Steinberg Cubase SX - 3.0, Cubase SL - 3.0 User Manual

Audio Effects and VST Instruments

Manual by Anders Nordmark Quality Control: C. Bachmann, H. Bischoff, S. Pfeifer, C. Schomburg

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

All product and company names are ™ or ® trademarks of their respective owners. Windows XP is a trademark of Microsoft Corporation. The Mac logo is a trademark used under license. Macintosh and Power Macintosh are registered trademarks.

5 The included effect

plug-ins

6 Introduction 7 Delay plug-ins 10 Distortion plug-ins 16 Dynamics plug-ins 37 Filter plug-ins 45 Modulation plug-ins 58 Other plug-ins 66 Restoration 67 Reverb plug-ins 72 Surround plug-ins

(Cubase SX only)

77 Tools

79 The included

VST Instruments

80 A1 Synthesizer 90 VB-1 Bass Synth 92 LM-7 Drum Machine 95 Embracer –

Surround Pad Synthesizer (Cubase SX only)

100 Monologue – Monophonic

Analog Modeling Synthesizer (Cubase SX only)

107 Index

CUBASE SX/SL

4 Table of Contents

The included effect plug-ins

Introduction

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

In Cubase SX/SL, 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.

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Delay plug-ins

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

DoubleDelay

This effect provides two separate delays that can be either tempo based or use freely specified delay time settings. Cubase SX/SL automatically provides the plug-in with the tempo currently used in the project.

The parameters are as follows:

Parameter Description

Mix Sets the level balance between the dry signal and the effect. If Double-

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

Tempo sync on/off

Delay Time 1 This is where you specify the base note value for the delay if tempo

Delay Time 2 As above.

Feedback This sets the number of repeats for both delays.

Tempo Sync 1 The note value multiplier (x1 to x10) for the first delay unit.

The buttons above the two Delay Time knobs are used to turn tempo sync on or off for the respective delay. If set to off the delay time can be set freely with the Delay Time knobs, without sync to tempo.

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

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Parameter Description

Tempo Sync 2 As above, but for the second delay unit.

Pan1 This sets the stereo position for the first delay.

Pan2 This sets the stereo position for the second delay.

You can also change parameters in the graphic display window. This works as follows:

•

If tempo sync is on, you can set the Tempo Sync 1 parameter by dragging the light blue handle left and right.

When tempo sync is off, this sets the Delay Time 1 parameter.

•

You can set the Pan 1 parameter by dragging the light blue handle up and down.

•

The dark blue handle works in the same way but for the corresponding second delay parameters.

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ModDelay

This is a delay effect that can either be tempo-based or use freely specified delay time settings. The delay repeats can also be modulated.

The parameters are as follows:

Parameter Description

Mix Sets the level balance between the dry signal and the effect. If ModDe-

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

Tempo sync on/off

Feedback This sets the number of repeats for the delay.

Delay Time This is where you specify the base note value for the delay if tempo sync

Tempo Sync knob

DelayMod. This controls the pitch modulation rate for the delay effect.

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

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

This is the note value multiplier (x1 to x10) for the delay when tempo sync is used.

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Distortion plug-ins

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

DaTube

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

The parameters are as follows:

Parameter Description

Drive Regulates the pre-gain of the “amplifier”. Use high values if you want an

overdriven sound just on the verge of distortion.

Balance This controls the balance between the signal processed by the Drive pa-

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

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

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Overdrive

Overdrive is a distortion-type effect, emulating the sound of a guitar amplifier. A selection of factory styles is available. Note that these are not stored parameter settings, but different basic overdrive algorithms, with the style names indicating the basic character of each algorithm.

The parameters are as follows:

Parameter Description

Input Sets the input level.

Output Sets the output level. As overdrive generates harmonics, it increases

the level of the processed signal. You can use the Output fader to compensate for the level increase.

Speaker simulation

Factory Styles Select one of six presets, which can be used as they are or as a basis

Bass Tone control for the low frequencies.

Mid Tone control for the mid frequencies.

Hi Tone control for the high frequencies.

Drive Governs the amount of overdrive. You can also adjust this by clicking

When this is activated, the effect simulates the sound of a speaker cabinet.

for further “tweaking”.

and dragging in the display.

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QuadraFuzz (Cubase SX only)

QuadraFuzz is a high-quality distortion effect divided into four frequency bands allowing for control over the level both before and after distortion. This high level of control can create a very wide selection of distortion effects, ranging from subtle to extreme. The user interface consists of two windows.

•

The main window features four Filterbank controls, the master Gain and Output controls and a preset selector.

•

In the editor window (which is opened by clicking the “Edit” button in the lower right corner) the main feature is a frequency band display.

This is where you set the width of the frequency bands as well as their level before distortion.

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How does QuadraFuzz work?

Here’s a short description of the three major factors that determine how QuadraFuzz sounds, and where you find the corresponding controls:

• The signal volume control before distortion.

You can use the Gain control on the left side of the QuadraFuzz main window to control the overall input level of the signal that is fed into the distortion stage. The signal is split up into four frequency bands in the editor window, with adjustable width and level controls. These control the input level before distortion.

• The distortion type, based on a selectable distortion characteristic.

• The signal volume control after distortion.

The Output control on the right side of the QuadraFuzz main window controls the overall output level. In addition, the Filterbank controls in the same window allow you to raise or lower the output volume of each separate frequency band that was defined in the editor window.

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Editing in the frequency band display

The signal is divided into four frequency bands before being passed to the distortion stage, as explained earlier. You adjust the level and width of these bands in the frequency band display.

The frequency band display

Two value scales as well as a number of rhomb- and diamond-shaped handles are available.

• The diamond-shaped handles at the bottom are used to define the corner frequencies of the different frequency bands.

• By using the rhomb-shaped handles on top of each frequency band you determine its relative level before distortion.

• The horizontal value scale below the Frequency band display indicates frequency. The maximum value on this scale corresponds to half the sample rate of the audio file used (Nyquist theorem).

• The vertical value scale to the right shows the approximate level of an edited frequency band.

• If you click and hold on one of the handles, its current value is displayed. Depending on the handle type, corner frequency or level is shown.

• The corner frequency handles can be moved by dragging horizontally. The level handles can be moved by dragging them up or down.

• To reset a level handle to 0 dB, hold down the [Shift] key on your computer keyboard and click on the handle.

• If you hold down the [Ctrl]/[Command] key and move a handle, the values will change in smaller steps.

• The “Solo” button above the frequency band display allows you to monitor individual frequency bands.

If Solo is activated, one of the four bands is highlighted indicating the selected band. You select other bands by clicking on them.

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The parameters

The following tables list all parameters available in QuadraFuzz.

The parameters in the main window are as follows:

Parameter Description

Gain dial This dial can be found in the lower left corner of the QudraFuzz win-

dow. You can use it to control the level of the overall input signal before distortion.

Filterbank dials: Low/Low Mid/ High Mid/High

Presets fader This is used to select one of the available presets. To select a new

Output dial This controls the overall output level.

Over LED When lit, this indicates that the total input signal level exceeds 0 dB.

Edit button By clicking on this button, located in the lower right corner of the

These dials are used to control the output level of the corresponding frequency band after distortion. Values between +/- 12 dB can be set for each band.

preset, click on the fader handle and drag horizontally.

This LED does not refer to the output level but solely to the input level before distortion. Levels above 0 dB are subject to strict limiting and cause signal clipping. As this is sometimes what you want, QuadraFuzz also offers this option.

main window, you can open or close the editor window.

The parameters in the editor window are as follows:

Parameter Description

Create If you click on this button, a dialog will open where you can add (and

name) a new preset to the preset set currently in memory. The presets are stored with the project – to make a preset available in other projects you use the File pop-up menu as usual.

Delete This deletes the selected preset from the preset set currently in mem-

ory. If you click on the button, a dialog appears where you can confirm or cancel the action.

Solo This mutes all frequency bands except the selected band.

Shape buttons The available distortion characteristics (from bottom to top) create ef-

fects from a slight distortion up to a trashy hardcore sound.

Frequency band display

Here you control the level and bandwidth for the four bands, see above.

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Dynamics plug-ins

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

SPL DeEsser (Cubase SX only)

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

The SPL DeEsser has the following parameters:

Parameter Description

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

you start with a value between 4 and 7.

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

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

Auto Threshold See separate description below.

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

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

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About the Auto Threshold function

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

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

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

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Dynamics

Compressor

Routing selector

AutoGate

Limiter

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

Activating the individual processors

You activate the individual processors by clicking on their labels. Activated processors have highlighted labels.

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The AutoGate section

Gating, or noise gating, is a method of dynamic processing that silences audio signals below a certain set threshold level. As soon as the signal level exceeds the set threshold, the gate opens to let the signal through. AutoGate offers all the features of a standard noise gate, plus some very useful additional features, such as auto-calibration of the threshold setting, a look-ahead predict function, and frequency selective triggering.

The available parameters are as follows:

Parameter Values Description

Threshold -60 - 0dB This setting determines the level where AutoGate

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

Attack 0,1 -100 ms or

“Predict mode”

Hold 0 - 1000 ms This determines how long the gate stays open af-

Release 10 - 1000 ms or

“Auto”

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

ter the signal drops below the threshold level.

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

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Trigger Frequency Range

AutoGate has a feature that allows the gate to be triggered only by signals within a specified frequency range. This is a most useful feature because it lets you filter out parts of the signal that might otherwise trigger the gate in places you don’t want it to, thus allowing more control over the gate function. The Trigger Frequency Range function is set using the control in the upper part of the AutoGate panel, and the buttons located below it.

