frequency of the filter. This parameter is linked by default to the expression pedal.


Selects the Cry Baby model.

Control ramp — Parameter transition

This effect is only available in the control signal generator category since it does not process any audio signals, either input or output. Its purpose is to create a signal that gradually changes from 0 to 1 or 1 to 0.

The transition is initiated as soon as the effect is activated. However, it can be restarted at any time by setting the Set position parameter.


Transition duration, in seconds.


Transition amplitude, in %.


Transition curve.

LinearSmooth transition
Acc 1Slow at the beginning, faster at the end
Acc 2
Acc 3
Acc 4
Sat 1Fast at the beginning, slower at the end
Sat 2
Sat 3
Sat 4
Fast 1Fast at the middle
Fast 2
Slow 1Slow at the middle
Slow 2

Sample and hold

This parameter adds steps to the transition curve by regularly blocking its value for a short period. It is expressed as a ratio between the blocked period and the total transition time.


Parameter that softens abrupt transitions. It determines the ratio between the rise time and the transition time. This parameter is made to work with Sample and hold.


Gives the transition direction. Standard direction is from 0 to 1. Inverted, it is from 1 to 0.

Set position

Sets the current point in the transition, as a percentage of progression. So it can be restarted if it was already completed. If you put an initial delay and want to restart the ramp immediately, set the position slightly above 0.

Initial delay

Time in seconds before the ramp kicks in after. The delay phase is started again when the position is set to exactly 0.


This parameter is meant to pause the transition.

LFO — Low frequency oscillator

This effect is only available in the control signal generator category since it does not process any audio signals, either input or output. Its purpose is to create an oscillating signal in order to modulate one or more parameters from another effect.

The LFO effect has a graphic display, showing one or more cycle of the current waveform. The first parameter in the graphic editor is the number of cycles shown. It is followed by the standard LFO parameters.


Oscillation frequency, en Hz.


Oscillation amplitude, percent.


Waveform generated by the LFO. Each form is described below:

This form regularly alternates between two out-of-phase sines. It is controlled by two additional parameters:

Variation 1TimeControls how an elementary period is split.
Variation 2PhasePhase between the two subwaves.

16 parts, phase quadrature:

32 parts, slight dephasing:

This form is an extension of the Biphase, consisting of a variable number of out-of-phase sines. The LFO moves regularly from one phase to the next one, each in turn. The phases of the subwave are fixed and evenly distributed. Therefore the phase shift depends on the number of phases.

Variation 1TimeControls how an elementary period is split.
Variation 2PhaseNumber of phases.

28 parts, 2 phases:

16 parts, 3 phases:

It is a wave that can embody different shapes located between the square, the triangle and the sawtooth. The precise shape can be controlled using the two auxiliary parameters.

Variation 1TimeDetermines the time position of the first extremum of the curve relative to the entire period.
Variation 2ShapeShape of the transitio between two extrema.

Time = 25%, Shape = 0%:

Time = 25%, Shape = 50%:

Time = 25%, Shape varying from 0 to 100%:

This is white noise filtered by a low pass at 12dB/octave, whose cutoff frequency depends on the Speed parameter.

Sample and hold

This parameter creates steps in the LFO curve by regularly blocking its value for a short period. It determines the ratio between the blocking period and the main oscillation period.


Parameter that softens abrupt transitions in the waveform. It determines the ratio between the rise time and the main oscillation period.

Chaos amount

This parameter introduces a chaotic element in the oscillations. It randomly accelerates or slows down the oscillator, ensuring that it always falls back on its feet so that the overall rhythm is maintained. It's a very simple way to add a little life to an oscillation. Below the same phase distortion applied to two waveforms:

Phase dist amount

Phase distorsion. This parameter is used to move the time position of the middle of the waveform. It corresponds to a pulse width setting. Here, on a sine:

Phase dist offset

Parameter that allows to change the fixed points as well as the one moved in the phase distortion. Below, the same deformation as previously by moving the point located at 3/4 (lower part of the wave):


Invert the LFO sign. It is used, for example, to transform a rising sawtooth into a descending one:


Switching to unipolar. This parameter moves up and compresses the curve so that the bottom is at zero. This mode can simplify the modulation rate settings because the minimum of the LFO corresponds to the basic value of the modulated parameter.

Variation 1, Variation 2

The meaning of these two parameters depends on the selected waveform.

Phase set

Manually sets the phase of the LFO, in degrees. The value 360 does not cause any change. This is useful when you want to apply a preset without changing the current phase.

V. Troubleshooting

Coming soon… (or not?)

VI. Change log

r14, 2018-08-26

r13, 2018-05-28

r12, 2018-04-03

r11, 2018-03-22

r10, 2017-08-01

r9, 2017-07-17

r8, 2017-06-14

r7, 2017-04-24

r6, 2017-04-22

r5, 2017-02-11

r4, 2017-02-11

r3, 2017-02-07

r2, 2017-02-05

r1, 2017-01-28

r0, 2016-11-29