Thank you for purchasing the ISA 430 MKII brought to you
by the Focusrite team – Ian, Trevor, Peter, Martin, Tom,
Mick A’C, Phil, Chris G, Micky, Pauline, Melissa, Chris W,
Rob J Snr, Simon J, Vernon, Giles, Rob J Jnr, Mick G, Tim,
Dave, Paul and Simon.
The chaps at Focusrite are a jolly hard working bunch and
take a great deal of pride in designing, building and delivering
products which are considered to be the best audio units
around; we hope your new Focusrite unit lives up to that
reputation and that you enjoy many years of productive
recording. If you would like to tell us about your recording
experiences, please email us at: sales@focusrite.com
Please read all of these instructions and save them for
future reference. Follow all warnings and instructions
marked on the unit.
•Do not obstruct air vents in the rear panel. Do not
insert objects through any apertures.
• Do not use a damaged or frayed power cord.
• Unplug the unit before cleaning. Clean with a damp
cloth only. Do not spill liquid on the unit.
•Unplug the unit and refer servicing to qualified service
personnel under the following conditions: if the power
cord or plug is damaged; if liquid has entered the unit; if
the unit has been dropped or the case damaged; if the
unit does not operate normally or exhibits a distinct
change in performance. Adjust only those controls that
are covered by the operating instructions.
•Do not defeat the safety purpose of the polarised or
grounding type plug. A polarised plug has two blades
with one wider than the other. A grounding type plug
has two blades and a third grounding prong. The wider
blade or the third prong are provided for your safety.
When the plug provided does not fit into your outlet,
consult an electrician for replacement of the obsolete
outlet.
WARNING: THIS UNIT MUST BE EARTHED
BY THE POWER CORD.
UNDER NO CIRCUMSTANCES SHOULD THE
MAINS EARTH BE DISCONNECTED FROM
THE MAINS LEAD.
This unit is capable of operating over a range of mains
voltages as marked on the rear panel. Ensure correct mains
voltage setting and correct fuse before connecting mains
supply. Do not change mains voltage settings while mains
supply is connected. To avoid the risk of fire, replace the
mains fuse only with the correct value fuse, as marked on
the rear panel. The internal power supply unit contains no
user serviceable parts. Refer all servicing to a qualified
service engineer, through the appropriate Focusrite dealer.
Power Connections
There is an IEC mains lead supplied with the unit which
should have the correct moulded plug for your country.
The wiring colour code used is:
For units shipped to the USA, Canada, Taiwan and Japan:
Live - Black Neutral - White Earth - Green
For units shipped to any other country:
Live - Brown Neutral - Blue Earth - Green and Yellow
Rear Panel Connections
2
(Optional A/D Card shown fitted)
XLR (Audio) Inputs and Outputs
All 3-pin XLR audio connectors (main and post-mic outputs,
mic, line & ADC inputs, insert sends & returns) are wired as
follows:
Pin 1Screen/chassis
Pin 2Audio 0°
Pin 3Audio 180°
INST. HI Z INPUT
1/4” jack wired as follows:
TipAudio 0°
SleeveScreen/chassis
KEY INPUTS
1/4” jack wired as follows:
TipAudio 0°
RingAudio 180°
SleeveScreen/chassis
DYNAMIC LINK
1/4” jack wired as follows:
TipCompressor sidechain link
RingGate sidechain link
SleeveScreen/chassis
MIC/LINE/INST INPUT
Any one of these inputs may be used as the main input to
the ISA 430 MKII’s main input. Signals routed to these
inputs are referred to as the ‘Internal’ or ‘Int’ signal path.
INSERT RETURN 1
This is used as the insert 1 input point, or as the input to
the EQ modules in ‘EQ Split mode’ (see ‘EQ Split’ diagram
on page 10).
INSERT SEND 1
This is used as the insert 1 output point, or as the analogue
output from the EQ module in ‘EQ Split mode’ (see ‘EQ
Split’ diagram on page 10).
