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
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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
43 Modulation plug-ins
57 Other plug-ins
66 Restoration plug-ins
74 Reverb plug-ins
76 Surround plug-ins
88 Tools plug-ins
95 The included VST Instruments
96 A1 Synthesizer
107 VB-1 Bass Synth
109 LM-7 Drum Machine
113 Nuendo 1.x effects and instruments
114 Introduction
115 Audio effect plug-ins
123 VST Instruments
125 Index
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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 Nuendo, 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. Nuendo automatically provides the plug-in with the tempo currently used in the project.
The parameters are as follows:
Parameter Values Description
Mix 0-100% Sets the level balance between the dry signal and
the effect. If DoubleDelay 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 1/1 - 1/32,
1/1 - 1/32 Triplet,
1/1 - 1/32 Dotted
or 0-9999ms
Delay Time 2 As above As above.
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.
This is where you specify the base note value for
the delay if tempo sync is on. If tempo sync is off,
it sets the delay time in milliseconds.
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Parameter Values Description
Feedback 0-100% This sets the number of repeats for both delays.
Tempo Sync 1 x1 to x10 The note value multiplier for the first delay unit.
Tempo Sync 2 x1 to x10 As above, but for the second delay unit.
Pan1 -100 to 100% This sets the stereo position for the first delay.
Pan2 -100 to 100% 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 Values Description
Mix 0-100% 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 0-100% This sets the number of repeats for the delay.
Delay Time 1/1 - 1/32,
1/1 - 1/32 Triplet,
1/1 - 1/32 Dotted
or 0-9999ms
Tempo Sync
knob
DelayMod. 0-100% This controls the pitch modulation rate for the delay
x1 to x10 This is the note value multiplier for the delay when
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.
This is where you specify the base note value for
the delay if tempo sync is on. If tempo sync is off,
it sets the delay time in milliseconds.
tempo sync is used.
effect.
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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 Values Description
Drive 0 - 100% Regulates the pre-gain of the “amplifier”. Use high values
if you want an overdriven sound just on the verge of distortion.
Balance 0 - 100% This controls the balance between the signal processed
by the Drive parameter and the dry input signal. For maximum drive effect, set this to its highest value.
Output -∞ - 0.000 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 Values Description
Input -15 to +15dB Sets the input level.
Output -15 to +15dB Sets the output level. As overdrive generates har-
monics, it increases the level of the processed signal. You can use the Output fader to compensate
for the level increase.
Speaker
simulation
Factory
Styles
Bass -15 to +15dB Tone control for the low frequencies.
Mid -15 to +15dB Tone control for the mid frequencies.
Hi -15 to +15dB Tone control for the high frequencies.
Drive 0-100% Governs the amount of overdrive. You can also ad-
On/Off Simulates the sound of a speaker cabinet.
Warm, Chordy,
Magic OD, Fat Drive,
Woody, Bluesy
Select one of six presets, which can be used as
they are or as a basis for further “tweaking”.
just 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
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 Values Description
S-Reduction 0 - 10 Controls the intensity of the de-essing effect. We recom-
mend 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 On/Off See separate description below.
Male/Female On/Off 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 AutoGate
Limiter
Routing
selector
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.
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
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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 Values Description
Attack 0 - 500 This is used for determining how long it should take for the
Gate to open after receiving a signal that triggers it.
Hold 0 - 3.000 Regulates how long the Gate remains open after a Note On
or Note Off message (see Hold Mode below).
Release 0 - 3.000 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 Note-On/
-100 <Off>+100
-100 <Off>+100
0 - 127 This controls to which extent the velocity values of the MIDI
Note-Off
The value you specify here determines how and to which extent the velocity values of the MIDI notes should affect the
Attack. If this is set to a positive value, the Attack time will increase with higher note velocities. Negative values will give
shorter Attack times with higher velocities. If you do not wish
to use this parameter, set it to the <Off> position.
The value you specify here determines how and to which extent the velocity values of the MIDI notes should affect the
Release. Positive values will increase the Release time and
negative values will decrease the Release time. If you do not
wish to use this parameter, set it to the <Off> position.
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.
Use this switch to set the Hold Mode. In Note-On mode, the
Gate will only remain open for the time set with the Hold and
Release parameters, regardless of the length of the MIDI
note that triggered the Gate. In Note-Off mode on the other
hand, the Gate will remain open for as long as the MIDI note
plays, and then apply the Hold and Release parameters.
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MultibandCompressor
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 sec-
ond 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 44.1 kHz soundfile is
used, the highest frequency will be 22 kHz.
❐
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 Nuendo. The range available is +/- 12 dB. 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 Nuendo never exceeds 0 dB. 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.
NuendoEQ2
The NuendoEQ2 plug-in is identical to the EQ section in the Channel
Settings window. As a plug-in, NuendoEQ2 can be applied in different
areas than the Channel EQ. For example, you could use it as an insert
effect, to EQ the output of another effect plug-in, etc. See the Operation Manual chapter “The mixer” for a description of the EQ parameters.
