Steinberg WaveLab Essential 6 Operation Manual

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Plug-in processor reference
About WaveLab Essential plug-ins
These plug-ins use WaveLab Essential’s own plug-in for-
mat, and cannot be used with other applications. Note:
ÖAs a rule, WaveLab Essential specific plug-ins can 
only be used in the Master Section (not as clip effects in 
the Audio CD Montage).
However, some WaveLab Essential effects are also included as VST 
plug-ins, available as clip effects in Audio CD Montages. This is indi-
cated for each effect below.
ÖYou can specify which plug-ins should be available in 
the Master Section by using the Organize Master Section 
Plug-ins function on the Options menu.
This also allows you to specify which plug-ins should be available in the 
Dithering Pane (post-master fader).
ÖPresets for WaveLab Essential plug-ins are handled 
like other presets in WaveLab Essential (processing func-
tions, etc.).
Auto Panner
The Auto Panner (only available in the Master Section) 
pans the signals continuously between the left and right 
channel in the stereo image. It has the following parame-
ters:
Chorus
The Chorus plug-in (only available in the Master Section) 
is a classic stereo chorus based on a sweeping delay, 
with the following parameters:
Parameter Description
LFO Freq 
(0.1Hz~50Hz)This parameter sets the speed of the panning. The 
higher the value, the faster the signal moves around 
in the stereo image. 
Width  (0~100%) Use this parameter to specify the “width” of the pan 
movements. The value 100% causes the signal to 
move from the extreme left to the extreme right, 
while 0% disables the panning effect.
Waveform
(Sine, Pulse)Allows you to specify the way in which the signal 
moves from left to right. Select Sine if you prefer 
fluid movements, or Pulse to create abrupt panning 
“jumps”.
Out Left, Out Right 
(-96dB~6dB)These two parameters allow you to adjust the level 
of the left or right channel, useful e.g. for correcting 
the stereo image or adjusting the overall gain. The 
setting 0dB means no change of level, while -96dB 
means turning the channel off completely.
Parameter Description
Delay (0.1~60ms) Use this parameter to specify the basic time delay 
for the chorused signal with respect to the “dry” 
signal. The higher the value, the more prominent the 
effect. Low settings (up to 7ms) create flanger-like 
effects. Settings up to 25ms are for classic chorus 
while settings above this value are mainly for spe-
cial effects.
Width (0~100%) Use this parameter to specify how much the delay 
time is allowed to vary with the modulation. It is this 
variation in delay time that causes the sweeping ef-
fect. Note that the value 0% should probably be 
avoided since it might create the impression that 
you are experiencing phase problems.
Frequency 
(0.01~25Hz)The Frequency parameter sets the speed of the 
sweep (the modulation). The higher the value, the 
faster the modulation. You will probably not use val-
ues above 7Hz except for special effects.
Feedback 
(0~100%)This parameter specifies how much of the output 
from the effect is fed back to the input (the feed-
back signal is also phase inverted). The higher the 
value, the more prominent the effect. At short delay 
times, this creates a flanger-like effect. At larger 
settings it creates more of a slapback repetition 
type of sound.
Fb Balance (0~100%) Use this parameter to set the volume of the Feed-
back signal (see above) in the mix. If this is set to 
100%, and combined with a Feedback setting be-
tween 65% and 100%, the effect goes into self-os-
cillation.
Glimmer 1, 
Glimmer 2 (0~100%)The two Glimmer parameters allow you to specify 
to what extent the Chorus signal should move 
around in the stereo image. They work in more or 
less the same way as the Auto Panner but only ap-
ply to the Chorus signals. Glimmer 1 processes the 
right channel signal, while Glimmer 2 processes a 
combination of the left and right channels (the ac-
tual left channel always remains at 0).
Stereo Spread 
(0~100%)This parameter specifies the width the Chorus ef-
fect will occupy in the stereo sound image. The 
value 0% creates a mono impression, and since left 
and right channel signals are then mixed together, 
the Chorus becomes louder.
Mix (0~100%) Use this parameter to specify the balance in level 
between the dry and the delayed signal. The value 
0% means that only the dry signal will be heard, 
while 100% actually means a 50/50 mix between 
dry and effect signal.
Output Lev
(-48dB~0dB)This is an attenuator that allows you to reduce the 
output level of the Chorus effect, avoiding clipping 
and hence distortion. If the Clip indicator lights up 
continuously, lower this value. 
