Plug-in Reference

Steinberg Documentation Team: Cristina Bachmann, Martina Becker, Heiko Bischoff, Lillie Harris, Christina
Kaboth, Dennis Martinez, Insa Mingers, Matthias Obrecht, Sabine Pfeifer
Translation: Ability InterBusiness Solutions (AIBS), Moon Chen, Jérémie Dal Santo, Rosa Freitag, GiEmme
Solutions, Josep Llodra Grimalt, Vadim Kupriianov, Roland Münchow, Boris Rogowski, Sergey Tamarovsky
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© Steinberg Media Technologies GmbH, 2024.
All rights reserved.
WaveLab Pro_12.0.0_en-US_2024-01-24
 Table of Contents

4 VST Plug-ins
4 Analyzer
33 Delay
36 Distortion
41 Dynamics
69 EQ
80 Filter
83 Mastering
105 Modulation
108 Pitch Shift
108 Restoration
114 Reverb
120 Spatial
123 Surround
125 Tools
130 Batch Processing Plug-ins
130 Audio Analyzer
132 Audio Injector
132 Audio Mixer
133 DC Remover
133 Delay Next Process Activation
133 Fade In/Fade Out
134 Instructor
135 Level Normalizer
135 Loudness Meta Normalizer
138 Loudness Restorer
138 Meta Leveler
140 Resizer
140 Stereo to Mono
141 Trimmer
142 Index

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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 in the the Organize tab of the Plug-ins
Preferences.
● VST plug-ins have their own preset handling. You can save or load effect programs (presets).

Analyzer

SuperVision
SuperVision is a professional tool suite for monitoring and analyzing your audio. The plug-in
comes with several different modules for level, spectral, phase, or waveform analysis. Up to nine
module slots allow you to create custom layouts for better overview.

SuperVision features two different processing modes: Maximum Audio Performance and
Sample-Accurate Display. You can choose which mode to use for each module independently.

Toolbar

Pause Measurement

Pauses/Continues the measurement for the selected module. Alt/Opt -click this button
to pause/continue all modules at the same time.

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NOTE

● You can also pause/continue the measurement for the selected module by right-
clicking it.
● If a module is paused, you can still adjust the graphical display that holds the last
measured values.
● For all modules that show a playback cursor, you can click on the paused display to
set the project cursor to this position. This is not possible during recording.

Hold Current Values on Stop

If this button is activated, the last measured values remain on the display when
playback is stopped.

Module selector

Allows you to select a module for the selected slot.

Open Module Settings

Opens the Module Settings window. It provides the settings for the selected module.

Channel selector

Allows you to select which channels are displayed. The available channel configurations
depend on the channel configuration of the track and on the selected module.
Mixdown allows you to display the average value of all channels within the track.

NOTE

● The channel selector is only available for configurations with two or more
channels.

Reset Module Values

Resets the measured values of the selected module. Alt/Opt -click to reset all modules
at the same time.

NOTE

You can also reset the measured values of a module by Ctrl/Cmd -clicking it.

Reset Module Values on Start

If this button is activated, all values are automatically reset when you start playback.

Split Horizontally

Splits the selected module slot horizontally.

NOTE

This button is not available if a module is maximized.

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Split Vertically

Splits the selected module slot vertically.

NOTE

This button is not available if a module is maximized.

Module Slot Controls

Each module slot shows the following controls in the top right corner if you move the mouse
over it:

Remove module slot

Removes the module slot from the current plug-in layout.

Split horizontally

Splits the module slot horizontally.

Split vertically

Splits the module slot vertically.

You can maximize a module by double-clicking it. To downsize it, double-click it again or click the
standard view button .

If your layout shows more than one module, you can change the focus by clicking a module or
pressing Tab .

In some modules, for example, Level, Loudness, or Time, you can press Ctrl/Cmd - S to copy
parameter values as text from the selected module to the clipboard for further use in other
applications.

You can show the current frames per second (fps) for all modules by pressing Alt/Opt - F .

RELATED LINKS
Module Settings Window on page 6
Signal Modules on page 8
Spectral Domain Modules on page 18
Phase Modules on page 24
Spatial Domain Modules on page 27
Waveform Modules on page 29
Other on page 32

Module Settings Window

In the Module Settings window, you can make individual settings for the selected module.

● To open the Module Settings window, click Open Module Settings on the plug-in
toolbar.

The settings on the Module Settings window toolbar are available for all modules:

Reset Settings

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Resets all parameter settings to the default values of the selected module.

Maximum Audio Performance/Sample-Accurate Display

Sets the processing mode for the selected module.

If this button is activated, Maximum Audio Performance is selected. In this mode,
the plug-in has no impact on the audio performance, but the analysis might not be
sample-accurate.
If this button is deactivated, Sample-Accurate Display is selected. In this mode, no
audio sample gets lost for analysis, but the audio performance might be slightly
reduced.

NOTE

Sample-Accurate Display is not available for all modules.

Enable Warnings

If this button is activated, a red border around the affected module indicates that the
displayed analysis result might not be completely sample-accurate.

NOTE

This setting is only available in Maximum Audio Performance mode.

Force Horizontal Display

If this button is activated, the module is always displayed horizontally when you resize
it.

NOTE

This setting is not available for all modules.

Force Vertical Display

If this button is activated, the module is always displayed vertically when you resize it.

NOTE

This setting is not available for all modules.

For the specific settings of a module, see the corresponding module description.

By default, only the top row of controls is displayed. Advanced shows/hides all settings for a
module.

RELATED LINKS
Signal Modules on page 8
Spectral Domain Modules on page 18
Phase Modules on page 24
Spatial Domain Modules on page 27
Waveform Modules on page 29
Other on page 32

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Signal Modules
The modules in this category allow you to visualize the level of the audio signal.

The following modules and module-specific settings are available:

Level
This module shows the level of your audio. It provides a multi-channel level meter and a
maximum level value display.

The following module-specific settings are available in the Module Settings window:

Scale
Allows you to select a scale according to different broadcast standards (Internal,
Digital, DIN, EBU, British, Nordic, K-20, K-14, K-12, +3 dB Digital, +6 dB Digital, or
+12 dB Digital).

Peak Hold
Specifies for how long the peak levels are held on the display.

Peak Fallback
Sets the speed of release for the level meters and peak indicators.

NOTE

● Alternatively, you can move the mouse pointer over the selected module, hold
Ctrl/Cmd , and use the mouse wheel to adjust this parameter.
● If this control is turned all the way to the left, the peak indicators are disabled.

Threshold
Sets a threshold level below which the display is masked.

NOTE

Alternatively, you can move the mouse pointer over the selected module and use the
mouse wheel to adjust this parameter.

Offset
Sets the offset between measured and displayed value in dB.
This parameter is only available for the DIN, EBU, British, and Nordic scale.

Clipping
Sets the clipping value for the Internal scale.

Minimum
Sets the minimum value for the Internal scale.

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Maximum
Sets the maximum value for the Internal scale.

Color
Sets the color of the meters. You can choose between the Scale color and the Track
color.

RMS AES17
Shows the level in accordance with AES17 (RMS +3 dB).

RMS Resolution
Sets the RMS resolution for the level display in milliseconds.

Max. Value
Sets the measurement mode for the maximum level value display. The following
modes are available:
● True Peak shows estimated real peak value for each channel.
● Peak Max. shows maximum sample value for each channel.
● RMS Max. shows the maximum RMS value for each channel.
● RMS Max. + True Peak shows the highest maximum RMS and the highest
estimated real peak value of all channels.
● RMS Max. + Peak Max. shows the highest maximum RMS and the highest
maximum sample value of all channels.

VU
This module shows the level of your audio on a classic VU meter. In addition to the VU meter
needle and the LED peak indicator, it provides a peak level indicator needle and a numeric
maximum level value display.

The following module-specific settings are available in the Module Settings window:

Scale
Allows you to select a scale according to different broadcast standards (Internal,
Digital, DIN, EBU, British, Nordic, K-20, K-14, K-12, +3 dB Digital, +6 dB Digital,
+12 dB Digital, VU dB, and VU dBFS).

Peak Hold
Specifies for how long the peak levels are held on the display.

Peak Fallback
Sets the speed of release for the level meters and peak indicators.

NOTE

● Alternatively, you can move the mouse pointer over the selected module, hold
Ctrl/Cmd , and use the mouse wheel to adjust this parameter.
● If this control is turned all the way to the left, the peak indicators are disabled.

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Meter Mode
Sets the behavior of the needle.

● VU mode imitates the physical behavior of an analog VU meter that shows the
current peak value.
● Peak mode shows the current peak value.
● RMS mode shows the current RMS value.

Offset
Sets the offset between measured and displayed value in dB.
This parameter is only available for the DIN, EBU, British, and Nordic scale.

Clipping
Sets the clipping value for the Internal scale.

Minimum
Sets the minimum value for the Internal scale.

Maximum
Sets the maximum value for the Internal scale.

Color
Sets the color of the meters. You can choose between the Track color and a Dark or
Light scheme.

RMS AES17
Shows the level in accordance with AES17 (RMS + 3 dB).

RMS Resolution
Sets the RMS resolution for the level display in milliseconds.

Max. Value
Sets the measurement mode for the maximum level value display. The following
modes are available:
● True Peak shows the estimated real peak value for each channel.
● Peak Max. shows the maximum sample value for each channel.
● RMS Max. shows the maximum RMS value for each channel.
● RMS Max. + True Peak shows the highest maximum RMS and the highest
estimated real peak value of all channels.
● RMS Max. + Peak Max. shows the highest maximum RMS and the highest
maximum sample value of all channels.

Level Histogram
This module shows a histogram for the peak or RMS value of the input level.

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The following module-specific settings are available in the Module Settings window:

Scale
Allows you to select a scale according to different broadcast standards (Internal,
Digital, DIN, EBU, British, Nordic, K-20, K-14, K-12, +3 dB Digital, +6 dB Digital, or
+12 dB Digital).

Meter Mode
Sets the displayed level value.

● Peak mode shows a histogram of the peak value.

● RMS mode shows a histogram of the RMS value.

Peak Fallback
Sets the speed of release for the level meters and peak indicators.

NOTE

● If you change this parameter during playback, you must click Reset Module
Values to update the display.
● Alternatively, you can move the mouse pointer over the selected module, hold
Ctrl/Cmd , and use the mouse wheel to adjust this parameter.
● If this control is turned all the way to the left, the peak indicators are disabled.

Offset
Sets the offset between measured and displayed value in dB.
This parameter is only available for the DIN, EBU, British, and Nordic scale.

Clipping
Sets the clipping value for the Internal scale.

Minimum
Sets the minimum value for the Internal scale.

Maximum
Sets the maximum value for the Internal scale.

RMS AES17
Shows the level in accordance with AES17 (RMS +3 dB).

RMS Resolution
Sets the RMS resolution for the level display in milliseconds.

Smooth
Smoothes the display of the level curve.

NOTE

Alternatively, you can move the mouse pointer over the selected module and use the
mouse wheel to adjust this parameter.

Measurement Modules
The modules in this category allow you to measure the loudness and intelligibility of the audio
signal.

The following modules and module-specific settings are available:

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Leq(m)/Leq(a)
These modules show the average volume over time, using a filter that emphasizes the mid
and upper range frequencies. The Leq(m) measurement uses an M-weighting, according to
the Trailer Audio Standards Association (TASA). It is mainly used to ensure that a film trailer
respects the volume limits for trailers in movie theaters. The similar Leq(a) measurement uses an
A-weighting and is used to verify the conformity of the sound for broadcasting.

The following module-specific setting is available in the Module Settings window:

Ref. Level
Sets the reference level above which the Leq(m)/Leq(a) value turns red, indicating that
the volume limit is exceeded.

