Steinberg Wavelab Pro - 10.0 Plug-in Reference

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Plug-in Reference

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Cristina Bachmann, Heiko Bischoff, Lillie Harris, Christina Kaboth, Insa Mingers, Matthias Obrecht, Sabine Pfeifer, Benjamin Schütte, Marita Sladek

This PDF provides improved access for vision-impaired users. Please note that due to the complexity and number of images in this document, it is not possible to include text descriptions of images.

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 specically 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. Registered licensees of the product described herein may print one copy of this document for their personal use.

All product and company names are ™ or ® trademarks of their respective owners. For more information, please visit www.steinberg.net/trademarks.

WaveLab Pro_10.0.0_en-US_2019-10-15

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4 WaveLab-specic Plug-ins

4 Resampler 4 Ducker 5 Leveler 6 Leveler Multi 6 MasterRig 23 Peak Master 24 RestoreRig 28 Silence 28 Stereo Expander

29 Steinberg VST 3 Plug-ins

29 AutoPan 30 Brickwall Limiter 31 Channel Extractor 31 Chorus 32 Compressor 34 CurveEQ 34 DeEsser 36 Distortion 37 DualFilter 37 EnvelopeShaper 38 Expander 40 Frequency 43 Gate 45 GEQ-10/GEQ-30 46 Limiter 47 L/R to M/S, M/S to L/R 47 Magneto II 48 Maximizer 49 Mix6to2 50 Mix8to2 51 MonoDelay 51 MonoToStereo 52 MultibandCompressor 54 MultibandEnvelopeShaper 56 MultibandExpander 58 Octaver 58 PingPongDelay 59 PostFilter 61 REVelation 63 RoomWorks 66 RoomWorks SE 66 StereoDelay 67 StereoEnhancer 68 Stereo Tools 69 StudioChorus 70 StudioEQ 72 TestGenerator 73 Tube Compressor 74 VintageCompressor

75 VSTDynamics

79 Legacy Plug-ins

80 Dithering Plug-ins

80 Internal Dithering 80 MBIT+™ Dithering 82 UV22HR

83 ASIO Plug-ins

83 External Gear 84 Audio Input

85 Batch Processing Plug-ins

85 Audio Analyzer 87 Audio Injector 87 Audio Mixer 88 DC Remover 88 Delay Next Process Activation 88 Fade In/Fade Out 89 Instructor 90 Level Normalizer 90 Loudness Meta Normalizer 91 Loudness Restorer 92 Meta Leveler 92 Resizer 92 Stereo to Mono 93 Trimmer

94 Index

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WaveLab-specic Plug-ins

WaveLab-specic plug-ins use the plug-in format of WaveLab, and cannot be used with other applications.

● WaveLab-specic plug-ins can only be used in the Master Section and in batch processes. WaveLab effects are also included as VST plug-ins, available as track or clip

● You can specify which plug-ins should be available on the Effects pane and the Final

●

Resampler

This plug-in is a professional sample rate converter providing exceptional transparency and preservation of the frequency content. It is only available in the Master Section.

However, some effects in audio montages.

Effects/Dithering pane of the Master Section by using the Plug-in Settings dialog.

When a multichannel conguration is used in the audio montage, only certain plug-ins can be used as master effects. All channels in the Master Section are affected equally.

Ducker

NOTE

This plug-in is very CPU consuming, especially in high quality modes.

Output Sample Rate

Denes the output sample rate while the input sample rate is determined by the sample rate of the active audio

Quality

Denes the quality of the algorithm that is used (Standard, High, Very High, Best). In

Standard mode, the CPU load is much lower than in Best mode but the sound

quality of the resulting audio is also lower.

This plug-in lets you control (modulate) the volume of clips placed on a track with the signal of one or more clips placed on the next adjacent track below it. The Ducker plug-in can only be used as a clip effect in the audio montage.

It uses the Route to options that can be found on the Track menu. You can use mono or stereo tracks for both the modulating and the upper track.

le or audio montage.

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WaveLab-specic Plug-ins Leveler

Threshold

Sets the loudness threshold that triggers the Ducker. Clips on the modulator track with levels above the threshold will cause the level of a clip on the upper track to be lowered.

Damping

Sets the amount of level reduction that is applied to the clip on the upper track.

Fall Time

Sets the time it takes for the level to change from 0 dB to the set damping level.

Leveler

Hold Time

When the modulating signal falls below the set threshold, this setting determines how long the level will stay reduced before it starts rising to normal level again.

Rise Time

Sets the time after which the reduced level rises to the normal level when the modulating signal falls below the set threshold (after the Hold Time).

Mix Mode

If this is activated, the Ducker outputs a mix of the two tracks. This is only useful if the Route to Upper Track Only option has been activated for the modulating track. Then this feature can be used for processing several clips through the same plug-in chain if more plug-ins have been assigned after the Ducker on the upper track.

Note that the mixed output is controlled by the upper track. If this is not playing a clip, both of the tracks will be silent.

This plug-in is useful for correcting an imbalance or adjusting levels between stereo channels, or for mixing down to mono.

Volume Left/Volume Right (-48 dB to 12 dB)

Governs how much of the signal is included in the left and/or right channel of the output bus.

Stereo Link

If this option is activated, Volume Right delivers the gain that is set for Volume Left.

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WaveLab-specic Plug-ins Leveler Multi

Mix to Mono

If this option is activated, a mono mix of the stereo channels is delivered to the output bus.

Leveler Multi

This plug-in takes multichannel input and applies a fader equally to all channels.

Volume (-48 dB to 12 dB)

Governs how much gain is applied to the signal before it is routed to the output bus.

MasterRig

MasterRig allows you to master audio material in an intuitive and creative way. It offers high-

class sound quality, accuracy, exibility, and control.

Main Layout

Module Chain

The module chain contains the mastering modules. You can add up to 8 modules.

The following settings are available for each module:

Bypass

Bypasses the module. This allows you to compare the sound of the unprocessed signal to that of the processed signal.

Solo

Solos the module. Only one module can be soloed at a time.

Remove

Removes the module from the module chain.

Scenes

You can save up to 4 MasterRig congurations as scenes. This allows you to compare different parameter settings and module combinations.

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WaveLab-specic Plug-ins MasterRig

● To copy the settings of a scene to another scene, click Copy Scene, and then click the

scene button of the scene where you want to paste the settings.

A copy of a scene is indicated by a (c) behind the scene name.

● To reset the settings of the selected scene, click Reset Scene.

● To rename a scene, double-click the scene name and type in another name.

Spectrum Display

The spectrum display in the upper half of the panel is where you set the width of the frequency bands. The vertical value scale to the left shows the gain level of each frequency band. The horizontal scale shows the frequency range.

● To dene the frequency range of the different frequency bands, use the handles at the

sides of each frequency band.

● To attenuate or boost the output level of each frequency band by ±15 dB, use the handles

on top of each frequency band.

Settings

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WaveLab-specic Plug-ins MasterRig

Parameter Linking

Links the parameters of the same type in all bands in a module. This allows you to edit parameter values of all bands in a module simultaneously. Two link modes are available:

● If Absolute Mode is activated and you edit a parameter value in one band, the

● If Relative Mode is activated and you edit a parameter value in one band, the

Auto Listen for Filters

If this option is activated and you edit a parameter of a module, the corresponding lter or band is soloed. This allows you to locate unwanted frequencies in your audio and helps you to focus on a particular band or lter. Once you stop editing the parameter,

Global Settings

Allows you to make global settings for MasterRig.

Undo/Redo

Undoes/Redoes the last operation. The undo/redo history is deleted when you select another scene.

Absolute and Relative.

corresponding parameter values in the other bands are set to the same value.

corresponding parameter values in the other bands keep their relation.

Solo is deactivated.

Input/Output Meter

The input/output meter provides a combined peak level, with peak-hold functionality and RMS meter. Between the meters for input and output is the gain reduction meter for the Limiter.

The maximum values for input/output peak level, RMS, and gain reduction are displayed above the meter display. To reset all maximum values, click any of the values.

Side-Chain Settings

The Compressor module and the Dynamic EQ module support side-chain. You can set up the side-chain routing for each band separately.

● To open the side-chain panel, click the SC button at the bottom left of each band section.

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WaveLab-specic Plug-ins MasterRig

Active

Activates the internal side-chain lter. The input signal can then be shaped according to the lter parameters.

SC FREQ

Sets the frequency of the side-chain lter.

Auto (Dynamic EQ only)

Deactivates the SC Frequency knob of the side-chain panel. Instead, the settings of the Frequency knob are used.

Listen

SC Q

Modules

Modules allow you to create a mastering chain. Some modules can be used only once and some in two instances in the module chain. You can rearrange modules in the module chain to change the processing order.

● To add a module to the module chain, click Add Module in the modules section and click a

● To remove a module, click the corresponding Remove button.

● To bypass a module, click the corresponding Bypass button.

● To solo a module, click the corresponding Solo button.

● To change the order of the modules, drag a module to another position in the module

Global Settings

● To open the Global Settings, click Global Settings above the spectrum display.

Allows you to solo the side-chain lter.

Sets the resonance or width of the lter.

module.

chain.

Spectrum Display

Show Spectrum

Activates/Deactivates the spectrum display.

Smooth

Determines the reaction time of the spectrum display. Lower values result in faster reaction times.

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WaveLab-specic Plug-ins MasterRig

Peak Hold

Freezes the peak values of the spectrum display.

Slope

Tilts the spectrum display around the 1 kHz pivot.

Two Channels

If this option is activated, the spectrum of the left and right channels are displayed separately.

EQ Curve

Show Curve

Shows/Hides the EQ curve in the spectrum display.

Filled

If this option is activated, the EQ curve is lled.

RMS

AES17 (+3 dB)

If this option is activated, the RMS value is raised by 3 dB to follow the AES17 standard.

Limiter

The Limiter module makes sure that the output level never exceeds a set output level, to avoid clipping in following devices.

Band Settings

On/Off

Activates/Deactivates the corresponding section.

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WaveLab-specic Plug-ins MasterRig

Balance

Mid/Side

Allow you to set the gain for the mid and side signal.

Solo Mid Signal/Solo Side Signal

Allow you to solo the mid or side signal.

Transients

If the Transients section is activated, you can set the following parameters:

ATT

Sets the gain of the attack phase of the signal for the corresponding band.

REL

Sets the gain of the release phase of the signal for the corresponding band.

Gain

Sets the output level for the corresponding band.

Harmonics

If the Harmonics section is activated, the Limiter module starts limiting the signal softly. At the same time, harmonics are generated, adding a warm, tube-like characteristic to the audio material.

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WaveLab-specic Plug-ins MasterRig

2nd HARM/3rd HARM

Allow you to control the second and third harmonic independently.

Drive

Allows you to adjust the amount of gain boost for the signal to raise the amount of soft-clipping.

Brickwall

Due to its fast attack time, Brickwall Limiter can reduce even short audio level peaks without creating audible artifacts. The limiting amount is displayed between the input and the output meter.

Release

Sets the time after which the gain returns to the original level when the signal drops below the threshold. If Auto Release is activated, the plug-in automatically nds the best release setting for the audio material.

Oversample

If this option is activated, Brickwall Limiter detects and limits signal levels between two samples to prevent distortion when converting digital signals into analog signals.

Stereo Link

If this option is activated, Brickwall Limiter uses the channel with the highest level to analyze the input signal. If the button is deactivated, each channel is analyzed separately.

Output

Sets the output level.

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WaveLab-specic Plug-ins MasterRig

Maximizer

Maximizer raises the loudness of audio material without the risk of clipping. The limiting

amount is displayed between the input and the output meter.

Optimize

Determines the loudness of the signal.

Output

Sets the output level.

Compressor

The Compressor module allows a signal to be split into four frequency bands. You can specify the level, bandwidth, and compressor characteristics for each band.

You can add two Compressor modules to the module chain, Compressor A and Compressor B.

Band Settings

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WaveLab-specic Plug-ins MasterRig

On/Off

Activates/Deactivates the corresponding section.

Soloing Frequency Bands

To solo a frequency band, activate the S button in each section. Only one band can be soloed at a time.

Channel Settings

Allow you to switch between left/right, stereo, and mid/side processing. In Left/ Right or Mid/Side processing mode, you can make different settings for the two

channels.

Add/Remove Band

Allow you to add and remove bands.

Standard

Allows you to create smooth compression effects.

THRESH (-60 to 0 dB)

Signal levels above the set threshold trigger the compressor.

ATT (0.1 to 100 ms)

Determines how fast the compressor responds. If the attack time is long, more of the initial part of the signal passes through unprocessed.

REL (10 to 1000 ms)

Sets the time after which the gain returns to its original level. If Auto Release is activated, the plug-in automatically

nds a suitable release setting for the audio.

Ratio

Sets the amount of gain reduction applied to signal above the set threshold.

Mix

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

Compressor curve display

Graphically illustrates the compressor curve that is shaped according to the Threshold and Ratio parameter settings.

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WaveLab-specic Plug-ins MasterRig

Output

Sets the output gain.

Side-Chain

Opens the Side-Chain settings.

