Steinberg Nuendo - 3.0 User Manual

Audio Effects and VST Instruments

Original Manual: Synkron
Revision and Quality Control for Nuendo 3:
Ashley Shepherd, Cristina Bachmann, Heiko Bischoff, Sabine Pfeifer, Claudia Schomburg
Thanks: Alexander Lerch

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

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

© Steinberg Media Technologies GmbH, 2004.
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
46 Modulation plug-ins
59 Other plug-ins
67 Restoration
75 Reverb plug-ins
80 Surround plug-ins
99 Tools

107 The included

VST Instruments

108 A1 Synthesizer
118 VB-1 Bass Synth
120 LM-7 Drum Machine
123 Embracer – Surround Pad

Synthesizer

128 Monologue – Monophonic

Analog Modeling Synthesizer

135 Mixconvert Appendix

136 Available conversions

141 Index

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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 Description

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

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

Tempo sync
on/off

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

Delay Time 2 As above.

Feedback This sets the number of repeats for both delays.

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

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

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

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

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Parameter Description

Pan1 This sets the stereo position for the first delay.

Pan2 This sets the stereo position for the second delay.

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

•

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

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

•

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

•

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

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ModDelay

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

The parameters are as follows:

Parameter Description

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

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

Tempo sync
on/off

Feedback This sets the number of repeats for the delay.

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

Tempo Sync
knob

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

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

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

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

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Distortion plug-ins

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

DaTube

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

The parameters are as follows:

Parameter Description

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

overdriven sound just on the verge of distortion.

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

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

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

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Overdrive

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

The parameters are as follows:

Parameter Description

Input Sets the input level.

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

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

Speaker
simulation

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

Bass Tone control for the low frequencies.

Mid Tone control for the mid frequencies.

Hi Tone control for the high frequencies.

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

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

for further “tweaking”.

and dragging in the display.

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QuadraFuzz

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

•

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

•

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

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

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How does QuadraFuzz work?

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

• The signal volume control before distortion.

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

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

• The signal volume control after distortion.

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

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Editing in the frequency band display

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

The frequency band display

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

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

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

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

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

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

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

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

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

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

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

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The parameters

The following tables list all parameters available in QuadraFuzz.

The parameters in the main window are as follows:

Parameter Description

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

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

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

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

Output dial This controls the overall output level.

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

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

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

preset, click on the fader handle and drag horizontally.

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

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

The parameters in the editor window are as follows:

Parameter Description

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

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

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

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

Solo This mutes all frequency bands except the selected band.

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

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

Frequency band
display

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

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Dynamics plug-ins

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

SPL DeEsser

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

The SPL DeEsser has the following parameters:

Parameter Description

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

you start with a value between 4 and 7.

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

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

Auto Threshold See separate description below.

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

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

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About the Auto Threshold function

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

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

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

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Dynamics

Compressor

Routing
selector

AutoGate

Limiter

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

Activating the individual processors

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

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The AutoGate section

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

The available parameters are as follows:

Parameter Values Description

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

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

Attack 0,1 -100 ms or

“Predict mode”

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

Release 10 - 1000 ms or

“Auto”

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

ter the signal drops below the threshold level.

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

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Trigger Frequency Range

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

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

1.

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

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

2.

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

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

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

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

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

3.

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

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

4.

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

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

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The Calibrate function

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

Use it as follows:

1.

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

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

2.

Play it back, and click on the Calibrate button.

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

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The Compressor section

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

The available parameters work as follows:

Parameter Values Description

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

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

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

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

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

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

Release 10-1000ms or

“Auto mode”

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

Compressor
Mode

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

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

loss, caused by compression.

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

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The Limiter section

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

The available parameters are the following:

Parameter Values Description

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

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

Release 10-1000ms or

“Auto mode”

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

The Routing section

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

• 1-2-3 (Compressor-Gate-Limit)

• 2-1-3 (Gate-Compressor-Limit)

• 1-3-2 (Compressor-Limit-Gate)

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Magneto

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

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

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

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

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

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

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

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

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

•

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

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

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About the Drive parameter and Magneto output levels

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

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

Metering Switch

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

•

Input

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

•

onTape

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

•

Output

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

Clip LEDs

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

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Input Level

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

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

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

Output Level

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

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

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

If “digital clipping” occurs

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

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The main parameters

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

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

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

Drive

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

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

Please use the following guidelines:

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

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

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

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

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

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Characteristics

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

Tape Speed

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

HF-Adjust

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

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

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MIDI Gate

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

Setting up

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

To set it up, proceed as follows:

1.