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Q
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.
• 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.
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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
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.
level.
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.
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• Individual step entries can be freely dragged up or down the vertical
axis, or directly set by clicking in an empty grid box. By click-dragging
left or right consecutive step entries will be set to the pointer position.
Setting filter cutoff values in the grid window.
• The horizontal axis shows the pattern steps 1-16 from left to right, and
the vertical axis determines the (relative) filter cutoff frequency and
resonance setting.
The higher up on the vertical axis a step value is entered, the higher the relative filter
cutoff frequency or filter resonance setting.
• By starting playback and editing the patterns for the cutoff and reso-
nance 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 song, 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
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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.
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 val-
ues 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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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 Values Description
Mix 0-100% Sets the level balance between the dry signal and the ef-
fect. 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 Saw/
Triangle/
Pulse
Frequency 0-5Hz This sets the modulation rate.
Delay 0-5ms This controls the depth of the Chorus effect.
Stages 1-4 This adds one to three more delay taps, producing a
This sets the modulation waveform. Triangle produces
smooth modulation, saw produces ramp shaped modulation and pulse waveform produces stepped modulation.
thicker, multi-layered 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 Values Description
Mix 0-100% Sets the level balance between the dry signal and
the effect. If the Flanger is used as a send effect, this
should be set to maximum as you can control the
dry/effect balance with the send.
Output 0-100% Sets the overall volume.
Tempo sync
on/off
Rate 1/1 - 1/32,
1/1 -1/32 Triplet,
1/1 -1/32 Dotted
or 0 - 5Hz
Tempo Sync
knob
Shape Sync
knob
Feedback 0-100% This determines the character of the flanger effect.
Depth 0-100% This sets the depth of the modulation sweep.
x1 to x10 This is the note value multiplier for the flanger sweep
1-16 This changes the shape of the modulating wave-
The button above the Rate knob is used to switch
tempo sync on or off. The button is yellow when
tempo sync is on, and gray when it is off.
If tempo sync is on, this is where you specify the base
note value for tempo syncing the flanger sweep.
If tempo sync is off, the sweep rate can be set freely
with the Rate knob, without sync to tempo.
when tempo sync is used.
form, altering the character of the flanger sweep.
Higher settings produce a more “metallic” sounding
sweep.
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Parameter Values Description
Delay 0-100ms This parameter affects the frequency range of the
modulation sweep, by adjusting the initial delay time.
Stereo Basis 0-100% This sets the stereo width of the effect. 0% is mono,
50% original stereo, and 100% maximum stereo enhancement.
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 Values Description
Mix 0-100% 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 0-100% Sets the overall volume.
Tempo sync
on/off
Speed 1/1 - 1/32,
1/1 - 1/32 Triplet,
1/1 - 1/32 Dotted
or 0 - 10Hz
On button On/Off Turns filter modulation on and off. When turned off,
Mono button On/Off Determines whether the output of the Metalizer will
Sharpness 0-100% Governs the character of the filter effect. The higher
The button above the Speed knob is used to switch
tempo sync on or off. The button is yellow when
tempo sync is on, and gray when it is off.
If tempo sync is on, this is where you specify the
base note value for tempo-syncing the effect. Note
that there is no note value modifier for this effect.
If tempo sync is off, the modulation speed can be set
freely with the Speed knob, without sync to tempo.
the Metalizer will work as a static filter.
be in stereo or mono.
the value, the narrower the affected frequency area,
producing sharper sound and a more pronounced
effect.
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Parameter Values Description
Tone 0-100% Governs the feedback frequency. The effect of this
will be more noticeable with high Feedback settings.
Feedback 0-100% Sets the amount of feedback. Higher values pro-
duce a more “metallic” sound.
• 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 Values Description
Mix 0-100% 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 0-100% Sets the overall volume.
Tempo sync
on/off
Rate 1/1 - 1/32,
1/1 - 1/32 Triplet,
1/1 - 1/32 Dotted
or 0 - 5Hz
Feedback 1-100% This sets the amount of feedback. A higher value pro-
Tempo Sync
knob
Stereo Basis 0-100% This sets the stereo width of the effect. 0% is mono,
x1 to x10 This is the note value multiplier for the Phaser sweep
The button above the Rate knob is used to switch
tempo sync on or off. The button is yellow when
tempo sync is on, and gray when it is off.
If tempo sync is on, this is where you specify the
base note value for tempo syncing the Phaser
sweep. If tempo sync is off, the sweep rate can be
set freely with the Rate knob, without sync to tempo.
duces a more pronounced effect.
when tempo sync is used.
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:
• 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 Values Description
Speed Stop/Slow/Fast This controls the speed of the Rotary.
MIDI Ctrl Controllers
[1] to [16]
Mode Switched/
Variable
Overdrive 0-100% Applies a soft overdrive or distortion.
Crossover
Freq.