INSERT SEND 2
This is used as the insert 2 output point, or as the analogue
output from the dynamics module in ‘Dyn Split mode’ (see
‘Dynamics Split’ diagram on page 10).
POST-MIC OUTPUT
This is used as an output from the point immediately after
the Phase section (i.e. before EQ and dynamics modules)
allowing direct recording of mic, line or instrument inputs. It
provides a direct-to-tape, ultra-short signal path output.
Taking a signal from the POST-MIC OUTPUT does NOT
interrupt a signal routed from the mic-pre to the EQ,
dynamics etc., so a direct feed to tape can be achieved
whilst simultaneously allowing processing of the same
source.
ADC INPUT 1
ADC Input 1 is used to route an external signal directly to
the optional A/D card via the limiter. When selected using
the ADC Input 1 button on the front panel, ADC Input 1
replaces the internal processed signal being fed to the left
side of the A/D card (i.e. whatever is connected to the mic,
line or instrument input no longer routes to the ADC). For
further detail see ADC Input 1 & 2 on page 13. It also
allows a signal to be summed with the processed signal from
the unit by pressing the Ext Sum switch and using the Ext
Level controls. This signal can be routed to the A/D card
and main output (See ADC Input 1 & 2 on page 13.)
ADC INPUT 2
ADC Input 2 is also used to route an external signal directly
to the optional A/D card via the limiter. When selected
using the ADC Input 2 button on the front panel, ADC
Input 2 replaces the internal unprocessed signal being fed
to the right side of the A/D card (i.e. whatever is connected
to the mic, line or instrument input). This still allows the
processed signal (i.e. whatever is connected to the mic, line
or instrument input) to route to the left side of the A/D
card. Hence there are four possible setups for the ADC 1 &
2 input switches. (See ADC Input 1 & 2 on page 13.)
INSERT RETURN 2
This is used as the insert 2 input point, or as the input to
the dynamics modules in ‘Dyn Split mode’ (see ‘Dynamics
Split’ diagram on page 10).
MAIN OUTPUT
This output is used as the main analogue internal signal
output, and is fed by whatever is connected to the MIC
INPUT, LINE INPUT or INST INPUT, after this signal has
been routed through the processing modules.
DYNAMIC LINK
3
You can connect two ISA 430 MKII units (using a standard
TRS 1/4” jack-to-jack lead between the DYNAMIC LINK
sockets) to allow both the compressor and gate sections to
behave as stereo pairs of processors. When connected in
this way, the compressors and gates behave as single stereo
devices. Both units respond to the higher of the two input
signal path levels. (The EQ channels can be matched visually
or aurally to be used as a stereo pair if required.) Hence, to
drive the compressors and gates as stereo pairs, set the
threshold pot of one of the compressors and one of the
ISA 430 MKII Front Panel Controls
gates to maximum (fully clockwise). The second unit will
then act as ‘master’, allowing one set of parameter changes
to affect both compressors or gates identically. NB: The deessers are not linked.
Retrofitting the Optional A/D Card
The optional A/D card can be retrofitted to a standard ISA
430 MKII at any time. Full fitting instructions for this option
are included with the Card.
Power
Applies power to the unit. Turn on the ISA 430 MKII before
powering up devices to which the outputs are connected.
Inst Input
Unbalanced instrument sources may either be connected via
the rear panel INST. HI Z INPUT, or via this duplicated
front panel jack. If both are connected, the front connection
overrides the rear connection. No DI box is required.
Metering
Meter Source
The VU meter can display input level, Insert 1 return level,
Insert 2 return level, compressor gain reduction or
sidechain Listen level. Press the Meter Source button to
step through Input, Insert Rtn 1, Insert Rtn 2 and Comp
sources as indicated by the corresponding LEDs. The
sidechain Listen level will automatically be displayed on the
VU meter when the Listen switches are pressed in the
compresser, gate or de-esser sections. The Listen LED will
also light when any of these switches are selected.
Meter 0VU Calibration
The metering calibration for the input level and insert
return levels can be shown over two different ranges:
• 0VU corresponds to +4dBu.