LFE
Sets the calibration of the low-frequency effects (LFE) channel during the
measurement. You can choose between +0 dB (default value) and +10 dB (value
according to ISO 21727) boost.

Loudness
This module shows the loudness of your audio in LU (Loudness Units) or LUFS (Loudness Units
referenced to Full Scale) according to EBU R 128.

TP (True Peak)
Shows the maximum true peak level in dB.

I (Integrated)
Shows the integrated loudness value. This is the average loudness value that is
measured over the whole audio range in LU or LUFS.

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S (Short-Term)
Shows the short-term loudness value that is measured every second on an audio block
of 3 seconds in LU or LUFS. This gives information about the loudest audio passages.

M Max. (Momentary Max.)

Shows the maximum value of all momentary loudness values that are measured every
100 ms in an audio range of 400 ms in LU or LUFS.

R (Range)
Shows the loudness range (LRA) that is measured over the whole audio range in LU.
The loudness range reports the ratio between the loudest and the quietest non-silent
sections. The audio is divided into small blocks. There is one audio block every second
and each block lasts 3 seconds, so that the analyzed blocks overlap. The top 10 % of the
quiet blocks and the top 5 % of the loud blocks are excluded from the final analysis. The
calculated loudness range is the ratio between the loudest and quietest remaining
audio blocks. This measurement helps you to decide how much compression or
expansion must be applied to the audio.
An asterisk (*) after a loudness range value indicates that less than 1 minute of audio
was analyzed.
Min. shows the minimum loudness range value in LU. Max. shows the maximum
loudness range value in LU.

NOTE

EBU R 128 does not recommend loudness range measurement for audio shorter than
1 minute due to too few data points.

PLR (Peak-to-Loudness)
Shows the peak-to-loudness ratio (PLR), also referred to as the crest factor, which is the
difference between the maximum true peak level value and the integrated loudness
value.

Time
Shows the overall duration of the loudness measurement.

The following module-specific settings are available in the Module Settings window:

Unit
Allows you to switch the meter scale between LUFS (absolute values) and LU (relative
values).

Scale
Allows you to set the meter to either EBU +9 scale (linear), EBU +18 scale (linear), or +23
scale (logarithmic).

Ref. Integrated
Sets a reference value for the integrated loudness. If higher values are detected, the
loudness meter indicates clipping.

Tol. Integrated
Sets a tolerance value for the integrated loudness.

Ref. True Peak

Sets a reference value for the true peak level. If higher values are detected, the
loudness meter indicates clipping.

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Tol. True Peak

Sets a tolerance value for the true peak level.

Ref. Short-Term
Sets a reference value for the short-term loudness. If higher values are detected, the
loudness meter indicates clipping.

Tol. Short-Term
Sets a tolerance value for the short-term loudness.

Ref. Momentary
Sets a reference value for the maximum momentary loudness. If higher values are
detected, the loudness meter indicates clipping.

Tol. Momentary
Sets a tolerance value for the maximum momentary loudness.

Ref. Range
Sets a reference value for the loudness range. If higher values are detected, the
loudness meter indicates clipping.

Tol. Range
Sets a tolerance value for the loudness range.

Loudness (Netflix)
This module provides a dialogue-gated loudness measurement that uses the Dolby Dialogue
Intelligence algorithm according to ITU-R BS.1770 and shows the loudness of your audio in LU
(Loudness Units) or LUFS (Loudness Units referenced to Full Scale). The Dialogue value allows
you to assess the percentage of speech-based sequences within the audio.

NOTE

When using the Loudness (Netflix) measurement, the Dolby Dialogue Intelligence algorithm
adds a latency of 2.048 seconds compared to the standard Loudness measurement.

TP (True Peak)
Shows the maximum true peak level in dB.

I (Integrated)
Shows the integrated loudness value. This is the average loudness value that is
measured over the whole audio range in LU or LUFS.

S (Short-Term)
Shows the short-term loudness value that is measured every second on an audio block
of 3 seconds in LU or LUFS. This gives information about the loudest audio passages.

M Max. (Momentary Max.)

Shows the maximum value of all momentary loudness values that are measured every
100 ms in an audio range of 400 ms in LU or LUFS.

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R (Range)
Shows the loudness range (LRA) that is measured over the whole audio range in LU.
The loudness range reports the ratio between the loudest and the quietest non-silent
sections. The audio is divided into small blocks. There is one audio block every second
and each block lasts 3 seconds, so that the analyzed blocks overlap. The top 10 % of the
quiet blocks and the top 5 % of the loud blocks are excluded from the final analysis. The
calculated loudness range is the ratio between the loudest and quietest remaining
audio blocks. This measurement helps you to decide how much compression or
expansion must be applied to the audio.
An asterisk (*) after a loudness range value indicates that less than 1 minute of audio
was analyzed.
Min. shows the minimum loudness range value in LU. Max. shows the maximum
loudness range value in LU.

NOTE

EBU R 128 does not recommend loudness range measurement for audio shorter than
1 minute due to too few data points.

Dlg. (Dialogue)
Shows the overall percentage of speech that is detected in the measured audio. The
speaker icon indicates that speech is detected at the current cursor position during
playback.

NOTE

If at least 15 % of speech is detected, a dialogue-gated measurement according to

ITU-R BS.1770-1 is used for visualization. If less speech is detected, a program-gated
measurement according to ITU-R BS.1770-3 is used.

Time
Shows the overall duration of the loudness measurement.

The following module-specific settings are available in the Module Settings window:

Unit
Allows you to switch the meter scale between LUFS (absolute values) and LU (relative
values).

Scale
Allows you to set the meter to either EBU +9 scale (linear), EBU +18 scale (linear), or +23
scale (logarithmic).

Ref. Integrated
Sets a reference value for the integrated loudness. If higher values are detected, the
loudness meter indicates clipping.

Tol. Integrated
Sets a tolerance value for the integrated loudness.

Ref. True Peak

Sets a reference value for the true peak level. If higher values are detected, the
loudness meter indicates clipping.

Tol. True Peak

Sets a tolerance value for the true peak level.

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Ref. Short-Term
Sets a reference value for the short-term loudness. If higher values are detected, the
loudness meter indicates clipping.

Tol. Short-Term
Sets a tolerance value for the short-term loudness.

Ref. Momentary
Sets a reference value for the maximum momentary loudness. If higher values are
detected, the loudness meter indicates clipping.

Tol. Momentary
Sets a tolerance value for the maximum momentary loudness.

Ref. Range
Sets a reference value for the loudness range. If higher values are detected, the
loudness meter indicates clipping.

Tol. Range
Sets a tolerance value for the loudness range.

Loudness Curve
This module shows loudness values as a time curve.

The following module-specific settings are available in the Module Settings window:

Duration
Sets the duration of the audio stream that is displayed.

NOTE

Alternatively, you can move the mouse pointer over the selected module, hold Ctrl/
Cmd , and use the mouse wheel to adjust this parameter.

Unit
Allows you to switch the meter scale between LUFS (absolute values) and LU (relative
values).

Scale
Allows you to set the meter to either EBU +9 scale (linear), EBU +18 scale (linear), or +23
scale (logarithmic).

Smooth
Smoothes the display of the loudness curve.

Momentary
Shows/Hides the maximum momentary loudness curve.

Short-Term
Shows/Hides the short-term loudness curve.

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Integrated
Shows/Hides the integrated loudness curve.

Range
Shows/Hides a gray area around the integrated loudness curve that represents the
loudness range.

Ref. Integrated
Sets a reference value for the integrated loudness.

Loudness Histogram
This module shows a histogram for the loudness or the loudness ratio value.

The following module-specific settings are available in the Module Settings window:

Unit
Allows you to switch the meter scale between LUFS (absolute values) and LU (relative
values).

Scale
Allows you to set the meter to either EBU +9 scale (linear), EBU +18 scale (linear), or +23
scale (logarithmic).

Meter Mode
Sets the displayed loudness or loudness ratio value.
● Momentary mode shows a histogram of the maximum value of all momentary
loudness values that are measured every 100 ms in an audio range of 400 ms.
● Short-Term shows a histogram of the short-term loudness value that is measured
every second on an audio block of 3 seconds.
● Integrated shows a histogram of the integrated loudness value.
● PLR shows a histogram of the peak-to-loudness ratio, also referred to as the crest
factor, which is the difference between the maximum true peak level value and
integrated loudness value.
● PSR shows a histogram of the peak-to-short-term-loudness ratio according to AES
Convention e-Brief 373.

Smooth
Smoothes the display of the loudness curve.

NOTE

Alternatively, you can move the mouse pointer over the selected module and use the
mouse wheel to adjust this parameter.

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Loudness Ratio
This module shows the peak-to-loudness ratio (PLR) and the peak-to-short-term-loudness ratio
(PSR) values according to AES specifications.

PSR
Shows the peak-to-short-term-loudness ratio according to AES Convention e-Brief 373.
The darker area of the meter indicates the minimum PSR value.

PLR
Shows the peak-to-loudness ratio, also referred to as the crest factor, which is the
difference between the maximum true peak level value and integrated loudness value.
The current PLR value is displayed numerically and also indicated by a thin bar on the
meter.

The following module-specific setting is available in the Module Settings window:

Ref. Level
Sets the reference level below which the PSR display turns red.

Time Smooth
Smoothes the temporal display of the PSR value.

NOTE

Alternatively, you can move the mouse pointer over the selected module, hold Alt , and
use the mouse wheel to adjust this parameter.

Spectral Domain Modules

The modules in this category allow you to visualize spectral information about the audio signal.

The following modules and module-specific settings are available:

Spectrum Curve
This module uses FFT (Fast Fourier Transform) techniques to show a frequency graph, providing
a precise and detailed real-time frequency analysis.

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The display shows the frequency spectrum as a linear graph. If you move the mouse pointer over
the display, a peak curve is shown in orange. Move the mouse pointer over the curves to show
the local maximum values in Hz. Press Ctrl/Cmd to show the maximum values in dB or press
Shift to show their pitch.

The following module-specific settings are available in the Module Settings window:

Time Smooth
Smoothes the temporal display.

NOTE

Alternatively, you can move the mouse pointer over the selected module, hold Alt , and
use the mouse wheel to adjust this parameter.

Peak Fallback
Sets the speed of release for the spectrum curve and the peak curve.

NOTE

● Alternatively, you can move the mouse pointer over the selected module, hold
Ctrl/Cmd , and use the mouse wheel to adjust this parameter.
● If this control is turned all the way to the left, the peak curve is disabled.

Freq. Smooth
Smoothes the frequency display of the spectrum curve.

NOTE

Alternatively, you can move the mouse pointer over the selected module and use the
mouse wheel to adjust this parameter.

FFT Window
Sets the block size of the window that is used for the analysis. If Multi is selected, 3
different block sizes are used at the same time.

Minimum
Sets the minimum value of the scale.

Maximum
Sets the maximum value of the scale.

Slope
Adds a slope to the frequency spectrum.

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Spectrum Bar
This module shows a graphical representation of the frequency spectrum, analyzed into separate
frequency bands, represented as vertical bars.

Move the mouse pointer over a bar to show the frequency range in Hz. Press Ctrl/Cmd to show
the current value in dB or press Shift to show its pitch range.

The following module-specific settings are available in the Module Settings window:

Time Smooth
Smoothes the temporal display.

NOTE

Alternatively, you can move the mouse pointer over the selected module, hold Alt , and
use the mouse wheel to adjust this parameter.

Peak Fallback
Sets the speed of release for the level meters and peak indicators.

NOTE

● Alternatively, you can move the mouse pointer over the selected module, hold
Ctrl/Cmd , and use the mouse wheel to adjust this parameter.
● If this control is turned all the way to the left, the peak indicators are disabled.

Threshold
Sets a threshold level below which the display is masked.

NOTE

Alternatively, you can move the mouse pointer over the selected module and use the
mouse wheel to adjust this parameter.

