Reaktor Manual 3x2v3o

]. • PsJ: Audio input for sound file position jitter control in ms. Typ. range: [0... ]. • Len: Audio input for setting the grain length in ms. Typ range: [10...100]. Default: 20 ms. • LnJ: Audio input for grain length jittercontrol in ms. Typ range: [10...100]. Default: 0 ms. • Att: Audio input for setting the attack time. Range: [0...1]. Default: 0.2. • Dec: Audio input for setting the decay time. Range: [0...1]. Default: 0.2. • Dist: Audio input for setting the delta time before starting the next grain in ms. Typ range: [5...100]. Default: 20. • DisJ: Audio input for delta time jitter control in ms. Typ range: [10...100]. Default: 20 ms. • Pan: Audio input for setting the position in the stereo field. Range: [-1(Left)...1(Right)]. • PnJ: Audio input for pan jitter control. Range: [0...1]. • A: Audio input for amplitude control. Typ range: [0...1]. Default: 1. • L: Polyphonic left channel audio output. Typ range: [-1...1]. • R: Polyphonic right channel audio output. Typ range: [-1...1]. • Lng: Event output for the length of samples in ms. Typ range: [0... ].

Reaktor English – 290

Beat Loop

Sampler

This is a real-time resynthesizer for the synchronized playback of beat-loop samples. The transposing of beat-loops can be set via the P input independent of playback speed. Beat Loop synchronizes itself to a 96th note clock source (24 ppq) that is connected to the C input or, if the C input is not connected, it can sync with the global REAKTOR clock. Therefore, by default all Beat Loops in REAKTOR run in sync with one another independent of the sample's internal speed. Furthermore, Beat Loop also allows you to easily link internal REAKTOR sequencer modules and MIDI clocks to rhythmic sample material. Sample management is carried out in the Sample Map Editor. Beat Loop requires samples that have been cut exactly, and which contain 2, 4, 8, 16, or 32, etc. beats. The speed of the beat loops used should be between 87 and 174 BPM. If the samples being used fulfill these conditions, Beat Loop can output them in good quality even if the playback speed is changed. By activating the


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• GTr: Event output for grain trigger. Outputs 1 when a new grain starts, 0 when it stops.

sample-related Pitched Sound option (accessible in the properties dialog box), possible pitch falsification of basslines (and similar) can be avoided. However, in doing so you will have to accept a deterioration in rhythmic precision. The loop range of the sample is configured using the Loop Start (LS) and Loop Length (LL) inputs. If LS and LL are not connected, the whole sample will be played back as a loop. Using positive Rst events, sample playback will be continued from the starting point (you can set this via the St input). If the No Stereo option is activated (you can set this in the properties dialog box), stereo samples will be treated as mono samples, i.e. the same signal is sent to the L and R outputs. In this case, Beat Loop uses only the left channel of stereo samples. This option has been made available because mono playback requires less processing capacity. All of Beat Loop's inputs, except C and Rst, are designed as audio inputs. During sample playback, changes to values take effect at intervals of sixteenths of a note value. • C: A positive event at this input switches the Beat Loop module by one 96th note value further. If this input is not connected, the module synchronizes itself with the global REAKTOR clock. • Rst: A positive event at this input resets playback to the position set at the St input. • P: Logarithmic audio control input for the playback rate (pitch). Furthermore, if the Sel input is not connected, P determines the selection of samples from the sample map. • Sel: Audio control input for selecting a sample from the sample map. If this input is not connected, the values present at the P input are used instead. • St: Audio control input for the starting point to be used when the next trigger occurs. The position is set in sixteenths of a note value from the start of the sample. • LS: Audio control input for the loop starting point in sixteenths of a note value from the start of the sample. If this input is not connected, the default is used: LS = 0.


Sample Lookup

Sampler

This module makes samples available as function value look-up tables. A position within the sample in milliseconds is set via the Pos input. The value of the sample at this point is sent to the outputs.


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• LL: Audio control input for the loop length in sixteenths of a note value. If this input is not connected, the default is used: LL = length of the sample. • SO: Audio control input for modulation of the sample position (Sample Offset) in sixteenths of a note value. This input is used to reach steps in the sample which, for instance, can be controlled by a sequencer module. • Sm: Audio control input for the Smoothness of the resynthesis process. The sound particles are adjusted using this. Very small values generally lead to a "cracking" sound every sixteenth interval. • Pan: Audio control input for the position in the stereo field (1 = Left, 0 = Center, 1 = Right). • A: Audio control input for the output amplitude. • L: Audio output for the left channel of the resynthesizer. • R: Audio output for the right channel of the resynthesizer. • Len: Polyphonic event output for the length of the current sample in milliseconds. • L16: Polyphonic event output for the length of the current sample in sixteenths of a note value. • P16: Polyphonic event output for the current position in the sample in sixteenths of a note value. • Ct16: Polyphonic event output for counting the sixteenths of a note value that have ed since the start / reset.

You can load sound files using the Load Sound... entry in the context menu. Sample management is carried out in the Sample Map Editor. The properties dialog box allows you to select one of three levels of quality that is to be used for interpolation during transposed sample playback (Poor, Good, Excellent). If set to Poor, the sample values are not interpolated during output. Higher playback quality is achieved at the expense of processing capacity. • Pos: Audio input for the position in the sample in milliseconds. • A: Audio input for amplitude modulation. • L: Audio output for the left channel of the sample. When processing mono samples, the same signal is output here as is sent to the R output. • R: Audio output for the right channel of the sample. When processing mono samples, the same signal is output here as is sent to the L output. • Lng: Polyphonic event output for the length of the current sample in milliseconds.

Sequencer Reaktor's sequencers include gated step-sequencers in four sizes as well as a selectable-position sequencer. You'll also find clocks to drive them here.


Sequencer

6-Step

Sequencer

Sequencer with 6 steps. The output value for each step (e.g. to control oscillator pitch) can be set independently. In addition a gate signal is output with the amplitude given by the current value of the gate amplitude input at the time the step advances. • C: Audio input for clock control. A positive zero crossing switches to the next step.Typically you would connect a Pulse Oscillator or MIDI Sync module here. • Rst: Audio input for a reset signal. A positive zero crossing puts the sequencer back to the first step. Typically you would connect a button or MIDI Start module here. • GA: Audio input for controlling the amplitude of the gate output. When the value is zero or the input is not connected, no signal appears on the gate output. • S1: Audio input for controlling the value of the first step. • S2: Audio input for controlling the value of the second step.


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Sequencer

• • • • • • •

S3: Audio input for controlling the value of the third step. S4: Audio input for controlling the value of the fourth step. S5: Audio input for controlling the value of the fifth step. S6: Audio input for controlling the value of the sixth step. Out: Event output for the sequencer step signal. G: Event output for the gate signal. St: Event output for the current step number.

8-Step

Sequencer

Like 6-Step Sequencer but with 8 steps.

12-Step

Sequencer

Like 6-Step Sequencer but with 12 steps.

16-Step Like 6-Step Sequencer but with 16 steps.


Sequencer

Sequencer

Value Selector, useful as a sequencer. An event at Pos addresses with its value one of the 16 inputs and the current value at this input is forwarded to Out. If the Pos input is driven by a signal which is incremented stepwise, the output Out provides a sequence of the values at the inputs 0...15. The values at the Pos input are folded into the number range [0 ... (Len - 1)] to allow a variable sequence length Len. The Pos input can also be driven by a control element, a Beat Loop module, a random event source, or by the master clock. • Pos: Input for controlling the selection. Each event at this input results in an event at the outputs. To run Select 16 as a sequencer, the position is set to the number of 16th notes that have ed since start or reset. • Len: Input to set the sequence length. Range: [ 1 ... 16 ].


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Multiplex 16

• 0-15: Audio input for controlling the value of the appropriate step. • Stp: Current sequence step number. The number at this output is calculated as Pos modulo Len. Range: [ 0 ... <sequence length> ]. • Bar: Current bar number. The number at this output is calculated as Integer(Pos / Len). • Out: Current sequence step output value.

LFO, Envelope This category includes a fully modulatable, multiwaveform LFO, a random control generator, and envelope generators of just about any description including time/level ramp generators in three sizes. All Envelopes can optionally display their curve in a graphic. This can be achieved with the Visible option on the Appearance page in the module Properties. The size of the display can be set in the Properties using the Size X and Size Y options. The timeline of the curve in the display is not scaled.

LFO

LFO, Envelope

Low Frequency Oscillator with Pulse, Triangle and Parabol Wave outputs. It is typically used as a source for modulation signals (for vibrato, tremolo etc.). The output signal is a stream of events at the Control Rate selected in the Settings menu. Since the LFO operates at the Control Rate, it is much more U efficient than an audio oscillator which could do the same job.


Slow Random

LFO, Envelope

Low Frequency Oscillator which produces a random step waveform. • F: Control input for the step frequency in Hz. • A: Control input for the amplitude. The output value is somewhere in the range –A to +A. • Out: Event output for the random signal.


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• F: Audio input for control of the oscillator frequency in Hz. For control with Tempo in BPM, you can compute the required value in Hz as follows: F = oscillations-per-beat x BPM/60. So, for example, for 3 oscillations per bar at 4 beats to the bar (that is ¾ oscillations per beat), you get F = (¾)/60 x BPM or 0.0125 x BPM. • A: Input for controlling the amplitude. The output signal moves between +A and –A. • W: Event input for control of pulse width, i.e. the ratio of the duration of the on-period to the off-period for the pulse output, and appropriate warping of the other waveforms. Range of values is -1 to 1. Symmetric waveforms at W = 0. • Snc: Event input for synchronization of the LFO waveform. A positive event synchronizes the oscillator, resetting it to the phase given by the Ph input. • Ph: Input for specifying the phase (position in the waveform) to which the oscillator is reset when synchronization occurs. Ph = 0: Phase = 0 deg (middle of rising slope), Ph = 0.5: Phase = 180 deg (middle of falling slope), Ph = 1: Phase = 360 deg (same as 0 deg). • Sin: Event output for the sine waveshape signal. • Tri: Event output for the triangle waveshape signal. • Pls: Event output for the pulse waveshape signal.

H - Env

LFO, Envelope

Envelope generator with hold characteristic. When the envelope is triggered the output value jumps to the current value of the amplitude input and stays there until the hold time has ed, after which the output jumps back to zero. The envelope can be triggered at any time, including retriggering during the hold time. • T: Audio input for triggering the envelope on a rising edge (value goes from zero in positive direction). • A: Audio input for the output value during the hold time. The input is sampled at the triggering instant after which the value is held at the output. • H: Logarithmic event input for controlling the hold time. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • Out: Audio output for the envelope signal.

HR - Env

LFO, Envelope

Envelope generator with hold-release characteristic. When the envelope is triggered the output value jumps to the current value of the amplitude input and stays there until the hold time has ed, after which the output decays exponentially with the release time back to zero. • T: Audio input for triggering the envelope on a rising edge (value goes from zero in positive direction).


• R: Logarithmic event input for controlling the release time. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • Out: Audio output for the envelope signal.

D - Env

LFO, Envelope

Envelope generator with decay characteristic. When the envelope is triggered the output value jumps to the current value of the amplitude input, after which the output decays exponentially with the decay time back to zero. • T: Audio input for triggering the envelope on a rising edge (value goes from zero in positive direction). • A: Audio input for the initial output value. The input is sampled at the triggering instant. • D: Logarithmic event input for controlling the decay time. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • Out: Audio output for the envelope signal.

DR - Env

LFO, Envelope


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• A: Audio input for the output value during the hold time. The input is sampled at the triggering instant after which the value is held at the output. • H: Logarithmic event input for controlling the hold time. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms).

Envelope generator with decay-release characteristic. When the envelope is triggered with a gate event the output value jumps to the amplitude value of the gate signal, after which the output decays exponentially with the decay time back to zero. A gate event with amplitude zero (at note off) sets the decaying time to the release time value which is normally set to be shorter. • G: Event input for the gate signal that triggers the envelope. The amplitude of the gate signal determines the initial output value. • D: Logarithmic event input for controlling the decay time. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • R: Logarithmic event input for controlling the release time. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • Out: Audio output for the envelope signal.

DSR - Env

LFO, Envelope

Envelope generator with decay-sustain-release characteristic. When the envelope is triggered with a gate event the output value jumps to the amplitude value of the gate signal, after which the output decays exponentially with the decay time to the sustain level (multiplied by the amplitude). After a gate event with amplitude zero (at note off) the output decays exponentially with the release time back to zero. • G: Event input for the gate signal that triggers the envelope. The amplitude of the gate signal determines the initial output value. • D: Logarithmic event input for controlling the decay time. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms).


• Out: Audio output for the envelope signal.

DBDR - Env

LFO, Envelope

Envelope generator with decay-breakpoint-release characteristic. When the envelope is triggered with a gate event the output value jumps to the amplitude value of the gate signal, after which the output decays exponentially with the decay-1 time until it reaches the breakpoint level (multiplied by the amplitude). Next it continues decaying exponentially with the decay-2 time back to zero. A gate event with amplitude zero (at note off) sets the decaying time to the release time value which is normally set to be shorter. • G: Event input for the gate signal that triggers the envelope. The amplitude of the gate signal determines the initial output value. • D1: Logarithmic event input for controlling the first decay time. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • B: Event input for controlling the breakpoint level. Range of values: 0 (never use decay-2 time) to 1 (immediately use decay-2 time). • D2: Logarithmic event input for controlling the second decay time. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • R: Logarithmic event input for controlling the release time. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms).


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• S: Event input for controlling the sustain level. Typical range of values is: 0 (decay to zero) to 1 (hold at initial level), but sustain can be greater than 1 or even less than 0. • R: Logarithmic event input for controlling the release time. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms).


DBDSR-Env

LFO, Envelope

Envelope generator with decay-breakpoint-decay-sustain-release characteristic. When the envelope is triggered with a gate event the output value jumps to the amplitude value of the gate signal, after which the output decays exponentially with the decay-1 time until it reaches the breakpoint level (multiplied by the amplitude). Next it continues decaying with the decay-2 time to the sustain level (multiplied by the amplitude). The output is held at the sustain level until a gate event with amplitude zero (at note off) arrives, after which the output decays exponentially with the release time back to zero. • G: Event input for the gate signal that triggers the envelope. The amplitude of the gate signal determines the initial output value. • D1: Logarithmic event input for controlling the first decay time. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • B: Event input for controlling the breakpoint level. Range of values: 0 (never use decay-2 time) to 1 (immediately use decay-2 time). • D2: Logarithmic event input for controlling the second decay time. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • S: Event input for controlling the sustain level. Typical range of values is: 0 (decay to zero) to 1 (hold at end of attack), but sustain can be greater than 1 or even less than 0. • R: Logarithmic event input for controlling the release time. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). Reaktor English – 304

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AD - Env

LFO, Envelope

Attack-Decay envelope, triggered by the positive slope of the T signal. The decay starts immediately, when the attack time is over. • T: Input for the trigger signal. A positive slope starts the envelope. • A: Control input for the output amplitude. • A: Control input for the attack time of the envelope. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • D: Control input for the decay time of the envelope. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • Out: Output for the envelope signal.

AR - Env

LFO, Envelope

Envelope generator with attack-release characteristic. When the envelope is triggered with a gate event the output value rises during the attack time with a linear slope to the amplitude value of the gate signal. The output is then held at the maximum level until a gate event with amplitude zero (at note off) arrives, after which the output decays exponentially with the release time back to zero. Reaktor English – 305

• G: Event input for the gate signal that triggers the envelope. The amplitude of the gate signal determines the maximum output value. • A: Logarithmic event input for controlling the attack time. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • R: Logarithmic event input for controlling the release time. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • Out: Audio output for the envelope signal.

ADR-Env

LFO, Envelope

Envelope generator with attack-decay-release characteristic. When the envelope is triggered with a gate event the output value rises during the attack time with a linear slope to the amplitude value of the gate signal, after which it decays with the decay time to zero. When a gate event with amplitude zero (at note off) arrives, the output decays exponentially with the release time back to zero. • G: Event input for the gate signal that triggers the envelope. The amplitude of the gate signal determines the maximum output value. • A: Logarithmic event input for controlling the attack time. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • D: Logarithmic event input for controlling the decay time. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • R: Logarithmic event input for controlling the release time. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • Out: Audio output for the envelope signal.


LFO, Envelope

Envelope generator with the classic attack-decay-sustain-release characteristic. When the envelope is triggered with a gate event the output value rises during the attack time with a linear slope to the amplitude value of the gate signal, after which it decays exponentially with the decay time to the sustain level (multiplied by the amplitude). The output is held at the sustain level until a gate event with amplitude zero (at note off) arrives, after which the output decays exponentially with the release time back to zero. • G: Event input for the gate signal that triggers the envelope. The amplitude of the gate signal determines the maximum output value. • A: Logarithmic event input for controlling the attack time. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • D: Logarithmic event input for controlling the decay time. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • S: Event input for controlling the sustain level. Typical range of values is: 0 (decay to zero) to 1 (hold at end of attack), but sustain can be greater than 1 or even less than 0. • R: Logarithmic event input for controlling the release time. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • Out: Audio output for the envelope signal.


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ADSR - Env

ADBDR - Env

LFO, Envelope

Envelope generator with attack-decay-breakpoint-decay-release characteristic. When the envelope is triggered with a gate event the output value rises during the attack time with a linear slope to the amplitude value of the gate signal, after which it decays exponentially with the decay-1 time until it reaches the breakpoint level (multiplied by the amplitude). Next it continues decaying exponentially with the decay-2 time back to zero. A gate event with amplitude zero (at note off) sets the decaying time to the release time value which is normally set to be shorter. • G: Event input for the gate signal that triggers the envelope. The amplitude of the gate signal determines the maximum output value. • A: Logarithmic event input for controlling the attack time. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • D1: Logarithmic event input for controlling the first decay time. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • B: Event input for controlling the breakpoint level. Range of values: 0 (never use decay-2 time) to 1 (immediately use decay-2 time). • D2: Logarithmic event input for controlling the second decay time. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • R: Logarithmic event input for controlling the release time. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • Out: Audio output for the envelope signal.


LFO, Envelope

Envelope generator with attack-decay-breakpoint-decay-sustainrelease characteristic. When the envelope is triggered with a gate event the output value rises during the attack time with a linear slope to the amplitude value of the gate signal, after which it decays according to the first decay time with a linear slope to the breakpoint level. From there it continues with the second decay time to the sustain level (the levels are multiplied by the amplitude). The output is held at the sustain level until a gate event with amplitude zero (at note off) arrives, after which the output decays exponentially with the release time back to zero. • G: Event input for the gate signal that triggers the envelope. The amplitude of the gate signal determines the maximum output value. • A: Logarithmic event input for controlling the attack time. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • D1: Logarithmic event input for controlling the first decay time. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • B: Event input for controlling the break point level. Typical range of values is: 0 to 1. • D2: Logarithmic event input for controlling the second decay time. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • S: Event input for controlling the sustain level. Typical range of values is: 0 to 1. • R: Logarithmic event input for controlling the release time. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • Out: Audio output for the envelope signal.


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ADBDSR-Env

AHDSR - Env

LFO, Envelope

Envelope generator with attack-hold-decay-sustain-release characteristic. When the envelope is triggered with a gate event the output value rises during the attack time with a linear slope to the amplitude value of the gate signal and stays there until the hold time has ed, after which it decays according to the decay time with a linear slope to the sustain level. The output is held at the sustain level until a gate event with amplitude zero (at note off) arrives, after which the output decays exponentially with the release time back to zero. • G: Event input for the gate signal that triggers the envelope. The amplitude of the gate signal determines the maximum output value. • A: Logarithmic event input for controlling the attack time. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • H: Logarithmic event input for controlling the hold time. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • D: Logarithmic event input for controlling the decay time. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • S: Event input for controlling the sustain level. Typical range of values is: 0 (decay to zero) to 1 (hold at end of attack), but sustain can be greater than 1 or even less than 0. • R: Logarithmic event input for controlling the release time. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • Out: Audio output for the envelope signal.


LFO, Envelope

Envelope generator with attack-hold-decay-breakpoint-decay-release characteristic. When the envelope is triggered with a gate event the output value rises during the attack time with a linear slope to the amplitude value of the gate signal and stays there until the hold time has ed, after which it decays exponentially with the decay-1 time until it reaches the breakpoint level (multiplied by the amplitude). Next it continues decaying exponentially with the decay-2 time back to zero. A gate event with amplitude zero (at note off) sets the decaying time to the release time value which is normally set to be shorter. • G: Event input for the gate signal that triggers the envelope. The amplitude of the gate signal determines the maximum output value. • A: Logarithmic event input for controlling the attack time. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • H: Logarithmic event input for controlling the hold time. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • D1: Logarithmic event input for controlling the first decay time. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • B: Event input for controlling the breakpoint level. Range of values: 0 (never use decay-2 time) to 1 (immediately use decay-2 time). • D2: Logarithmic event input for controlling the second decay time. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • R: Logarithmic event input for controlling the release time. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms).


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AHDBDR - Env


4-Ramp

LFO, Envelope

4-stage envelope generator with linear slopes. When the envelope is triggered with a gate event the output value rises to the first level, reaching it within the time set for the first stage. Then it continues to the second level within the time set for the second stage, and so on. The third level is the sustain level, at which the output is held until a gate event with amplitude zero (at note off) arrives, after which the output returns to zero within the last timeperiod. • G: Event input for the gate signal that triggers the envelope. The amplitude of the gate signal combines with all the level values to determine the actual levels reached. • T1: Logarithmic event input for controlling the time for the first stage. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • L1: Input for controlling the level which is reached at the end of the first stage. • T2: Logarithmic event input for controlling the time for the second stage. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • L2: Input for controlling the level which is reached at the end of the second stage.


• LS: Input for controlling the level which is reached at the end of the third stage. This is the sustain level. • TR: Logarithmic event input for controlling the time for the last stage which is the release. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • St: Event output for the envelope’s current stage (1, 2 ...). After the end of the release stage and before a new trigger, the value is 0. With appropriate processing this value can be used to chain other envelopes, or for the envelope to retrigger itself. • Out: Audio output for the envelope signal.

5-Ramp

LFO, Envelope

5-stage envelope generator with linear slopes. When the envelope is triggered with a gate event the output value rises to the first level, reaching it within the time set for the first stage. Then it continues to the second level within the time set for the second stage, and so on. The fourth level is the sustain level, at which the output is held until a gate event with amplitude zero (at note off) arrives, after which the output returns to zero within the last timeperiod.


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• T3: Logarithmic event input for controlling the time for the third stage. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms).

• G: Event input for the gate signal that triggers the envelope. The amplitude of the gate signal combines with all the level values to determine the actual levels reached. • T1: Logarithmic event input for controlling the time for the first stage. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • L1: Input for controlling the level which is reached at the end of the first stage. • T2: Logarithmic event input for controlling the time for the second stage. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • L2: Input for controlling the level which is reached at the end of the second stage. • T3: Logarithmic event input for controlling the time for the third stage. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • L3: Input for controlling the level which is reached at the end of the third stage. • T4: Logarithmic event input for controlling the time for the fourth stage. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • LS: Input for controlling the level which is reached at the end of the fourth stage. This is the sustain level. • TR: Logarithmic event input for controlling the time for the last stage, which is the release. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • St: Event output for the envelope’s current stage (1, 2 ...). After the end of the release stage and before a new trigger, the value is 0. With appropriate processing this value can be used to chain other envelopes, or for the envelope to retrigger itself. • Out: Audio output for the envelope signal.


LFO, Envelope

6-stage envelope generator with linear slopes. When the envelope is triggered with a gate event the output value rises to the first level, reaching it within the time set for the first stage. Then it continues to the second level within the time set for the second stage, and so on. The fifth level is the sustain level, at which the output is held until a gate event with amplitude zero (at note off) arrives, after which the output returns to zero within the last timeperiod. • G: Event input for the gate signal that triggers the envelope. The amplitude of the gate signal combines with all the level values to determine the actual levels reached. • T1: Logarithmic event input for controlling the time for the first stage. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • L1 :Input for controlling the level which is reached at the end of the first stage. • T2: Logarithmic event input for controlling the time for the second stage. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • L2: Input for controlling the level which is reached at the end of the second stage.


ENGLISH

6-Ramp

• T3: Logarithmic event input for controlling the time for the third stage. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • L3: Input for controlling the level which is reached at the end of the third stage. • T4: Logarithmic event input for controlling the time for the fourth stage. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • L4: Input for controlling the level which is reached at the end of the fourth stage. • T5: Logarithmic event input for controlling the time for the fifth stage. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • LS: Input for controlling the level which is reached at the end of the fifth stage. This is the sustain level. • TR: Logarithmic event input for controlling the time for the last stage, which is the release. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • St: Event output for the envelope’s current stage (1, 2 ...). After the end of the release stage and before a new trigger, the value is 0. With appropriate processing this value in can be used to chain other envelopes, or for the envelope to retrigger itself. • Out: Audio output for the envelope signal.

Filter Filters make up Reaktor's largest collection of signal processors. You'll find 22 of them here covering everything from standard low, high-, and band, to emulations of classic synth filters from Sequential and Moog. There's also an all filter for reverb and dispersion circuits as well as integrating and differentiating filters.


All the filters and EQs can optionally display their frequency response in a graph on the . This can be achieved with the Visible option on the Appearance page in the module Properties. The size of the display can be set in the Properties using the Size X and Size Y options. The graph’s frequency axis is logarithmic and ranges from 10 Hz to 20 kHz.

HP/LP 1-Pole

Filter

1-pole filter with high and low outputs (6 dB/octave falloff) and logarithmic control of cutoff frequency. • P: Logarithmic event input for controlling the cutoff frequency. Value in semitones (69 = 440 Hz). • In: Audio signal input for the signal to be filtered • HP: Audio output for the high filtered signal • LP: Audio output for the low filtered signal

HP/LP 1-Pole FM

Filter


ENGLISH

All of REAKTOR’s filters can operate at any frequency, from 0 Hz (constant signal) through the entire audio range right up to the limit set by the sample rate. This means they are all equally suited for audio processing or for smoothing control signals (e.g. portamento). When using a filter to process the input to a port which is of the type that only accepts events (e.g. P as opposed to F) you need to insert an A to E (perm) module for conversion.

1-pole filter with high and low outputs (6 dB/octave falloff), logarithmic and linear control of cutoff frequency. • P: Logarithmic event input for controlling the cutoff frequency. Value in semitones (69 = 440 Hz). • F: Audio input for linear control of the cutoff frequency. Value in Hz. P and F together determine the oscillator frequency. • In: Audio signal input for the signal to be filtered • HP: Audio output for the high filtered signal • LP: Audio output for the low filtered signal

All 1-Pole

Filter

First order all filter. This type of filter has a flat amplitude response, but the phase shift between input and output grows from 0 degrees at low frequencies to -180 degrees at high frequencies. At the corner frequency controlled by the P input the phase is shifted for -90 degrees. • P: Logarithmic control input for the corner frequency, where the phase shift is -90 degrees. Scale: 1 Semitone per unit, 43 = G1 = 98 Hz, 84 = C5 = 1047 Hz. • In: Input for the signal to be all filtered (phase-shifted). • Out: Output for the all filtered (phased-shifted signal).