						

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Plug-in processor reference
Crystal Resampler
The Crystal Resampler plug-in (only available in the Mas-
ter Section) is a professional sample rate converter pro-
viding exceptional transparency and preservation of the 
frequency content:
Echo
The Echo plug-in (only available in the Master Section) is 
a stereo echo effect with two separate delay lines. It has 
the following parameters:
EQ-1
EQ-1 is available both as a VST and a WaveLab Essential 
plug-in. It can be used as a clip effect in the Audio CD 
Montage, or as a global effect in the Master Section. This is a three-band equalizer with high and low shelving 
filters and a full parametric mid-frequency band. You can 
turn off each band separately by clicking the correspond-
ing button, making it easy to compare the signal with and 
without EQ. The following parameters are available:
Leveler
This plug-in is available both as a VST and a WaveLab Es-
sential plug-in. It can be used as a clip effect in the Audio 
CD Montage, or as a global effect in the Master Section. 
This “effect” simply reduces or boosts the signal level. 
This is useful for matching levels between effects. You 
may want to patch in the Leveler after an equalizer plug-in, 
for example. The parameters consist of volume settings for 
the left and right channels, and a Stereo Link setting 
(when activated, the Volume Left parameters control the 
level of both channels). Finally, a Mix to mono setting al-
lows you to mix an incoming stereo signal to mono (much 
like the Mono button in the Master Section).
Noise Gate
This plug-in is available both as a VST and a WaveLab Es-
sential plug-in. It can be used as a clip effect in the Audio 
CD Montage, or as a global effect in the Master Section.
Parameter Description
Sample Rate 
(6 - 96 kHz)This defines the output sample rate while the input 
sample rate is determined by the sample rate of the 
active audio file or Audio CD Montage.
Quality (Preview (fast), 
Standard)This defines the quality of the algorithm which is 
used. In Preview mode the CPU load is much lower 
than in Standard mode, but as a trade off the sonic 
quality of the resulting audio is also lower.
Parameter Description
Delay 1 
(0.5~1000ms)This sets the delay time of Delay 1, with respect to 
the incoming signal. Please note that the minimum 
value (0.5ms) creates an out-of-phase impression.
Feedback 1 
(0~100%)This sets the amount of delayed signal fed back 
into the Delay 1 block, to create repetitions. The 
value 100% means that the echo signal is repeated 
indefinitely, while 0% means there is only one repe-
tition.
Link 1-2 
(Off, Linked)Select Off if you wish to use Delay 1 and Delay 2 as 
two independent blocks. Select Linked if the output 
of Delay 1 is to be connected to the Input of Delay 
2.
Delay 2, Feedback 2 See Delay 1 (0.5~1000ms) and Feedback 1 
(0~100%) above. These parameters are identical 
to these but apply to the second Delay block.
Del. Balance 
(0~100%)This sets the stereo width of Delay 1 and Delay 2. 
When set to 100%, Delay 1 is assigned to the left 
channel, while Delay 2 is assigned to the right 
channel. The value 0% means that both Delay 
blocks are spread across the stereo field.
Vol Left, Vol Right 
(-96dB~0dB)Use these parameters to correct volume imbal-
ances brought about by the Delay effects. These 
parameters only apply to the Echo effect, the dry 
signal is not affected by these settings.Parameter Description
High Gain Determines the boost or cut (in dB) of the high 
shelving filter.
High Frequency Sets the frequency of the high shelving filter. Fre-
quencies above this value will gradually be in-
creased or reduced in level, according to the High 
Gain setting.
Mid  Gain Determines the boost or cut (in dB) of the mid band 
EQ.
Mid Frequency Sets the center frequency of the mid band eq. Fre-
quencies around this value will be affected by the 
Mid Gain.
Mid Q Use this parameter to set the width of the Mid 
band, e.g. how wide a frequency range around the 
Mid Frequency should be affected by the mid band 
EQ. The higher this value, the “narrower” the mid 
band.
Low Gain Determines the boost or cut (in dB) of the low 
shelving filter.
Low Frequency Sets the frequency of the low shelving filter. Fre-
quencies below this value will gradually be in-
creased or reduced in level, according to the Low 
Gain setting. 
						

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Plug-in processor reference
The Noise Gate plug-in mutes any signal that falls below a 
specified threshold level. This can be useful for removing 
unwanted residual noise from audio material without hav-
ing to manually clean up or mute soundfiles. Other appli-
cations include gating reverb “tails” and tightening 
percussion tracks.
Peak Master
This plug-in is available both as a VST and a WaveLab Es-
sential plug-in. It can be used as a clip effect in the Audio 
CD Montage, or as a global effect in the Master Section. 
This plug-in provides a safe and transparent way of boost-
ing the perceived loudness of audio material. By limiting 
transients and simultaneously raising the general level by 
compression, the Peak Master will increase the subjective 
loudness of the signal without risk of distortion inducing 
peaks.
