Steinberg Nuendo - 5.0 Plug-in Reference

Plug-in Reference
Cristina Bachmann, Heiko Bischoff, Marion Bröer, Sabine Pfeifer, Heike Schilling
The information in this document is subject to change without notice and does not represent a commitment on the part of Steinberg Media Technologies GmbH. The software described by this document is subject to a License Agreement and may not be copied to other media except as specifically allowed in the License Agreement. No part of this publica­tion may be copied, reproduced or otherwise transmitted or recorded, for any purpose, without prior written permission by Steinberg Media Technologies GmbH.
All product and company names are ™ or ® trademarks of their respective owners. Windows XP is a trademark of Microsoft Corporation. Windows Vista and Windows 7 are registered trademarks or trademarks of Microsoft Corpora­tion in the United States and/or other countries. The Mac logo is a trademark used under license. Macintosh and Power Macintosh are registered trademarks. MP3SURROUND and the MP3SURROUND logo are registered trademarks of Thomson SA, registered in the US and other countries, and are used under license from Thomson Licensing SAS.
Release Date: April 13, 2010
© Steinberg Media Technologies GmbH, 2010.
All rights reserved.

Table of Contents

5 The included effect plug-ins
6 Introduction 6 Delay plug-ins 9 Distortion plug-ins 10 Dynamics plug-ins 19 EQ plug-ins 21 Filter plug-ins 27 Generator plug-ins 29 Modulation plug-ins 36 Other plug-ins 38 Pitch Shift plug-ins 39 Restoration plug-ins 43 Reverb plug-ins 50 Spatial + Panner plug-ins 53 Surround plug-ins 61 Tools – MultiScope
63 MIDI effects
64 Introduction 64 Arpache 5 65 Arpache SX 66 Auto LFO 67 Beat Designer (Nuendo Expansion Kit only) 72 Chorder 74 Compressor 75 Context Gate 76 Density 76 Micro Tuner 76 MIDI Control 77 MIDI Echo 78 MIDI Modifiers 78 MIDI Monitor 79 Note to CC 80 Quantizer 80 StepDesigner 82 Track Control 84 Transformer
85 MixConvert Appendix
86 Available conversions
88 Index
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Table of Contents
1

The included effect plug-ins

Introduction

Delay plug-ins

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.
Most of the included effects are compatible with VST3, this is indicated by an icon in front of the name of the plug­in as displayed in plug-in selection menus (for further infor mation, see the chapter “Audio effects” in the Operation Manual).
This section contains descriptions of the plug-ins in the “Delay” category.

ModMachine

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ModMachine combines delay modulation and filter fre­quency/resonance modulation and can provide many interesting modulation effects. It also features a Drive parameter for distortion effects.
The following parameters are available:
Parameter Description
Delay This is where you specify the base note value for the de-
Delay – Sync button
Rate The Rate parameter sets the base note value for tempo
Rate – Sync button
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6
lay if tempo sync is on (1/1–1/32, straight, triplet, or dot­ted). If tempo sync is off, the delay time can be set freely in milliseconds.
The button below the Delay knob is used to switch tempo sync for the Delay parameter on or off.
syncing the delay modulation (1/1 to 1/32, straight, triplet, or dotted). If tempo sync is off, the rate can be set freely.
The button below the Rate knob is used to switch tempo sync for the Rate parameter on or off.
Parameter Description
Width Sets the amount of delay pitch modulation. Note that al-
Feedback Sets the number of repeats for the delay.
Drive Adds distortion to the feedback loop. The longer the
Mix Sets the level balance between the dry signal and the ef-
Nudge button Clicking the Nudge button once will momentarily speed
Signal path graphic and Filter position
Filter type (in graphic display)
Freq Sets the cutoff frequency for the filter. It is only available
Speed Sets the speed of the filter frequency LFO modulation.
Speed – Sync button
Range Lo/Hi These knobs specify the range (in Hz) of the filter fre-
Spatial Introduces an offset between the channels to create a
Q-Factor Controls the resonance of the filter. It is only available if
Speed Sets the speed of the filter resonance LFO modulation.
Speed – Sync button
though the modulation affects the delay time, the sound is mostly perceived as a vibrato or chorus-like effect.
Feedback, the more the delay repeats become distorted over time.
fect. If ModMachine is used as a send effect, set this to the maximum value (100 effect balance with the send.
up the audio coming into the plug-in, simulating an ana log tape nudge type sound effect.
The filter can either be placed in the feedback loop of the delay or in the output path of the effect (after the Drive and Feedback parameters). To switch between the “loop” and “output” positions, click on the Filter section displayed in the graphic or click on the Position field at the bottom right of the graphic.
The Type button allows you to select a filter type. A low­pass, band-pass, and high-pass filter are available.
if tempo sync for the Speed parameter (see below) is deactivated and the parameter is set to “0”.
When using tempo sync, the Speed parameter sets the base note value for tempo syncing the modulation (1/1 to 1/32, straight, triplet, or dotted). If tempo sync is off, the speed can be set freely.
The button below the Speed knob is used to switch tempo sync for the Speed parameter on or off.
quency modulation. Both positive (e. g. Lo set to 50 and Hi set to 10000) and negative (e. Hi set to 500) ranges can be set. If tempo sync is off and the Speed is set to zero, these parameters are inactive and the filter frequency is controlled by the Freq parame ter instead.
stereo panorama effect for the filter frequency modula tion. Turn clockwise for a more pronounced stereo effect.
filter resonance LFO tempo sync is deactivated and the Speed parameter (see below) is set to “0”. When using tempo sync, the resonance is controlled by the Speed and Range parameters.
When using tempo sync, the Speed parameter sets the base note value for tempo syncing the modulation (1/1 to 1/32, straight, triplet, or dotted). If tempo sync is off, the speed can be set freely.
The button below the Speed knob is used to switch tempo sync for the Speed parameter on or off.
%) as you can control the dry/
g. Lo set to 5000 and
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Parameter Description
Range Lo/Hi These knobs specify the range of filter resonance mod-
Spatial Introduces an offset between the channels to create a

MonoDelay

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ulation. Both positive (e. g. Lo set to 50 and Hi set to
100) and negative (e. ranges can be set. If tempo sync is off and the Speed is set to zero, these parameters are inactive and the filter resonance is controlled by the Q-Factor parameter in stead.
stereo panorama effect for the filter resonance modu lation. Turn clockwise for a more pronounced stereo effect.
g. Lo set to 100 and Hi set to 50)
This is a mono delay effect that can either be tempo­based or use freely specified delay time settings.
The following parameters are available:
Parameter Description
Delay This is where you specify the base note value for the delay
Sync button The button below the Delay knob is used to switch tempo
Feedback Sets the number of repeats for the delay.
­Filter Lo This filter affects the feedback loop of the effect signal
Filter Hi This filter affects the feedback loop of the effect signal
Mix Sets the level balance between the dry signal and the ef-
if tempo sync is on (1/1–1/32, straight, triplet, or dotted). If tempo sync is off, it sets the delay time in milliseconds.
sync on or off.
and allows you to roll off low frequencies from 10
Hz. The button below the knob activates/deactivates
800 the filter.
and allows you to roll off high frequencies from 20 down to 1.2 deactivates the filter.
fect. If MonoDelay is used as a send effect, set this to the maximum value as you can control the dry/effect balance with the send.
kHz. The button below the knob activates/
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Hz up to
kHz
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The delay can also be controlled from another signal source via the side-chain input. When the side-chain sig­nal exceeds the threshold, the delay repeats are silenced. When the signal drops below the threshold, the delay re­peats reappear. For a description of how to set up side­chain routing, see the chapter “Audio effects” in the Oper ation Manual.
The delay can also be controlled from another signal source via the side-chain input. When the side-chain sig­nal exceeds the threshold, the delay repeats are silenced. When the signal drops below the threshold, the delay re­peats reappear. For a description of how to set up side-
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chain routing, see the chapter “Audio effects” in the Oper ation Manual.
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PingPongDelay