Multi 2-Pole


Filter

The band gain is always 1 (0 dB), whereas the gain at the cutoff frequency increases with the resonance. Caution: Very large amplitudes are generated when Res is almost 1. • P: Logarithmic event input for controlling the cutoff frequency. Value in semitones (69 = 440 Hz). • Res: Event input for controlling the filter's resonance/damping. Range of values 0 (maximal damping, no resonance) to 1 (no damping, maximal resonance, self oscillation!). At high resonance the filter’s Q-factor = frequency [Hz] / bandwidth [Hz] ≅ 1 / (2 – 2 Res). • In: Audio signal input for the signal to be filtered • HP: Audio output for the high filtered signal • BP: Audio output for the band filtered signal • LP: Audio output for the low filtered signal

Multi 2-Pole FM

Filter

2-pole filter with high and low outputs (12 dB/octave falloff), band (above and below each 6 dB/octave falloff), variable resonance, logarithmic and linear control of cutoff frequency. The band gain is always 1 (0 dB), whereas the gain at the cutoff frequency increases with the resonance. Caution: Very large amplitudes are generated when Res is almost 1. • P: Logarithmic event input for controlling the cutoff frequency. Value in semitones (69 = 440 Hz).


ENGLISH

2-pole filter with high and low outputs (12 dB/octave falloff), band (above and below each 6 dB/octave falloff), variable resonance, and logarithmic control of cutoff frequency.

• F: Audio input for linear control of the cutoff frequency. Value in Hz. P and F together determine the oscillator frequency. • Res: Event input for controlling the filter's resonance/damping. Range of values 0 (maximal damping, no resonance) to 1 (no damping, maximal resonance, self oscillation!). At high resonance the filter’s Q-factor = frequency [Hz] / bandwidth [Hz] ≅ 1 / (2 – 2 Res). • In: Audio signal input for the signal to be filtered • HP: Audio output for the high filtered signal • BP: Audio output for the band filtered signal • LP: Audio output for the low filtered signal

Multi/Notch 2-Pole

Filter

2-pole filter with band reject (notch) output, high and low outputs (12 dB/octave falloff), band (above and below each 6 dB/octave falloff), variable resonance and logarithmic control of cutoff frequency. The gain at the cutoff frequency is always 1 (0 dB), whereas the band gain decreases with increasing resonance. At the notch filter output the selected frequency is completely removed. • P: Logarithmic event input for controlling the cutoff frequency. Value in semitones (69 = 440 Hz). • Res: Event input for controlling the filter's resonance/damping. Range of values 0 (maximal damping, no resonance) to 1 (no damping, maximal resonance, self oscillation!). At high resonance the filter’s Q-factor = frequency [Hz] / bandwidth [Hz] ≅ 1 / (2 – 2 Res). • In: Audio signal input for the signal to be filtered


HP: Audio output for the high filtered signal BP: Audio output for the band filtered signal LP: Audio output for the low filtered signal N: Audio output for the band reject (notch) filtered signal

Multi/Notch 2-Pole FM

Filter

2-pole filter with band reject (notch) output, high and low outputs (12 dB/octave falloff), band (above and below each 6 dB/octave falloff), variable resonance, logarithmic and linear control of cutoff frequency. The gain at the cutoff frequency is always 1 (0 dB), whereas the band gain decreases with increasing resonance. At the notch filter output the selected frequency is completely removed. • P: Logarithmic event input for controlling the cutoff frequency. Value in semitones (69 = 440 Hz). • F: Audio input for linear control of the cutoff frequency. Value in Hz. P and F together determine the oscillator frequency. • Res: Event input for controlling the filter's resonance/damping. Range of values 0 (maximal damping, no resonance) to 1 (no damping, maximal resonance, self oscillation!). At high resonance the filter’s Q-factor = frequency [Hz] / bandwidth [Hz] ≅ 1 / (2 – 2 Res). • In: Audio signal input for the signal to be filtered • HP: Audio output for the high filtered signal • BP: Audio output for the band filtered signal • LP: Audio output for the low filtered signal • N: Audio output for the band reject (notch) filtered signal


ENGLISH

• • • •

Multi/LP 4-Pole

Filter

4-pole filter with low (24 dB/oct falloff), band (12/ 12 dB/oct) and band/low (6/18 dB/oct) outputs, 2-pole high and low (12 dB/oct) and band (6/6 dB/oct) outputs, variable resonance and logarithmic control of cutoff frequency. The band gain is always 1 (0 dB), whereas the gain at the cutoff frequency increases with the resonance. Caution: Very large amplitudes are generated when Res is almost 1. • P: Logarithmic event input for controlling the cutoff frequency. Value in semitones (69 = 440 Hz). • Res: Event input for controlling the filter's resonance/damping. Range of values 0 (maximal damping, no resonance) to 1 (no damping, maximal resonance, self oscillation!). • In: Audio signal input for the signal to be filtered • HP2: Audio output for the 2-pole high filtered signal • BP2: Audio output for the 2-pole band filtered signal • LP2: Audio output for the 2-pole low filtered signal • BP4: Audio output for the 4-pole band filtered signal • BL4: Audio output for the 4-pole band/low filtered signal • LP4: Audio output for the 4-pole low filtered signal


Filter

4-pole filter with low (24 dB/oct falloff), band (12/ 12 dB/oct) and band/low (6/18 dB/oct) outputs, 2-pole high and low (12 dB/oct) and band (6/6 dB/oct) outputs, variable resonance, logarithmic and linear control of cutoff frequency. The band gain is always 1 (0 dB), whereas the gain at the cutoff frequency increases with the resonance. Caution: Very large amplitudes are generated when Res is almost 1. • P: Logarithmic event input for controlling the cutoff frequency. Value in semitones (69 = 440 Hz). • F: Audio input for linear control of the cutoff frequency. Value in Hz. P and F together determine the oscillator frequency. • Res: Event input for controlling the filter's resonance/damping. Range of values 0 (maximal damping, no resonance) to 1 (no damping, maximal resonance, self oscillation!). • In: Audio signal input for the signal to be filtered • HP2: Audio output for the 2-pole high filtered signal • BP2: Audio output for the 2-pole band filtered signal • LP2: Audio output for the 2-pole low filtered signal • BP4: Audio output for the 4-pole band filtered signal • BL4: Audio output for the 4-pole band/low filtered signal • LP4: Audio output for the 4-pole low filtered signal


ENGLISH

Multi/LP 4-Pole FM

Multi/HP 4-Pole

Filter

4-pole filter with high (24 dB/oct falloff), band (12/ 12 dB/oct) and band/high (18/6 dB/oct) outputs, 2-pole high and low (12 dB/oct) and band (6/6 dB/oct) outputs, variable resonance and logarithmic control of cutoff frequency. The band gain is always 1 (0 dB), whereas the gain at the cutoff frequency increases with the resonance. Caution: Very large amplitudes are generated when Res is almost 1. • P: Logarithmic event input for controlling the cutoff frequency. Value in semitones (69 = 440 Hz). • Res: Event input for controlling the filter's resonance/damping. Range of values 0 (maximal damping, no resonance) to 1 (no damping, maximal resonance, self oscillation!). • In: Audio signal input for the signal to be filtered • HP2: Audio output for the 2-pole high filtered signal • BP2: Audio output for the 2-pole band filtered signal • LP2: Audio output for the 2-pole low filtered signal • HP4: Audio output for the 4-pole high filtered signal • BH4: Audio output for the 4-pole band/high filtered signal • BP4: Audio output for the 4-pole band filtered signal


Filter

4-pole filter with high (24 dB/oct falloff), band (12/ 12 dB/oct) and band/high (18/6 dB/oct) outputs, 2-pole high and low (12 dB/oct) and band (6/6 dB/oct) outputs, variable resonance, logarithmic and linear control of cutoff frequency. The band gain is always 1 (0 dB), whereas the gain at the cutoff frequency increases with the resonance. Caution: Very large amplitudes are generated when Res is almost 1. • P: Logarithmic event input for controlling the cutoff frequency. Value in semitones (69 = 440 Hz). • F: Audio input for linear control of the cutoff frequency. Value in Hz. P and F together determine the oscillator frequency. • Res: Event input for controlling the filter's resonance/damping. Range of values 0 (maximal damping, no resonance) to 1 (no damping, maximal resonance, self oscillation!). • In: Audio signal input for the signal to be filtered • HP2: Audio output for the 2-pole high filtered signal • BP2: Audio output for the 2-pole band filtered signal • LP2: Audio output for the 2-pole low filtered signal • HP4: Audio output for the 4-pole high filtered signal • BH4: Audio output for the 4-pole band/high filtered signal • BP4: Audio output for the 4-pole band filtered signal


ENGLISH

Multi/HP 4-Pole FM

Pro-52 Filter

Filter

Filter taken from the Pro-52 virtual analog synthesizer. It is a 4pole low filter (24 dB/oct falloff) with variable resonance and logarithmic and linear control of cutoff frequency. The filter goes into self-oscillation when Res approaches the value 1. The amplitude of the self-oscillation is approximately 1. • P: Logarithmic event input for controlling the cutoff frequency. Value in semitones (69 = 440 Hz). • F: Audio input for linear control of the cutoff frequency. Value in Hz. P and F together determine the oscillator frequency. • Res: Event input for controlling the filter's resonance/damping. Range of values 0 (maximal damping, no resonance) to 1 (no damping, maximal resonance, self-oscillation). • In: Audio signal input for the signal to be filtered • Out: Audio output for the 4-pole low filtered signal

Ladder Filter

Filter

Same as Ladder Filter FM but without the frequency modulation input F. See next module description.


Filter

Filter modelled on the classic ladder circuit patented by Bob Moog. It is a 4-pole filter with different low outputs: 24 dB/ oct, 18 dB/oct , 12 dB/oct and 6 dB/oct falloff. It also has variable resonance and logarithmic and linear control of cutoff frequency. The saturation characteristic of the analog circuit can optionally be simulated by turning on Distortion in the properties. When Distortion is enabled, the filter goes into self-oscillation when the value of Res is 1 or more. The amplitude of the self-oscillation is approximately 1 when Res is just above 1, but can be much larger when driving Res even higher. The filter also has options for selecting the quality of the simulation: Standard, High and Excellent. The effect is particularly noticeable when distortion is turned on. Of course, the higher quality settings come at the price of increased U usage. • P: Logarithmic event input for controlling the cutoff frequency. Value in semitones (69 = 440 Hz). • F: Audio input for linear control of the cutoff frequency. Value in Hz. P and F together determine the oscillator frequency. • Res: Event input for controlling the filter's resonance/damping. Range of values 0 (no resonance) to 1 (maximal resonance, self-oscillation). Values above 1 are possible in Distortion mode. • In: Audio signal input for the signal to be filtered • LP1: Audio output for the 6 dB/oct low filtered signal • LP2: Audio output for the 12 dB/oct low filtered signal • LP3: Audio output for the 18 dB/oct low filtered signal • LP4: Audio output for the 24 dB/oct low filtered signal


ENGLISH

Ladder Filter FM

Peak EQ

Filter

Parametric equalizer with adjustable boost/cut, band width and logarithmic frequency control. With the Peak EQ a specific frequency in the signal and a narrow or wide band of frequencies around it can be amplified or attenuated. More distant frequencies are not affected. • P: Logarithmic event input for controlling the cutoff frequency. Value in semitones (69 = 440 Hz). • Res: Event input for controlling the filter's resonance (Q-factor). Range of values 0 (minimal resonance, maximal band width) to 1 (maximal resonance, minimal band width). At high resonance the filter’s Q-factor = frequency [Hz] / bandwidth [Hz] ≅ 1 / (2 – 2 Res). • B: Event input for controlling boost/cut in dB. At value B = 0 the signal remains unchanged. • In: Audio signal input for the signal to be equalized • Out: Audio output for the equalized signal

Peak EQ FM


Filter

• P: Logarithmic event input for controlling the cutoff frequency. Value in semitones (69 = 440 Hz). • F: Audio input for linear control of the cutoff frequency. Value in Hz. P and F together determine the oscillator frequency. • Res: Event input for controlling the filter's resonance (Q-factor). Range of values 0 (minimal resonance, maximal band width) to 1 (maximal resonance, minimal band width). At high resonance the filter’s Q-factor = frequency [Hz] / bandwidth [Hz] ≅ 1 / (2 – 2 Res). • B: Event input for controlling boost/cut in dB. At value B = 0 the signal remains unchanged. • In: Audio signal input for the signal to be equalized • Out: Audio output for the equalized signal

High Shelf EQ

Filter

Parametric equalizer with high shelving characteristic, adjustable boost/cut, band width and logarithmic frequency control. The level of frequencies above the corner frequency is raised or reduced by the set amount. Low frequencies are not affected. • P: Logarithmic event input for controlling the cutoff frequency. Value in semitones (69 = 440 Hz). • B: Event input for controlling boost/cut in dB. At value B = 0 the signal remains unchanged. • In: Audio signal input for the signal to be equalized • Out: Audio output for the equalized signal


ENGLISH

Parametric equalizer with adjustable boost/cut, band width and logarithmic and linear frequency control. With the Peak EQ a specific frequency in the signal and a narrow or wide band of frequencies around it can be amplified or attenuated. More distant frequencies are not affected.

High Shelf EQ FM

Filter

Parametric equalizer with high shelving characteristic, adjustable boost/cut, band width and logarithmic and linear frequency control. The level of frequencies above the corner frequency is raised or reduced by the set amount. Low frequencies are not affected. • P: Logarithmic event input for controlling the cutoff frequency. Value in semitones (69 = 440 Hz). • F: Audio input for linear control of the cutoff frequency. Value in Hz. P and F together determine the oscillator frequency. • B: Event input for controlling boost/cut in dB. At value B = 0 the signal remains unchanged. • In: Audio signal input for the signal to be equalized • Out: Audio output for the equalized signal

Low Shelf EQ

Filter

Parametric equalizer with low shelving characteristic, adjustable boost/cut, band width and logarithmic frequency control. The level of frequencies below the corner frequency is raised or reduced by the set amount. High frequencies are not affected. • P: Logarithmic event input for controlling the cutoff frequency. Value in semitones (69 = 440 Hz).


Low Shelf EQ FM

Filter

Parametric equalizer with low shelving characteristic, adjustable boost/cut, band width and logarithmic and linear frequency control. The level of frequencies below the corner frequency is raised or reduced by the set amount. High frequencies are not affected. • P: Logarithmic event input for controlling the cutoff frequency. Value in semitones (69 = 440 Hz). • F: Audio input for linear control of the cutoff frequency. Value in Hz. P and F together determine the oscillator frequency. • B: Event input for controlling boost/cut in dB. At value B = 0 the signal remains unchanged. • In: Audio signal input for the signal to be equalized • Out: Audio output for the equalized signal

Differentiator

Filter


ENGLISH

• B: Event input for controlling boost/cut in dB. At value B = 0 the signal remains unchanged. • In: Audio signal input for the signal to be equalized • Out: Audio output for the equalized signal

The differentiator gives the slope of the input signal in change units per millisecond. The effect is like a high filter where the amplification is proportional to frequency. Unity gain at 159 Hz. • In: Audio signal input for the signal to be differentiated • Out: Audio output for the differentiated signal

Integrator

Filter

Integrator with Reset input. The output value changes with a slope given by the input in change units per millisecond. The effect is like a low filter where amplification is proportional to 1/frequency. Unity gain at 159 Hz. • In: Audio signal input for the signal to be integrated • Set: Event input for the reset. When an event is received, the output signal is set to the value of the event. • Out: Audio output for the integrated signal

Delay Reaktor provides six types of delay to accomplish most delay functions. Those include two tap delays, two granular delays, a diff (handy for reverb circuits), and a unit delay useful in loop-back processing and physical modeling.

Single Delay


Delay

The upper limit for the delay time can be adjusted in the module's properties dialog window in the Max Delay Buffer field (default: 1 second) and affects the memory usage. How big this value can be set depends on the amount of available RAM storage in the computer. At a sample rate of 44.1 kHz you need 172 kB of RAM for each second of buffer length and for each voice. For one minute you need 10 MB. If the Delay is used as an Event Delay (the Inport is red indicating that the module is in Event processing mode) you can set the Max Count Of Buffered Events at the same place. This module replaces several modules of former REAKTOR version: • It behaves like a REAKTOR 3 Static Delay if you connect an audio signal to the In-input and an event signal to the Dly-input. If the delay time does not correspond to an integer number of samples, the output signal is generated by interpolation. The interpolation method can be chosen in the properties. Linear interpolation may reduce high frequency components in the sound somewhat. • It behaves like a REAKTOR 3 Modulation Delay if you connect an audio signal to the In-input and an audio signal to the Dlyinput.The delay time can be continuously modulated by an audio signal. If the delay time does not correspond to an integer number of samples, the output signal is generated by interpolation. The interpolation method can be chosen in the properties. Linear interpolation may reduce high frequency components in the sound somewhat. • It behaves like a REAKTOR 3 Event Delay if you connect an event signal to the In-input.

Ports • Dly: Hybrid input for controlling the delay time. Value in milliseconds. • In: Hybrid input for the signal to be delayed. Reaktor English – 333

ENGLISH

Polyphonic delay line for audio and event signals. The input signal appears at the output with a delay corresponding to the set time. The delay time is controlled at the Dly input.

• Out: Hybrid output for the delayed signal.

M u l t i - Ta p D e l a y

Delay

Multi tap delay line for audio signals. If the set delay time corresponds to a non-integer number of sample, interpolation occurs. The interpolation method can be set in the properties. The outputs are usually connected to a mixer or scanner. • In: Audio input for the signal to be delayed. • Dly1...8: Audio inputs for control of the delay time in milliseconds. Typ range: [0...1000]. • 1...8: Audio outputs for the delayed input signal. 1 is delayed by time Dly1, 2 by Dly2 etc.

Diff Delay


Delay

The delay time is controlled at the Dly input. The delay time can be continuously modulated by an audio signal. If the delay time does not correspond to an integer number of samples, the output signal is generated by interpolation. The interpolation method can be chosen in the properties. Linear interpolation may reduce high frequency components in the sound somewhat. The amount of internal is controlled at the Dffs input. When the delay time is set to zero, the Diff functions as a 1pole all filter. Like this you can construct a phaser, for example, by connecting several diffs in series and modulating the Dffs parameter. The upper limit for the delay time can be adjusted in the module's properties dialog window (default: 200 ms) and affects the memory usage. • Dly: Hybrid input for controlling the delay time. Value in milliseconds. • Dffs: Event input for controlling the diffusion coefficient. Range of values: -1 ... 1. Dffs = 0 : the module is a pure delay, Dffs = 1 : output = input, Dffs = -1 : output = -input. The most useful values for Dffs are near 0.5. • In: Audio input for the signal to be diffused. • Out: Audio output for the diffused signal.


ENGLISH

The diff is an all- filter containing a delay line with . Its function is to smear (decorrelate) the input signal without emphasizing any frequency components. Its typical use is as a building block for reverb type effects, where you would connect several of these modules in series and give them different delay times of a few milliseconds.

Grain Delay

Delay

A delay line and pitch-shifter for audio signals. The input signal appears at the outputs with a delay corresponding to the set time at the Dly input, transposed by the amount set at the P input in semitones. The input signal is “chopped-up” into sound particles, whose size is controlled by the Granularity input (Gr). The “roughness” of the resulting sound can be controlled via the Smoothness input (Sm). The position of the sound particles in the stereo field is set at the Pan input. The delay time of the Grain Delay can be varied without affecting the pitch. Interesting effects are possible in combination with random/noise generators. In the properties dialog window the playback quality and also the upper limit of the delay time can be set. The available maximum delay time can deviate from the set amount by up to 50 %. • P: Logarithmic audio control input for transposing in semitones (Pitch). • Dly: Audio control input for the delay time in milliseconds. • Gr: Audio control input for the granularity of the re-synthesis process, in milliseconds. This parameter sets the size of the sound particles used for the re-synthesis. • Sm: Audio control input for the Smoothness of the re-synthesis process. This affects the formation of the sound particles. Low settings result generally in a rougher sound. • Pan: Audio control input for stereo field positioning (-1 = Left, 0 = Middle, 1 = Right).


A: Audio control input for the output amplitude. In: Input for the audio signal to be delayed. L: Audio output for the left channel of the delay. R: Audio output for the right channel of the delay. Dly: Polyphonic event output, which is set to the current delay time. This output always produces an event whenever a new sound particle is made.

Grain Cloud Delay

Delay

The Grain Cloud Delay module is similar to the Grain Cloud module (Samplers section) except that instead of operating on audio files loaded into its memory, it processes a constantly changing audio buffer that is fed by the module's bottom input port (labeled "In").


ENGLISH

• • • • •

• Trig: Event input for triggering the next grain. Values >0 trigger the next grain. Values =-1 suppress the next grain (see the Dist input). • Frz: Event input to freeze the audio buffer. Values >0 freeze the buffer. • P: Audio input for logarithmic control of pitch-shift in semitones. The pitch is independent of the speed of traversal. Typical range -20 to 20. • D/F: Audio input setting the direction (if the P input is wired) or linear frequency (if the P input is not wired) of grain playback. The audio buffer plays at the original pitch when F=1, in reverse direction when F=-1. Typical range -4 to 4. Default 1. • PJ: Audio input for pitch jitter (in semitones). Typical range 0 to 3. • PS: Audio input for logarithmic pitch shift of current grain in semitones. Typical range -3 to 3. • Dly: Audio input for setting the delay time in ms. Allowed range 0 to buffer length. • DlJ: Audio input for delay-time jitter in ms. Allowed range 0 to buffer length. • Len: Audio input for the grain length in ms. Typical range 10 to 100. Default 30. • LnJ: Audio input for grain-length jitter in ms. Typical range 10 to 100. Default 0. • Att: Audio input for setting individual grain playback attack time. Range 0 to 1. Default 0.2. • Dec: Audio input for setting individual grain playback decay time. Range 0 to 1. Default 0.2. • Dist: Audio input for setting the delta-time between grains in ms. Grains are automatically triggered at this rate. Typical range 5 to 100. Default 20. • DisJ: Audio input for delta-time jitter in ms. Typical range 5 to 100. Default 20. • Pan: Audio input for setting the pan position in the stereo field. Range -1 to 1. • PnJ: Audio input for setting the pan jitter. Range 0 to 1.


Unit Delay

Delay

Delays audio signal by one sample period (1/sample rate). Every structure that uses some kind of must have a unit delay in the loop. If no explicit unit delay is there, the program will insert an invisible unit delay somewhere in the loop. • In: Input for the signal to be delayed by one sample period. • Out: Output for the delayed signal.

Audio Modifier Reaktor's audio processing modules provide various forms of distortion (shaping, clipping, and mirroring, for example). There are also slew limiter, peak detector, sample and hold, and frequency divider modules.

Saturator

Audio Modifier


ENGLISH

• A: Audio input for amplitude control. Typical range 0 to 1. Default 1. • In: Audio input for the audio signal to be delayed. • L: Polyphonic left channel audio output. • R: Polyphonic right channel audio output. • Dly: Polyphonic delay time output at every grain start. • Gtr: Polyphonic Grain Trigger: 1 when a new grain starts, 0 when it stops.

Distortion with a smoothly rounded input-output curve for soft transition to saturation. The output value is limited to ±2 (reached for input values bigger than ±4). Very small input values are not changed. • In: Audio input for signal to be saturated • Out: Audio output for saturated signal

Saturator 2

Audio Modifier

Saturator 2 is a fourth-order asymmetric parabolic saturator and offers control over the saturation curve. • LA: Level asymmetry. At LA = 0 the saturation levels for positive and negative signals are equal. For LA > 0 the positive level is reduced. At LA = 1 it becomes zero. For LA < 0 the negative level is reduced. At LA = -1 it becomes zero. • KH: Knee hardness. At KH = 0 the saturation rises as soft as possible. The full range between zero and the saturation level is used for the rounded curve. With growing values of KH the curve range is reduced to (1 - KH) of the saturation level. At KH = 1 the signal is clipped hard at the saturation level. • KHA: Knee hardness asymmetry. At KHA = 0 the knee hardness for positive and for negative signals is the same. For KHA > 0 the knee hardness for positive signals is reduced. At KHA = 1 it becomes zero. For KHA < 0 the knee hardness for negative signals is reduced. At KHA = -1 it becomes zero. • Offs: This input adds an offset to the input signal and shifts it relative to the saturator curve. For a zero signal the offset is fully compensated at the output. • In: Input for the signal to be distorted. • Out: Output for the distorted signal. Reaktor English – 340

Audio Modifier

Distortion with a hard clipping and adjustable upper and lower limit. When the input signal exceeds the upper limit it is clipped to the limit value, similarly for the lower limit. Signal values between the limits are ed unchanged. • • • •

Max: Audio input for controlling the upper limit of the signal Min: Audio input for controlling the lower limit of the signal In: Audio input for signal to be clipped Out: Audio output for clipped signal

Mod. Clipper

Audio Modifier

Distortion with a hard clipping and variable limit. When the absolute value of the input signal exceeds limit it is clipped to that value. Smaller signal values are ed unchanged. • M: Audio input for controlling the limit on the absolute value of the signal • In: Audio input for signal to be clipped • Out: Audio output for clipped signal


ENGLISH

Clipper

Mirror 1 Level

Audio Modifier

Distortion by signal value mirroring with adjustable mirror level. Signal values above the mirror level are “reflected” at the level to end up smaller. Signal values below the mirror level are ed unchanged. • Max: Audio input for controlling the mirror level • In: Audio input for signal to be modified • Out: Audio output for the modified signal

Mirror 2 Levels

Audio Modifier

Distortion by double signal value mirroring with adjustable mirror levels. Signal values above the upper mirror level are “reflected” at the level to end up smaller, those below the lower mirror level are ”reflected” at that level to end up larger. Signal values in the middle are ed unchanged. • • • •

Max: Audio input for controlling the upper mirror level Min: Audio input for controlling the lower mirror level In: Audio input for signal to be modified Out: Audio output for the modified signal


Audio Modifier

Chopper Modulator that switches amplification of the input signal between a variable factor and one. When the modulation signal is positive, the input signal is multiplied by the value X. When the modulation signal is negative, the input is unchanged. • M: Audio input for the modulation signal (only the sign is relevant). • X: Audio input for controlling the amplification factor used when M is positive. • In: Audio input for the signal to be chopped. • Out: Audio output for the chopped signal

Shaper 1 BP

Audio Modifier

Signal shaper with piecewise linear input/output curve and one breakpoint. The slope of the curve above the breakpoint can be adjusted. Signal values below the breakpoint are ed unchanged. • Bp: Audio input for controlling the level of the breakpoint • Sl: Audio input for controlling the slope of the upper part of the input/output curve (1 = no change in signal). • In: Audio input for the signal to be shaped. • Out: Audio output for the shaped signal.


ENGLISH

Chopper

Shaper 2 BP

Audio Modifier

Signal shaper with piecewise linear input/output curve and two breakpoint. The slope of the curve above the upper and below the lower breakpoint can be adjusted. Signal values between the breakpoints are ed unchanged. • Bp1: Audio input for controlling the level of the upper breakpoint • Sl1: Audio input for controlling the slope of the upper part of the input/output curve (1 = no change in signal). • Bp2: Audio input for controlling the level of the lower breakpoint. • Sl2: Audio input for controlling the slope of the lower part of the input/output curve (1 = no change in signal). • In: Audio input for the signal to be shaped. • Out: Audio output for the shaped signal.

Shaper 3 BP


Audio Modifier

The slope of the curve above the upper breakpoint, below the lower breakpoint and between the breakpoints and zero can be adjusted. Signal values between the breakpoints are ed unchanged. • Bp1: Audio input for controlling the level of the upper breakpoint • Sl1: Audio input for controlling the slope of the upper part of the input/output curve • Bp2: Audio input for controlling the level of the lower breakpoint • Sl2: Audio input for controlling the slope of the lower part of the input/output curve (1 = no change in signal). • Sl+: Audio input for controlling the slope of the input/output curve between zero and the upper breakpoint (1 = no change in signal). • Sl-: Audio input for controlling the slope of the input/output curve between the lower breakpoint and zero (1 = no change in signal). • In: Audio input for the signal to be shaped • Out: Audio output for the shaped signal

Shaper Parabolic

Audio Modifier

Signal shaper with parabolic (2nd order polynomial) input/output curve. The linear and square parts of the signal can be adjusted ( Out = Lin In + Par In2 ).