Select the audio to be affected by the MIDI Gate.

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

2.

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

The MIDI Gate control panel opens.

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

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

4. Open the Output (“out:”) pop-up menu for the MIDI track and select
the MIDI Gate option.

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

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

Make sure the MIDI track is selected and start playback.

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

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

The following MIDI Gate parameters are available:

Parameter Description

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

after receiving a signal that triggers it.

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

message (see Hold Mode below).

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

value set with the Hold parameter).

Note To
Attack

Note To
Release

Velocity To
VCA

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

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

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

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

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

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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]/[Command] and move a handle, the values will
change in smaller steps.

Adding and removing frequency bands

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

About the Frequency scale

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

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

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The Solo button

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

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

Using the Characteristics editor

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

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

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

About breakpoints

• Clicking anywhere on the line will add a breakpoint.

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

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

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

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

About the Compressor type (MODE)

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

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

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The Output dial

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

The SoftClip function

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

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VST Dynamics

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

page 18), but with the following differences:

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

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

Activating the individual processors

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

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

Clicking the button again activates the same configuration of processors.

Auto Gate section

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

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Auto Level section

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

Parameter Description

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

Reaction time buttons
(Slow, Mid, Fast)

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

Compressor section

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

Soft Clip section

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

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

Limiter section

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

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Filter plug-ins

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

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.

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.

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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.

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

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.

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Parameter Description

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

Left/Stereo/Right/Mono
Modes

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

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StepFilter

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

General operation

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

Setting step values

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

• Individual step entries can be freely dragged up or down the vertical

axis, or directly set by clicking in an empty grid box. By click-dragging
left or right consecutive step entries will be set to the pointer position.

Setting filter cutoff values in the grid window.

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

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

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

Selecting new patterns

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

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

• To select new patterns you use the pattern selector.

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

Pattern Selector

Using pattern copy and paste to create variations

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

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

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

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StepFilter parameters

Parameter/Value Description

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

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

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

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

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

values to change more smoothly.

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

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

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

Mix Adjusts the mix between dry and processed signal.

Output Sets the overall volume.

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

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Tonic – Analog Modeling Filter

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

The Tonic Analog Modeling Filter has the following properties:

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

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

12dB band pass and 12dB high pass modes.

• Adjustable drive and resonance up to self-oscillation.

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

• Audio and MIDI trigger modes.

• Powerful step LFO with smoothing and morphing.

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

Tonic parameters.

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Filter

Parameter Description

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

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

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

governed by the filter type.

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

puts the filter into self-oscillation.

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

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

Mix Sets the balance between dry and effect signal.

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

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

Env Mod

Parameter Description

Mode Tonic offers three types of envelope modulation:

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

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

in slower rise times when the envelope is triggered.

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

sult in slower envelope tails.

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

level.

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

A rather stunning effect…

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LFO Mod

Parameter Description

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

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

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

level.

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

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

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

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

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

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

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

are grayed out in the step window.

Preset Offers a number of step LFO waveform patterns. Choices include:

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

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

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

X/Y Pad

Parameter Description

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

of Tonic’s parameters are available as destinations

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

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

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

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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 Description

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

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

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

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

Frequency This sets the modulation rate.

Delay This controls the depth of the Chorus effect.

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

ered chorus effect.

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

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Flanger

Flanger is a classic flanger effect with stereo enhancement.

The parameters are as follows:

Parameter Description

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

Output Sets the overall volume.

Tempo sync
on/off

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

Tempo Sync
knob

Shape Sync
knob

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

Depth This sets the depth of the modulation sweep.

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

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

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

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

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

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

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

duce a more “metallic” sounding sweep.

adjusting the initial delay time.

and 100% maximum stereo enhancement.

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

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

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

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

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

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

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Metalizer

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

Parameter Description

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

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

Output Sets the overall volume.

Tempo sync
on/off

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

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

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

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

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

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

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

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

work as a static filter.

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

able with high Feedback settings.

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

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Phaser

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

Parameter Description

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

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

Output Sets the overall volume.

Tempo sync
on/off

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

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

Tempo Sync
knob

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

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

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

nounced effect.

