Steinberg Nuendo - 4.0 Plug-in Reference

Plug-in Reference

Original manual by Anders Nordmark

Revision: Cristina Bachmann, Heiko Bischoff, Marion Bröer, Sabine Pfeifer

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 is either a registered trademark or trademark 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.

Release Date: April 02, 2008

5 The included effect plug-ins

6 Introduction 6 Delay plug-ins 9 Distortion plug-ins 11 Dynamics plug-ins 20 EQ plug-ins 22 Filter plug-ins 28 Mastering – UV 22 HR 29 Modulation plug-ins 35 Other plug-ins 37 Restoration plug-ins 41 Reverb plug-ins 43 Spatial plug-ins 44 Surround plug-ins 53 Tools plug-ins

57 MIDI effects

58 Introduction 58 Arpache 5 59 Arpache SX 60 Autopan 61 Chorder 62 Compress 63 Context Gate 64 Density 64 Micro Tuner 64 MIDIControl 65 MIDIEcho 66 Note to CC 67 Quantizer 67 Step Designer 69 Track Control 71 Track FX 71 Transformer

72 Mixconvert Appendix

73 Available conversions

75 Index

Table of Contents

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.

Ö Most of the included effects are compatible with VST3, this is indicated by an icon in front of the name of the plug-in as displayed in plug-in selection menus (for further information, see the chapter “Audio Effects” in the Operation Manual).

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

ModMachine

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 parameters are as follows:

Parameter Description

Delay This is where you specify the base note value for the de-

Tempo sync Delay on/off

Rate The Rate parameter sets the base note value for tempo

The included effect plug-ins

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 turns tempo sync for the delay parameter on or off. If set to off, the delay time can be set freely with the Delay knob.

syncing the delay modulation (1/1 to 1/32, straight, triplet or dotted). If tempo sync is off, the rate can be set freely with the Rate knob.

Parameter Description

Tempo sync Rate on/off

Width This sets the amount of delay pitch modulation. Note that

Feedback This sets the number of repeats for the delay.

Drive This parameter adds distortion to the feeback loop. The

Mix Sets the level balance between the dry signal and the ef-

Nudge Clicking the Nudge button once will momentarily speed

Signal path graphic

Output/Loop The Filter can either be placed in the feedback loop of the

Filter type This toggle button allows you to select a filter type. Low-

Freq This sets the cutoff frequency for the filter. This is avail-

Speed This sets the speed of the filter frequency LFO modula-

Range Lo/Hi These knobs specify the range (in Hz) of the filter fre-

Spatial This introduces an offset between the channels to create

Q-Factor This controls the resonance of the filter. This is available

Speed This sets the speed of the filter resonance LFO modula-

The button below the Rate knob turns tempo sync for the rate parameter on or off. If set to off, the rate can be set freely with the Rate knob.

although the modulation affects the delay time, the sound is mostly perceived as a vibrato or chorus-like effect.

longer the Feedback, the more the delay repeats become distorted over time.

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

up the audio coming into the plug-in, simulating an analog tape nudge type sound effect.

You can click on the Filter sections displayed in the graphic in the center of the plug-in to place the Filter section either before or after the Drive and Feedback parameters in the signal path.

delay or in its output path (see above).

pass/bandpass/hipass filter types are available.

able only, if filter frequency LFO tempo sync is deactivated and the Speed parameter (see below) is set to “0".

tion. If tempo sync is activated 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 rate can be set freely with the Speed knob.

quency modulation. Both positive (e.g. Lo set to 50 and Hi set to 10000) and negative (e.g. Lo set to 5000 and 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 instead controlled by the Freq parameter.

a stereo panorama effect for the filter frequency modulation. Turn clockwise for a more pronounced stereo effect.

only, if filter resonance LFO tempo sync is deactivated and the Speed parameter (see below) is set to “0". If tempo sync is on, the resonance is controlled by the Speed and Range parameters.

tion. If tempo sync is activated, 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 rate can be set freely with the Speed knob.

Parameter Description

Range Lo/Hi These knobs specify the range of filter resonance modu-

Spatial This introduces an offset between the channels to create

lation. Both positive (e.g. Lo set to 50 and Hi set to 100) and negative (e.g. Lo set to 100 and Hi set to 50) 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 instead.

a stereo panorama effect for the filter resonance modulation. Turn clockwise for a more pronounced stereo effect.

The included effect plug-ins

MonoDelay

PingPongDelay

This is a mono delay effect that can either be tempo-based or use freely specified delay time settings. The delay can also be controlled from another signal source via the SideChain input.

