Steinberg Nuendo - 5.0 Plug-in Reference

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Plug-in Reference

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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

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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

Table of Contents

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The included effect plug-ins

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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

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

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 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.

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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

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

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/

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.

chain routing, see the chapter “Audio effects” in the Oper ation Manual.

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

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

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.

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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.

 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

A de-esser is used to reduce excessive sibilance, primarily for vocal recordings. Basically, it is a special type of com-

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

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

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

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.

 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

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.

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.

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.

Bypassing frequency bands

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

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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)

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

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.

Parameter Description

Release (10 to 1000 ms or Auto mode)

Graphical display

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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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

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

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.

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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.

Q 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 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

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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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

range by up to 24

Band Mode.

dB.

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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.

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.

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

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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Page 27

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.

g. to

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.

 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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4. Make sure that the frame rate in the receiving device

matches the frame rate set in the SMPTE Generator.

5. Activate the “Link to Transport” button.

The SMPTE Generator now outputs timecode that corresponds to the Nuendo time display.

6. On the Nuendo Transport panel, click Play.

The external device is now synchronized and will follow any position changes set with the Nuendo transport controls.

Parameter Description

Frequency section

Gain section Allows you to set the amplitude of the signal. The higher

Allows you to set the frequency of the generated signal. You can select one of the preset values (100, 440, 1000,

Hz), or use the slider to set a value between

or 10000

Hz and 20000 Hz.

the value (up to 0 select one of the preset values (e. slider to set a value between -81 and 0

dB), the stronger the signal. You can

g. -20 dB), or use the

dB.

TestGenerator

This utility plug-in allows you to generate an audio signal, which can be recorded as an audio file. The resulting file can then be used for a number of purposes:

• For testing the specifications of audio equipment.

• For measurements of various kinds, such as calibrating tape recorders.

• For testing signal processing methods.

• For educational purposes.

The TestGenerator is based on a waveform generator which can generate a number of basic waveforms such as sine and saw as well as various types of noise. Furthermore, you can set the frequency and amplitude of the generated signal.

As soon as you add the TestGenerator as an effect on an audio track and activate it, a signal is generated. You can then activate recording as usual to record an audio file according to the signal specifications:

Parameter Description

Waveforms and noise section

Allows you to set the basis for the signal generated by the waveform generator. You can select between four basic waveforms (sine, triangle, square, and sawtooth) and three types of noise (white, pink, and brownian).

Mastering – UV22HR

The UV22HR is a dithering plug-in, based on an advanced algorithm developed by Apogee. For an introduction to the concept of dithering, see the chapter “Audio effects” in the Operation Manual.

The following parameters are available:

Option Description

Bit Resolution The UV22HR supports dithering to multiple resolutions:

Hi Try this first, it is the most “all-round” setting.

Lo This applies a lower level of dither noise.

Auto black When this is activated, the dither noise is gated (muted)

Dithering should always be applied post-fader on an output bus.

8, 16, 20 or 24 bits. You select the desired resolution by clicking the corresponding button.

during silent passages in the material.

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Modulation plug-ins

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

AutoPan

This is a simple auto-pan effect. It can use different waveforms to modulate the left-right stereo position (pan), either using tempo sync or manual modulation speed settings.

The following parameters are available:

Parameter Description

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

Sync button The button below the Rate knob is used to switch tempo

Width Sets the depth of the auto-pan effect.

Waveform Shape selector

 The Width parameter can also be controlled from another signal source via the side-chain input. For a description of how to set up side-chain routing, see the chapter “Audio effects” in the Operation Manual.

note value for tempo-syncing the effect (1/1 to 1/32, straight, triplet, or dotted). If tempo sync is off, the auto-pan speed can be set freely with the Rate knob.

sync on or off.

Allows you to select the modulation waveform. A sine and a triangle waveform are available.

Chorus

This is a single stage chorus effect. It works by doubling whatever is sent into it with a slightly detuned version (see also “StudioChorus” on page 34).

The following parameters are available:

Parameter Description

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

Sync button The button below the Rate knob is used to switch tempo

Width Determines the depth of the chorus effect. Higher set-

Waveform Shape selector

Spatial Sets the stereo width of the effect. Turn clockwise for a

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

Delay Affects the frequency range of the modulation sweep by

Filter Lo/Hi Allow you to roll off low and high frequencies of the effect

note value for tempo syncing the chorus sweep (1/1 to 1/32, straight, triplet, or dotted). If tempo sync is off, the sweep rate can be set freely with the Rate knob.

sync on or off.

tings produce a more pronounced effect.

Allows you to select the modulation waveform, altering the character of the chorus sweep. A sine and a triangle waveform are available.

wider stereo effect.

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

adjusting the initial delay time.

signal.

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 The modulation can also be controlled from another signal source via the side-chain input. When the side-chain signal exceeds the threshold, the modulation is controlled by the side-chain signal’s envelope. For a description of how to set up side-chain routing, see the chapter “Audio ef fects” in the Operation Manual.

Cloner

The Cloner plug-in adds up to four detuned and delayed voices to the signal, for rich modulation and chorus effects.

The following parameters are available:

Parameter Description

Voices Allows you to select the number of voices (up to four). For

Spatial Spreads the added voices across the stereo spectrum.

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

Output Allows you to reduce or increase the output gain by up to

Detune slider 1–4

Delay slider 1–4

Detune Governs the overall depth of the detuning for all voices. If

Natural button By clicking the Natural button below the Detune knob,

Detune – Humanize

each added voice, a Detune and a Delay slider are added in the right half of the panel.

Turn clockwise for a deeper stereo effect.

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

dB.

Controls the relative detune amount for each voice. Positive and negative values can be set, from -100 to 100. A value of zero means no detune for that voice.

Controls the relative delay amount for each voice. A value of zero means no delay for that voice.

this is set to zero, no detuning takes place, regardless of the Detune slider settings.

you can change the pitch algorithm.

Controls the amount of detune variation when Static Detune is deactivated. With Humanize, the detune is constantly modulated for a more natural effect. The value range is from 0 to 100 (strongest detune variation).

Parameter Description

Static Detune button

Delay Governs the overall depth of the delay for all voices. If set

Delay – Humanize

Static Delay button

Use this button to activate/deactivate the Static Detune function. If activated, the set detune amount is static, and the Humanize knob is grayed out.

to zero, no delay takes place regardless of the Delay slider settings.

Controls the amount of delay variation when Static Detune is deactivated. With Humanize, the delay is constantly modulated for a more natural effect. The value range is from 0 to 100 (strongest delay variation).

Use this button to activate/deactivate the Static Delay function. If activated, the set delay amount is static, and the Humanize knob is grayed out.

Flanger

Flanger is a classic flanger effect with added stereo enhancement.

The following parameters are available:

Parameter Description

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

Sync button The button below the Rate knob is used to switch tempo

Range Lo/Hi Set the frequency boundaries for the flanger sweep.

Feedback Determines the character of the flanger effect. Higher

Spatial Sets the stereo width of the effect. Turn clockwise for a

note value for tempo syncing the flanger sweep (1/1 to 1/32, straight, triplet, or dotted). If tempo sync is off, the sweep rate can be set freely with the Rate knob.

sync on or off.

settings produce a more “metallic” sounding sweep.

wider stereo effect.

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Parameter Description

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

Waveform Shape selector

Delay Affects the frequency range of the modulation sweep by

Manual knob Allows you to change the sweep position manually when

Manual button Use this button to activate/deactivate the Manual func-

Filter Lo/Hi Allow you to roll off low and high frequencies of the effect

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

Allows you to select the modulation waveform, altering the character of the flanger sweep. A sine and a triangle waveform are available.

adjusting the initial delay time.

the Manual button is deactivated. The value range is from 0 to 100.

tion. If activated, the flanger sweep is static, i. e. no modulation takes place.

signal.

 The modulation can also be controlled from another signal source via the side-chain input. When the sidechain signal exceeds the threshold, the modulation is con trolled by the side-chain signal’s envelope. For a description of how to set up side-chain routing, see the chapter “Audio effects” in the Operation Manual.

Metalizer

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

Parameter Description

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

Sharpness Governs the character of the filter effect. The higher the

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

On button Turns filter modulation on and off. When turned off, Met-

Mono button When this is activated, the output of Metalizer is mono.

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

Sync button The button above the Speed knob is used to switch

Output slider Sets the overall volume.

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

value, the narrower the affected frequency area, produc ing a sharper sound and a more pronounced effect.

more noticeable with high Feedback settings.

alizer works as a static filter.

note value for tempo-syncing the effect (1/1 to 1/32, straight, triplet, or dotted). Note that there is no note value modifier for this effect. If tempo sync is off, the modulation speed can be set freely with the Speed knob.

tempo sync on (button lights up) or off.

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

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Phaser

Phaser produces the well-known “swooshing” phasing effect with additional stereo enhancement.

The following parameters are available:

Parameter Description

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

Sync button The button below the Rate knob is used to switch tempo

Width Determines the width of the modulation effect between

Feedback Determines the character of the phaser effect. Higher

Spatial When using multi-channel audio, the Spatial parameter

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

Manual knob Allows you to change the sweep position manually when

Manual button Use this button to activate/deactivate the Manual func-

Filter Lo/Hi Allow you to roll off low and high frequencies of the effect

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

sync on or off.

higher and lower frequencies.

settings produce a more pronounced effect.

creates a 3-dimensional impression by delaying modula tion in each channel.

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

the Manual button is deactivated. The value range is from 0 to 100.

tion. If activated, the flanger sweep is static, i. e. no modulation takes place.

signal.

RingModulator

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

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

The following parameters are available:

Parameter Description

Oscillator – LFO Amount

Oscillator – Env. Amount

Oscillator – Waveform buttons

Oscillator – Range slider

Controls how much the oscillator frequency is affected by the LFO.

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

Allows you to select the oscillator waveform; square, sine, saw, or triangle.

Determines the frequency range of the oscillator in Hz.

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Parameter Description

Oscillator – Frequency

Oscillator – Roll-Off

LFO – Speed Sets the LFO speed.

LFO –

Amount

Env.

LFO – Waveform

LFO – Invert Stereo

Envelope Generator section – Attack and Decay

Lock L<R button

Output slider Sets the overall volume.

Mix slider Adjusts the mix between dry and processed signal.

Sets the oscillator frequency +/- 2 octaves within the selected range.

Cuts high frequencies in the oscillator waveform, to soften the overall sound. This is best used when harmon ically rich waveforms are selected (e. g. square or saw).

Controls how much the input signal level – via the envelope generator – affects the LFO speed. Positive and negative values can be set, at 0 plied. With negative values, a loud input signal slows down the LFO, whereas positive values are used to speed it up at loud input signals.

Allows you to select the LFO waveform; square, sine, saw, or triangle.

Inverts the LFO waveform for the right channel of the oscillator, which produces a wider stereo perspective for the modulation.

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

When this button is enabled, the L and R input signals are merged, and produce the same envelope output level for both oscillator channels. When disabled, each chan nel has its own envelope, which affects the two channels of the oscillator independently.

% no modulation is ap-

Rotary

The Rotary plug-in simulates the classic effect of a rotating speaker. A rotary speaker cabinet features speakers rotating at variable speeds to produce a swirling chorus effect, commonly used with organs. Rotary features all the parameters associated with the real thing.

The following parameters are available:

Parameter Description

Speed selector (Stop/

Slow/Fast)

Speed Change Mode

Speed Mod When the Slow/Fast setting is set to variable control, this

MIDI controller pop-up menu

Overdrive Applies a soft overdrive or distortion.

CrossOver Sets the crossover frequency (200 to 3000 Hz) between

Horn – Slow Allows for a fine adjustment of the high rotor Slow speed.

Horn – Fast Allows for a fine adjustment of the high rotor Fast speed.

Horn – Accel. Allows for a fine adjustment of the high rotor acceleration

Horn – Amp

Mod

Horn –

Mod

Freq

Bass – Slow Allows for a fine adjustment of the low rotor Slow speed.

Bass – Fast Allows for a fine adjustment of the low rotor Fast speed.

Bass – Accel. Allows for a fine adjustment of the low rotor acceleration

Bass – Amp

Mod

Bass – Level Adjusts the overall bass level.

Microphones – Phase

Microphones – Angle

Microphones – Distance

Output Allows you to adjust the overall output level.

Mix Allows you to adjust the mix between dry and processed

Allows you to control the speed of the Rotary in three steps.

Allows you to select whether the Slow/Fast setting is a switch (left) or a variable control (right). When switch mode is selected and Pitchbend is the controller, the speed will switch with an up or down flick of the bender. Other controllers switch at MIDI value 64.

allows you to select the rotary speed, from 0 (Stop) to 100 (Fast).

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.

the low and high frequency loudspeakers.

time.

Controls the high rotor amplitude modulation.

Controls the high rotor frequency modulation.

time.

Adjusts the modulation depth of the amplitude.

Allows you to adjust the phasing amount in the sound of the high rotor.

Sets the simulated microphone angle. 0 = mono, 180 = one mic on each side.

Sets the simulated microphone distance from the speaker in inches.

signals.

Directing MIDI to the Rotary

For realtime MIDI control of the Speed parameter, MIDI must be directed to the Rotary.

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• Whenever Rotary 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 Rotary is selected on the Output Routing menu, MIDI is directed to the plug-in from the selected track.

StudioChorus

The StudioChorus plug-in is a two stage chorus effect which adds short delays to the signal and pitch modulates the delayed signals to produce a “doubling” effect. The two separate stages of chorus modulation are completely independent and are processed serially (cascaded).

For each stage the following parameters are available:

Parameter Description

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

Sync button The button below the Rate knob is used to switch

Width Determines the depth of the chorus effect. Higher set-

Waveform Shape selector

Spatial Sets the stereo width of the effect. Turn clockwise for a

Mix Sets the level balance between the dry signal and the

Delay Affects the frequency range of the modulation sweep

Filter Lo/Hi Allow you to roll off low and high frequencies of the ef-

note value for tempo syncing the chorus sweep (1/1 to 1/32, straight, triplet, or dotted). If tempo sync is off, the sweep rate can be set freely with the Rate knob.

tempo sync on or off.

tings produce a more pronounced effect.

Allows you to select the modulation waveform, altering the character of the chorus sweep. A sine and a trian gle waveform are available.

wider stereo effect.

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

by adjusting the initial delay time.

fect signal.

Tranceformer

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

The following parameters are available:

Parameter Description

Waveform buttons

Tone Sets the frequency (pitch) of the modulating oscillator

Depth Governs the depth of the pitch modulation.

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

Sync button The button above the Speed knob is used to switch

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

Mono button Governs whether the output is stereo or mono.

Allow you to select a pitch modulation waveform.

to 5000 Hz).

note value for tempo-syncing the effect (1/1 to 1/32, straight, triplet, or dotted). Note that there is no note value modifier for this effect. If tempo sync is off, the mod ulation speed can be set freely with the Speed knob.

tempo sync on (button lights up) or off.

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Parameter Description

Output slider Allows you to adjust the output level of the effect.

Mix slider Sets the level balance between the dry signal and the

effect.

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

Tremolo

Tremolo produces amplitude (volume) modulation. The following parameters are available:

Parameter Description

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

Sync button The button below the Rate knob is used to switch tempo

Depth Governs the depth of the amplitude modulation.

Spatial Adds a stereo effect to the modulation.

Output Allows you to adjust the output volume.

note value for tempo-syncing the effect (1/1 to 1/32, straight, triplet, or dotted). If tempo sync is off, the modulation speed can be set freely with the Rate knob.

sync on or off.

Vibrato

The Vibrato plug-in produces pitch modulation. The following parameters are available:

Parameter Description

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

Sync button The button below the Rate knob is used to switch tempo

Depth Governs the depth of the pitch modulation.

Spatial Adds a stereo effect to the modulation.

note value for tempo-syncing the effect (1/1 to 1/32, straight, triplet, or dotted). If tempo sync is off, the modulation speed can be set freely with the Rate knob.

sync on or off.

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Other plug-ins

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

BitCrusher

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

The following parameters are available:

Parameter Description

Mode Allows you to select one of the four operating modes of

Sample Divider Sets the amount by which the audio samples are deci-

Depth Defines the bit resolution. A setting of 24 gives the highest

Output slider Governs the output level from BitCrusher. Drag the slider

Mix slider Regulates the balance between the output from Bit-

BitCrusher. In each mode the plug-in sounds differently. Modes I and III are nastier and noisier, while modes II and IV are more subtle.

mated. At the highest setting (65), nearly all of the information describing the original audio signal is eliminated, turning the signal into unrecognizable noise.

audio quality, while a setting of 1 creates mostly noise.

upwards to increase the level.