Bands/Oct.
Sets the number of bands per octave.

Minimum
Sets the minimum value of the scale.

Maximum
Sets the maximum value of the scale.

Slope
Adds a slope to the frequency spectrum.

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Spectrum Intensity
This module represents the frequency magnitude of the audio. The more intensely a bar is
colored, the higher the magnitude at this frequency.

The following module-specific settings are available in the Module Settings window:

Time Smooth
Smoothes the temporal display.

NOTE

Alternatively, you can move the mouse pointer over the selected module, hold Alt , and
use the mouse wheel to adjust this parameter.

FFT Window
Sets the block size of the window that is used for the analysis. If Multi is selected, 3
different block sizes are used at the same time.

Color
Allows you to choose a color scheme.

Minimum
Sets the minimum value of the scale.

Maximum
Sets the maximum value of the scale.

Slope
Adds a slope to the frequency spectrum.

Spectrum Keyboard
This module represents the frequency magnitude of the audio mapped to the keys of a piano
keyboard. The more intensely a key is colored, the higher the magnitude at its frequency.

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The following module-specific settings are available in the Module Settings window:

Time Smooth
Smoothes the temporal display.

NOTE

Alternatively, you can move the mouse pointer over the selected module, hold Alt , and
use the mouse wheel to adjust this parameter.

Color
Allows you to choose a color scheme.

Minimum
Sets the minimum value of the scale.

Maximum
Sets the maximum value of the scale.

Slope
Adds a slope to the frequency spectrum.

Spectrogram
This module shows the last seconds of the audio stream. This allows you to detect disturbances
in the spectrogram and to monitor the noise level and frequencies, for example.

NOTE

This module runs in Maximum Audio Performance mode.

The following module-specific settings are available in the Module Settings window:

FFT Window
Sets the block size of the window that is used for the analysis. This allows you to adjust
the trade-off between temporal resolution and frequency resolution. If you specify a
higher value, more frequencies are analyzed but they are located less accurately in the
time domain.

Duration
Sets the duration of the audio stream that is displayed.

NOTE

Alternatively, you can move the mouse pointer over the selected module, hold Ctrl/
Cmd , and use the mouse wheel to adjust this parameter.

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Color
Allows you to choose a color scheme.

Minimum
Sets the minimum value of the scale.

Maximum
Sets the maximum value of the scale.

NOTE

Alternatively, you can move the mouse pointer over the selected module and use the mouse
wheel to adjust the Minimum and Maximum parameters simultaneously.

Chromagram
This module displays a chromagram of your audio.

NOTE

This module runs in Maximum Audio Performance mode.

The following module-specific settings are available in the Module Settings window:

Duration
Sets the duration of the audio stream that is displayed.

NOTE

Alternatively, you can move the mouse pointer over the selected module, hold Ctrl/
Cmd , and use the mouse wheel to adjust this parameter.

Color
Allows you to choose a color scheme.

Minimum
Sets the minimum value of the scale.

Maximum
Sets the maximum value of the scale.

NOTE

Alternatively, you can move the mouse pointer over the selected module and use the mouse
wheel to adjust the Minimum and Maximum parameters simultaneously.

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Phase Modules
The modules in this category allow you to visualize the phase or directional relationship between
the channels of the audio signal.

The following modules and module-specific settings are available:

Phasescope
This module uses a vectorscope display to show the phase and amplitude relationship between
the left and right stereo channels. This provides you with directional information about a stereo
audio signal.

Hold Shift and move the mouse pointer over the display to measure the angle.

The following module-specific settings are available in the Module Settings window:

Zoom
Allows you to zoom on the graphical display.

NOTE

Alternatively, you can move the mouse pointer over the selected module and use the
mouse wheel to adjust this parameter.

Auto Zoom
If this button is activated, the zoom factor is adapted automatically.

Mode
Sets the display mode. The following modes are available: Lines, Dots, Envelope.

Peak Fallback
Sets the speed of release for the peak envelope in Envelope mode.

NOTE

If this control is turned all the way to the left, the peak envelope is disabled.

Scale
Activates/Deactivates the axis labeling.

NOTE

This option is only available if Auto Zoom is deactivated.

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Panorama
This module uses a polar coordinate display to show the phase and amplitude relationship
between the left and right stereo channels. This provides you with directional information about
a stereo audio signal.

The following module-specific settings are available in the Module Settings window:

Zoom
Allows you to zoom on the graphical display.

NOTE

Alternatively, you can move the mouse pointer over the selected module and use the
mouse wheel to adjust this parameter.

Auto Zoom
If this button is activated, the zoom factor is adapted automatically.

Mode
Sets the display mode. The following modes are available: Lines, Dots, Envelope.

Peak Fallback
Sets the speed of release for the peak envelope in Envelope mode.

NOTE

If this control is turned all the way to the left, the peak envelope is disabled.

Scale
Activates/Deactivates the axis labeling.

NOTE

This option is only available if Auto Zoom is deactivated.

Multipanorama
This module provides you with frequency-dependent directional information about a stereo
audio signal.

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The following module-specific settings are available in the Module Settings window:

Time Smooth
Sets the time for which an energy impulse is displayed.

Bands/Oct.
Sets the number of bands per octave.

Color
Allows you to choose a color scheme.

Correlation
This module visualizes the phase-correlation between the left and right channel. This allows you,
for example, to check the mono compatibility of a stereo recording.

The following module-specific setting is available in the Module Settings window:

Time Smooth
Smoothes the temporal display of the correlation.

NOTE

Alternatively, you can move the mouse pointer over the selected module, hold Alt , and
use the mouse wheel to adjust this parameter

Multicorrelation
This module visualizes the phase-correlation between the left and right channel for different
frequency bands.

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Move the mouse pointer over a bar to display its frequency range in Hz. Hold Ctrl/Cmd to
display its current value. Hold Shift to display its pitch range.

The following module-specific settings are available in the Module Settings window:

Time Smooth
Smoothes the temporal display of the correlation.

NOTE

Alternatively, you can move the mouse pointer over the selected module, hold Alt , and
use the mouse wheel to adjust this parameter

Bands/Oct.
Sets the number of bands per octave.

Balance
This module visualizes the balance between the left and right channel.

The following module-specific setting is available in the Module Settings window:

Time Smooth
Smoothes the temporal display of the correlation.

NOTE

Alternatively, you can move the mouse pointer over the selected module, hold Alt , and
use the mouse wheel to adjust this parameter

Spatial Domain Modules

The modules in this category allow you to visualize the spatial dimension of the audio signal.

The following modules and module-specific settings are available:

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Surround
This module visualizes the level and correlation of the different speakers in a surround speaker
setup.

If all channels have the same level, a perfect circle is shown in the center of the display.

NOTE

This module is only available for channel-based surround configurations. Speaker configurations
with top speakers and Ambisonics channels are not supported.

The following module-specific settings are available in the Module Settings window:

Zoom
Allows you to zoom on the graphical display.

NOTE

Alternatively, you can move the mouse pointer over the selected module and use the
mouse wheel to adjust this parameter.

Peak Fallback
Sets the speed of release for the peak envelope.

NOTE

● Alternatively, you can move the mouse pointer over the selected module, hold
Ctrl/Cmd , and use the mouse wheel to adjust this parameter.
● If this control is turned all the way to the left, the peak envelope is disabled.

Scale
Activates/Deactivates the axis labeling for the correlation display.

Ambisonics
This module allows you to visualize the energy distribution of an Ambisonics signal.

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A plane representation of the spherical Ambisonics sound field shows a grid of hexagons. The
color of a hexagon indicates the RMS level at this position. A filter allows you to smooth the
visualization.

NOTE

This module is only available for Ambisonics channel configurations.

The following module-specific settings are available in the Module Settings window:

Attack
Sets the attack time of the smoothing filter.

Release
Sets the release time of the smoothing filter.

Minimum
Sets the minimum value of the signal intensity scale.

Maximum
Sets the maximum value of the signal intensity scale.

Threshold
Sets the minimum signal level that is displayed. This value is indicated by the lower
triangle in the color legend on the right. If you change this value, the Fade Range
value is adjusted accordingly.

Fade Range
Sets the level at which a hexagon is displayed fully opaque. This value is indicated by
the upper triangle in the color legend on the right.

Color
Allows you to choose a color scheme.

Resolution
Sets the resolution of the grid.

Waveform Modules
The modules in this category allow you to visualize the waveform of the audio signal.

The following modules and module-specific settings are available:

Oscilloscope
This module shows a highly magnified view of the waveform.

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The following module-specific settings are available in the Module Settings window:

Zoom
Allows you to zoom on the graphical display by adjusting the amplitude.

NOTE

Alternatively, you can move the mouse pointer over the selected module, hold Alt/
Opt , and use the mouse wheel to adjust this parameter.

Frequency
Allows you to zoom on the graphical display by adjusting the frequency.

NOTE

Alternatively, you can move the mouse pointer over the selected module, hold Ctrl/
Cmd , and use the mouse wheel to adjust this parameter.

Trigger
Sets the channel that is used to synchronize the audio signal.

NOTE

Alternatively, you can click the waveform of the corresponding channel.

Scale
Activates/Deactivates the axis labeling.

NOTE

This option is only available if Auto Zoom is deactivated.

Phase
Allows you to shift the zero-crossing position.

Wavescope
This module displays the real-time waveform of the audio signal.

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Move the mouse pointer over a waveform position to display the corresponding project time.

The following module-specific settings are available in the Module Settings window:

Zoom
Allows you to zoom on the graphical display.

NOTE

Alternatively, you can move the mouse pointer over the selected module and use the
mouse wheel to adjust this parameter.

Duration
Sets the duration of the audio stream that is displayed.

NOTE

Alternatively, you can move the mouse pointer over the selected module, hold Ctrl/
Cmd , and use the mouse wheel to adjust this parameter.

Tempo Sync
If this button is activated, you can set Duration in beats.

NOTE

The equivalent duration is limited to a minimum of 0.5 s and a maximum of 30 s.

Scale
Activates/Deactivates the axis labeling.

NOTE

This option is only available if Auto Zoom is deactivated.

Station. Cursor
If this button is activated, the waveform moves continuously under the stationary
cursor. If this button is deactivated, the waveform is refreshed when the cursor moves
over it.

Wavecircle
This module displays the real-time waveform of the audio signal as a circle.

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Move the mouse pointer over a waveform position to display the corresponding project time.

The following module-specific settings are available in the Module Settings window:

Zoom
Allows you to zoom on the graphical display.

NOTE

Alternatively, you can move the mouse pointer over the selected module and use the
mouse wheel to adjust this parameter.

Duration
Sets the duration of the audio stream that is displayed.

NOTE

Alternatively, you can move the mouse pointer over the selected module, hold Ctrl/
Cmd , and use the mouse wheel to adjust this parameter.

Tempo Sync
If this button is activated, you can set Duration in beats.

NOTE

The equivalent duration is limited to a minimum of 0.5 s and a maximum of 30 s.

Reverse
Changes the rotation direction.

Station. Cursor
If this button is activated, the waveform moves continuously under the stationary
cursor. If this button is deactivated, the waveform is refreshed when the cursor moves
over it.

Other
This category provides a time display.

Time
This module shows the current time position of the project cursor.

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Delay

NOTE

If the project cursor is located outside the locator range, the color of the time display turns gray.

The following module-specific setting is available in the Module Settings window:

Mode
Allows you to select one of the following display modes: Time, Sample, Beats, or
Timecode.

Delay

MonoDelay
This is a mono delay effect. The delay line uses freely specified delay time settings.

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

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

Delay
Sets the delay time 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
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.

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Delay

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 control activates/deactivates the filter.

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

Delay
Sets the delay time 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
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.

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

Start Left/Start Right

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

StereoDelay
StereoDelay has two independent delay lines with freely specified delay time settings.

