T-RackS 6

MODULE PROCESSORS MANUAL

T-RackS 6 Module Processors

T-RackS 6 is available in different versions.

The difference between the different versions is in the number of
gear models included and in the features available.
While consulting this manual, please refer to the specific gear models
and features available in the T-RackS 6 version you are using.

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Table of Contents
Contents3
1 – Vintage Tube Compressor/Limiter model 6707
2 – EQP-1A Vintage Tube Program Equalizer9
3 – Opto Compressor11
4 – Brickwall Limiter13
5 – Linear Phase EQ15
6 – Classic Compressor17
7 – Classic Multiband Limiter19
8 – Classic Clipper21
9 – Classic Equalizer23
10 – Black 76 Limiting Amplifier25
11 – White 2A Levelling Amplifier27
12 – British Channel29
13 – Quad Comp33
14 – Quad Lim35
15 – Quad Image37
16 – De-Esser39
17 – CSR Plate Reverb41
17.1 – Easy42
17.2 – I/O Levels42
17.3 – Time42
17.4 – Reverb43
17.5 – Color43
17.6 – Reflections43
17.7 – Echo44
17.8 – Modulation Matrix44
17.8.1 – Modulation45
17.8.2 – Sources (modulation sources)45
17.8.3 – Destinations46
17.9 – MACROS47

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17.9.1 – MATRIX47
17.9.2 – Sources (modulation sources)48
17.9.3 – Destinations48
17.10 – MACRO NAME50
18 – CSR Hall Reverb51
18.1 – Easy52
18.2 – I/O Levels52
18.3 – Time52
18.4 – Reverb53
18.5 – Color53
18.6 – Reflections53
18.7 – Modulation Matrix, MARCOS and MACRO NAME54
19 – CSR Room Reverb55
19.1 – Easy56
19.2 – I/O Levels56
19.3 – Time56
19.4 – Reverb57
19.5 – 4.19.5 - Color57
19.6 – Reflections57
19.7 – Modulation Matrix, MARCOS and MACRO NAME57
20 – CSR Inverse Reverb58
20.1 – Easy59
20.2 – I/O Levels (Advanced Mode)59
20.3 – Time59
20.4 – Reverb60
20.5 – Color60
20.6 – Reflections60
20.7 – Modulation Matrix, MARCOS and MACRO NAME61
21 – White Channel62
22 – Bus Compressor66
23 – Tape Echo (EchoFlex)68
24 – Precision Comp/Limiter70
25 – Master EQ 43272
26 – EQ 7374

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27 – EQ 8176
28 – EQ PA78
29 – EQ PB80
30 – EQ PG82
31 – Stealth Limiter84
32 – Mic Room87
33 – Saturator X89
34 – ONE92
35 – Dyna-Mu93
36 – EQual94
37 – Master Match X97
38 – Leslie®99
39 – Tape Machine 24104
40 – Tape Machine 80106
41 – Tape Machine 440108
42 – Tape Machine 99110
43 – Sunset Sound Studio Reverb112
44 – Space Delay115
45 – Comprexxor118
46 – Teac A-6100 MKII120
47 – Teac A-3340S122
48 – TASCAM 388124
49 – TASCAM PORTA ONE126
50 – FAME Studio Reverb128
51 – Joe Chiccarelli Vocal Strip131
52 – The Farm Stone Room134
53 – Bass One137
54 – Dual Spring139
55 – Lo-Fi Punch141
56 – Delay Lab143

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57 – Pusher145
58 – Filter Fusion147
59 – Channel Strip X149
60 – Support151
60.1 – IK Product Manager151

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T-RackS 6 Module Processors

1 – Vintage Tube Compressor/Limiter model 670

This processor is based on the “Holy Grail” of compressors/limiters, the Fairchild® 670. With a faithful
reproduction of every control, this is an incredibly accurate model that captures every nuance of one of the best
Fairchild units available.

Controls:

• AGC: sets the compressor channels mode. Set it to Left/Right for unlinked stereo (independent channels)
operation. Set it to Link for linked stereo operation (both channels will have identical settings except for
Input Gain, and the same amount of gain reduction will always be applied to both channels to ensure stereo
image stability. Set it to Lat/Vert to have the compressor two channels to work on Mid and Side portions of
the Stereo signal. With this mode you’ll be able to process independently your Center (Lat) and Sides (Vert)
image of the stereo program.
• LEFT-LAT DC Threshold: exactly like on the original unit, set it fully counterclockwise for maximum softness
of the compression knee; Set it fully clockwise for maximum hardness of the compression knee.
• RIGHT-VERT: same as LEFT-LAT DC Threshold for right channel.
• LEFT-LAT TIME CONSTANT: sets the compressors release time. Positions 1 from 4 are single time
constant, from quick to slow, positions 5 and 6 are dual time constants and are especially useful for mixed
stereo material.
• RIGHT-VERT TIME CONSTANT: same as LEFT-LAT TIME CONSTANT for right channel.
• LEFT CHANNEL INPUT GAIN: sets the compressor input gain on Left channel. When no compression is
happening set this knob to 10 for unity I/O gain.
• RIGHT CHANNEL INPUT GAIN: same as LEFT CHANNEL INPUT GAIN, for right channel.
• LEFT-LAT THRESHOLD: sets the amount of compression. Fully counterclockwise means no compression,
fully clockwise give the maximum possible amount of compression for that signal.
• RIGHT-VERT THRESHOLD: same as LEFT-LAT THRESHOLD for right channel.

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• IN/OUT/GR METERS: when set to IN the VU meters will indicate the audio level at the inputs of the
compressor. When set to OUT they will indicate the audio level at the outputs of the compressor. The most
important indication is GR (default) where the amount of compression in dB is shown. Always use the GR
position when dialing the desired compression amount with THRESHOLD knobs.
• OUTPUT: sets the output level within a +/- 15 dB range.
• BYPASS: completely bypasses the whole unit.
• RESET: brings the compressor back to its default state where a compression of around 2 dB is happening
on moderate level mixes.

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2 – EQP-1A Vintage Tube Program Equalizer

This processor is based on what is universally regarded as one of the best and most musical program EQs ever
made, the Pultec® EQP-1A. Extremely accurate modeling of two real world units known to have “the sound”
with sonic performance that is indistinguishable from the originals. Stereo or MS operation.

Controls:

• LOW FREQUENCY: sets the frequency intervention point for the low shelving boost and atten filters: 20,
30, 60 or 100 Hz. Please note that because the Pultec’s original circuit was based on a passive equalization
network the boost and atten filters will not cover the same frequency span and will not cancel one each
other if set identically.
• LOW FREQUENCY BOOST: applies a low frequencies boost with a shelving shape on the selected
frequency range. Set it to 0 for no boost, 10 for max boost.
• LOW FREQUENCY ATTEN: applies a low frequencies attenuation with a shelving shape on the selected
frequency range. Set it to 0 for no attenuation, 10 for max attenuation.
• BANDWIDTH: widens (clockwise position) or narrows (counterclockwise position) the range of frequencies
covered by the high frequency bell curve. This control does not affect the sound of the low frequency filter
nor the one of the high frequency attenuation.
• HIGH FREQUENCY: sets the center frequency of the high frequency bell boost filter: 3, 4, 5, 8, 10, 12, 16
kHz. (The high frequency boost and attenuation filter have independent frequency settings).
• HIGH FREQUENCY BOOST: applies high frequency boost with a bell shape, centered on the selected
frequency. Set it to 0 for no boost, 10 for max boost.
• HIGH FREQUENCY ATTEN: attenuate high frequencies with a shelving shape starting from the selected
value on ATTEN SEL. Set it to 0 for no attenuation, 10 for max attenuation.
• ATTEN SEL: sets the frequency intervention point for the high frequency attenuation filter: 5, 10 or 20 kHz.
• OUTPUT: sets the global equalizer gain, from -15 to + 15 dB.

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• ON/OFF: completely bypasses the unit when on the OFF position.

• RESET: resets the equalizer to its initial, flat, state.
• L=R: linked channels mode. Set it this way when you want to control both left and right channels at the
same time, by applying the exact same equalization to them.
• L/R: click on L or R icons when you want to tweak the equalization independently on the Left or Right
channels.
• M/S: this option uses a Mid-Side matrix to insert the equalizers on your processing chain. This means that
you will be able to equalize independently the Center or the Sides of your stereo program. The L and R
icons will be replaced by M and S when in this mode. Click on M to equalize the Center (Mid) of your stereo
image, or on S to equalize the Sides.

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3 – Opto Compressor

Optical compression has a special sonic character that true audiophiles appreciate. This analog-modeled
module faithfully recreates that particular (and extremely musical) “movement” that characterizes only the best
opto compressors.

Controls:

• INPUT: sets the input level of the compressor, from -24 to +24 dB.
• RATIO: sets the Ratio of the compressor. Use low values for gentle and invisible mastering applications and
higher values for more powerful and obvious compression. Ratio range is from 1:1 (no compression) to 30:1
(almost limiting).
• ATTACK: sets the compressor attack time from 0 to 50 ms. Extremely fast attack values could deliver
slight distortion on certain sounds.
• RELEASE: sets the compressor release time from 30 ms to 5 seconds.
• COMPRESSION: sets the amount of compression that is applied to the signal. Fully counterclockwise
means no compression, fully clockwise means full compression.
• IN/OUT/GR METERS: when set to IN the VU meters will indicate the audio level at the inputs of the
compressor. When set to OUT they will indicate the audio level at the outputs of the compressor. The most
important indication is GR (default) where the amount of compression in dB is shown. Always use the GR
position when dialing in the desired compression amount in with the COMPRESSION knob.
• OUTPUT: sets the compressor output level within a range of +/- 24 dB.
• BYPASS: completely bypasses the opto compressor module.
• RESET: brings the compressor back to its default state where a compression of around 2 dB is happening
on moderate level mixes.
• L=R: the same settings will be applied to L and R compressors. The L and R compression will be exactly the
same if the LINK button is ON, to avoid stereo image shifts.

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• L/R: click on L or R icons if you want to dial in specific compression setting for L and R channels,
independently.
• M/S: this option uses a Mid-Side matrix to insert the stereo compressor on your processing chain. This
means that you will be able to compress independently the Center or the Sides of your stereo program. The
L and R icons will be replaced by M and S when in this mode. Click on M to set the compression for the
Center (Mid) of your stereo image, or on S to set compression for the Sides.

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4 – Brickwall Limiter

This Brickwall Limiter can really make your masters loud without squashing them to a lifeless sound. Accurate
and transparent, digital brickwall limiting doesn’t get any better than this. Selectable algorithms allow you to
match the best limiting style for each program type.
IMPORTANT: this is the processor that should be always used at the end of your T-RackS mastering chain
because it has special features to keep the final output level of your master under control.

Controls:

• GAIN REDUCTION: shows the amount of limiting that is happening in the gain reduction element of the
limiter. Limiting obtained by saturation or clipping is NOT shown by this meter.
• INPUT: sets the input level for the limiter, and therefore the volume increase the limiter will give.
• ATTACK TIME: sets the limiter attack time. Depending on how much limiting you’re applying it may be
necessary to increase this value to preserve percussive material’s sense of impact.
• RELEASE TIME: sets the limiter release time. The longer the release time the softer the limiter will be. Short
and very short release times will make the limiter more suitable for rock, R&B, pop etc, while longer release
times will make the limiter more suitable for music that is generally mastered quieter, like jazz, classic, fusion
etc.
• STYLE: sets the style the limiter will use. Options are:
• Clean: this is the cleanest and most transparent style, useful for genres where the complete absence of
distortion is a must.
• Advanced 1, 2, 3, 4: these styles uses a “mix” between warm and smooth saturation and digital limiting.
All offer more punch and color than “Clean”.
• Sat 1, 2, 3: these styles use various forms of saturation as limiting. The Gain Reduction meter will not
show reduction because the gain reduction element is not working on these styles.
• Clipping: straight digital 0 dBFS clipping, no gain reduction shown.