This is a stereo delay effect that alternates each delay re­peat between the left and right channels. The effect can either be tempo-based or use freely specified delay time settings.
The following parameters are available:
Parameter Description
Delay This is where you specify the base note value for the delay
Sync button The button below the Delay Time knob is used to switch
Feedback Sets the number of repeats for the delay.
Filter Lo This filter affects the feedback loop and allows you to roll
Filter Hi This filter affects the feedback loop and allows you to roll
Spatial Sets the stereo width for the left/right repeats. Turn clock-
Mix Sets the level balance between the dry signal and the ef-
if tempo sync is on (1/1–1/32, straight, triplet, or dotted). If tempo sync is off, it sets the delay time in milliseconds.
tempo sync on or off.
off low frequencies up to 800 knob activates/deactivates the filter.
off high frequencies from 20 button below the knob activates/deactivates the filter.
wise for a more pronounced stereo “ping-pong” effect.
fect. If PingPongDelay is used as a send effect, set this to the maximum value as you can control the dry/effect bal ance with the send.
Hz. The button below the
kHz down to 1.2 kHz. The

StereoDelay

StereoDelay has two independent delay lines which either use tempo-based or freely specified delay time settings.
The following parameters are available:
Parameter Description
Delay 1 & 2 Using these controls you specify the base note value for
Sync button The buttons below the Delay knobs are used to turn
Feedback 1 & 2
Filter Lo 1 & 2
Filter Hi 1 & 2
Pan 1 & 2 These controls are used to set the stereo position for
Mix 1 & 2 Use these controls to set the level balance between the
The delay can also be controlled from another signal
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source via the side-chain input. When the side-chain sig­nal exceeds the threshold, the delay repeats are silenced. When the signal drops below the threshold, the delay re peats reappear. For a description of how to set up side­chain routing, see the chapter “Audio effects” in the Oper­ation Manual.
the delay if tempo sync is on (1/1–1/32, straight, triplet, or dotted). If tempo sync is off, they set the delay time in milliseconds.
tempo sync on or off for the respective delay.
The Feedback controls set the number of repeats for each delay.
These filters affect the feedback loop and allow you to roll off low frequencies up to 800 knobs activate/deactivate the filter.
These filters affect the feedback loop and allow you to roll off high frequencies from 20 buttons below the knobs activate/deactivate the filter.
each delay.
dry signal and the effect. If StereoDelay is used as a send effect, set them to the maximum value (100 control the dry/effect balance with the send.
Hz. The buttons below the
kHz down to 1.2 kHz. The
%) as you can
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Distortion plug-ins

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

AmpSimulator

AmpSimulator is a distortion effect, emulating the sound of various types of guitar amp and speaker cabinet combi­nations. A wide selection of amp and cabinet models is available.
The following parameters are available:
Parameter Description
Amplifier pop-up menu
Drive Controls the amount of amp overdrive.
Bass Tone control for the low frequencies.
Middle Tone control for the mid frequencies.
Treble Tone control for the high frequencies.
Presence Boosts or dampens the higher frequencies.
Volume Controls the overall output level. Cabinet
pop-up menu
Damping Lo/Hi Further tone controls for shaping the sound of the selected
This pop-up menu is opened by clicking on the amplifier name shown at the top of the amp section. It allows you to select an amplifier model. The amp section can be by passed by selecting “No Amp”.
This pop-up menu is opened by clicking on the cabinet name shown at the top of the cabinet section. It allows you to select a speaker cabinet model. This section can be bypassed by selecting “No Speaker”.
speaker cabinet. Click on the values, enter a new value and press the [Enter] key.

DaTube

This effect emulates the characteristic warm, lush sound of a tube amplifier.
The following parameters are available:
Parameter Description
Drive Regulates the pre-gain of the “amplifier”. Use high values
Balance Controls the balance between the signal processed by the
Output Adjusts the post-gain, or output level, of the “amplifier”.
if you want an overdriven sound just on the verge of distortion.
Drive parameter and the dry input signal. For maximum drive effect, set this to its highest value.

Distortion

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Distortion will add crunch to your tracks.
The following parameters are available:
Parameter Description
Boost Increases the distortion amount.
Feedback Feeds part of the output signal back to the effect input,
Tone Lets you select a frequency range to which to apply the
Spatial Changes the distortion characteristics of the left and right
Output Raises or lowers the signal going out of the effect.
increasing the distortion effect.
distortion effect.
channel, thus creating a stereo effect.
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SoftClipper

This effect adds soft overdrive, with independent control over the second and third harmonic.
The following parameters are available:
Parameter Description
Input Regulates the pre-gain. Use high values if you want an
Mix Setting Mix to 0 means that no processed signal is added
Output Adjusts the post-gain, or output level.
Second Allows you to adjust the amount of the second harmonic
Third Allows you to adjust the amount of the third harmonic in
overdriven sound just on the verge of distortion.
to the original signal.
in the processed signal.
the processed signal.

Dynamics plug-ins

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

Compressor

Compressor reduces the dynamic range of the audio, mak­ing softer sounds louder or louder sounds softer, or both. Compressor features separate controls for threshold, ratio, attack, hold, release and make-up gain parameters. Com pressor features a separate display that graphically illus­trates the compressor curve shaped according to the Threshold and Ratio parameter settings. Compressor also features a Gain Reduction meter that shows the amount of gain reduction in dB, Soft knee/Hard knee compression modes and a program-dependent Auto feature for the Re lease parameter.
The following parameters are available:
Parameter Description
Threshold (-60 to 0 dB)
Ratio (1:1 to 8:1)
Soft Knee button
Make-up (0 to 24 dB or Auto mode)
Determines the level where Compressor “kicks in”. Signal levels above the set threshold are affected, but signal lev els below are not processed.
Sets the amount of gain reduction applied to signals over the set threshold. A ratio of 3:1 means that for every 3 the input level increases, the output level will increase by only 1
dB.
If this button is off, signals above the threshold are com­pressed instantly according to the set ratio (hard knee). When Soft Knee is activated, the onset of compression is more gradual, producing a less drastic result.
This parameter is used to compensate for output gain loss, caused by compression. If the Auto button is activated, the knob becomes dark and the output is automatically ad justed for gain loss.
dB
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Parameter Description
Attack (0.1 to
ms)
100
Hold (0 to
ms)
5000
Release (10 to 1000
ms or
Auto mode)
Analysis (0 to 100) (Pure Peak to Pure RMS)
Live button When this button is activated, the “look ahead” feature of
Determines how fast Compressor will respond to signals above the set threshold. If the attack time is long, more of the early part of the signal (attack) passes through unproc essed.
Sets the time the applied compression will affect the signal after exceeding the threshold. Short hold times are useful for “DJ-style” ducking, while longer hold times are required for music ducking, e. when working on a documentary film.
Sets the amount of time it takes for the gain to return to its original level when the signal drops below the threshold level. If the Auto button is activated, Compressor will auto matically find an optimal release setting that varies de­pending on the audio material.
Determines whether the input signal is analyzed according to peak or RMS values (or a mixture of both). A value of 0 is pure peak and 100 pure RMS. RMS mode operates using the average power of the audio signal as a basis, whereas Peak mode operates more on peak levels. As a general guideline, RMS mode works better on material with few transients such as vocals, and Peak mode better for per cussive material, with a lot of transient peaks.
Compressor is disengaged. Look ahead produces more accurate processing, but adds a certain amount of latency as a trade-off. When Live mode is activated, there is no la tency, which might be better for “live” processing.
g.
The compression can also be controlled from another signal source via the side-chain input. When the side-chain signal exceeds the threshold, the compression is triggered. For a description of how to set up side-chain routing, see the chapter “Audio effects” in the Operation Manual.