Puncher
This plug-in is available both as a VST and a WaveLab Es-
sential plug-in. It can be used as a clip effect in the Audio 
CD Montage, or as a global effect in the Master Section.The Puncher plug-in generates additional harmonics 
which are added to the audio material. The result is a more 
dynamic, and “punchier” sound, particularly when applied 
to drums and percussive material. Compared to the Peak 
Master the Puncher plug-in could be described as operat-
ing in almost the opposite way. Puncher leaves quieter 
parts untouched but will add power to the louder portions 
without causing clipping. The plug-in is optimized for peak 
signal levels between -10 and 0dB, the closer to 0dB, the 
better.
Silence
The Silence plug-in (only available in the Master Section) 
lets you add silent portions at the start and/or end of a file. 
This can be useful for example in conjunction with effects 
such as reverb and delay which produce audio “tails” – i.e. 
the sound of the effect lingers after the end of the file – 
since the sound of the effect would otherwise be muted at 
the end of the file. To remedy this, just place the Silence 
plug-in before the other plug-in in the Master Section and 
specify the length of the silent portion as necessary, so 
that the sound of the effect is allowed to decay naturally.
The Silence plug-in only has two parameters which let you 
define the length of the silent portions at the start and end 
of the file.
StereoExpander
The StereoExpander plug-in (only available in the Master 
Section) narrows or enlarges the stereo width of an exist-
ing stereo signal. This is set by the single “Width” param-
eter. 
A value of 0% produces two equal output channels (the 
original stereo image is lost). Values between 1 and 49% 
correspond to a narrower stereo image. A value of 50% 
corresponds to the original signal. Values between 51 and 
100% enlarge the stereo image.
Parameter Description
Threshold
(-144~-12dB)This setting determines at what level the Noise 
Gate is activated. Any signal or portion of a signal 
that falls below the chosen threshold will be muted.
Rel Time (1~5000ms) Determines how long the gate stays open after a 
signal below the Threshold level has been de-
tected.
Rel Sens (1~100) This setting is used to prevent the gate from being 
triggered on/off inadvertently when the signal is 
close to the threshold level.
Attack Sens (1~100) Determines the time it takes for the gate to open. A 
low setting provides a fast transient response but a 
high setting will soften or mute the early portion of 
the sound which is triggering the gate.
Parameter Description
Input Gain
(-12~+24dB)This allows you to adjust the input level to Peak 
Master. Use this to (typically) raise the loudness of 
the signal. Use extreme boost settings with caution 
as they can induce distortion.
Out Ceiling
(-18~0dB)This setting determines the maximum level at the 
Peak Master outputs.
Softness (-5~5) This parameter affects the way the Peak Master op-
erates. A high setting will maximize the perceived 
loudness effect but can in some cases result in a 
slight harshness of the sound. Adjust this parame-
ter to optimize the balance between sound quality 
and the desired effect.
Parameter Description
Density (Soft, Medium, 
Hard)The difference between these 3 settings lies in the 
number of added harmonics. The setting you use 
depends on the audio material.
Effect (0~100%) This adjusts the balance between the processed 
and the dry signal.
Input Gain
(-12~24dB)This sets the input level. Boosting the signal may 
cause clipping, so use this with caution. With no 
boost, Puncher will never cause clipping. 
						

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Plug-in processor reference
VST Plug-ins
About VST Plug-ins
These plug-ins use Steinberg’s widely adapted VST plug-
in format. As a rule, VST plug-ins can be used by any VST-
compatible application, although some plug-ins may still 
be limited to use with certain programs. Note:
ÖVST Plug-ins can be used in the Master Section or as 
clip effects in the Audio CD Montage.
ÖAs with WaveLab Essential plug-ins, you can specify 
which VST plug-ins should be available in the Master Sec-
tion by using the Organize Master Section Plug-ins func-
tion on the Options menu.
This also allows you to specify which plug-ins should be available in the 
Dithering Pane (post-master fader).
ÖIt’s also possible to exclude VST plug-ins completely 
from WaveLab Essential, thereby removing them from the 
clip and track effects lists as well.
ÖVST plug-ins have their own preset handling. 
When you click the Preset button for this type of effect, a pop-up menu 
appears, allowing you to save or load effect programs (presets) or com-
plete banks containing several programs.
Autopan
The AutoPan plug-in pans the signals continuously be-
tween the left and right channel in the stereo image. It has 
the following parameters:
Choirus2
Choirus2 is a chorus effect, used for making the sound 
“warmer”, etc. It has the following parameters:
Parameter Description
LFO Freq 
(0.1Hz~10Hz)This parameter sets the speed of the pan move-
ments. The higher the value, the faster the pro-
cessed signal moves around in the stereo image. 
Width (0~100%) Use this parameter to specify the “width” of the pan 
movements. Setting this to its maximum value 
causes the signal to move from the extreme left to 
the extreme right, while lowering it completely dis-
ables the panning effect.
Waveform Determines the shape of the panning curve. Avail-
able curve shapes are Sine, Triangle, Sawtooth, 
and Pulse.
Out Levl  The stereo output level of the effect.
