M6 Docs MuLab Modules   


For a detailed description of MuSynth, MuDrum, MuPad, MuSampla, MultiSampla and MuVerb, MuEcho, click here.

Common Info

  • Every module that has a parameter also has an event input for receiving parameter events that automate the parameter. So even if a certain module has nothing to do with events, for example the audio amplifier, it still has an event input to receive automation events.
  • Every parameter of every MuTools module is renamable. This may be handy when you use the parameter on a MUX front panel, but also to make more explicit patches.
  • Note that the M6 sound system uses automatic mono/stereo signal management. Whenever an input is expecting a mono signal and you route a stereo signal to it, the stereo signal is locally (within the jack) converted to mono. And the other way around too of course.

Audio Generators


Generates a timbre using a digital waveform.

This oscillator works anti-aliased so even with high pitches you get a crystal clear sound You can draw your own waveforms by click-dragging in the waveform display. You can also import almost any wave file as a waveform via a right-click on the waveform display -> Open, or via drag-dropping a wave file from the OSX finder / Windows explorer. Note that the full imported wave file is regarded as a single cycle.

The oscillator needs note events so it knows at what pitch it should play. So best to connect its event input. Otherwise it defaults to playing a C3.

AIPS means Add Inverted Phase-Shifted. Simply put: with a Saw, this causes Block PWM, cfr the picture below where waveform 2 is the inverted and phase-shifted version of waveform 1. But instead of limiting it only to saw waveforms to create traditional PWM, the MuTools oscillator's AIPS can be used with any waveform! And the phase-shifting can be modulated, e.g. by an LFO.

Via the "Layers" button you can make the oscillator generate multiple layers of the same waveform. This way you can easily create super fat sounds and make chord sounds etc. The "Layers" button pops up the layer editor, which can be used in "Quick Edit" and "Deep Edit" modes. Oscillator layers setups can also be saved+recalled as presets.

By default the oscillator starts at a random start phase to get a more analog natural sound. If you want a more digital sound, you have the option to fix the start phase from 0 to 359 degrees. This is also editable via the "Layers" button.

To apply pitch modulation, make sure the Pitch Range parameter is non zero for the pitch range defines the maximum pitch modulation depth. For example when the incoming modulation signal is 100% and pitch range is 1200 cents then the modulation is 1200 cents.

The portamento section has 4 aspects: (hold mouse to see the tool tips)

  • Time = The time to glide from one pitch to another.
  • Legato Glide = When enabled portamento is only applied when two notes are played legato i.e. the second note starts before the first ends.
  • Poly Glide = When enabled the glide is applied to the last voice only, otherwise the glide is applied to all voices, which definitely is a different effect.
  • Curve = The pitch glide curve. For example linear, or fast exponential or slow exponential, or any other of the many curve options.


Multi Wave Oscillator

See this separate doc page: Multi Wave Oscillator


Noise Generator

Generates white noise.


Sample Player

Plays a sample taking the Start-Loop-End markers into account.

  • To load a sample, left-click on the sample display.
  • You can also drag-drop a sample file onto that display.
  • Double-click the sample display to edit the sample.
  • Once you've loaded a sample, you can use the previous/next sample buttons to step thru the samples on your system! So practical!
  • You can choose a start, loop and end marker and a loop mode.
  • Managing these markers is done via the audio editor's time bar, see the audio lab doc page for more details.
  • Note that you can also use markers that are in reversed order for reversed playback!
  • Also read the info on the oscillator module as it includes common info that also applies to this module.


Multi Sample Player

Plays a multisample. A multisample is a collection of key and/or velocity zones that each play a sample. For detailed info on editing multisamples, click here.

Dropping a sample onto the Multi Sample Editor will create a new full range zone for that sample.

Also read the info on the oscillator module as it includes common info that also applies to this module.


Test Sine Generator

Generates a monophonic sine signal.



The PolySynth is a modular synth engine featuring 48 note polyphony. The PolySynth is essential when you want to build polyphonic synth patches. It also features a special Mono Mode for creating monophonic sounds.

For all details on the Poly Synth, click here.

Audio Processors


1P Lowpass

The One Pole Lowpass filter is a soft first order lowpass filter with a -6 dB/octave slope.