Tube

This versatile compressor with integrated tube-simulation allows you to produce smooth and warm compression effects.

Input

In combination with the Output setting, this parameter determines the compression amount. The higher the input gain setting and the lower the output gain setting, the more compression is applied.

ATT (0.1 to 100 ms)

Determines how fast the compressor responds. If the attack time is long, more of the initial part of the signal passes through unprocessed.

REL (10 to 1000 ms)

Sets the time after which the gain returns to its original level. If Auto Release is activated, the plug-in automatically nds the best release setting for the audio.

Drive

Controls the amount of tube saturation.

Mix

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

Output

Sets the output gain.

Side-Chain

Opens the Side-Chain settings.

Vintage

Vintage Compressor is modeled after vintage type compressors.

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WaveLab-specic Plug-ins MasterRig

Input

In combination with the Output setting, this parameter determines the compression amount. The higher the input gain setting and the lower the output gain setting, the more compression is applied.

ATT (0.1 to 100 ms)

Determines how fast the compressor responds. If the attack time is long, more of the initial part of the signal passes through unprocessed.

REL (10 to 1000 ms)

Sets the time after which the gain returns to its original level. If Auto Release is activated, the plug-in automatically material.

Ratio

Sets the amount of gain reduction that is applied to signals above the set threshold.

Attack Mode (Punch)

If this option is activated, the early attack phase of the signal is preserved, retaining the original punch in the audio material, even with short Attack settings.

Mix

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

Output

Sets the output gain.

Side-Chain

Opens the Side-Chain settings.

nds the best release setting for the audio

Maximizer

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WaveLab-specic Plug-ins MasterRig

Optimize

Determines the loudness of the signal.

Mix

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

Output

Sets the output gain.

Equalizer

The Equalizer module is a high-quality 8-band parametric stereo equalizer with 8 fully parametric mid-range bands. The low and high bands can act as either shelving lter, as peak lter (band-pass), or as cut lter (low-pass/high-pass, band 1 and 8 only).

You can add two Equalizer modules to the module chain, Equalizer A and Equalizer B.

Band Settings

On/Off

Activates/Deactivates the corresponding section.

Channel Settings

Allow you to switch between left/right, stereo, and mid/side processing. In Left/ Right or Mid/Side processing mode, you can make different settings for the two

channels.

IMPORTANT

When using Mid/Side processing mode, we recommend that you activate Linear Phase in order to avoid unwanted sound colorization.

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WaveLab-specic Plug-ins MasterRig

Linear Phase

Activates/Deactivates linear phase mode for the corresponding band.

Linear phase mode avoids unwanted frequency dependent phase shifts of the audio signal that might occur with standard minimum phase equalizing.

NOTE

● Linear phase mode leads to an increase in latency.

● In rare cases, for example, when using low cut ltering with a high slope for

Equalizer Section

bass signals, also an unwanted pre-ringing effect may be audible.

Type

You can choose between the EQ types Low Shelf, Peak, High Shelf, and Notch. For band 1 and 8, you can also select the types Cut 12, Cut 24, and Cut 48.

● Low Shelf boosts or attenuates frequencies below the cutoff frequency by the specied amount.

● High Shelf boosts or attenuates frequencies above the cutoff frequency by the specied amount.

● Peak boosts or attenuates frequencies at the set frequency value with a bell shaped lter.

● Notch boosts or attenuates frequencies at the set frequency value with a very narrow

● Cut attenuates frequencies below (band 1) or above (band 8) the set frequency. You can choose between different slopes: 6  dB, 12  dB, 24  dB, 48  dB, or 96  dB per octave.

FREQ (20 to 20000 Hz)

Sets the frequency of the corresponding band.

Controls the width of the corresponding band.

Gain (-15 to +15 dB)

Sets the amount of attenuation/boost for the corresponding band.

lter.

Dynamic EQ

Dynamic EQ allows you to adjust frequencies and lets you determine when and how the EQ is

applied depending on the dynamics of the audio material.

You can add two Dynamic EQ modules to the module chain, Dynamic EQ A and Dynamic EQ B.

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WaveLab-specic Plug-ins MasterRig

Band Settings

On/Off

Activates/Deactivates the corresponding section.

Channel Settings

Allow you to switch between left/right, stereo, and mid/side processing. In Left/ Right or Mid/Side processing mode, you can make different settings for the two

channels.

Equalizer Section

Type pop-up menu

Allows you to select the EQ types.

● Low Shelf boosts or attenuates frequencies below the cutoff frequency by the specied amount.

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WaveLab-specic Plug-ins MasterRig

● Peak boosts or attenuates frequencies at the set frequency value with a bell

● High Shelf boosts or attenuates frequencies above the cutoff frequency by the

FREQ (20 to 20000 Hz)

Sets the frequency of the corresponding band.

Controls the width of the corresponding band.

Gain (-15 to +15 dB)

Sets the amount of attenuation/boost for the corresponding band.

THRESH (-50 to 0 dB)

Determines the threshold level. Only signal levels above the threshold are processed.

ATT (0.1 to 100 ms)

Determines how fast Dynamic EQ responds to signals above the threshold. If the attack time is long, more of the early part of the signal passes through unprocessed.

REL (10 to 1000 ms)

Sets the time after which Dynamic EQ returns to its original level when the signal drops below the threshold.

shaped lter.

specied amount.

Saturator

Ratio

The higher the level of the input signal is above the threshold, the more ltering occurs. Low ratio values mean that the lter starts to boost or attenuate smoothly above the threshold. High ratio values mean that the lter starts to work almost immediately.

Side-Chain

Opens the Side-Chain settings.

Imager

Mix

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

Output

Sets the output gain.

The Imager module allows you to expand or reduce the stereo width of your audio in up to four bands. This way you can independently adjust the stereo image in dened frequency domains.

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WaveLab-specic Plug-ins Peak Master

Band Settings

On/Off

Activates/Deactivates the corresponding section.

Soloing Frequency Bands

To solo a frequency band, activate the S button in each section. Only one band can be soloed at a time.

Add/Remove Band

Allow you to add and remove bands.

Imager Section

Width

Pan

Output

Peak Master

This is a basic plug-in that minimizes peaks in your audio le, allowing a louder mix without clipping. It is useful in taming dynamic instruments.

It is primarily used as a brickwall limiter. For example, you can limit audio peaks without altering the rest of the audio signal. In this case, set Input Gain to 0 dB and Out Ceiling to 0 dB, to achieve a clip-free audio signal. When used in this way, Peak Master is an excellent plug-in to succeed a resampler plug-in, and to proceed a dithering plug-in.

Allows you to control the stereo width per band.

Allows you to pan the signal left/right.

Sets the output level for each band.

Input Gain

Values range from -12 dB to 24 dB.

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WaveLab-specic Plug-ins RestoreRig

Out Ceiling

This is the maximum level of the output signal. Values range from -18 dB to 0 dB.

Softness

This governs the speed at which the signal becomes unaffected after limiting has been triggered on some samples. Values range from -5 to +5.

RestoreRig

RestoreRig allows you to remove noise from an audio recording with different restoration

modules. The noise can be an impulsive noise (DeClicker), an ambient noise (DeNoiser), or a low tonal noise (DeBuzzer)

Main Layout

Input Signal/Output Signal

Displays the input signal and the output signal of the restored signal. The vertical value scale to the left shows the gain level of the input and output signals. The horizontal scale shows the frequency range.

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WaveLab-specic Plug-ins RestoreRig

Master

The input/output meter provides a peak level meter.

The maximum values for input/output peak level are displayed above the meter display. To reset all maximum values, click any of the values.

Settings

Modules

Filled Curve

Allows you to ll the curves of the input signal and the output signal.

Smooth Metering

Determines the reaction time of the display. Lower values result in faster reaction times.

Gain Control

The Gain Control allows you to set the master gain for the modules.

The modules DeClicker, DeNoiser, and DeBuzzer allow you to remove different kinds of noises.

To activate or deactivate a module, click Activate/Deactivate on the left of the module

● name.

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WaveLab-specic Plug-ins RestoreRig

● To only listen to the sound that has been removed from the audio, click the Noise

Listening Mode button of the module that you want to hear.

DeClicker

DeClicker allows you to remove clicks from audio material.

Activate/Deactivate DeClicker

Activates/Deactivates the module.

DeNoiser

Noise Listening Mode

Allows you to listen to the signal that has been removed from the original audio material.

Meters

Allow you to monitor the quantity of impulsive noise that is removed from the signal.

NOTE

Avoid letting the meter reach the red region, as this can produce destructive artifacts.

Crackle

Allows you to remove very short impulsive noise from the audio signal.

Click

Allows you to remove medium-sized impulsive noise from the audio signal.

Pop

Allows you to remove long impulsive noise from the audio signal.

DeNoiser allows you to remove noise from the audio material.

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WaveLab-specic Plug-ins RestoreRig

Activate/Deactivate DeNoiser

Activates/Deactivates the module.

Noise Listening Mode

Allows you to listen to the signal that has been removed from the original audio material.

Dynamic Level

Allows you to remove noise that evolves over time from the audio signal.

Static Level

Allows you to remove noise that does not evolve over time from the audio signal. The

Learn option allows you to dene the stationary noise.

Noise

The Noise options allow you to dene a section in an audio le that contains a static noise that you want to remove. When you then render the audio le, you can remove the recorded static noise from the audio signal.

1. Play back the audio section that contains the noise that you want to remove

2. Use the Static Level dial to set the level.

3. To remove the recorded static noise in the audio le, render the audio le.

and click Learn.

RestoreRig records the audio for a few seconds to detect the static noise.

DeBuzzer

If you want to record the static noise at another audio section, click Reset, play back another audio section, and click Learn again.

Algorithm

Allows you to select different DeNoiser algorithms. Depending on the audio material, different modes can affect the DeNoiser quality.

● Smooth is sucient for most uses.

● Use Musical for harmonic content with low rhythmic or transient components.

● Use Rhythmic for drum and percussive content.

● Use Strong if the noise level reduction is more important than the accuracy of

the noise reduction.

● Use Speech for vocal content.

DeBuzzer allows you to remove harmonic noise with a fundamental frequency that should be around 50 to 60  Hz.

Activate/Deactivate DeBuzzer

Activates/Deactivates the module.

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WaveLab-specic Plug-ins Silence

Noise Listening Mode

Allows you to listen to the signal that has been removed from the original audio material.

Level

Allows you to dene the reduction of the noise in dB.

Sensitivity

Allows you to dene how sensitive the reduction will be to the current audio level. At 0  %, sensitivity values, the level is dynamically dened in a range between 0  dB and the Level value. This reduces the buzz when the audio level is low and does not affect the audio when the audio level is high.

Frequency

Allows you to dene the value of the fundamental frequency.

Auto

If this option is activated, DeBuzzer automatically detects the fundamental frequency of the current most prominent harmonic tone.

NOTE

DeBuzzer reduces the current harmonic noise with the Level value. At higher

Once you have detected the frequency that you want to remove, deactivate Auto.

Silence

This plug-in provides a simple way of inserting a precise period of silence at the start or at the end of an audio le. Use this plug-in to add silence at the end of a le, so that the tail of a reverb plug-in does not cut immediately at the end of the

Start

Use the slider to insert from 0 to 60,000 ms of silence at the start of the le.

End

Use the slider to insert from 0 to 60,000 ms of silence at the end of the le.

Stereo Expander

This plug-in is a stereo width enhancer that makes a stereo signal sound wider. It gives better results from real stereo material, as opposed to mono channels panned to different positions in the stereo image.

le.

Width

Higher values result in a greater stereo width. Usually, you set Width to values between 0 % and 20 %. Higher values can be used for special effects.

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Steinberg VST 3 Plug-ins

In WaveLab there is no limitation to the use of VST plug-ins. They can be used wherever plug-ins can be inserted.

● You can specify which VST plug-ins should be available in the Effects pane and Final Processing/Dithering pane of the Master Section by using the Plug-in Settings dialog.

● VST plug-ins have their own preset handling. You can save or load effect programs (presets).

AutoPan

This auto-pan effect provides several parameters to modulate the left/right stereo position. You can use presets or create individual curves for the modulation waveform. AutoPan also allows for chopping effects by linking the modulation of left and right channel.

NOTE

The panning effect of this plug-in works only on stereo tracks.

Waveform display

Shows the shape of the modulation waveform and allows you to manually adjust it. To draw an individual curve, click a node and move the mouse. To draw a straight line, Shift-click a node and move the mouse.

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Steinberg VST 3 Plug-ins Brickwall Limiter

Waveform preset buttons

Phase

Rate

Allow you to select presets for the modulation waveform.

● Sine creates a smooth sweep.

● Triangle creates a ramp, that is, a linear movement from full right to full left

and back.

● Square creates an instant jump to full right, then to full left, and then back to

center.

● Random One Shot creates a random curve. Click this button again to create a

new random curve.

● Random Continuous automatically creates a new random curve after each

period.

Sets the offset for the starting point of the curve. If multiple AutoPan plug-ins are used on different tracks, for example, different offset settings for each track allow for a more organic overall sound.

Sets the auto-pan speed in Hertz and shows the movement within the panorama.