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• OUTPUT CEILING: sets the top level the audio will never be allowed to exceed. Typically set to -0.1 or 0.2
dB. If “D/A Distortion Protection” is selected in the Preferences this can be left to 0 dB.
• OUTPUT: peak indication of the limiter output audio level.
• BYPASS: completely bypasses the Brickwall limiter.
• RESET: brings the limiter back to its default state.
• L=R: click on the L or R icons if you want to tweak the limiter’s L or R channels independently.

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5 – Linear Phase EQ

This is accurate, high definition, high precision EQ that can be trusted for delicate Mastering tasks with 6
completely identical bands that can be set to all available filter types and that can span the whole spectrum
range (any kind of overlap is possible). Switchable between minimum and linear phase characteristics. Stereo or
MS operation.

Controls:

• GAIN/RES 1-6: sets the gain from -15 to +15 dB when the band is set to Peaking or Shelving. When the
band is set to High or Low pass filters this knob will control the resonance.
• FREQ 1-6: sets the center or cutoff frequency for the band, all bands can span from 10 Hz to 20 kHz.
• Q 1-6: sets the bandwidth of the Peaking filters, Q ranges from 0.2 to 20. When bands are set to Shelving
this knob will set the shape of the shelving filter from gentle to steep and resonant, spanning from 0.2 to 3.
• BAND ON/OFF: each band can be turned On or Off independently. All bands are Off at startup. When
the band is ON the button is green.
• BAND FILTER TYPE: each band can be set to work as a Peaking filter (default for bands from 2 to 5),
Shelving filters (defaults for bands 1 and 6) and high or low pass filers.

• OUTPUT: sets the equalizer output level, from -15 to +15 dB.
• OUTPUT METER: peak meter for the equalizer output audio signal.
• LINEAR PHASE: sets the phase characteristic of the equalizer. When Off, the equalizer performs with

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minimum phase characteristics, when On the equalizer is completely linear phase.

• CURVE DISPLAY: this display shows the resulting equalization curve for both channels. Individual bands EQ
points can be dragged to set Gain and Frequency points without using the equalizer knobs.
• BYPASS: completely bypasses the whole equalizer.
• RESET: brings the equalizer back to its default, flat, state.
• L=R: linked channels mode. Set it this way when you want to control both left and right channels at the
same time, by applying the exact same equalization to them.
• L/R: click on L or R icons when you want to tweak the equalization independently on the Left or Right
channels. Left channel curve will be displayed in green, right channel in red.
• M/S: this option uses a Mid-Side matrix to insert the equalizers on your processing chain. This means that
you will be able to equalize independently the Center or the Sides of your stereo program. The L and R
icons will be replaced by M and S when in this mode. Click on M to equalize the Center (Mid) of your stereo
image, or on S to equalize the Sides.

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6 – Classic Compressor

The T-RackS classic tube compressor emulates that classic, analog, hi-end vintage gear used in mastering
applications. This will give you that big, warm sound typical of highly acclaimed hardware devices. This
compressor has some unique features that make it very flexible but always very musical and creative.

Controls:

• SIDECHAIN HPF: this control adds a high pass filter to the detector stage of the compressor. The
detector stage is where the compressor analyzes the incoming signal to determine how gain reduction
must be applied. This variable high pass filter prevents, when turned up, that compression pumping effect
mainly caused by the material’s low end content. The higher this knob the softer and more inaudible the
compression character will be. At the minimum or at very low settings the compression will be much more
aggressive and noticeable. Use lower settings when you want to “hear” the compression effect and use
higher values for a gentle compression, as you may want in acoustic mixes.
• ATTACK TIME: adjust this control to change the speed of compression in the attack phase. With higher
values, transients will pass unaltered through the compressor. With lower values the gain reduction will
respond quicker and transients will be more affected. Value range is from 15 ms to 80 ms. While adjusting,
check the release time value above the knob to see the attack time value you’re setting.
• RATIO: use this control to define the strength of the compression. Lower values give a gentler and less
noticeable compression effect, more suitable for 2-tracks mixes and acoustic sounds. Higher values give
a more effective compression effect with, at times, all the side-effects you might expect from a over-
compression process. These effects are sometimes desirable, that’s why T-RackS’ compressor ratio control
sweeps up to a value of 5. If you are not sure which ratio control value is suitable for your needs, start setting
this control at 2 - 3.
• RELEASE TIME: adjust this control to change the speed of compression in the release phase. Using higher
values, the recovery time will be longer and the compression will be less noticeable. Using lower values, the
recovery time will be shorter and the average loudness will be higher. Value range is from 70 ms to 1.5 s.

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• INPUT DRIVE: this knob controls the amount of signal that drives the compressor’s input. This is no
threshold point leveller so the amount of compression is determined by the strength of input signal. Range is
from -18 dB to +18 dB.
• STEREO ENHANCEMENT: this control will affect the stereo imaging of your mix. Turning it up you will
increase the stereo width and vice versa. While adjusting this knob check the value you’re setting (in dB).
Value range is from -5 to +5 dB, where 0 dB has no effect.
• GAIN REDUCTION VU: this analog VU shows the amount of Gain reduction in dB. When compression
does not occur, the VU shows 0 dB. When compression is affecting the signal the VU indicates the level of
compression in dB.
• OUTPUT: this knob controls the compressor output level.
• BYPASS: this switch allows you to bypass the Compressor module. When the LED is lit, the Compressor is
ON. To turn OFF the Compressor module, click the switch so that the LED is OFF.
• RESET: brings the compressor back to its default state.

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7 – Classic Multiband Limiter

This three-band peak limiter can make the mix even louder by reducing unwanted peaks. The peak limiting
action is made separately on the three bands (low, mid, high). You will have a very loud master, without clipping,
though as with any of these processors, use care not to over-do it. Leaving the output knob set to 0 dB means
that the limiter output signal can’t go beyond –0.05 dBfs.

Controls:

• SINGLE BAND LEVEL: these 3 knobs set the level (volume) of the three bands. This is very useful to get the
correct level of each band after the limiting. Use this as a very broad tone control tool on the mix. This is a
global, three band, equalizer too.
• SINGLE BAND THRESHOLD: these 3 knobs set the limiting amount over the individual band. Lowering
the value the threshold will be lower and the limiting more frequent. Raising the threshold will make the
limiter compress that band less. You can check this by looking at the analog VU on the right, this shows the
average compression of the three bands.
• CROSS OVER POINTS: these two knobs set the two frequency split points, the low-mid split and the
mid-high split. The default positions are fine in general but you can experiment with different settings. For
example lower the low-mid point to, say, 70 Hz to only limit and tighten-up the low end.
• RELEASE TIME: adjust this control to change the speed of limiting in the release phase. This is the amount
of time the limiter will take to return to normal gain after a peak limiting has occurred. While adjusting, check
the current Limiter numeric display to see the release time you’re setting. Value range is from 60 ms to 1.6
seconds.
• INPUT DRIVE: this knob controls the amount of signal that drives the multiband limiter input. Unlike the
compressor module, this one has an exact threshold point where it starts to limit the signal. If you drive
more signal through the limiter input, more peaks will be limited and the sound will be louder. If you want
to preserve more peaks, set this control to a lower value. Values range is from -10 dB to + 15 dB. While
adjusting Input Drive, check the values (in dB) by looking at the value above the Drive knob.

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• OVERLOAD: this control will affect the way Limiter reduces peaks. Setting this knob to a lower value will
result in more frequent “pure” gain reduction by the limiter. Setting this knob to a higher value will result in
less frequent gain reduction, but in more frequent clipping. Obviously setting this control at higher values will
give you more loudness, but more clipping across the 0 dB level.
• GAIN REDUCTION VU: this analog VU shows the amount of Gain reduction in dB. When limiting does not
occur the VU shows 0 dB. When limiting is affecting the signal, the VU indicates the level of limiting in dB.
This VU shows the average value of limiting by all three spectral bands.
• OUTPUT: this knob controls the limiter output level.
• BYPASS: this switch allows you to bypass the Limiter module. When the LED is lit, the Limiter is ON. To turn
OFF the Limiter module, click the switch so that the LED is OFF.
• RESET: brings the limiter back to its default state.

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8 – Classic Clipper

This clipping stage module is useful every time you need to cut out all the peaks that keep the average level of
a signal too low. Peak clipping is very often used in mastering because it’s in most cases more transparent than
peak limiting.

Controls:

• GAIN: the Gain control adjusts the pre-level of this clipping stage. This is the level of the signal injected into
the clipper and will determine the quantity of saturation applied to the audio. TIP: leave the OUTPUT now at
0 dB because doing this the clipper limiter will never go beyond –0.05 dBfs, regardless of the Gain level you
dial in.
• GAIN REDUCTION METER: shows instant by instant the amount of signal that is attenuated by the
clipping/saturation effect.
• SLOPE: the clipping shape of this stage is adjustable. It’s continuously variable from a straight digital hard
clip (fully clockwise) to ultra soft non-clipping mode (fully counterclockwise). Depending on music styles, the
clipping can be less noticeable either with softer or harder shapes; that’s why this control is continuously
adjustable. Only your ears can detect the optimal clipping shape for the piece of music you’re mastering. If
you are unsure, start with a middle value, say 3 dB, as a starting point.
• SAT: when the signal is in the non-linear zone (saturation) the SAT LED lights up.
• OUTPUT: this is internally calibrated to ensure that the output of the clipper will never overload. This means
that, regardless the input level of the clipper and the position of the GAIN knob, you’ll never get samples
over 0dB out from this module because the internal clipping has a –0.05 dBfs ceiling. This is only valid if the
OUTPUT knob is set at 0 dB.
• OUTPUT METER: shows the peak output level on the same bar for L and R channels, L is on the top and R
on the bottom.
• OVER: these two LEDs, separate for left and right channels, light up ONLY when a digital overload at the
clipper outputs has occurred. In theory they should never turn on. Adjust the OUTPUT control to ensure they

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stay off, with as high a level as possible.

• BYPASS: this completely bypasses the Clipper module.
• RESET: brings the Clipper back to its default state.

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9 – Classic Equalizer

The T-RackS Classic Equalizer consists of a high quality six-band parametric EQ specially designed to achieve
high-end performance in a mastering environment. It consists of:

1) 4th order High-pass filter, from 16 Hz to 5.3 kHz.

2) Low shelving type filter, from 30 Hz to 200 Hz.
3) Low-mid peaking type filter, from 33 Hz to 5.4 kHz, variable Q.
4) High-mid peaking type filter, from 200 Hz to 17.5 kHz, variable Q.
5) High shelving type filter, from 750 Hz to 8.5 kHz.
6) 4th order Low-pass filter, from 200 Hz to 18 kHz.

Controls:

• LOW CUT band ON/OFF: use this button to turn On and Off the band filter. When the filter is active the
button is highlighted in yellow.
• LOW CUT: the knob adjusts this frequency of the high pass filter. Sweep range is from 16 Hz to 5.3 kHz.
• LOW band ON/OFF: use this button to turn the filter On and Off. When the filter is active the button is
highlighted in yellow.
• LOW FREQ: the knob adjusts the frequency range of low-shelving filter. All frequencies up to this value will
be boosted or reduced. You can set the cutoff frequency anywhere from 30 Hz to 200 Hz.
• LOW GAIN: use this control to boost or reduce the level of this band. The range is from -15 dB to +15 dB.
• LOW MID band ON/OFF: use this button to turn the filter On and Off. When the filter is active the button is
highlighted in yellow.
• LOW MID FREQ: this knob adjusts the center frequency of the Low-Mid peaking filter. Range is 33 Hz to 5.4
kHz.