Mic Angle 0-180 degrees Sets the simulated microphone angle. 0 = mono,
200-3000Hz Sets the crossover frequency between the low and
Selects the MIDI continuous controller for the Speed
parameter. See page 53.
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.
high frequency loudspeakers.
180 = one mic on each side.
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Parameter Values Description
Mic Distance 1-36 Inches Sets the simulated microphone distance from the
speaker.
Low Rotor
Amp Mod.
Low Rotor
Mix Level
Hi Rotor
Amp Mod.
Hi Rotor
Freq. Mod.
Phasing -100 to 100 Adjusts the amount of phasing in the sound of the
Hi Slow 0.0-720 rpm Fine adjustment of the high rotor Slow speed.
Hi Rate 0.0-720 rpm/sec Fine adjustment of the high rotor acceleration time.
Hi Fast 0.0-720 rpm Fine adjustment of the high rotor Fast speed.
Lo Slow 0.0-720 rpm Fine adjustment of the low rotor Slow speed.
Lo Rate 0.0-720 rpm/sec Fine adjustment of the low rotor acceleration time.
Lo Fast 0.0-720 rpm Fine adjustment of the low rotor Fast speed.
Output Adjusts the overall output level.
Mix Adjusts the mix between dry and processed signal.
0-100% Adjusts amplitude modulation depth.
0-200% Adjusts overall bass level.
0-100% High rotor amplitude modulation.
0-100% High rotor frequency modulation.
high rotor.
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 Values Description
Mix 0-100% Sets the level balance between the dry signal and
the effect. If Symphonic 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
1/1 - 1/32,
1/1 - 1/32 Triplet,
1/1 - 1/32 Dotted
The button below the Temp sync knob is used to
switch tempo sync on or off. The button is yellow
when tempo sync is on, and gray when it is off.
If tempo sync is on, this is where you specify the
base note value for tempo syncing the auto-panning.
Tempo Sync
knob
Delay 0-100 ms This determines the delay time and thus the char-
Depth 0-100% This controls the depth of the chorus effect.
Rate 0-100% This sets the modulation rate for the chorus effect,
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x1 to x10 This is the note value multiplier, determining the
timing of the auto-panning.
acter of the chorus effect, if activated.
If you only want to use Symphonic as an auto-panner or a stereo enhancer, set this to 0%.
if activated.
Parameter Values Description
Stereo Basis 0-100% When the Auto-panner is activated, this sets the
stereo width of the 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.
Output 0-100% Adjusts the output level of the effect.
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 Values Description
Mix 0-100% Sets the level balance between the dry signal and the
effect.
Output 0-100% Adjusts the output level of the effect.
Tone 1 - 5000Hz Sets the frequency (pitch) of the modulating oscillator.
Tempo sync
on/off
Speed 1/1 - 1/32
(straight, triplet
or dotted) or
0 - 10Hz
On button On/Off Turns modulation of the pitch parameter on or off.
Mono button On/Off Governs whether the output will be stereo or mono.
Depth 0-100% Governs the depth of the pitch modulation.
Waveform
buttons
Sine, Square,
Saw, Reverse
Saw, Triangle
The button above the Speed knob is used to switch
tempo sync on or off. The button is yellow when tempo
sync is on, and gray when it is off.
If tempo sync is on, this is where you specify the base
note value for tempo-syncing the effect. 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.
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 as follows:
Parameter Values Description
Mode I, II,I II, IV Select one of four operating modes for the Bitcrusher. Each
mode will produce a different sounding result. Modes I and III
are nastier and noisier, while modes II and IV are more subtle.
Depth 0-24 Use this to set the desired bit resolution. A setting of 24
gives the highest audio quality, while a setting of 1 will create
mostly noise.
Sample
Divider
1-65 This sets the amount by which the audio samples are deci-
mated. At the highest setting (65), nearly all of the information describing the original audio signal will be eliminated,
turning the signal into unrecognizable noise.
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Parameter Values Description
Mix 0-100% This slider regulates the balance between the output from
the Bitcrusher 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.
Output 0-100% Governs the output level from the Bitcrusher. Drag the slider
upwards to 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 Values Description
Mix 0-100% 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 1/1 - 1/32,
1/1 - 1/32 Triplet,
1/1 - 1/32 Dotted
or 0 - 50Hz
Stereo/Mono
button
Waveform
buttons
Depth 0-100% Sets the depth of the Chopper effect. This can also
Stereo/Mono Determines whether the Chopper will work as an
Sine, Square,
Saw, Reverse
Saw, Triangle
The button above the Speed knob is used to switch
tempo sync on (yellow button) or off (gray button).
If tempo sync is on, this is where you specify the
base note value for tempo-syncing the effect. 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.
auto-panner (button set to “Stereo”) or a tremolo
effect (button set to “Mono”).
Sets the modulation waveform.
be set by clicking in the graphic display.
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Apogee UV 22 HR
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 85).
