Moebius Lab v2.1

Moebius Lab is a laboratory.
A laboratory for Audio...
A very huge laboratory!!!!
17 effects!!!
11 slots for an AUv3 or IAA Apps !!
A Sampler / File player
A awesome monophonic Synthesizer!!!
A MIDI Keyboard which allows you to control the 11 others Apps that you added to the slots (If you choose to plug only Apps instead Effects)
A routing which allows you to route the Time Domain nodes to the Spectrum Domain nodes & vice versa...
And sure all the features that you can find in all Ape Apps (Snapshots Pad / Accelerometer /....)
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Input module: Input or Sampler or Synth + 2 slots for an AUv3 or IAA App
Output module: Plate Reverb + Dynamic Processor + 1 slot for an AUv3 or IAA App
Time Domain Nodes: Exciter + Ringer + Klamper + Swarmer (Chorus) + Filter + Splitdel (Delay) + Reduktor + Shifter + Reverso + 8 slots for an AUv3 or IAA App
​Spectrum Domain Nodes: Brikwall Filter + Spectral runner (Spectral Delay with Feedback) + Grip (samples and holds spectral components of the incoming signal) + Blur (A (noisy) “spectral reverb) + Degrader (spectral decimator) + Flutter (a sort of animated brickwall)

What can we do with it?
So many things...
You can use it for the Awesome monophonic Synthesizer only
To built a loop
As a Mastering tool
To control all your synth from one place
The possibilities are infinites !
This App is Awesome!
A Killer App! A Must Have !

Download on the App Store

Price $12.99 Author Amazing Noises Type Synthesizer / Sampler / Multi Effects Supports Inter-App Audio (IAA), Audiobus )Version2.1 Released 06/29/2016 Updated 08/05/2019

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Direct link to other page
​Page 2
​Spectrum Domain Nodes | The monophonic Synthesizer
Page 3
The monophonic Synthesizer suite | The MIDI Keyboard | The Sampler | Plugins Support | The Presets | The Snapshot Pad | Settings | Sync | Others windows

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Input module

In this window you can select the Input source; you can also add two external audio apps by tapping on the “plus” icons in the middle If you add an effect it will process the input before routing it to the Moebius Ring If you add an instrument you can mix it with the internal input
Input - Sampler - Synth Switch
You can choose between the internal sample player and a live input via Microphone, Audiobus or Inter-App-Audio or a Built-In Synth
Input Recall
When this toggle is on, the source Input, the Sampler and the Tempo value are recalled when a new preset is loaded
Please note that even if this switch is disabled, all data (Source Input,
Sampler and Tempo) are still saved in the presets
If the Recall Input toggle is switched on and the Sampler is subscribed for SYNC (i e the sync icon is switched on), the MASTER tempo will not be affected

Output module

In the Output Window you can activate two “post-Moebius” effects; a Reverb (“PlateVerb”) and a Dynamics Processor

PLATEVERB
As the name suggests this is a classic Plate Reverb In the Output Window
tap on the icon next to the “Output Gain” knob to show the PlateVerb Edit Window
Balance‘ with this parameter you can mix the dry and wet (reverberated) signals
Pre Delay‘ you can add a short delay to the reverb effect to separate it from the dry signal
Low Cut and High Cut" with these parameters you can equalize the sound before sending it to the reverberation circuit The equalizer doesn’t affect the dry sound
Pre Delay‘ you can add a short delay to the reverb effect to separate it from the dry signal
Low Cut and High Cut" with these parameters you can equalize the sound before sending it to the reverberation circuit The equalizer doesn’t affect the dry sound
Damping- with this parameter you can attenuate the high frequency reflections faster than the low frequency ones The higher the parameter is, the more high frequencies are attenuated
Size: the size of the reverberation space If you set this parameter to 1 and the Damping parameter to 0, you have an “almost infinite” reverb tail

