Europa Shapeshifting Synthesizer : Panel reference

Panel reference
Sound Engines On/Off and Edit Focus section
Engine Select
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The Oscillator section
Here is where you choose oscillator waveform and set the wave shape and pitch for the selected Sound Engine.
On
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Oct
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Range: 5 octaves.
Semi
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Range: 12 semitones (one octave).
Tune
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Range: +/- 50 cents (down or up half a semitone).
Kbd
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Range: 0% (no tracking (constant pitch)) to 100% (1 semitone per note).
Waveform display
The interactive Waveform display shows the waveform shape in real-time.
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See “Recording display movements in the sequencer” for tips about automating display movements.
Waveform selector
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The wave shapes are shown in the display above and are updated in real-time according to the current settings and modulations. A great way to understand how the sound actually “looks”.
The waveforms are:
A pure sinewave at Shape=0%, gradually transformed via triangle and square towards a sawtooth wave at Shape=100%.
A square wave at Shape=0%, gradually transformed towards a sawtooth wave at Shape=100%.
A negative ramp sawtooth wave at Shape=0%, gradually transformed via triangle towards a positive ramp sawtooth wave at Shape=100%.
A 0% duty cycle pulse wave (silence) at Shape=0 gradually transformed via a 50% duty cycle square wave towards a 100% duty cycle pule wave (silence) at Shape=100%.
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A lo-fi “early computer game” type of signal. Turn the Shape knob to change the overtone contents and the octave transposition.
A pure sinewave at Shape=0%. As the Shape is increased, the pitch of the synced sinewave oscillator is raised.
A cosine window modulated by a sinewave. Turn the Shape knob to change the sinewave frequency and thus sweep through the generated formants.
This is a simulation of an electric piano. A soft/mellow tone at Shape=0% gradually transformed towards an
agitated signal at Shape=100%, with natural sound at the 12 o’clock position (50%).
A simulation of a vocal cord with a bit of noise modulation. Change the overtone content with the Shape knob.
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A physical model of a “string”, generated by sending a short pulse through pitched delay lines. At Shape=0% there is no damping and at Shape=100% there is full damping, which results in just a short clicking sound.
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Try this together with the “Stretch” algorithm in the Harmonics section to create realistic metallic sounds.
This is a special mode where you can manually draw your waveforms in the Envelope 3 and Envelope 4 windows and then gradually crossfade between the drawn waveforms using the Shape knob. See “Using the Envelope 3 and Envelope 4 curves as Sound Engine waveforms” for information on how to draw your own waveforms.
These are frequency modulated sine waves with different frequency ratios between the carrier (C: ) and modulator ( :M) signals. Set the frequency modulation amount with the Shape knob.
A pure sinewave signal at Shape=0% gradually fed back internally at an 1:1 ratio. The feedback signal is filtered before fed back to the carrier signal. If you modulate the Shape parameter from e.g. an LFO you will get a similar result as when using the FM FB Noise waveform without Shape modulation, see “Noise > FM FB Noise” below.
A sample & hold modulated noise. Change the sample & hold rate with the Shape knob. If you play high up on the keyboard at high Shape values, you get a kind of “pitched noise” sound.
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A pure sinewave signal modulated by low frequency noise. At Shape=0% the noise has its lowest frequency and at Shape=100% the noise frequency is higher (but still low-frequent). The character of the signal is similar to the Band Noise in the Thor synthesizer.
This generates a random lo-fi “digital” bit noise. At Shape=0% the signal is completely silent and with increasing Shape values the signal is modulated faster and in a wider frequency range.
A pure sinewave signal at Shape=0% gradually fed back internally at an 1:1 ratio. The feedback signal unfiltered before fed back to the carrier signal which gives the signal a noisy character.
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This signal produces a range of noises, from tonal noise up to almost white noise, by amplitude modulating the signal’s partials with noise.
The Wave Tables sub-menu contains a selection of very useful wave tables. Each wave table features eight waveforms that you could crossfade between with the Shape knob.
