Steinberg WaveLab 3 Operation Manual

							WAVELAB
Plug-in Processor Reference 33 – 781
Q
Q is a high-quality 4-band parametric stereo equalizer with two fully para-
metric midrange bands. The low and high bands can act as either stan-
dard shelving filters or fixed-gain high/low-cut filters.
Making settings
1.Click the corresponding On button below the EQ curve display to acti-
vate any or all of the Low, Mid 1, Mid 2 or High equalizer bands.
When a band is activated, a corresponding EQ point appears in the EQ curve display.
2.Set the parameters for an activated EQ band.
This can be done in several ways:
• By using the knobs.
• By clicking a value field and entering values numerically.
• By using the mouse to drag points in the EQ curve display window.
By using this method, you control both the Gain and Frequency parameters simultaneously. 
The knobs turn accordingly when you drag points. In addition, if the Mid 1 and Mid 2 bands 
(M1 and M2) are activated there will be two points on each side of the Gain/Frequency 
point that control the width (Q) parameter.
If you press [Shift] while dragging, values can be set in finer increments.
Parameters
Parameter Description
Low Freq
(20-2000Hz)This sets the frequency of the Low band.
Low Gain (+/-20dB) This sets the amount of cut/boost for the Low band.
Low Cut If this button is activated for the Low band, it will act as a Low Cut fil-
ter. The Gain parameter will be fixed.
Mid 1 Freq 
(20-20000Hz)This sets the center frequency of the Mid 1 band. 
Mid 1 Gain 
(+/- 20dB)This sets the amount of cut/boost for the Mid 1 band.
Mid 1 Width 
(0.05-5.00 Octaves)This sets the width of the Mid 1 band, in octaves. The lower this value, 
the “narrower” the bandwidth.
Mid 2 Freq
(20-20000Hz)This sets the center frequency of the Mid 2 band. 
Mid 2 Gain
(+/- 20dB)This sets the amount of cut/boost for the Mid 2 band. 
						

							WAVELAB
33 – 782 Plug-in Processor Reference
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 filtering 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. 
Mid 2 Width 
(0.05-5.00 Octaves)This sets the width of the Mid 2 band, in octaves. The lower this value, 
the “narrower” the bandwidth.
High Freq 
(200-20000Hz)This sets the frequency of the High band.
High Gain (+/-20dB) This sets the amount of cut/boost for the High band.
High Cut If this button is activated for the High band, it will act as a High Cut fil-
ter. The Gain parameter will be fixed.
Output (+/- 20dB) This parameter allows you to adjust the overall output level.
Left/Stereo/Right/
Mono ModesFor stereo signals you can set independent curves for the left and 
right channels by clicking the corresponding button. If the Stereo 
mode is activated, the curve will be applied to both channels. 
When channel independent curves have been set, the left/right chan-
nel curves will be colored green and red, respectively. The currently 
non-selected channel is shown with a dotted curve. If you activate 
Stereo mode after independent curves have been set, the currently 
active curve will be applied to both channels.
Mono mode is automatically activated for mono signals and is other-
wise unavailable. Parameter Description 
						

							WAVELAB
Plug-in Processor Reference 33 – 783
• 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 frequencies are; 
firstly, higher harmonics are most often perceived as “too high”, and se-
condly, their amplitude normally doesn’t follow that of the fundamental in a 
natural way.
• Please note the relation between the Frequency parameter and the har-
monics 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 fre-
quencies starting at 12000Hz.
Another part of this process is giving the added harmonics appropriate 
amplitude curves. The amplitude of the harmonics is based on the ampli-
tude 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 apparent on a psycho-acoustical 
level. To hear what is actually added to the signal, use the Solo button.
Parameters
The parameters are as follows:
Parameter Description
Solo When this is activated, the output will only contain the added Harmon-
ics. The original unprocessed signal will not be heard on the output.
This mode is used as a diagnostic tool to monitor what the current set-
tings 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 frequency lower than this setting will not be af-
fected 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. 
						

