Manual

main menu

The above icons are visible at the top of the MikroWave screen. From left to right, these buttons are:

1. play / pause starts and stops the sequencer and drum machine.
2. instrument menu, opens the page holding all the available instruments.
3. mixer page, opens the mixer where you can adjust the volume levels of all individual instruments, along with the master volume.
4. configuration screen, where you can set general application settings such as the audio buffer size of your device. More information on buffer size.
5. load song, load a previously saved song
6. save song, what it says on the tin ( not available in the FREE version )
7. Export song, export your song as either WAV or MIDI, or upload to SoundCloud ( not available in the FREE version )
8. online help/manual (i.e. this page you're viewing right now!)
9. exit the application

instrument menu

The main page and the page that the application will return to whenever you press the BACK button. The instrument menu shows four different columns, each representing a different instrument. The first three columns each represent a different synthesizer, each with its own unique colour. These synthesizer columns are each subdivided in three rows:

1. grid opens the grid view in which you can sequence notes. This gives the synthesizer something to play and makes it audible.
2. keys opens the keyboard view in which you can play the synthesizer live. The keyboard is multi-timbral (can play more than one note simultaneously). On tablets the keyboard shares the screen with the effects modules.
3. fx shows all the effects modules for the given synthesizer. Als see shaping a basic sound.

The last column directs you to the drum machine, this is where you craft a beat.

sequencer use

The sequencer can be seen as the clock that watches over your musical composition and plays notes when it is "their time". When the sequencer is running it will synchronize to play both the synthesizer and drum machine instruments.

The synthesizers sequencer is accessed by tapping the grid button in the respective synthesizers column on the instrument selector screen. When opening a new set it is completely empty. This grid sequencer is a representation of a "sheet of music" in both time and pitch, where pitch is represented vertically (the note names appear on the left) and time horizontally (where each "slot" is a sixteenth note within a measure).

By tapping inside the grid, you create a new note at the given position with the given pitch. If you want to create longer notes, you press and hold the note with one finger, and tap (or drag) with another finger along the horizontal axis to extend the note length. You can navigate the measures by dragging inside the grid window.

If you aren't satisfied with a note, you can remove it by double tapping it. You can also change its location (and thus position in time and its pitch) by holding down the note and dragging it across the visible grid area.

At the bottom right of the grid screen you can find a green button that will open the sequencer controls.

sequencer controls

The sequencer controls offer control over song length, tempo, looping and recording. From left to right:

measures the amount of measures/bars the song contains. You can add as much measures as you please. If you lower the amount of measures but have entered note data in a deleted measure, you can still access these notes by increasing the amount again (for instance you have four bars of music, but lowered the amount of bars to three, after increasing it back to four, all the previously entered notes are still available). Note that when saving only the notes in the entered range are saved.

record button when the sequencer is running, all sound will be recorded until this button is toggled again. After recording you will be presented with the option to save the recording to your device or discard it. You can record for as long as your device has disk space.

loop toggling this control will loop the current measure. This allows you to precisely work on a single measure in a multi-measure song.

tempo control changes the tempo. For precise round number increments you can use the +/- buttons to the left of the control. The tempo range spans 40 to 300 BPM (beats per minute). The tempo change is executed upon release of the dial, rather than realtime.

shaping a basic synthesizer sound

When working with a synthesizer, you start out with selecting your basic waveform as it is the source of all generated sound. In MikroWave you have six of these waveforms as demonstrated in the image above. These are:

1. sine wave the purest tone available, can sound bell-like at high pitches.
2. sawtooth a more jagged sound, works well on bass lines
3. square wave a sound you'd associate with 80s videogames, works nice on chords.
4. pulse width modulated wave, once more 80s retro, this times reminiscent of a Commodore 64 where the basic wave has a filter sweeping through a fixed range.
5. noise what it says on the tin : a tuned burst of randomness, not so much musical but can be shaped to create subtle ambient effects
6. string synth, a Karplus-Strong algorithm creating nice twangy sounds.