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

Chopper

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

Parameter Description

Waveform buttons

Depth Sets the depth of the Chopper effect. This can also be

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

Sync button The button above the Speed knob is used to switch

Stereo/Mono button

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

Set the modulation waveform.

set by clicking in the graphical display.

note value for tempo-syncing the effect (1/1 to 1/32, straight, triplet, or dotted). Note that there is no note value modifier for this effect. If tempo sync is off, the tremolo/auto-pan speed can be set freely with the Speed knob.

tempo sync on (button lights up) or off.

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

fect. If Chopper is used as a send effect, this should be set to the maximum value.

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Octaver

This plug-in can generate two additional voices that track the pitch of the input signal one octave and two octaves below the original pitch, respectively. Octaver is best used with monophonic signals. The following parameters are available:

Parameter Description

Direct Adjusts the mix of the original signal and the generated

Octave 1 Adjusts the level of the generated signal one octave be-

Octave 2 Adjusts the level of the generated signal two octaves be-

voice(s). A value of 0 means only the generated and transposed signal is heard. By raising this value, more of the original signal is heard.

low the original pitch. Set to 0 means the voice is muted.

Tuner

This is a guitar tuner. Simply connect a guitar or other instrument to an audio input and select the Tuner as an insert effect (make sure you deactivate any other effect that alters pitch, like chorus or vibrato). When the instrument is connected, proceed as follows:

• Play a note.

The key is shown in the middle of the display. In addition, the frequency in Hz is shown in the bottom left corner and the octave range in the bottom right corner. If the key is wrong (e. the key is shown as Fb), first tune the string so that the correct key is shown.

• The two arrows indicate any deviation in pitch by their position. If the pitch is flat, they will be positioned in the left half of the display, if the pitch is sharp they will be in the right half.

The deviation is also shown (in Cent) in the upper area of the display.

• Tune the instrument so that the two arrows are in the middle.

Repeat this procedure for each string.

g. if you wish to tune the E string and

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Pitch Shift plug-ins

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

PitchCorrect

PitchCorrect automatically detects, adjusts and fixes slight pitch and intonation inconsistencies in monophonic vocal and instrumental performances in realtime. The ad vanced algorithms of this plug-in preserve the formants of the original sound thus allowing for natural sounding pitch correction without the typical “Micky Mouse” effect.

Furthermore, you can use PitchCorrect creatively. You can create backing vocals, for example, by modifying the lead vocals or vocoder sounds by using extreme values. You can use an external MIDI controller, a MIDI track or the vir tual keyboard to “play” a note or a scale of target pitches that determine the current scale notes to which the audio is shifted. This allows you to change your audio in a very quick and easy way, which is extremely useful for live per formances. In the keyboard display, the original audio will be displayed in blue while the changes are displayed in orange.

The following parameters are available:

Parameter Description

Correction – Speed

Correction – Tolerance

Determines the smoothness of the pitch change. Higher values cause the pitch shift to occur immediately. 100 is a very drastic setting that is designed mainly for special effects (e.

g. the famous “Cher” effect).

Determines the sensitivity of analysis. A low Tolerance value lets PitchCorrect find pitch changes quickly. When the Tolerance value is high, pitch variations in the audio

g. vibrato) will not be immediately interpreted as note

(e. changes.

Parameter Description

Correction – Transpose (-12 to 12)

Scale Source – Internal

Scale Source – External MIDI Scale

Scale Source – External MIDI

Note

Formant – Shift (-60 to 60)

Formant – Optimize (Gen eral, Male, Fe-

male)

Formant – Preservation (On/Off)

Master Tuning Detunes the output signal. The default setting is 440 Hz.

With this parameter you can adjust (or “retune”) the pitch of the incoming audio in semitone steps. You can set positive and negative values from -12 to 12. A value of zero means the signal is not transposed.

If you choose the Internal option from the Scale Source pop-up menu, you can use the pop-up menu next to it to decide to which scale the source audio will be adapted. The following options are available: Chromatic: The audio will be pitched to the closest semitone. Major/Minor: The audio will be pitched to the major/minor scale specified in the pop-up menu to the right. This will be reflected on the keyboard display. Custom: The audio will be pitched to the notes that you specify by clicking the desired keys on keyboard display. To reset the keyboard, click on the orange line below the display.

Select this option if you want the audio to be shifted to a scale of target pitches, using an external MIDI controller, the Virtual Keyboard or a MIDI track. Note that you have to assign the audio track as the output of your MIDI track and that the Speed parameter has to be set to a value other than Off.

Select this option if you want the audio to be shifted to a target note, using an external MIDI controller, the Virtual Keyboard or a MIDI track. Note that you have to assign the audio track as the output of your MIDI track and that the Speed parameter has to be set to a value other than Off.

Changes the natural timbre, i. e. the characteristic frequency components of the source audio.

Allows you to specify the sound characteristics of the

sound sources. While General is the default setting, Male is designed for low pitches and Female for high pitches.

When set to Off, formants are raised and lowered with the pitch, provoking strange vocal effects. Higher pitch correction values result in “Micky Mouse” effects, lower pitch correction values in “Monster” sounds. When set to On, the formants are kept, maintaining the character of the audio.

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PitchDriver

PitchDriver was created for sound design purposes in postproduction. This plug-in can be used for extreme up or down pitching of voices or effect samples (e. g. to create eerie monster sounds). Shifting the pitch with this plug-in will not keep the formants.

The following parameters are available:

Parameter Description

Detune Lets you detune the pitch of the incoming audio. Positive

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

Spatial The Spatial parameter is used to create an ambience ef-

Output Allows you to adjust the output volume.

 To avoid hearing artifacts, it is recommended to set the ASIO buffer for your audio card to at least 128 sam ples. The buffer size can be set on the card driver’s control panel (opened via the Device Setup dialog in Nuendo).

and negative values can be set.

fect.

fect. It introduces a light pitch offset to the incoming signal. Different offset values are used for the individual input channels in order to create a panorama effect. Note that the created panorama effect can be unstable. For a stable panorama, turn of the Spatial parameter. In that case the incoming signals are summed up to a mono signal.

Restoration plug-ins

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

DeClicker

The DeClicker plug-in is specifically designed to eliminate single “clicks” or “pops” in a recording. A typical applica tion is to clean up recordings made from vinyl records, but you may also find it useful for removing pops from microphone switches, oxidized connector noises, clicks from sync problems when transferring material digitally, etc.

 Note that the DeClicker module is not optimized for crackles (a series of short clicks). However, as it is often hard to distinguish between clicks and crackles, you might also be able to use it to improve your recording in this re spect.

 If the recording also contains background noise (hiss), you may want to combine DeClicker with the DeNoiser plug-in.

How DeClicker works

The DeClicker process is divided in two tasks:

• Analysis – when the audio signal passes through DeClicker, the selected analysis algorithm finds the clicks in the recording. You provide input to the analysis parame ters by selecting a Mode and setting the Threshold and DePlop parameters.

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• Removal – a de-click algorithm is applied to the audio,

removing the clicks.

In many cases, the original audio material “hidden” underneath a click cannot be restored. This means there will be a gap once the click has been removed. DeClicker has the ability to automatically “redraw” the hence missing parts of the waveform. This feature can also be used to remove tape dropouts with a length of up to 60 samples (just above one millisecond at 44.1

kHz).

The whole de-clicking process can be visually monitored in the Input and Output displays in the DeClicker panel (showing the incoming audio and the processed, i.

e. declicked, audio). This helps you adjust the parameters. Furthermore, if you activate the Audition button, only the removed material will be heard (and shown in the Output display).

Make sure that no low-pass filter has been applied to your audio material before you edit it with DeClicker. This may affect the detection of clicks.

Parameters

Parameter Description

Audition button

Classic button When this button is activated, DeClicker attempts to re-

Quality section Here you can determine the quality of the click removal

Mode section Which mode to select depends on the source material.

When this button is activated, only the removed material will be heard. The Output display will also show the waveform image of the removed material in this mode.

move both audible clicks and crackle noise. When deactivated, only single clicks are removed while crackles (rapidly repeated clicks) are ignored. Which mode to choose, depends on the source material. Note also that Classic mode requires less CPU power.

and audio restoration, with “4” being the best quality set ting. Please note that selecting higher quality settings means that more processing power is consumed. Also, note that in some situations it might be more productive to use a lower Quality value. One example of this is when two clicks follow each other in quick succession or when you tackle a click in a low level part that is fol lowed by a loud part.

Standard mode is suitable for a wide variety of source material – try this option first. Vintage mode is suitable for restoring “antique” recordings (with limited high fre quency content), while Modern mode is best suited for contemporary recordings with a wide frequency range (putting greater emphasis on distinguishing clicks from other strong impulses in the audio material).

Parameter Description

Threshold slider

DePlop slider Controls a special high-pass filter which works on signals

Determines the amplitude (level) required for a click to be detected. In many cases, DeClicker’s sensitive algorithms identify a lot more clicks than you can actually hear. To avoid wasting processing power for removing inaudible clicks, raise this parameter to a high value, and then lower it until all the artifacts that you actually want removed are detected. The lower the setting, the more clicks will be detected, but also the higher the risk of audible artifacts. If in doubt, activate Audition mode and check that the re moved material does not contain any actual musical or rhythmical information, etc.

below 150 times appears after eliminating a click. The slider adjusts the filter frequency (Off–150 Note: this function is best applied to older recordings, which often use a narrow frequency range. Be careful when applying it to modern recordings, as you may risk removing parts of the useful signal!

Hz. It cuts away the “plop noise” which some-

Hz).

Tips and Tricks

• By combining Vintage Mode and extreme Threshold and DePlop settings, you can create an interesting effect which “softens” material with particularly sharp attacks, e. g. percussion or brass.

• If you have material with digital distortion (clipping), try applying DeClicker. While it cannot do miracles, it can at least make some improvement to the overall “hardness” introduced by the distortion.

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Noise Reduction

Noise Floor

Ambient Analysis

Transient Analysis

Input

Output

Level

Noise Reduction

Ambience

DeNoiser

The DeNoiser plug-in lets you suppress noise without affecting the general sound quality. Or, in tech talk, the DeNoiser removes broad band noise from arbitrary audio material without leaving any “spectral finger print”. The al gorithm that this plug-in is based on has the ability to track and adjust itself to variations in background noise. This means the noise can be diminished without side effects, preserving the spatial impression, and without letting the result become “colorless”. Many years of research were invested in developing the methods used.

Typical applications for the DeNoiser plug-in include cleaning or remastering recordings from old tape or vinyl, or noisy live recordings.

How DeNoiser works

DeNoiser is based on spectral subtraction. Each section of the frequency spectrum that has an amplitude below the estimated noise floor is reduced in intensity by use of a spectral expander. The result is a noise reduction that does not affect the phase of the signal.

The figure below shows the signal flow:

The solid line represents the actual audio signal, while the dotted lines represent control signals.

The signal is continuously analyzed by the first module in the chain, to estimate the noise floor at any given time. This is sufficient when the noise level is constant or mod

ulates slowly. When the noise level varies rapidly, the Ambience and Transient analysis helps adjust the response of the noise reduction unit, allowing transient-rich material to maintain its liveliness and natural ambience.

 When processing audio with DeNoiser, the plug-in needs a short time (less than a second) to analyze the material and set its internal parameters. Since you would not want to include this short “startup sequence” in the final result, you should make it a habit to first play back a short section of the audio, thereby letting DeNoiser “learn” the noise floor, and then stop and start over again from the be ginning. The plug-in then remembers the settings internally.

The Noisefloor Display

The display on the left of the DeNoiser panel is crucial when making settings. It contains the following three elements:

• The dark green spectral graph.

This shows the spectrum of the audio being played back. The horizontal axis shows the frequency (linear scale). The low frequencies are visible on the left side, the high ones on the right side. The vertical axis shows the signal amplitudes, thus the level (displayed as a logarithmic dB scale).

• The yellow line.

This is a spectral estimation of the noise floor. The average of this value is shown numerically below the display.

• The light green line.

This is simply a graphical representation of the Offset parameter.

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The light green Offset line should be adjusted so that it appears as close above the yellow noise floor graph as possible. The dark green spectrum plot is there to help you finetune the Offset setting, so that only the noise is removed, not parts of the signal (ideally, the light green line should be between the yellow line and the spectrum plot).

Parameters

Parameter Description

Freeze button This button is used to “freeze” the noise floor detection

Classic button When this is activated, a less CPU-intensive version of

A/B/Store buttons

Reduction slider

Ambience slider

Offset slider This parameter serves as a threshold, governing the over-

process. The yellow noise floor graph in the display will hold its current value (as will the numeric noise floor value display below) until you deactivate Freeze. This allows you to take a closer look at the readings.

the DeNoiser algorithm is used. Use Classic mode if you are short on processing power. However, for optimum noise suppression, we recommend that you deactivate Classic mode.

These buttons are described below this table.

Governs the amount of noise reduction. The display above this slider shows the amount of dB by which the noise level is being reduced. The final result also depends on the Ambience parameter, and on the automatic Ambi ence and Transient analysis of the original material, as described above.

This parameter is used to specify a balance between the noise suppression and the amount of natural ambience, which is essential for a natural result. With a low Ambi ence setting, the sound can become somewhat lifeless and sterile. A high setting, on the other hand, preserves more of the ambient character of the sound, but the noise suppression is less effective.

all level at which the noise reduction is performed. For optimal noise reduction with a minimum of sound colora tion, this parameter should be set to a value slightly above the noise floor level. To help you do this, the offset value is shown as a light green line in the noisefloor display, while the noise floor is shown as a yellow line.

Using the A/B setups

With the A/B buttons you can make instantaneous switches between two different DeNoiser setups, allowing you to quickly try out and compare different configura

tions. You can also use this feature for separate settings for two different sections of an audio recording. Proceed as follows:

1. Make the settings you want for setup A.

2. Click the Store button and then the A button.

3. Make the settings you want for setup B.

4. Click the Store button and then the B button.

Now the two setups are stored, and you can switch between them simply by clicking the A or B button.

Grungelizer

Grungelizer adds noise and static to your recordings – kind of like listening to a radio with bad reception, or a worn and scratched vinyl record. The following parame ters are available:

Parameter Description

Crackle Adds crackle to create that old vinyl record sound. The

RPM switch When emulating the sound of a vinyl record, this switch

Noise Regulates the amount of static noise added.

Distort Adds distortion.

EQ Turn this knob to the right to cut off the low frequencies,

AC Emulates a constant, low hum of AC current.

Frequency switch

Timeline Regulates the amount of overall effect. The farther to the

farther to the right you turn the knob, the more crackle is added.

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

and create a more hollow, lo-fi sound.

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

right (1900) you turn the knob, the more noticeable the effect.

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Reverb plug-ins

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

REVerence

REVerence is a convolution tool that allows you to apply room characteristics (reverb) to the audio. This is done by processing the audio signal according to an impulse re sponse – a recording of an impulse in a room or another location that is used to recreate the characteristics of the room. As a result, the processed audio will sound as if it were played in the same location. Included with the plugin are top quality samples of real spaces to create rever beration.

 REVerence can be very demanding in terms of RAM. This is because the impulse responses that you load into the program slots are preloaded into RAM to guarantee an artifact-free switching between programs. Therefore you should always load only those programs that you need for a given task.

Using the program matrix

A program is the combination of an impulse response and its settings. These include reverb settings (see “Changing

the reverb settings” on page 44), EQ settings (see “Making EQ settings” on page 45), pictures (see “Loading pictures” on page 46), and output settings (see “Making output settings” on page 46). The program matrix allows

you to load programs and to view the name of the current program, i. e. the impulse response (see “Working with

custom impulse responses” on page 46).

The following parameters are available:

Parameter Description

Program name In the upper left corner of the plug-in panel, either the

Browse button This button opens a browser window showing the avail-

Import button Click this button to load your own impulse response files

Program slots (1 to 36)

Smooth Parameter Changes button

Store button Stores the active impulse response and its settings as a

name of the loaded impulse response file or the name of the program is shown. After loading an impulse response, its number of channels and the length in seconds are dis played for a few seconds.

able programs. When you select a program in the browser, it is loaded into the active program slot. To be able to filter the list of impulse responses in the browser window, e. can activate the Filters section (by clicking the “Set Up Window Layout” button at the bottom left of the window).

from disk. The files should have a maximum length of 10 seconds. Longer files are automatically cut. For more infor mation, see “Working with custom impulse responses” on

page 46.

Into these slots you can load all the impulse responses (programs) that you want to work with in a session. The selected program slot is indicated by a (blinking) white frame. Occupied slots are shown in a different color. Double-clicking an empty program opens a browser win dow, showing the available programs. Double-clicking an occupied program slot loads the corresponding impulse response into REVerence (“Recall”).