• 0VU corresponds to +18dBu.
Either can be selected via the Meter 0VU Calibration
button. For the compressor, the meter indicates the
amount of gain reduction applied, from 0VU (no
compression) to -20VU (corresponding to 20dB of gain
reduction). NB: The effect upon ‘Listen’ of calibration is
similar to that on input and insert returns. However, as the
signal being monitored by the listen circuit is a sidechain
source, the meter acts more as a visual indication of the
attack and decay of a signal, rather than true level.
Listen LED
This illuminates when Listen is selected on the compressor,
expander/gate or de-esser, and indicates that the unit is
monitoring the selected sidechain frequencies. N.B. When
attempting to listen to the dynamics in ‘Split Dynamics’
mode, pressing the Listen switch will only result in visual
monitoring; as the dynamics are split from the monitoring
path, they can be viewed in the meter but not heard.
Audio O/L LED
This LED illuminates when the peak signal level reaches or
exceeds +20dBu, or when the peak signal level reaches 6dB
below the clip point. The signal is monitored at five points:
post-the input gain Trim, post-Insert 1, post-the EQ
3
module, post-Insert 2, and post-the Dynamics module (since
4
each module could cause clipping if incorrectly set up).
Occasional short-duration peaks which may cause the LED
to blink will not normally cause audible distortion, but if the
LED is lit constantly, the level in the appropriate module
should be reduced to prevent overloading.
Digital Output Meters
legend). The gain range is split between two gain modes
depending upon the status of the 30-60 switch:
Mode 1: Mic Gain Range 0-30
With the 30-60 switch off, the rotary gain knob operates
over a gain range of 0dB to +30dB, the level of gain chosen
being indicated on the front panel by the outer arc of yellow
numbers around the gain knob.
Two LED bargraph meters
monitor the level being fed
to the output (left meter)
and input (right meter) of
the unit.
They also display the
channels of the optional
ADC card when the ADC
Input 1 and ADC Input 2
switches are selected. The
meters provide a wide
range from –42dBFS (20dBu) to 0dBFS (+22dBu).
The output meter monitoring point is just before the ADC
input; this means that if the limiter is switched in the action
of the limiter can be seen on the meter. This monitor point
is also post-the output level control. When ADC Input 1 is
selected the meter is fed from ADC Input 1. The input
meter monitoring point is post-the phase switch. When
ADC Input 2 is selected the meter is fed from the ADC
Input 2.
Input Stage
Three input options are
provided to give
compatibility with mic,
line or instrument
sources. An immediatelypost-input-stage, balanced
output (POST-MIC
OUTPUT) is provided on
the rear panel giving an ultra-short signal path to allow for
the cleanest possible recordings. N.B. The POST-MIC
OUTPUT can be used in conjunction with the main output,
allowing the user to record a dry signal for archive/safety
purposes, at the same time as recording the processed
output.
Select
Pressing Select steps through each of the three inputs as
indicated by the corresponding LEDs. When the Mic LED is
lit, the mic input is active etc. Only one of the Mic, Line or
Instrument inputs can be used.
Mic Input Gain
With the mic input selected, the user has access to the full
gain range in 10dB steps from 0dB to +60dB (yellow
Mode 2: Mic Gain Range 30-60
With the 30-60 switch on (illuminated), the rotary gain
knob operates over a gain range of 30dB to 60dB, the level
of gain chosen being indicated on the front panel by the
outer arc of yellow numbers around the gain knob.
An additional 20dB of gain can be applied to the signal after
the Mic/Line gain knob using the Trim knob. See Trim
control text below for full explanation.
Line Input Gain
With the line input selected, the user has access to gain
settings ranging from –20dB to +10dB, indicated on the
front panel by the arc of white numbers around the gain
knob. The 30-60 switch is inactive when the line input is
selected, as the gain range for Line level inputs is restricted
to –20dB to +10dB in 10dB steps.
An additional 20dB of gain can be applied to the signal after
the Mic/line gain knob using the Trim knob. See Trim
control text below for full explanation.