ENGLISH

Signal shaper with piecewise linear input/output curve and three breakpoint.

• Lin: Audio input for controlling the level of the linear, undistorted part • Par: Audio input for controlling the level of the square, distorted part • In: Audio input for the signal to be shaped • Out: Audio output for the shaped signal

Shaper Cubic

Audio Modifier

Signal shaper with cubic parabolic (3rd order polynomial) input/ output curve. The linear, square and cubic parts of the signal can be adjusted ( Out = Lin In + Par In2 + Cub In3 ). • Lin: Audio input for controlling the level of the linear, undistorted part • Par: Audio input for controlling the level of the square, distorted part • Cub: Audio input for controlling the level of the cubic, distorted part • In: Audio input for the signal to be shaped • Out: Audio output for the shaped signal

Slew Limiter


Audio Modifier

• Up: Audio input for controlling the maximum slew rate for rising signals. Value in units of 1/sec • Dn: Audio input for controlling the maximum slew rate for falling signals. Value in units of 1/sec • In: Audio input for signal to be slew rate limited • Out: Audio output for slew rate limited signal

Peak Detector

Audio Modifier

Detector for the peak amplitude. The input signal is rectified and smoothed with an adjustable release time. The attack time is zero. The result of the Peak Detector’s action is that the output value follows the amplitude envelope of the input – use this module as an envelope follower. • Rel: Logarithmic event input for controlling the release time. 0 = 1 ms, 20 = 10 ms, 40 = 100 ms (i.e. in dB1ms). • In: Audio input for signal to be detected • Out: Audio output for the signal

Sample & Hold

Audio Modifier


ENGLISH

Slew rate limiter with separately adjustable maximum rate for rising and falling signals. The output signal tracks the input signal, but for fast movement and jumps at the input, the output follows with a limited rate of change (ramp) until its value reaches that of the input signal.

Sample & Hold with clock input. When the clock signal rises above zero, the current input value is ed to the output and held there until the next clock pulse. The result is a step waveform at the output. • C: Audio input for the clock signal. The input is sampled on a rising edge here. • In: Audio input for the signal to be sampled • Out: Audio output for the sampled signal

Frequency Divider

Audio Modifier

Frequency divider (pulse sub-oscillator) with independently controllable high and low level duration. A pulse wave is generated by counting the zero crossings of the input signal. The frequency of the output waveform will be a fraction of the frequency of the input waveform. The frequency ratio can be adjusted ( fout = 2 fin / ( C+ + C– ) ). An asymmetric waveform results when C+ and C- have different values. • C+: Audio input for controlling the number of zero crossings of the input signal during the high phase of the output. • C-: Audio input for controlling the number of zero crossings of the input signal during the low phase of the output. • A: Audio input for controlling the amplitude. The output signal moves between +A and –A. • In: Audio input for the signal to be frequency-divided • Out: Audio output for the frequency divided signal.


Oscillator

Holds a table of data values. The table can be read out as an audio signal, audio can be stored in the table and the table’s content can be displayed and edited graphically. The table can be 1-dimensional (a row of values) adressed by X, or 2-dimensional (a matrix of rows and columns, or a set of independent rows) adressed by X and Y. The value at the output is taken from the table by reading at the position given by the inputs RX and RY. A signal at the module’s In-port are stored in individual cells of the table according to the write position given by the inputs WX and WY. X is the horizontal position from left to right and Y is the vertical position from top to bottom. The count always starts at 0 for the first element. The module’s display can show all the data or a limited region of it. Many options in the properties allow customizing the behaviour. For full details on properties, menus and keyboard shortcuts, please see the section Table Modules on page 165. • RX: Audio input for the X-position of a table cell from which the data is read. • RY: Audio input for the Y-position of a table cell from which the data is read. This is used in 2D-mode or for addressing the row number if more than one row exists.


ENGLISH

A u d i o Ta b l e

• WX: Audio input for the X-position of a table cell in which the data is written. • WY: Audio input for the Y-position of a table cell in which the data is written. • W: Audio input for activating table write operation. • In: Audio input for the signal to be written into the table. When the value at W is bigger than 0, the value at In is written into the table at the position given by WX and WY. • XO: Event input for the horizontal offset of the displayed data region. XO controls the data position that appears in the display (according to View Alignment). The value of XO is in the units specified in properties. • XR: Event input for the horizontal range of the displayed data region. XR controls how many units of data fit in the display, i.e. it lets you zoom into the data. • YO: Event input for the vertical offset of the displayed data region. YO controls the data position that appears in the display (according to View Alignment). The value of YO is in the units specified in properties. • YR: Event input for the vertical range of the displayed data region in 2D-mode. YR controls how many units of data fit in the display, i.e. it lets you zoom into the data. • Out: Audio output for signal read from the table at the position controlled by the RX and RY inputs. • DX: Event output for the size of the horizontal table rows in units. • DY: Event output for the size of the vertical table columns in units.

Event Processing Event modules are used for counting, for logical operations, for splitting and merging control signals, and for timing. You'll also find modules here for shaping and randomizing control signals. Finally there's a full-featured lookup table module-the control equivalent of the audio table in the oscillator section.


Event Processing

Accumulator (sum) for event values. The value of each event at the input is added to the total value stored in the module. The value of the updated sum is sent as an event at the output. • In: Input for the events to be accumulated. • Set: Event input for (re)setting the internal sum. The value of an event at this input determines the new value of the internal sum. • Out: Event output for the accumulated total value.

Counter

Event Processing

Counter controlled by events. A positive event at the appropriate input increases or decreases the output value by one. • Up: Input for counting up. Only events with a positive, nonzero value have an effect here, increasing the output by one. • Dwn: Input for counting down. Only events with a positive, non-zero value have an effect here, decreasing the output by one. • Set: Event input for (re)setting the counter value. The value of an event at this input determines the new value of the counter. • Out: Event output for the counter value.


ENGLISH

Accumulator

Randomizer

Event Processing

Event randomizer with adjustable spread. The arriving events are modified with a random positive or negative offset in the given range (Out = In + X, where –Rng ≤ X ≤ Rng). • Rng: Audio input for controlling the range of the random signal modification • In: Event input for signal to be randomized • Out: Event output for randomized signal

Frequency Divider

Event Processing

The frequency of the output events is the frequency of the input events divided by a number controlled by the input N. The output signal returns to zero after a certain number of input events, depending on the pulse length which is set with the input W. • N: Control input for the number of input events per output period.(N < 2 : no division, 2 ... 2.99 : division by 2, 3 ... 3.99 : division by 3, etc.). • W: Control input for the pulse width of the output signal. Range of values: 0 ... 1 (0% ... 100%). W = 0 : the gate signal at Out returns to zero already at the next event at In, W = 0.5 : the gate signal at Out returns to zero after half the period, W = 1 : the gate signal at Out stays on all the time.


Ctrl. Shaper 1 BP

Event Processing

Event signal shaper with piecewise linear input/output curve and one fixed breakpoint. The output produced by linear interpolation between the given points. • • • • •

100: Output value at In = 1 (100%). 50: Output value at the breakpoint at In = 0.5 (50%). 0: Output value at In = 0 (0%). In: Event input for the signal to be shaped Out: Event output for the shaped signal

Ctrl. Shaper 2 BP

Event Processing


ENGLISH

• In: Input for the event (clock) signal to be frequency-divided (e.g. MIDI Clock pulses). Only events with a positive, non-zero value have an effect here. • Rst: Event input for resetting the internal counter in order to force a synchronous start (connect to MIDI Start signal if using the Event Freq. Divider for MIDI synchronization). Requires an event with positive non-zero value. • Out: Event output for the frequency divided signal

Event signal shaper with piecewise linear input/output curve and two fixed breakpoints. The output produced by linear interpolation between the given points. • • • • • •

100: Output value at In = 1 (100%). 66: Output value at the upper breakpoint at In = 0.66 (66%). 33: Output value at the lower breakpoint at In = 0.33 (33%). 0: Output value at In = 0 (0%). In: Event input for the signal to be shaped Out: Event output for the shaped signal

Ctrl. Shaper 3 BP

Event Processing

Event signal shaper with piecewise linear input/output curve and three fixed breakpoints. The output produced by linear interpolation between the given points. • • • • • • •

100: Output value at In = 1 (100%). 75: Output value at the upper breakpoint at In = 0.75 (75%). 50: Output value at the middle breakpoint at In = 0.5 (50%). 25: Output value at the lower breakpoint at In = 0.25 (25%). 0: Output value at In = 0 (0%). In: Event input for the signal to be shaped Out: Event output for the shaped signal


Event Processing

Logic gate for event signals. The output is the logic AND of the two input values, i.e. the output is 1 when both inputs are positive, otherwise the output is 0. The inputs treat values of zero and below as the logic False state, values above zero are logic True.

Logic OR

Event Processing

Logic gate for event signals. The output is the logic OR of the two input values, i.e. the output is 1 when one or both inputs are positive, otherwise the output is 0. The inputs treat values of zero and below as the logic False state, values above zero are logic True.

Logic EXOR

Event Processing

Logic gate for event signals. The output is the logic EXOR of the two input values, i.e. the output is 1 when one but not both inputs are positive, otherwise the output is 0. So in effect one input inverts the logic value of the other input.


ENGLISH

Logic AND

The inputs treat values of zero and below as the logic False state, values above zero are logic True.

Logic NOT

Event Processing

Logic gate for event signals. The output is the logic complement of the input value, i.e. the output is 0 when the input is positive, otherwise the output is 1. The input treats values of zero and below as the logic False state, values above zero are logic True.

Order

Event Processing

Event-Order. An event arriving at the input is transmitted with the same value on all the outputs, but in a well defined order: first it goes to 1, then to 2 and finally to 3. The event travels through ALL the modules in the chain connected to 1, before going to the first module connected to 2 etc. • • • •

In: Input for events to be re-transmitted in a defined order. 1: An event is transmitted on this output first. 2: An event is transmitted on this output second. 3: An event is transmitted on this output third.


Event Processing

An event arriving at the Trg input is ed to the output and triggers a series of N additional output events. Each subsequent event has the value of its predecessor incremented by the value at the Inc input. All events are sent before the next audio sample is processed. This module can be used where an iterative event calculation is needed.It helps to avoid the construction of event loops which can lead to unstable operation of Reaktor. • Trg: Trigger input. An event at this input triggers N+1 events at the output. • Inc: Increment for the value of each subsequent event. • N: Number of additional output events. The value has to be greater than or equal to the integer number (decimal places will be cut).

Separator

Event Processing

All received events are compared to the threshold value and are sent to either one or the other output. One use for the separator would be to convert a gate signal to a trigger signal by filtering out the zero events (Thld = 0, Hi = trigger output). • Thld: Audio control input for the threshold value • In: Input for the events to be separated and sent to the two outputs.


ENGLISH

Iteration

• Hi: Output for those events whose value is greater than the threshold level. • Lo: Output for those events whose value is less than or equal to the threshold level.

Va l u e

Event Processing

Value-changer for events. Events arriving at the input In have their value replaced with the current value at the Val input, and are then sent with the new value from the output. This module can also be used as an event-controlled sample&hold module. • In: Input for events to be changed in value. • Val: Input for specifying the new value for the events. • Out: Event output for the value-changed events.

Merge

Event Processing

Merger for event signals. When more than one wire is connected at the input, the output value is that of the last received input event, irrespective of which wire it came from. The module has a dynamic in-port managment. The number of inports can be defined with Min Num Port Groups on the Function page of the Properties. Unlike in previous REAKTOR versions subsequent events with the same value will not be filtered out anymore. Therefore use the Step Filter module.


Event Processing

An event is only ed if the input value is larger/smaller than the previous input value +/- tolerance. • In: Input for events to be filtered. • Tol: Input for the tolerance level. • Out: Event output.

Router M->1

Event Processing

Router multiple to one. The events at the selected input will be let through other events will be filtered. The inputs are selected by the Pos input. When Wrap mode is selected in the Properties, Pos wraps around so that Max+1 is the same as 0, Max+2 is 1 etc.. The module has a dynamic in-port managment. The number of inports can be defined with Min Num Port Groups on the Function page of the Properties. • Pos: Input for selecting the thru input. • In: Signal input. • Out: Output for the selected input signal.

Router 1,2

Event Processing


ENGLISH

Step Filter

On/off switch for one or toggle switch for two event signals, with control input. The last event ed through is held at the output even while off. The module has a dynamic in-port managment. The number of inports can be defined with Min Num Port Groups on the Function page of the Properties between 1 and 2. • Ctrl: Hybrid input for control of switching. While the value is greater than zero all input events are ed to the output. • In: Event input for the signal to be switched • Out: Event output for the switched signal. While Ctrl is smaller that zero, the last value is held at the output.

Router 1->M

Event Processing

Router one to multiple. The events at the input will be let through to the selected output. The output is selected by the Pos input. When Wrap mode is selected in the Properties, Pos wraps around so that Max+1 is the same as 0, Max+2 is 1 etc.. The module has a dynamic out-port managment. The number of out-ports can be defined with Min Num Port Groups on the Function page of the Properties. • Pos: Input for selecting the thru output. • In: Signal input. • Out: Output for the input signal.

Ti m e r


Event Processing

• In: Polyphonic input for the events whose distance in time is to be measured. • F: Polyphonic event output for the frequency of input events in Hz. • T: Polyphonic event output for the time between events in milliseconds.

Hold

Event Processing

Hold envelope. When the module is triggered by a positive event at the G input, the event’s value is held as the output value until the hold time has ed, after which the output jumps back to zero. The module can be triggered at any time, including retriggering during the hold time. • G: Event input for triggering the envelope. Only events with a positive values (not zero) have an effect here. • H: Input for controlling the hold time in milliseconds. • Out: Event output for the envelope signal.


ENGLISH

The elapsed time between the two last received events is measured and output as an event. Also, the frequency whose period of oscillation is equal to the measured duration is calculated and output.

E v e n t Ta b l e

Event Processing

Holds a table of data values. Event values can be read from the table, event values can be stored in the table and the table’s content can be displayed and edited graphically . The table can be 1-dimensional (a row of values) adressed by X, or 2-dimensional (a matrix of rows and columns, or a set of independent rows) adressed by X and Y. The value at the output is taken from the table by reading at the position given by the inputs RX and RY. Values arriving at the module’s W input are stored in individual cells of the table according to the write position given by the inputs WX and WY. X is the horizontal position from left to right and Y is the vertical position from top to bottom. The count always starts at 0 for the first element. The module’s display can show all the data or a limited region of it. Many options in the properties allow customizing the behaviour. For full details on properties, menus and keyboard shortcuts, please see the section Table Modules on page 165. • RX: Audio input for the X-position of a table cell from which the data is read. • RY: Audio input for the Y-position of a table cell from which the data is read. This is used in 2D-mode or for addressing the row number if more than one row exists.



ENGLISH

• R: Event input for triggering table read operation at the position given by RX and RY. Each event here produces an event at the Out output. • WX: Audio input for the X-position of a table cell in which the data is written. • WY: Audio input for the Y-position of a table cell in which the data is written. • In: Event input for writing into the table at the position given by WX and WY. The value of the data written into the table cell is the value of the event arriving at In. • XO: Event input for the horizontal offset of the displayed data region. XO controls the data position that appears in the display (according to View Alignment). The value of XO is in the units specified in properties. • XR: Event input for the horizontal range of the displayed data region. XR controls how many units of data fit in the display, i.e. it lets you zoom into the data. • YO: Event input for the vertical offset of the displayed data region. YO controls the data position that appears in the display (according to View Alignment). The value of YO is in the units specified in properties. • YR: Event input for the vertical range of the displayed data region in 2D-mode. YR controls how many units of data fit in the display, i.e. it lets you zoom into the data. • Out: Event output for data from the cell controlled by the RX, RY and R inputs. • DX: Event output for the size of the horizontal table rows in units. • DY: Event output for the size of the vertical table columns in units.

A uxiliary If you can't find it anywhere else, it's probably here. First you'll find tape decks for recording and playing back audio files. Then there are modules for managing polyphonic voices, for converting audio signals to control signals, and for reporting the status of various Reaktor processes such as tuning and tempo.

Ta p e d e c k 1 - C h

Auxiliary

1-channel Tapedeck module for recording and playback of audio signals. Audio files can be read from and write to either memory or harddisk depending on the properties setting. In harddisk mode the files are written into the folder you specify in the Reaktor preferences. Reaktor does sample rate converting on the fly and writes the file in the sample rate which is currently used by your audio system. In memory mode you can display the waveform of a loaded audio file in the by ticking the checkbox Picture in the Appearance tab in the properties. The whole waveform for the audio file will fit automatically to the Size of the picture which you enter in the Appearance tab.


By ticking the checkbox Keep audio only in memory in the properties the tapedeck module starts working in memory mode and the Memory section with the Save, Save as... and Reload buttons becomes active. The maximum recording time is set with Max Recording Size (sec) in seconds. How big this value can be depends on the amount of available RAM storage in the computer. At a sample rate of 44.1 kHz you need 86 kB of RAM for each second of buffer length, for one minute you need 5 MB. If the buffer memory for the Tapedeck has been chosen too large, virtual memory swaps by the operating system can cause significant hard disk activity during recording. This can prevent REAKTOR from processing audio smoothly. Audio files can be imported for playback with the button Select File... in the properties of the Tapedeck module. A file that is already imported can be loaded again by clicking on the Reload button, e.g. if it has been changed using a sample editor. When importing an audio file, the length of the tapedeck’s memory buffer is adjusted to match the loaded data. If you later want to make a recording which is longer than this, you will first need to set a bigger value under Max Recording Size (sec) in the module's properties dialog window. The loaded file will automatically set to the current sample rate of Reaktor. Note: Be aware that Reaktor converts all audio files stored in the tapedecks in memory mode to the new sample rate whenever Reaktor switches to a new rate. So you should avoid changing the sample rate if your ensemble contains memory based audio samples. If the audio file is stored on harddisk you can use the Relaod button after changing the sample rate to import the file again. A recording can be exported using Save in the properties. If the file has already been exported, you will be asked if you want to overwrite the file, e.g. if you have made a new recording in the Tapedeck. With Save as... you can store the file under a new name.


ENGLISH

Memory mode

Harddisk mode By ticking the checkbox Stream audio from/to harddisk in the properties the tapedeck module starts working in harddisk mode. Audio files are written to and read from harddisk directly. You can define an audio file for playback using the Select File... button. With the New File button you create a new file named "untitled" ( if a file with the name is already existing in your Audio folder you have specified in the Reaktor Preferences the new name will contain an additional mumber). You can edit this name directly in the name field inside the Properties. If Value is activated in the properties, a display for the file name will appear in the . By opening the context menu on this box files can also be imported and exported. • Rec: Event input for switching recording mode on and off, e.g. with a gate signal. Start of recording on an event with a positive value. End of recording on an event with negative or zero value or when the maximum recording time is reached. • Play: Event input for switching playback mode on and off, e.g. with a gate signal. Start of playback on an event with a positive value, playback stops on an event with negative or zero value. • Lp: Event input for switching loop playback mode on and off. When this input receives a positive value, playback starts from the beginning when the end of playback is reached. The result is an infinite loop while playback is switched on. • In: Monophonic audio input for the signal to be recorded. Maximum value ± 1. To record a polyphonic signal a Voice Combiner has to be inserted. • Pse: Pause control input. A value greater than zero pauses, else continues. Typ. range: [0 ... 1]. • Pos: Input for setting playback position in ms. Typ. range: [0 ... 20000]. • Spd: Variable playback speed of the Tape. Values must be greater than 0. A value of 1 means normal speed. Typ. range: [0.8 ... 1.2]. • Out: Audio output for the playback signal or the signal being recorded. • Rec: 1 = Tapedeck is recording, else 0. Typ. range: [0 ... 1].


Ta p e d e c k 2 - C h

Auxiliary

This works just like Tapedeck 1-Ch but has two channels for recording and playback of audio signals. It imports and exports stereo files. • In L: Monophonic audio input for the signal to be recorded on the left channel. Maximum value ± 1. To record a polyphonic signal a Voice Combiner has to be inserted. • In R: Monophonic audio input for the signal to be recorded on the right channel. Maximum value ± 1. To record a polyphonic signal a Voice Combiner has to be inserted. • L: Audio output for the playback signal or the signal being recorded on the left channel. • R: Audio output for the playback signal or the signal being recorded on the right channel.


ENGLISH

• Play: 1 = Tapedeck is playing back, else 0. Typ. range: [0 ... 1]. • Wrp: An Event with value 1 is sent each time the tape is wrapping around the loop. • Pos: 1 = Tapedeck is paused, else 0. Typ. range: [0 ... 1]. • Time: current playback/recording position in ms [0 ... ]. • Lng: Length of recording in ms. Typ. range: [0 ... ].

Aud io Voice Combi ner

Auxiliary

Audio Voice Combiner. Converts a polyphonic audio signal to monophonic by summing all the voices. • In: Polyphonic audio input for the signal to be combined to mono • Out: Monophonic audio output for the combined signal

E v e n t V. C . A l l

Auxiliary

Event Voice Combiner. Converts a polyphonic event signal to monophonic by sending the events of all the voices to the same monophonic voice. • In: Polyphonic event input for the signal to be combined to mono • Out: Monophonic event output for the combined signal

E v e n t V. C . M a x

Auxiliary

Event Voice Combiner with maximum value selection. Converts a polyphonic event signal to monophonic by finding the voice with the highest current value and sending it to the mono output. A polyphonic gate signal input is necessary to recognize active voices.


E v e n t V. C . M i n

Auxiliary

Event Voice Combiner with minimum value selection. Converts a polyphonic event signal to monophonic by finding the voice with the lowest current value and sending it to the mono output. A polyphonic gate signal input is necessary to recognize active voices. • In: Polyphonic event input for the signal whose minimum value is to be output • G: Polyphonic event input for the gate signal of the polyphonic instrument • Out: Monophonic event output for the minimum value signal

A to E

Auxiliary

Audio to event converter. The audio signal is sampled using the Control Rate specified in the Settings menu and the values are sent out as a stream of events.


ENGLISH

• In: Polyphonic event input for the signal whose maximum value is to be output • G: Polyphonic event input for the gate signal of the polyphonic instrument • Out: Monophonic event output for the maximum value signal

A t o E ( Tr i g )

Auxiliary

Audio to event converter with trigger input. When the trigger input signal goes from zero to positive values (rising edge) the audio signal is sampled and output as an event. • T: Audio input for triggering the conversion. Triggers on a rising edge • In: Audio input for signal to be sampled and output as events • Out: Event output for the sampled signal

A to E (Perm)

Auxiliary

Audio to event converter with permanent conversion at regular intervals and adjustable sampling frequency. The audio signal is sampled with the given frequency and output as a stream of events. When running this module at a high frequency (above 1000 Hz) the U-Load caused by event processing can rise significantly. Typically F = 200 Hz is sufficient. • F: Audio input for controlling the sample frequency. Value in Hz • In: Audio input for signal to be sampled and output as events • Out: Event output for the sampled signal


Auxiliary

Audio to gate event converter with gate amplitude input. When the trigger signal goes from zero to positive values (rising edge), the gate signal is switched on with an amplitude of the current value of the amplitude input. When the trigger input returns to zero or negative values (falling edge) the gate signal is switched off. • T: Audio input for triggering the gate signal • A: Audio input for controlling the amplitude of the gate events • Out: Event output for the gate event signal.

To V o i c e

Auxiliary

Monophonic input signals are transmitted to one voice of the polyphonic output signal. The voice number is given by the current value of the V input. Signals are discarded when the value at V is not a valid voice number. This is useful for making polyphonic sequencers, for example. • V: Monophonic input for defining the number of the voice to which an event is to be sent. Typ. Range: [ 1 ... 10 ]. • In: Monophonic hybrid input for signals to be sent to one of the polyphonic voices. • Out: Polyphonic hybrid output. The input signals are transmitted (with their original value) on one voice of this polyphonic signal.


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A to Gate

F r o m Vo i c e

Auxiliary

One voice is selected from the polyphonic input signal by the current value of the V input. Only events on the selected voice are transmitted to the monophonic output. All events on other voices are discarded. • V: Monophonic input for defining the number of the voice from which events are allowed through. Typ. Range: [ 1 ... 10 ]. • In: Hybrid input. One voice of this polyphonic signal is sent to the output. • Out: Monophonic hybrid output. The input signal belonging to one voice are transmitted (with their original value) on this monophonic output.

Audio Smoother

Auxiliary

Smoother for monophonic event signals with audio output. Typically, a smoother is connected after a fader or button to get smooth transitions. The jumps in the value of the input event signal are smoothed to ramps. The Transition Time (in milliseconds) can be adjusted in the properties. Exactly after this time has elapsed, the output reaches the same value as the input, assuming that no further jumps occurred at the input. The larger the Transition Time, the stronger the smoothing effect. • In: Mono event input for the signal to be smoothed. • Out: Mono audio output for the smoothed signal.


Auxiliary

Smoother for monophonic event signals. Typically, a smoother is connected after a fader or button to get smooth transitions. The jumps in the value of the input event signal are smoothed to ramps. The Transition Time (in milliseconds) can be adjusted in the properties. Exactly after this time has elapsed, the output reaches the same value as the input, assuming that no further jumps occurred at the input. The larger the Transition Time, the stronger the smoothing effect. During the transition, events are output with the rate selected as the Control Rate in the Settings menu. The higher the rate, the higher the resolution of the smoothing transition. • In: Mono event input for the signal to be smoothed. • Out: Mono event output for the smoothed signal.

M a s t e r Tu n e / L e v e l

Auxiliary

Controls the overall level and tuning. • Tun: Control input for the master tuning. Scale: 1 semitone per unit. At 0.0 the note a3 is tuned to 440 Hz Typ. Range: [ 1 ... 1 ]. • Lvl: Control input for the master level at the output converters. Scale: 1 dB per unit 0.0 = unity gain = 0.0 dB Typ. Range: [ 60 ... 0 ]. • Tun: Output for for the master tuning. • Lvl: Output for for the master level.


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Event Smoother

Te m p o I n f o

Auxiliary

Source for the current Tempo measured in Beats per Second. To get the BPM value, multiply by 60. • Out: Event output for the current tempo in Beats per Second (Hz).

Vo i c e I n f o

Auxiliary

Voice Info module. • V: Polyphonic output for the ID Number of each Voice [ 1, 2, 3 ... Voices ]. • Vcs: Output for the current Number of Voices in the instrument. • Min: Output for the Min Unison Voices setting of the Instrument. This is the minimum number of voices assigned to one key for playing in unison. • Max: Output for the Max Unison Voices setting of the Instrument. This is the maximum number of voices assigned to one key for playing in unison.

Tu n i n g I n f o


Auxiliary

• Tun: Control for tuning the instrument in semitones. • Spr: Control for the amount of unison spread in semitones. • Shft: Control for the shifting incoming MIDI notes in semitones. • Tun: Output for the tuning of the instrument in semitones. • Spr: Output for the tuning spread amount in semitones. • Shft: Output for the amount of the note shift of incoming notes in semitones.

System Info

Auxiliary

Source for information about the system: Sampling rate (in samples/sec), control rate (in Hz) and U load (in %). • SR: Output for the current sampling rate in samples per second. • CR: Output for the current control rate in Hz. • DClk: Output for the current display rate in frames per second. Event is sent just before each display update. • U: Output for the current U load in percentages.