The “Smooth Parameter Changes” button is located between the program slots and the Store/Recall/Erase buttons. If it is activated, a crossfade is performed when switching programs. Leave this button deactivated while looking for a suitable program or an appropriate setting for an impulse response. Once you have set up the program matrix to your liking, ac tivate the button to avoid hearing artifacts when switching between programs.

program.

g. by room type or the number of channels, you

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Parameter Description

Recall button Reloads the selected program. Use this to reset a pro-

Erase button Removes the selected program from the matrix.

gram to its default settings.

• When automating a project and loading a REVerence program, only two automation events are written.

If load a plug-in preset instead (which contains a lot more settings than a program), a lot of unnecessary automation data (for the settings that you did not use) is written.

Programs vs. presets

You can save your REVerence settings as VST plug-in presets or programs. The differences between the two and the advantages are described in the following.

Both presets and programs use the file extension .vstpreset and appear in the same category in the MediaBay (Plug-In Presets), but they are represented by different icons:

Icon Description

A REVerence preset contains all settings and parameters for the plug-in, that is all the loaded impulse responses along with their parameter settings and positions in the program matrix. Presets are loaded via the Presets pop-up menu at the top of the plug-in panel.

A REVerence program only contains the settings related to a single impulse response. Programs are loaded and managed via the program matrix.

Presets

Presets are useful in the following situations:

• When you want to save a complete setup with different impulse responses for later use (e. explosion sounds that can be reused for other scenes or movies).

• When you want to save different parameter sets for the same impulse response so that you can later choose the set that best suits your needs.

Programs

Programs offer the following advantages:

• Up to 36 programs can be loaded into the program matrix for instant recall.

• A program provides a quick and easy way to save and recall a subset of the plug-in parameters (i. e. the settings for a single impulse response), allowing for short loading times.

g. different setups for

Setting up programs

Proceed as follows:

1. In the program matrix, click on a program slot to select it.

A blinking white frame indicates that this program slot is selected.

2. Click the Browse button or click on the empty slot again to load one of the included programs.

You can also import a new impulse response file, see “Importing impulse

responses” on page 47.

3. In the browser that appears, select the program containing the impulse response that you want to use and click OK.

The name of the loaded impulse response is shown in the upper left corner of the REVerence panel.

4. Set up the REVerence parameters as needed and click the Store button to save the impulse response with the current settings as a new program.

5. Set up as many programs as you need (up to 36) by following the steps above.

 If you want to use your set of programs in other projects, save your settings as a plug-in preset using the Presets pop-up menu at the top of the plug-in panel.

Changing the reverb settings

The reverb settings allow you to change the characteristics of the room.

The following parameters are available:

Parameter Description

Front All values shown in the top row are for the front speakers.

Rear

Auto Gain button

If you are working with surround tracks up to 5.1, you can use this row to set up an offset for the rear channels.

When this button is activated, the impulse response is automatically normalized.

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Parameter Description

Reverse button Reverses the impulse response.

Pre-Delay Controls the amount of time between the dry signal and

Time Scaling Controls the reverb time.

Size Determines the size of the simulated room.

Level A level control for the impulse response. This governs the

ER Tail Split Sets a split point between the early reflections and the

ER Tail Mix Allows you to set up the relation of early reflections and

the onset of the reverb. With higher pre-delay values you can simulate larger rooms.

volume of the reverb.

tail, allowing you to determine where the reverb tail be gins. A value of 60 means that the early reflections will be heard for 60

tail. Values above 50 attenuate the early reflections and values below 50 attenuate the tail.

ms.

The impulse response display

The Display section allows you to view the impulse response details and to change the length of the response (trimming).

The following parameters are available:

Parameter Description

Play button/ Time Scaling wheel

Time Domain display

Spectrogram display

Information display

When clicking the play button to apply the loaded impulse response, a short click is played. This provides a neutral test sound that makes it easier for you to know how different settings influence the reverb characteris tics. The Time Scaling wheel lets you adjust the reverb time.

Shows the waveform of the impulse response.

Shows the analyzed spectrum of the impulse response. Time is displayed along the horizontal axis, frequency along the vertical axis, and volume is represented by the color.

Shows additional information, e. g. the name of the program and the loaded impulse response, the number of channels, the length, and Broadcast Wave File informa tion.

Parameter Description

Activate Impulse Trimming button

Trim slider Allows you to trim the start and end of the impulse re-

Use this button at the bottom right of the Impulse display section to activate impulse trimming. The Trim slider is shown below the Impulse display.

sponse. Drag the front handle to trim the start of the impulse response, or the end handle to trim the reverb tail. You can also use the mouse wheel for trimming. Note that the impulse response will be cut without any fading.

Making EQ settings

In the Equalizer section you can tune the sound of the reverb.

The following parameters are available:

Parameter Description

EQ curve display Shows the EQ curve. You can use the EQ parameters

Activate EQ button

Low Shelf On button

Low Freq (20 to 500)

Low Gain (-24 to +24)

Mid Peak On button

Mid Freq (100 to 10000)

Mid Gain (-12 to +12)

below the display to change the EQ curve, or modify the curve manually by dragging the curve points.

This button to the right of the EQ parameters activates the EQ for the effect plug-in.

Activates the low shelf filter that boosts or cuts frequencies below the cutoff frequency by the specified amount.

Sets the frequency of the low band.

Sets the amount of cut/boost for the low band.

Activates the mid peak filter that creates a peak or notch in the frequency response.

Sets the center frequency of the mid band.

Sets the amount of cut/boost for the mid band.

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Parameter Description

Hi Shelf On button

Hi Freq (5000 to 20000)

Hi Gain (-24 to +24)

Activates the high shelf filter that boosts or cuts frequencies above the cutoff frequency by the specified amount.

Sets the frequency of the high band.

Sets the amount of cut/boost for the high band.

Loading pictures

In the Pictures section you can load graphics files to illustrate the setting, i. e. the recording location or microphone arrangement of the loaded impulse response. Up to five pictures can be loaded.

The following parameters are available:

Parameter Description

Add button Opens a file dialog where you can navigate to the graph-

Next button If several pictures are loaded, you can click this button to

Remove button Deletes the active picture. Note that this will not remove

 Pictures are only referenced by the plug-in and will not be copied to the project folder.

ics file that you want to import. JPG, GIF, and PNG file formats are supported.

display the next image.

the graphics file from your hard disk.

Making output settings

In the Output section you can control the overall level and determine the dry/wet mix.

The following parameters are available:

Parameter Description

Output activity meter

Output slider Allows you to adjust the overall output level. Out

(-24 to +12) Mix

(0 to 100)

Indicates the overall level of the impulse response and its settings.

Raises or lowers the signal output of the plug-in.

Sets the level balance between the dry and the wet signal.

Working with custom impulse responses

In addition to working with the impulse responses included with REVerence, you can import your own impulse responses and save these as programs or presets. WAV, AIF, and AIFF files with a mono, stereo, true-stereo, or multi-channel (up to 5.0) configuration are supported. If a multi-channel file contains an LFE channel, this channel is ignored.

REVerence uses the same channel width as the track it is inserted on. When importing impulse response files with more channels than the corresponding track, the plug-in only reads as many channels as needed. If the impulse re sponse file contains less channels than the track, REVerence generates the missings channels (e. g. the center channel as a sum of the left and right channels). If the rear channels are missing (when importing a stereo response file onto a 4.0 track, for example), the left and right channels are also used for the rear channels. In this case you can use the Rear offset parameter to create more spatiality.

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Importing impulse responses

To import impulse responses, proceed as follows:

1. In the program matrix, click the Import button.

2. Navigate to the file that you want to import, and click

Open.

The file is loaded into REVerence. The channels from an interleaved file are imported in the same order as in other areas of Nuendo (e. Connections window), see below.

g. the VST

3. Make the appropriate settings and add a picture, if available.

Pictures residing in the same folder as the impulse response file or in the parent folder are automatically found and displayed.

4. Click the Store button to save the impulse response and its settings as a program. That way you can recall the setup at any time.

The program slot turns blue, indicating that a program is loaded.

 When saving a program, the impulse response file itself is only referenced. It still resides in the same location as before and is not modified in any way.

5. Repeat these steps for any impulse response files that you want to work with.

REVerence reads input channels in the following order:

No. of input channels

1 L

2 L/R

3 L/R/C

4 L/R/LS/RS (if inserted on a track with a 4.0 channel con-

4 LL/LR/RL/RR (if inserted on a track with a stereo config-

5 L/R/C/LS/RS

6 L/R/C/LFE/LS/RS (LFE is being ignored.)

Channel order in REVerence

figuration, see below)

uration, see below)

True stereo

Impulse responses recorded as trues-stereo files enable you to create a very realistic impression of the corresponding room. REVerence can only process true-stereo impulse response files with the following channel configuration (in exactly that order): LL, LR, RL, RR.

The channels are defined as follows:

Channel The signal from this

source…

LL left source left microphone

LR left source right microphone

RL right source left microphone

RR right source right microphone

…was recorded with this microphone

 If your true-stereo impulse responses are only available as separate mono files, you can use the Export Audio Mixdown function in Nuendo to create REVerence compli ant interleaved files (see the chapter “Export Audio Mixdown” in the Operation Manual).

By default, REVerence automatically works in true-stereo mode when the plug-in is inserted on a stereo track and you load a 4-channel impulse response.

Therefore, if you are working with surround files, that is, 4-channel impulse responses recorded with a Quadro configuration (L/R, LS/RS), you need to insert the plug-in on an audio track with a 4.0 configuration. On a stereo track these files would be processed in true-stereo mode, too.

So how can you prevent REVerence from unintenionally processing surround files in true-stereo mode? The answer is a “Recording Method” attribute that can be written to the iXML chunk of the corresponding impulse response file. Whenever you load an impulse response with a 4-channel configuration on a stereo track, REVerence searches the iXML chunk of the file. If the plug-in finds the Recording Method attribute, the following happens:

• If the attribute is set to “TrueStereo”, the plug-in works in true-stereo mode.

• If the attribute is set to “A/B” or “Quadro”, the plug-in works in normal stereo mode and processes only the L/R channels of the surround file.

 You can use the Attribute Inspector in the MediaBay to tag your own impulse response files with the Recording Method attribute. For more information, see the chapter “MediaBay” in the Operation Manual.

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Relocating content

Once you have imported your own impulse responses in REVerence you can comfortably work with them on your computer. But what if you need to transfer your content to another computer, for example because you work some

times with a PC and sometimes with a notebook, or you need to hand over a project to a colleague in the studio?

The factory content will not be a problem since it is also present on the other computer. For these impulse responses you just need to transfer your REVerence programs and presets to be able to access your setups.

User content is a different matter, though. If you have transferred your audio files to an external drive or a differ ent hard disk location on the other computer, REVerence cannot access the impulse responses any more since the old file paths have become invalid.

To access your impulse responses again, proceed as follows:

1. Transfer you audio files to a location that you will be able to access from the second computer (i. e. an external hard disk).

If you keep the files in the same folder structure as on the first computer, REVerence will automatically find all files contained in this structure.

2. Transfer any REVerence presets or programs that you need to the second computer.

If you are unsure where the presets need to be stored, you can find the paths in the MediaBay (see the chapter “The MediaBay” in the Operation Manual).

3. Open REVerence on the second computer and try to load the preset or program that you want to work with.

The Locate Impulse Response dialog opens.

4. Navigate to the folder that contains your impulse responses. Click Open.

REVerence is now able to access all the impulse responses stored in this location.

The new path to these audio files has not been saved yet. To make the files permanently available without having to use the Locate dialog, you need to save your programs or presets under a different name.

RoomWorks

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

verberation.

The following parameters are available:

Parameter Description

Input – Lo Freq Determines the frequency at which the low-shelving filter

Input – Hi Freq Determines the frequency at which the high-shelving filter

Input –

Gain

Input – Hi Gain Controls the amount of boost or cut for the high-shelving

Reverb – Pre-Delay

Reverb – Reverb Time

Reverb – Size Alters the delay times of early reflections to simulate

Reverb – Diffusion

Reverb – Width

Reverb – Variation button

Reverb – Hold button

Damping –

Freq

takes effect. Both the high and low settings filter the input signal prior to reverb processing.

Controls the amount of boost or cut for the low-shelving filter.

filter.

Controls how much time passes before the reverb is applied. This allows you to simulate larger spaces by increasing the time it takes for first reflections to reach the listener.

Allows you to set the reverb time in seconds.

larger or smaller spaces.

Affects the character of the reverb tail. Higher values lead to more diffusion and a smoother sound, while lower val ues lead to a clearer sound.

Controls the width of the stereo image. 100 % gives you full stereo reverb. At 0

Pressing this button generates a new version of the same reverb program using altered reflection patterns. This is helpful when certain sounds are causing odd ringing or undesirable results. Creating a new variation will often solve these issues. There are 1000 possible variations.

Pressing this button freezes the reverb buffer in an infinite loop (yellow circle around button). You can create some interesting pad sounds using this feature.

Determines the frequency below which low-frequency damping will occur.

%, the reverb is all in mono.

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Parameter Description

Damping – High Freq

Damping – Low Level

Damping – High Level

Envelope – Amount

Envelope – Attack

Envelope – Release

Surround – Distance

Surround – Rotate button

Surround – Balance

Output – Mix Determines the balance of dry (unprocessed) and wet

Output –

only

Wet button

Output – Efficiency

Determines the frequency above which high-frequency damping will occur.

Affects the decay time of low frequencies. Normal room reverb decays quicker in the high- and low-frequency range than in the mid-range. Lowering the level percent age causes low frequencies to decay quicker. Values above 100 slowly than the mid-range frequencies.

Affects the decay time of high frequencies. Normal room reverb decays quicker in the high- and low-frequency range than in the mid-range. Lowering the level percent age causes high frequencies to decay quicker. Values above 100 slowly than the mid-range frequencies.

Determines how much the envelope attack and release controls affect the reverb itself. Lower values have a more subtle effect while higher values lead to a more drastic sound.

The envelope settings in RoomWorks control how the reverb will follow the dynamics of the input signal in a fashion similar to a noise gate or downward expander. Attack determines how long it takes for the reverb to reach full volume after a signal peak (in milliseconds). This is similar to a pre-delay but the reverb is ramping up instead of starting all at once.

Determines how long after a signal peak the reverb can be heard before being cut off, similar to a gate’s release time.

This control is only available for surround configurations. With this parameter you can control where the virtual lis tening position is within the room. Positive values position the listener closer to the front of the room and negative values place the listener towards the rear of the room.

This button is only available for surround configurations. When active, the perspective of the room is shifted 90°.

This control is only available for surround configurations. Balance controls the relative levels between the forward and rear speakers. Positive values favor the front speak ers and negative values favor the rear speakers. When the Rotate option is activated, these relationships will shift 90°.

(processed) signal. When RoomWorks is used as an in sert for an FX channel, you will most likely want to set this to 100

This button defeats the mix parameter, setting the effect to 100 mally be pressed when RoomWorks is being used as a send effect for an FX or group channel.

Determines how much processing power is used for RoomWorks. The lower the value, the more CPU re sources will be used, and the higher the quality of the reverb. Interesting effects can be created with very high Efficiency settings (>90

% cause low frequencies to decay more

% cause high frequencies to decay more

% or use the Send button.

% wet or affected signal. This button should nor-

%). Experiment for yourself.

Parameter Description

Output – Export button

Output – Output meter

RoomWorks SE

Determines if during audio export RoomWorks will use the maximum CPU power for the highest quality reverb. During export you may wish to keep a higher efficiency setting to achieve a specific effect. If you want the high est quality reverb during export, make sure this button is activated.

Indicates the level of the output signal.

RoomWorks SE is a “lite” version of the RoomWorks plug-in. This plug-in delivers high quality reverberation, but has fewer parameters and is less CPU demanding than the full version. The following parameters are available:

Parameter Description

Pre-Delay Controls how much time passes before the reverb is ap-

Reverb Time Allows you to set the reverb time in seconds.

Diffusion Affects the character of the reverb tail. Higher values lead

Hi Level Affects the decay time of high frequencies. Normal room

Lo Level Affects the decay time of low frequencies. Normal room

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

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

to more diffusion and a smoother sound, while lower val ues lead to a clearer sound.

reverb decays quicker in the high- and low-frequency range than in the mid-range. Lowering the level percentage causes high frequencies to decay quicker. Values above 100

% cause high frequencies to decay more slowly than

the mid-range frequencies.

reverb decays quicker in the high- and low-frequency range than in the mid-range. Lowering the level percent age causes low frequencies to decay quicker. Values above 100 slowly than the mid-range frequencies.

(processed) signal. When using RoomWorks SE inserted in an FX channel, you will most likely want to set this to 100

% cause low frequencies to decay more

% or use the Send button.

