Steinberg Cubase Artist - 6.0, Cubase 6.0 Plug-in Reference

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Plug-in Reference

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Cristina Bachmann, Heiko Bischoff, Marion Bröer, Sabine Pfeifer, Heike Schilling 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. Registered licensees of the product described herein may print one copy of this document for their personal use.

All product and company names are ™ or ® trademarks of their respective owners. Windows 7 is 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. MP3SURROUND and the MP3SURROUND logo are registered trademarks of Thomson SA, registered in the US and other countries, and are used under license from Thomson Licensing SAS.

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5 The included effect plug-ins

6 Introduction 6 Delay plug-ins 9 Distortion plug-ins 15 Dynamics plug-ins 23 EQ plug-ins 25 Filter plug-ins 30 Modulation plug-ins 37 Pitch Shift plug-ins 39 Reverb plug-ins 46 Spatial + Panner plug-ins 47 Surround plug-ins (Cubase only) 48 Tools plug-ins

74 The included VST instruments

75 Introduction 75 Embracer – Surround Pad Synthesizer

(Cubase only)

77 Groove Agent ONE 81 HALion Sonic SE 82 LoopMash 90 Monologue – Monophonic Analog Modeling

Synthesizer (Cubase only)

92 Mystic 100 Prologue 110 Spector 118 Diagrams

52 MIDI effects

53 Introduction 53 Arpache 5 54 Arpache SX 55 Auto LFO 56 Beat Designer 61 Chorder 63 Compressor 64 Context Gate 65 Density 65 Micro Tuner 65 MIDI Control 66 MIDI Echo 67 MIDI Modifiers 67 MIDI Monitor 68 Note to CC 68 Quantizer 69 StepDesigner 71 Track Control 73 Transformer

120 Index

Table of Contents

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The included effect plug-ins

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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 following parameters are available:

Parameter Description

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

Delay – Sync button

Rate The Rate parameter sets the base note value for tempo

Rate – Sync button

Width Sets the amount of delay pitch modulation. Note that al-

Feedback Sets the number of repeats for the delay. Drive Adds distortion to the feedback loop. The longer the

Mix Sets the level balance between the dry and the wet sig-

Nudge button Clicking the Nudge button once will momentarily speed

Signal path graphic and Filter position

Filter type (in graphic display)

Freq Sets the cutoff frequency for the filter. It is only available

Speed Sets the speed of the filter frequency LFO modulation.

Speed – Sync button

note value for the delay (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 is used to switch tempo sync for the Delay parameter on or off.

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

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

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

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

nal. If ModMachine is used as a send effect, set this to the maximum value (100 effect balance with the send.

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

The filter can either be placed in the feedback loop of the delay or in the output path of the effect (after the Drive and Feedback parameters). To switch between the “loop” and “output” positions, click on the Filter section displayed in the graphic or click on the Position field at the bottom right of the graphic.

The Type button allows you to select a filter type. A lowpass, band-pass, and high-pass filter are available.

if tempo sync for the Speed parameter (see below) is deactivated and the parameter is set to “0”.

When using tempo sync, 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 speed can be set freely.

The button below the Speed knob is used to switch tempo sync for the Speed parameter on or off.

%) as you can control the dry/

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Parameter Description

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

Spatial Introduces an offset between the channels to create a

Q-Factor Controls the resonance of the filter. It is only available if

Speed Sets the speed of the filter resonance LFO modulation.

Speed – Sync button

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

Spatial Introduces an offset between the channels to create a

quency modulation. Both positive (e. g. Lo set to 50 and Hi set to 10000) and negative (e. 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 controlled by the Freq parame ter instead.

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

filter resonance LFO tempo sync is deactivated and the Speed parameter (see below) is set to “0”. When using tempo sync, the resonance is controlled by the Speed and Range parameters.

When using tempo sync, 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 speed can be set freely.

The button below the Speed knob is used to switch tempo sync for the Speed parameter on or off.

modulation. Both positive (e. to 100) and negative (e.

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 pa rameter instead.

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

g. Lo set to 5000 and

g. Lo set to 50 and Hi set

g. Lo set to 100 and Hi set to

MonoDelay

This is a mono delay effect that can either be tempobased or use freely specified delay time settings.

The following parameters are available:

Parameter Description

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

Sync button The button below the Delay knob is used to switch tempo

Feedback 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 and the wet sig-

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

sync on or off.

and allows you to roll off low frequencies from 10

Hz. The button below the knob activates/deactivates

800 the filter.

and allows you to roll off high frequencies from 20 down to 1.2 deactivates the filter.

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

kHz. The button below the knob activates/

Hz up to

kHz

Ö The delay can also be controlled from another signal source via the side-chain input. When the side-chain signal exceeds the threshold, the delay repeats are silenced. When the signal drops below the threshold, the delay re peats reappear. For a description of how to set up sidechain routing, see the chapter “Audio effects” in the Oper ation Manual.

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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 following parameters are available:

Parameter Description

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

Sync button The button below the Delay Time knob is used to switch

Feedback 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 Sets the stereo width for the left/right repeats. Turn

Mix Sets the level balance between the dry and the wet sig-

Ö The delay can also be controlled from another signal source via the side-chain input. When the side-chain sig nal exceeds the threshold, the delay repeats are silenced. When the signal drops below the threshold, the delay re peats reappear. For a description of how to set up sidechain routing, see the chapter “Audio effects” in the Oper ation Manual.

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

tempo sync on or off.

off low frequencies up to 800 knob activates/deactivates the filter.

off high frequencies from 20 button below the knob activates/deactivates the filter.

clockwise for a more pronounced stereo “ping-pong” ef fect.

nal. If PingPongDelay is used as a send effect, set this to the maximum value as you can control the dry/effect bal ance with the send.

Hz. The button below the

kHz down to 1.2 kHz. The

StereoDelay

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

The following parameters are available:

Parameter Description

Delay 1 & 2 If tempo sync is on, this is where you specify the base

Sync button The buttons below the Delay knobs are used to turn

Feedback 1 & 2

Filter Lo 1 & 2

Filter Hi 1 & 2

Pan 1 & 2 These controls are used to set the stereo position for

Mix 1 & 2 Use these controls to set the level balance between the

Ö The delay can also be controlled from another signal source via the side-chain input. When the side-chain signal exceeds the threshold, the delay repeats are silenced.

When the signal drops below the threshold, the delay re peats reappear. For a description of how to set up sidechain routing, see the chapter “Audio effects” in the Oper ation Manual.

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

tempo sync on or off for the respective delay. The Feedback controls set the number of repeats for

each delay. These filters affect the feedback loop and allow you to roll

off low frequencies up to 800 knobs activate/deactivate the filter.

These filters affect the feedback loop and allow you to roll off high frequencies from 20 buttons below the knobs activate/deactivate the filter.

each delay.

dry and the wet signal. If StereoDelay is used as a send effect, set them to the maximum value (100 control the dry/effect balance with the send.

Hz. The buttons below the

kHz down to 1.2 kHz. The

%) as you can

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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 following parameters are available:

Parameter Description

Amplifier pop-up menu

Drive Controls 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 Boosts or dampens the higher frequencies. Volume Controls the overall output level. Cabinet

pop-up menu

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

This pop-up menu is opened by clicking on the amplifier name shown at the top of the amp section. It allows you to select an amplifier model. The amp section can be by passed by selecting “No Amp”.

This pop-up menu is opened by clicking on the cabinet name shown at the top of the cabinet section. It allows you to select a speaker cabinet model. 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.

BitCrusher

If you are into lo-fi sound, BitCrusher is the effect for you. It offers the possibility of decimating and truncating the in put audio signal by bit reduction, to get a noisy, distorted sound. You can for example make a 24-bit audio signal sound like an 8 or 4-bit signal, or even render it completely garbled and unrecognizable.

The following parameters are available:

Parameter Description

Mode Allows you to select one of the four operating modes of

Sample Divider Sets the amount by which the audio samples are deci-

Depth Defines the bit resolution. A setting of 24 gives the high-

Output slider Governs the output level from BitCrusher. Drag the slider

Mix slider Regulates the balance between the output from Bit-

BitCrusher. In each mode the plug-in sounds differently. Modes I and III are nastier and noisier, while modes II and IV are more subtle.

mated. At the highest setting (65), nearly all of the information describing the original audio signal is eliminated, turning the signal into unrecognizable noise.

est audio quality, while a setting of 1 creates mostly noise.

upwards to increase the level.

Crusher and the original audio signal. Drag the slider upwards for a more dominant effect, and downwards if you want the original signal to be more prominent.

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DaTube

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

The following parameters are available:

Parameter Description

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

Balance Controls the balance between the signal processed by

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

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

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

Parameter Description

Spatial Changes the distortion characteristics of the left and right

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

channel, thus creating a stereo effect.

Grungelizer

Distortion

Distortion will add crunch to your tracks. The following parameters are available:

Parameter Description

Boost Increases the distortion amount. Feedback Feeds part of the output signal back to the effect input,

increasing the distortion effect.

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

distortion effect.

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Grungelizer adds noise and static to your recordings – kind of like listening to a radio with bad reception, or a worn and scratched vinyl record. The following parameters are available:

Parameter Description

Crackle Adds crackle to create that old vinyl record sound. The

RPM switch When emulating the sound of a vinyl record, this switch

Noise Regulates the amount of static noise added. Distort Adds distortion. EQ Turn this knob to the right to cut off the low frequencies,

AC Emulates a constant, low hum of AC current. Frequency

switch Timeline Regulates the amount of overall effect. The farther to the

farther to the right you turn the knob, the more crackle is added.

lets you set the RPM (revolutions per minute) speed of the record (33/45/78 RPM).

and create a more hollow, lo-fi sound.

Sets the frequency of the AC current (50 or 60 Hz), and thus the pitch of the AC hum.

right (1900) you turn the knob, the more noticeable the effect.

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SoftClipper (Cubase only)

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

The following parameters are available:

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 Allows you to adjust the amount of the second harmonic

Third Allows you to adjust the amount of the third harmonic in

overdriven sound just on the verge of distortion.

to the original signal.

in the processed signal.

the processed signal.

VST Amp Rack

The VST Amp Rack is a powerful guitar amp simulator. It offers a choice of amplifiers and speaker cabinets that can be combined with stomp box effects. The plug-in processes the mono sum of the channel and outputs a mono or stereo signal, depending on the track configuration.

At the top of the plug-in panel there are six buttons, arranged according to the position of the corresponding elements in the signal chain. These buttons open different pages in the Display section of the plug-in panel: PreEffects, Amplifiers, Cabinets, Post-Effects, Microphone Position, and Master.

Below the Display section, the selected amplifier is shown. The color and texture of the area below the amplifier indicate the selected cabinet.

Pre/Post-Effects

On the Pre-Effects and the Post-Effects pages, you can select up to six common guitar effects. On both pages the same effects are available, the only difference being the position in the signal chain (before and after the amplifier). On each page, every effect can be used once.

Each effect features an On/Off button known from stompbox effects, as well as individual parameters. The following effects and parameters are available:

Effect Option Description

Wah Wah Pedal Controls the filter frequency sweep. Volume Pedal Controls the level of the signal passing

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through the effect.

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Effect Option Description

Compressor Intensity Changes the intensity of the compressor ef-

Chorus Rate Allows you to set the sweep rate.

Width Determines the depth of the chorus effect.

Phaser Rate Allows you to set the sweep rate.

Width Determines the width of the modulation effect

Flanger Rate Allows you to set the sweep rate.

Feedback Determines the character of the flanger effect.

Mix Sets the level balance between the dry and

Tremolo Rate Allows you to set the modulation speed.

Depth Governs the depth of the amplitude modula-

Octaver Direct Adjusts the mix of the original signal and the

Octave 1 Adjusts the level of the signal that is gener-

Octave 2 Adjusts the level of the signal that is gener-

Delay Delay Sets the delay time in milliseconds.

Feedback Sets the number of repeats for the delay. Mix Sets the level balance between the dry and

Tape Delay Delay Tape Delay creates a delay effect known from

Feedback Sets the number of repeats for the delay. Mix Sets the level balance between the dry and

fect.

This parameter can be synchronized to the project tempo, see

Higher settings produce a more pronounced effect.

This parameter can be synchronized to the project tempo, see

between higher and lower frequencies.

This parameter can be synchronized to the project tempo, see

Higher settings produce a more “metallic” sounding sweep.

the wet signal.

This parameter can be synchronized to the project tempo, see

tion.

generated voices. A value of 0 means only the generated and transposed signal is heard. By raising this value, more of the original signal is heard.

ated one octave below the original pitch. A setting of 0 means that the voice is muted.

ated two octaves below the original pitch. A setting of 0 means that the voice is muted.

This parameter can be synchronized to the project tempo, see

the wet signal.

tape machines. The Delay parameter sets the delay time in milliseconds. This parameter can be synchronized to the project tempo, see

the wet signal.

“Sync mode” on page 13.

Effect Option Description

Tape Ducking Delay

Overdrive Drive Overdrive creates a tube-like overdrive effect.

Fuzz Boost Fuzz creates a rather harsh distortion effect.

Gate Threshold Determines the level where Gate is activated.

Equalizer Low Changes the level of the low-frequency por-

Reverb Type A convolution-based reverb effect. The Type

Delay Tape Ducking Delay creates a delay effect

Feedback Sets the number of repeats for the delay. Duck Works like an automatic mix parameter. If the

Tone Works as a filter effect on the added harmon-

Level Adjusts the output level.

Tone Works as a filter effect on the added harmon-

Level Adjusts the output level.

Release Sets the amount of time it takes for the gate to

Middle Changes the level of the mid-frequency por-

High Changes the level of the high-frequency por-

Mix Sets the level balance between the dry and

known from tape machines with a ducking pa rameter. The Delay parameter sets the delay time in milliseconds. This parameter can be synchronized to the project tempo, see

level of the input signal is high, the portion of the effect signal is lowered, i. internal mix value). If the level of the input signal is low, the portion of the effect signal is raised (high internal mix value). This way the delayed guitar signal stays rather dry during loud or intensely played passages.

The higher the Drive value, the more harmon ics are being added to the output signal of this effect.

ics.

The higher the Boost value, the more distor tion is being created.

ics.

Signal levels above the set threshold trigger the gate to open, and signal levels below the set threshold close the gate.

close.

tion of the incoming signal (from -12 dB to 12

dB).

tion of the incoming signal (from -12 dB to

dB).

tion of the incoming signal (from -12 dB to

dB).

parameter allows you to switch between dif ferent reverb types (Studio, Hall, Plate, and Room).

the wet signal.

“Sync mode” on page 13.

e. ducked (low

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Sync mode

For some controls, the sync mode can be activated to synchronize the corresponding parameter with the tempo of the host application. These plug-in parameters are then used to specify the base note value for tempo syncing (1/1 to 1/32, straight, triplet, or dotted).

The names of these parameters are underlined. Click on a control knob to activate/deactivate tempo sync. An LED at the top right of the knob indicates that Sync mode is active. You can then select a base note value for tempo syncing from the pop-up menu above the control.

Using effects

• To insert a new effect, click the plus button that appears when hovering with the mouse over an empty plug-in slot or over one of the arrows before or after a used effect slot.

• To remove an effect from an effect slot, click on the effect name and select “None” from the pop-up menu.

• To change the order of the effects in the chain, click on an effect and drag it to another position.

• To activate or deactivate an effect, click on the pedallike button below the effect’s name.

When an effect is active, the LED next to the button is lit.

Pre-effects are always mono, while post-effects can be mono or stereo, depending on the track configuration.

Ö Using the Quick Controls in Cubase, you can conveniently set up an external MIDI device such as a foot controller to control the VST Amp Rack effects. For more information, see the chapter “Track Quick Controls” in the Operation Manual.

Amplifiers

The amps available on the Amplifiers page were modeled on real-life amplifiers. Each amp features settings typical for guitar recording, such as gain, equalizers, and master volume. The sound-related parameters (Bass, Middle, Tre ble, and Presence) have a significant impact on the overall character and sound of the corresponding amp.

The following amp models are available:

• Plexi – Classic British rock tone; extremely transparent sound,

very responsive.

• Plexi Lead – British rock tone of the 70’s and 80’s.

• Diamond – The cutting edge hard rock and metal sounds of

the 90’s.

• Blackface – Classic American clean tone.

• Tweed – Clean and crunchy tones; originally developed as a

bass amp.

• Deluxe – American crunch sound coming from a rather small

amp with a big tone.

• British Custom – Produces the sparkling clean or harmoni-

cally distorted rhythm sounds of the 60’s.

The different amps keep their settings when you switch models. However, if you want to use the same settings after reloading the plug-in, you need to set up a preset.

Using amplifiers

• To switch amps on the Amplifiers page, simply click on the model that you want to use.

• Select “No Amplifier” if you only want to use the cabinets and effects.

Cabinets

The cabinets available on the Cabinets page simulate real-life combo boxes or speakers. For each amp, a corresponding cabinet type is available. However, you can combine amps and cabinets at will.

Using cabinets

• To switch cabinets on the Cabinets page, simply click on the model that you want to use. Select “No Cabinet” if you only want to use the amps and effects.

• If you select “Link Amplifier & Cabinet Choice”, the plug-in automatically selects the cabinet corresponding to the selected amp model.

Microphone Position

On the Microphone Position page, you can choose be-

tween 7 positions to place the microphone. These positions result from two different angles (center and edge) and three different distances from the speaker, as well as an additional center position at an even greater distance from the speaker.

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You can choose between two microphone types: a largediaphragm condenser microphone and a dynamic microphone. Crossfading between the characteristics of the two microphones is also possible.

Placing the microphone

• To select a microphone position, simply click on the corresponding ball in the graphic.

The selected position is marked in red.

• To select one of the microphone types or blend between the two types, turn the Mix control between the two microphones.

View settings

Two differents views for the VST Amp Rack plug-in panel are available: the default view and a compact view, which takes up less screen space.

In the default view, you can use the top buttons to open the corresponding page in the Display section above the amp controls. In the compact view the page display is hid den from view. You can still change the amp settings and switch amps or cabinets using the mouse wheel.

• In the default view, you can horizontally resize the plugin panel by clicking and dragging the edges or corners.

Master

Use the Master page to fine-tune the sound. There is an additional three-band Equalizer, a Tuner, and a Master level control for the output of the plug-in.

Using the Master controls

• To activate/deactivate the Equalizer, click the pedal-like On/Off button.

When the Equalizer is active, the LED next to the button is lit.

• To activate/deactivate an equalizer band, click the corresponding Gain knob.

When a band is active, the LED to the left of the Gain knob is lit.

• To tune your guitar strings, click the pedal-like On/Off button to activate the Tuner and play a string.

When the correct pitch is displayed and the row of LEDs below the digital is green, the string is tuned correctly. The more red LEDs on the left/ right are lit, the lower/higher the pitch.

• To mute the output signal of the plug-in, click the pedallike Master button.

When the LED is off, the output is muted. Use this to tune your guitar in silence, for example.

• To change the volume of the output signal, use the Level control in the Master section.

Using the hover controls

Hover controls are buttons that become visible on the plug-in frame if the mouse pointer is positioned somewhere on the plug-in panel.

Switching between default and compact view

• To toggle between the different views, click the down/ up arrow button (Show/Hide Extended Display) at the top center of the plug-in frame.

Changing the amplifier and cabinet selection in the compact view

In the compact view, a hover control on the lower border of the plug-in frame allows you to select different amplifier and cabinet models.

• To select a different amplifier or cabinet, click the name and select a different model from the pop-up menu.

• To lock the amplifier and cabinet combination, activate the “Link/Unlink Amplifier & Cabinet Choice” button.

If you now select another amp model, the cabinet selection follows. However, if you select a different cabinet model, the lock is deactivated.

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Previewing effect settings

In both views, you can show a preview of the pre- and post-effects that you selected on the corresponding pages:

• Click and hold the Show Pre-Effects or Show Post-Effects button at the bottom left or right (respectively) of the plug-in frame.

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. Com pressor 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 Re lease parameter.

The following parameters are available:

Parameter Description

Threshold (-60 to 0 dB)

Ratio (1:1 to 8:1)

Soft Knee button

Make-up (0 to 24 dB or Auto mode)

Attack (0.1 to

ms)

100

Hold (0 to

ms)

5000

Release (10 to

ms or

1000 Auto mode)

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

Live button When this button is activated, the “look ahead” feature of

Ö The compression can also be controlled from another

Determines the level where Compressor “kicks in”. Signal levels above the set threshold are affected, but signal lev els below are not processed.

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

dB.

If this button is off, signals above the threshold are compressed instantly according to the set ratio (hard knee). When Soft Knee is activated, the onset of compression is 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 automatically ad justed for gain loss.

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) passes through unpro cessed.

Sets the time the applied compression will affect the signal after exceeding the threshold. Short hold times are useful for “DJ-style” ducking, while longer hold times are required for music ducking, e. when working on a documentary film.

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 auto matically find an optimal release setting that varies depending on the audio material.

Determines whether the input signal is analyzed 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 per cussive material, with a lot of transient peaks.

Compressor is disengaged. Look ahead produces more accurate processing, but adds a certain amount of latency as a trade-off. When Live mode is activated, there is no la tency, which might be better for “live” processing.

signal source via the side-chain input. When the sidechain signal exceeds the threshold, the compression is triggered. For a description of how to set up side-chain routing, see the chapter “Audio effects” in the Operation Manual.

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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.

The following parameters are available:

Parameter Description

Reduction Controls the intensity of the de-essing effect. Threshold When the Auto Threshold option is deactivated, you can

Auto The Auto Threshold function automatically and continu-

Release Sets the amount of time it takes for the de-essing effect

Level meters Indicate the dB values of the input (IN) and output (OUT)

use this control to set a threshold for the incoming signal level, above which the plug-in starts to reduce the sibilants.

ally chooses an optimum threshold setting independent of the input signal. The Auto Threshold function does not work for low-level signals (< -30 such a file, set the threshold manually.

to return to zero when the signal drops below the thresh old value.

signals as well as the value by which the level of the sibi lant (or s-frequency) is reduced (GR). The gain reduction meter shows values between 0

dB (the s-frequency level is lowered by 20 dB).

-20

db peak level). To reduce the sibilants in

dB (no reduction) and

Positioning the DeEsser in the signal chain

When recording a voice, the de-esser’s position in the signal chain is usually located after the microphone preamp and before a compressor/limiter. This keeps the compressor/limiter from unnecessarily limiting the overall signal dynamics.

EnvelopeShaper

EnvelopeShaper can be used to cut or boost the gain of the Attack and Release phase of audio material. You can either use the knobs or drag the breakpoints in the graph ical 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 to 20 dB) Changes the gain of the Attack phase of the

Length (5 to 200 ms) Determines the length of the Attack phase.

Release (-20 to 20 dB) Changes the gain of the Release phase of the

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

signal.

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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 graphical display to change the Threshold and the Ratio parameter values.

The following parameters are available:

Parameter Description

Threshold (-60 to 0 dB)

Ratio (1:1 to 8:1)

Soft Knee button

Attack (0.1 to

ms)

100

Hold (0 to

ms)

2000 Release

(10 to 1000

ms or

Auto mode)

Determines the level where expansion “kicks in”. Signal levels below the set threshold are affected, but signal lev els above are not processed.

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

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

Determines how fast Expander responds to signals below the set threshold. If the attack time is long, more of the early part of the signal (attack) passes through unpro cessed.

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.

Parameter Description

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

Live button When this button is activated, the “look ahead” feature of

Expander is disengaged. Look ahead produces more ac curate processing, but adds a certain amount of latency as a trade-off. When Live mode is activated, there is no la tency, which might be better for “live” processing.

Ö The expansion can also be controlled from another signal source via the side-chain input. When the side-chain signal exceeds the threshold, 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 set threshold level. As soon as the signal level exceeds the set threshold, the gate opens to let the signal through.

The following parameters are available:

Parameter Description

Threshold (-60 to 0 dB)

State LED Indicates whether the gate is open (LED lights up in

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 close the gate.

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

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Parameter Description

Filter buttons (LP, BP, and HP)

Side-Chain button

Center (50 Hz to 20000

Q-Factor (0.01 to 10000)

Monitor button Allows you to monitor the filtered signal. Attack (0.1 to

ms)

1000

Hold (0 to 2000 ms)

Release (10 to 1000 ms or Auto mode)

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

Live button When this button is activated, the “look ahead” feature of

When the Side-Chain button (see below) is activated, you 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 sidechain filter. The input signal can then be shaped accord ing to set filter parameters. Internal side-chaining can be useful for tailoring how the Gate operates.

When the Side-Chain button is activated, this sets the

Hz)

center frequency of the filter. When the Side-Chain button is activated, this sets the

resonance of the filter.

Sets the time it takes for the gate to open after being triggered. If the Live button (see below) is deactivated, it ensures 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.

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

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 au dio material.

Determines whether the input signal is analyzed according to Peak or RMS values (or a mixture of both). A value of 0 is pure Peak and 100 pure RMS. RMS mode oper ates 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 mate rial with few transients such as vocals, and Peak mode better for percussive material, with a lot of transient peaks.

Gate is disengaged. Look ahead produces more accurate processing, but adds a certain amount of latency as a trade-off. When Live mode is activated, there is no la tency, which might be better for “live” processing.

Limiter

Limiter is designed to ensure that the output level never exceeds a set output level, to avoid clipping in following devices. Limiter can adjust and optimize the Release pa rameter 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 following parameters are available:

Parameter Description

Input (-24 to +24 dB)

Output (-24 to +6 dB)

Release (0.1 to 1000 ms or Auto mode)

Allows you to adjust the input gain.

Determines the maximum output level.

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 var ies depending on the audio material.

