Steinberg Cubase 4.0 Plug-in Reference

Plug-in Reference

Original manual by Anders Nordmark

Revision: Cristina Bachmann, Heiko Bischoff, Marion Bröer, Sabine Pfeifer

The information in this document is subject to change without notice and does not represent a commitment on the part of Steinberg Media Technologies GmbH. The software described by this document is subject to a License Agreement and may not be copied to other media except as specifically allowed in the License Agreement. No part of this publication may be copied, reproduced or otherwise transmitted or recorded, for any purpose, without prior written permission by Steinberg Media Technologies GmbH.

All product and company names are ™ or ® trademarks of their respective owners. Windows XP is a trademark of Microsoft Corporation. Windows Vista is either a registered trademark or trademark of Microsoft Corporation in the United States and/or other countries. The Mac logo is a trademark used under license. Macintosh and Power Macintosh are registered trademarks.

Release Date: October 19, 2007

Table of Contents

5 The included effect plug-ins

6 Introduction 6 Delay plug-ins 9 Distortion plug-ins 10 Dynamics plug-ins 19 EQ plug-ins 21 Filter plug-ins 25 Mastering – UV 22 HR (Cubase only) 25 Modulation plug-ins 32 Other plug-ins 34 Restoration plug-ins 34 Reverb plug-ins 36 Spatial plug-ins 37 Surround plug-ins (Cubase only) 37 Tools plug-ins

41 The included VST Instruments

42 Introduction 42 Prologue 42 Sound parameters 47 Modulation and controllers 52 Spector (Cubase only) 52 Sound parameters 54 Modulation and controllers 60 Mystic (Cubase only) 60 Sound parameters 62 Modulation and controllers 67 HALionOne 69 Embracer – Surround Pad Synthesizer

(Cubase only)

71 Monologue – Monophonic Analog Modeling

Synthesizer (Cubase only)

73 Diagrams

75 MIDI effects

76 Introduction 76 Arpache 5 77 Arpache SX 78 Autopan 79 Chorder 80 Compress 81 Context Gate 82 Density 82 Micro Tuner 82 MIDIControl 83 MIDIEcho 84 Note to CC 85 Quantizer 85 Step Designer 87 Track Control 89 Track FX 89 Transformer

90 Index

Table of Contents

The included effect plug-ins

Introduction

This chapter contains descriptions of the included plug-in effects and their parameters.

In Cubase, the plug-in effects are arranged in a number of different categories. This chapter is arranged in the same fashion, with the plug-ins listed in separate sections for each effect category.

Ö Most of the included effects are compatible with VST3, this is indicated by an icon in front of the name of the plug-in as displayed in plug-in selection menus (for further information, see the chapter “Audio Effects” in the Operation Manual).

Delay plug-ins

This section contains descriptions of the plug-ins in the “Delay” category.

ModMachine (Cubase only)

ModMachine combines delay modulation and filter frequency/resonance modulation and can provide many interesting modulation effects. It also features a Drive parameter for distortion effects.

The parameters are as follows:

Parameter Description

Delay This is where you specify the base note value for the de-

Tempo sync Delay on/off

Rate The Rate parameter sets the base note value for tempo

Tempo sync Rate on/off

Width This sets the amount of delay pitch modulation. Note that

Feedback This sets the number of repeats for the delay.

Drive This parameter adds distortion to the feeback loop. The

Mix Sets the level balance between the dry signal and the ef-

Nudge Clicking the Nudge button once will momentarily speed

Signal path graphic

Output/Loop The Filter can either be placed in the feedback loop of the

Filter type This toggle button allows you to select a filter type. Low-

Freq This sets the cutoff frequency for the filter. This is avail-

Speed This sets the speed of the filter frequency LFO modula-

lay if tempo sync is on (1/1–1/32, straight, triplet or dotted). If tempo sync is off, the delay time can be set freely in milliseconds.

The button below the Delay knob turns tempo sync for the delay parameter on or off. If set to off, the delay time can be set freely with the Delay knob.

syncing the delay modulation (1/1 to 1/32, straight, triplet or dotted). If tempo sync is off, the rate can be set freely with the Rate knob.

The button below the Rate knob turns tempo sync for the rate parameter on or off. If set to off, the rate can be set freely with the Rate knob.

although the modulation affects the delay time, the sound is mostly perceived as a vibrato or chorus-like effect.

longer the Feedback, the more the delay repeats become distorted over time.

fect. If ModMachine is used as a send effect, this should be set to maximum (100%) as you can control the dry/effect balance with the send.

up the audio coming into the plug-in, simulating an analog tape nudge type sound effect.

You can click on the Filter sections displayed in the graphic in the center of the plug-in to place the Filter section either before or after the Drive and Feedback parameters in the signal path.

delay or in its output path (see above).

pass/bandpass/hipass filter types are available.

able only, if filter frequency LFO tempo sync is deactivated and the Speed parameter (see below) is set to “0".

tion. If tempo sync is activated the Speed parameter sets the base note value for tempo syncing the modulation (1/ 1 to 1/32, straight, triplet or dotted). If tempo sync is off, the rate can be set freely with the Speed knob.

The included effect plug-ins

Parameter Description

Range Lo/Hi These knobs specify the range (in Hz) of the filter fre-

Spatial This introduces an offset between the channels to create

Q-Factor This controls the resonance of the filter. This is available

Speed This sets the speed of the filter resonance LFO modula-

Range Lo/Hi These knobs specify the range of filter resonance modu-

Spatial This introduces an offset between the channels to create

quency modulation. Both positive (e.g. Lo set to 50 and Hi set to 10000) and negative (e.g. Lo set to 5000 and Hi set to 500) ranges can be set. If tempo sync is off and the Speed is set to zero, these parameters are inactive and the filter frequency is instead controlled by the Freq parameter.

a stereo panorama effect for the filter frequency modulation. Turn clockwise for a more pronounced stereo effect.

only, if filter resonance LFO tempo sync is deactivated and the Speed parameter (see below) is set to “0". If tempo sync is on, the resonance is controlled by the Speed and Range parameters.

tion. If tempo sync is activated, the Speed parameter sets the base note value for tempo syncing the modulation (1/1 to 1/32, straight, triplet or dotted). If tempo sync is off, the rate can be set freely with the Speed knob.

lation. Both positive (e.g. Lo set to 50 and Hi set to 100) and negative (e.g. Lo set to 100 and Hi set to 50) ranges can be set. If tempo sync is off and the Speed is set to zero, these parameters are inactive and the filter resonance is controlled by the Q-Factor parameter instead.

a stereo panorama effect for the filter resonance modulation. Turn clockwise for a more pronounced stereo effect.

MonoDelay

This is a mono delay effect that can either be tempo-based or use freely specified delay time settings. The delay can also be controlled from another signal source via the SideChain input.

The parameters are as follows:

Parameter Description

Delay This is where you specify the base note value for the delay

Tempo sync on/off

Feedback This sets the number of repeats for the delay.

Filter Lo This filter affects the feedback loop of the effect signal

Filter Hi This filter affects the feedback loop of the effect signal

Mix Sets the level balance between the dry signal and the ef-

Side-Chain on/off

if tempo sync is on (1/1–1/32, straight, triplet or dotted). If tempo sync is off, it sets the delay time in milliseconds.