The parameters are as follows:

Parameter Description

Delay This is where you specify the base note value for the delay

Tempo sync on/off

Feedback This 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-

Side-Chain on/off

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.

The button below 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.

and allows you to roll off low frequencies from 10Hz up to 800Hz. The button below the knob activates/deactivates the filter.

and allows you to roll off high frequencies from 20kHz down to 1.2kHz. The button below the knob activates/ deactivates the filter.

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

When this is activated, the delay can be controlled by a signal routed to the Side-Chain input. When the sidechain signal exceeds the threshhold the delay repeats are silenced. When the signal drops below the threshold the delay repeats reappear. For a description on how to set up Side-Chain routing, see the chapter “Audio effects” in the Operation Manual.

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 parameters are as follows:

Parameter Description

Delay This is where you specify the base note value for the delay

Tempo sync on/off

Feedback This 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 This parameter sets the stereo width for the left/right re-

Mix Sets the level balance between the dry signal and the ef-

Side-Chain on/off

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.

The button below 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.

off low frequencies up to 800Hz. The button below the knob activates/deactivates the filter.

off high frequencies from 20kHz down to 1.2kHz. The button below the knob activates/deactivates the filter.

peats. Turn clockwise for a more pronounced stereo “ping-pong” effect.

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

When this is activated, the delay can be controlled by a signal routed to the Side-Chain input. When the sidechain signal exceeds the threshhold, the delay repeats are silenced. When the signal drops below the threshold, the delay repeats reappear. For a description on how to set up Side-Chain routing, see the chapter “Audio effects” in the Operation Manual.

The included effect plug-ins

StereoDelay

Distortion plug-ins

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

This effect features two independent delay lines which can either use tempo-based or freely specified delay time settings.

The parameters are as follows:

Parameter Description

Delay 1 This is where you specify the base note value for the delay,

Delay 2 As above.

Tempo sync on/off

Feedback 1 & 2

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

Pan1 & 2 This sets the stereo position for each delay.

Mix Sets the level balance between the dry signal and the ef-

Side-Chain on/off

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.

The buttons below each respective Delay knob 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.

This sets the number of repeats for each delay.

off low frequencies up to 800 Hz. The button below the knob activates/deactivates the filter.

off high frequencies from 20kHz down to 1.2kHz. The button below the knob activates/deactivates the filter.

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

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 parameters are as follows:

Parameter Description

Drive Governs 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 Use this to boost or damp the higher frequencies.

Volume This controls the overall output level.

Amplifier This allows you to select between various amplifier mod-

Cabinet Various speaker cabinet models. Click on the currently

Damping Lo/Hi Further tone controls for shaping the sound of the se-

els. Click on the currently selected amplifier name to open a pop-up with all the available amplifier models. This section can be bypassed by selecting “No Amp".

selected cabinet name to open a pop-up with all the available amplifier models.This section can be bypassed by selecting “No Speaker".

lected speaker cabinet. Click on the values, enter a new value and press the [Enter] key.

The included effect plug-ins

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

Balance This controls the balance between the signal processed

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

if you want an overdriven sound just on the verge of distortion.

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

Distortion

SoftClipper

This effect adds soft overdrive, with independent control over the second and third harmonic.

The parameters are as follows:

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 This allows you to adjust the amount of the second har-

Third This allows you to adjust the amount of the third harmonic

overdriven sound just on the verge of distortion.

to the original signal.

monic in the processed signal.

in the processed signal.

Distortion will add crunch to your tracks.

The parameters are as follows:

Parameter Description

Drive Increases the distortion amount.

Feedback This parameter feeds part of the output signal back to the

Tone Lets you select a frequency range to which to apply the

Spatial Changes the distortion characteristics of the left and

Output Raises or lowers the signal going out of the effect.

effect input, increasing the distortion effect.

distortion effect.

right channel, thus creating a stereo effect.

The included effect plug-ins

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. Compressor 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 Release parameter.

The available parameters work as follows:

Parameter Description

Threshold (-60 to 0dB)

Ratio (1:1 to 8:1)

Soft Knee (On/Off)

Make-up (0–24dB or “Auto mode")

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

If this is off, signals above the threshold will be compressed instantly according to the set ratio (hard knee). When Soft Knee is activated, the onset of compression will be 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 instead automatically adjusted for gain loss.

Parameter Description

Attack (0.1–100ms)

Hold (0– 2000ms)

Release (10– 1000ms or “Auto mode”)

Analysis (0–100) (Pure Peak to Pure RMS)

Live mode (On/Off)

Side-Chain (On/Off)

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.