Ö The gate can also be controlled from another signal source via the side-chain input. When the side-chain signal exceeds the threshold, the gate opens. For a description of how to set up side-chain routing, see the chapter “Audio effects” in the Operation Manual.

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Maximizer

Maximizer is 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 tube-like distortion to the signal.

The following parameters are available:

Parameter Description

Output (-24 to +6 dB)

Optimize (0 to 100)

Soft Clip button

Determines the maximum output level. Should normally be set to 0 (to avoid clipping).

Determines the loudness of the signal.

When this button is activated, Maximizer starts limiting (or clipping) the signal “softly”, at the same time generating harmonics which add a warm, tube-like characteristic to the audio material.

MIDI Gate

Gating, in its fundamental form, silences audio signals below a set threshold level. 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, how ever, is not triggered by threshold levels, but MIDI notes. Hence it needs both audio and MIDI data to function.

Setting up

To set up MIDI Gate, proceed as follows:

1. Select the audio to be affected by 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 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 effect.

This can be an empty MIDI track or a MIDI track containing data, it does not matter. However, if you wish to use MIDI Gate in realtime – as op posed to using a recorded part – 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 effect.

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

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5. Make sure the MIDI track is selected, and start playback.

6. 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 Determines how long it takes for the gate to open after

Hold Regulates how long the gate remains open after a note-

Release Determines how long it takes for the gate to close (in ad-

Note To Attack Determines to which extent the velocity values of the MIDI

Note To Release

Velocity To VCA

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

receiving a signal that triggers it.

on or note-off message (see Hold Mode below).

dition to the value set with the Hold parameter).

notes affect the attack. The higher the value, the more the attack time increases with high note velocities. Negative values give shorter attack times with high velocities. If you do not wish to use this parameter, set it to the 0 position.

Determines to which extent the velocity values of the MIDI notes affect the release. The higher the value, the more the release time increases. If you do not wish to use this parameter, set it to the 0 position.

Controls to which extent the velocity values of the MIDI notes determine the output volume. At a value of 127 the volume is controlled entirely by the velocity values, and at a value of 0 the velocities have no effect on the volume.

the gate only remains 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, the gate remains open for as long as the MIDI note plays, and then the Hold and Release parameters are applied.

MultibandCompressor (Cubase

The MultibandCompressor allows a signal to be split into a maximum of 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 charac teristics 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 num ber 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 cor ner 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 +/- 15 dB after compression.

only)

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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 break

point from which the line deviates from the straight diagonal will be the threshold point.

For each of the four bands the following compressor parameters are available:

Parameter Description

Threshold (-60 to 0 dB)

Ratio (1000 to 8000) (1:1 to 8:1)

Attack (0.1 to 100 ms)

Release (10 to 1000 ms or Auto mode)

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

Determines the amount of gain reduction applied to signals over the set threshold. A ratio of 3000 (3:1) means that for every 3 level increases by only 1

Determines how fast the compressor responds 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 thresh old level. If the Auto button is activated, the compressor will automatically find an optimal release setting that var ies depending on the audio material.

dB the input level increases, the output

dB.

The Output control

The Output knob controls the total output level that the MultibandCompressor passes on to Cubase. The range is from -24 to +24

dB.

VintageCompressor (Cubase only)

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

The available parameters work as follows:

Parameter Description

Input (-24 to 48 dB)

Output

(-48 to 24 dB) Attack

(0.1 to 100 ms)

Punch (On/Off)

Release (10 to 1000 ms or Auto mode)

In combination with the Output setting, this parameter determines the compression amount. The higher the in put gain setting and the lower the output gain setting, the more compression is applied.

Sets the output gain.

Determines how fast the compressor responds. If the attack time is long, more of the early part of the signal (attack) passes 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 set ting that varies depending on the audio material.

Ö The compression can also be controlled from another signal source via the side-chain input. When the sidechain signal exceeds the threshold, the compression is triggered. For a description of how to set up side-chain routing, see the chapter “Audio effects” in the Operation Manual.

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VSTDynamics

Limiter

Module Configuration

Gate

Compressor

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, con taining 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 set threshold level. As soon as the signal level exceeds the set threshold, the gate opens to let the signal through. The Gate trigger in put can also be filtered using an internal side-chain.

The following parameters are available:

Parameter Description

Threshold (-60 to 0 dB)

State LED Indicates whether the gate is open (LED lights up in

Side-Chain button

LP (low-pass), BP (band-pass), HP (high-pass)

Center (50 to 22000

Hz)

Q-Factor (0.001 to 10000)

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 close the gate.

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

This button activates the internal side-chain filter. You can use this to filter out parts of the signal that might otherwise trigger the gate in places you not 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.

Sets the center frequency of the filter.

Sets the resonance or width of the filter.

Parameter Description

Monitor (On/Off)

Attack (0.1 to 100 ms)

Hold (0 to 2000 ms)

Release (10 to 1000 ms or Auto mode)

Input gain meter Shows the input gain.

Allows you to monitor the filtered signal.

Sets the time it takes for the gate to open after being triggered.

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

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 material.

The Compressor section

The compressor reduces the dynamic range of the audio,

making softer sounds louder or louder sounds softer, or both. It works like a standard compressor with separate controls for threshold, ratio, attack, release and make-up gain. The compressor features a separate display that graphically illustrates the compressor curve shaped ac cording to the Threshold, Ratio and Make-Up Gain parameter settings. It also features meters for input gain and gain reduction and a program-dependent Auto feature for the Release parameter.

The available parameters work as follows:

Parameter Description

Threshold (-60 to 0 dB)

Ratio (1:1 to 8:1)

Make-Up (0 to 24 dB)

Attack (0.1 to 100 ms)

Release (10 to 1000 ms or Auto mode)

Graphical display

Determines the level where the compressor “kicks in”. Signal levels above the set threshold are affected, but signal levels below are not processed.

Determines the amount of gain reduction applied to signals above the set threshold. A ratio of 3:1 means that for every 3

dB the input level increases, the output level in-

creases by only 1 dB. This parameter is used to compensate for output gain

loss, caused by compression. When the Auto button is activated, gain loss is being compensated automatically.

Determines how fast the compressor responds to signals above the set threshold. If the attack time is long, more of the early part of the signal (attack) passes through un processed.

Use the graphical display to graphically set the Threshold and Ratio values. To the left and right of the graphical dis play you will find two meters that show the amount of input gain and gain reduction in dB.

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The Limiter section

The limiter is designed to ensure that the output level never exceeds a set threshold, 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. The limiter adjusts and optimizes these parameters automatically according to the audio material. You can also adjust the Release parameter manually.

The following parameters are available:

Parameter Description

Output (-24 to +6 dB)

Soft Clip button

Release (10 to 1000 ms or Auto mode)

Meters The three meters show the input gain (IN), the gain re-

Determines the maximum output level. Signal levels above the set threshold are affected, but signal levels be low are left unaffected.

If this button is activated, the limiter acts differently. When the signal level exceeds -6 iting (or clipping) the signal “softly”, at the same time generating harmonics which add a warm, tube-like char acteristic to the audio material.

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

duction (GR) and the output gain (OUT).

dB, Soft Clip starts lim-

The Module Configuration button

Using the Module Configuration button in the bottom right corner of the plug-in panel, you can set the signal flow or der 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)

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

dB, allowing for fine control of the frequency re-

to 12 sponse. 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 fre quency bands, or enter values numerically by clicking on a gain value at the top of the display.

• At the bottom of the window the individual frequency bands are shown in Hz.

• At the top of the display the amount of cut/boost is shown in dB.

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

Parameter Description

Output Controls the overall gain of the equalizer. Flatten button Resets all the frequency bands to 0 dB. Range Allows you to relatively adjust how much a set curve cuts

Invert button Inverts the current response curve. Mode pop-up

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

The filter mode set here determines how the various frequency band controls interact to create the response curve, see below.

dB.

About the filter modes

On the pop-up menu 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. The following filter modes are available:

Filter mode Description

True Resp Applies serial filters with an accurate frequency re-

Digi Stand In this mode the resonance of the last band depends on

Classic Applies a classic parallel filter structure where the re-

Variable Q Applies parallel filters where the resonance depends on

ConstQ u Applies parallel filters where the resonance of the first

ConstQ s Applies parallel filters where the resonance is raised

Resonant Applies serial filters where a gain increase of one band

sponse.

the sample rate.

sponse does not follow the set gain values accurately.

the amount of gain.

and last bands depends on the sample rate.

when boosting the gain and vice versa.

will lower the gain in adjacent bands.

StudioEQ

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

Making settings

1. Click the corresponding On button on the left of the plug-in panel to activate any or all of the 4 equalizer bands (Low, Mid 1, Mid 2, and High).

When a band is activated, the 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 on the numeric values and typing in new values.

• By using the mouse to drag points in the EQ curve display. When using the mouse to change the parameter settings,

the following modifier keys can be used:

Modifier key Description

– When no modifier key is pressed and you drag an EQ

[Shift] Keep the [Shift] key pressed and drag the mouse to

[Alt]/[Option] Keep the [Alt]/[Option] key pressed and drag the

[Ctrl]/[Command] Keep the [Ctrl]/[Command] key pressed and drag the

point in the display, the Gain and Frequency parame ters are adjusted simultaneously.

change the Q-factor of the corresponding EQ band.

mouse to change the frequency of the corresponding EQ band.

mouse to change the gain value of the corresponding EQ band.

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The following parameters are available:

Parameter Description

Band 1 Gain (-20 to +24 dB)

Band 1 Inv button Inverts the gain value of the filter. Use this button to fil-

Band 1 Freq (20 to 2000 Hz)

Band 1 Q-Factor (0.5 to 10)

Band 1 Filter mode

Band 2 Gain (-20 to +24 dB)

Band 2 Inv button Inverts the gain value of the filter (see the description

Band 2 Freq (20 to 20000 Hz)

Band 2 Q-Factor (0.5 to 10)

Band 3 Gain (-20 to +24 dB)

Band 3 Inv button Inverts the gain value of the filter (see the description

Band 3 Freq (20 to 20000 Hz)

Band 3 Q-Factor (0.5 to 10)

Band 4 Inv button Inverts the gain value of the filter (see the description

Band 4 Gain (-20 to +24 dB)

Band 4 Freq (200 to 20000 Hz)

Band 4 Q-Factor (0.5 to 10)

Sets the amount of cut/boost for the low band.

ter out unwanted noise. While looking for the frequency to omit, it sometimes helps to boost it in the first place (set the filter to positive gain). After you have found it, you can use the Inv button to cancel it out.

Sets the frequency of the low band.

Controls the width or resonance of the low band.

For the low band, you can select between three types of shelving filters, a Peak (band-pass), and a Cut (lowpass/high-pass) filter. When Cut mode is selected, the Gain parameter is fixed.

-Shelf I adds resonance in the opposite gain direction slightly above 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. Sets the amount of cut/boost for the mid 1 band.

of the Invert button for Band 1). Sets the center frequency of the mid 1 band.

Sets the width of the mid 1 band: the higher this value, the “narrower” the bandwidth.

Sets the amount of cut/boost for the mid 2 band.

of the Invert button for Band 1). Sets the center frequency of the mid 2 band.

Sets the width of the mid 2 band: the higher this value, the “narrower” the bandwidth.

of the Invert button for Band 1). Sets the amount of cut/boost for the high band.

Sets the frequency of the high band.

Controls the width or resonance of the high band.

Parameter Description

Band 4 Filter mode

Output (-24 to +24 dB)

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

For the high band, you can select between three types of shelving filters, a Peak, and a Cut filter. When Cut mode is selected, the Gain parameter is fixed.

-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 knob on the top right of the plug-in panel allows

you to adjust the overall output level.

adjusted, keeping the output level constant regardless of the EQ settings.

Filter plug-ins

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

DualFilter

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

The following parameters are available:

Parameter Description

Position Sets the filter cutoff frequency. If you set this to a nega-

Resonance Sets the sound characteristic of the filter. With higher val-

tive value, DualFilter will act as a low-pass filter. Positive values cause DualFilter to act as a high-pass filter.

ues, a ringing sound is heard.

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StepFilter

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

General operation

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

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 are set at the pointer position.

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

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.

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 settings are saved together in the 8 pattern slots.

• Use the Pattern Selector below the Resonance grid to select a new pattern.

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

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 slot, which is useful for creating variations on a pattern.

• Select the pattern you wish to copy, click the Copy button, select another pattern slot, and click Paste.

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

StepFilter parameters

The following parameters are available:

Parameter Description

Base Cutoff Sets the base filter cutoff frequency. Values set in the

Base Resonance Sets the base filter resonance. Values set in the Reso-

Glide This will apply glide between the pattern step values,

Filter mode Use this slider to select a filter mode: low-pass (LP),

Sync button When the Sync button to the right of the Sync pop-up

Sync pop-up menu (1/1 to 1/32, straight, triplet, or dotted)

Output slider Sets the overall volume. Mix slider Adjusts the mix between dry and processed signal.

Cutoff grid are relative to the Base Cutoff value.

nance grid are relative to the Base Resonance value. Note that very high Base Resonance settings can pro duce loud ringing effects at certain frequencies.

causing values to change more smoothly.

band-pass (BP), or high-pass (HP) (from left to right).

menu is activated (yellow), the pattern playback is syn chronized with the project tempo.

Use this pop-up menu to set the pattern beat resolution, i. e. what note values the pattern will play in relation to the tempo.

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ToneBooster

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 Sets the center filter frequency. Gain Allows you to adjust the gain of the selected frequency

range by up to 24 Width Sets the resonance of the filter. Mode selector Sets the basic operational mode of the filter; Peak or

Band Mode.

dB.

Tonic (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 cre ative tool rather than a tool to fix audio problems, it can add color and punch to your tracks while being light on CPU usage.

Tonic has the following properties:

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

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

low-pass, 12 dB band-pass, and 12 dB 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.

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Page 28

Filter

In the Filter section at the center of the plug-in panel, the following parameters are available:

Parameter Description

Mode pop-up menu

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

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

Drive Adds a soft, tube-like saturation to the sound. As with an

Mix Sets the balance between dry and effect signal. Channel

selector (Ch.).

Sets the filter type. Available filter types are: 24 dB low-

dB low-pass, 12 dB low-pass, 6 dB low-pass,

pass, 18

dB band-pass, and 12 dB high-pass.

ates is governed by the filter type.

onance puts the filter into self-oscillation.

analog filter, the amount of saturation also depends on

the input signal level.

Allows you to choose between mono or stereo operation.

When set to mono, the output signal of Tonic is mono re

gardless of the input signal.

Env Mod

In the Env Mod section, the following parameters are available:

Parameter Description

Mode pop-up menu

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, the envelope level modulates the

Tonic offers three types of envelope modulation:

“Follow” tracks the input signal’s volume envelope for dy-

namic 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 ad

dition, velocities higher than 80 add an accent to the en-

velope 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 Routing pop-up menu for

the track.

times result in slower rise times when the envelope is trig

gered.

times result in slower envelope tails.

filter cutoff level.

LFO speed. A rather stunning effect.

X/Y Pad

In the X/Y Pad at the bottom left of the plug-in panel, the following parameters are available:

Parameter Description

X Par pop-up menu

Y Par pop-up menu

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

Sets the parameter to be modulated on the x-axis of the XY Pad. All of Tonic’s parameters are available as desti nations.

Sets the parameter to be modulated on the y-axis of the XY Pad.

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

LFO Mod

In the LFO Mod section, the following parameters are available:

Parameter Description

Mode pop-up menu

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

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

Smooth Controls the smoothing of the LFO steps. This works like

Morph Controls the playback value of the LFO step sequencer. It

Steps pop-up menu

Presets pop-up menu

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

Sets the direction of the step LFO modulation. The available modes are: Forward, Reverse, Alternating, and Random.

ter cutoff level.

is always in sync with the project tempo. An example: at a rate of 4.00 steps per beat in a 4/4 time signature, the step sequencer advances in 16th notes. At a rate of 4.00 beats per step in a 4/4 time signature the LFO advances only one step per bar. Note that the current LFO Rate is shown in the field below the Env Mod section.

a glide effect applied to the filter cutoff.

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 returns to its original setting.

Sets the number of steps played in sequence. Deactivated steps are grayed out in the Step Matrix.

Offers a number of step LFO waveform patterns. 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.

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Page 29

WahWah

Mastering – UV22HR

WahWah is a variable slope band-pass filter that can be auto-controlled by a side-chain signal or via MIDI model

ing 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 lies at 50.

The following parameters are available:

Parameter Description

Pedal Controls the filter frequency sweep. Pedal Control

(MIDI) pop-up menu

Freq Lo/Hi Set the frequency of the filter for the Lo and Hi Pedal po-

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

Gain Lo/Hi Set the gain of the filter for the Lo and Hi Pedal positions. Filter Slope

selector

Allows you to choose the MIDI controller that is used to control the plug-in. Set this to “Automation” if you do not want to use MIDI realtime control.

sitions.

Pedal positions.

Allows you to choose between two filter slope values:

dB or 12 dB.

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

The UV22HR 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 Manual.

The following parameters are available:

Option Description

Bit Resolution The UV22HR supports dithering to multiple resolutions:

Hi Try this first, it is the most “all-round” setting. Lo This applies a lower level of dither noise. Auto black When this is activated, the dither noise is gated (muted)

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

during silent passages in the material.

Dithering should always be applied post-fader on an output bus.

MIDI control

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

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

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

The included effect plug-ins

Page 30

Modulation plug-ins

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

AutoPan

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

The following parameters are available:

Parameter Description

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

Sync button The button below the Rate knob is used to switch tempo

Width Sets the depth of the auto-pan effect. Waveform

Shape selector

Ö The Width parameter can also be controlled from another signal source via the side-chain input. For a description of how to set up side-chain routing, see the chapter “Audio effects” in the Operation Manual.

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.

sync on or off.

Allows you to select the modulation waveform. A sine and a triangle waveform are available.

Chopper

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

The following parameters are available:

Parameter Description

Waveform buttons

Depth Sets the depth of the Chopper effect. This can also be

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

Sync button The button above the Speed knob is used to switch

Stereo/Mono button

Mix Sets the level balance between the dry and the wet sig-

Set the modulation waveform.

set by clicking in the graphical display.

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 tremolo/auto-pan speed can be set freely with the Speed knob.

tempo sync on (button lights up) or off. Determines whether the Chopper works as an auto-pan-

ner (button set to “Stereo”) or a tremolo effect (button set to “Mono”).

nal. If Chopper is used as a send effect, this should be set to the maximum value.

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Page 31

Chorus

This is a single-stage chorus effect. It works by doubling whatever is sent into it with a slightly detuned version (see

“StudioChorus (Cubase only)” on page 35).

also The following parameters are available:

Parameter Description

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

Sync button The button below the Rate knob is used to switch tempo

Width Determines the depth of the chorus effect. Higher set-

Waveform Shape selector

Spatial Sets the stereo width of the effect. Turn clockwise for a

Mix Sets the level balance between the dry and the wet sig-

Delay Affects the frequency range of the modulation sweep by

Filter Lo/Hi Allow you to roll off low and high frequencies of the effect

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.

sync on or off.

tings produce a more pronounced effect. Allows you to select the modulation waveform, altering

the character of the chorus sweep. A sine and a triangle waveform are available.

wider stereo effect.

nal. If Chorus is used as a send effect, set this to the maximum value as you can control the dry/effect balance with the send.

adjusting the initial delay time.

signal.

Ö The modulation can also be controlled from another signal source via the side-chain input. When the sidechain signal exceeds the threshold, the modulation is con trolled 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 following parameters are available:

Parameter Description

Voices Allows you to select the number of voices (up to four). For

Spatial Spreads the added voices across the stereo spectrum.

Mix Sets the level balance between the dry and the wet sig-

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

Detune slider 1–4

Delay slider 1–4

Detune Governs the overall depth of the detuning for all voices. If

Natural button By clicking the Natural button below the Detune knob,

Detune – Humanize

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

Turn clockwise for a deeper stereo effect.

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

dB.

12 Controls the relative detune amount for each voice. Posi-

tive and negative values can be set, from -100 to 100. A value of zero means no detune for that voice.

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

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

you can change the pitch algorithm. Controls the amount of detune variation when Static De-

tune is deactivated. With Humanize, the detune is constantly modulated for a more natural effect. The value range is from 0 to 100 (strongest detune variation).

The included effect plug-ins

Page 32

Parameter Description

Static Detune button

Delay Governs the overall depth of the delay for all voices. If set

Delay – Humanize

Static Delay button

Use this button to activate/deactivate the Static Detune function. If activated, the set detune amount is static, and the Humanize knob is grayed out.

to zero, no delay takes place regardless of the Delay slider settings.

Controls the amount of delay variation when Static Detune is deactivated. With Humanize, the delay is constantly modulated for a more natural effect. The value range is from 0 to 100 (strongest delay variation).

Use this button to activate/deactivate the Static Delay function. If activated, the set delay amount is static, and the Humanize knob is grayed out.

Flanger

Parameter Description

Mix Sets the level balance between the dry and the wet sig-

Waveform Shape selector

Delay Affects the frequency range of the modulation sweep by

Manual knob Allows you to change the sweep position manually when

Manual button Use this button to activate/deactivate the Manual func-

Filter Lo/Hi Allow you to roll off low and high frequencies of the effect

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

Allows you to select the modulation waveform, altering the character of the flanger sweep. A sine and a triangle waveform are available.

adjusting the initial delay time.

the Manual button is deactivated. The value range is from 0 to 100.

tion. If activated, the flanger sweep is static, i. e. no modulation takes place.

signal.

Flanger is a classic flanger effect with added stereo enhancement.

The following parameters are available:

Parameter Description

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

Sync button The button below the Rate knob is used to switch tempo

Range Lo/Hi Set the frequency boundaries for the flanger sweep. Feedback Determines the character of the flanger effect. Higher

Spatial Sets the stereo width of the effect. Turn clockwise for a

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.

sync on or off.

settings produce a more “metallic” sounding sweep.

wider stereo effect.

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Page 33

Metalizer

Phaser

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

The following parameters are available:

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 be

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

Mono button When this is activated, the output of Metalizer is mono. Speed If tempo sync is on, this is where you specify the base

Sync button The button above the Speed knob is used to switch

Output slider Sets the overall volume. Mix slider Sets the level balance between the dry and the wet sig-

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

more noticeable with high Feedback settings.

alizer works 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.

tempo sync on (button lights up) or off.

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

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

The following parameters are available:

Parameter Description

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

Sync button The button below the Rate knob is used to switch tempo

Width Determines the width of the modulation effect between

Feedback Determines the character of the phaser effect. Higher

Spatial When using multi-channel audio, the Spatial parameter

Mix Sets the level balance between the dry and the wet sig-

Manual knob Allows you to change the sweep position manually when

Manual button Use this button to activate/deactivate the Manual func-

Filter Lo/Hi Allow you to roll off low and high frequencies of the effect

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.

sync on or off.

higher and lower frequencies.

settings produce a more pronounced effect.

creates a 3-dimensional impression by delaying modula tion in each channel.

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

the Manual button is deactivated. The value range is from 0 to 100.

tion. If activated, the flanger sweep is static, i. e. no modulation takes place.

signal.

The included effect plug-ins

Page 34

RingModulator

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.

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

The following parameters are available:

Parameter Description

Oscillator – LFO Amount

Oscillator – Env. Amount

Oscillator – Waveform buttons

Oscillator – Range slider

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). Pos itive 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.

Allows you to select the oscillator waveform; square, sine, saw, or triangle.

Determines the frequency range of the oscillator in Hz.

Parameter Description

Oscillator – Frequency

Oscillator – Roll-Off

LFO – Speed Sets the LFO speed. LFO –

Amount

Env.

LFO – Waveform

LFO – Invert Stereo

Envelope Generator section – Attack and Decay

Lock L<R button

Output slider Sets the overall volume. Mix slider Adjusts the mix between dry and processed signal.

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 harmon ically 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, at 0 plied. With negative values, a loud input signal slows down the LFO, whereas positive values are used to speed it up at loud input signals.

Allows you to select the LFO waveform; square, sine, saw, or triangle.

Inverts the LFO waveform for the right channel of the oscillator, which produces a wider stereo perspective for the modulation.

The Envelope Generator section controls how the input signal is converted to envelope data, which can then be used to control oscillator pitch and LFO speed. It has two main controls: Attack controls how fast the envelope output level rises in response to a rising input signal. Decay controls how fast the envelope output level falls in response to a falling input signal.

When this button is enabled, the L and R input signals are merged, and produce the same envelope output level for both oscillator channels. When disabled, each chan nel has its own envelope, which affects the two channels of the oscillator independently.

% no modulation is ap-

Rotary

The Rotary plug-in simulates the classic effect of a rotating speaker. A rotary speaker cabinet features speakers rotating at variable speeds to produce a swirling chorus effect, commonly used with organs. Rotary features all the parameters associated with the real thing.

The included effect plug-ins

Page 35

The following parameters are available:

Parameter Description

Speed selector (Stop/ Slow/Fast)

Speed Change Mode

Speed Mod When the Slow/Fast setting is set to variable control, this

MIDI controller pop-up menu

Overdrive Applies a soft overdrive or distortion. CrossOver Sets the crossover frequency (200 to 3000 Hz) between

Horn – Slow Allows for a fine adjustment of the high rotor Slow speed. Horn – Fast Allows for a fine adjustment of the high rotor Fast speed. Horn – Accel. Allows for a fine adjustment of the high rotor acceleration

Horn – Amp

Mod

Horn –

Mod

Freq Bass – Slow Allows for a fine adjustment of the low rotor Slow speed. Bass – Fast Allows for a fine adjustment of the low rotor Fast speed. Bass – Accel. Allows for a fine adjustment of the low rotor acceleration

Bass – Amp

Mod Bass – Level Adjusts the overall bass level. Microphones –

Phase Microphones –

Angle Microphones –

Distance Output Allows you to adjust the overall output level. Mix Allows you to adjust the mix between dry and processed

Allows you to control the speed of the Rotary in three steps.