The button below the Delay Time knob is used to turn tempo sync on or off. If set to off, the delay time can be set freely with the Delay Time knob, without sync to tempo.

and allows you to roll off low frequencies from 10Hz up to 800Hz. The button below the knob activates/deactivates the filter.

and allows you to roll off high frequencies from 20kHz down to 1.2kHz. The button below the knob activates/ deactivates the filter.

fect. If MonoDelay is used as a send effect, this should be set to maximum as you can control the dry/effect balance with the send.

When this is activated, the delay can be controlled by a signal routed to the Side-Chain input. When the sidechain signal exceeds the threshhold the delay repeats are silenced. When the signal drops below the threshold the delay repeats reappear. For a description of how to set up Side-Chain routing, see the chapter “Audio effects” in the Operation Manual.

The included effect plug-ins

PingPongDelay

This is a stereo delay effect that alternates each delay repeat between the left and right channels. The effect can either be tempo-based or use freely specified delay time settings.

The parameters are as follows:

Parameter Description

Delay This is where you specify the base note value for the delay

Tempo sync on/off

Feedback This sets the number of repeats for the delay.

Filter Lo This filter affects the feedback loop and allows you to roll

Filter Hi This filter affects the feedback loop and allows you to roll

Spatial This parameter sets the stereo width for the left/right re-

Mix Sets the level balance between the dry signal and the ef-

Side-Chain on/off

if tempo sync is on (1/1–1/32, straight, triplet or dotted). If tempo sync is off, it sets the delay time in milliseconds.

The button below the Delay Time knob is used to turn tempo sync on or off. If set to off, the delay time can be set freely with the Delay Time knob, without sync to tempo.

off low frequencies up to 800 Hz. The button below the knob activates/deactivates the filter.

off high frequencies from 20kHz down to 1.2kHz. The button below the knob activates/deactivates the filter.

peats. Turn clockwise for a more pronounced stereo “ping-pong” effect.

fect. If PingPongDelay is used as a send effect, this should be set to maximum as you can control the dry/effect balance with the send.

When this is activated, the delay can be controlled by a signal routed to the Side-Chain input. When the sidechain signal exceeds the threshhold, the delay repeats are silenced. When the signal drops below the threshold, the delay repeats reappear. For a description of how to set up Side-Chain routing, see the chapter “Audio effects” in the Operation Manual.

StereoDelay

StereoDelay has two independent delay lines which either use tempo-based or freely specified delay time settings.

The parameters are as follows:

Parameter Description

Delay 1 This is where you specify the base note value for the delay,

Delay 2 As above.

Tempo sync on/off

Feedback 1 & 2

Filter Lo This filter affects the feedback loop and allows you to roll

Filter Hi This filter affects the feedback loop and allows you to roll

Pan1 & 2 This sets the stereo position for each delay.

Mix Sets the level balance between the dry signal and the ef-

Side-Chain on/off

if tempo sync is on (1/1–1/32, straight, triplet or dotted). If tempo sync is off, it sets the delay time in milliseconds.

The buttons below each respective Delay knob are used to turn tempo sync on or off for the respective delay. If set to off, the delay time can be set freely with the Delay Time knobs.

This sets the number of repeats for each delay.

off low frequencies up to 800Hz. The button below the knob activates/deactivates the filter.

off high frequencies from 20kHz down to 1.2kHz. The button below the knob activates/deactivates the filter.

fect. If StereoDelay is used as a send effect, this should be set to maximum (100%) as you can control the dry/effect balance with the send.

The included effect plug-ins

Distortion plug-ins

This section contains descriptions of the plug-ins in the “Distortion” category.

AmpSimulator

AmpSimulator is a distortion effect, emulating the sound of various types of guitar amp and speaker cabinet combinations. A wide selection of amp and cabinet models is available.

The parameters are as follows:

Parameter Description

Drive Governs the amount of amp overdrive.

Bass Tone control for the low frequencies.

Middle Tone control for the mid frequencies.

Treble Tone control for the high frequencies.

Presence Use this to boost or damp the higher frequencies.

Volume This controls the overall output level.

Amplifier This allows you to select between various amplifier mod-

Cabinet Various speaker cabinet models. Click on the currently

Damping Lo/Hi Further tone controls for shaping the sound of the se-

els. Click on the currently selected amplifier name to open a pop-up with all the available amplifier models. This section can be bypassed by selecting “No Amp".

selected cabinet name to open a pop-up with all the available amplifier models.This section can be bypassed by selecting “No Speaker".

lected speaker cabinet. Click on the values, enter a new value and press the [Enter] key.

DaTube

This effect emulates the characteristic warm, lush sound of a tube amplifier.

The parameters are as follows:

Parameter Description

Drive Regulates the pre-gain of the “amplifier”. Use high values

Balance This controls the balance between the signal processed

Output Adjusts the post-gain, or output level, of the “amplifier”.

if you want an overdriven sound just on the verge of distortion.

by the Drive parameter and the dry input signal. For maximum drive effect, set this to its highest value.

Distortion

Distortion will add crunch to your tracks.

The parameters are as follows:

Parameter Description

Drive Increases the distortion amount.

Feedback This parameter feeds part of the output signal back to the

Tone Lets you select a frequency range to which to apply the

Spatial Changes the distortion characteristics of the left and

Output Raises or lowers the signal going out of the effect.

effect input, increasing the distortion effect.

distortion effect.

right channel, thus creating a stereo effect.

The included effect plug-ins

SoftClipper (Cubase only)

This effect adds soft overdrive, with independent control over the second and third harmonic.

The parameters are as follows:

Parameter Description

Input Regulates the pre-gain. Use high values if you want an

Mix Setting Mix to 0 means that no processed signal is added

Output Adjusts the post-gain, or output level.

Second This allows you to adjust the amount of the second har-

Third This allows you to adjust the amount of the third harmonic

overdriven sound just on the verge of distortion.

to the original signal.

monic in the processed signal.

in the processed signal.

Dynamics plug-ins

This section contains descriptions of the plug-ins in the “Dynamics” category.

Compressor

Compressor reduces the dynamic range of the audio, making softer sounds louder or louder sounds softer, or both. Compressor features separate controls for threshold, ratio, attack, hold, release and make-up gain parameters. Compressor features a separate display that graphically illustrates the compressor curve shaped according to the Threshold and Ratio parameter settings. Compressor also features a Gain Reduction meter that shows the amount of gain reduction in dB, Soft knee/Hard knee compression modes and a program-dependent Auto feature for the Release parameter.

The available parameters work as follows:

Parameter Description

Threshold (-60 to 0dB)

Ratio (1:1 to 8:1)

Soft Knee (On/Off)

Make-up (0–24dB or “Auto mode")

This setting determines the level where Compressor “kicks in”. Signal levels above the set threshold are affected, but signal levels below are not processed.

Ratio determines the amount of gain reduction applied to signals over the set threshold. A ratio of 3:1 means that for every 3dB the input level increases, the output level will increase by only 1 dB.

If this is off, signals above the threshold will be compressed instantly according to the set ratio (hard knee). When Soft Knee is activated, the onset of compression will be more gradual, producing a less drastic result.

This parameter is used to compensate for output gain loss, caused by compression. If the Auto button is activated, the knob becomes dark and the output is instead automatically adjusted for gain loss.