The User Wave options let you use the external sample you have loaded in the User Wave section (see “The User Wave and Mixer section”). The oscillator then generates and plays back wavetables (grains) of that sample. The “User Wave Smooth” algorithm uses a crossfaded loop within each grain, which produces a smoother character to the sound. Set the playback position in the sample with the Shape knob. Modulate the Shape parameter, for example from a negative Envelope ramp, for continuous movement in the sample.
Shape
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The wave shapes are shown in the display above and are updated in real-time according to the current Shape settings.
Shape Modulation
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The “Inverted” sub-menu contains inverted variations of all modulation sources.
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Phase Sync
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When active, the sound character will be the same each time you play the same note. When inactive, the sound character will vary more or less each time you play the same note.
The Modifiers section
The two Modifiers can be used for modifying the currently selected waveform in various ways. The two Modifiers are identical in functionality and can be used alone or together (or not at all).
Modifier On/Off
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Modifier selector
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The available Modifier types are:
This is oscillator sync but with a crossfade at the sync positions. This makes the effect a little smoother (less bright) than with regular hard sync, see below.
Oscillator hard sync is when one oscillator restarts the period of another oscillator, so that they will have the same base frequency. If you change or modulate the frequency of the synced oscillator you get the characteristic sound associated with oscillator sync. Control the frequency of the synced oscillator (and thereby the overtone spectrum) with the Amount knob.
This inverts the waveform phase at a variable position within the waveform cycle. Set the phase angle with the Amount knob.
This mirrors the waveform cycle (in the time line) at a variable position in the waveform cycle. Set the mirroring
position in the waveform cycle with the Amount knob. At Amount=50% the waveform is completely symmetric.
This lets you quantize the waveform in time, i.e. reduce the sample rate. Set the sample rate reduction amount with the Amount knob.
This lets you truncate the signal’s bit depth, thus making it possible to achieve that noisy, characteristic “8-bit sound” for example. Set the bit-reduction amount with the Amount knob.
This distorts the waveform by modulating the start phase of the waveform cycle. This generally creates a brighter tone towards the extremes of the Amount range (0% and 100%). At Amount=50% the signal is unaffected.
This multiplies a copy of the waveform with the original waveform. Set the phase angle of the copied waveform with the Amount knob.
This modulates the waveform with low frequency noise. Perfect for adding e.g. “breath noise” to a signal. Set the noise modulation amount with the Amount knob.
This amplifies the signal above the available headroom and then wraps the peaks down into the available headroom. This adds quite an aggressive distortion to the sound.
This amplifies the signal above the available headroom and then “mirrors” the peaks down into the available headroom. Fold is similar to the Wrap shaping but is generally less “aggressive”.
This amplifies the signal above the available headroom and then clips the peaks that are above the headroom. Generally, a signal that is clipped to the maximum would result in a pulse/square shaped waveform.
Soft clip is similar to hard clip described above, but has a smoother shape at the clipping points and thus generates less overtones.
This generates sine shaping distortion to the signal.
This distorts the waveform by introducing a short low-frequency noise glitch in the waveform cycle, but only in parts of the waveform that go from zero to positive level.
This is similar to Glitch 1 described above, but introduces a more high-frequent noise glitch in the waveform cycle.
This makes it possible to gradually crossfade between the original signal and a copy of the signal one octave above. Set the crossfade amount with the Amount knob.
This makes it possible to gradually crossfade between “one octave up” and “one octave+one fifth up”. As soon as you turn on the Fifth modifier you automatically raise the pitch by one octave. The reason for this is that this is done by multiplying frequencies, i.e. you crossfade between the double and triple of the original frequency. Set the crossfade amount with the Amount knob.
This makes it possible to gradually crossfade between the original signal and copies of the signal at the 16 first harmonics above the original frequency. Set the position in the harmonic spectrum with the Amount knob.
This is the same as “16 Harmonics” described above, except this always keeps the original signal mixed in with the selected harmonic.
This gradually adds copies of the original signal at the first 16 harmonics. Turning up the Amount knob will add on the harmonics one by one until all 16 harmonics are present in the signal.
This multiplies the waveform with a sinewave signal to generate a ring modulator effect. Set the modulator frequency with the Amount knob.