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Spatial + Panner plug-ins

This section contains descriptions of the plug-ins in the “Spatial + Panner” category.

MonoToStereo

This effect will turn a mono signal into a “pseudo-stereo” signal. The plug-in must be inserted on a stereo track playing a mono file.

The following parameters are available:

Parameter Description

Width Controls the width or depth of the stereo enhancement.

Delay Increases the amount of differences between the left and

Color Generates additional differences between the channels

Mono button Switches the output to mono, to check for possible un-

Turn clockwise to increase the enhancement.

right channels to further increase the stereo effect.

to increase the stereo effect.

wanted coloring of the sound which sometimes can occur when creating an artificial stereo image.

StereoEnhancer

This plug-in will expand the stereo width of (stereo) audio material. It cannot be used with mono files.

The following parameters are available:

Parameter Description

Width Controls the width or depth of the stereo enhancement.

Delay Increases the amount of differences between the left and

Color Generates additional differences between the channels

Mono button Switches the output to mono, to check for possible un-

Turn clockwise to increase the enhancement.

right channels to further increase the stereo effect.

to increase the stereo enhancement.

wanted coloring of the sound which sometimes can occur when enhancing the stereo image.

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SurroundPan

The SurroundPanner plug-in allows you to position mono or stereo audio in the surround field. It consists of an im age of the speaker arrangement, as defined by the output bus selected on the Output Routing pop-up menu, with the sound source indicated as a gray ball.

Although the plug-in can be used as an insert effect, it is most often inserted in the output of a track or channel. By default the SurroundPanner V5 is used for new tracks or channels, but you can switch to the SurroundPan plug-in, if needed. For more information about this, see the chap ter “Surround sound” in the Operation Manual.

 The SurroundPan plug-in was used as the default panner before Nuendo 5. In has now been replaced by the SurroundPanner V5 plug-in. However, projects created with a previous version of Nuendo still use the old Sur roundPan plug-in.

Mode – Standard/Position/Angle

The Standard Mode/Position Mode/Angle Mode switch allows you to work in three modes:

• In both Standard and Position mode, the speakers in the front

are aligned, as they would normally be in a cinema-type situation. This means that the front speakers are at a varying distance from the center. Standard mode (default) is the best mode for moving sources between speakers without level attenuation.

• Angle Mode is the traditional surround sound mixing definition.

Note that here the speakers are defined as being at equal dis tance from the center. This is not really a true representation of for example a cinema, but has still proven to work well in many situations.

Speakers

The speakers in the panel represent the chosen surround configuration.

You can turn speakers on and off by clicking them with [Alt]/[Option] pressed. When a speaker is turned off, no audio will be routed to that surround channel.

Positioning and levels

The text below assumes that the Mono/Stereo popup is set to “Mono Mix”. For information on the other modes, see below.

A sound source is positioned either by clicking or by dragging the gray “ball” around in the panel (or by using key commands, see below).

• In Standard Mode, the signal levels from the individual speakers are indicated by colored lines from the speakers to the center of the display.

Exactly how levels are handled may require some explanation:

• When you move a source around, a number will indicate the loudness in each speaker.

• This is a value in dB (decibel) and is relative to the nominal level of the source. In other words, 0.0 (dB) represents full level.

• If you position the source far enough away from a speaker, its level will drop to zero (indicated by a negative infinity symbol).

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• In Standard Mode, the signal levels from the individual

speakers are indicated by colored lines from the speakers to the center of the display.

• In Position Mode, the concentric circles will help you determine the level of the signal at a certain position.

• The yellow circle represents -3 dB below nominal level, the red circle is at -6

dB and the blue is located at -12 dB.

These are affected by attenuation, see below.

• In Angle Mode, a white arc helps you determine the perceived “range” of a source (white and blue for stereo tracks). The sound will be at its loudest in the middle of the arc and will have dropped in level towards the ends.

You can use modifier keys to restrict movement in various ways:

In Standard and Position Mode:

Key Movement restriction

[Ctrl]/[Command] Vertically only

[Ctrl]/[Command][Shift]

[Alt]/[Option] Diagonally (up left, down right)

[Ctrl]/[Command][Alt]/[Option]

[Shift] Mouse movements are scaled to allow very fine

Horizontally only

Diagonally (up right, down left)

movements.

In Angle Mode:

Key Movement restriction

[Shift] From center to perimeter only

[Ctrl]/[Command] Along the perimeter only (at current distance from

center)

The LFE encoder (all modes)

If the selected surround setup includes an LFE channel, a separate LFE level encoder is available in the SurroundPanner window. Use this to set the signal amount sent to the LFE channel. For further possibilities to set the LFE level, see the chapter “Surround sound” in the Operation Manual.

Mono/Stereo pop-up menu (all modes)

If you have a mono channel, the Mono/Stereo (Mo./St.) pop-up menu is set to Mono Mix by default. The panner will then behave as described above.

If you have a stereo channel, you have the option of using one of the three Mirror modes. Two gray balls will then appear, one for each channel (L/R). This will allow you to move the two channels symmetrically, by dragging one of them. The three modes allow you to select which axis should be used for mirroring.

• The default mode for stereo channels is the Y-Mirror mode.

• If you run a stereo signal through the panner in Mono Mix

mode, the two channels are mixed together before entering the plug-in.

• If you run a mono signal through the plug-in using one of the

stereo modes, the signal is split before entering the plug-in.

Additional parameters (Standard mode)

There is also a special set of key commands for working in the SurroundPanner window.

For a complete list of the available key commands, click on the SurroundPanner logo and then click again!

The included effect plug-ins

• Center level.

The Center control determines how center source signals are reproduced by the front speakers. With a value of 100 the center source. With a value of 0 ghost image created by the left and right speakers. Other values will pro duce a mix between these two methods.

%, the center speaker provides

%, the center source is provided by the

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• Divergence controls.

The three divergence controls determine the attenuation curves used when positioning sound sources for X-axis front (Front), X-axis back (Rear), and Y-axis (F/R, front/rear), respectively. If all three divergence controls are set to 0 sets all other speakers to zero level (-×) (except for the center speaker which depends on the center level). With higher values, the other speak ers receive a percentage of the sound source.

% (default), positioning a sound source on a speaker

Additional parameters (Position and Angle modes)

• Attenuate.

Attenuate can be used to amplify or weaken the source. Exactly what effect this has on the level in each speaker can be determined by the level read outs, the concentric circle (Position mode) and the arc (Angle mode).

• Normalize.

Normalize is a function for controlling the overall loudness from all speakers. When this is set to 1.0 (full normalization), the level from all speakers together is always exactly 0 or attenuated accordingly.

dB. The individual levels will then be boosted

Please note that this is not a dynamic feature, like compression or limiting. It is instead just a tool for scaling the nominal output levels from the surround channels.

SurroundPanner V5

For a description of the SurroundPanner V5 plug-in, see the chapter “Surround sound” in the Operation Manual.

Surround plug-ins

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

MatrixDecoder

The MatrixDecoder reverses the Encoder process performed by the MatrixEncoder (see below). It is used for monitoring how an encoded mix will sound when played back on a Pro Logic compatible system. When an encoded mix is played back via the decoder, the Lt/Rt channels are again converted to four outputs (LRCS).

This manual does not attempt to explain the full background on how Pro Logic works, but focuses on how you can use the MatrixEncoder/Decoder to produce a mix that is compatible with this standard.

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MatrixEncoder

The MatrixEncoder is intended for the Pro Logic compatible encoding of multi-channel files. This is a process where a 4-channel surround mix is “packed” into two channels for broadcasting or a two-channel version for DVDs, for exam ple. The MatrixEncoder takes four separate inputs (LRCS = Left, Right, Center, and Surround) and creates two final outputs: Left-total and Right-total (Lt and Rt).

Setting up

1. In the VST Connections window, create an output bus with the “LRCS” channel configuration and route it to the physical outputs of your audio hardware.

This is if you want to make a four-channel surround mix. If you want to make a five-channel mix, see

round format” on page 55.

2. Place the MatrixEncoder in the first “post fader” insert slot (#7) for the output bus, followed by the MatrixDecoder (#8).

Using the MatrixEncoder/Decoder

1. Set up the mix roughly the way you want it.

Use the SurroundPanner V5 to place channels in the surround mix, or assign channels to the individual LRCS outputs.

2. Activate the MatrixEncoder.

What you now hear is the encoded stereo mix, the way it will sound when played back on a normal stereo reproducer. On the MatrixEncoder con trol panel, you can adjust the Gain of the Lt/Rt output by using the fader.

“Using the MatrixEncoder with the 5.0 sur-

3. Activate the MatrixDecoder, open the control panel and click the Steering Mode button.

Now you can hear how the mix will be reproduced in surround on a Pro Logic compatible system.

• The “Steering” display shows an ‘x’ within the surround field. The position of this x sign indicates the dominant direction of the mix, sometimes referred to as the “dominance vector”. Part of the processing that is applied for various technical reasons results in the dominant channel being enhanced and the non-dominant channels being re

duced in gain.

4. By activating and deactivating the Bypass button in the MatrixDecoder, you can compare the decoded mix with the encoded stereo mix, and make adjustments in the Mixer as necessary.

The main goal is to produce a mix that sounds good in both the encoded and the decoded version. To compare the encoded or decoded mix with the unprocessed mix, switch off both the MatrixEncoder and the Decoder.

The encoding/decoding process will produce significant signal loss compared to the unprocessed mix. This is normal, and does not indicate that something is not working properly. However, with careful tweaking of the mix you can decrease the signal degradation to a much more acceptable level. You have to adjust levels and other settings before the signal runs through the MatrixEncoder, since neither the encoder or decoder can “control” the mix in any way.

5. When you are satisfied with the result, bypass the MatrixDecoder, or remove it from its effect slot.

6. Connect a master recording device to the stereo mix output and perform a mixdown as usual.

The resulting encoded stereo mix will be compatible with common home

systems that use the Pro Logic standard.

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Using the MatrixEncoder with the 5.0 surround format

There are situations when you may want to mix for several surround formats. For example, you might need to mix the same material for 5.1 and LRCS.

5.1 is similar to LRCS. Omitting the LFE channel is easy, but more of a problem is that LRCS only has one surround channel whereas 5.1 has two.

For this reason the MatrixEncoder sums up the surround channels to a mono signal.

Proceed as follows:

1. Create your mix for 5.1.

2. In the VST Connections window, create an output bus

with a “5.0” channel configuration and route it to the physical outputs of your audio hardware.

3. Run the mix through the MatrixEncoder.

First, the two surround channels are merged to make the mix compatible with LRCS. Then the four resulting signals are encoded as usual. This way, far fewer adjustments are necessary when working with 5.1 and LRCS at the same time.

Using the MatrixDecoder with the 5.0 surround format

Normally two surround speakers are used even when playing back LRCS. The two speakers then simply use the same material. The MatrixDecoder simulates this by deliv ering the surround channel to two outputs. This allows you to move between formats and listening situations with less repatching of speaker channels.

Mix6To2

Mix6To2 lets you quickly mix down your surround mix format to stereo. You can control the levels of up to six surround channels and decide for each channel up to which level it will be included in the resulting mix.

 Mix6To2 does not simulate a surround mix or add any psycho-acoustical artifacts to the resulting output – it is simply a mixer. The plug-in should be placed in one of the post-fader insert effect slots for the output bus.

For each of the surround channels the following parameters are available:

• Two volume faders that govern how much of the signal will be

included in the left and/or right channel of the output bus.

• A Link button that links the two volume faders.

• Two Invert buttons that allow you to invert the phase of the left

and right channel of the surround bus.

For the Output bus the following parameters are available:

• A Link button that links the two Output faders.

• A Normalize button. If activated, the mixed output is normal-

ized, i. e. the output level is automatically adjusted so that the loudest signal is as loud as possible without clipping.

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Mix8To2

MixConvert

Mix8To2 lets you quickly mix down your surround mix format to stereo. You can control the levels of up to eight surround channels and decide for each channel up to which level it will be included in the resulting mix.

 Mix8To2 does not simulate a surround mix or add any psycho-acoustical artifacts to the resulting output – it is simply a mixer. The plug-in should be placed in one of the post-fader insert effect slots for the output bus.

For each of the surround channels the following parameters are available:

• Two volume faders that govern how much of the signal will be included in the left and/or right channel of the output bus.

• A Link button that links the two volume faders.

• Two Invert buttons allow you to invert the phase of the left and right channel of the surround bus.

For the Output bus the following parameters are available:

• A Link button that links the two Output faders.

• A Normalize button. If activated, the mixed output is normalized, i. e. the output level is automatically adjusted so that the loudest signal is as loud as possible without clipping.

The MixConvert plug-in is similar to the Mix6To2 plug-in in that it can be used to quickly convert a multi-channel mix into another format that uses less channels when used as insert (for example converting a 5.1 surround mix to a ste reo mix). MixConvert converts surround mixes into other surround formats, for example to mix down a 7.1 Cinema surround format to a 5.1 home theater format.

There are several obvious applications for this:

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

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

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

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

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MixConvert is unique as a plug-in since it is used automati-

Indicates that MixConvert is inserted instead of the panner.

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

cally by Nuendo in certain situations (like SurroundPanner). Nuendo will substitute MixConvert for the panner in either the main channel or in the aux send panner position when an upmix or a downmix is needed. These are the possible scenarios:

• Whenever a multi-channel audio track, group channel, or FX channel (with more than three audio paths) is routed to an output bus or group channel with a different number of audio paths (e.

g. 5.1 to stereo), the MixConvert plug-in

is inserted instead of the panner in that channel.

• Whenever a multi-channel audio track, group channel, FX channel, or output bus has an aux send that is routed to a group channel or output bus with a different number of audio paths, MixConvert will be inserted instead of the aux send’s panner.

Interface

The plug-in panel has three different sections. On the left you will find the Input Configuration section with all corre sponding parameters. In the middle section the level parameters for the upmix/downmix are displayed. Above this, the preset controls can be found. On the right you will find the Output Configuration section with all corresponding parameters. Additionally, on the far left there is a Gain fader.

In the following sections all controls are explained in detail. Note that when you move the mouse pointer over a con trol, a tooltip is displayed at the bottom of the MixConvert window.

Gain section

In this section the following parameters are available:

Parameter Description

Global Gain fader

Max Output Level field

Max Output Level LED

Attenuates or increases all channels to compensate for clipping or low levels in the converted signal. Gain de pends on the input signal, the number of loudspeakers and a number of downmix parameters (see

Downmix parameters” on page 58). You can use this

fader to globally adjust the gain by ±12 dB for all channels.

This field above the Gain slider shows the maximum output level.

The LED to the right of the field indicates whether this maximum level is above 0 reset the value field and the indicator.

dB (clipping). Click the LED to

“Upmix/

Input Configuration

The input configuration is determined by the channel width of the track, group or output bus MixConvert is inserted in.

In this section the following parameters are available:

Parameter Description

Mute button – front or surround channels

Solo button – front or surround channels

Phase Shift buttons (0°, 90°, 180°, 270°)

Solo to Center button

Rear to Front button

Mutes all front or surround channels.

Soloes all front or surround channels (“Solo mode” on

page 59).

Shift the phase of the front left or right channel, or the surround left or right channel. Click the corresponding button to increase the phase by 90°. Right-click/[Ctrl]click to reset to 0°. (For more information on phase shifting, see “Phase

shifting” on page 59).

When this button is enabled, all speakers that are soloed are heard on the center channel (if available). If no center channel is present (as with stereo), the signal from the soloed channel is distributed equally to the left and right speakers.

Solos the rear channels and routes them to the front speakers.

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Parameter Description

Speaker symbols and LFE

Width controls The front and back Width controls are used to set the

Click on a speaker symbol to solo the speaker. If you hold down [Alt]/[Option] while clicking, the channel is muted. Holding down [Ctrl]/[Command] activates the exclusive solo (mutes all other channels even if they are also solo). Clicking again (without a modifier key) re sets the channel.

width of the audible panorama. At minimal width (0 the panorama is very narrow. In most cases, a setting of 50

% will be appropriate as it results in unaltered signals. Values above 50 % create an artificial widening of the panorama; similar to phase shifting. Be careful about modifying the panorama width when you want to generate matrixed downmixes. Drag the Width controls (the colored lines at the top and bottom of the Input Configuration display) to set the width. You can also click on the name of the control to open a pop-up menu from which you can select set

%, 25 %, 50 % and 100 %)

values (0

Parameter Description

Memory button You can use the Memory, Toggle, and Clear buttons to

Toggle button Using the Toggle button you can switch between the

Clear Memory Clears the temporary parameter buffer.

Surround fader Sets the level of the surround channel.