Note Range Info

Auxiliary

Note Range Info module.


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Control for and information about the settings of the instrument's tuning parameters.

• Upr: Input for selecting the upper limit for received midi notes. • Lwr: Input for selecting the lower limit for received midi notes. • Upr: Output for the upper note limit for received midi notes. • Lwr: Output for the lower note limit for received midi notes.

MIDI Channel Info

Auxiliary

Midi Channel Info module. • ICh: Input for selecting the input midi channel. • OCh: Input for selecting the output midi channel. • ICh: Output for the current input midi channel of the instrument. • OCh: Output for the current output midi channel of the instrument.

Snapshot


Auxiliary

Appearance The representation of this module changes depending on the chosen style on the properties’ Appearance page. The following styles are available: • Button: Each module in-port creates a button. All buttons will be arrayed vertically in the instrument . The currently activated button will be displayed in the Indicator color of the Instrument. • Menu: Each module in-port creates a new entry in a drop down list. • Text : Each module in-port creates a new entry in a list which displays multiple entries at the same time. If you have created more entries than fit into the text display specified by the Size X and Size Y fields on the Appearance tab, you will get scrollbars in the . • Spin: Each module in-port creates a new entry in a list. You can switch through the list using a + and a - button at the right side of the list entry display. The Size X and Size Y fields are controlling the display size of the control element in the .

Ports • Snp: Audio input for selecting the snapshot to be recalled or stored. Range: 1 ... 128. • Bnk: Audio input for selecting the snapshot bank. Range: 1 ... 16. • Recl: A positive event recalls the snapshot selected by the Snp and Bnk inputs. • St: A positive event stores the snapshot to a position selected by the Snp and Bnk inputs.


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The Snapshot module allows you to change snapshots and morph between snapshots from within Reaktor. The module has a builtin list processor that works exactly like the List module (see the modules section).

• A: A positive event takes the current values from the Snp and Bnk inputs to select a snapshot for the Morph position A. • B: A positive event takes the current values from the Snp and Bnk inputs to select a snapshot for the Morph position B. • Mrph: This input controls the morph position between the snapshots loaded for A and B. Value <= 0.0: A, Value 0.0 -1.0 : morphing between A and B. Value >= 1.0: B. • Sw: Events with negative and zero values set the switches/buttons to their position in snapshot A. Positive events set the switches/buttons to their position in snapshot B. • MT: Event input for the morph time in ms. • Rnd: A positive event triggers the snapshot randomize function. • Amt: Input for the randomization amount. Range: 0.0 ... 1.0 (1.0 = 100%) • Mrg: A positive event triggers the random merge function. • Snp: Output for the index of the current snapshot. An event is sent every time a snapshot is recalled or stored. • Bnk: Output for the index of the current snapshot bank. An event is sent every time a snapshot is recalled or stored. • Recl: An event with value = 1.0 is sent when a snapshot was recalled. • St: An event with value = 1.0 is sent when a snapshot was stored. • A: Output for the index of the snapshot of morph position A. An event is sent when a snapshot was recalled or selected for position A. • B: Output for the index of the snapshot of morph position B. An event is sent when a snapshot was recalled or selected for position B. • Mrph: Output for the morph position. Value = 0.0 : A, Value 0.0 - 1.0: morphing between A and B, Value = 1.0: B. • Sw: Output for Switch Position A/B. Value = 0.0 if the switches/buttons are set to their position in snapshot A. Value = 1.0 if the switches/buttons are set to their position in snapshot B. • MT: Output for the morph time in ms.


Set Random

Auxiliary

Sets the random seed for the pseudo-random number generator used in all Event Randomize, Slow Random and Geiger modules. Only modules in the same instrument are affected. Each unique value starts a different pseudo-random sequence.

Unison Spread

Auxiliary

Polyphonic event source for a constant value used to spread out parameters for unison voices. In unison mode, the different voices that play the same note need to have their parameters spread out a little to make them slightly different so that their sum results in a sound that is fatter, and not just louder. For note pitch this happens automatically and is controlled by the Unison Spread parameter in the instrument properties. Other parameters can be spread out by adding an offset generated by the Unison Spread module. The value at the module’s input determines the amount of spread. At the output you get a value which is different for each of the voices playing the same note. The value only changes once a new note is played.


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• Rnd: An event (with value = 1.0) is sent when the randomize function was triggered. • Amt: Output for the randomization amount. Range: 0.0 ... 1.0 (1.0 = 100%) • Mrg: An event (with value = 1.0) is sent when the random merge function was triggered.

S n a p Va l u e

Auxiliary

This module stores the input value with a snapshot and outputs the value, when the snapshot is recalled. • In: Input for the value to be stored in a snapshot. The input signal is sampled in the moment when the snapshot is stored. If connected to an event source, input events are ed to the output. • Out: Output for the value stored in the snapshot that has been recalled most recently.

Terminal Terminals are used to route audio and control signals in and out of Reaktor's Instrument and Macro sub-structures. You can create terminals automatically when you drag a cable to an Instrument or Macro within the structure window while holding down the Ctrl key.

In Port

Te r m i n a l

Terminal for audio and event signals to create in-ports for Instruments and Macros.

Out Port

Te r m i n a l

Terminal for audio and event signals to create out-ports for Instruments and Macros.


Te r m i n a l

Send terminal for audio and event signals to create a cableless connection within an Instrument. Each inserted Send module creates an entry in a list of available signal sources in the Properties of a Receive module.

Receive

Te r m i n a l

Receive terminal for audio and event signals to create a cableless connection within an Instrument.

Properties - Function page For each Send module inserted in the same Instrument a list entry is created. The list entry name will be created according to the label of the Send module. The Up and Down buttons above the list serves for moving a selected entry up or down. The list contains the following columns which can be sorted by a click on the appropriate header entry: • #: Indicates the list position of the Send module. The default value refers to this number. • Label: The name of the Send module. If you rename a Send module the label in the list will be updated. • State: Indicates if a connection to this send module is possible. Only establish a connection if this field displays OK. • Use: Click on this field to set a connection between to the appropriate Send module. An existing connection is indicated by a cross.


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Send

The Mouse Resolution only applies to the control if you choose the Spin style on the Appearance page, where you can click on the control entry and drag up or down to change the entry. The Default value is used whenever an initialization of the control happens. In this case the entry # in the list will be selected according to the Default value entered here.

Appearance The representation of this module changes depending on the chosen style on the properties’ Appearance page. The following styles are available: • Button: Each module in-port creates a button. All buttons will be arrayed vertically in the instrument . The currently activated button will be displayed in the Indicator color of the Instrument. • Menu: Each module in-port creates a new entry in a drop down list. • Text : Each module in-port creates a new entry in a list which displays multiple entries at the same time. If you have created more entries than fit into the text display specified by the Size X and Size Y fields on the Appearance tab, you will get scrollbars in the . • Spin: Each module in-port creates a new entry in a list. You can switch through the list using a + and a - button right hand of the list entry display. The Size X and Size Y fields are controlling the display size of the control element in the .

OSC Send


Te r m i n a l

Multiple connections to the OSC Send module result in OSC messages with multiple arguments. For example, if you wire the MX and MY outputs of an XY Module to an OSC Send module, each OSC message will have both the X and Y value. You must wire multiple outputs from the OSC Receive Module to receive multiple arguments-one for each argument up to a maximum of 10. If you have created more than one in-port for the OSC Send module, always the first in-port of the module will trigger the OSC message. OSC messages can also be sent and received directly from some Reaktor Modules, for example controls. The send and receive settings are the same as in the OSC terminal modules. The module has a dynamic in-port managment. The number of inports can be defined with Min Num Port Groups on the Function page of the Properties.

OSC Receive

Te r m i n a l

See OSC Send module.


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OSC messages can be communicated using the OSC Send and OSC Receive modules. Both modules are dynamic - new in- or out-ports are added by wiring to blank regions in the port areas while holding down the Ctrl key (left edge if the Send and right edge of the Receive Module).


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23 Reaktor 4 Library Synthesizers Carbon

Instrument and Presets: Mike Daliot; Design: Christophe Stoll The Carbon synthesizer is a fantastic-sounding machine with over a hundred sounds from a different planet. At its heart, Carbon is a powerful subtractive synthesizer with four oscillators, copious modulation, and a filter. We even wouldn't hesitate to say that Carbon has one of the best-sounding digital filters you've heard. In fact, Carbon has a choice of eleven different filter types and designs, each painstakingly shaped for silky smoothness across the entire frequency range. Carbon's LFOs include innovative features such as the possibility to vary their frequency slightly in each voice for an organic, living sound. Each one of Carbon's breathtaking pads, cutting leads, ripping basses, and quivering atmospheres will convince you that all life is based on Carbon.

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Quick Start Even though Carbon's may show many controls, its clear layout and function grouping make operation easy. Simply step through some presets at random to get a feel for the type of sounds that Carbon can make. In the filter section, try modifying Cutoff and Resonance. You can even try choosing different filter types in the menu to the left of Cutoff to see how the different filters change the sound.

Signal Flow and Structure Carbon's sound is routed as follows: Oscillators lay

Filter Saturator Four-band EQ Chorus De-

Carbon features four oscillators plus a noise generator. The level of each oscillator is set with the large knob on top of each oscillator channel, while LFO modulation of the amplitude can be switched on or off directly underneath this knob. Each of Carbon's four oscillators offer two different modes, with each mode making a unique sound. Oscillator 1 is focused on pulse-width modulation, switchable between square or sinusoidal waveforms. Oscillator 2 concentrates on saw waves, switchable between a massive 7-saw (seven parallel saw oscillators, each slightly detuned), or a single sawtooth waveform. Oscillator 3 is based on interrupting the waveform - with a quantized sine wave being "interrupted" in amplitude to create low-fi effects, while a sawtooth oscillator is "interrupted" in time for the familiar sync effect. Oscillator 4 is mainly (but not always) used as a sub oscillator, offering standard pulse and sine waveforms with pulse width modulation. Below each oscillator mode selector, the tuning of each oscillator can be set in semitones and cents in the PitchShift section. The pitch of each oscillator can be modulated by either the second envelope (Env 2), or LFO 1. Oscillators 1 and 4 can have the width of their waveforms also modulated by Env 2 or LFO 1. The additional parameters in the oscillator section, such as "odet" (oscillator de-


All four oscillators plus noise are summed and sent to Carbon's outstanding filter .The filter provides eight different filter modes where nearly every mode uses straight forward technology, mainly to emulate the warm and powerful sound of analogue filters. The modes Va, CLa and J/S offer an additional alternative mode which can be switched on by the Alt control below the main mode selector, increasing the effective number of modes to 11. Many of the filter types are very similar and offer a quick and easy way to modify the sound of each preset - simply select a preset at random, then choose a new filter for it. The main parameters of each filter mode - cutoff frequency, frequency bandwidth, resonance and - can be set at the right of the mode selector. Below each parameter knob there are additional controls to modulate the respective value, including the possibility to map the MIDI velocity and the modulation wheel onto the filter's sound. The sound comes out of the filter and goes right into a saturator and four-band EQ for even more fine-control over the sound's frequency spectrum. Besides a low shelf EQ, a high shelf EQ, and peak EQ with adjustable center frequency, there is also an additional high filter used to increase the sound's brightness.

Modulation Carbon has two envelopes and two LFOs. Envelope 1 always controls the amplitude, while Envelope 2 can control different parameters, such as oscillator volume, oscillator detune, filter cutoff, filter resonance, filter bandwidth, and filter delay. There is no modulation matrix in Carbon - all modulations are set in their respective sections. For instance, to modulate filter cutoff by Envelope 2, simply look underneath the Cutoff knob in the filter section and you'll see a small knob for E2 and a button to activate the modulation amount. Carbon's LFOs offer valuable soundshaping functions. Using the FSpr knob in either LFO section adds a frequency spread to the LFO frequency across multiple


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tune), "LCo," (loudness correction) etc., are used only for very fine shaping of the amplitude and range of modulation of the oscillators across the frequency spectrum. We warned you that Carbon was from another planet!

voices. This means that if you play a chord, each note would have slightly different LFO speeds for a shimmering, organic sound. Also, the modulation of the LFO can be delayed, for instance to program a vibrate which takes a few moments before coming in.

Global Parameters Parameters effecting the entire synth are set in the Global section, such as Mono/Poly (set with the Mono button), tuning, unison spread, and glide. Hold lets you specify how long each note will last for, no matter how long the actual MIDI note was held down for (value set in mSec underneath the Hold button). Old controls how much pitch randomness is introduced to emulate vintage hardware.

Green Matrix

Instruments: Erik Wiegand; Presets: Howard Scarr; Design: Christophe Stoll Green Matrix is the MIDI-playable brother of Blue Matrix. Both use the same analog-style synthesis engine and modulation matrix to generate some amazing and challenging sounds. Its sound engine features two multiple-waveforms oscillators, a multi-mode filter, multi-mode distortion, and finally a second


Quick Start Plug in a MIDI keyboard and flip through the presets! Go ahead and experiment with the synthesizer controls - try changing LFO speed or cutoff frequency, for example.

Structure and Signal Flow Green Matrix's synth engine is based on traditional analog synthesis. It has three oscillators, two filters, distortion, and effects. Both Green Matrix and its sequencer-driven brother Blue Matrix are created from Reaktor 4's Classic Modular macro library which puts the sound quality and capabilities of the classic modular synths into your computer. For more information on building your own instruments with the Classic Modular macro library, please check your Reaktor 4 's guide. The Classic Modular library is included on the Reaktor 4 CD. Oscillators 1, 2, and 3 each offer a different waveform selection. Osc 1 offers the traditional sawtooth, pulse, triangle, sine, and impulse waveforms, while Osc 2 lets you choose between bipolar ramp, bipolar pulse, pulse, parabolic, and noise. The bipolar waveforms' shape can be modified in realtime with the small Shape knob. Osc 3 Osc 1 can be frequency modulated by Osc 2 or 3 for FM sounds (with the small FM knob and switch below), while Osc 2 and 3 can be ring modulated by Osc 1 for metallic and "gong" sounds (with the small Ring switch). Osc 1 can also be synced to Osc 2, with the choice between hard, soft, and MIDI gate-activated modes. Since Osc 1, 2, and 3 each offer unique features, they can be switched on or off to save U with the small on switch on their left sides. The base pitch and fine tuning of each oscillator is done with their respective Pitch and Fine controls.


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multi-mode filter. A beat-synced delay and a diffusion delay round out the effects. A complete modulation matrix lets you flexibly route the two envelopes, LFO, four channels of sequenced modulation, and the usual MIDI controllers to every important synthesis parameter.

The two oscillators feed into Filter 1 which offers a choice of 12 or 24 dB/octave low, high, band, peak EQ, and notch modes. Cutoff and Reson adjust the cutoff frequency and resonance, respectively, and the FM knobs controls how much Osc 1 or Osc 2 will frequency-modulate the cutoff frequency (settable with the small switch to the right of the FM knob). Pkey determines how the filter tracks the pitch of the notes from the sequencer. When Pkey is set to 1, then the filter cutoff will exactly follow the pitch of the sequencer notes, meaning that higher notes will be brighter, and lower notes will be duller. If set to zero, then the cutoff frequency won't change. The other knobs, GainC, Limit, F foll, and Rel all control the fine-tuned shaping of the filter across the frequency and resonance spectrum. GainC controls how much loudness reduction will be applied at high resonance values. Limt, F foll, and Rel control a built-in resonance limiter to avoid unwanted harmonic thumps with high resonance values. Filter 2 is similar in layout to Filter 1. There are a choice of four low filters, two band filters, and a combined band/ low filter (BL4). In addition to the controls you're now familiar with from Filter 1, Filter 2 also offers the additional possibility to trigger self-resonance with a built in click. The Click knob controls how loud the click is, and the Vel knob determines how the click responds to velocity. After Filter 2, the sound es through a distortion unit then a chorus/delay effect. Note that an extended version of the chorus/ delay effect is available in the Fusion Reflections ensemble. The Delay/Chorus consists of four stereo modulation delays. The main delay time is controlled by the Delay knob. Diffusion sets the inaccuracy of the delay times to "spread" the sound.

Modulation Green Matrix features three standard ADSR envelopes and a two LFOs that syncs to tempo. All three envelopes are velocity sensitive: Vel>A sets how much the velocity will affect attack time. The speed of the LFO can be set in musical note values, or freely. When Unit is set to 0, then the Frequency of the LFO is set in Hz. When Unit is set to any other value, the frequency of the LFO takes on the musical units set by Unit. For example, if Unit is 96 Reaktor Library English – 390

All of Green Matrix's modulation sources are routed in a comprehensive modulation matrix: The LFO, two envelopes, four modulation channels from the sequencer, and typical MIDI controls like mod/pitchbend/velocity can be routed to synthesizer destinations using the comprehensive modulation matrix. Switch to the "B" view of the synth to see the modulation matrix. You can easily switch to the B view by clicking the B button in the title bar of the synth. You can route the modulation to the following synth parameters: Oscillator pitch (osc 1, osc 2, or osc 1 + 2), osc 1 FM, osc 2 shape, osc ½ mix, filter cutoff (osc 1, osc 2, or osc 1 + 2), filter 1 FM, filter resonance (1 or 2), envelope parameters, and even distortion, delay time, and delay . In short, nearly everything! The modulation sources (LFOs, envelopes, etc) are arranged vertically. The modulation destinations (synth parameters such as filter cutoff, oscillator pitch, etc) are arranged vertically. Simply activate the switches for the combination you'd like to modulate. Underneath each modulation column is a switch to turn the modulation on or off, and a knob to set the modulation amount.


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and frequency is 1, then the LFO will be set to the very fast speed of 1/96th note per cycle. The display underneath the Freq knob depends on how Unit is set: if Unit is set to 0, then the display automatically shows values in Hz, and if Unit is set to a musical value, then the Freq display will automatically update. Width and Phase control the shape of the LFO, which can be selected by the switch to the right. Key S determines if the LFO will synchronize to note-on events, as delivered by the sequencer.

Junatik

Instrument: NI, Joerg Holzamer; Sounds: Easy Sounds, Joerg Holzamer, ear2ear, Uwe G. Hoenig; Demo: Joerg Holzamer 1982 , about one year before the era of FM Synthesis began, polyphonic analogous synthesizers finally became (more) affordable. This was mainly due to a japanese company bringing an attractively priced synthesizer to market which offered six voices and was named after a very important roman goddess. Although the first version of that synth had no memory locations, it convinced with its charismatic and full sound. In fact, the synthesizer was not complicated at all and sounds could be set up very easily and quickly, therefore the absence of memory locations (which were rather expensive at that time) wasn't very tragical. The fat sound of the synth did not come from its oscillator/sub oscillator combination, but rather from a good sounding low filter and especially an integrated chorus effect. Unfortunately the chorus was very noisy, but it broadened the somehow sterile sound of the oscillator/sub oscillator significantly and made a big contribution to the popularity of the synth. Junatik offers a surprisingly authentic recreation of the still very popular synth. The symphatically straight sound architecture of Junatik was modelled on the original and carefully supplemented with some important functions. Besides an authentic sounding filter, Junatik therefore offers an improved oscillator Reaktor Library English – 392

Kaleidon

Instrument and Presets: Mike Daliot; Design: Christophe Stoll Kaleidon is the biggest synth in the entire Reaktor Library. Its sounds cover an exceptionally wide range, from realistic mallets, organs, reeds, and saxophones to house chords, techno basses, and otherwordly atmospheres. Its specs are formidable: six oscillators plus noise and sub-osc with sync, FM, and ring modulaReaktor Library English – 393

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section. With its threefold and detunable saw wave it is now even capable of producing detuned and very fat sounds. An optional velocity sensitivity, which was not available in the original, has also been implemented. Additionally, the sound generation has been supplemented by a 3-band EQ with a semiparametric middle band, a very good sounding distortion unit and a tempo based stereo delay with filter – effects which perfectly complement the strong sound of Junatik. It goes without saying that the unrenounceable chorus effect of the original has also been implemented – yet without the noise.

tion; two multimode filters; three feature-rich envelopes; three LFOs (one which lets you draw you own waveforms); graphical key-tracking; and a full effects section featuring many unique effects, from rich reverb and chorus to a graphically sequenced delay line and quad ring-modulator.

Quick Start Kaleidon is Carbon's big brother, so you may want to take a few moments to get familiar with the Carbon synth as well. Instead of Carbon's four oscillators plus noise, Kaleidon offers six plus sub and noise. Instead of Carbon's single filter-design, Kaleidon features two filters with flexible routing. Instead of Carbon's integrated effects, Kaleidon offers full-featured effects in rack format. Due to its complexity, Kaleidon takes advantage of both view "A" and view "B" - view "B" is designed to be the day-to-day interface of the synth, with control over the most important synth parameters. View "A" is a tweak-head's delight, as it shows every single parameter directly. A monitor with 1280x1024 resolution is recommended to use the full view "A." The effects are laid out in rack-style underneath the synth. You can maximize or minimize each effect, as needed. Note that all controls in the reduced"B" view are also duplicated in the full "A" view.

Signal Flow and Structure Kaleidon's sound begins with three oscillator sections, with each section consisting of two independent oscillators. Each oscillator has a choice of pulse, sawtooth, triangle, sine, bi-saw, parabolic with PWM, and noise. All three oscillator sections are nearly identical, with oscillator sections 2 and 3 giving enhanced sync and waveform options. When the Qnt button is activated in Osc section 2 or 3, additional harmonics are added to the waveform by reducing the bit depth. Osc sections 2 and 3 also offer a three-saw waveform where three sawtooth waveforms are slightly detuned to create a rich sound. The amplitude, pitch, and width of each oscillator section (acting on both of the oscillators in the section)


The oscillators can be flexibly routed to the two filters with the Filter In Routing and Filter Out Routing controls. Here you can set which oscillators go to which filters, and you can set serial or parallel filter routing. The Prefilter loudness correction fine-tunes the amplitude of the oscillators before they hit the filters, to avoid unwanted resonant peaks. The two filters are nearly identical, with each offering multiple modes, including various flavors of low, band, and high. Each filter has a slightly different selection of filter types. Four large knobs control the important parameters of each filter: Cutoff, resonance, bandwidth, and delay. delay is only applicable to the Grobian and V4 filter types. Underneath each of the large knobs are the modulation possibilities for that knob. The small knob adjusts the amount of modulation, and the switch turns the modulation on or off for its labeled parameter. After the filters, the sound es through Kaleidon's innovative Pan modulator. Here, the sound is spread in stereo across the frequency range. For FX Matrix A, you can choose between chorus, the Vierring sequenced ring modulator, (also available separately in the library), and diffusion delays (also available as the Fusion Reflections effect). After FX Matrix A, the sound es through a saturator, and then through FX Matrix B. For the B effect, you have a choice of a tempo delay, an innovative sequenced "unit" delay, and a reverb. The reverb core is taken from the SpaceMaster reverb, available in the Library. The Sequenced Unit Delay effect is unique to Kaleidon. It consists of four delays whose delay time can be sequenced graphically. An additional filter rounds out the effect to create dynamically shifting atmospheres. The "Lurker" patch is an excellent demonstration of what the Unit Delay can do.


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can be modulated with the pull-down menus at the bottom of each oscillator. An additional noise generator and sub oscillator are also available underneath the oscillators. The A and B oscillators of each oscillator section can be ring modulated with the /RM button.

Three envelopes can control many of Kaleidon's parameters, and their shape can also be modulated by velocity or the mod wheel. The modulation is controlled in the same manner as the filters.

Global Parameters Kaleidon's global parameters are nearly identical to Carbon's. Parameters effecting the entire synth are set in the Global section, such as Mono/Poly (set with the Mono button), tuning, unison spread, and glide. Hold lets you specify how long each note will last for, no matter how long the actual MIDI note was held down for (value set in mSec underneath the Hold button). Old controls how much pitch randomness is introduced to emulate vintage hardware.

Nanowave

Instrument: Uwe G. Hoenig; Sounds: Sound Burst; Demo: Sound Burst Nanowave is an homage to the legendary WaveTable synthesizers made by PPG and Waldorf. The sound generation – called "WaveSet" synthesis here – is structured similarly to the classic subtractive synthesis used in analog synthesizers, but it has some special features on the oscillator level which result in a significantly wider variety of sounds.


Both of the Nanowave oscillators have an identical structure: Each gives you the option to select the WaveSet, the octave and the pitch tuning in semitones. With the slide switch you can manually move through the WaveSet, or alternatively the waveform position can be modulated by a separate wave envelope, by attack, keytracking or by the LFO. Additionally frequency modulation is possible. A couple of other sound sources are also available: a simple sine oscillator, a noise generator with selectable tone colour and the ring-modulation signal produced from waves 1 and 2. Nanowave’s multi-mode filter works with a slope of 12 dB/oct and can be modulated in various ways. The LFO starts with note onset, and consequently allows envelope-like modulations. It also has a separate envelope for the LFO amplitude. The three envelopes in Nanowave can be modulated by the keyboard velocity and have a special parameter which enhances the exponential characteristic of this process – very good for extremely percussive sounds.


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The WaveSet oscillators of Nanowave are not based on the common waveforms like sawtooth or square. They use so-called WaveSets which contain many different waveforms. 43 of these WaveSets are included with Nanowave. Within a WaveSet, a particular waveform can be chosen manually but the really crucial feature in WaveSet Synthesis is the option to "travel" in the WaveSet, i.e. switching dynamically between the waveforms of a WaveSet.

P-Bass

Instruments, Presets, and Design: Cornelius Lejeune P-bass is a bass synthesizer that uses physically (hence the "P") modeled string resonance to generate its tones. It sounds like a combination between an electric bass and an analog synthesizer.

Quick Start Play some notes in the low ranges of your MIDI keyboard and check out the presets. Change the Impulse, Sustain, and Bright faders to see how the sound changes in a natural way.

Structure and Signal Flow P-Bass uses physical modeling synthesis to generate its warm bass tones. A string resonator, with parameters adjustable in the String macro let you specify the brightness and resonant character of the string. You can draw in the two dimensional Damp & Freq graph to make simultaneous changes to the damping and damping frequency. The horizontal access controls the amount of damping (up to 24 dB/octave) of all frequencies above the one adjusted along the vertical axis. This filtering occurs within the resonating string model.


The Envelope controls how the sound changes over time with tradional attack, decay, sustain, and release controls. Finally, tremolo and vibrato can be added in the Vib & Trem macro.

SteamPipe

Instruments: Martijn Zwartjes; Presets: Martijn Zwartjes and Paul Swenenhuis; Design: Leonhard Lass SteamPipe is a physical-modeling synthesizer that uses a tuned resonator to create bowed, blown, and plucked sounds, as well as many strange new and hybrid sounds. SteamPipe effectively models air, or steam, being blown through a tuneable pipe. In addition to a large number of controls for the "shape" of the pipe, and a tuned all filter, there is a mod wheel-controlled filter to achieve damping and breath noise effects. Giving dimension to all of this is an excellent-sounding reverb unit.


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In the Body macro, the sound can be further filtered with the Bright knob, and another two dimensional graph adjusting the filter boost amount and frequency. The horizontal axis of this control sets the amount of boost, and the vertical axis sets the frequency.

Quick Start Since SteamPipe is such a potentially expressive instrument, you'll want to plug in your favorite MIDI keyboard to try it out. Step through the presets to get an idea of the different kinds of sounds this thing can make. Give the mod wheel a twirl or two and hear how this can add expression to the sound.