Parameter Description
Time  Use this parameter to specify the basic time delay 
for the chorused signal with respect to the “dry” 
signal. The higher the value, the more prominent the 
effect. Low settings create flanger-like effects, me-
dium settings provides classic chorus while higher 
settings are mainly for special effects.
Width  Use this parameter to specify how much the delay 
time is allowed to vary with the modulation. It is this 
variation in delay time that causes the sweeping ef-
fect. Note that the value 0% should probably be 
avoided since it might create the impression that 
you are experiencing phase problems.
Lfo Freq The Frequency parameter sets the speed of the 
sweep (the modulation). The higher the value, the 
faster the modulation. You will probably not use 
higher values (above 7Hz) except for special ef-
fects.
Feedback  This parameter specifies how much of the output 
from the effect is fed back to the input (the feed-
back signal is also phase inverted). The higher the 
value, the more prominent the effect. At short delay 
times, this creates a flanger-like effect. At larger 
settings it creates more of a slapback repetition 
type of sound.
Feed Bal  Use this parameter to set the volume of the Feed-
back signal (see above) in the mix. If this is set to 
100%, and combined with a Feedback setting be-
tween 65% and 100%, the effect goes into self-os-
cillation.
Glimmer 1, 
Glimmer 2 The two Glimmer parameters allow you to specify 
to what extent the Chorus signal should move 
around in the stereo image. They work in more or 
less the same way as the Auto Panner but only ap-
ply to the Chorus signals. Glimmer 1 processes the 
right channel signal, while Glimmer 2 processes a 
combination of the left and right channels (the ac-
tual left channel always remains at 0).
Out Levl  The stereo output level of the effect. 
						

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Plug-in processor reference
CleanComp
CleanComp is a simple compressor that allows you to 
limit loud sounds, while at the same time boosting the 
overall loudness of the audio material.
DeClicker
The DeClicker plug-in is specifically designed to eliminate 
single “clicks” or “pops” in a recording. One typical appli-
cation is to clean up recordings made from vinyl records, 
but you may also find it useful for removing pops from mi-
crophone switches, oxidized connector noises, clicks from 
sync problems when transferring material digitally, etc.
ÖNote that the DeClicker module is not optimized for 
crackles (a series of short clicks).
However, as it is often hard to distinguish between clicks and crackles, 
you might also be able to use it to improve your recording in this respect.
ÖIf the recording also contains background noise (hiss), 
you may want to combine DeClicker with the DeNoiser 
plug-in.
How DeClicker works
The Declicker process is divided into two steps:
Analysis – when the audio signal passes through De-
Clicker, the selected analysis algorithm finds the clicks in 
the recording. You provide input to the analysis parame-
ters by selecting a Mode and the Threshold and DePlop 
parameters.Removal – a de-click algorithm is applied to the audio, 
removing the clicks.
In many cases, the original audio material “hidden” underneath a click 
cannot be restored. This means there will be a gap once the click has 
been removed. DeClicker has the ability to automatically “redraw” the 
hence missing parts of the waveform. This feature can also be used to 
remove tape dropouts with a length of up to 60 samples (just above one 
millisecond at 44.1kHz).
The whole Declicking process can be visually monitored in 
the Input and Output displays of the DeClicker window 
(showing the incoming audio and the processed - De-
Clicked - audio, respectively). This helps you to adjust the 
parameters. Furthermore, if you activate the Audition but-
ton, only the removed material will be heard (and shown in 
the Output display).
Parameters
Parameter Description
Ceiling (0dB~-24) This setting determines the maximum level at the 
CleanComp outputs.
Softness (-5~5) This parameter affects the way CleanComp oper-
ates. A high setting will maximize the perceived 
loudness effect but can in some cases result in a 
slight harshness of the sound. Adjust this parame-
ter to optimize the balance between sound quality 
and the desired effect.
Out Gain (0~+24dB) This allows you to adjust the output level from 
CleanComp. Use this to (typically) raise the loud-
ness of the signal. Use extreme boost settings with 
caution as they can induce distortion.
!Make sure that no low-pass filter has been applied to 
your audio material before you edit it with DeClicker. 
This may affect the detection of clicks.
Parameter Description
Audition button When this is activated, only the removed material 
will be heard. The Output display will also show the 
waveform image of the removed material in this 
mode.
Classic When this is activated, the DeClicker attempts to 
remove both audible clicks and crackle noise. 
When it’s deactivated, only single clicks will be re-
moved while crackles (rapidly repeated clicks) are 
ignored. Which mode to choose depends on the 
source material. Note also that Classic mode re-
quires less CPU power.