Link

If this button is activated, the left and right channel are modulated simultaneously. This results in a chopping effect instead of auto-panning.

In this mode, Width sets the intensity of the volume modulation.

Width

Sets the amount of deection to the left and right side of the stereo panorama. If

Link is activated, this parameter sets the intensity of the volume modulation.

Smooth

Allows you to smooth the transition between individual steps of the panorama curve.

Brickwall Limiter

Brickwall Limiter ensures that the output level never exceeds a set limit.

Due to its fast attack time, Brickwall Limiter can reduce even short audio level peaks without creating audible artifacts. However, this plug-in creates a latency of 1ms. Brickwall Limiter features separate meters for input, output, and the amount of limiting. Position this plug-in at the end of the signal chain, before dithering.

Page 31

Steinberg VST 3 Plug-ins Channel Extractor

Threshold (-20 to 0 dB)

Release (3 to 1000 ms or Auto mode)

Link

Detect Intersample Clipping

NOTE

Brickwall Limiter is designed for the reduction of occasional peaks in the signal. If the Gain Reduction meter indicates constant limiting, try raising the threshold or lowering the overall

level of the input signal.

Determines the level where the limiter kicks in. Only signal levels above the set threshold are processed.

Sets the time after which the gain returns to the original level when the signal drops below the threshold. If the Auto button is activated, the plug-in automatically nds the best release setting for the audio material.

If this button is activated, Brickwall Limiter uses the channel with the highest level to analyze the input signal. If the button is deactivated, each channel is analyzed separately.

If this option is activated, Brickwall Limiter uses oversampling to detect and limit signal levels between two samples to prevent distortion when converting digital signals into analog signals.

Channel Extractor

This plug-in allows you to only keep the left or the right channel of a stereo stream.

Channel

Lets you select whether to keep the left or the right channel of the stereo stream.

Chorus

This plug-in is a single-stage chorus effect. It doubles the audio that is sent into it with a slightly detuned version.

Delay

Affects the frequency range of the modulation sweep by adjusting the initial delay time.

Page 32

Steinberg VST 3 Plug-ins Compressor

Width

Spatial

Mix

Rate

Waveform Shape

Lo Filter/Hi Filter

Sets the depth of the chorus effect. Higher settings produce a more pronounced effect.

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

Sets the level balance between the dry signal and the wet signal. If the effect is used as a send effect, set this parameter to the maximum value, as you can control the dry/effect balance with the send level.

Allows you to set the sweep rate freely with the Rate knob.

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

Allow you to roll off low and high frequencies of the effect signal.

NOTE

If side-chaining is supported, the modulation can also be controlled from another signal source via the side-chain input. If 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

Compressor

Compressor reduces the dynamic range of the audio, making softer sounds louder or louder

sounds softer, or both.

Operation Manual.

Compressor features a separate display that graphically illustrates the compressor curve that is

shaped according to the Threshold and Ratio parameter settings. Compressor also features a

Page 33

Steinberg VST 3 Plug-ins Compressor

Gain Reduction meter that shows the amount of gain reduction in dB, Soft knee/Hard knee compression modes, and a program-dependent auto feature for the Release parameter.

Threshold (-60 to 0 dB)

Ratio

Soft Knee

High Ratio

Make-Up (0 to 24 dB or Auto mode)

Determines the level where the compressor kicks in. Only signal levels above the set threshold are processed.

Sets the amount of gain reduction applied to signals above the set threshold. A ratio of 3:1 means that for every 3 dB the input level increases, the output level increases by 1 dB.

If this button is deactivated, signals above the threshold are compressed instantly according to the set ratio (hard knee). If Soft Knee is activated, the onset of compression is more gradual, producing a less drastic result.

Sets the ratio to a xed value of 20:1.

Compensates for output gain loss caused by compression. If Auto Make-Up Gain is activated, the output is automatically adjusted for gain loss.

Dry Mix

Mixes the dry input signal to the compressed signal.

Attack (0.1 to 100 ms)

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 passes through unprocessed.

Hold (0 to 5000 ms)

Sets the time the applied compression affects the signal after exceeding the threshold. Short hold times are useful for DJ-style ducking, while longer hold times are required for music ducking, for example, when working on a documentary lm.

Release (10 to 1000 ms or Auto mode)

Sets the time after which the gain returns to its original level when the signal drops below the threshold. If Auto Release is activated, the plug-in automatically nds the best release setting for the audio material.

Analysis (Pure Peak to Pure RMS)

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

Peak mode

Live

If this button is activated, the look-ahead feature of the effect is deactivated. Lookahead produces more accurate processing, but adds a a trade-off. If Live mode is activated, there is no latency, which is better for live processing.

specic amount of latency as

Page 34

Steinberg VST 3 Plug-ins CurveEQ

CurveEQ

Voxengo CurveEQ is a spline equalizer for professional music and audio production applications. CurveEQ shows the lter response you are designing by means of a spline, that is, a smooth

curvy line. This way you can see how the EQ alters the sound.

CurveEQ implements spectrum matching technology that allows you to transfer the spectral shape of one recording to another. In other words, you can copy the frequency balance of existing time-proven mixes so that other mixes can be improved. The switched between linear-phase and minimum-phase modes. CurveEQ also features a customizable spectrum analyzer. Furthermore, you can display, save, and load static spectrum plots for comparison and matching purposes.

For detailed information about CurveEQ and its parameters, refer to the documentation provided by Voxengo at http://www.voxengo.com.

DeEsser

DeEsser reduces excessive sibilance, primarily for vocal recordings. It is a special type of

compressor that is tuned to be sensitive to the frequencies produced by the s-sound.

lters of CurveEQ can be

Close proximity microphone placement and equalizing can lead to situations where the overall sound is just right, but there is a problem with sibilants.

Display

Shows the spectrum of the input signal.

● To adjust the frequency band, drag the border lines or click in the middle of the band and drag.

● To change the width of the frequency band, hold Shift and drag to the left or right.

Page 35

Steinberg VST 3 Plug-ins DeEsser

Filter

Lo/Hi

Solo

Diff

Sets the left and right border of the frequency band. You can set the frequency either in Hz or as a note value. If you enter a note value, the frequency is automatically displayed in Hz accordingly. For example, a note value of A3 sets the frequency to 440 Hz. When you enter a note value, you can also enter a cent offset. For example, enter A5 -23 or C4 +49.

NOTE

Make sure that you enter a space between the note and the cent offset. Only in this case, the cent offsets are taken into account.

Solos the frequency band. This helps you to nd the appropriate position and width of that band.

Plays back what DeEsser removed from the signal. This help you to adjust the frequency band, threshold, and reduction parameters, so that only sharp s-sounds are removed, for example.

Dynamics

Reduction

Controls the intensity of the de-essing effect.

Threshold (-50 to 0 dB)

If the Auto option is deactivated, you can use this control to set a threshold for the incoming signal level above which the plug-in starts to reduce the sibilants.

Release (1 to 1000 ms)

Sets the time after which the de-essing effect returns to zero when the signal drops below the threshold.

Auto

Automatically and continually sets an optimum threshold setting independent of the input signal. The Auto option does not work for low-level signals (< -30 db peak level). To reduce the sibilants in such a

le, set the threshold manually.

Side-Chain

Freq (25 Hz to 20 kHz)

If the Side-Chain button is activated, this sets the frequency of the lter. You can set the frequency either in Hz or as a note value. If you enter a note value, the frequency is automatically displayed in Hz accordingly. For example, a note value of A3 sets the frequency to 440 Hz. When you enter a note value, you can also enter a cent offset. For example, enter A5 -23 or C4 +49.

NOTE

Make sure that you enter a space between the note and the cent offset. Only in this case, the cent offsets are taken into account.

Q-Factor

If the Side-Chain button is activated, this sets the resonance or width of the lter.

Page 36

Steinberg VST 3 Plug-ins Distortion

Side

Monitor

Live

Positioning the DeEsser in the Signal Chain

When recording a voice, the position of DeEsser in the signal chain is usually located after the microphone pre-amp and before a compressor/limiter. This keeps the compressor/limiter from unnecessarily limiting the overall signal dynamics.

Distortion

Activates the internal side-chain lter. You can now shape the input signal according to the lter parameters. Internal side-chaining can be useful for tailoring how the gate operates.

Allows you to monitor the ltered signal.

If this button is activated, the look-ahead feature of the effect is deactivated. Lookahead produces more accurate processing, but adds a specic amount of latency as a trade-off. If Live mode is activated, there is no latency, which is better for live processing.

Distortion adds crunch to your tracks.

Boost

Increases the distortion amount.

Oversampling

Activates/Deactivates oversampling. Oversampling results in less artifacts for higher distortion.

NOTE

If this parameter is activated, the effect requires more processing power.

Page 37

Steinberg VST 3 Plug-ins DualFilter

Mix

Tone

Feedback

Spatial

Output

DualFilter

DualFilter lters out specic frequencies while allowing others to pass through.

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

Changes the tonal characteristic of the output signal.

Feeds part of the output signal back to the effect input. Higher settings increase the distortion effect.

Changes the distortion characteristics of the left and right channels, thus creating a stereo effect.

Sets the output level.

Position

Sets the lter cutoff frequency. If you set this to a negative value, DualFilter acts as a low-pass lter. Positive values cause DualFilter to act as a high-pass lter.

Resonance

Sets the sound characteristic of the lter. With higher values, a ringing sound is heard.

EnvelopeShaper

EnvelopeShaper can be used to attenuate or boost the gain of the attack and release phase of

audio material.

You can use the knobs or drag the breakpoints in the graphical display to change parameter values. Be careful with levels when boosting the gain and if needed reduce the output level to avoid clipping.

Page 38

Steinberg VST 3 Plug-ins Expander

Attack (-20 to 20 dB)

Length (5 to 200 ms)

Sets the gain of the attack phase of the signal.

Sets the length of the attack phase.

Expander

Release

Sets the gain of the release phase of the signal.

Output

Sets the output level.

Expander reduces the output level in relation to the input level for signals below the set threshold. This is useful if 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.

Page 39

Steinberg VST 3 Plug-ins Expander

Threshold

Ratio

Soft Knee

Fall (0.1 to 100 ms)

Hold (0 to 2000 ms)

Rise (10 to 1000 ms or Auto mode)

Determines the level where the expansion kicks in. Only signal levels below the set threshold are processed.

Sets the amount of gain boost applied to signals below the threshold.

If this button is deactivated, signals below the threshold are expanded instantly according to the set ratio (hard knee). If expansion is more gradual, producing less drastic results.

Determines how fast the expander responds to signals below the set threshold. If the fall time is long, more of the early part of the signal passes through unprocessed.

Sets the time the applied expansion affects the signal below the threshold.

Sets the time after which the gain returns to its original level when the signal exceeds the threshold. If the Auto Rise button is activated, the plug-in automatically nds the best rise setting for the audio material.

Soft Knee is activated, the onset of

Analysis (Pure Peak to Pure RMS)

Live

Peak mode

Page 40

Steinberg VST 3 Plug-ins Frequency

Frequency

Frequency is a high-quality equalizer with 8 fully parametric bands. The bands can act as either

shelving lter, as peak or notch lter (band-pass), or as cut lter (low-pass/high-pass).

Main Layout

Reset

Alt-click this button to reset all parameter values.

Auto Listen for Filters

If this option is activated and you edit a parameter of a band, the corresponding frequency range is isolated. This helps you to focus on a particular frequency range and allows you to locate unwanted frequencies in your audio.

Global Settings

Opens the settings dialog for the spectrum display.

Show/Hide Keyboard

Shows/Hides the keyboard below the graphical editor.

On the keyboard, color indicators reect the center frequencies of all active equalizer bands. You can adjust the frequency of a band by dragging its color indicator. If you drag the color indicator of a band to a key, the band is set to its exact frequency.

Output

Adjusts the overall output level.

Page 41

Steinberg VST 3 Plug-ins Frequency

Band Settings

Activate/Deactivate Band

Activates/Deactivates the corresponding band.

NOTE

● To activate/deactivate a band, you can also double-click the corresponding

handle in the graphical editor.

● If a band is deactivated, you can still modify its parameters.

Switch Processing

Allow you to switch between left/right, stereo, and mid/side processing. In Left/ Right or Mid/Side processing mode, you can make different settings for the two

channels.

IMPORTANT

When using Mid/Side processing mode, we recommend that you activate Linear Phase in order to avoid unwanted sound colorization.

NOTE

This setting is only available for stereo tracks.

Linear Phase

Activates/Deactivates linear phase mode for the corresponding band.

Linear phase mode avoids unwanted frequency dependent phase shifts of the audio signal that might occur with standard minimum phase equalizing.

NOTE

● Linear phase mode leads to an increase in latency.

● In rare cases, for example, when using low cut ltering with a high slope for

bass signals, also an unwanted pre-ringing effect may be audible.

Filter type

You can choose between the lter types Low Shelf, Peak, High Shelf, and Notch. For band 1 and 8, you can also select the types Cut 6, Cut 12, Cut 24, Cut 48, and Cut

96.

● Low Shelf boosts or attenuates frequencies below the cutoff frequency by the

specied amount.