Instrument Input Gain
With the instrument input selected, gain is applied to the
input signal by using the Trim control only, which allows
+10dB to +40dB of gain range. The level of gain chosen is
indicated on the front panel by the outer arc of yellow
numbers around the gain knob. This input is suitable for
high impedance sources such as guitar or bass pickups
(which may be connected directly without the need for an
external DI box), or vintage synthesizers with high
impedance outputs.
Trim
The Trim control provides additional variable gain of 0dB to
+20dB when mic or line inputs are selected. The level of
gain chosen is indicated on the front panel by the inner arc
of white numbers around the gain knob. The additional
20dB of gain that can be applied to the Mic or Line signal is
very useful for two reasons:
When high gain is required
The trim used in conjunction with the Mic gain of 60dB will
give a total of up to 80dB of pre-amp gain, making it very
useful for getting good digital recording levels from very low
output devices, such as dynamic and ribbon microphones.
Gain adjustment during recording
When small amounts of gain adjustment are needed to
correct for performance level variations during recording,
use the trim knob rather than the stepped Mic/Line gain
knob, as switching the 10dB gain steps will be much too
intrusive. It is therefore good practice to apply some Trim
5
gain before using the 10dB stepped gain knob to find the
optimum recording level so that the Trim control can be
used to gently add or take away gain later, if so required.
+48V
Pressing the +48V switch provides +48V phantom power,
suitable for condenser microphones, to the rear panel XLR
microphone connector. This switch does not affect the
other inputs. If you are unsure whether your microphone
requires phantom power, refer to its handbook, as it is
possible to damage some microphones (most notably
ribbon microphones) by providing phantom power.
Phase
Pressing Phase inverts the phase of the selected input. This
is primarily used to correct phase problems when using
multiple microphones on a single source.
EQ Module
Impedance
Pressing Impedance steps through each of the four
transformer pre-amp input impedance values, as indicated
by their corresponding LEDs. By selecting different values
for the impedance of the ISA 430 MKII transformer input,
the performance of both the ISA 430 MKII pre-amp and the
microphone connected can be tailored to set the desired
level and frequency response.
Mic Air
Pressing Mic Air increases the impedance effect of the
transformer on high frequencies. Transformers impart an
effect which is often referred to as ‘air’, adding a
spaciousness to the sound of the mic pre-amp. Mic Air
further emphasises this effect. For further deails see FAQ
number 18 on page 22.
All EQ
Pressing All EQ activates all sections of the EQ module
(including the Hi and Lo Pass Filters), placing the whole
module in the audio path. Toggling All EQ allows an A/B
comparison of EQ’d/flat settings, without having to switch
each section of the EQ individually, and without using
Bypass which switches both EQ and dynamics modules in or
out of the audio path. The Comp and Gate switches on
individual EQ sections (see below) operate independently
from the All EQ switch.
Filter In
Press Filter In to make the Hi and Lo Pass Filters active in
the audio path. This selection is cancelled if Comp or Gate
is pressed. Both filters provide 18dB/octave roll-off, and
since the filter frequencies overlap they may be configured
as a very tight bandpass filter for creative compression and
gating; use when you wish to select a specific instrument or
narrow frequency band from a complex signal, then feed to
the sidechain of the compressor or gate.
Low Pass Filter
Sets a roll-off frequency from 400Hz to 22kHz.
High Pass Filter
Sets a roll-off frequency from 20Hz to 1k6Hz.
Comp
When the Comp switch is pressed, both Hi and Lo pass
filters are inserted in the compressor sidechain to enable
frequency-selective (aka frequency-conscious) compression.
This selection is cancelled if Filter In or Gate is pressed.
Gate
When the Gate switch is pressed both filters are inserted in
the gate sidechain to enable accurate drum gating. This
selection is cancelled if Filter In or Comp is pressed.