NOTE

This plug-in works only on stereo tracks.

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Delay

Feedback
Set the number of repeats for each delay.

Delay
Sets the delay time in milliseconds.

Mix
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.

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

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 control activates/deactivates the filter.

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Distortion

Distortion

Distortion
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.

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

Tone
Changes the tonal characteristic of the output signal.

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

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

Output
Sets the output level.

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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 High 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 filter.

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Quadrafuzz v2
Quadrafuzz v2 is a multi-band distortion and multi-effect plug-in for processing drums and
loops but also for treatment of vocals, for example. You can distort up to 4 bands. 5 different
distortion modes with several sub-modes are available.

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

● To define 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.

Global Settings
SB
Switches between multi band and single band mode.

Scenes
You can save up to 8 different settings. If the default setting of a scene is active, the
selected scene button lights up yellow.
If you change the default settings, the button lights up green, indicating that this scene
has customized settings.

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To copy the settings of a scene to another scene, select the scene that you want to
copy, click Copy, and click one of the numbered buttons.
You can automate the selection of scenes.

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

Output (-24 to 24 dB)

Sets the output level.

Band Settings
Mute Band

Mutes/Unmutes a frequency band.

Solo Band

Solos the corresponding frequency band.

Bypass Band

Bypasses a frequency band.

In/Out meter
Display the input and output level.

Gate
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.

Tape
This band mode simulates the saturation and compression of recording on analog tape
machines.

Drive
Controls the amount of tape saturation.

Tape Mode Dual

Simulates the use of two machines.

Tube
This band mode simulates the saturation effects using analog tubes.

Drive
Controls the amount of tube saturation.

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Tubes
Determine the number of tubes that are simulated.

Dist
This band mode adds distortion to your tracks.

Drive
Controls the amount of distortion.

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

Amp
This band mode simulates the sound of various types of guitar amps.

Drive
Controls the amount of amp overdrive.

Amp Types
You can select the following types of guitar amps:
● Amp Clean
● Amp Crunch
● Amp Lead

Dec
This band mode allows you to decimate and truncate the input audio signal to create a noisy,
distorted sound.

Decimator
Controls the resulting bit-resolution. The lower the resolution, the higher the distortion
effect.

Mode
Allows you to select one of the four operating modes. In each mode, the effect sounds
differently. Modes I and III are nastier and noisier, while modes II and IV are more
subtle.

S&H
Sets the amount by which the audio samples are decimated. At the highest setting,
nearly all of the information describing the original audio signal is eliminated, turning
the signal into unrecognizable noise.

Delay
To open the Delay section, click the Delay button.

Time
Sets the delay time.

Mode
If this option is activated, the delay signal is routed back into the distortion unit to
create a feedback with distortion.

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NOTE

High FBK values and low Duck values in combination with activated Mode can lead to
unwanted noise.

FBK
Determines the number of repeats for each delay.

Duck
Determines how much the delay signal ducks when an audio signal is present.

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

Sliders
Width
Sets the stereo width for the corresponding band.

Out
Sets the output gain for the corresponding band.

Pan
Sets the stereo position for the corresponding band.

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

Dynamics

Black Valve
Black Valve is a highly adaptive compressor with a vintage tube sound. It is based on a classic
analog device containing a tube pre-amp in conjunction with a classic compressor.

Drive
Controls the amount of tube saturation.

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Gain
Sets the input gain for the compressor section to adjust the gain after the tube stage,

Reduction
The LEDs show the gain reduction.

Peak Reduction
Combines the threshold and ratio parameters from common compressors. The higher
the value, the more compression occurs.

Dry/Wet
Adjusts the mix between the dry signal and the wet signal, preserving the transients of
the input signal. This allows for parallel compression.

Output
Sets the output gain.

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.

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

Release (3 to 1000 ms or Auto mode)

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 finds the
best release setting for the audio material.

Link Channels
If this option is activated, the plug-in uses the input from the channel with the highest
level. If the option is deactivated, each channel is analyzed separately.

Detect Intersample Clipping

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

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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.

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

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
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
Determines the level where the compressor kicks in.

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.

Soft Knee
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.

High Ratio
Sets the ratio to a fixed value of 20:1.

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

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

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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 film.

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

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

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

You can use it, for example, when close proximity microphone placement and equalizing lead to
situations where the overall sound is just right, but where unwanted sibilants occur.

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

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.

Filter
Lo/Hi
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.

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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.

Solo
Solos the frequency band. This helps you to find the appropriate position and width of
that band.

Diff
Plays back what DeEsser removed from the signal. This helps 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 file, set the threshold manually.

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

Freq (25 Hz to 20 kHz)

If Side-Chain is activated, this sets the frequency of the filter. 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 Side-Chain is activated, this sets the resonance or width of the filter.

Monitor
Allows you to monitor the filtered signal.

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

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.

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

You can use the controls or drag the breakpoints on 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.

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.

Output
Sets the output level.

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

You can either use the controls or drag the breakpoints on the graphical display to change the
Threshold and the Ratio parameter values.

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Threshold
Determines the level where the expansion kicks in.

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

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

Fall (0.1 to 100 ms)

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)

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

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

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Gate
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.

NOTE

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

Hold (0 to 2000 ms)

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

Release (10 to 1000 ms or Auto mode)

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

Threshold
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.

State LED
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 Peak mode works better for percussive material
with a lot of transient peaks.

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Range
Adjusts the attenuation of the gate when it is closed. If Range is set to minus infinite
, the gate is completely closed. The higher the value, the higher the level of the
signal that passes through.

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

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

Monitor
Allows you to monitor the filtered signal.

Center
If Side-Chain is activated, this sets the center frequency of the filter.

Q-Factor
If Side-Chain is activated, this sets the resonance or width of the filter.

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

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

Limiter
Limiter is designed to ensure that the output level never exceeds a set output level, to avoid
clipping in following devices.

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Limiter can adjust and optimize the Release parameter automatically according to the audio
material, or it can be set manually. Limiter 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 finds the best release setting for the audio
material.

Output
Sets the maximum output level.

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.

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Mix
Sets the level balance between the dry 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.

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

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. This allows you to set the
plug-in into ducking mode per 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.

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● To define 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 bands by ±15 dB after compression, use the
handles at the top of each frequency band.

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

Output
Sets the output level.

Bypass Band

Bypasses a frequency band.

Solo Band

Solos the corresponding frequency band.

Compressor Section
You can specify the Threshold and Ratio by moving breakpoints or using the corresponding
controls. The threshold is marked by the first breakpoint where the line deviates from the
straight diagonal.

Threshold
Determines the level where the compressor kicks in.

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)

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 finds the
best release setting for the audio material.

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.

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Frequency
If Side-Chain is activated, this sets the frequency of the side-chain filter.

Q-Factor
If Side-Chain is activated, this sets the resonance or width of the filter.

Side-Chain
Activates the internal side-chain filter. The side-chain signal can then be shaped
according to the filter parameter.

Monitor
Allows you to monitor the filtered signal.

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.

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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.

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

Output
Sets the output level.

Bypass Band

Bypasses a frequency band.

Solo Band

Solos the corresponding frequency band.

Shaper Section
You can specify the Attack, Length, and Release by moving breakpoints or using the
corresponding controls. 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.

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

Output
Sets the output level.

Bypass Band

Bypasses a frequency band.

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Solo Band

Solos the corresponding frequency band.

Expander Section
You can specify the Threshold and Ratio by moving breakpoints or using the corresponding
controls. The first breakpoint from which the line deviates from the straight diagonal is the
threshold point.

Threshold
Determines the level where the expansion kicks in.

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

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

Fall (0.1 to 100 ms)

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.

Rise (10 to 1000 ms or Auto mode)

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 finds the
best rise setting for the audio material.

Output
Sets the output level.

Side-Chain Section

Frequency
If Side-Chain is activated, this sets the frequency of the side-chain filter.

Q-Factor
If Side-Chain is activated, this sets the resonance or width of the filter.

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

Monitor
Allows you to monitor the filtered signal.

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Raiser
Raiser is a versatile limiter plug-in, that can increase the loudness of the audio material to a high
extent. Its range goes from smooth limiting of solo tracks and full mixes to a rather aggressive
limiting, best suited for percussive material.

Raiser can adjust and optimize the Release parameter automatically according to the audio
material, or you can set it manually. Raiser features separate meters for the input, output, and
the amount of limiting (middle meters).

Stationary Cursor
If this button is activated, the waveform moves continuously under the stationary
cursor. If this button is deactivated, the waveform is refreshed when the cursor moves
over it.

Gain
Sets the input gain.

Reduction
Shows the amount of gain reduction.

Release Mode
This pop-up menu sets the mode for the release time:

● Manual mode allows you to set the release time freely with the Time control.
● Auto mode automatically applies the best release setting for the audio material.
● Restricted Min. mode automatically applies the best release setting for the audio
material but allows you to define a minimum release time with the Release
control.
● Restricted Max. mode automatically applies the best release setting for the audio
material but allows you to define a maximum release time with the Release
control.

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● Aggressive mode applies aggressive limiting and preserves the transients of the
input signal without creating pumping effects.

Fast
If this option is activated, an initial fast release phase is added before the second phase
that is set by the Time parameter. We recommend using this option for percussive
material.

Release Time
Sets the time after which the gain returns to the original level when the signal drops
below the threshold. This parameter is only available if Manual, Restricted Min., or
Restricted Max. mode is selected.

Link Channels
If this option is activated, the plug-in uses the input from the channel with the highest
level. If the option is deactivated, each channel is analyzed separately.

Detect Intersample Clipping

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

Ceiling
Sets the maximum output level. If Detect Intersample Clipping is activated, the
maximum output level is the true peak level.

Compare
Sets the output level to the level of the dry input signal. This allows you to monitor the
limiting effect without the level boost.

Squasher
Squasher is a multi-band compressor that combines upward and downward compression,
allowing you to squash the audio signal. You can set up different parameters for up to three
frequency bands and use internal side-chaining to control the amount of compression for each
band.

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Main section
This section provides general settings for upward and downward compression for each
frequency band.

Frequency/Compressor display
Displays the frequency spectrum. It allows you to edit the band range, the output level,
the upward/downward ratio, and the upward/downward threshold for each band.
You can switch the display between a frequency spectrum view and a compressor
characteristics view by clicking the corresponding buttons left of the display:

Frequency spectrum

Compressor characteristics

You can edit the output level of a band or the cutoff frequency between two bands by
dragging the corresponding handle on the frequency spectrum display.

Show/Hide Full Frequency Band View

Shows/Hides the frequency spectrum/compressor display.

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Input meter
Shows the level of the overall input signal.

Input
Sets the overall input level.

Bands
Sets the number of frequency bands.

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

Parameter Link
Links the parameters of the same type in all bands. 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 of one band, the
corresponding parameter values of the other bands are set to the same value.
● If Relative Mode is activated and you edit a parameter value of one band, the
relation of the corresponding parameter values of the other bands remains the
same.

Activate/Deactivate Band
Activates/Deactivates the corresponding frequency band.

Solo Band
Solos the corresponding frequency band.

Up Ratio/Down Ratio
Set the amount of upward and downward compression. These parameters allow you to
adjust the squashing effect.

In
Shows the input level for the corresponding frequency band.

Up Threshold/Down Threshold
The left triangle sets the threshold for upward compression. If the input signal is below
this threshold, the upward compressor raises the level according to the Up Ratio value.
The right triangle sets the threshold for downward compression. If the input signal
is above this threshold, the downward compressor lowers the level according to the
Down Ratio value.
You can drag the darker area between both handles to adjust Up Threshold and Down
Threshold simultaneously.

Output meter
Shows the level of the overall output signal.

Output
Sets the overall output level.

Squash Parameter section

This section provides additional compression and filter settings for each frequency band.

Show/Hide Squash Parameter Section

Shows/Hides the squash parameter section.