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

reo, and 100% maximum stereo enhancement.

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

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

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

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

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

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

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Ringmodulator

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

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

Parameter Description

Oscillator LFO
Amount

Oscillator Env.
Amount

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

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

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

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

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

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

range.

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

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Parameter Description

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

LFO Speed Sets the LFO Speed.

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

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

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

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

Envelope Generator
(Attack and
Decay dials)

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

Mix Adjusts the mix between dry and processed signal.

Output Sets the overall volume.

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

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

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Rotary

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

The parameters are as follows:

Parameter Description

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

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

See page 55.

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

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

Overdrive Applies a soft overdrive or distortion.

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

high frequency loudspeakers.

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

each side.

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

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Parameter Description

Low Rotor Amp
Mod.

Low Rotor Mix
Level

Hi Rotor Amp
Mod.

Hi Rotor Freq.
Mod.

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

Hi Slow Fine adjustment of the high rotor Slow speed.

Hi Rate Fine adjustment of the high rotor acceleration time.

Hi Fast Fine adjustment of the high rotor Fast speed.

Lo Slow Fine adjustment of the low rotor Slow speed.

Lo Rate Fine adjustment of the low rotor acceleration time.

Lo Fast Fine adjustment of the low rotor Fast speed.

Output Adjusts the overall output level.

Mix Adjusts the mix between dry and processed signal.

Adjusts amplitude modulation depth.

Adjusts overall bass level.

High rotor amplitude modulation.

High rotor frequency modulation.

Directing MIDI to the Rotary

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

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

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

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Symphonic

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

The parameters are as follows:

Parameter Description

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

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

Tempo sync
on/off

Tempo Sync
pop-up

Tempo Sync
knob

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

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

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

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

Output Adjusts the output level of the effect.

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

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

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

effect, if activated.

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

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

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

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

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

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

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

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Tranceformer

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

Parameter Description

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

Output Adjusts the output level of the effect.

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

Tempo sync
on/off

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

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

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

Depth Governs the depth of the pitch modulation.

Waveform
buttons

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

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

Sets the pitch modulation waveform.

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

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Other plug-ins

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

Bitcrusher

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

Parameter Description

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

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

Sample
Divider

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

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

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

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

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

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

increase the level.

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Chopper

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

Parameter Description

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

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

Tempo sync
on/off

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

Stereo/Mono
button

Waveform
buttons

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

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

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

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

Sets the modulation waveform.

the graphic display.

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Apogee UV 22 HR

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 96).

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.

The included effect plug-ins 1 – 61

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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 63.

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 63.

To set up for use, proceed as follows:

1. Select a source for the modulator.

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

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

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

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

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

4. Select a MIDI track.

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

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

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

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

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

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

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

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

Setting up – using an external carrier

There are two modes for using an external carrier:

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

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

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

This is described on page 64.

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

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

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

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

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

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

it on another empty audio track.

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

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

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

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

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

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

modulator channel!

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

Setting up – using an external carrier plus MIDI

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

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

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

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Vocoder parameters

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

Parameter Description

Number of
Bands

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

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

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

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

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

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

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

Output This controls the output level of the Vocoder.

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

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

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

the Vocoder, respectively.

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

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

original input signal while notes are played.

output while notes are played.

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

ting high frequencies pass.

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Vocoder synth parameters

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

Parameter Description

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

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

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

Noise Adds white noise to the sound.

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

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

P.Drift Adds random pitch variation to the oscillators.

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

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

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

tors. It does not affect the white noise generator.

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

LFO Rate Controls the LFO rate (for vibrato).

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

Mod Wheel.

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Restoration

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

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.

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.

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• 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.

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.

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Parameter Description

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 Description

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

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

Noise This dial regulates the amount of static noise added.

Distort Use this dial to add distortion.

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

more hollow, lo-fi sound.

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

Frequency
switch

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

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

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

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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 Description

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

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

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

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

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

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

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

verb sound softer.

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

reduce low frequency “rumble”.

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Reverb B

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

Parameter Description

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

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

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

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

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

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

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

and smoother sounding reverb.

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RoomWorks

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

Parameter Description

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

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

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

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

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

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

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

Time Reverb Time in milliseconds.

Hold Pressing this button freezes the reverb buffer in an infinite loop

(yellow circle around button). You can create some interesting
pad sounds using this feature.