DeEsser

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A de-esser is used to reduce excessive sibilance, primarily for vocal recordings. Basically, it is a special type of com-
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pressor 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
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problem with sibilants.
The following parameters are available:
Parameter Description
Reduction Controls the intensity of the de-essing effect.
Threshold When the Auto Threshold option is deactivated, you can
Auto The Auto Threshold function automatically and continu-
Release Sets the amount of time it takes for the de-essing effect
Level meters Indicate the dB values of the input (IN) and output (OUT)
use this control to set a threshold for the incoming signal level, above which the plug-in starts to reduce the sibilants.
ally chooses an optimum threshold setting independent of the input signal. The Auto Threshold function does not work for low-level signals (< -30 such a file, set the threshold manually.
to return to zero when the signal drops below the tresh old value.
signals as well as the value by which the level of the sibi lant (or s-frequency) is reduced (GR). The gain reduction meter shows values between 0
dB (the s-frequency level is lowered by 20 dB).
-20
db peak level). To reduce the sibilants in
dB (no reduction) and
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Positioning the DeEsser in the signal chain
When recording a voice, the de-esser’s position in the signal chain is usually located after the microphone pre­amp and before a compressor/limiter. This keeps the compressor/limiter from unnecessarily limiting the overall signal dynamics.

EnvelopeShaper

EnvelopeShaper can be used to cut or boost the gain of the Attack and Release phase of audio material. You can either use the knobs or drag the breakpoints in the graph ical display to change parameter values. Be careful with levels when boosting the gain and if needed reduce the Output level to avoid clipping.
The following parameters are available:
Parameter Description
Attack (-20 to 20 dB) Changes the gain of the Attack phase of the sig-
Length (5 to 200 ms) Determines the length of the Attack phase.
Release (-20 to 20 dB) Changes the gain of the Release phase of the
Output (-24 to 12 dB) Sets the output level.
nal.
signal.

Expander

Expander reduces the output level in relation to the input level for signals below the set threshold. This is useful when you want to enhance the dynamic range or reduce the noise in quiet passages. You can either use the knobs or drag the breakpoints in the graphical display to change the Threshold and the Ratio parameter values.
The following parameters are available:
Parameter Description
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Threshold (-60 to 0 dB)
Ratio (1:1 to 8:1)
Soft Knee button
Attack (0.1 to 100
ms)
Hold (0 to
ms)
2000 Release
(10 to
ms or
1000 Auto mode)
Determines the level where expansion “kicks in”. Signal levels below the set threshold are affected, but signal lev els above are not processed.
Determines the amount of gain boost applied to signals below the set threshold.
If this button is off, signals below the threshold are ex­panded instantly according to the set ratio (“hard knee”). When Soft Knee is activated, the onset of expansion is more gradual, producing a less drastic result.
Determines how fast Expander responds to signals below the set threshold. If the attack time is long, more of the early part of the signal (attack) passes through unprocessed.
Sets the time the applied expansion will affect the signal below the Threshold.
Sets the amount of time it takes for the gain to return to its original level when the signal exceeds the threshold level. If the Auto button is activated, Expander will automatically find an optimal release setting that varies depending on the audio material.
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Parameter Description
Analysis (0 to 100) (Pure Peak to Pure RMS)
Live button When this button is activated, the “look ahead” feature of
Determines whether the input signal is analyzed according to peak or RMS values (or a mixture of both). A value of 0 is pure peak and 100 pure RMS. RMS mode operates using the average power of the audio signal as a basis, whereas Peak mode operates more on peak levels. As a general guideline, RMS mode works better on material with few transients such as vocals, and Peak mode better for per cussive material, with a lot of transient peaks.
Expander is disengaged. Look ahead produces more ac curate processing, but adds a certain amount of latency as a trade-off. When Live mode is activated, there is no la tency, which might be better for “live” processing.
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The expansion can also be controlled from another sig­nal source via the side-chain input. When the side-chain signal exceeds the threshold, the expansion is triggered. For a description of how to set up side-chain routing, see the chapter “Audio effects” in the Operation Manual.

Gate

Parameter Description
Filter buttons (LP, BP, and HP)
Side-Chain button
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Center (50 Hz to 20000
­Q-Factor (0.01
to 10000)
Monitor button Allows you to monitor the filtered signal.
Attack (0.1 to
ms)
1000
Hold (0 to 2000 ms)
Release (10 to 1000 ms or Auto mode)
Analysis (0 to 100) (Pure Peak to Pure RMS)
Live button When this button is activated, the “look ahead” feature of
When the Side-Chain button (see below) is activated, you can use these buttons to set the filter type to either low-pass, band-pass, or high-pass.
This button (below the Center knob) activates the side­chain filter. The input signal can then be shaped accord ing to set filter parameters. Internal side-chaining can be useful for tailoring how the Gate operates.
When the Side-Chain button is activated, this sets the
Hz)
center frequency of the filter.
When the Side-Chain button is activated, this sets the resonance of the filter.
Sets the time it takes for the gate to open after being trig­gered. If the Live button (see below) is deactivated, it en­sures that the gate will already be open when a signal above the threshold level is played back. Gate manages this by “looking ahead” in the audio material, checking for signals loud enough to pass the gate.
Determines how long the gate stays open after the signal drops below the threshold level.
Sets the amount of time it takes for the gate to close (af­ter the set hold time). If the Auto button is activated, Gate will find an optimal release setting, depending on the au dio material.
Determines whether the input signal is analyzed according to Peak or RMS values (or a mixture of both). A value of 0 is pure Peak and 100 pure RMS. RMS mode operates using the average power of the audio signal as a basis, whereas Peak mode operates more on peak levels. As a general guideline, RMS mode works better on material with few transients such as vocals, and Peak mode better for per cussive material, with a lot of transient peaks.
Gate is disengaged. Look ahead produces more accurate processing, but adds a certain amount of latency as a trade-off. When Live mode is activated, there is no la tency, which might be better for “live” processing.
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Gating, or noise gating, silences audio signals below a set threshold level. As soon as the signal level exceeds the set threshold, the gate opens to let the signal through.
The following parameters are available:
Parameter Description
Threshold (-60 to 0 dB)
State LED Indicates whether the gate is open (LED lights up in
Determines the level where Gate is activated. Signal lev­els above the set threshold trigger the gate to open, and signal levels below the set threshold will close the gate.
green), closed (LED lights up in red) or something in be tween (LED lights up in yellow).
The included effect plug-ins
The gate can also be controlled from another signal source via the side-chain input. When the side-chain sig nal exceeds the threshold, the gate opens. For a descrip­tion of how to set up side-chain routing, see the chapter “Audio effects” in the Operation Manual.
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Limiter