Center fader Sets the level of the center channel.

LFE fader Sets the level of the LFE channel.

Norm button Normalizes all speaker channels.

LP button Enables/disables the low-pass filter (120 Hz) applied to

toggle between two different sets of downmix parameters for direct comparison. Click the Memory button to write all current parameters to the temporary parameter buffer. Note this does not include the output configuration, which must be identical for both parameter sets.

buffered parameter set and the (changed) current para meter set.

the LFE channel.

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

Upmix/Downmix parameters

The faders in the middle section of the plug-in panel control the levels for the surround channels, front center channel and LFE channel in the upmix/downmix. The surround channels cannot be modified individually. For center and surround channels, the level can be changed between -×

dB. For the LFE channel it can be changed between

and +6

-× and +10

dB, since in some mixes the LFE channel may be attenuated by 10 dB (see “LFE channel” on page 59). The names Surround, Center and LFE refer to the corresponding channels in the input configuration.

In this section the following parameters are available:

Parameter Description

Preset pop-up menu

Save Preset button

Allows you to load a preset (see “Loading and saving pre-

sets” on page 58).

Allows you to save a preset or delete the preset shown in the Preset pop-up menu.

Output Configuration

When Nuendo automatically replaces the panner by Mix-

Convert, the output configuration is determined by the

destination of the channel or aux send. However, the output configuration can be modified when used as an insert

effect. You either change it directly in the pop-up menu at

the top of the Output Configuration section or indirectly by loading a preset.

In this section the you will find the same parameters as in the Input Configuration section (see above), except for the Width controls, and the “Solo to Center” and “Rear to Front” buttons.

General Notes

Loading and saving presets

Full presets are only available for MixConvert when it is used as an insert effect. When Nuendo automatically places MixConvert instead of a panner, the preset menu displays only presets for the current input/output configu ration.

Presets are selected and managed at the top of the middle section of the plug-in panel. The name of the selected preset is displayed in the text field. Click the symbol next to the text field to open a pop-up menu from which you can select a different preset. Which presets are available from this pop-up menu, depends on the downmix options available for the current input configuration. You save a new set of parameters by entering a new name in the text

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field and selecting Save Preset from the pop-up menu that

appears when you click the Save button. You can save up to 64 presets for every input/output configuration. To de lete a user preset, select Delete Preset from the Save pop-up menu. Note that the factory-defined presets cannot be deleted.

Phase shifting

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

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

Level

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

• The input signals must be scaled to avoid clipping.

• The number of speakers used influences the overall volume.

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

LFE channel

The LFE channel is automatically filtered using a low-pass filter. The cutoff frequency of this low-pass filter is 120 Hz, the filter slope is 12

dB/Oct. An LFE channel present in the input configuration, but not present in the output con figuration, is mixed evenly to the front-left and front-right channels since it is assumed that these will be the chan

nels using the speakers with the widest frequency range.

Solo mode

Since there is no dedicated solo bus, all solos are inplace,

e. all other (non-solo) channels are muted.

Available conversions

Not all theoretically possible combinations are actually

available in MixConvert since the plug-in is limited to

channels with 8 audio paths (this means that 10.2 or 8.1 are not supported). For a list of all available combinations,

“MixConvert Appendix” on page 85.

see

MixConvert-ControlRoom

The MixConvert-ControlRoom plug-in is identical to the MixConvert plug-in. It can convert surround mixes into other surround formats such as mixing a 7.1 Cinema sur round format down to a 5.1 home theater format. The de-

cisive difference to the MixConvert plug-in is, that this

plug-in has no latency.

MixerDelay

MixerDelay allows you to adjust and manipulate each individual channel in a surround track, group or bus.

• Above the individual channel controls you will find global buttons for turning off Mute, Solo and Input Phase switches for all channels.

For each channel the following controls are available:

Parameter Description

Mute button Allows you to mute individual channels.

Solo button Allows you to solo individual channels.

Inv button Lets you invert the phase or polarity for individual chan-

Delay slider Allows you to delay individual speaker channels. The delay

Level slider Allows you to fine-tune the volume balance between the

nels.

times are shown in milliseconds and centimeters, making this feature very useful for distance compensation when playing back surround mixes on different speaker setups, etc.

surround channels.

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Parameter Description

Volume meter Shows the level of the input signal.

Routing section

Lets you select/switch the desired outputs for the channels quickly. You can assign the same output to several channels by holding down the [Alt]/[Option] key while se lecting. Note that there are also several channel routing presets available.

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

ms) and fine adjustments are best made by numerically entering the delay time in centimeters for speaker alignment.

The MixerDelay is not a mixer – the number of outputs is the same as the number of inputs. If you need to mix down a surround signal to stereo, use the Mix6to2, Mix8to2 or MixConvert plug-ins.

SurroundDither

hence distortion. For example, when “truncating bits” as a result of moving from 24- to 16-bit resolution, quantization errors are added to an otherwise immaculate recording. By adding a special kind of noise at an extremely low level, the

effect of these errors is minimized. The added noise could be perceived as a very low-level hiss under exacting listen ing conditions. However, this is hardly noticeable and much preferred to the distortion that otherwise occurs.

When should I use SurroundDither?

• Basically anytime you mix down to a lower resolution, either in realtime (playback) or with the Export Audio Mixdown function, you should consider dithering.

• Since SurroundDither is capable of dithering up to eight channels at the same time, it is recommended to use this plug-in for surround channels.

If not, you may want to use the UV22HR instead, see “Mastering –

UV22HR” on page 28.

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

Dithering Type

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

Option Description

Off No dithering is applied.

Type 1 Try this first, it is the most “allround” type.

Type 2 This method emphasizes higher frequencies more than

Type 1.

Noise Shaping Options (Off, Type 1–3)

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

Ditherbits

SurroundDither is not an “effect” as such. Dithering is a method for controlling the noise produced by quantization errors in digital recordings. The theory behind this is that during low-level passages, only a few bits are used to rep resent the signal, which leads to quantization errors and

The included effect plug-ins

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

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

If the selected channel has less than eight sub-channels, the additional channel buttons are grayed out.

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• Above each button there is a value field that displays the bit resolution the file will be converted to.

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

An example

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

1. For the output bus, add SurroundDither to a postfader insert effect slot.

This can be one of the last two slots.

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

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

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

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

Tools – MultiScope

MultiScope can be used for viewing the waveform, phase linearity or frequency content of a signal. There are three different modes:

• Oscilloscope (Ampl.)

• Phase Correlator (Scope)

• Frequency Spectrum Analyzer (Freq.)

 The Freeze button can be used to freeze the display in all three modes. Click it again to exit freeze mode.

Oscilloscope mode (Ampl.)

• To view a signal waveform, open the MultiScope control panel and make sure that the “Ampl.” button in the lower left corner is lit.

• If the source signal is stereo you can now select either the Left or Right channel for viewing, or Stereo for both channels to be shown in the window. If it is a mono signal, this does not matter.

• If MultiScope is used with a multi-channel track or output bus, you can select any speaker channel for viewing, or All Channels to view them all at once.

• You can now adjust the Amplitude knob to increase/ decrease the vertical size of the waveform, and the Frequency knob to select the frequency area for viewing.

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Frequency Spectrum Analyzer mode (Freq.)

Phase Correlator mode (Scope)

• Click the Freq button so that it lights up.

MultiScope now divides the frequency spectrum into separate vertical bands, which allows you to get a visual overview of the different frequen cies’ relative amplitude. The frequency bands are shown left to right, starting with the lower frequencies.

• If MultiScope is used with a multi-channel track or output bus, you can select any speaker channel for viewing, or All Channels to view them all at once.

• Adjust the Amplitude knob to increase/decrease the vertical range of the bands.

• By adjusting the Frequency knob, you can divide the frequency spectrum into 8, 15, or 31 bands, or you set it to “Spectrum”, which gives you a high-resolution view.

• Use the Mode A and Mode B buttons to switch between different view modes.

Mode A is more graphically detailed, showing a solid, blue amplitude bar for each band. Mode B is less detailed, showing a continuous blue line that displays the peak levels for each band. These view modes do not have any effect if you have set the Frequency knob to “Spectrum”.

• Click the Scope button so that it lights up.

The phase correlator indicates the phase and amplitude relationship be-

tween channels in a stereo pair or a surround configuration.

For stereo pairs, the indications work in the following way:

• A vertical line indicates a perfect mono signal (the left and right channels are the same).

• A horizontal line indicates that the left channel is the same as the right, but with an inverse phase.

• A random but fairly round shape indicates a well balanced stereo signal. If the shape “leans” to the left, there is more energy in the left channel and vice versa (the extreme case of this is if one side is muted, in which case the phase meter will show a straight line, angled 90° to the other side).

• A perfect circle indicates a sine wave on one channel, and the same sine wave shifted by 90° on the other.

• Generally, the more you can see a “thread”, the more bass in the signal, and the more “spray-like” the display, the more high frequencies in the signal.

When MultiScope is used with a surround channel in Scope mode, the pop-up menu to the right of the Scope button determines the result:

• If “Stereo (Front)” is selected, the display will indicate

the phase and amplitude relationship between the front stereo channels.

• If “Surround” is selected, the display indicates the

energy distribution in the surround field.

The included effect plug-ins

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MIDI effects

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Introduction

This chapter describes the included MIDI realtime effects and their parameters.

How to apply and handle MIDI effects is described in the chapter “MIDI realtime parameters and effects” in the Operation Manual.

Arpache 5

A typical arpeggiator accepts a chord (a group of MIDI notes) as input, and plays back each note in the chord separately, with the playback order and speed set by the user. The Arpache 5 arpeggiator does just that, and more. Before describing the parameters, let’s look at how to cre ate a simple, typical arpeggio:

1. Select a MIDI track and activate monitoring (or record enable it) so that you can play “thru” the track.

Make sure that the track is properly set up for playback to a suitable MIDI instrument.

2. Select and activate the arpeggiator.

For now, use it as an insert effect for the selected track.

3. In the arpeggiator panel, use the Step Size setting to set the arpeggio speed.

The speed is set as a note value, relative to the project tempo. For example, setting Step Size to “16” means the arpeggio will be a pattern of sixteenth notes.

4. Use the Length setting to set the length of the arpeggio

notes.

This allows you to create staccato arpeggios (Length value smaller than the Step Size setting) or arpeggio notes that overlap each other (Length value greater than Step Size).

5. Set the Key Range parameter to 12.

This will make the notes arpeggiate within an octave.

6. Play a chord on your MIDI instrument.

Now, instead of hearing the chord, you will hear the notes of the chord played one by one, in an arpeggio.

7. Try the different arpeggio modes by clicking the Play

Order buttons.

The symbols on the buttons indicate the playback order for the notes (Invert, Up Only, etc.). The settings are described below.

Parameters

The Arpache 5 has the following settings:

Setting Description

Play Order buttons

Step Size Determines the speed of the arpeggio, as a note value re-

Length Sets the length of the arpeggio notes, as a note value re-

Key Range Determines the arpeggiated note range, in semitones

Allows you to select the playback order for the arpeggiated notes. The options are Normal, Invert, Up only, Down only, Random, User. If you select User, you can set the playback order manually using the 12 Play Order slots that are now shown at the bottom of the dialog.

lated to the project tempo. The range is 32T (1/32 note triplets) to “1.” (dotted note values).

lated to the project tempo. The range is the same as for the Step Size setting.

counted from the lowest key you play. This works as follows: – Any notes you play that are outside this range will be transposed in octave steps to fit within the range. – If the range is more than one octave, octave-transposed copies of the notes you play will be added to the arpeggio (as many octaves as fit within the range).

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Setting Description

Play Order slots

MIDI Thru If this is activated, the notes sent to the arpeggiator (i. e.

If the User play order is selected, you can use these “slots” to specify a custom playback order for the arpeggio notes: Each of the 12 slots corresponds to a position in the arpeggio pattern. For each slot, you specify which note should be played on that position by selecting a number. The numbers correspond to the keys you play, counted from the lowest pressed key. So, if you play the notes C3-E3-G3 (a C major chord), “1” would mean C3, “2” would mean E3, and “3” would mean G3. Note that you can use the same number in several slots, creating arpeggio patterns that are not possible us ing the standard play modes. Please note that you need to begin with the left-most slot and then fill the slots to the right.

the chord you play) will pass through the plug-in (sent out together with the arpeggiated notes).

Arpache SX

This is an even more versatile and advanced arpeggiator, capable of creating anything from traditional arpeggios to complex, sequencer-like patterns. The Arpache SX has two different modes: Classic and Sequence.

Classic vs. Sequence mode

The Classic mode determines the basic behavior of the Arpache SX. When Sequence mode is selected, the Ar pache SX uses the events of an additional MIDI part as a pattern. This pattern then forms the basis for the arpeggio, in conjunction with the MIDI input.

Classic mode

The following parameters are available:

Parameter Description

Direction This allows you to choose how the notes in the chord you

One Shot Mode

Transpose When a setting other than “Off” is selected, the arpeggio

Repeats The “Repeats” setting sets the number of transposed re-

Pitch Shift The “Pitch Shift” setting determines the transposition of

MIDI Thru If this is activated, the notes sent to the arpeggiator (i. e.

Step Size Determines the resolution of the arpeggio, i. e. its “speed”

Length Determines the length of the arpeggio notes (in fixed note

Max. Polyphony

Sort by When you play a chord into the Arpache SX, the arpeg-

Velocity Determines the velocity of the notes in the arpeggio. Us-

play should be arpeggiated. In Classic mode you can choose a value from a pop-up menu, in Sequence mode you will find additional options, see below.

Activate this option if you want the phrase to be played only once. When this option is deactivated, the phrase will be looped.

will be expanded upwards, downwards or both (depend ing on the mode). This is done by adding transposed repeats of the basic arpeggio pattern.

peats.

each repeat.

the chord you play) will pass through the plug-in (sent out together with the arpeggiated notes).

(in fixed note values or PPQ, if the PPQ button is acti vated). In Sequence mode you can also activate the “from sequence” option, see below.

values or PPQ, if the PPQ button is activated). In Se quence mode you can also activate the “from sequence” option, see below.

Determines how many notes should be accepted in the input chord. The “All” setting means there are no limita tions.

giator will sort the notes in the chord in the order specified here. For example, if you play a C-E-G chord, with “Note Lowest” selected, C will be the first note, E will be the second and G the third. This affects the result of the Arp Style setting.

ing the slider you can set a fixed velocity, or you can activate the “via Input” button to use the velocity values of the corresponding notes in the chord you play. In Sequence mode you can also activate the “from sequence” option, see below.

Sequence mode

In Sequence mode you can import a MIDI part into the Arpache SX by dragging it from the Project window and dropping it in the “Drop MIDI Sequence” field on the right of the Arpache SX panel.

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Now, the notes in the dropped MIDI part will be sorted internally, either according to their pitch (“MIDI Seq. sort by pitch” checkbox activated) or according to their play order in the part. This results in a list of numbers. For example, if the notes in the MIDI part are C E G A E C and they are sorted according to pitch, the list of numbers will read 1 2 3 4 2 1. Here, there are 4 different notes/numbers and 6 trigger positions.

The MIDI input (the chord you send into the Arpache SX) will generate a list of numbers, with each note in the chord corresponding to a number depending on the “Sort by” setting.

Furthermore, the two lists of numbers will be matched – the Arpache SX tries to play back the pattern from the dropped MIDI part but using the notes from the MIDI input (chord). The result depends on the Play Mode setting:

Option Description

Trigger The whole pattern from the dropped MIDI file will be played

Trigger Cnt. As above, but even when all keys are released, the phrase

Sort Normal Matches the notes in the MIDI input with the notes in the

Sort First As above, but if there are fewer notes in the MIDI input,

Sort Any As above, but if there are fewer notes in the MIDI input,

Arp. Style As above, but if there are fewer notes in the MIDI input,

Repeat In this mode, the chords played will not be separated into

back, but transposed according to one of the notes in the MIDI input. Which note is used for transposing depends on the Sort by setting.

continues playing from the last position (where it stopped), when a new key is pressed on the keyboard. This is typi cally used when playing “live” through the Arpache SX.

dropped MIDI part. If there are fewer notes (numbers) in the MIDI input, some steps in the resulting arpeggio will be empty.

the missing notes will be replaced by the first note.

the missing notes will be replaced by any (random) note.

the missing notes will be replaced by the last valid note in the arpeggio.

notes. Instead they will be used as is, and only the rhythm of the dropped MIDI part is used for playback.

Note also that you can choose to keep the original note timing, note length and note velocities from the dropped MIDI part, by selecting “from sequence” for the Step Size, Length and Velocity options.