Structure and Signal Flow SteamPipe simulates the ing of air through a pipe where the size and resonance can be modified. It uses a type of synthesis called physical modeling, which uses contoured noise ing through delays that can be tuned and filtered. The ensemble is basically separated into three parts: Steam, Pipe, and R66, the reverb module. The Steam module generates shaped and filtered noise. The Pipe module gives the "wind" pitch and resonance. R66 provides the space. You can think of the Steam module as SteamPipe's oscillator, or at least part of it. Steam provides the sound energy that will be pitch-formed by the Pipe module. To this end there are an ADSR volume envelope, modulated by velocity and keyboard scaling (meaning that the envelope can be set to be shorter or longer overall as effected by velocity), and low filter with key- and velocity-tracking. The Gen section of the Steam module is where the timbral shaping of the DC/Noise source takes place. The low filter works in 1-pole or 2-pole mode, though the resonance control only applies to the 2-pole filter. After the noise is filtered, the signal is fed into the Pipe module. The Pipe module is made up of a number of submodules for creating pitch and resonance. The noise signal is fed from a single, tuned delay, which provides pitch, into the All module for resonance. A Saturator receives the signal next and applies edge and breakup. The MW Filter finishes off the signal chain, giving one final overall tone shaping stage. The and PushPull sections act on signals diverted from the main signal chain and ed back into it via loops.


The All filter receives the tuned signal from the single delay. It can be turned on and off with the Power button in the All section. The All can be tuned to create resonant effects. You can make glassy, metallic and bell-like sounds by detuning the All filter against the Delay. By adjusting the Diffusion knob, you can also create a variety of reverb sounds - the simulation of air echoing along a pipe's hard surface. The Saturation module morphs between saturation and clipping, overdriving or breaking up the signal before it hits the MW Filter. The MW Filter, controlled by the mod wheel, features a 1-pole high followed by a 1-pole low filter. Each filter allows you to set a wheel-down and a wheel-up setting, making it possible to set up complex timbre changes and damping effects. Each filter can have its own key track setting. The lone Polarity switch inverts the pipe polarity, changing the timbre of the sound, often turning high frequency tones to deep ones and vice versa. The module processes the in the signal chain. The RT knob extends or shortens the reverb generated from the signal. The reverb signal can be muffled with the Damp control. Damping can be modified by K-Track amount know. High K-Track values result in more damping on higher pitches. This allows SteamPipe to emulate struck or plucked instruments like pianos, harps, and acoustic guitars.


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The Del Tune module contains the tuned delay that provides pitch to the Steam. The Tune and Fine knobs allow you to dial in the fundamental pitch of the signal. The A440 oscillator at the bottom of the ensemble is there to give a stable pitch to tune the pipe to. The Err knob introduces small amounts of detuning to the pitch, to give a more human quaver. The Delay pitch can be swept negatively or positively by the mod wheel, with the amount of modulation set by the MW knob.

R66 The final step in SteamPipe's processing chain is the R66 reverb module. R66 is a full featured and rich-sounding reverb with lots of parameters to adjust. First, the Del module uses predelay time and left/right panning offset to position the signal in time and space. The Diffusion module has controls for size and reverb time and has a low filter and low and high frequencies damping in order to model a convincing space. There are also controls to modulate reverb time and pan position with a sine wave or random LFO, or both. The Pos knob lets you set the reverb reflections early or late, further coloring the sound of your virtual space.

Sum Synth

Instruments and Presets: Lazyfish; Design: Ian Warner


Quick Start Get your MIDI keyboard warmed up and check out the presets. Pay special attention to the Tones, Step, Detune, and Harmonics controls since they have a big effect on Sum Synth's sound. See how the four controls interact to change the width and quality of the sound.

Structure and Signal Flow Sum Synth uses additive synthesis techniques to generate its deep tones. It has sine or square oscillator waveshapes with the capabiltity to produce up to 128 additive waves, and a soundshaping section consisting of a stereo 4-pole low filter, an overdrive effect, and the Diff Chorus. Two envelopes further shape the sound -- one for volume, and one for filter frequency, which can also modulate oscillator pitch. The oscillator can be switched between sine and square (with adjustable pulse width) waveforms. The Pitch Control module contains controls to divide the oscillator pitch into detunable tones, select the initial octave, and add glide between notes. The Step knob lets you separate the tones from each other by musical intervals. Detune, will, as you can guess, detune the intervals from each other to give chorusing or atonal sounds. The Harmonics knob generates ring mod-like effects between the tones. Along with a pulse width control for the pulse oscillator, there is an Unphase knob, which causes the oscillator tones to trigger successively out of phase with each other, thickening the sound.


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Sum Synth is an additive synthesizer with a few tricks up its sleeve. A unique oscillator section makes it easy to control the frequencies of the oscillators and shape of the sound, while an informative graphical display shows the current pitch of each oscillator. The number of oscillators can easily be varied simply by changing the polyphony of the instrument in the instrument title bar.

The oscillator feeds into the Pan/Overdrive module. The Drive knob allows you to overdrive the stereo amplifier that the oscillator is fed through. Use the Pan control to widen the stereo soundfield. After the Overdrive circuit, the signal reaches the stereo low filter. The Filter module gives control over cutoff frequency, resonance, and envelope contour amount. The filter modules are taken from the Pro-52 and give out a rich, creamy sound. The Filter/Freq AHD envelope works the same way as the Amplitude envelope. Hold/sustain can be activated by velocity, allowing you create dynamic bass drum patterns and expressive lead lines. Velocity can also be routed to filter contour amount. The AmP knob routes the Filter envelope signal to the oscillator pitch. This can be useful for making the little blip at the beginning of those classic bass drums. The Amplitude ADHSR envelope has a hold stage that can be activated by velocity - click the bottom button to the left of the amplitude envelope display to turn it on. Less velocity will result in more hold time, which can create interesting gating effects in your playing. Velocity can also be routed to volume by clicking the topmost of the two buttons. The Diff Chorus consists of four stereo modulation delays. The main delay time is controlled by the Delay knob. Diffusion sets the inaccuracy of the delay times to "spread" the sound. The Dly Mod control adds an LFO to modulate delay time. The Speed knob sets the rate of modulation. The Stereo knob offers still one more method to spread the sound across the stereo field. A twoband equalizer lets you shape the chorus tone by attenuating the bass and treble frequencies. Clicking the buttons above the Bass and Treble faders engages the EQ bands.


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Uranus

Instrument: Olivier Gerber; Sounds: Olivier Gerber; Demo: M.S.Zanx Uranus offers an extensive voice architecture with three oscillators, a shaping/distortion unit, two multimode filter, four envelope generators, two LFOs, and a very "expensive" sounding chorus effect. Uranus is well suited for the whole range of analog sounds: basses, leads, effects, and especially pads. Many useful presets give you a good demonstration of Uranus' flexibility. Each of the two main oscillators in Uranus offers eight waveforms; the third oscillator offers five waves and noise. Oscillator 2 can be frequency modulated by oscillator 1. Before being fed to the filters, the oscillator mix can be tweaked with one of four different effects in the distortion unit. The filter section offers two multimode resonance filters that can be used in parallel or serial configuration. Each filter has its own set of parameters and a separate envelope generator. In serial mode, a saturation effect for emulating Moog-like filter distortion is employed between the two filters.


The modulation section features two LFOs an extra envelope with flexible routing capabilities. All envelopes work as classic ADSR types. The chorus effect plays a large part in creating Uranus' elegant and "high-class" sound. It features a quad-delay configuration with independently adjustable modulation rates.


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Synths - Sequenced Blue Matrix

Instrument: Erik Wiegand; Presets: Howard Scarr; Design: Christophe Stoll Blue Matrix is an incredible-sounding sequenced synthesizer with a classic analog-style sound engine and an integrated pitch, gate, and modulation sequencer whose capabilities rival those of many stand-alone programs. Its sound engine features two multiple-waveform oscillators, a multi-mode filter, multi-mode distortion, and finally a second multi-mode filter. A beat-synced delay and a diffusion delay round out the effects. The full-featured sequencer offers independent control over gate and pitch for smooth pitch glide, and also offers four channels of graphical modulation sequencing. A complete modulation matrix lets you flexibly route the two envelopes, LFO, four channels of sequenced modulation, and the usual MIDI controllers to every important synth engine parameter.


Quick Start Start the system clock and step through the presets! Feel free to experiment with the synth controls - try changing oscillator pitch or cutoff frequency, for instance. If you draw in the tables, you will permanently change that preset if you save the ensemble, so it's probably better to read the Sequencer section of this guide first.

Structure and Signal Flow Blue Matrix's synth engine is based on traditional analog-style synthesis. It has two unique oscillators, and a sound-shaping section consisting of Filter 1 Distortion Filter 2. Both Blue Matrix and its polyphonic, MIDI-playable brother Green Matrix are created from Reaktor 4's Classic Modular macro library which puts the sound quality and capabilities of the classic modular synths into your computer. For more information on building your own instruments with the Classic Modular macro library, please check your Reaktor 4 's guide. The Classic Modular library is included on the Reaktor 4 CD.

Osc1 and Osc2 each offer a different waveform selection. Osc 1 offers the traditional sawtooth, pulse, triangle, sine, and impulse waveforms, while Osc 2 lets you choose between bipolar ramp, bipolar pulse, pulse, parabolic, and noise. The bipolar waveforms' shape can be modified in realtime with the small Shape knob. Osc 1 can be frequency modulated by Osc 2 for FM sounds (with the small FM knob and switch above), while Osc 2 can be ring modulation by Osc 1 for metallic and "gong" sounds (with the small Ring switch). Osc 1 can also be synced to Osc 2, with the choice between hard, soft, and MIDI gate-activated modes. Since Osc 1 and 2 each offer unique features, they can both be switch on or off to save U with the small switch on the right. The base pitch and fine tuning of each oscillator is done with their respective Pitch and Fine knobs. The two oscillators feed into Filter 1 which offers a choice of 12 or 24 dB/octave low, high, band, peak EQ, and notch modes. Cutoff and Reson adjust the cutoff frequency and resonance, respectively, and the FM knobs controls how much Reaktor Library English – 408

The Distortion section between the two multimode filters also features multiple modes of operation. While the clipper mode provides a relatively harsh distortion sound and the saturator mode results in warm overdrive, the several wrapping modes (marked by the name of the waveform used for wrapping) produce unique sounds from subtle to extreme. A visual display of the distortion function helps to see what's going on inside. Note that the same Filter 1 Distortion Filter 2 section of Blue Matrix is available as an effect in the Analogic Filter Box.

Filter 2 is similar in layout to Filter 1, but its primary function to to further shape the sound after the Distortion unit. There are a choice of four low filters, two band filters, and a combined band/low filter (BL4). In addition to the controls you're now familiar with from Filter 1, Filter 2 also offers the additional possibility to trigger self-resonance with a built in click. The Click knob controls how loud the click is, and the Vel knob determines how the click responds to velocity. After Filter 2, the sound es serially through two delay-based effects: The Beat Sync delay and the Diff Chorus. Note that extended versions of these excellent effects are available in the Fusion Reflections and Echomania ensembles. The Beat Sync delay's large Delay knob sets the delay time in sixteenth notes. The Div knob to its right divides the delay time to create dotted and triplet times, while the display above the Div knob shows the


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Osc 1 or Osc 2 will frequency-modulate the cutoff frequency (settable with the small switch to the right of the FM knob). Pkey determines how the filter tracks the pitch of the notes from the sequencer. When Pkey is set to 1, then the filter cutoff will exactly follow the pitch of the sequencer notes, meaning that higher notes will be brighter, and lower notes will be duller. If set to zero, then the cutoff frequency won't change. The other knobs, GainC, Limit, F foll, and Rel all control the fine-tuned shaping of the filter across the frequency and resonance spectrum. GainC controls how much loudness reduction will be applied at high resonance values. Limt, F foll, and Rel control a built-in resonance limiter to avoid unwanted harmonic thumps with high resonance values.

actual delay time. For instance, a Delay of seven sixteenths and a Div of two will produce a delay time of 3.5 sixteenths. The small Q switch turns quantization on and off in respect to modulation coming from the sequencer - yes, it is possible to sequence the delay time! You can find more information on the sequencer below. Smth sets an internal smoother on the delay time. Naturally, sets the level, while the HP P and LP P set the frequency of a high and low filter in the path. The delay time can be modulated with an internal LFO, with A (amplitude) controlling the amount of modulation, P (pitch) controlling the frequency, and L-R controlling the stereo spread. The Diff Chorus consists of four stereo modulation delay. The main delay time is controlled by the Diff Dly knob. Diff sets the inaccuracy of the delay times to "spread" the sound. Like the Beat Sync delay, an LFO with similar controls modulates the delay time.

The Sequencer Blue Matrix's sequencer consists of three areas where sequencer data can be drawn in - the uppermost handles the pitch of the notes in the sequence, the field below displays the trigger and velocity information of those notes, and the last one contains four independent tracks where additional modulation data can be stored and edited. There are a number of horizontal and vertical control bars for zooming, selecting the length of the sequence, and selecting the "edit" range for commands such as randomize, quantize, and copy/paste/insert. Important Note!! As the complete sequencer is based on Reaktor's event tables, please note that the table data and snapshot data are independent of each other. The snapshots correctly store the position of all knobs relating to the sequencer, but they don't store any of the table data. Naturally, it is possible for each snapshot to have its own sequence, but you must take tare that the global Seq knob is set to the same number as the current snapshot. Otherwise, if different presets share the same table Seq, you could end up changing a table sequence and inadvert-


Pitch-table: To the right of pitch edit-table are three vertical bars. The first is the vertical zoom/scroll bar. Clicking and dragging near the top or bottom of this bar will act as a zoom, while dragging in the middle of the bar scrolls the display. The next vertical bar sets the trigger threshold. Any notes below this threshold won't be triggered. The third bar sets the edit range. The edit range can be extended by clicking and dragging near the top or bottom, or the edit range can be dragged up or down by clicking and dragging in the middle of it. An additional, horizontal editor bar is provided to select notes in time. The copy/ paste/cut/insert functions to the right of the vertical bars do the usual functions (for the area defined by the horizontal and vertical edit bars), with a couple neat tricks. P Inc sets how much the pitch will be incremented when data is inserted or pasted, in semitones. If you wanted to transpose a figure by five semitones, for instance, simply arrange the horizontal and vertical edit bars to cover the length of that figure, hit copy, then insert. Likewise, V inc sets the velocity increment of pasted or inserted data. Randomizing and Quantizing data: In addition to copying and pasting the range denoted by the edit bars, you can also randomize data within this range. Simply click on Rand, and the edit areas will be filled with random pitches whose properties are determined by the Hold, Retrig, and N. On functions. Data in the edit range can also be quantized with the Quant button, with resolution set by Step. An undo function is available for copy/paste, randomizing, quantization, clear, and recording data into the table.


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ently changing the sound of all other snapshots that use the same table. If each snapshot has its unique Seq number, you will avoid this problem. Please note, however, that table data is excluded from the snapshot functions such as morph, randomize, and compare. To be really safe, if you spent a lot of time working on a sequence that you love, save the entire ensemble using the Save As… command.

Gate table and Glide: Underneath the Pitch table is a table where you can draw in gate (note-on) events. The height of the event corresponds to velocity. Note that in Blue Matrix's sequencer, pitch and gate are separate. The sequencer still sends out pitch information even if there is no gate signal. The means that you can create custom glides. The Glide time is set with the small Glide knob to the left of the pitch table. Small values of glide mean the oscillator pitch reflects what's shown in the pitch table. Large glide values create a ramp, or smoothing, from one pitch to the next. In Leg (Legato) mode, then glide is only active for overlapping notes. Modulation Table: The third table in the sequencer is a four channel modulation sequencer, where you can independently draw and edit four different modulations. You can route the modulations to synth parameters in the matrix modulation section. Similar to the pitch sequencer, to the right of the mod sequencer there are three vertical bars, each with a different function. The first one is a zoom/scroll bar. The second selects the vertical edit range, while the third bar selects which channel of modulation is shown in the display. You can either view and edit a single channel, or you can view (but not edit) all four channels simultaneously. To switch between these two modes, simply click on the View button underneath the three vertical control bars. When there is a vertical grid shown, then you can draw in your modulation data. After drawing in a shape, click on View again and you will see the two-dimensional representation, where blue represents minimum values and red represents high values. In two-dimensional mode, you can view all four modulation channels simultaneously. Global loop length and zoom controls: Underneath the modulation table are three horizontal global control bars that affect all tables. The first one is the edit-range bar, that is also linked to the edit range bar for the pitch table. The middle horizontal bar is the sequence loop length bar. Similar to the other controls you're now familiar with, this bar can be resized by dragging near the beginning or end, and can be moved by dragging in the middle. You can set the quantize value for the loop bar with the Bar


Recording data into the sequencer: It's possible to save MIDI data sent to the instrument, e. g. by a keyboard. The Rec> buttons at the left of the data fields select and activate the different sequencer parts for recording. Note that both upper data fields can only be recorded together (as MIDI events always contain both pitch and trigger information) while each modulation track can be recorded separately. The actual recording starts when the Rec button is pushed. If at this moment another region of the sequence is playing than the region selected by the horizontal edit bar, the W lamp will light up to mark the sequencer's waiting status; when the global read-out pointer enters the edit region, the R lamp is will light to show that the instrument is recording.

Two hints for using the tables: 1) If you accidentally draw into a table, you can always use the table's Undo command. 2) Use the global copy and paste commands to work on a copy of a sequence, or to build a new sequence up from an existing one. Set the horizontal edit bar (either the one under the pitch table or the modulation table, they're always linked) to the length of the entire sequence. You may need to zoom out (bottommost bar underneath the modulation sequencer). Use the global copy and paste command (large buttons to the left of the trigger sequencer): Select the sequence that you want to copy using the Seq knob, press Copy, then use the Seq knob to go to the empty or destination sequence. Press Paste and viola!


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control to its left. The bottom-most bar is the horizontal zoom/ range bar, which operates the same as all the other zoom/range bars: dragging the beginning or end acts as zoom, and dragging the middle scrolls.

Modulation Blue Matrix features two identical standard ADSR envelopes and a single LFO that syncs to tempo. Both envelopes are velocity sensitive: Vel>A sets how much the velocity will affect attack time, while Velo sets how much velocity will control the amplitude of the entire envelope. The speed of the LFO can be set in musical note values, or freely. When Unit is set to 0, then the Frequency of the LFO is set in Hz. When Unit is set to any other value, the frequency of the LFO takes on the musical units set by Unit. For example, if Unit is 96 and frequency is 1, then the LFO will be set to the very fast speed of 1/96th note per cycle. The display underneath the Freq knob depends on how Unit is set: if Unit is set to 0, then the display automatically shows values in Hz, and if Unit is set to a musical value, then the Freq display will automatically update. Width and Phase control the shape of the LFO, which can be selected by the switch to the right. Key S determines if the LFO will synchronize to note-on events, as delivered by the sequencer. All of Blue Matrix's modulation sources are routed in a comprehensive modulation matrix: The LFO, two envelopes, four modulation channels from the sequencer, and typical MIDI controls like mod/pitchbend/velocity can be routed to synthesizer destinations using the comprehensive modulation matrix. Switch to the "B" view of the synth to see the modulation matrix. You can easily switch to the B view by clicking the B button in the title bar of the synth. You can route the modulation to the following synth parameters: Oscillator pitch (osc 1, osc 2, or osc 1 + 2), osc 1 FM, osc 2 shape, osc ½ mix, filter cutoff (osc 1, osc 2, or osc 1 + 2), filter 1 FM, filter resonance (1 or 2), envelope parameters, and even distortion, delay time, and delay . In short, nearly everything! The modulation sources (LFO, envelope, sequencer modulation, etc) are arranged vertically. The modulation destinations (synth parameters such as filter cutoff, oscillator pitch, etc) are arranged vertically. Simply activate the switches for the combination you'd like to modulate. Underneath each modulation column is a switch to turn the modulation on or off, and a knob to set the modulation amount.


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DSQ-32

Instrument: Martin Brinkmann; Sequences/Sounds: Rob Acid; Demo: NI DSQ-32 is a virtual drum computer with seven instruments: Bass drum, snare, closed and open hi-hat, tom 1and 2 as well as crash cymbals. The sequencer of the DSQ-32 uses the popular sequencer grid programming, the sound generation of the instruments is modelled after classic analog beatboxes. As the name already indicates, the DSQ-32 offers a maximum pattern length of 32 steps. The length of a pattern can be set with the controllers St (starting point) and End (end point). The buttons on the seven tracks allow you to switch the individual beats on or off for the respective instruments. Track eight is available to set an accent for all instruments on the given step. The two controllers lo and hi are used to set the volume for the unaccented and accented steps respectively. As a special feature of the DSQ-32, each step provides an adjustable morph value. This value is used to control those parameters in an instrument which are duplicated and labelled "... 0" and "... 1". Each of the two controls determines the value at one end of the morph range for the given parameter. The actual value used then dependends on the morph value which changes with time. For example, if P0 and P1 (for pitch control) are set to different


values, a sequence of notes in the given range can be programmed for the instrument using the morph knobs in the sequencer. To exclude a certain parameter from the morphing process just set the "... 0" and "... 1" knobs to the same value. By activating the man button of the sequencer the morph value can be controlled manually with the morph knob. Another component of the DSQ-32 is a simple delay which automatically synchronizes with the currently used tempo and the delay time can be set in steps. The generated beat patterns can be captured with the integrated recorder and exported as audio files.

Vierring

Instrument and Presets: Mike Daliot Vierring is a creative rhythm generator where you can run any audio through and get percolating percussive patterns. Vierring divides the incoming audio into four frequency bands, and each band can be graphically sequenced for amplitude and ring modulation. Two delay effects enhance the output.


Play some audio through the effect* and jump through the pesets. Note that vierring works best with harmonically rich material, such as bright sustained pads. For an interesting result, try running a complete classical music CD or record through!

Structure and Signal Flow Before sound enters the Vierring ring modulation engine, the filters in the Freq Shift section split up the sound into four bands. Hicut and locut will remove high and low frequencies respectively, while the three band (BP) knobs set the crossover points to create four frequency ranges. The controls for each of the four band are identical. Lev sets the volume of the band, and Pan controls the position in the stereo field. The BW and Res controls set the parameters for the band filter used to feed audio into the band. The RM button turns ring modulation on, and the RM knobs adjusts the amount of ring modulation. You can adjust the attack and decay characteristics of the trigger envelope with the att and dec knobs, respectively. The Length-64 parameter sets the pattern length of that band. Note that it's possible to have an 8-step lowband, and a 7-step midband, for instance! With the event table to the right of the band controls, you can sequence he amplitude and ring modulation amount (top sequence) and the effect send (bottom sequencer). The 1<>2 and wetL knobs determine to which delay the signal to, and the volume of the wet signal. The snd and viaT2 knobs adjust the amount of delay send, and the amount of delay send as specified by the second table. Underneath the two tables is a small horizontal bar that shows the frequency range that that band represents. In the Global section, you can set the active set of tables with the Pattern knob. The attack, dec, res, and BW knobs perform the same function as the similarly-labeled knobs in each band, but they act globally. Note that the filter bandwidth and the global filter frequency can be modulated by the graphical mod sequencer.


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Quick Start

* You can play a sound through the built-in loop-player, through the realtime audio inputs, or you can process audio in realtime by using Reaktor as an effect plugin. Please check your Reaktor or Reaktor Session 's guide for helpful information.

WaveWeaver

Instrument and Presets: Uwe G. Hoenig; Design: Timothy Lamb The WaveWeaver ensemble uses wavetable synthesis to produce a range of subtle or harsh tones and rhythmic pads. The oscillator in WaveWeaver is actually several sets of short, looping soundfiles, each used as an oscillator waveform. The soundfiles in each wavetable can be can be blended into each other, and you can use an XY pad to simultaneously select among them and modify the loop start point. A built-in sequencer with scale correction allows you to create some unique, shifting grooves. MoDelay and FilterDelay let you add some effects to your sounds.


Although you can play WaveWeaver from a keyboard, it's a good idea to start the system clock and run through some of the snapshots. This way you can get a feel for what WaveWeaver's controls do while sound is being made.

Structure and Signal Flow WaveWeaver's wavetable, which can be extensively modulated and morphed in tone, feeds through a distortion circuit and a bit reducer before ing an auxiliary sine wave oscillator and a noise source at the 4-pole low filter. After the filter, the signal travels through the ModDelay and the FilterDelay. Along the way, an LFO, two ADSR envelopes, and the sequencer modulate the wavetable and filter. The Wavetable Oscillator section, in the upper left corner, is where the controls for sound generation are located. There are actually two wavetables, which can be detuned from one another for a wider sound. You can crossfade between the various soundfiles of the waveset and set loop start points by moving the cursor in the XY pad. Soundfiles are located on the vertical, or Y, axis and the start is on the horizontal, or X, axis. The Octave knob to the left of the Wavetable pad lets you set the octave shift for both the main oscillator pair and the auxiliary oscillator. The Interval knob, immediately below this, controls the pitch shift, in semitones, of the second wavetable. The Detune knob allows you to fine tune the interval of the two oscillators. You can also set the amount of pitch glide for the oscillators using the Glide knob. The four knobs to the right of the XY pad control the intensity and polarity of different modulation sources that can have an effect on the wavetable. At the top, the W< Mod >S knob sets the amount of position modulation within the waveset when turned to the left, and modulates the selection of the soundfile when turned to the right. Below this knob, the - Envelope +, - Gate +, - and LFO + knobs control the mixing of the various modulation sources. You could, for example, mix the negative output of the filter envelope with the positive output of the LFO to modulate soundfile selection. The signal from the wavetable es through a distortion


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Quick Start

circuit and a bit reducer. The Drive and LoFi knobs control control the amount of processing by these effects. In addition, the wavetables can be frequency modulated by the auxiliary sine wave oscillator, for extreme FM sounds. You can set the amount with the FM knob. Pressing the Gsync switch causes the two wavetable oscillators to trigger in sync with one another on gate signals. Otherwise the oscillators start at their current free-running position. Playing with the Drive, LoFi, and FM controls can yield a variety of sounds ranging from spiky to metallic to liquid. Before reaching the filter, the wavetables can be mixed with the aux sine wave, which features tuning relative to the main oscillators in octaves, and a noise source with adjustable filter color and envelope decay. The sine oscillator can ring modulate the wavetables, and the resulting signal can also be mixed into the filter input with the Ringmod level knob. All of these sources are fed into a 4-pole low filter, whose cutoff frequency and resonance can be set with the Filter XY pad. Much like in the Wavetable section, the knobs along the right edge of the Filter XY pad control the amount of positive or negative filter modulation from the envelope, LFO, and gate signals. These modulation sources reside on the right side of the ensemble. The LFO allows you to mix sine, triangle, and square wave shapes to achieve complex modulations. LFO frequency and waveform symmetry can be controlled with the Rate and Symm knobs, respectively. You can also set the start point of the LFO wave when it is retriggered using the Phase knob. When the Gysinc button is switched on, the LFO is retriggered with each gate signal; it restarts at the point in the waveform set by the Phase knob. An ADSR envelope modulates the wavetable and/or the filter frequency. Another ADSR controls the amplitude contour. The final elements in the signal chain are the MoDelay and the FilterDelay. The MoDelay produces dimensional chorus and flanging sounds with its delay time LFO. The Time/FB XY pad allows you to smoothly adjust delay time and . The Amt/ Rate XY pad lets you set the amount and frequency of the LFO modulating delay time.