Limiter is designed to ensure that the output level never exceeds a set output level, to avoid clipping in following devices. Limiter can adjust and optimize the Release pa­rameter automatically according to the audio material, or it can be set manually. Limiter also features separate meters for the input, output and the amount of limiting (middle meters).
The following parameters are available:
Parameter Description
Input (-24 to +24 dB)
Output (-24 to +6 dB)
Release (0.1 to 1000 ms or Auto mode)
Allows you to adjust the input gain.
Determines the maximum output level.
Sets the amount of time it takes for the gain to return to its original level. If the Auto button is activated, Limiter will automatically find an optimal release setting that var ies depending on the audio material.

Maximizer

Maximizer is used to raise the loudness of audio material without the risk of clipping. Optionally, there is a soft clip function that removes short peaks in the input signal and introduces a warm tube-like distortion to the signal.
The following parameters are available:
Parameter Description
Output (-24 to +6 dB)
Optimize (0 to 100)
Soft Clip button
-
Determines the maximum output level. Should normally be set to 0 (to avoid clipping).
Determines the loudness of the signal.
When this button is activated, Maximizer starts limiting (or clipping) the signal “softly”, at the same time generating harmonics which add a warm, tube-like characteristic to the audio material.
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MIDI Gate

Gating, in its fundamental form, silences audio signals be­low a set threshold level. When a signal rises above the set level, the gate opens to let the signal through while signals below the set level are cut off. MIDI Gate, how­ever, is not triggered by threshold levels, but MIDI notes. Hence it needs both audio and MIDI data to function.
Setting up
To set up MIDI Gate, proceed as follows:
1. Select the audio to be affected by MIDI Gate.
This can be audio material from any audio track, or even a live audio input (provided you have a low latency audio card).
2. Select MIDI Gate as an insert effect for the audio track.
The MIDI Gate control panel opens.
3. Select a MIDI track to control the MIDI Gate effect.
This can be an empty MIDI track or a MIDI track containing data, it does not matter. However, if you wish to use MIDI Gate in realtime – as opposed to using a recorded part – the track has to be selected for the effect to re ceive the MIDI output.
4. Open the Output Routing pop-up menu for the MIDI track and select the MIDI Gate option.
The MIDI output from the track is now routed to the MIDI Gate effect.
What to do next depends on whether you are using live or recorded audio and whether you are using realtime or re corded MIDI. We will assume for the purposes of this manual that you are using recorded audio, and play the MIDI in realtime.
5. Make sure the MIDI track is selected, and start play­back.
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6. Play a few notes on your MIDI keyboard.
As you can hear, the audio track material is affected by what you play on your MIDI keyboard.
The following MIDI Gate parameters are available:
Parameter Description
Attack Determines how long it takes for the gate to open after
Hold Regulates how long the gate remains open after a note-
Release Determines how long it takes for the gate to close (in ad-
Note To Attack Determines to which extent the velocity values of the MIDI
Note To Release
Velocity To VCA
Hold Mode Use this switch to set the Hold Mode. In Note-On mode,
receiving a signal that triggers it.
on or note-off message (see Hold Mode below).
dition to the value set with the Hold parameter).
notes affect the attack. The higher the value, the more the attack time increases with high note velocities. Negative values give shorter attack times with high velocities. If you do not wish to use this parameter, set it to the 0 position.
Determines to which extent the velocity values of the MIDI notes affect the release. The higher the value, the more the release time increases. If you do not wish to use this parameter, set it to the 0 position.
Controls to which extent the velocity values of the MIDI notes determine the output volume. At a value of 127 the volume is controlled entirely by the velocity values, and at a value of 0 the velocities have no effect on the volume.
the gate only remains open for the time set with the Hold and Release parameters, regardless of the length of the MIDI note that triggered the gate. In Note-Off mode, the gate remains open for as long as the MIDI note plays, and then the Hold and Release parameters are applied.
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MultibandCompressor

The MultibandCompressor allows a signal to be split into a maximum of four frequency bands, each with its own freely adjustable compressor characteristic. The signal is processed on the basis of the settings that you have made in the Frequency Band and Compressor sections. You can specify the level, bandwidth and compressor charac teristics for each band by using the various controls.
The Frequency Band editor
The Frequency Band editor in the upper half of the panel is where you set the width of the frequency bands as well as their level after compression. Two value scales and a num ber of handles are available. The vertical value scale to the left shows the input gain level of each frequency band. The horizontal scale shows the available frequency range.
The handles provided in the Frequency Band editor can be dragged with the mouse. You use them to set the cor ner frequency range and the input gain levels for each fre­quency bands.
• The handles at the sides are used to define the frequency range of the different frequency bands.
• By using the handles on top of each frequency band, you can cut or boost the input gain by +/- 15 dB after compression.
Soloing frequency bands
A frequency band can be soloed using the S button in each compressor section. Only one band can be soloed at a time.
Using the Compressor section
By moving breakpoints or using the corresponding knobs, you can specify the Threshold and Ratio. The first break­point from which the line deviates from the straight diago­nal will be the threshold point.
For each of the four bands the following compressor pa­rameters are available:
Parameter Description
-
Threshold (-60 to 0 dB)
Ratio (1000 to 8000) (1:1 to 8:1)
Attack (0.1 to 100 ms)
Release (10 to 1000 ms or Auto mode)
Determines the level where Compressor “kicks in”. Sig­nal levels above the set threshold are affected, but signal levels below are not processed.
Determines the amount of gain reduction applied to sig­nals over the set threshold. A ratio of 3000 (3:1) means that for every 3 level increases by only 1
Determines how fast the compressor responds to sig­nals above the set threshold. If the attack time is long, more of the early part of the signal (attack) will pass through unprocessed.
Sets the amount of time it takes for the gain to return to its original level when the signal drops below the thresh old level. If the Auto button is activated, the compressor will automatically find an optimal release setting that var ies depending on the audio material.
dB the input level increases, the output
dB.
The Output control
­The Output knob controls the total output level that the
MultibandCompressor passes on to Nuendo. The range is from -24 to +24
dB.
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Bypassing frequency bands
Each frequency band can be bypassed using the B button in each compressor section.
The included effect plug-ins
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VintageCompressor

Limiter
Module Configuration
Gate Compressor
This is modelled after vintage type compressors. This com­pressor features separate controls for input and output gain, attack, and release. In addition, there is a Punch mode which preserves the attack phase of the signal and a pro­gram-dependent Auto feature for the Release parameter.
The available parameters work as follows:
Parameter Description
Input (-24 to 48 dB)
Output (-48 to 24 dB)
Attack (0.1 to 100 ms)
Punch (On/Off)
Release (10 to 1000 ms or Auto mode)
The compression can also be controlled from another signal source via the side-chain input. When the side-chain signal exceeds the threshold, the compression is triggered. For a description of how to set up side-chain routing, see the chapter “Audio effects” in the Operation Manual.
In combination with the Output setting, this parameter determines the compression amount. The higher the in put gain setting and the lower the output gain setting, the more compression is applied.
Sets the output gain.
Determines how fast the compressor responds. If the at­tack time is long, more of the early part of the signal (at­tack) passes through unprocessed.
When this is activated, the early attack phase of the sig­nal is preserved, retaining the original “punch” in the au­dio material, even with short Attack settings.
Sets the amount of time it takes for the gain to return to its original level. If the Auto button is activated, Vintage Compressor will automatically find an optimal release set ting that varies depending on the audio material.