These modifiers let you frequency modulate the currently selected Waveform at various ratios. The carrier signal is the currently selected Waveform (C: ) and the modulator ( :M) is the modifier signal. Set the frequency modulation amount with the Amount knob.
Here, an internally fed back sinewave signal at an 1:1 ratio modulates the waveform (same signal type as in the “FM > FM Feedback” Waveform).
This simulates 2 copies of the original signal. The Amount knob controls the detuning amount and rate.
This simulates 6 copies of the original signal. The Amount knob controls the detuning amount and rate.
This simulates a variable number of copies of the original signal. The Amount knob controls the number of copies, the detuning amount and rate.
This simulates 2 copies of the original signal at +1 and +2 octaves relative to the original signal. The Amount knob controls the detuning amount and rate.
This simulates a formant (body) filter, which produces multiple peaks and notches in the frequency spectrum of the signal. The Amount knob controls the formant transposition in the frequency spectrum. At Amount=50% the signal is unaffected. Below 50% the formant is transposed down and above 50% it’s transposed up.
To make the formant static in the frequency spectrum, regardless of which note you play, modulate the Amount parameter using the “-KEY” (inverted) modulation source with a fairly high value (see “Amount Modulation” below). This is especially useful if you are using an acoustic instrument sample as User Wave in the Oscillator section.
Amount
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The wave shapes are updated in real-time and shown in the Waveform display.
Amount Modulation
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The “Inverted” sub-menu contains inverted variations of all modulation sources.
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The Spectral Filter
The signal from the Oscillator section can then be processed by the Spectral Filter. The Spectral Filter features a wide variety of algorithms that affect the partials of the signal.
Spectral Filter On/Off
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Spectral Filter display
The interactive Spectral Filter display shows the filter shape in real-time.
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See “Recording display movements in the sequencer” for tips about automating display movements.
Spectral Filter selector
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The available filter types are:
This simulates a standard 12dB/octave lowpass filter. Set the cutoff frequency with the Freq knob and the
resonance amount with the Reso knob.
This simulates a standard 24dB/octave lowpass filter. Set the cutoff frequency with the Freq knob and the
resonance amount with the Reso knob.
This simulates a standard 24dB/octave highpass filter. Set the cutoff frequency with the Freq knob and the
resonance amount with the Reso knob.
This simulates a standard 12dB/octave bandpass filter. Set the center frequency with the Freq knob and the
resonance amount with the Reso knob.
This simulates a standard single band parametric equalizer with a fixed bandwidth. Set the center frequency with the Freq knob and the gain/attenuation with the Reso knob.
This simulates two 12dB/octave bandpass filter routed in parallel. Set the center frequency of the first bandpass filter with the Freq knob and the peak separation with the Reso knob.
This simulates the formants of the vocal tract by using multi-peak+notch filters. Change the formant with the Freq and Reso knobs.
This simulates a non-resonant lowpass filter with a variable attenuation slope. Set the cutoff frequency with the Freq knob and the attenuation slope with the Reso knob.
This simulates a non-resonant highpass filter with a variable attenuation slope. Set the cutoff frequency with the Freq knob and the attenuation slope with the Reso knob.
This simulates a multi notch filter, great for phaser types of effects. Set the cutoff frequency of the first notch with the Freq knob and the attenuation amount - and consequently the bandwidth - of the notches with the Reso knob. The difference between “Comb +” and “Comb –” (see below) is in the position of the peaks in the spectrum. The main audible difference is that the “Comb –” version causes a bass cut.
This simulates a comb filter with a positive feedback loop - but without feed forward - ideal for flanger and phaser types of effects. Set the cutoff frequency of the second peak with the Freq knob and the resonance amount with the Reso knob. The difference between “Comb +” (see above) and “Comb –” is in the position of the peaks in the spectrum. The main audible difference is that the “Comb –” version causes a bass cut.
The three Resonator algorithms contain formant filter tables that simulate various body resonances (multi-peak+notch filters). Set the position in the formant tables with the Freq knob and the resonance with the Reso knob.
This is a special mode where you can manually draw your own filter curve in the Envelope 4 window. You then control the cutoff/center frequency with the Freq knob and the resonance with the Reso knob. See “Using the Envelope 4 curve as a Spectral Filter curve” for information on how to draw your own filter curves.