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 per-
cussive 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 Values Description
Number of
Bands
Bandwidth 0-100% This sets the bandwidth for the frequency bands,
Min./Max. Freq. 40-8000Hz These parameters set the minimum and maximum fre-
log/lin 0-100% Log/Lin controls how the frequency bands are spaced
Env.Speed 10-19699ms/
High Thru 0-100% This lets through high frequencies around the “S” fre-
Talk Thru 0-100% Adjusts the level of the original input signal passed to
Gap Thru 0-100% Gap Thru (only available in MIDI mode) sets the level
Output 0-100% This controls the output level of the Vocoder.
Emphasis 0-100% This is a highpass filter, gradually cutting lower fre-
2-24 This governs how many frequency bands the modula-
tor signal is divided into. Fewer bands will provide a
thinner more resonant sound, whereas using more
bands will make the sound fuller and more intelligible.
which affects the overall timbre. Very narrow bandwidth settings will produce a thin, whistle-like sound.
quency limits for the Vocoder, respectively.
between the minimum and maximum frequencies. Log
= equal spacing in octaves, Lin = equal spacing in Hz.
This affects the basic timbre of the Vocoder.
This determines the attack and release times of the
HOLD
Vocoder envelope. 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.
quency from the original input signal while notes are
played.
the Vocoder output while notes are played.
of the original input 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.
quencies while letting 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 Values Description
Voices 1-8 This sets the number of voices for the synth.
Fine Tune -100 to +100
Cents
Pitch Bend 1-12 Semitones Sets the up/down range of the Pitch Bend in semi-
Noise 0-100% Adds white noise to the sound.
NoiseMod 0-100% This makes the oscillators modulate the noise level.
P.Drift 0-100% Adds random pitch variation to the oscillators.
P.Glide 0-100% This makes the pitch glide between notes played.
P.Bright 0-100% This is a lowpass filter that can be used to soften the
P.Detune 12.00 to 0.00 Allows you to detune one of the oscillators in cent
LFO Rate 1-23Hz Controls the LFO rate (for vibrato).
Vibrato 0-100% Adds vibrato to the oscillators. This can also be con-
Tunes the oscillators in cent (100th of a semitone)
steps.
tone steps.
This gives the noise a rasping sound, turning “sss”
into “zzz”.
The parameter controls the time it takes for the pitch
to glide from one note to the next.
tone of the oscillators. It does not affect the white
noise generator.
steps.
trolled by using the Mod Wheel.
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Restoration plug-ins
This section contains descriptions of the plug-ins in the “Restoration”
category.
DeClicker
The DeClicker plug-in is specifically designed to eliminate single
“clicks” or “pops” in a recording. One typical application is to clean up
recordings made from vinyl records, but you may also find it useful for
removing pops from microphone switches, oxidized connector noises,
clicks from sync problems when transferring material digitally, etc.
• Note that the DeClicker module is not optimized for crackles (a series of
short clicks).
However, as it is often hard to distinguish between clicks and crackles, you might also
be able to use it to improve your recording in this respect.
• If the recording also contains background noise (hiss), you may want to
combine DeClicker with the DeNoiser plug-in.
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How DeClicker works
The Declicker process is divided in two tasks:
• Analysis – when the audio signal passes through DeClicker, the selected analysis algorithm finds the clicks in the recording. You provide input to the analysis
parameters by selecting a Mode and the Threshold and DePlop parameters.
• Removal – a de-click algorithm is applied to the audio, removing the clicks.
In many cases, the original audio material “hidden” underneath a click can not be restored. This means there will be a gap once the click has been removed. DeClicker has
the ability to automatically “redraw” the hence missing parts of the waveform. This feature can also be used to remove tape dropouts with a length of up to 60 samples (just
above one millisecond at 44.1kHz).
The whole Declicking process can be visually monitored in the Input
and Output displays of the DeClicker window (showing the incoming
audio and the processed – DeClicked – audio, respectively). This
helps you adjust the parameters. Furthermore, if you activate the Audition button, only the removed material will be heard (and shown in the
Output display).
❐
Make sure that no low-pass filter has been applied to your audio material
before you edit it with DeClicker. This may affect the detection of clicks.
Parameters
Parameter Description
Audition When this is activated, only the removed material will be heard. The Out-
put display will also show the waveform image of the removed material in
this mode.
Classic When this is activated, the DeClicker attempts to remove both audible
clicks and crackle noise. When it’s deactivated, only single clicks will be
removed while crackles (rapidly repeated clicks) are ignored. Which
mode to choose depends on the source material. Note also that Classic
mode requires less CPU power.
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Parameter Description
Threshold This setting determines the amplitude (level) required for a click to be de-
tected. In many cases, DeClicker’s sensitive algorithms identify a lot more
clicks than you can actually hear. To avoid wasting processing power to
remove inaudible clicks, raise this parameter to a high value, and then
lower it until all the artefacts that you actually want removed are detected.
The lower the setting, the more clicks will be detected but also the higher
the risk of audible artefacts. If in doubt, activate Audition mode and check
that the removed material doesn’t contain any actual musical or rhythmical information, etc.