DYNAMICS PROCESSOR
After the PlateVerb, you can change the output sound using a compressor and expander effect
This is used to either reduce or expand the dynamic range of an audio signal
The Dynamics Processor controls set high (compression) and low (expansion) threshold volume levels When the incoming audio is between these levels, it is left unchanged Audio that rises above the upper threshold is cut in volume to bring it back down to that level (compression) Audio that falls below the lower threshold is cut in volume to make quiet sounds even quieter (downward expansion)
​Additional controls adjust the attack and release time for the plugin's volume envelopes, the expansion ratio, the compression headroom, and a master gain to boost or cut volume on the Dynamics Processor’s output
Threshold: when the signal is above the threshold it is compressed
Headroom: upper limit for the compressor, a limiter prevents the signal to go above this threshold
Attack and Release‘ activation and de-activation times for the compression when it is above the threshold
Exp Threshold: when the signal is below the Exp Threshold it is backward expanded
Exp Ratio: expansion ratio

Global routing

You have 7 faders to route the signal globally
1 — Master DryNVet- with this fader you mix the input and the output signal
When it is at the center the two signals are both at maximum level Moving the fader to the right lowers the dry signal, moving it to the lelt lowers the wet signal
2 — Gain Input 1- this is the gain of the signal sent to the Time Domain Chain
3 — Gain Output 1' this is the gain of the signal coming from the Time Domain Chain
4 — Gain Input 2- this is the gain of the signal sent to the Spectral Chain
5 — Gain Output 2' this is the gain of the signal coming from the Spectral Chain
6 — Feedback Chain 1' with this fader you can send the signal from the Time Domain Chain to the Spectral Chain
7 — Feedback Chain 2- with this fader you can send the signal from the Spectral Chain to the Time Domain Chain
Each node presents a vertical dry/wet fader

Time Domain Nodes

1 Exciter

The Exciter Node distorts the signal below (Saturation) or above (Excitation) a given frequency
Sat/Exc' When this parameter is O, only the signal below the split frequency is distorted (see below), when it is 1, only the signal above the split frequency is distorted Other values cause different mixes of distortion above and below the split frequency
Split Freq: this is the parameter used to separate the two distortion bands (Saturation and Excitation)
Amount: the amount of saturation/excitation
Feedback: the feedback reintroduces the distorted signal back into the circuit, for even more distortion

2 Ringer

A ring modulator with distortion and feedback
Frequency: Frequency of the modulator
Freq/Pitch: you can set the modulator frequency in hertz or in MIDI note values
Shape: the shape of the modulator; 0 = sine, 1 = saturated sine
Lowpass" a low-pass filter for the ring modulation
Feedback‘ the amount of modulated signal sent back into the modulator

3 Klamper

Clamps the signal with a few special features
Klamp Level- when the signal amplitude is above this level, it is “klamped”;
i e if the Klamp Level is 0 5, all signal values above 0 5, or below -0 5 are transformed according to the Mode parameter (see below)
Mode: you can choose among 6 modes of “klamping” for the signal exceeding the Klamp Level
Klamp- the signal is clipped
Fold‘ the signal is folded back into the limits
Wrap- the signal is wrapped in the opposite direction
Annihil- the signal is annihilated (reduced to O)
Sine- the signal is replaced with a sinusoidal oscillation
Square‘ the signal is replaced with a square wave oscillation
Oscil Freq: frequency parameter for the "Sine" and “Square” modes above
Split Freq: you can klamp the signal above or below the split frequency (see the Spec Balance below)
Split Freq: you can klamp the signal above or below the split frequency (see the Spec Balance below)
Spec Balance: when this parameter is 0, only the signal below the split frequency is klamped, when it is 1, only the signal above the split frequency is klamped
Other values cause different mixes of klamping above and below the split frequency
Normalize: the klamped signal can be normalized, i e raised to the level specified by the Norm Level parameter below
Norm Level: when the Normalize switch is on, this is the level for the klamped signal

4 Swarmer

A “superchorus”, excellent to create super fat synth sounds (and more) It is composed by a chain of modulated delay lines
Activity: how fast the delay lines are modulated
Depth: sets the amount of delay in the delay lines
Tank: this is the feedback parameter, sets the amplitude of the signal sent back to the circuit
Lowpass' a low-pass filter for the feedback signal.