The included effect plug-ins

Parameter Description

Attack (0.1–100ms)

Hold (0– 2000ms)

Release (10– 1000ms or “Auto mode”)

Analysis (0–100) (Pure Peak to Pure RMS)

Live mode (On/Off)

Side-Chain (On/Off)

This determines how fast Compressor will respond to signals above the set threshold. If the attack time is long, more of the early part of the signal (attack) will pass through unprocessed.

Sets the time the applied compression will affect the signal after exceeding the Threshold.

Sets the amount of time it takes for the gain to return to its original level when the signal drops below the Threshold level. If the “Auto” button is activated, Compressor will automatically find an optimal release setting that varies depending on the audio material.

This parameter determines whether the input signal is analysed according to peak or RMS values (or a mixture of both). A value of 0 is pure peak and 100 pure RMS. RMS mode operates using the average power of the audio signal as a basis, whereas Peak mode operates more on peak levels. As a general guideline, RMS mode works better on material with few transients such as vocals, and Peak mode better for percussive material, with a lot of transient peaks.

When activated, Live mode disengages the “look ahead” feature of the Compressor. Look ahead does produce more accurate processing but will add a certain amount of latency as a trade-off. When Live mode is activated, there is no latency, which might be better for “live” processing.

When this is activated, the compression can be controlled by a signal routed to the Side-Chain input. When the sidechain signal exceeds the threshhold, the compression is triggered. For a description of how to set up Side-Chain routing, see the chapter “Audio effects” in the Operation Manual.

SPL DeEsser (Cubase only)

A de-esser is used to reduce excessive sibilance, primarily for vocal recordings. Basically, it is a special type of compressor that is tuned to be sensitive to the frequencies produced by the “s” sound, hence the name de-esser. Close proximity microphone placement and equalizing can lead to situations where the overall sound is just right, but there is a problem with sibilants. Conventional compression and/or equalizing will not easily solve this problem, but a de-esser can.

The SPL DeEsser has the following parameters:

Parameter Description

S-Reduction Controls the intensity of the de-essing effect. We recom-

Level display Indicates the dB value by which the level of the sibilant or

Auto Threshold See separate description below.

Male/Female This sets the s-frequency and sibilant recognition to the

mend that you start with a value between 4 and 7.

s-frequency is reduced. The display shows values between 0dB (no reduction) and minus 20dB (the s-frequency level is lowered by 20dB). Each segment in the display represents a level reduction of 2 dB.

characteristic frequency ranges of the female or male voice. The center frequency of the bandwidth at which the SPL DeEsser operates is located in the 7 kHz range for the female voice and in the 6kHz range for the male voice.

About the Auto Threshold function

Conventional de-essing devices all have a threshold parameter. This is used to set a threshold for the incoming signal level, above which the device starts to process the signal. The SPL DeEsser however has been designed for utmost ease-of-use. With Auto Threshold on (the button lights up) it automatically and constantly readjusts the threshold to achieve an optimum result. If you still wish to determine for yourself at which signal level the SPL DeEsser should start to process the signal, deactivate the Auto Threshold button. The SPL DeEsser will then use a fixed threshold.

When recording a voice, usually the de-esser's position in the signal chain is located after the microphone pre-amp and before a compressor/limiter. This is useful, as it keeps the compressor/limiter from unnecessarily limiting the overall signal dynamics by reacting to excessive sibilants and s-frequencies.

The Auto Threshold function keeps the processing on a constant level. The input threshold value is automatically and constantly adjusted to the audio input level. Even level differences of say 20dB do not have a negative impact on the result of the processing. The input levels may vary, but processing remains constant.

The included effect plug-ins

EnvelopeShaper (Cubase only)

EnvelopeShaper can be used to cut or boost the gain of the Attack and Release phase of the audio material. You can either use the knobs or drag the breakpoints in the graphic display to change parameter values. Be careful with levels when boosting the gain and if needed reduce the Output level to avoid clipping.

The following parameters are available:

Parameter Description

Attack (-20–20dB) Changes the gain of the Attack phase of the signal.

Length (5–200ms) This determines the length of the Attack phase.

Release (-20–20dB) Changes the gain of the Release phase of the signal.

Output (-24–12dB) Sets the output level.

Expander (Cubase only)

Expander reduces the output level in relation to the input level for signals below the set threshold. This is useful, when you want to enhance the dynamic range or reduce the noise in quiet passages. You can either use the knobs or drag the breakpoints in the graphic display to change the Threshold and the Ratio parameter values.

The following parameters are available:

Parameter Description

Threshold (-60–0dB)

Ratio (1:1–8:1)

Soft Knee (On/Off)

Attack (0.1–100ms)

Hold (0–2000ms)

Release (10–1000ms or Auto mode)

Analysis (0–100) (Pure Peak to Pure RMS)

This setting determines the level where expansion “kicks in”. Signal levels below the set threshold are affected, but signal levels above are not processed.

Ratio determines the amount of gain boost applied to signals below the set threshold.

If this is off, signals below the threshold will be expanded instantly according to the set ratio ("hard knee"). When Soft Knee is activated, the onset of expansion will be more gradual, producing a less drastic result.

This determines how fast Expander will respond to signals below the set threshold. If the attack time is long, more of the early part of the signal (attack) will pass through unprocessed.

Sets the time the applied expansion will affect the signal below the Threshold.

Sets the amount of time it takes for the gain to return to its original level when the signal exceeds the Threshold level. If the “Auto” button is activated, Expander will automatically find an optimal release setting that varies depending on the audio material.

The included effect plug-ins

Parameter Description

Live mode (On/Off)

Side-Chain (On/Off)

When activated, Live mode disengages the look ahead feature of Expander. Look ahead does produce more accurate processing but will add a certain amount of latency as a trade-off. When Live mode is activated, there is no latency.

When this is activated, the expansion can be controlled by a signal routed to the Side-Chain input. When the side-chain signal exceeds the threshhold, the expansion is triggered. For a description of how to set up Side-Chain routing, see the chapter “Audio effects” in the Operation Manual.

Gate

Gating, or noise gating, silences audio signals below a certain set threshold level. As soon as the signal level exceeds the set threshold, the gate opens to let the signal through.

The available parameters are as follows:

Parameter Description

Threshold (-60–0dB)

state LED This indicates whether the gate is open (LED lights up in

Filter buttons When the Side-chain button (see below) is activated, you

Side-chain (Off/On)

Center (50Hz– 20000Hz)

Q-Factor (0.01–10000)

This setting determines the level where Gate is activated. Signal levels above the set threshold trigger the gate to open, and signal levels below the set threshold will close the gate.

green), closed (LED lights up in red) or something in between (LED lights up in yellow).

can use these buttons to set the filter type to either Low Pass, Band Pass or High Pass.

This button (below the Center knob) activates the filter. The input signal can then be shaped according to set Center and Q-Factor parameters which may be useful in tailoring how the Gate operates.

Sets the center frequency of the filter.

Sets the Resonance of the filter.

Parameter Description

Monitor (Off/On)

Attack (0.1–1000 ms)

Hold (0–2000ms)

Release (10–1000ms or “Auto”)

Analysis (0–100) (Pure Peak to Pure RMS

Live mode (On/Off)

Allows you to monitor the filtered signal.

This parameter sets the time it takes for the gate to open after being triggered. If the Live button (see below) is deactivated, it will ensure that the gate will already be open when a signal above the threshold level is played back. Gate manages this by “looking ahead” in the audio material, checking for signals loud enough to pass the gate.