This utilizes a filter generated from FFT analyses of the external sample you have loaded in the User Wave section (see “The User Wave and Mixer section”). Transpose the formant up/down in the frequency spectrum with the Freq knob and change the filter’s position in the sample with the Reso knob.
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Freq
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Reso
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Frequency Modulation
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At 0% the filter is static and doesn’t track the keyboard at all. At 100% the filter tracks the keyboard 1 semitone per note. At values above 0% you can also see the filter curve move sideways in the Spectral Filter Display
depending on where on the keyboard you play.
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The “Inverted” sub-menu contains inverted variations of all modulation sources.
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The Harmonics section
The Harmonics section offers extensive modulation possibilities of the partials of the signal. For most algorithms the partials’ characteristics is displayed in the Spectral Filter display, see “Spectral Filter display”.
Harmonics On/Off
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Harmonics selector
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The available Harmonics types are:
This alters the gain for each of the partials in the signal in a random fashion. Turn the Pos knob to change the
randomization “pattern” and the Amount knob to change the partial gain levels in the “pattern”.
This alters the gain/attenuation for the first eight partials in the signal. Turn the Pos knob to crossfade between the partials and the Amount knob to change the partial gain/attenuation level. Amount levels below 50% attenuate all partials but the one selected with the Pos knob. Amount levels above 50% attenuate the partial selected with the Pos knob.
This alters the gain/attenuation of the partials in the signal. At Pos=0% the Amount knob controls the mix strictly between the odd and even partials in the signal. At other Pos values, the gain/attenuation is not strictly on odd and even partials. At Amount=50% the Pos value has no effect.
This stretches or squeezes all partials (overtones) - except for the fundamental - in the signal, up or down in the frequency range. Perfect for turning a harmonic signal into a more inharmonic one. Change the stretch amount with the Amount knob. The Pos knob controls the start phase of all the overtones. At Pos=0% all partials start at the same phase. When Amount is set fairly high the Pos parameter have little or no influence on the sound.
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This is the perfect algorithm for really dense pad sounds. The Ensemble algorithm simulates a type of chorus
effect by utilizing noise modulation of the partials. Set the noise frequency with the Pos knob and the mix level with the Amount knob.
The Ensemble Sparse algorithm also utilizes noise modulation of the partials, but here a lot of noise frequency bands are cut out. This makes Ensemble Sparse sound more animated and less smooth than the Ensemble algorithm described above. Set the noise frequency with the Pos knob and the mix level with the Amount knob.
This amplitude modulates the high frequencies in the signal with (high-frequency) noise, perfect for adding “breath noise” to the signal, for example. Set the noise frequency with the Pos knob and the noise mix level with the Amount knob.
The Harmonic Lag A-R algorithm is designed especially for use with the User Wave algorithm in the Spectral Filter (see “User Wave”) to create vocoder effects. The Harmonic Lag A-R algorithm controls the Spectral Filter - so the Spectral Filter has to be on for this to work!
Note that the Harmonic Lag A-R algorithm works on the filter partials - not the oscillator’s signal partials. Set the Attack time of the filter partials with the Pos knob and the Release time with the Amount knob. These controls work similarly to the Attack and Decay parameters on the BV512 Vocoder device.
Pos
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The frequency spectrum is updated in real-time and shown in the Spectral Filter display.
Amount
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The frequency spectrum is updated in real-time and shown in the Spectral Filter display.
The Unison section
The Unison function generates detuned duplicates of the signal in pairs on either side of the original signal’s pitch.
Unison On/Off
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Unison display
The Unison display shows the unison characteristics, as set with the controls in the Unison section. Note, though, that this display is not interactive like the Waveform and Spectral Filter displays.
Unison Type selector
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The available Unison types are:
This generates duplicates of the signal on either side of the original signal’s pitch.
This generates duplicates of the signal on either side of the fourth above the original signal’s pitch.
This generates duplicates of the signal on either side of the fifth above the original signal’s pitch.
This generates duplicates of the signal on either side of the original signal’s pitch - one octave down.