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Att.
Sets the compression attack time for both the upward and downward compressor.

Rel.
Sets the compression release time for both the upward and downward compressor.

Drive
Sets the amount of saturation. This parameter adds harmonics to the output signal.

Gate
Sets the threshold for the internal gating effect. Signal levels above this threshold
trigger the gate to open. Signal levels below this threshold close the gate.

NOTE

You can control this parameter via side-chaining.

Mix
Adjusts the mix between dry signal and wet signal for the corresponding band.

Output
Sets the output level for the corresponding band.

NOTE

Alternatively, you can edit this parameter by dragging the corresponding handle on the
frequency display.

Side-Chain section
This section provides settings for internal side-chaining for each frequency band. It is only
available if the parameter section is shown.

Show/Hide Side-Chain Section

Shows/Hides the side-chaining section.

Activate/Deactivate Side-Chaining for Band

Activates/Deactivates side-chaining for the corresponding band.

Side-Chain Filter Listen

Lets you monitor the side-chain signal and the applied filter.

Freq
Sets the frequency of the side-chain filter.

Q
Sets the Q factor of the side-chain filter.

Send to
This pop-up menu allows you to send the side-chain signal to the compressor section
(Squasher) or to the internal gate.

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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 filter the trigger signal.

VU Meter
Shows the amount of gain reduction.

In/Out Meters
Show the highest peaks of all available input and output channels.

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

Drive
Controls the amount of tube saturation.

Output
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 finds the best release setting for the audio
material.

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

Ratio
Toggles between a low and a high ratio value.

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

Side-chain section
Filter Type (Low-Pass/Band-Pass/High-Pass)
If Side-Chain is activated, these buttons allow you to set the filter type to low-pass,
band-pass, or high-pass.

Center
If Side-Chain is activated, this sets the center frequency of the filter.

Q-Factor
If Side-Chain is activated, this sets the resonance or width of the filter.

Monitor
Allows you to monitor the filtered 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 Punch mode which preserves the attack phase of the signal and a program-
dependent Auto Release function.

VU Meter
Shows the amount of gain reduction.

In/Out Meters
Show the highest peaks of all available input and output channels.

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

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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.

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 finds the best release setting for the audio
material.

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

Output (-48 to 24 dB)

Sets the output gain.

VoxComp
VoxComp is a compressor designed for processing vocals. It uses a highly adaptive algorithm,
which makes it easy to achieve the desired results.

Meter section
Shows the input, output, and gain reduction levels.

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

Dry/Wet
Adjusts the mix between the dry signal and the wet signal, preserving the transients of
the input signal. This allows for parallel compression.

Threshold
Determines the level where the compressor kicks in.

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Output
Sets the output gain.

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

The window is divided into three sections containing controls and meters for each effect. Activate
the individual effects using the Gate, Compressor, and Limiter buttons. You can select between
three different routing options using the Module Configurator button.

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 filtered using an internal side-chain signal.

The following parameters are available:

Input meter
Shows the level of the input 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.

Threshold
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.

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State LED
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).

Release (10 to 1000 ms or Auto mode)

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

Hold (0 to 2000 ms)

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 closed. If Range is set to minus infinite
, the gate is completely closed. The higher the value, the higher the level of the
signal that passes through.

Side-Chain
Activates the internal side-chain filter. The input signal can then be shaped according
to the filter 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 filter type to low-pass,
band-pass, or high-pass.

Center
If Side-Chain is activated, this sets the center frequency of the filter.

Q-Factor
If Side-Chain is activated, this sets the resonance or width of the filter.

Monitor
Allows you to monitor the filtered 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.

Gain Reduction meter

Shows the amount of gain reduction.

Threshold
Determines the level where the compressor kicks in.

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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.

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

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

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 (attack) passes through
unprocessed.

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 finds the
best release setting for the audio material.

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 finds the best release setting for the audio
material.

Output section
Output meter
Shows the level of the output signal.

Module Configurator
Changes the signal flow through the three effects. Changing the order of the effects
can produce different results, and the available routing configurations allow you to
quickly compare what works best for a given situation. Click Module Configurator to
toggle between the following routing configurations:

● G-C-L (Gate-Compressor-Limiter)
● C-L-G (Compressor-Limiter-Gate)

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EQ

● C-G-L (Compressor-Gate-Limiter)

EQ

CurveEQ
Voxengo CurveEQ is a spline equalizer for professional music and audio production applications.
CurveEQ shows the filter 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 filters of CurveEQ
can be 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.

EQ-M5
EQ-M5 is a characterful vintage tube EQ, especially suited for mid processing. It offers low, mid,
and high peak filters.

Low Frequency
Sets the frequency of the Low Boost filter.

Low Boost
Sets the boost of the low peak filter.

Mid Frequency
Sets the frequency of the Mid Attenuate filter.

Mid Attenuate
Sets the attenuation of the mid peak filter.

High Frequency
Sets the frequency of the High Boost filter.

High Boost
Sets the boost of the high peak filter.

Output
Sets the output gain.

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EQ-P1A
EQ-P1A is unique sounding vintage tube EQ with a famous low frequency boost and attenuation,
based on a classic analog equalizer. It offers low and high shelving filters and an additional high
peak filter.

Low Boost
Sets the boost of the low shelving filter.

Low Attenuate
Sets the attenuation of the low shelving filter.

Low Boost/Attenuate Frequency

Sets the frequency of the Low Boost and Low Attenuate filters.

High Boost Bandwidth

Sets the bandwidth of the High Boost filter.

High Boost
Sets the boost of the high peak filter.

High Boost Frequency

Sets the frequency of the High Boost filter.

High Attenuate
Sets the attenuation of the high shelving filter.

High Attenuate Frequency

Sets the frequency of the High Attenuate filter.

Output
Sets the output gain.

Frequency 2
Frequency 2 is a high-quality equalizer with 8 fully parametric bands. The bands can act as
shelving filter, as peak or notch filter, or as low-pass/high-pass cut filter.

The plug-in supports separate internal or external side-chaining for each band. For Low Shelf,
High Shelf, and Peak filters, dynamic filtering lets you determine when and how the EQ is
applied, depending on the dynamics of the audio material.

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Main Section
Reset

Alt/Opt -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.

Display Gain Range

Allows you to adjust the gain range that is shown in the spectral display.

Keys
Shows/Hides the keyboard below the graphical editor.
On the keyboard, color indicators reflect 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.

View
Toggles between single-band and multi-band view. Single-band view provides
additional parameters for each band.

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NOTE

To toggle between single-band and multi-band view, you can also double-click on the
top of a band section.

Output
Adjusts the overall output level.

Output meter
Shows the level of the overall output signal.

Band Settings

Multi-band view

Single-band view

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.
● You can modify the parameters of deactivated bands.

Switch Processing buttons

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.

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IMPORTANT

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

NOTE

This setting is only available for stereo tracks.

Linear Phase Processing

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.
Activating this option deactivates dynamic filtering for the corresponding band.

NOTE

● Linear phase mode leads to an increase in latency.

● In rare cases, for example, when using low-cut filtering with a high slope for bass
signals, you may hear an unwanted pre-ringing effect.

Filter Type
You can choose between the filter 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
specified amount.
● Peak boosts or attenuates frequencies at the set frequency value with a bell-
shaped filter.
● High Shelf boosts or attenuates frequencies above the cutoff frequency by the
specified amount.
● Notch boosts or attenuates frequencies at the set frequency value with a very
narrow filter.
● 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
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, A5 -23 or C4 +49.

NOTE

● You can adjust the Freq parameter of a band in the graphical editor by Alt/Opt -
clicking the corresponding handle and moving the mouse left and right.
● To ensure that the cent offsets are taken into account, enter a space between the
note and the cent offset.

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

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Q
For Peak and Notch filters, this parameter controls the width of the band. For Low
Shelf and High Shelf filters, 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 to the handle and turn the mouse wheel.
● This parameter is not available for Cut filters.

Gain
Sets the amount of attenuation/boost for the corresponding band. If Dynamic
Filtering is activated, this is also the target gain value.

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 filters.

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

Show Dynamics Parameters

Shows/Hides parameters for dynamic filtering in multi-band view.

NOTE

This setting is only available for the filter types Low Shelf, Peak, High Shelf, and
Notch.

Activate/Deactivate Dynamic Filtering

Activates/Deactivates dynamic filtering for the corresponding band. If this option is
activated, the filtering of the band depends on the dynamics of the input signal.
Activating this setting deactivates linear phase mode for the corresponding band.

NOTE

This setting is only available for the filter types Low Shelf, Peak, High Shelf, and
Notch.

Threshold
Determines the threshold level. Only signal levels above the threshold are dynamically
filtered.

Input meter
Shows the level of the input signal.

NOTE

In multi-band view, the input level meter is combined with the threshold handle.

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Start
Allows you to adjust the starting point for the gain. Dynamic filtering takes place from
this point to the set Gain of the EQ band.

NOTE

This setting is only available in single-band view.

Ratio
The higher the level of the input signal is above the threshold, the more filtering
occurs. Low ratio values mean that the filter starts to boost or attenuate smoothly
above the threshold. High ratio values mean that the target gain is reached almost
immediately.

Attack
Determines how fast the 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.

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

Side-Chain
Activates/Deactivates internal side-chaining for the corresponding band. This allows
you to shape the input signal according to the side-chain filter parameters.

NOTE

● This setting is only available in single-band view.

● Side-chaining is only available if dynamic filtering is activated.
● If side-chaining is activated, this is indicated by showing SC on the top of a band
section.

Side-Chain Filter Auto

Activates/Deactivates automatic filtering of the side-chain signal. If this parameter is
activated, the SC Freq and SC Q parameters are deactivated. Instead, the Freq and Q
values of the corresponding band are used.

NOTE

This setting is only available in single-band view.

Side-Chain Filter Listen

Allows you to solo the side-chain filter. This way, you can quickly check the part of the
signal that is filtered out using the current settings.

NOTE

This setting is only available in single-band view.

SC Freq
Sets the frequency of the side-chain filter for 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.

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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.
● This setting is only available in single-band view.

SC Q
Sets the resonance or width of the side-chain filter for the corresponding band.

NOTE

This setting is only available in single-band view.

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

Spectrum Display
Show Spectrum
Activates/Deactivates the spectrum display.

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

Smooth
Determines the reaction time of the spectrum display. Lower values result in faster
reaction times and smoother transitions.

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

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

Slope
Tilts the spectrum display around the 1 kHz pivot.

EQ Curve
Show Curve
Shows/Hides the EQ curve on the spectrum display.

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

GEQ-10/GEQ-30

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GEQ-30

GEQ-30

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

You can draw response curves on 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 filter 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.

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True Response
Applies serial filters 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 filter structure where the response does not follow the set
gain values accurately.

VariableQ
Applies parallel filters where the resonance depends on the amount of gain.

ConstQ asym
Applies parallel filters where the resonance is raised when boosting the gain and vice
versa.

ConstQ sym
Applies parallel filters where the resonance of the first and last bands depends on the
sample rate.

Resonant
Applies serial filters where a gain increase of one band lowers the gain in adjacent
bands.

StudioEQ
Studio EQ is a high-quality 4-band parametric stereo equalizer. All four bands can act as fully
parametric peak filters. In addition, the low and high bands can act as either shelving filters
(three types) or as cut filters (low-pass/high-pass).

Main Layout
Reset
Alt/Opt -click this button to reset all parameter values.

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Show Input/Output Spectrum

Shows the spectrum before and after filtering.

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.

Band Settings

Activate/Deactivate Band
Activates/Deactivates the corresponding band.

NOTE

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

Freq
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, 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.
● To ensure that the cent offsets are taken into account, enter a space between the
note and the cent offset.