Sets the time the applied compression will affect the signal after exceeding the Threshold.

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.

This parameter determines whether the input signal is analysed 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 percussive material, with a lot of transient peaks.

When activated, Live mode disengages the “look ahead” feature of the Compressor. Look ahead does produce more accurate processing but will add a certain amount of latency as a trade-off. When Live mode is activated, there is no latency, which might be better for “live” processing.

When this is activated, the compression can be controlled by a signal routed to the Side-Chain input. When the sidechain signal exceeds the threshhold, the compression is triggered. For a description on how to set up Side-Chain routing, see the chapter “Audio effects” in the Operation Manual.

The included effect plug-ins

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 recom-

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

Auto Threshold See separate description below.

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

mend that you start with a value between 4 and 7.

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

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.

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.

The included effect plug-ins

EnvelopeShaper

EnvelopeShaper can be used to cut or boost the gain of the Attack and Release phase of the audio material. You can either use the knobs or drag the breakpoints in the graphic 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 (-20–20dB) Changes the gain of the Attack phase of the signal.

Length (5–200ms) This determines the length of the Attack phase.

Release (-20–20dB) Changes the gain of the Release phase of the signal.

Output (-24–12dB) Sets the output level.

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 graphic display to change the Threshold and the Ratio parameter values.

The following parameters are available:

Parameter Description

Threshold (-60–0dB)

Ratio (1:1–8:1)

Soft Knee (On/Off)

Attack (0.1–100ms)

Hold (0–2000ms)

Release (10–1000ms or Auto mode)

Analysis (0–100) (Pure Peak to Pure RMS)

This setting determines the level where expansion “kicks in”. Signal levels below the set threshold are affected, but signal levels above are not processed.

Ratio determines the amount of gain boost applied to signals below the set threshold.

If this is off, signals below the threshold will be expanded instantly according to the set ratio ("hard knee"). When Soft Knee is activated, the onset of expansion will be more gradual, producing a less drastic result.

This determines how fast Expander will respond to signals below the set threshold. If the attack time is long, more of the early part of the signal (attack) will pass 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.

The included effect plug-ins

Parameter Description

Live mode (On/Off)

Side-Chain (On/Off)

When activated, Live mode disengages the look ahead feature of Expander. Look ahead does produce more accurate processing but will add a certain amount of latency as a trade-off. When Live mode is activated, there is no latency.

When this is activated, the expansion can be controlled by a signal routed to the Side-Chain input. When the side-chain signal exceeds the threshhold the expansion is triggered. For a description on how to set up Side-Chain routing, see the chapter “Audio effects” in the Operation Manual.

Gate

Gating, or noise gating, 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.

The available parameters are as follows:

Parameter Description

Threshold (-60–0dB)

state LED This indicates whether the gate is open (LED lights up in

Filter buttons When the Side-chain button (see below) is activated, you

Side-chain (Off/On)

Center (50Hz– 20000Hz)

Q-Factor (0.01–10000)

This setting 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 between (LED lights up in yellow).

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 filter. The input signal can then be shaped according to set Center and Q-Factor parameters which may be useful in tailoring how the Gate operates.

Sets the center frequency of the filter.

Sets the Resonance of the filter.

Parameter Description

Monitor (Off/On)

Attack (0.1–1000 ms)

Hold (0–2000ms)

Release (10–1000ms or “Auto”)

Analysis (0–100) (Pure Peak to Pure RMS

Live mode (On/Off)

Allows you to monitor the filtered signal.

This parameter sets the time it takes for the gate to open after being triggered. If the Live button (see below) is deactivated, it will ensure 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.

This determines how long the gate stays open after 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, Gate will find an optimal release setting, depending on the audio program material.

This parameter determines whether the input signal is analysed 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 percussive material, with a lot of transient peaks.

When activated, Live mode disengages the “look ahead” feature of the Gate. Look ahead does produce more accurate processing but will add a certain amount of latency as a trade-off. When Live mode is activated, there is no latency, which might be better for “live” processing.

The included effect plug-ins

Limiter

Maximizer

Limiter is designed to ensure that the output level never exceeds a certain 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 available parameters are the following:

Parameter Description

Input (-24–+24dB)

Output (-24–+6dB)

Release (0.1–1000ms or Auto mode)

Allows you to adjust the input gain.

This setting determines the maximum output level.

This parameter 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 varies depending on the audio material.

Maximizer can be 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 tubelike distortion to the signal.