Parametric EQ
Two separate bands of parametric EQ, hi-mid and low-mid,
are provided, each with continuously variable boost/cut
with centre detent, sweep control with two ranges, and
fully variable Q. The first band covers the range 40Hz to
400Hz (120Hz to 1k2Hz when x3 is pressed) and the
second band covers 600Hz to 6kHz (1k8Hz to 18kHz when
x3 is pressed).
Shelving EQ In
6
Press in to switch the shelving EQ into the signal path. This
selection is cancelled if Gate or Comp is pressed.
Hi Range
The frequency controls have two ranges, the higher being
selected when the Hi Range switch is pressed (frequencies
shown in yellow on the panel).
Param EQ In
Press in to switch the parametric EQ into the signal path.
This selection is cancelled if Gate or Comp is pressed.
x3
The Sweep controls have two ranges, the higher being
selected when the x3 switch is pressed ( x3 frequencies are
shown in yellow on the panel).
Comp
When the Comp switch is pressed, the parametric EQ is
inserted into the compressor sidechain, allowing frequencyselective (frequency-conscious) compression. This selection
is cancelled if Param EQ In or Gate is pressed.
Gate
When the Gate switch is pressed, the parametric EQ is
inserted into the gate sidechain for e.g. accurate drum
gating. This selection is cancelled if Param EQ In or Comp is
pressed.
Shelving EQ
High and low frequency-shelving sections are available, each
with continuously variable boost/cut with center detent, and
a four position rotary switch for selection of roll-off
frequency. In addition, the Hi Range switch allows for two
ranges of roll-off frequency per band – resulting in eight
frequency selections in total.
LOW SHELF
33
HI RANGEHI RANGE
20
460
0Hz
HIGH SHELF
655
2k2
1k5
18k
15k
Low frequency shelving
The frequency range steps are 20Hz, 56Hz, 160Hz and
460Hz (33Hz, 95Hz, 270Hz and 655Hz when Hi Range
switch is engaged).
High frequency shelving
The frequency range steps are 1.5kHz, 3.3kHz, 6.8kHz and
15kHz (2.2kHz, 4.7kHz, 10kHz and 18khz when Hi Range is
engaged).
Comp
When the Comp switch is pressed, the shelving EQ is
inserted into the compressor sidechain, allowing frequencyselective (frequency-conscious) compression. This selection
is cancelled if Shelving EQ In or Gate is pressed.
Gate
When the Gate switch is pressed, the shelving EQ is
inserted into the gate sidechain for e.g. accurate drum
gating. This selection is cancelled if Shelving EQ In or Comp
is pressed.
Compressor
Comp In
Press Comp In to switch the compressor into the signal
path. Note that the VU meter can be selected to display the
compressor gain reduction (see Metering section, page 3).
Listen
7
Press to allow audio monitoring of the compressor
sidechain to assist accurate frequency adjustment when
setting up frequency-conscious compression. The Listen
LED below the main VU meter illuminates to show that
Listen mode is active. Note that the VU meter automatically
displays the compressor sidechain listen level when a Listen
switch is pressed in (see Metering section, page 3).
Vintage
The Vintage switch activates the vintage compressor. This
has two modes, Comp and Lim, as indicated by the LEDs to
the right of the Vintage switch. The Vintage switch steps
through the different modes of compressor operation.
In Comp mode the compressor is a vintage opto
compressor. In Lim mode the compressor acts more like a
vintage opto limiter with a harder ratio and knee. When
neither the Comp or Lim LEDs are illuminated the
compressor is in the classic Focusrite discrete Class A VCA
mode.
N.B. When in Vintage Comp or Lim mode the Attack,
Release, Auto Release and Ext Key controls are inactive.
Attack
Attack determines how quickly compression is applied once
the level of the source signal has risen above the threshold.
When turned anticlockwise the response is very fast, which
tends to make the compressor react to the peak levels of
the signal. This is sometimes desirable, but can cause
unwanted “pumping” of steadier low level components of
the signal by short transients. A slower attack will cause the
compressor to ignore short transients and respond more to
the average loudness of the signal; however this may seem
to increase relative volume of the transients.