This determines how long the gate stays open after the signal drops below the threshold level.

This parameter sets the amount of time it takes for the gate to close (after the set hold time). If the “Auto” button is activated, Gate will find an optimal release setting, depending on the audio program material.

This parameter determines whether the input signal is analysed according to Peak or RMS values (or a mixture of both). A value of 0 is pure Peak and 100 pure RMS. RMS mode operates using the average power of the audio signal as a basis, whereas Peak mode operates more on peak levels. As a general guideline, RMS mode works better on material with few transients such as vocals, and Peak mode better for percussive material, with a lot of transient peaks.

When activated, Live mode disengages the “look ahead” feature of the Gate. Look ahead does produce more accurate processing but will add a certain amount of latency as a trade-off. When Live mode is activated, there is no latency, which might be better for “live” processing.

The included effect plug-ins

Limiter

Maximizer

Limiter is designed to ensure that the output level never exceeds a certain set output level, to avoid clipping in following devices. Limiter can adjust and optimize the Release parameter automatically according to the audio material, or it can be set manually. Limiter also features separate meters for the input, output and the amount of limiting (middle meters).

The available parameters are the following:

Parameter Description

Input (-24–+24dB)

Output (-24–+6dB)

Release (0.1–1000ms or Auto mode)

Allows you to adjust the input gain.

This setting determines the maximum output level.

This parameter sets the amount of time it takes for the gain to return to its original level. If the “Auto” button is activated, Limiter will automatically find an optimal release setting that varies depending on the audio material.

Maximizer can be used to raise the loudness of audio material without the risk of clipping. Optionally, there is a soft clip function that removes short peaks in the input signal and introduces a warm tubelike distortion to the signal.

The available parameters are the following:

Parameter Description

Output (-24–+6dB)

Optimize (0–100)

Soft Clip (On/Off)

This setting determines the maximum output level. Should normally be set to 0 (to avoid clipping).

This setting determines the loudness of the signal.

Soft Clipper starts limiting (or clipping) the signal “softly”, at the same time generating harmonics which add a warm, tubelike characteristic to the audio material.

The included effect plug-ins

MIDI Gate

Gating, in its fundamental form, silences audio signals below a certain set threshold level. That means, when a signal rises above the set level, the Gate opens to let the signal through while signals below the set level are cut off. MIDI Gate, however, is a Gate effect that is not triggered by threshold levels, but instead by MIDI notes. Hence it needs both audio and MIDI data to function.

Setting up

MIDI Gate requires both an audio signal and a MIDI input to function.

To set it up, proceed as follows:

1. Select the audio to be affected by the MIDI Gate.

This can be audio material from any audio track, or even a live audio input (provided you have a low latency audio card).

2. Select the MIDI Gate as an insert effect for the audio track.

The MIDI Gate control panel opens.

3. Select a MIDI track to control the MIDI Gate.

This can be an empty MIDI track, or a MIDI track containing data, it doesn’t matter. However, if you wish to play the MIDI Gate in real-time – as opposed to having a recorded part playing it – the track has to be selected for the effect to receive the MIDI output.

4. Open the Output Routing pop-up menu for the MIDI track and select the MIDI Gate option.

The MIDI Output from the track is now routed to the MIDI Gate.

What to do next depends on whether you are using live or recorded audio and whether you are using real-time or recorded MIDI. We will assume for the purposes of this manual that you are using recorded audio, and play the MIDI in real-time.

Make sure the MIDI track is selected and start playback.

5. Now play a few notes on your MIDI keyboard.

As you can hear, the audio track material is affected by what you play on your MIDI keyboard.

The following MIDI Gate parameters are available:

Parameter Description

Attack This is used for determining how long it should take for

Hold Regulates how long the Gate remains open after a Note

Release This determines how long it takes for the Gate to close

Note To Attack

Note To Release

Velocity To VCA

Hold Mode Use this switch to set the Hold Mode. In Note-On mode,

the Gate to open after receiving a signal that triggers it.

On or Note Off message (see Hold Mode below).

(in addition to the value set with the Hold parameter).

The value you specify here determines to which extent the velocity values of the MIDI notes should affect the Attack. The higher the value, the more the Attack time will increase with high note velocities. Negative values will give shorter Attack times with high velocities. If you do not wish to use this parameter, set it to the 0 position.

The value you specify here determines to which extent the velocity values of the MIDI notes should affect the Release. The higher the value, the more the Release time will increase. If you do not wish to use this parameter, set it to the 0 position.

This controls to which extent the velocity values of the MIDI notes determine the output volume. A value of 127 means that the volume is controlled entirely by the velocity values, while a value of 0 means that velocities will have no effect on the volume.

the Gate will only remain open for the time set with the Hold and Release parameters, regardless of the length of the MIDI note that triggered the Gate. In Note-Off mode on the other hand, the Gate will remain open for as long as the MIDI note plays, and then apply the Hold and Release parameters.

The included effect plug-ins

MultibandCompressor (Cubase only)

The MultibandCompressor allows a signal to be split in up to four frequency bands, each with its own freely adjustable compressor characteristic. The signal is processed on the basis of the settings that you have made in the Frequency Band and Compressor sections. You can specify the level, bandwidth and compressor characteristics for each band by using the various controls.

The Frequency Band editor

The Frequency Band editor in the upper half of the panel is where you set the width of the frequency bands as well as their level after compression. Two value scales and a number of handles are available. The vertical value scale to the left shows the input gain level of each frequency band. The horizontal scale shows the available frequency range.

The handles provided in the Frequency Band editor can be dragged with the mouse. You use them to set the corner frequency range and the input gain levels for each frequency bands.

• The handles at the sides are used to define the frequency range of the different frequency bands.

• By using the handles on top of each frequency band, you can cut or boost the input gain by +/- 15dB after compression.

Bypassing frequency bands

Each frequency band can be bypassed using the “B” button in each compressor section.

Soloing frequency bands

A frequency band can be soloed using the “S” button in each compressor section. Only one band can be soloed at a time.

Using the Compressor section

By moving breakpoints or using the corresponding knobs, you can specify the Threshold and Ratio. The first breakpoint from which the line deviates from the straight diagonal will be the threshold point. The compressor parameters for each of the four bands are as follows:

Parameter Description

Threshold (-60–0dB)

Ratio (1000–8000) (1:1 to 8:1)

Attack (0.1– 100ms)

Release (10– 1000ms or “Auto”)

This setting determines the level where Compressor “kicks in”. Signal levels above the set threshold are affected, but signal levels below are not processed.

Ratio determines the amount of gain reduction applied to signals over the set threshold. A ratio of 3000 (3:1) means that for every 3dB the input level increases, the output level will increase by only 1dB.

This determines how fast the compressor will respond to signals above the set threshold. If the attack time is long, more of the early part of the signal (attack) will pass through unprocessed.

Sets the amount of time it takes for the gain to return to its original level when the signal drops below the Threshold level. If the “Auto” button is activated, the compressor will automatically find an optimal release setting that varies depending on the audio material.

The Output dial

The Output dial controls the total output level that the MultibandCompressor passes on to Cubase. The range available is +/- 24dB.

The included effect plug-ins

VintageCompressor (Cubase only)

VSTDynamics

This is modelled after vintage type compressors. Compressor features separate controls for input gain, attack, release and output gain parameters. In addition, there is a Punch mode which preserves the attack phase of the signal and a program dependent Auto feature for the Release parameter.