The available parameters are the following:

Parameter Description

Output (-24–+6dB)

Optimize (0–100)

Soft Clip (On/Off)

This setting determines the maximum output level. Should normally be set to 0 (to avoid clipping).

This setting determines the loudness of the signal.

Soft Clipper starts limiting (or clipping) the signal “softly”, at the same time generating harmonics which add a warm, tubelike characteristic to the audio material.

The included effect plug-ins

MIDI Gate

Gating, in its fundamental form, silences audio signals below a certain set threshold level. That means, 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 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.

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

Hold Regulates how long the Gate remains open after a Note

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

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 to open after receiving a signal that triggers it.

On or Note Off message (see Hold Mode below).

(in addition to the value set with the Hold parameter).

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.

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

The included effect plug-ins

MultibandCompressor

The MultibandCompressor allows a signal to be split in up to 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 characteristics 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 number 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 corner 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 +/- 15dB after compression.

Bypassing frequency bands

Each frequency band can be bypassed using the “B” button in each compressor section.

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. The compressor parameters for each of the four bands are as follows:

Parameter Description

Threshold (-60–0dB)

Ratio (1000–8000) (1:1 to 8:1)

Attack (0.1– 100ms)

Release (10– 1000ms or “Auto”)

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 determines the amount of gain reduction applied to signals over the set threshold. A ratio of 3000 (3:1) means that for every 3dB the input level increases, the output level will increase by only 1dB.

This determines how fast the 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.

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, the compressor will automatically find an optimal release setting that varies depending on the audio material.

The Output dial

The Output dial controls the total output level that the MultibandCompressor passes on to Nuendo. The range available is +/- 24dB.

The included effect plug-ins

VintageCompressor

Limiter

Routing selector

Gate

Compressor

This is modelled after vintage type compressors. Compressor features separate controls for input gain, attack, release and output gain parameters. 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 gain (-24–48dB)

Output gain (-48–24dB)

Attack (0.1–100ms)

Punch (Off/On)

Release (10–1000ms or “Auto mode”)

Side-Chain (On/Off)

This setting, together with the Output gain parameter determines the compression amount. The higher the Input gain setting, and the lower the Output gain setting, the more compression is applied.

Sets the output gain.

This determines how fast Compressor will respond. If the attack time is long, more of the early part of the signal (attack) will pass 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 setting that varies depending on the audio material.

When this is activated, the compression can be controlled by a signal routed to the Side-Chain input. When the sidechain signal exceeds the threshhold the compression is triggered. For a description on how to set up Side-Chain routing, see the chapter “Audio effects” in the Operation Manual.

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 controls 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 certain set threshold level. As soon as the signal level exceeds the set threshold, the gate opens to let the signal through. The Gate trigger input can also be filtered using an internal side-chain.

The available parameters are as follows:

Parameter Description

Threshold (-60–0dB)

state This indicates whether the gate is open (LED lights up in

Side-chain (On/Off)

LP (Lowpass), BP (Bandpass), HP (Highpass)

Center (50–22000Hz)

Q-Factor (0.001–10000)

This setting 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 between (LED lights up in yellow).

This button activates the internal side-chain filter. This lets you filter out parts of the signal that might otherwise trigger the gate in places you don’t 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.

This sets the center frequency of the filter.

This sets the resonance or width of the filter.

The included effect plug-ins

Parameter Description

Monitor (Off/On)

Attack (0.1–100ms)

Hold (0–2000ms)

Release (10–1000ms or “Auto”)

Allows you to monitor the filtered signal.

This parameter sets the time it takes for the gate to open after being triggered.

This determines how long the gate stays open after the signal drops below the threshold level.

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 Description

Threshold (-60–0dB)

Ratio (1:1–8:1)

Make-Up (0–24dB)

Attack (0.1–100ms)

Release (10–1000ms or “Auto”)

Graphic display

This setting determines the level where Compressor “kicks in”. Signal levels above the set threshold are affected, but signal levels below are not processed.

This parameter is used to compensate for output gain loss, caused by compression. When Auto is on, gain loss will be compensated automatically.

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.

Use the graphic display to graphically set the Threshold or the Ratio value.

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 prevent the output level from 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 Description

Output (-24–+6dB)

Soft Clip (On/Off)

Release (10–1000ms or “Auto”)

This setting determines the maximum output level. Signal levels above the set threshold are affected, but signal levels below are left unaffected.

Soft Clipper acts differently compared to the 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.

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 Module Configuration button

In the bottom right corner of the plug-in panel you will find a button with which 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. Simply click the Module Configuration button 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)

The included effect plug-ins

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 12dB allowing for fine control of the frequency response. 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 frequency bands or enter values numerically by clicking on a gain value at the top of the display.