The Sequencer The three-part sequencer is started and tempo-controlled by the global MIDI clock. Swing can be added with the Shuffle control. You can adjust the sequence start point and loop length with the Start and range knobs. Pressing the Reset button in the Clock section forces the sequence to reset itself to the position determined by the Start knob. The Skip knob sets the number of steps in the sequencer's event table left out after each time the sequence advances. The Bidir and Inv buttons change the order in which the sequence steps play back. Bidir causes the sequence to run forward, then backward, while Inv inverts the playback order. These controls are extremely useful for nondestructively altering the character of a sequence in performance, for example. You can also modulate the current playback position of the sequencer over time with the two Position Mod LFOs. Each can have the modulation amount inverted and have control over LFO waveform symmetry with the Symm knobs. The top sequencer row controls the length of each note. The middle row lets you program a pitch sequence, while the bottom row sets gate value, which is wired to effect velocity, or volume, for each step. The pitch can be offset to transpose the sequence using the Pitch knob. Engaging the MIDI switch next to Pitch allows you to transpose the base pitch of the sequence and play chords with a MIDI keyboard. You can use these controls to create complex, modulating arpeggiations. The Scale Correction section gives you the option of mapping pitch data onto a specific scale, for example a c major or pentatonic scale, or a Dorian mode. This can create subtle and moving effects when you transpose the sequence pitch using a keyboard.


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The FilterDelay is a tempo-synced delay with a resonant low filter to alter the sound output. You can change the filter cutoff and resonance with the Cut/Res XY pad. Since the delay time is automatically synced to the system clock, you can use the Time knob to change delay time by beat divisions.

Samplers and Transformers BeatSlicer 2

Instrument and Documentation: James Walker-Hall; Design; Christophe Stoll BeatSlicer will separate any waveform into smaller component 'slices', which can then be individually tailored by adjusting pitch, envelope and a range of FX parameters. BeatSlicer is primarily designed for drum-loop manipulation, but the extensive range of parameters offer many creative options with any material.

Quick start Right-click (PC) / Ctrl-click (Mac) on the large central window, select 'File', 'Load data into table …' and then choose an audio loop from your sound library. Now, press the 'slice' button at the top of the instrument . The loop will be scanned and MIDI notes from C-2 (by default) will be assigned to the detected slices.


Tune allows you to transpose the pitch of the entire drum kit in semitones, and Gain controls the overall output level. Pb determines the amount that the pitchbend wheel affects the pitch of the entire loop, and Vel specifies the amount to which note velocity determines amplitude, again, for all slices. Note, however that both pitchbend and velocity can be assigned to slices individually using the modulation matrix, in which case, it is probably best to set these two global knobs to zero. Root selects the root key for the loop - i.e. the first slice will be assigned to this key. (60=Middle C, 48 C-1 etc.) Rst! (reset) sets all per-slice settings to default. BeatSlicer can slice loops by employing either a transient detection algorithim, or by dividing the loop into equal-length sections. To scan a loop for transients press Detect!. If more or less slices are required, adjust the Sens (sensitivity) knob (higher values result in more slices) and press Detect! again. Thr (threshold) specifies the lowest permitted volume (in decibels) of slices. You will not normally need to edit this parameter. To cut a loop into equal length slices, set the desired number of slices in the drop down menu, and then press Slice!. For example, to slice a 4 bar loop (4/4 time) into 16th's, select 64 slices (as 4 * 16 = 64). Good results will only usually be attained if the loop is an exact number of bars in length. Note that if you keep the mouse key depressed after pressing either Detect! or Slice!, detected slices will be revealed graphically. After using either method, additional slices can be added manually with the Add! button. Additional slices are placed at the end of the loop, so the position will require adjustment using the start and length controls (see below). Add can be used to compensate for slices missed by the detection algorithm, and also for creating multiple versions of the same slice, but with different FX parameters.


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Global controls

Loop section The large window displays the waveform of the current loop. To load a loop, Right-click (PC) / Ctrl-click (Mac) on this window and select 'File', 'Load data into table …'. The large slider bar at the top of the section indicates the current slice selected for editing. Clicking and dragging on this bar selects the slice you wish to edit (and also auditions slices). Slices can also be selected with the Sel (select) knob. When Trk (track) is enabled, the current slice is determined by the most recent MIDI note input. Disabling Trk is often preferable when BeatSlicer is being 'played' by a sequencer, and you want to select and tweak one slice in particular. Enabling the Solo button filters all MIDI input except for the current slice. The Zoom button is useful for making fine adjustments to the slice position and length. Usually, changes to any parameter apply to the current slice only. However, when the All button is enabled, you can apply changes to the entire loop simultaneously.

Slice tweaking BeatSlicer enables you to customise the sound of each slice in your loop, as summarised in the following table: Start and Length: This section allows you to manually 'nudge' the position and length of slices. The top row buttons are for coarse adjustments, whereas the second row buttons are for fine adjustments. The fine nudge distance can be specified in milliseconds using the knob at the bottom of each section. The coarse nudge distance is 10x this amount. Pitch: P transposes pitch in semitones, and the < button reverses playback. Envelope: A specifies the envelope attack time, as a proportion of the length of the slice. Thus, if A is set to 0.5, the envelope will reach peak value halfway through the slice. Similarly, D determines decay time as a proportion of slice length, so a setting of 0.5 will start the decay period (the descent to zero amplitude)


Compress: A digital compressor / wave-shaper that reduces variations in amplitude in order to obtain a louder, denser sound. Thr (threshold) determines the level (db) at which compression begins, and R (ratio) determines the extent of compression. Output gain is automatically increased so that peak input and output remain the same. Drive: Overdrive distortion with a smooth-curved saturation characteristic. Db sets the input drive amount in decibels. Output gain is automatically reduced so that peak input and output remain the same. Filter: A LP / HP resonant filter. Cut determines the filter cutoff point, and Res determines the amount of gain at that point. HP toggles between high- and low- mode. Hq employs an advanced filter algorithm that may sound better with certain settings (most typically high resonance). In HQ mode the 24 switch changes both the LP and the HP filter to 24db mode, in standard mode the 24 switch changes the LP filter only. Aux: The 2 aux sends enable you to connect external effects to BeatSlicer, and set the send levels individually for each slice. For example, you could route some hi-hats to a tempo delay unit, but not the kick drum. Normally, the aux send level is post-fader (i.e. relative to the gain slider in the Mix section.) However, when Pr (Pre-fader) is enabled, it is independent of the gain slider.Note that setting the send level to minimum (-40db) disables the aux send. Out: Gain determines the slice output level. Setting the gain to minimum (-40db) mutes the output.

Modulation BeatSlicer allows various slice parameters to the modulated by several sources.


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half way through the slice. Therefore, the overall shape of the envelope is not only determined by the attack and decay times, but also the slice length.Enabling the s button switches to key sustain mode. If the MIDI note is released before the end of the slice has been reached, the envelope will terminate prematurely.

In both modulation sections (A and B) the left text box displays the current source, and the adjacent button cycles through other available sources. The right text box displays the current destination, and the adjacent button cycles through other destinations. The slider bar specifies the amount (and direction) to which the source modulates the destination. For example, to assign velocity to amplitude, select 'Velocity' as the source, 'Amp' as the destination, and then set the slider bar to the full right hand position. Modulation sources • • • • • •

Velocity: MIDI note on velocity (range 0 to 1). PB: MIDI pitchbend wheel (range -1 to 1) PB/H: As above, but held for the duration of the slice. CC1: MIDI CC1 - the modulation wheel (range 0 to 1) CC1/H: As above, but held for the duration of the slice. Random: The randomiser generates a value between -1 and 1 every time a slice is triggered. Modulation destinations • • • • • • • • •

Gain: Slice amplitude(-100% to 100%) Pitch: Slice pitch(-12 to +12 semitones) Attack: Envelope attack time(-1 to +1) Decay: Envelope hold time(-1 to +1) Ratio: Compression ratio(-20 to +20) Drive: Overdrive amount(-60 to +60 decibels) Cutoff: Filter cutoff(-60 to +60 semitones) Aux 1: Aux 1 send level(-100% to 100%) Aux 2: Aux 2 send level(-100% to 100%)

Sync Delay and Drum16 The BeatSlicer2 ensemble also features a tempo sync delay fx unit, connected to BeatSlicers auxiliary outputs. To use it, ensure that the aux section is enabled in BeatSlicer, and the delay module is switched on.


Delay module Delay: The upper menu specifies the delay time in musical-time. Medium times (e.g. 3/16) are useful for rhythmical delay, whereas short times (e.g. 1/96) can obtain electronic sounds. The lower menu specifies the filter mode - for standard delay-type sounds, either Off or LP is recommended. The other filter modes can obtain interesting sounds, but should be used with caution as they can lead to loops with high resonance.Input specifies the initial feed level into the delay unit. The output is then re-fed into the unit at the level specified by FB (), which thus determines the rate of decay and the length of the tail.Cut and Res control the cutoff and resonance for the filter. Be careful with high resonance! Mix: Gain specifies post-delay output level, Pan the stereo output position of the channel, and Wet mixes between the dry and wet signal. To use this module as a stereo effect, pan channel 1 hardleft and channel 2 hard-right. LFO / Route: The two tempo-synced LFO's can modulate various parameters via the Route section. Most typically perhaps, stereo pan to create ping-pong delay.

Drum16 Arrange: The pattern length can be specified with the Bars and Beats controls.When Seq is activated, the pattern sequencer controls the current pattern. Select / Copy: When Seq is disabled, the top slider selects the current pattern to play (Pressing Clear will erase this pattern. The lower slider selects the copy destination. Pressing Copy will overwrite this pattern.


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Additionally, a 16-channel sequencer is included, which is useful for driving BeatSlicer in Reaktor standalone mode (but will also sync to the host clock in plug-in mode).

Matrix: Here, in addition to drawing the actual pattern sequence, you can select the MIDI note of each channel, and Mute and Solo channels with the left and right blue buttons respectively. Pitchbend: By default, values in the Pitchbend grid snap to 1/24 range of the wheel. If BeatSlicers pitchbend range is set to 12 semitones, each step on the grid will correspond to one semitone.

GrainStates

Instrument and Presets: Martin Brinkmann; Design: Ian Warner GrainStates is a granular texture maker that works wonders for creating dense, breathing atmospheres. Taking advantage of Reaktor 4's grain cloud delay module, GrainStates lets you create granular soundscapes in realtime. You can even freeze the live audio - imagine playing a guitar into Grainstates, freezing the audio, then playing a counterpoint to the granular texture. Eight scenes - each scene storing information about grain size, density, pitch, pitch spread, and more - are sequentially recalled in sync with the master tempo. A dual-frequency delay adds depth to the sound by letting you specify independent delay and times for the high and low frequencies. Grainstates consists of two sister ensembles: GrainStatesFX and GrainStatesSP. GrainStatesFX is an effect using the grain cloud delay that works on live input, while GrainStatesSP is centered around the grain cloud sampler module. The FX ensemble has the advantages that you can easily process audio without loading anything into a sampler and you can freeze the incoming audio


Quick Start GrainStatesFX: Start the system clock and run some audio through the ensemble.* You'll notice a graphical representation of the sound "marching through" the granular buffer. To freeze the buffer, press the Freeze button to the left of this graphical display. If you stop the system clock, the sound will continue, but the scenes won't advance. GrainStatesSP: Start the system clock and step through the presets. If you stop the system clock, the sound will continue, but the scenes won't advance.

Structure and Signal Flow At the heart of GrainStates is the granular grain cloud module. GrainStatesFX is based on two (for true stereo operation) grain cloud delay modules, while GrainStatesSP is based on a single grain cloud sampler module. Both the grain cloud delay and grain cloud sampler have identical controls, with the grain cloud delay adding the ability to freeze the sound. All of GrainState's controls (with the exception of the 2Band Delay) are used to control the grain cloud. A master sequencer runs through eight scenes are run through sequentially, with each scene providing control over various granular parameters. Every scene can have its own length, settable by the Ln slider, whose units are set in the Seq Control macro. You can also set the total number of scenes (NrSt), and if you want to disable the scene sequencer simply click on "man" and you can select a scene manually with the SelS knob.


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stream - great for live performance! Since the sound es through the ensemble, however, there's no way to save the sound data with the preset. GrainStatesSP stores the sample with the preset so it can easily be recalled at a later time, but you must first load your sound into the grain cloud sampler.

Each scene provide control over pitch jitter (PJ: amount of pitch randomization, in semitones), pitch shift (PS: in semitones), transposition (TP: in semitones), volume (Lvl), and an XY lets you set two parameters graphically at once: The horizontal axis sets the start position of the grain (relative to the graphical display on top of the internal buffer), while the vertical axis sets the length of the grain. Three additional knobs provide control over the smoothness of scene transitions, and the grain density smear (Smr). GrainStatesFX only: The sound is filtered by the Filtor macro with independently adjustably high (HP) and low (LP) frequencies. Res adjusts the resonance of both filters, while Byps disables the filter. The output of the filters is fed back into the grain cloud delay, with independently adjustable per scene with the FB slider to the right of the XY control. The is only active then the grain cloud delay is not frozen. When the grain cloud delay is frozen, it ignores any signal to its inputs. The sound then es through a 2Band Delay, which gives independent control over delay time in sixteenth notes (StepsL and StepsH), (FBL and FBH), and a filter-modulating LFO. The cutoff and resonance of the filter that splits the two bands is determined by the Frq and Res knobs. Finally, a D/W knob sets the mix level. To by the filter, simply set D/W to zero.

Additional Controls Global Params. In the global parameters, you can set the global attack and decay of the grains, and the amount of pan jitter (stereo randomization). In GrainStatesSP, this is also where you select the active sample, with Sel. GrainStatesFX only: the Move macro controls a built-in ramp oscillator that controls the delay time. The Steady knob is the amount of delay modulation - when at zero, then the ramp oscillator does not change the delay time.


In addition to automated sequencer control, GrainStates also lets perform with a MIDI keyboard. In the MIDI macro, you have control over the MIDI functionality. If "m TP" is activated, then MIDI notes will pitch the sound, like in a conventional sampler. In GrainStates, however, all notes are the same length, regardless of pitch. When "mSel" is active, each scene is mapped onto a note pitch between 48 and 59 (only the white keys of a keyboard); by pressing one of the notes the respective scene is selected. With the Split knob you can specify another keyrange that recalls scenes. "m-tg" toggles MIDI triggering of sound on and off. GrainStatesFX only: "m-frz" lets you toggle the freeze effect on/off via MIDI. GrainStatesSP only: If you play GrainStatesSP over MIDI, you can activate the envelope (env macro) to also control the amplitude of the sound with MIDI, according to the settings of the envelope. * You can play a sound through the built-in loop-player, through the realtime audio inputs, or you can process audio in realtime by using Reaktor as an effect plugin. Please check your Reaktor or Reaktor Session 's guide for helpful information.


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MIDI Control

Memory Drum

Instrument and Documentation: James Walker-Hall; Design: Christophe Stoll Memory Drum is comprehensive MIDI drum sampler that enables the independent configuration of up to 60 drum samples in a single, easy-to-use interface. Specifically designed for drum sampling, it features an Attack-Hold-Decay envelope, compressor, saturator, multimode filter, multiple output channels, 2 auxsends, and complex modulation possibilities. The simple interface is designed so that kits can be constructed quickly and easily, yet the extensive range of effects and modulation options offer vast creative possibilities for generating new sounds from your existing samples.


Double-click on the sampler window, open the Reaktor sample map editor, and load Memory Drum with a few drum samples. As you trigger the samples from your keyboard, notice that the blue box near the top of the instrument moves to indicate the current MIDI note. Any parameter that you tweak will be ed for that note. For example, apply an envelope to a sample - click the top-left blue square in the Envelope A section, and adjust the attack and release times. Now press another key on your keyboard, and adjust some parameters for that sample, and so on…

Global controls The master controls are located at the top of the instrument . Tune allows you to transpose the pitch of the entire drum kit in semitones, and Gain controls the overall output level. Pb determines the amount that the pitchbend wheel affects the pitch of the entire kit, and Vel specifies the amount to which note velocity determines amplitude, again, for all samples. (Note, however that both pitchbend and velocity can be assigned to samples individually using the modulation matrix, in which case, it is probably best to set these two global knobs to zero). Shift transposes MIDI note input up or down as required. Rst! (reset) sets all per-note settings to default. By default, Memory Drum has 2 stereo output channels, but these can be used as 4 discrete mono channels by activating the M (mono) button.

Edit section The indicator in the top-left of this section displays the current MIDI note selected for editing. Clicking and dragging on the large slider bar allows you to select (and audition) other MIDI notes. You can select a range of notes to edit simultaneously by dragging at the bottom of the slider.


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Quick start

The lower slider scrolls the viewable area up and down the MIDI note range C-2 to C2. Clicking once on this bar toggles the view area size. When Trk (track) is enabled, the current sample is determined by the most recent MIDI note input, which is useful during initial kit creation. However, disabling Trk is often preferable when Memory Drum is being 'played' by a sequencer, and you want to select and tweak one sample in particular. Enabling the Solo button filters all MIDI input except for the current sample. When the All button is enabled, sample parameter changes are applied to the entire kit simultaneously.

The sampler The sample window is used to load samples (double click on the window). The playback start position can be specified using the slider bar below the sampler window, as can the playback direction (the button labelled <). The sound of each sample in your kit can be tailored extensively, as detailed in the following table: Sample parameters Pitch: The main slider in this section sets pitch transposition in semitones. The Fine buttons allow fine-tuning of pitch in steps of +/-0.05 semitones. Envelope A: A (attack) specifies the time taken to reach full amplitude, H (hold) specifies the time held at full amplitude, and D (decay) specifies the time taken for amplitude to fall back to zero. Enabling the s button switches to key sustain mode. If the MIDI note is released before the end of the slice has been reached, the envelope will terminate prematurely. Compress: A digital compressor / wave-shaper that reduces variations in amplitude in order to obtain a louder, denser sound. Thr (threshold) determines the level (db) at which compression begins, and R (ratio) determines the extent of compression. Output gain is automatically increased so that peak input and output remain the same.


Filter: A LP / HP resonant filter. Cut determines the filter cutoff point, and Res determines the amount of gain at that point. HP toggles between high- and low- mode. Hq employs an advanced filter algorithm that may sound better with certain settings (most typically high resonance). In HQ mode the 24 switch changes both the LP and the HP filter to 24db mode, in standard mode the 24 switch changes the LP filter only. Aux: The 2 aux sends enable you to connect external effects to Memory Drum, and set the send levels individually for each sample. For example, you could route some hi-hats to a tempo delay unit, but not the kick drum. Normally, the aux send level is postfader (i.e. relative to the gain slider in the Mix section.) However, when Pr (Pre-fader) is enabled, it is independent of the gain slider.Note that setting the send level to minimum (-40db) disables the aux send. Out: Here you can adjust the gain (in decibels) and pan for each sample. The out button cycles through available outputs. Setting the gain to minimum (-40db) mutes the output.

Modulation Memory Drums' advanced modulation routing allows various parameters to be modulated by a variety of sources (both MIDI and internal). In each modulation section (A, B and C) the top text box displays the current source, and the adjacent buttons cycle through other available sources. The bottom text box displays the current destination, and similarly the adjacent buttons cycle through other destinations. The slider bar specifies the amount (and direction) to which the source modulates the destination. So, to assign velocity to amplitude, select 'Velocity' as the source, 'Amp' as the destination, and then set the slider bar to the full right hand position.


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Drive: Overdrive distortion with a smooth-curved saturation characteristic. Db sets the input drive amount in decibels. Output gain is automatically reduced so that peak input and output remain the same.

Some modulation sources have a 2nd variation denoted by /H. This option holds the value of the source for the entire duration of the sample. Try asg Pitchbend to Pan and repeatedly triggering the sample whilst modulating the PitchBend wheel. Now, change the source to 'PB/H' and note the difference. Modulation sources • • • • • •

Velocity: MIDI note on velocity (range 0 to 1). PB: MIDI pitchbend wheel (range -1 to 1) CC1: MIDI CC1 - the modulation wheel (range 0 to 1) CC7: MIDI CC7 - the volume slider (range 0 to 1) CC10: MIDI CC10 - the pan knob (range 0 to 1) Envelope A: Note that the amp envelope (Envelope A) can always be used as a modulation source, regardless of whether it is enabled in the Envelope A section. • Envelope B: A three-stage attack-decay-breakpoint-decay envelope for general-purpose modulation. A (attack) specifies the time taken to reach the envelope peak, D1 (decay #1) specifies the time taken to reach the breakpoint level (set by BP), and D2 (decay #2) determines the time taken to reach 0. The total time of the envelope is always the sum of A, D1 and D2, and is unaffected by MIDI note off events.Vel specifies the extent to which MIDI note velocity influences the peak level. Therefore, when it is set to 0, the envelope peak always equals 1. When it is set to 1, peak = velocity and when it is set to -1, peak is inversely related to velocity. • LFO A / B: The 2 LFO's automatically sync to the song tempo and position. A variety of shapes can be selected using the upper menu, and for the pulse and random shapes, Sm (smooth) denotes that the output is smoothed by a 20ms ramp. Frequency can be set in musical time by the lower menu. The output is shown visually at the bottom of the section, but also, clicking and dragging on this bar distorts the LFO shape.The LFO's are monophonic and thus the above controls are global parameters for the entire kit. • Random: The randomiser generates a value between -1 and 1 every time a note is triggered.


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Modulation destinations • • • • • • • • • • •

Gain: Sample amplitude(-100% to 100%) Pan: (-1 to +1) Pitch: Sample pitch(-24 to 24 semitones) Start: Sample start position(-1 to +1) Attack: Envelope attack time(-70 to +70) Hold: Envelope hold time(-70 to +70) Ratio: Compression ratio(-20 to +20) Drive: Input drive(-60 to +60 decibels) Cutoff: Filter cutoff(-60 to +60 semitones) Aux 1: Aux 1 send level(-100% to 100%) Aux 2: Aux 2 send level(-100% to 100%)

Sync Delay and Drum16 The Memory Drum ensemble also features a tempo sync delay fx unit, connected to Memory Drums auxiliary outputs. To use it, ensure that the aux section is enabled in Memory Drum, and the delay module is switched on. Additionally, a 16-channel sequencer is included, which is useful for driving Memory Drum in Reaktor standalone mode (but will also sync to the host clock in plug-in mode).

Delay module Delay: The upper menu specifies the delay time in musical-time. Medium times (e.g. 3/16) are useful for rhythmical delay, whereas short times (e.g. 1/96) can obtain electronic sounds. The lower menu specifies the filter mode - for standard delay-type sounds, either Off or LP is recommended. The other filter modes can obtain interesting sounds, but should be used with caution as they can lead to loops with high resonance.Input specifies the initial feed level into the delay unit. The output is then


re-fed into the unit at the level specified by FB (), which thus determines the rate of decay and the length of the tail.Cut and Res control the cutoff and resonance for the filter. Be careful with high resonance! Mix: Gain specifies post-delay output level, Pan the stereo output position of the channel, and Wet mixes between the dry and wet signal. To use this module as a stereo effect, pan channel 1 hardleft and channel 2 hard-right. LFO / Route: The two tempo-synced LFO's can modulate various parameters via the Route section. Most typically perhaps, stereo pan to create ping-pong delay.

Drum16 Arrange: The pattern length can be specified with the Bars and Beats controls.When Seq is activated, the pattern sequencer controls the current pattern. Select / Copy: When Seq is disabled, the top slider selects the current pattern to play (Pressing Clear will erase this pattern. The lower slider selects the copy destination. Pressing Copy will overwrite this pattern. Matrix: Here, in addition to drawing the actual pattern sequence, you can select the MIDI note of each channel, and Mute and Solo channels with the left and right blue buttons respectively. Pitchbend: By default, values in the Pitchbend grid snap to 1/24 range of the wheel. If Memory Drum's pitchbend range is set to 12 semitones, each step on the grid will correspond to one semitone.


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Tips Voice allocation If you are triggering many samples simultaneously, or using long samples with long hold and/or release times, you may need to increase the number of voices on the instrument title bar. (Of course, increasing the number of voices places greater demand on the U.) Minimising U demand Switch to the 'B' view after configuring your drum kit, as this relieves Reaktor of updating the many XY displays. If any section is not being used, then disable it by deactivating the button in the bottom left corner. Only use the 24db and HQ filter modes if necessary. Disable and disconnect any unused aux send effects, and if you are not using multiple outputs disconnect them by deleting the wires in the ensemble structure. If U usage is still too high, use a lower sample rate (until mixdown), and try using fewer voices. Creative tips Using the compress and drive sections, you can obtain very dense and 'fat' drum sounds quickly and easily. Note that using light settings in both sections simultaneously often obtains a more desirable sound than heavier settings in only one section. Also, try routing velocity to compressor ratio and or drive as an alternative to gain. Asg the LFO's to pan is an easy way to breath life into dull drum loops, and more experimental loop sounds can be obtained by asg the LFO's to pitch or cutoff, for example.


Travelizer III

Instruments and Presets: Uwe G. Hoenig; Design: Cornelius Lejeune Travelizer III is the latest version of the classic granular texture maker. Travelizer lets you scrub through any sample using the grain cloud module. It can be played over MIDI, allowing you to create ethereal pads and leads. Travelizer III differs from its earlier versions because grain length can be quantized to 16th notes, allowing textures and granular rhythms to be synced to MIDI clock.

Quick Start Step through the presets with the system clock on and feel free to experiment! Try using the travel xy pad to control sample playback position and grain length while a sample is playing. If you click on the pad, the volume Attack-Release envelope will trigger and start the sound. Click the Gate button to continuously trigger the volume envelope so you can make adjustments while the sample is looping.


Travelizer III granulates any sample loaded into it, giving you a variety of controls over the quality of the granulation. The sample grains through a 3-voice Resonator with independent controls for tuning and the capability to track midi notes. The Resonator is followed by a stereo delay with paired with a high- filter. An attack-release envelope allows you to contour the loudness of the sound. Travelizer III's grain cloud module allows you to break the sample down into grains, specifying certain qualities of the sound, like grain size and smoothness. The Pitch module has controls to tune the sample and set whether it will track incoming MIDI notes. The MIDI switch turns note-tracking of sample pitch on. The Jitter control introduces small amounts of pitch modulation. The Slide switch next to this makes the sample pitch glide from note to note. Slide intensity can be positive or negative. An LFO allows you to modulate Travelizer III's pitch - turn it on and off with the small switch above the Shp fader. The LFO can be crossfaded between triangle wave and sine wave shapes with the Shp fader. Clicking and dragging on the Amount and Rate x-y pad lets you adjust modulation depth and speed. The LFO can be set to retrigger on MIDI gate events with the Gate button. The Position module allows you to modulate the playback position of the sample. You can smoothly, or roughly, sweep through the sample using a crossfadeable sine/triangle LFO like the one in the Pitch module. The Inertia control allows you to smooth out the LFO movement, in effect limiting its swing. The Jitter control adds small amounts of random position modulation to achieve a glitchy rhythmic edge. Like the Pitch LFO, the Position LFO can be retriggered by MIDI gate events by clicking the Gate button. The Smoothness control lets you adjust volume crossfading between sample grains. A high Smoothness amount will result in a more defined sound with less obvious granulation. The lower the Smoothness control is set, the more granulated the output will sound.


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Structure and signal flow

The sample is now fed into the Resonator. The Resonator uses three voices to create tuned atmospheric and spatial effects. Each of the three voices can be tuned in semitones relative to each other. Clicking the MIDI switch lets you control the root pitch using MIDI notes. You can control damping and decay of the Resonator with the xy pad. A rhythmic loop can be turned into a harmonic soundscape by manipulating the damping and decay controls and creating chords with the voice tunings. From the Resonator, the sample es through the Delay module, which includes a high- filter. You can use this to manipulate the sonic character of the sample before it hits the delay by removing low frequencies with the filter. The Delay xy pad lets you set 16th note-quantized delay times for the left and right channels. Finally, an attack-release envelope allows you to contour the loudness of your sound. Use the sample control to select a sample or double-click on the grain cloud module to load your own.