This generates duplicates of the signal on either side of the original signal’s pitch. The Detune parameter (see below) now controls the phases of the signal copies, instead of the detuning. This is great for creating wide stereo sounds without lots of detuning.
If Phase Sync is active in the Oscillator (see “Phase Sync”), all signals in the Unison function get the same relative start phases each time you play the same note. The original signal always has the phase 0 degrees.
Count
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Range: 1-7.
Note that for even numbers, the original signal is represented by two duplicates.
Blend
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For even “Count” numbers (see above), one of the duplicates represents the original signal in the mix.
Detune
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If the “Phase Only” Unison type is selected (see above), the Detune knob controls the phases of the signal duplicates instead of the pitch detuning.
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Spread
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The User Wave and Mixer section
The User Wave and Mixer section is where you can sample (or load an external sample) to use in the Oscillators (see “User Wave/User Wave Smooth”) and/or in the Spectral Filter (see “User Wave”). In the Mixer you can mix and pan the signals from the three Sound Engines before sending them to the Filter, Amp and Multi FX sections.
Sample Select/Load/Sampling/Edit buttons
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See the “Sampling” chapter for more information about setting up and using Reason for sampling.
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See “Editing samples” for more information about editing samples in Reason.
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Level
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Range: -Inf to +6.0dB.
Pan
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The Filter section
Routing buttons
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If deactivated, the signal bypasses the Filter and goes straight to the Amp section, see “The Amplifier section”.
Drive
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Filter Type selector
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A state variable (SVF) highpass filter with a 12dB/octave slope. This filter is similar to the State Variable Filter in the Thor synthesizer.
A state variable (SVF) bandpass filter with 12dB/octave slopes. This filter is similar to the State Variable Filter in the Thor synthesizer.
A state variable (SVF) lowpass filter with a 12dB/octave slope. This filter is similar to the State Variable Filter in the Thor synthesizer.
A state variable (SVF) notch filter. This filter is similar to the State Variable Filter in the Thor synthesizer.
A ladder-type lowpass filter with a 24dB/octave slope. The resonance peak more narrow in this filter type than in the MFB LP 24dB filter (see below). The filter can be driven to self-oscillate.
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A multiple feedback (MFB) lowpass filter with a 12dB/octave slope.
A multiple feedback (MFB) lowpass filter with a 24dB/octave slope. The resonance peak is wider in this filter type that in the Ladder filter (see above). The filter can be driven to self-oscillate.
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A multiple feedback (MFB) highpass filter with a 24dB/octave slope.
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An “early MS-20 type” of lowpass filter with a 12dB/octave slope. The filter can be driven to self-oscillate.
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Reso
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In the SVF Notch filter, the Reso knob controls the width of the notch - from wide to narrow.
Freq
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Frequency Modulation
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At 0% the filter is static and doesn’t track the keyboard at all. At 100% the filter tracks the keyboard 1 semitone per note.
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The “Inverted” sub-menu contains inverted variations of all modulation sources.
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Use one of the Envelopes (see “The Envelopes section”) as modulation source to create a Filter envelope.
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The Amplifier section
The Amplifier section contains a standard ADSR envelope, which controls the amplitude of the signals from all three Sound Engines equally.
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The picture below shows the various stages of the ADSR envelope:
The ADSR envelope stages.
A(ttack)
When you play a note on your keyboard, the envelope is triggered. This means it starts rising from zero to the value set with the Gain knob (see below). How long this should take, depends on the Attack setting. If the Attack is set to “0”, the Gain level is reached instantly. If the Attack value is raised, it will take longer time before the Gain level is reached.
D(ecay)
After the Gain level has been reached, the level starts to drop. How long this should take is governed by the Decay parameter.
If you want to emulate the volume envelope of a note played on a piano for example, the Attack should be set to “0”, the Decay parameter should be set to a medium value and the Sustain level should be set to “0”, so that the volume gradually decreases down to silence, even if you keep holding the key down. Should you want the decay to drop to some other value than zero, you raise the Sustain parameter.
S(ustain)
The Sustain parameter determines the level the envelope should rest at, after the Decay stage. If you set Sustain to full level, the Decay setting is of no importance since the volume of the sound is never lowered.