The basic operation of the Trigger Frequency Range function is as follows:

While playing back audio, click the “Listen” button.

You will now monitor the audio signal, and the gate will be bypassed.

While listening, drag the two handles in the Trigger Frequency Range display to set the frequency range you want to use to trigger the gate.

You will hear the audio being filtered as you move the handles.

• Dragging the left handle to the right will progressively cut frequencies starting from the low end of the frequency spectrum.

• Dragging the right handle to the left will progressively cut frequencies starting from the high end of the frequency spectrum.

The frequency range between the two handles will be used to trigger the gate.

After setting the frequency range, click the “On” button.

AutoGate will now use the selected frequency range as the trigger input.

To disable the Trigger Frequency Range function, click the “Off” button.

AutoGate will now use the unfiltered audio signal as the trigger input.

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The Calibrate function

This function, activated by using the Calibrate button located below the Threshold knob, is used to automatically set the threshold level. It is especially useful for material with consistent inherent background noise, like tape hiss. This may most of the time be masked by the audio content, but becomes noticeable during silent passages.

Use it as follows:

Find a part of the audio material, preferably not too short, where only the background noise is heard.

If you can only find a short background noise section, try looping it.

Play it back, and click on the Calibrate button.

The button will blink for a few seconds, and then automatically set the threshold so that the noise will be silenced (gated) during passages where there is no other signal present.

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The Compressor section

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

The available parameters work as follows:

Parameter Values Description

Threshold -60 - 0dB This setting determines the level where Compressor

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

Ratio 1:1 - 8:1 Ratio determines the amount of gain reduction applied

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

Attack 0.1-100 ms This determines how fast Compressor will respond to

signals above the set threshold. If the attack time is long, more of the early part of the signal (attack) will pass through unprocessed.

Release 10-1000ms or

“Auto mode”

MakeUp Gain 0 - 24dB This parameter is used to compensate for output gain

Compressor Mode

RMS/Peak RMS mode operates using the average power of the

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

loss, caused by compression.

audio signal as a basis, whereas Peak mode operates more on peak levels. As a general guideline, RMS mode works better on material with few transients such as vocals, and Peak mode better for percussive material, with a lot of transient peaks.

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The Limiter section

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

The available parameters are the following:

Parameter Values Description

Threshold -12 - 0dB This setting determines the maximum output level.

Signal levels above the set threshold are affected, but signal levels below are left unaffected.

Release 10-1000ms or

“Auto mode”

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

The Routing section

In the Routing section you can set the signal flow order for the three processors. Changing the order of the processors can produce different results, and the available options allow you to quickly compare what works best for a given situation. Beside each processor label is a number. These numbers are used to represent the signal flow options shown in the Routing section. There are three routing options:

• 1-2-3 (Compressor-Gate-Limit)

• 2-1-3 (Gate-Compressor-Limit)

• 1-3-2 (Compressor-Limit-Gate)

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Magneto (Cubase SX only)

Magneto brings the positive qualities of analog recording to your digital system, by providing the following capabilities:

• Simulates “tape saturation” and “tape overdrive” in a very realistic manner.

• Adds warmth, punch, and brilliance to a sound.

• Allows you to emphasize the “small details” in the sound.

• Works great on bass and guitar recordings as well as on drums, including individual samples and drum loops.

• Makes sampled drums and percussion sound much more “natural” and “warm”.

• Removes the “hardness” otherwise associated with digital audio recording.

All this makes Magneto suitable for processing both single sounds and complete recordings. In other words; practically any recording that you want to make sound warmer or more “natural”.

The algorithm behind Magneto is based on extensive studies and measurements of analog tape recorders. Special care has been taken to transfer the results of these studies into the digital domain.

•

If your audio material has been recorded digitally with Emphasis, it contains a disproportionate amount of high frequencies. This will disturb the audio analysis in Magneto.

We recommend that you convert such material (removing Emphasis) before processing it with Magneto.

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About the Drive parameter and Magneto output levels

• Magneto is different from analog tape recorders in one respect: On an analog tape machine, you will get a lower output level when overdriving the tape “too far”. This is known as the “saturation” effect. In Magneto, high Drive settings do not have this effect on the Output level.

• Magneto needs headroom to perform its “magic”. For this reason you may note a decrease in Output level (compared to the Input level) when using very low Drive parameter settings (when the onTape meter shows levels below approximately +10dB). Since low Drive settings is not a normal situation (since the plug-in then practically doesn’t have any audible effect), this is not something you would normally encounter. However, if for some application a low Drive setting is required, you can compensate for the loss in level with the Output Level parameter, see later in this text.

Metering Switch

Use the “Level” buttons to switch the meters between three modes:

•

Input

In this mode, the level of the input signal is shown. This should never exceed 0dB, as mentioned above and described in more detail below.

•

onTape

In this mode, the meters show an equivalent of the level recorded on the simulated “tape”. See the description of the Drive parameter for more details.

•

Output

This shows the output level for the entire plug-in. This should never exceed 0dB, see below.

Clip LEDs

The Input and Output Clip LEDs, located on the corresponding “Level” meter buttons, show if the signal is too “hot” (clipping occurs) at the input or output. The advantage of these is that they indicate clipping regardless of the mode the meters are switched to.

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Input Level

This is used to make sure the input signal is strong enough, without exceeding full level (so that clipping is avoided).

• If your input is already normalized, or sufficiently hot, leave this knob at 0.0 dB.

• If you need to adjust the input level, switch the Level metering to Input. Then adjust the knob until the signal peaks are as close as possible to 0dB without ever exceeding that level!

Output Level

• Under normal conditions, the Output Level control should be left at 0.0dB. The DSP algorithm in Magneto includes an “auto-gain” function which tries to keep the output level as close as possible to 0dBfs, at high Drive settings.

• At very low drive settings (if onTape metering indicates peak levels at 7dB or less – see the Drive parameter description for more info) you might need to amplify the signal using the Output Level control. However, always do this with the Level metering Output button activated, so that you can check that clipping doesn’t occur.

• At very high HF-Adjust settings, you might need to back off a bit on the Output level. Again, use Output metering to check.

If “digital clipping” occurs

If clipping occurs, (if the sound is heavily distorted), start by switching to input metering and check the input levels. If the input levels seem OK, switch to Output metering and adjust the Output Level as needed.

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The main parameters

You can change the Magneto parameters in realtime – i.e. while the audio material is played back – and the changes take effect more or less immediately (depending on your system). This allows you to experiment to get a feeling for how the settings interact.

Input level, Output Level, “Level” buttons and Meters

These are used to adjust the level throughout the signal chain as described on the previous pages.

Drive

This is the main parameter. It is used to set the simulated analog tape “recording level”. The value corresponds to how far above normal working level (0dB) you want to “record” on the “analog tape”. For example, a setting of 7 means the “tape” is “overdriven” by 7dB.

The higher you set this, the more of the “tape saturation” effect you will get.

Please use the following guidelines:

• Start out with a Drive setting of 10dB. Then adjust to taste.

• The effect of this parameter varies drastically with the frequency content and other characteristics of the material. There is no “best setting” for all types of recordings.

• If the material you are processing is already compressed or has been recorded on analog tape, a high Drive setting is not recommended, since it will give the sound an unnatural character.

• When processing complete mixes, you will have to be more careful with the Drive settings than when processing individual recordings. If all you want is to add some “warmth” or “punch” to a complete mix, adjust the Drive setting carefully.

• Always use the onTape meter to check out the effect of the setting on the material. This meter has to go pretty far above the 0dB level for Magneto to have any audible effect on the sound. If the meter displays levels close to, or even below 0dB, you get no “overdrive” or tape saturation effect at all! If this occurs, you need to raise the Drive setting or adjust the input level.

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Characteristics

This affects the tonal characteristics of the “tape saturation” effect controlled by the Drive parameter, as described above.

Tape Speed

This switches the tape simulation between 15 and 30 ips (inches per second) tape speed. There are slight differences in the harmonic character of the two. How much you will actually be able to hear of this difference depends on the frequency content of the material.

HF-Adjust

Various types of tape, recording and playback equalizers and the general design of various tape machines has an overall impact on the character of the sound. This control is used to adjust the High frequency content of the material to simulate those differences. It also has an effect on the perceived “warmth” of the sound.

This parameter can be used to compensate for the loss in high frequency that the overdrive effect introduces. Unlike on a real tape recorder it can also be used to boost the high frequency contents, compared to the original!

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MIDI Gate

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

Setting up

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

To set it up, proceed as follows:

Select the audio to be affected by the MIDI Gate.

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

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

The MIDI Gate control panel opens.

Select a MIDI track to control the MIDI Gate.

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

Open the Output (“out:”) 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.

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

Make sure the MIDI track is selected and start playback.

Now play a few notes on your MIDI keyboard.

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

The following MIDI Gate parameters are available:

Parameter Description

Attack This is used for determining how long it should take for the Gate to open

after receiving a signal that triggers it.

Hold Regulates how long the Gate remains open after a Note On or Note Off

message (see Hold Mode below).

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

value set with the Hold parameter).

Note To Attack

Note To Release

Velocity To VCA

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

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

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

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

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

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MultibandCompressor (Cubase SX only)

The MultibandCompressor allows a signal to be split in up to five 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 Characteristics editors. You can specify the level, bandwidth and compressor characteristics for each band by using the various controls.

The Frequency Band editor

The Frequency Band editor is where you set the width of the frequency bands as well as their level before compression. Two value scales and a number of diamond-shaped handles are available. The vertical value scale to the right gives you a clue to the approximate input gain level of each frequency band.