Allows you to select whether the Slow/Fast setting is a switch (left) or a variable control (right). When switch mode is selected and Pitchbend is the controller, the speed will switch with an up or down flick of the bender. Other controllers switch at MIDI value 64.

allows you to select the rotary speed, from 0 (Stop) to 100 (Fast).

Allows you to choose the MIDI controller that is used to control the plug-in. Set this to “Automation” if you do not want to use MIDI realtime control.

the low and high frequency loudspeakers.

time. Controls the high rotor amplitude modulation.

Controls the high rotor frequency modulation.

time. Adjusts the modulation depth of the amplitude.

Allows you to adjust the phasing amount in the sound of the high rotor.

Sets the simulated microphone angle. 0 = mono, 180 = one mic on each side.

Sets the simulated microphone distance from the speaker in inches.

signals.

Directing MIDI to the Rotary

For realtime MIDI control of the Speed parameter, MIDI must be directed to the Rotary.

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

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

StudioChorus (Cubase only)

The StudioChorus plug-in is a two stage chorus effect which adds short delays to the signal and pitch modulates the delayed signals to produce a “doubling” effect. The two separate stages of chorus modulation are completely independent and are processed serially (cascaded).

For each stage the following parameters are available:

Parameter Description

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

Sync button The button below the Rate knob is used to switch

Width Determines the depth of the chorus effect. Higher set-

Waveform Shape selector

Spatial Sets the stereo width of the effect. Turn clockwise for a

Mix Sets the level balance between the dry and the wet sig-

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.

tempo sync on or off.

tings produce a more pronounced effect. Allows you to select the modulation waveform, altering

the character of the chorus sweep. A sine and a trian gle waveform are available.

wider stereo effect.

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

The included effect plug-ins

Page 36

Parameter Description

Delay Affects the frequency range of the modulation sweep

Filter Lo/Hi Allow you to roll off low and high frequencies of the ef-

by adjusting the initial delay time.

fect signal.

Ö The modulation can also be controlled from another signal source via the side-chain input. When the sidechain signal exceeds the threshold, the modulation is 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.

Parameter Description

Sync button The button above the Speed knob is used to switch

On button Turns modulation of the pitch parameter on or off. Mono button Governs whether the output is stereo or mono. Output slider Allows you to adjust the output level of the effect. Mix slider Sets the level balance between the dry and the wet sig-

tempo sync on (button lights up) or off.

nal.

Ö Note that clicking and dragging in the display allows you to adjust the Tone and Depth parameters at the same time!

Tranceformer

Tranceformer is a ring modulator effect, in which the incoming audio is ring modulated by an internal, variable frequency oscillator, producing new harmonics. A second oscillator can be used to modulate the frequency of the first oscillator, in sync with the Song tempo if needed.

The following parameters are available:

Parameter Description

Waveform buttons

Tone Sets the frequency (pitch) of the modulating oscillator

Depth Governs the depth of the pitch modulation. Speed If tempo sync is on, this is where you specify the base

Allow you to select a pitch modulation waveform.

to 5000 Hz).

Tremolo

Tremolo produces amplitude (volume) modulation. The following parameters are available:

Parameter Description

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

Sync button The button below the Rate knob is used to switch tempo

Depth Governs the depth of the amplitude modulation. Spatial Adds a stereo effect to the modulation. Output Allows you to adjust the output volume.

note value for tempo-syncing the effect (1/1 to 1/32, straight, triplet, or dotted). If tempo sync is off, the modulation speed can be set freely with the Rate knob.

sync on or off.

The included effect plug-ins

Page 37

Vibrato

The Vibrato plug-in produces pitch modulation. The following parameters are available:

Parameter Description

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

Sync button The button below the Rate knob is used to switch tempo

Depth Governs the depth of the pitch modulation. Spatial Adds a stereo effect to the modulation.

note value for tempo-syncing the effect (1/1 to 1/32, straight, triplet, or dotted). If tempo sync is off, the modulation speed can be set freely with the Rate knob.

sync on or off.

Pitch Shift plug-ins

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

Octaver

This plug-in can generate two additional voices that track the pitch of the input signal one octave and two octaves below the original pitch, respectively. Octaver is best used with monophonic signals.

The following parameters are available:

Parameter Description

Direct Adjusts the mix of the original signal and the generated

Octave 1 Adjusts the level of the signal that is generated one oc-

Octave 2 Adjusts the level of the signal that is generated two oc-

voices. A value of 0 means only the generated and trans posed signal is heard. By raising this value, more of the original signal is heard.

tave below the original pitch. A setting of 0 means that the voice is muted.

taves below the original pitch. A setting of 0 means that the voice is muted.

The included effect plug-ins

Page 38

Pitch Correct

Pitch Correct automatically detects, adjusts and fixes slight pitch and intonation inconsistencies in monophonic vocal and instrumental performances in realtime. The ad vanced algorithms of this plug-in preserve the formants of the original sound thus allowing for natural sounding pitch correction without the typical “Mickey Mouse” effect.

Furthermore, you can use Pitch Correct creatively. You can create backing vocals, for example, by modifying the lead vocals or vocoder sounds by using extreme values. You can use an external MIDI controller, a MIDI track or the virtual keyboard to “play” a note or a scale of target pitches that determine the current scale notes to which the audio is shifted. This allows you to change your audio in a very quick and easy way, which is extremely useful for live performances. In the keyboard display, the original au dio will be displayed in blue while the changes are displayed in orange.

The following parameters are available:

Parameter Description

Correction – Speed

Correction – Tolerance

Correction – Transpose (-12 to 12)

Determines the smoothness of the pitch change. Higher values cause the pitch shift to occur immediately. 100 is a very drastic setting that is designed mainly for special effects (e.

g. the famous “Cher” effect).

Determines the sensitivity of analysis. A low Tolerance value lets Pitch Correct find pitch changes quickly. When the Tolerance value is high, pitch variations in the audio (e.

g. vibrato) will not be immediately interpreted as

note changes. With this parameter you can adjust (or “retune”) the pitch

of the incoming audio in semitone steps. You can set positive and negative values from -12 to 12. A value of zero means that the signal is not transposed.

Parameter Description

Scale Source – Internal

Scale Source – External MIDI Scale

Scale Source – External MIDI Note

Formant – Shift (-60 to 60)

Formant – Optimize (Gen eral, Male, Female)

Formant – Preservation (On/Off)

Master Tuning Detunes the output signal. The default setting is 440 Hz.

If you choose the Internal option from the Scale Source pop-up menu, you can use the pop-up menu next to it to decide to which scale the source audio will be adapted. The following options are available: Chromatic: The audio will be pitched to the closest semitone. Major/Minor: The audio will be pitched to the major/minor scale specified in the pop-up menu to the right. This will be reflected on the keyboard display. Custom: The audio will be pitched to the notes that you specify by clicking the desired keys on keyboard display. To reset the keyboard, click on the orange line below the display.

Select this option if you want the audio to be shifted to a scale of target pitches, using an external MIDI controller, the Virtual Keyboard or a MIDI track. Note that you have to assign the audio track as the output of your MIDI track and that the Speed parameter has to be set to a value other than Off.

Select this option if you want the audio to be shifted to a target note, using an external MIDI controller, the Virtual Keyboard or a MIDI track. Note that you have to assign the audio track as the output of your MIDI track and that the Speed parameter has to be set to a value other than Off.

Changes the natural timbre, i. e. the characteristic frequency components of the source audio.

Allows you to specify the sound characteristics of the sound sources. While General is the default setting,

Male is designed for low pitches and Female for high pitches.

When set to Off, formants are raised and lowered with the pitch, provoking strange vocal effects. Higher pitch correction values result in “Mickey Mouse” effects, lower pitch correction values in “Monster” sounds. When set to On, the formants are kept, maintaining the character of the audio.

The included effect plug-ins

Page 39

Reverb plug-ins

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

REVerence (Cubase only)

REVerence is a convolution tool that allows you to apply room characteristics (reverb) to the audio. This is done by processing the audio signal according to an impulse response – a recording of an impulse in a room or another location that is used to recreate the characteristics of the room. As a result, the processed audio will sound as if it were played in the same location. Included with the plugin are top quality samples of real spaces to create rever beration.

Ö REVerence can be very demanding in terms of RAM. This is because the impulse responses that you load into the program slots are preloaded into RAM to guarantee an artifact-free switching between programs. Therefore you should always load only those programs that you need for a given task.

Using the program matrix

A program is the combination of an impulse response and its settings. These include reverb settings (see

the reverb settings” on page 40), EQ settings (see “Making EQ settings” on page 41), pictures (see “Loading pic- tures” on page 42), and output settings (see “Making

“Changing

output settings” on page 42). The program matrix allows

you to load programs and to view the name of the current program, i.

e. the impulse response (see “Working with

custom impulse responses” on page 42).

The following parameters are available:

Parameter Description

Program name In the upper left corner of the plug-in panel, either the

Browse button This button opens a browser window showing the avail-

Import button Click this button to load your own impulse response files

Program slots (1 to 36)

Smooth Parameter Changes button

Store button Stores the active impulse response and its settings as a

Recall button Reloads the selected program. Use this to reset a pro-

Erase button Removes the selected program from the matrix.

name of the loaded impulse response file or the name of the program is shown. After loading an impulse response, its number of channels and the length in seconds are dis played for a few seconds.

able programs. When you select a program in the browser, it is loaded into the active program slot. To be able to filter the list of impulse responses in the browser window, e. you can activate the Filters section (by clicking the “Set Up Window Layout” button at the bottom left of the win dow).

from disk. The files should have a maximum length of 10 seconds. Longer files are automatically cut. For more infor mation, see “Working with custom impulse responses” on

page 42.

Into these slots you can load all the impulse responses (programs) that you want to work with in a session. The selected program slot is indicated by a (blinking) white frame. Occupied slots are shown in a different color. Double-clicking an empty program opens a browser win dow, showing the available programs. Double-clicking an occupied program slot loads the corresponding impulse response into REVerence (“Recall”).

The “Smooth Parameter Changes” button is located between the program slots and the Store/Recall/Erase buttons. If it is activated, a crossfade is performed when switching programs. Leave this button deactivated while looking for a suitable program or an appropriate setting for an impulse re sponse. Once you have set up the program matrix to your liking, activate the button to avoid hearing artifacts when switching between programs.

program.

gram to its default settings.

g. by room type or the number of channels,

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Programs vs. presets

You can save your REVerence settings as VST plug-in presets or programs. The differences between the two and the advantages are described in the following.

Both presets and programs use the file extension .vstpreset and appear in the same category in the MediaBay (Plug-In Presets), but they are represented by different icons:

Icon Description

A REVerence preset contains all settings and parameters for the plug-in, that is all the loaded impulse responses along with their parameter settings and positions in the program matrix. Presets are loaded via the Presets pop-up menu at the top of the plug-in panel.

A REVerence program only contains the settings related to a single impulse response. Programs are loaded and managed via the program matrix.

Presets

Presets are useful in the following situations:

• When you want to save a complete setup with different impulse responses for later use (e. g. different setups for explosion sounds that can be reused for other scenes or movies).

• When you want to save different parameter sets for the same impulse response so that you can later choose the set that best suits your needs.

Programs

Programs offer the following advantages:

• Up to 36 programs can be loaded into the program matrix for instant recall.

• A program provides a quick and easy way to save and recall a subset of the plug-in parameters (i. for a single impulse response), allowing for short loading times.

• When automating a project and loading a REVerence program, only two automation events are written.

If load a plug-in preset instead (which contains a lot more settings than a program), a lot of unnecessary automation data (for the settings that you did not use) is written.

e. the settings

Setting up programs

Proceed as follows:

1. In the program matrix, click on a program slot to select it.

A blinking white frame indicates that this program slot is selected.

2. Click the Browse button or click on the empty slot again to load one of the included programs.

You can also import a new impulse response file, see “Importing impulse

responses” on page 42.

3. In the browser that appears, select the program containing the impulse response that you want to use and click OK.

The name of the loaded impulse response is shown in the upper left corner of the REVerence panel.

4. Set up the REVerence parameters as needed and click the Store button to save the impulse response with the current settings as a new program.

5. Set up as many programs as you need (up to 36) by following the steps above.

Ö If you want to use your set of programs in other projects, save your settings as a plug-in preset using the Presets pop-up menu at the top of the plug-in panel.

Changing the reverb settings

The reverb settings allow you to change the characteristics of the room.

The following parameters are available:

Parameter Description

Front All values shown in the top row are for the front speakers.

Rear

Auto Gain button

Reverse button Reverses the impulse response. Pre-Delay Controls the amount of time between the dry signal and

Time Scaling Controls the reverb time. Size Determines the size of the simulated room.

If you are working with surround tracks up to 5.1, you can use this row to set up an offset for the rear channels.

When this button is activated, the impulse response is automatically normalized.

the onset of the reverb. With higher pre-delay values you can simulate larger rooms.

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Parameter Description

Level A level control for the impulse response. This governs the

ER Tail Split Sets a split point between the early reflections and the

ER Tail Mix Allows you to set up the relation of early reflections and

volume of the reverb.

tail, allowing you to determine where the reverb tail be gins. A value of 60 means that the early reflections will be heard for 60

tail. Values above 50 attenuate the early reflections and values below 50 attenuate the tail.

ms.

The impulse response display

The Display section allows you to view the impulse response details and to change the length of the response (trimming).

The following parameters are available:

Parameter Description

Play button/ Time Scaling wheel

Time Domain display

Spectrogram display

Information display

Activate Impulse Trimming button

Trim slider Allows you to trim the start and end of the impulse re-

When clicking the play button to apply the loaded impulse response, a short click is played. This provides a neutral test sound that makes it easier for you to know how differ ent settings influence the reverb characteristics. The Time Scaling wheel lets you adjust the reverb time.

Shows the waveform of the impulse response.

Shows the analyzed spectrum of the impulse response. Time is displayed along the horizontal axis, frequency along the vertical axis, and volume is represented by the color.

Shows additional information, e. g. the name of the program and the loaded impulse response, the number of channels, the length, and Broadcast Wave File informa tion.

Use this button at the bottom right of the Impulse display section to activate impulse trimming. The Trim slider is shown below the Impulse display.

sponse. Drag the front handle to trim the start of the impulse response, or the end handle to trim the reverb tail. You can also use the mouse wheel for trimming. Note that the impulse response will be cut without any fading.

Making EQ settings

In the Equalizer section you can tune the sound of the reverb.

The following parameters are available:

Parameter Description

EQ curve display Shows the EQ curve. You can use the EQ parameters

Activate EQ button

Low Shelf On button

Low Freq (20 to 500)

Low Gain (-24 to +24)

Mid Peak On button

Mid Freq (100 to 10000)

Mid Gain (-12 to +12)

Hi Shelf On button

Hi Freq

(5000 to 20000) Hi Gain

(-24 to +24)

below the display to change the EQ curve, or modify the curve manually by dragging the curve points.

This button to the right of the EQ parameters activates the EQ for the effect plug-in.

Activates the low shelf filter that boosts or cuts frequencies below the cutoff frequency by the specified amount.

Sets the frequency of the low band.

Sets the amount of cut/boost for the low band.

Activates the mid peak filter that creates a peak or notch in the frequency response.

Sets the center frequency of the mid band.

Sets the amount of cut/boost for the mid band.

Activates the high shelf filter that boosts or cuts frequencies above the cutoff frequency by the specified amount.

Sets the frequency of the high band.

Sets the amount of cut/boost for the high band.

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Loading pictures

In the Pictures section you can load graphics files to illustrate the setting, i. e. the recording location or microphone arrangement of the loaded impulse response. Up to five pictures can be loaded.

The following parameters are available:

Parameter Description

Output activity meter

Output slider Allows you to adjust the overall output level. Out

(-24 to +12) Mix

(0 to 100)

Indicates the overall level of the impulse response and its settings.

Raises or lowers the signal output of the plug-in.

Sets the level balance between the dry and the wet signal.

• To lock the dry/wet balance while browsing through the available presets and programs, activate the Lock button (padlock symbol) next to the MIX parameter.

The following parameters are available:

Parameter Description

Add button Opens a file dialog where you can navigate to the graph-

Next button If several pictures are loaded, you can click this button to

Remove button Deletes the active picture. Note that this will not remove

ics file that you want to import. JPG, GIF, and PNG file formats are supported.

display the next image.

the graphics file from your hard disk.

Ö Pictures are only referenced by the plug-in and will not be copied to the project folder.

Making output settings

In the Output section you can control the overall level and determine the dry/wet mix.

Working with custom impulse responses

In addition to working with the impulse responses included with REVerence, you can import your own impulse responses and save these as programs or presets. WAVE and AIFF files with a mono, stereo, true-stereo, or multichannel (up to 5.0) configuration are supported. If a multichannel file contains an LFE channel, this channel is ignored.

REVerence uses the same channel width as the track it is inserted on. When importing impulse response files with more channels than the corresponding track, the plug-in only reads as many channels as needed. If the impulse re sponse file contains less channels than the track, REVerence generates the missings channels (e. g. the center channel as a sum of the left and right channels). If the rear channels are missing (when importing a stereo response file onto a 4.0 track, for example), the left and right chan nels are also used for the rear channels. In this case you can use the Rear offset parameter to create more spatiality.

Importing impulse responses

To import impulse responses, proceed as follows:

1. In the program matrix, click the Import button.

2. Navigate to the file that you want to import, and click

Open.

The file is loaded into REVerence. The channels from an interleaved file are imported in the same order as in other areas of Cubase (e. Connections window), see below.

g. the VST

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3. Make the appropriate settings and add a picture, if available.

Pictures residing in the same folder as the impulse response file or in the parent folder are automatically found and displayed.

4. Click the Store button to save the impulse response and its settings as a program. That way you can recall the setup at any time.

The program slot turns blue, indicating that a program is loaded.

Ö When saving a program, the impulse response file itself is only referenced. It still resides in the same location as before and is not modified in any way.

5. Repeat these steps for any impulse response files that you want to work with.

REVerence reads input channels in the following order:

No. of input channels

1 L 2 L/R 3 L/R/C 4 L/R/LS/RS (if inserted on a track with a 4.0 channel con-

4 LL/LR/RL/RR (if inserted on a track with a stereo config-

5 L/R/C/LS/RS 6 L/R/C/LFE/LS/RS (LFE is being ignored.)

Channel order in REVerence

figuration, see below)

uration, see below)

True stereo

Impulse responses recorded as trues-stereo files enable you to create a very realistic impression of the corre

sponding room. REVerence can only process true-stereo impulse response files with the following channel configu ration (in exactly that order): LL, LR, RL, RR.

The channels are defined as follows:

Channel The signal from this

LL left source left microphone LR left source right microphone RL right source left microphone RR right source right microphone

source…

…was recorded with this microphone

Ö If your true-stereo impulse responses are only available as separate mono files, you can use the Export Audio Mixdown function in Cubase to create REVerence compli ant interleaved files (see the chapter “Export Audio Mixdown” in the Operation Manual).

By default, REVerence automatically works in true-stereo mode when the plug-in is inserted on a stereo track and you load a 4-channel impulse response.

Therefore, if you are working with surround files, that is, 4-channel impulse responses recorded with a Quadro configuration (L/R, LS/RS), you need to insert the plug-in on an audio track with a 4.0 configuration. On a stereo track these files would be processed in true-stereo mode, too.

So how can you prevent REVerence from unintentionally processing surround files in true-stereo mode? The answer is a “Recording Method” attribute that can be written to the iXML chunk of the corresponding impulse response file. Whenever you load an impulse response with a 4-channel configuration on a stereo track, REVerence searches the iXML chunk of the file. If the plug-in finds the Recording Method attribute, the following happens:

• If the attribute is set to “TrueStereo”, the plug-in works in true-stereo mode.

• If the attribute is set to “A/B” or “Quadro”, the plug-in works in normal stereo mode and processes only the L/R channels of the surround file.

Ö You can use the Attribute Inspector in the MediaBay to tag your own impulse response files with the Recording Method attribute. For more information, see the chapter “MediaBay” in the Operation Manual.

Relocating content

Once you have imported your own impulse responses in REVerence you can comfortably work with them on your computer. But what if you need to transfer your content to another computer, for example because you work some times with a PC and sometimes with a notebook, or you need to hand over a project to a colleague in the studio?

The factory content will not be a problem since it is also present on the other computer. For these impulse re sponses you just need to transfer your REVerence programs and presets to be able to access your setups.

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User content is a different matter, though. If you have

transferred your audio files to an external drive or a different hard disk location on the other computer, REVerence cannot access the impulse responses any more since the old file paths have become invalid.

To access your impulse responses again, proceed as follows:

1. Transfer you audio files to a location that you will be able to access from the second computer (i. e. an external hard disk).

If you keep the files in the same folder structure as on the first computer, REVerence will automatically find all files contained in this structure.

2. Transfer any REVerence presets or programs that you need to the second computer.

If you are unsure where the presets need to be stored, you can find the paths in the MediaBay (see the chapter “The MediaBay” in the Operation Manual).

3. Open REVerence on the second computer and try to load the preset or program that you want to work with.

The Locate Impulse Response dialog opens.

4. Navigate to the folder that contains your impulse responses. Click Open.

REVerence is now able to access all the impulse responses stored in this location.

The new path to these audio files has not been saved yet. To make the files permanently available without having to use the Locate dialog, you need to save your programs or presets under a different name.

RoomWorks (Cubase only)

RoomWorks is a highly adjustable reverb plug-in for creating realistic room ambience and reverb effects in stereo and surround formats. The CPU usage is adjustable to fit the needs of any system. From short room reflections to cavern-sized reverb, this plug-in delivers high quality re verberation.

The following parameters are available:

Parameter Description

Input – Lo Freq Determines the frequency at which the low-shelving filter

Input – Hi Freq Determines the frequency at which the high-shelving filter

Input – Lo

Gain

Input – Hi Gain Controls the amount of boost or cut for the high-shelving

Reverb – Pre-Delay

Reverb – Reverb Time

Reverb – Size Alters the delay times of early reflections to simulate

Reverb – Diffusion

Reverb – Width

Reverb – Variation button

Reverb – Hold button

Damping –

Freq

Lo Damping –

High Freq Damping –

Low Level

Damping – High Level

Envelope – Amount

takes effect. Both the high and low settings filter the input signal prior to reverb processing.

Controls the amount of boost or cut for the low-shelving filter.

filter. Controls how much time passes before the reverb is ap-

plied. This allows you to simulate larger spaces by increasing the time it takes for first reflections to reach the listener.

Allows you to set the reverb time in seconds.

larger or smaller spaces. Affects the character of the reverb tail. Higher values lead

to more diffusion and a smoother sound, while lower val ues lead to a clearer sound.

Controls the width of the stereo image. 100 % gives you full stereo reverb. At 0

Pressing this button generates a new version of the same reverb program using altered reflection patterns. This is helpful when certain sounds are causing odd ringing or undesirable results. Creating a new variation will often solve these issues. There are 1000 possible variations.

Pressing this button freezes the reverb buffer in an infinite loop (yellow circle around button). You can create some interesting pad sounds using this feature.

Determines the frequency below which low-frequency damping will occur.

Determines the frequency above which high-frequency damping will occur.

Affects the decay time of low frequencies. Normal room reverb decays quicker in the high- and low-frequency range than in the mid-range. Lowering the level percent age causes low frequencies to decay quicker. Values above 100 slowly than the mid-range frequencies.

Affects the decay time of high frequencies. Normal room reverb decays quicker in the high- and low-frequency range than in the mid-range. Lowering the level percent age causes high frequencies to decay quicker. Values above 100 slowly than the mid-range frequencies.

Determines how much the envelope attack and release controls affect the reverb itself. Lower values have a more subtle effect while higher values lead to a more drastic sound.

% cause low frequencies to decay more

% cause high frequencies to decay more

%, the reverb is all in mono.

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Parameter Description

Envelope – Attack

Envelope – Release

Surround – Distance

Surround – Rotate button

Surround – Balance

Output – Mix Determines the balance of dry (unprocessed) and wet

Output – Wet

only

button

Output – Efficiency

Output – Export button

Output – Output meter

The envelope settings in RoomWorks control how the reverb will follow the dynamics of the input signal in a fashion similar to a noise gate or downward expander. Attack determines how long it takes for the reverb to reach full volume after a signal peak (in milliseconds). This is similar to a pre-delay but the reverb is ramping up instead of starting all at once.

Determines how long after a signal peak the reverb can be heard before being cut off, similar to a gate’s release time.

This control is only available for surround configurations. With this parameter you can control where the virtual lis tening position is within the room. Positive values position the listener closer to the front of the room and negative values place the listener towards the rear of the room.

This button is only available for surround configurations. When active, the perspective of the room is shifted 90°.

This control is only available for surround configurations. Balance controls the relative levels between the forward and rear speakers. Positive values favor the front speak ers and negative values favor the rear speakers. When the Rotate option is activated, these relationships will shift 90°.

(processed) signal. When RoomWorks is used as an in sert for an FX channel, you will most likely want to set this to 100

% or use the wet only button.

This button defeats the mix parameter, setting the effect

% wet or affected signal. This button should nor-

to 100 mally be pressed when RoomWorks is being used as a send effect for an FX or group channel.