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• LOW MID GAIN: use this control to boost or reduce the effect of this filter. Range is from -15 dB to 15 dB.
• LOW MID Q: sets the bandwidth of the peaking filter, from 0.2 (broad) to 20 (sharp).
• HI MID band ON/OFF: use this button to turn the filter On and Off. When the filter is active the button is
highlighted in yellow.
• HI MID FREQ: this knob adjusts the center frequency of the Hi-Mid peaking filter. Range is from 200 Hz to
17.5 kHz.
• HI MID GAIN: use this control to boost or reduce the effect of this filter. Look at the value below the knob to
check the value (in dB) you’re setting. The range is from -15 dB to 15 dB.
• HI MID Q: sets the bandwidth of the peaking filter, from 0.2 (broad) to 20 (sharp).
• HI band ON/OFF: use this button to turn the filter On and Off. When the filter is active the button is
highlighted in yellow.
• HI FREQ: this knob adjusts the frequency range of the high shelving filter. Range is from 750 to 8.3 kHz.
• HI GAIN: use this control to boost or reduce the effect of this filter. Its range is from -15 dB to +15 dB.
• HI CUT band ON/OFF: use this button to turn On and Off the band filter. When the filter is active the button
is highlighted in yellow.
• HI CUT: this knob adjusts the cut-off frequency of the lo-pass filter. Range is from 200 to 18 kHz.
• OUTPUT: sets the equalizer output level from -15 to +15 dB.
• CURVE DISPLAY: this display shows the resulting equalization curve for both channels. Individual bands EQ
points can be dragged to set Gain and Frequency points without using the equalizer knobs.
• BYPASS: completely bypasses the equalizer.
• RESET: brings the Equalizer back to its default, flat, state.
• L=R: linked channels mode. Set it this way when you want to control both left and right channels at the
same time, by applying the exact same equalization to them.
• L/R: click on L or R icons when you want to tweak the equalization independently on the Left or Right
channels. Left channel curve will be displayed in green, right channel in white.
• M/S: this option uses a Mid-Side matrix to insert the equalizers on your processing chain. This means that
you will be able to equalize independently the Center or the Sides of your stereo program. The L and R
icons will be replaced by M and S when in this mode. Click on M to equalize the Center (Mid) of your stereo
image, or on S to equalize the Sides.

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T-RackS 6 Module Processors

10 – Black 76 Limiting Amplifier

Black 76 is based on the most famous FET limiting amplifiers. The strong sonic character of Black 76 makes
it one of the most versatile compressor - limiters. Based on all discrete transistor circuitry with true peak
limiting and fast attack time, it is the go-to workhorse compressor for all instruments. Black 76 can be used
on everything: drums, vocals, bass, guitars, piano, choirs, brass, strings... literally everything! Vocals, for
example, will gain presence, consistency, clarity and punch when run through Black 76. Black 76 gives drums
power, density, with total control on the level of “snap” you want, and it gives guitars a magic midrange zone
that makes them really sing. This dynamics processing legend has been carefully modeled with IK’s maniacal
attention to detail right down to the design of the VU meter for incredible accuracy.

Controls:

• INPUT: determines the amount of signal going into the Black 76.
• OUTPUT: determines the final level going out of the Black 76. Once the desired amount of limiting or
compression is achieved with the use of the Input control, the Output control can be used to make up any
gain loss due to the gain reduction process.
• ATTACK: sets the amount of time it takes the Black 76 to respond to an incoming signal and begin gain
reduction. The Black 76 attack time is adjustable from 20 microseconds to 800 microseconds (both
extremely fast). The attack time is fastest when the Attack knob is in its fully clockwise position, and it is
slowest when in its fully counterclockwise position.
• OFF: when this switch is pressed the compression is disabled and the processor works as a signal amplifier.
• RELEASE: sets the amount of time it takes the Black 76 to return to its initial (pre-gain reduction) level. The
release time is adjustable from 50 milliseconds to 1100 milliseconds. The release time is fastest when the
Release knob is in its fully clockwise position, and it is slowest when in its fully counterclockwise position.
• RATIO: these controls set the behavior of the compressor related to the signal hitting the threshold. A setting
of 4:1 means that signal exceeding the set threshold will be reduced by four times (4 dBs over the threshold
will reduced to 1dB). Likewise, 8:1. Higher ratios will make the Black 76 work as a limiter.

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T-RackS 6 Module Processors

• ALL: this is the typical “ALL BUTTONS IN” setting that makes the Black 76 behave like no standard
compressor due to the fact that some parameters like attack and release times are not fixed, but change
constantly. This results in more distortion and very pronounced “pumping” effect that can be effectively used
for some creative effects on room or ambient drum tracks, or to add distortion and sonic character in parallel
compression to bass, guitars and vocal tracks.
• METER: these buttons will allow to monitor either the applied gain reduction or the plug-in’s output level.
When the +4 button is selected a reading of 0dB on the metering will correspond to a -12dBFS level. When
the +8 button is selected the 0dB on the meter will correspond to a -8dBFS level.
• OFF: this button power off the processor. Note that the signal flow will be interrupted.
• RESET: clicking once on this switch brings all controls back to their original default state where a moderate
amount of compression is already applied (depending on the level of the input signal).
• BYPASS: completely bypasses the processing in the plug-in to compare with the unprocessed signal.
• L=R: links the two channels for stereo operation: all parameters affect the two channels in the same way to
maintain a consistent stereo image.
• L/R: click on the L or R icons if you want to dial in specific compression settings for L and R channels
independently.
• M/S: this option uses a Mid-Side matrix to insert the stereo compressor on your processing chain. This
means that you will be able to independently compress the Center or the Sides of your stereo program. The
L and R icons will be replaced by M and S when in this mode. Click M to set the compression for the Center
(Mid) of your stereo image or S to set compression for the Sides.

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T-RackS 6 Module Processors

11 – White 2A Levelling Amplifier

Based on the most famous vacuum tube optical limiter Leveling Amplifier, the White 2A compressor is an
impressive reproduction of one of the most popular and great sounding dynamics processing units in music
recording history. Its tube driven electro-optical gain reduction circuit is well-known for a very smooth and
natural compression, yet it is very vibey and incredibly musical when driven hard. This character has been
faithfully captured in the White 2A plug-in — right down to the accurate movement of the meter — in order to
provide you with one of the most useful tools for your compression needs. It is a must-have for tracking and
mixing vocals and anything that needs gentle, warm, consistent compression. White 2A is also great for winds,
bass, guitar, piano, organ, synths and more!

Controls:

• GAIN: the gain control should be set after the desired amount of compression is determined using the Peak
Reduction control. Once the Peak Reduction control is set, adjust the Gain Control to set the desired output
level.
• METER SELECTOR: sets the Metering display to show the actual output level with two different scales:
+10dB and +4dB. When set to Gain Reduction, you can monitor the amount of compression applied.
• PEAK REDUCTION: sets the amount of compression applied to the signal processed by the plug-in. There
is no compression when the knob is turned fully counterclockwise, and there is maximum compression
when it is turned fully clockwise.
• RESET: clicking once on this switch put all the controls back to their original default state where a moderate
amount of compression is applied (depending on the level of the input signal).
• LIMIT/COMPRESS: changes the compression ratio between two fixed settings: COMPRESS has a more
gentle curve, while LIMIT has a more pronounced effect.
• BYPASS: completely bypasses the processing in the plug-in to compare with the unprocessed signal.
• L=R: links the two channels for stereo operation: all parameters affect the two channels in the same way to
maintain a consistent stereo image.

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T-RackS 6 Module Processors

• L/R: click on the L or R icons if you want to dial in specific compression setting for L and R channels
independently.
• M/S: this option uses a Mid-Side matrix to insert the stereo compressor on your processing chain. This
means that you will be able to independently compress the Center or the Sides of your stereo program. The
L and R icons will be replaced by M and S when in this mode. Click M to set the compression for the Center
(Mid) of your stereo image or S to set compression for the Sides.

­­­­­­­­­28 White 2A Levelling Amplifier

T-RackS 6 Module Processors

12 – British Channel

The British Channel module is based on one of the most popular and universally acclaimed British mixing
consoles which hit the market with great impact in the 80’s and forever changed the way records were made.
The British Channel carefully reproduces all of its op-amp driven EQ and VCA Compressor/Expander-Gate
behavior and, as for the real thing, it can be used on all kinds of sources across a DAW to shape and finely
craft the mix’s sound. Its wealth of controls will make getting into each mix’s detail easy, and all instruments will
gain their own sound and dimension thanks to a super-precise EQ section (with selectable circuit mode) and a
dynamics section capable of going from subtle level control to extreme squashing and effective gating.

Controls:

EQ Section:
LOW Band:
• Gain - dB: determines the amount of boost or cut in Decibels for this frequency band.
• Freq - Hz: sets the frequency in Hertz at which the boost or cut happens.
• PEAK: when this switch is pressed, the LOW band changes from low shelving type to paeking type with
fixed Q.
LOW-MID Band:
• Gain - dB: determines the amount of boost or cut in Decibels for this frequency band.
• Freq - KHz: sets the frequency in Kilohertz at which the boost or cut happens.
• Q: sets the width of the frequency band where extreme left results in a narrow and sharper band, while
extreme right results in a wide and broader band.
HI-MID Band:
• Gain - dB: determines the amount of boost or cut in Decibels for this frequency band.
• Freq - KHz: sets the frequency in Kilohertz at which the boost or cut happens.

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T-RackS 6 Module Processors

• Q: sets the width of the frequency band where extreme left results in a narrow and sharper band, while
extreme right results in a wide and broader band.
HI Band:
• Gain - dB: determines the amount of boost or cut in Decibels for this frequency band.
• Freq - KHz: sets the frequency in Kilohertz at which the boost or cut happens.
• PEAK: when this switch is pressed, the HI band changes from high shelving type to peaking type with
fixed Q..

FILTERS Section (LOW-PASS/HI-PASS):

• FLT IN: this switch enables the filter section which can be used independently from the rest of the EQ
section.
• Freq - Hz: sets the frequency of the High Pass filter rolloff point. The filter has a 12 dB/Octave cut (18
when in Black mode). In the OUT position the filter is inactive: this effectively allows to use the two filters
independently.
• Freq - KHz: sets the frequency of the Low Pass filter rolloff point. The filter has a 12 dB/Octave cut. Just like
the other, in the OUT position the filter is inactive.
• FLT S/C: this switch makes the filter section act as the detector for the sidechain circuit of the dynamics
section. When engaged the HP and LP filters are not usable for their primary purpose.
• IN: this switch turns the whole EQ section (minus filters) on or off.
• B: this switch allows to choose which EQ mode the British Channel will turn in: when disengaged the mode
is Brown, meaning that the EQ behavior will be the one of the Brown EQ circuit, while when engaged the EQ
mode will be the one of the Black EQ circuit with its distinctive frequency shifts and boost/cut/Q differences.
This two different circuit designs offer sonically distinct results, and having them both available at the press
of a switch is a very handy resource for the ultimate tone tweaking.