Auto LFO

This plug-in works like an LFO in a synthesizer, allowing you to send out continuously changing MIDI controller messages. One typical use for this is automatic MIDI panning, but you can select any MIDI continuous controller event type. The Auto LFO effect has the following parameters:

Waveform

These settings determine the shape of the controller curves sent out. You can click on a waveform symbol, or choose a value from the pop-up menu.

Wavelength

This is where you set the speed of the Auto LFO, or rather the length of a single controller curve cycle. Using the slider or by choosing an entry from the pop-up menu, you can set this to rhythmically exact note values (or PPQ val ues if the PPQ button is activated). The lower the note value, the slower the speed. For example, if you set this to “1/8”, the waveform will be repeated every eighth note.

Controller Type

Determines which continuous controller type is sent out. Typical choices would include pan, volume and brightness, but your MIDI instrument may have controllers mapped to various settings, allowing you to modulate the synth para meter of your choice – check the MIDI implementation chart for your instrument for details!

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Density

Step display

Flam position settings, see

“Adding flams”

on page 70.

Pattern display. Here the 12 patterns are displayed for the 4 subbanks. Click on a “key” to select a pattern and on a number to select a subbank.

Swing settings, see

“The Swing setting”

on page 69.

Swing and Offset controls

Lane Name fields

Jump mode

This determines the density of the controller curves sent out. The value can be set to “small”, “medium”, or “large”, or to rhythmically exact note values (by choosing from the pop-up menu). The higher the note value, the smoother the controller curve. For example, if you set this to “1/16”, a new controller event will be sent out at every 1/16 note position.

Value Range

These two sliders are used to determine the range of controller values sent out, i. e. the “bottom” and “top” of the controller curves.

Beat Designer (Nuendo Expansion Kit only)

The Beat Designer is a MIDI pattern sequencer that allows you to create your own drum parts or “patterns” for a project. With the Beat Designer, you can quickly and eas ily set up the drums for a project, by experimenting and creating new drum sequences from scratch.

Normally, you will work on a short sequence, adjusting and modifying it while playing it back in a loop until you get the desired result. The drum patterns can then either be con verted to MIDI parts on a track or triggered using MIDI notes during playback, see

“Converting patterns into MIDI

parts” on page 71 and “Triggering patterns” on page 71.

To use the Beat Designer, select it as MIDI insert effect for a MIDI track (routed to a VSTi or an external device) or an instrument track.

Overview

When you open the control panel for the Beat Designer for the first time, it shows a display with 8 empty lanes, each containing 16 steps.

Patterns and subbanks

The Beat Designer patterns are saved as pattern banks. One pattern bank contains 4 subbanks which in turn contain 12 patterns each.

In the pattern display in the lower part of the Beat Designer, subbanks and patterns are displayed graphically. To select a subbank, click on a number (1 to 4) at the top of the dis play. To select a pattern within this subbank, click on a “key” in the keyboard display below.

Initial settings

The steps represent the beat positions in the pattern. You can specify the number of steps and the step resolution globally for a pattern:

• Click in the “Number of steps for this pattern” value field

and enter the desired value.

The maximum number of steps is 64. By default, 16 steps are shown.

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• The playback length, i. e. the note value for the steps,

Step resolutionNumber of steps for this pattern

can be specified in the Step resolution pop-up menu next to the Number of Steps setting.

On this menu, you can also set triplet values. These also affect the Swing setting, see

“The Swing setting” on page 69. The default setting is 1/16.

Selecting drum sounds

To specify a drum sound, click in the drum name field for a lane and select the desired drum sound from the pop-up menu. The available drum sounds depend on the selected drum map. If no drum map is selected for the track, the GM (General MIDI) drum names are used.

• To find the right sound, you can audition the selected drum sound by clicking the Preview Instrument button (the speaker icon).

Entering drum steps

To enter a drum step, click on the step field where you want to add a beat. You could e. g. add a snare drum on each downbeat for a lane and a bass drum on a second lane. When you click in an empty field, it becomes “filled”, indicating that you will hear a drum beat on this step.

You can also click and drag to enter a continuous range of drum steps.

 When working on drum patterns, it is a good idea to play back a section of the project in a loop while inserting the drum sounds, as this allows you to hear the result im mediately.

Removing steps

• To remove a drum step, simply click on the corresponding field again.

• To remove a range of drum steps, click and drag over them.

Setting the velocity

When entering a drum step, the velocity setting of this step is determined by where you click: Click in the upper part of a step for the highest velocity setting, in the middle section for a medium velocity and in the lower part for the lowest velocity setting. This is a quick way of roughly set ting the velocity on the fly while entering drum sounds. In the display, the different velocity settings are indicated by different colors.

• You can fine-tune the velocity setting for an existing

drum step by clicking on it and dragging up or down.

The current velocity is indicated numerically while you drag, allowing you to find the desired setting easily. The available range is from 1 to 127.

• You can also fine-tune the velocity for a range of drum

steps. Click on the first step, drag up or down to enter into velocity edit mode, and then drag sideways and up or down to modify the velocity for all the steps.

• If you hold down [Shift] while dragging up or down, you

can change the velocity for all steps on a lane.

 If you change the velocity for several steps at the same time, the relative velocity differences will be kept for as long as possible (until the minimum or maximum setting is reached).

The velocity for the steps will be increased or decreased by the same amount.

• You can also create a crescendo (or decrescendo) for

an existing range of drum steps by holding down [Alt]/ [Option], clicking on the first step, dragging up or down and then dragging to the left or right.

Editing operations

• You can move all drum steps on a lane by holding down

[Shift], clicking on the lane and dragging to the left or right.

• You can also “invert” a lane, i. e. add drum sounds for all

steps that were empty while removing all existing drum steps. This lets you create unusual rhythmic patterns. To do so, hold down [Alt]/[Option] and drag the mouse over the lane.

• You can copy the content of a lane onto another lane by

holding down [Alt]/[Option], clicking in the section to the left of the lane you want to copy and dragging to the de sired position.

When you drag, a vertical line and a plus symbol will be displayed.

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Lane handling

If you find that you have too many or too few lanes in the Beat Designer, you can add or remove them.

• To add a lane, click on the “Add Instrument Lane” button at the bottom right of the last lane shown.

• To remove a lane, click on the “Remove Instrument Lane” button in the controls section at the far right of the lane.

• You can change the order of the drum lanes by clicking in an empty area in the section to the left of a lane (i.

e. not

on a button) and dragging it to another position.

• You can mute or solo a lane by clicking the respective buttons to the left of the step display.

The lane operations always affect all patterns in the Beat Designer instance, not only the one you edit.

The Pattern Functions menu

This menu contains the following editing functions:

Option Description

Shift Left This moves all steps of the current pattern (all steps on all

Shift Right This moves all steps of the current pattern (all steps on all

Reverse Reverses the pattern, so that it plays backwards.

Copy Pattern

Paste Pattern

Clear Pattern

Insert Pattern at Cursor

lanes) to the left.

lanes) to the right.

This copies the pattern to the clipboard. Copied patterns can be pasted into another pattern subbank (see below), and even directly into the project. The default key command for this is [Ctrl]/[Command]-[C].

Allows you to paste a complete pattern, e. g. into another pattern subbank, even into another instance of the Beat De signer. This is handy when you want to create variations based on existing patterns. The default key command for this is [Ctrl]/[Command]-[V].

This resets the current pattern.

This creates a MIDI part for the current pattern and inserts it in the Project window, at the position of the project cursor

“Converting patterns into MIDI parts” on page 71).

(see also

Option Description

Insert Subbank at Cursor

Insert Pattern at Left Locator

Insert Subbank at Left Locator

Fill Loop with Pattern

This creates a number of MIDI parts (one for each used pattern in the subbank) and inserts them one after the other, starting at the project cursor (see also

into MIDI parts” on page 71).

This creates a MIDI part for the current pattern and inserts it in the Project window, at the left locator (see also

ing patterns into MIDI parts” on page 71).

This creates a number of MIDI parts (one for each used pattern in the subbank) and inserts them one after the other, starting at the left locator (see also

into MIDI parts” on page 71).

This creates a MIDI part for the current pattern and inserts it in the Project window as often as needed to fill the current loop area (the space between the left and right locators), see also

“Converting patterns into MIDI parts” on page 71.

“Converting patterns

• You can set up key commands for the Insert options

and the Fill Loop command in the Key Commands dialog.

How to set up and use key commands is described in the chapter “Key Commands” in the Operation Manual.

The Swing setting

This parameter can be used to create a swing or shuffle rhythm, which allows you to add a more human feel to drum patterns that might otherwise be too static. This is done by offsetting every second drum step for a lane. If a triplet step resolution is used, every third drum step will be offset instead.

In the lower right section of the Beat Designer panel, you can find two Swing sliders. Dragging a slider to the right will delay every second (or third, see above) drum step in the pattern. Dragging to the left will make them play a little earlier.

You can set up two swing settings with these sliders and then quickly switch between these during playback. By de fault, the first swing setting is used (activated) in all lanes, but the slider is set to zero (middle position). Change the setting for this slider to hear how the pattern’s feel changes.

Drag the upper fader to set swing setting I and the lower fader to set swing setting II.

“Convert-

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You can switch between the two swing settings using the

Click here to add up to three flams to the step.

With these sliders, you can specify the velocity for the separate flams.

Here, you can specify the flam positions for all steps containing one, two and three flams, respectively.

Swing buttons to the right of the step display.

Click on the buttons to select the respective swing setting or click on a selected button to deactivate swing for this lane.

Adding flams

The Flam parameter lets you add flams (short secondary drum hits just before or after the actual main drum beat).

You can add up to three flams for each pattern step:

1. Click in the lower left corner of the step you want to add a flam to.

Little squares appear in the step when you point with the mouse at the step. After you clicked, the first square becomes filled to indicate that you added a flam.

2. Click again to add the second and third flam, if needed.

3. In the lower left section of the Beat Designer panel you can make settings for the flams you created.

• When you add flams before the very first drum step in a

pattern, this is indicated in the display by a small arrow in the top left corner of this step. This indicates that you have to treat this pattern with special care in playback and ar

ranging. Starting playback at the normal pattern start would result in these flams not being played.

• Use the vertical sliders to the right of the flam sliders to set the velocity for the flams.

4. Start playback to hear the flams you created.

Offsetting lanes

To the right of the step display, you can find the Offset sliders for the lanes. These allow you to offset all drum steps on this lane. Drag a slider to the left to make the drum steps start a little earlier and to the right to let them start later.

Playing e. g. the bass drum or snare a little earlier allows you to add more “urgency” to the drums, delaying these drum sounds will result in a more relaxed drum pattern. Experiment with the settings to find out which fit best in your project.

Note that this function can also be used to correct faulty drum samples: If a drum sound has an attack that is slightly late, simply adjust the Offset slider for the lane.

Saving and loading presets

You can save all 48 Beat Designer patterns as a pattern bank. This can then be loaded in other projects. Pattern banks contain all the step and lane settings for a pattern (Mute and Solo, number and order of the lanes, pitch, etc.).

To save a pattern bank, proceed as follows:

1. In the Beat Designer, click on the Preset Management button to the right of the preset name field.

• The first (topmost) Position slider specifies the flam position for all steps containing one single flam, the second slider the flam positions for all steps containing two flams, and the third slider the flam position for all steps containing three flams.

• Drag a Position slider to the left to add the flams before the drum step and to the right to add them after the step.

2. On the pop-up menu select “Save Preset”.

A dialog appears.

3. Enter a name for the preset and click OK.

The preset will now be available on the Preset browser, in the MediaBay and on the Load Track Preset pop-up menu in the Inspector.

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Pattern banks are handled much like track presets in the

Click here and drag to convert this subbank into separate MIDI parts.

Click here and drag to convert this pattern into a MIDI part.

Jump mode is activated.

MediaBay. For further information, refer to the chapters “The MediaBay” and “Working with track presets” in the Operation Manual.

Using the drum patterns in your project

You can use the drum patterns created with the Beat Designer in two ways: either by converting them to MIDI parts on a MIDI or instrument track or by triggering the dif ferent patterns using MIDI notes.

Converting patterns into MIDI parts

You can convert the drum patterns created in the Beat Designer into a MIDI part by dragging them into the Project window.

Proceed as follows:

1. Set up one or more patterns of the same subbank.

2. In the lower part of the window, click on a pattern or

subbank and drag it at the desired position onto a MIDI or instrument track in the Project window.

If you drag the pattern or subbank to an empty area in the Project window, a new MIDI track is created. This will be an exact copy of the original track for which you opened the Beat Designer.

• If you drag a single pattern into the Project window, one MIDI part is created containing the drum sounds of the pattern.

• If you drag a subbank into the Project window, several MIDI parts (one for each used pattern in the subbank) are created and inserted one after the other in the project.

You can also use the Pattern Functions menu to insert patterns or subbanks into the project, see “The Pattern

Functions menu” on page 69.

When you have created MIDI parts for your drum patterns this way, make sure to deactivate the Beat Designer, to avoid doubling of the drums. The Beat Designer will continue to play as long as it is acti

vated.

• If you import patterns that sound before the first step (due to flams or lane offsets), the MIDI part will be length ened accordingly.

The inserted MIDI parts can now be edited as usual in the project. You can e.

g. fine-tune your settings in the Drum

Editor.

 Once a pattern is converted into a MIDI part, it cannot be opened in the Beat Designer again.

Triggering patterns

When you want to be able to modify your drum patterns in the Beat Designer while working on the project, you cannot convert them into parts, as these cannot be opened again in the Beat Designer. Instead, you can trigger the patterns from within the project.

You can trigger the patterns in the Beat Designer using Note On events. These can either be events on a MIDI track or be played live via a MIDI keyboard. Which pattern will be triggered depends on the pitch of the MIDI notes. The trigger range is four octaves starting with C1 (i. to B4).

Proceed as follows:

1. Open the Beat Designer for a track.

Again, this can be a MIDI or an instrument track.

2. Click on the Jump field to activate Jump mode.

In this mode, a MIDI note-on event will trigger a new pattern.

e. C1

Only the used patterns in a subbank are inserted, i. e. if you did not enter drum steps in a pattern, this will not be converted into a MIDI part.

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• When you want to trigger the patterns using a MIDI part containing trigger events, you can specify whether the pattern will be switched directly (at the moment the event is received) or at the next bar: Click on the field to the right (where it says “Now”) to activate the immediate switching of patterns. When Now is deactivated, patterns will switch at the beginning of the next bar in the project.

• When you want to trigger the patterns “live” via a MIDI keyboard, the new patterns are always played when the next bar in the project is reached.

Switching immediately would always produce an undesirable interruption in playback.

Now, you can trigger the patterns in the following way:

1. Play back the project and press a key on your MIDI keyboard to trigger the next pattern.

The pattern will start at the next bar line.

2. Create a MIDI part and enter notes at the positions in the project where you want to switch patterns.

Depending on the Jump mode setting, the new pattern will be played directly or start at the following bar.

• You can also drag a pattern or subbank into the Project when Jump mode is active to automatically create MIDI parts containing the trigger events.

 When triggering a pattern that contains sound before the first step (due to flams or lane offsets), these are taken into account as well.

Chorder

The Chorder is a MIDI chord processor, allowing you to assign complete chords to single keys in a multitude of variations. These can then be played back live or using re corded notes on a MIDI track.

There are three main operating modes: “All Keys”, “One Octave”, and “Global Key”. You can switch between these modes using the Chords pop-up, see below.

For every key you can record up to eight different chords or variations on so-called “layers”. This is described in detail in the section “Using Layers” on page 73.

Operating modes

In the lower left section of the Chorder window, you can choose an option from the Chords pop-up menu to decide which keys in the keyboard display will be used to record your chords.

Global Key

In this mode, you can assign chords to each key on the keyboard display. When you play any of these keys, you will hear the assigned chords instead.

One Octave

The One Octave mode is similar to the All Keys mode, but you can only set up chords for each key of a single octave (that is, up to eight different chords on twelve keys). When you play a note (e. a transposed version of the chords set up for this key.

Global Key

In Global Key mode, you can set up chords for a single key only. These chords (that you recorded on C3) are then played by all keys on the keyboard, but transposed ac cording to the note you play.

g. C) on a different octave, you will hear

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The chord indicator lane

The chord indicator lane in One Octave mode with chords set up for 5 of the 12 available trigger keys.

At the top of the keyboard display you will find a thin lane with a small rectangle for each key that you can use to record a chord. These rectangles are shown in blue for all keys that already have chords assigned to them.

 In Global Key mode the C3 key has a special marking instead since this is the only key used in this mode.

Entering chords

To enter chords you need to switch to Learn mode. In this mode a transparent red bar indicates which element is ready for “learning” a note or chord. When you choose the trigger note for a chord, for example, the keyboard display is shown in red.

