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For a detailed description of MuSynth, MuDrum, MuPad, MuSampla, MultiSampla and MuVerb, MuEcho, click here.
- Every module that has automatable parameters also has an event input for receiving parameter automation events.
Even if a certain module has nothing to do with events, for example the audio amplifier, it still has an event input to receive parameter 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 M8 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.
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 MacOS finder / Windows explorer. Note that the full imported wave file is regarded as a single cycle.
You can also edit a waveform's harmonics (via drawbar sliders) via right-click -> Edit Harmonics.
Note that the first time you tweak a harmonic slider this might give a bigger change in the original source waveform as only the first 64 harmonics are preserved and the original harmonic phases are discarded.
The oscillator needs note events so it knows at what pitch it should play.
When used in the PolySynth, oscillators do not generate any sound until they received a first note-on event. So make sure to connect its event input.
When used outside a PolySynth, oscillators default to playing a middle C (C4).
Pitch Modulation
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.
Note that when doing audio-rate pitch modulation, there may be a shift in pitch perception.
This is no bug, it is mathematically correct and it even depends on the modulator waveform shape.
There are 2 solutions: Or you compensate the pitch shift by re-tuning the oscillator.
Or use phase modulation (see below): Audio-rate phase modulation results in a steady pitch.
Phase Shift
The Phase Shift section lets you control the phase of the waveform.
There are 4 phase shift modes:
- Off = No phase shift
- Source Only = Only the source waveform itself is phase-shifted, no extra layer.
- Add Layered = A second version of the waveform is phase-shifted and added to the original source.
- Invert Layered = A second version of the waveform is phase-shifted, then inverted and then added to the original source. (In M6 and before this was known as AIPS)
When modulating the phase on audio-rate you'll get real phase modulation, which can produce very rich timbres like the Yamaha DX and Casio CZ series etc.
By using the "Invert Layered" phase shift mode on a sawtooth waveform and modulating the phase-shift parameter with an LFO, you can generate Block PWM, cfr the picture below where sawtooth 2 is the inverted and phase-shifted version of sawtooth 1.
Super Layers
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.
The Drift parameter is expressed in cents and when for example set to 10 the pitch of that layer will drift between -10 and +10 cents around the main pitch each time a new note is played.
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 can fix the start phase from 0 to 359 degrees.
This is also editable via the "Layers" button.
Note that when playing sustained notes and meanwhile changing the layer setup, you won't immediately hear the changes.
The new layer setup only becomes effective upon playing new notes.
Portamento
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.
Notes
- Via the options menu -> Properties, you can change the Key Follow property. For example when Key Follow is set to 0, the pitch is independent from the key.
- Waveform options menu -> Save As: When you choose a .txt file the waveform will be exported as numerical values. (Integer 16 Bit / Float 32 Bit)
See this separate doc page: Multi-Form Oscillator
Generates white noise.
Plays a sample taking the Start-Loop-End positions into account:
It starts playing at the Start position, then plays until the End position, then if loop is enabled it starts looping between the Loop and End positions.
When the End position comes before the Start position the sample is played in a reversed way.
Note that the Start-Loop-End positions features 2 different ways: Straight Sample Index or Marker.
For details see this doc section.
Time stretching
You can also apply time stretching to the sample.
Time stretching means that the sample is played faster or slower but without changing the pitch.
See detailed info here: Time Stretching.
Notes
- To load a sample, left-click on the sample selector field.
- You can also drag-drop a sample file onto that field or on the sample 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! Very 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.
- Via the options menu -> Properties, you can change the Key Follow property. For example when Key Follow is set to 0, the pitch is independent from the key.
- Spectral stretching very short slices may sound a bit chorussy/blurry. To avoid that use longer slices or use granular stretching.
Plays a multi-sample.
A multi-sample is a collection of key and/or velocity zones that each play a sample.
For detailed info on editing multi-samples, click here.
Notes
- Dropping a sample onto the Multi-Sample Editor will create a new full range zone for that sample.
- Dropping a multiple samples onto the Multi-Sample Editor will create as many new zones as the number of dropped samples.
- The maximum number of simultaneous layers for a single key is 12.