DYNAMICS Section:
• IN: this switch enables the dynamics section.
GATE/EXP:
• EQ PRE: this switch modifies the sequence of EQ and dynamics sections. When engaged the
compressor section is shifted before the EQ setion, effectively changing how the whole module reacts to
the incoming signal in relation to the parameters settings.
• EQ S/C: this switch enables the sidechain circuit, meaning that the EQ section is used as a detector
for the dynamics section. This is a very effective way of compressing a particular frequency which is
selected with the EQ section. When this switch is enabled the EQ section will not work the standard way
and could not be used to EQ the incoming signal (you will have to instantiate another British Channel EQ/
Dyn plug-in or use another EQ).
• THRESHOLD: this control sets the level (in dB) below which the incoming signal is attenuated by the
gate/expander circuit.
• RANGE: this parameter controls the amount of gain reduction (in dB) applied to the incoming signal
when this falls below the threshold level. The extreme counterclockwise position will have none to
minimal gain reduction, while more effective gating (for more noisy or problematic tracks) is obtained
turning progressively this control towards the 40 dB value (fully clockwise).

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T-RackS 6 Module Processors

• RELEASE - S: this control determines the time (in seconds) it takes the expander/gate to release the
signal to its normal level once it has gone above the threshold.
• FAST ATT: this control switches the attack time it takes to the gate/expander circuit to open, to 1ms;
this can be useful to produce some very particular effects like snappy and percussive sounds with very
marked attack transients.
• EXP: this control activates the expansion mode for the gate/expander section.
COMP:
• THRESHOLD: this control sets the level in dB which has to be passed by the input signal coming into
the British Channel module for the compression to start working. At the 0 position the module is not
compressing. Turning the knob clockwise will lower the threshold and, depending on the input signal,
start compressing. Turning the knob counterclockwise towards the +10 value will lift the thershold
so that hotter signals can be easily managed. This control and its behavior can be affected by the
Master section’s I/O level controls. As an example, try driving the compressor’s input level harder with
the Master’s section IN knob and set the threshold low for some serious squashing and pumping
compression effects. The OUT knob will make calibrating the final output of the module easy in order
to avoid clipping. A note on Auto Makeup Gain: this compressor circuit design is provided of an auto
makeup gain control; this will progressively compensate the output level in relation to the threshold
control avoiding a drop in overall output level as a result of gain reduction. The lower the threshold, the
more you will hear it raise the output level, so pay attention about setting a correct gain staging among
EQ, Compressor and Master I/O levels.
• RATIO: this control determines the gain reduction amount applied to the signal passing above the
level set by the threshold control. Lower values of this control (1:1, 1.5:1) will result in none to very
light compression, while turning the knob clockwise will progressively make for a more pronounced
compression effect as you approach higher values; the extreme clockwise position will result in a limiting
effect.
• FAST ATT: this switch shifts the attack time it takes the circuit to start compressing once the incomig
signal has passed the set threshold to a faster 1 ms. It can be quite handy to emphasize attack transients
on percussive instruments like snare or kick drums and the likes.
• RELEASE - S: this control sets the time(in seconds) it takes the compressor to release the signal back
to uncompressed state after it has gone below the threshold level. Shorter release times will make the
compressor snappy to downright “pumping” when lots of compression is applied. Longer release times
will result in a more natural sounding effect and increased sustain on particular instruments.
• HARD: this switch changes the detecting mode of the compressor from RMS to peak.

MASTER Section:
• Ø (Phase): this control shifts the phase of the incoming signal by 180°, useful to check phase compatibility
of sources or to reverse phase in one of two coupled tracks (like snare drum top and bottom).
• IN: this control lets you trim the input level of the signal going into the British Channel module. It can be used
to correct the level of the incoming signal when needed (in case of a particularly weak or hot track) since
it has a range of +/- 15 dB. As described above (in the compressor’s threshold parameter section) it can
be used to drive the compressor’s section harder by boosting the input signal, thus producing some very
particular compression effects which can be further emphasized by ratio, attack and release time settings.
• OUT: this control lets you trim the master output level of the plug in. It can be used to avoid clipping and
overloading of the output stage in case some very radical EQ boosting is applied, or to simply control the
level of a track that needs to be adjusted.

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T-RackS 6 Module Processors

• RESET: this control brings the setting of all of the parameters back to their default status once clicked.
• L=R: links the two channels for stereo operation: all parameters affect the two channels in the same way to
maintain a consistent stereo image.
• L/R: click on the L or R icons if you want to dial in specific compression settings for L and R channels
independently.
• M/S: this option uses a Mid-Side matrix to insert the stereo compressor on your processing chain. This
means that you will be able to independently compress the Center or the Sides of your stereo program. The
L and R icons will be replaced by M and S when in this mode. Click M to set the compression for the Center
(Mid) of your stereo image or S to set compression for the Sides.

­­­­­­­­­32 British Channel

T-RackS 6 Module Processors

13 – Quad Comp

The Quad Comp is one of the four modules in the Multiband serie of IK Multimedia plug ins for T-RackS.
Multiband processing at the core of this plug-in splits the incoming signal into up to four frequency bands and
treats each one separately with a dedicated set of controls. The width of each band can be finely adjusted so
that the processing can be very selective (in case of a narrow band) or spread across a wider range. Each band
is loaded with an optical compressor of particular effectiveness but with a very smooth sound and a full set of
adjustable parameters.

Controls:

• FREQUENCY Display: this big display shows the whole frequency spectrum and how it’s divided in the
four bands by the vertical frequency bars. To modify the width of a band click and drag on the top square
marker and adjust to the desired frequency (which will appear during movement). Clicking on a band’s area
will automatically select it and this will be reflected by the set of underlying controls which will change values
accordingly. Click and drag on the central dot to adjust the band’s gain (it will be reflected by the band’s
underlying gain control).
• METERING Display: this display shows the gain reduction (in dBs) for each band and the master output
level. When the Quad-Comp is used in STEREO mode the gain reduction will be shown for each single band
independently. When the plug in is used in M/S (Mid Side) mode the gain reduction meter of each band will
switch to a different visualization where two separate bar of LEDs will show Mid gain reduction (red LEDs)
and Side gain reduction (grey LEDs).
• MASTER OUTPUT Fader: use this master fader to adjust the final output of the plug in in order to optimize
level and avoid clipping (in case of too hot signals) or to bring back the signal to a more consistent level after
compression has been applied.
• ON/BYPASS Switch: this control enables the plug in processing when in the “ON” status.
• THRESHOLD: this control sets the threshold level for the selected band at which the plug in will start
compressing the incoming signal. Turning the knob counterclockwise will progressively lower the threshold

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T-RackS 6 Module Processors

resulting in more compression.

• RATIO: this control sets the amount of gain reduction applied to the incoming signal once it has passed
above the threshold level. Lower values of this control (when turned fully counterclockwise) will result in
none to light compression of the selected frequency band, while progressively turning the knob clockwise
will increase gain reduction. The extreme clockwise position with a ratio of 30:1 will result in very high
compression.
• ATTACK: this control sets the time it takes the compressor to react and apply gain reduction once the
incoming signal has passed above the set threshold. The extreme counterclockwise values will result in a
very fast attack (0.1 ms) which will emphasize transients on percussive sources.
• RELEASE: this control determines the amount of time it takes the compressor to bring the signal back to its
uncompressed state after it has fallen below the threshold. Short release times will result in a more audible
compression effect while longer ones (towards the extreme clockwise position) will result in a more natural
compression and in increased sustain of several sources.
• GAIN: this control sets the output level of the selected band. It can be used as a makeup gain to
compensate for level loss after compression or to adjust the balance among the four bands. Adjustments of
this control will be reflected by the “gain dot” on selected band and the shape of the horizontal white gain
line.
• SOLO: this button puts the selected band in solo mode to listen to its frequency range and gain reduction
applied.
• BYPASS: this control bypasses the compression for the selected band.
• BANDS: this control lets you select the number of active bands the frequency spectrum will be divided into.
• SLOPE: This control determines the steepness of the filters which will be used to split the frequency
spectrum in bands. At its minimum (6dB/Oct) the slope of the filter is quite gentle and allows for more
overlapping among the bands. At its maximum value (48 dB/Oct) the filter is more selective and separation
among bands becomes sharper.
• LINK BANDS: this button links the controls for all bands so that the values set for a selcted band will be
mirrored on all of the other bands. This can be handy when all control must be set identical for all bands. The
only exception is the gain control which can be still independently modified for each band by using the Gain
Dot on the frequency display.
• STEREO: this control links the two channels for stereo operation: all parameters affect the two channels in
the same way to maintain a consistent stereo image.
• MID/SIDE: this option uses a Mid-Side matrix to independently compress the Center or the Sides of your
stereo program. Click MID to set the compression for the Center (Mid) of your stereo image or SIDE to set
compression for the Sides.
• RESET Switch: clicking on this control will reset all of the control to their initial default state. Clicking again
after the reset will bring them back to the last setting they were left, in case you accidentally hit the reset
button, or changed your mind.

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T-RackS 6 Module Processors

14 – Quad Lim

The Quad Limiter is one of the four plug-ins in the Multiband series for T-RackS. Multiband processing at the
core of this plug-in splits the incoming signal into up to four frequency bands and treats each one separately
with a dedicated set of controls. The width of each band can be finely adjusted so that the processing can be
very precise (in case of a narrow band) or spread across a wider range. Each band is loaded with a peak limiter
which controls dynamics with extreme precision and accuracy across the whole spectrum, with the added
flexibility of selectable frequency zones. In addition to the four limiters the Quad Lim is loaded with a master
brickwall limiter located post processing so that no overs can escape and clipping is avoided. This will let you
produce very loud masters with a controlled frequency balance. Of course, be careful with limiting amount as
too much will produce flat tracks without dynamics … unless you want that!

Controls:

• FREQUENCY Display: this big display shows the whole frequency spectrum and how it’s divided in the
four bands by the vertical frequency bars. To modify the width of a band click and drag on the top square
marker and adjust to the desired frequency (which will appear during movement). Clicking on a band’s area
will automatically select it and this will be reflected by the set of underlying controls which will change values
accordingly. Click and drag on the central gain dot to adjust the band’s gain (it will be reflected by the band’s
underlying gain control).
• METERING Display: this display shows the gain reduction (in dBs) for each band and the master output
level. When the Quad-Lim is used in STEREO mode the gain reduction will be shown for each single band
independently. When the plug in is used in M/S (Mid Side) mode the gain reduction meter of each band will
switch to a different visualization where two separate bar of LEDs will show Mid gain reduction (red LEDs)
and Side gain reduction (grey LEDs).
• MASTER OUTPUT Fader: use this master fader to adjust the final output of the plug in. This fader works
pre Brickwall Limiter so it can be used to boost the overall level and amount of limiting after the settings on
the four bands have been adjusted without the risk of clipping or overload.
• ON/BYPASS Switch: this control enables the plug in processing when in the “ON” status.

