The six submixers each have two inputs with multiple modes for the mix and modulation kind.
Switch to set submixer mode:
mixes input 1 (top) and input 2 (bottom). Matrix modulations are smoothed to 10Hz to reduce CPU usage.
same as SLO, but modulations are at full event rate.
output is input 1 * input 2, with mix controlling output level. The result is like ring modulation with oscillator and similar sources, or it can be used to modulate at audio rate with input 1 as LFO or envelope. Level control and matrix modulations use less CPU but are smoothed to 10Hz.
Like am1, but matrix modulations are at full event rate (increasing CPU usage).
output is just input 2 with low mix, or input 1 multiplied by input 2 with high mix. Matrix modulations are at full event rate. another.
The three state-variable filter units all provide 19 different kinds of filtering mode. In some modes, the filter characteristics or wet/dry mix can be set and modulated.
The INPUT switch sets the audio input, and the KIND switch sets the type of filter. Setting input to OFF disables the filter entirely (and turns off the XY panel), saving CPU. The following filter types are available:
Low-pass and high-pass filters in parallel.
Comb filters. The filter P controls set the comb delay, and the filter Q controls set the amount of feedback.
A 2-pole low-pass filter with feedback, followed by a 2-pole low-pass filter.
Equalizers. The P controls set the corner frequency and the Q controls set the boost/cut factor (0.5 no boost/cut; higher values are boost, lower values are cut).
The source audio is directly passed to the output.
The FmSrc switch sets the FM source. Internal logic reduces CPU if this switch is set to OFF. The FM knob controls the amount of FM modulation. FM modulation depth is set in semitones around current filter pitch.
PSRC selects the pitch source for frequency tracking.
takes pitch from global pitch, pre glide but post transposition and remapping. The CENTER control in the global pitch panel also allows changing of filter response over pitch.
take pitch from oscillators, post glide, and after any pitch modulation and transposition for the oscillators themselves.
take pitch from the envelopes directly, before any modifications to global pitch.
sets the center frequency.
- For linear tracking, set PCNTR to middle C (C3=60).
- To skew pitch tracking so that lower frequencies cause greater pitch change, lower PCNTR.
- To skew pitch tracking so that higher frequencies cause greater pitch change, increase PCNTR.
Specifically, the PCNTR control sets the middle point (0.5) for pitch tracking. Lower pitches than PCNTR linearly slope down so that MIDI 10 has PTRK value of 0. higher pitches than PCNTR linearly slope up so that MIDI 110 has PTRK value of 1.
If global pitch is the source, then Pcntr further modifies the frequency response after the CENTER control in the global pitch module. This permits multisegment tracking response.
PTRCK and QTRK set the amount of pitch tracking for filter frequency and resonance, respectively. If set to 0, there is no pitch tracking.
For more complex pitch response curves, set PTRCK and QTRK to 0; then route a pitch source to frequency pitch or resonance via the modulation matrix.
On filters 1 and 2, the MODE switch selects the filter gain and saturation mode:
- DIRECT: gain is adjusted by filter's pitch and resonance at event rate
- SAT: filter output also passes through saturator
- S+L: filter output passes through saturator, then through unity-gain limiter.
- S+A: filter output passes through saturator, then through AGC.
- LMT: filter output passes through unity-gain limiter.
- AGC: filter output level is fixed by automatic gain circuit to be unity gain.
- If SAT is off, this sets mix of dry signal (fully left) and filtered signal (fully right).
- If SAT is on, all of the input signal passes through the filter, and MIX instead sets the mix of filtered signal (fully left) and saturated filter (fully right).
In addition, the MIX knob provides the following special control modes:
- For LH and feedback filters, MIX sets the separation between the filter stages.
- For peak EQ, MIX sets the bandwidth.
The XY control displays the current frequency and resonance for each voice, after shaping and modulation. The XY control crosshair sets the base frequency and resonance for the filter.
When on, resonance has a more logarithmic characteristic, providing higher degree of control at higher resonances. When off, resonance is linear.
On filter 1 and 2, the A and B listboxes select two modulation sources for the filter. See the section on matrix modulations for a description of the modulation sources.