The available parameters work as follows:

Parameter Description

Input gain (-24–48dB)

Output gain (-48–24dB)

Attack (0.1–100ms)

Punch (Off/On)

Release (10–1000ms or “Auto mode”)

Side-Chain (On/Off)

This setting, together with the Output gain parameter determines the compression amount. The higher the Input gain setting, and the lower the Output gain setting, the more compression is applied.

Sets the output gain.

This determines how fast Compressor will respond. If the attack time is long, more of the early part of the signal (attack) will pass through unprocessed.

When this is activated, the early attack phase of the signal is preserved, retaining the original “punch” in the audio material, even with short Attack settings.

Sets the amount of time it takes for the gain to return to its original level. If the “Auto” button is activated, Vintage Compressor will automatically find an optimal release setting that varies depending on the audio material.

When this is activated, the compression can be controlled by a signal routed to the Side-Chain input. When the sidechain signal exceeds the threshhold the compression is triggered. For a description of how to set up Side-Chain routing, see the chapter “Audio effects” in the Operation Manual.

Gate

Compressor

Limiter

Routing selector

VSTDynamics is an advanced dynamics processor. It combines three separate processors: Gate, Compressor and Limiter, covering a variety of dynamic processing functions. The window is divided into three sections, containing controls and meters for each processor.

Activating the individual processors

You activate the individual processors using the buttons at the bottom of the plug-in panel.

The Gate section

Gating, or noise gating, is a method of dynamic processing that silences audio signals below a certain set threshold level. As soon as the signal level exceeds the set threshold, the gate opens to let the signal through. The Gate trigger input can also be filtered using an internal side-chain.

The available parameters are as follows:

Parameter Description

Threshold (-60–0dB)

state This indicates whether the gate is open (LED lights up in

Side-chain (On/Off)

LP (Lowpass), BP (Bandpass), HP (Highpass)

Center (50–22000Hz)

Q-Factor (0.001–10000)

This setting determines the level where Gate is activated. Signal levels above the set threshold trigger the gate to open, and signal levels below the set threshold will close the gate.

green), closed (LED lights up in red) or something in between (LED lights up in yellow).

This button activates the internal side-chain filter. This lets you filter out parts of the signal that might otherwise trigger the gate in places you don’t want it to, or to boost frequencies you wish to accentuate, allowing for more control over the gate function.

These buttons set the basic filter mode.

This sets the center frequency of the filter.

This sets the resonance or width of the filter.

The included effect plug-ins

Parameter Description

Monitor (Off/On)

Attack (0.1–100ms)

Hold (0–2000ms)

Release (10–1000ms or “Auto”)

Allows you to monitor the filtered signal.

This parameter sets the time it takes for the gate to open after being triggered.

This determines how long the gate stays open after the signal drops below the threshold level.

The Compressor section

Compressor reduces the dynamic range of the audio, making softer sounds louder or louder sounds softer, or both. Compressor functions like a standard compressor with separate controls for threshold, ratio, attack, release and make-up gain parameters. Compressor features a separate display that graphically illustrates the compressor curve shaped according to the Threshold, Ratio and MakeUp Gain parameter settings. Compressor also features a Gain Reduction meter that shows the amount of gain reduction in dB, and a program dependent Auto feature for the Release parameter.

The available parameters work as follows:

Parameter Description

Threshold (-60–0dB)

Ratio (1:1–8:1)

Make-Up (0–24dB)

Attack (0.1–100ms)

Release (10–1000ms or “Auto”)

Graphic display

This setting determines the level where Compressor “kicks in”. Signal levels above the set threshold are affected, but signal levels below are not processed.

This parameter is used to compensate for output gain loss, caused by compression. When Auto is on, gain loss will be compensated automatically.

This determines how fast Compressor will respond to signals above the set threshold. If the attack time is long, more of the early part of the signal (attack) will pass through unprocessed.

Use the graphic display to graphically set the Threshold or the Ratio value.

The Limiter section

Limiter is designed to ensure that the output level never exceeds a certain set output level, to avoid clipping in following devices. Conventional limiters usually require very accurate setting up of the attack and release parameters, to prevent the output level from going beyond the set threshold level. Limiter adjusts and optimizes these parameters automatically, according to the audio material. You can also adjust the Release parameter manually.

The available parameters are the following:

Parameter Description

Output (-24–+6dB)

Soft Clip (On/Off)

Release (10–1000ms or “Auto”)

This setting determines the maximum output level. Signal levels above the set threshold are affected, but signal levels below are left unaffected.

Soft Clipper acts differently compared to the limiter. When the signal level exceeds -6dB, SoftClip starts limiting (or clipping) the signal “softly”, at the same time generating harmonics which add a warm, tubelike characteristic to the audio material.

This parameter sets the amount of time it takes for the gain to return to its original level when the signal drops below the threshold level. If the “Auto” button is activated, Limiter will automatically find an optimal release setting that varies depending on the audio material.

The Module Configuration button

In the bottom right corner of the plug-in panel you will find a button with which you can set the signal flow order for the three processors. Changing the order of the processors can produce different results, and the available options allow you to quickly compare what works best for a given situation. Simply click the Module Configuration button to change to a different configuration. There are three routing options:

• C-G-L (Compressor-Gate-Limit)

• G-C-L (Gate-Compressor-Limit)

• C-L-G (Compressor-Limit-Gate)

The included effect plug-ins

EQ plug-ins

This section describes the plug-ins in the “EQ” category.

GEQ-10/GEQ-30 (Cubase only)

These graphic equalizers are identical in every respect except for the number of available frequency bands (10 and 30 respectively). Each band can be cut or boosted by up to 12dB allowing for fine control of the frequency response. In addition there are several preset modes available which can add “color” to the sound of the GEQ-10/GEQ-30.

• You can draw response curves in the main display by

click-dragging with the mouse.

Note that you have to click on one of the sliders first before dragging across the display. You can also point and click to change individual frequency bands or enter values numerically by clicking on a gain value at the top of the display.

• At the bottom of the window the respective frequency

bands are shown in Hz.

• At the top of the display, the amount of cut/boost is

shown in dB.

Apart from the frequency bands, the following parameters are available:

Parameter Description

Output This controls the overall gain of the equalizer.

Range This allows you to relatively adjust how much a set curve

Flatten button Resets all the frequency bands to 0dB.

Invert range This will invert the current response curve.

Mode The filter mode set here determines how the various fre-

cuts or boosts the signal. If the Range parameter is turned fully clockwise, +/- 12dB is the available range.

quency band contrrols interact to create the response curve. See also below.

About the filter modes

On the pop-up in the lower right corner there are several different EQ modes available. These modes can add color or character to the equalized output in various ways, which is sometimes desirable. As always, let your ears be the judge! Here follow brief descriptions of the filter modes:

• True Response – serial filters with accurate frequency response.

• Digi Standard – resonance of last band depends on sample rate.

• Variable Q – parallell filters where the resonance depends on the amount of gain. Musical sounding.

• Constant Q u – parallell filters where the resonance of the first and last bands depends on the sample rate (u=unsymmetric).

• Constant Q s – parallell filters where the resonance is raised when boosting the gain and vice versa (s=symmetric).

• Resonant – serial filters where a gain increase of one band will lower the gain in adjacent bands.