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T-RackS 6 Module Processors

• THRESHOLD: this control sets the threshold level for the selected band at which the plug in will start limiting
the incoming signal. Turning the knob counterclockwise will progressively lower the threshold resulting in
more limiting.
• ATTACK: this control sets the time it takes the limiter to react and apply gain reduction once the incoming
signal has passed above the set threshold. The extreme counterclockwise values will set a very fast attack
(0.05 ms) which will result in a very fast response from the plug in catching the most instantaneous transient
peaks.
• RELEASE: this control determines the amount of time (in ms) it takes the limiter to bring the signal back
to its normal state after it has fallen below the threshold. As a general guideline for a correct peak limiting
action, release time should be set as short as possible, in order to “clamp” transient peaks without touching
the overall RMS level. Anyway, too short release times will cause audible distortion and artifacts (especially at
low frequencies), so take the time to adjust this parameter carefully.
• GAIN: this control sets the output level of the selected band. Working in conjunction with the threshold
parameter it can be used to raise the gain of the selected band effectively making it louder as transient
peaks are limited and RMS value of the track increased. Adjustments of this control will be reflected by the
“gain dot” on the selected band and the shape of the horizontal white gain line.
• SOLO: this button puts the selected band in solo mode to listen to its frequency range and limiting applied.
• BYPASS: this control bypasses the limiting action for the selected band.
• BANDS: this control selects the number of active bands the frequency spectrum will be divided into.
• SLOPE: this control determines the steepness of the filters which will be used to split the frequency
spectrum in bands. At its minimum (6dB/Oct) the slope of the filter is quite gentle and allows for more
overlapping among the bands. At its maximum value (48 dB/Oct) the filter is more selective and separation
among bands becomes sharper.
• LINK BANDS: this button links the controls for all bands so that the values set for a selected band will be
mirrored on all of the other bands. This can be handy when all controls must be set identical for all bands.
The only exception is the gain control which can be still independently modified for each band by using the
“gain dot “ on the frequency display.
• BRICKWALL: this control activates the brickwall limiter placed at the back end of the four bands. Consider
this additional limiter as an extreme protection to avoid any possible clipping or overloading of the program;
its fixed parameters are carefully adjusted to result in maximum transparency and effective overload
protection. Working in conjunction with the master output fader it can also be used to globally increase the
level of the track after the limiters on all four bands have been set.
• STEREO: this control links the two channels for stereo operation: all parameters affect the two channels in
the same way to maintain a consistent stereo image.
• MID/SIDE: this option uses a Mid-Side matrix to independently limit the Center or the Sides of your stereo
program. Click MID to set the limiting for the Center (Mid) of your stereo image or SIDE to set limiting for the
Sides.
• RESET Switch: clicking on this control will reset all of the controls to their initial default state. Clicking again
after the reset will bring them back to the last setting they were left in, in case you accidentally hit the reset
button or changed your mind.

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T-RackS 6 Module Processors

15 – Quad Image

The Quad Image is one of the four modules in the Multiband series of IK Multimedia plug ins for T-RackS.
Multiband processing at the core of this plug-in splits the incoming signal into up to four frequency bands and
treat each one separately with a dedicated set of controls. The width of each band can be finely adjusted so that
the processing can be very selective (in case of a narrow band) or spread across a wider range. Each band is
loaded with an M/S matrix processor which modifies its stereo width.

Controls:

• FREQUENCY Display: this big display shows the whole frequency spectrum and how it’s divided in the
four bands by the vertical frequency bars. To modify the width of a band click and drag on the top square
marker and adjust to the desired frequency (which will appear during movement). Clicking on a band’s area
will automatically select it, and this will be reflected by the set of underlying controls which will change values
accordingly. Click and drag on the central gain dot to adjust the band’s gain (it will be reflected by the band’s
underlying gain control). The horizontal white line will display the percentage of stereo widening applied to
each band with 0% being mono (or all M signal) and 200% being wide open stereo (or all S signal). The four
underlying “radar” displays will monitor the stereo width of each band.
• METERING Display: this set of led meters will display the M and S level (with different colors) for each band.
• MASTER OUTPUT Fader: this control will regulate the master output of the plug in which is shown on the
adjacent led meter.
• WIDTH: this control regulates the amount of stereo spread for the selected band.
• GAIN: this control sets the amount of gain of the selected band.
• SOLO: this button puts the selected band in solo mode allowing to listen to its frequency range and stereo
width applied.
• BYPASS: this control bypasses the stereo widening action for the selected band.
• BANDS: this control selects the number of active bands the frequency spectrum will be divided into.

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T-RackS 6 Module Processors

• SLOPE: this control determines the steepness of the filters which split the frequency spectrum in bands.
At its minimum (6dB/Oct) the slope of the filter is quite gentle and allows for more overlapping among the
bands. At its maximum value (48 dB/Oct) the filter is more precise and separation among bands becomes
sharper.
• LINK BANDS: this button links the controls for all bands so that the values set for a selcted band will be
mirrored on all of the other bands. This can be handy when all controls must be set identical for all bands.
The only exception is the gain control which can be still independently modified for each band by using the
“gain dot “ on the frequency display.
• RESET Switch: clicking on this control will reset all of the controls to their initial default state. Clicking again
after the reset will bring them back to the last setting they were left in, in case you accidentally hit the reset
button or changed your mind.
• ON/BYPASS Switch: this control enables the plug in processing when in the “ON” status.

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T-RackS 6 Module Processors

16 – De-Esser

The De-Esser is one of the four processors in the Multiband series for T-RackS. With this processor, the
de-essing action and excessive sibilance or high frequency control can be adjusted in a dual operating mode
depending on the number of bands selected. A wealth of controls adjusts the de-essing action to suit any
particular situation.

Controls:

• FREQUENCY Display: this big display shows the whole frequency spectrum and how it’s divided with the
vertical bars. The frequency range of this module is centered in a different area than the other processors
in the Multiband Series (its range goes form 1 to 13KHz). This is because de-essing process is only used in
this range of the spectrum. To modify the width of the bands, use the underlying slider control.
• DE-ESS: this control determines the amount of de-essing applied. The resulting reduction in HF content is
displayed on the adjacent led meter. Be careful to adjust this control as too much de-essing action could
result in unnatural sound, especially when used on vocal tracks.
• RELEASE: this control sets the release time it takes the De-Esser to bring the signal back to its normal state
once the signal has fallen below the set de-essing action.
• DE-ESS ACTION: use this control to listen to the effective amount of de-essing applied to the track.
• NORMAL: this control sets the listening mode so that you’ll hear the track while the de-essing action is
applied.
• DE-ESS BAND: this control allows you to put the De-Ess band in solo mode.
• SOFT: this control sets the intensity of the de-essing action, when selected the operating mode is more
gentle with a less pronounced and slower HF reduction, while when de-selected it’s faster and more marked
action.
• BANDS: this drop down menu allows to select the number of bands the frequency spectrum will be divided
into: when set to 2 bands the de-essing action will be performed on the higher band, while when 3 bands is

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T-RackS 6 Module Processors

selected it will be possible to set the higher and lower margin so that the de-essing action will be performed
on a selected area.
• SLOPE: this control sets the steepness of the filters used to split the frequency spectrum into bands. Higher
values of this control will result in harder separation with less interaction of the de-essing action on adjacent
bands.
• RESET Switch: clicking on this control will reset all of the controls to their initial default state. Clicking again
after the reset will bring them back to the last setting they were left in, in case you accidentally hit the reset
button or changed your mind.
• ON/BYPASS Switch: this control enables the plug in processing when in the “ON” status.

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17 – CSR Plate Reverb

Plate reverbs have been a staple in pop music recording since the late fifties. They consisted of a large metal
sheet, suspended from a heavy steel frame, with a small transducer (like a small speaker) transmitting the sound
energy onto the plate. A pickup (or 2 if stereo) was used to capture the sound. This was all housed in a 4 x 8
foot box. When originally designed, this was considerably smaller than the forerunning “echo chamber”.

Since the creation of digital reverbs, to further exemplify its popularity, the plate reverb has been found on just
about every rack-mounted reverb box. It is exceptional on vocals, having a bright, slightly metallic character,
but warm enough to sit well in most mixes without muddying up the singer. It is also often used on drums and
percussion. Its smooth attack and bright characteristic make it a great choice for adding depth to a snare drum
without burying it in the reverberation.

There are two operating

modes for this module:
“Easy” for a basic control
of the most commonly
used parameters, and
“Advanced” to tweak
every detail and have deep
control of the reverb shape
and tone. The diagram
below shows the controls
breakdown of Classik
Studio Reverb’s “Classik
Plate”.

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T-RackS 6 Module Processors

17.1 – Easy
• MIX: controls the ratio of reverb and original sound. 0% is all unprocessed sound. 100% is all reverb. Use
this control to increase or decrease the amount of reverb in the sound. If you are using your reverb as a send
effect, this control will probably be set to 100%.
• DIFFUSION: determines the rate that the echo density increases over time. A small diffusion setting
simulates an acoustic space with smooth walls, like a bathroom. A larger diffusion setting creates a more
open and dense sounding reverb.
• RVB TIME (Reverb Time): adjusts the length of the reverb. This is a commonly used control that shortens
and lengthens the time that the reverb tail lasts.
• LOW TIME: controls the length of the reverb for low frequencies (determined by the Crossover control) of
the RVB Time.
• HIGH DAMP (High Dampening): shortens the high frequency decay time of the reverb. This allows you to
emulate the way different materials in a real room, such as carpet and furnishings, absorb sound.
• HIGH FREQ (High Frequency): changes the frequency of high frequency dampening.

17.2 – I/O Levels

• IN LEVEL: adjusts the input level of the reverb.
• OUT LEVEL: adjusts the output level from the reverb.
• MIX: controls the ratio of reverb and original sound. 0% is all unprocessed sound. 100% is all reverb. Use
this control to increase or decrease the amount of reverb in the sound. If you are using your reverb as a send
effect, this control will probably be set to 100%.
• IN IMAGE: controls the stereo image of the input signal before it is processed through the effect. This control
allows you to change the signal from normal stereo to mono and manipulate the phase of the signal. For
example a value of -45 swaps the left and right channels, 0 is mono, and 45 is normal stereo. With a setting
of 90, all mono information will be excluded from the reverb sound.
• OUT IMAGE: controls the stereo image of the affected signal. With a value of -360 to +360, this control
extends the stereo field of the processed sound. This control can be used to create mono, stereo, or
surround type effects. It can also be changed in real-time or automated using the modulation or macro
matrix.

17.3 – Time
• RVB TIME (Reverb Time): adjusts the length of the reverb. In other words, this shortens and lengthens the
time that the reverb tail lasts.
• LOW TIME: controls the length of the reverb for low frequencies (determined by the Crossover parameter) of
the RVB Time.
• CROSSOVER: changes the crossover frequency for the Low time control.
• HIGH FREQ (High Frequency): changes the frequency for the low-pass filter used for High Frequency
damping.
• HIGH DAMP (High Dampening): dampens (attenuates) the high frequency being re-circulated into the

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T-RackS 6 Module Processors

reverberation. Using High frequency dampening often gives a more natural sounding reverb.
• PREDELY (Predelay): determines the amount of time between the initial input signal and the reverberation.
Increasing the predelay can create a sense of distance and, to some degree, change the size of the acoustic
space.

17.4 – Reverb
• SIZE: controls the approximate size of the acoustic space from small to large. This control is measured in
meters and is approximately equal to the longest dimension of the space.
• DIFFUSION: determines the rate that the echo density increases over time. A small diffusion setting
simulates an acoustic space with smooth walls, like a bathroom. A larger diffusion setting creates a more
open and dense sounding reverb.
• BUILDUP: changes the speed of the reverb’s buildup. When this control is turned down, it causes a fast
attack in the reverb. To create a slower buildup in the reverberation, turn up the buildup control. Increasing
the Buildup will create the sense of a large room/hall, but the spatial information can be lost - this can be
restored with the reflection parameters (delay and level).
• DISPERSE: disperse works in conjunction with the Buildup control to determine the initial envelope of the
reverberation. You can use Disperse to vary the apparent size and character of the early reflection.
• MODULATN (Modulation): creates movement in the reverberation tail. Modulation continually changes the
timbre of the reverberation. If used in smaller amounts, 10%-50%, it can result in a more natural sounding
reverb. If used higher than that, it can cause acoustic instruments timbre to become unstable.