If you want to emulate the volume envelope of an organ, you theoretically only really need to use the Sustain parameter set to full level, as a basic organ volume envelope instantly goes to the maximum level (Attack “0”) and stays there (Decay “0”), until the key is released and the sound instantly stops (Release “0”).
But often a combination of Decay and Sustain is used to generate envelopes that rise up to the Gain level, then gradually decreases to finally land to rest on a level somewhere in-between zero and the Gain level. Note that Sustain
represents a level, whereas the other envelope parameters represent times.
R(elease)
The Release parameter works just like the Decay parameter, except it determines the time it takes for the volume to drop back to zero after you release the key.
Pan
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Gain
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This is the maximum level the envelope will reach after the Attack stage is completed (see above).
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Velo
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The Envelopes section
The Envelopes section features four separate polyphonic (one per voice) general purpose envelope generators, that can be assigned to control selectable parameter(s) in the Modulation Bus section.
The Envelopes are extremely flexible, and you can draw your own custom modulation shapes by clicking and drawing in the display area. There are also a number of preset shapes that you can use as starting points (or use as is). If you use Loop mode, you could turn the envelope into a kind of LFO.
See “The Modulation Bus section” for details on how to assign the Envelopes to the desired destination(s).
Envelope 1, 2, 3 and 4
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Preset
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Let’s select a standard ADSR style of envelope curve:
 
Adding a Sustain stage
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The vertical red marker that appears indicates at what level (and where) the envelope will stay sustained until you
release the key.
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Adding and removing envelope points
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Changing the envelope curve shape
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Looping the envelope
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If there was previously a sustain stage in the envelope, this will automatically be disabled when you click the Loop button.
Here we have edited a stepped curve from the Presets. We have also enabled Beat Sync and set the length/rate to 4/4. This means that each step in the curve now represents an 1/8th note.
Editing levels only
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In this mode you cannot change the time positions of the envelope points, only their levels (height). This is extra useful with a stepped Preset curve, because dragging up or down will change the value of an entire segment, turning the Envelope into a pseudo-sequencer.
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Adjusting the level of a segment.
Creating “free form” envelope curves
In the Edit Y-Pos mode, you can also draw “free form” curves:
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Using the Envelope 3 and Envelope 4 curves as Sound Engine waveforms
As a special feature you can use the Envelope 3 and Envelope 4 curves as waveforms for the Sound Engines:
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Using the Envelope 4 curve as a Spectral Filter curve
Another special feature is that you could use the Envelope 4 curve as a filter curve in the Spectral Filter:
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At Reso=0% the curve is completely flat (no gain or attenuation) and at Reso=100% the resonance corresponds exactly to the Envelope 4 curve.
The LFO section
An LFO (Low Frequency Oscillator) is used for generating cyclic modulation. A typical example is to have an LFO modulate the pitch of a signal to produce vibrato, but there are countless other applications for LFOs.
The LFO section features three separate general purpose LFOs, that can be assigned to control selectable parameter(s) in the Modulation Bus section, see “The Modulation Bus section”.
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Besides the standard waveforms (sine, triangle, pulse, etc.) there are random, slope and stepped waveforms. The shape of the waveforms are shown in the display.
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The Rate parameter now controls the time divisions.
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Turn clockwise for longer delay times.
The Effects section
The Effects section features six different effect modules that can be freely reordered by dragging & dropping. Most of the effect parameters are also available as destinations in the Modulation Bus, see “The Modulation Bus section”.
At the top of the Effects section are six Effect buttons. Click any of these to bring up the control panel for the corresponding effect. Below the Effect buttons are the On/Off buttons for the individual effects. Click these to activate the effects.
Reordering the effects
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Moving the Reverb effect to another position in the effects chain.
You can reorder the effects at any time.
Reverb
This is a stereo reverb, routed as a send effect.
This governs the length of the reverb effect.
Sets the emulated room size, from small room to large hall. Middle position is the default room size.
Lowering this parameter results in a closer and gradually more “canned” sound. Raising the parameter results in a more spacey sound, with longer pre-delay.
Raising the Damp value cuts off the high frequencies of the reverb, thereby creating a smoother, warmer effect.