The diamond-shaped handles provided in the Frequency Band editor can be dragged with the mouse. You use them to set the corner frequencies and the input gain levels for up to five frequency bands. The width of each frequency band can be adjusted by dragging horizontally.

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The Level handles can be moved by dragging them up or down. If you click and hold on a handle, its current value is displayed. Depending on the handle type, corner frequency or level is shown.

• The diamond-shaped handles at the bottom are used to define the corner frequencies of the different frequency bands.

• By using the diamond-shaped handles on top of each frequency band you can cut or boost the input gain by +/- 12dB before compression.

• To reset a Level handle to 0 dB, hold down the [Shift] key on your computer keyboard and click on the handle.

• If you hold down the [Shift] key and click on the corner frequency handles, they will be set to the same bandwidth (in octaves). The exact bandwidth they will be set to is dependent on the number of bands currently used.

If you hold down [Ctrl] (Win) or [Command] (Mac) and move a handle, the values will change in smaller steps.

Adding and removing frequency bands

To add a frequency band, drag the leftmost or rightmost corner frequency handle towards the middle of the window, and a new band will automatically appear (given that you have less than the maximum number of five bands active). To remove a frequency band, drag the second leftmost or second rightmost handle out of the left or right edge of the window respectively.

About the Frequency scale

The horizontal value scale below the Frequency band display indicates frequency. The maximum value on this scale corresponds to half the sample rate of the audio file used. Hence, if a 44.1kHz soundfile is used, the highest frequency will be 22kHz.

In the digital domain, only frequencies of up to half the sample rate used can be reproduced (Nyquist theorem). The values available in the Frequency band display do therefore depend on the sample rate of the audio material used.

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The Solo button

The Solo button in the lower right part of the MultibandCompressor panel can be used to separately monitor each of the frequency bands. This function is useful both when editing bandwidth settings and compressor characteristics.

• To select another band while solo is active, click somewhere in the (dark) area of the frequency band that you wish to monitor.

Using the Characteristics editor

By adding breakpoints and drawing curves you set the compressor characteristic. Before you start using the Characteristics editor, you have to select the frequency band you want to process. This is done in the Frequency Band editor by clicking in the area inside the frequency band.

• A selected band is highlighted for editing both in the Frequency Band and the Characteristics editors.

If you select another frequency band, the previously edited band characteristic is still shown in the Characteristics window, but it is no longer highlighted or editable until you select it again.

About breakpoints

• Clicking anywhere on the line will add a breakpoint.

• To remove a breakpoint, hold down [Shift] and click on it.

• The first breakpoint from which the line deviates from the straight diagonal will be the threshold point.

• Creating a curve in the area below the diagonal input/output line will cause compression. Compression decreases the output level in relation to the input level.

• Creating a curve in the area above the diagonal input/output line will cause expansion. Expansion increases the output level in relation to the input level.

About the Compressor type (MODE)

• Classic mode works like a standard compressor with fixed attack and release parameters.

• Complex mode features a new compression approach with a program adaptive circuit. The program adaptive compression automatically optimizes parameters according to the audio material.

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The Output dial

The Output dial controls the total output level that the MultibandCompressor passes on to Cubase SX/SL. The range available is +/- 12dB. If the SoftClip function (see below) is active, the Output dial instead controls the amount of soft clipping.

The SoftClip function

The SoftClip function is positioned at the very last stage of the internal signal path, right after the Output dial. When active, it will ensure that the total output to Cubase SX/SL never exceeds 0dB. It works by clipping the signal gently, generating harmonics which add a warm, tube-like characteristic to the signal.

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VST Dynamics

The VST Dynamics plug-in is similar to the Dynamics plug-in (see

page 18), but with the following differences:

• VST Dynamics has two additional modules: Auto Level and Soft Clip.

• The signal flow is fixed, in the order AutoGate-AutoLevel-CompressorSoftClip-Limiter.

Activating the individual processors

You activate the individual processors by clicking on their labels. Activated processors have highlighted labels. You can activate as many processors as you want, but remember that not all processors are designed to work together. For example, “Limit” and “SoftClip” are both designed to ensure that the output never exceeds 0dB, but achieves this in different ways. To have both of them activated would be unnecessary.

• To turn off all activated VST Dynamics processors, click the lit On button to the right in the panel.

Clicking the button again activates the same configuration of processors.

Auto Gate section

This is exactly the same section as the AutoGate in the Dynamics plug-in. See page 19 for details.

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Auto Level section

Auto Level reduces signal level differences in audio material. It can be used to process recordings where the level unintentionally varies. It will boost low levels and attenuate high level audio signals. Only levels above the set threshold will be processed, so low level noise or rumble will not be boosted. If the input level is greater than 0dB, Auto Level will react very fast, because it “looks ahead” in the audio material for strong signal levels and can attenuate levels before they occur, thus reducing the risk of signal clipping. Auto Level has the following parameters:

Parameter Description

Threshold Only levels stronger than the set threshold will be processed.

Reaction time buttons (Slow, Mid, Fast)

Here you can set the amount of time it takes for Auto Level to adjust the gain. Set this according to whether the program level changes suddenly or over a length of time.

Compressor section

This is exactly the same section as the Compressor section in the Dynamics plug-in. See page 22 for details.

Soft Clip section

Soft Clip is designed to ensure that the output level never exceeds 0dB, like a limiter. Soft Clip, however, acts differently compared to a conventional limiter. When the signal level exceeds -6dB, SoftClip starts limiting (or clipping) the signal “softly”, at the same time generating harmonics which add a warm, tubelike characteristic to the audio material. Soft Clip is simplicity itself to use as it has no control parameters. The meter indicates the input signal level, and thus the amount of “softclipping”. Levels in the green area (weaker than -6dB) are unaffected, while levels in the yellow-orange-red area indicate the degree of “softclipping”. The deep red meter area to the right indicates input levels higher than 0dB.

• Avoid feeding Soft Clip with excessively high signal levels as audible distortion may occur, although the output level will never exceed 0dB.

Limiter section

This is exactly the same section as the Limiter in the Dynamics plug-in. See page 23 for details.

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Filter plug-ins

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

Q (Cubase SX only)

Q is a high-quality 4-band parametric stereo equalizer with two fully parametric midrange bands. The low and high bands can act as either standard shelving filters or fixed-gain high/low-cut filters.

Making settings

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

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

2. Set the parameters for an activated EQ band.

This can be done in several ways:

• By using the knobs.

• By clicking a value field and entering values numerically.

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

By using this method, you control both the Gain and Frequency parameters simultaneously. The knobs turn accordingly when you drag points. In addition, if the Mid 1 and Mid 2 bands (M1 and M2) are activated there will be two points on each side of the Gain/Frequency point that control the width (Q) parameter. If you press [Shift] while dragging, values can be set in finer increments.

Parameters

Parameter Description

Low Freq (20-2000Hz) This sets the frequency of the Low band.

Low Gain (-20 to +20 dB) This sets the amount of cut/boost for the Low band.

Low Cut If this button is activated for the Low band, it will act as a

Low Cut filter. The Gain parameter will be fixed.

Mid 1 Freq (20-20000Hz) This sets the center frequency of the Mid 1 band.

Mid 1 Gain (+/- 20dB) This sets the amount of cut/boost for the Mid 1 band.

Mid 1 Width (0.05-5.00 Octaves)

Mid 2 Freq (20-20000Hz) This sets the center frequency of the Mid 2 band.

Mid 2 Gain (-20 to +20 dB) This sets the amount of cut/boost for the Mid 2 band.

Mid 2 Width (0.05-5.00 Octaves)

High Freq (200-20000Hz) This sets the frequency of the High band.

High Gain (-20 to +20 dB) This sets the amount of cut/boost for the High band.

High Cut If this button is activated for the High band, it will act as a

Output (-20 to +20 dB) This parameter allows you to adjust the overall output level.

Left/Stereo/Right/Mono Modes

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

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

High Cut filter. The Gain parameter will be fixed.

For stereo signals you can set independent curves for the left and right channels by clicking the corresponding button. If the Stereo mode is activated, the curve will be applied to both channels. When channel independent curves have been set, the left/ right channel curves will be colored green and red, respectively. The currently non-selected channel is shown with a dotted curve. If you activate Stereo mode after independent curves have been set, the currently active curve will be applied to both channels. Mono mode is automatically activated for mono signals and is otherwise unavailable.

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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 will be set to the pointer position.

Setting filter cutoff values in the grid window.

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

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

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

Selecting new patterns

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

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

• To select new patterns you use the pattern selector.

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

Pattern Selector

Using pattern copy and paste to create variations

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

• Click the Copy button with the pattern you wish to copy selected, select another pattern memory location, and click Paste.

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

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StepFilter parameters

Parameter/Value Description

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

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

Base Resonance This sets the base filter resonance. Resonance values set in the

Resonance grid window are values relative to the Base Resonance value. Note that very high Base Resonance settings can produce loud ringing effects at certain frequencies.

Glide This will apply glide between the pattern step values, causing

values to change more smoothly.

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

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

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

Mix Adjusts the mix between dry and processed signal.

Output Sets the overall volume.

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

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Tonic – Analog Modeling Filter (Cubase SX only)

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

The Tonic Analog Modeling Filter has the following properties:

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

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

12dB band pass and 12dB high pass modes.

• Adjustable drive and resonance up to self-oscillation.

• Envelope follower for dynamic filter control with an audio signal.

• Audio and MIDI trigger modes.

• Powerful step LFO with smoothing and morphing.

• X/Y matrix pad for additional realtime modulation with access to all

Tonic parameters.

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Filter

Parameter Description

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

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

Cutoff Sets the filter cutoff frequency. How this parameter operates is

governed by the filter type.