							WAVELAB
33 – 784 Plug-in Processor Reference
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 doubled.
The Stereo Echo has the following parameters:
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 set-
ting 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 definitely be avoided. If this happens, please lower the Input level 
and/or Gain.
Meter This allows you to check your adjustments of the Input and Gain con-
trols, so that the signal does not change drastically in level when pass-
ing 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 channels for mono operation, in which case the 
maximum 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 re-
peats.
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 receives both its normal source and the 
complete output of the left channel. Parameter Description 
						

							WAVELAB
Plug-in Processor Reference 33 – 785
StereoExpander
The StereoExpander plug-in narrows or enlarges the stereo 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 applications. 
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 automatically moves the other as well, but you can make separate 
adjustments for the channels by pressing [Alt] and dragging.
The two Phase switches let you invert the phase of the left or right chan-
nel (or both).
The Algorithm buttons let you adjust the stereo sound image. 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 resembles a signal re-
corded 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 micro-
phone, and the ambience using a second microphone positioned 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 assigned to the right 
side and the right channel to the left side.
Volume L The output level of the left channel delay.
Volume R The output level of the right channel delay. Parameter Description 
						

							WAVELAB
33 – 786 Plug-in Processor Reference
Voice Attenuator
This plug-in can be used to remove lead vocals from a recording, to pro-
duce a “karaoke” effect. The principle concept 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 vocal completely, 
without using very complex processing beyond 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 Frequency 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 apply-
ing 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 processors; Auto-
Gate, Compress, AutoLevel, Limit and SoftClip, covering a variety of Dy-
namic Processing functions. The VST Dynamics window is divided into 
five sections, containing controls and meters for each processor. You ac-
tivate the VST Dynamics panel by clicking the “On” button in the lower 
right corner. Once VST Dynamics is activated, you can turn the individual 
processors on and off by clicking on their labels. Activated processors 
have highlighted labels.
You can activate as many processors as you want, but remember that not 
all processors are designed to work together. 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. 
						

							WAVELAB
Plug-in Processor Reference 33 – 787
The following processors are available on the VST Dynamics panel (click 
an item in the list for more information about the corresponding processor):
•“AutoGate” on page 787
•“AutoLevel” on page 789
•“Compress” on page 789
•“SoftClip” on page 790
•“Limit” on page 790
AutoGate
Gating, or noise gating, is a method of dynamic processing that silences 
audio signals below a certain set threshold level. As soon as the signal 
level exceeds the set threshold, the gate opens to let the signal through. 
AutoGate offers all the features of a standard noise gate, plus some very 
useful additional features, such as auto calibration of the threshold set-
ting, a look-ahead predict function, and frequency selective triggering. 
Available parameters are as follows:
Parameter Explanation
Threshold This setting determines the level where AutoGate is activated. Signal 
levels above the set threshold trigger the gate to open, but signal levels 
below the set threshold will close the gate.
Attack This parameter sets the time it takes for the gate to open after being 
triggered. If the Predict button is activated, it will ensure that the gate 
will already be open when a signal above the threshold level is played 
back. AutoGate manages this by “looking ahead” in the audio material, 
checking for signals loud enough to pass the gate.
Hold This determines how long the gate stays open after the signal drops be-
low the threshold level.
Release This parameter sets the amount of time it takes for the gate to close (af-
ter the set hold time). If the “Auto” button is activated, AutoGate will find 
an optimum release setting, depending on the audio program material. 
						

							WAVELAB
33 – 788 Plug-in Processor Reference
Trigger Frequency Range 
AutoGate has a feature that allows the gate to be triggered only by sig-
nals within a specified frequency range. This is a most useful feature be-
cause it lets you filter out parts of the signal that might otherwise trigger 
the gate in places you don’t want it to, thus allowing more control over the 
gate function. The Trigger Frequency Range function is controlled using 
the control in the upper part of the AutoGate panel, and the slider located 
below it. The basic operation of the Trigger Frequency Range function is 
as follows:
1.While playing back audio, drag the slider to the “Listen” position.
You will now monitor the audio signal, and the gate will be bypassed.
2.While listening, drag the two handles in the Trigger Frequency window to 
set the frequency range you wish to use to trigger the gate.
You will hear the audio being filtered as you move the handles.
• Dragging the left handle to the right will progressively cut frequencies starting 
from the low end of the frequency spectrum.
• Dragging the right handle to the left will progressively cut frequencies starting 
from the high end of the frequency spectrum.
3.After setting the frequency range, drag the slider to the “On” position.
AutoGate will now use the selected frequency range as the trigger input.
4.To disable the Trigger Frequency Range function, drag the slider to “Off”.
AutoGate will now use the unfiltered audio signal as the trigger input.
Calibrate Function
This function, activated by using the Calibrate button located below the 
Threshold knob, is used to automatically set the threshold level. It is espe-
cially useful for material with consistent inherent background noise in the 
audio material, like tape hiss for example. This may be masked by the au-
dio content for most of the time, but becomes noticeable during silent 
passages. Use as follows:
1.Find a part of the audio material, preferably not too short, where only the 
background noise is heard. 
If you can find only a short section with background noise, try looping it.
2.Play it back, and click on the Calibrate button.
The button will blink for a few seconds, and then automatically set the threshold so that the 
noise will be silenced (gated) during passages where there is no other signal present. 
						