Use this parameter to adjust the send level to the Reverb effect.
If you play a note, have a long delay Decay time and turn down Amount, the reverberation will continue.
Delay
This is a stereo delay, routed as a send effect.
Activate Sync to sync the delay time to the main sequencer Tempo.
This sets the time between the delay repeats. If Sync is active (see above), the Time parameter now controls the time divisions.
Activate Ping Pong to have the delay repeats alternating between left and right in the stereo panorama. The effect is also dependent on the Pan parameter (see below).
Sets the panning of the delay repeats in the stereo panorama. If Ping Pong is active (see above) the Pan knob controls the panning of the initial delay repeat as well as the total stereo spread of the remaining repeats.
The FB (feedback) parameter determines the number of delay repeats.
Use this parameter to adjust the send level to the Delay effect.
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Distortion
The Distortion effect features six different types of distortion.
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“Dist” produces a dense, rich analog type of distortion.
“Scream” produces a less bright type of distortion.
“Tube” emulates a tube type of distortion.
“Sine” is a sine shaping distortion.
“S/H” gives the effect of sample rate reduction.
“Ring” is a ring modulator effect.
Sets the overdrive/feedback level of the selected distortion.
This is a lowpass filter and sets the tone of the selected distortion.
Sets the Dry/Wet amount of the distortion.
Compressor
This is a stereo compressor.
This governs how quickly the compressor will apply its effect when signals rise above the set threshold. If you raise this value, the response will be slower, allowing more of the signal to pass through the compressor unaffected. Typically, this is used for preserving the attacks of the sounds.
When the signal level drops below the set threshold, this determines how long it takes before the compressor lets the sound through unaffected. Set this to short values for intense, “pumping” compressor effects, or to longer values for a smoother change of the dynamics.
This is the threshold level above which the compression sets in. Signals with levels above the threshold will be affected, signals below it will not. In practice, this means that the lower the Threshold setting, the more the compression effect.
This specifies the amount of gain reduction applied to the signals above the set threshold.
Phaser/Flanger/Chorus
This is a stereo Phaser/Flanger/Chorus.
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The selected effect type is displayed on the Effect button.
Sets the depth of the selected effect. To get a static sound, set Depth to zero.
Sets the rate/speed of the modulation.
Sets the stereo width of the effect.
Sets the Dry/Wet amount of the effect.
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EQ
The EQ effect is a single band parametric equalizer with adjustable Q-value and Gain.
Sets the center frequency of the EQ band.
Sets the bandwidth of the EQ band, from wide to narrow.
Sets the gain/attenuation of the EQ band, from -18dB to +18dB.
The Modulation Bus section
The Modulation Bus section is used for routing a modulation Source to one or two modulation Destinations each. This creates a very flexible routing system that complements the “pre-wired” routing in Europa.
The Modulation Bus section in Europa is derived from the one in the Reason Thor Polysonic Synthesizer device, so if you are familiar with Thor, you will quickly find your way around in Europa’s modulation bus.
There are eight “Source –> Destination 1 –> Destination 2 –> Scale” busses, of which the first four have pre-assigned sources. However, these four pre-assigned sources can be easily changed if you like.
A Source parameter can modulate two different Destination parameters per bus (with variable Amount settings). Each bus also has a Scale parameter that affects the relative modulation Amount for both Destinations.
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The following parameters can be used as modulation Sources:
Modulation Bus Source parameters.
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Assigning LFO 1 Rate as a Destination for Envelope 1.
As you hover over a valid destination control on the panel, the parameter name is automatically displayed in the Destination box in the Modulation Bus.
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The following parameters can be used as modulation Destinations:
Modulation Bus Destination parameters.
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8.
The available Scale parameters are the same as the Source parameters, see “Modulation Bus Source parameters.”.
9.
Both positive and negative Scale Amount values can be set (+/- 100%). If you, for example, are using the Mod Wheel as Scale parameter and don’t want any modulation when the Mod Wheel is set to zero, set the Scale Amount parameter to 100%. Then, there will be no effect when the Mod wheel is set to zero, and full modulation when the Mod Wheel is all the way up.
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Europa Shapeshifting Synthesizer : Panel reference