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

Q
For Peak filters, this parameter controls the width of the band. For Shelf filters, it adds
a drop or a boost, depending on the gain setting of the band. For Cut filters, 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 to the handle and turn the mouse wheel.

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

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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 filters.

Filter type
For the low and high band, you can choose between three types of shelving filters, a
peak filter, and a cut filter. If Cut mode is selected, the Gain parameter is fixed.
● Shelf I adds resonance in the opposite gain direction slightly above the set
frequency.
● Shelf II adds resonance in the gain direction at the set frequency.
● Shelf III is a combination of Shelf I and II.

Filter

DualFilter
DualFilter filters out specific frequencies while allowing others to pass through.

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

Resonance
Sets the sound characteristic of the filter. With higher values, a ringing sound is heard.

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

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

Graphical display
Visualizes the settings for all parameters.

Level meter
Shows the output level, giving you an indication of how the filtering affects the overall
level of the edited audio.

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

Allows you to eliminate low-frequency noise. The filter 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.

Low-Cut Slope
Allows you to choose a slope value for the low-cut filter.

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Filter

Low-Cut Preview
Use this button between the Low-Cut controls and the graphical display to switch the
filter to a complementary high-cut filter. This deactivates any other filters, allowing you
to listen only to the frequencies that you want to filter out.

Notch Freq
Sets the frequency of the notch filter. 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 filter out.

Notch Gain Invert

This button inverts the gain value of the notch filter. Use this button to filter out
unwanted noise. When looking for the frequency to omit, it sometimes helps to boost
it first (set the notch filter to positive gain). After you have found the frequency of the
noise, you can use the Invert button to cancel it out.

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

Notch Preview
Use this button between the notch filter controls and the graphical display to create
a band-pass filter with the peak filter's frequency and Q. This deactivates any other
filters, allowing you to listen only to the frequencies you want to filter out.

Notches buttons (1, 2, 4, 8)

These buttons add additional notch filters to filter out harmonics.

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

This high-cut filter allows you to remove high-frequency noise. The filter is inactive if
the curve point is located all the way to the right. 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.

High-Cut Slope
Allows you to choose a slope value for the high-cut filter.

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

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Mastering

Internal Dithering
This is a WaveLab-specific 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.

NOTE

Internal Dithering is only available on the second slot of the Final Effects / Dithering pane.

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 final 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 file will still be 24 bit in size, although only 16 bits of
information will have significance. When rendering to a 16-bit file, specify the file
resolution to avoid wasting space.

Leveler
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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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.

Lin Pro Dither

Lin Pro Dither is a dithering plug-in that uses advanced algorithms and offers additional noise
shaping to increase the apparent signal-to-noise ratio by altering the spectrum of the low-level
audio signal. Different noise shaping types allow you to adapt the dithering effect to the audio
material to be processed.

Output Bit Depth

Sets the bit depth of the output signal.

NOTE

Dithering changes the bit depth, but not the sample size. For example, when dithering
24 bit to 16 bit, the file is still 24 bit in size, although only 16 bits of information
have significance. When processing a 16-bit file, specify the Output Bit Depth value
accordingly to avoid the creation of unnecessarily large files.

Dither Control
If Auto Blanking is activated, the dither noise is gated during silent passages.

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Dither Type
Allows you to choose between Triangular and Triangular Hi-Pass dither types.

● If None is selected, no dithering is applied but the signal is still rendered to the set
Output Bit Depth.

Noise Shaping
Activates/Deactivates noise shaping. Noise shaping increases the apparent signal-to-
noise ratio. The pop-up menu allows you to choose between different algorithms which
move the noise to different frequency ranges.

NOTE

Noise shaping is not available if None is selected as Dither Type.

MasterRig
MasterRig allows you to master audio material in an intuitive and creative way. It offers high-
class sound quality, accuracy, flexibility, 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 configurations as scenes. This allows you to compare different
parameter settings and module combinations.

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● 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 define 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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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: Absolute and Relative.

● If Absolute Mode is activated and you edit a parameter value in one band, the
corresponding parameter values in the other bands are set to the same value.
● If Relative Mode is activated and you edit a parameter value in one band, the
corresponding parameter values in the other bands keep their relation.

Auto Listen for Filters

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

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.

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

SC FREQ
Sets the frequency of the side-chain filter.

Auto (Dynamic EQ only)

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

Listen
Allows you to solo the side-chain filter.

SC Q
Sets the resonance or width of the filter.

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
module.
● 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 chain.

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

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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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 filled.

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 prevents clipping by ensuring that the output level does not exceed a set
value.

Band Settings

On/Off
Activates/Deactivates the corresponding section.

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Balance

Mid/Side
Allows you to set the gain for the mid and side signal.

Solo Mid Signal/Solo Side Signal

Allows you to solo the mid or side signal.

Transients
The Transients section allows you to set the following parameters:

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

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

Gain
Sets the output level for the corresponding band.

Harmonics
With the Harmonics section 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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2nd HARM/3rd HARM

Allows 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
Thanks 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; that is, the signal drops
below the threshold. With Auto Release activated, the plug-in automatically identifies
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 to analog signals.

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

Output
Sets the output level.

Maximizer
Maximizer raises the loudness of audio material without any risk of clipping. The limiting
amount is displayed between the input and the output meter.

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Maximizer Type set to Classic Maximizer Type set to Modern

Type
Classic provides the algorithms of previous versions of this module and is suited for all
styles of music.
Modern is particularly suited for contemporary styles of music. The algorithm allows
for more loudness than Classic. Modern provides additional settings to control the
release phase:
Release sets the overall release time.
Recover allows for faster signal recovering at the beginning of the release phase.

Optimize
Determines the loudness of the signal.

Output
Sets the output level.

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

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.

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Band Settings

On/Off
Activates/Deactivates the corresponding section.

Solo Band

Solos the corresponding frequency band.

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.

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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 finds 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 and the wet signal.

Compressor curve display

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

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.

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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 finds the best release setting for the audio.

Drive
Controls the amount of tube saturation.

Mix
Sets the level balance between the dry 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.

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 finds the best release setting for the audio
material.

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

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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 and the wet signal.

Output
Sets the output gain.

Side-Chain
Opens the Side-Chain settings.

Maximizer

Optimize
Determines the loudness of the signal.

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

Output
Sets the output gain.

RELATED LINKS
Side-Chain Settings on page 87

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 filter, as peak
filter (band-pass), or as cut filter (low-pass/high-pass, band 1 and 8 only).

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

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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.

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 filtering with a high slope for bass
signals, also an unwanted pre-ringing effect may be audible.

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Equalizer Section

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
specified amount.
● High Shelf boosts or attenuates frequencies above the cutoff frequency by the
specified amount.
● Peak boosts or attenuates frequencies at the set frequency value with a bell
shaped filter.
● Notch boosts or attenuates frequencies at the set frequency value with a very
narrow filter.
● 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.

Q
Controls the width of the corresponding band.

Gain (-15 to +15 dB)

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

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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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
specified amount.

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● Peak boosts or attenuates frequencies at the set frequency value with a bell
shaped filter.
● High Shelf boosts or attenuates frequencies above the cutoff frequency by the
specified amount.

FREQ (20 to 20000 Hz)

Sets the frequency of the corresponding band.

Q
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.

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

Side-Chain
Opens the Side-Chain settings.

RELATED LINKS
Side-Chain Settings on page 87

Saturator
The Saturator module allows you to simulate the sound of analog tubes, and the saturation and
compression effect when recording on analog tape machines.

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

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Band Settings

On/Off
Activates/Deactivates the corresponding section.

Solo Band

Solos the corresponding frequency band.

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.

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Saturator Section

Tape/Tube
Allows you to switch between tube saturation and tape saturation.
● Tube saturation simulates the saturation of analog tube compressors.
● Tape saturation simulates the saturation and compression effect of analog tape
machine recordings.

Drive
Controls the amount of saturation.

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

Output
Sets the output gain.

Imager
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 defined frequency domains.

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Band Settings

On/Off
Activates/Deactivates the corresponding section.

Solo Band

Solos the corresponding frequency band.

Add/Remove Band
Allow you to add and remove bands.

Imager Section

Width
Allows you to control the stereo width per band.

Pan
Allows you to pan the signal left/right.

Output
Sets the output level for each band.

Peak Master
This is a basic plug-in that minimizes peaks in your audio file, 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.

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Input Gain
Values range from -12 dB to 24 dB.

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.

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.

NOTE

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

Output Sample Rate

Defines the output sample rate while the input sample rate is determined by the
sample rate of the active audio file or audio montage.

Quality
Defines 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.

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Modulation

Modulation

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.

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.

Waveform preset buttons

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.

Phase
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.

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

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Modulation

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 deflection 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.

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.

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
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.

Rate
Sets the sweep rate.

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

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

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Modulation

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
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.

Rate
Sets the sweep rate.

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

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

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Pitch Shift

Pitch Shift

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
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.

Octave 1
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.

Restoration

DeReverb
DeReverb allows you to remove reverb from the input signal in real time. The plug-in uses an
impulse response that is generated by a physical model based on adjustable parameters and a
learning function.

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Restoration

When you activate the learning function, the plug-in records a short range of the input signal and
automatically adjusts the parameters of the physical model.

Frequency display
Sets the spectral envelope of the impulse response. You can edit the four bands of
the spectral envelope range and the output level for each band. You can edit the level
of a band or the cutoff frequency between two bands by dragging the corresponding
handle.

Algorithm
Sets the restoration algorithm. We recommend to try different algorithms and choose
the algorithm that provides the best results.

Play Mode
Sets the playback mode:
● Input plays back the wet input signal.
● Reverb plays back only the removed reverb portion.
● Output plays back the processed output signal.

Reduction
Sets the amount of reverb that is removed from the wet input signal.

IR Physical Model section

Duration
Sets the duration of the impulse response.

Pre-Delay
Sets a pre-delay for the impulse response.

Learn
Activates the learning function.

Play IR
Allows you to playback the generated impulse response.

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Restoration

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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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

Filled Curve
Allows you to fill 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.

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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● 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.

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
DeNoiser allows you to remove noise from the audio material.

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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 define the stationary noise.

Noise
The Noise options allow you to define a section in an audio file that contains a static
noise that you want to remove. When you then render the audio file, 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 and
click Learn.
RestoreRig records the audio for a few seconds to detect the static noise.
2. Use the Static Level dial to set the level.
3. To remove the recorded static noise in the audio file, render the audio file.
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 sufficient 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
DeBuzzer allows you to remove harmonic noise with a fundamental frequency that should be
around 50 to 60 Hz.

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Activate/Deactivate DeBuzzer
Activates/Deactivates the module.

Noise Listening Mode

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

Level
Allows you to define the reduction of the noise in dB.

Sensitivity
Allows you to define how sensitive the reduction will be to the current audio level. At
0 %, DeBuzzer reduces the current harmonic noise with the Level value. At higher
sensitivity values, the level is dynamically defined 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 define 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

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

Reverb

REVelation
REVelation produces a high-quality algorithmic reverb with early reflections and reverb tail.

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Reverb

The early reflections are responsible for the spatial impression in the first milliseconds of
the reverb. For emulating different rooms, you can choose between different early reflections
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 first reflections to reach
the listener.

Early Reflections
Here, you select an early reflections pattern. The early reflections pattern contains the
most important delays that deliver the key information for the spatial impression of the
room.

ER/Tail Mix
Sets the level balance between the early reflections and the reverb tail. At a setting of
50 %, early reflections and tail have the same volume. Settings below 50 % raise the
early reflections and lower the tail, as a result the sound source moves towards the
front of the room. Settings above 50 % raise the tail and lower the early reflections, as a
result the sound source moves towards the back of the room.

Size
Adjusts the length of the early reflections 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 reflections pattern is compressed and the room is perceived
smaller.

Low Cut
Attenuates the low frequencies of the early reflections. The higher this value, the less
low frequencies are present in the early reflections.