VSTDynamics

VSTDynamics is an advanced dynamics processor. It com­bines three separate processors: Gate, Compressor and Limiter, covering a variety of dynamic processing functions. The window is divided into three sections, containing con trols and meters for each processor.
Activating the individual processors
You activate the individual processors using the buttons at the bottom of the plug-in panel.
The Gate section
Gating, or noise gating, is a method of dynamic process­ing that silences audio signals below a set threshold level. As soon as the signal level exceeds the set threshold, the
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gate opens to let the signal through. The Gate trigger in put can also be filtered using an internal side-chain.
The following parameters are available:
Parameter Description
Threshold (-60 to 0 dB)
State LED Indicates whether the gate is open (LED lights up in
Side-Chain
­button
LP (low-pass), BP (band-pass), HP (high-pass)
Center (50 to
Hz)
22000
Determines the level where Gate is activated. Signal lev­els above the set threshold trigger the gate to open, and signal levels below the set threshold close the gate.
green), closed (LED lights up in red) or something in be tween (LED lights up in yellow).
This button activates the internal side-chain filter. You can use this to filter out parts of the signal that might other wise trigger the gate in places you not want it to, or to boost frequencies you wish to accentuate, allowing for more control over the gate function.
These buttons set the basic filter mode.
Sets the center frequency of the filter.
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Parameter Description
Q-Factor (0.001 to
10000) Monitor
(On/Off) Attack
(0.1 to 100 ms) Hold
(0 to 2000 ms) Release
(10 to 1000 ms or Auto mode)
Sets the resonance or width of the filter.
Allows you to monitor the filtered signal.
Sets the time it takes for the gate to open after being triggered.
Determines how long the gate stays open after the sig­nal drops below the threshold level.
Sets the amount of time it takes for the gate to close (after the set hold time). If the Auto button is activated, Gate will find an optimal release setting, depending on the audio material.
The Compressor section
The compressor reduces the dynamic range of the audio, making softer sounds louder or louder sounds softer, or both. It works like a standard compressor with separate controls for threshold, ratio, attack, release and make-up gain. The compressor features a separate display that graphically illustrates the compressor curve shaped ac cording to the Threshold, Ratio and Make-Up Gain pa­rameter settings. It also features Gain Reduction meters and a program-dependent Auto feature for the Release parameter.
The available parameters work as follows:
Parameter Description
Threshold (-60 to 0 dB)
Ratio (1:1 to 8:1)
Make-Up (0 to 24 dB)
Attack (0.1 to 100 ms)
Determines the level where the compressor “kicks in”. Signal levels above the set threshold are affected, but signal levels below are not processed.
Determines the amount of gain reduction applied to sig­nals above the set threshold. A ratio of 3:1 means that for
dB the input level increases, the output level in-
every 3 creases by only 1 dB.
This parameter is used to compensate for output gain loss, caused by compression. When the Auto button is activated, gain loss is being compensated automatically.
Determines how fast the compressor responds to signals above the set threshold. If the attack time is long, more of the early part of the signal (attack) passes through un processed.
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Parameter Description
Release (10 to 1000 ms or Auto mode)
Graphical display
Sets the amount of time it takes for the gain to return to its original level when the signal drops below the thresh old level. If the Auto button is activated, the compressor will automatically find an optimal release setting that var ies depending on the audio material.
Use the graphical display to graphically set the Threshold and Ratio values. To the left and right of the graphical dis play you will find two meters that show the amount of gain reduction in dB.
The Limiter section
The limiter is designed to ensure that the output level never exceeds a set threshold, to avoid clipping in follow ing devices. Conventional limiters usually require very ac­curate setting up of the attack and release parameters to prevent the output level from going beyond the set thresh old level. The limiter adjusts and optimizes these parame­ters automatically according to the audio material. You can also adjust the Release parameter manually.
The following parameters are available:
Parameter Description
Output (-24 to +6 dB)
Soft Clip button
Release (10 to 1000 ms or Auto mode)
Determines the maximum output level. Signal levels above the set threshold are affected, but signal levels be low are left unaffected.
If this button is activated, the limiter acts differently. When the signal level exceeds -6 clipping) the signal “softly”, at the same time generating harmonics which add a warm, tube-like characteristic to the audio material.
Sets the amount of time it takes for the gain to return to its original level when the signal drops below the thresh old level. If the Auto button is activated, the limiter will au­tomatically find an optimal release setting that varies depending on the audio material.
dB, Soft Clip starts limiting (or
The Module Configuration button
Using the Module Configuration button in the bottom right corner of the plug-in panel, you can set the signal flow or der for the three processors. Changing the order of the pro­cessors can produce different results, and the available options allow you to quickly compare what works best for a given situation. Simply click the Module Configuration but ton to change to a different configuration. There are three routing options:
• C-G-L (Compressor-Gate-Limit)
• G-C-L (Gate-Compressor-Limit)
• C-L-G (Compressor-Limit-Gate)
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The included effect plug-ins
18

EQ plug-ins

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

GEQ-10/GEQ-30

These graphic equalizers are identical in every respect ex­cept for the number of available frequency bands (10 and 30 respectively). Each band can be cut or boosted by up to
dB, allowing for fine control of the frequency response.
12 In addition there are several preset modes available which can add “color” to the sound of the GEQ-10/GEQ-30.
You can draw response curves in the main display by click-dragging with the mouse.
Note that you have to click on one of the sliders first before dragging across the display. You can also point and click to change individual fre quency bands, or enter values numerically by clicking on a gain value at the top of the display.
At the bottom of the window the individual frequency bands are shown in Hz.
At the top of the display the amount of cut/boost is shown in dB.
Apart from the frequency bands, the following parameters are available:
Parameter Description
Output Controls the overall gain of the equalizer.
Flatten button Resets all the frequency bands to 0 dB.
Range Allows you to relatively adjust how much a set curve cuts
or boosts the signal. If the Range parameter is turned fully clockwise, the range is +/-12
dB.
Parameter Description
Invert button Inverts the current response curve.
Mode pop-up menu
The filter mode set here determines how the various fre­quency band controls interact to create the response curve, see below.
About the filter modes
On the pop-up menu in the lower right corner there are several different EQ modes available. These modes can add color or character to the equalized output in various ways. Here follow brief descriptions of the filter modes:
• True Response – serial filters with accurate frequency
response.
• Digi Standard – resonance of last band depends on sample
rate.
• Variable Q – parallel filters where the resonance depends on
the amount of gain. Musical sounding.
• Constant Q u – parallel filters where the resonance of the first
and last bands depends on the sample rate (u=unsymmetric).
• Constant Q s – parallel filters where the resonance is raised
when boosting the gain and vice versa (s=symmetric).
• Resonant – serial filters where a gain increase of one band will
lower the gain in adjacent bands.