Page 42

Steinberg VST 3 Plug-ins Frequency

Freq

● Peak boosts or attenuates frequencies at the set frequency value with a bell

shaped lter.

● High Shelf boosts or attenuates frequencies above the cutoff frequency by the

specied amount.

● Notch boosts or attenuates frequencies at the set frequency value with a very

narrow lter.

● Cut attenuates frequencies below (band 1) or above (band 8) the set

frequency. You can choose between different slopes: 6 dB, 12 dB, 24 dB, 48 dB, or 96 dB per octave.

Sets the frequency of the corresponding band. You can set the frequency either in Hz or as a note value. If you enter a note value, the frequency is automatically changed to Hz. For example, a note value of A3 sets the frequency to 440 Hz. When you enter a note value, you can also enter a cent offset. For example, enter A5 -23 or C4 +49.

NOTE

● You can adjust the Freq parameter of a band in the graphical editor by Alt-

clicking the corresponding handle and moving the mouse left and right.

● Ensure that you enter a space between the note and the cent offset. Only in

this case, the cent offsets are taken into account.

If the band is active, the frequency value is reected as a highlighted key on the keyboard below the graphical editor.

For Peak and Notch lters, this parameter controls the width of the band. For Low Shelf and High Shelf lters, it adds a drop or a boost, depending on the gain setting

of the band.

NOTE

● You can adjust the Q parameter of a band in the graphical editor by Shift-

clicking the corresponding handle and moving the mouse up and down. Alternatively, you can point on the handle and turn the mouse wheel.

● This parameter is not available for Cut lters.

Gain

Sets the amount of attenuation/boost for the corresponding band.

NOTE

● You can adjust the Gain parameter of a band in the graphical editor by Ctrl/

Cmd-clicking the corresponding handle and moving the mouse up and down.

● This parameter is not available for Cut lters.

Invert Gain

Inverts the value of the gain parameter. Positive gain values become negative and vice versa.

Global Settings

● To open the Global Settings, click Global Settings above the spectrum display.

Page 43

Steinberg VST 3 Plug-ins Gate

Spectrum Display

Show Spectrum

Peak Hold

Smooth

Bar Graph

Two Channels

Slope

Activates/Deactivates the spectrum display.

Holds the peak values of the spectrum display for a short time.

Determines the reaction time of the spectrum display. Lower values result in faster reaction times.

If this option is activated, the frequency spectrum is analyzed into 60 separate bands that are displayed as vertical bars.

If this option is activated, the spectrum of the left and right channels are displayed separately.

Tilts the spectrum display around the 1 kHz pivot.

Gate

EQ Curve

Show Curve

Shows/Hides the EQ curve in the spectrum display.

Filled

If this option is activated, the EQ curve is lled. Amount allows you to specify the degree of coverage between 10 and 80 %.

Gating, or noise gating, silences audio signals below a set threshold. As soon as the signal level exceeds the threshold, the gate opens to let the signal through.

Attack (0.1 to 1000 ms)

Sets the time after which the gate opens when it is triggered.

Page 44

Steinberg VST 3 Plug-ins Gate

Hold (0 to 2000 ms)

Release (10 to 1000 ms or Auto mode)

Threshold

State LED

NOTE

Deactivate the Live button to make sure that the gate is already open when a signal above the threshold is played back.

Determines how long the gate remains open after the signal drops below the threshold level.

Sets the time after which the gate closes after the set Hold time. If Auto Release is activated, the plug-in automatically nds the best release setting for the audio material.

Determines the level at which the gate is activated. Signal levels above the set threshold trigger the gate to open, and signal levels below the set threshold close the gate.

Indicates whether the gate is open (LED lights up in green), closed (LED lights up in red), or in an intermediate state (LED lights up in yellow).

Analysis (Pure Peak to Pure RMS)

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 percussive material with a lot of transient peaks.

Live

Peak mode works better for

specic amount of latency as

Side-Chain Section

Side-Chain

Activates the internal side-chain lter. The input signal can then be shaped according to the lter parameters. Internal side-chaining is useful for tailoring how the gate operates.

Monitor

Allows you to monitor the ltered signal.

Center (50 to 20000 Hz)

If the Side-Chain button is activated, this sets the center frequency of the lter.

Q-Factor

If the Side-Chain button is activated, this sets the resonance or width of the lter.

Filter Type (Low-Pass/Band-Pass/High-Pass)

If Side-Chain is activated, these buttons allow you to set the lter type to low-pass, band-pass, or high-pass.

Page 45

Steinberg VST 3 Plug-ins GEQ-10/GEQ-30

GEQ-10/GEQ-30

These graphic equalizers are identical, except for the number of available frequency bands (10 and 30).

Each band can be attenuated or boosted by up to 12 dB, allowing for ne control of the frequency response. In addition, there are several preset modes available that can add color to the sound of GEQ-10/GEQ-30.

You can draw response curves in the main display by clicking and dragging with the mouse. You have to click one of the sliders before you drag across 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 attenuation/boost is shown in dB.

Output

Sets the overall gain of the equalizer.

Flatten

Resets all the frequency bands to 0 dB.

Range

Allows you to adjust how much a set curve cuts or boosts the signal.

Invert

Inverts the current response curve.

Mode pop-up menu

Allows you to set the lter mode that determines how the various frequency band controls interact to create the response curve.

EQ Modes

The Mode pop-up menu in the lower right corner allows you to select an EQ mode, which add color or character to the equalized output in various ways.

True Response

Applies serial lters with an accurate frequency response.

Digital Standard

In this mode, the resonance of the last band depends on the sample rate.

Classic

Applies a classic parallel lter structure where the response does not follow the set gain values accurately.

Page 46

Steinberg VST 3 Plug-ins Limiter

VariableQ

ConstQ asym

ConstQ sym

Resonant

Limiter

Limiter is designed to ensure that the output level never exceeds a set output level, to avoid

clipping in following devices.

Applies parallel lters where the resonance depends on the amount of gain.

Applies parallel lters where the resonance is raised when boosting the gain and vice versa.

Applies parallel lters where the resonance of the rst and last bands depends on the sample rate.

Applies serial lters where a gain increase of one band lowers the gain in adjacent bands.

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).

Input (-24 to 24 dB)

Sets the input gain.

Release (0.1 to 1000 ms or Auto mode)

Sets the time after which the gain returns to its original level. If Auto Release is activated, the plug-in automatically material.

Output

Sets the maximum output level.

nds the best release setting for the audio

Page 47

Steinberg VST 3 Plug-ins L/R to M/S, M/S to L/R

L/R to M/S, M/S to L/R

This plug-in allows you to convert a stereo signal into a M/S signal and vice versa.

The L/R to M/S tool converts a L/R signal that is divided into a left and a right signal into a M/S signal that is divided into a mid signal (L+R) and side signals (L-R).

The M/S to L/R tool reconverts the M/S signal into a L/R signal.

Magneto II

Magneto II simulates the saturation and compression of recording on analog tape machines.

Saturation

Determines the amount of saturation and the generation of overtones. This leads to a small increase in input gain.

Saturation On/Off

Activates/Deactivates the saturation effect.

Dual Mode

Simulates the use of two machines.

Frequency Range Low/High

These parameters set the frequency range of the spectrum band to which the tape effect is applied.

For example, to avoid the saturation of lower frequencies, set the Low value to 200 Hz or 300 Hz. To avoid the saturation of very high frequencies, set the parameter to values below 10 kHz.

Solo

Allows you to hear only the set frequency range including the tape simulation effect. This helps you to determine the appropriate frequency range.

HF-Adjust

Sets the amount of high frequency content of the saturated signal.

HF-Adjust On/Off

Activates/Deactivates the HF-Adjust lter.

High

Page 48

Steinberg VST 3 Plug-ins Maximizer

Maximizer

Maximizer raises the loudness of audio material without the risk of clipping. The plug-in provides two modes, Classic and Modern, that offer different algorithms and parameters.

Classic

Classic mode provides the classic algorithms from previous versions of this plug-in.

This mode is suited for all styles of music.

Modern

In Modern mode, the algorithm allows for more loudness than in Classic mode. This mode is particularly suited for contemporary styles of music.

Modern mode also provides additional settings to control the release phase:

● Release sets the overall release time.

● Recover allows for a faster signal recovering at the beginning of the release

phase.

Optimize

Determines the loudness of the signal.

Mix

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

Output

Sets the maximum output level.

Soft Clip

If this button is activated, Maximizer starts limiting or clipping the signal softly. At the same time, harmonics are generated, adding a warm, tube-like characteristic to the audio material.

Page 49

Steinberg VST 3 Plug-ins Mix6to2

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 is included in the resulting mix.

NOTE

This plug-in does not simulate a surround mix or add any psycho-acoustical artifacts to the resulting output – it is simply a mixer. The plug-in is only available in the surround audio montage is active.

Master Section and if a

Surround Channels

Volume faders

Determine how much of the signal is included in the left and/or right channel of the output bus.

Link

Links the volume faders of a surround channel.

Invert Phase

Inverts the phase of the corresponding surround bus channel.

Output Bus

Volume faders

Set the volume of the mixed output.

Link

Links the Output faders.

Normalize

If this option is activated, the mixed output is normalized. For example, the output level is automatically adjusted so that the loudest signal is as loud as possible without clipping.

Page 50

Steinberg VST 3 Plug-ins Mix8to2

Mix8to2

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 is included in the resulting mix.

NOTE

This plug-in does not simulate a surround mix or add any psycho-acoustical artifacts to the resulting output – it is simply a mixer. The plug-in is only available in the 8 channel audio montage is active.

Master Section and if a

Surround Channels

Volume faders

Determine how much of the signal is included in the left and/or right channel of the output bus.

Link

Link the volume faders.

Invert Phase

Inverts the phase of the corresponding surround bus channel.

Output Bus

Volume faders

Set the volume of the mixed output.

Link

Links the Output faders.

Normalize

If this option is activated, the mixed output is normalized. For example, the output level is automatically adjusted so that the loudest signal is as loud as possible without clipping.

Page 51

Steinberg VST 3 Plug-ins MonoDelay

MonoDelay

This is a mono delay effect that can either be tempo-based or use freely specied delay time settings.

Lo Filter

Hi Filter

Affects the feedback loop of the effect signal and allows you to roll off low frequencies. The button below the knob activates/deactivates the lter.

Affects the feedback loop of the effect signal and allows you to roll off high frequencies. The button below the knob activates/deactivates the lter.

Delay

If Tempo Sync is activated, this sets the base note value for the delay. If Tempo Sync is deactivated, the delay time can be set freely in milliseconds.

Feedback

Sets the amount of the signal that is sent back into the delay input. The higher this value, the higher the number of repeats.

Mix

MonoToStereo

MonoToStereo turns a mono signal into a pseudo-stereo signal. The plug-in can be used on a

mono le or a stereo le with equal channels.

NOTE

This plug-in works only on stereo tracks.

Delay

Increases the amount of differences between the left and right channels to further increase the stereo effect.

Page 52

Steinberg VST 3 Plug-ins MultibandCompressor

Width

Controls the width or depth of the stereo enhancement. Turn clockwise to increase the enhancement.

Mono

Switches the output to mono, to check for possible unwanted coloring of the sound which sometimes can occur when creating an

Color

Generates additional differences between the channels to increase the stereo enhancement.

MultibandCompressor

MultibandCompressor allows a signal to be split into four frequency bands. You can specify the

level, bandwidth, and compressor characteristics for each band.

articial stereo image.

NOTE

To compensate for output gain loss that is caused by compression, MultibandCompressor uses an automatic make-up gain. If side-chaining is activated for a frequency band in the side-chain section, the automatic make-up gain is deactivated for this band.

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. The vertical value scale to the left shows the gain level of each frequency band. The horizontal scale shows the available frequency range.

Page 53

Steinberg VST 3 Plug-ins MultibandCompressor

● To dene the frequency range of the different frequency bands, use the handles at the

● To attenuate or boost the gain of the frequency bands by ±15 dB after compression, use

Live

Bypassing Frequency Bands

Soloing Frequency Bands

Output (-24 to 24 dB)

Compressor Section

sides of each frequency band.

the handles at the top of each frequency band.

To bypass each frequency band, activate the Bypass Band button in each section.

To solo a frequency band, activate the S button in each section. Only one band can be soloed at a time.

Sets the output level.

You can specify the Threshold and Ratio by moving breakpoints or using the corresponding knobs. The threshold is marked by the diagonal.

Threshold (-60 to 0 dB)

Determines the level where the compressor kicks in. Only signal levels above the set threshold are processed.

Ratio

Sets the amount of gain reduction applied to signals above the set threshold. A ratio of 3:1 means that for every 3 dB the input level increases, the output level increases by 1 dB.

Attack (0.1 to 100 ms)

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 passes through unprocessed.

Release (10 to 1000 ms or Auto mode)

rst breakpoint where the line deviates from the straight

Side-Chain Section

To open the side-chain section, click the SC button at the bottom left of the plug-in window.

IMPORTANT

To be able to use the side-chain function for the bands, global side-chain must be activated for the plug-in.