5 Filter

The Filter node has two configurations-
Dub: dual low-pass high-pass state variable filter with resonance and overlap factor
Morph: a triple low-pass/band-pass/high-pass state variable filter with resonance and continuous morphing between the filter types
You can switch between the two configurations using the radio buttons on the right of the edit window
​Dub Parameters:
Cutoff‘ this parameter sets the cut off frequency and the filter type From -100 to 0 it is a low pass filter with cutoff frequency gradually increasing from 0 Hz to 15000 Hz, from 0 to 100 it becomes an high pass filter with cutoff frequency gradually increasing from 0 Hz to 15000 Hz
Overlap- the overlap factor influences the Filter behaviour When it is 0 the Filter parameter has the effect described above Increasing the overlap factor makes the two filter types (low pass and high pass) to gradually overlap (i e the high pass effect begins before the low pass section has completed its path), and when the overlap is at its maximum, the filter behaves like a band pass (because the low-pass and the high-pass sections have nearly the same cut off frequency)
Morph Parameters:
Frequency: from 10 to 11 000 Hz
Type: the filter type can be morphed gradually from low-pass (0) to band- pass (0 5) to high-pass (1)
Other Parameters:
Resonance‘ this is the amount of resonance the filter has at its cut off frequency
Order‘ this parameter gradually changes the filter order from 12 dB to 24 dB
Env Depth: this parameter sets the amount of variation the internal envelope follower has on the filter cutoff frequency
Env Speed: this parameter sets the speed of the envelope follower, i e how fast it follows the input signal amplitude

6 Splitdel

A dual band delay with feedback
The delay time can be expressed in milliseconds or in note values (1/4, 1/8, etc) To switch between the two, tap the chain on the upper right of the edit window
Split Freq: signal components below the split frequency are sent to Delay 1 the components above are sent to Delay 2
Delay 1: the delay time of the first delay line
Feedback 1: the feedback factor of the first delay line
Delay 2: the delay time of the second delay line
Feedback 2: the feedback factor of the second delay line

7 Reduktor

This node can be used to decimate and do other “bad things” to your signal
Bit Depth: you can change the bit depth of the signal
Resample: this parameter is a factor of the actual sampling rate' i e if the SR is 44100 Hz, a Resample of 0 5 would give a 22050 Hz sampling rate, o 1 would giv 4410 Hz and so on
int Freq: you can randomly interrupt your signal This parameter sets how many interruption statistically you have per second
Int Ratio: this is the ratio between the interruption and the sound If set to 0 5 the sound has the same duration of the interruption, if set to 0 1 the sound has 1/10 of the duration and the interruption 9/10 and so on
Int Amount: this is the depth of the interruption If it is 1 the sound is completely interrupted (silenced) If it is 0 5 the sound amplitude is halved during the interruption and so on
Single/Dual/Alternate Switch: Single = both channels are interrupted at the same time; Dual = the two channels are interrupted independently; Alternate = the cannels are interrupted alternatively
None/\Nhite/Pink Switch: None = during the interruption the sound is silenced; White = the sound is replaced with white noise; Pink = the sound is replaced with pink noise
Split Freq: you can decimate the signal above or below the split frequency (see the Spec Balance below).
Spec Balance: when this parameter is 0, only the signal below the split frequency is decimated, when it is 1, only the signal above the split frequency is decimated. Other values cause different mixes of decimation above and below the split frequency.

8 Shifter

A dual band frequency shifter
Split Freq: the signal components below the split frequency are shifted by
Shift 1, the signal components above the split frequency are shifted by Shift
2
Shift 1: frequency shift parameter for the signal components below the split frequency
Shift 2: frequency shift parameter for the signal components above the split frequency
Shifter Stereo Flip- when this switch is on, the right channel shift parameters are inverted (Shift 2 for the lower frequencies and Shift 1 for the higher)

9 Reverso

Plays the signal in reverse at regular rate
Switch Rate L/R‘ it is expressed in note values (1/4, 1/2 and so on) For each given value (e g 1/4), the first half (1/8) is played normally and the
second half (1/8) is played in reverse
Reverse Prob L/R: the probability that a segment is played in reverse
Inertia: the time it takes for the internal delay line to change the length of the reverse signal when the Switch Rate parameter is changed It afiects the pitch of the sound
Stereo Flip‘ when switched on, the reverse processing is applied independently (and with difierent parameters) to the two channels

10 to 17 to host an AUv3 or IAA FX, FX MIDI, Instrument or Generator

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