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

puts the filter into self-oscillation.

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

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

Mix Sets the balance between dry and effect signal.

Ch. Chose between mono or stereo operation. When set to mono, the

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

Env Mod

Parameter Description

Mode Tonic offers three types of envelope modulation:

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

Attack Controls the attack time of the envelope. Higher attack times result

in slower rise times when the envelope is triggered.

Release Controls the release time of the envelope. Higher release times re-

sult in slower envelope tails.

Depth Controls the amount of envelope control applied to the filter cutoff

level.

LFO Mod Using this parameter, envelope level modulates the LFO speed.

A rather stunning effect…

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LFO Mod

Parameter Description

Mode Sets the direction of the step LFO modulation. The available

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

Depth Controls the amount of LFO modulation applied to the filter cutoff

level.

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

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

Smooth The smooth parameter controls the smoothing of the LFO steps.

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

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

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

Steps Sets the number of steps played in sequence. Deactivated steps

are grayed out in the step window.

Preset 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).

Step Matrix Click into the step matrix to set the level for each of the 16 LFO

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

X/Y Pad

Parameter Description

X Par Sets the parameter to be modulated on the x axis of the XY Pad. All

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

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

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of Tonic’s parameters are available as destinations

nation. 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.

Modulation plug-ins

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

Chorus

The Chorus plug-in adds short delays to the signal, and pitch modulates the delayed signals to produce a “doubling” effect.

The parameters are as follows:

Parameter Description

Mix Sets the level balance between the dry signal and the effect. If Chorus

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

Shapes This sets the modulation waveform. Triangle produces smooth modu-

lation, saw produces ramp shaped modulation and pulse waveform produces stepped modulation.

Frequency This sets the modulation rate.

Delay This controls the depth of the Chorus effect.

Stages This adds one to three more delay taps, producing a thicker, multi-lay-

ered chorus effect.

• Note that clicking and dragging in the display allows you to adjust the Frequency and Delay parameters at the same time!

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Flanger

Flanger is a classic flanger effect with stereo enhancement.

The parameters are as follows:

Parameter Description

Mix Sets the level balance between the dry signal and the effect. If the Flanger

Output Sets the overall volume.

Tempo sync on/off

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

Tempo Sync knob

Shape Sync knob

Feedback This determines the character of the flanger effect. Higher settings pro-

Depth This sets the depth of the modulation sweep.

Delay This parameter affects the frequency range of the modulation sweep, by

Stereo Basis This sets the stereo width of the effect. 0% is mono, 50% original stereo,

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

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

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

This is the note value multiplier (x1 to x10) for the flanger sweep when tempo sync is used.

This changes the shape of the modulating waveform, altering the character of the flanger sweep.

duce a more “metallic” sounding sweep.

adjusting the initial delay time.

and 100% maximum stereo enhancement.

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You can also change parameters in the graphic display. This works as follows:

• If tempo sync is on, you can set the base note value by clicking the waveform and dragging left and right.

When tempo sync is off, this sets the Rate parameter.

• You can set the Depth parameter by clicking the waveform and dragging up and down.

This means you can freely adjust Rate and Depth at the same time by clicking and dragging.

• By click-dragging the green/blue line in the display left or right you can change the Stereo Basis parameter.

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Metalizer

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

Parameter Description

Mix Sets the level balance between the dry signal and the effect. If Metalizer

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

Output Sets the overall volume.

Tempo sync on/off

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

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

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

Sharpness Governs the character of the filter effect. The higher the value, the nar-

Tone Governs the feedback frequency. The effect of this will be more notice-

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

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

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

work as a static filter.

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

able with high Feedback settings.

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

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Phaser

The Phaser plug-in produces the classic “swooshing” sound that characterizes phasing. It works by shifting the phase of the signal and adding it back to the original signal, causing partial cancellation of the frequency spectrum.

Parameter Description

Mix Sets the level balance between the dry signal and the effect. If the

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

Output Sets the overall volume.

Tempo sync on/off

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

Feedback This sets the amount of feedback. A higher value produces a more pro-

Tempo Sync knob

Stereo Basis This sets the stereo width of the effect. 0% is mono, 50% original ste-

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

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

nounced effect.

This is the note value multiplier (x1 to x10) for the Phaser sweep when tempo sync is used.

reo, and 100% maximum stereo enhancement.

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You can also change parameters in the graphic display. This works as follows:

• If tempo sync is on, you can set the base note value by clicking the waveform and dragging left and right.

When tempo sync is off, this sets the Rate parameter.

• You can set the Feedback parameter by clicking the waveform and dragging up and down.

This means you can freely adjust the Rate and Feedback at the same time by clicking and dragging.

• By click-dragging the blue/green line in the display left or right you can change the Stereo Basis parameter.

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Ringmodulator

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

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

Parameter Description

Oscillator LFO Amount

Oscillator Env. Amount

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

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

Oscillator Frequency Sets the oscillator frequency +/- 2 octaves within the selected

Roll-Off Cuts high frequencies in the oscillator waveform, to soften the

LFO Amount controls how much the oscillator frequency is affected by the LFO.

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

range.

overall sound. This is best used when harmonically rich waveforms are selected (e.g. square or saw).

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Parameter Description

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

LFO Speed Sets the LFO Speed.

LFO Env. Amount Controls how much the input signal level – via the envelope gen-

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

Invert Stereo This inverts the LFO waveform for the right channel of the oscilla-

tor, which produces a wider stereo perspective for the modulation.

Envelope Generator (Attack and Decay dials)

Lock L<R When this button is enabled, the L and R input signals are

Mix Adjusts the mix between dry and processed signal.

Output Sets the overall volume.

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 sets 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.

merged, and produce the same envelope output level for both oscillator channels. When disabled, each channel has its own envelope, which affects the two channels of the oscillator independently.

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Rotary

The Rotary plug-in simulates the classic effect of a rotary speaker. A rotary speaker cabinet features variable speed rotating speakers to produce a swirling chorus effect, commonly used with organs. Rotary features all the parameters associated with the real thing. The included presets provide good starting points for further tweaking of the numerous parameters.

The parameters are as follows:

Parameter Description

Speed This controls the speed of the Rotary in three steps: Stop/Slow/Fast.

MIDI Ctrl Selects the MIDI continuous controller for the Speed parameter.

See page 54.

Mode Selects whether the Slow/Fast speed setting is a switch (left button

is lit), or a variable control (right button lit). When switch mode is selected and Pitch Bend is the controller, the speed will switch with an up or down flick of the bender. Other controllers switch at 64.

Overdrive Applies a soft overdrive or distortion.

Crossover Freq. Sets the crossover frequency (200-3000Hz) between the low and

high frequency loudspeakers.

Mic Angle Sets the simulated microphone angle. 0 = mono, 180 = one mic on

each side.

Mic Distance Sets the simulated microphone distance from the speaker in inches.

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Parameter Description

Low Rotor Amp Mod.

Low Rotor Mix Level

Hi Rotor Amp Mod.

Hi Rotor Freq. Mod.

Phasing Adjusts the amount of phasing in the sound of the high rotor.

Hi Slow Fine adjustment of the high rotor Slow speed.

Hi Rate Fine adjustment of the high rotor acceleration time.

Hi Fast Fine adjustment of the high rotor Fast speed.

Lo Slow Fine adjustment of the low rotor Slow speed.

Lo Rate Fine adjustment of the low rotor acceleration time.

Lo Fast Fine adjustment of the low rotor Fast speed.

Output Adjusts the overall output level.

Mix Adjusts the mix between dry and processed signal.

Adjusts amplitude modulation depth.

Adjusts overall bass level.

High rotor amplitude modulation.

High rotor frequency modulation.

Directing MIDI to the Rotary

For real-time MIDI control of the Speed parameter, MIDI must be directed to the Rotary.

• Whenever the Rotary has been added as an insert effect (for an audio track or an FX channel), it will be available on the output (“out:”) popup menu for MIDI tracks.

If Rotary is selected on the “out:” menu, MIDI will be directed to the plug-in from the selected track.

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Symphonic

The Symphonic plug-in combines a stereo enhancer, an auto-panner synchronized to tempo and a chorus-type effect. For best results, apply the Symphonic effect to stereo signals.

The parameters are as follows:

Parameter Description

Mix Sets the level balance between the dry signal and the effect. If Sym-

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

Tempo sync on/off

Tempo Sync pop-up

Tempo Sync knob

Delay This determines the delay time and thus the character of the chorus

Depth This controls the depth of the chorus effect. If you only want to use

Rate This sets the modulation rate for the chorus effect, if activated.

Stereo Basis When the Auto-panner is activated, this sets the stereo width of the

Output Adjusts the output level of the effect.

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

If tempo sync is on, this is where you specify the base note value for tempo syncing the auto-panning (1/1 - 1/32, straight, triplet or dotted).

This is the note value multiplier (x1 to x10), determining the timing of the auto-panning.

effect, if activated.

Symphonic as an auto-panner or a stereo enhancer, set this to 0%.

panning. When the Auto-panner is deactivated (Tempo sync off), this determines the depth of the stereo enhancer effect. 0% is mono, 50% original stereo, and 100% maximum stereo enhancement.

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You can also change parameters in the graphic display. This works as follows:

• You can set the Rate parameter by clicking the waveform and dragging left and right.

• You can set the Depth parameter by clicking the waveform and dragging up and down.

This means you can freely adjust Rate and Depth at the same time by clicking and dragging.

• By click-dragging the green/blue line in the display left or right you can change the Stereo Basis parameter.

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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.