Threshold This setting determines the amplitude (level) re-
quired for a click to be detected. In many cases, 
DeClicker’s sensitive algorithms identify a lot more 
clicks than you can actually hear. To avoid wasting 
processing power to remove inaudible clicks, raise 
this parameter to a high value, and then lower it un-
til all the artefacts that you actually want removed 
are detected. The lower the setting, the more clicks 
will be detected but also the higher the risk of audi-
ble artefacts. If in doubt, activate Audition mode 
and check that the removed material doesn’t con-
tain any actual musical or rhythmical information, 
etc. 
						

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Plug-in processor reference
Tips and Tricks
 By combining Vintage Mode and extreme Threshold and De-
Plop settings, you can create an interesting effect which “soft-
ens” material with particularly sharp attacks, e.g. percussion or 
brass.
 If you have material with digital distortion (clipping), try apply-
ing DeClicker. While it can’t do miracles, it can at least make 
some improvement to the overall “hardness” introduced by the 
distortion.
DeNoiser
The DeNoiser plug-in lets you suppress noise without af-
fecting the general sound quality. Or, in tech talk, the De-
Noiser removes broad band noise from arbitrary audio 
material without leaving any “spectral finger print”. The al-
gorithm that this plug-in is based on has the ability to track 
and adjust itself to variations in background noise. This means the noise can be diminished without side effects, 
preserving the spatial impression, and without letting the 
result become “colorless”. Many years of research were 
invested in developing the methods used.
Typical applications for the DeNoiser include cleaning or 
remastering recordings from old tape or vinyl, or noisy live 
recordings.
How DeNoiser works
DeNoiser is based on spectral subtraction. Each section 
of the frequency spectrum that has an amplitude below 
the estimated noise floor, is reduced in intensity by use of 
a spectral expander. The result is a noise reduction that 
does not affect the phase of the signal. 
The figure below shows the signal flow:
The solid line represents the actual audio signal, while the dotted lines 
represent control signals
The signal is continuously analyzed by the first module in 
the chain, to estimate the noise floor at any given time. 
This is sufficient when the noise level is constant or mod-
ulates slowly. When the noise level varies rapidly, the Am-
bience and Transient analyses help adjust the response of 
the noise reduction unit, allowing transient-rich material to 
maintain its liveliness and natural ambience.
ÖWhen you process audio in DeNoiser, the plug-in will 
need a short time (less than a second) to analyze the ma-
terial and set its internal parameters.
Since you would not want to include this short “startup sequence” in the 
final result, you should make it a habit to first play back a short section of 
the audio, thereby letting DeNoiser “learn” the noisefloor, and then stop 
and start over again from the beginning. The plug-in then remembers the 
settings internally.
DePlop This setting controls a special highpass filter which 
works on signals below 150 Hz. It cuts away the 
“plop noise” which sometimes appears after elimi-
nating a click. The slider adjusts the filter frequency 
(off - 150 Hz).
Note: This function is best applied to older record-
ings, which often use a narrow frequency range. Be 
careful when applying this function to modern re-
cordings, as you may risk removing parts of the 
useful signal! 
Quality This determines the quality of the click removal and 
audio restoration, with “4” being the best quality 
setting. Please note that selecting higher quality 
settings also means that more processing power is 
consumed.
Also, note that in some situations it might be more 
productive to use a lower Quality value. One exam-
ple of this is when two clicks follow each other in 
quick succession or when you tackle a click in a 
low level part that is followed by a loud part.
Mode Which Mode to select depends on the source ma-
terial. Standard mode is suitable for a wide variety 
of source material - try this option first. Vintage 
mode is suitable for restoring “antique” recordings 
(with limited high frequency content), while Modern 
mode is best suited for contemporary recordings 
with a wide frequency range (putting greater em-
phasis on distinguishing clicks from other strong 
impulses in the audio material).
Bypass This will bypass the effect, allowing you to compare 
the DeClicked and unprocessed material. Parameter Description
Noise 
Reduction
Noise Floor  Ambient 
AnalysisTransient 
Analysis
InputOutput
Level
Noise Reduction
Ambience 
						

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Plug-in processor reference
The Noisefloor Display
The display to the left in the DeNoiser window is crucial 
when making settings. It contains the following three ele-
ments:
The dark green spectral graph.
This shows the spectrum of the audio currently being played back. The 
horizontal axis shows the frequency (linear scale). The low frequencies 
are visible on the left side, the high ones on the right side. The vertical 
axis shows the signal amplitudes, thus the level (displayed as a logarith-
mic dB scale).
The yellow line.
This is a spectral estimation of the noise floor. The average of this value is 
shown numerically below the display.
 The light green line.
This is simply a graphic representation of the Offset parameter.
The light green Offset line should be adjusted so that it 
appears as close above the yellow noise floor graph as 
possible. The dark green spectrum plot is there to help 
you fine-tune the Offset setting, so that only the noise is 
removed, not parts of the signal (ideally, the light green 
line should be between the yellow line and the spectrum 
plot).