Determines how much processing power is used for RoomWorks. The lower the value, the more CPU re sources will be used, and the higher the quality of the reverb. Interesting effects can be created with very high Efficiency settings (>90

Determines if during audio export RoomWorks will use the maximum CPU power for the highest quality reverb. During export you may wish to keep a higher efficiency setting to achieve a specific effect. If you want the high est quality reverb during export, make sure this button is activated.

Indicates the level of the output signal.

%). Experiment for yourself.

RoomWorks SE

RoomWorks SE is a “lite” version of the RoomWorks plugin. This plug-in delivers high quality reverberation, but has fewer parameters and is less CPU demanding than the full version.

The following parameters are available:

Parameter Description

Pre-Delay Controls how much time passes before the reverb is ap-

Reverb Time Allows you to set the reverb time in seconds. Diffusion Affects the character of the reverb tail. Higher values lead

Hi Level Affects the decay time of high frequencies. Normal room

Lo Level Affects the decay time of low frequencies. Normal room

Mix Determines the blend of dry (unprocessed) signal to wet

plied. This allows you to simulate larger spaces by increasing the time it takes for first reflections to reach the listener.

to more diffusion and a smoother sound, while lower val ues lead to a clearer sound.

reverb decays quicker in the high- and low-frequency range than in the mid-range. Lowering the level percent age causes high frequencies to decay quicker. Values above 100 slowly than the mid-range frequencies.

reverb decays quicker in the high- and low-frequency range than in the mid-range. Lowering the level percent age causes low frequencies to decay quicker. Values above 100 slowly than the mid-range frequencies.

(processed) signal. When using RoomWorks SE inserted in an FX channel, you will most likely want to set this to 100

% cause high frequencies to decay more

% cause low frequencies to decay more

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Spatial + Panner plug-ins

This section contains descriptions of the plug-ins in the “Spatial + Panner” category.

MonoToStereo

This effect will turn a mono signal into a “pseudo-stereo” signal. The plug-in must be inserted on a stereo track playing a mono file.

The following parameters are available:

Parameter Description

Width Controls the width or depth of the stereo enhancement.

Delay Increases the amount of differences between the left and

Color Generates additional differences between the channels

Mono button Switches the output to mono, to check for possible un-

Turn clockwise to increase the enhancement.

right channels to further increase the stereo effect.

to increase the stereo effect.

wanted coloring of the sound which sometimes can occur when creating an artificial stereo image.

StereoEnhancer

This plug-in will expand the stereo width of (stereo) audio material. It cannot be used with mono files.

The following parameters are available:

Parameter Description

Width Controls the width or depth of the stereo enhancement.

Delay Increases the amount of differences between the left and

Color Generates additional differences between the channels

Mono button Switches the output to mono, to check for possible un-

Turn clockwise to increase the enhancement.

right channels to further increase the stereo effect.

to increase the stereo enhancement.

wanted coloring of the sound which sometimes can occur when enhancing the stereo image.

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Surround plug-ins (Cubase only)

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

Mix6To2

Mix6To2 lets you quickly mix down your surround mix format to stereo. You can control the levels of up to six surround channels and decide for each channel up to which level it will be included in the resulting mix.

Ö Mix6To2 does not simulate a surround mix or add any psycho-acoustical artifacts to the resulting output – it is simply a mixer. The plug-in should be placed in one of the post-fader insert effect slots for the output bus.

For each of the surround channels the following parameters are available:

• Two volume faders that govern how much of the signal will be included in the left and/or right channel of the output bus.

• A Link button that links the two volume faders.

• Two Invert buttons that allow you to invert the phase of the left and right channel of the surround bus.

For the Output bus the following parameters are available:

• A Link button that links the two Output faders.

• A Normalize button. If activated, the mixed output is normalized, i. e. the output level is automatically adjusted so that the loudest signal is as loud as possible without clipping.

MixerDelay

MixerDelay allows you to adjust and manipulate each individual channel in a surround track, group or bus.

• Above the individual channel controls you will find global buttons for turning off Mute, Solo and Invert Phase switches for all channels.

For each channel the following controls are available:

Parameter Description

Mute button Allows you to mute individual channels. Solo button Allows you to solo individual channels. Inv button Lets you invert the phase or polarity for individual chan-

Delay slider Allows you to delay individual speaker channels. The de-

Level slider Allows you to fine-tune the volume balance between the

Volume meter Shows the level of the input signal. Routing

section

Ö It is common for the center channel in a 5.1 speaker configuration to be closer to the mix position in order to accommodate large video monitors or projection screens. In cases like this, MixerDelay can be used to compensate for the center channel being too close. Simply adjust the delay for the center channel by the difference in distance (in cm) between it and the other speakers to the mix posi tion. You must delay the closer speaker so that the sound from it arrives at the same time as the sound from the more

nels.

lay times are shown in milliseconds and centimeters, making this feature very useful for distance compensation when playing back surround mixes on different speaker setups, etc.

surround channels.

Lets you select/switch the desired outputs for the channels quickly. You can assign the same output to several channels by holding down the [Alt]/[Option] key while se lecting. Note that there are also several channel routing presets available.

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distant speakers. Note that MixerDelay has a wide range

(up to 1000 ms) and fine adjustments are best made by numerically entering the delay time in centimeters for speaker alignment.

The MixerDelay is not a mixer – the number of outputs is the same as the number of inputs. If you need to mix down a surround signal to stereo, use the Mix6to2, Mix8to2 or MixConvert plug-ins.

Tools plug-ins

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

MultiScope (Cubase only)

MultiScope can be used for viewing the waveform, phase linearity or frequency content of a signal. There are three different modes:

• Oscilloscope (Ampl.)

• Phase Correlator (Scope)

• Frequency Spectrum Analyzer (Freq.)

Ö The Freeze button can be used to freeze the display in all three modes. Click it again to exit freeze mode.

Oscilloscope mode (Ampl.)

• To view a signal waveform, open the MultiScope control panel and make sure that the “Ampl.” button in the lower left corner is lit.

• If the source signal is stereo you can now select either the Left or Right channel for viewing, or Stereo for both channels to be shown in the window. If it is a mono signal, this does not matter.

• If MultiScope is used with a multi-channel track or output bus, you can select any speaker channel for viewing, or All Channels to view them all at once.

• You can now adjust the Amplitude knob to increase/ decrease the vertical size of the waveform, and the Fre

quency knob to select the frequency area for viewing.

Frequency Spectrum Analyzer mode (Freq.)

• Click the Freq button so that it lights up.

MultiScope now divides the frequency spectrum into separate vertical bands, which allows you to get a visual overview of the different frequen cies’ relative amplitude. The frequency bands are shown left to right, starting with the lower frequencies.

• If MultiScope is used with a multi-channel track or output bus, you can select any speaker channel for viewing, or All Channels to view them all at once.

• Adjust the Amplitude knob to increase/decrease the vertical range of the bands.

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• By adjusting the Frequency knob, you can divide the

frequency spectrum into 8, 15, or 31 bands, or you set it to “Spectrum”, which gives you a high-resolution view.

• Use the Mode A and Mode B buttons to switch be-

tween different view modes.

Mode A is more graphically detailed, showing a solid, blue amplitude bar for each band. Mode B is less detailed, showing a continuous blue line that displays the peak levels for each band. These view modes do not have any effect if you have set the Frequency knob to “Spectrum”.

When MultiScope is used with a surround channel in Scope mode, the pop-up menu to the right of the Scope button determines the result:

• If “Stereo (Front)” is selected, the display will indicate the phase and amplitude relationship between the front stereo channels.

• If “Surround” is selected, the display indicates the energy distribution in the surround field.

Phase Correlator mode (Scope)

• Click the Scope button so that it lights up.

The phase correlator indicates the phase and amplitude relationship between channels in a stereo pair or a surround configuration.

For stereo pairs, the indications work in the following way:

• A vertical line indicates a perfect mono signal (the left and right channels are the same).

• A horizontal line indicates that the left channel is the same as the right, but with an inverse phase.

• A random but fairly round shape indicates a well balanced stereo signal. If the shape “leans” to the left, there is more energy in the left channel and vice versa (the extreme case of this is if one side is muted, in which case the phase meter will show a straight line, angled 90° to the other side).

• A perfect circle indicates a sine wave on one channel, and the same sine wave shifted by 90° on the other.

• Generally, the more you can see a “thread”, the more bass in the signal, and the more “spray-like” the display, the more high frequencies in the signal.

SMPTEGenerator (Cubase only)

This plug-in is not a real audio effect. It sends out SMPTE timecode to an audio output, allowing you to synchronize other equipment to Cubase (provided that the equipment can sync directly to SMPTE timecode). This can be very useful if you do not have access to a MIDI-to-timecode converter.

The following parameters are available:

Parameter Description

Main timecode display

Frame rate display and pop-up menu

This display shows the current timecode. When “Link to Transport” is deactivated, the generator is in “free run” mode. You can then use the timecode dis play to set the SMPTE start time. When “Link to Transport” is activated, you cannot change any of the values. This display shows the current time code in sync with the Transport panel. Where applicable, the offset defined in the offset timecode display is taken into account (see below).

The frame rate shown to the right of the timecode display defaults to the frame rate set in the Project Setup dialog. To generate timecode in a different frame rate (e. stripe a tape), select another format on the pop-up menu (only available if “Link to Transport” is deactivated). Note that for another device to synchronize correctly to Cubase, the same frame rate has to be set in the Project Setup dialog, the SMPTE Generator and the receiving device.

g. to

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Parameter Description

Offset timecode display

Generate Code button

Link to Transport button

Timecode in Still Mode button

This display is only available if “Link to Transport” is activated. It allows you to set an offset with regard to the timecode used by Cubase. The offset affects the gener ated SMPTE signal, the current cursor position in Cubase remains unaffected. For example, use this when playing back video using an external device, where the video starts at a different time code position than in Cubase. A scenario could be as follows: Your have placed the same video several times on the Cubase timeline, in order to record different audio versions for that video one after the other. However, since video playback is done via an external machine (replaying the same video) you need an offset to match the different timecode positions in Cubase with the (unchanging) start position on the external machine.

When you activate this button, the plug-in generates SMPTE timecode in “free run” mode, meaning that it out puts continuous timecode independent from the Transport panel. Use this mode if you want to stripe tape with SMPTE.

When you activate this button, the timecode is synchronized to the Transport panel.

When you activate this button, the plug-in also generates SMPTE timecode in stop mode. However, note that this will not be continuous timecode, but timecode generated at the current cursor position. For example, this can be useful when working with video editing software that interprets the absence of timecode as a stop command. By using this option, the video soft ware can enter still mode instead so that a still frame is shown instead of a blank screen.

Ö To change one of the timecode values (main and offset timecode displays), double-click on any of the timecode fields and enter a new value.

Example – Synchronizing a device to Cubase

1. Use the SMPTE Generator as an insert effect on an

audio track, and route that track to a separate output.

Make sure that no other insert or send effect is used on this track. You should also disable any EQ.

2. Connect the corresponding output on the audio hard-

ware to the timecode input on the device you wish to synchronize to Cubase.

Make all necessary settings for the external device so that it synchronizes to incoming timecode.

3. If needed, adjust the level of the timecode, either in

Cubase or in the receiving device.

Activate the Generate Code button (make the device send the SMPTE timecode in “free run” mode) to test the level.

4. Make sure that the frame rate in the receiving device matches the frame rate set in the SMPTE Generator.

5. Activate the “Link to Transport” button.

The SMPTE Generator now outputs timecode that corresponds to the Cubase time display.

6. On the Cubase Transport panel, click Play.

The external device is now synchronized and will follow any position changes set with the Cubase transport controls.

TestGenerator (Cubase only)

This utility plug-in allows you to generate an audio signal, which can be recorded as an audio file. The resulting file can then be used for a number of purposes:

• For testing the specifications of audio equipment.

• For measurements of various kinds, such as calibrating tape recorders.

• For testing signal processing methods.

• For educational purposes.

The TestGenerator is based on a waveform generator which can generate a number of basic waveforms such as sine and saw as well as various types of noise. Further more, you can set the frequency and amplitude of the generated signal.

As soon as you add the TestGenerator as an effect on an audio track and activate it, a signal is generated. You can then activate recording as usual to record an audio file ac cording to the signal specifications:

Parameter Description

Waveforms and noise section

Allows you to set the basis for the signal generated by the waveform generator. You can select between four basic waveforms (sine, triangle, square, and sawtooth) and three types of noise (white, pink, and brownian).

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Parameter Description

Frequency section

Gain section Allows you to set the amplitude of the signal. The higher

Allows you to set the frequency of the generated signal. You can select one of the preset values (100, 440, 1000,

Hz), or use the slider to set a value between

or 10000

Hz and 20000 Hz.

the value (up to 0 select one of the preset values (e. slider to set a value between -81 and 0

dB), the stronger the signal. You can

g. -20 dB), or use the

dB.

Tuner

This is a guitar tuner. Simply connect a guitar or other instrument to an audio input and select the Tuner as an insert effect (make sure you deactivate any other effect that alters pitch, like chorus or vibrato).

When you play a note, the pitch is shown in the middle of the display. In addition, the frequency in Hz is shown in the bottom left corner and the octave range in the bottom right corner.

The two arrows indicate any deviation in pitch. If the pitch is flat, they are positioned in the left half of the display, if the pitch is sharp they are in the right half. The deviation is also shown (in Cent) in the upper area of the display.

• If a string is out of tune (e. g. if the pitch for the E string is shown as Eb), tune the string so that the correct pitch is shown and the two arrows are in the middle.

Repeat this procedure for each string.

• To mute the output signal so that you can tune the strings in silence, activate the Mute button at the bottom middle of the plug-in panel.

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MIDI effects

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Introduction

This chapter describes the included MIDI realtime effects and their parameters.

How to apply and handle MIDI effects is described in the chapter “MIDI realtime parameters and effects” in the Operation Manual.

Arpache 5

A typical arpeggiator accepts a chord (a group of MIDI notes) as input, and plays back each note in the chord separately, with the playback order and speed set by the user. The Arpache 5 arpeggiator does just that, and more. Before describing the parameters, let’s look at how to cre ate a simple, typical arpeggio:

1. Select a MIDI track and activate monitoring (or record enable it) so that you can play “thru” the track.

Make sure that the track is properly set up for playback to a suitable MIDI instrument.

2. Select and activate the arpeggiator.

For now, use it as an insert effect for the selected track.

3. In the arpeggiator panel, use the Step Size setting to set the arpeggio speed.

The speed is set as a note value, relative to the project tempo. For example, setting Step Size to “16” means the arpeggio will be a pattern of sixteenth notes.

4. Use the Length setting to set the length of the arpeggio

notes.

This allows you to create staccato arpeggios (Length value smaller than the Step Size setting) or arpeggio notes that overlap each other (Length value greater than Step Size).

5. Set the Key Range parameter to 12.

This will make the notes arpeggiate within an octave.

6. Play a chord on your MIDI instrument.

Now, instead of hearing the chord, you will hear the notes of the chord played one by one, in an arpeggio.

7. Try the different arpeggio modes by clicking the Play

Order buttons.

The symbols on the buttons indicate the playback order for the notes (Invert, Up Only, etc.). The settings are described below.

Parameters

The Arpache 5 has the following settings:

Setting Description

Play Order buttons

Step Size Determines the speed of the arpeggio, as a note value re-

Length Sets the length of the arpeggio notes, as a note value re-

Key Range Determines the arpeggiated note range, in semitones

Allows you to select the playback order for the arpeggiated notes. The options are Normal, Invert, Up only, Down only, Random, User. If you select User, you can set the playback order manually using the 12 Play Order slots that are now shown at the bottom of the dialog.

lated to the project tempo. The range is 32T (1/32 note triplets) to “1.” (dotted note values).

lated to the project tempo. The range is the same as for the Step Size setting.

counted from the lowest key you play. This works as follows: – Any notes you play that are outside this range will be transposed in octave steps to fit within the range. – If the range is more than one octave, octave-transposed copies of the notes you play will be added to the arpeggio (as many octaves as fit within the range).

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Setting Description

Play Order slots

MIDI Thru If this is activated, the notes sent to the arpeggiator (i. e. the

If the User play order is selected, you can use these “slots” to specify a custom playback order for the arpeggio notes: Each of the 12 slots corresponds to a position in the arpeggio pattern. For each slot, you specify which note should be played on that position by selecting a number. The numbers correspond to the keys you play, counted from the lowest pressed key. So, if you play the notes C3-E3-G3 (a C major chord), “1” would mean C3, “2” would mean E3, and “3” would mean G3. Note that you can use the same number in several slots, creating arpeggio patterns that are not possible using the standard play modes. Please note that you need to begin with the left-most slot and then fill the slots to the right.

chord you play) will pass through the plug-in (sent out to gether with the arpeggiated notes).

Arpache SX

This is an even more versatile and advanced arpeggiator, capable of creating anything from traditional arpeggios to complex, sequencer-like patterns. The Arpache SX has two different modes: Classic and Sequence.

Classic vs. Sequence mode

The Classic mode determines the basic behavior of the Arpache SX. When Sequence mode is selected, the Arpache SX uses the events of an additional MIDI part as a pattern. This pattern then forms the basis for the arpeggio, in conjunction with the MIDI input.

Classic mode

The following parameters are available:

Parameter Description

Direction This allows you to choose how the notes in the chord you

One Shot Mode

Transpose When a setting other than “Off” is selected, the arpeggio

Repeats The “Repeats” setting sets the number of transposed re-

Pitch Shift The “Pitch Shift” setting determines the transposition of

MIDI Thru If this is activated, the notes sent to the arpeggiator (i. e.

Step Size Determines the resolution of the arpeggio, i. e. its “speed”

Length Determines the length of the arpeggio notes (in fixed note

Max. Polyphony

Sort by When you play a chord into the Arpache SX, the arpeg-

Velocity Determines the velocity of the notes in the arpeggio. Us-

play should be arpeggiated. In Classic mode you can choose a value from a pop-up menu, in Sequence mode you will find additional options, see below.

Activate this option if you want the phrase to be played only once. When this option is deactivated, the phrase will be looped.

will be expanded upwards, downwards or both (depend ing on the mode). This is done by adding transposed repeats of the basic arpeggio pattern.

peats.

each repeat.

the chord you play) will pass through the plug-in (sent out together with the arpeggiated notes).

(in fixed note values or PPQ, if the PPQ button is acti vated). In Sequence mode you can also activate the “from sequence” option, see below.

values or PPQ, if the PPQ button is activated). In Se quence mode you can also activate the “from sequence” option, see below.

Determines how many notes should be accepted in the input chord. The “All” setting means there are no limita tions.

giator will sort the notes in the chord in the order specified here. For example, if you play a C-E-G chord, with “Note Lowest” selected, C will be the first note, E will be the second and G the third. This affects the result of the Arp Style setting.

ing the slider you can set a fixed velocity, or you can activate the “via Input” button to use the velocity values of the corresponding notes in the chord you play. In Sequence mode you can also activate the “from sequence” option, see below.

Sequence mode

In Sequence mode you can import a MIDI part into the Arpache SX by dragging it from the Project window and dropping it in the “Drop MIDI Sequence” field on the right of the Arpache SX panel.

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Now, the notes in the dropped MIDI part will be sorted internally, either according to their pitch (“MIDI Seq. sort by pitch” checkbox activated) or according to their play order in the part. This results in a list of numbers. For example, if the notes in the MIDI part are C E G A E C and they are sorted according to pitch, the list of numbers will read 1 2 3 4 2 1. Here, there are 4 different notes/numbers and 6 trigger positions.

The MIDI input (the chord you send into the Arpache SX) will generate a list of numbers, with each note in the chord corresponding to a number depending on the “Sort by” setting.

Furthermore, the two lists of numbers will be matched – the Arpache SX tries to play back the pattern from the dropped MIDI part but using the notes from the MIDI input (chord). The result depends on the Play Mode setting:

Option Description

Trigger The whole pattern from the dropped MIDI file will be played

Trigger Cnt. As above, but even when all keys are released, the phrase

Sort Normal Matches the notes in the MIDI input with the notes in the

Sort First As above, but if there are fewer notes in the MIDI input,

Sort Any As above, but if there are fewer notes in the MIDI input,

Arp. Style As above, but if there are fewer notes in the MIDI input,

Repeat In this mode, the chords played will not be separated into

back, but transposed according to one of the notes in the MIDI input. Which note is used for transposing depends on the Sort by setting.

continues playing from the last position (where it stopped), when a new key is pressed on the keyboard. This is typi cally used when playing “live” through the Arpache SX.

dropped MIDI part. If there are fewer notes (numbers) in the MIDI input, some steps in the resulting arpeggio will be empty.

the missing notes will be replaced by the first note.

the missing notes will be replaced by any (random) note.

the missing notes will be replaced by the last valid note in the arpeggio.

notes. Instead they will be used as is, and only the rhythm of the dropped MIDI part is used for playback.

Note also that you can choose to keep the original note timing, note length and note velocities from the dropped MIDI part, by selecting “from sequence” for the Step Size, Length and Velocity options.

Auto LFO

This plug-in works like an LFO in a synthesizer, allowing you to send out continuously changing MIDI controller mes sages. One typical use for this is automatic MIDI panning, but you can select any MIDI continuous controller event type. The Auto LFO effect has the following parameters:

Waveform

These settings determine the shape of the controller curves sent out. You can click on a waveform symbol, or choose a value from the pop-up menu.

Wavelength

This is where you set the speed of the Auto LFO, or rather the length of a single controller curve cycle. Using the slider or by choosing an entry from the pop-up menu, you can set this to rhythmically exact note values (or PPQ val ues if the PPQ button is activated). The lower the note value, the slower the speed. For example, if you set this to “1/8”, the waveform will be repeated every eighth note.

Controller Type

Determines which continuous controller type is sent out. Typical choices would include pan, volume and brightness, but your MIDI instrument may have controllers mapped to various settings, allowing you to modulate the synth para meter of your choice – check the MIDI implementation chart for your instrument for details!

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Step display

Flam position settings, see “Adding flams” on

page 59.

Pattern display (12 patterns on 4 subbanks)

Swing settings, see

“The Swing setting”

on page 58.

Swing and Offset controls

Lane Name fields

Jump mode

Step resolution

Number of steps for this pattern

Density

This determines the density of the controller curves sent out. The value can be set to “small”, “medium”, or “large”, or to rhythmically exact note values (by choosing from the pop-up menu). The higher the note value, the smoother the controller curve. For example, if you set this to “1/16”, a new controller event will be sent out at every 1/16 note position.

Value Range

These two sliders are used to determine the range of controller values sent out, i. e. the “bottom” and “top” of the controller curves.

Beat Designer

The Beat Designer is a MIDI pattern sequencer that allows you to create your own drum parts or “patterns” for a project. With the Beat Designer, you can quickly and easily set up the drums for a project, by experimenting and creating new drum sequences from scratch.

Normally, you will work on a short sequence, adjusting and modifying it while playing it back in a loop until you get the desired result. The drum patterns can then either be con verted to MIDI parts on a track or triggered using MIDI notes during playback, see “Converting patterns into MIDI

parts” on page 60 and “Triggering patterns” on page 60.

To use the Beat Designer, select it as MIDI insert effect for a MIDI track (routed to a VSTi or an external device) or an instrument track.

Overview

When you open the control panel for the Beat Designer for the first time, it shows a display with 8 empty lanes, each containing 16 steps.

Patterns and subbanks

The Beat Designer patterns are saved as pattern banks. One pattern bank contains 4 subbanks which in turn con tain 12 patterns each.

In the pattern display in the lower part of the Beat Designer, subbanks and patterns are displayed graphically. To select

a subbank, click on a number (1 to 4) at the top of the dis-

play. To select a pattern within this subbank, click on a “key” in the keyboard display below.

Initial settings

The steps represent the beat positions in the pattern. You can specify the number of steps and the step resolution globally for a pattern:

• Click in the “Number of steps for this pattern” value field and enter the desired value.

The maximum number of steps is 64. By default, 16 steps are shown.

• The playback length, i. e. the note value for the steps, can be specified in the Step resolution pop-up menu next to the Number of Steps setting.

On this menu, you can also set triplet values. These also affect the Swing setting, see

“The Swing setting” on page 58. The default setting is 1/16.

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Selecting drum sounds

To specify a drum sound, click in the drum name field for a lane and select the desired drum sound from the pop-up menu. The available drum sounds depend on the selected drum map. If no drum map is selected for the track, the GM (General MIDI) drum names are used.

• To find the right sound, you can audition the selected drum sound by clicking the Preview Instrument button (the speaker icon).

Entering drum steps

To enter a drum step, click on the step field where you want to add a beat. You could e. each downbeat for a lane and a bass drum on a second lane. When you click in an empty field, it becomes “filled”, indicating that you will hear a drum beat on this step.

You can also click and drag to enter a continuous range of drum steps.

Ö When working on drum patterns, it is a good idea to play back a section of the project in a loop while inserting the drum sounds, as this allows you to hear the result immediately.

g. add a snare drum on

Removing steps

• To remove a drum step, simply click on the corresponding field again.

• To remove a range of drum steps, click and drag over them.

Setting the velocity

When entering a drum step, the velocity setting of this step is determined by where you click: Click in the upper part of a step for the highest velocity setting, in the middle section for a medium velocity and in the lower part for the lowest velocity setting. This is a quick way of roughly set ting the velocity on the fly while entering drum sounds. In the display, the different velocity settings are indicated by different colors.

• You can fine-tune the velocity setting for an existing drum step by clicking on it and dragging up or down.

The current velocity is indicated numerically while you drag, allowing you to find the desired setting easily. The available range is from 1 to 127.

• You can also fine-tune the velocity for a range of drum steps. Click on the first step, drag up or down to enter into velocity edit mode, and then drag sideways and up or down to modify the velocity for all the steps.