Parameter Description

Mix Sets the level balance between the dry signal and the effect.

Output Adjusts the output level of the effect.

Tone Sets the frequency (pitch) of the modulating oscillator (1 to 5000 Hz).

Tempo sync on/off

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

On button Turns modulation of the pitch parameter on or off.

Mono button Governs whether the output will be stereo or mono.

Depth Governs the depth of the pitch modulation.

Waveform buttons

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

Sets the pitch modulation waveform.

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

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Other plug-ins

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

Bitcrusher

If you’re into lo-fi sound, Bitcrusher is the effect for you. It offers the possibility of decimating and truncating the input 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 parameters are:

Parameter Description

Mode Select one of four operating modes for the Bitcrusher. Each mode will

Depth Use this to set the desired bit resolution. A setting of 24 gives the high-

Sample Divider

Mix This slider regulates the balance between the output from the Bitcrusher

Output Governs the output level from the Bitcrusher. Drag the slider upwards to

produce a different sounding result. Modes I and III are nastier and noisier, while modes II and IV are more subtle.

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

This sets the amount by which the audio samples are decimated. At the highest setting (65), nearly all of the information describing the original audio signal will be eliminated, turning the signal into unrecognizable noise.

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

increase the level.

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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 parameters are as follows:

Parameter Description

Mix Sets the level balance between the dry signal and the effect. If Chopper

is used as a send effect, this should be set to maximum.

Tempo sync on/off

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

Stereo/Mono button

Waveform buttons

Depth Sets the depth of the Chopper effect. This can also be set by clicking in

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

tempo-syncing the effect (1/1 - 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, without sync to tempo.

Determines whether the Chopper will work as an auto-panner (button set to “Stereo”) or a tremolo effect (button set to “Mono”).

Sets the modulation waveform.

the graphic display.

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Apogee UV 22 HR (Cubase SX only)

The UV22 HR is a dithering plug-in, based on an advanced algorithm developed by Apogee (for an introduction to the concept of dithering, please refer to the chapter “Audio Effects” in the Operation Manual). You can use the UV22 HR plug-in for all dithering situations, except when working with surround audio. This is because the UV22 HR is a standard “stereo in” – “stereo out” plug-in (as opposed to the SurroundDither plug-in, see page 74).

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

Option Description

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

Low This applies a lower level of dither noise.

Autoblack When this is activated, the dither noise is gated (muted) during silent

passages in the material.

Bit Resolution The UV22 HR supports dithering to multiple resolutions: 8, 16, 20 or

24 bits. You select the desired resolution by clicking the corresponding button.

Dither should always be applied post output bus fader.

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Vocoder

The Vocoder can apply sound/voice characteristics taken from one signal source, called the “modulator” and apply this to another source, called the “carrier”. A typical application of a vocoder is to use a voice as a modulator and an instrument as a carrier, making the instrument “talk”. A vocoder works by dividing the source signal (modulator) into a number of frequency bands. The audio attributes of these frequency bands can then be used to modulate the carrier.

The Vocoder has a built-in carrier (basically a simple polyphonic synthesizer) but you can also use an external carrier, see page 62.

Setting up – using MIDI

In this mode, the Vocoder is set up slightly differently than other plugin effects. This is because this setup requires both an audio signal (as the modulator source) and a MIDI input (to play the carrier) to function. To set up for using an external carrier, see page 62.

To set up for use, proceed as follows:

1. Select a source for the modulator.

The modulator source can be audio material from any audio track, or even a live audio input routed to an audio track (provided you have a low latency audio card).

• Good modulator source material are talking or singing voices or percussive sounds, e.g. drum loops.

Static pads or soft ambient material are generally less appropriate for use as modulators, but there are no absolute rules as to what could be used as a modulator source.

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2. Select the Vocoder as an insert effect for the audio channel with the modulator signal.

3. Make sure that the Vocoder Mode is set to “MIDI”.

4. Select a MIDI track.

This can be an empty MIDI track, or a MIDI track containing data, it doesn’t matter. However, if you wish to play the Vocoder in real-time – as opposed to having a recorded part playing it – the track has to have monitoring activated (or be record enabled) for the Vocoder to receive the MIDI output.

5. Select “Vocoder” from the MIDI “out:” pop-up menu for the MIDI track.

The MIDI Output from the track is now routed to the Vocoder. There is an indicator on the Vocoder panel below the Mode switches that blinks when receiving MIDI.

That concludes setting up – you are now ready to start vocoding!

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

6. Make sure the MIDI track is record enabled and start playback.

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

As you can hear, the audio track material, or rather its formant characteristics, is now applied to the Vocoder’s built-in sound source!

Setting up – using an external carrier

There are two modes for using an external carrier:

• “Ext” mode is when the carrier and the modulator can be any two audio sources.

The synth section is disabled and grayed out when this mode is selected. MIDI input and the Gap Thru Vocoder parameter are also disabled.

• “MIDI+Ext” mode mixes the audio carrier with the Vocoder’s synth sound.

This is described on page 63.

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To use an external carrier instead of the built-in synth (“Ext mode”), you set up as follows:

1. Create a Group channel from the Add Track submenu on the Project menu.

2. Open an audio file you wish to use as the carrier source and place it on an empty audio track.

3. Pan the audio channel full right in the Mixer or in the Inspector.

4. Route the output of the audio channel to the group.

5. Open an audio file you wish to use as the modulator source and place

it on another empty audio track.

Events on the two audio tracks (carrier and modulator) have to play back simultaneously for the Vocoder to work.

6. Pan the modulator audio channel full left in the Mixer or in the Inspector.

7. Route the output of the modulator audio channel to the group.

8. Select the Vocoder as an insert effect for the group channel.

9. Open the Vocoder panel and activate the “Ext.” Mode button.

10.If you now start playback, the carrier channel will be modulated by the

modulator channel!

Note that the synth section on the left half of the Vocoder panel and the “Gap Thru” parameter are now disabled.

Setting up – using an external carrier plus MIDI

Setting up is the same as for using an external carrier, except that a MIDI track with its output routed to the Vocoder should also be present. The MIDI track can either play the Vocoder synth in real time or from prerecorded parts. Make sure that monitoring (or record enable) is activated for the track so that the Vocoder synth will receive MIDI played in real time.

• Set up as described, and activate “MIDI+Ext.” mode on the Vocoder panel.

Any incoming MIDI now triggers the Vocoder synth, and the synths output is mixed with the audio carrier signal.

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Vocoder parameters

The Vocoder parameters govern the general sound quality of the vocoded sound.

Parameter Description

Number of Bands

Bandwidth This sets the bandwidth for the frequency bands, which affects the

Min./Max. Freq. These parameters set the minimum and maximum frequency limits for

log/lin Log/Lin controls how the frequency bands are spaced between the

Env.Speed This determines the attack and release times of the Vocoder envelope.

High Thru This lets through high frequencies around the “S” frequency from the

Talk Thru Adjusts the level of the original input signal passed to the Vocoder

Gap Thru Gap Thru (only available in MIDI mode) sets the level of the original in-

Output This controls the output level of the Vocoder.

Emphasis This is a highpass filter, gradually cutting lower frequencies while let-

This governs how many frequency bands the modulator signal is divided into (2-24). Fewer bands will provide a thinner more resonant sound, whereas using more bands will make the sound fuller and more intelligible.

overall timbre. Very narrow bandwidth settings will produce a thin, whistle-like sound.

the Vocoder, respectively.

minimum and maximum frequencies. Log = equal spacing in octaves, Lin = equal spacing in Hz. This affects the basic timbre of the Vocoder.

Fast settings will cause the modulator signal to trigger the Vocoder instantly, longer settings will gradually increase the attack/release times, providing a more subtle Vocoder effect. If set to “HOLD” the modulator is “frozen”, and doesn’t affect the carrier synth at all.

original input signal while notes are played.

output while notes are played.

put signal that is passed to the Vocoder output when no MIDI notes are being played. This lets you apply the Vocoder to a vocal track adding vocoded parts just where you want them.

ting high frequencies pass.

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Vocoder synth parameters

If the built-in synthesizer is the carrier, it is the sound of this instrument that the modulator source is applied to. The synth is polyphonic with up to 8 voices and features 2 oscillators per voice. The synth has the following parameters:

Parameter Description

Voices This sets the number of voices for the synth (1-8).

Fine Tune Tunes the oscillators ± a semitone, in cents (100th of a semitone) steps.

Pitch Bend Sets the up/down range of the Pitch Bend in semitone steps (1-12).

Noise Adds white noise to the sound.

NoiseMod This makes the oscillators modulate the noise level. This gives the noise

a rasping sound, turning “sss” into “zzz”.

P.Drift Adds random pitch variation to the oscillators.

P.Glide This makes the pitch glide between notes played. The parameter con-

trols the time it takes for the pitch to glide from one note to the next.

P.Bright This is a lowpass filter that can be used to soften the tone of the oscilla-

tors. It does not affect the white noise generator.

P.Detune Allows you to detune one of the oscillators in cent steps.

LFO Rate Controls the LFO rate (for vibrato).

Vibrato Adds vibrato to the oscillators. This can also be controlled by using the

Mod Wheel.

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Restoration

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

Grungelizer

The 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 available parameters are as follows:

Parameter Description

Crackle This adds crackle to create that old vinyl record sound. The farther to

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

Noise This dial regulates the amount of static noise added.

Distort Use this dial to add distortion.

EQ Turn this dial to the right to cut off the low frequencies, and create a

AC This emulates a constant, low hum of AC current.

Frequency switch

Timeline This dial regulates the amount of overall effect. The farther to the right

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the right you turn the dial, the more crackle is added.