- Also read the info on the oscillator module as it includes common info that also applies to the Multi-Sample Player.
- Via the options menu -> Properties, you can change the Key Follow property. For example when Key Follow is set to 0, the pitch is independent from the key.
This module plays a sample grain = a fraction of a sample.
The great thing is that the start and length of the grain can change dynamically, creating all kinds of special effects.
The sample start-end markers define the region in which the grain can start.
The Grain Start parameter works within this region.
The Grain Length parameter goes from almost zero to Max Length.
There are 4 modes:
- Playback I = Standard sample playback as if you're playing a sample, but than in short snippets = the grains.
- Playback II = Same as Playback I but the grain length is independent of the played pitch, so the grain loop will always have the same rhythm.
- Waveform = The grain is handled as a waveform. So you can play the grains in a melodic way, just like an oscillator.
- Resonator = The grain length is set so that when the grain is looped it will sound at the relevant pitch.
It is a bit similar as Waveform mode, but it's different. Note that in this mode the Grain Length parameter is unused.
The Crossfade amount defines how much crossfade smoothing must be applied while grains are dynamically changing. Without crossfade, intense changes in grain start/length can cause clicks.
Grain Random defines the jitter on the timing of the grain. Expressed in microseconds so it can be used subtly.
The Attack and Decay parameters apply an amplitude envelope on the grain.
The Attack parameter goes from 0% to 100% of the grain length, while the Decay parameter goes from 0% to 1200% of the grain length, where the very maximum value maps to no decay at all.
Note that when the Decay is eg 500% it does not mean that the grain will continue playing polyphonically with the next grain, it only means that the decay slope is less, thus resulting in a more constant amplitude than eg Decay 100%.
That way when the Decay parameter increases towards maximum, the sound will change naturally towards no decay at all at the maximum. Note that the decay only starts when the attack has finished.
Also see this video: www.youtube.com/watch?v=zTRq8trsHgw
Notes
- Via the options menu -> Properties, you can change the Key Follow property. For example when Key Follow is set to 0, the pitch is independent from the key.
Generates a monophonic sine signal.
The PolySynth is a modular synth engine featuring 64 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.
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.
Passes all incoming audio frequencies equally, but changes the phase relationship between the various frequencies.
As with all parameters you can double-click the parameter value to set it by text input.
If you enter "0.5" it will be 0.5 seconds. If you enter "0.5 ms" it will be 0.5 milliseconds.
Amplifies and pans the incoming audio.
Displays the audio level.
Limits the incoming audio level using the defined curve.
The X axis is the input level, the Y axis is the output level, the grid lines show the 0 dB levels.
When applying a soft gentle curve it can be used to smooth peaks.
When limiting the audio more drastically this will result in distortion, which may be a useful effect.
Inverts the polarity of the incoming audio signal.
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!
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.
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.
Routes the incoming audio input to 1 or more of the outputs based on the incoming note events.
The Root Key defines which key will select the first audio output, all other keys will select the other audio outputs relatively to the root key.
The Attack and Release Time define how fast the audio starts/ends when a certain audio output is (de)selected.
Solo Mode:
- OFF = You can play multiple notes at the same time and the audio will go to multiple outputs.
- Gated = The last incoming note will select a single audio output. After the last note has been released there will be no audio output anymore.
- Continuous = The last incoming note will select a single audio output. After the last note has been released the last selected output will continue to play.
When Use Velocity is on, the velocity of the incoming notes will also define the gain for the audio output.
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.
This audio filter has a very steep cutoff slope and a distinctive sound.
It features lowpass, highpass, bandpass and bandstop filters with in 7th, 11th, 15th, 19th and 25th order versions.
| Frequency Spectrum Analyser |
The Frequency Spectrum Analyser shows the frequency spectrum of the incoming audio.
The left side border is fixed at 20.0 Hz, the right side is fixed at the Nyquist frequency which is half the samplerate.
Note that digital signals cannot contain frequencies higher than the Nyquist frequency so it would also make no sense to show higher frequencies than the Nyquist frequency.
You can zoom in and out the level scale by scrolling the mouse wheel.
Right-click the display for more setup options, eg to set the display to dual channel mode, or to enable the mouse info display that shows the frequency and level at the mouse cursor.