The A>P and A>Q knobs set the amount that modulation source A is applied to frequency and resonance, respectively. The B>P and B>Q knobs perform the same function for modulation source B.
The pitch and modulation amounts for source A and source B are normally positive. To change their polarity, press the "-" button next to them.
- Use parallel filters panned to different positions for chorus effects. Or pan the original signal to one side and the filtered to the other.
- Use a comb filter with PTRK=1 to add sonic richness.
- Use a comb filter with PTRK=0, to create a chorus effect.
- Use a comb filter with high resonance to simulate a string or tube.
- Use LH2 or LH4 with initial narrow separation, and increase the separation with envelope over time to create a "fly-in" effect.
The distortion modules are single-in, single-out polyphonic audio modules that can be placed between any other two audio modules.
The input switch selects the audio source for distortion.
The mode switch selects the type of distortion:
- SMIRROR - 2-way mirror. LVL sets one mirror point, SYM the other.
- MIRROR - 1-way mirror. LVL sets the mirror point.
- CLIP - asymmetric high/low clip. LVL sets one clip level, SYM the other.
- CLIP - symmetric high/low clip. LVL sets the clip level.
- SSAT - asymmetric saturation. LVL sets the amount of saturation, SYM the asymmetry.
- SAT - saturation. LVL sets the amount of saturation.
- SQNT - asymmetric quantization. LVL sets the quantization level, SYM sets the ERR amount.
- QNT - quantization. LVL sets the quantization level.
- SAW - saw wrapper. LVL sets wrapping range, SYM the step asymmetry.
- TRI - triangle wrapper, SYM the step asymmetry.
- SINE - sine wrapper, SYM the step asymmetry.
- BYPASS - source audio passes directly to output, useful for monitoring
Sets the level or mix of the distorted signal, as described above.
Sets the asymmetry for asymmetric distortion modes, as described above.
The LOCUT switch enables a high-pass filter to reduce sub-audio-frequency rumble.
The three channels of the output mixer take their inputs from envelopes 1-3, respectively. Turn off the envelope AUDIO to reduce CPU usage.
These buttons enable/mute the respective channels.
Click and drag up and down in the TRIM numeric display to adjust the relative amplitude of the three envelope outputs so they are all equal, then use the LVL knob to fade in and out the envelope outputs.
The pan control adjusts channel mix to left and right outputs. The internal design removes the -6dB drop in output level when pan is centered.
The send levels to the effects units are post-fader by default. Enable PRE to use Pre-fader levels.
RET1 sets the return level from echo, and RET2 the return level from chorus.
The OUT slider sets the output volume. The MASTER knob works like the channel TRIM controls, letting you set the relative overall volume output for your speakers or audio recording equipment.
The LIMIT switch enables monophonic limiting on the final output. The REL knob controls the limiter release/respond time when LIMIT is enabled.
The HICUT switch enables hi-cut equalization on the final output. The FREQ numeric control sets the corner pitch of the output in MIDI note values.
The B panel contains a simple audio recorder on the output of the main mixer.
The ON switch enables audio recording. The WAV-file textfield lets you set a file to which the audio recording is saved.
When the recorder is switched on, the RECORD and PLAY buttons let you capture audio and audition the recording.
FX1 is a delay effect with ping-pong, multitap, and reverb modes.
Switch to set delay type:
- MSec : mode is a ping-pong dual delay with time set in milliseconds
- SYNC : also a ping-pong dual delay, but time is set in tempo units.
- 8X : an 8-tap delay line, with the output from the 8 taps selected by the audio modulation source.
- VRB : a simple reverb unit.
TIME sets the delay time and FBK the amount of feedback.
The CUT switch enables a low-pass filter in the MSEC and SYNC feedback paths, and a high-shelf EQ in the RVB feedback path. CUT sets the filter cutoff in MIDI note units.
Pan sets the panpot of the delay output.
For modes other than 8X, AUDIOMOD enables audio-rate modulation of the delay time, and MOD sets the depth of audio-rate modulation.
For 8X mode, AUDIOMOD selects the source for modulation of the 8-tap output delay. MOD sets the amount of modulation across the 8 taps.
Choir is a monophonic chorus unit with optional feedback.