ParametersUsing the A/B setups
With the A/B buttons you can make instantaneous 
switches between two different DeNoiser setups, allowing 
you to quickly try out and compare different configura-
tions. You can also use this feature for separate settings 
for two different sections of an audio recording. Proceed 
as follows:
1.Make the settings you want for setup A.
2.Click on [Store] and then on the [A] button.
3.Make the settings you want for setup B.
4.Click on [Store] and then on the [B] button.
Now the two setups are stored, and you can switch between them sim-
ply by clicking [A] or [B].
NaturalVerb
NaturalVerb is a high-quality reverb that adds ambience, 
or room-quality, to the sound. In addition to the standard 
size and decay parameters, NaturalVerb also features low- 
and high-pass filters, plus a gate for gated reverb effects.
To change the parameters, either drag the sliders up and 
down, or click in a slider area to set the slider. If you hold 
down [Shift], you can change the parameters with a higher 
degree of precision. If you hold down [Ctrl] and click in a 
slider area, the slider is reset to its default value.
Parameter Description
Freeze If you activate this button, you “freeze” the noise 
floor detection process. The yellow noise floor 
graph in the display will hold its current value (as 
will the numeric noise floor value display below) un-
til you deactivate Freeze. This allows you to take a 
closer look at the readings.
Reduction Governs the amount of noise reduction. The display 
below this fader shows the amount of dB by which 
the noise level is being reduced. The final result 
also depends on the Ambience parameter, and on 
the automatic Ambience and Transient analysis of 
the original material, as described above.
Ambience This parameter is used to specify a balance 
between the noise suppression and the amount of 
natural ambience, which is essential for a natural 
result. With a low Ambience setting, the sound can 
become somewhat lifeless and sterile. A high set-
ting, on the other hand, preserves more of the am-
bient character of the sound, but the noise 
suppression is less effective.
Offset This parameter serves as a threshold, governing the 
overall level at which the noise reduction is per-
formed. For optimal noise reduction with a minimum 
of sound coloration, this parameter should be set to 
a value slightly above the noise floor level. To help 
you do this, the offset value is shown as a light 
green line in the noisefloor display, while the noise 
floor is shown as a yellow line.
A/B/Store These are described below this table. 
Classic When this is activated, a less CPU-intensive ver-
sion of the DeNoiser algorithm is used. Use Classic 
mode if you are short on processing power. How-
ever, for optimum noise suppression, we recom-
mend that you deactivate Classic mode.
Bypass When this is activated, the signal passes through 
the plug-in but you don’t hear the results of the pro-
cessing. Use this to compare the sound with and 
without processing. Note: the analysis is always 
performed, regardless of the Bypass switch. This 
allows you to monitor the noise floor, spectrum and 
level in the spectrum display. Parameter Description 
						

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Plug-in processor reference
If you click on the logo a diagram of the signal-chain is 
shown. The following parameters are available:
Spectralizer
The Spectralizer is a type of audio “enhancer” or “exciter”. 
It can be put to many uses:
 To restore lost harmonics in a recording.
 To improve the clarity and transparency of a recording.
 To make a recording sound “warmer”.
How Spectralizer works
Most audio equipment introduces a slight low-pass filter-
ing to the audio signal. This means you lose “high end” or 
“clarity”. The lost high frequency components often have a 
level close to the noise floor. This means that simply using 
EQ to boost the high frequencies does not create the de-
sired effect – the noise is amplified as much as the signal.
Spectralizer can actually re-synthesize lost harmonics 
based on existing lower frequencies in the material. This 
can create an acoustically more pleasing result than EQ-
ing.
Spectralizer basically works by generating 2nd and 3rd 
harmonics or overtones. 
Parameter Description
Pre-Delay This governs the start time of the first “early reflec-
tion”, i.e. how the sound is “bounced” off the walls 
in the simulated room environment. The value range 
is 0-100 milliseconds. The lower the value, the 
sooner the early reflection is heard. 
HPF This is a high-pass filter that only affects the reverb 
signal into the NaturalVerb, not the original audio 
signal. A high-pass filter lets high-frequency signals 
through while cutting off low-frequency signals. 
The slider allows you to set the frequency for the fil-
ter, and only sounds above the set frequency will 
be heard.
LPF This is a low-pass filter that only affects the reverb 
signal into the NaturalVerb, not the original audio 
signal. A low-pass filter lets low-frequency signals 
through while cutting off high-frequency signals. 
The slider allows you to set the frequency for the fil-
ter, and only sounds below the set frequency will 
be heard.
Room Size This regulates the size of the simulated room, and 
thereby the spaciousness of the reverb. The value 
range is 1-30, and the higher the value you specify, 
the bigger the room. 
Decay This lets you specify the length of the reverberation. 
The value range goes from 26 milliseconds to 
11.63 seconds.