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

more hollow, lo-fi sound.

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

(1900) you turn this dial, the more noticeable the effect.

Reverb plug-ins

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

Reverb A

Reverb A is a reverb plug-in which provides smooth, dense reverb effects. Reverb A has the following parameters:

Parameter Description

Mix Sets the level balance between the dry signal and the effect (wet). If

Reverb A is used as a send effect, this should be set to maximum wet, as you can control the dry/wet balance with the send.

Room Size This setting determines the “size” of the simulated room environment.

Predelay This parameter sets a delay between the direct sound and the reverb

effect output. A short predelay before the reverb reduces reverb “clutter” which blurs the sound, and makes the reverb effect more naturalsounding.

Reverb Time This parameter sets the length of the reverb time.

Filter HighCut This filters out high frequencies for the reverb, which can make the re-

verb sound softer.

Filter LowCut This filters out the lower frequencies for the reverb. It can be used to

reduce low frequency “rumble”.

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Reverb B

The Reverb B provides reverb with low processor demands. It has the following parameters:

Parameter Description

Mix Sets the level balance between the dry signal and the effect. If Reverb B

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

Room Size Governs the “size” of the simulated room environment.

Predelay This parameter sets a delay between the direct sound and the reverb

effect output. A short predelay before the reverb reduces reverb “clutter” which blurs the sound, and makes the reverb effect more naturalsounding.

Reverb Time This parameter sets the length of the reverb effect.

Damp This parameter “dampens” the higher frequencies, producing a rounder

and smoother sounding reverb.

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RoomWorks

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 reverberation. RoomWorks has the following parameters:

Parameter Description

High Freq Frequency at which the high shelving filter takes effect. Both the

high and low filters EQ the input signal prior to reverb processing.

High Shelf Gain The amount of boost or cut for the high shelving filter.

Low Freq Frequency at which the low shelving filter takes effect.

Low Shelf Gain The amount of boost or cut for the low shelving filter.

Predelay The amount of time before the onset of reverb. This allows you to

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

Time Reverb Time in milliseconds.

Hold 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.

Size This alters the delays times of early reflections to simulate larger

or smaller spaces.

Diffusion This affects the character of the reverb tail. Higher diffusion is

smoother while less diffusion can be clearer. This emulates changing the types of surfaces in a room (brick vs. carpet for instance).

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Parameter Description

Width This controls the width of the stereo image. At 100%, you get full

stereo reverb. At 0%, the reverb is all in mono.

Efficiency This unique control determines how much of the CPU is used for

RoomWorks. The lower the percentage of efficiency, the more CPU resources will be used. This will yield a higher quality reverb than higher percentage settings. Interesting effects can be created with very high Efficiency settings (>90%). Experiment for yourself.

Export This button determines if during audio export RoomWorks will

use the maximum CPU power for the highest quality reverb or not. You may wish to keep a higher efficiency setting for a desired effect during export. If you want the highest quality reverb during export make sure this is selected (yellow circle around button).

Variation Pressing this button will generate a new version of the same re-

verb 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.

High Damping Freq This determines the frequency above which high frequency

damping will occur.

High Damping Amount

Low Damping Freq This determines the frequency below which low damping will

Low Damping Amount

Envelope Amount This determines how much effect the envelope attack and re-

This affects the decay time of high frequencies. Normal room reverb decays quicker in the high and low frequency range than in the midrange. Lowering the damping percentage will cause high frequencies to decay quicker. Damping percentage values above 100% will cause high frequencies to decay longer than the midrange.

occur.

The amount of damping applied to the low frequencies. At 100%, no damping occurs. Values lower than 100% increase the amount of damping, reducing low frequencies over time. Values above 100% have the opposite effect.

lease controls have on the reverb itself. Lower numbers have a more subtle effect while higher numbers sound more drastic.

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Parameter Description

Envelope Attack 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 in milliseconds it takes for the reverb to reach full volume after a signal peak. This is similar to a predelay but the reverb is ramping up instead of starting all at once.

Envelope Release The release determines how long after a signal peak the reverb

can be heard before being cut off, similar to a gate’s release time.

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

cessed) signal. When using RoomWorks inserted in an FX channel, you will most likely want to set this to 100% or use the Send button.

Send This button defeats the mix parameter, setting the effect to 100%

wet or affected signal. This button should normally be pressed when RoomWorks is being used as a send effect inserted on an FX or group channel.

Rotate When active, the perspective of the room is shifted 90°.

Pos The position control is only available for surround configurations.

With this parameter you can control where the virtual listening 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.

Bal (Balance, only available for surround channels)

Balance controls the relative levels between the forward and rear speakers. Positive values favor the front speakers and negative values favor the rear speakers. Note that when the Rotate option is activated, these relationships will shift 90°.

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Surround plug-ins (Cubase SX only)

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

Mix6To2 (Cubase SX only)

The Mix6To2 effect allows you to control the levels of up to six surround channels, and to mix these down to a stereo output. The pop-up menu contains a number of speaker arrangement presets that correspond to some default surround formats. The Mix6To2 lets you quickly mix down your surround mix format to stereo, and to include parts of the surround channels in the resulting mix.

• Note that Mix6To2 does not simulate a surround mix or add any psycho-acoustical artifacts to the resulting output – it is simply a mixer. Also note that the Mix6To 2 should be placed in one of the post fader insert effect slots for the output bus.

Each of the surround channels has the following parameters:

• Two volume faders that govern the levels of the surround bus to the left and right side of the (master) bus.

• A Link button that links the two volume faders.

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

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The Master bus has the following parameters:

• A Link button that links the two Master faders.

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

SurroundDither (Cubase SX only)

SurroundDither is not an “effect” as such. Dithering is a method for controlling the noise produced by quantization errors in digital recordings. The theory behind this is that during low level passages, only a few bits are used to represent the signal, which leads to quantization errors and hence distortion. For example, when “truncating bits”, as a result of moving from 24- to 16-bit resolution, quantization errors are added to an otherwise immaculate recording. By adding a special kind of noise at an extremely low level, the effect of these errors is minimized. The added noise could be perceived as a very low-level hiss under exacting listening conditions. However, this is hardly noticeable and much preferred to the distortion that otherwise occurs.

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When should I use SurroundDither?

• Basically anytime you mix down to a lower resolution, either in real-time (playback) or with the Export Audio Mixdown function, you should consider dithering.

• Since SurroundDither is capable of dithering up to six channels at the same time, it is recommended if you’re using surround channels. If not, you may want to use the UV22 HR instead, see page 60.

The following options can be set in the SurroundDither control panel:

Dithering Type

There are no hard and fast rules for the following options, it all depends on the type of material you are processing. We recommend that you experiment and let your ears be the final judge:

Option Description

Off No dithering is applied.

Type 1 Try this first, it is the most “all-round” type.

Type 2 This method emphasizes higher frequencies more than Type 1.

Noise Shaping Options (Off, Type 1 - 3)

This parameter alters the character of the noise added when dithering. Again, there are no fixed general rules, but you may notice that the higher the number selected here, the more the noise is moved out of the ear’s most sensitive range, the mid-range.

Ditherbits

This is used to specify the intended bit resolution for the final result.

• The section has six buttons, one for each channel.

• Above each button there are six corresponding value fields that display

the bit resolution the files will be converted to.

Clicking a button several times cycles through the available bit resolution values.

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An Example

Say you have set up a project to record 24-bit files. After completion, you want to create a digital 16-bit master for CD burning. Proceed as follows:

1. Add SurroundDither to a post fader insert effect slot for the output bus.

I.e. in one of the last two slots.

2. Open the control panel for SurroundDither, and select the Dithering and Noise Shaping Type.

3. Set the Ditherbit destination to “16” for all the master mix outputs currently used, as defined in the VST Connections dialog.

If you are not using Surround channels, this will be Channel 1 and 2.

4. When you now play back the Project, the digital outputs of your audio hardware will output the mix with 16-bit resolution, with dithering applied.

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SurroundPan (Cubase SX only)

The SurroundPan plug-in provides a graphical overview representing the speaker arrangement and the sound source, allowing you to dynamically position the audio in the surround field.

This plug-in is described in detail in the Operation Manual chapter “Surround Sound”.

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Tools

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

SMPTE Generator (Cubase SX only)

This plug-in is not an effect device. It sends out SMPTE time code to an audio output, allowing you to synchronize other equipment to Cubase SX (provided that the equipment can sync directly to SMPTE time code). This can be very useful if you don’t have access to a MIDIto-time code converter.

The following items and parameters are available:

• Generate Button

Activate this to make the device generate SMPTE time code.

• Link Button

This synchronizes the time code output to the Transport time positions. When Link is activated, the time code output will exactly match the play position in Cubase SX. Activating the Generate button makes the device send the SMPTE time code in “free run” mode, meaning that it will output continuous time code, independently from the transport status in Cubase SX. If you wish to “stripe” a tape with SMPTE, you should use this mode.

• Start Time

This sets the time at which the SMPTE Generator starts, when activated in “free run” mode (Link button off). To change the Start time, click on a digit and move the mouse up or down.

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• Current Time

When Link is on this shows the current position in Cubase SX. If Link is off it shows the current time of the SMPTE Generator in “free run” mode. This cannot be set manually.

• Framerate

This defaults to the frame rate set in the Project Setup dialog. If you wish to generate time code in another frame rate than the Project is currently set to (for example to stripe a tape), you can select another format on the Framerate pop-up (provided that “Link” is off). Note, however, that for the other device to synchronize correctly with Cubase SX, the framerate has to be the same in the Project Setup dialog, the SMPTE Generator and in the receiving device.