High Cut
Attenuates the high frequencies of the early reflections. The lower this value, the less
high frequencies the early reflections will have.

Delay
Delays the onset of the reverb tail.

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 infinitely 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.

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Reverb

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 Low Time parameter.

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.

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

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

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.

Mix
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.

Lock Mix Value

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
Specifies the frequency of the pitch modulation.

Modulation Depth
Sets the intensity of the pitch modulation.

Modulation Activate
Activates/Deactivates the chorusing effect.

RoomWorks
RoomWorks is a highly adjustable reverb plug-in for creating realistic room ambience and reverb
effects in stereo and surround formats. The CPU usage is adjustable to fit the needs of any

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system. From short room reflections to cavern-sized reverb, this plug-in delivers high quality
reverberation.

Input Filters
Low Frequency
Determines the frequency at which the low-shelving filter takes effect. Both the high
and low settings filter the input signal prior to reverb processing.

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

Low Gain
Sets the amount of boost or attenuation for the low-shelving filter.

High Gain
Sets the amount of boost or attenuation for the high-shelving filter.

Reverb Character
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 first reflections to reach
the listener.

Size
Alters the delay times of the early reflections to simulate larger or smaller spaces.

Reverb Time
Sets 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 reflection patterns. This is helpful if some sounds are causing odd ringing or

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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 infinite 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.

High Level
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.

Envelope
Amount
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.

Attack
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.

Release
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.

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Efficiency
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 Efficiency settings (>90 %).

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 efficiency setting
to achieve a specific 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.

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 first reflections to reach
the listener.

Reverb Time
Sets 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.

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.

High Level
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.

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Spatial

Mix
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 %.

Spatial

VST AmbiDecoder
VST AmbiDecoder allows you to convert Ambisonics audio for playback on headphones or
multi-channel speaker setups.

For a description of VST AmbiDecoder, see the Operation Manual.

MixConvert V6
The MixConvert V6 plug-in can be used to quickly convert a multi-channel mix to a format with a
different channel configuration, for example, to mix down a 5.1 surround mix into a stereo mix.

For a description of MixConvert V6, see the Operation Manual.

Imager
Imager allows you to expand or reduce the stereo width of your audio in up to 4 bands. This
way, you can independently adjust the stereo image in defined frequency domains.

Bands
Sets the number of frequency bands.

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Live
If this button is activated, a more analog-style filter bank is used. This mode introduces
no latency and is better suited for live performances. If this button is deactivated,
a more neutral-sounding linear phase filter bank is used, at the cost of introducing
latency.

Frequency display
Shows a spectrum display and allows you to edit the band range and the output level
for each band.
You can edit the output level of a band or the cutoff frequency between two bands by
dragging the corresponding handle.

Output meter
Shows the level of the overall output signal.

Activate/Deactivate Band
Activates/Deactivates the corresponding frequency band.

Solo Band

Solos the corresponding frequency band.

Phase display
A phase scope for each band indicates the phase and amplitude relationship between
the stereo channels. The phase scope works as follows:
● A vertical line indicates a perfect mono signal (the left and right channels are the
same).
● A horizontal line indicates that the left channel is the same as the right, but with an
inverse phase.
● A fairly round shape indicates a well balanced stereo signal. If the shape leans to
one side, there is more energy in the corresponding channel.
● A perfect circle indicates a sine wave on one channel, and the same sine wave
shifted by 45° on the other.
Generally, the more visible a thread, the more bass is in the signal. The more spray-like
the display, the more high frequencies are in the signal.
The phase correlation meters below work as follows:
● The vertical bar shows the current phase correlation.
● With a mono signal, the meter shows +1, indicating that both channels are
perfectly in phase.
● If the meter shows -1, the two channels are the same, but one is inverted.

Show/Hide Phase Scope

Shows/Hides the phase scopes and phase correlation meters for all bands.

Width
Sets the stereo width for the corresponding band.

Pan
Sets the left-right panning for the corresponding band.

Output
Sets the output level for the corresponding band.

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Spatial

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

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

Mono
Sets the output to mono. This allows you to check for unwanted coloring of the sound
which can occur when creating an artificial stereo image.

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

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

NOTE

This plug-in works only on stereo tracks.

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Surround

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

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 unwanted coloring of the sound, which can
occur when enhancing the stereo image.

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

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.

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.

Surround

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 Master Section and if a
surround audio montage is active.

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Surround

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.

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 Master Section and if a
8 channel audio montage is active.

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Tools

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.

Tools

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.

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Tools

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.

SampleAlign
The SampleAlign plug-in allows you to delay one audio channel in a stereo file, relative to the
other channel; that is, to modify the samples of either one of the channels or both the left and
the right audio channel.

The resulting shift is applied to the entire waveform, with all frequencies shifted by the same
value.

The SampleAlign plug-in serves the following purposes:

● To correct misalignments: If your audio sounds weak or hollow on playback, this can be the
consequence of positioning microphones at varying distances from the sound source during
recording: The spatial discrepancy results in time delays between the microphones, which, in
turn, lead to sample issues in the mix, with the delayed channel playing back a number of
silent samples to compensate for the delay. Shifting the samples of one channel enables you
to correct this misalignment and to improve the sound quality.
● To create special sound effects: In audio production, sample shifting is often used for sound
design, such as the creation of “swirling” or “spinning” sounds, which you can use in music or
in movies and video games.
● To separate microphone channels: You can use the plug-in to emulate stereo microphone
channel separation.
● To improve stereo imaging: You can use sample shifting to manipulate the stereo image.
For example, by shifting the samples of one channel, you can create the illusion of a sound
source being farther to the left or to the right than it actually is. This can be useful in music
production, where creating a wide and engaging stereo image is often an expressed goal.
● To conduct psycho-acoustic experiments: Sample shifts are used in binaural beats, where
two sine waves with slightly different frequencies are presented to each ear separately.
The brain processes the difference in frequencies, which can have various effects on
consciousness, from relaxation to focus stimulation.

Left/Mid Delay
Allows you to modify the delay for the left or mid channel.
The farther you move the slider to the right, the more samples are shifted.
You can invert the phase by activating Invert Phase below the slider.

Right/Side Delay
Allows you to modify the delay for the right or side channel.
The farther you move the slider to the right, the more samples are shifted.

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Tools

You can invert the phase by activating Invert Phase below the slider.

Mid/Side
Activating this option allows you to modify the phase of the mid and side channel.

NOTE

The plug-in is mainly designed to perform micro-shifts, which is why the editing unit is set to
samples rather than milliseconds.

The tool is primarily designed for stereo files. When it is applied to mono files, it only takes effect
on the delay and phase inversion parameters of the left channel.

IMPORTANT

While sample shifting can be useful, we recommend to apply it with caution. Otherwise, it can
lead to phase cancelation, with the left and right channels canceling each other out, significantly
reducing the overall loudness or causing other unintended negative effects.

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 file. Use this plug-in to add silence at the end of a file, so that the tail of a reverb
plug-in does not cut immediately at the end of the file.

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

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

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 files that you do not want to convert to mono or for fixing a
problem with the stereo file, for example.

Invert Left Phase/Invert Right Phase

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

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Tools

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.
● 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.

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

The resulting file can then be put to multiple uses, such as the following:

● Testing the specifications of audio equipment

● Measurements of various kinds, such as calibrating tape recorders
● Testing signal processing methods
● Educational purposes

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 TestGenerator to an
audio track as an effect and activate it, a signal is generated. You can then start recording an
audio file as usual, based on the signal specifications.

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Tools

Interval
Activates/Deactivates the modulation of the generated signal using a pulse signal of
1 kHz with a duration of 200 ms.

Signal type section

Allows you to set the basis for the signal generated by the waveform generator. You
can choose from different waveforms (Sine, Triangle, Square, a sweep signal, and
Sawtooth) and types of noise (White, Pink, Brown, Blue, Gray, Violet).

Frequency section
Allows you to set the frequency of the generated signal. You can select one of the
preset values, set a value by using the slider or a quick control, or enter a value into the
value field. In the value field, 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, A5 -23 or C4 +49.

NOTE

Enter a space between the note and the cent offset to ensure that 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, set a value by using the slider or a quick
control, or enter a value into the value field.

NOTE

The minimum gain value of -120 dB can only be set by entering the value into the value
field.

Input Gain
Allows you to add the input signal to the generated signal.

Sweep section
Allows you to adjust the start frequency, the end frequency, and the duration of the
sweep signal.

Output buttons
Allow you to activate/deactivate the output signal for each channel separately.

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In the Batch Processor window, you can add a sequence of plug-ins that can be used to
process a batch of audio files. These plug-ins can be standard plug-ins available from the Master
Section, offline 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 files with statistics about the audio files in a batch
process.

This monopass plug-in is exclusively available in the Batch Processor window.

To analyze files without writing anything, select No Audio Output via the Output tab of the
Batch Processor window.

Information to Report
This column allows you to select which of the following information to include in the output:

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Audio Analyzer

● Sample Rate
● Number of Channels
● Real Bit Depth
● Number of Samples/Duration
● Digital Silence at Head and Tail

NOTE

We recommend that you activate this option if a silent section at the beginning and at the
end of audio material intended for public release is mandatory in your professional context,
as it facilitates verification for quality control purposes.

● Digital Peaks
● True Peaks
● Integrated Loudness
● Loudness Range
● Short-term Loudness: Maximum
● Short-term Loudness: Minimum
● Momentary Loudness: Maximum
● Momentary Loudness: Minimum
● Loudness Balance
● RMS
● DC Offset

Output
This column allows you to set up the output of the Audio Analyzer. The following options are
available:

Send Analysis Results to XML Processor

Submits the analysis results to the XML or HTML output of the batch processor as
parameters.

Create One Report per Audio File

Generates a report for each audio file in the batch process. The audio file name is used
as the file name.

Merge Results into Global Report

Consolidates the results of the analysis into a global report. The audio file name is used
as the file name.

Open Global Report at End of Batch Process

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

File Name Suffix

Allows you to specify a file name suffix. This is necessary when using the Audio
Analyzer in a batch process multiple times; for example, to see the stats before and
after adding specific plug-ins.
We recommend using a different suffix for each instance of the Audio Analyzer plug-in
in the processing chain.

Output format
Allows you to choose from the following output formats:

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Audio Injector

● Pure text
● HTML
● Adobe PDF
● Open Office
● Spreadsheet
● XML

Audio Injector
This plug-in allows you to insert an audio file at the beginning and/or end of the audio file being
processed. The inserted file can also be crossfaded with the original audio file, if required.

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

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

Specifies the audio file to be added before the main audio file.

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

Specifies the audio file to be added after the main audio file.

Perform Crossfade
Allows you to select a crossfade time and shape for the crossfade between the main
audio file and the injected audio file.

Audio Mixer
This plug-in allows you to mix an audio file with other audio files. The mix-in happens from a
specified time and can optionally be repeated at a specified interval.

For example, you can insert a spectral watermark into the audio spectrum or you can insert beep
sounds to mark an audio file 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 the audio file that you want to mix into other audio files. The audio
file must be no longer than 30 seconds.

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DC Remover

Time Position
Allows you to specify the time position at which the audio file will be mixed in.

Repeat Every
If this option is activated, you can specify the time after which the audio file is
repeatedly mixed in.

DC Remover
This plug-in allows you to remove any DC Offset from an audio file.

It is useful to apply this plug-in first in a batch before other plug-ins to avoid further processing a
file containing any DC Offset. For example, an audio file 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 offline processor in
the Audio Editor.

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
Specifies the duration until the wet signal starts to fade-in the dry signal.

Crossfade Time
Specifies 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
file. 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.

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Instructor

Duration
Determines the duration of the fade.

Start Gain/End Gain

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

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 Instructor plug-in can help in this situation,
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 first 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 first instance of the Instructor plug-in.