The 2X mode enables 2-band chorus; the 4X mode enables better sounding but more CPU intensive 4-band chorus; the 8X mode enables a further four bands, for only slightly more CPU than the 4-band mode.
The FLG2 and FLG4 are similar to 2X and 4X, but also provide feedback and cross-feedback options.
DEL sets the chorus delay time, DEEP the depth, SPEED the rate of modulation, and SEP the stereo separation.
For flange modes only, sets the level of feedback and cross-feedback. Be careful with high levels of feedback!
The three LFOs are all polyphonic. Envelopes 1-3 also contain envelopes 4-6 respectively, sharing the same gate and sync sources. The envelopes in the LFOs cause the LFO to fade in and/or fade out.
This is useful for monitoring the sound without the LFO being applied. Turn unused LFOs off to save CPU.
Frequency is adjustable over a scaled range of 0 to 110Hz. Frequencies above 10Hz are generally not effective for the output mixer, because of its internal smoothing (to reduce clicks on large value changes).
When above zero, sets the amount that the frequency is attenuated for each individual voice, so that the LFOs are all at different frequencies.
The LFOs provide sine (actually parabolic), triangle, saw, pulse, and random outputs.
The XY display provides attenuation of overall output amplitude.
The width control does not affect saw waveform output. Otherwise it provides width ratio control in the range 0% to 100%. The random waveform is sampled by the LFO's pulse output, so changing the width introduces a shuffle into the timing of the random waveform output.
Sets the phase at which the LFO starts upon sync and envelope start.
When above zero, causes all the LFO voices to have a different phase.
Selects the sync and gate source for the envelope.
- Use pitch tracking to modulate frequency or amplitude, so higher pitches have deeper modulation.
- Use an envelope with predelay to modulate LFO amplitude for fade-in or fade-out effects.
The waveshaper module ("shaper") is probably the most unique module in the instrument. The shaper can be used in three different ways (simultaneously, if desired):
- audio shaper
- event modulation source
You can draw your own oscillators in the EDIT table. To do so, select "Shaper" in the WAVE switches of oscillators 1-3 and draw the waves, selecting "step," "ramp", or "curve" for each. The oscillator CONTOUR then morphs to the next waveform as follows:
- Osc1 morphs from wave1 when contour=0, to wave2 when contour=1.
- Osc2 morphs from wave2 when contour=0, to wave3 when contour=1.
- Osc3 morphs from wave3 when contour=0, to wave1 when contour=1.
For the oscillators, FM, hard sync, soft sync, and contour are available for the oscillator waveforms you draw.
You can use the waveshaper as an audio shaper. The AUDIO switch in the shaper panel selects the audio input, and the audio output is available as SHAPER in all the audio switches. The audio controls in the shaper panel set drive, saturation, and AGC.
In this mode the waveshaper defines a curve for audio passing into it. When the input audio value is -1, it is mapped to the value at the left edge of the edit table. When the input audio value is +1, it is mapped to the value at the right edge of the table. Intermediate input audio values between -1 and +1 are mapped to corresponding middle points in the EDIT table. The shaper module's SCENE value sets which of the three waveforms are used; intermediate values provide mixes of the three waveforms, and the SCENE can be modulated from matrix 1.
The shaper waveform is also available as an event modulation source. In this mode the shaper essentially provides an additional sequencer that can loop or be triggered by various sources. The sequencer output can be ramped or smoothed, and can morph between the three waves with the SCENE control, in the same way as for audio waveshaping.
The AUDIO switch selects the audio source for the waveshaper. If off, the shaper is still available as an event modulation source.
For event modulation, sets the rate at which the shaper curve is scanned.
For event modulation sets the trigger mode for the shaper:
- LOOP: the shaper loops from left to right, then reverses, continuously.
- MIDI: When any MIDI note on is received, the event shaper restarts at the beginning and plays from left to right once, monophonically (identical on all voices).
- ENV1/2/3: The envelopes trigger the event shaper in the same way as for MIDI, but polyphonically (this means for example the event shaper can apply arbitrary curves to pitch or filter frequency, or can modulate say an LFO amplitude to fade in and out in different ways as a note plays).