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

or smaller spaces.

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

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

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Parameter Description

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

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

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

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

Export This button determines if during audio export RoomWorks will

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

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

verb program using altered reflection patterns. This is helpful when
certain sounds are causing odd ringing or undesirable results.
Creating a new variation will often solve these issues. There are
1000 possible variations.

High Damping Freq This determines the frequency above which high frequency

damping will occur.

High Damping
Amount

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

Low Damping
Amount

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

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

occur.

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

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

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Parameter Description

Envelope Attack The envelope settings in RoomWorks control how the reverb will

follow the dynamics of the input signal in a fashion similar to a
noise gate or downward expander. Attack determines how long
in milliseconds it takes for the reverb to reach full volume after a
signal peak. This is similar to a predelay but the reverb is ramping
up instead of starting all at once.

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

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

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

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

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

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

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

Pos The position control is only available for surround configurations.

With this parameter you can control where the virtual listening po­sition is within the room. Positive values position the listener closer
to the front of the room and negative values place the listener to­wards the rear of the room.

Bal (Balance, only
available for surround
channels)

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

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Surround plug-ins

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

Matrix Decoder

The Matrix Decoder reverses the Encoder 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

82.

• 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.

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.

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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 per­form a mixdown as usual.

The resulting encoded stereo mix will now be compatible with common home systems
that use the Pro Logic standard.

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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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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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 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 pop­up 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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Mixconvert

The Mixconvert plug-in is similar to the Mix6To2 and Mix8To2 plug-ins
in that it is used to quickly convert a multichannel mix into another for­mat that uses less channels when used as insert (for example convert­ing a 5.1 surround mix to a stereo mix). Mixconvert can convert
surround mixes into other surround formats such as mixing a 7.1 Cin­ema surround format down to a 5.1 home theater format.

There are several obvious applications for this:

• Auditioning what an automatically generated downmix will sound like at the
customer’s location.

• Quickly generating an additional mix that uses a different number of channels
or a different speaker configuration.

• Outputting several mix configurations simultaneously in various surround for­mats for broadcast purposes.

Users can use presets with standard upmix/downmix setups for spe­cific configurations. It is possible to save up to 64 user-defined pre­sets for each input/output configuration.

Mixconvert is unique as a plug-in since it is used automatically by
Nuendo in certain situations (like SurroundPanner). Nuendo will sub­stitute Mixconvert for the panner in either the main channel or in the
aux send panner position when an upmix or a downmix is needed.
These are the possible scenarios:

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• Whenever a multichannel audio track (more than three audio paths), group
channel or FX channel is routed to an output bus or group channel with a dif­ferent number of audio paths (e.g. 5.1 to stereo), a Mixconvert plug-in will be
inserted in place of the panner in that channel.

Indicates that Mixconvert is
inserted in place of the panner.

• Whenever a multichannel audio track, group channel, FX channel or Output
bus has an aux send that is routed to a Group channel or Output bus with a
different number of audio paths, a Mixconvert plug-in will be inserted in place
of the aux send’s panner.

Indicates that Mixconvert is inserted
in the aux send panner position.

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Interface

Overview

The plug-in’s interface has three different sections. On the left you
find the input Configuration display with all parameters that directly af­fect the input configuration. In the middle section the level parameters
for the upmix/downmix are displayed. Above this, the preset controls
can be found. On the right the output configuration is displayed with
all parameters that affect the output configuration. Additionally, on the
far left there is a gain fader.

The following sections explain all controls in detail. Note that when
you move the mouse pointer over a control, a tooltip is displayed at
the bottom of the MixConvert window.

Global Gain fader

Gain depends on the input signal, the number of loudspeakers and a
number of downmix parameters (see page 91). You can use this fader
to globally adjust gain by ±12dB for all channels.

Max Output Level

This field shows the maximum output level. The LED display on the
right hand side of the field indicates whether this maximum level is
above 0dB (clipping). Click the LED to reset the value field and the in­dicator.

Input Configuration

The Input Configuration is determined by the channel width of the
track, group or output bus Mixconvert is inserted in.

Output Configuration

The Output Configuration can only be modified when used as an insert
effect. When Nuendo automatically replaces the panner by Mixconvert,
the Output configuration is determined by the destination of the chan­nel or aux send. When used as an insert effect, the Output configura­tion can be changed either directly in the pop-up menu above the
Output Configuration display or indirectly by loading a preset.