• If you hold down [Shift] while dragging up or down, you can change the velocity for all steps on a lane.

Ö If you change the velocity for several steps at the same time, the relative velocity differences will be kept for as long as possible (until the minimum or maximum setting is reached).

The velocity for the steps will be increased or decreased by the same amount.

• You can also create a crescendo (or decrescendo) for an existing range of drum steps by holding down [Alt]/ [Option], clicking on the first step, dragging up or down and then dragging to the left or right.

Editing operations

• You can move all drum steps on a lane by holding down [Shift], clicking on the lane and dragging to the left or right.

• You can also “invert” a lane, i. e. add drum sounds for all steps that were empty while removing all existing drum steps. This lets you create unusual rhythmic patterns. To do so, hold down [Alt]/[Option] and drag the mouse over the lane.

• You can copy the content of a lane onto another lane by holding down [Alt]/[Option], clicking in the section to the left of the lane you want to copy and dragging to the de sired position.

When you drag, a vertical line and a plus symbol will be displayed.

Lane handling

If you find that you have too many or too few lanes in the Beat Designer, you can add or remove them.

• To add a lane, click on the “Add Instrument Lane” button at the bottom right of the last lane shown.

• To remove a lane, click on the “Remove Instrument Lane” button in the controls section at the far right of the lane.

• You can change the order of the drum lanes by clicking in an empty area in the section to the left of a lane (i. e. not on a button) and dragging it to another position.

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• You can mute or solo a lane by clicking the respective

buttons to the left of the step display.

The lane operations always affect all patterns in the Beat Designer instance, not only the one you edit.

The Pattern Functions menu

This menu contains the following editing functions:

Option Description

Shift Left This moves all steps of the current pattern (all steps on all

Shift Right This moves all steps of the current pattern (all steps on all

Reverse Reverses the pattern, so that it plays backwards. Copy

Pattern

Paste Pattern

Clear Pattern

Insert Pattern at Cursor

Insert Subbank at Cursor

Insert Pattern at Left Locator

Insert Subbank at Left Locator

Fill Loop with Pattern

lanes) to the left.

lanes) to the right.

This copies the pattern to the clipboard. Copied patterns can be pasted into another pattern subbank (see below), and even directly into the project. The default key command for this is [Ctrl]/[Command]-[C].

Allows you to paste a complete pattern, e. g. into another pattern subbank, even into another instance of the Beat De signer. This is handy when you want to create variations based on existing patterns. The default key command for this is [Ctrl]/[Command]-[V].

This resets the current pattern.

This creates a MIDI part for the current pattern and inserts it in the Project window, at the position of the project cursor

“Converting patterns into MIDI parts” on page 60).

(see also This creates a number of MIDI parts (one for each used pat-

tern in the subbank) and inserts them one after the other, starting at the project cursor (see also

into MIDI parts” on page 60).

This creates a MIDI part for the current pattern and inserts it in the Project window, at the left locator (see also

ing patterns into MIDI parts” on page 60).

This creates a number of MIDI parts (one for each used pattern in the subbank) and inserts them one after the other, starting at the left locator (see also

into MIDI parts” on page 60).

This creates a MIDI part for the current pattern and inserts it in the Project window as often as needed to fill the current loop area (the space between the left and right locators),

“Converting patterns into MIDI parts” on page 60.

Chorder

The Chorder is a MIDI chord processor, allowing you to assign complete chords to single keys in a multitude of variations. These can then be played back live or using re corded notes on a MIDI track.

There are three main operating modes: “All Keys”, “One Octave”, and “Global Key”. You can switch between these modes using the Chords pop-up, see below.

For every key you can record up to eight different chords or variations on so-called “layers”. This is described in detail in the section “Using layers” on page 62.

Operating modes

In the lower left section of the Chorder window, you can choose an option from the Chords pop-up menu to decide which keys in the keyboard display will be used to record your chords.

Global Key

In this mode, you can assign chords to each key on the keyboard display. When you play any of these keys, you will hear the assigned chords instead.

One Octave

The One Octave mode is similar to the All Keys mode, but you can only set up chords for each key of a single octave (that is, up to eight different chords on twelve keys). When you play a note (e. a transposed version of the chords set up for this key.

Global Key

In Global Key mode, you can set up chords for a single

key only. These chords (that you recorded on C3) are then played by all keys on the keyboard, but transposed according to the note you play.

g. C) on a different octave, you will hear

The chord indicator lane

At the top of the keyboard display you will find a thin lane with a small rectangle for each key that you can use to record a chord. These rectangles are shown in blue for all keys that already have chords assigned to them.

Ö In Global Key mode the C3 key has a special marking instead since this is the only key used in this mode.

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Entering chords

To enter chords you need to switch to Learn mode. In this mode a transparent red bar indicates which element is ready for “learning” a note or chord. When you choose the trigger note for a chord, for example, the keyboard display is shown in red.

The keyboard display in Learn mode

The second layer in Learn mode

Proceed as follows:

1. Click the Learn button at the top of the Chorder window to activate Learn mode.

The chord indicator lane is now tinted red, indicating that it is active.

2. Select the key to which you want to assign a chord by clicking on it on the keyboard display, or by pressing the key on a connected MIDI keyboard.

The red bar will now move to the first layer, indicating that you are ready to record the first chord.

Ö In Global Key mode you do not have to choose a trigger key. The first layer is activated directly.

3. Play a chord on the MIDI keyboard and/or use the mouse to enter or change the chord in the layer display.

Any notes you enter are immediately shown in the Chorder display. The notes are shown in different colors, depending on the pitch.

• If you are entering chords via a MIDI keyboard, the Chorder will learn the chord as soon as you release all keys of your MIDI keyboard simultaneously.

As long as a key is pressed, you can continue looking for the right chord.

• If more than one layer is shown, the Chorder will jump automatically to the next layer where you can record another chord.

When all the layers for a key are filled, the red bar will jump back to the keyboard display so that you can choose a different trigger key (in Global Key mode the Learn mode is deactivated instead).

• If you are entering chords with the mouse, the Chorder will not jump to the next layer automatically.

You can select/deselect as many notes as you wish and then click on another layer or deactivate the Learn mode to continue.

4. Repeat the above with any other keys you wish to use.

Using layers

The Layers pop-up menu at the bottom right of the window allows you to set up chord variations in the layer display above the keyboard. This works with all three modes and provides up to eight variations for each assignable key (that is, a maximum of 8 different chords in Global Key mode,

x 8 chords in One Octave mode and 128 x 8 chords in

12 All Keys mode).

The different layers can be triggered by velocity or interval. Proceed as follows to set up your layers:

1. Open the Layers pop-up menu and select Velocity or Interval. Set this to Single Mode if you want to set up only one chord per key.

2. Use the slider below the Layers pop-up menu to specify how many variations (layers) you want to use.

3. Enter the chords as described above.

4. Now you can play the keyboard and trigger the varia-

tions according to the selected layer mode.

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The layer modes work as follows:

Trigger mode Description

Velocity The full velocity range (1–127) is divided into “zones”,

Interval In this mode, the Chorder will play one chord at a time –

Single Mode Select this if you do not wish to use different layers.

according to the number of layers you specified. For ex ample, if you are using two variations (Number of Layers is set to 2) there will be two velocity “zones”: 1–63 and 64–127. Playing a note with velocity 64 or higher will trigger the second layer, while playing a softer note will trigger the first layer. Using the “Velocity spread” slider at the bottom right of the window, you can change the velocity ranges of the layers so that a different layer will be activated using the same velocity value.

you cannot play several different chords simultaneously. When the Interval mode is selected, you press two keys on your keyboard to trigger the desired layer, with the lower key determining the base note for the chord. The layer number will be the difference, i. tween the two keys. To select layer 1, press a key one semitone higher than the base note, for layer 2, press a key two semitones higher, and so on.

e. the interval, be-

Playstyle

From the Playstyle pop-up menu at the bottom of the pane you can choose one of seven different styles that determine

in which order the individual notes of the chords are played back.

The following options are available:

Playstyle Description

simultaneous In this mode all notes are played back simultaneously. fast up In this mode a small arpeggio is added, starting with

slow up Similar to “fast up”, but using a slower arpeggio. fast down Similar to “fast up”, but starting with the highest note. slow down Similar to “slow up”, but starting with the highest note. fast random In this mode the notes are played back in a rapidly

slow random Similar to “fast random”, but the note changes occur

the lowest note.

changing random order.

more slowly.

Empty layers

If you enter less chords than layers present for a key, these layers will be filled automatically when you end the Learn mode.

This works according to the following rules:

• Empty layers are filled from bottom to top.

• If there are empty layers below the first layer with a

chord, these are filled from top to bottom. An example: If you have a setup with 8 layers, and you enter the chord

C in layer 3 and G7 in layer 7, you get the following result: chord C in layers 1 to 6 and G7 in layers 7 and 8.

Resetting layers

In Learn mode, you can use the “Reset layers” button at the top left of the Chorder window to delete all notes in the different layers for the selected trigger key.

Compressor

This MIDI compressor is used for evening out or expanding differences in velocity. Though the result is similar to what you get with the Velocity Compression track para meter, the Compress plug-in presents the controls in a manner more like regular audio compressors.

The following parameters are available:

Parameter Description

Threshold Only notes with velocities above this value will be affected

Ratio This determines the rate of compression applied to the

by the compression/expansion.

velocity values above the threshold level. Ratios greater than 1:1 result in compression (i. locity) while ratios lower than 1:1 result in expansion (i. e. greater difference in velocity). What actually happens is that the part of the velocity value that is above the threshold value is divided by the ratio value.

e. less difference in ve-

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Parameter Description

Gain This adds or subtracts a fixed value from the velocities.

Since the maximum range for velocity values is 0–127, you may need to use the Gain setting to compensate, keeping the resulting velocities within the range. Typi cally, you would use negative Gain settings when expanding and positive Gain settings when compressing.

Context Gate

The Context Gate allows for selective triggering/filtering of MIDI data. It features two modes: in Poly Mode the Context Gate recognizes certain chords that are played and in Mono Mode only certain MIDI notes are let through. These modes can be used for context selective control of MIDI devices and are, for example, very useful in certain live scenarios.

The following parameters are available:

Poly Mode – Polyphony Gate

This allows you to filter MIDI according to the number of pressed keys within a given key range. This can be used independently or in conjunction with the Chord Gate function.

• The Key Range Limit sliders are used to set the key range.

Only notes within this range will be let through.

• The “Minimum Polyphony” value field allows you to specify the minimum number of notes required to open the gate.

Poly Mode – Chord Gate

When Chord Gate is activated, only notes in recognized chords are let through.

• Two Recognition modes are available: Simple and Normal. In Simple mode, all standard chords (major/minor/b5/ dim/sus/maj7 etc.) are recognized, whereas Normal mode takes more tensions into account.

Mono Mode – Channel Gate

When this is activated, only single note events in a specified MIDI channel are let through, which can be used with MIDI controllers that can send MIDI over several channels simultaneously, for example guitar controllers which send data for each string over a separate channel.

• You can set Mono Channel to a specific channel (1 to

16), or to “Any”, i. e. no channel gating.

Mono Mode – Velocity Gate

This can be used independently or in conjunction with the Channel Gate function. Played notes will sound (no noteoff message) until a note is played inside the set range (and additionally the set Channel Gate channel, if checked).

• The Key Range Limit sliders are used to set the key range.

Only notes within this range will be let through.

• Notes below the Minimum Velocity threshold value will be gated.

Auto Gate Time

If there is no input activity, all resounding notes are sent a note-off message after the set time, in seconds or milliseconds.

Panic Reset button

Sends an “All Notes Off” message over all channels, in case of hanging notes.

Learn Reset button

When this is activated, you can specify a Reset trigger event via MIDI. Whenever this specific MIDI event is sent, it triggers an “All Notes Off” message. When you have set the Reset event, the Learn button should be deactivated.

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Application examples

Poly Mode

In Poly mode, you could use the Context Gate to accompany yourself during a live guitar performance using a VST instrument. To do this, you might use a guitar to MIDI converter: You could then program the Context Gate, for example, to allow only those notes to pass the gate that are part of a four-note chord. During your performance you would then play a four-note chord every time that you want to trigger the VST instrument. The instrument will play until the Auto Gate Time is reached and fade out. For more complex performances this can be combined with an arpeggiator, without having to use external pedals to trigger the effect.

Micro Tuner

Mono Mode

In Mono Mode you could use the Context Gate to trigger variations played with a drum machine/VST instrument. To do this, you will need a guitar to MIDI converter: You could then filter the MIDI channel using the Input Transformer (optional) and program the Context Gate to allow only certain notes on your guitar to pass the gate (e. ning at the 12th band). When you now play one of these notes, the note-off command will not be send out and the corresponding note will sound until the note is played again, a new note is let through, or the Auto Gate Time is reached. This way you can trigger lots of different effects or notes using the high notes on you guitar without having to use an additional MIDI instrument.

g. begin-

Density

This generic control panel affects the “density” of the notes being played from (or thru) the track. When this is set to 100 Density setting below 100 “mute” notes. Raising the setting above 100 randomly add notes that have been played before.

%, the notes are not affected. Lowering the

% will randomly filter out or

% will instead

The Micro Tuner lets you set up a different microtuning scheme for the instrument, by detuning each key.

• Each Detune slider corresponds to a key in an octave (as indi-

cated by the keyboard display). Adjust a Detune field to raise or lower the tuning of that key, in cents (hundreds of a semitone).

• By keeping the [Alt]/[Option] key pressed, you can adjust all

keys by the same amount.

The Micro Tuner comes with a number of presets, including both classical and experimental microtuning scales.

MIDI Control

This generic control panel allows you to select up to eight different MIDI controller types, and use the value fields or sliders (which are displayed when you click on a value field while holding down the [Alt]/[Option] key) to set values for these. A typical use for this would be if you are using a MIDI instrument with parameters that can be controlled by MIDI controller data (e.

g. filter cutoff, resonance, levels,

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etc.). By selecting the correct MIDI controller types, you can use the plug-in as a control panel for adjusting the sound of the instrument from within Cubase, at any time.

• To select a controller type, use the pop-up menus on the right.

• To deactivate a controller slider, set it to “Off” (drag the slider all the way down).

For example, setting this to -2 will cause the first echo note to have a pitch two semitones lower than the original note, the second echo note two semitones lower than the first echo note, and so on.

Repeats

This is the number of echoes (1 to 12) from each incoming note.

MIDI Echo

This is an advanced MIDI Echo, which will generate additional echoing notes based on the MIDI notes it receives. It creates effects similar to a digital delay, but also features MIDI pitch shifting and much more. As always it is impor tant to remember that the effect does not “echo” the actual audio, but the MIDI notes which will eventually produce the sound in the synthesizer.

The following parameters are available:

Velocity Offset

This parameter allows you to raise or lower the velocity values for each repeat so that the echo fades away or in creases in volume (provided that the sound you use is velocity sensitive). For no change of velocity, set this to 0 (middle position).

Pitch Offset

If you set this to a value other than 0, the repeating (echoing) notes will be raised or lowered in pitch, so that each successive note has a higher or lower pitch than the pre vious. The value is set in semitones.

Beat Align

During playback, the Beat Align parameter quantizes the position of the first echo note. You can either set this to “rhythmically exact” values (displayed as note values – see the table below) or activate the PPQ button and choose a PPQ value.

Setting this to “1/8”, for example, will cause the first echo note to sound on the first eighth position after the original note.

Ö The echo time can also be affected by the Delay Decay parameter.

Ö During live mode, this parameter has no effect since the first echo will always be played together with the note event itself.

Delay

The echoed notes will be repeated as set up with this parameter. You can either set this to “rhythmically exact” val-

ues (displayed as note values – see the table below) or activate the PPQ button and choose a PPQ value. This makes it easy to find rhythmically relevant delay values, but still allows experimental settings in between.

Delay Decay

This parameter lets you adjust how the echo time should be changed with each successive repeat. The value is set as a percentage.

• When set to 100 % (middle position) the echo time will be the

same for all repeats (as set with the Delay parameter).

• If you raise the value above 100 %, the echoing notes will play

with gradually longer intervals (i. e. the echo will become slower).

• If you lower the value below 100 %, the echoing notes will be-

come gradually faster, like the sound of a bouncing ball.

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Length

This sets the length of the echoed notes. This can either be identical with the length of the original notes (parameter set to its lowest value) or the length you specify manually. You can either set this to “rhythmically exact” values (displayed as note values – see the table below) or activate the PPQ button and choose a PPQ value.

Ö The length can also be affected by the Length Decay parameter.

Length Decay

This parameter lets you adjust how the length of the echoed notes should change with each successive re

peat. The higher the setting (25–100), the longer the echoed notes will be, compared to their original notes.

About ticks and note values

The timing and position-related parameters (Delay, Length and Beat Align) can all be set in ticks (or PPQ which denotes the same thing here). There are 480 ticks to each quarter note. While the parameters allow you to step between the rhythmically relevant values (displayed as note values), the following table can also be of help, showing you the most common note values and their correspond

ing number of ticks:

Note Value Ticks

1/32 note 60 1/16 note triplet 90 1/16 note 120 1/8 note triplet 160 1/8 note 240 Quarter note triplet 320 Quarter note 480 Half note 960

MIDI Modifiers

This plug-in is essentially a duplicate of the MIDI Modifiers section in the Inspector. This can be useful, for example, if you need extra Random or Range settings.

The MIDI Modifiers effect also includes the “Scale Transpose” function that is not available among the track parameters.

Scale Transpose

This allows you to transpose each incoming MIDI note, so that it fits within a selected musical scale. The scale is specified by selecting a key (C, C#, D, etc.) and a scale type (major, melodic or harmonic minor, blues, etc.).

Ö To turn Scale Transpose off, select “No Scale” from the Scale pop-up menu.

MIDI Monitor

The MIDI Monitor is used to monitor incoming MIDI events. You can choose whether to analyze live or playback events and which types of MIDI data are to be monitored. Use this, for example, to analyze which MIDI events are being gener ated by a MIDI track, or to find “suspicious” events, such as notes with velocity 0 that certain MIDI devices might fail to interpreted as note-off events.

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Inputs section

In this section you can choose whether to monitor Live Events or Playback Events.

Show section

Here, you can activate/deactivate the different types of MIDI events, e. choose the Controller option you can also define which type of controller to monitor.

Data table

In the table in the lower section of the window, you will see detailed information about the monitored MIDI events.

Buffer pop-up menu

In the Buffer pop-up menu you can set the buffer size to 100, 1000 or 10000 events. This is the maximum number of events that is kept in the list of monitored events. Once this list is full, the oldest entries will be deleted when new events are received.

Ö The larger the buffer, the more processing resources are required. To avoid a negative impact on your system’s performance, make sure to use the smallest possible buf fer size.

g. notes or program change events. If you

Note to CC

This effect will generate a MIDI continuous controller event for each incoming MIDI note. The value of the controller event corresponds to the velocity of the MIDI note, which is then used to control the selected MIDI controller (by default CC 7, Main Volume). For each note end, another controller event with the value 0 is sent. The incoming MIDI notes pass through the effect unaffected.

The purpose of this plug-in is to generate a gate effect. This means that the notes played are used to control something else. For example, if Main Volume (CC 7) is se lected, notes with low velocity will lower the volume in the MIDI instrument, while notes with a high velocity will raise the volume.

Note that a controller event is sent out each time a new note is played. If high and low notes are played simultaneously, this may lead to confusing results.

Therefore, the Note to CC effect is best applied to monophonic tracks (playing one note at a time).

Export function

Click the Export button to export the monitoring data as a simple text file.

Record events button

Use this button to the left of the Inputs section to start or stop the monitoring of MIDI events.

Clear list button

The Clear List button to the left of the Show section allows you to clear the table of recorded MIDI events.

MIDI effects

Quantizer

Quantizing is a function that changes the timing of notes by moving them towards a “quantize grid”. This grid may consist of e. notes would all get perfect sixteenth note timing), but could also be more loosely related to straight note value positions (applying a “swing feel” to the timing, etc.).

Ö The main Quantize function in Cubase is described in the Operation Manual.

g. straight sixteenth notes (in which case the

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Number of steps

SwingStep

size

Pattern selector

Controller pop-up menu

Shift octave up/down

Shift steps left/right

While the Quantize function on the Edit menu applies the timing change to the actual notes on a track, the Quantizer effect allows you to apply quantizing “on the fly”, changing the timing of the notes in real time. This makes it easier to try out different settings when creating grooves and rhythms. Note however, that the main Quantize function contains settings and features that are not available in the Quantizer.

The Quantizer has the following parameters:

Parameter Description

Quantize Note This sets the note value on which the quantize grid is

Swing This allows you to offset every second position in the grid,

Strength This determines how close the notes should be moved to

Delay This delays (positive values) or advances (negative val-

Realtime quantize

based. Straight notes, triplets and dotted notes are avail able. For example, “16” means straight sixteenth notes and “8T” means eighth note triplets.

creating a swing or shuffle feel. The value is a percentage – the higher you set this, the farther to the right every even grid position is moved.

the quantize grid. When set to 100 forced to the closest grid position; lowering the setting will gradually loosen the timing.

ues) the notes in milliseconds. Unlike the Delay setting in the Track Parameters, this delay can be automated.

During live mode this option can be used to change the timing of the notes played so that they fit the quantize grid.

%, all notes will be

StepDesigner

The StepDesigner is a MIDI pattern sequencer that sends out MIDI notes and additional controller data according to the pattern you set up. It does not make use of the incom ing MIDI, other than automation data (such as recorded pattern changes).

Creating a basic pattern

1. Use the Pattern selector to choose which pattern to create.

Each StepDesigner can hold up to 200 different patterns.

2. Use the “Step size” setting to specify the “resolution” of the pattern.

In other words, this setting determines how long each step is. For example, if this is set to “1/16” each step will be a sixteenth note.

3. Specify the number of steps in the pattern with the “Number of steps” setting.

As you can see in the note display, the maximum number of steps is 32. For example, setting “Step size” to 16 and “Number of steps” to 32 would create a two bar pattern with sixteenth note steps.

4. Click in the note display to insert notes.

You can insert notes on any of the 32 steps, but the StepDesigner will only play back the number of steps set with the Step size parameter.

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• The display spans one octave (as indicated by the pitch list to the left). You can scroll the displayed octave up or down by clicking in the pitch list and dragging up or down.

This way you can insert notes at any pitch. Note that each step can contain one note only – the StepDesigner is monophonic.

Click and drag to view other octaves.

• To remove a note from the pattern, click on it again.

5. On the Controller pop-up menu, select Velocity.

This pop-up menu determines what is shown in the lower controller display.

6. Adjust the velocity of the notes by dragging the velocity bars in the controller display.

7. To make notes shorter, select “Gate” on the Controller pop-up menu and lower the bars in the controller display.

When a bar is set to its maximum value (fully up), the corresponding note will be the full length of the step (as set with the Step size parameter).

8. To make notes longer, you can tie two notes together. This is done by inserting two notes and clicking in the Tie column for the second note.

When two notes are tied, the second note will not be triggered – the previous note is lengthened instead. Also, the tied (second) note will au tomatically get the same pitch as the first note. You can add more notes and tie them in the same way, creating longer notes.

9. If you now start playback in Cubase, the pattern will play as well, sending out MIDI notes on the track’s MIDI output and channel (or, if you have activated the StepDe signer as a send effect, on the MIDI output and channel selected for the send in the Inspector).

Adding controller curves

The Controller pop-up menu has two more items: two controller types.

• You can select which two controller types (filter cutoff, resonance, volume, etc.) should be available on the popup menu by clicking the Setup button and selecting controllers from the lists that appears.

This selection is global, i. e. it applies to all patterns.

• To insert controller information in a pattern, select the desired controller from the pop-up menu and click in the controller display to draw events.

The MIDI controller events will be sent out during playback along with the notes.

Ö If you drag a controller event bar all the way down, no controller value is sent out on that step.

Other pattern functions

The following functions make it easier to edit, manipulate and manage patterns:

Function Description

Shift Octave up/down

Shift Steps left/right

Reverse Reverses the pattern, so that it plays backwards. Copy/Paste Allows you to copy the current pattern and paste it in an-

Reset Clears the pattern, removing all notes and setting control-

Randomize Generates a completely random pattern – useful for ex-

Swing The Swing parameter allows you to offset every second

Presets Handling of presets is described in the chapter “MIDI real-

These buttons allow you to shift the entire pattern up or down in octave steps.

Moves the pattern one step to the left or right.

other pattern location (in the same StepDesigner instance or another).

ler values to default.

perimenting.

step, creating a swing or shuffle feel. The value is a per centage – the higher you set this, the farther to the right every even step is moved.

time settings” in the Operation Manual. Note that a stored Preset contains all 200 patterns in the StepDesigner.

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Automating pattern changes

You can create up to 200 different patterns in each StepDesigner – just select a new pattern and add notes and controllers as described above.

Typically, you want the pattern selection to change during the project. You can accomplish this by automating the Pattern selector, either in real time by activating the Write automation and switching patterns during playback or by drawing in the automation track for the StepDesigner’s MIDI track. Note that you can also press a key on your MIDI keyboard to change patterns. For this, you have to set up the StepDesigner as an insert effect for a record enabled MIDI track. Press C1 to select pattern 1, C#1 to select pattern 2, D1 to select pattern 3, D#1 to select pat tern 4 and so on. If you want, you can record these pattern changes as note events on a MIDI track.

Proceed as follows:

1. Select the desired MIDI track or create a new one and activate the StepDesigner as an insert effect.

2. Set up several patterns as described above.

3. Press the Record button and press the desired keys

on your keyboard to select the corresponding patterns.