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Live Tools goBox

Instrument and Presets: Uwe G. Hoenig; Design: Timothy Lamb goBox is a monophonic sampler specially designed for live use. It features some hands controls for altering patterns, sounds, and modulations during a performance. A series of event tables form an easy to use interface that make it easy to see what's going on. Apart from sample modulation capabilities, there are a tempo-synced Filter Delay, a Mod Delay, and the Sync-ro-nizer module, which can play patterns of short samples metronomically.

Quick Start Start the system clock and check out some of the presets. As the sequence plays, try adjusting some of the sample modulation parameters, like Attack, Loop, or Octave. Try out the tap 'n' drag slider located under the waveform display to manually modulate a sample as it plays.


Structure and Signal Flow goBox is a sequenced sampler of an unusual sort. Instead of playing loops it triggers one short sample at a time, monophonically. This can lead to bizarre junky/funky beats, uncertain patterings, and minimalist dance floor workouts. Two event tables sequence the pitch and trigger state of each of four sampler modules. The Start point, Octave, and Pan position can be set using event tables for each of the samples. Start determines where in the sample the loop will start, while Octave tunes each sample in octave steps. Pan lets you position the sample in the stereo field. The samples have their amplitudes contoured by four ADR envelopes. The tables for these are located in the second row in the sample modulation area. Overall volume for each of the four sample channels can be set using the Gain control. The Reduktor performs sample rate- and bit depth-reduction for each sample. Use this to add some dirt to your samples or to reduce them to staticy noise. All of the envelopes can be scaled, that is, generally shortened or lengthened in time, with the Scale button below the Morph XY pad. Use the Scale control to tighten up beats or to make beats seem louder by increasing the overall decay and release times. The Loop table controls how much of the sample loops as it is triggered - beginning at the point set by the Start control. Odd loop lengths can create stuttering and staccato rhythm effects. After being contoured, the samples can be fed into the effects modules in differing amounts with the Filter Delay and the Mod Delay tables. The Filter Delay's delay time is automatically synchronized to the system clock in 16th notes. Filter frequency and resonance can be adjusted with the XY pad. Use the Envelope knob to modulate filter cutoff with the sample envelope. A Mod Delay for chorus and reverb-like effects accompanies the Filter Delay. Two xy pads let you adjust the Time/ balance and the Amount/Rate of delay time modulation. The Sync-ro-nizer module sits on the end of the signal chain after the effects. It offers two synced sample players for generating simple, metronomic beats to help keep everything in line and lock down your rhythms. Select a sample using the Select knob. The Reaktor Library English – 444

The Sequencer Event tables are used to sequence the pitch and trigger values for the sample loop module. The Pitch table sets the pitch for each step of a 16th note grid. The Trigger table selects which of the four possible samples plays on each step of the grid. You can place a rest into a sequence by selecting no sample for that point on the grid. As the clock runs, the sample selections play on the steps they are set for. A Transpose fader to the right of the Pitch and Trigger tables shifts the pitch sequence up or down. You can select a sample for each of the four samplers using the Choose table. Load new sample maps into the sampler module by double clicking on the waveform display. You can label your sample categories by double clicking on the text field to the left of the Choose table and typing in the corresponding areas. The clock controls for running and modulating the sequence all appear in the upper left hand corner. You can use the Scene knob to store sequencer and sample settings independently of the snapshot menu. You can store up to eight scenes. The Reset selector allows you to determine when the sequencer will reset its start point to the step chosen with the Start selector. From top to bottom, the sequence can be set to: reset every bar, reset every 2nd bar, reset every fourth bar, or no reset at all. The Grid selector, below Reset, is used to set the beat resolution of the sequence grid, i.e. 1/32, 1/16, 1/8, 1/4. Set the length of the sequencer pattern using the Range selector. A range of "0" will mute the sequencer. The T button will instantly switch the sequencer resolution to triplets of the chosen grid value, introducing interesting rhythmic twists. The 1/2 button will set the sequence playback to half time. Since this is a "nondestructive" event, not changing the timing, goBox will always stay on the beat. The Reverse switch, labeled by an arrow pointing to the left, has the effect of reversing the sequence play-


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sample pitch can be adjusted using the Tune control. The Style knob selects a timing value on a preset sample trigger grid (i.e. 1/ 8th notes, 1/16th notes, 1/32nd notes, etc). You can turn each sample on and off using the mute buttons below the level meters.

back. This is also a "nondestructive" effect, instantly switching on the beat back to forward play. The Bi-Directional sequence playback button, to the right of Reverse, causes the sequence to alternate forward and reverse playback. Finally, the Shuffle knob allows you to add varying degrees of swing to the sample triggering. goBox can shift the sequencer patterns automatically to keep your beats fresh by using the Position Mod LFO. Position Mod is a low frequency square wave with adjustable width and rate that modulate the current sequencer step position. You can specify the range of modulation in sequencer steps by using the Steps control. Along goBox's bottom edge are a variety of hands-on real-time sequence modulation controls. The Skip buttons causes the sequencer to skip 1, 2, or 3 steps - adding dramatic rhythmic variations. The Trig buttons cause the currently playing sample to retrigger, or roll, at 1/8, 1/16, or 1/32 notes. Try these buttons while the sequencer is running to drop rolls and trills into the sequence as fills or turn-arounds. The Hold button repeats the current sequence step when depressed. This is perfect for creating breaks to heighten drama in a performance. Porta introduces a small amount of pitch glide to the samples. The Rev button creates a sort of reverse playback. The Legat button forces consecutive sequencer steps with the same gate value NOT to retrigger the envelope. This causes some samples not to sound on their sequencer steps, stripping back the beat.

Morph The Morph pad allows you to make overall changes to goBox's settings with one easy to use XY control. It can be used to momentarily transform your sequence and/or sample settings, or to add small amounts of almost random change to your performance.


Scenario

Instruments: Lazyfish; Presets: Jake Mandell; Design: Timothy Lamb Scenario is a complete live-performance environment with realtime timestretching, performance-oriented effects, and memory and instant recall of thousands of scenes. Have you ever played live with a computer and been frustrated that you don't have enough control over the audio? Or - at the oppositve end of the spectrum - have you ever been performing and been overwhelmed with the possibilities? Scenario solves both of these seemingly opposing problems in an elegant and ingenious way. First, let's take a look at what Scenario is, and then we'll see why it's an ideal live-performance tool. Scenario consists of a Loop Engine that contains four identical time-stretching loop players that fits each loop to the system tempo. Each loop player lets you perReaktor Library English – 447

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The Y-axis morphs the sequence parameters, while the X-axis handles the sampler parameters. Clicking on the pad introduces the Morph effect at that position for as long as the mouse button is held down. Settings return to their previous positions when the mouse button is disengaged. Clicking the M On button holds the Morph setting as long as the button is on.

form in realtime with the loop length, loop start, pitch, and animated filter. A powerful performance-oriented effects block is also included, which lets you do things never possible before since the Scene Effects and Loop Engine are controlled by the same sample-accurate system clock. For instance, you can reshuffle and even reverse the music, rhythmically gate the sound exactly on the beat, create super-tight loops, and more. The Miniseq instrument can shuffle and reorder each loop. Just draw a pattern into Miniseq, then activate the shuffle function by clicking the small button in the Loop Engine at the bottom of each loop player. And now, the kicker: All of Scenario's settings can be stored into a "scene". You can store more than sixteen thousand scenes in the built-in Scene Memory. Scenes can be recalled sequentially for "one-touch" performance, but you still have full control over all parameters within each scene. That's what makes Scenario so revolutionary - it gives you the power to lay out an entire live set in advance so you're assured that things go smoothly. Not only you determine when to advance to the next scene (this would be quite boring!), but you also have complete control over every scene. You can adjust the pitch of the loops, the loop length, the filtering, and the effects - every single parameter on the screen. When it's time, switch to the next scene, and - Bam! - it comes in perfectly on the beat, every knob perfectly recalled, including the effects.

Quick Start For an easy tour of what Scenario can do, take a look at the demo live-set that is included in it. Start the system clock from the toolbar - you should hear some music now - and then after a few bars, press the Next button on the Scene Memory instrument on the bottom. You just advanced to the next scene. Keep on pressing Next every few bars - notice that every single parameter of Scenario is updated, including all effect settings. Feel free to play with any of the knobs to see how they influence the sound. Notice that each effect must be first enabled in order to hear it - simply click on each effect's Enable button. You can only hear a cell when it's active - to activate or mute a cell, simply click on the button below that cell's level meters.


The Scene Loop Engine makes all the noise. It consists of four identical loop players and a handy bar/beat counter so you can always keep track of where you are without having to count: The three marching bars on top denote the number of 4-bars, bars, and beats. Every sixteen bars it's reset to zero. The summed output of all four loopers goes through the Scene Effects, and then to the output. Let's take a look at an individual looper in the Scene Loop Engine. Each looper gives precise control over the loop length, pitch, and filter of the loop. The tempo of the loopers are all synchronized to the master system clock, whose tempo is settable in the toolbar. The audio will stretch or compress in realtime to fit at the given tempo. The time stretching algorithm is very high tuned for rhythmical material. SampleSel selects which loop to play. Each looper can hold a Map of 128 different loops, and SampleSel simply selects the desired loops. See the section below, "Loading Samples," for more information on how to easily load your sounds into "Scenario." Offset controls the sample start time, in sixteenth notes - where the sample will start from at the downbeat. LoopLng sets the loop length, also in sixteenth notes. Pitch lets you modify the pitch of the sample without changing its time. Wdth, Filter, Reso, rnd, and Bps all control the looper's internally-animated filter. The filter is a band consisting of two two-pole filters (a high- filter that cuts the lows, and a low- filter that cuts the highs). The width between the two filters is controlled by Wdth, and the combined resonance of both filters is set by Reso. Filter sets the center frequency of the two filters, and Bps byes the filter completely. Rnd controls how animated the filter is. If set to zero, then the filter will stay at the frequency set by Filter. But if Rnd is high, then the filter will slowly move around to create movement. Fit changes the pitch of the loop to "fit" it to the current tempo, just like a turntable changes the pitch when it slows or speeds a record up. Finally, two more controls: Gain controls the volume of the looper, and the unlabelled button underneath the level meters activates (on) or mutes (off) the track. The small button underneath this activation button activates the retrigger sequencer (Miniseq).


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Structure and Signal Flow

The output of the Scene Loop Engine gets fed into the Scene Effects block. Scene Effects consists of four effects (Loop, Slicemanipulator, InfinityFFB, and Gate), routed serially, and a master two-channel crossfader. The first effect is a looper - it simply loops the incoming audio, but it does it very precisely. The Loop knob controls how long the loop will be (in bars), and the Loop button activates and deactivates the loop. Slicemanipulator is an extremely unusual, yet performance-friendly effect. It divides the audio into "slices" then lets you rearrange and even reverse the audio in time. Split controls how many slices there will be per bar. Shift A moves even slices forward in time (in sixteenth notes), while Shift B moves odd slices forward. Rev A reverses the even slices, while Rev B reverses the odd slices. Enable turns the effect on or off. InfinityFFB is a delay effect with a finely-tuned path. This means that you can freely perform with the Delay, , and Filter knobs without risk of overloading or running into distortion. Delay sets the delay time in sixteenth notes, while Smth controls how smooth changes in delay time will be. Fback sets the amount of within the delay, while Filter controls the filter frequency of a filter built into the delay path. Wdth and Reso also control that filter - the structure of this filter is also a dual-two pole filter like the filter for each looper channel. Enable turns the effect on or off. The last effect, Gate, rhythmically gates the audio with the frequency set by Freq (in sixteenth notes). Offs sets the latency of the gate - a higher offset value lets more audio through before the gate kicks in. Hold controls how long the gate will be open for - at small values hardly any audio will through. The usual Enable button turns the effect on or off. Finally, we have a two band crossfader as a master dry/wet control. We have independent control over the high and low frequency ranges. Low controls the mix of the low frequencies, and High controls the highs. When the crossfader is fully up, the effect is active for that frequency range. The third instrument is the Miniseq, where you can draw in a retrigger sequencer for the Loop players. All loop players share the same sequence, but the retrigger sequencer can be turned on and off for each Loop player independently. The small bottom-most


The fourth instrument in Scenario is the Scene Memory, the brain that doesn't make any sound. This brain stores all scenes into a giant table. Don't be frightened by the complex-looking table - it's there just to show if a scene has information in it or not. The current scene is shown in two ways: by the read-only PartN and SceneN numerical displays to the bottom left, and by the two draggable rectangles above the table. You can change the scene in two ways: By pressing Prev/Next to go the previous or next scene, respectively, or by dragging on one of the two rectangles above the table to go directly to a specific scene. There are two settings to adjust the size of the table: Parts and Scenes. The total number of scenes is Part * Scenes - each part can contain from 16 to 128 scenes. Since there can also be from 16 to 128 parts, you can adjust the size of the scene memory from 128 (16 * 16) to 16,384 (128 * 128). Generally, depending on the style of music, 128 scenes should be sufficient for a 30 to 60 minute live set. Working with high numbers of scenes has the disadvantage that the draggable part and scene-selector rectangles (above the scene memory table) become very small. The Write button (all the way on the right) does exactly what you'd expect - the current positions of every parameter is magically written into the table at the current scene position. The Flow button is an important feature in organizing your scenes - when Flow is on, anytime you go to a new scene (either with the Prev/Next buttons or with the draggable scene and part selectors above the table), then the new scene will be immediately active. This is probably what you want for live performance, but Flow can be turned off for when you're composing your set. With flow off, when you go to a new scene, it does not load. This is very useful for copying and pasting scenes from one location to another.


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button underneath each Loop player activates the retrigger sequencer. The bottom-central Hold and Smooth knob adjust the hold time and smoothness (envelope time) of the retrigger sequencer for all loops.

Loading Samples The most convenient way to load samples into Scenario is with the Browser. Simply double click on the visual display of the sample to enter the sample map editor. If you want to clear the map to start fresh, simply double twice on the delete button. Using drag and drop, just load your samples into the sample map editor. You probably don't want to use the graphical keyboard sample map editor unless you're very careful to align one sample per key. The text-based sample map editor will take care of this automatically. Please see your 's manual for more helpful information about loading samples and using the sample-map editor. Because of Scenario's advanced time-stretching algorithm, all samples must first be analyzed before they can be used. When you drag the samples into the sample-map editor, it will first analyze them for you. You will then be asked if you want to save the analysis data into the samples - it's recommended to do so, so you don't have to analyze them again the next time the live set is loaded.

Tips and Tricks There are many tricks you can use Scenario for. No one every said you had to use it live, either. You can also use it in the studio to create complete tracks with. Filter-splitting - create one track out of four. You can take advantage of Scenario's filters to combine several tracks into one. Like the bass drum from one track and the synth from another and the high hat from a third? Use Scenario to create track mutations. Simply load a different loop into each of Scenario's loop players in the Loop Engine, and tweak the filter settings to focus on a different frequency range in each loop. Make good quality loops. Loops have to be cut exactly to 1, 2, 4, 8, 16, etc bars in order for the beat looper to play them correctly. The loops can be at any tempo, however, and Scenario will automatically fit all loops to the same tempo. All common sound editing programs let you bounce audio with exact bar sizes, so check your manual for detailed information.


Use the two-band crossfader in the Scene Effects. The two-band crossfader is a well-kept secret for music played on a powerful PA system. By only effecting the high frequencies, the bass stays clear and doesn't get muddy. MIDI control. Any of the Scenario's parameters can be MIDI controlled. Due to its complex internal structure (using internal OSC communication between the three instruments), you can't use MIDI learn on the parameters, or they will not be ed by the Scene Memory any more. If your MIDI controller is programmable, such as a Peavey PC-1600 (www.peavey.com) or Bitsream (www.wave-idea.com), then it's better to program your MIDI controller instead of using MIDI learn. Saving live sets. Each live set should be saved as its own ensemble so it can be recalled with all samples and scene memories intact. Simply use Save As… and give your live set a name.


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Make good scenes, then freely improvise. For an effective live performance, it often pays to spend a lot of time making good scenes, and having a good flow between the scenes. You can copy and paste scenes in the Scene Memory simply by activating a scene by going to that scene with Flow enabled, then by turning off Flow and going to the position you want to write the scene to. Press Write, and you just copied and pasted a scene from one part to another. When performing, you can be assured that if any live improvisation gets out of hand, you can just go to the next scene and everything will be on track again.

Effects Analogic Filter Box

Instruments and Presets: Erik Wiegand; Design: Ian Warner The Analogic Filter Box sandwiches a rich and meaty distortion unit between two hearty analog-style filters to create a soundshaping tool for every appetite. Juicy modulation is also provided on the side: An envelope, LFO, and envelope follower can be freely routed to the most important filter and distortion parameters. Analogic Filter Box can handle everything from fat disco-loop tweaking to full-on mangling of any sound source imaginable.

Quick Start Select Preset 1 (Default) - this resets the Analogic Filter Box to its default settings, with the sound ing through the effect unchanged.* You must activate either filter 1, 2, or the distortion in order to hear the effect. To turn on filter 2 (the one after the distortion), for instance, simply select a filter type from the pulldown menu in the bottom-right corner of the Filter 2 macro. Now, by moving the Cutoff P and Reson knobs, you should immediately hear your freshly filtered sound. Filter 1 and Distortion are activated in the same way.


Structure and Signal Flow The signal is routed from input to Filter 1, to the Distortion, to Filter 2, and then out. Both filters offer multiple operation modes. For the first filter, you can choose between Low, High, Band, Peak EQ, and Notch filters, with a choice of a 12 or 24 dB/octave slope per filter type. The second filter is designed for shaping the sound after the distortion, so it offers four different low filters, three band filters, and a band/low combo. Both filters were very carefully designed to produce warm, analog sounds even at extreme resonance and cutoff settings. Beside the normal cutoff frequency control available in both filters, Filter 1 also provides fast modulation of its frequency by an additional oscillator (which can itself be modulated by the LFO, envelope, or envelope follower!).


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To get some motion into the sound, we'd want to take advantage of the Analogic Filter's copious modulation options. To modulate the cutoff of filter 2 by the LFO, for instance, select "Lfo" in the 2 P menu in the filter modulation section. Make sure that Filter 2 is activated - you may want to select Patch 1 (Default) again and activate Filter 2 by hand. Underneath the modulation source menu there's a small number box where you can click and drag to specify how much you want the LFO to modulation the cutoff of the second filter by. For a very obvious effect, just click and drag until the number box shows 100. Note that you can also enter negative numbers in the modulation amount if you want the LFO to decrease the filter frequency. One last step - you must activate the modulation by clicking on the small button next to the modulation amount. Now - assuming filter 2 is activated and audio is running through the effect - you should hear the LFO controlling the filter cutoff.

The Distortion section between the two multimode filters also features multiple modes of operation. While the clipper mode provides a relatively harsh distortion sound and the saturator mode results in warm overdrive, the several wrapping modes (marked by the name of the waveform used for wrapping) produce unique sounds from subtle to extreme. An additional quantize mode converts the incoming signal into a step waveform, for familiar bit-reduction effects to mimic the character of vintage samplers, for instance. A visual display of the distortion function helps to see what's going on inside.

Modulation Analogic Filter offers six modulation sources (A built-in LFO, envelope follower, enveloper, MIDI note pitch, modulation wheel, and pitch bend wheel). The modulation sources and the flexible matrix signal routing system at the bottom of this effect transforms it into an incredibly powerful machine. The modulation signal routing system provides a source selector for each parameter to be modified; among those modifiable parameters are the cutoff frequencies and resonance settings of both filters and the distortion amount and symmetry control of the distortion section. In addition to gate information from MIDI noteon events that can even be used to trigger the filter, you can also use the pitch and mod wheels as modulation sources. An internal LFO, envelope follower, and auto-trigger envelope can add movement to the sound without the need any external MIDI controllers. The LFO offers different waveforms and can also be synchronized to the global tempo or MIDI clock (the small Unit knob syncs the LFO to MIDI clock and sets the musical note-units that are shown under Freq). The Envelope Follower calculates its modulation amount from the incoming signal: At high levels there is a high modulation level, and at low levels it's low. The Interval knob controls the response time to fast level changes. Use the cutoff controls of the internal high and low filters to select a specific frequency band of the incoming signal to trigger


It's even possible to combine any two MIDI controllers to make one dependent on the other - for instance, to have the amount of LFO modulation dependent on MIDI pitch. You can define custom mix modulation combinations in the Define Mix 1 and 2 areas as the bottom of the instrument. * You can play a sound through the built-in loop-player, through the realtime audio inputs, or you can process audio in realtime by using Reaktor as an effect plugin. Please check your Reaktor or Reaktor Session 's guide for helpful information.

Banaan Electrique Amp Simulation

Instruments and Presets: Martijn Zwartjes; Design; Leonhard Lass


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the envelope follower. The Envelope is a standard attack-decaysustain-release envelope generator, triggered by MIDI note on events. However, an additional auto-trigger feature allows it to be triggered by the incoming audio, settable with the Tresh slider.

Banaan Electrique is a sophisticated guitar and bass amp simulator with built in effects. It's like having a pedal board full of vintage pedals and a vintage amp, with the added advantages of not needing to cable anything together or risk electrocuting yourself by running too much current through the amp. You can still get shocked by its lush sound, however. Naturally, Banaan is happy to eat whatever types of sounds you feed it, whether they're vocals, drum loops, synths, or scallops.

Getting Started Banaan Electrique was designed to be so easy that even guitar players can use it! Just play some audio through it and let a rip.* Check out the presets to see the variety of tones that Banaan can achieve.

Structure and Signal Flow The audio runs through Banaan Electrique linearly:

mono gate 3-band EQ Compressor 1 Amp Simu Phaser Rotor Dual Delay Reverb Compressor 2

Input lator

The incoming signal is converted to mono, amplified, gated, equalized and compressed to control the input's amplitude level and to enhance those sounds wanted to be in the recording, getting rid of any unwanted noise. Then, a guitar amplifier is simulated, including distortion, overdrive and filtering. Its output signal is sent to a rotor module, placing the mono sound in the stereo field dynamically. This signal is then routed to a phaser effect and, afterwards, to a stereo echo unit whose delay times can be synced to master tempo or MIDI clock. A high-quality reverb enhances the spatial sound once more, feeding its output into the final compressor. Not all modules have to be active at the same time - in fact, it's a good idea to turn off any modules that you're not using to save U. You can turn modules on and off with their respective Power buttons.


Classic Vocoder

Instrument and Presets: Lazyfish/Classic 2-VCO Synth: NI The Classic Vocoder was designed to faithfully emulate the wellknown tones of singing robots made popular in the seventies. The instrument combines a vocoding engine, a vintage-type synthesizer, and a four-band dynamics processor for a warm, smooth sound.

Quick Start The audio input is vocoded with the built-in synthesizer. Play some audio into the vocoder.* You don't have to sing into it - the Classic Vocoder also gives great results with drum loops or other sounds.

Structure and Signal Flow Audio input 1 is the modulator, and is vocoded with the Classic 2VCO synth, which is the carrier. The entire signal flow is mono. If you are using a stereo signal, only the left channel will be used. The output of the vocoder is fed into the four-band normalizer to smooth-out the sound and remove any uncomfortable signal peaks that could come with vocal sibilants or drum transients.


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The Vocoding Engine The vocoding engine consists only of four controls and a large graphic display of the frequency bands. Using an elegant function selector, you can adjust the important settings for the band filters that divide the carrier and modular into multiple frequency bands. The pitch, resonance, peak release, smoothness, and amplitude of the filters can be set. By dragging the vertical bar on the left side of the graphic display, you can choose which settings to adjust. For instance, to modify filter pitch, bring the bar up to filter pitch, and then the "offset" and "slope" knobs will adjust the filter pitch settings across the bands. The "dry/wet" and "car/mod" knobs are always active, no matter where the parameter selector is. Dry/wet mixes between the pure input signal and the vocoder output, while car/mod mixes between the audio input and the built-in synth.

Classic 2-VCO A simple analog-style synthesizer with two oscillators, one filter and filter envelope, one LFO, and one amplitude envelope. Each oscillator is limited to either a pulse or a sawtooth waveform.

Four Band Normalizer A four-band normalizer smooths out the sound before the output. The Split knob controls the filter frequencies that split the incoming audio into four bands - the approximate frequencies splits are shown on top of the instrument with the three horizontal meters. Each band provides four controls and before/after meters. Each band gives you control over: Limiter on/off (Lim), volume of the band (Amp), and the transient response of the limiter (Smth, and Rel).


Sonically, Vocoding uses the characteristics of one sound to control another. To achieve the popular robotic-singing effect, a voice (technically called the modulator) is vocoded with a constant sound, such as a synth or string sound (the carrier). The frequency content of the voice is split up into many different bands - the number of bands has an obvious impact on the sound, with fewer bands leading to more synthetic voices, and higher bands make the voice easier to understand. You can adjust the number of voices of the vocoder instrument to change how many bands are used. Up to 128 voices (bands) are possible. All changes in number of bands are immediately shown in the graphical display. The amplitude of each frequency band of the voice is linked to the frequency bands in the string or synth sound. The re-shaped bands of the carrier signal are mixed together, providing the output signal of the vocoder. If you're interested in vocoding, you may also want to do check out another NI product - the VOKATOR. The VOKATOR features vocoding up to 1024 bands, a built in synthesizer and granular sampler, and many high-end vocoding features. www.ni-vokator.com. * You can play a sound through the built-in loop-player, through the realtime audio inputs, or you can process audio in realtime by using Reaktor as an effect plugin. Please check your Reaktor or Reaktor Session 's guide for helpful information.

Echomania

Instrument and Presets: Erik Wiegand; Design: Ian Warner Reaktor Library English – 461

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What is Vocoding?

Echomania is an advanced and spectacular-sounding delay box that excels at tight, tempo-synced rhythms. It includes two LFOs to to modulate the delay time and built-in EQ. The delay time is handily synced to the global MIDI clock, for creating stretchy rhythmic effects. A drive and noise circuit combined with a offset control recreate vintage sounds. Get dubbing!

Quick Start Play some audio through Echomania.* Flip through the snapshots and discover the gritty (or crystalline) depths of Tapedelay.

Structure and Signal Flow The delay time is set in 16th note increments, synced to the global MIDI clock. The Unit and Divisor windows, to the right of the Time control, allow you to fine tune the beat division and create different rhythmic feels -- to be precise, the enumerator divided by the denominator scales the delay time fraction. There is also an LFO, with blendable sine/triangle/square/slow random waveshapes that is hardwired to modify delay time. You can set the amount of time modulation with the Amount knob. The Freq knob controls the rate of the LFO in 16th notes. The Unit knob divides the speed of the global clock by the Unit amount. For example, if the Unit knob is set to "6", then the LFO Freq amount will be in 16th notes. The Width control morphs between sine and pulse-like, or, if the Tri button is engaged, triangle and saw tooth waveshapes. The starting phase of the LFO can be adjusted positively or negatively using the Phse knob. If you engage the Snc button, the right-side LFO can be phase-offset from the left side by the amount set with the Right knob. This can create a variety of woozy stereo spinning and phasing effects.