Page 54

Steinberg VST 3 Plug-ins MultibandEnvelopeShaper

Frequency

If the Side-Chain button is activated, this sets the frequency of the side-chain lter.

Q-Factor

If the Side-Chain button is activated, this sets the resonance or width of the lter.

Side-Chain

Activates the internal side-chain lter. The side-chain signal can then be shaped according to the

Monitor

Allows you to monitor the ltered signal.

lter parameter.

MultibandEnvelopeShaper

MultibandEnvelopeShaper allows a signal to be split into four frequency bands. You can

attenuate or boost the gain of the attack and release phase of audio material for each band.

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. The vertical value scale to the left shows the gain level of each frequency band. The horizontal scale shows the available frequency range.

Page 55

Steinberg VST 3 Plug-ins MultibandEnvelopeShaper

● To dene the frequency range of the different frequency bands, use the handles at the sides of each frequency band.

● To attenuate or boost the gain of the frequency band, use the handles at the top of each frequency band.

Live

Bypassing Frequency Bands

To bypass each frequency band, activate the Bypass Band button in each section.

Soloing Frequency Bands

To solo a frequency band, activate the S button in each section. Only one band can be soloed at a time.

Output (-24 to 24 dB)

Sets the output level.

Shaper Section

You can specify the Attack, Length, and Release by moving breakpoints or using the corresponding knobs. Be careful with levels when boosting the gain. You can reduce the output level to avoid clipping.

Attack (-20 to 20 dB)

Sets the gain of the attack phase of the signal.

Length (5 to 200 ms)

Sets the length of the attack phase.

Release

Sets the gain of the release phase of the signal.

Sensitivity (-40 to -10 dB)

Sets the sensitivity of the detection.

Output

Sets the output level.

Page 56

Steinberg VST 3 Plug-ins MultibandExpander

MultibandExpander

MultibandExpander allows a signal to be split into four frequency bands. You can reduce the

output level in relation to the input level for signals below the set threshold for each band. This is useful if you want to enhance the dynamic range or reduce the noise in quiet passages.

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 expansion. The vertical value scale to the left shows the gain level of each frequency band. The horizontal scale shows the available frequency range.

● To dene the frequency range of the different frequency bands, use the handles at the sides.

● To attenuate or boost the gain of the frequency band after expansion, use the handles on top of each frequency band.

Live

Bypassing Frequency Bands

To bypass each frequency band, activate the Bypass Band button in each section.

Soloing Frequency Bands

To solo a frequency band, activate the S button in each section. Only one band can be soloed at a time.

specic amount of latency as

Page 57

Steinberg VST 3 Plug-ins MultibandExpander

Output (-24 to 24 dB)

Expander Section

You can specify the Threshold and Ratio by moving breakpoints or using the corresponding knobs. The rst breakpoint from which the line deviates from the straight diagonal is the threshold point.

Threshold

Ratio

Maximum Reduction

Fall (0.1 to 100 ms)

Sets the output level.

Determines the level where the expansion kicks in. Only signal levels below the set threshold are processed.

Sets the amount of gain boost applied to signals below the threshold.

Sets the maximum amount by which the level is reduced when the signal falls below the set threshold.

Determines how fast the expander responds to signals below the set threshold. If the fall time is long, more of the early part of the signal passes through unprocessed.

Hold (0 to 2000 ms)

Sets the time the applied expansion affects the signal below the threshold.

Rise (10 to 1000 ms or Auto mode)

Output

Sets the output level.

Side-Chain Section

Frequency

If the Side-Chain button is activated, this sets the frequency of the side-chain lter.

Q-Factor

If the Side-Chain button is activated, this sets the resonance or width of the lter.

Side-Chain

Activates the internal side-chain lter. The side-chain signal can then be shaped according to the lter parameters. Side-chaining is useful for tailoring how the effect operates.

Monitor

Allows you to monitor the ltered signal.

Page 58

Steinberg VST 3 Plug-ins Octaver

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. Octaver is best used with monophonic signals.

Direct

Octave 1

Sets the level balance between the dry signal and the wet signal. A value of 0 means that only the generated and transposed signal is heard. By raising this value, more of the original signal is heard.

Adjusts the level of the generated signal one octave below the original pitch. A setting of 0 means that the voice is muted.

Octave 2

Adjusts the level of the generated signal two octaves below the original pitch. A setting of 0 means that the voice is muted.

PingPongDelay

This is a stereo delay effect that alternates each delay repeat between the left and right channels.

NOTE

This plug-in works only on stereo tracks.

Lo Filter

Affects the feedback loop of the effect signal and allows you to roll off low frequencies. The button below the knob activates/deactivates the

Hi Filter

Affects the feedback loop of the effect signal and allows you to roll off high frequencies. The button below the knob activates/deactivates the lter.

lter.

Page 59

Steinberg VST 3 Plug-ins PostFilter

Delay

Feedback

Mix

Spatial

Start Left/Start Right

PostFilter

If Tempo Sync is activated, this sets the base note value for the delay. If Tempo Sync is deactivated, the delay time can be set freely in milliseconds.

Sets the amount of the signal that is sent back into the delay input. The higher this value, the higher the number of repeats.

Sets the stereo width for the left/right repeats. Turn clockwise for a more pronounced stereo ping-pong effect.

Determines whether the delay repeat starts on the left or the right channel.

This effect allows quick and easy ltering of unwanted frequencies, creating room for the important sounds in your mix.

PostFilter combines a low-cut lter, a notch lter, and a high-cut lter. You can change the settings by dragging the curve points in the graphical display, or by adjusting the controls below the display section.

Graphical display

Visualizes the settings for all parameters.

Page 60

Steinberg VST 3 Plug-ins PostFilter

Level meter

Low-Cut Freq (20 Hz to 1 kHz, or Off)

Low-Cut Slope

Low-Cut Preview

Shows the output level, giving you an indication of how the ltering affects the overall level of the edited audio.

Allows you to eliminate low-frequency noise. The lter is inactive if the curve point is located all the way to the left. You can set the frequency either in Hz or as a note value. If you enter a note value, the frequency is automatically changed to Hz. For example, a note value of A3 sets the frequency to 440 Hz. When you enter a note value, you can also enter a cent offset. For example, enter A5 -23 or C4 +49.

NOTE

Ensure that you enter a space between the note and the cent offset. Only in this case, the cent offsets are taken into account.

Allows you to choose a slope value for the low-cut lter.

Use this button between the Low-Cut controls and the graphical display to switch the

lter to a complementary high-cut lter. This deactivates any other lters,

allowing you to listen only to the frequencies that you want to lter out.

Notch Freq

Sets the frequency of the notch lter. You can set the frequency either in Hz or as a note value. If you enter a note value, the frequency is automatically changed to Hz. For example, a note value of A3 sets the frequency to 440 Hz. When you enter a note value, you can also enter a cent offset. For example, enter A5 -23 or C4 +49.

NOTE

Ensure that you enter a space between the note and the cent offset. Only in this case, the cent offsets are taken into account.

Notch Gain

Adjusts the gain of the selected frequency. Use positive values to identify the frequencies that you want to lter out.

Notch Gain Invert

This button inverts the gain value of the notch lter. Use this button to lter out unwanted noise. When looking for the frequency to omit, it sometimes helps to boost it of the noise, you can use the Invert button to cancel it out.

Notch Q-Factor

Sets the width of the notch lter.

Notch Preview

Use this button between the notch lter controls and the graphical display to create a band-pass lter with the peak lter's frequency and Q. This deactivates any other

lters, allowing you to listen only to the frequencies you want to lter out.

rst (set the notch lter to positive gain). After you have found the frequency

Notches buttons (1, 2, 4, 8)

These buttons add additional notch lters to lter out harmonics.

High-Cut Freq (3 Hz to 20 kHz, or Off)

This high-cut lter allows you to remove high-frequency noise. The lter is inactive if the curve point is located all the way to the right. You can set the frequency either in

Page 61

Steinberg VST 3 Plug-ins REVelation

High-Cut Slope

High-Cut Preview

REVelation

REVelation produces a high-quality algorithmic reverb with early reections and reverb tail.

Hz or as a note value. If you enter a note value, the frequency is automatically changed to Hz. For example, a note value of A3 sets the frequency to 440 Hz. When you enter a note value, you can also enter a cent offset. For example, enter A5 -23 or C4 +49.

NOTE

Ensure that you enter a space between the note and the cent offset. Only in this case, the cent offsets are taken into account.

Allows you to choose a slope value for the high-cut lter.

This button between the High-Cut controls and the graphical display allows you to switch the lter to a complementary low-cut lter. This deactivates any other lters, allowing you to listen only to the frequencies you want to lter out.

The early reections are responsible for the spatial impression in the rst milliseconds of the reverb. For emulating different rooms, you can choose between different early reections patterns and adjust their size. The reverb tail, or late reverberation, offers parameters for controlling the room size and the reverb time. You can adjust the reverb time individually in 3 frequency bands.

Pre-Delay

Determines how much time passes before the reverb is applied. This allows you to simulate larger rooms by increasing the time it takes for the rst reections to reach the listener.

Early Reections

Here, you select an early reections pattern. The early reections pattern contains the most important delays that deliver the key information for the spatial impression of the room.

Page 62

Steinberg VST 3 Plug-ins REVelation

ER/Tail Mix

Size

Low Cut

High Cut

Delay

Sets the level balance between the early reections and the reverb tail. At a setting of 50 %, early reections and tail have the same volume. Settings below 50 % raise the

reections and lower the tail, as a result the sound source moves towards the

early front of the room. Settings above 50 % raise the tail and lower the early reections, as a result the sound source moves towards the back of the room.

Adjusts the length of the early reections pattern. At a setting of 100 %, the pattern is applied with its original length and the room sounds the most natural. At settings below 100 %, the early smaller.

Attenuates the low frequencies of the early reections. The higher this value, the less low frequencies are present in the early

Attenuates the high frequencies of the early reections. The lower this value, the less high frequencies the early reections will have.

Delays the onset of the reverb tail.

reections pattern is compressed and the room is perceived

reections.

Room Size

Controls the dimensions of the simulated room. At a setting of 100 %, the dimensions correspond to a cathedral or a large concert hall. At a setting of 50 %, the dimensions correspond to a medium-sized room or studio. Settings below 50 % simulate the dimensions of small rooms or a booth.

Main Time

Controls the overall reverb time of the tail. The higher this value, the longer the reverb tail will decay. At a setting of 100 %, the reverb time is innitely long. The

Main Time parameter also represents the mid band of the reverb tail.

High Time

Controls the reverb time for the high frequencies of the reverb tail. With positive values, the decay time of the high frequencies is longer. With negative values, it is shorter. Frequencies are affected depending on the High Freq parameter.

Low Time

Controls the reverb time for the low frequencies of the reverb tail. For positive values, low frequencies decay longer and vice versa. Frequencies will be affected depending on the Low Freq parameter.

High Freq

Sets the cross-over frequency between the mid and the high band of the reverb tail. You can offset the reverb time for frequencies above this value from the main reverb time with the High Time parameter.

Low Freq

Sets the cross-over frequency between the low and the mid band of the reverb tail. The reverb time for frequencies below this value can be offset from the main reverb time with the

Shape

Controls the attack of the reverb tail. At a setting of 0 %, the attack is more immediate, which is a good setting for drums. The higher this value, the less immediate the attack.

Low Time parameter.

Page 63

Steinberg VST 3 Plug-ins RoomWorks

Density

High Cut

Width

Mix

Lock Mix Value

Adjusts the echo density of the reverb tail. At a setting of 100 %, single reections from walls cannot be heard. The lower this value, the more single reections can be heard.

Attenuates the high frequencies of the reverb tail. The lower this value, the less high frequencies the reverb tail will have.

Controls the width of the stereo image. At a setting of 0 %, the output of the reverb is mono, at 100 % it is stereo.

Activate this button (padlock symbol) next to the Mix parameter to lock the dry/wet balance while browsing through the available presets.

Modulation

Modulation allows you to enrich the reverb tail through subtle pitch modulations.

Modulation Rate

Modulation Depth

Modulation Activate

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 t the needs of any system. From short room reverberation.

Species the frequency of the pitch modulation.

Sets the intensity of the pitch modulation.

Activates/Deactivates the chorusing effect.

reections to cavern-sized reverb, this plug-in delivers high quality

Page 64

Steinberg VST 3 Plug-ins RoomWorks

Input Filters

Low Frequency

High Frequency

Low Gain

High Gain

Reverb Character

Pre-Delay

Determines the frequency at which the low-shelving lter takes effect. Both the high and low settings

Determines the frequency at which the high-shelving lter takes effect. Both the high and low settings lter the input signal prior to reverb processing.

Sets the amount of boost or attenuation for the low-shelving lter.

Sets the amount of boost or attenuation for the high-shelving lter.

Determines how much time passes before the reverb is applied. This allows you to simulate larger rooms by increasing the time it takes for the rst reections to reach the listener.

lter the input signal prior to reverb processing.

Size

Alters the delay times of the early reections to simulate larger or smaller spaces.

Reverb Time

Allows you to set the reverb time in seconds.

Diffusion

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

Width

Controls the width of the stereo image. At a setting of 0 %, the output of the reverb is mono, at 100 % it is stereo.