The included effect plug-ins

StudioEQ (Cubase only)

This is a high-quality 4-band parametric stereo equalizer with two fully parametric midrange bands. The low and high bands can act as either shelving filters (three types) or as a Peak (bandpass) or Cut (lowpass/highpass) filter.

Making settings

1. Click the corresponding On button to the left of the EQ curve display to activate any or all of the Low, Mid 1, Mid 2 or High equalizer bands.

When a band is activated, a corresponding eq point appears in the EQ curve display.

2. Set the parameters for an activated EQ band.

This can be done in several ways:

• By using the knobs.

• By clicking a value field and entering values numerically.

• By using the mouse to drag points in the EQ curve display window.

By using this method, you control both the Gain and Frequency parameters simultaneously. The knobs turn accordingly when you drag points.

The following parameters are available:

Parameter Description

Low Freq (20 to 2000Hz)

Low Gain (-20 to +24dB)

Low Q-Factor This controls the width or resonance of the Low band.

Low Filter mode

Mid 1 Freq (20 to 20000Hz)

Mid 1 Gain (+/- 24dB)

Mid 1 Q-Factor (0.5 to 10)

Mid 2 Freq (20 to 20000Hz)

Mid 2 Gain (-20 to +24dB)

Mid 2 Q-Factor (0.5 to 10)

High Freq (200 to 20000Hz)

High Gain (-20 to +24dB)

High Q-Factor This parameter controls the width or resonance of the

High Filter mode

Output (-24 to +24dB)

Auto Gain When this is activated, the gain is automatically adjusted,

This sets the frequency of the Low band.

This sets the amount of cut/boost for the Low band.

For the Low band, you can select between three types of shelving filters or Peak (bandpass) or Cut (lowpass/highpass) filters. The Gain parameter will be fixed if Cut mode is selected.

-Shelf I adds resonance in the opposite gain direction slightly over the set frequency.

-Shelf II adds resonance in the gain direction at the set frequency.

-Shelf III is a combination of Shelf I and II.

This sets the center frequency of the Mid 1 band.

This sets the amount of cut/boost for the Mid 1 band.

This sets the width of the Mid 1 band. The higher this value, the “narrower” the bandwidth.

This sets the center frequency of the Mid 2 band.

This sets the amount of cut/boost for the Mid 2 band.

This sets the width of the Mid 2 band. The higher this value, the “narrower” the bandwidth.

This sets the frequency of the High band.

This sets the amount of cut/boost for the High band.

High band.

For the High band, you can select between three types of shelving filters, and Peak or Cut filters. The Gain parameter will be fixed if Cut mode is selected.

-Shelf I adds resonance in the opposite gain direction slightly below the set frequency.

-Shelf II adds resonance in the gain direction at the set frequency.

-Shelf III is a combination of Shelf I and II.

This parameter allows you to adjust the overall output level.

keeping the output level constant regardless of the EQ settings.

The included effect plug-ins

Filter plug-ins

This section contains descriptions of the plug-ins in the “Filter” category.

General operation

StepFilter can produce two simultaneous 16-step patterns for the filter cutoff and resonance parameters, synchronized to the sequencer tempo.

DualFilter

This effect filters out certain frequencies while allowing others to pass through.

The following parameters are available:

Parameter Description

Position This parameter sets the filter cutoff frequency. If you set

Resonance Sets the sound characteristic of the filter. With higher

this to a negative value, DualFilter will act as a low-pass filter. Positive values cause DualFilter to act as a highpass filter.

values, a ringing sound is heard.

StepFilter

Setting step values

• Setting step values is done by clicking in the pattern grid windows.

• Individual step entries can be freely dragged up or down the vertical axis, or directly set by clicking in an empty grid box. By click-dragging left or right, consecutive step entries will be set to the pointer position.

Setting filter cutoff values in the grid window.

• The horizontal axis shows the pattern steps 1–16 from left to right, and the vertical axis determines the (relative) filter cutoff frequency and resonance setting.

The higher up on the vertical axis a step value is entered, the higher the relative filter cutoff frequency or filter resonance setting.

• By starting playback and editing the patterns for the cutoff and resonance parameters, you can hear how your filter patterns affect the sound source connected to StepFilter directly.

Selecting new patterns

• Created patterns are saved with the project, and up to 8 different cutoff and resonance patterns can be saved internally.

Both the cutoff and resonance patterns are saved together in the 8 Pattern memories.

• To select new patterns you use the pattern selector.

New patterns are all set to the same step value by default.

StepFilter is a pattern-controlled multimode filter that can create rhythmic, pulsating filter effects.

The included effect plug-ins

Pattern Selector

Using pattern copy and paste to create variations

You can use the Copy and Paste buttons below the pattern selector to copy a pattern to another pattern memory location, which is useful for creating variations on a pattern.

• Select the pattern you wish to copy, click the Copy but-

ton, select another pattern memory location and click Paste.

The pattern is copied to the new location, and can now be edited to create variations using the original pattern as a starting point.

ToneBooster

StepFilter parameters

Parameter/ Value

Base Cutoff This sets the base filter cutoff frequency. Cutoff values

Base Resonance

Glide This will apply glide between the pattern step values,

Filter Mode This slider selects between lowpass (LP), bandpass (BP)

Sync 1/1 to 1/32 (Straight, Triplet or Dotted)

Output Sets the overall volume.

Mix Adjusts the mix between dry and processed signal.

Description

set in the Cutoff grid window are values relative to the Base Cutoff value.

This sets the base filter resonance. Resonance values set in the Resonance grid window are values relative to the Base Resonance value. Note that very high Base Resonance settings can produce loud ringing effects at certain frequencies.

causing values to change more smoothly.

or highpass (HP) filter modes (from left to right respectively).

This sets the pattern beat resolution, i.e. what note values the pattern will play in relation to the tempo.

ToneBooster is a filter that allows you to raise the gain in a selected frequency range. It is particularly useful when inserted before AmpSimulator in the plug-in chain (see

“AmpSimulator” on page 9), greatly enhancing the tonal

varieties available.

The following parameters are available:

Parameter Description

Tone This sets the center filter frequency.

Gain Allows you to adjust the gain of the selected frequency

Width This sets the resonance of the filter.

Mode This sets the basic operational mode of the filter; Peak or

range by up to 24dB.

Bandpass.

The included effect plug-ins

Tonic – Analog Modeling Filter (Cubase only)

Tonic is a versatile and powerful analog modeling filter plug-in based on the filter design of the Monologue monophonic synthesizer. Its variable characteristics plus the powerful modulation functions make it an excellent choice for all current music styles. Designed to be more a creative tool rather than a tool to fix audio problems, it can add color and punch to your tracks while being light on CPU usage.

The Tonic Analog Modeling Filter has the following properties:

• Dynamic multimode analog modeling filter (mono/stereo).

• 24dB low pass, 18dB low pass, 12 dB low pass, 6dB

low pass, 12dB band pass and 12dB high pass modes.

• Adjustable drive and resonance up to self-oscillation.

• Envelope follower for dynamic filter control with an

audio signal.

• Audio and MIDI trigger modes.

• Powerful step LFO with smoothing and morphing.

• X/Y matrix pad for additional realtime modulation with

access to all Tonic parameters.

Filter

Parameter Description

Mode Sets the filter type. Available filter types are: 24dB Low

Cutoff Sets the filter cutoff frequency. How this parameter oper-

Res Changes the resonance of the multi-mode filter. Full res-

Drive Drive adds a soft, tube-like saturation to the sound. Like

Mix Sets the balance between dry and effect signal.