DePlop This setting controls a special highpass filter which works on signals be-
low 150 Hz. It cuts away the “plop noise” which sometimes appears after
eliminating a click. The slider adjusts the filter frequency (off - 150Hz).
Note: this function is best applied to older recordings, which often use a
narrow frequency range. Be careful when applying this function to modern recordings, as you may risk removing parts of the useful signal!
Quality This determines the quality of the click removal and audio restoration,
with “4” being the best quality setting. Please note that selecting higher
quality settings also means that more processing power is consumed.
Also, note that in some situations it might be more productive to use a
lower Quality value. One example of this is when two clicks follow each
other in quick succession or when you tackle a click in a low level part
that is followed by a loud part.
Mode Which Mode to select depends on the source material. Standard mode
is suitable for a wide variety of source material – try this option first. Vintage mode is suitable for restoring “antique” recordings (with limited high
frequency content), while Modern mode is best suited for contemporary
recordings with a wide frequency range (putting greater emphasis on
distinguishing clicks from other strong impulses in the audio material).
Tips and Tricks
• By combining Vintage Mode and extreme Threshold and DePlop settings, you
can create an interesting effect which “softens” material with particularly
sharp attacks, e.g. percussion or brass.
• If you have material with digital distortion (clipping), try applying DeClicker.
While it can’t do miracles, it can at least make some improvement to the overall “hardness” introduced by the distortion.
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DeNoiser
The DeNoiser plug-in lets you suppress noise without affecting the
general sound quality. Or, in tech talk, the DeNoiser removes broad
band noise from arbitrary audio material without leaving any “spectral
finger print”. The algorithm that this plug-in is based on has the ability
to track and adjust itself to variations in background noise. This means
the noise can be diminished without side effects, preserving the spatial
impression, and without letting the result become “colorless”. Many
years of research were invested in developing the methods used.
Typical applications for the DeNoiser include cleaning or remastering
recordings from old tape or vinyl, or noisy live recordings.
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How DeNoiser works
DeNoiser is based on spectral subtraction. Each section of the frequency spectrum, that has an amplitude below the estimated noise
floor, is reduced in intensity by use of a spectral expander. The result
is a noise reduction that does not affect the phase of the signal.
The figure below shows the signal flow:
Noise Reduction
Level
Noise
Floor
Input Output
The solid line represents the actual audio signal, while the dotted lines represent
control signals.
Ambient
Analysis
Transient
Analysis
Noise
Reduction
Ambience
The signal is continuously analyzed by the first module in the chain, to
estimate the noise floor at any given time. This is sufficient when the
noise level is constant or modulates slowly. When the noise level varies rapidly, the Ambience- and Transient-analysis help adjust the response of the noise reduction unit, allowing transient-rich material to
maintain its liveliness and natural ambience.
• When you process audio in DeNoiser, the plug-in will need a short time
(less than a second) to analyze the material and set its internal parameters.
Since you would not want to include this short “startup sequence” in the final result,
you should make it a habit to first play back a short section of the audio, thereby letting
DeNoiser “learn” the noisefloor, and then stop and start over again from the beginning.
The plug-in then remembers the settings internally.
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The Noisefloor Display
The display to the left in the DeNoiser window is crucial when making
settings. It contains the following three elements:
• The dark green spectral graph.
This shows the spectrum of the audio currently being played back. The horizontal axis
shows the frequency (linear scale). The low frequencies are visible on the left side, the
high ones on the right side. The vertical axis shows the signal amplitudes, thus the level
(displayed as a logarithmic dB scale).
• The yellow line.
This is a spectral estimation of the noise floor. The average of this value is shown numerically below the display.
• The light green line.
This is simply a graphic representation of the Offset parameter.
The light green Offset line should be adjusted so that it appears as
close above the yellow noise floor graph as possible. The dark green
spectrum plot is there to help you fine-tune the Offset setting, so that
only the noise is removed, not parts of the signal (ideally, the light
green line should be between the yellow line and the spectrum plot).
Parameters
Parameter Description
Freeze If you activate this button, you “freeze” the noise floor detection pro-
cess. The yellow noise floor graph in the display will hold its current
value (as will the numeric noise floor value display below) until you deactivate Freeze. This allows you to take a closer look at the readings.
Reduction Governs the amount of noise reduction. The display above this fader
shows the amount of dB by which the noise level is being reduced. The
final result also depends on the Ambience parameter, and on the automatic Ambience and Transient analysis of the original material, as described above.
Ambience This parameter is used to specify a balance between the noise sup-
pression and the amount of natural ambience, which is essential for a
natural result. With a low Ambience setting, the sound can become
somewhat lifeless and sterile. A high setting, on the other hand, preserves more of the ambient character of the sound, but the noise suppression is less effective.