The pivot key for the key tracking is note number 60 = middle C. Note that if you want to use the key tracking, the filter needs note events so it can track the pitch.


Allpass Filter

Passes all incoming audio frequencies equally, but changes the phase relationship between the various frequencies.



Amplifies and pans the incoming audio.


Audio Level Meter

Displays the audio level.


Audio Limiter

Limits the incoming audio using a logarithmic clipper.


Audio Inverter

Inverts the polarity of the incoming audio signal.


Audio Pos/Neg Splitter

This module splits the incoming audio so that the positive part is on output 1 and the negative part on output 2. This can be used, for example, to apply different distortions to positive and negative part.

Or you could hook up an Audio Inverter on audio 2, then mix that with out 1 of the Pos/Neg Splitter, then you have an unipolar positive audio signal that can be fed into the Ring Modulator. Several possible applications, and maybe you'll find even more!


Balancer 1->2

Routes the incoming audio to the outputs 1 and 2 based on the balance parameter. The curve parameter defines the crossfade curve. More on curves on the Curves doc page.


Balancer 2->1

Routes the incoming audio inputs 1 and 2 to the output based on the balance parameter. The curve parameter defines the crossfade curve. More on curves on the Curves doc page.


Bit Reducer

This module will degrade the used number of bits to encode the signal level. By degrading the number of bits you can get that vintage dirty sound from game consoles in the 70s and 80s.


Level Compressor

Compresses the dynamics of the incoming audio.

The compressor has 2 audio inputs. When both inputs are connected, then input 1 = the real audio signal, input 2 = the control audio signal (aka side-chain) which will drive the compressor. When only in 1 is connected real audio and control audio are the same.

Below the threshold the audio is untouched. Above the threshold the audio will be compressed. The softer the 'knee' the less explicit the compression will be.


Mixer Strip

Combines a volume slider, a pan control, a mute and a level meter.


Mono Echo (Long)

Echoes the incoming audio.


Mono Echo (Short)

Echoes the incoming audio.


Modular Feedback Delay

Delays the incoming audio and supports a feedback loop thru a MUX patch in which you can put any module you like:

When sync mode is on the delay time is expressed in MIDI Clocks, which is 1/24th of a beat. The table below shows some typical values:

MIDI ClocksMusical
11/96 = 1/64T
21/48 = 1/32T
41/24 = 1/16T
81/12 = 1/8T
161/6 = 1/4T
321/3 = 1/2T


Multi Mode Filter

Filters certain frequencies in the audio signal.

The pivot key for the key tracking is note number 60 = middle C. Note that if you want to use the key tracking, the filter needs note events so it can track the pitch.



The oscilloscope shows the audio waveform. Scroll the mouse-wheel over the display to zoom in/out on the waveform. Right-click its display to set the time range etc.

It also supports a special sync mode: When the event input is connected the oscilloscope will sync its display to the incoming note events, resulting in a stable waveform display for that note frequency. This is very handy to see how eg a filter is modifying a waveform.


Pure Delay

Delays the incoming audio. Maximum delay is 2 seconds, so in case the samplerate is 48000 Hz the maximum delay in frames is 96000. This delay module can be used to manually compensate the process delay of other plug-in modules in a sample-accurate way. It supports both mono and stereo modes.


TF Lowpass Filter

24 dB/oct lowpass filter with support for high resonance. Sounds different, more "vintage synth" than the Multi Mode Filter.

The pivot key for the key tracking is note number 60 = middle C. Note that if you want to use the key tracking, the filter needs note events so it can track the pitch.


Ring Modulator

Multiplies the audio inputs 1 and 2.


Samplerate Reducer

This module will degrade the samplerate to encode the signal over time. By degrading the samplerate you can get that vintage dirty sound from game consoles in the 70s and 80s.


Stereo Combinor

Audio input 1 will become the left side of the stereo signal, audio input 2 the right side.

Note that the M6 sound system uses automatic mono/stereo signal management. If you route a stereo signal to one of the inputs, it's first converted to mono.


Stereo Splitter

Splits the left and right sides of the incoming audio into mono outputs 1 and 2.

Note that the M6 sound system uses automatic mono/stereo signal management. In case you would route a mono signal to the Stereo Splitter then it's interpreted as a stereo signal and so the mono input will appear at both outputs.