Ch. Choose between mono or stereo operation. When set to

pass, 18dB Low pass, 12 dB Low pass, 6 dB Low pass, 12dB Band pass and 12 dB High pass.

ates is governed by the filter type.

onance puts the filter into self-oscillation.

for an analog filter, the amount of saturation also depends on the input signal level.

mono, the output signal of Tonic will be mono regardless of the input signal.

Env Mod

Parameter Description

Mode Tonic offers three types of envelope modulation:

Attack Controls the attack time of the envelope. Higher attack

Release Controls the release time of the envelope. Higher release

Depth Controls the amount of envelope control applied to the

LFO Mod Using this parameter, envelope level modulates the LFO

“Follow” tracks the input signal’s volume envelope for dynamic control of the filter cutoff. “Trigger” uses the input signal to trigger the envelope and have it run through a single envelope cycle. “MIDI” uses any MIDI note to trigger the envelope. The filter cutoff tracks the keys played on the keyboard. In addition velocities higher than 80 will add an accent to the envelope by increasing the envelope depth and reducing the decay time. For MIDI control, set up a separate MIDI control track and select “Tonic” from the output pop-up menu for the track.

times result in slower rise times when the envelope is triggered.

times result in slower envelope tails.

filter cutoff level.

speed. A rather stunning effect.

The included effect plug-ins

X/Y Pad

Parameter Description

X Par Sets the parameter to be modulated on the x axis of the

Y Par Sets the parameter to be modulated on the y axis of the

XY Pad Use the mouse to control any two of Tonic’s parameters

XY Pad. All of Tonic’s parameters are available as destinations

XY Pad.

in combination. By moving the mouse horizontally, you can control the x parameter, by moving it vertically, you can control the y parameter. You can also record controller movements as automation data.

LFO Mod

Parameter Description

Mode Sets the direction of the step LFO modulation. The avail-

Depth Controls the amount of LFO modulation applied to the fil-

Rate Controls the speed of the LFO modulation. The LFO rate

Smooth The smooth parameter controls the smoothing of the LFO

Morph Morph controls the playback value of the LFO step se-

Steps Sets the number of steps played in sequence. Deacti-

Preset Offers a number of step LFO waveform patterns.

Step Matrix Click into the step matrix to set the level for each of the

able modes are: Forward, Reverse, Alternating, and Random.

ter cutoff level.

is always in sync with the song tempo. For example: a rate of 4.00 steps per beat advances the step sequencer in 16th notes at a 4/4 time signature. A rate of 4.00 beats per step would advance the LFO at only one step per bar in a 4/4 time signature.

steps. This works like a glide effect applied to the filter cutoff.

quencer. It makes the LFO steps drift about randomly. Experiment freely with the morph parameter. As you return the knob to its zero position the step pattern will return to its original setting.

vated steps are grayed out in the step window.

Choices include: Sine, Sine+, Cosine, Triangle, Sawtooth, Square, Random and User (which is the pattern saved with the respective program).

16 LFO steps. A higher amount results in a deeper filter cutoff modulation. Click and drag along the matrix to “draw” a waveform.

WahWah

WahWah is a variable slope bandpass filter that can be auto-controlled by a side-chain signal or via MIDI modeling the well-known analog pedal effect (see below). You can independently specify the frequency, width and the gain for the Lo and Hi Pedal positions. The crossover point between the Lo and Hi Pedal positions is at 50.

The parameters are as follows:

Parameter Description

Pedal This controls the filter frequency sweep.

Freq Lo/Hi Sets the frequency of the filter for the Lo and Hi Pedal

Width Lo/Hi Sets the width (resonance) of the filter for the Lo and Hi

Gain Lo/Hi Sets the gain of the filter for the Lo and Hi Pedal posi-

Slope Specifies the slope of the filter; 6dB or 12dB.

Side-Chain On/Off

MIDI control

For real-time MIDI control of the Pedal parameter, MIDI must be directed to the WahWah plug-in.

• Whenever the WahWah has been added as an insert effect (for an audio track or an FX channel), it will be available on the Output Routing pop-up menu for MIDI tracks.

If WahWah is selected on the Output Routing menu, MIDI will be directed to the plug-in from the selected track.

positions.

Pedal positions.

tions.

A signal routed to the Side-Chain input of the effect can control the Pedal parameter when this is activated. The louder the signal, the more the filter frequency (Pedal) is raised so the plug-in acts as an “auto-wha” effect. For a description of how to set up Side-Chain routing, see the chapter “Audio effects” in the Operation Manual.

The included effect plug-ins

Mastering – UV 22 HR (Cubase only)

The UV22 HR is a dithering plug-in, based on an advanced algorithm developed by Apogee. For an introduction to the concept of dithering, see the chapter “Audio Effects” in the Operation Manunal.

The following options can be set in the UV 22 HR control panel:

Option Description

Hi Try this first, it is the most “all-round” setting.

Low This applies a lower level of dither noise.

Auto black When this is activated, the dither noise is gated (muted)

Bit Resolution The UV22 HR supports dithering to multiple resolutions:

Dither should always be applied post output bus fader.

during silent passages in the material.

8, 16, 20 or 24 bits. You select the desired resolution by clicking the corresponding button.

Modulation plug-ins

This section contains descriptions of the plug-ins in the “Modulation” category.

AutoPan

This is a simple autopan effect. It can use different waveforms to modulate the left-right stereo position (pan), either using tempo sync or manual modulation speed settings.

The parameters are as follows:

Parameter Description

Rate If tempo sync is on, this is where you specify the base

Tempo sync on/off

Width Sets the depth of the Autpan effect.

Shape Sets the modulation waveform. Sine and Triangle wave-

Side-Chain On/Off

note value for tempo-syncing the effect (1/1 to 1/32, straight, triplet or dotted). If tempo sync is off, the auto-pan speed can be set freely with the Rate knob, without sync to tempo.

The button below the Rate knob is used to switch tempo sync on (the button lights up) or off.

forms are available.

A signal routed to the Side-Chain input of the effect can control the Width parameter when this is activated. For a description of how to set up Side-Chain routing, see the chapter “Audio effects” in the Operation Manual.

The included effect plug-ins

Chorus

This is a single stage chorus effect. It works by doubling whatever is sent into it with a slightly detuned version. See also “StudioChorus” on page 30.

The parameters are as follows:

Parameter Description

Tempo sync on/off

Rate If tempo sync is on, this is where you specify the base

Width This determines the depth of the chorus effect. Higher

Spatial This sets the stereo width of the effect. Turn clockwise

Mix Sets the level balance between the dry signal and the ef-

Delay This parameter affects the frequency range of the modu-

Shape This changes the shape of the modulating waveform, al-

Filter Lo/Hi These parameters allow you to roll off low and high fre-

Side-Chain On/Off

The button below the Rate knob is used to switch tempo sync on or off. The button is lit when tempo sync is on.

note value for tempo syncing the chorus sweep (1/1 to 1/32, straight, triplet or dotted). If tempo sync is off, the sweep rate can be set freely with the Rate knob, without sync to tempo.

settings produce a more pronounced effect.

for a wider stereo effect.

fect. If StudioChorus is used as a send effect, this should be set to maximum as you can control the dry/effect balance with the send.

lation sweep, by adjusting the initial delay time.

tering the character of the chorus sweep. Sine and triangle waveforms are available.

quencies of the effect signal, respectively.