For example, this means that the DeNoiser plug-in has time to sufficiently analyze the audio
stream before the second stream start is injected. The “badly” processed first 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.

NOTE

Do not set a value that is longer than the shortest file in the batch, otherwise a short file 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 first instance of the Instructor plug-in.

Reverse Stream
If this option is activated, the start of the stream is injected first 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.

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Level Normalizer

Duration
Specifies how much audio to replicate or skip.

Level Normalizer
This multipass plug-in allows you to raise or lower levels so that the signal peaks exactly at the
specified value just before it is converted to a file.

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 file gets the specified peak level.
This ensure a clip-less mix.

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

Loudness Meta Normalizer

The Loudness Meta Normalizer batch plug-in allows you to change the loudness level of a batch
of files, while maintaining a consistent loudness ratio among them.

The Loudness Meta Normalizer plug-in operates by modifying gain values. It does not apply
any audio compression, thus ensuring that the original sound quality is not compromised.

The Loudness Meta Normalizer serves the following purposes:

● To apply the same gain to all files, while at the same time, ensuring that a specified peak
level is not exceeded in any file. If required to prevent clipping, the gain that you specify
is automatically reduced by the plug-in, after analyzing all files in the batch, and prior to
applying the gain to the files in the batch.
● To set the loudness of a reference file to a specific value, and to offset the gain level of all
other files in the batch by the amount that was applied to the reference file to obtain the
desired loudness level.

The Loudness Meta Normalizer batch plug-in is exclusively available on the Metapass Plug-ins
panel of the Batch Processor window.

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Loudness Meta Normalizer

You can choose one of the following options from the Loudness pop-up menu at the top:

Match Loudest Audio File

Applies the same amount of gain to all audio files, so that they match the loudness
level of the loudest of them as closely as possible.

Match Maximum Achievable Loudness

Detects the maximum achievable loudness among all files, without exceeding the
specified peak value, before applying the same amount of gain to all audio files, so
that they match the maximum loudness level.

Set Same Specific Loudness

Applies a specific amount of gain to all audio files, to match the specified loudness
value as closely as possible.

Set Loudness of Named File, Shift Others

Allows you to define a single file as the reference, or “master”, file. This file is set
to the loudness value that you specify in the Loudness entry field below, and the
same amount of gain is applied to all audio files, regardless of their original individual
loudness levels.
When you select this option, a new entry field appears, which prompts you to enter
Part of Master File Name. You can either type in the entire file name or a fragment of
it, so that the plug-in can identify the file and use it as the “master” or reference file.
The option is particularly suited for stem mastering.
Use case example:
You have the following files:

● Main Mix
● Drum Stem
● Bass Stem
● Guitar Stem
● Keyboard Stem

To set the Main Mix file as the “master” file, you can type in the entire file name or a
part of it, such as “mix” or “main”.

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Loudness Meta Normalizer

NOTE

Make sure that the name fragment that you enter into the field is unique. If you intend
to set Guitar Stem as the “master” file, for example, typing in “stem” is not sufficient
for the plug-in to clearly identify a file. As a consequence, no “master” file is set.

The option allows you to set the Main Mix file to a specific value, such as -18 LUFS
Integrated Max, before it automatically adds the same amount of gain to the loudness
level of all stem files.

In the Loudness entry field, you can specify the loudness value to be matched; for example, -23
LUFS to adhere to the provisions of the EBU R-128 standard in this respect.

You can choose one of the following options from the Reference pop-up menu:

File Integrated Loudness

The loudness of all files, using the standard Integrated Loudness value, in accordance
with the EBU R-128 standard

Top of Loudness Range

The peak of a loudness range (LRA); that is, the average loudness level of the loudest
audio segments

NOTE

The loudest 10% of the audio segments with a duration of less than three seconds are
excluded from the calculation, to ensure that exceptionally loud single sounds do not
affect the outcome.

Maximum Short-Term Loudness

The highest value resulting from the short-term loudness analysis

You can choose from the following Maximum Peak Level options:

● Control Digital Peaks

Limits the sample values, with the peak measured in the digital domain as the reference.

● Control True Peaks

Limits the reconstructed analog sample values, with the peak measured by simulating an
analog signal as the reference.

● Maximum Peak
Determines the maximum peak value not to be exceeded.

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Loudness Restorer

Loudness Restorer
Loudness Restorer captures the loudness at a specific 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.

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

Mode - Capture
The first 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 file (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).

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

Meta Leveler
The Meta Leveler plug-in allows you to adjust the levels of all files in a batch by the same value.

The Meta Leveler operates by modifying gain values. It does not apply any audio compression,
thus ensuring that the original sound quality is not compromised.

The main purpose of this plug-in is to apply the same gain to all files, while at the same time,
ensuring that a specified peak level is not exceeded in any file. If required, to prevent clipping,
the gain that you specify is automatically reduced by the plug-in, after analyzing all files in the
batch, prior to actually applying the gain to the files.

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Meta Leveler

The Meta Leveler batch plug-in is exclusively available on the Metapass Plug-ins panel of the
Batch Processor window.

Peak Normalize Predominant File, Shift Others

Adjusts the file with the highest peak (the “predominant” file), so that its peak
corresponds to the Maximum Peak value, as defined in the entry field below.
All other files are offset by the same amount.
Use case example:
You have the following files:
● File “A” with a peak of -4 dB.
● File “B” with a peak of -6 dB.
● File “C” with a peak of -7 dB.
The “predominant” file is file “A”.
With Maximum Peak set to -1 dB, a gain of +3 dB is applied to file “A”, so that it
reaches -1 dB. The same amount of gain (+3 dB) is applied to all other files.
This results in the following peak levels:
● -1 dB for file “A”.
● -3 dB for file “B”.
● -4 dB for file “C”.

NOTE

With this option activated, it is not possible to set a specific value for Gain.

Gain
Allows you to define a specific gain to be applied to all audio files.

NOTE

The actual gain that is applied to individual files may be lower than the value you enter,
or even negative, if required, to ensure that the value specified for Maximum Peak is
not exceeded.

Maximum Peak
Allows you to set the maximum peak level to a specific value.

● With Peak Normalize Predominant File, Shift Others activated, the Maximum
Peak value specifies the maximum peak level of the “predominant” file; that is, the
file with the highest peak.

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Batch Processing Plug-ins
Resizer

● With Peak Normalize Predominant File, Shift Others deactivated, the Maximum
Peak value specifies the maximum peak level to be set for all files at the end of the
process.

Reference
Allows you to choose from the following options for analyzing the batch of files:

● True Peaks: The analysis is based on analog reconstructed values.

● Digital Peaks: The analysis is based on sample values.

Resizer
This plug-in allows you to specify the duration of all audio files 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
peak level that the stereo file contains, or set the gain to be applied and/or the maximum level to
be reached in the resulting mono file.

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 file is the same as the
peak level of the original stereo file.

Gain
Specifies the increase or decrease in peak level for the resulting mono file, in relation
to the original stereo file.

Maximum Level of Mono Signal

Specifies the peak level that the resulting mono file must not exceed. This ensure that
the output file never clips. This way, the result never exceeds 0 dB, regardless of the
specified Gain value.

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Batch Processing Plug-ins
Trimmer

Trimmer
This plug-in allows you to remove a specified duration (from 0 ms to 60 s) of audio from the head
and/or tail of an audio file.

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

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 Index

A CurveEQ 69

Aligning
Samples 126 D
Amp simulation DC Remover 133
Quadrafuzz v2 38 DeBuzzer 113
Analyzers RestoreRig 110
SuperVision 4 DeClicker 112
Audio Analyzer 130 RestoreRig 110
Audio Injector 132 DeEsser 45
Audio Mixer 132 Delay Next Process Activation 133
AutoPan 105 Delays
MonoDelay 33
B PingPongDelay 34
Quadrafuzz v2 38
Batch processing plug-ins 130 StereoDelay 34
Audio Analyzer 130 DeNoiser 112
Audio Injector 132 RestoreRig 110
Audio Mixer 132 DeReverb 108
DC Remover 133 Distortion 36
Delay Next Process Activation 133 Dithering
Fade In/Fade Out 133 Lin Pro Dither 84
Instructor 134 Dithering plug-ins
Level Normalizer 135 Internal dithering 83
Loudness Meta Normalizer 135 DualFilter 80
Loudness Restorer 138 Ducking effects 43, 52
Meta Leveler 138 Dynamic EQ
Resizer 140 MasterRig 98
Stereo to Mono 140
Trimmer 141
Black Valve 41 E
Brickwall Limiter 42 EBU R-128
Audio Analyzer 130
C Loudness Meta Normalizer 135
Envelope shapers
Channel Extractor 125 EnvelopeShaper 47
Chopper effects MultibandEnvelopeShaper 54
AutoPan 105 EQ-M5 69
Chorus 106 EQ-P1A 70
Chorus effects Equalizer
Chorus 106 MasterRig 96
StudioChorus 107 Equalizers
Compressor 43 CurveEQ 69
MasterRig 92 EQ-M5 69
Compressors EQ-P1A 70
Black Valve 41 Frequency 2 70
Compressor 43 GEQ-10 76
DeEsser 45 GEQ-30 76
Maximizer 51 StudioEQ 78
MultibandCompressor 52 Expanders
Squasher 59 Expander 47
Tube Compressor 63 MultibandExpander 56
Vintage Compressor 64
VoxComp 65
VSTDynamics 66

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Index

F Modulation effects
AutoPan 105
Fade In/Fade Out 133 Chorus 106
Filter effects StudioChorus 107
DualFilter 80 MonoDelay 33
PostFilter 81 MonoToStereo 122
Frequency 2 70 MultibandCompressor 52
MultibandEnvelopeShaper 54
MultibandExpander 56
G
Gates O
Gate 49
Quadrafuzz v2 38 Octaver 108
VSTDynamics 66
P
I
Peak Master 103
Imager 120 PingPongDelay 34
MasterRig 102 Plug-ins
Instructor 134 Batch processing 130
Internal dithering 83 VST 3 4
PostFilter 81
L
Q
L/R to M/S 126
Level Normalizer 135 Quadrafuzz v2 38
Leveler 83
Leveler Multi 84
Limiter 50 R
MasterRig 89 Raiser 58
Limiters Resampler 104
Brickwall Limiter 42 Resizer 140
Limiter 50 Restoration
Maximizer 51 DeReverb 108
Raiser 58 RestoreRig 110
VSTDynamics 66 Layout 110
Lin Pro Dither 84 Modules 111
Loudness Meta Normalizer 135 REVelation 114
Loudness Restorer 138 Reverb effects
REVelation 114
M RoomWorks 116
RoomWorks SE 119
M/S to L/R 126 RoomWorks 116
Magneto II 37 RoomWorks SE 119
MasterRig 85
Compressor 92
Dynamic EQ 98 S
Equalizer 96 Sample rate
Imager 102 Resampler 104
Layout 85 SampleAlign 126
Limiter 89 Samples
Modules 88 Aligning 126
Saturator 100 Shifting 126
Settings 88 Saturation effects
Maximizer 51 Magneto II 37
Meta Leveler 138 Quadrafuzz v2 38
Meters Saturator
SuperVision 4 MasterRig 100
Mix6to2 123 Shifting
Mix8to2 124 Samples 126
MixConvert V6 120 Silence 127
Squasher 59

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Index

Stereo Expander 123

Stereo to Mono 140
Stereo Tools 127
StereoDelay 34
StereoEnhancer 122
StudioChorus 107
StudioEQ 78
SuperVision 4

T
TestGenerator 128
Tools
SuperVision 4
TestGenerator 128
Trimmer 141
Tube Compressor 63

V
Vintage Compressor 64
VoxComp 65
VST AmbiDecoder 120
VSTDynamics 66

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WaveLab Pro 12.0.0