Steinberg Cubase SX - 2.0 User Manual
Audio Effects and VST Instruments
Manual by Ludvig Carlson, Anders Nordmark, Roger Wiklander
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.
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© Steinberg Media Technologies GmbH, 2003.
All rights reserved.
Table of Contents
5 The included effect plug-ins
6 Introduction
7 Delay plug-ins
10 Distortion plug-ins
16 Dynamics plug-ins
37 Filter plug-ins
42 Modulation plug-ins
55 Other plug-ins
67 Reverb plug-ins
69 Surround plug-ins (Cubase SX only)
73 The included VST Instruments
74 A1 Synthesizer
85 VB-1 Bass Synth
87 LM-7 Drum Machine
91 VST plug-ins from previous Cubase versions
92 Introduction
93 Cubase 5 audio effect plug-ins
114 Earlier audio effect plug-ins
123 Cubase 5 VST Instruments
135 Index
CUBASE SX/SL
4 Table of Contents
1
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.
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 buttons are gray)
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.
Tempo Sync 1 The note value multiplier (x1 to x10) for the first delay unit.
The included effect plug-ins 1 – 7
CUBASE SX/SL
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 ModDelay
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 (gray button) 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.
When this is activated, the effect simulates the sound of a speaker
cabinet.
for further “tweaking”.
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
and dragging in the display.
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QuadraFuzz
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
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.
before
distortion.
• The distortion type, based on a selectable distortion characteristic.
after
• The signal volume control
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.
distortion.
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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
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.
after
distortion. Values between +/- 12 dB can be set
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
7 kHz 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 is blue) 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 20 dB 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:
1.
While playing back audio, click the “Listen” button.
You will now monitor the audio signal, and the gate will be bypassed.
2.
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.
3.
After setting the frequency range, click the “On” button.
AutoGate will now use the selected frequency range as the trigger input.
4.
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:
1.
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.
2.
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
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:
1. Select the audio to be affected by the MIDI Gate.
This can be audio material from any audio track, or even a live audio input (provided
you have a low latency audio card).
2. Select the MIDI Gate as an insert effect for the audio track.
The MIDI Gate control panel opens.
3. Select a MIDI track to control the MIDI Gate.
This can be an empty MIDI track, or a MIDI track containing data, it doesn’t matter. However, if you wish to play the MIDI Gate in real-time – as opposed to having a recorded
part playing it – the track has to be selected for the effect to receive the MIDI output.
4. Open the Output (“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.
5. Now play a few notes on your MIDI keyboard.
As you can hear, the audio track material is affected by what you play on your MIDI keyboard.
The following MIDI Gate parameters are available:
Parameter Description
Attack This is used for determining how long it should take for 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.
CUBASE SX/SL
1 – 30 The included effect plug-ins
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