Variation

Clicking this button generates a new version of the same reverb program using altered reection patterns. This is helpful if some sounds are causing odd ringing or undesirable results. Creating a new variation often solves these issues. There are 1000 possible variations.

Hold

Activating this button freezes the reverb buffer in an innite loop. You can create some interesting pad sounds using this feature.

Damping

Low Frequency

Determines the frequency below which low-frequency damping occurs.

High Frequency

Determines the frequency above which high-frequency damping occurs.

Low Level

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

Page 65

Steinberg VST 3 Plug-ins RoomWorks

High Level

Envelope

Amount

Attack

Release

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

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 follows 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 release time of a gate.

Output

Mix

Sets the level balance between the dry signal and the wet signal. If RoomWorks is used as an insert effect for an FX channel, you most likely want to set this to 100 % or use the wet only button.

Wet only

This button deactivates the Mix parameter, setting the effect to 100 % wet or affected signal. This button should normally be activated if RoomWorks is used as a send effect for an FX channel or a group channel.

Eciency

Determines how much processing power is used for RoomWorks. The lower the value, the more CPU resources are used, and the higher the quality of the reverb. Interesting effects can be created with very high

Export

Determines if during audio export RoomWorks uses the maximum CPU power for the highest quality reverb. During export, you may want to keep a higher eciency setting to achieve a specic effect. If you want the highest quality reverb during export, make sure this button is activated.

Output meter

Shows the level of the output signal.

Eciency settings (>90 %).

Page 66

Steinberg VST 3 Plug-ins RoomWorks SE

RoomWorks SE

RoomWorks SE is a smaller version of the RoomWorks plug-in. RoomWorks SE delivers high

quality reverberation, but has fewer parameters and is less CPU demanding than the full version.

Pre-Delay

Determines how much time passes before the reverb is applied. This allows you to simulate larger rooms by increasing the time it takes for the rst reections to reach the listener.

Reverb Time

Diffusion

Low Level

High Level

Mix

StereoDelay

Allows you to set the reverb time in seconds.

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

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

RoomWorks SE inserted in an FX channel, you most likely want to set this to 100 %.

StereoDelay has two independent delay lines with freely specied delay time settings.

NOTE

This plug-in works only on stereo tracks.

Page 67

Steinberg VST 3 Plug-ins StereoEnhancer

Feedback

Delay

Set the number of repeats for each delay.

If Tempo Sync is activated, this sets the base note value for the delay. If Tempo Sync is deactivated, the delay time can be set freely in milliseconds.

Mix

Lo Filter

Affects the feedback loop of the effect signal and allows you to roll off low frequencies. The button below the knob activates/deactivates the lter.

Pan

Sets the stereo position.

Hi Filter

Affects the feedback loop of the effect signal and allows you to roll off high frequencies. The button below the knob activates/deactivates the

StereoEnhancer

StereoEnhancer expands the stereo width of (stereo) audio material. It cannot be used with

mono les.

NOTE

This plug-in works only on stereo tracks.

lter.

Page 68

Steinberg VST 3 Plug-ins Stereo Tools

Delay

Width

Mono

Increases the amount of differences between the left and right channels to further increase the stereo effect.

Controls the width or depth of the stereo enhancement. Turn clockwise to increase the enhancement.

Switches the output to mono, to check for possible unwanted coloring of the sound which sometimes can occur when enhancing the stereo image.

Color

Stereo Tools

Stereo Tools allows you to pan or position both the left and right channels independently of one

another. You can use it for stereo les that you do not want to convert to mono or for xing a problem with the stereo

Invert Left Phase/Invert Right Phase

Generates additional differences between the channels to increase the stereo enhancement.

le, for example.

Inverts the polarity of an audio channel. You can use it for removing the center information or for xing a channel that has been inverted, for example.

Swap Stereo Channels

Swaps the left and right channels.

Transform

Determines the conversion method:

● Nothing: No conversion of the signal.

● Left/Right -> Mid/Side: Converts a stereo signal into a mid/side signal.

Page 69

Steinberg VST 3 Plug-ins StudioChorus

● Mid/Side -> Left/Right: Converts a mid/side signal into a stereo signal.

Left/Mid Gain (dB)

Sets the gain of the left stereo signal or the mid signal of the M/S signal.

Right/Side Gain (dB)

Sets the gain of the right stereo signal or of the side signals of the M/S signal.

Left/Mid Panning (%)

Pans the left stereo signal or the mid signal of the M/S signal.

Right/Side Panning (%)

Pans the right stereo signal or the side signals of the M/S signal.

StudioChorus

StudioChorus is a two-stage chorus effect that adds short delays to the signal and modulates

the pitch of the delayed signals to produce a doubling effect. The two separate stages of chorus modulation are independent and are processed serially (cascaded).

Delay

Affects the frequency range of the modulation sweep by adjusting the initial delay time.

Width

Sets the depth of the chorus effect. Higher settings produce a more pronounced effect.

Spatial

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

Mix

Rate

Allows you to set the sweep rate freely with the Rate knob.

Waveform Shape

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

Page 70

Steinberg VST 3 Plug-ins StudioEQ

Lo Filter/Hi Filter

NOTE

StudioEQ

Studio EQ is a high-quality 4-band parametric stereo equalizer. All four bands can act as fully

parametric peak lters. In addition, the low and high bands can act as either shelving lters (three types) or as cut lters (low-pass/high-pass).

Allow you to roll off low and high frequencies of the effect signal.

Operation Manual.

Main Layout

Reset

Alt-click this button to reset all parameter values.

Show Input/Output Spectrum

Shows the spectrum before and after ltering.

Output

Adjusts the overall output level.

Auto Gain

If this button is activated, the gain is automatically adjusted, keeping the output level nearly constant regardless of the EQ settings.

Page 71

Steinberg VST 3 Plug-ins StudioEQ

Band Settings

Activate/Deactivate Band

Freq

Activates/Deactivates the corresponding band.

NOTE

● If a band is deactivated, you can still modify its parameters.

NOTE

● You can adjust the Freq parameter of a band in the graphical editor by Alt-

clicking the corresponding handle and moving the mouse left and right.

● Ensure that you enter a space between the note and the cent offset. Only in

this case, the cent offsets are taken into account.

Inv

Inverts the gain value of the lter. Use this button to lter out unwanted noise. When looking for the frequency to omit, it sometimes helps to boost it in the the lter to positive gain). After you have found the frequency of the noise, you can use the Inv button to cancel it out.

For Peak lters, this parameter controls the width of the band. For Shelf lters, it adds a drop or a boost, depending on the gain setting of the band. For Cut lters, it adds a resonance.

NOTE

● You can adjust the Q parameter of a band in the graphical editor by Shift-

clicking the corresponding handle and moving the mouse up and down. Alternatively, you can point on the handle and turn the mouse wheel.

Gain

Sets the amount of attenuation/boost for the corresponding band.

rst place (set

NOTE

● You can adjust the Gain parameter of a band in the graphical editor by Ctrl/

Cmd-clicking the corresponding handle and moving the mouse up and down.

● This parameter is not available for Cut lters.

Filter type

For the low and high band, you can choose between three types of shelving lters, a peak lter (band-pass), and a cut lter (low-pass/high-pass). If Cut mode is selected, the Gain parameter is xed.

Page 72

Steinberg VST 3 Plug-ins TestGenerator

● Shelf I adds resonance in the opposite gain direction slightly above the set

● Shelf II adds resonance in the gain direction at the set frequency.

● Shelf III is a combination of Shelf I and II.

TestGenerator

This utility plug-in allows you to generate an audio signal, which can be recorded as an audio le.

The resulting le can then be used for a number of purposes:

● Testing the specications of audio equipment

● Measurements of various kinds, such as calibrating tape recorders

● Testing signal processing methods

● Educational purposes

frequency.

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

Waveforms and noise section

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

Frequency section

Allows you to set the frequency of the generated signal. You can set the frequency either in Hz or as a note value. If you enter a note value, the frequency is automatically changed to Hz. For example, a note value of A3 sets the frequency to 440 Hz. When you enter a note value, you can also enter a cent offset. For example, enter A5 -23 or C4 +49.

NOTE

Ensure that you enter a space between the note and the cent offset. Only in this case, the cent offsets are taken into account.

Gain section

Allows you to set the amplitude of the signal. The higher the value, the stronger the signal. You can select one of the preset values, or use the slider to set a value between -81 and 0 dB.

le according to the signal specications.

Page 73

Steinberg VST 3 Plug-ins Tube Compressor

Tube Compressor

This versatile compressor with integrated tube-simulation allows you to achieve smooth and warm compression effects. The VU meter shows the amount of gain reduction. Tube

Compressor features an internal side-chain section that lets you lter the trigger signal.

Drive (1.0 to 6.0 dB)

Controls the amount of tube saturation.

Input

Determines the compression amount. The higher the input gain, the more compression is applied.

Ratio

Toggles between a low and a high ratio value.

Output (-12 to 12 dB)

Sets the output gain.

Character

Keeps the bass tight and preserves its attacks by decreasing the tube saturation for lower frequencies, and adds brilliance by creating harmonics for higher frequencies.

Attack (0.1 to 100 ms)

Determines how fast the compressor responds. If the attack time is long, more of the initial part of the signal passes through unprocessed.

Release (10 to 1000 ms or Auto mode)

Sets the time after which the gain returns to its original level. If Auto Release is activated, the plug-in automatically nds the best release setting for the audio material.

Mix

Adjusts the mix between dry signal and wet signal, preserving the transients of the input signal.

Page 74

Steinberg VST 3 Plug-ins VintageCompressor

In/Out Meters

VU Meter

Side-Chain

Side-chain section

Filter Type (Low-Pass/Band-Pass/High-Pass)

Center (50 to 20000 Hz)

Q-Factor

Show the highest peaks of all available input and output channels.

Shows the amount of gain reduction.

Activates the internal side-chain lter. The input signal can then be shaped according

lter parameters. Internal side-chaining is useful for tailoring how the gate

to the operates.

If Side-Chain is activated, these buttons allow you to set the lter type to low-pass, band-pass, or high-pass.

If the Side-Chain button is activated, this sets the center frequency of the lter.

If the Side-Chain button is activated, this sets the resonance or width of the lter.

Monitor

Allows you to monitor the ltered signal.

VintageCompressor

VintageCompressor is modeled after vintage type compressors.

This compressor features separate controls for Input and Output gain, Attack, and Release. In addition, there is a dependent Auto feature for the Release parameter.

Input

Determines the compression amount. The higher the input gain, the more compression is applied.

Punch mode which preserves the attack phase of the signal and a program-

Output (-48 to 24 dB)

Sets the output gain.

Attack (0.1 to 100 ms)

Determines how fast the compressor responds. If the attack time is long, more of the initial part of the signal passes through unprocessed.

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Steinberg VST 3 Plug-ins VSTDynamics

Punch

If 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.

Release (10 to 1000 ms or Auto mode)

Sets the time after which the gain returns to its original level. If Auto Release is activated, the plug-in automatically material.

Mix

Adjusts the mix between dry signal and wet signal, preserving the transients of the input signal.

VU Meter

Shows the amount of gain reduction.

In/Out Meters

Show the highest peaks of all available input and output channels.

VSTDynamics

nds the best release setting for the audio

VSTDynamics is an advanced dynamics processor. It combines three separate effects: Gate, Compressor, and Limiter, covering a variety of dynamic processing functions.

Gate

Gating, or noise gating, is a method of dynamic processing that silences audio signals below a set threshold. As soon as the signal level exceeds the threshold, the gate opens to let the signal through. The gate trigger input can also be ltered using an internal side-chain signal.

The following parameters are available:

Input meter

Shows the level of the input signal.

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Steinberg VST 3 Plug-ins VSTDynamics

Attack (0.1 to 100 ms)

Threshold

State LED

Release (10 to 1000 ms or Auto mode)

Hold (0 to 2000 ms)

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 passes through unprocessed.

Determines the level at which the gate is activated. Signal levels above the set threshold trigger the gate to open, and signal levels below the set threshold close the gate.

Indicates whether the gate is open (LED lights up in green), closed (LED lights up in red), or in an intermediate state (LED lights up in yellow).

Sets the time after which the gate closes after the set Hold time. If Auto Release is activated, the plug-in automatically nds the best release setting for the audio material.

Determines how long the gate remains open after the signal drops below the threshold level.

Range

Adjusts the attenuation of the gate when it is shut. If Range is set to minus innite

, the gate is completely shut. The higher the value, the higher the level of the

signal that passes through the shut gate.

Side-Chain

Activates the internal side-chain lter. The input signal can then be shaped according to the lter parameters. Internal side-chaining is useful for tailoring how the gate operates.

Filter Type (Low-Pass/Band-Pass/High-Pass)

If Side-Chain is activated, these buttons allow you to set the lter type to low-pass, band-pass, or high-pass.

Center (50 to 20000 Hz)

If the Side-Chain button is activated, this sets the center frequency of the lter.

Q-Factor

If the Side-Chain button is activated, this sets the resonance or width of the lter.

Monitor

Allows you to monitor the ltered signal.