Example - Synchronizing a device to Cubase SX

Proceed as follows:

1. Connect the SMPTE Generator as an insert effect on an audio channel, and route the output of that channel to a separate output.

Make sure that no other insert or send effects are used on the time code channel. You should also disable EQ, if this is active.

2. Connect the corresponding output on the audio hardware to the time code input on the device you wish to synchronize to Cubase SX.

Make all necessary settings in the other device, so that it is set to synchronize to incoming timecode.

3. Adjust the level of the time code if needed, either in Cubase SX or in the receiving device.

Activate Generate button (make the device send the SMPTE time code 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 button.

The SMPTE Generator will now output time code that matches the position of the Cubase SX Transport panel.

• Press Play on the Cubase SX Transport panel.

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

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The included VST Instruments

A1 Synthesizer

The A1 is a dual oscillator software synthesizer with the following main features:

• The A1 is polyphonic with up to 16 voices.

• Multimode filter.

Lowpass, bandpass, highpass and notch filter types are available.

• PWM (Pulse Width Modulation).

• FM (Frequency Modulation).

• Ring Modulator.

• Built-in stereo chorus/flanger effect.

• The A1 receives MIDI in Omni mode (on all MIDI channels).

You don’t need to select a MIDI channel to direct MIDI to the A1.

• The A1 responds to MIDI Controller messages.

See page 89.

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A1 Parameters

Oscillator 1 and 2 section

This section contains parameters affecting the oscillators.

Parameter Description

Octave Clicking on the outer ring of the dial allows you to tune the oscillator in

octave steps.

Semitone Clicking on the inner ring of the dial allows you to tune the oscillator in

semitone steps.

Detune Tunes the oscillator in cent (100th of a semitone) steps.

Shape This sets the waveform for the oscillator (sine, triangle, sawtooth or

pulse).

PW Sets the width of the waveform when a Pulse waveform is selected.

Turning the dial clockwise gradually produces a narrower pulse waveform. Note that a PW setting of 100% will lead to complete cancellation of the waveform (i.e. silence), if no modulation (see PW Mod) is applied.

PW Mod This parameter determines the amount of Pulse Width Modulation

(PWM) by the LFO. Positive and negative values can be set. A Pulse waveform must be selected for PW Mod to function.

Pitch Mod This parameter determines the amount of oscillator 1 pitch modulation

(or vibrato) by the LFO. Positive and negative values can be set.

FM (Osc 1 only)

FM Env (Osc 1 only)

Governs the amount of frequency modulation. See page 89.

This governs how much the Filter Envelope parameters affects the FM amount. Positive and negative values can be set. See page 89.

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LFO section

This section contains the LFO (Low Frequency Oscillator) parameters. LFOs are used to modulate parameters like pitch (vibrato) or the filter cutoff.

Parameter Description

LFO Wave This sets the LFO waveform for modulating parameters:

Sine and triangle waves have a smooth waveform, suitable for normal vibrato. Saw produces a ramp up or down cycle. S&H produces stepped random modulation. Square waves produce cycles that abruptly change between two values. Random produces smooth random modulation.

LFO Sync If this is activated, the LFO rate will be synchronized to the se-

quencer tempo in various bar/beat divisions that can be set with the LFO Speed parameter.

LFO Speed Governs the modulation rate of the LFO.

LFO Speed (tempo sync on)

If the “LFO Sync” parameter is activated, the LFO rate will be synchronized to the sequencer tempo, according to the different beat divisions that can be specified here.

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Filter section

This section contains the filter parameters:

Parameter Description

Filter type Sets the filter type to either lowpass, highpass, bandpass or notch.

Cutoff Controls the filter frequency or “cutoff”. If a lowpass filter is used, it

Resonance The Resonance control for the filter. Raise this for a more pronounced

Drive This parameter can overdrive the filter to produce distortion effects.

Filter Envelope Controls how much the filter cutoff should be affected by the Filter

Filter Velocity Determines how the filter cutoff will be affected by velocity, i.e. how hard

The filter types are described on page 88.

could be said to control the opening and closing of the filter, producing the classic “sweeping” synthesizer sound. How this parameter operates is governed by the filter mode (see page 88).

filter sweep effect.

Envelope. Negative values will invert the filter envelope settings.

or soft you strike a key. Positive values will increase the cutoff frequency the harder you strike a key. Negative values will invert this relationship.

Filter Envelope Attack/Decay/ Sustain/ Release

Cutoff Mod This controls how much the filter cutoff is modulated by the LFO (low

Keytrack If this parameter is set to values over 0, the filter cutoff frequency will

The Filter Envelope Attack, Decay, Sustain and Release parameters. Use these parameters to determine how the filter cutoff should open and close with time, when a note is played. Values can be changed using the dials or by dragging the breakpoints in the graphic display.

frequency oscillator).

increase the further up on the keyboard you play. Negative values invert this relationship.

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Amplifier section

This section contains the Amplifier parameters:

Parameter Description

Amplifier Attack/ Decay/Sustain/ Release

Velocity This determines how much the Amplifier Envelope should be af-

Mono When this is activated, the A1 will be monophonic, i.e. only play

The Amplifier Attack, Decay, Sustain and Release parameters. Use these parameters to determine how the volume should change with time, when a note is played. Values can be changed using the dials or by dragging the breakpoints in the graphic display.

fected by velocity, i.e. by how hard or soft you strike a note on the keyboard.

one voice at a time.

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The Chorus/Flanger section

Adding chorus will introduce a wide stereo effect and generally “fatten” sounds. With higher Feedback settings, more metallic sounding flanging effects are produced. The section contains the following parameters.

Parameter Description

Speed Controls the modulation rate of the effect.

Feedback Increasing the Feedback parameter value results in a more pro-

nounced sweeping metallic sound. Positive and negative feedback values can be set.

Depth Controls the depth of the modulation.

Quad Adds more delay taps, producing richer chorus/flanger effects.

On This turns the chorus/flanger effect on or off.

The Glide section

This section contains the glide parameters.

Parameter Description

On If set to “On” the pitch will glide up or down between notes played.

Speed Controls the time it takes for the pitch to glide from one note to the

next when using Glide.

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The Mixer section

This section controls the relative levels of Oscillator 1 and 2. Here you also set the levels of the Ring Modulator and Noise Generator outputs.

Parameter Description

Osc 1 Sets the volume of oscillator 1.

Ring Mod Controls the level of the ring modulator. See page 89.

Osc 2 Sets the volume of oscillator 2.

Noise Noise is commonly used to create wind and percussion type sounds.

To hear the noise generator output on its own, turn down the osc 1 and 2 output in the Mixer.

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Mod Wheel section

This section controls how the modulation wheel affects certain parameters. Positive and negative values can be set. For example, this can be used to set up so that moving the mod wheel gradually removes Filter Cutoff LFO modulation and instead introduces vibrato.

Parameter Description

Pitch Mod Governs the amount of LFO modulation of the oscillator frequency

(vibrato) using the mod wheel.

Cutoff Mod Governs the amount of LFO modulation of the Filter Cutoff para-

meter using the mod wheel.

Cutoff Governs how much the mod wheel affects the Filter Cutoff fre-

quency. Positive values raise the cutoff frequency when moving the mod wheel forward. Negative values invert this relationship.

The Parameter display

The Parameter display is located in the middle of the A1 panel. The Parameter display shows information about any A1 parameter control when you point at it with the mouse. The information is presented in the following way (from left to right):

• “Section” shows what A1 section the parameter belongs to.

• “Parameter” shows the name of the parameter.

• “Value” shows the current value of the parameter.

• “Ctrl” shows the MIDI Controller assigned to the parameter, see page 89.

The included VST Instruments 2 – 87

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Setting the number of voices

A1 can have up to 16 voices, but you can freely set the number of voices for each program by changing the value in the “Voices” field.

Keyboard section

The keyboard shows incoming MIDI note data as played by “invisible hands”. The keyboard can be “played” by clicking on it with mouse. Note that the velocity produced will be fixed and that you cannot record anything by clicking the keyboard.

• “Bend Range” is the only parameter that can be set in this section.

A value of “1” equals a semitone bend range, “2” equals a range of two semitones etc.

About the Filter types

The A1 features a multimode filter. The various filter modes are selected with the Filter Type buttons, and are as follows:

• Lowpass

Lowpass filters let low frequencies pass and cuts out the high frequencies. This is the most commonly used filter type in analog synthesizers.

• Bandpass

A bandpass filter cuts frequencies above and below the cutoff frequency, allowing a specific range of frequencies to pass while attenuating all others.

• Highpass

A highpass filter is the opposite of a lowpass filter, cutting out the lower frequencies and letting the high frequencies pass.

• Notch

A notch filter cuts out frequencies in a narrow midrange band, letting the frequencies below and above through.

Filter Slope

You can also select between 12 or 24 dB filter slopes for all filter types. A 12 dB Lowpass filter leaves more of the harmonics in the filtered sound compared to a 24 dB Lowpass filter.

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Ring Modulator

Ring modulators basically multiply two audio signals together. In the A1, Oscillator 1 is multiplied with Oscillator 2 to produce sum and difference frequencies. Ring modulation can be used to create complex, bell-like sounds.

• To hear the output of Ring Modulator on its own, turn down the osc 1 and 2 output in the Mixer.

• If the oscillators are tuned to the same frequency, and no modulation is applied to either the oscillator 1 or 2 frequency, the ring modulated output will sound fairly similar to the “normal” sound of the oscillators. It is when the frequencies of osc 1 and osc 2 differ, that you get the more complex timbres associated with ring modulation.