Tapedelay's module provides an offset control, labeled FB > 1, which boosts and shapes the feedbacl signal, making it seem to get louder and louder (but without degenerating into uncontrollable noise). The Mixer lets you add a tape saturation-like Dist Drive effect and Noise to give everything that just-pulled-out-of-the-closet feeling. The Dry Wet section allows you to balance the amount of dry and delayed signal. You can also use the Tap buttons to select whether the delay tap comes before or after the saturation/noise circuit. * You can play a sound through the built-in loop-player, through the realtime audio inputs, or you can process audio in realtime by using Reaktor as an effect plugin. Please check your Reaktor or Reaktor Session 's guide for helpful information.

Flatblaster Multiband Dynamics

Instrument: Martijn Zwartjes, Presets: Jaap Wajer; Design: Ian Warner


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The EQ module processes the delayed signal. It is essentially a parametric EQ that contains an LFO identical to the one in the Delay module. You can create synchronized filter sweeps, fizzing hi-frequency delay tails, and all manner of dubby effects by boosting and modulating select frequency bands. The Eq Res control lets you dial in the peak width of the frequency, while EQ Boost lets you crank it up.

Flatblaster is a high-end finalizing and multiband dynamic shaping tool. Flatblaster combines four frequency-specific compressors with a full-spectrum peak-limiter. It is an excellent finalstep mastering plug-in, but it can also be used while mixing since it doesn't introduce any delay to the signal. Each of the compressors has a saturator, so you could saturate just the upper-mid frequencies, for instance, without muddying the bass. It also makes an excellent de-esser and sibilant reducer.

Quick Start Even though Flatblaster gives control over many sound-shaping parameters, there's no need to be intimidated by its complexity. A full range of presets shows off its capabilities and gives good starting points for tweaking the effect for your sound. Simply play some audio through Flatblaster* and step throught the presets. Try experimenting with muting, soloing, and bying each individual band so you can carefully hear what the 'blaster's doing. Be careful when adjusting the saturation of each band! The sound can potentially get very loud if you don't first reduce the make-up gain (the knob labeled Gain to the right of the Sat knob).

Structure and Signal Flow The input signal is divided into four bands - the three crossover frequencies are adjustable independently. Each frequency band is processed by an independent, identical compressor. Each band can be muted, soloed, and byed (no compression). The signal is summed before going through a full-band peak limiter, which can also be independently byed. The master by for the effect is located to the right of the input meters, above the crossover settings.

Frequency-Specific Compressor Each of the four compressors are absolutely identical. In fact, they have to be! If they weren't, then unwanted phase shifts could creep in. Each compressor gives control over saturation, saturation makeup gain, threshold, compression ratio, adjustable knee, Reaktor Library English – 464

Full-Band Peak Limiter The peak limiter effects the full frequency range of the audio, after each of the four frequency bands has been compressed separately. The Threshold slider controls when the peak limiter will start working. With Threshold at 0, the peak limiter will have no effect. For mastering, it's recommended to have the Threshold set to around -3 or -4 dB. Severe Threshold settings will lead to pumping, which may or may not be desirable. The Attack and Release knobs control how the peak limiter responds to transient signals. The final Peak slider sets the output level of the signal. When Peak is set to its maximum 0 dB, then the audio will be as loud as possible. There's really no reason to ever set Peak lower than this, unless you needed to ensure a certain amount of headroom. * You can play a sound through the built-in loop-player, through the realtime audio inputs, or you can process audio in realtime by using Reaktor as an effect plugin. Please check your Reaktor or Reaktor Session 's guide for helpful information.


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attack, release, and output makeup gain. Note that the Ratio has to be higher than 0 for the compressor to have any effect - at a Ration of 1 (maximum), the compressor acts as a limiter. The red meters show the amount of peak reduction. The Attack and Release knobs control how the compressor responds to transient signals.

Fusion Reflections

Instrument and Presets: Erik Wiegand; Design: Ian Warner Fusion Reflections is a delay-based effect that can create early reflections, shimmering choruses, fluttering delays, and even ambient reverbs. Two distinct diffusion engines are chained together to create an extremely wide range of effects. Each finelytuned diffusion engine consists of four stereo modulation delays and an innovative graphical display that shows the actual delay time for each delay. Just five controls control the core parameters of each diffusion engine.

Quick Start Play some audio through the effect* and select Preset 1 Long Decay Echo. This preset only uses the Chor Fusion diffusion engine. Play with the Diff Dly knob - note how the graphic display constantly updates to show you the current delay times. Stop the incoming audio to listen to the sound's decay. Play with Dly Mod and Speed to see what effect they have on the sound and the graphic. Be sure to check out all the other presets to see how versatile this effect can be!


The sound es through two diffusion engines serially. It's important to note that the two engines are similar but not identical they each offer unique sound-shaping capabilities. The first diff, Chor Fusion, is the simpler of the two and is designed for early reflections, choruses, and atmosphere. If offers a high and low shelf EQ, with a graphical display of the EQ curve. The second diffusion engine, Echo Fusion, adds a delay before the diffusion delays. The diffusion delays in the Chor and Echo engines are identical. Echo Fusion is perfectly tailored to late reflections, long delays, and long reverbs. A high filter (HP) cuts the low frequencies after the delay and before the diffusors, while a low filter (LP) can reduce the brightness at the last step. The input to the Echo Fusion engine can be switched between the dry signal, and the signal coming out of Chor Fusion. In the Mixer section the signals of Chor Fusion> and Echo Fusion are combined and mixed with the dry signal. Each section can also be switched on or off to save U.

Diffusion Delays Both the Chor and Echo Fusion engines contain identical diffusion delays. As mentioned above, Echo fusion adds a single feeback delay before the diffusion delays, but the operation of the diffusion delay section is identical per effect. Each diffusion delay consists of four stereo modulation delays and an innovative graphical display that shows the actual delay time of each delay. The main delay time is controlled by the Diff Dly knob. The Dly Mod knob controls the amount of delay time modulation by the internal LFOs. Speed controls the speed of the LFO. Stereo sets the stereo spread of the delays, and Diffusion sets the inaccuracy of the delay times. The sum effect of these five knobs is graphically shown in the display underneath, where each "pendulum" represents the delay time of a single delay. * You can play a sound through the built-in loop-player, through the realtime audio inputs, or you can process audio in realtime by using Reaktor as an effect plugin. Please check your Reaktor or Reaktor Session 's guide for helpful information.


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Structure and Signal Flow

SpaceMaster

Instrument and Presets: Martijn Zwartjes; Design: Leonhard Hass The SpaceMaster series of reverbs breaks new ground in reverb modelers for Reaktor. Spacemaster uses two different Diffusion modules to achieve stunningly convincing room sounds. And to fully exploit SpaceMaster's lushness, there are three versions SpaceMaster stereo, SpaceMaster Quad, and SpaceMaster 5.1 Surround.

Quick Start To really get a feel for the kinds of lush atmospherics SpaceMaster can provide, it's a good idea to either hook a beat looper or external sound source up to it, or, to utilize it in plugin mode in your favorite audio sequencer. Stepping through the presets should give you a good impression of the kinds of real and imaginary spaces SpaceMaster can emulate. Adjusting the controls of the Early and Late Diffusion modules will have the most effect on the sound, and will give you an idea of how the two main components interact in creating ambiences - especially since they can be arranged in serial or parallel signal paths.

Structure and Signal Flow This guide will use the Stereo SpaceMaster ensemble to outline the various controls for shaping the reverb signal, since most of them are the same among the three types of SpaceMasters. Read on below to learn about the specifics of the Quad and Surround flavors of SpaceMaster reverb.


SpaceMaster Stereo is composed of a PreDelay module, an Early Diffusion module, a Late Diffusion module, and a post EQ. The Diffusion modules can be combined together to create a complex impression of space. By adjusting the balance of the Early and Late diffusion modules, you can precisely move the origin of the reflections from near to far (or front to back), making SpaceMaster perfect for Surround mixing situations in which a truly roomfilling reverb can be created. The Input section, at the fat left, allows you to trim the input gain to avoid overloading the audio signal. The input is next processed by the PreDelay. Use this to add an initial delay to the wet signal. You can also use the L/R offset knob to add some perceived stereo width by altering the delay time of the left-side single delay module. You can by the PreDelay with the Byps button. Next in the signal path, the Early Diffusion module, which is actually a series of up to 12 diffusion delays, provides the near reverb processing. The Size control allows you to determine the range of space that the close reflections will be generated by. It changes delay time in milliseconds. The Diffusion knob lets you adjust the density of the reverb signal. To further adjust the reverb depth of the Early Diffusion module, the Mode switch allows you to select 6 or 12 diffusion modules. Watch your U load carefully to make sure your computer can handle the strain of processing with 12 (or 24) diffusion modules. The Damp knob controls the frequency of a 1-pole low filter for attenuating high frequencies. As in the PreDelay section, clicking the Byps button will take the Early Diffusion module out of the signal path. The Routing switch, located between the Early and Late Diffusion modules, lets you determine how the input signal will be routed through the two modules. The Ser button engages Serial mode, where the Early Diffusion module is simply routed directly into the Late module. The Par switch engages Parallel mode, allowing the Early and Late modules to maintain separate signal paths, until they are mixed at the EQ module. You can also use the Early/Late


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SpaceMaster Stereo:

knob to balance the amount of early and late reverb signal. This is a great way to change the perception of location within a space, by shifting between the early and late reverb sounds (to make it seem like a sound is moving around inside of the "room"). The Late Diffusion module provides the capabilities for creating a larger and more richly defined space. The Size and Diffusion knobs work the same way as those in the Early section, but there a few new options. The RT knob controls a loop, allowing you to stretch out the apparent reverb time, or the time it takes for the echo to return to the point of origin. There are also controls for high and low shelving EQs, called Hi Damp and Lo Damp, respectively, that allow you to shape the frequency range of the reverb signal. You can modulate the delay time and apparent position of the reverb signal by using the modulation controls. The Spin knob adjusts the amplitude of a sine wave LFO, while the Dizzy knob controls amplitude for a Slow Random LFO. You can create complex modulations by adjusting the balance of sine wave and random LFOs. The Freq knob controls the rate of both LFOs together. As in the Early section, 6 or 12 diffusion modules can be selected with the Mode switch. This section can also be byed with the Byps button. Both the Early and Late Diffusion sections feed into the EQ. The EQ consists of stereo low shelf, parametric, and high shelf filters. Starting at the left, the Lo knob lets you control attenuation or boost of the low frequency set with the Frq immediately above. The Mid knob controls the parametric EQ band. The Frq and Q knobs above it let you adjust the frequency and bandwidth of the parametric. The Hi knob controls attenuation of the high shelf filter. Use the Frq immediately above the Hi knob to adjust the frequency. As in the other modules, the EQ can be taken offline with the Byps button. Finally, the mixture of wet and dry signal can be adjusted to taste with the Mix knob in the output section.


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SpaceMaster Quad

Instrument and Presets: Martijn Zwartjes; Design: Leonhard Hass The SpaceMaster Quad ensemble works in the same way as the stereo version, but it splits the reverb signal into four discrete outputs for using in a four-speaker system. The signal for the forward speakers is referred to as the Front signal, while the signal for the rear speakers is referred to as the Surround, or Sur, signal. In the Input section, you can adjust the gain for Front and Sur signals to determine how much signal sources reaches each signal chain. The PreDelay section is simplified from its stereo brother by use of an XY pad to finely adjust Front and Surround L/R offset. The overall PreDelay time can be controlled by the Time fader. The Early Diffusion section is also simplified, offering only Size and high frequency Cut controls for both Front and Surround diffusion signals. The Late Diffusion module is effectively the same as the one found in SpaceMaster Stereo. Refer to the description above the learn how it works. The Mix section allows you to balance the Front and Surround signal amount and to adjust the wet/dry mix for each.


SpaceMaster 5.1 Surround

Instrument and Presets: Martijn Zwartjes; Design: Leonhard Hass The Surround version of SpaceMaster is effectively the same as the Quad version, except that it adds a Center channel. The PreDelay XY pad adds control for the center channel. SpaceMaster Surround also adds an Output Gain section that allows you to use precise metering to control the relative volumes of the Center, Sub, Front, and Surround reverb signals. The balance of Early/ Late signal can also be adjusted for each channel using the Mix module, which has familiar controls for wet/dry levels. Finally, SpaceMaster Surround does not include the EQ module found in the Stereo and Quad versions.

Spring Tank

Instrument and Presets: Tim Schwerdtfeger


Quick Start Spring Tank is meant to be an experimental effect, rather than a realistic one, play some audio through the effect* and experiment! It allows control over the elements of the "spring's" morphology, so you can design your own spring.

Structure and Signal Flow The input source is amplified or attenuated with the Level knob at the far left of the instrument. High levels here can introduce transducer saturation (which is not such a bad thing). The signal is then fed into the spring tank. Here, you can adjust the spring's physical characteristics. The Damp knob sets high frequency damping for the spring. Turning the knob clockwise increases the damping. Stiffness, which mimics the flexibility of the spring, gives brighter, more resonant sounds at higher settings. The Shape knob allows you to crossfade between a round spring shape to the left, and a rectangular one to the right. The round shape emphasized a ringing sound, while the rectangular shape creates less ringing but a more diffuse sound. The Thickness of the spring determines the virtual diameter of spring winding, and therefore the overall length of the spring. Longer spring settings result in reduced brightness. The Length setting reflects global length. Turn this up to increase the shattering decay sounds. The Decay knob changes the length of the decays. It alters the amount of in the system. Switching the Mono button off engages an additional short delay in one channel, resulting in the simulation of stereo.


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Spring Tank is not your typical room simulator - instead, it pays homage to the trashy, unrealistic spring reverbs of the past. While Spring Tank isn't exactly a physical model of a spring, it goes a long way toward recreating the spring reverb characteristics: dull, transducer-saturated, and boingy, with the familiar nonlinear resonating decay.

The Suspension section allows you to mimic the type of spring suspension in the system. When you turn the knob to the left, you increase the "softness" of the suspension material. This will lengthen the decay rate of lower frequencies. What this is actually doing is adjusting the cutoff of a high filter in the loop. Suspension will therefore also have an effect on the sound of the Transducer saturation. The Color knob lets you adjust a global tone control. Turning this clockwise results in a brighter sound, and vice versa. The Level knob controls the volume of the wet signal. You can adjust the balance of wet signal to dry signal with the Mix knob. Finally, you can by the effect all together by clicking the On button. * You can play a sound through the built-in loop-player, through the realtime audio inputs, or you can process audio in realtime by using Reaktor as an effect plugin. Please check your Reaktor or Reaktor Session 's guide for helpful information.

Two Knees Compressor

Instrument and Presets: Erik Wiegand; Design: Ian Warner Two Knees Compressor is a simple compressor with an important quirk - it has two separate adjustable thresholds and ratio controls. 2-Knees Compressor can perform as a precisely accurate compressor/limiter, a distorting sound shaper, a transient modifier, or anything in between. It includes a compression curve display to show the relationship between the amplitude of the input signal and the amplitude of the processed signal.


Play some audio through Two Knees*. Try to input a beat and run through the snapshots to hear a clear example of how it can alter transients.

Structure and Signal Flow Set the input level with the In fader to achieve 0 dB on the input meters. The top set of Threshold and Ratio knobs refer to the upper threshold of the compressor. The upper compression ration is applied if the amplitude of the input signal exceeds the upper threshold. The ratio determines how many dB the signal must exceed the threshold to result in a 1dB increase of the output signal. The lower threshold and ration controls are directly beneath the upper controls. The lower compression ratio is applied if the amplitude signal is within the range between the upper and lower thresholds. The Thresh setting shows up on the display just below the compression curve display. The upper threshold setting will be on the right side of the display and the lower will appear on the left. This will help you to dial in your threshold range to get particular hardknee or soft-knee sounds. Try adjusting the degree of separation between the upper and lower thresholds and watching the display to see where the compression crosses from sharp-angled hardknee style to rounded soft-knee. You can adjust the attack time of the compression in milliseconds using the Attack knob. The Attack setting determines how fast the compressor responds to (turns down, basically) a signal that exceeds the threshold limits. Slower attack times tend to let more transients through. The Release knob lets you set how fast the compressor returns to unity gain (or zero gain) after falling below the threshold limits. You can manipulate the apparent sustain time of some sounds with the Threshold adjustment. Long release times tend to sound more "natural". Finally, you can make up for gain reduction after compression by adjusting the Out level. You can also by the compressor by turning off the On switch.


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Quick Start



Symbols ? Menu 72

Numerics 1 / Square Root 243 1/96 Clock 92, 234 12-Step 296 16-Step 296 2D Table Default 102 2D Table Max 102 2D Table Min 102 4-Ramp 269, 312 4-Step 268 5-Ramp 313 6-Ramp 315 6-Step 295 8-Step 296

A a * b + c 237 A to E 369 A to E (Perm) 370 A to E (Trig) 370 A to Gate 371 About 72 Accumulator 351 Activate MIDI In 91 Activate OSC 26 AD - Env 305 ADBDR - Env 308 ADBDSR-Env 309 Add 235 ADSR - Env 306, 307, 310, 311 Akai Import 163 All Controls Visible in Ensemble 100 All 1-Pole 318 Alpha Channel 144 Amp 249 Animation 143 Animation Height 144 Animation Width 144

Appearance page 89, 101 Append 148 AR - Env 305 Arccos 245 Arcsin 244 Arctan 245 ASIO 5 Audio + MIDI Settings 63 Audio Modifier 339 Audio signals 126 Audio Smoother 372 Audio Table 349 Audio Units 6 Audio Voice Combiner 368 Audio-In Module 84 Audio-In module 33 Audio-Out Module 85 Audio-Out module 33 Auto Save 21 Auto Voice Reduction 99 Automatic Layout 60 Automatic Layout 90 Automatic Voice Reduction 64 Automation ID 20 Auxiliary 364

B Background Bitmap 103 Bank 150 base ID 20 Basic Operation 51 Batch Processing 56 Beat Loop 291 Beatloop 155 Bi-Pulse 265 Bi-Saw 253 BMP 144 Browser 76 Button 132, 137, 210

C Ch. Aftertouch 227, 232 Choose Color 90, 102 Chopper 343

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Index

Classic Modular 178 Clipper 341 Clock Oscillator 271 Clock Start 60 Clock Stop 61 Clock Sync 27 Color Scheme 90 Color scheme 102 Compare 148, 239 Compare/Equal 240 Compress 20 Connection 134 Connection page 91, 104 Constant 235 Constant Sources 118 Cont/Note 134 Context menu 52 Control Rate 59 Control Sources 116 Controller 134, 227, 232 Copy 57 Core Audio 6 Counter 351 U Usage 64 Create backup file before saving 63 Create Constant 114 Create Control 113 Creating Instruments 93 Creating Macros 108 Crossfade 247 Ctrl. Shaper 1 BP 353 Ctrl. Shaper 2 BP 353 Ctrl. Shaper 3 BP 354 Cubase 7 Customized s 143 Custom-made control 140 Cut 57

D D - Env 301 DBDR - Env 303 Default 148

Delay 332 Delete 58 Differentiator 331 Diff Delay 334 Directories 65 DirectSound 6 Disable Automation 135 Display 180 Distributor 248 Divide x/y 237 Double-click 52 DR - Env 301 Drop-down Menu 138 Drum-Maps 157 DSR - Env 302 DXi 13 DXI 2 6 Dynamic ports 206

E Edit Menu 57 Ensemble 83 Ensemble Properties 86 Ensemble Structure 84 Ensemble window 85 Envelope 298 Event Loop Prevention 126 Event Processing 350 Event Signals 124 Event Smoother 373 Event Table 362 Event V.C. All 368 Event V.C. Max 368 Event V.C. Min 369 Exit 56 Expon. (A) 241 Expon. (F) 241 External Sync 60, 92

F Fader 131, 137, 207 Favorites 76 File Menu 54 File missing 154

G Gate 138, 211, 225 Geiger 272 Grain Cloud 289 Grain Cloud Delay 337 Grain Delay 336 Graph BG 102 Graph Fill 102 Graph Line 102 Grid 102

H H - Env 300 High Shelf EQ 329 High Shelf EQ FM 330 Hints 122 Hold 361 Hold Control 105 Horizontal 144 HP/LP 1-Pole 317 HP/LP 1-Pole FM 317 HR - Env 300 Hybrid modules 205

I Ignore Tempo Change 55, 61 Import MIDI File 55 Impulse 266 Impulse FM 266 Impulse Sync 267 In Port 380 Incremental 133 Indicator 102 Info page 89, 101 Insert 148 Insert Menu 59 Instrument 93 Instrument Header 95 Instrument Properties 97

Integrator 332 Invert 236 Iteration 357

K Key Control 140 Knob 131, 137, 210

L Ladder Filter 326 Ladder Filter FM 327 Lamp 218 Level 88 Level Lamp 218 Level Meter 221 LFO 32, 298 Local Identifier 27 Local IP Address 27 Local Port 27 Log (A) 241 Log (F) 242 Logic 10 Logic AND 355 Logic EXOR 355 Logic NOT 356 Logic OR 355 Loop editor 163 loop length 274 Loop MIDI File 61 Low Shelf EQ 330 Low Shelf EQ FM 331 Lower Limit 118 Lower Note Limit 105

M Macro Collection 178 Master Tune/Level 373 Math 235 Max 118, 131 Max Automation ID 135 Max Unison V 99 Maximum processor usage 64 Measure U Usage 61 Merge 358 Meter 220

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Filter 316 First Steps 30 Frequency Divider 348, 352 From Voice 372 Function page 87, 97

MIDI Channel Info 376 MIDI Clock Out 60 MIDI In 224 MIDI In Device 91 MIDI Learn 59, 74, 141 MIDI Note 134 MIDI Out 231 MIDI Out Device 92 MIDI Sources 117 Min 118, 131 Min Unison V 99 Minimize 96 Mirror 1 Level 342 Mirror 2 Levels 342 Mixer 249 MME 6 Mod. Clipper 341 Modulo 238 Mono 58, 114 Moving 51 MRPH Time 152 Multi 2-Pole 318 Multi 2-Pole FM 319 Multi Picture 222 Multi Text 223 Multi/HP 4-Pole 324 Multi/HP 4-Pole FM 325 Multi/LP 4-Pole 322 Multi/LP 4-Pole FM 323 Multi/Notch 2-Pole 320 Multi/Notch 2-Pole FM 321 Multi-Picture 144 Multiplex 16 297 multiplier 236 Multiply 236 Multi-Ramp 269 Multi-Sampling 156 Multi-Sine 260 Multi-Step 268 Multi-Tap Delay 334 Mute 58, 94, 95, 114 Mute All 88

Mute Port 114

N New Ensemble 54 Noise 271 Note Pitch 224 Note Pitch/Gate 231 Note Range Info 375 Note Shift 105 Nuendo 8 Num Animations 103, 144 Number of Undos 63

O Off Velocity 226 On Velocity 226 on/off switch 360 Open 54 Open Sound Control (OSC) 25 Options 63 Order 356 OSC Memeber list 28 OSC Monitor 29 OSC Receive 383 OSC Send 382 OSC Settings 63, 106 OSC Source 134 OSC Synchronization 27 OSC system setup 26 OSC Target 106, 134 Oscillator 250 oscillator mode 274 Out Port 380 Overwrite 148

P 131 Color 90 Edit 95 Editing 130 Lock 95 Operation 137 to MIDI 133 Panner 248 Par FM 256

Q Quantize 240

R Random 272 Random Isolate 151 Randomize 151 Randomizer 352 Range of Values 117 Reassign 100 Recall by MIDI 88, 100, 147 Recall by Parent 100, 147 Receive 381 Receive Channel 91

Receive External MIDI 133 Recent Ensembles 56 Reciprocal 237 Recorder Settings 69 Recorderbox 68 Rect./Sign 239 Rectifier 238 Redo 57 relay 360 Relay 1,2 247 Reload last ensemble at start-up 63 Remap to single key 162 Remote to MIDI 134 Resizability 143 Resizeability 145 Restore Browser State from Ensemble 64 Resynth 155 Right mouse button 52 Rnd Amt 151 Root Key 157 Router 1,2 359 Router 1->M 360 Router M->1 359 RTAS 6, 14 Run/Stop Audio 61

S Sample & Hold 347 Sample Lookup 293 Sample Management 153 Sample Map Editor 158 Sample Rate 59, 74 Sample-Analysis 155 Sample-Maps 156 Sampler 275 Sampler FM 276 Sampler Loop 277 Samples 153 Saturator 339 Saturator 2 340 Save Ensemble 54

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Par PWM 258 Par Sync 257 Parabol 256 Parametric equalizer 328, 329 Paste 58 Peak Detector 347 Peak EQ 328 Peak EQ FM 328 Picture 144, 221 Picture Index 103 Picture Properties 143 Pitch Former 155 Pitchbend 224, 231 Play MIDI File 55, 61 Playerbox 67 Plugin Size 24 Poly Aftertouch 134, 228 Ports 94, 108 Power x y 242 Preferences 63 Preview 143 Pro-52 Filter 326 Program Change 228, 233 Protection 74 Pulse 262 Pulse 1-ramp 264 Pulse 2-ramp 265 Pulse FM 262 Pulse Sync 263

Save Ensemble As 54 Save Instrument As 94 Saw FM 251 Saw Pulse 252 Saw Sync 251 Sawtooth 250 Scanner 246 Scope 217 Screen-sets 66 Sel. Note Gate 226 Sel. Poly AT 228, 233 Select A 152 Select All 58 Select B 152 Select Picture 143 Selection 51 Selector 246 Send 381 Send Channel 92 Separator 357 Sequencer 294, 295 Set By Trg 160 Set Protected 60 Set Random 379 Set to Custom 102 Set transpose to zero 162 Set Unprotected 60 Settings Menu 59 Shaper 1 BP 343 Shaper 2 BP 344 Shaper 3 BP 344 Shaper Cubic 346 Shaper Parabolic 345 Show Hints 74 Show Label 131 Show Sorting 62 Show Value 131 Signal Path 246 Signal Processing 123 Sine 243, 258 Sine FM 259 Sine Sync 259

Sine/Cos 244 Single Delay 332 Single Trig. Gate 225 Slew Limiter 346 Slow Random 299 Small Design 131 Snap Isolate 150 Snap Value 380 Snapshot 96, 376 Snapshot ID 147 Snapshot Morphing 152 Snapshots 146 Soft Takeover 133 Solo 94, 95 Sonar 13 Song Pos 230, 234 Sort and compress 20 Sources 116 Spin Control 139 Square Root 243 Standalone Version 16 Start/Stop 229, 234 Step 131 Step Filter 359 Stepsize 118 Stereo Amp 249 Stereo Pan 248 Store by Parent 100 Store Map with Ensemble 155, 164 Structure 46 Structure Icon 103 Structures 110 Subtract 236 Sustain Control 105 Switch 132, 138, 213 Switches 119 Sync Clock 92, 230 System Info 375 System Menu 61

T Table Modules 165 Tapedeck 1-Ch 364

U Undo 57 Unison 96 Unison mode 99 Unison Spread 379 Unit Delay 339 Upper Limit 118 Upper Note Limit 105

V Value 358 View 66 Visible 103 Voice Allocation 99 Voice Assign 99 Voice Combiner 94 Voice Info 374 Voices 96 VST 2.0 Plug-in 7

W Wire 39 Wires 51, 120

X XY 215 XY Field 139

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Tapedeck 2-Ch 367 Targa 144 Tempo 74, 374 Tempo Info 374 Terminal 380 Terminals 94, 119 Text 223 Text box 139 TGA 144 Time Sync 27 Timer 360 To Voice 371 Toggle 138, 211 Toolbar 73 Total Recall 20 Transparency 143 Tri FM 254 Tri Sync 254 Tri/Par Symm 255 Triangle 253 Trigger 137, 211 Tune 88 Tuning Info 374

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