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.
The RMS switch defines whether the control audio level is measured by peak level or by RMS level.
When RMS is on it uses a 1 ms RMS window on the control audio signal to measure its level.
Combines a volume slider, a pan control, a mute and a level meter.
Echoes the incoming audio.
Echoes the incoming audio.
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 Clocks | Musical |
| 1 | 1/96 = 1/64T |
| 2 | 1/48 = 1/32T |
| 3 | 1/32 |
| 4 | 1/24 = 1/16T |
| 6 | 1/16 |
| 8 | 1/12 = 1/8T |
| 9 | 1/16D |
| 12 | 1/8 |
| 16 | 1/6 = 1/4T |
| 18 | 1/8D |
| 24 | 1/4 |
| 32 | 1/3 = 1/2T |
| 36 | 1/4D |
| 48 | 1/2 |
| 72 | 1/2D |
| 96 | 1/1 |
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.
Delays the incoming audio.
Maximum delay is 2 seconds, so in case the samplerate is 48000 Hz the maximum delay in samples 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.
The module resonates the incoming audio.
Besides the pitch parameter it also responds to incoming note events and so the resonation frequency can be played by a keyboard or sequence.
Channel Mode and Key Follow are self-explanatory properties.
Reset Mode defines when the Resonator should reset any past resonations and restart with a clean sheet.
Invert FB = Invert Feedback = defines whether resonations are inverted in amplitude each time they bounce the wall.
Also see This Video
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.
Multiplies the audio inputs 1 and 2.
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.
Audio input 1 will become the left side of the stereo signal, audio input 2 the right side.
Note that the M8 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.
Splits the left and right sides of the incoming audio into mono outputs 1 and 2.
Note that the M8 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.
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.
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.
This module is practical for MUX patches that need a first (note) event in order to work as expected, eg. to trigger a looped envelope.
The defined event is sent out whenever this module (re)starts processing:
- When it is inserted into a MUX patch
- When it is loaded from a preset file
- When its process switch (the green on/off led) is switched back on
| Parameter Event Generator |
The Parameter Event Generator generates a new parameter event whenever the (modulated) parameter value changes.
It's recommended to first connect this module's event output to the target module's event input before choosing which parameter to select, for it's that connection that makes it possible for this Parameter Event Generator to know what parameters you want to choose from.
| 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.
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.
Both "Momentary" and "Latch" modes are supported.
| Parameter Value Randomizer |
Upon receiving a note-on, this module randomizes the selected parameters within a defined range.
It's recommended to first connect the module to the target module you want to randomize so that this module knows about the target parameters.
This module can also randomize any VST plug-in.
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.
The Pitch Bend Generator generates a new Pitch Bend event whenever the (modulated) parameter value changes, taking a 1000 step resolution per polarity into account, thus 2001 possible values in total.
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.
Start/Stop Modes
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.
Sync Modes
There are 2 start modes:
- Start = Start from sequence start.
- Sync = Start from within the sequence, synced to the 'master' play position.
Then there are 3 sync timing options
- Instant = The sync action is done immediately.
- Next Beat = The sync action is done upon the next beat.
- Next Loop = The sync action is done upon the next step sequence loop.
All together this results in the 6 options listed in the Sync Mode field.
Auto Transpose
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.
You can also drag-drop a sample on a Sequence Player module. This will generate an audio sequence for that sample.
Arpeggiator Mode
Please see the Step Sequencer info. It uses the same Arpeggiator Mode.
Notes
- When used in inside the PolySynth and only connecting the Sequence Player to the event input, the Arpeggiator mode will not work as you might expect because the event input only inputs the note event that created the polyphonic voice.
The Step Sequencer module is a very creative tool to experiment with and generate all kinds of sequences like drum patterns, melodic riffs, chord progressions etc.
It features 12 lanes of 32 steps. Each lane can be a single note, a (strummed) chord, or even a micro sequence on its own.
The Step Sequencer plays based on the note events it receives. When playing the Step Sequencer via MIDI the step sequence is automatically transposed so you can play melodies of sequences, a great way to generate new musical ideas.