Tanh Distortion

Applies a TanH math function on the incoming audio, resulting in interesting distortion effects.



Normally audio signals vary within the -100% to +100% range and so it's a bipolar signal. This module will make a the audio signal vary within the 0% to 100% range and thus make it a unipolar signal. This is done by applying the formula out = (in + 100%) / 2. So if the in signal = -100% it becomes 0%. If input is +50% it becomes +75%. No clipping is applied in case the input signal is outside the -100% to +100% range. If you want to ensure that an audio signal is within the -100% to +100% range you can use an Audio Limiter module with Threshold to 0 dB, Intensity to 100% and Out Gain to 0 dB.

Event Generators


Audio Envelope Follower

The Audio Envelope Follower tracks the incoming audio and generates an output modulation signal from it depending on the Attack and Release settings. The Attack and Release parameters define how fast the modulation envelope reacts on the incoming audio stream. At the same time it also outputs a note-on event whenever the envelope goes beyond the threshold level, and a note-off whenever the envelope goes below the threshold level:

Note: Try to avoid the case where the Audio Envelope Follower's attack and release times are zero while RMS window is off, cause that means that each sample will create a new event and envelope point which is not ok. So or leave the RMS window on, or set attack or release to at least 25ms.


Parameter Event Generator

The Parameter Event Generator generates a new parameter event whenever the (modulated) parameter value changes, taking the defined resolution into account. The higher the resolution the more possible values and the more accurate the ouput but also the more events are generated which will have an impact on CPU consumption. The impact on CPU consumption also depends on what type of module and parameter you're targetting.


MIDI Controller Generator

The MIDI Controller Generator generates a new MIDI controller event whenever the (modulated) parameter value changes, taking the 128 step MIDI resolution into account.


Note Event Pad

The Note Event Pad module generates a note event as long as it is clicked. It's like a single drum pad. Also useful to (re)trigger effect patches that need a key trigger. It can be integrated into a MUX front panel and it is recordable.


Piano Keyboard

The Piano Keyboard module simply generates note events as you play them on the keyboard.

Important note wrt recording the keyboard: By default the resulting track will target this keyboard module specificly so that the played back notes sound exactly the same as how you recorded them. But in the special case that this keyboard module's event input is directly and exclusively routed from the event input of the main module/rack it belongs too, then that main module/rack will be the target module, which is more comfortable. For in this special case it is sure that the played back notes arrive unaltered at the keyboard module and so everything will still sound the same as during recording. In all other cases it is not sure that the playbacked notes would arrive unaltered at the keyboard module (eg there could be a key transposer in between), and so that's why that specific keyboard module must be targetted directly then.

The above is just background info, you don't need to think about it when making music, the app will automatically make the correct choice.


Sequence Player

The Sequence Player plays a sequence. You can choose one of the sequences in your project. To edit the sequence double-click it or click the edit button at the right.

The Start Mode defines when the sequence is (re)started:

  • On Start Playing = On the first note on the sequence starts playing, then when new note ons are received while the sequence is already playing then the sequence is not restarted.
  • On Every Note On = Each received note on will restart the sequence.

The Stop Mode defines when the sequence stops playing:

  • On Same Note On = The sequence will stop if a note on is received on the same key as the last key.
  • On Last Note Off = The sequence will stop when all notes are released.
  • Never = The sequence is never stopped.

When Sync is enabled then when the sequence starts playing it is synced to the host's sequencer. If the host's sequencer is not playing, no syncing can occur of course, and so the sequence starts playing at its start.

The Auto Transpose defines whether the sequence should be transposed by the input key. The base key is C3 = note nr 60, so when the Sequence Player receives a D3 it will transpose the sequence 2 semitones up.

Drag-dropping a MIDI or MuSequence file onto the sequence field will load that file into the sequence.

Event Processors


Drum Note Processor

Incoming notes are dispatched to 1 of the 12 drum pads and outputted to 1 of the 12 event outputs. Playing the drum pads also generates the relevant notes.


Event Delay

Delays the incoming events between the minimum and maximum delay times.

The random varies around the actual delay time. The amount of variation is defined by the Random parameter. The maximum variation (100%) is the smallest of these 2: (Max Delay Time - Actual Delay Time) and (Actual Delay Time - Min Delay Time)


Event Monitor

Bypasses and displays the incoming events.