StudioEQ

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This is a high-quality 4-band parametric stereo equalizer with two fully parametric mid-range bands. The low and high bands can act as either shelving filters (three types), or as a Peak (band-pass) or Cut (low-pass/high-pass) filter.
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Making settings
1. Click the corresponding On button on the left of the plug-in panel to activate any or all of the 4 equalizer bands (Low, Mid 1, Mid 2, and High).
When a band is activated, the corresponding EQ point appears in the EQ curve display.
2. Set the parameters for an activated EQ band.
This can be done in several ways:
• By using the knobs.
• By clicking on the numeric values and typing in new values.
• By using the mouse to drag points in the EQ curve display.
When using the mouse to change the parameter settings, the following modifier keys can be used:
Modifier key Description
When no modifier key is pressed and you drag an EQ
[Shift] Keep the [Shift] key pressed and drag the mouse to
[Alt]/[Option] Keep the [Alt]/[Option] key pressed and drag the
[Ctrl]/[Command] Keep the [Ctrl]/[Command] key pressed and drag the
point in the display, the Gain and Frequency parame ters are adjusted simultaneously.
change the Q-factor of the corresponding EQ band.
mouse to change the frequency of the corresponding EQ band.
mouse to change the gain value of the corresponding EQ band.
The following parameters are available:
Parameter Description
Band 1 Gain (-20 to +24 dB)
Band 1 Inv button Inverts the gain value of the filter. Use this button to fil-
Band 1 Freq (20 to 2000 Hz)
Band 1 Q-Factor (0.5 to 10)
Band 1 Filter mode
Sets the amount of cut/boost for the low band.
ter out unwanted noise. While looking for the frequency to omit, it sometimes helps to boost it in the first place (set the filter to positive gain). After you have found it, you can use the Inv button to cancel it out.
Sets the frequency of the low band.
Controls the width or resonance of the low band.
For the low band, you can select between three types of shelving filters, a Peak (band-pass), and a Cut (low­pass/high-pass) filter. When Cut mode is selected, the Gain parameter is fixed.
-Shelf I adds resonance in the opposite gain direction slightly above the set frequency.
-Shelf II adds resonance in the gain direction at the set frequency.
-Shelf III is a combination of Shelf I and II.
Parameter Description
Band 2 Gain (-20 to +24 dB)
Band 2 Inv button Inverts the gain value of the filter (see the description
Band 2 Freq (20 to 20000 Hz)
Band 2 Q-Factor (0.5 to 10)
Band 3 Gain (-20 to +24 dB)
Band 3 Inv button Inverts the gain value of the filter (see the description
Band 3 Freq (20 to 20000 Hz)
Band 3 Q-Factor (0.5 to 10)
Band 4 Inv button Inverts the gain value of the filter (see the description
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Band 4 Gain (-20 to +24 dB)
Band 4 Freq (200 to 20000 Hz)
Band 4 Q-Factor (0.5 to 10)
Band 4 Filter mode
Output (-24 to +24 dB)
Auto Gain button When this button is activated, the gain is automatically
Sets the amount of cut/boost for the mid 1 band.
of the Invert button for Band 1).
Sets the center frequency of the mid 1 band.
Sets the width of the mid 1 band: the higher this value, the “narrower” the bandwidth.
Sets the amount of cut/boost for the mid 2 band.
of the Invert button for Band 1).
Sets the center frequency of the mid 2 band.
Sets the width of the mid 2 band: the higher this value, the “narrower” the bandwidth.
of the Invert button for Band 1).
Sets the amount of cut/boost for the high band.
Sets the frequency of the high band.
Controls the width or resonance of the high band.
For the high band, you can select between three types of shelving filters, a Peak, and a Cut filter. When Cut mode is selected, the Gain parameter is fixed.
-Shelf I adds resonance in the opposite gain direction slightly below the set frequency.
-Shelf II adds resonance in the gain direction at the set frequency.
-Shelf III is a combination of Shelf I and II.
This knob on the top right of the plug-in panel allows you to adjust the overall output level.
adjusted, keeping the output level constant regardless of the EQ settings.
The included effect plug-ins
20

Filter plug-ins

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

DualFilter

PostFilter

The DualFilter effect filters out certain frequencies while allowing others to pass through.
The following parameters are available:
Parameter Description
Position Sets the filter cutoff frequency. If you set this to a nega-
Resonance Sets the sound characteristic of the filter. With higher val-
tive value, DualFilter will act as a low-pass filter. Positive values cause DualFilter to act as a high-pass filter.
ues, a ringing sound is heard.
The PostFilter is the filter plug-in to use if you are working on a post-production mix, but of course you can use it in music production, too, as an alternative to complex EQ configurations. It allows quick and easy filtering of un
­wanted frequencies, creating room for the important sounds in your mix.
The PostFilter plug-in combines a low-cut filter, a notch fil­ter and a high-cut filter. You can either make settings by dragging the handles in the graphical display, or by adjust­ing one of the controls below the display section.
Use the Preview buttons to compare the result of your fil­tering and the filtered frequencies.
The following parameters are available:
Parameter Description
Level meter The meter to the right of the EQ display shows the out-
Low Cut Freq (20 Hz to 1 kHz, or Off)
Low Cut Slope pop-up menu
Low Cut Preview button
put level, giving you an indication of how the filtering af­fects the overall level of the edited event.
Use this low-cut filter to eliminate low-frequency noise. The filter is off when the handle/knob is moved all the way to the left.
Allows you to choose a slope value for the low-cut filter.
Use the Preview button (found between the Low Cut Freq button and the graphical display) to switch the fil ter to a complementary high-cut filter. This deactivates any other filters, allowing you to listen only to the fre quencies you want to filter out.
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Parameter Description
Notch Freq Sets the frequency of the notch filter.
Notch Gain Allows you to adjust the gain of the selected frequency.
Notch Gain Invert button
Notch Q-Factor Sets the width of the notch filter.
Notch Preview button
Notches buttons (1, 2, 4, 8)
High Cut Freq (3 Hz to 20 kHz, or Off)
High Cut Slope pop-up menu
High Cut Preview button
Use positive values to identify the frequencies that you want to filter out.
Inverts the gain value of the notch filter. Use this button to filter out unwanted noise. While looking for the fre quency to omit, it sometimes helps to boost it in the first place (set notch filter to positive gain). After you have found it, you can use the Invert button to cancel it out.
Use the Preview button (found between the notch filter buttons and the graphical display) to create a band­pass filter with the peak filter’s frequency and Q. This deactivates any other filters, allowing you to listen only to the frequencies you want to filter out.
These buttons add additional notch filters to filter out harmonics.
Use this high-cut filter to eliminate high-frequency noise. Filter is Off when the handle/knob is moved all the way to the right.
Allows you to choose a slope value for the high-cut fil­ter.
Use the Preview button (found between the High Cut Freq button and the graphical display) to switch the fil ter to a complementary low-cut filter. This deactivates any other filters, allowing you to listen only to the fre quencies you want to filter out.
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Q
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Q is a high-quality 4-band parametric stereo equalizer with two fully parametric mid-range bands. The low and
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high bands can act as either standard shelving filters or fixed-gain high/low-cut filters.
Making settings
1. Click the corresponding On button below the EQ curve display to activate any or all of the Low, Mid 1, Mid 2, or High equalizer bands.
When a band is activated, a corresponding EQ point appears in the EQ curve display.
2. Set the parameters for an activated EQ band.
This can be done in several ways:
• By using the knobs.
• By clicking a value field and entering values numerically.
• By using the mouse to drag points in the EQ curve display
window.
By using this method, you control both the Gain and Frequency parame­ters simultaneously. The knobs turn accordingly when you drag points. In addition, if the Mid 1 and Mid 2 bands (M1 and M2) are activated there will be two points on each side of the Gain/Frequency point that control the width (Q) parameter. If you press [Shift] while dragging, values can be set in finer increments.
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The following parameters are available:
Parameter Description
Low Freq (20 to 2000 Hz)
Low Gain (-20 to +20 dB)
Low Cut If this button is activated for the low band, it acts as a
Mid 1 Freq (20 to 20000 Hz)
Mid 1 Gain (+/-20 dB)
Mid 1 Width (0.05 to 5.00 Octaves)
Mid 2 Freq (20 to 20000 Hz)
Mid 2 Gain (-20 to +20 dB)
Mid 2 Width (0.05 to 5.00 Octaves)
High Freq (200 to 20000 Hz)
High Gain (-20 to +20 dB)
High Cut If this button is activated for the High band, it acts as a
Output slider (-20 to +20 dB)
Left/Stereo/Right/ Mono modes
Sets the frequency of the low band.
Sets the amount of cut/boost for the low band.
low cut filter, and the Gain parameter is fixed.
Sets the center frequency of the mid 1 band.
Sets the amount of cut/boost for the mid 1 band.
Sets the width of the mid 1 band in octaves. The lower this value, the “narrower” the bandwidth.
Sets the center frequency of the mid 2 band.
Sets the amount of cut/boost for the mid 2 band.
Sets the width of the mid 2 band in octaves. The lower this value, the “narrower” the bandwidth.
Sets the frequency of the high band.
Sets the amount of cut/boost for the high band.
high cut filter, and the Gain parameter is fixed.
Allows you to adjust the overall output level.
For stereo signals you can set independent curves for the left and right channels by clicking the correspond ing button. If the Stereo button is activated, the curve is applied to both channels. When channel-independent curves have been set, the curves for the left and right channel are colored green and red, respectively. The channel that is not selected is shown with a dotted curve. If you activate the Stereo button after independent curves have been set, the active curve is applied to both channels. Mono mode is automatically activated for mono sig­nals and is otherwise unavailable.