The two parameter controls are attack and decay which add a amplitude (volume) envelope, allowing you to fade in and out the sound to create subtle swell effects.

secondary oscillator

Depending on which waveform you selected for the first oscillator, chances are things still sound a bit too thin for your liking. For spicing up a simple waveform you can kick in the secondary oscillator which basically operates like the first oscillator, however :

Simply doubling a waveform won't actually affect the sound (other than increasing its loudness). You could choose a different waveform for the secondary oscillator to add harmonic complexity to the sound, or you could choose to alter the pitch of the secondary oscillator using the three controls. All pitch changes are relative to the pitch of the first oscillator.

octave shift the pitch in full octave increments. You can shift down two octaves and up two octaves, this can add extra weight or a floating layer to the sound.

shift shift the pitch in semitone (half-note) increments. You can shift down seven semitones and up seven semi-tones. You can use this control to harmonize between the oscillators.

detune shifts the pitch in cents, you can shift down by 50 cents and up by 50 cents. This creates a very subtle change in frequency and can work wonders on twin sawtooth oscillators to widen the sound.

arpeggiator

The arpeggiator is a tool that modulates the pitch of the oscillators. An arpeggio is basically a sequence of notes that outline the notes within a chord; playing one after another instead of simultaneously.

The arpeggiator sequence can contain up to eight steps (the amount which can be selected using the steps-dial). By selecting one of the eight slots and using the pitch- dial you can specify a note shift of up to 12 semitones (up or down) for each step in the arpeggio sequence. This shift is once more relative to the note played by oscillator 1.

The arpeggiator operates at different speeds. Using the size-dial you can specify the total length (duration) of each step from a whole measure to a 64th note. Note that if the sequencer holds a note that has a duration shorter than the arpeggio step size, you will not hear the arpeggiator "work" as the note has finished playing before the arpeggiator "steps" to the next pitch slot. You can use the latter situation to your benefit by playing held keyboard notes over a staccato bassline programmed in the grid sequencer.

routeable oscillator / frequency modulator

The routeable oscillator aids in modulating your basic sound by introducing another wave which has a cycle that repeats at a lower frequency ( it operates at a range from .1 to 10 Hz ).

In itself, the secondary oscillator doesn't do anything to the audio signal, but by routing it to a target, it's effect becomes clear:

1. to filter will modulate the Jennifer Lo-Pass filter (see below) by adding movement to the cutoff frequency of the filter.
2. by selecting to FM, the oscillator will function as a frequency modulator on the main waveform, this allows you to increase the harmonic complexity of the sound.

by tapping the wave-button you can change the basic oscillator waveform between sine, triangle, sawtooth and square wave allowing you to set the depth of the effect.

jennifer lo-pass filter

The Jennifer Lo-Pass is, as it's punny name suggests, a low pass filter (a filter being an amplifier circuit that works on specific frequencies, allowing you to cut or boost at a specific range, greatly changing the timbre of the original sound).

A low pass filter rolls of the high frequency content of the incoming audio.

1. the cutoff-parameter selects the frequency at which the cut begins
2. the resonance-parameter sets the depth of the cut

By routing this effect to the secondary oscillator you can create a moving effect creating sweeps use-able for creating anything from 70s sleaze funk to dub step basses.

mangler

This is where you go to if you just don't like pure tones.

The Mangler is essentially a bit crusher, an effect that distorts audio by reducing the resolution and bandwidth of the incoming signal. You can use it to subtly accentuate harmonic content or to completely tear the sound source apart until it sounds like gravel.

1. bit resolution controls the bit rate of the outgoing signal, the range is 1 - 16 bits. The lower the resolution, the higher the distortion amount.
2. distortion mix acts as an attenuation control, the higher the distortion, the higher the output volume, which in turn can cause more distortion as the output signal clips, which might not be entirely what you're after!

phaser

A phaser is another flavour of filter. A phaser differs from the Jennifer Lo-Pass as it filters using multiple peaks and troughs within the frequency spectrum. The position of these notches is modulated by a low frequency oscillator for a sweeping effect.