17.5 – Color
• LO CUT F (Low Cut Frequency): changes the frequency of the low cut filter in the reverberation path. This
control ranges from 10Hz to 20 kHz.
• LO CUT G (Low Cut Gain): changes the gain of the low cut filter. This is adjustable from -24dB to 0dB.
• HI CUT F (High Cut Frequency): changes the frequency of the high cut filter in the reverberation path. This
control ranges from 10Hz to 20 kHz.
• HI CUT G (High Cut Gain): changes the gain of the high cut filter. This is adjustable from -24dB to 0dB.

17.6 – Reflections
• TIME LEFT: changes the delay between the input signal and the early reflections in the left channel.
• LEVEL LEFT: changes the level of the reflections in the left channel.
• TIME RIGHT: changes the delay between the input signal and the early reflections in the right channel.
• LEVEL RIGHT: changes the level of the reflections in the right channel.

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T-RackS 6 Module Processors

17.7 – Echo
• TIME LEFT: increases and decreases the length of the echo in the left channel. This echo is routed to the
reverb engine and the feedback path as well.
• FEED LEFT: controls the amount of delay that is being sent to the feedback paths. To hear significant
changes in this control, you may have to change this level dramatically.
• TIME RIGHT: increases and decreases the length of the echo in the right channel. This echo is routed to the
reverb engine and the feedback path as well.
• FEED RIGHT: controls the amount of delay that is being sent to the feedback paths. To hear significant
changes in this control, you may have to change this level dramatically.

17.8 – Modulation Matrix

This section of controls is common to all the CSR modules.
The modulation matrix is a flexible system for adding continuous movement to the parameters and sound of
Classik Studio Reverb modules. It allows you to route 4 modulation sources to a selection of parameters in the
reverb engine. You can achieve subtle or extreme variations in the reverb by applying the modulation sources to
the different parameters (destinations) in Classik Studio Reverb.
Here is a diagram of the modulation system and its sources and destinations.

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T-RackS 6 Module Processors

17.8.1 – Modulation
The matrix window displays your modulations sources, destinations, minimum and maximum values, and curve
controls.

17.8.2 – Sources (modulation sources)

LFO1 and LFO2
Low frequency oscillators generate Sine, Triangle, Sawtooth, Square Wave, and Noise all with adjustable rates.
These signals can be used to modulate any of the available destinations in the matrix.

• TYPE: selects the waveform of the oscillator.

• RATE: selects the speed of the waveform.

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T-RackS 6 Module Processors

ENV1 and ENV2

Envelope generators featuring attack and release. They can be triggered from Input L, R, or summed and
Reverb L, R, or summed. These envelopes can be used to modulate any of the available destinations in the
matrix.

• INPUT: selects the input trigger of the envelope generator.

• GAIN: adjusts the amount of gain required to trigger the envelope.
• ATTACK: changes the length of time that the envelope ramps up to full open.
• RELEASE: changes the amount of time that the envelope returns to off after it has been opened.

Any of the modulation sources can be routed multiple times up until all 8 inputs of the modulation matrix are full.

17.8.3 – Destinations
Destinations are the parameters in the reverb engine that can be modulated. Almost every parameter in the CSR
reverb engine can be chosen.

• MIN: assigns the minimum value of modulation that can be applied to the destination. This value will change
depending on the selected parameter.
• MAX: assigns the maximum value of modulation that can be applied to the destination. This value will
change depending on the selected parameter.
• CURVE: changes the response curve between the source and the destination.

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T-RackS 6 Module Processors

There are three available curves:

• LIN (Linear): provides a linear response between the source and the destination.
• LOG (Logarithmic): applies a logarithmic scale to the response between the source and destination.
• EXP (Exponential): provides an exponential scale of response between the source and the destination.

17.9 – MACROS
This section of controls is common to all the CSR modules.
The MACRO controls are the 4 sliders on the far right hand side of the interface. These sliders can be assigned
to multiple parameters. So you can add additional single controls to the front panel, giving you more easy
control to a preset. You can also create complex morphing controls, making it possible to move from one type
of reverb sound to a totally different reverb sound in a single slider movement.

17.9.1 – MATRIX
The MACROS are configured in much the same way as the modulation matrix.

There is an 8x8 matrix that allows you to route the 4 MACRO controls to multiple destinations. Here is a diagram
of the MACRO controls matrix.

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T-RackS 6 Module Processors

The MACRO matrix functions like the modulation matrix in that it can route any of the modulation sources
multiple times up until all 8 inputs of the matrix are full. The controls of the MACRO matrix are the same as well.

17.9.2 – Sources (modulation sources)

MACRO 1-4: slider controls that can control multiple functions from the front panel, making it possible to create
morphing effects using a single control.

17.9.3 – Destinations
Destinations are the parameters in the reverb engine that can be controlled. You can choose almost every
parameter as a destination source.

• MIN: assigns the minimum value the MACRO can apply to the destination. This value will change depending
on the selected parameter. If the min value is set higher than the maximum value the macro control will move
the parameter in the opposite direction.
• MAX: assigns the maximum value the MACRO can apply to the destination. This value will change

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T-RackS 6 Module Processors

depending on the selected parameter.

• CURVE: changes the response curve between the source and the destination.

There are three available curves:

• LIN (Linear): provides a linear response between the source and the destination.
• LOG (Logarithmic): applies a logarithmic scale to the response between the source and destination.
• EXP (Exponential): provides an exponential scale of response between the source and the destination.

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T-RackS 6 Module Processors

17.10 – MACRO NAME

In addition to assigning the MACRO controls in the matrix, you can assign a name to the MACRO control that
will appear whenever the reverb patch is opened. This is done by clicking the MACROS button in the upper right
of the MACRO matrix window .

This will open up the naming window. Now click on one of the boxes under “LABEL NAME”. This will open
up a new dialog box.Type in a short name and click OK. Now this name will appear on the front interface of
CSR after you save the preset.

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T-RackS 6 Module Processors

18 – CSR Hall Reverb

Hall reverb is a simulation of a large acoustic hall. This type of effect has been widely used to create large
ambient spaces. Digital reverbs made this sound popular, simply because it was not until the introduction of
digital technology that such complex artificial reverbs were possible, unless you had the keys to your local
cathedral or concert hall...

Many people would describe this as a vocal reverb. This is useful on vocals if you are looking for a thick, full
reverb. This is typically a very pronounced reverb effect. It often adds a “larger than life” characteristic to the
affected sound. Hall reverbs are useful in creating a large space around any musical sound, as well as very
useful for strange or moody film sound effects.

There are two operating

modes for this module:
“Easy” for a basic control
of the most commonly
used parameters, and
“Advanced” to tweak
every detail and have deep
control of the reverb shape
and tone.

The diagram shows the

controls breakdown of
Classik Studio Reverb’s
“Classik Hall”.

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T-RackS 6 Module Processors

18.1 – Easy
• MIX: controls the ratio of reverb and original sound. 0% is all unprocessed sound. 100% is all reverb.
Use this control to increase or decrease the amount of reverb in the sound. If you are using your reverb
as a send effect, this control will probably be set to 100%.
• DIFFUSION: determines the rate that the echo density increases over time. A small diffusion setting
simulates an acoustic space with smooth walls, like a bathroom. A larger diffusion setting creates a more
open and dense sounding reverb.
• RVB TIME (Reverb Time): adjusts the length of the reverb. This is a commonly used control that shortens
and lengthens the time that the reverb tail lasts.
• LOW TIME: controls the length of the reverb for low frequencies (determined by the Crossover parameter) of
the RVB Time.
• HIGH FREQ (High Frequency): changes the frequency of high frequency dampening.
• HIGH DAMP (High Dampening): shortens the high frequency decay time of the reverb. This allows you to
emulate the way different materials in a real room, such as carpet and furnishings, absorb sound.

18.2 – I/O Levels

• IN LEVEL: adjusts the input level of the reverb.
• OUT LEVEL: adjusts the output level from the reverb.
• MIX: controls the ratio of reverb and original sound. 0% is all unprocessed sound. 100% is all reverb.
Use this control to increase or decrease the amount of reverb in the sound. If you are using your reverb
as a send effect, this control will probably be set to 100%.
• IN IMAGE: controls the stereo image of the input signal before it is processed through the effect. This control
allows you to change the signal from normal stereo to mono and manipulate the phase of the signal. For
example a value of -45 swaps the left and right channels, 0 is mono, and 45 is normal stereo. With a setting
of 90, all mono information will be excluded from the reverb sound.
• OUT IMAGE: controls the stereo image of the effected signal. With a value of -360 to +360, this control
extends the stereo field of the processed sound. This control can be used to create mono, stereo, or
surround type effects. It can also be changed in real-time or automated using the modulation or macro
matrix.

18.3 – Time
• RVB TIME (Reverb Time): adjusts the length of the reverb. In other words, this shortens and lengthens the
time that the reverb tail lasts.
• LOW TIME: controls the length of the reverb for low frequencies (determined by the Crossover parameter) of
the RVB Time.
• CROSSOVER: changes the crossover frequency for the Low time control.
• HIGH FREQ (High Frequency): changes the frequency for the low-pass filter used for High Frequency
damping.
• HIGH DAMP (High Dampening): damps (attenuates) the high frequency being recirculated into the

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T-RackS 6 Module Processors

reverberation. Using High frequency damping often gives a more natural sounding reverb.
• PREDELY (Predelay): determines the amount of time between the initial input signal and the reverberation.
Increasing the predelay can create a sense of distance and, to some degree, change the size of the acoustic
space.

18.4 – Reverb
• SIZE: controls the approximate size of the acoustic space from small to large. This control is measured in
meters and is approximately equal to the longest dimension of the space.
• DIFFUSION: determines the rate that the echo density increases over time. A small diffusion setting
simulates an acoustic space with smooth walls, like a bathroom. A larger diffusion setting creates a more
open and dense sounding reverb.
• BUILDUP: changes the speed of the reverb’s buildup. When this control is turned down, it causes a fast
attack in the reverb. To create a slower buildup in the reverberation, turn up the buildup control. Increasing
the Buildup will create the sense of a large room/hall, but the spatial information can be lost - this can be
restored with the reflection parameters (delay and level).
• DISPERSE: disperse works in conjunction with the Buildup control to determine the initial envelope of the
reverberation. You can use Disperse to vary the apparent size and character of the early reflection.
• MODULATN (Modulation): creates movement in the reverberation tail. Modulation continually changes the
timbre of the reverberation. If used in smaller amounts, 10%-50%, it can result in a more natural sounding
reverb. If used higher than that, it can cause acoustic instruments timbre to become unstable.
• MODMOVE (Modulation Movement): changes the time that early reflection will move in the reverberation.
This can be used to create more accurate spatialization of the reverb effect.

18.5 – Color
• LO CUT F (Low Cut Frequency): changes the frequency of the low cut filter in the reverberation path. This
control ranges from 10Hz to 20 kHz.
• LO CUT G (Low Cut Gain): changes the gain of the low cut filter. This is adjustable from -24dB to 0dB.
• HI CUT F (High Cut Frequency): changes the frequency of the high cut filter in the reverberation path. This
control ranges from 10Hz to 20 kHz.
• HI CUT G (High Cut Gain): changes the gain of the high cut filter. This is adjustable from -24dB to 0dB.

18.6 – Reflections
• TIME LEFT: changes the delay between the input signal and the early reflections in the left channel.
• LEVEL LEFT: changes the level of the reflections in the left channel.
• TIME RIGHT: changes the delay between the input signal and the early reflections in the right channel.
• LEVEL RIGHT: changes the level of the reflections in the right channel.