- LFO1/2/3: when one of the voices on the corresponding LFO passes from 0 to one, the shaper plays from left to right once, then stops.
- Seq1/2/3/4: when the corresponding sequencer advances one step, the shaper plays the same shape on all voices from left to right once, then stops.
For audio shaping only, sets the drive level passed into the audio waveshaper. Higher levels result in more clipping.
Selects the wave for event and audio shaping. Non-integer morph values provide mixes between the three curves. Note that shaper morph modulation (from matrix 1) can change between the various curves dynamically.
With the EDIT subpanel, you can select whether to edit user wave 1, 2, or 3.
The SCOPE displays the current output curve of the shaper, after morph and point modulation.
There are two sample and hold sources. The panel for sample and hold lets you set the modulation source and the sampling source for each.
The S&H component provides modulation shaping, as well as sample & hold modes. Also, the S&H component provides additional modulation sources.
In the S&H panel, the first listbox sets the modulation source, which may be any of the LFOs, sequencers, envelope, and MIDI.
S&H sources are the same as for matrix sources, and also include some additional sources: envelope pitch 1/2/3. See the MATRIX SOURCE help for more information.
The second listbox sets the trigger/gate source. In HIGH/LOw, and GATE modes, triggers from the sampling source cause the modulation source values to be resampled. For example, if the modulation source is an LFO and the sampling source is an envelope, then whenever that envelope is triggered, the LFO values are resampled for that source.
The following sampling sources are available:
- Env1/2/3 : sampling occurs for each voice individually upon envelope gate-on events.
- S1/2/3/4step : sampling occurs for each voice individually when the respective sequencer advances one step.
- S1/2/3/4loop : the sampling occurs for all voices when the respective sequencer starts its first step.
- LFO1/2/3 : the sampling occurs for each voice individually when the value for the voice in the corresponding LFO passes from 0 to one.
It's possible to select the same source for sampling and modulation, for example an LFO could sample itself upon crossing zero. This could be useful to generate triggers for other modules, but generally would not be used. However setting the same source introduces an event loop, which could crash Reaktor. The sample& hold logic therefore includes a single-cycle event delay (2.5msec by default) to stop Reaktor from crashing should you set the same modulation and sampling source.
- HI TRIG : source is sampled when gate goes high, source is sampled.
- LO TRIG : source is sampled when gate goes low.
- GATE : when gate is >0, source is passed continuously. When gate goes below 0, output goes low.
LOG : Output=(source EXP(0.5)). Source values range between 0 and 1, so:
- EXP: output+(source*source). Source values are between 0 and 1, so:
The scope has an input list to monitor voice 1 of any source. Turn off the scope to reduce CPU usage.
The "Auto" switch automatically adjusts the amplitude so it fits in the window, otherwise the Y gain can be adjusted manually. FREQ controls the refresh rate.
In the B panel, you can select snapshots from any bank, and also overwrite the existing snapshot with the OVERWRITE button.
The instrument includes snaps contributed by beta testers in separate banks.
These modules on the B panel are snap -isolated shapers for incoming MIDI velocity and aftertouch data, allowing you to adapt the instrument to the characteristics of your performance keyboard.
This button back selects between the various shaping modes:
- OFF: output = input.
- LIN: output is linearly scaled between the LO and HI settings. the crosshair in the SHAPE graph additionally lets you set a breakpoint in the linear mapping.
- CRV: the output is instead shaped by a curve, so that higher values provide progressively lower output, or vice versa. To see the available curves, drag the crosshair in the SHAPE area to its four corners.
- BI: the output is shaped by a binomial curve. Adjust the BICURV knob in this mode to adjust the inflection amount, and the crosshair again adjusts the curve slopes.
These set the low and high output values to which input values are mapped. drag the crosshairs up and down to the desired point.
These adjust the interpolation method used when mapping input to output values, as described under MODE above.
To the right of the HI bar, a meter displays the last output value from the unit. When adjusting velocity characteristics, tap the MIDI keyboard slowly and quickly, then adjust the shaper parameters for the desired response. Similarly, when adjusting aftertouch, change the pressure on a key and see if the output range is as desired.