The keyboard display in Learn mode

The second layer in Learn mode

Proceed as follows:

1. Click the Learn button at the top of the Chorder window to activate Learn mode.

The chord indicator lane is now tinted red, indicating that it is active.

2. Select the key to which you want to assign a chord by clicking on it on the keyboard display, or by pressing the key on a connected MIDI keyboard.

The red bar will now move to the first layer, indicating that you are ready to record the first chord.

 In Global Key mode you do not have to choose a trigger key. The first layer is activated directly.

3. Play a chord on the MIDI keyboard and/or use the mouse to enter or change the chord in the layer display.

Any notes you enter are immediately shown in the Chorder display. The notes are shown in different colors, depending on the pitch.

• If you are entering chords via a MIDI keyboard, the Chorder will learn the chord as soon as you release all keys of your MIDI keyboard simultaneously.

As long as a key is pressed, you can continue looking for the right chord.

• If more than one layer is shown, the Chorder will jump automatically to the next layer where you can record an

other chord.

When all the layers for a key are filled, the red bar will jump back to the keyboard display so that you can choose a different trigger key (in Global Key mode the Learn mode is deactivated instead).

• If you are entering chords with the mouse, the Chorder will not jump to the next layer automatically.

You can select/deselect as many notes as you wish and then click on another layer or deactivate the Learn mode to continue.

4. Repeat the above with any other keys you wish to use.

Using Layers

The Layers pop-up menu at the bottom right of the window allows you to set up chord variations in the layer display above the keyboard. This works with all three modes and provides up to eight variations for each assignable key (that is, a maximum of 8 different chords in Global Key mode, 12 x 8 chords in One Octave mode and 128 x 8 chords in All Keys mode).

The different layers can be triggered by velocity or interval. Proceed as follows to set up your layers:

1. Open the Layers pop-up menu and select Velocity or Interval. Set this to Single Mode if you want to set up only one chord per key.

2. Use the slider below the Layers pop-up menu to specify how many variations (layers) you want to use.

3. Enter the chords as described above.

4. Now you can play the keyboard and trigger the varia-

tions according to the selected layer mode.

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The layer modes work as follows:

Trigger mode Description

Velocity The full velocity range (1–127) is divided into “zones”,

Interval In this mode, the Chorder will play one chord at a time –

Single Mode Select this if you do not wish to use different layers.

according to the number of layers you specified. For ex ample, if you are using two variations (Number of Layers is set to 2) there will be two velocity “zones”: 1–63 and 64–127. Playing a note with velocity 64 or higher will trigger the second layer, while playing a softer note will trigger the first layer. Using the “Velocity spread” slider at the bottom right of the window, you can change the velocity ranges of the layers so that a different layer will be activated using the same velocity value.

you cannot play several different chords simultaneously. When the Interval mode is selected, you press two keys on your keyboard to trigger the desired layer, with the lower key determining the base note for the chord. The layer number will be the difference, i. tween the two keys. To select layer 1, press a key one semitone higher than the base note, for layer 2, press a key two semitones higher, and so on.

e. the interval, be-

Playstyle

From the Playstyle pop-up menu at the bottom of the pane you can choose one of seven different styles that determine

in which order the individual notes of the chords are played back.

The following options are available:

Playstyle Description

simultaneous In this mode all notes are played back simultaneously.

fast up In this mode a small arpeggio is added, starting with

slow up Similar to “fast up”, but using a slower arpeggio.

fast down Similar to “fast up”, but starting with the highest note.

slow down Similar to “slow up”, but starting with the highest note.

fast random In this mode the notes are played back in a rapidly

slow random Similar to “fast random”, but the note changes occur

the lowest note.

changing random order.

more slowly.

Empty layers

If you enter less chords than layers present for a key, these layers will be filled automatically when you end the Learn mode.

This works according to the following rules:

• Empty layers are filled from bottom to top.

• If there are empty layers below the first layer with a

chord, these are filled from top to bottom.

An example:

If you have a setup with 8 layers, and you enter the chord C in layer 3 and G7 in layer 7, you get the following result: chord C in layers 1 to 6 and G7 in layers 7 and 8.

Resetting layers

In Learn mode, you can use the “Reset layers” button at the top left of the Chorder window to delete all notes in the different layers for the selected trigger key.

Compressor

This MIDI compressor is used for evening out or expanding differences in velocity. Though the result is similar to what you get with the Velocity Compression track para meter, the Compress plug-in presents the controls in a manner more like regular audio compressors.

The following parameters are available:

Parameter Description

Threshold Only notes with velocities above this value will be affected

Ratio This determines the rate of compression applied to the

by the compression/expansion.

velocity values above the threshold level. Ratios greater than 1:1 result in compression (i. locity) while ratios lower than 1:1 result in expansion (i. e. greater difference in velocity). What actually happens is that the part of the velocity value that is above the threshold value is divided by the ratio value.

e. less difference in ve-

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Parameter Description

Gain This adds or subtracts a fixed value from the velocities.

Since the maximum range for velocity values is 0–127, you may need to use the Gain setting to compensate, keeping the resulting velocities within the range. Typi cally, you would use negative Gain settings when expanding and positive Gain settings when compressing.

Context Gate

The Context Gate allows for selective triggering/filtering of MIDI data. It features two modes: in Poly Mode the Context Gate recognizes certain chords that are played and in Mono Mode only certain MIDI notes are let through. These modes can be used for context selective control of MIDI devices and are, for example, very useful in certain live scenarios.

The following parameters are available:

Poly Mode – Polyphony Gate

This allows you to filter MIDI according to the number of pressed keys within a given key range. This can be used independently or in conjunction with the Chord Gate function.

• The Key Range Limit sliders are used to set the key range.

Only notes within this range will be let through.

• The “Minimum Polyphony” value field allows you to specify the minimum number of notes required to open the gate.

Poly Mode – Chord Gate

When Chord Gate is activated, only notes in recognized chords are let through.

• Two Recognition modes are available: Simple and Normal. In Simple mode, all standard chords (major/minor/b5/ dim/sus/maj7 etc.) are recognized, whereas Normal mode takes more tensions into account.

Mono Mode – Channel Gate

When this is activated, only single note events in a specified MIDI channel are let through, which can be used with MIDI controllers that can send MIDI over several channels simultaneously, for example guitar controllers which send data for each string over a separate channel.

• You can set Mono Channel to a specific channel (1 to

16), or to “Any”, i. e. no channel gating.

Mono Mode – Velocity Gate

This can be used independently or in conjunction with the Channel Gate function. Played notes will sound (no noteoff message) until a note is played inside the set range (and additionally the set Channel Gate channel, if checked).

• The Key Range Limit sliders are used to set the key range.

Only notes within this range will be let through.

• Notes below the Minimum Velocity threshold value will be gated.

Auto Gate Time

If there is no input activity, all resounding notes are sent a note-off message after the set time, in seconds or milliseconds.

Panic Reset button

Sends an “All Notes Off” message over all channels, in case of hanging notes.

Learn Reset button

When this is activated, you can specify a Reset trigger event via MIDI. Whenever this specific MIDI event is sent, it triggers an “All Notes Off” message. When you have set the Reset event, the Learn button should be deactivated.

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Application examples

Poly Mode

In Poly mode, you could use the Context Gate to accompany yourself during a live guitar performance using a VST instrument. To do this, you might use a guitar to MIDI converter: You could then program the Context Gate, for example, to allow only those notes to pass the gate that are part of a four-note chord. During your performance you would then play a four-note chord every time that you want to trigger the VST instrument. The instrument will play until the Auto Gate Time is reached and fade out. For more complex performances this can be combined with an arpeggiator, without having to use external pedals to trigger the effect.

Mono Mode

In Mono Mode you could use the Context Gate to trigger variations played with a drum machine/VST instrument. To do this, you will need a guitar to MIDI converter: You could then filter the MIDI channel using the Input Transformer (optional) and program the Context Gate to allow only certain notes on your guitar to pass the gate (e. ning at the 12th band). When you now play one of these notes, the note-off command will not be send out and the corresponding note will sound until the note is played again, a new note is let through, or the Auto Gate Time is reached. This way you can trigger lots of different effects or notes using the high notes on you guitar without having to use an additional MIDI instrument.

g. begin-

Micro Tuner

The Micro Tuner lets you set up a different microtuning scheme for the instrument, by detuning each key.

• Each Detune slider corresponds to a key in an octave (as indi-

cated by the keyboard display). Adjust a Detune field to raise or lower the tuning of that key, in cents (hundreds of a semitone).

• By keeping the [Alt]/[Option] key pressed, you can adjust all

keys by the same amount.

The Micro Tuner comes with a number of presets, including both classical and experimental microtuning scales.

MIDI Control

Density

This generic control panel affects the “density” of the notes being played from (or thru) the track. When this is set to 100 Density setting below 100 % will randomly filter out or “mute” notes. Raising the setting above 100 randomly add notes that have been played before.

%, the notes are not affected. Lowering the

% will instead

This generic control panel allows you to select up to eight different MIDI controller types, and use the value fields or sliders (which are displayed when you click on a value field while holding down the [Alt]/[Option] key) to set values for these. A typical use for this would be if you are using a MIDI instrument with parameters that can be controlled by

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MIDI controller data (e. g. filter cutoff, resonance, levels, etc.). By selecting the correct MIDI controller types, you can use the plug-in as a control panel for adjusting the sound of the instrument from within Nuendo, at any time.

• To select a controller type, use the pop-up menus to the right.

• To deactivate a controller slider, set it to “Off” (drag the slider all the way down).

For example, setting this to -2 will cause the first echo note to have a pitch two semitones lower than the original note, the second echo note two semitones lower than the first echo note, and so on.

Repeats

This is the number of echoes (1 to 12) from each incoming note.

MIDI Echo

This is an advanced MIDI Echo, which will generate additional echoing notes based on the MIDI notes it receives. It creates effects similar to a digital delay, but also features MIDI pitch shifting and much more. As always it is impor tant to remember that the effect does not “echo” the actual audio, but the MIDI notes which will eventually produce the sound in the synthesizer.

The following parameters are available:

Velocity Offset

This parameter allows you to raise or lower the velocity values for each repeat so that the echo fades away or increases in volume (provided that the sound you use is velocity sensitive). For no change of velocity, set this to 0 (middle position).

Pitch Offset

If you set this to a value other than 0, the repeating (echoing) notes will be raised or lowered in pitch, so that each successive note has a higher or lower pitch than the previous. The value is set in semitones.

Beat Align

During playback, the Beat Align parameter quantizes the position of the first echo note. You can either set this to “rhythmically exact” values (displayed as note values – see the table below) or activate the PPQ button and choose a PPQ value.

Setting this to “1/8”, for example, will cause the first echo note to sound on the first eighth position after the original note.

 The echo time can also be affected by the Delay Decay parameter.

 During live mode, this parameter has no effect since the first echo will always be played together with the note event itself.

Delay

The echoed notes will be repeated as set up with this parameter. You can either set this to “rhythmically exact” val-

ues (displayed as note values – see the table below) or

activate the PPQ button and choose a PPQ value. This makes it easy to find rhythmically relevant delay values, but still allows experimental settings in between.

Delay Decay

This parameter lets you adjust how the echo time should be changed with each successive repeat. The value is set as a percentage.

• When set to 100 % (middle position) the echo time will be the

same for all repeats (as set with the Delay parameter).

• If you raise the value above 100 %, the echoing notes will play

with gradually longer intervals (i. e. the echo will become slower).

• If you lower the value below 100 %, the echoing notes will be-

come gradually faster, like the sound of a bouncing ball.

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Length

This sets the length of the echoed notes. This can either be identical with the length of the original notes (parameter set to its lowest value) or the length you specify manually. You can either set this to “rhythmically exact” values (displayed as note values – see the table below) or activate the PPQ button and choose a PPQ value.

 The length can also be affected by the Length Decay parameter.

Length Decay

This parameter lets you adjust how the length of the echoed notes should change with each successive repeat. The higher the setting (25–100), the longer the echoed notes will be, compared to their original notes.

About ticks and note values

The timing and position-related parameters (Delay, Length and Beat Align) can all be set in ticks (or PPQ which denotes the same thing here). There are 480 ticks to each quarter note. While the parameters allow you to step between the rhythmically relevant values (displayed as note values), the following table can also be of help, showing you the most common note values and their correspond ing number of ticks:

Note Value Ticks

1/32 note 60

1/16 note triplet 90

1/16 note 120

1/8 note triplet 160

1/8 note 240

Quarter note triplet 320

Quarter note 480

Half note 960

MIDI Modifiers

This plug-in is essentially a duplicate of the MIDI Modifiers section in the Inspector. This can be useful, for example, if you need extra Random or Range settings.

The MIDI Modifiers effect also includes an additional function that is not available among the track parameters:

Scale Transpose

This allows you to transpose each incoming MIDI note, so that it fits within a selected musical scale. The scale is specified by selecting a key (C, C#, D, etc.) and a scale type (major, melodic or harmonic minor, blues, etc.).

 To turn Scale Transpose off, select “No Scale” from the Scale pop-up menu.

MIDI Monitor

The MIDI Monitor is used to monitor incoming MIDI events. You can choose whether to analyze live or playback events and which types of MIDI data are to be monitored. Use this, for example, to analyze which MIDI events are being gener ated by a MIDI track, or to find “suspicious” events, such as notes with velocity 0 that certain MIDI devices might fail to interpreted as note-off events.

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Inputs section

In this section you can choose whether to monitor Live Events or Playback Events.

Show section

Here, you can activate/deactivate the different types of MIDI events, e.

g. notes or program change events. If you choose the Controller option you can also define which type of controller to monitor.

Data table

In the table in the lower section of the window, you will see detailed information about the monitored MIDI events.

Buffer pop-up menu

In the Buffer pop-up menu you can set the buffer size to 100, 1000 or 10000 events. This is the maximum number of events that is kept in the list of monitored events. Once this list is full, the oldest entries will be deleted when new events are received.

 The larger the buffer, the more processing resources are required. To avoid a negative impact on your system’s performance, make sure to use the smallest possible buffer size.

Export function

Click the Export button to export the monitoring data as a simple text file.

Note to CC

This effect will generate a MIDI continuous controller event for each incoming MIDI note. The value of the controller event corresponds to the velocity of the MIDI note, which is then used to control the selected MIDI controller (by default CC 7, Main Volume). For each note end, another controller event with the value 0 is sent. The incoming MIDI notes pass through the effect unaffected.

The purpose of this plug-in is to generate a gate effect. This means that the notes played are used to control something else. For example, if Main Volume (CC 7) is selected, notes with low velocity will lower the volume in the MIDI instrument, while notes with a high velocity will raise the volume.

Note that a controller event is sent out each time a new note is played. If high and low notes are played simultaneously, this may lead to confusing results. Therefore, the Note to CC effect is best applied to monophonic tracks (playing one note at a time).

Record events button

Use this button to the left of the Inputs section to start or stop the monitoring of MIDI events.

Clear list button

The Clear List button to the left of the Show section allows you to clear the table of recorded MIDI events.

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Number of steps

SwingStep

size

Pattern selector

Controller pop-up menu

Shift octave up/down

Shift steps left/right

Quantizer

Quantizing is a function that changes the timing of notes by moving them towards a “quantize grid”. This grid may consist of e. notes would all get perfect sixteenth note timing), but could also be more loosely related to straight note value positions (applying a “swing feel” to the timing, etc.).

 The main Quantize function in Nuendo is described in the Operation Manual.

While the Quantize function on the MIDI menu applies the timing change to the actual notes on a track, the Quantizer effect allows you to apply quantizing “on the fly”, changing the timing of the notes in real time. This makes it easier to try out different settings when creating grooves and rhythms. Note however, that the main Quantize function contains settings and features that are not available in the Quantizer.

The Quantizer has the following parameters:

Parameter Description

Quantize Note This sets the note value on which the quantize grid is

Swing This allows you to offset every second position in the grid,

Strength This determines how close the notes should be moved to

Delay This delays (positive values) or advances (negative val-

Realtime quantize

g. straight sixteenth notes (in which case the

based. Straight notes, triplets and dotted notes are avail able. For example, “16” means straight sixteenth notes and “8T” means eighth note triplets.

creating a swing or shuffle feel. The value is a percentage – the higher you set this, the farther to the right every even grid position is moved.

the quantize grid. When set to 100 forced to the closest grid position; lowering the setting will gradually loosen the timing.

ues) the notes in milliseconds. Unlike the Delay setting in the Track Parameters, this delay can be automated.

During live mode this option can be used to change the timing of the notes played so that they fit the quantize grid.

%, all notes will be

StepDesigner

The StepDesigner is a MIDI pattern sequencer that sends out MIDI notes and additional controller data according to the pattern you set up. It does not make use of the incom ing MIDI, other than automation data (such as recorded pattern changes).