The pattern changes will be recorded on the MIDI track.

4. Stop recording and play back the MIDI track.

You will now hear the recorded pattern changes.

Ö This will only work for the first 92 patterns.

Track Control

The Track Control effect contains three ready-made control panels for adjusting parameters on a GS or XG compatible MIDI device. The Roland GS and Yamaha XG protocols are extensions of the General MIDI standard, allowing for more sounds and better control of various instrument settings. If your instrument is compatible with GS or XG, the Track Controls effect allows you to adjust sounds and effects in your instrument from within Cubase.

Selecting a control panel

At the top of the Track Controls effect window you will find a pop-up menu. This is where you select which of the available control panels to use:

Control panel Description

GS 1 Effect sends and various sound control parameters for

XG 1 Effect Sends and various sound control parameters for

XG 2 Global settings (affecting all channels) for instruments

use with instruments compatible with the Roland GS standard.

use with instruments compatible with the Yamaha XG standard.

compatible with the Yamaha XG standard.

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About the Reset and Off buttons

Regardless of the selected mode, you will find two buttons labeled “Off” and “Reset” at the top of the control panel:

• Clicking the Off button will set all controls to their lowest value, without sending out any MIDI messages.

• Clicking the Reset button will set all parameters to their default values, and send out the corresponding MIDI messages.

For most parameters, the default values will be zero or “no adjustment”, but there are exceptions to this. For example, the default “Send 1” setting is 64.

GS 1

The following controls are available when the GS 1 Controls mode is selected:

Control Description

Send 1 Send level for the reverb effect. Send 2 Send level for the chorus effect. Send 3 Send level for the “variation” effect. Attack Adjusts the attack time of the sound. Lowering the value

Decay Adjusts the decay time of the sound. Lowering the value

Release Adjusts the release time of the sound. Lowering the value

Cutoff Adjusts the filter cutoff frequency. Resonance Adjusts the filter resonance. Express Allows you to send out expression pedal messages on

Ch. Press. Allows you to send out aftertouch (channel pressure)

Breath Allows you to send breath control messages on the

Modul. Allows you to send modulation messages on the track’s

shortens the attack, while raising it gives a slower attack. Middle position (64) means no adjustment is made.

shortens the decay, while raising it makes the decay lon ger.

shortens the release, while raising it makes the release time longer.

the track’s MIDI channel.

messages on the track’s MIDI channel. This is useful if your keyboard cannot send aftertouch, but you have sound modules that respond to aftertouch. The default value for this parameter is zero.

track’s MIDI channel.

MIDI channel (just as you normally do with a modulation wheel on a MIDI keyboard).

XG 1

The following controls are available when the XG 1 mode is selected:

Control Description

Send 1 Send level for the reverb effect. Send 2 Send level for the chorus effect. Send 3 Send level for the “variation” effect. Attack Adjusts the attack time of the sound. Lowering this value

Release Adjusts the release time of the sound. Lowering this value

Harm.Cont Adjusts the harmonic content of the sound. Bright Adjusts the brightness of the sound. CutOff Adjusts the filter cutoff frequency. Resonance Adjusts the filter resonance.

shortens the attack, while raising it gives a slower attack. Middle position means no adjustment is made.

shortens the release, while raising it makes the release time longer. Middle position means no adjustment is made.

XG 2

In this mode, the parameters affect global settings in the instrument(s). Changing one of these settings for a track will in fact affect all MIDI instruments connected to the same MIDI output, regardless of the MIDI channel setting of the track. Therefore, to avoid confusion it might be a

good idea to create an empty track and use this only for these global settings.

The following controls are available:

Control Description

Eff. 1 This allows you to select which type of reverb effect

Eff. 2 This allows you to select which type of chorus effect

Eff. 3 This allows you to select one of a large number of “varia-

Reset Sends an XG reset message. MastVol This is used to control the Master Volume of an instrument.

should be used: No effect (the reverb turned off), Hall 1–2, Room 1–3, Stage 1–2 or Plate.

should be used: No effect (the chorus turned off), Chorus 1–3, Celeste 1–3 or Flanger 1–2.

tion” effect types. Selecting “No Effect” is the same as turning off the variation effect.

Normally you should leave this in its highest position and set the volumes individually for each channel (with the vol ume faders in the Cubase mixer or in the Inspector).

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Transformer

The Transformer is a realtime version of the Logical Editor. With this you can perform very powerful MIDI processing on the fly, without affecting the actual MIDI events on the track.

The Logical Editor is described in the corresponding chapter in the Operation Manual. As the parameters and functions are almost identical, the descriptions for the Logical Editor also apply to the Transformer. Where there are differences between the two, this is clearly stated.

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The included VST instruments

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Introduction

This chapter contains descriptions of the included VST instruments and their parameters.

Ö Most of the included instruments are compatible with VST3, this is indicated by an icon in front of the name (for further information, see the section “About VST 3” in the chapter “Audio effects” in the Operation Manual).

Embracer – Surround Pad Synthesizer (Cubase only)

Embracer is a simple but powerful polyphonic synthesizer designed entirely for producing pads and accompaniment sounds. With its easy-to-use envelope and tone controls, it gives you fast access to the sounds you need without having to search through thousands of presets. However, the most powerful feature of Embracer is its surround out put. With a single switch, you can turn the instrument from stereo to surround and the width control allows you to spread your pad sound anywhere from mono to stereo to full 360° surround. The unique “eye” controller gives you an exact idea of how the sound is placed in a mix.

If you have never worked with a surround system before, now is the time to start exploring these possibilities.

• “Eye” controller for simultaneous tone and width control.

• Full MIDI control implementation.

Osc 1 and 2

Parameter Description

Wave Selects the waveform for each oscillator. Available wave-

Tone Embracer offers a high pass and low pass filter for each

Width Controls the spatial spread of the signal. A value of 0 %

Coarse (Oscillator 2 only)

Fine (Oscillator 2 only)

forms are: Carpet, DigiPad, Choir, Ensemble, Metal Phaze, Phase Strings, Sing Sing, Soft Wave, Spit Strynx, Step floor, Submerged, Wave Bellz. Note: If you want to use only one oscillator, set the waveform to OFF. In this case only one voice per key is used.

oscillator. Both filters are controlled via a single Tone knob. In the 50 Reducing the tone value adds low pass filtering. Values above 50 also be controlled by the “eye” controller.

puts the signal mono into the center position. In stereo mode, a value of 100 width. In surround mode, a value of 100 360° surround image. The width parameter can be con trolled by a variety of modulation sources, as well as by the “eye” controller.

Changes the pitch in semitones. Maximum range is +1/24 semitones = 2 octaves.

Changes the pitch in fine steps with a range of up to ±50 cents. Note: If you want to create a slight detune effect between the oscillators, make sure to set the master tune parame ter to a negative value of the same amount to keep the instrument in tune.

% center position, the signal is not filtered.

% add high pass filtering. This parameter can

% results in a maximum stereo

% creates a full

The Embracer Surround Pad Synthesizer has the following properties:

• Embracer is a Polyphonic surround pad synthesizer.

• 2 oscillators with 12 waveforms.

• Independent envelope and tone controls.

• Stereo and surround outputs.

• Up to 32 voices of polyphony per instance.

• Dynamic width control for exciting 3D sounds.

The included VST instruments

Envelope and Level

Parameter Description

Attack Controls the attack time of each oscillator. Higher values

Attack Vel Sets the amount of velocity control of the attack time.

Level Controls the oscillator output level. Level Vel Sets the amount of velocity control of the oscillator level.

create slower attacks.

Higher values increase the velocity sensitivity.

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Master

Parameter Description

Release Controls the overall release time of the volume envelope.

Mode Sets the output mode of Embracer. You can choose be-

Width Ctr Use this parameter to select a modulation source for the

Max Poly Sets the total number of voices available. Each oscillator

Fine Tune Use this to adjust the pitch of the whole instrument.

Master Out Sets the overall output volume of the instrument.

Higher values result in longer release times.

tween “Stereo” and “Surround”. In Stereo Mode, Embracer has one stereo output in the VST Mixer. In Surround Mode, Embracer has either a quadraphonic 4-channel out put or two independent stereo outputs in the Mixer. See below for more details on using Embracer in a surround mixer setup.

width parameter. Available sources are: Mod Wheel, Af tertouch, Velocity and Envelope. Both oscillators are controlled simultaneously. However, modulation depth is controlled independently by the re spective width parameter of each oscillator.

uses one voice per note played. Thus, a two-oscillator sound with 8 voices results in 4-voice polyphony. The de fault value for Max Poly is 16.

Range is ±50 cents. Use Fine Tune in combination with the Fine Tune parameter of OSC 2 to create smooth de tune effects.

The “Eye”

The Embracer’s unique “Eye” controller offers a creative new way of controlling the sound’s overall character and shape. This controller gives you access to several para

meters at the same time. For each oscillator, there is a circle representing the tone

and width of the sound. Click and drag the corresponding circle to change its shape. There are also two (numbered) oscillator handles. You can drag these vertically to change the tone or horizontally to change the width of the respec tive oscillator. When you drag a handle, the respective Tone and Width knobs of the oscillator are adjusted ac

cordingly. Play a note while editing to hear the effect. The “eye” cannot only be used as a controller for the tone

and width parameters, but also works as a surround scope for monitoring the spatial integration of the current sound. The display represents the sound’s position in the stereo or surround sound field. In stereo mode, the sound position is shown only in the upper half of the display and

represents the front part of the sound field. In surround mode, the sound position is shown in the upper and lower half of the display and represents the front and rear part of the sound field.

• You can use Embracer’s automation feature to record the movements of the mouse within the “eye” controller!

Using Embracer in Surround Mode

When you want to enjoy Embracer in 3D, set it up in surround mode and listen to it on a surround system. Let’s assume you have a surround monitoring system set up

with your VST mixer and your VST output connections are properly set up. Proceed as follows:

1. Open an instance of Embracer in the VST instruments rack and set it to surround mode.

2. When you open the mixer you see two separate stereo channels for the Embracer. The first is titled “Embracer” and the second “Embracer rear”.

3. Assign both channel outputs to the surround output bus.

The two channel strips now show independent surround panners. By default, the first output pair is assigned to the front left and right channels and the second output pair to the rear left and right channels. The sur round width can be controlled with the “width” parameter.

4. Double-click on the surround panner to open its control panel. Set the “Mono/Stereo” parameter to either “Y-Mirror”, “X-mirror” or “XY-mirror”. You can now freely adjust the surround panning to your taste.

5. If your surround configuration includes a center or LFE channel, you can also add some of Embracer’s signal to the center or LFE channels. Feel free to experiment to find

out what works best in a given project and mix.

The included VST instruments

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Groove Agent ONE

Groove Agent ONE is an easy-to-use sample-based MPC-style virtual drum machine for creating beats and reconstructing loops.

Audio samples can be associated with the Groove Agent ONE pads. Each pad is associated with a MIDI pitch, allowing you to trigger individual pads via MIDI notes.

To facilitate the creation of your own drum patterns, Groove Agent ONE provides a number of advanced functions.

Groups and pads

The pads and all functions related to the associating and auditioning of sounds can be found in the right half of the Groove Agent ONE panel.

Groove Agent ONE provides up to 128 pads, organized in eight groups of 16 pads. You can switch between the different groups by clicking on the corresponding group buttons (labeled 1 to 8) above the pads. Each pad is mapped to a particular MIDI note (C-2 to G8, which equals 128 notes).

• The button of the active group is highlighted. If one or more pads of a group have samples mapped to them, an additional red frame is displayed around group buttons.

By default, group 3 is active when you open Groove Agent ONE.

Pad functions

• The pads show the associated MIDI note in the top right corner.

You can change the MIDI note by right-clicking it and selecting a different note from the pop-up menu.

• You can assign up to eight samples to a pad.

See “Drag & drop of audio material” on page 78.

• If one or more samples have been assigned to a pad, the name of the first of these samples is displayed at the bottom of the pad.

To change the name, right-click it, enter a new name and press [Enter]. This allows you, e. this pad.

g., to indicate that more than one sample is mapped to

• To remove a sample assignment, click on the pad and drag the associated sample(s) to the trash icon in the LCD display to the left (see

Note that the trash icon is found only on either the Voice, Filter or Amplifier pages.

“Editing sounds” on page 79).

• The pad status is indicated by different colors.

During playback, a pad lights up yellow for as long as a sample mapped to this pad is played back. When either the Voice, Filter or Amplifier but ton is activated in the Pad Edit section and you click on a pad, it turns light green to indicate that it is selected for editing. Unselected pads not playing back any samples are gray.

• To select multiple pads for sound editing, [Ctrl]/[Command]-click on the pads.

The pad that has been selected first lights up light green, the rest of the selected pads turn dark green (see

“Editing sounds” on page 79).

• To mute or solo a pad, click the corresponding icon in the upper left corner of a pad.

The icon lights up to indicate that the pad is muted or soloed. If you solo a pad, all other pads are muted automatically. To unmute or unsolo the pad, click once more on the icon.

• You can drag a sample from one pad to another pad.

If the second pad already has a sample mapped to it, the sample assignment is swapped. Note that you can also swap the MIDI notes of the two pads by pressing [Shift] when dropping the sample.

• You can drag and drop samples between groups.

Click on a pad that has a sample mapped to it, keep the mouse button pressed and move the mouse pointer over the button of another group. When the pad display now changes to display the pads of the other group, drag and drop the sample on the desired pad.

Velocity

• The velocity is determined by where on the pad you click: it is lowest at the bottom of the pad and highest at the top.

• You can force all pads to a velocity value of 127 by activating the V-Max button in the Global section in the top right corner of the Groove Agent ONE panel.

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Resetting pads

You can find a Reset button in the Global section in the top right corner of the Groove Agent ONE panel. It allows you to clear all pad assignments of the current instance of Groove Agent ONE.

As a safety precaution, the Reset button is locked by default. Clicking the Reset button when it is locked has no effect.

To unlock the Reset button, hold down the [Shift] key while clicking. The button color changes to red. When you click Reset now, all pad assignments are reset.

The Reset button is re-locked automatically five seconds after unlocking it.

Drag & drop of audio material

Groove Agent ONE provides advanced drag & drop support. You can drag one or more samples at the same time from Cubase onto Groove Agent ONE. Samples can either be mapped to the same pad, or to different pads.

You can drag files to Groove Agent ONE from the following Cubase locations:

•MediaBay

• Project window

•Pool

• Sample Editor (regions)

• Audio Part Editor

• LoopMash (slices)

Layering samples on the same pad

When you select between one and eight samples and drag them to Groove Agent ONE, dropping them onto a pad (or onto the Layer indicator – see below) automatically creates a corresponding number of layers for this pad.

Drag & drop to several pads

Rather than dropping several samples to the same pad, you can also let Groove Agent ONE distribute samples across the available pads in one or several groups. To do so, select the desired samples, drag them to the Groove Agent ONE window, press [Shift] and drop the samples

onto a pad. The samples are mapped to the available pads, starting with the pad on which you initially dropped the samples, and then upwards according to the MIDI pitches of the pads.

How many samples can be dropped to several pads depends on the number of pads available in your current instance of Groove Agent ONE. If Groove Agent ONE cannot supply a sufficient number of free pads for the number of dropped samples, a dialog is displayed in which you can confirm or cancel the operation.

Replacing individual samples

To replace a sample mapped to one pad with another sample, proceed as follows:

• Drag the new sample to the pad, press [Alt]/[Option] and drop it.

To replace a sample in a pad layer with another sample, proceed as follows:

• Drag the new sample to the Layer indicator, press [Alt]/ [Option] and drop it onto the required layer.

Slicing a loop and triggering individual sounds via MIDI

Drag & drop to several pads has a number of uses. For example, it allows you to trigger individual sounds from an audio loop via MIDI. Proceed as follows:

1. Slice up a drum loop using the Sample Editor. Open the resulting audio part in the Audio Part Editor and press [Ctrl]/[Command]-[A] to select all audio events.

See the Operation Manual for details about slicing.

2. In the Audio Part Editor, click on one of the selected events and drag it to the Groove Agent ONE window.

3. Press the [Shift] key.

4. Point the mouse pointer at an empty pad and let go of

the mouse button.

The individual samples from the audio part are now mapped to the available pads of Groove Agent ONE.

Now look at the Exchange section (to the left of the pads): the MIDI Export pad (the field displaying a double arrow) at the bottom of the section is lit. When mapping several samples to several pads, Groove Agent ONE creates a MIDI file containing all MIDI information to trigger these pads, and maps this file to the MIDI Export pad.

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5. Drag this MIDI file from the MIDI Export pad onto the

Cubase Project window.

Dropping the file onto the Project window creates a new MIDI track. You can also drop the MIDI file to an existing MIDI or instrument track.

6. Play back the MIDI file.

The unedited MIDI file plays the same groove as the original audio loop. By editing the MIDI file you can change the original groove.

Ö You can also use LoopMash to slice an audio loop, and drag an individual slice directly from LoopMash to a Groove Agent One sample pad. For further information about LoopMash, see

“LoopMash” on page 82.

Saving the Groove Agent ONE setup

You can save the current configuration of Groove Agent ONE either as a plug-in preset or as a combination of a Groove Agent ONE archive (.gak) and a plug-in preset.

These presets and archives are useful in cases where you want to use your current settings and samples on a different computer.

• When a sample belonging to a preset cannot be found, Groove Agent ONE prompts you to locate the missing files. You can click either Ignore to skip this message, click Locate File to navigate to a specific folder containing the missing file(s), or click Search Folder to browse a spe cific folder and any subfolders that might contain the missing file(s).

Saving a GAK archive

You can save all Groove Agent ONE settings, and the sample files referenced by the current configuration, as a Groove Agent ONE kit. The file name extension of these kit files is “*.gak”. Proceed as follows:

1. Set up Groove Agent ONE the way you want it.

2. In the Exchange section, click the Export button.

The “Export Groove Agent ONE kit” dialog opens in which you can specify a location and a name for the new archive.

3. Click Save.

The archive is created and the dialog is closed.

Saving plug-in presets

You can save your current Groove Agent ONE configuration, including all settings for samples, pads and groups, as a plug-in preset.

1. At the top of the Groove Agent ONE window, click the VST Sound button to the right of the Presets pop-up menu and select “Save Preset”.

The Save Preset dialog opens.

2. Enter a name for the new preset and click OK.

The preset is saved in the User Content folder on your system.

Loading plug-in presets

To load an existing plug-in preset, proceed as follows:

1. At the top of the Groove Agent ONE window, click the VST Sound button and select “Load Preset” from the pop-up menu.

The Presets browser opens.

2. The Presets browser shows all presets it finds in the VST 3 Presets folder for Groove Agent ONE. Double-click the desired preset to load it.

The Presets browser is closed and the preset is loaded into Groove Agent ONE.

Note that a plug-in preset file is created alongside the .gak file. This plug-in preset references the samples inside the .gak file. It can be browsed in the MediaBay, giving you access to all Groove Agent ONE settings (including all samples) from within Cubase.

Loading a GAK archive

To load the GAK file, proceed as follows:

1. In the Exchange section, click the Import button.

Navigate to the GAK file.

2. Click Open.

The saved settings and all samples are imported into Groove Agent ONE.

Editing sounds

All sound editing functions can be found in and below the LCD display in the left half of the panel.

The LCD display can show four different sound editing pages, selected by clicking one of the four buttons in the Pad Edit section.

The information on the Play page refers to this instance of Groove Agent ONE as a whole. When the Play button is activated, the LCD display shows the name of the loaded VST preset and information on the number of samples and

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pads used by this instance of Groove Agent ONE. The Size parameter indicates the amount of RAM occupied by the currently loaded samples. The Polyphony counter shows the number of pads currently playing.

• Click on a pad for sound editing.

It turns light green and the display shows its sample parameters.

• To adjust a parameter, either use one of the quick controls below the display, or click on the parameter in the display and adjust it by dragging your mouse.

• You can select multiple pads for sound editing by [Ctrl]/ [Command]-clicking on them, and adjust their parameters in one go with the quick controls below the display.

The first selected pad lights up light green, all other selected pads turn dark green. The display shows the parameters of the first selected pad.

• By default, the parameters of the selected samples are adjusted in relation to their previous settings. If you want to set a specific value for all selected samples, [Ctrl]/ [Command]-click the quick control to set an initial value, release [Ctrl]/[Command] and adjust the value.

The parameter will be set to the same value for all selected sample pads.

On the Voice, Filter, and Amplifier pages, sample-specific data is displayed:

Parameter Description

Brightness slider

VST Preset The name of a loaded VST Preset is displayed in the top

Sample/Pad The name of the sample (and the pad to which it is as-

Trash icon You can remove the current sample assignment by click-

MIDI input off When the MIDI symbol button in the top right corner of the

Layer indicator The long bar near the top of the LCD display shows the

Layer number The layer number indicates which is the active layer of the

Use the little slider at the very top of the LCD display to set the display brightness.

left of the LCD display.

signed).

ing on a pad or on the Layer indicator (see below) and dragging it onto the trash icon.

LCD display is activated, the LCD display shows the waveform and parameter values of the currently playing sample. When this button is deactivated, the display shows only the data for the currently edit selected sample.

active layer for the current pad. If more than one layer ex ist for the selected pad, the bar is divided accordingly. You can drag the dividing line between layers to change the velocity ranges of the layers. You can drag a new sample from the MediaBay and drop it directly onto the Layer indicator bar (this is the same as dropping a sample on a pad). You can drag layers to a different position on the bar.

current pad.

Parameter Description

Sample This is the name of the sample file. Velocity Here you can specify a velocity range for the current

Coarse Here you can tune the sample by up to ±12 semitones. Fine This parameter lets you fine-tune the sample by up to

Volume Sets the sample volume. Waveform

display s/e locators in

waveform display

layer.

±100 cents.

The waveform of the current sample.

You can define the sample start and end points by dragging the s and e locators in the waveform display. When you click on a locator and press [Ctrl], this will zoom in on the waveform and center the display around the locator. Note that the locators automatically snap to zero cross ings.

Depending on the selected page (Play, Voice, Filter, Amplifier), up to six quick controls with different pad-specific parameter assignments are displayed.

Play parameters

The parameter controls on the Play page are copies of the parameters on the Voice, Filter, and Amplifier pages.

The row of parameter controls below the LCD display shows six parameters:

Parameter Description

Volume The volume of the pad currently selected for editing. Pan The panorama setting of the pad currently selected for

Coarse Use this control to tune the pad by up to ±12 semitones. Cutoff Sets the filter cutoff frequency. Q Sets the filter resonance. Output Groove Agent ONE provides up to 16 stereo outputs. You

editing.

can route pads to individual outputs using this control.

Voice parameters

The row of parameter controls below the LCD display

shows six parameters:

Parameter Description

Mode Here you can reverse the currently selected sample so

Coarse Use this control to tune the pad by up to ±12 semitones. Fine Use this control to fine-tune the pad by up to ±100

that you hear it backwards.

cents.

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Parameter Description

Mute Gr. With this control you can assign a pad to one of eight

Tr. Mode The sample of the currently selected pad is played either

Output Groove Agent ONE provides up to 16 stereo outputs.

mute groups. Pads within a mute group never play back simultaneously. New notes cancel previous notes.

from start to finish (One Shot) or only for as long as you hold the mouse button/key (Key Hold). Key Hold can also be determined by the length of the corresponding MIDI note on your track.

You can route pads to individual outputs using this con trol. See the Operation Manual for information on how to use multitimbral instruments in Cubase.

Filter parameters

The row of parameter controls below the LCD display shows four parameters used to edit the Groove Agent ONE filter:

Parameter Description

Type Sets the filter type: low-pass (LP), high-pass (HP) or band-

Cutoff Sets the filter cutoff frequency. Q Sets the filter resonance. Mod This parameter determines the influence that velocity has

pass (BP). When you set this knob to OFF, the settings on this editing page have no effect.

on the cutoff frequency. When set to 0 no effect. When set to any other value, the cutoff fre quency changes depending on the velocity.

%, the setting has

Amplifier parameters

The row of parameter controls below the LCD display shows six parameters:

Parameter Description

Volume The volume of the pad currently selected for editing. Pan The panorama setting of the pad currently selected for

Attack Controls the amplifier envelope attack time. Release Controls the amplifier envelope release time. Reduce the

Amp Mod This parameter determines the influence that velocity has

Attack Mod This parameter determines the influence that velocity has

editing.

release time to shorten the decay of sounds played in one-shot mode.

on the pad volume setting. When set to 100 sounds louder the higher the velocity. When set to 0 velocity has no effect on the pad volume.

on the Attack setting. When set to 0 effect on the attack. When set to 100 pad with high velocity, the Attack time is increased by

%. The higher the Attack Mod setting, the longer the

50 additional attack time for a pad.

%, the pad

%, velocity has no

% and playing a

Master volume

In the Master section in the lower left of the Groove Agent ONE panel you can find a master volume slider that sets the output volume of the instrument.

The Exchange section

This section is used to import or export data to/from Groove Agent ONE.

Importing MPC files

Clicking the Import button opens a file dialog in which you can navigate to a PGM file (.pgm is the AKAI MPC ex change format).

Ö Groove Agent ONE imports only the mapping data from the PGM file. Any additional information (on MPC effects, etc.) cannot be imported into Groove Agent ONE.

The MIDI Export pad is described in detail in the section

“Slicing a loop and triggering individual sounds via MIDI”

on page 78. The function of the Export button is described in detail in

the section

“Saving a GAK archive” on page 79.

Automation of Groove Agent ONE parameters

When opening an automation subtrack for a track that uses Groove Agent ONE, you can select the following plug-in parameters from the Add Parameters dialog:

•Volume

•Pan

•Mute

•Cutoff

• Resonance These parameters are available for the pads C1 to B4.