Makeup
Compression results in an overall reduction in level. The
Makeup control allows you to restore the compressor’s
output level back to the original level.
Ratio
The Ratio control determines the rate at which
compression is applied to the signal with increasing input,
and is the ratio of change in input level compared to change
in output level. The control gives a range of 1.5 to 10.
Higher ratio settings will produce more noticeable
compression, so for the least obtrusive result, the ratio
should be set at the minimum necessary for the application.
For example, using low threshold and low ratio will produce
a less noticeable effect than a high threshold and high ratio,
even though the total amount of compression may be the
same.
Release
Release determines how quickly compression is removed
once the level of the source signal has fallen below the
threshold. When in the anticlockwise position, the
compression releases very quickly, which may be
appropriate on rapidly varying signals to avoid compressing
the beats that follow, but can result in excessive distortion
on more sustained material. Clockwise rotation increases
the release time, giving a smoother effect, but which at the
same time may result in transients causing audible
“pumping”.
Theshold
Threshold determines the level at which compression
begins, with a range of -28dB to +12dB. The lower the
threshold, the more the signal is compressed. Setting a
higher threshold allows quieter passages in music or speech
to remain unaffected; only passages that exceed the
threshold will be compressed.
in the key input signal by adding gain to the sibilant
8
frequencies. This will cause those frequencies to be heavily
compressed in the original signal. The ISA 430 MKII’s
dedicated de-esser can take care of primary de-essing
duties, whilst key input compression deals with a secondary
problem occurring at a separate frequency.
Blend and Mix
Press the Blend switch to activate the blend circuit and
make the Mix control active. The Mix control allows the
user to mix a blend of uncompressed signal and compressed
signal (from totally uncompressed to fully compressed).
Equal amounts of compressed and uncompressed signal are
present when the mix control is set to its mid point.
Auto Release
Press the Auto Release switch to set the release rate to
auto, substituting an adaptive attack/release circuit which
varies the release rate to suit the dynamics of the signal.
This enables the use of fast attack times without any
“pumping” type artefacts, especially effective on complex
programme material. The release rate is probably the most
important variable when recording rock/pop music, since it
directly affects loudness. Loudness is increased by the
maintenance of high mean levels: compression increases the
proportion of high level signal content, and as the diagram
shows, the faster the unit releases, the more low-level signal
is brought to a higher level. Therefore, the faster the
release rate, the higher the perceived loudness of the
recording.
Ext Key
Pressing Ext Key switches control of the compressor to an
external signal fed into the rear panel COMP KEY INPUT
jack socket. Note that the key input is not output from the
unit (it is the original input that is compressed and output)
but the compression is applied as if the key input signal were
being compressed. N.B. Ext Key is not available in Vintage
mode.
Example: one of the problems in compressing a mixed
programme is that gain reduction tends to be controlled by
one dominant instrument or sound. For more natural
compression, you need to attenuate the sound of the
dominant instrument, but this is probably not acceptable
since it would affect the mix.
N.B. When attempting to use the compressor blend and the
gate simultaneously, the gate will be less effective. This is
because the original input signal and the processed (gated)
signal are being blended, so the more input signal there is,
the less audible the effect of the gate will be.
Gate
Gate In
Press Gate In to switch the gate into the signal path.
Expand
Pressing Expand causes the gate to function as an expander,
which has a similar effect to gating, but instead of cutting off
any signal below the threshold, it proportionately decreases
it (see diagram below). This may give a more natural sound
when recording non-percussive sources. Pressing Expand a
second time causes the switch to revert to being unlit, and
causes the section to act as a gate once more. The adjacent
LED meter indicates in dB the amount of gain reduction
applied by the expander/gate, and provides additional visual
indication of the effect of the Range control (see below).
Using the key input allows a second feed of the input signal
to be routed through an equaliser to attenuate the
dominant instrument. That signal is then fed into the key
input. Now you can use the attenuated key input to control
the compression of the original signal. The original signal is
compressed as if the dominant instrument had been
attenuated. This technique can be very useful when
compressing bass-heavy dance music. By attenuating the
bass in the key input signal, the bass in the original retains
more of its dynamics.