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Parameter Description
Offset This parameter serves as a threshold, governing the overall level at
which the noise reduction is performed. For optimal noise reduction
with a minimum of sound coloration, this parameter should be set to a
value slightly above the noise floor level. To help you do this, the offset
value is shown as a light green line in the noisefloor display, while the
noise floor is shown as a yellow line.
A/B/Store These buttons are described below this table.
Classic When this is activated, a less CPU-intensive version of the DeNoiser al-
gorithm is used. Use Classic mode if you are short on processing
power. However, for optimum noise suppression, we recommend that
you deactivate Classic mode.
Using the A/B setups
With the A/B buttons you can make instantaneous switches between
two different DeNoiser setups, allowing you to quickly try out and compare different configurations. You can also use this feature for separate
settings for two different sections of an audio recording. Proceed as
follows:
1. Make the settings you want for setup A.
2. Click on [Store] and then on the [A] button.
3. Make the settings you want for setup B.
4. Click on [Store] and then on the [B] button.
Now the two setups are stored, and you can switch between them simply by clicking
[A] or [B].
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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 Values Description
Crackle N/A This adds crackle to create that old vinyl record sound.
The farther to the right you turn the dial, the more crackle
is added.
RPM switch 33-45-78 When emulating the sound of a vinyl record, this switch
lets you set the RPM (revolutions per minute) speed of
the record.
Noise N/A This dial regulates the amount of static noise added.
Distort N/A Use this dial to add distortion.
EQ N/A Turn this dial to the right to cut off the low frequencies,
and create a more hollow, lo-fi sound.
AC N/A This emulates a constant, low hum of AC current.
Frequency
switch
Timeline Today - 1900 This dial regulates the amount of overall effect. The far-
50-60 Hz This sets the frequency of the AC current, and thus the
pitch of the AC hum.
ther to the right (1900) you turn this dial, the more noticeable the effect.
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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 Values Description
Mix Dry/Wet 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 20-100 This setting determines the “size” of the simulated
room environment.
Predelay 0-100ms 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
natural-sounding.
Reverb Time 0,2s - forever This parameter sets the length of the reverb time.
Filter HighCut -15 to 0dB This filters out high frequencies for the reverb, which
can make the reverb sound softer.
Filter LowCut -15 to 0dB 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 Values Description
Mix 0-100% Sets the level balance between the dry signal and the ef-
fect. 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 0-100% Governs the “size” of the simulated room environment.
Predelay 0-100% 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 natural-sounding.
Reverb Time 0-100% This parameter sets the length of the reverb effect.
Damp 0-100% This parameter “dampens” the higher frequencies, produc-
ing a rounder and smoother sounding reverb.
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Surround plug-ins
This section contains descriptions of the plug-ins in the “Surround”
category.
Matrix Encoder
The Matrix Encoder is intended for Pro Logic compatible encoding of
multichannel files. This is a process where a 4 channel Surround mix is
“packed” into two channels for broadcasting or distribution on video
tape, for example. The Matrix Encoder takes four separate inputs; left,
right, center, and surround (LRCS), and creates two final outputs, lefttotal and right-total (Lt and Rt).
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Matrix Decoder
The Matrix Decoder reverses the process performed by the Matrix Encoder (see above). It is used for monitoring how an encoded mix will
sound when played back on a Pro Logic compatible system. When an
encoded mix is played back via the decoder, the Lt/Rt channels are
again converted to four outputs (LRCS).
❐
This manual does not attempt to explain the full background on how Pro
Logic works, but focuses on how you can use the Matrix Encoder/Decoder to produce a mix that is compatible with this standard.
Setting up
• Create an output bus with the “LRCS” speaker arrangement, in the
VST Connections window, and route it to the physical outputs on your
audio hardware.
This is if you want to make a four-channel surround mix. If you want to make a five-channel
mix, see “Using the Matrix Encoder/Decoder with the 5.0 Surround format” on page 80.
• The Encoder should be placed in the first “post fader” insert slot (#7)
for the output bus, followed by the Decoder.
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Using the Matrix Encoder/Decoder
1. Set up the mix roughly the way you want it.
Use the Surround Panner to place channels in the Surround mix, or assign channels to
the individual LRCS outputs.
2. Activate the Matrix Encoder.
What you now hear is the encoded stereo mix, the way it will sound when played back
on a normal stereo reproducer. If you open the Matrix Encoder control panel you can
adjust the Gain of the Lt/Rt output by using the fader.
3. Activate the Matrix Decoder, open the control panel and click on the
Steering “On” button.
Now you can hear how the mix will be reproduced in surround on a Pro Logic compatible system.
• The “Steering” display shows a ball within the LRCS axis. The position
of this ball indicates the dominant direction of the mix, sometimes referred to as the “dominance vector”. Part of the processing that is applied, for various technical reasons, results in the dominant channel
being enhanced and the non-dominant channels being reduced in gain.
4. By switching the Matrix Decoder “Bypass” button on and off, you can
compare the decoded mix with the encoded stereo mix, and make adjustments in the Mixer as necessary.