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
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down the vertical axis, or directly set by clicking in an empty grid box. By click-dragging left or right, consecutive step entries are set at the pointer position.
The horizontal axis shows the pattern steps 1 to 16 from left to right, and the vertical axis determines the (relative) filter cutoff frequency and resonance settings.
The higher up on the vertical axis a step value is entered, the higher the relative filter cutoff frequency or filter resonance setting.
By starting playback and editing the patterns for the cut­off and resonance parameters, you can hear how your filter patterns affect the sound source connected to StepFilter.
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23
Selecting new patterns
Created patterns are saved with the project, and up to 8 different cutoff and resonance patterns can be saved inter­nally.
Both the cutoff and resonance settings are saved together in the 8 pattern slots.
Use the Pattern Selector below the Resonance grid to select a new pattern.
New patterns are all set to the same step value by default.

ToneBooster

Using pattern copy and paste to create variations
You can use the Copy and Paste buttons below the Pat­tern Selector to copy a pattern to another pattern slot, which is useful for creating variations on a pattern.
Select the pattern you wish to copy, click the Copy but­ton, select another pattern slot, and click Paste.
The pattern is copied to the new slot, and can now be edited to create variations using the original pattern as a starting point.
StepFilter parameters
Parameter Description
Base Cutoff Sets the base filter cutoff frequency. Values set in the
Base Resonance Sets the base filter resonance. Values set in the Reso-
Glide This will apply glide between the pattern step values,
Filter mode Use this slider to select a filter mode: low-pass (LP),
Sync button When the Sync button to the right of the Sync pop-up
Sync pop-up menu (1/1 to 1/32, straight, triplet, or dotted)
Output slider Sets the overall volume.
Mix slider Adjusts the mix between dry and processed signal.
Cutoff grid are relative to the Base Cutoff value.
nance grid are relative to the Base Resonance value. Note that very high Base Resonance settings can pro duce loud ringing effects at certain frequencies.
causing values to change more smoothly.
band-pass (BP), or high-pass (HP) (from left to right).
menu is activated (yellow), the pattern playback is syn chronized with the project tempo.
Use this pop-up menu to set the pattern beat resolution, i.
e. what note values the pattern will play in relation to the
tempo.
ToneBooster is a filter that allows you to raise the gain in a selected frequency range. It is particularly useful when in serted before AmpSimulator in the plug-in chain (see
“AmpSimulator” on page 9), greatly enhancing the tonal
varieties available.
The following parameters are available:
Parameter Description
Tone Sets the center filter frequency.
Gain Allows you to adjust the gain of the selected frequency
Width Sets the resonance of the filter.
Mode selector Sets the basic operational mode of the filter; Peak or
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range by up to 24
Band Mode.
dB.
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The included effect plug-ins
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Tonic