Damping Damping can be used for attenuating the high fre-
quencies of the reverb, thereby creating a softer, 
warmer sound. The higher the value, the more the 
high frequencies will be attenuated.
Stereo Mix This parameter is used for balancing the reverb sig-
nal between the left and right channel inputs to the 
NaturalVerb. The value range is 0-100%. A setting 
of 0 means that the reverb signals for both chan-
nels are completely independent of each other (de-
fault), while a setting of 100 means that the reverb 
signals for both channels are equally mixed with 
each other (50/50). In between these, settings from 
1-99% mean that each channel signal will contain 
that percentage of the other channel’s signal. 
Wet/Dry This regulates the balance between the effect 
sound (wet) and the original, unprocessed audio 
signal (dry). If the slider is in the middle position 
(default), the output will be balanced equally. With 
higher values, the original signal will be more pro-
nounced, and with lower values, the effect sound 
will be more dominant.
Gate button Clicking this button turns the Gate section on and 
off. Gating cuts off signals below a certain set 
threshold level. That is, the Gate only opens to let 
signals above the set threshold through. Note that 
the three sliders directly above this button (Sensi-
tivity, Threshold and Fade-Out) control the Gate ef-
fect, and therefore have no functionality when this 
button is in the Off position. Also note that the Gate 
only affects the reverb, not the original audio signal.
Sensitivity This parameter determines how fast the Gate will 
open to let a trigger signal pass. The value range is 
1-100 milliseconds. In order for this to have any ef-
fect, the Gate button must be in the On position.
Threshold This is used for setting the reference signal level (in 
dB) for the Gate. Signal levels above the set 
threshold open the Gate and pass through, but sig-
nal levels below the set threshold close the Gate 
and are cut off. In order for this to have any effect, 
the Gate button must be in the On position.
Fade-Out This parameter determines how long it should take 
for the Gate to close again after being triggered to 
let a signal through. The value range is 0-200 milli-
seconds. With higher values, more signal “residue” 
will be allowed to pass through the Gate before it 
closes, thereby producing a smoother cut-off. In or-
der for this to have any effect, the Gate button must 
be in the On position. Parameter Description 
						

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Plug-in processor reference
 The second harmonic is a signal at twice the frequency (one 
octave) above the basic frequency (the fundamental). 
 The 3rd harmonic is three times the fundamental (one octave 
and a fifth above it).
The reasons for limiting the processor to these two fre-
quencies are; firstly, higher harmonics are most often per-
ceived as “too high”, and secondly, their amplitude 
normally doesn’t follow that of the fundamental in a natural 
way.
ÖPlease note the relation between the Frequency pa-
rameter and the harmonics generated. 
If the Frequency parameter for example is set to 4000, the 2nd harmonic 
generator will only create frequencies from 8000Hz upwards and the 3rd 
harmonic generator will add frequencies starting at 12000Hz.
Another part of this process is giving the added harmonics 
appropriate amplitude curves. The amplitude of the har-
monics is based on the amplitude of the existing material, 
but you can control it to some extent using the Density 
and Kick parameters, see “Parameters” below.
The amplitude of the added harmonics is usually very low. 
The difference Spectralizer makes is sometimes only ap-
parent on a psycho-acoustical level. To hear what is actu-
ally added to the signal, use the Solo button.
Parameters
The parameters are as follows:
Stereo Echo
The Stereo Echo is a delay with separate settings for the 
left and right channel. It can also be used as a single mono 
delay, in which case the maximum delay time will be dou-
bled.
The Stereo Echo has the following parameters:
Parameter Description
Solo When this is activated, the output will only contain 
the added Harmonics. The original unprocessed 
signal will not be heard on the output.
This mode is used as a diagnostic tool to monitor 
what the current settings actually add to the signal.
Kick When this is activated, even more Harmonics will 
be added when a transient (attack) occurs in the 
signal.
Frequency This adjusts the frequency of the high-pass filter 
that appears just after the input. Signals with a fre-
quency lower than this setting will not be affected 
by the processing. In other words, there will be no 
harmonics added to the frequencies that are too 
low to pass through the filter.
Density This controls the amplitude “envelope” of the 
added harmonics. The higher the value, the more 
prominent the effect.
Input This adjusts the overall input to the processor. Use 
this to both maximize the signal level and to make 
sure that internal clipping does not occur. Use the 
Meter and “Int. Clip” indicator to check the levels.
Gain This adjusts the level of the signal just before it 
reaches the harmonic generators.
As you increase this, you will most probably have to 
lower the Input setting to avoid clipping.
2nd This sets the level of the 2nd harmonics in the mix.
3rd This sets the level of the 3rd harmonics in the mix. 
Mix The Mix parameter adjusts the balance between the 
unprocessed signal and the added harmonics.