1. the rate-parameter controls the speed of the low frequency oscillator, allowing you to go from a slow sweep to a fast "underwater"-warble.
2. the depth-parameter controls the slope of the filter accents.
3. the feedback-parameter determines the amount of effected signal which is going into the next phaser stage(s).

speak formant filter

A formant filter is yet another type of filter, this time round the accentuated frequencies are carefully selected to mimic the spectral peaks of a human voice.

What it boils down to : the speak-module recreates the sound of vowels by applying multiple formants to recreate their characteristic overtones.

By using the vowel-parameter you can travel (and interpolate) between A, E, I, O, U-type vowel sounds.

degrader

The degrader acts as a decimator, which is basically another "destruction"-type effect where the sample rate of the incoming signal is reduced.

As the degrader introduces a lot of high frequency "chirping", it is best applied for momentary effects, it also works wonderfully with delay.

By using the depth-parameter, you can alter the amount of frequency reducing.

polly delay line

Delay can basically be seen as an echo effect. The incoming signal is recorded and played back after a small delay, the playback is mixed in with the original signal allowing you to overlay the repeats.

Note that the delay effect adds to the signal, instead of modifying the input signal. As such, it is possible to precisely set the delay to repeat a certain motif, while constantly making changes to the incoming signal (i.e. playing different notes, changing effect device properties, etc.)

1. time specifies the delay between the original sound and the playback of the echo, the range if from 5 to 2000 ms. At lower settings the sound can mimic a reverb or act as a comb filter.
2. mix specifies the playback level of the echo(es)
3. the feedback-dial allows you to feed repeats back into the input line of the delay, creating a large loop of accumulating echoes which can provide anything from rich textures... to NOISE!

compressor

Compression is a touchy subject. At its heart it aims to reduce the difference in amplitude between subsequent peaks of the incoming signal. What this means : you can "squash" a signal and keep the subjective volume level the same, or optionally just maximize volume or create energetic "pumping" effects.

1. attack specifies the delay between registering a signal exceeding the threshold and actually applying the compression
2. release specifies the delay between registering a signal as being within the treshold and actually stopping compressing the signal
3. threshold specifies the volume level (in decibels) at which the signal should be compressed, once the incoming signal exceeds the threshold the compressor actually starts working (keeping the attack and release-rates in mind).
4. ratio specifies the amount of volume reduction/attenuation applied to the signal

pitchshifter

The pitchshifter allows you to "bend" the pitch of the incoming signal, which can add subtle articulations (or wild mayhem) to your sound.

The pitch-dial allows you to shift the pitch by 2 semitones, both up and down. Note that the dial will return to the 0 position (restoring original pitch) when released.

Pitchshifting is an expensive process on the CPU, if use of the pitchshifter results in unstable audio, increase your buffer size.

buffer size ?

MikroWave's audio engine was written explicitly to process audio live in as little time as is possible on a mobile device. Which is kinda like demanding to run through water as there are a lot of different device configurations to deal with.

What we're battling is what is known as latency, this is the time between an action and hearing the result. When you press a key on the keyboard, or turn a dial on an effects device you'd ideally want an instant result, which in technical terms means : you want the generation of sound (synthesis) and application of the effects (digital signal processing) to happen in as little time as possible. To do this audio is processed as a continuous feed of small snippets, known as a circular buffer. The exact size of a single buffer is what is of importance.

On first launch MikroWave reads your device configuration and makes an estimate of what would be the ideal buffer size. In reality, you could achieve a better balance between response time and audio stability by experimenting with altering this "recommended" buffer size.

Long story short: for faster response, use a lower buffer size. However, setting the buffer size too low can lead to stress on your device's processor, where the CPU is working as hard as it can, but cannot deliver the results in time leading to glitches. If audio sounds unstable (stutters or drops out), use a higher buffer size.