Event Recorder

When switched on, the event recorder records the incoming events into a new sequence. The event recorder is synced to the main record button in the transport panel. So when the event recorder is enabled and you start recording, all incoming events are recorded and when stopping recording, a new sequence is created. When switching the event recorder off while recording, you can temporarily mute recording the incoming events, but recording mode stands by until the main record button in the transport panel is hit again. Only when the main record button in the transport panel is switched off, then the event recording is finished and a new sequence is created. The target module of the new recording is the first module after the event recorder itself because that way when playing back the recorded sequence it will sound the same as before.

A typical example: Insert an arpeggiator in rack slot 1, an event recorder in slot 2 and a synth in slot 3. Now switch off the event recorder so it's only bypassing events. Setup the arp as you want. Now when you want to record the arp in a real sequence, switch on the event recorder, hit record in the transport and record as long as you want, then stop recording. A new sequence part with the rendered arp is added to the composition.


MIDI Channel Remapper

Using this module, each of the 16 MIDI channels can be remapped to another one.


MIDI Channel Splitter

Events are dispatched to one of the 16 outputs based on their MIDI channel.


Monophonic Note Tracker

Processes the incoming notes in such a way that only a single note is outputted, keeping track of any simultaneous notes and outputting new relevant note on/off events when you add/release notes. Can be used to create mono-type sounds, even for VSTs. Note: There is another way to create monophonic sounds: By limiting the Polyphony of the PolySynth to 1. For more details, click here.


Note Dispatcher

This module dispatches the incoming notes (events) to one of the outputs depending on the mode:

  • Random: Whenever a note-on is received, a random output is selected.
  • Round Robin: Whenever a note-on is received, the next output is selected. And when a note-on with Special Key is received, the output is reset to the first one.
  • Keyed: Whenever a note-on with Special Key is received, the next output is selected.
  • Key Range: The note-ons in the key range from Special Key to Special Key + Num Outs define the output.


Note Key Ranger

Will make sure that the incoming notes are inside the defined range. There are two strategies:

  • Round Robin: Output key = Min Key + (Input Key modulo (Max Key + 1 - Min Key)). For example if the key range is C3-C4 all C keys will become C3, all F keys will become F3 etc.
  • Proportional: Output key = Min Key + ((Input Key/127) * (Max Key - Min Key)). The result is rounded to the nearest note. For example if the key range is C3-C4 then C5 will become 60 + ((84/127) * 12) = rounded to 68 = G#3.


Note Key Splitter

Dispatches the incoming events to 1 of the 12 outputs based on the note key, i.e. C notes go to output 1, G# notes to output 9 etc.

This module can be very useful to create drum patches where each of the 12 keys triggers another sound. And there may be more creative uses for this module!

All other non-note events are routed to output 1.


Note Key/Vel Filter

This module only bypasses the notes that are inside the defined key and velocity range. All other notes are blocked. By combining a couple of these modules you can create keyboard splits and/or velocity dependent synth layers, for example.


Note Mapper

Maps the incomming notes to 1 or more out notes. Can be used to apply note scaling or to generate chords.

This module only maps the 12 keys of a single octave, the C3 octave. Then all keys of all other octaves also use the same relative key mapping but taking their own octave into account. For example if you map the C3 key to a major chord, then C2 will play a C2 major, C4 will play a C4 major and so on.

The "Key Transpose" transposes the output notes for the selected key. That makes it easy to select a chord and quickly try it out on different keys.


  • The whole Note Mapper setup can also be saved as a preset file. This is handy to create preset chord combinations, for example.
  • The copy-paste keyboard shortcuts also copy-paste the selected chord.
  • You can drag-drop a MuChord file on the input keyboard to apply that chord to that input key.
  • Even when you have chosen/dropped a MuChord file from outside the library, the previous/next chord buttons will select the previous/next chord in that relevant folder(s).


Note Modifier

Modifies the key and velocity of the incoming note events.

Modulation Generators


ADSR Envelope

Generates a Attack-Decay-Sustain-Release modulation envelope.

Note that the ADSR won't do much if you don't feed it with note events, as the ADSR curve is triggered by a 'Note On' event!