HALion Sonic SE

This VST instrument is described in detail in the separate PDF document “HALion Sonic SE”.

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LoopMash

LoopMash is one of a kind: a powerful tool for the slicing and instant re-assembling of any kind of rhythmic audio material. With LoopMash, you can preserve the rhythmic pattern of one audio loop, but you can replace all sounds of this loop with the sounds from up to seven other loops.

LoopMash provides dozens of possibilities to influence the way the slices are re-assembled, thus giving you full control over the results of your performance. You can choose from a variety of effects and apply them to single slices or to your overall performance. Finally, you can store your configuration as scenes on scene pads, and trigger these scene pads with your MIDI keyboard. All this turns LoopMash into a really powerful instrument for live perfor mances and recordings!

LoopMash is fully integrated into Cubase, which allows you to drag and drop audio loops from the MediaBay or Project window directly onto the LoopMash panel. Furthermore, you can drag and drop slices from LoopMash to the sam ple pads of Groove Agent One. This allows you to extract certain sounds that you like from LoopMash and use them with Groove Agent One. LoopMash supports the undo and redo functionality of Cubase, so that you can see and mod ify your steps in the Edit History dialog, as long as the LoopMash panel is open (for further information about the Edit History dialog, see the Operation Manual).

Getting started

To give you a first impression of what you can do with LoopMash, we have created a tutorial preset. Proceed as follows:

1. In Cubase, create an instrument track with LoopMash as the associated VST instrument.

2. In the Inspector for the new track, click the Edit Instrument button to open the LoopMash panel.

It has two main areas: the track section in the upper part of the panel, and the parameter section at the bottom.

3. At the top of the plug-in panel, click on the icon to the right of the preset field and select Load Preset from the pop-up menu.

4. The Presets browser opens, showing presets found in the VST 3 Presets folder for LoopMash.

5. Select the preset called “A Good Start…(Tutorial) 88”.

The preset is loaded into LoopMash.

6. At the bottom of the panel, make sure that the sync button in the transport controls is off, and start playback by clicking the play button.

In the LoopMash panel, you can see a sliced loop waveform in the top (red) track. This track is selected (which is indicated by the track’s background color and the lit button to the left of the waveform display).

The selected track holds the master loop. The rhythmic pattern of the LoopMash output is governed by the master

e. what you hear is the rhythmic pattern of this loop.

loop – i.

7. Look at the 24 pads below the track section: the pad labeled “Original” is selected. Select the pad named “Clap”.

A new loop is displayed on the second track in the track display, and you hear that the snare drum sound of the first loop has been replaced with a handclap sound.

8. Select the pad labeled “Trio”, and then the pad labeled “Section”. Each time you click, a new loop is added to the mash.

Note how the rhythmic pattern of the music stays the same, although an increasing number of sounds is taken from the other loops.

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9. Select other pads to find out how different parameter

Slices 1 to 4 selected for playback.

Master track slices for playback steps 1 to 4.

settings influence the LoopMash output. For a detailed description of the available parameters, see the section

“LoopMash parameters” on page 84.

Some of the pads have the same label, e. g. “Original” and “Replaced”. The scenes that are associated with these pads form the basis for varia tions of that scene. The variations of a scene are associated with the scene pads to the right of the original scene, i. “SliceFX” is a variation of the scene labeled “Original” and shows an ex ample for the usage of slice effects (see “Applying slice selection modi-

fiers and slice effects” on page 86).

e., the scene labeled

On the left of each track, you find the similarity gain sliders. These sliders are the most important control elements of LoopMash: the further to the right you move the similarity gain slider of a track, the more slices will be played back from this track.

How does LoopMash work?

Whenever you import a loop into LoopMash, the plug-in analyzes the audio material. It generates so-called “perceptual descriptors” (information on tempo, rhythm, spectrum, timbre, etc.) and then slices the loop into eighthnote segments.

This means that after you have imported several loops, LoopMash knows the rhythmic pattern of each loop and the location of various sounds that make up this pattern within each loop. During playback, LoopMash uses the perceptual descriptors to determine how similar each slice is to the current slice of the master track.

Note that LoopMash does not categorize the sounds, but looks for overall similarity in the sound. For example, Loop Mash might replace a low snare drum sound with a kick drum sound, even though a high snare sound is also avail able. LoopMash always tries to create a loop acoustically similar to the master loop, but using other sounds.

The similarity is shown by the brightness of each slice on each track, and also by the position of each slice on the similarity gain slider to the left of each track (when you click on a slice, its position is highlighted on the similarity gain slider). The brighter a slice, the more similar a slice is to the current master track slice, and the further to the right it is displayed on the similarity gain slider. Darker slices are less similar and can be found further to the left on the slider.

The similarity gain settings of the various tracks determine which slice gets playback priority. This creates a new loop, over and over again, but with the rhythmic pattern of the original master loop.

In the following figure you can see four tracks. The track at

the top is the master track. During playback, LoopMash moves through the master loop step-by-step (which is indicated by a rectangle in the track’s color around the cur-

rent slice) and automatically selects four slices from these tracks to replace the slices of the master track. The currently playing slice is indicated by a white rectangle around the slice.

The following figure shows the result of the selection process for each playback step.

For best performance, use audio files that have the same sample rate as your project (to avoid sample rate conversion when loading presets or storing scenes).

Experiment with the provided LoopMash presets, and with your own loops of different lengths and with different rhythms, containing many different sounds – LoopMash is like an instrument, and we very much encourage you to play it!

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LoopMash parameters

Similarity gain sliders

Ruler showing bars and beats

Track volume

Loop range selector

Master track on/off

VU meter

Track transposition value

Similarity threshold control

Track display

You can influence the process of constantly assembling a new loop with the various functions and parameter controls of LoopMash.

Ö Note that many of LoopMash’s parameters can be automated. See the description for the automation of VST instrument parameters in the chapter “VST instruments and instrument tracks” in the Operation Manual.

The track section

The track section contains the track display with the track controls for setting the track volume and a transposition value to the right of each track. To the left of the track dis play you will find the similarity gain sliders. With the button between the similarity gain slider and the track, you can define the master track that serves as the reference for rhythm and timbre. At the top of the track display you will find a ruler that shows bars and beats and the loop range selector.

Importing and removing loops

You can import up to eight audio loops onto the eight tracks in the track display. Proceed as follows:

1. Locate the audio loop that you want to import in one of the following locations: MediaBay and the MediaBay related browsers (e. g. the Mini Browser), Project window, Pool, Sample Editor (regions), Audio Part Editor, or the Explorer/Finder.

The quickest way to find the LoopMash content is to use the MediaBay: Navigate to the LoopMash content via the VST Sound node.

2. Drag the loop file onto a track in LoopMash.

Dragging a loop to a track already occupied replaces the original loop.

LoopMash separates the loop into slices, analyzes them, and displays them as a waveform on the track. One track can hold up to 32 slices. Even if a long loop were to con tain more than 32 slices, LoopMash imports only the first

32. Ideally, you would use a loop file cut at bar boundaries. When you import your file from the MediaBay, LoopMash uses the tempo information supplied by the MediaBay for the slicing of the loop.

• To remove a loop from a LoopMash track, right-click the track and select “Clear track”.

Defining the master loop

One track is always selected. This is the master track: it provides the rhythmic pattern that you hear, and it is the sounds of this loop that are replaced by slices selected from the other loops in the current LoopMash configura

tion.

• To make a track the master track, activate the button to the left of the track display.

Auditioning slices

To audition the slices, proceed as follows:

• Click on the slice that you want to hear.

• Use the Step function in the transport controls (see

“Transport controls” on page 87) to step through the

slices.

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Playback and master slice indicators

A rectangle in the track color around a slice indicates the current position within the master loop, i. e. the master slice. The slice currently selected for playback is indicated by a white rectangle.

Setting a loop range

At the top of the track display, a ruler showing bars and beats (using the project’s time signature) is displayed. In the ruler, you also find the loop range selector (the bracket) that defines the play length.

• To shorten the play length, click and drag the handles of the loop range selector (the bracket) at the top of the track display.

This allows you to select even a very small range within your master loop for playback – the rest of the loop is not taken into consideration. Note that short loop ranges (less than 1 bar) may conflict with the jump interval setting (see

“Storing your configuration as scenes” on page 87).

• To change the playback range, click the loop range selector and drag it to a different position as a whole.

Setting track transposition value and track volume

The track controls to the right of each track allow you to set a track transposition value and the track volume for each track individually.

• To set a track transposition value, click the button to the right of the track and select the desired transposition interval from the pop-up menu.

The set value is displayed on the button.

Ö This function is tied to the setting for the Slice Timestretch parameter (see “Audio Parameters” on page

88). When Slice Timestretch is deactivated, transposition

is created by increasing/decreasing the playback speed of the slices (transposing a track up by one octave corresponds to playing the slices twice as fast). With Slice Timestretch on, you get true pitch shifting, i.

e., there is no

change in playback speed.

• You can change the relative volumes of your tracks with the volume controls on the far right of each track.

This is useful for level adjustments between tracks. A VU meter to the left of the volume control provides visual feedback of the current volume.

Setting the similarity

With the similarity gain slider (to the left of each track) you can determine how important a particular track is for the “mashing up” of the master loop. By moving the slider, you specify that a track is more/less similar to the master track, thus “overruling” the result of the LoopMash analysis. As a result, more/less slices from this track are included in the current mash.

• Move the slider to the right to select more slices from the corresponding track for playback, and to the left to re duce the number of slices for playback (set to middle position by default).

The brightness of the slices changes when moving the similarity gain slider. The further to the right, the lighter the color, and the higher the playback priority for these slices.

The vertical lines on the similarity gain slider correspond to the slices in this loop. The changing pattern of slices indicates similarity of each slice, on all tracks, to the current master track slice. The further to the right a line is, the greater the similarity of this slice to the master slice.

• Drag the similarity threshold control (the thin line with handles at the top and bottom intersecting all similarity gain sliders) to the left or right to determine a minimum similarity that slices must match to be considered for play back.

Slices with a similarity below (i. e. to the left of) this threshold are not played.

On the Slice Selection page at the bottom of the LoopMash panel, you can make further settings for influencing which slices will be played (see

“Slice Selection” on page

88).

Creating composite tracks

LoopMash allows you to build composite tracks, i. e., as soon as you drag a slice to a different position on the same track or another track, you will be asked if you want to create a composite track.

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To build a composite track, proceed as follows:

Move this slider to specify the number of slices that the track will include.

Composite

track

1. Import the loop that you want to extract sounds from.

2. Audition the slices and drag the slices that you want to

use onto an empty track.

A dialog opens asking you to confirm that you want to create a composite track, and to determine the number of slices that the track will contain. If you enter a higher number of slices than the track actually contains, the track will be filled up with empty slices.

3. Click OK.

The destination track of the dragged slice becomes composite, indicated by a “C” to the left of the track.

You can use this feature in a very versatile way:

• You can assemble a combination of sounds that you like most on one track.

• You can define a certain rhythmic pattern by combining slices from different loops on a composite track and mak ing this track the master loop.

• You can use a composite track as a clipboard, allowing you to include sounds from more than eight loops into your mash.

You can use one track for importing and removing the loops that you want to search for sounds, and use the remaining seven tracks as com posite tracks. This allows for including up to 32 sounds from up to 32 different loop files on each of the seven composite tracks.

Ö Composite tracks are quantized according to the set tempo (see

“Transport controls” on page 87).

Applying slice selection modifiers and slice effects

Right-clicking a slice opens a context menu where you can influence the selection of individual slices and which effect is applied to them. The upper part of the context menu shows the slice selection modifiers. The following options are available:

Option Description

Always Only available for master track slices. The slice is played

Always Solo Only available for master track slices. The slice is played

Exclude The slice is never selected for playback. Boost Increases the similarity for this particular slice, so that it is

always.

always and exclusively (independent of the Voices parameter that you set on the Slice Selection page, see

Selection” on page 88).

played back more often.

Below the selection modifiers, the context menu shows the slice effects. The following options are available:

Option Description

Mute Mutes the slice. Reverse Plays the slice in reverse. Staccato Shortens the slice. Scratch A, B Plays the slice as if scratched. Backspin 4 Simulates a turntable backspin lasting over 4 slices. Slowdown Applies a slowdown. Tapestart Simulates a tapestart, i. e., speeds the slice up. Tapestop 1, 2 Simulates a tapestop, i. e., slows the slice down. Slur 4 Stretches the slice over 4 slice lengths. Slur 2 Stretches the slice over 2 slice lengths.

Stutter 2, 3, 4, 6, 8

Plays only the initial portion of a slice, and repeats it 2, 3, 4, 6, or 8 times during one slice length, respectively.

The best way to hear the results of the effects is to try them out!

Ö You can also apply effects to your overall performance

“Performance Controls” on page 89).

(see

“Slice

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Transport controls

Tempo field Play Locate Step left/right

Save scene Remove scene

Jump interval

Pad with associated scene

Selected scene

Empty scene pad

The transport controls can be found at the bottom of the LoopMash panel.

Button Description

Play Click the Play button to start or stop playback. Locate Click the Locate button to return to the beginning of the

Step Clicking the Step left/right button steps backwards/for-

Setting the LoopMash tempo

During playback, LoopMash can be synchronized to the tempo set in Cubase, or can follow its own tempo setting:

• Click the sync button (to the right of the Play button) to activate or deactivate synchronization to the project tempo set in Cubase.

When sync is on, you can start playback using the Cubase transport controls. With sync off, LoopMash starts playing only when you click the Play button in LoopMash.

• When the sync button is deactivated, the current LoopMash tempo (in BPM) is displayed in the tempo field to the left of the master button. To change the “local” tempo, click in the tempo field, enter a new value, and press [Enter].

• When the sync button is deactivated, you can click the master button (to the right of the tempo field) to copy the tempo of the current master loop into the tempo field.

The sync on/off parameter can be automated. This is useful to control LoopMash in a Cubase project – with sync off, the playback of Loop Mash within a project is paused.

Controlling transport functions with your MIDI keyboard

You can control the start, stop, sync on, and sync off function with your MIDI keyboard.

Function Key

Start C2 Stop D2 Sync on E2 Sync off F2

loop (bar 1/beat 1). Playback always starts automatically when clicking this button.

wards through the timeline, playing one slice at a time.

Ö If you do not have a MIDI keyboard connected to your computer, you can make use of the Virtual Keyboard feature in Cubase (see the Operation Manual).

Storing your configuration as scenes

On the Slice Selection and the Audio Parameters pages you will find a row of 24 pads. To each of these pads, you can save one “scene”, i. tracks with all parameter settings. By triggering the pads, you can quickly change between different scenes during your performance.

• To save the current settings as a scene, click the round button and then a pad.

This saves your setup to that pad.

• To recall a scene, click on the corresponding scene pad.

• To remove a scene from a pad, click the x button and then the desired pad.

• To edit a scene pad label, double-click on the scene pad and enter a name.

• To rearrange the scene pads, click on a scene pad and drag it to a new position.

Once you have set up a LoopMash configuration, you should save it to a scene pad. Changing scenes with-

The included VST instruments

out saving means discarding any unsaved changes.

Setting a jump interval

You can determine a point at which LoopMash changes to the next scene during playback when you trigger a pad. Proceed as follows:

• Click the Jump interval button and select an option from the pop-up menu that opens.

Ö The option “e: End” means that the current loop is played to the end before switching scenes. When you set up a short loop range, you may need to set the interval to “e: End” to ensure that the jump point is reached.

e. a combination of up to eight

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Triggering scene pads with your MIDI keyboard

As you can see, the scene pads are arranged according to the keys on a MIDI keyboard. You can trigger the 24 scene pads with a connected MIDI keyboard starting from C0 and ending with B1. You can also make use of the Vir tual Keyboard for triggering the scene pads (see the Operation Manual).

Slice Selection

Click the Slice Selection button (above the transport controls) to open the Slice Selection page. The options on this page allow you to further influence which slices will be selected for playback.

The following parameters are available:

Parameter Description

Number of Voices

Voices per Track This is the maximum number of slices that can be se-

Selection Offset Move this slider to the right to allow slices that are less

Random Selection

Selection Grid Here you can determine how often LoopMash will look

Here you can set the total number of slices from all tracks that are used to replace the master slice (ac cording to the current similarity gain settings). The range is from one (left) to four (right) voices, i. from up to four loops can play simultaneously. Increas ing the number of voices increases the CPU load.

lected from a single track. The range is from one to four. The less slices can be picked from the same track, the more variety you get in the LoopMash output.

similar to be selected for playback. This setting affects all tracks of this scene (see

as scenes” on page 87).

Move this slider to the right to allow more variation when selecting slices for playback, adding a more “ran dom” feel to the selection process. This setting affects all tracks of this scene (see

as scenes” on page 87).

for similar slices during playback: always (left position), or only every 2nd, 4th, or 8th (right position) step. For example, if you set the Selection Grid to every 8th step (right position), LoopMash will replace similar slices ev ery 8th step. Between two “replacement” steps it plays back the tracks of the slices that have been selected in the last replacement step, resulting in longer playback sequences on one track.

“Storing your configuration

e. sounds

Parameter Description

Similarity Method

Here you can modify the criteria that LoopMash considers when comparing the slices for similarity. There are three similarity methods: Standard – This is the standard method, where all slices on all tracks are compared and various charac teristics regarding rhythm, tempo, spectrum, etc. are taken into account. Relative – This method does not only consider the overall similarity of all slices on all tracks, but also takes the relation to the other slices on the same track into account, e. est sound on one track with the loudest, lowest sound on another track. Harmonic – This method only takes the analyzed tonal information into account, so that a slice is replaced by a harmonically similar slice, rather than by a rhythmically similar slice. With this method, also the track transposi tion value is considered, i. e., a master slice with a C D the transposition value of the track of the slice with the D ity gain sliders in a low position when you work with this method, because otherwise you may produce dishar monies. You can modify the transposition values to play back more slices of a specific track.

g., LoopMash may replace the loudest, low-

major chord will not be replaced by a slice with a major chord. But it will indeed be replaced if you set

major chord to “-2”. It is advisable to keep the similar-

Audio Parameters

Click the Audio Parameters button (above the transport controls) to open the Audio Parameters page. With the options on this page you can influence the sound of the LoopMash audio output.

The following parameters are available:

Option Description

Adapt Mode With Adapt Mode, you can adapt the sound of the

Adapt Amount Move this slider to the right to increase the adapta-

Slice Quantize Move this slider to the right to apply quantizing to

selected slice to the sound of the master slice. The available options are: Volume – changes the overall volume of the selected slice. Envelope – modifies volume changes within the slice. Spectrum – modifies the spectrum of the slice (equalization). Env + Spectrum – this is a combination of the Envelope and Spectrum modes.

tion specified with the Adapt Mode parameter.

the slices, i. note grid. When the slider is all the way to the left, the slices follow the rhythmic pattern defined by the original master loop.

e. the slices are aligned to a eighth-

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Option Description

Slice Timestretch Use this option to apply realtime timestretching to

Staccato Amount When you move this slider to the right, the length

Dry/Wet Mix This sets the balance between the volumes of the

the slices, filling gaps or avoiding overlaps be tween slices that are not played back at their original tempo, or when combining slices with different original tempos. Applying timestretch increases the CPU load and may affect the sound quality. Reduce the need for timestretching by using loops with similar original tempos. See also the descrip tion of the track transposition value (see “Setting

track transposition value and track volume” on page 85).

of the slices is gradually reduced, giving the output a staccato feel.

master loop and the selected slices from the other tracks.

Performance Controls

Click the Performance Controls button to open the Performance Controls page. On this page you find a row of buttons that are arranged according to the keys on a MIDI keyboard.

• By clicking these buttons during playback, you can apply effects to your overall performance.

An effect will be applied as long as you keep the button activated.

Most of the available effects correspond to the effects that you can apply to single slices, with the green buttons corresponding to the stutter and slur effects and the red buttons to the Mute, Reverse, Staccato effects, etc. (see

“Applying slice selection modifiers and slice effects” on page 86).

Ö Effects triggered with the Performance Controls buttons override the slice effects.

With the blue buttons and the yellow button, you can apply additional effects that cannot be applied to single slices:

Button Description

Cycle 4, 2, 1 Temporally sets up a short cycle over 4, 2, 1 slice,

Continue Plays back the tracks of the currently selected

respectively. This short cycle is always set up within the loop range that is set in the ruler (see

“Setting a loop range” on page 85). Setting up a

cycle over 1 slice means that this slice is repeated until you release the button.

slices continuously until you release the button.

Ö You cannot save these global effects in scenes. To apply effects and save them in scenes, you should use slice effects.

Triggering the Performance Controls with your MIDI keyboard

You can trigger the Performance Controls with your MIDI

keyboard starting from C3 upwards. You can also make use of the Virtual Keyboard for triggering the Performance Controls (for information about the Virtual Keyboard see the Operation Manual).

Saving and loading VST presets

You can save all current scenes as a VST preset. Proceed as follows:

1. At the top of the LoopMash window, click the icon to the right of the Preset field and select “Save Preset” from the pop-up menu.

The Save Preset dialog opens.

2. Enter a name for the new preset and click OK.

The preset is saved in the User Content folder on your system. Make sure that you tag your presets for better handling in the MediaBay.

To load an existing VST preset, proceed as follows:

1. At the top of the LoopMash window, click the icon to the right of the Preset field and select “Load Preset” from the pop-up menu.

The Presets browser opens.

2. The Presets browser shows all presets it finds in the VST 3 Presets folder for LoopMash. Double-click the desired preset.

The Presets browser is closed and the preset is loaded into LoopMash.

• When a loop belonging to a preset cannot be found, LoopMash displays a standard file dialog in which you can navigate to the file.

Ö The “Empty” preset clears all settings of the current LoopMash instance.

Loading VST presets saved with a former version of LoopMash

When you load a VST preset that was saved with a previous version of LoopMash, all new parameters will automatically be set to values that match the behavior of the previous LoopMash version.

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Ö To ensure MIDI control compatibility, the saved scenes are shifted to the scene pads 13-24, i. e., the scene on pad 1 is shifted to pad 13, the scene on pad 2 to pad 14, and so on.

Monologue – Monophonic Analog Modeling Synthesizer (Cubase only)

Monologue is a monophonic analog synthesizer based on physical modeling technology. It offers full, rich and color ful sounds without consuming a lot of CPU power. The Monologue synthesizer is the perfect tool for bass, lead and sequenced sounds.

The Monophonic Analog Modeling Synthesizer has the following properties:

• 2 oscillators with sawtooth, square and triangle waveforms.

• An additional noise generator for white noise.

• Monologue has two filters: a high pass filter and a versatile multimode filter.

• Monologue has a single LFO.

• Monologue has 4-stage ADSR mod and amp envelopes.

• Monologue has an effects section with chorus, phaser, and flanger effects, plus separate delay and overdrive units.

• Monologue has a X/Y matrix pad for additional realtime modulation with access to all Monologue parameters.

Osc 1 and 2

In the OSC 1 and OSC 2 sections, the following parameters are available:

Parameter Description

Waveform (pop-up menu)

Coarse Sets the coarse pitch in semitones. The available range is

Fine Allows you to fine-tune the pitch in cent increments. The

Depth Controls the pitch modulation depth for the mod source

Mod Src Defines the pitch modulation source. Available sources

PWM (OSC2 only)

Sync (OSC2 only)

This is where you select the waveform: Saw, Square and Sub for oscillator 1 and Saw, Square and Triangle for Os cillator 2.

±1 octave.

available range is ±50 cents.

defined in the “mod src” field. The available range is ±1 octave.

are: Modwheel, Aftertouch, Pitchbend, Velocity, LFO and Mod Env.

Controls the pulse width of the square wave. In the center position, pulse width is 50/50. Turning the PWM knob clockwise or counter clockwise creates a positive or neg ative pulse, respectively.

Activating the sync button synchronizes the pitch of oscillator 2 to the pitch of oscillator 1. When this is active, changing or modulating the pitch of oscillator 2 changes the tone and not the pitch. For the typical sync sound, turn osc 1 down in the mix and use osc 2 only.

Mix

In the Mix section, the following parameters are available:

Parameter Description

Osc 1 Sets the pre-filter level for oscillator 1. Noise Sets the pre-filter noise level. Osc 2 Sets the pre-filter level for oscillator 2.

Filter

In the Filter section, the following parameters are available:

Parameter Description

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

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

High Pass Sets the cutoff frequency of the additional high-pass filter. Res Changes the resonance of the multi-mode filter. Full res-

pass, 18

dB Low pass, 12 dB Low pass, 6 dB Low pass,

dB Band pass and 12 dB High pass.

ates is governed by the filter type.

onance puts the filter into self-oscillation.

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Parameter Description

Key Track Determines the amount of key tracking applied to the filter

Mod Src (A+B)

Depth (A+B)

cutoff frequency. The available range is 0 to 100 range of 100 keyboards pitch 1:1.

Defines the filter modulation source. The available sources are: Modwheel, Aftertouch, Pitchbend, Velocity, LFO, and Mod Env.

Controls the filter modulation depth for the mod source set in the “mod src” field.

% tunes the filter cutoff frequency to the

%. A

Envelope

In the Envelope section, the following parameters are available:

Parameter Description

A – (Attack) Sets the attack time. D – (Decay) Sets the decay time. S – (Sustain) Sets the sustain level. R – (Release) Sets the release time. Mod Src

(A+B)

Depth (A+B) Controls the envelope modulation depth for the mod

Defines the envelope modulation source. You can select: Modwheel, Aftertouch, Pitchbend, Velocity, LFO and Mod Env.

source defined in the “mod src” field.