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T-RackS 6 Module Processors

18.7 – Modulation Matrix, MARCOS and MACRO NAME

This section of controls is common to all the the CSR modules. Refer to paragraphs from 4.17.8 to 4.17.10 for
detailed description.

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T-RackS 6 Module Processors

19 – CSR Room Reverb

Room reverb is often used to create the ambience of a small to medium sized room. This type of reverb is very
effective on any instrument where you want to add the illusion of recording the sound with room ambience. This
might seem like a simple thing, but in reality most close-miked instruments lack very much ambience in the
recorded sound. Adding a little bit of room reverb can make the sound sit nicer in the mix and give the illusion
that the sound came from a nice sounding room, even if it was recorded in a small boxy closet!

This is useful on any sound needing a small to medium size acoustic space. Often used on drums, acoustic
guitar, and even vocals, this is a very effective reverb sound on many different instruments.

There are two operating

modes for this module:
“Easy” for a basic control
of the most commonly
used parameters, and
“Advanced” to tweak
every detail and have deep
control of the reverb shape
and tone.

The diagram shows the

controls breakdown of
Classik Studio Reverb’s
“Classik Room”.

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T-RackS 6 Module Processors

19.1 – Easy
• MIX: controls the ratio of reverb and original sound. 0% is all unprocessed sound. 100% is all reverb. Use
this control to increase or decrease the amount of reverb in the sound. If you are using your reverb as a send
effect, this control will probably be set to 100%.
• DIFFUSION: determines the rate that the echo density increases over time. A small diffusion setting
simulates an acoustic space with smooth walls, like a bathroom. A larger diffusion setting creates a more
open and dense sounding reverb.
• DECAY TIME: controls the length of the reverb tail.
• DECAY LVL (Decay Level): controls the level of the reverb tail. When turned off, only the early reflections are
in the signal.
• HIGH FREQ (High Frequency): changes the frequency of high frequency damping.
• HIGH DAMP (High Dampening): shortens the high frequency decay time of the reverb. This allows you to
emulate the way different materials in a real room, such as carpet and furnishings, absorb sound.

19.2 – I/O Levels

• IN LEVEL: adjusts the input level of the reverb.
• OUT LEVEL: adjusts the output level from the reverb.
• MIX: controls the ratio of reverb and original sound. 0% is all unprocessed sound. 100% is all reverb. Use
this control to increase or decrease the amount of reverb in the sound. If you are using your reverb as a send
effect, this control will probably be set to 100%.
• IN IMAGE: controls the stereo image of the input signal before it is processed through the effect. This control
allows you to change the signal from normal stereo to mono and manipulate the phase of the signal. For
example a value of -45 swaps the left and right channels, 0 is mono, and 45 is normal stereo. With a setting
of 90, all mono information will be excluded from the reverb sound.
• OUT IMAGE: controls the stereo image of the effected signal. With a value of -360 to +360, this control
extends the stereo field of the processed sound. This control can be used to create mono, stereo, or surround
type effects. It can also be changed in real-time or automated using the modulation or macro matrix.

19.3 – Time
• DECAY TIME: controls the length of the reverb tail.
• DECAY LVL (Decay Level): controls the level of the reverb tail. When turned off, only the early reflections are
in the signal.
• HIGH FREQ (High Frequency): changes the frequency for the low-pass filter used for High Frequency
damping.
• HIGH DAMP (High Dampening): dampens (attenuates) the high frequency being recirculated into the
reverberation. Using High frequency damping often gives a more natural sounding reverb.
• PREDELY (Predelay): determines the amount of time between the initial input signal and the reverberation.
Increasing the predelay can create a sense of distance and, to some degree, change the size of the acoustic
space.

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T-RackS 6 Module Processors

19.4 – Reverb
• SIZE: controls the approximate size of the acoustic space from small to large. This control is measured in
meters and is approximately equal to the longest dimension of the space.
• DIFFUSION: determines the rate that the echo density increases over time. A small diffusion setting
simulates an acoustic space with smooth walls, like a bathroom. A larger diffusion setting creates a more
open and dense sounding reverb.
• MODULATN (Modulation): creates movement in the reverberation tail. Modulation continually changes the
timbre of the reverberation. If used in smaller amounts, 10%-50%, it can result in a more natural sounding
reverb. If used higher than that, it can cause acoustic instruments timbre to become unstable.
• MODMOVE (Modulation Movement): changes the time that early reflection will move in the reverberation.
This can be used to create more accurate spatialization of the reverb effect.

19.5 – 4.19.5 - Color

• LO CUT F (Low Cut Frequency): changes the frequency of the low cut filter in the reverberation path. This
control ranges from 10Hz to 20 kHz.
• LO CUT G (Low Cut Gain): changes the gain of the low cut filter. This is adjustable from -24dB to 0dB.
• HI CUT F (High Cut Frequency): changes the frequency of the high cut filter in the reverberation path. This
control ranges from 10Hz to 20 kHz.
• HI CUT G (High Cut Gain): changes the gain of the high cut filter. This is adjustable from -24dB to 0dB.

19.6 – Reflections
• TIME LEFT: changes the delay between the input signal and the early reflections in the left channel.
• LEVEL LEFT: changes the level of the reflections in the left channel.
• TIME RIGHT: changes the delay between the input signal and the early reflections in the right channel.
• LEVEL RIGHT: changes the level of the reflections in the right channel.

19.7 – Modulation Matrix, MARCOS and MACRO NAME

This section of controls is common to all the the CSR modules. Refer to paragraphs from 4.17.8 to 4.17.10 for
detailed description.

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T-RackS 6 Module Processors

20 – CSR Inverse Reverb

Reverse reverbs were popularized in the 1980’s. This type of reverb can be heard on many rock snare drums
from this time period. A Reverse reverb applies an envelope (slow attack, fast decay) to the early reflections. The
main parameter that affects the sound of this reverb is time. Time can be adjusted for the buildup and cut off of
the early reflections. This affects the sound as if it were being reversed in the reverberation, but in actuality it is
simply an envelope effect.

This type of reverb has been heavily used on drums and percussion. It is also useful with vocals or any sound
that needs an unnatural but interesting ambience. Try combining this reverb with another reverb effect to create
a complex moving ambience around the sound.

There are two operating

modes for this module:
“Easy” for a basic control
of the most commonly
used parameters, and
“Advanced” to tweak
every detail and have deep
control of the reverb shape
and tone.

The diagram shows the

controls breakdown of
Classik Studio Reverb’s
“Classik Inverse”.

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T-RackS 6 Module Processors

20.1 – Easy
• MIX: controls the ratio of reverb and original sound. 0% is all unprocessed sound. 100% is all reverb. Use
this control to increase or decrease the amount of reverb in the sound. If you are using your reverb as a send
effect, this control will probably be set to 100%.
• DIFFUSION: determines the rate that the echo density increases over time. A small diffusion setting
simulates an acoustic space with smooth walls, like a bathroom. A larger diffusion setting creates a more
open and dense sounding reverb.
• REV TIME (Reverb Time): adjusts the length of the reverb. This is a commonly used control that shortens
and lengthens the time that the reverb tail lasts.
• MID SLOPE: controls the middle and high frequency envelopes. The actual frequencies affected are
determined by the Crossover. Positive values will cause the middle to high frequencies to swell from soft to
loud over time. The higher the value the more pronounced the swell. Negative values make the mid to high
frequencies become quieter over time. More extreme settings will make this attenuation more prominent.
• LOW SLOPE: controls the shape of the low frequency envelope for the reverb. A value above 0 creates
a smooth rise from soft to loud over time. A negative value causes the low frequencies in the reverb to
become quieter over time before dropping off.
• BUILDUP: changes the speed of the reverb’s buildup. When this control is turned down, it causes a fast
attack in the reverb. To create a slower buildup in the reverberation, turn up the buildup control. Increasing
the Buildup will create the sense of a large room/hall, but the spatial information can be lost - this can be
restored with the reflection parameters (delay and level).

20.2 – I/O Levels (Advanced Mode)

• IN LEVEL: adjusts the input level of the reverb.
• OUT LEVEL: adjusts the output level from the reverb.
• MIX: controls the ratio of reverb and original sound. 0% is all unprocessed sound. 100% is all reverb. Use
this control to increase or decrease the amount of reverb in the sound. If you are using your reverb as a send
effect, this control will probably be set to 100%.
• IN IMAGE: controls the stereo image of the input signal before it is processed through the effect. This control
allows you to change the signal from normal stereo to mono and manipulate the phase of the signal. For
example a value of -45 swaps the left and right channels, 0 is mono, and 45 is normal stereo. With a setting
of 90, all mono information will be excluded from the reverb sound.
• OUT IMAGE: controls the stereo image of the effected signal. With a value of -360 to +360, this control
extends the stereo field of the processed sound. This control can be used to create mono, stereo, or
surround type effects. It can also be changed in real-time or automated using the modulation or macro
matrix.

20.3 – Time
• MID SLOPE: controls the middle and high frequency envelopes. The actual frequencies affected are
determined by the Crossover. Positive values will cause the middle to high frequencies to swell from soft to
loud over time. The higher the value the more pronounced the swell. Negative values make the mid to high

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frequencies become quieter over time. More extreme settings will make this attenuation more prominent.
• LOW SLOPE: controls the shape of the low frequency envelope for the reverb. A value above 0 creates
a smooth rise from soft to loud over time. A negative value causes the low frequencies in the reverb to
become quieter over time before dropping off.
• CROSSOVER: changes the crossover frequency between the LOW and MID DAMP controls.
• HIGH FREQ: changes the frequency for the low-pass filter used for High Frequency damping.
• HIGH DAMP: damps (attenuates) the high frequency being recirculated into the reverberation. Using High
frequency damping often gives a more natural sounding reverb.
• PREDELY (Predelay): determines the amount of time between the initial input signal and the reverberation.
Increasing the predelay can create a sense of distance and, to some degree, change the size of the acoustic
space.

20.4 – Reverb
• REV TIME (Reverb Time): adjusts the length of the reverb. In other words, this shortens and lengthens the
time that the reverb tail lasts.
• DIFFUSION: determines the rate that the echo density increases over time. A small diffusion setting
simulates an acoustic space with smooth walls, like a bathroom. A larger diffusion setting creates a more
open and dense sounding reverb.
• BUILDUP: changes the speed of the reverb’s buildup. When this control is turned down, it causes a fast
attack in the reverb. To create a slower buildup in the reverberation, turn up the buildup control. Increasing
the Buildup will create the sense of a large room/hall, but the spatial information can be lost - this can be
restored with the reflection parameters (delay and level).

20.5 – Color
• LO CUT F (Low Cut Frequency): changes the frequency of the low cut filter in the reverberation path. This
control ranges from 10Hz to 20 kHz.
• LO CUT G (Low Cut Gain): changes the gain of the low cut filter. This is adjustable from -24dB to 0dB.
• HI CUT F (High Cut Frequency): changes the frequency of the high cut filter in the reverberation path. This
control ranges from 10Hz to 20 kHz.
• HI CUT G (High Cut Gain): changes the gain of the high cut filter. This is adjustable from -24dB to 0dB.

20.6 – Reflections
• TIME LEFT: changes the delay between the input signal and the early reflections in the left channel.
• LEVEL LEFT: changes the level of the reflections in the left channel.
• TIME RIGHT: changes the delay between the input signal and the early reflections in the right channel.
• LEVEL RIGHT: changes the level of the reflections in the right channel.