If the ADSR is unsufficient for what you want to do, then replace it by a Multi-Point Envelope, see below.



Generates a low frequency modulation signal.


  • Frequency/Rate: controls how fast the waveshape is running.
  • Start Phase: controls the start phase of the waveshape. It can also be set to Random.
  • Amplitude: controls the height of the waveshape, and thus also the intensity of modulation on the connected modules.
  • Start Fade: controls fading the amplitude in/out when the LFO starts, eg when a note-on is received.
  • End Fade: controls fading the amplitude in/out when the LFO received a note-off.

Both the Start Fade and the End Fade can do a fade-in or a fade-out. When the fade parameter value starts with < then it's doing a fade-in, when the fade parameter starts with > then it's doing a fade-out. The End Fade will always continue fading from where the Start Fade ended. So if you release the note in the middle of the start fade, then the end fade will start its fade-in or fade-out at 50% of the amplitude. There is one special case: When Start Fade is an immediate fade out (> 0 ms = "Inactive"), then the End Fade is the only applied fade and it will start the waveshape with the start phase and do its fade-in from 0% or its fade-out from 100% of the defined amplitude.

The LFO waveshape is fully editable and can be saved/reloaded via MuWaveShape preset files. Double-clicking the waveshape display opens the waveshape editor.

A wave shape envelope is expressed in frames, but the time positions in a wave shape envelope are simply relative to the loop end point. It is the loop end point that defines the full cycle. So setting the loop end to eg 360 frames makes it easier for editing as then it matches the 0 - 360 degrees of a waveform. But you're free to set it as you want. For this reason there is the context function "Resize Shape". Note that the wave shape length has no effect on the final quality, the final quality is always perfect. That's because the wave shape is vectorial using 64 bit floating point values.

The LFO also is syncable to the tempo, if you want. When Tempo Sync is enabled, then the frequency parameter is expressed in Cpb = Cycles Per Beat. To set an exact value double-click the parameter value or right-click and choose "Edit Value". Then you can enter "2" to set it to 2 Cpb. You can also input "1.5 bpc" (Beats per cycle) which will automatically do the conversion calculation to Cpb. (= 0.66667 Cpb) You can also input "1/4", "3/8" etc which will automatically be converted to the correct Cpb value.

More info tips:

  • Right-clicking the waveshape display shows the context menu for this waveshape.
  • Clicking the left/right buttons blow waveform display steps forward thru the preset waveshapes in the factory and user library (MuWaveShape sub-folder). If you have just changed one these library folders, close+reopen the editor to refresh the list of preset waveforms to step thru.
  • Remember that for doing fine adjustements of parameter values, hold [Ctrl], cfr the User Interface doc page.
  • Each time the LFO receives a note on event, it restarts the waveform cycle.
  • By putting an LFO outside the PolySynth and using its signal inside the PolySynth, you can create a global LFO that is common to all voices.


Multi-Point Envelope

The Multi-Point Envelope can generate very complex modulation signals. You can add as many points in the modulation envelope as you want. For all details about the how to shape the envelope, click here.

To define a section that must be looped, right-click the points that must be the start and end points of the loop and choose "Set Loop Start" + "Set Loop End". And when you have a looped section you have these options for what must happen on note-off:

  • Keep looping: The note-off has no influence, the loop keeps looping.
  • Finish Looping: The current loop will be played until loop end, then the Multi-Point Envelope continues the section beyond the loop, as if there was no loop.
  • Stop Looping: The loop is immediately stopped, and the Multi-Point Envelope continues the section beyond the loop.

The Multi-Point Envelope has 4 speed parameters:

  • Global Speed: Controls the global speed for the whole modulation envelope.
  • Attack Speed: Controls the speed from the start until loop start.
  • Loop Speed: Controls the speed from loop start until loop end.
  • Release Speed: Controls the speed from loop end until end.


  • To create a 'sustain' section like in an ADSR, put a loop on a single point.
  • The 'Finish Looping' mode should not be combined with curve types that do not start on 0% and end on 100%.
  • If the Multi-Point Envelope is not connected to an event source it will start playing automatically, else it needs a note event trigger.