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Faders for Surround, Center and LFE

These faders control the levels for the surround channels, front center
channel and LFE channel in the upmix/downmix. The surround chan­nels cannot be modified individually. For center and surround chan­nels, the level can be changed between -∞ and +6dB. For the LFE
channel it can be changed between -∞ and +10dB, since in some
mixes the LFE channel may be attenuated by 10dB (see page 91).
The names Surround, Center and LFE refer to the corresponding
channels in the Input Configuration.

Solo and Mute buttons

Using the Solo and Mute buttons (on the left of the Input Configuration
and the right of the Downmix Configuration sections) you can mute or
solo all front or surround channels simultaneously (see page 92).

Soloing or muting individual speakers

If you want to solo or mute a single loudspeaker in the Input Configura­tion or Output Configuration displays, you can click on it. Simply click­ing will solo the channel. When you hold down the [Alt]/[Option] key
while clicking, the channel will be muted. Holding down the [Ctrl]/[Com­mand] key while clicking will also mute all channels currently in solo
mode. Clicking again (without a modifier key) will reset the channel.

Phase shift

You can shift the phase of the front left/right channels and the sur­round left/right channels in steps of 90˚. Clicking the button once will
increase the phase by a further 90˚. You can reset the phase value by
right-clicking (Windows) or [Ctrl]-clicking (Mac) on the button.

Phase shifting can be used for various purposes. In a downmix from 2
channels to 1 channel it may be useful to introduce a 90° phase shift
on one channel to avoid level increases in the downmix signal (caused
by frequencies present in both channels). Also, phase shifts can be
used to create “virtual” reverberation by cancelling all center informa­tion, leaving the resulting ambience.

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As a general rule, you should be careful when using phase shifts, as
they might have negative repercussions on the frequency spectrum and
the level of the downmix. Also, when you generate matrixed downmixes,
you should avoid introducing additional phase shifts, since these would
prevent the decoding of the mix for different speaker configurations.

Toggling between parameter sets

You can use the Memory, Toggle and Clear buttons to toggle be­tween two different sets of downmix parameters, for direct compari­son. Click the Memory button to write all current parameters to the
temporary parameter buffer. This buffer is cleared when clicking the
Clear button. Using the Toggle button, you can switch between the
buffered parameter set and the (changed) current parameter set. Note
that here the Output Configuration is not a parameter, but must be
identical for both parameter sets.

Modifying the width

The front and back Width controls are used to set the width of the au­dible panorama. At minimal width (0%) the panorama is very narrow.
In most cases, the default setting will be 50%. The 50% setting re­sults in unaltered signals. Values above 50% will create an artificial
widening of the panorama; similar to phase shifting. You should be
careful when modifying the panorama width when you want to gener­ate matrixed downmixes.

Drag the Width controls (the colored lines at the top and bottom of
the input Configuration display) to set the width. You can also click on
the name of the control to open a pop-up menu from which you can
select set values (0%, 25%, 50% and 100%).

Any signals that are equally in either the surround channels or the main
left and right channels will be completely out of phase (180°) when the
width parameter is set to 100%. This will cause those signals to be com­pletely cancelled when played over a mono system, such as AM radio
broadcast or mono television. Always check for mono compatibility with
mixes that are to be broadcast.

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Loading and saving presets

Full presets are only available for Mixconvert when it is used as an in­sert effect. When Nuendo automatically places Mixconvert in place of
a panner, the preset menu displays only presets for the current input/
output configurations.

Presets are selected and managed at the top of the middle section of
the plug-in interface. The name of the currently selected preset is dis­played in the text field. Click the symbol next to the text field to open a
pop-up menu from which you can select a different preset. Which pre­sets are available from this pop-up menu depends on the downmix op­tions available for the current input configuration. You save a new set of
parameters by entering a new name in the text field and selecting Save
Preset from the pop-up menu that appears when you click the Save
button. You can save up to 64 presets for every input/output configura­tion. To delete a user preset, select Delete Preset from the Save pop­up menu. Note that the factory-defined presets cannot be deleted.

General Notes

Level

The volume of the downmixed signal can be different from the volume
of the original mix. There are several reasons for this:

• The input signals must be scaled to avoid clipping.