Compressor

Compressor reduces the dynamic range of the audio, making softer sounds louder or louder

sounds softer, or both. It features a separate display that graphically illustrates the compressor curve shaped according to your settings.

Input meter

Shows the level of the input signal.

Graphical display

Visualizes the settings for Threshold and Ratio and allows you to adjust them by dragging the handles.

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Steinberg VST 3 Plug-ins VSTDynamics

Gain Reduction meter

Threshold (-60 to 0 dB)

Ratio

Make-Up (0 to 24 dB or Auto mode)

Attack (0.1 to 100 ms)

Release (10 to 1000 ms or Auto mode)

Shows the amount of gain reduction.

Determines the level where the compressor kicks in. Only signal levels above the set threshold are processed.

Sets the amount of gain reduction applied to signals above the set threshold. A ratio of 3:1 means that for every 3 dB the input level increases, the output level increases by 1 dB.

Compensates for output gain loss caused by compression. If Auto Make-Up Gain is activated, the output is automatically adjusted for gain loss.

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 unprocessed.

Limiter

A limiter ensures that the output level never exceeds a set threshold to avoid clipping in effects following in the chain. Conventional limiters usually require a very accurate setup of the attack and release parameters to prevent the output level from going beyond the set threshold level.

Limiter adjusts and optimizes these parameters automatically according to the audio material.

Input meter

Shows the level of the input signal.

Gain Reduction meter

Shows the amount of gain reduction.

Soft Clip

If this button is activated, the signal is limited when the signal level exceeds -6 dB. At the same time, harmonics are generated, adding a warm, tube-like characteristic to the audio material.

Output

Sets the maximum output level.

Release (10 to 1000 ms or Auto mode)

Sets the time after which the gain returns to its original level. If Auto Release is activated, the plug-in automatically nds the best release setting for the audio material.

Output section

Output meter

Shows the level of the output signal.

Module Congurator

Changes the signal ow through the three effects. Changing the order of the effects can produce different results, and the available routing congurations allow you to

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Steinberg VST 3 Plug-ins VSTDynamics

● G-C-L (Gate-Compressor-Limiter)

● C-L-G (Compressor-Limiter-Gate)

● C-G-L (Compressor-Gate-Limiter)

quickly compare what works best for a given situation. Click Module Congurator to toggle between the following routing congurations:

Page 79

Legacy Plug-ins

Under Windows, a set of plug-ins is provided for compatibility with audio projects that referenced these effects when using earlier versions of WaveLab. An audio montage which referenced these plug-ins would otherwise require cumbersome user intervention to open, for example.

Their use with new audio projects is not recommended and they are not documented.

Page 80

Dithering Plug-ins

Dithering plug-ins add small quantities of noise to a signal to reduce the audibility of low level distortion in a digital recording. A small amount of random noise is added to the analog signal before the sampling stage, reducing the effect of quantization errors.

Internal Dithering

This is a WaveLab-specic plug-in that provides a simple way of adding a small amount of noise to the rendered signal to improve the apparent signal-to-noise ratio of the output.

The following parameters are available when selecting Internal Dithering.

Noise Type

Sets the noise type for adding to the signal.

● In No Noise mode, no dithering is applied.

● The Noise Type 1 mode is the most all-round method.

● The Noise Type 2 mode emphasizes higher frequencies more than Noise

Type 1.

Noise Shaping

Increases the apparent signal to noise ratio by altering the spectrum of the low-level audio signal which results from lowering the number of bits. The higher the number you select here, the more the noise is moved out of the ear’s mid-range.

Bit Resolution

Allows you to specify the intended bit resolution for the nal audio, after dithering, regardless of whether you want to render the settings or play back in real-time.

Dithering changes the sample resolution, but not the sample size. For example, when dithering 24 bit to 16 bit, the le will still be 24 bit in size, although only 16 bits of information will have signicance. When rendering to a 16-bit le, specify the le resolution to avoid wasting space.

MBIT+™ Dithering

This plug-in allows you to convert and dither to 24, 20, 16, 12, or 8 bits. This is useful for mastering a track for a CD (16-bit) from a 24-bit source, for example.

The MBIT+™ dither algorithm reduces quantization distortion with minimal perceived noise and produces smooth and quiet conversions.

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Dithering Plug-ins MBIT+™ Dithering

Bit Quantization

Sets the bit depth to which you are dithering. MBIT+™ produces 32-bit oating-point output, but all low-order bits will be zero and should be truncated.

Type

Sets the type of dithering. MBIT+™ contains two traditional dithering types and a proprietary MBIT+™ dithering type.

● Type 1 is a traditional dither based on a rectangular probability distribution

function (PDF).

● Type 2 is a traditional dither based on a triangular PDF.

● MBIT+™ offers superior results when used with all types of source material.

Dither Amount

When using MBIT+™ dither, this controls the amount of dithering. The None and Low settings can leave some non-linear quantization distortion or dither noise

modulation, while higher settings completely eliminate the non-linear distortion at the expense of a slightly increased noise cases.

When using Type 1 or Type 2 dither, this controls the number of bits used to perform dithering. In most cases, 1 bit is suce, but over-dithering with 2 bits can be useful in same cases.

Noise Shaping

When using MBIT+™ dither, this controls the amount of noise shaping. The choices range from no noise shaping to very aggressive noise shaping, providing approximately 14 dB of audible noise suppression, at the expense of a higher noise

oor.

When using Type 1 or Type 2 dither, this controls the noise shaping. Dithering noise can be shaped in order to make it less audible. Simple noise shaping performs simple high-pass ltering on the noise. Clear noise shaping aggressively moves the noise toward the Nyquist frequency. noise away from audible bands, and Psych 9 is a 9th-order lter with similar characteristics.

Auto-Blanking

If this option is activated, MBIT+™ mutes dither output when the input is completely silent for at least 0.7 seconds.

oor. The Normal setting is suce for most

Psych 5 is a 5th-order lter designed to move

Minimize Peaks

If this option is activated, spurious peaks in the noise-shaped dither are suppressed.

Harmonics Suppression

If this option is activated, the truncation rules are slightly altered, moving the harmonic quantization distortion away from overtones of audible frequencies. This option does not create any random dithering noise truncation, but with better tonal quality in the resulting signal. This option is

oor. Instead it works more like

Page 82

Dithering Plug-ins UV22HR

UV22HR

applicable only in the modes without dithering noise and without aggressive noise shaping.

UV22HR is an advanced version of Apogee's renowned UV22 dithering algorithm, capable of dithering to 8, 16, 20, or 24 bits.

8, 16, 20, 24 bit

These buttons allow you to select the intended bit resolution for the nal audio. As when using the internal dithering, it is important to set this to the correct resolution.

Applies a normal dither gain.

Applies a lower level of dither noise.

Auto black

If this option is activated, the dither noise is gated during silent passages.

Page 83

ASIO Plug-ins

External Gear

This Master Section plug-in allows you to process audio les using external hardware processors. One or more ASIO outputs are used to send the audio signal to your processor, and corresponding ASIO inputs are used to return the signal from the external processor.

By default, this plug-in is located in the ASIO submenu of the Master Section effects. An ASIO driver must be used, and only one instance of this plug-in is allowed in the Master Section plugin chain.

External Gear Plug-in

In the Effects pane of the Master Section window, select the External Gear plug-in from the ASIO submenu.

Num. Outputs

Here you can set the number of outputs to use. Normally this is the same as the number of inputs (the conguration in which case you set this parameter to 2 with the slider.

Latency

External gear may introduce latency. WaveLab can automatically compensate for this if you select compensation yourself (in milliseconds). The latency introduced by the ASIO driver is automatically taken into account by WaveLab.

Send Level

Allows you to adjust the send level. This should normally be set to 0 dB. If necessary, adjust the input level on the external effect.

Using External Gear

PROCEDURE

1. Select File > Preferences > VST Audio Connections.

2. Set the Audio Device to ASIO.

3. Select the ASIO Plug-ins tab.

4. Select the channels to be used for device output (to gear) and device input (from gear), and click OK.

As Inputs option). However, you can use a mono out/stereo in

Auto (only active during rendering), or you can set this latency

Page 84

ASIO Plug-ins Audio Input

5. In the Effects pane of the Master Section window, select the External Gear plug-in from

6. Make your settings.

AFTER COMPLETING THIS TASK

Now you can process a signal through the external processor, just as if it was a software plug-in effect. If you render a le using the External Gear plug-in, playback is not available during the rendering.

Audio Input

This is a special Master Section plug-in that allows you to render a signal coming in to the inputs of a sound card together with any Master Section effects. This signal can be anything your sound card accepts, such as a feed from a mixer, a recorder, or a microphone.

These should be different I/O channels than the ones you use for playback and recording. The available outputs in this plug-in equals the number of inputs (up to 8).

the ASIO submenu.

The External Gear plug-in window opens.

When the Audio Input plug-in is loaded, wave playback is not possible.

Setting Up the Audio Input Plug-in

PROCEDURE

1. Select File > Preferences > VST Audio Connections.

2. Set the Audio Device to ASIO.

3. Select the ASIO Plug-ins tab.

4. Select the channels to be used for device input, and optionally name them.

5. In the top effect slot of the Effects pane of the Master Section window, select the Audio Input plug-in from the ASIO submenu.

6. In the Audio Input plug-in window, set the number of inputs and the sample rate.

You have to match the number of inputs in the VST Audio Connections tab to the number of inputs selected here.

7. Start playback.

The cursor does not move, but the Play button is lit and you can now monitor the input source. Pressing

8. If you change the settings in the control panel, click Stop, and restart playback to apply them.

9. In the Master Section, click Render.

10. Select a name, an audio format, and a location for the le to be rendered.

11. Click Start.

Recording/rendering starts, recording the external input from the output of the Master Section, including all real-time processing. You can monitor the recording as it happens.

12. Click Stop to stop the recording/rendering.

Stop ends input monitoring.

Page 85

Batch Processing Plug-ins

In the Batch Processor window, you can add a sequence of plug-ins that can be used to process a batch of audio les. These plug-ins can be standard plug-ins available from the Master Section, oine processes available in the Audio Editor, and plug-ins that are only available within batch processing.

The following batch processor plug-ins are described in the WaveLab Pro Operation Manual:

● Loudness Normalizer

Pitch Quantize

●

Pitch Correction

●

● Pan Normalizer

●

Time Stretch

Audio Analyzer

This plug-in allows you to generate text les with statistics about the audio les in a batch process.

This monopass plug-in is exclusive to the Batch Processor window.

If you want to analyze les without writing anything, select No Output in the Output tab of the Batch Processor window.

Page 86

Batch Processing Plug-ins Audio Analyzer

Information to Report

In this section you specify which information to include in the output. The following information can be included:

● Sample rate

● Number of channels

● Real bit resolution

● Number of samples/duration

● Digital peaks

● True peaks

● Integrated loudness

● Loudness range

● Short-term loudness (maximum)

● Short-term loudness (minimum)

● Momentary loudness (maximum)

● Momentary loudness (minimum)

● Loudness balance

● DC Offset

Output

In this section you set up the output of the Audio Analyzer. The following options are available:

Send Analysis Results to XML Processor

If this option is activated, the analysis results are passed as parameters to the XML or HTML output of the batch processor.

Create One Report per Audio File

If this option is activated, one report is created for each audio le in the batch process. The audio

Merge Results into Global Report

If this option is activated, the results of the analysis are merged into a global report. The audio

Open Global Report at End of Batch Process

If this option is activated, a global report opens after the batch process.

File Name Sux

Lets you specify a le name sux. This is necessary when you are using this plug-in multiple times in a batch process, for example, to see the stats before and after specic plug-ins.

Use different suxes for each instance of the Audio Analyzer plug-in used in the processing chain.

le name is used as the report le name.

Output format

Lets you select the output format. The following formats are available:

● Pure text

● HTML

● Adobe PDF

● Open Oce

● Spreadsheet

Page 87

Batch Processing Plug-ins Audio Injector

● XML

Audio Injector

This plug-in allows you to insert an audio le at the beginning and/or end of the audio le being processed. The inserted

This monopass plug-in is exclusive to the Batch Processor window.

Audio File to Inject at the Start (No Longer Than 60 Seconds)

Species the audio le to be added before the main audio le.

Audio File to Inject at the End (No Longer Than 60 Seconds)

Species the audio le to be added after the main audio le.

le can also be crossfaded with the original audio le, if required.

Perform Crossfade

Audio Mixer

This plug-in allows you to mix an audio le with other audio les. The mix-in happens from a specied time and can optionally be repeated at a specied interval.

For example, you can insert a spectral watermark into the audio spectrum or you can insert beep sounds to mark an audio le as demo material.

This monopass plug-in is exclusive to the Batch Processor window.

Audio File to Mix (No Longer Than 30 Seconds)

Allows you to select a crossfade time and shape for the crossfade between the main audio

le and the injected audio le.

Allows you to select the audio le that you want to mix into other audio les. The audio le must be no longer than 30 seconds.

Time Position

Allows you to specify the time position at which the audio le will be mixed in.

Repeat Every

If this option is activated, you can specify the time after which the audio le is repeatedly mixed in.