Another application of this technique is to create a second
de-esser. When de-essing, sibilant frequencies need to be
heavily compressed. To achieve this, accentuate the sibilants
Listen
Press to allow audio monitoring of the gate sidechain to
enable accurate frequency adjustment during setup. The
Listen LED below the main VU meter illuminates to show
that Listen mode is active. Note that the VU meter
automatically displays the gate sidechain listen level when a
9
Listen switch is pressed in (see Metering section, page 3).
Range
Range determines how much the signal is attenuated when
the gate is closed. The gate can be set up to cut (80dB
attenuation, control clockwise) or to more gently lessen
attenuation (min 0dB). Maximum attenuation may give an
unnatural sound, so keep Range at a low value unless it is
essential to reduce high levels of background noise, or
unless an obviously gated effect is desired.
Threshold
Threshold determines the level at which the gate opens (or
at which gain reduction finishes when in Expander mode).
The higher the threshold, the more low-level noise is
reduced, and the more extreme the effect.
De-Esser
The de-esser uses an optical design, allowing the user to
remove excessive sibilance from a vocal performance
(where “ess” sounds are over-emphasised).
De-Ess In
Press in to activate the De-Esser.
Hold
Hold controls the variable delay before the gate release
starts. This allows the gate to be held open until the signal
has decayed sufficiently, so that the rapid onset of gain
reduction isn’t noticeable. Alternatively the signal can be
deliberately truncated before its natural end to create a
special effect.
Release
This control sets the Release time, the rate at which the
gate attenuation increases, fading out the signal. This release
period begins as soon as the signal drops below the
threshold. On transient signals, a fast release will be
appropriate (control anticlockwise), but with other material,
a slower release (control clockwise) may give a more
natural sound. Ideally the release needs to be slightly slower
than the natural decay rate of the signal to avoid audibly
cutting it short.
Fast Attack
Fast Attack determines how quickly the gate opens once
the level of the source signal has risen above the threshold.
When the switch is pressed, the response is fast, which may
be necessary on some signals to avoid “missing” an initial
transient, but which could also introduce an undesirable
click on smooth, sustained sounds when using a high
threshold setting. On such signals a slower attack (switch
released) may give a more natural sound.
Hyst
Press to introduce wider hysteresis. This increases the level
difference between the gate switching on and off, which
prevents the gate from oscillating (‘chattering’) with
particular combinations of input signal and threshold setting.
This function is particularly useful when gating a signal with
a very long decay time and large amounts of level
modulation (e.g. a grand piano).
Ext Key
Pressing Ext Key switches control of the gate to an external
signal on the rear panel GATE KEY INPUT jack.
Threshold
Threshold determines how much de-essing is being applied
to the selected frequency. The lower the threshold (control
anticlockwise) the more de-essing is applied.
Freq
This control selects the frequency to be removed; between
2k2Hz and 9k2Hz.
De-Ess Listen
Press to allow monitoring of only those signals which will
trigger activation of the de-esser, rather than hearing the
overall effect in a complex full bandwidth signal. The Listen
LED below the main VU meter illuminates to show that
‘Listen’ mode is active. Note that the VU meter
automatically displays the de-ess listen level when the Listen
switch is pressed in (see Metering section, page 3).
Active LED
The Active LED illuminates when the de-esser is active at
the chosen frequency, and shines more brightly with
increasing level reduction.
Setting up the De-Esser
Press De-Ess Listen with Threshold at maximum and slowly
reduce until the selected frequency begins to trigger the deesser. Vary the frequency control to find the exact area of
the signal which you wish to remove. Once located, switch
off De-Ess Listen and adjust Threshold for the amount of
reduction required. No further adjustment of Freq should
be required, as the hot spot will have been precisely found
using De-Ess Listen.
Output
Ext Sum
Press to route a sum of both the
internal signal and external line input
signal (from the EXT SUM INPUT) to
the Main Output. External signals could
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