• The number of speakers used influences the overall volume.

• The level of the downmixed signal depends on the correlation of all added sig­nals, which is why phase shifting can influence the volume level.

LFE channel

The LFE channel is automatically filtered using a low-pass filter. The
cutoff frequency of this low-pass filter is 120Hz, the filter slope is
12dB/Oct. An LFE channel present in the input configuration, but not
present in the output configuration, is mixed evenly to the front-left
and front-right channels since it is assumed that these will be the
channels using the speakers with the widest frequency range.

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Keyboard shortcuts

The plug-in interface is designed for mouse operation. There are two
commands for which keyboard shortcuts exist:

• Store Parameter Memory: [M] (for “memory”)

• Toggle Parameters: [S] (for “swap”)

Solo mode

Since there is no dedicated solo bus, all solos are inplace, i.e. all other
(non-solo) channels are muted.

Functionality and available conversions

The speaker configuration of the input mix (Input Configuration) is
defined by the width of the channel it is inserted in. It is displayed au­tomatically. The speaker configuration of the output mix (Output Con­figuration) is automatically selected when Mixconvert is inserted in the
panner position of a channel or aux send. If it is used as an insert ef­fect, the output configuration can be selected either from the corre­sponding menu or by loading a preset.

Note, however, that not all theoretically possible combinations are ac­tually available. Mixconvert is limited to channels with 8 audio paths
(this means that 10.2 or 8.1 are not supported). In the appendix on

page 136 you find the tables listing all available and all unavailable

combinations.

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Brief description of Mixconvert parameters

Parameter name Description

Width Modifies the panorama

- 0% (minimum width)

- 50% (normal width, unaltered)

- 100% (maximum width)

Global Gain Attenuates or increases all channels to compensate for clipping

or low levels in the converted signal

Surround level Level of the surround channel

LFE level Level of the LFE channel

Center level Level of the front center channel

Phase shift Phase shift of a channel (settings: 0˚, 90˚, 180˚, 270˚), available

for front and surround left/right. Click once for shifting a further
90˚. Right-click/[Ctrl]-click to reset to 0˚.

Speaker Click a speaker symbol to set the speaker to mute or solo mode.

Click once for activating the Solo mode. [Alt]/[Option]-click for
activating the Mute mode. [Ctrl]/[Command]-click for activating
the exclusive solo (mute all other channels even if they are also
solo). Click again on a speaker to reset the channel.

Solo button Soloes all front and surround channels

Mute button Mutes all front and surround channels

Output Config Only available when used as insert. Sets the output speaker con-

figuration

Store Memory Temporarily saves the current parameter set

Toggle Memory Toggles between the current and the temporary parameter set

Clear Memory Clears the temporary parameter buffer

Save Preset Saves or deletes the preset specified in the preset text field

Preset pop-up menu Loads a preset

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MixerDelay

The MixerDelay is a tool that allows you to adjust and manipulate each
individual channel in a surround track, group or bus. Each channel has
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 or 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.

It is common for the center channel in a 5.1 speaker configuration to be closer to the
mix position in order to accommodate large video monitors or projection screens. In
cases like this, Mixerdelay can be used to compensate for the center channel being
too close. Simply adjust the delay for the center channel by the difference in distance
(in cm) between it and the other speakers to the mix position. You must delay the
closer speaker so that the sound from it arrives at the same time as the sound from the
more distant speakers. Note that Mixerdelay has a wide range (up to 1000ms) and fine
adjustments are best made by numerically entering the delay time in centimeters for
speaker alignment.

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• The channel routing section lets you select/switch the desired outputs for the
channels quickly. You can assign the same output to several channels by hold­ing down the [Alt]/[Option] key while selecting. 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 menu 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, Mix8to2 or Mixconvert plug-ins.

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 real-time
(playback) or with the Export Audio Mixdown function, you should con­sider dithering.

• Since SurroundDither is capable of dithering up to six channels at the
same time, it is recommended if you’re using surround channels.

If not, you may want to use the UV22 HR instead, see page 61.

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

Dithering Type

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

Option Description

Off No dithering is applied.

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

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

Noise Shaping Options (Off, Type 1 - 3)

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

Ditherbits

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

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

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

the bit resolution the files will be converted to.

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

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An Example

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

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

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

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

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

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

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

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SurroundPan

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

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

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Tools

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

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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