• At the bottom of the window the respective 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 This controls the overall gain of the equalizer.

Range This allows you to relatively adjust how much a set curve

Flatten button Resets all the frequency bands to 0dB.

Invert range This will invert the current response curve.

Mode The filter mode set here determines how the various fre-

cuts or boosts the signal. If the Range parameter is turned fully clockwise, +/- 12dB is the available range.

quency band contrrols interact to create the response curve. See also below.

About the filter modes

On the pop-up 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, which is sometimes desirable. As always, let your ears be the judge! 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 – parallell filters where the resonance depends on the amount of gain. Musical sounding.

• Constant Q u – parallell filters where the resonance of the first and last bands depends on the sample rate (u=unsymmetric).

• Constant Q s – parallell 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.

The included effect plug-ins

StudioEQ

This is a high-quality 4-band parametric stereo equalizer with two fully parametric midrange bands. The low and high bands can act as either shelving filters (three types) or as a Peak (bandpass) or Cut (lowpass/highpass) filter.

Making settings

1. Click the corresponding On button to the left of 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.

The following parameters are available:

Parameter Description

Low Freq (20 to 2000Hz)

Low Gain (-20 to +24dB)

Low Q-Factor This parameter controls the width or resonance of the

This sets the frequency of the Low band.

This sets the amount of cut/boost for the Low band.

Low band.

Parameter Description

Low Filter mode

Mid 1 Freq (20 to 20000Hz)

Mid 1 Gain (+/- 24dB)

Mid 1 Q-Factor (0.5 to 10)

Mid 2 Freq (20 to 20000Hz)

Mid 2 Gain (-20 to +24dB)

Mid 2 Q-Factor (0.5 to 10)

High Freq (200 to 20000Hz)

High Gain (-20 to +24dB)

High Q-Factor This parameter controls the width or resonance of the

High Filter mode

Output (-24 to +24dB)

Auto Gain When this is activated, the gain is automatically adjusted,

For the Low band, you can select between three types of shelving filters or Peak (bandpass) or Cut (lowpass/highpass) filters. The Gain parameter will be fixed if Cut mode is selected.

-Shelf I adds resonance in the opposite gain direction slightly over 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 sets the center frequency of the Mid 1 band.

This sets the amount of cut/boost for the Mid 1 band.

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

This sets the center frequency of the Mid 2 band.

This sets the amount of cut/boost for the Mid 2 band.

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

This sets the frequency of the High band.

This sets the amount of cut/boost for the High band.

High band.

For the High band, you can select between three types of shelving filters, and Peak or Cut filters. The Gain parameter will be fixed if Cut mode is selected.

-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 parameter allows you to adjust the overall output level.

keeping the output level constant regardless of the EQ settings.

The included effect plug-ins

Filter plug-ins

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

DualFilter

This effect filters out certain frequencies while allowing others to pass through.

The following parameters are available:

Parameter Description

Position This parameter sets the filter cutoff frequency. If you set

Resonance Sets the sound characteristic of the filter. With higher

this to a negative value, DualFilter will act as a low-pass filter. Positive values cause DualFilter to act as a highpass filter.

values, a ringing sound is heard.

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.

The included effect plug-ins

PostFilter

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 unwanted 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 graphic display, or by adjusting one of the parameter 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 Displays the output level, giving you an indication of how

Lo Cut Freq (20–1000Hz)

Lo Cut Preview Use the Preview button (found between the Lo Cut Freq

Notch Freq Sets the frequency of the notch filter.

the filtering affects the overall level of the edited event.

Use this low-cut filter to eliminate low-frequency noise. Filter is Off when the handle/knob is moved all the way to the left.

button and the graphic display) to switch the filter to a complementary high-cut filter. This deactivates any other filters, allowing you to listen only to the frequencies you want to filter out.

Parameter Description

Notch Gain Allows you to adjust the gain of the selected frequency.

Notch Q-Factor Sets the width of the notch filter.

Notch filter Preview

Notches (1, 2, 4, 8)These buttons add one or more additional notch filters to

Hi Cut Freq (3–20kHz)

Hi Cut Preview Use the Preview button found between the Hi Cut Freq

Use positive values to identify the frequencies that you want to filter out.

Use the Preview button (found between the notch filter buttons and the graphic display) to create a band-pass 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.

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.

button and the graphic display) to switch the filter to a complementary low-cut filter. This deactivates any other filters, allowing you to listen only to the frequencies you want to filter out.

The included effect plug-ins

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