The main goal is to produce a mix that sounds good in both the encoded and the decoded version. If you wish to compare the encoded or decoded mix with the unprocessed mix, you should switch off both the Matrix Encoder and the Decoder.
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❐
The encoding/decoding process will produce significant signal loss compared to the unprocessed mix. This is normal, and does not indicate that
something isn’t working properly. You can however, with careful tweaking of the mix decrease the signal degradation to a much more acceptable level. You have to adjust levels and other settings before the Matrix
Encoder, neither the encoder or decoder can “control” the mix in any way.
5. When you are satisfied with the result, Bypass the Matrix Decoder, or
remove it from its effect slot.
6. Connect a master recording device to the stereo mix output and per-
form a mixdown as usual.
The resulting encoded stereo mix will now be compatible with common home systems
that use the Pro Logic standard.
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Using the Matrix Encoder with the 5.0 surround format
There are situations when you may want to mix for several Surround
formats. For example, you might need to mix the same material for 5.1
and one for LRCS.
5.1 is similar to LRCS. Omitting the LFE channel is easy, but more of
a problem is that LRCS only has one Surround Channel whereas 5.1
has two.
For this reason there are two Surround Channels in the Matrix Encoder, making a total of 5 Channels. This is meant to be used in conjunction with the 5.0 surround format. Proceed as follows.
1. Create your mix for 5.1.
2. Create an output bus with the “5.0” speaker arrangement, in the VST
Connections window, and route it to the physical outputs on your audio hardware.
3. Run the mix through the Matrix Encoder.
Now, the two Surround channels will first be merged together to make
the mix compatible with LRCS. Then the four resulting signals will be
encoded as usual. This will require much fewer adjustment when moving between 5.1 and LRCS.
Using the Matrix Decoder with the 5.0 surround format
The Matrix Decoder also has five channels. This is for similar reasons.
Normally two surround speakers are used even when playing back
LRCS. The two speakers then simply use the same material. The Matrix decoder simulates this by delivering the Surround channel to two
outputs. This allows you to move between formats and listening situations with less repatching of speaker channels.
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Mix6To2
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 the 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.
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.
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Mix8To2
The Mix8To2 effect allows you to control the levels of up to eight 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 Mix8To2 allows you to
quickly mix down your surround mix format to stereo, and to include
parts of the surround channels in the resulting mix.
• Note that the Mix8To2 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 Mix8To 2 should be placed in one of the post fader
insert effect slots for the output bus.
Each of the surround channels have 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.
The Master bus has the following parameters:
• A Link button that links the two Master faders.
• A Normalize button that will normalize the mixed output if activated.
Normalize is a function for controlling the overall loudness of the output. When this is
activated, the level of the mixed output will be boosted to exactly 0dB.
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MixerDelay
The MixerDelay is a tool that allows you to adjust and manipulate each
individual channel in a surround track or bus. The channels have the
following controls:
• Level faders allow you to fine-tune the volume balance between the surround
channels.
• Mute and Solo buttons are useful for listening to individual channels, etc.
• Phase switches let you invert the phase polarity for individual channels.
• Delay controls allow you to delay individual speaker channels. The delay times
are shown in milliseconds and centimeters, making this feature very useful for
distance compensation when playing back surround mixes on different speaker
setups, etc.
• The channel routing section lets you select/switch the desired outputs for the
channels quickly. Note that there are also several channel routing presets
available. (Simply click the “Select Presets” button on the common panel to
open a pop-up mennu listing the available presets.)
Finally there is a common panel to the right with global buttons for
turning off Mute, Solo and Input Phase switches for all channels.
• The MixerDelay is not a mixer – the number of outputs is the same as the
number of inputs.
If you need to mix down a surround signal to stereo, you should use the Mix6to2 or
Mix8to2 plug-ins.
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SurroundDither
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 realtime (playback) or with the Export Audio Mixdown function, you should
consider dithering.
• Since SurroundDither is capable of dithering up to eight 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 8 buttons, one for each channel.
• Above each button there are 8 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 cur-
rently 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
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 plug-ins
This section contains descriptions of the plug-ins in the “Tools” category.
MultiScope
The MultiScope can be used for viewing the waveform, phase linearity
or frequency content of a signal. There are three different modes:
• Oscilloscope (Ampl.)
• Phase Correlator (Scope)
• Frequency Spectrum analyzer (Freq.)
Ampl (Oscilloscope) mode
• To view a signal waveform, open the MultiScope control panel and
make sure that the button “Ampl.” in the lower left corner is lit.
• If the source signal is stereo you can now select either the Left or
Right channel for viewing, or Stereo for both channels to be shown in
the window. If it is a Mono signal, this won’t matter.
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• If the MultiScope is used with a multi-channel track or output bus, you
can select any speaker channel for viewing, or All Channels to view
them all at once.
• You can now adjust the Amplitude knob to increase/decrease the vertical size of the waveform, and the frequency knob to select the frequency area for viewing.
• The “Freeze” button can be used to freeze the display for all three
Scope modes.