When this is activated, the modulation can be controlled by a signal routed to the Side-Chain input. When the side-chain signal exceeds the threshhold the modulation will be controlled by the side-chain signal’s envelope. For a description of how to set up Side-Chain routing, see the chapter “Audio effects” in the Operation Manual.

Cloner (Cubase only)

The Cloner plug-in adds up to four detuned and delayed voices to the signal, for rich modulation and chorus effects.

The parameters are as follows:

Parameter Description

Voices This allows you to select the number of voices (up to

Spatial This spreads the added voices across the stereo spec-

Mix Sets the level balance between the dry signal and the ef-

Output Allows you to reduce or increase the output gain by up to

Detune slider 1–4

Delay slider 1–4

Master Detune This parameter governs the overall depth of the detuning

Humanize Delay knob

Humanize Detune knob

Master Delay This parameter governs the overall depth of the delay for

four). For each added voice, a Detune and a Delay slider are added in the right half of the panel.

trum. Turn clockwise for a deeper stereo effect.

fect. If Cloner is used as a send effect, this should be set to maximum as you can control the dry/effect balance with the send.

+/- 12dB.

This controls the relative detune amount for each voice. Positive and negative values can be set, from -100 to

100. A value of zero means no detune for that voice.

This controls the relative delay amount for each voice. A value of zero means no delay for that voice.

for all voices. If this is set to zero, no detuning takes place, regardless of the Detune slider settings.

Humanize is turned on and off with the Static Delay button button below this knob. When activated the delay settings are subtly varied, for a richer effect. Values range from 0 to 100 (strongest delay variation). If deactivated, the set delay amount is static, and the knob is blacked out.

Humanize is turned on and off with the Static Detune button below this knob. When activated, the detune settings are subtly varied, for a richer effect. Values range from 0 to 100 (strongest detune variation). If deactivated, the set detune amount is static, and the knob is blacked out.

all voices. If this is set to zero, no delay takes place, regardless of the Delay slider settings.

The included effect plug-ins

Flanger

Metalizer

Flanger is a classic flanger effect with added stereo enhancement.

The parameters are as follows:

Parameter Description

Tempo sync on/ off

Rate If tempo sync is on, this is where you specify the base

Range Lo/Hi This sets the frequency boundaries for the flanger sweep.

Feedback This determines the character of the flanger effect.

Spatial This sets the stereo width of the effect. Turn clockwise

Mix Sets the level balance between the dry signal and the ef-

Shape This changes the shape of the modulating waveform, al-

Delay This parameter affects the frequency range of the modu-

Manual If this is activated, the flanger sweep will be static, i.e. no

Filter Lo/Hi These parameters allow you to roll off low and high fre-

Side-Chain On/Off

The button below the Rate knob is used to switch tempo sync on or off. The button is lit when tempo sync is on.

note value for tempo syncing the flanger sweep (1/1 to 1/32, straight, triplet or dotted). If tempo sync is off, the sweep rate can be set freely with the Rate knob, without sync to tempo.

Higher settings produce a more “metallic” sounding sweep.

for a wider stereo effect.

fect. If the Flanger is used as a send effect, this should be set to maximum as you can control the dry/effect balance with the send.

tering the character of the flanger sweep.

lation sweep, by adjusting the initial delay time.

modulation. You can instead change the sweep position manually by turning this knob.

quencies of the effect signal, respectively.

The Metalizer feeds the audio signal through a variable frequency filter, with tempo sync or time modulation and feedback control.

Parameter Description

Feedback The higher the value, the more “metallic” the sound.

Sharpness Governs the character of the filter effect. The higher the

Tone Governs the feedback frequency. The effect of this will

On button Turns filter modulation on and off. When turned off, the

Mono button When this is on, the output of the Metalizer will be in mono.

Speed If tempo sync is on, this is where you specify the base

Tempo sync on/off

Output Sets the overall volume.

Mix Sets the level balance between the dry signal and the ef-

value, the narrower the affected frequency area, producing sharper sound and a more pronounced effect.

be more noticeable with high Feedback settings.

Metalizer will work as a static filter.

note value for tempo-syncing the effect (1/1 to 1/32, straight, triplet or dotted). Note that there is no note value modifier for this effect. If tempo sync is off, the modulation speed can be set freely with the Speed knob, without sync to tempo.

The button above the Speed knob is used to switch tempo sync on or off. The button is lit when tempo sync is on.

fect. If Metalizer is used as a send effect, this should be set to maximum as you can control the dry/effect balance with the send.

The included effect plug-ins

Phaser

Phaser produces the well-known “swooshing” phasing effect with additional stereo enhancement.

The parameters are as follows:

Parameter Description

Tempo sync on/off

Rate If tempo sync is on, this is where you specify the base

Width The width of the modulation effect between higher and

Feedback This determines the character of the phaser effect.

Spatial When using multi-channel audio, Spatial creates a 3-di-

Mix Sets the level balance between the dry signal and the ef-

Manual If this is activated, the phaser sweep will be static, i.e. no

Filter Lo/Hi These parameters allow you to roll off low and high fre-

Side-Chain On/Off

The button below the Rate knob is used to switch tempo sync on or off. The button is lit when tempo sync is on.

note value for tempo syncing the phaser sweep (1/1 to 1/32, straight, triplet or dotted). If tempo sync is off, the sweep rate can be set freely with the Rate knob, without sync to tempo.

lower frequencies.

Higher settings produce a more pronounced effect.

mensional impression by delaying modulation in each channel.

fect. If the Phaser is used as a send effect, this should be set to maximum as you can control the dry/effect balance with the send.

modulation. You can instead change the sweep position manually by turning this knob.

quencies of the effect signal, respectively.

Ringmodulator

The Ringmodulator can produce complex, bell-like enharmonic sounds. Ring modulators work by multiplying two audio signals. The ring modulated output contains added frequencies generated by the sum of, and the difference between, the frequencies of the two signals.

The Ringmodulator has a built-in oscillator that is multiplied with the input signal to produce the effect.

Parameter Description

Oscillator LFO Amount

Oscillator Env. Amount

Oscillator Wave Selects the oscillator waveform; square, sine, saw or

Oscillator Range Determines the frequency range of the oscillator in Hz.

Oscillator Frequency

Oscillator Roll-Off

LFO Speed Sets the LFO Speed.

LFO Env. Amount

LFO Waveform Selects the LFO waveform; square, sine, saw or triangle.

Controls how much the oscillator frequency is affected by the LFO.

Controls how much the oscillator frequency is affected by the envelope (which is triggered by the input signal). Positive and negative values can be set, with center position representing no modulation. Left of center, a loud input signal will decrease the oscillator pitch, whereas right of center the oscillator pitch will increase when fed a loud input.

triangle.

Sets the oscillator frequency +/- 2 octaves within the selected range.

Cuts high frequencies in the oscillator waveform, to soften the overall sound. This is best used when harmonically rich waveforms are selected (e.g. square or saw).

Controls how much the input signal level – via the envelope generator – affects the LFO speed. Positive and negative values can be set, with center position representing no modulation. Left of center, a loud input signal will slow down the LFO, whereas right of center a loud input signal will speed it up.

The included effect plug-ins

+ 63 hidden pages