Wobble Generator

Generates a randomly changing modulation signal. Changes are within the bounds you indicate. This module can be used to add an 'analog feel' to your patches, for example by routing a bit of wobble to the oscillator pitch modulation input. (make sure the oscillator pitch modulation depth is non zero, cfr the oscillator info)

More in detail: The 2 "Freq" parameters define the amount of time needed to travel from one top to another. When a new top is reached a new random time (between min and max) is set to reach the next top. And so on. Alt Offset means that the offset will alternate between positive and negative with each new wobble. So for example, if Amplitude is 0% and Offset is 100% and Alternating is on, then this results in a positive top of 100% followed by a negative top of -100% followed by a positive top of 100% and so on. The curve defines how the value slides from a previous top to the next one. So when you want a steady amplitude but a random timing, set Amplitude to 0% and Offset to 100% (or some other value) and enable Alt Offset.


Constant Modulator

Generates a constant modulation signal. This can be handy for example to map a meta-parameter to a modulation signal. Therefore map the meta-parameter to the Value parameter. Then when you tweak the meta-parameter, the modulation output will change accordingly.

Modulation Processors


Modulation Mapper

With the Modulation Value Mapper, you can transform the incoming modulation values so that the output modulation values are smaller/larger and/or have inversed polarity:

In the above picture example, we've set the Offset to 0 % and the Maximum to -50%.
This way the level of the input signal is reduced to 50% AND inversed.

First the curve is applied, then the offset and amplitude.

Event To Modulation Converters

The Note/Controller/Aftertouch/PitchBend To Modulation Converters convert the incoming events into a modulation signal.

The Note To Modulation Converter has a Low and High Key which define the note key range. Notes lower than low key will generate the minimum value, notes higher than high key will generate the maximum value. If you don't want that notes outside the defined key range cause a modulation value update, insert a Note Filter module before the Note To Modulation Converter.



Audio Input

Where the audio comes into MUX Modular. By adding audio input modules, this MUX also gets more audio input jacks!


Audio Output

Where the audio goes out MUX Modular. By adding audio output modules, this MUX also gets more audio output jacks!


Event Input

Where the events comes into MUX Modular. By adding event input modules, this MUX also gets more event input jacks!

Note that MUX Modular needs at least 1 event input jack in order to receive parameter automation events.


Event Output

Where the events go out MUX Modular. By adding event output modules, this MUX also gets more event output jacks!


Modulation Input

Where the modulation comes into MUX Modular. By adding modulation input modules, this MUX also gets more modulation input jacks!

Tip: You can use a modulation input in the PolySynth to input a global LFO modulation signal which will be applied to all voices at the same time!


Modulation Output

Where the modulation go out MUX Modular. By adding modulation output modules, this MUX also gets more modulation output jacks!



Audio File Recorder

The Audio File Recorder simply records the incoming audio signal.

To actually record, make sure the audio recorder is enabled (aka armed), and click the Record button in the transport panel. (or press the proper shortcut).

Double-click the Audio File Recorder to edit its settings: From left to right:

  • Drag-drop handle: Can be used to drag-drop the audio recorder eg on a track, so to setup track based audio recording.
  • Enable (aka arm) button: Switch on to actually get ready for recording.
  • Monitor switch: 3 options: Off, Monitor When Enabled, Monitor Always. Monitoring means that the input signal is streaming thru to the output.
  • Name: The name of this audio recorder. This name will also be used as base name for the recordings, then also a YYYYMMDD-HHMMSS timestamp is added.
  • Record From: The input source.
  • Output To: The output destination.
  • Chans: Defines whether the recorded file must be mono or stereo.
  • AN: Auto-Normalize defines whether the recorded file must be normalized to maximum volume after recording.

Also read the doc page about Recording.

Extra notes:

  • The Audio File Recorder is only available in the Project Modular Area, not (yet) in the MUX modular areas.
  • The Audio File Recorder has 2 audio inputs: The first is the audio input that is recorded (and monitored). The second audio input is purely bypass thru the audio output and is used for playing back track audio parts.



Racks are described in detail here.



The incoming signal simply goes to output 1, but part of the signal also goes to output 2.


Patch Point

Is a generic plug-in that simply bypasses the incoming signals.
Can be used as a target module from which you can then dispatch the audio/events to one or many other target modules.

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