About FM

Frequency Modulation, or FM, is when the frequency of one oscillator (called the “carrier”) is modulated by the frequency of another oscillator (called the “modulator”). Using FM can produce a wide range of harmonic and non-harmonic timbres.

• In the A1, Oscillator 1 is the carrier and Oscillator 2 the modulator.

When using FM, you should turn the master volume for Oscillator 2 down to zero in the Mixer to hear the “pure” sound of FM. The output of oscillator 2 is internally routed to oscillator 1 anyway when using FM.

• Changing the frequency of Oscillator 2 also changes the timbre of the FM sound.

The waveform selected for both oscillators also affects the timbre.

MIDI Controller Messages

The A1 responds to MIDI Controller Messages. All A1 parameters are assigned controller numbers. To find out what controller number is assigned to a parameter, simply point at the parameter and you can see the associated controller number assigned to it in the Parameter display (see page 87).

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VB-1 Bass Synth

The VB-1 is a virtual bass instrument built on real-time physical modelling principles. It has the following properties:

• VB-1 is polyphonic with up to 4 voices.

• VB-1 receives MIDI In Omni mode (on all MIDI channels).

You don’t need to select a MIDI channel to direct MIDI to the VB-1.

• VB-1 responds to the following MIDI messages:

MIDI Note On/Off (velocity governs volume), Volume and Pan.

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VB-1 Parameters

Parameter Description

Pick-up To change the Pick-up position, click and drag the lower end of the Pick-

up. Positioning the pick-up position towards the left produces a hollow sound that emphasizes the upper harmonics of the plucked string. When placed towards the right position, the tone is fuller and warmer.

Pick This determines where along the length of the string the initial pluck is

made. This controls the “roundness” of the tone, just like on a real bass. Click-drag the Pick to change position.

Shape This knob selects the basic waveform used to drive the plucked string

model. This parameter can drastically change the sound character. The control smoothly morphs through the waves. It is possible to create sounds that have no relation to a bass guitar with this control.

Volume This knob regulates the VB-1 volume.

Damper This parameter controls the length of time the string vibrates after being

plucked.

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LM-7 Drum Machine

(

Volume and Tune faders (for each drum sound).

Pad (one for each drum sound). Press to audition the drum sound assigned to the Pad, or to select a sound for adjusting pan.

This adjusts the Pan for the individual drums. The setting is applied to the currently selected drum, indicated by a lit yellow LED over the Pad button.

the position in the stereo image)

This sets the global velocity sensitivity for LM-7.

Master Volume

The LM-7 is a 24-bit drum machine. It has the following properties:

• LM-7 is polyphonic with up to 12 voices.

• LM-7 receives MIDI in Omni mode (on all MIDI channels).

You don’t need to select a MIDI channel to direct MIDI to LM-7.

• LM-7 responds to the following MIDI messages:

MIDI Note On/Off (velocity governs volume).

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LM-7 Parameters

Parameter Description

Velocity This sets the global velocity sensitivity for LM-7. The higher the value,

the more sensitive LM-7 will be to incoming velocity data. If set to “0”, the sounds will play back with a fixed velocity value.

Volume sliders The volume sliders are used to adjust the volume for each individual

drum sound.

Tune sliders The tune sliders are used to tune each individual drum sound, up or

down 1 octave.

Pad The Pads are used for two things: To audition the individual drum

sounds, and to select a sound for adjusting pan.

Panorama This is used to position an individual sound in the stereo image. The

setting applies to the currently selected sound, indicated by a lit yellow LED over the Pad button.

Drum sounds

LM-7 comes with six sets of drum sounds. “Compressor”, “909” and “Percussion” are loaded as the default sets when launching LM-7. “Modulation”, “Fusion” and “DrumNbass” can be loaded by selecting “Load Bank” from the File menu and opening the lm7_second_set.fxb file (which is located in the Vstplugins/Drums subfolder).

• You switch between the three loaded sets by using the pop-up menu (just like you switch between effect programs).

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MIDI note mapping

The table below shows how the drum sounds are assigned to note values on your MIDI keyboard. The mapping is GM compatible:

Drum sound Note Comment

Bd C1

Rim C#1 Compressor only.

Snare D1

Clap D#1 909 only.

Hi-Hat F#1

O-Hi-Hat A#1

Tom 1 A1

Tom 2 B2

Tom 3 D2

Crash C#2

Ride D#2 Compressor only.

Tambourine F#2 Percussion only.

Cowbell G#2 Percussion only.

Hi Bongo C3 Percussion only.

Lo Bongo C3# Percussion only.

Conga Mute D3 Percussion only.

Conga Open D#3 Percussion only.

Conga Lo E3 Percussion only.

Timbale Lo G3 Percussion only.

Timbale Hi G#3 Percussion only.

Cabasa A3 Percussion only.

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Embracer – Surround Pad Synthesizer (Cubase SX only)

Embracer is a simple but powerful polyphonic synthesizer designed entirely for producing pads and accompaniment sounds. With its easy-touse 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 output. 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 will be placed in a mix.

If you’ve never worked with a surround system before, now is the time to start exploring these possibilities.

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.

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• Up to 32 voices of polyphony per instance.

• Dynamic width control for exciting 3D sounds.

• Unique “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 waveforms are:

Carpet, DigiPad, Choir, Ensemble, Metal Phaze, Phase Strings, Sing Sing, Soft Wave, Spit Strynx, Stepfloor, 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 will be used.

Tone Embracer offers a high pass and low pass filter for each oscillator.

Both filters are controlled via a single Tone knob. In the 50% center position, the signal will not be filtered. Reducing the tone value adds low pass filtering. Values above 50% add high pass filtering. This parameter can also be controlled by the “eye” controller.

Width Controls the spatial spread of the signal. A value of 0% puts the sig-

nal mono into the center position. In stereo mode, a value of 100% results in a maximum stereo width. In surround mode, a value of 100% creates a full 360° surround image. The width parameter can be controlled by a variety of modulation sources, as well as by the “eye” controller.

Coarse (Oscillator 2 only)

Fine (Oscillator only)

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 parameter to a negative value of the same amount to keep the instrument in tune.

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Envelope and Level

Parameter Description

Attack Controls the attack time of each oscillator. Higher values create

slower attacks.

Attack Vel Sets the amount of velocity control of the attack time. Higher values

increase the velocity sensitivity.

Level Controls the oscillator output level.

Level Vel Sets the amount of velocity control of the oscillator level. Higher

values increase the velocity sensitivity.

Master

Parameter Description

Release Controls the overall release time of the volume envelope. Higher

values result in longer release times.

Mode Sets the output mode of Embracer. You can choose between “Ste-

reo” and “Surround”. In Stereo Mode, Embracer has one stereo output in the VST Mixer. In Surround Mode, Embracer has either a quadraphonic 4-channel output or two independent stereo outputs in the VST Mixer. See below for more details on using Embracer in a surround mixer setup.

Width Ctr Use this parameter to select a modulation source for the width

parameter. Available sources are: Mod Wheel, Aftertouch, Velocity and Envelope. Both oscillators are controlled simultaneously. However, modulation depth is controlled independently by the respective width parameter of each oscillator.

Max Poly Sets the total number of voices available. Each oscillator uses one

voice per note played. Thus, a two-oscillator sound with 8 voices results in 4-voice polyphony. The default value for Max Poly is 16.

Fine Tune Use this to adjust the pitch of the whole instrument. Range is +/- 50

Cents. Use Fine Tune in combination with the Fine Tune parameter of OSC 2 to create smooth detune effects.

Master Out Sets the overall output volume of the instrument.

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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 parameters 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 respective oscillator. When you drag a handle, the respective Tone and Width knobs of the oscillator are adjusted accordingly. Play a note while editing to hear the effect.

The “eye” can not 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!

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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.

1. Open an instance of Embracer in the VST instruments rack and set it to surround mode.

2. When you open the VST mixer you will 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 will 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 surround 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.

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Monologue – Monophonic Analog Modeling Synthesizer (Cubase SX only)

Monologue is a monophonic analog synthesizer based on physical modeling technology. It offers full, rich and colorful 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.

• Monologue has 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.

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Steinberg Cubase SX - 3.0, Cubase SL - 3.0 User Manual

Specifications and Main Features

Frequently Asked Questions

User Manual

Table of Contents

The included effect plug-ins

Introduction

Delay plug-ins

DoubleDelay

ModDelay

Distortion plug-ins

DaTube

Overdrive

QuadraFuzz (Cubase SX only)

Dynamics plug-ins

SPL DeEsser (Cubase SX only)

Dynamics

Magneto (Cubase SX only)

MIDI Gate

MultibandCompressor (Cubase SX only)

VST Dynamics

Filter plug-ins

Q (Cubase SX only)

StepFilter

Tonic – Analog Modeling Filter (Cubase SX only)

Modulation plug-ins

Chorus

Flanger

Metalizer

Phaser

Ringmodulator

Rotary

Symphonic

Tranceformer

Other plug-ins

Bitcrusher

Chopper

Apogee UV 22 HR (Cubase SX only)

Vocoder

Restoration

Grungelizer

Reverb plug-ins

Reverb A

Reverb B

RoomWorks

Surround plug-ins (Cubase SX only)

Mix6To2 (Cubase SX only)

SurroundDither (Cubase SX only)

SurroundPan (Cubase SX only)

Tools

SMPTE Generator (Cubase SX only)

The included VST Instruments

A1 Synthesizer

VB-1 Bass Synth

LM-7 Drum Machine

Embracer – Surround Pad Synthesizer (Cubase SX only)

Monologue – Monophonic Analog Modeling Synthesizer (Cubase SX only)