Creating a basic pattern

1. Use the Pattern selector to choose which pattern to create.

Each StepDesigner can hold up to 200 different patterns.

2. Use the “Step size” setting to specify the “resolution” of the pattern.

In other words, this setting determines how long each step is. For exam-

ple, if this is set to “1/16” each step will be a sixteenth note.

3. Specify the number of steps in the pattern with the “Number of steps” setting.

As you can see in the note display, the maximum number of steps is 32. For example, setting “Step size” to 16 and “Number of steps” to 32 would create a two bar pattern with sixteenth note steps.

4. Click in the note display to insert notes.

You can insert notes on any of the 32 steps, but the StepDesigner will only play back the number of steps set with the Step size parameter.

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• The display spans one octave (as indicated by the pitch list to the left). You can scroll the displayed octave up or down by clicking in the pitch list and dragging up or down.

This way you can insert notes at any pitch. Note that each step can contain one note only – the StepDesigner is monophonic.

Click and drag to view other octaves.

• To remove a note from the pattern, click on it again.

5. On the Controller pop-up menu, select Velocity.

This pop-up menu determines what is shown in the lower controller display.

6. Adjust the velocity of the notes by dragging the velocity bars in the controller display.

7. To make notes shorter, select “Gate” on the Controller pop-up menu and lower the bars in the controller display.

When a bar is set to its maximum value (fully up), the corresponding note will be the full length of the step (as set with the Step size parameter).

8. To make notes longer, you can tie two notes together. This is done by inserting two notes and clicking in the Tie column for the second note.

When two notes are tied, the second note will not be triggered – the previous note is lengthened instead. Also, the tied (second) note will au tomatically get the same pitch as the first note. You can add more notes and tie them in the same way, creating longer notes.

9. If you now start playback in Nuendo, the pattern will play as well, sending out MIDI notes on the track’s MIDI output and channel (or, if you have activated the StepDe signer as a send effect, on the MIDI output and channel selected for the send in the Inspector).

Adding controller curves

The Controller pop-up menu has two more items: two controller types.

• You can select which two controller types (filter cutoff, resonance, volume, etc.) should be available on the popup menu by clicking the Setup button and selecting controllers from the lists that appears.

This selection is global, i. e. it applies to all patterns.

• To insert controller information in a pattern, select the desired controller from the pop-up menu and click in the controller display to draw events.

The MIDI controller events will be sent out during playback along with the notes.

 If you drag a controller event bar all the way down, no controller value is sent out on that step.

Other pattern functions

The following functions make it easier to edit, manipulate and manage patterns:

Function Description

Shift Octave up/down

Shift Steps left/right

Reverse Reverses the pattern, so that it plays backwards.

Copy/Paste Allows you to copy the current pattern and paste it in an-

Reset Clears the pattern, removing all notes and setting control-

Randomize Generates a completely random pattern – useful for ex-

Swing The Swing parameter allows you to offset every second

Presets Handling of presets is described in the chapter “MIDI real-

These buttons allow you to shift the entire pattern up or down in octave steps.

Moves the pattern one step to the left or right.

other pattern location (in the same StepDesigner instance or another).

ler values to default.

perimenting.

step, creating a swing or shuffle feel. The value is a per centage – the higher you set this, the farther to the right every even step is moved.

time settings” in the Operation Manual. Note that a stored Preset contains all 200 patterns in the StepDesigner.

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Automating pattern changes

You can create up to 200 different patterns in each StepDesigner – just select a new pattern and add notes and controllers as described above.

Typically, you want the pattern selection to change during the project. You can accomplish this by automating the Pattern selector, either in real time by activating the Write automation and switching patterns during playback or by drawing in the automation track for the StepDesigner’s MIDI track. Note that you can also press a key on your MIDI keyboard to change patterns. For this, you have to set up the StepDesigner as an insert effect for a record enabled MIDI track. Press C1 to select pattern 1, C#1 to select pattern 2, D1 to select pattern 3, D#1 to select pat tern 4 and so on. If you want, you can record these pattern changes as note events on a MIDI track.

Proceed as follows:

1. Select the desired MIDI track or create a new one and activate the StepDesigner as an insert effect.

2. Set up several patterns as described above.

3. Press the Record button and press the desired keys

on your keyboard to select the corresponding patterns.

The pattern changes will be recorded on the MIDI track.

4. Stop recording and play back the MIDI track.

You will now hear the recorded pattern changes.

 This will only work for the first 92 patterns.

Track Control

The Track Control effect contains three ready-made control panels for adjusting parameters on a GS or XG compatible MIDI device. The Roland GS and Yamaha XG protocols are extensions of the General MIDI standard, allowing for more sounds and better control of various instrument settings. If your instrument is compatible with GS or XG, the Track Controls effect allows you to adjust sounds and effects in your instrument from within Nuendo.

Selecting a control panel

At the top of the Track Controls effect window you will find a pop-up menu. This is where you select which of the available control panels to use:

Control panel Description

GS 1 Effect sends and various sound control parameters for

XG 1 Effect Sends and various sound control parameters for

XG 2 Global settings (affecting all channels) for instruments

use with instruments compatible with the Roland GS standard.

use with instruments compatible with the Yamaha XG standard.

compatible with the Yamaha XG standard.

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About the Reset and Off buttons

Regardless of the selected mode, you will find two buttons labeled “Off” and “Reset” at the top of the control panel:

• Clicking the Off button will set all controls to their lowest value, without sending out any MIDI messages.

• Clicking the Reset button will set all parameters to their default values, and send out the corresponding MIDI messages.

For most parameters, the default values will be zero or “no adjustment”, but there are exceptions to this. For example, the default “Send 1” setting is 64.

GS 1

The following controls are available when the GS 1 Controls mode is selected:

Control Description

Send 1 Send level for the reverb effect.

Send 2 Send level for the chorus effect.

Send 3 Send level for the “variation” effect.

Attack Adjusts the attack time of the sound. Lowering the value

Decay Adjusts the decay time of the sound. Lowering the value

Release Adjusts the release time of the sound. Lowering the value

Cutoff Adjusts the filter cutoff frequency.

Resonance Adjusts the filter resonance.

Express Allows you to send out expression pedal messages on

Ch. Press. Allows you to send out aftertouch (channel pressure)

Breath Allows you to send breath control messages on the

Modul. Allows you to send modulation messages on the track’s

shortens the attack, while raising it gives a slower attack. Middle position (64) means no adjustment is made.

shortens the decay, while raising it makes the decay longer.

shortens the release, while raising it makes the release time longer.

the track’s MIDI channel.

messages on the track’s MIDI channel. This is useful if your keyboard cannot send aftertouch, but you have sound modules that respond to aftertouch. The default value for this parameter is zero.

track’s MIDI channel.

MIDI channel (just as you normally do with a modulation wheel on a MIDI keyboard).

XG 1

The following controls are available when the XG 1 mode is selected:

Control Description

Send 1 Send level for the reverb effect.

Send 2 Send level for the chorus effect.

Send 3 Send level for the “variation” effect.

Attack Adjusts the attack time of the sound. Lowering this value

Release Adjusts the release time of the sound. Lowering this value

Harm.Cont Adjusts the harmonic content of the sound.

Bright Adjusts the brightness of the sound.

CutOff Adjusts the filter cutoff frequency.

Resonance Adjusts the filter resonance.

shortens the attack, while raising it gives a slower attack. Middle position means no adjustment is made.

shortens the release, while raising it makes the release time longer. Middle position means no adjustment is made.

XG 2

In this mode, the parameters affect global settings in the instrument(s). Changing one of these settings for a track will in fact affect all MIDI instruments connected to the same MIDI output, regardless of the MIDI channel setting of the track. Therefore, to avoid confusion it might be a good idea to create an empty track and use this only for these global settings.

The following controls are available:

Control Description

Eff. 1 This allows you to select which type of reverb effect

Eff. 2 This allows you to select which type of chorus effect

Eff. 3 This allows you to select one of a large number of “varia-

Reset Sends an XG reset message.

MastVol This is used to control the Master Volume of an instrument.

should be used: No effect (the reverb turned off), Hall 1–2, Room 1–3, Stage 1–2 or Plate.

should be used: No effect (the chorus turned off), Chorus 1–3, Celeste 1–3 or Flanger 1–2.

tion” effect types. Selecting “No Effect” is the same as turning off the variation effect.

Normally you should leave this in its highest position and set the volumes individually for each channel (with the vol ume faders in the Nuendo mixer or in the Inspector).

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Transformer

The Transformer is a realtime version of the Logical Editor. With this you can perform very powerful MIDI processing on the fly, without affecting the actual MIDI events on the track.

The Logical Editor is described in the corresponding chapter in the Operation Manual. As the parameters and functions are almost identical, the descriptions for the Logical Editor also apply to the Transformer. Where there are differences between the two, this is clearly stated.

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MixConvert Appendix

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Available conversions

The following tables list all combinations when MixConvert is used. Each column is an output configuration and each row is an input configuration. When MixConvert is used as an insert effect, only downmix is possible. In this case, the number of outputs can be less than or equal to the num ber of inputs.

• D = Direct connection (1 to 1)

• M = MixConvert is used

• P = Standard Panner is used (Stereo Dual Panner/Stereo Combined Panner/Stereo Balance Panner)

• S = SurroundPanner is used

• - = Direct connection is used (trying to match the speaker configuration, for example L-> L or C->C)

Output Config.

_____________

Input Config.

Mono D P S S S S S S S S S S S S

Stereo P P S S S S S S S S S S S S

LRS M M D M M M M M M M M M M M

LRS+Lfe M M M D M M M M M M M M M M

LRC M M M M D M M M M M M M M M

LRC+Lfe M M M M M D M M M M M M M M

LRCS M M M M M M D M M M M M M M

LCRS+Lfe M M M M M M M D M M M M M M

Quadro M M M M M M M M D M M M M M

Quadro+Lfe M M M M M M M M M D M M M M

5.0 M M M M M M M M M M D M M M

5.1 M M M M M M M M M M M D M M

6.0 Cine M M M M M M M M M M M M D M

6.0 Music M M M M M M M M M M M M M D

6.1 Cine M M M M M M M M M M M M M M

6.1 Music M M M M M M M M M M M M M M

7.0 Cine M M M M M M M M M M M M M M

7.0 Music M M M M M M M M M M M M M M

7.1 Cine M M M M M M M M M M M M M M

7.1 Music M M M M M M M M M M M M M M

8.0 Cine M M M M M M M M M M M M M M

8.0 Music M M M M M M M M M M M M M M

8.1 Cine - - - - - - - - - - - - - -

8.1 Music - - - - - - - - - - - - - -

10.2 - - - - - - - - - - - - - -

Mono Stereo LRS LRS

+Lfe

LRC LRC

+Lfe

LRCS LCRS

+Lfe

Quadro Quadro

+Lfe

5.0 5.1 6.0 Cine 6.0 Music

MixConvert Appendix

Page 87

Output Config. ______________

Input Config.

Mono S S S S S S S S S S S

Stereo SSSSSSSSSSS

LRS M M M M M M M M - - -

LRS+Lfe MMMMMMMM - - -

LRC M M M M M M M M - - -

LRC+Lfe MMMMMMMM - - -

LRCS M M M M M M M M - - -

LCRS+Lfe MMMMMMMM - - -

Quadro M M M M M M M M - - -

Quadro+Lfe MMMMMMMM - - -

5.0 M M M M M M M M - - -

5.1 MMMMMMMM - - -

6.0 Cine M M M M M M M M - - -

6.0 Music MMMMMMMM - - -

6.1 Cine D M M M M M M M - - -

6.1 Music M D MMMMMM - - -

7.0 Cine M M D M M M M M - - -

7.0 Music MM M D MMMM - - -

7.1 Cine M M M M D M M M - - -

7.1 Music MMMMMD MM - - -

8.0 Cine M M M M M M D M - - -

8.0 Music MMMMMMMD ---

8.1 Cine - - - - - - - - D - -

8.1 Music ---------D -

10.2 - - - - - - - - - - D

6.1 Cine 6.1 Music 7.0 Cine 7.0 Music 7.1 Cine 7.1 Music 8.0 Cine 8.0 Music 8.1 Cine 8.1 Music 10.2

MixConvert Appendix

Page 88

Index

Page 89

AmpSimulator 9 Apogee UV22HR 28 Arpache 5 64 Arpache SX 65 Arpeggiator 64, 65 Auto LFO (MIDI effect) 66 AutoPan 29

Beat Designer (MIDI effect) 67 BitCrusher 36

Chopper 36 Chorder (MIDI effect) 72 Chorus 29 Cloner 30 Compressor 10 Compressor (MIDI effect) 74

DaTube 9 DeClicker 39 DeEsser 11 Delay plug-ins 6 DeNoiser 41 Density (MIDI effect) 76 Distortion 9 Distortion plug-ins 9 Dither 28 DualFilter 21 Dynamics plug-ins 10

EnvelopeShaper 12 EQ plug-ins 19 Expander 12

Filter plug-ins 21 Flanger 30

Gate 13 Generator plug-ins 27 GEQ-10 19 GEQ-30 19 Grungelizer 42 GS Control Panel 82

Limiter 14

Mastering plug-ins 28 MatrixDecoder 53 MatrixEncoder 54 Maximizer 14 Metalizer 31 Micro Tuner (MIDI effect) 76 MIDI Context Gate (MIDI effect) 75 MIDI Control (MIDI effect) 76 MIDI Echo (MIDI effect) 77 MIDI Gate 15 MIDI Modifiers (MIDI effect) 78 MIDI Monitor (MIDI effect) 78 MIDI Step Sequencer 80 Mix6To2 55 Mix8To2 56 MixConvert 56 MixConvert-ControlRoom 59 MixerDelay 59 ModMachine 6 Modulation plug-ins 29 MonoDelay 7 MonoToStereo 50 MultibandCompressor 16 MultiScope 61

Note to CC (MIDI effect) 79

Octaver 37 Other plug-ins 36

Pattern Sequencer 80 Phase shift (MixConvert) 59 Phaser 32 PingPongDelay 8 Pitch shift plug-ins 38 PitchCorrect 38 PitchDriver 39 PostFilter 21

Q 22 Quantizer (MIDI effect) 80

Restoration plug-ins 39 Reverb plug-ins 43 REVerence 43 RingModulator 32 Roland GS Control Panel 82 RoomWorks 48 RoomWorks SE 49 Rotary 33

SMPTE Generator 27 SoftClipper 10 Spatial plug-ins 50 StepDesigner (MIDI effect) 80 StepFilter 23 StereoDelay 8 StereoEnhancer 50 StudioChorus 34 StudioEQ 19 Surround plug-ins 53 SurroundDither 60 SurroundPan 51 SurroundPanner V5 53

Index

Page 90

TestGenerator 28 ToneBooster 24 Tonic 25 Tools plug-ins 61 Track Control (MIDI effect) 82 Tranceformer 34 Transformer (MIDI effect) 84 Tremolo 35 Tuner 37

UV22HR 28

Vibrato 35 VintageCompressor 17 VSTDynamics 17

WahWah 26

XG Control Panel 82

Yamaha XG Control Panel 82

Index

Steinberg Nuendo - 5.0 Plug-in Reference

Specifications and Main Features

Frequently Asked Questions

User Manual

Table of Contents

The included effect plug-ins

Introduction

Delay plug-ins

ModMachine

MonoDelay

PingPongDelay

StereoDelay

Distortion plug-ins

AmpSimulator

DaTube

Distortion

SoftClipper

Dynamics plug-ins

Compressor

DeEsser

EnvelopeShaper

Expander

Gate

Limiter

Maximizer

MIDI Gate

MultibandCompressor

VintageCompressor

VSTDynamics

EQ plug-ins

GEQ-10/GEQ-30

StudioEQ

Filter plug-ins

DualFilter

PostFilter

StepFilter

ToneBooster

Tonic

WahWah

Generator plug-ins

SMPTEGenerator

TestGenerator

Modulation plug-ins

AutoPan

Chorus

Cloner

Flanger

Metalizer

Phaser

RingModulator

Rotary

StudioChorus

Tranceformer

Tremolo

Vibrato

Other plug-ins

BitCrusher

Chopper

Octaver

Tuner

Pitch Shift plug-ins

PitchCorrect

PitchDriver

Restoration plug-ins

DeClicker

DeNoiser

Grungelizer

Reverb plug-ins

REVerence

RoomWorks

RoomWorks SE

Spatial + Panner plug-ins

MonoToStereo

StereoEnhancer

SurroundPan

SurroundPanner V5

Surround plug-ins

MatrixDecoder

MatrixEncoder

Mix6To2