At the same time the Step Sequencer also features multiple patterns, each with its own loop, and state of the art sync modes that allow realtime switching between patterns and come up with original dynamic sequences.
The Step Sequencer also features Modulation Groups which let you group certain steps and vary (and automate!) the pitch, velocity and length of the grouped steps on the fly!
More specs of the Step Sequencer:
- Each step can be easily toggled on/off, transposed, etc.
- Supports multiple MIDI channels hence multi-timbral sequences.
- Unlimited number of pattern, each with its own lane setup, loop length and step length.
- Seamless pattern switching, recordable.
- Includes 4 lanes for sequencing MIDI Controller/Aftertouch/Pitchbend. Pitchbend features an auto-glide function.
- Arpeggiator mode with the same creative power and flexibility.
- Can be driven by another step sequencer, which can be driven by another one... Unlimited creativity.
Details
- There are 2 panes: The note pane and the controller pane. The note pane has 12 lanes. The controller pane has 4 lanes.
- Double-click a lane header to set it up. A note lane can be setup to up to 5 notes, thus also supporting strummed chords or micro sequences!
- The definition of the lanes is per pattern. So each pattern can have a different lane setup. Note that there is a pattern context function "Copy Lane Setup To All Patterns" in case you want to have the same lane setup in all patterns.
- [Alt]+click on a lane header monitors that lane.
- Both the note and controller pane have several tab buttons that define which step values are being edited.
- Click a step to (de)active it.
- Step values can be changed using the scroll wheel, or click-dragging a step value vertically.
- Horizontal drag (de)activates multiple steps in that lane.
- You can create, select, duplicate, copy-paste, open, save patterns via the pattern selector and options button.
- Each pattern has its own loop and "Steps/Beat" setup. The loop points can be moved by drag-dropping the loop start/end indicators.
- [Alt]-drag loop start/end will move the entire loop.
- Patterns can be selected via MIDI by the lowest notes: Note nr 0 selects pattern 0, note nr 1 selects pattern 1, ...
That way you can play patterns on the fly, even multiplex them on the fly, all using the selected sync mode.
- Note that real-time pattern switching is not available when the step sequencer is inside a PolySynth. When playing new notes they will play the current selected pattern until that voice has ended.
- You can use the Step Sequencer outside a PolySynth or inside a PolySynth. It results in different behavior. When used outside a PolySynth the notes generated by the Step Sequencer will each generate new voices in the synth that follows (unless that's a mono synth of course), but when used inside a PolySynth then the notes generated by the Step Sequencer will be used within a single voice.
- When used in inside the PolySynth and only connecting the Step Sequencer to the event input, the Arpeggiator mode will not work as you might expect because the event input only inputs the note event that created the polyphonic voice.
- When using the step sequencer inside a PolySynth the play LEDs will show the position of the first active voice.
So when playing a note (voice) and then hitting a second note you'll see the LEDs for the first voice.
If you then release the first note and it ends (Main Envelope has finished) then the LEDs will show the position of that second voice which has become the first active voice now.
- Each step can be set to one of 5 Modifier Groups A - E. Each Modifier Group has an automatable parameter for modifying the pitch, velocity and length of the respective steps.
- Auto Transpose is not available when Arpeggiator mode is on because then the input notes are used to make the arpeggio.
- Pressing [Insert] or [Delete] while hovering the mouse over the step LEDs will insert/delete a step in all lanes.
- Pressing [Insert] or [Delete] while hovering the mouse over a lane step will insert/delete a step in that lane.
- When inserting a new step sequencer module and there exists a "New.MuStepSequencer" preset file in the user or factory library, that preset file is automatically opened as default setup.
- The maximum value for Length and Offset values is the pattern's loop length. The knob for these values uses a non-linear scale so that the most frequently used values have enough editing space, while still reserving space for the whole value range up to the maximum, which is the pattern's loop length.
- The 100% step length value is shown as the 50% display value. The values above 100% up to the maximum are shown in a proportianal way.
Sync Modes
Same as the Sync Modes of the Sequence Player module.
Arpeggiator Mode
In arpeggiator mode, the step sequencer notes are replaced by the incoming notes. The replacement is done like this:
- C4 is replaced by the 1st input note.
- C#4 is replaced by the 2nd input note.