Click it again to exit freeze mode.
Phase Correlator mode
To select the phase correlator, click the “Scope” button so that it
lights up. The phase correlator indicates the phase and amplitude relationship between channels in a stereo pair or a surround configuration.
For stereo pairs, the indications work in the following way:
• A vertical line indicates a perfect mono signal (the left and right channels are
the same).
• A horizontal line indicates that the left channel is the same as the right, but
with an inverse phase.
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• A random but fairly round shape indicates a well balanced stereo signal. If the
shape “leans” to the left, there is more energy in the left channel and vice
versa (the extreme case of this is if one side is muted, in which case the Phase
Meter will show a straight line, angled 90° to the other side).
• A perfect circle indicates a sine wave on one channel, and the same sine
wave shifted by 90° on the other.
• Generally, the more you can see a “thread”, the more bass in the signal, and
the more “spray-like” the display, the more high frequencies in the signal.
When the MultiScope is used with a surround channel in Scope
mode, the pop-up menu to the right of the Scope button determines
the result:
• If “Stereo (Front)” is selected, the display will indicate the phase and
amplitude relationship between the front stereo channels, as described above.
• If “Surround” is selected, the display indicates the energy distribution
in the surround field.
Frequency Spectrum Analyzer
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• Click on the “Freq” button so that it lights up in yellow.
The MultiScope is now in Frequency Spectrum analyze mode, and will divide the frequency spectrum into separate vertical bands, which allows you to get a visual overview of the different frequencies’ relative amplitude. The frequency bands are shown
left to right, starting with the lower frequencies.
• If the source signal is stereo you can now select either the Left or
Right channel for viewing, or Stereo for both channels to be shown in
the window. If it is a Mono signal, this won’t matter.
• If the MultiScope is used with a multi-channel track or output bus, you
can select any speaker channel for viewing, or All Channels to view
them all at once.
• Adjust the Amplitude knob to increase/decrease the vertical range of
the bands.
• By adjusting the Frequency knob, you can divide the frequency spec-
trum into 8, 15, or 31 bands, or you can select “Spectrum”, which
shows a high resolution view.
• Use the Mode A and Mode B buttons to switch between different
view modes.
Mode A is more graphically detailed, showing a solid, blue amplitude bar for each
band. Mode B is less detailed, showing a continuous blue line that displays the peak
levels for each band. These view modes don’t have any effect if you have selected
“Spectrum” with the Frequency knob.
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SMPTE Generator
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 Nuendo (provided that the equipment can sync directly to SMPTE time
code). This can be very useful if you don’t have access to a MIDI-totime 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
Nuendo.
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 Nuendo. 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.
• Current Time
When Link is on this shows the current position in Nuendo. If Link is off it shows the
current time of the SMPTE Generator in “free run” mode. This cannot be set manually.
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• 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 Nuendo, 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 Nuendo
Proceed as follows:
1. Connect the SMPTE Generator as an insert effect on an audio chan-
nel, 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 Nuendo.
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 Nuendo 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
Nuendo Transport panel.
• Press Play on the Nuendo Transport panel.
The other device is now synchronized and will follow any position changes set with the
Nuendo transport controls.
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Test Generator
This utility allows you to generate an audio signal, which can be recorded as an audio file. The resulting file can then be used for a number of purposes:
• For testing the specifications of audio equipment.
• For measurements of various kinds, including calibrating tape recorders.
• For testing signal processing methods.
• For educational purposes.
The Test Generator is based on a waveform generator which can generate a number of basic waveforms such as sine and saw and various
types of noise. In addition, you can also set the frequency and amplitude of the generated signal.
As soon as you add the Test Generator as an effect to an audio track
and activate it, a signal is generated. You can then activate recording
as usual to record an audio file according to the signal specifications:
Parameter Description
Waveforms By clicking these buttons, you select the basis for the signal gen-
erated by the waveform generator. You can select between four
basic waveforms: Sine, Square, Sawtooth and Triangle, or three
types of noise (white, brown and pink noise – from left to right).
Frequency This controls the frequency of the generated signal, from 1 Hz to
Gain This controls the amplitude of the signal. The higher the value (up
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20000 Hz.
to 0 dB) the stronger the signal.
2
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 106.
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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 (Oscillator
1 only)
FM Env (Osc 1
only)
Governs the amount of frequency modulation. See page 106.
This governs how much the Filter Envelope parameters affects the FM
amount. Positive and negative values can be set. See page 106.
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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. The
filter types are described on page 105.
Cutoff Controls the filter frequency or “cutoff”. If a lowpass filter is used, it
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 105).
Resonance The Resonance control for the filter. Raise this for a more pronounced
filter sweep effect.
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
Envelope. Negative values will invert the filter envelope settings.
Filter Velocity Determines how the filter cutoff will be affected by velocity, i.e. how
hard 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-
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.
Mono When this is activated, the A1 will be monophonic, i.e. only play
one voice at a time.
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