LFO

In the LFO section, the following parameters are available:

Parameter Description

Waveform (pop-up menu)

Rate Adjusts the frequency of the LFO, thus changing the rate

Sync When “Sync” is “on” the LFO speed is synchronized to

Mod Src Defines the LFO modulation source. Available sources

Depth Controls the LFO modulation depth for the mod source

Here, you can select the waveform for the low frequency oscillator. Available waveforms are: Triangle, Square, Sawtooth, Sample & Hold and Random.

of the modulation. Depending on the LFO sync parame ter, you can edit the rate in Hertz or in note values.

the sequencer’s tempo. This also affects the LFO rate format.

are: Modwheel, Aftertouch, Pitchbend, Velocity, LFO and Mod Env.

defined in the “mod src” field.

X/Y Pad

In the X/Y Pad section, the following parameters are available:

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 Monologue’s param-

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

XY Pad.

eters 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.

Effects

In the Effects section, the following parameters are available:

Parameter Description

FX Type (pop-up menu)

Rate Use this to adjust the rate of the effect modulation. Depth Use this to adjust the depth of the effect modulation. FBK Controls the feedback of the effect. Mix Controls the balance between dry and wet (effect) signal.

Overdrive Controls the amount of overdrive (distortion) added to the

Delay Sets the delay time in musical values. The delay effect is

Spread Controls the stereo spread of the delay signal. If you set

Tone Adds a low pass filter to the delay. Increasing “tone”

FBK Controls the amount of feedback of the delay. High feed-

Mix Controls the balance between dry and wet (effect) signal.

Selects the effect type for Monologue’s pitch effects. The available types are Chorus, Flanger and Phaser.

Set to 0, the effect is off. Set to 50, the balance between dry and wet signal is 50/50.

slight amount of overdrive creates punch and

signal. A bottom. Higher amounts add distortion.

always in sync with the song tempo.

this to 0, the delay is centered mono. Higher amounts of spread shift the left and right delay channels. If you set this to 100, the delays “ping-pong” between the left and right channels at an even rate.

makes every delay repetition darker in tone.

back levels create infinite delays. Use this parameter with caution.

Set to 0, the effect is off. Set to 50, the balance between dry and wet signal is 50/50.

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Master

In the Master section, the following parameters are available:

Parameter Description

Glide Mode The available modes are: “held”, “on” and “off”. With

Rate Controls the glide rate – the time it takes for a note to

PB Range Controls the range of a pitch bend MIDI controller. Range

Env Trigger When set to “Multi”, each keystroke re-triggers the enve-

Note Priority Defines which note is played when multiple keys are held.

Oct Controls the master pitch of Monologue in octave steps.

Master Out Controls the master output level that is sent to the VST

Keyboard Pressing the “keyboard” button reveals a six octave virtual

“held” selected, a glide effect only occurs for notes played legato.

reach its destination pitch.

can be set between 1 and 24 semitones for a total of two octaves.

lopes. When set to “single”, legato notes do not retrigger the envelopes, effectively holding the envelopes on the sustain level until all keys are released before a new note is triggered.

Options are: First, Lowest, Highest, and Last.

Range is ±4 octaves.

mixer. Use it to adjust the balance between different pre sets. Use the VST mixer channel volume to control or automate the Monologue master volume.

keyboard. Pressing the “keyboard” button again hides the keyboard and displays the master section again.

Mystic

The synthesis method used by Mystic is based on three parallel comb filters with feedback. A comb filter is a filter with a number of “notches” in its frequency response, with the notch frequencies harmonically related to the frequency

of the fundamental (lowest) notch. A typical example of comb filtering occurs if you are using

a flanger effect or a delay effect with very short delay time. As you probably know, raising the feedback (the amount of signal sent back into the delay or flanger) causes a res onating tone – this tone is basically what the Mystic produces. This astonishingly simple synthesis method is capable of generating a wide range of sounds, from gentle plucked-string tones to weird, non-harmonic timbres.

The basic principle is the following:

• You start with an “impulse sound”, typically with a very short decay.

The spectrum of the impulse sound largely affects the tonal quality of the final sound. To set up an impulse sound on the Mystic you use a slightly simplified version of the synthesis found on the Spector synth.

• The impulse sound is fed into the three comb filters, in parallel. Each of these has a feedback loop.

This means the output of each comb filter is fed back into the filter. This results in a resonating feedback tone.

• When the signal is fed back into the comb filter, it goes via a separate, variable low pass filter.

This filter corresponds to the damping of high frequencies in a physical instrument – when set to a low cutoff frequency it causes high harmonics to decay faster than the lower harmonics (as when plucking a string on a guitar, for example).

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• The level of the feedback signal is governed by a feed-

back control.

This determines the decay of the feedback tone. Setting this to a negative value simulates the traveling wave in a tube with one open end and one closed end. The result is a more hollow, square wave-like sound, pitched one octave lower.

• A detune control offsets the fundamental frequencies of

the three comb filters, for chorus-like sounds or drastic special effects.

Finally you have access to the common synth parameters – two LFOs, four envelopes and an effect section.

• By default, envelope 2 controls the level of the impulse

sound – this is where you set up the short impulse decay when emulating string sounds, etc.

Ö The signal flow of the Mystic synth is illustrated in the section

“Diagrams” on page 118.

Sound parameters

The Impulse Control section

This is where you set up the impulse sound – the sound fed into the comb filters, serving as a starting point for the sound. The Impulse Control has two basic waveforms that are filtered through separate spectrum filters with adjust able base frequency; the output is an adjustable mix between the two waveform/spectrum filter signals.

• To set up the contour, click in one of the displays and drag the mouse to draw the desired curve. Note that this produces the inverse contour in the other display, for max imum sonic versatility.

To set up the contour independently for the two filters, hold down [Shift] and click and drag the mouse in either display.

• Use the Preset pop-up menu to select a preset contour if you like.

• If you want to random calculate a spectrum filter curve, you can choose the Randomize function from the Preset pop-up menu.

Each time you choose this function, a new randomized spectrum appears.

Waveform pop-up menu

The pop-up menu at the bottom of the waveform section (the central box at the top of the panel) allows you to select a basic waveform to be sent through filter contour A. The options are especially suited for use with the spectrum filter.

Cut

This offsets the frequency of the filter contour, working somewhat like a cutoff control on a standard synth filter.

To use the filter contour in its full frequency range, set Cut

to its maximum value.

Spectrum displays

The displays allow you to draw a filter contour with your mouse for spectrum filters A & B.

The included VST instruments

Morph

Adjusts the mix between the two signal paths: waveform A spectrum contour A and waveform B spectrum contour B.

Coarse

This offsets the pitch for the impulse sound. In a typical “string setup”, when the impulse sound is very short, this does not change the pitch of the final sound, but the tonal color.

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Raster

This removes harmonics from the impulse sound. As the harmonic content of the impulse sound is reflected in the comb filter sound, this changes the final timbre.

Comb filter sound parameters

Damping

This is a 6 dB/oct low pass filter that affects the sound being fed back into the comb filters. This means the sound becomes gradually softer when decaying, i.e. high harmonics to decay faster than the lower harmonics (as when plucking a string on a guitar, for example).

• The lower the Damping, the more pronounced this effect.

If you open the filter completely (turn Damping up to max) the harmonic content is static – i.e. the sound does not get softer when decaying.

Level

This determines the level of the impulse sound being fed into the comb filters. By default, this parameter is modu lated by envelope 2. That is, you use envelope 2 as a level envelope for the impulse sound.

• For a string-type sound, you want an envelope with a quick attack, a very short decay and no sustain (an “impulse” in other words), but you can also use other envelopes for other types of sounds.

Try raising the attack for example, or raising the sustain to allow the impulse sound to be heard together with the comb filter sound.

Crackle

This allows you to send noise directly into the comb filters. Small amounts of noise produce a “crackling”, erratic effect; higher amounts give a more pronounced noise sound.

Feedback

This determines the amount of signal sent back into the comb filters (the feedback level).

• Setting Feedback to zero (twelve o’clock) effectively turns off

the comb filter sound, as no feedback tone is produced.

• Setting Feedback to a positive value creates a feedback tone,

with higher settings generating longer decays.

• Setting Feedback to a negative value creates a feedback tone

with a more hollow sound, pitched one octave lower. Lower settings generate longer decays.

Detune

This offsets the notch frequencies of the three parallel comb filters, effectively changing the pitches of their feedback tones. At low settings, this creates a chorus-like detune effect. Higher settings detunes the three tones in wider intervals.

Pitch and Fine

Overall pitch adjustment of the final sound. This changes the pitch of both the impulse sound and the final comb filter sound.

Key Tracking

This button determines whether the impulse sound should track the keyboard or not. This affects the sound of the comb filters in a way similar to a key track switch on a regular subtractive synth filter.

Portamento

This parameter makes the pitch glide between the notes you play. The parameter setting determines the time it takes for the pitch to glide from one note to the next. Turn the knob clockwise for longer glide time.

The “Mode” switch allows you to apply glide only when you play a legato note (when switch is set to Legato). Legato is when you play a note without releasing the previously played note. Note that Legato mode only works with monophonic parts.

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Master Volume and Pan

The master Volume controls the master volume (amplitude) of the instrument. By default this parameter is controlled by Envelope 1, to generate an amplitude envelope for the oscillators.

The Pan knob controls the position in the stereo spectrum for the instrument. You can use Pan as a modulation des tination.

Modulation and controllers

The lower half of the control panel displays the various modulation and controller assignment pages available as well as the effect page. You switch between these pages using the buttons above this section.

The following pages are available:

• The LFO page has two low frequency oscillators (LFOs) for modulating parameters – see below.

• The Envelope page contains the four Envelope generators which can be assigned to control parameters – see

“Envelope page” on page 97.

• The Event page contains the common MIDI controllers (Mod wheel, Aftertouch, etc.) and their assignments – see

“Event page” on page 98.

• The Effect page has three separate effect types available; Distortion, Delay and Modulation – see “Effects

(EFX) page” on page 99.

LFO page

The LFO page is opened by clicking the LFO button at the top of the lower half of the control panel. The page contains all parameters and the modulation and velocity destinations for two independent LFOs.

Depending on the currently selected preset, there may already be modulation destinations assigned, in which case these are listed in the “Mod Dest” box for each LFO – see

“Assigning LFO modulation destinations” on page 96.

A low frequency oscillator (LFO) is used for modulating parameters, for example the pitch of an oscillator (to produce vibrato), or for any parameter where cyclic modulation is desired.

The two LFOs have identical parameters:

Parameter Description

Speed This governs the rate of the LFO. If MIDI Sync is activated

Depth This controls the amount of modulation applied by the

Waveform This sets the LFO waveform. Sync mode

(Part/MIDI/ Voice/Key)

About the sync modes

The Sync modes determine how the LFO cycle affects the notes you play:

Parameter Description

Part In this mode, the LFO cycle is free running and affects all

MIDI In this mode the LFO rate is synced in various beat incre-

(see below), the available rate values are selectable as note values, so the rate is synced to the sequencer tempo in Cubase in various beat increments.

LFO. If set to zero, no modulation is applied.

This sets the sync mode for the LFO. See below for a description.

the voices in sync. “Free running” means that the LFO cy cles continuously, and does not reset when a note is played.

ments to MIDI clock.

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Parameter Description

Voice In this mode each voice in the Part has its own indepen-

Key Same as Voice except that it is not free running – for

dent LFO cycle (the LFO is polyphonic). These cycles are also free running – each key down starts anywhere in the LFO cycle phase.

each key down the LFO cycle starts over.

About the waveforms

Most standard LFO waveforms are available for LFO modulation. You use Sine and Triangle waveforms for smooth modulation cycles, Square and Ramp up/down for differ ent types of stepped modulation cycles and Random or Sample for random modulation. The Sample waveform is different:

• In this mode, the LFO actually makes use of the other LFO as well.

For example, if LFO 2 is set to use Sample the resulting effect also depends on the speed and waveform of LFO 1.

Assigning LFO modulation destinations

To assign a modulation destination for an LFO, proceed as follows:

1. Click in the “Mod Dest” box for one of the LFOs.

A pop-up menu appears in which all possible modulation destinations are shown. All Sound parameters as well as most LFO and Envelope pa rameters are available as destinations.

2. Select a destination, e.g. Cut.

The selected modulation destination is now shown in the list. Beside the destination, a default value (50) has been set. The value represents the modulation amount.

• You can set positive and negative modulation values by clicking on the value in the list, typing in a new value and pressing the Enter key.

To enter negative values type a minus sign followed by the value.

3. Select a suitable LFO Waveform, Speed, Depth, and Sync mode.

You should now hear the Cut parameter being modulated by the LFO.

4. Using the same basic method, you can add any number of modulation destinations for the LFO.

They are all listed in the “Mod Dest” box.

• To remove a modulation destination click on its name in the list and select “Off” from the pop-up menu.

Assigning LFO velocity destinations

You can also assign LFO modulation that is velocity controlled (i.e. governed by how hard or soft you strike a key). Proceed as follows:

1. Click in the “Vel Dest” box for one of the LFOs.

A pop-up menu appears in which all possible velocity destinations are shown.

2. Select a destination.

The selected velocity destination is now shown in the list. Beside the destination, a default value (50) has been set. The value represents the modulation amount. See below for an example of how velocity modula

tion works.

• You can set positive and negative values by clicking on the value in the list, typing in a new value and pressing the Enter key.

To enter negative values type a minus sign followed by the value.

3. Using the same basic method, you can add any number of velocity destinations for the LFO.

They are all listed in the “Vel Dest” box.

• To remove a modulation destination click on its name in the list and select “Off” from the pop-up menu.

LFO modulation velocity control – an example: If you follow the steps above and select the Cut parameter

as a Velocity destination, the following happens:

• The harder you strike the key, the more the Cut parameter is modulated by the LFO.

• If you enter a negative value for the velocity modulation amount, the opposite happens; the harder you play the less the Cut parameter is modulated by the LFO.

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Envelope page

The Envelope page is opened by clicking the ENV button at the top of the lower half of the control panel. The page contains all parameters and the modulation and velocity destinations for the four independent envelope genera tors.

Envelope generators govern how a parameter value changes when a key is pressed, when a key is held and finally when a key is released.

On the Envelope page, the parameters for one of the four envelope generators is shown at a time.

• You switch between the four envelopes in the section to the left.

Clicking on either of the four mini curve displays 1 to 4 selects it and displays the corresponding envelope parameters to the right. The mini curve displays also reflect the envelope settings for each corresponding enve lope.

• Envelope generators have four parameters; Attack, Decay, Sustain, and Release (ADSR).

See below for a description of these.

• You can set envelope parameters in two ways; either by using the sliders or by click-dragging the curve in the Envelope curve display.

You can also do this in the mini curve displays.

• By default Envelope 1 is assigned to the master volume, and therefore acts as an amplitude envelope. The ampli tude envelope is used to adjust how the volume of the sound changes from the time you press a key until the key is released.

If no amplitude envelope were assigned, there would be no output.

• Envelope 2 is by default assigned to the Level parameter.

See “Level” on page 94.

The Envelope parameters are as follows:

Attack

The attack phase is the time it takes from zero to the maximum value. How long this takes is governed by the Attack setting. If the Attack is set to “0”, the maximum value is reached instantly. If this value is raised, it takes time before the maximum value is reached. Range is from 0.0 millisec onds to 91.1 seconds.

Decay

After the maximum value has been reached, the value starts to drop. How long this takes is governed by the Decay time parameter. The Decay time has no effect if the Sustain parameter is set to maximum. Range is from 0.0 millisec onds to 91.1 seconds.

Sustain

The Sustain parameter determines the level the envelope rests at after the Decay phase. Note that Sustain represents a level, whereas the other envelope parameters represent times. Range is from 0 to 100.

Release

Release determines the time it takes for the value to fall back to zero after releasing the key. Range is from 0.0 mil liseconds to 91.1 seconds.

Punch

When Punch is activated, the start of the decay phase is delayed a few milliseconds (the envelope “stays” at top level for a moment before moving on to the decay phase). The result is a punchier attack similar to a compressor ef fect. This effect is more pronounced with short attack and decay times.

Retrigger

When Retrigger is activated, the envelope re-triggers each time you play a new note. However, with certain tex tures/pad sounds and a limited number of voices it is recommended to leave the button deactivated, due to click noises that might occur, when the envelope is ended up abruptly. This is caused by the incoming re-trigger that forces the envelope to start over again.

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Assigning Envelope modulation destinations

To assign a modulation destination for an Envelope, proceed as follows:

1. Click in the “Mod Dest” box for one of the Envelopes.

A pop-up menu appears in which all possible modulation destinations are shown. All Sound parameters as well as most LFO and Envelope pa rameters are available as destinations.

2. Select a destination, e.g. Cut.

The selected modulation destination is now shown in the list. Beside the destination, a default value (50) has been set. The value represents the modulation amount.

• You can set positive and negative modulation values by clicking on the value in the list, typing in a new value and pressing the Enter key.

To enter negative values type a minus sign followed by the value.

3. Select a suitable envelope curve for the modulation.

You should now hear the Cut parameter being modulated by the envelope as you play.

4. Using the same basic method, you can add any number of modulation destinations for the envelope.

They are all listed in the “Mod Dest” box.

• To remove a modulation destination click on its name in the list and select “Off” from the pop-up menu.

Assigning Envelope velocity destinations

You can also assign Envelope modulation that is velocity controlled (i.e. governed by how hard or soft you strike a key). Proceed as follows:

1. Click in the “Vel Dest” box for one of the envelopes.

A pop-up menu appears in which all possible velocity destinations are shown.

2. Select a destination.

• You can set positive and negative values by clicking on the value in the list, typing in a new value and pressing the Enter key.

To enter negative values type a minus sign followed by the value.

3. Using the same basic method, you can add any number of velocity destinations for the Envelope.

They are all listed in the “Vel Dest” box.

• To remove a modulation destination click on its name in the list and select “Off” from the pop-up menu.

Envelope modulation velocity control – an example: If you follow the steps above and select the Cut parameter

as a Velocity destination, the following happens:

• The harder you strike the key, the more the parameter is modulated by the Envelope.

• If you enter a negative value for the velocity modulation amount, the opposite happens; the harder you play the less the Cut parameter is modulated by the Envelope.

Event page

The Event page is opened by clicking the EVENT button at the top of the lower half of the control panel. This page contains the most common MIDI controllers and their respective assignments.

The following controllers are available:

Controller Description

Modulation Wheel

Velocity Velocity is used to control parameters according to how

Aftertouch Aftertouch, or channel pressure, is MIDI data sent when

Key Pitch Tracking

The modulation wheel on your keyboard can be used to modulate parameters.

hard or soft you play notes on your keyboard. A common application of velocity is to make sounds brighter and louder if you strike the key harder.

pressure is applied to a keyboard after the key has been struck, and while it is being held down or sustained. Af tertouch is often routed to control filter cutoff, volume, and other parameters to add expression. Most (but not all) MIDI keyboards send Aftertouch.

This can change parameter values linearly according to where on the keyboard you play.

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To assign any of these controllers to one or several parameters, proceed as follows:

1. Click in the “Mod Dest” box for one of the controllers.

A pop-up menu appears in which all possible modulation destinations are shown. All Sound parameters as well as most LFO and Envelope pa rameters are available as destinations.

2. Select a destination.

The selected modulation destination is now shown in the list. Beside the destination, a default value (50) has been set. The value represents the modulation amount when the controller is at its full range.

• You can set positive and negative modulation values by clicking on the value in the list, typing in a new value and pressing the Enter key.

To enter negative values type a minus sign followed by the value.

3. Using the same basic method, you can add any number of modulation destinations for the controllers.

They are all listed in the “Mod Dest” box for the respective controller.

• To remove a modulation destination click on its name in the list and select “Off” from the pop-up menu.

Effects (EFX) page

This page features three separate effect units: Distortion, Delay and Modulation (Phaser/Flanger/Chorus). The Effect page is opened by clicking the EFX button at the top of the lower half of the control panel.

• Each separate effect section is laid out with a row of buttons that determine the effect type or characteristic and a row of sliders for making parameter settings.

• To activate an effect, click the “Active” button so that a dot appears.

Clicking again deactivates the effect.

• Tube Emulation produces distortion similar to valve amplifiers. The parameters are as follows:

Parameter Description

Drive Sets the amount of distortion by amplifying the input sig-

Filter This parameter sets the crossover frequency of the dis-

Tone This parameter controls the relative amount of low-pass

Level This controls the output level of the effect.

nal.

tortion filter. The distortion filter consists of a low pass filter and a high pass filter with a cutoff frequency equal to the crossover frequency.

and high-pass filtered signal.

Delay

You can select between 3 basic delay characteristics:

• Stereo Delay has two separate delay lines panned left and

right.

• In Mono Delay the two delay lines are connected in series for

monophonic dual tap delay effects.

• In Cross Delay the delayed sound bounces between the ste-

reo channels.

The parameters are as follows:

Parameter Description

Song Sync This switches tempo sync of the delay times on or off. Delay 1 Sets the delay time ranging from 0 ms to 728 ms. If MIDI

Delay 2 Same as Delay 1. Feedback This controls the decay of the delays. With higher set-

Filter A low pass filter is built into the feedback loop of the de-

Level This controls the output level of the effect.

sync is activated the range is from 1/32 to 1/1; straight, triplet or dotted.

tings the echoes repeat longer.

lay. This parameter controls the cutoff frequency of this feedback filter. Low settings result in successive echoes sounding darker.

Distortion

You can select between 4 basic distortion characteristics:

• Distortion provides hard clipping distortion.

• Soft Distortion provides soft clipping distortion.

• Tape Emulation produces distortion similar to magnetic tape saturation.

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Modulation

You can select between 3 basic modulation characteristics:

• The Phaser uses an 8-pole allpass filter to produce the classic phasing effect.

• The Flanger is composed of two independent delay lines with feedback for the left and the right channel respectively. The delay time of both delays is modulated by one LFO with ad

justable frequency.

• Chorus produces a rich chorus effect with 4 delays modulated by four independent LFOs.

The parameters are as follows:

Parameter Description

Song Sync This switches tempo sync of the Rate parameter on or off. Rate Sets the rate of the LFOs modulating the delay time. If

Depth This parameter controls the depth of the delay time mod-

Delay This parameter sets the delay time of the four delay lines. Feedback The feedback parameter controls the amount of positive

Level This controls the output level of the effect.

Song Sync is activated the rate is synced to various beat increments.

ulation.

or negative feedback for all four delay lines. The adjust able range is from -1 to 1.

SR parameters

With these buttons you can change the sample rate. Lower sample rates basically reduce the high frequency content and sound quality, but the pitch is not altered. This is a great way to emulate the “lo-fi” sounds of older digital synths!

• If button “F” is active, the selected Part’s program plays back with the sample rate set in the host application.

• If button “1/2” is active, the selected Part’s program plays back with half the original sample rate.

• If button “1/4” is active, the selected Part’s program plays back with a quarter of the original sample rate.

• A bonus effect of using lower sample rates is that it re-

duces the load on the computer CPU, allowing more simultaneous voices to be played, etc.

Prologue

Prologue is modelled on subtractive synthesis, the method used in classic analog synthesizers. It has the following ba sic features:

• Multimode filter

Variable slope low pass and high pass, plus band pass and notch filter modes – see

• Three oscillators, each with 4 standard waveforms plus an assortment of specialized waveforms.

See “Selecting Waveforms” on page 101.

• Frequency modulation.

See “About frequency modulation” on page 103.

• Ring Modulation.

See “Ring modulation” on page 103.

• Built-in effects.

See “Effects (EFX) page” on page 109.

• Prologue receives MIDI in Omni mode (on all MIDI channels).

You do not have to select a MIDI channel to direct MIDI to the Prologue.

Ö The signal flow of the Prologue synth is illustrated in the section “Diagrams” on page 118.

“About the filter types” on page 104.

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Steinberg Cubase Artist - 6.0, Cubase 6.0 Plug-in Reference

Specifications and Main Features

Frequently Asked Questions

User Manual

Table of Contents

The included effect plug-ins

Introduction

Delay plug-ins

ModMachine (Cubase only)

MonoDelay

PingPongDelay

StereoDelay

Distortion plug-ins

AmpSimulator

BitCrusher

DaTube

Grungelizer

Distortion

SoftClipper (Cubase only)

VST Amp Rack

Dynamics plug-ins

Compressor

DeEsser (Cubase only)

EnvelopeShaper

Expander (Cubase only)

Gate

Limiter

Maximizer

MIDI Gate

VintageCompressor (Cubase only)

VSTDynamics

EQ plug-ins

GEQ-10/GEQ-30 (Cubase only)

StudioEQ

Filter plug-ins

DualFilter

StepFilter

ToneBooster

Tonic (Cubase only)

WahWah

Mastering – UV22HR

Modulation plug-ins

AutoPan

Chopper

Chorus

Cloner (Cubase only)

Flanger

Metalizer

Phaser

RingModulator

Rotary

StudioChorus (Cubase only)

Tranceformer

Tremolo

Vibrato

Pitch Shift plug-ins

Octaver

Pitch Correct

Reverb plug-ins

REVerence (Cubase only)

RoomWorks (Cubase only)

RoomWorks SE

Spatial + Panner plug-ins

MonoToStereo

StereoEnhancer

Surround plug-ins (Cubase only)

Mix6To2

MixerDelay

Tools plug-ins

MultiScope (Cubase only)

SMPTEGenerator (Cubase only)

TestGenerator (Cubase only)

Tuner

MIDI effects

Introduction

Arpache 5

Arpache SX

Auto LFO

Beat Designer

Chorder