- D4 is replaced by the 3rd input note.
- ...
- C3 is replaced by the 1st input note, but then 1 octave down.
- D5 is replaced by the 3rd input note, but then 1 octave up.
If there are less input notes than used notes in the step sequence, the input notes are re-used using the 'modulo' function.
For example if the step sequence uses C4, C#4, D4 and D#4 in some rhythmic pattern, thus targetting the first 4 input notes, but you only play 2 notes to the step sequencer, then D4 will be replaced by the first input note again and D#4 will be replaced by the 2nd input note again.
This all happens on the fly.
More Properties
Via the options button at the top of the editor -> "Properties" you can set important properties for this step sequencer module.
Note that these properties are for the step sequencer module and thus common to all patterns of this step sequencer.
Start/Stop Modes
The Start/Stop modes act the same as the Start/Stop Modes of the Player module.
Pattern Notes = Select & Play
When ON then when receiving a note that selects a pattern then this note will not only select the pattern but also immediately play the pattern.
This mode is not compatible with Auto Transpose or any Arpergiator mode.
Monophonic Editing
When ON then there can only be 1 note per step. Toggling another note will automatically switch off any other notes for that step.
Show Velocity Knobs
When ON then the velocities will be shown as a knob instead of a bar.
Step LED Color
Let you choose the color for the step LEDs at the top of the grid.
Generates MIDI controller events for the X and Y axes.
Using the options button in the top right you can setup this module.
Generates parameter events for the X and Y axes.
Using the options button in the top right you can setup this module.
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.
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)
Bypasses and displays the incoming events.
The [Delete] key clears the display.
"BT" = Block Time. Audio apps work in audio blocks, requested by the audio driver.
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.
Using this module, each of the 16 MIDI channels can be remapped to another one.
Events are dispatched to one of the 16 outputs based on their MIDI channel.
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.
This module dispatches the incoming notes (events) to one of the outputs depending on the mode:
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.
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.
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.
This module lets you define key zones and each such zone can be assigned to one of the event outputs.
So, for example, zone C3 to B3 goes to output 1, C4 to B4 goes to output 2 and C5 to B5 goes to output 3.
These event outputs will only send out the note events. The last output will bypass any non-note events like pitch-bend etc.
That way you have precise control to what should happen to these events.
For example you can also connect it to all outputs so to send thru pitch-bend.
| MIDI Controller Event Pad |
Generates a MIDI controller event when clicking the pad, and another MIDI controller event when releasing the pad.
Both "Momentary" and "Latch" modes are supported.
Maps the incomming notes to 1 or more out notes.
Can be used to apply note scaling or to generate chords.
Can also be used to remap notes for example complex drum setups.
Input Mode = Octave Mode
In this mode this module only maps the 12 keys of a single octave, the C4 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 C4 key to a major chord, then C3 will play a C3 major, C5 will play a C5 major and so on.
Input Mode = Full Range
In this mode you can map all 128 keys of the keyboard.
Key Transpose
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.
Key Name
Each key can be named, eg a chord name, drum sound, or any other relevant name.
This will be visible in the sequence editor and makes it more easy to recognize mapped keys.
Tips
- 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).
Modifies the key and velocity of the incoming note events.
This module will create a new note-on event whenever a note-off event is received.
You can define the length of that new note-on.
When set to "Relative" the note-on will last as long as the note that triggered it, or half/twice as long etc, as defined by the Length parameter.
So Millisecs and MIDI Clocks will result in a fixed new note length, Relative will result in a variable new note length.
You can also transpose the new note-on.
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.
Note that when an ADSR envelope is retriggered before it has ended, it re-starts at its current level, so to avoid unwanted clicking.
This cannot happen when the ADSR is used inside the PolySynth, but it can happen when used in an effect patch.
Generates a low frequency modulation signal.
Parameters:
- 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 samples, 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 samples 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.
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.
Notes:
- 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.
- When inserting a new MPE module and there exists a "New.Mudulator" preset file in the user or factory library, that preset file is automatically opened as default setup.
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.
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.
Generates modulation signals for the X and Y axes.
Using the options button in the top right you can setup this module.
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.
Displays the modulation signal level.