Page 88

Batch Processing Plug-ins DC Remover

DC Remover

This plug-in allows you to remove any DC Offset from an audio le.

It is useful to apply this plug-in rst in a batch before other plug-ins to avoid further processing a le containing any DC Offset. For example, an audio le that has a DC offset is not at its loudest possible volume when normalized, because the offset consumes headroom.

This multipass plug-in is available in the Batch Processor window and as an oine processor in

Audio Editor.

the

Delay Next Process Activation

This plug-in allows you to delay the processing of the next VST plug-in in the plug-in chain for a certain time.

This monopass plug-in is exclusive to the Batch Processor window.

Delay

Species the duration until the wet signal starts to fade-in the dry signal.

Crossfade Time

Species the duration of the crossfade time.

Fade In/Fade Out

This plug-in allows you to fade the beginning (Fade In) or the end (Fade Out) of a batch audio

le. You can choose the length and shape of the fade, its duration, and the gain you want it to start/end at.

The fade plug-ins are exclusive to the Batch Processor window. Fade In is a monopass plug-in and Fade Out is a multipass plug-in.

Shape

Determines the shape of the fade.

Duration

Determines the duration of the fade.

Start Gain/End Gain

Determines the gain with which the fade starts. It ends with 0 dB.

Page 89

Batch Processing Plug-ins Instructor

Instructor

Instructor is a special utility plug-in that allows you to instruct the next plug-in in the batch with

information about the audio it needs to process. This is useful for situations where you want to use monopass plug-ins that require an analysis stage that is not available at this point.

In effect, the Instructor plug-in turns a monopass plug-in into a dual pass one. Some monopass plug-ins, such as DeNoiser or DeBuzzer, need to learn about the audio they are to process before they can begin processing correctly. The because it can teach the next plug-in in the audio chain about the audio it is about to process.

The Instructor plug-in must be used as a pair:

1. The rst instance replicates the start of the audio stream. This means the next plug-in in the chain receives the start of the audio stream twice.

2. The second instance of the plug-in comes after the plug-in being instructed. It cuts out the extra audio injected by the

For example, this means that the DeNoiser plug-in has time to suciently analyze the audio stream before the second stream start is injected. The “badly” processed rst part of the stream is skipped by the second instance of the Instructor plug-in.

You can set the Instructor plug-in to replicate up to 20 seconds of audio.

Instructor plug-in can help in this situation,

rst instance of the Instructor plug-in.

NOTE

Do not set a value that is longer than the shortest le in the batch, otherwise a short le is over truncated by the second instance of the plug-in.

This monopass plug-in is exclusive to the Batch Processor window.

Replicate Stream Start

Injects the start of the audio stream twice into the next plug-ins. This action must be selected for the

Reverse Stream

If this option is activated, the start of the stream is injected rst in reverse sample order, then in normal sample order. This changes nothing from the spectrum analysis point of view, but it improves the transition between the repeated streams.

Trim Stream Start

Skips the start of the audio stream. This action must be selected for the second instance of the Instructor plug-in.

rst instance of the Instructor plug-in.

Duration

Species how much audio to replicate or skip.

Page 90

Batch Processing Plug-ins Level Normalizer

Level Normalizer

This multipass plug-in allows you to raise or lower levels so that the signal peaks exactly at the specied value just before it is converted to a le.

Peak Level

Specify the highest level of any audio sample.

Stereo Link

Applies the gain to both channels.

Mix to Mono

Mixes the left and right channels. The resulting mono le gets the specied peak level. This ensure a clip-less mix.

Only if Clipping

Only applies a gain change if the audio le is beyond the reference peak level at some point. If not, the signal is untouched.

Loudness Meta Normalizer

This plug-in allows you to normalize a batch of les to the same loudness, while taking the EBU R-128 loudness measurement and a true peak analysis into account.

The purpose of this plug-in is to achieve the same loudness in all les (the highest loudness found, if possible), while being certain that no le clips. For each le, a specic gain is computed by the plug-in once all les have been analyzed and prior to actually applying any gain to achieve the common loudness. If it is not possible to match the highest found loudness, the level of the le with the highest loudness is reduced, so that other les can get the same loudness. Because no peak compression is used, the dynamics is preserved and no distortion is introduced.

This metapass plug-in is exclusive to the Batch Processor window.

Match Loudness

Select what loudness the clip should get. The following options are available:

● Match loudest audio le

Page 91

Batch Processing Plug-ins Loudness Restorer

● Match maximum achievable loudness

● Match specic loudness

Loudness

Determines the specic loudness to match. For example, -23 LUFS if you want to follow the EBU R-128 recommendation for broadcast.

Reference

Select whether WaveLab should use as reference the loudness of the entire clip (EBU R-128 recommendation), the average loudest 3 second audio section (

Loudness Range), or the loudest 3 seconds audio section (Maximum Short-Term Loudness).

Peaks

Select whether WaveLab should refer to sample values (digital peaks) or to analog reconstructed values (true peaks).

Loudness Restorer

Loudness Restorer captures the loudness at a specic point in the audio chain and restores that

loudness at another point. For this reason, the Loudness Restorer must be inserted in pairs into the signal chain: one plug-in for capturing and one plug-in for restoring.

Top of

This multipass plug-in is exclusive to the Batch Processor window.

Mode - Capture

The rst instance in the plug-in pair must be set to this mode. This makes the plug-in read the signal at this position in the audio chain.

Mode - Restore Peak Level/Restore Loudness (EBU R-128 Weighted)

The second instance in the plug-in pair must be set to one of these modes. Select one of these options if you want to use peak levels as a basis for determining what is considered equal level. Restore Loudness (EBU R-128 Weighted) produces a more natural result than Restore Peak Level.

Reference menu

Select whether WaveLab should use as reference the loudness of the entire le (EBU R-128 recommendation), the average loudest 3 second audio section (Top of

Loudness Range), or the loudest 3 seconds audio section (Maximum Short-Term Loudness).

Peak menu

Select whether WaveLab should use sample values (Digital Peaks) or analog reconstructed values (True Peaks).

Page 92

Batch Processing Plug-ins Meta Leveler

Limit to 0 dB

If this option is activated, the restoration process will never result in levels above 0 dB.

Meta Leveler

This plug-in allows you to change the level of a batch of les consistently.

The core purpose of this plug-in is to apply the same gain to all les, while being certain that a specic peak level never exceeds in any le. The unique gain that you want to apply is (possibly) reduced by the plug-in, once all les in the batch have been analyzed, and prior to actually applying the gain across the batch.

Resizer

This metapass plug-in is exclusive to the Batch Processor window.

Gain

Applies the specied gain to each le. The actual gain can be lower and even negative, to not exceed the value specied in the Maximum Peak eld.

Maximum Peak

Species the maximum peak level that any audio le should get at the end of the process.

Peaks

Select whether WaveLab should refer to sample values (digital peaks) or to analog reconstructed values (true peaks).

This plug-in allows you to specify the duration of all audio les in the batch, and to choose whether to insert silence after the end of the chosen duration.

This monopass plug-in is exclusive to the Batch Processor window.

Stereo to Mono

This plug-in allows you to mix a stereo signal down to a mono signal while being certain not to clip while mixing channels because of the multipass approach. You can choose to use the same

Page 93

Batch Processing Plug-ins Trimmer

peak level that the stereo le contains, or set the gain to be applied and/or the maximum level to be reached in the resulting mono le.

This multipass plug-in is exclusive to the Batch Processor window.

Keep Same Peak Level as Stereo File

If this option is activated, the peak level of the resulting mono le is the same as the peak level of the original stereo le.

Gain

Species the increase or decrease in peak level for the resulting mono le, in relation to the original stereo

Maximum Level of Mono Signal

Species the peak level that the resulting mono le must not exceed. This ensure that the output le never clips. This way, the result never exceeds 0 dB, regardless of the

le.

specied Gain value.

Trimmer

This plug-in allows you to remove a specied duration (from 0 ms to 60 s) of audio from the head and/or tail of an audio le.

This monopass plug-in is exclusive to the Batch Processor window.

Page 94

Index

ASIO

Audio Input 84 External Gear 83

Plug-ins 83 Audio Analyzer 85 Audio Injector 87 Audio Input 84 Audio Mixer 87 AutoPan 29

Batch processing plug-ins 85

Audio Analyzer 85

Audio Injector 87

Audio Mixer 87

DC Remover 88

Delay Next Process Activation 88

Fade In/Fade Out 88

Instructor 89

Level Normalizer 90

Loudness Meta Normalizer 90

Loudness Restorer 91

Meta Leveler 92

Resizer 92

Stereo to Mono 92

Trimmer 93 Brickwall Limiter 30

Compressors (continued)

Vintage Compressor 74 VSTDynamics 75

CurveEQ 34

DC Remover 88 DeBuzzer

DeClicker 26

DeEsser 34 Delay Next Process Activation 88 Delays

DeNoiser 26

Distortion 36 Dithering

Dithering plug-ins 80

DualFilter 37 Ducker 4 Dynamic EQ

RestoreRig 24

MonoDelay 51 PingPongDelay 58 StereoDelay 66

RestoreRig 24

UV22HR 82

Internal dithering 80 MBIT+™ dithering 80

MasterRig 18

Channel Extractor 31 Chopper Effects

AutoPan 29 Chorus Effects

Chorus 31

StudioChorus 69 Clip effects

Ducker 4 Compressor

MasterRig 13 Compressors

Compressor 32

DeEsser 34

Maximizer 48

MultibandCompressor 52

Tube Compressor 73

EBU R-128

Audio Analyzer 85 Loudness Meta Normalizer 90

Envelope Shapers

EnvelopeShaper 37 MultibandEnvelopeShaper 54

Equalizer

MasterRig 17

Expanders

Expander 38 MultibandExpander 56

External Gear 83

Fade In/Fade Out 88 Frequency 40

Page 95

Index

Gates

Gate 43

VSTDynamics 75 GEQ-10 GEQ-30 45

Imager

MasterRig 22 Instructor 89 Internal dithering 80 iZotope

MBIT+™ dithering 80

L/R to M/S 47 Legacy plug-ins 79 Level Normalizer 90 Leveler 5 Leveler Multi 6 Limiter

MasterRig 10 Limiters

Brickwall Limiter 30

Limiter 46

Maximizer 48

VSTDynamics 75 Loudness Meta Normalizer 90 Loudness Restorer 91

M/S to L/R 47 Magneto II 47 MasterRig 6

Compressor 13

Dynamic EQ 18

Equalizer 17

Imager 22

Layout 6

Limiter 10

Modules 9

Saturator 20

Settings 9 Maximizer 48 MBIT+™ dithering 80 Meta Leveler 92 Mix6to2 49 Mix8to2 50 MonoDelay 51 MonoToStereo 51 MultibandCompressor 52 MultibandEnvelopeShaper 54 MultibandExpander 56

Peak Master 23 PingPongDelay 58 Plug-ins

ASIO 83 Batch processing 85 Dithering 80 Legacy 79 VST 3 29 WaveLab-specic 4

PostFilter

Recording

Audio Input 84 Resampler 4 Resizer 92 RestoreRig 24

Layout 24

Modules 25 REVelation 61 RoomWorks 63 RoomWorks SE 66

Sample rate

Resampler 4 Saturation

Magneto II 47 Saturator

MasterRig 20 Silence 28 Stereo Expander 28 Stereo to Mono 92 Stereo Tools 68 StereoDelay 66 StereoEnhancer 67 StudioChorus 69 StudioEQ 70

TestGenerator 72 Trimmer 93 Tube Compressor 73

UV22HR 82

Vintage Compressor 74 VSTDynamics 75

Octaver 58

Steinberg Wavelab Pro - 10.0 Plug-in Reference

Specifications and Main Features

Frequently Asked Questions

User Manual

Table of Contents

Resampler

Ducker

Leveler

Leveler Multi

MasterRig

Main Layout

Modules

Global Settings

Limiter

Compressor

Equalizer

Dynamic EQ

Saturator

Imager

Peak Master

RestoreRig

Main Layout

Modules

DeClicker

DeNoiser

DeBuzzer

Silence

Stereo Expander

Steinberg VST 3 Plug-ins

AutoPan

Brickwall Limiter

Channel Extractor

Chorus

Compressor

CurveEQ

DeEsser

Distortion

DualFilter

EnvelopeShaper

Expander

Frequency

Global Settings

Gate

GEQ-10/GEQ-30

Limiter

L/R to M/S, M/S to L/R

Magneto II

Maximizer

Mix6to2

Mix8to2

MonoDelay

MonoToStereo

MultibandCompressor

MultibandEnvelopeShaper

MultibandExpander

Octaver

PingPongDelay

PostFilter

REVelation

RoomWorks

RoomWorks SE

StereoDelay

StereoEnhancer

Stereo Tools

StudioChorus

StudioEQ

TestGenerator

Tube Compressor

VintageCompressor

VSTDynamics

Legacy Plug-ins

Dithering Plug-ins

Internal Dithering

MBIT+™ Dithering

UV22HR

ASIO Plug-ins

External Gear

External Gear Plug-in

Using External Gear

Audio Input