							WAVELAB
Plug-in Processor Reference 33 – 789
AutoLevel
AutoLevel reduces signal level differences in audio material. It can be 
used to process recordings where the level unintentionally varies. It will 
boost low levels and attenuate high level audio signals. Only levels above 
the set threshold will be processed, so low level noise or rumble will not 
be boosted. If the input level is greater than 0dB, AutoLevel will react very 
fast, because it looks ahead in the audio material for strong signal levels 
and can attenuate levels before they occur, thus reducing the risk of sig-
nal clipping.
Compress
Compress reduces the dynamic range of the audio, so that softer sounds 
get louder or louder sounds get softer, or both. Compress functions like a 
standard compressor with separate controls for threshold, ratio, attack, 
release and make-up gain parameters. 
There is also a separate display that graphically illustrates the compressor 
curve shaped according to the Threshold, Ratio and MakeUp Gain pa-
rameter settings. Compress also features a Gain Reduction meter that 
shows the amount of gain reduction in dB, and a program dependent 
Auto feature for the Release parameter.
Parameter Description
Threshold Only levels stronger than the set threshold will be processed.
Reaction Time 
SwitchThis parameter sets the amount of time it takes for AutoLevel to adjust 
the gain. Set this according to whether the program level changes sud-
denly or over a length of time.
Parameter Description
Threshold This setting determines the level where Compress “kicks in”. Signal lev-
els above the set threshold are affected, but signal levels below are not 
processed.
Ratio Ratio determines the amount of gain reduction applied to signals over 
the set threshold. A ratio of 3:1 means that for every three dB the input 
level increases, the output level will increase by only one dB.
Attack This determines how fast Compress will respond to signals above the 
set threshold. If the attack time is long, more of the early part of the sig-
nal (attack) will pass through unprocessed. 
						

							WAVELAB
33 – 790 Plug-in Processor Reference
SoftClip
SoftClip is designed to ensure that the output level never exceeds 0dB, 
like a limiter. SoftClip, however, acts differently compared to a conven-
tional limiter. When the signal level exceeds -6dB, SoftClip starts limiting 
(or clipping) the signal “softly”, at the same time generating harmonics 
which add a warm, tubelike characteristic to the audio material. SoftClip 
is simplicity itself to use as it has no control parameters. The meter indi-
cates the input signal level, and thus the amount of “softclipping”. Levels 
in the green area (weaker than -6dB) are unaffected, while levels in the 
yellow-orange-red area indicate the degree of “softclipping”. The deep 
red meter area to the right indicates input levels higher than 0dB. 
• Avoid feeding SoftClip with excessively high signal levels as audible distor-
tion may occur, although the output level will never exceed 0dB.
Limit
Limit is designed to ensure that the output level never exceeds a certain 
set output level, to avoid clipping in following devices. Conventional limi-
ters usually require very accurate setting up of the attack and release pa-
rameters, to totally avoid the possibility of the output level going beyond 
the set threshold level. Limit adjusts and optimizes these parameters au-
tomatically, according to the audio material. However, should you want to, 
you can adjust the Release parameter manually.
Release Sets the amount of time it takes for the gain to return to its original level 
when the signal drops below the Threshold level. If the “Auto” button is 
activated, Compress will automatically find an optimum release setting, 
that varies depending on the audio program material.
MakeUp Gain This parameter is used to compensate for output gain loss, caused by 
compression.
Parameter Description
Threshold This setting determines the maximum output level. Signal levels above 
the set threshold are affected, but signal levels below are left unaffected
Release This parameter sets the amount of time it takes for the gain to return to 
its original level when the signal drops below the threshold level. If the 
“Auto” button is activated, Limit will automatically find an optimum re-
lease setting that varies depending on the audio program material. Parameter Description