Samples and holds the input value upon every incoming note-on.
The Transition Time and Transition Curve properties control the transition between values.
This makes it possible to apply smoother values transitions which sometimes is necessary for nicely modulating certain parameters without clicking/artifacts.
| Audio To Modulation & Modulation To Audio Converters |
By default modulations, like an LFO to pitch to create vibrato, are done on a low frequency rate, typically in the 0.1 - 30.0 Hz range.
Doing faster and/or more complex modulations can result in sonic artifacts and inefficient CPU use (a warning alert pops up in that case) as the standard modulation system is not designed for high frequency modulations.
However using the Audio To Modulation Converter and Modulation To Audio Converter modules, modulations can also be done on audio rate.
Almost all parameters effectively support audiorate modulation with only a few exceptions. (eg ADSR Attack/Release Speed, LFO Frequency)
Being able to do audiorate modulations has several advantages:
- More sonic options as you can route audio to modulation and vice versa.
- This allows for unlimited FM, PM, AM, filter FM, scan wavetables at audio rate, etc...
- All audio processor modules (filters, delays, ...) can also be used to process audiorate modulation signals.
Notes
- Note that audio rate modulation uses more CPU.
- Audio-rate modulation of VST parameters is not recommended, unless it's sure that the VST can handle such intense situation, which will be rather exceptional.
| 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.
Where the audio comes into MUX Modular.
By adding audio input modules, this MUX also gets more audio input jacks!
The "Project Audio Input" is specific for the project modular area.
Where the audio goes out MUX Modular.
By adding audio output modules, this MUX also gets more audio output jacks!
The "Project Audio Output" is specific for the project modular area.
The "PolySynth Audio Output" is specific for the PolySynth.
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.
The "Project Event Input" is specific for the project modular area.
The "PolySynth Event Input" is specific for the PolySynth.
Where the events go out MUX Modular.
By adding event output modules, this MUX also gets more event output jacks!
The "Project Event Output" is specific for the project modular area.
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!
The "Project Modulation Input" is specific for the project modular area.
The "PolySynth Modulation Input" is specific for the PolySynth.
Where the modulation go out MUX Modular.
By adding modulation output modules, this MUX also gets more modulation output jacks!
The "Project Modulation Output" is specific for the project modular area.
The Audio File Recorder module simply records the incoming audio signal into a audio file which is named like this audio recorder with a YYYYMMDD-HHMMSS timestamp appended.
To actually record, make sure the audio recorder is enabled (aka armed), and click the main record button in the transport panel. (or press the proper shortcut).
Double-click the Audio File Recorder to edit its settings:
- 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.
- Record From: The input source.
- Chans: Defines whether the recorded file must be mono or stereo.
Notes
- When Record From is set to Direct Audio Input, you can directly select one of the audio device inputs. This is a handy shortcut way to avoid having to use an Audio Input module.
- Recorded files are not automatically put on a track. If you want audio track recording, see this doc page: Recording.
- When an audio recorder is linked to a track (cfr Audio Track Recording), then these context functions are not available: Rename, Delete, Replace. Because the linked track is in control over that audio recorder.
- The Audio File Recorder module is only available in the Project Modular Area, not (yet) in the deeper MUX modular areas.
Racks are described in detail here.
This module is a slot for hosting another module.
It's especially useful for integration into a MUX front panel where the front panel user can choose eg an audio processing module without having to dive into the modular area.
It is the modular patch designer that sets up the Plug-In Slot inputs and outputs via right-click on the module in the modular area -> Setup Inputs/Outputs.
Then whatever module is plugged in will be automatically be connected to the available inputs/outputs of the Plug-In Slot.
You can setup the properties of the plug-in slot via its context menu in the modular area.
The properties include the number of inputs and outputs, the preferred type of plug-in module, etc.
The incoming signal simply goes to output 1, but part of the signal also goes to output 2.
This module can be used to manually define a latency for a plug-in or hardware synth/fx that doesn't report its latency, and so it gets included in the APLC system. (Automatic Plug-in Latency Compensation)
This module also is a handy tool to check APLC in MuLab or any other host DAW.
You can control whether the latency must be reported and whether the incoming audio and events must be effectively delayed.
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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