Tonic is a versatile and powerful analog modeling filter plug-in based on the filter design of the Monologue mono­phonic synthesizer. Its variable characteristics plus the powerful modulation functions make it an excellent choice for all current music styles. Designed to be more a cre ative tool rather than a tool to fix audio problems, it can add color and punch to your tracks while being light on CPU usage.
Tonic has the following properties:
Dynamic multimode analog modeling filter (mono/stereo).
24 dB low-pass, 18 dB low-pass, 12 dB low-pass, 6 dB
low-pass, 12 dB band-pass, and 12 dB high-pass modes.
Adjustable drive and resonance up to self-oscillation.
Envelope follower for dynamic filter control with an
audio signal.
Audio and MIDI trigger modes.
Powerful step LFO with smoothing and morphing.
X/Y matrix pad for additional realtime modulation with
access to all Tonic parameters.
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Filter
In the Filter section at the center of the plug-in panel, the following parameters are available:
Parameter Description
Mode pop-up menu
Cutoff Sets the filter cutoff frequency. How this parameter oper-
Res Changes the resonance of the multi-mode filter. Full res-
Drive Adds a soft, tube-like saturation to the sound. As with an
Mix Sets the balance between dry and effect signal.
Channel selector (Ch.).
Sets the filter type. Available filter types are: 24 dB low-
dB low-pass, 12 dB low-pass, 6 dB low-pass,
pass, 18
dB band-pass, and 12 dB high-pass.
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ates is governed by the filter type.
onance puts the filter into self-oscillation.
analog filter, the amount of saturation also depends on the input signal level.
Allows you to choose between mono or stereo operation. When set to mono, the output signal of Tonic is mono re gardless of the input signal.
Env Mod
In the Env Mod section, the following parameters are available:
Parameter Description
Mode pop-up menu
Attack Controls the attack time of the envelope. Higher attack
Release Controls the release time of the envelope. Higher release
Depth Controls the amount of envelope control applied to the
LFO Mod Using this parameter, the envelope level modulates the
Tonic offers three types of envelope modulation: “Follow” tracks the input signal’s volume envelope for 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 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 Routing pop-up menu for the track.
times result in slower rise times when the envelope is trig gered.
times result in slower envelope tails.
filter cutoff level.
LFO speed. A rather stunning effect.
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X/Y Pad
In the X/Y Pad at the bottom left of the plug-in panel, the following parameters are available:
Parameter Description
X Par pop-up menu
Y Par pop-up menu
XY Pad Use the mouse to control any two of Tonic’s parameters
Sets the parameter to be modulated on the x-axis of the XY Pad. All of Tonic’s parameters are available as desti nations.
Sets the parameter to be modulated on the y-axis of the XY Pad.
in combination. By moving the mouse horizontally you control the x parameter, and by moving it vertically you control the y parameter. You can also record controller movements as automation data.
LFO Mod
In the LFO Mod section, the following parameters are available:
Parameter Description
Mode pop-up menu
Depth Controls the amount of LFO modulation applied to the fil-
Rate Controls the speed of the LFO modulation. The LFO rate
Smooth Controls the smoothing of the LFO steps. This works like
Morph Controls the playback value of the LFO step sequencer. It
Steps pop-up menu
Presets pop-up menu
Step Matrix Click into the Step Matrix to set the level for each of the
Sets the direction of the step LFO modulation. The avail­able modes are: Forward, Reverse, Alternating, and Ran­dom.
ter cutoff level.
is always in sync with the project tempo. An example: at a rate of 4.00 steps per beat in a 4/4 time signature, the step sequencer advances in 16th notes. At a rate of 4.00 beats per step in a 4/4 time signature the LFO advances only one step per bar. Note that the current LFO Rate is shown in the field be­low the Env Mod section.
a glide effect applied to the filter cutoff.
makes the LFO steps drift about randomly. Experiment freely with the Morph parameter. As you return the knob to its zero position, the step pattern returns to its original setting.
Sets the number of steps played in sequence. Deacti­vated steps are grayed out in the Step Matrix.
Offers a number of step LFO waveform patterns. Choices include: Sine, Sine+, Cosine, Triangle, Sawtooth, Square, Random, and User (which is the pattern saved with the respective program).
16 LFO steps. A higher amount results in a deeper filter cutoff modulation. Click and drag along the matrix to “draw” a waveform.

WahWah

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WahWah is a variable slope band-pass filter that can be auto-controlled by a side-chain signal or via MIDI model ing the well-known analog pedal effect (see below). You can independently specify the frequency, width and the gain for the Lo and Hi Pedal positions. The crossover point between the Lo and Hi Pedal positions lies at 50.
The following parameters are available:
Parameter Description
Pedal Controls the filter frequency sweep.
Pedal Control (MIDI) pop-up menu
Freq Lo/Hi Set the frequency of the filter for the Lo and Hi Pedal po-
Width Lo/Hi Set the width (resonance) of the filter for the Lo and Hi
Gain Lo/Hi Set the gain of the filter for the Lo and Hi Pedal positions.
Filter Slope selector
When the side-chain input is activated, a signal routed to the side-chain input of the effect can control the Pedal parameter. The louder the signal, the more the filter fre­quency (Pedal) is raised so that the plug-in acts as an “auto-wha” effect. For a description of how to set up side­chain routing, see the chapter “Audio effects” in the Oper­ation Manual.
MIDI control
For realtime MIDI control of the Pedal parameter, MIDI must be directed to the WahWah plug-in.
Whenever WahWah has been added as an insert effect (for an audio track or an FX channel), it is available on the Output Routing pop-up menu for MIDI tracks.
If WahWah is selected on the Output Routing menu, MIDI data is directed to the plug-in from the selected track.
Allows you to choose the MIDI controller that is used to control the plug-in. Set this to “Automation” if you do not want to use MIDI realtime control.
sitions.
Pedal positions.
Allows you to choose between two filter slope values:
dB or 12 dB.
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The included effect plug-ins
26

Generator plug-ins

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

SMPTEGenerator

This plug-in is not a real audio effect. It sends out SMPTE timecode to an audio output, allowing you to synchronize other equipment to Nuendo (provided that the equipment can sync directly to SMPTE timecode). This can be very useful if you do not have access to a MIDI-to-timecode converter.
The following parameters are available:
Parameter Description
Main timecode display
Frame rate display and pop-up menu
This display shows the current timecode. When “Link to Transport” is deactivated, the generator is in “free run” mode. You can then use the timecode dis play to set the SMPTE start time. When “Link to Transport” is activated, you cannot change any of the values. This display shows the current time code in sync with the Transport panel. Where applicable, the offset defined in the offset timecode display is taken into account (see below).
The frame rate shown to the right of the timecode display defaults to the frame rate set in the Project Setup dialog. To generate timecode in a different frame rate (e. stripe a tape), select another format on the pop-up menu (only available if “Link to Transport” is deactivated). Note that for another device to synchronize correctly to Nuendo, the same frame rate has to be set in the Project Setup dialog, the SMPTE Generator and the receiving device.
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Parameter Description
Offset timecode display
Generate Code button
Link to Transport button
Timecode in Still Mode button
This display is only available if “Link to Transport” is acti­vated. It allows you to set an offset with regard to the time­code used by Nuendo. The offset affects the generated SMPTE signal, the current cursor position in Nuendo re mains unaffected. For example, use this when playing back video using an external device, where the video starts at a different time code position than in Nuendo. A scenario could be as fol­lows: Your have placed the same video several times on the Nuendo timeline, in order to record different audio versions for that video one after the other. However, since video playback is done via an external machine (replaying the same video) you need an offset to match the different timecode positions in Nuendo with the (unchanging) start position on the external machine.
When you activate this button, the plug-in generates SMPTE timecode in “free run” mode, meaning that it out puts continuous timecode independent from the Trans­port panel. Use this mode if you want to stripe tape with SMPTE.
When you activate this button, the timecode is synchro­nized to the Transport panel.
When you activate this button, the plug-in also generates SMPTE timecode in stop mode. However, note that this will not be continuous timecode, but timecode generated at the current cursor position. For example, this can be useful when working with video editing software that interprets the absence of timecode as a stop command. By using this option, the video soft ware can enter still mode instead so that a still frame is shown instead of a blank screen.
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To change one of the timecode values (main and off­set timecode displays), double-click on any of the time­code fields and enter a new value.
Example – Synchronizing a device to Nuendo
1. Use the SMPTE Generator as an insert effect on an audio track, and route that track to a separate output.
Make sure that no other insert or send effect is used on this track. You should also disable any EQ.
2. Connect the corresponding output on the audio hard­ware to the timecode input on the device you wish to syn­chronize to Nuendo.
Make all necessary settings for the external device so that it synchronizes to incoming timecode.
3. If needed, adjust the level of the timecode, either in Nuendo or in the receiving device.
Activate the Generate Code button (make the device send the SMPTE timecode in “free run” mode) to test the level.
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The included effect plug-ins
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