Int. clip When this lights up, the signal has exceeded the 
maximum level that Spectralizer can handle. This 
will lead to unpleasant distortion and should defi-
nitely be avoided. If this happens, please lower the 
Input level and/or Gain.
Meter This allows you to check your adjustments of the In-
put and Gain controls, so that the signal does not 
change drastically in level when passing through 
the Spectralizer.
Parameter Description
Delay 1  The delay time for the left channel. The maximum 
delay time is 1486 ms, unless you link both chan-
nels for mono operation, in which case the maxi-
mum delay time is 2972 ms - see below (1000ms 
= 1 second).
Feedback 1  This sets the amount of delayed signal fed back 
into the Delay 1 block, to create repetitions. Higher 
values result in a higher number of echo repeats.
Link 1-2
(Off, Linked)Select Off if you wish to use Delay 1 and Delay 2 as 
two independent blocks. Select Linked if the output 
of Delay 1 is to be connected to the Input of Delay 
2.
Delay 2,
Feedback 2These parameters are identical to these but apply 
to the second Delay block.
Del2 Bal This parameter determines how much of the left 
channel output is sent to the right channel input. 
When set to 0.0 (fully left), then none of the left 
channel output is added to the right channel input; 
when it is set to 1.0 (fully right), the right input re-
ceives both its normal source and the complete 
output of the left channel.
Volume L The output level of the left channel delay.
Volume R The output level of the right channel delay. Parameter Description 
						

							200
Plug-in processor reference
StereoExpander
The StereoExpander plug-in narrows or enlarges the ste-
reo width of an existing stereo signal. There is only one 
parameter, the horizontal stereo effect slider. Setting this 
to a value of -100% produces two equal output channels 
(the original stereo image is lost). Values between -99 and 
-1 correspond to a narrower stereo image. A value of 0 
corresponds to the original signal, while values between 1 
and 100 enlarge the stereo image.
Tools One
Tools One is an extremely useful “effect” for various appli-
cations. 
The level faders allow you to adjust the level of the left and 
right channel respectively. You can [Shift]-drag to make 
detailed settings. [Ctrl]-clicking a fader resets it to 0 dB 
(no level adjustment). Normally, adjusting one fader auto-
matically moves the other as well, but you can make sepa-
rate adjustments for the channels by pressing [Alt] and 
dragging.
The two Phase switches let you invert the phase of the left 
or right channel (or both).
The Algorithm buttons let you adjust the stereo sound im-
age. When none of the buttons are activated, the stereo 
image will be preserved as is. MS process mode can be 
used in one of two ways:
 To transform an incoming “regular” stereo signal so that it re-
sembles a signal recorded according to the M-S (middle/side) 
principle. This technique is often used in broadcasting to 
record the direct signal source (usually a voice) using one mi-
crophone, and the ambience using a second microphone po-
sitioned at a 90° angle. 
 To transform an incoming MS signal into a “regular” stereo 
signal (to simulate an “XY” recording technique, where neither 
microphone is placed directly in front of the signal source).
Channel Swap, finally, means that the left channel is as-
signed to the right side and the right channel to the left 
side.
Voice Attenuator
This plug-in can be used to remove lead vocals from a re-
cording, to produce a “karaoke” effect. The principle con-
cept is based on the fact that vocals are usually mixed to 
center position in the stereo field, and that the human 
voice occupies a limited area of the frequency spectrum. 
Note, however, that it is nearly impossible to remove a vo-
cal completely, without using very complex processing be-
yond the scope of this plug-in. 
 If the Remove Mono button is activated, the plug-in will sum 
the right and the left channels (with one of the channels out of 
phase), in the frequency range set by the Low and High Fre-
quency parameters. This method will only work with stereo 
material.
 If the Notch Filter button is activated, the plug-in will filter out 
the signals within the frequency range set with the Low and 
High Frequency parameters, by applying a notch (band reject) 
filter. This method can be used with both stereo and mono 
material.
 The Gain parameter allows you to adjust the output level of 
the plug-in.
VSTDynamics
General Information
The VST Dynamics plug-in combines five separate pro-
cessors; AutoGate, Compress, AutoLevel, Limit and Soft-
Clip, covering a variety of Dynamic Processing functions. 
The VST Dynamics window is divided into five sections, 
containing controls and meters for each processor. You 
activate the VST Dynamics panel by clicking the “On” but-
ton in the lower right corner. Once VST Dynamics is acti-
vated, you can turn the individual processors on and off by 
clicking on their labels. Activated processors have high-
lighted labels.
You can activate as many processors as you want, but re-
member that not all processors are designed to work to-
gether. For example, “Limit” and “SoftClip” are both 
designed to ensure that the output never exceeds 0dB, 
but achieve this in different ways. To have both of them 
activated would be unnecessary. The internal signal flow 
is printed in the lower right part of the Dynamics panel.