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20.7 – Modulation Matrix, MARCOS and MACRO NAME

This section of controls is common to all the the CSR modules. Refer to paragraphs from 4.17.8 to 4.17.10 for
detailed description.

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21 – White Channel

The White Channel module is based on the EQ and dynamics sections of one of the most widely used,
modern large format-British mixing consoles. It can be considered the evolution of the most popular 80’s
mixing consoles, of which IK has already released a killer module, the British Channel. By the late ‘90s, this
console had become a defining tool for shaping countless hits, particularly in the Hip Hop/R & B music
scene, due to its incredibly wide bandwidth and pristine sound. It also has all of the automation features that
made it one of the most, if not the most flexible console for mixing jobs in top facilities the world over. IK’s
module faithfully recreates its very clean and transparent circuit design and the resulting sound and analog
feel, with the flexibility of mode-switching EQ and a feature-rich dynamics section.

Controls:

EQ Section:
LOW Band:
• Gain - dB: determines the amount of boost or cut in decibels (dB) for this frequency band.
• Freq - Hz: sets the frequency in hertz (Hz) at which the boost or cut happens.
• PEAK: when this switch is pressed, the LOW band changes from low-shelving type to peaking type with
a fixed Q (bandwidth).
LOW-MID Band:
• Gain - dB: determines the amount of boost or cut in decibels for this frequency band.
• Freq - KHz: sets the frequency in kilohertz (KHz) at which the boost or cut happens.
• Q: sets the width of the frequency band where extreme left results in a narrow and sharper band, while
extreme right results in a wide and broader band.
HI-MID Band:
• Gain - dB: determines the amount of boost or cut in decibels for this frequency band.

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• Freq - KHz: sets the frequency in kilohertz at which the boost or cut happens.
• Q: sets the width of the frequency band where extreme left results in a narrow and sharper band, while
extreme right results in a wide and broader band.
HI Band:
• Gain - dB: determines the amount of boost or cut in decibels for this frequency band.
• Freq - KHz: sets the frequency in kilohertz at which the boost or cut happens.
• PEAK: when this switch is pressed, the HI band changes from high-shelving type to peaking type with
fixed Q.

FILTERS Section (LOW-PASS/HI-PASS):

• FLT IN: this switch enables the filter section, which can be used independently from the rest of the EQ
section.
• Freq - Hz: sets the frequency of the high-pass filter roll-off point. The filter has a 12 dB/octave cut (18
when in B mode). In the OUT position, the filter is inactive: this effectively allows the use of the two filters
independently.
• Freq - KHz: sets the frequency of the low-pass filter roll-off point. The filter has a 12 dB/Octave cut. Just like
the other, in the OUT position the filter is inactive.
• FLT S/C: this switch makes the filter section act as the detector for the sidechain circuit of the dynamics
section. When engaged the HP and LP filters are not usable for their primary purpose.
• IN: this switch turns the whole EQ section (minus filters) on or off.
• B: this switch allows you to choose which EQ mode the White Channel will work in: when disengaged, the
EQ behavior will be the one of the “E” EQ circuit in the original console. When engaged, the EQ mode will be
the “G” EQ circuit with its distinctive frequency shifts and boost/cut/Q differences. These two different circuit
designs offer sonically distinct results, and having them both available at the press of a switch is a very
handy resource for ultimate tone tweaking.

DYNAMICS Section:
• IN: this switch enables the dynamics section.
GATE/EXP:
• EQ PRE: this switch modifies the sequence of EQ and dynamics sections. When engaged the
compressor section is shifted to before the EQ section (when disengaged, it’s after the EQ section),
effectively changing how the whole module reacts to the incoming signal in relation to the parameter
settings.
• EQ S/C: this switch enables the sidechain circuit, meaning that the EQ section is used as a detector for
the dynamics section. This is a very effective way of compressing a particular frequency that is selected
with the EQ section. When this switch is enabled the EQ section will not work the standard way and
could not be used to EQ the incoming signal (you will have to instantiate another White Channel EQ/Dyn
plug-in or use another EQ).
• THRESHOLD: this control sets the level (in dB) below which the incoming signal is attenuated by the
gate/expander circuit.
• RANGE: this parameter controls the amount of gain reduction (in dB) applied to the incoming signal
when it falls below the threshold level. The extreme counterclockwise position will have none-to-

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minimal gain reduction, while more effective gating (for more noisy or problematic tracks) is obtained by
progressively turning this control towards the 40 dB value (fully clockwise).
• RELEASE - S: this control determines the time (in seconds) it takes the expander/gate to release the
signal to its normal level once it has gone above the threshold.
• FAST ATT: this control switches the attack time it takes to the gate/expander circuit to open, to 1ms;
this can be useful to produce some very particular effects like snappy and percussive sounds with very
marked attack transients.
• EXP: this control activates the expansion mode for the gate/expander section.
• HOLD: this control sets the amount of time (in ms) the gate will stay open after the signal has fallen
below the threshold.
COMP:
• THRESHOLD: this control sets the level in dB that must be exceeded by the input signal coming into
the White Channel module for the compression to start working. At the 0 position the module is not
compressing. Turning the knob clockwise will lower the threshold and, depending on the input signal,
start compressing. Turning the knob counterclockwise towards the +10 value will lift the threshold
so that hotter signals can be easily managed. This control and its behavior can be affected by the
Master section’s I/O level controls. As an example, try driving the compressor’s input level harder with
the Master’s section IN knob and set the threshold low for some serious squashing and pumping
compression effects. The OUT knob will make calibrating the final output of the module easy in order
to avoid clipping. A note on Auto Makeup Gain: this compressor circuit design is provided of an auto
makeup gain control; this will progressively compensate the output level in relation to the threshold
control avoiding a drop in overall output level as a result of gain reduction. The lower the threshold, the
more you will hear it raise the output level, so pay attention about setting a correct gain staging among
EQ, Compressor and Master I/O levels.
• RATIO: this control determines the gain reduction amount applied to the signal passing above the level set
by the threshold control. Lower values of this control (1:1, 1.5:1) will result in no compression or very light
compression, while turning the knob clockwise will progressively make for a more pronounced effect as
you approach higher values. The extreme clockwise position will result in a limiting effect.
• FAST ATT: this switch shifts the attack time it takes the circuit to start compressing once the incoming
signal has passed the set threshold to a faster 1 ms value. It can be quite handy to emphasize attack
transients on instruments like snare or kick drums and other percussive sources.
• RELEASE - S: this control sets the time (in seconds) it takes the compressor to release the signal back
to uncompressed state after it has gone below the threshold level. Shorter release times will make the
compressor snappy to downright “pumping” when lots of compression is applied. Longer release times
will result in a more natural sounding effect and increased sustain on particular instruments.
• HARD: this switch changes the detecting mode of the compressor from RMS to peak.

MASTER Section:
• Ø (Phase): this control shifts the phase of the incoming signal by 180°, useful to check phase compatibility
of sources or to reverse phase in one of two coupled tracks (like snare drum top and bottom).
• IN: this control lets you trim the input level of the signal going into the White Channel module. It can be
used to correct the level of the incoming signal when needed (in case of a particularly weak or hot track)
since it has a range of +/- 15 dB. As described above (in the compressor’s threshold parameter section) it
can be used to drive the compressor section harder by boosting the input signal, thus producing some very

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particular compression effects, which can be further emphasized by ratio, attack and release time settings.
• OUT: this control lets you trim the master output level of the plug in. It can be used to avoid clipping and
overloading of the output stage in case some very radical EQ boosting is applied, or to simply control the
level of a track that needs to be adjusted.
• RESET: when clicked, this control brings the settings of all of the parameters back to their default.
• L=R: links the two channels for stereo operation: all parameters affect the two channels in the same way to
maintain a consistent stereo image.
• L/R: click on the L or R icons if you want to dial in specific compression settings for L and R channels
independently.
• M/S: this option uses a Mid-Side matrix to insert the stereo compressor on your processing chain. This
means that you will be able to independently compress the Center or the Sides of your stereo program.
The L and R icons will be replaced by M and S, when in this mode. Click M to set the compression for the
Center (Mid) of your stereo image, or S to set compression for the sides.

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22 – Bus Compressor

The Bus Compressor is modeled after the stereo master-bus compressor built into one of the most popular
British analog mixing consoles, which was a staple in the 80’s and 90’s pop and rock music scene (the same
console our acclaimed British Channel is derived from). This compressor was so in-demand that it was also
released as a separate, outboard rack unit. Due to its VCA circuit design and operation, it is known to impart
a very defined sonic signature to the audio program, making all tracks in a mix “blend” together in a very
musical way. As a result, its nickname among the audio engineering and recording studio insiders is the “glue
compressor.” Most often used on the mix bus or on the drum sub-master, this is an essential piece of kit to use
for imparting punch and boldness to any mix.

Controls:

• THRESHOLD: this control sets the minimum level above which the signal is compressed. Lower settings
of this control result in a more compressed signal. This control has no fixed settings; it offers continuously
variable values from +20 to -20.
• RATIO: this control sets the behavior of the compressor related to the signal hitting the threshold. A setting
of 4:1 means that signal exceeding the set threshold will be reduced by four times (4 dB over the threshold
will reduced to 1dB).
• ATTACK: this control sets the time it takes the compressor to start its action once the signal has exceeded
the threshold.
• RELEASE: this control sets the time it takes the compressor to release the signal back to its uncompressed
state once its level falls below the set threshold.
• MAKEUP: this control is used to compensate for the loss of level in the program that can be produced by
particularly heavy compression settings.
• SC HI PASS: this control sets the corner frequency (in Hz) for the hi-pass filter, which can be used before
the detection circuit of the compressor to prevent too much bass-frequency content from modulating
the compression. The hi-pass filter slope is a fixed 24 db/octave. This control is particularly useful when

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compressing mixes with heavy bass content. Setting it fairly high prevents elements like kick/bass from
making the compressor “pump”. At lower settings, “pumping” (which is a popular effect in dance and electro
mixes) is more likely.
• COMPRESSION IN: this control turns the compression circuit ON or OFF.
• GRIT: this control engages second harmonic distortion when pressed. It is a very subtle effect but
perceivable particularly with “cleaner” mix material. When engaged, the compressor behaves like an out-of-
calibration hardware unit, imparting a gritty sound that has become popular among recording engineers.
• RESET: this switch brings the module back to its default state when pressed.
• L=R: links the two channels for stereo operation: all parameters affect the two channels in the same way to
maintain a consistent stereo image.
• L/R: click on the L or R icons if you want to dial in specific compression setting for L and R channels
independently.
• M/S: this option uses a Mid-Side matrix to insert the stereo compressor on your processing chain. This
means that you will be able to independently compress the center or the sides of your stereo program. The
L and R icons will be replaced by M and S when in this mode. Click M to set the compression for the Center
(mid) of your stereo image or S to set compression for the sides.

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23 – Tape Echo (EchoFlex)

The Tape Echo (EchoFlex) is based on the classic hardware Echoplex EP-3 tape echo unit, which was one of the
first echo devices used in audio applications and musicians’ setups. Based on a standard 1/4” magnetic tape
reel, just like the one used in standard studio multitrack recorders, it uses 3 tape heads, one of which can be
moved with the linked “time bar,” to set the length of the echo produced. These factors contribute to the classic
“vintage” sound of the analog echo, which has a natural high-frequency damping as the repeats decay. The
simple interface makes it a very intuitive and fast echo device, which can also be used for very creative effects
when pushed to extreme settings. The Tape Echo has been modeled down to the tape behavior and anomalies
like wow and flutter, tape wear, hum/noise and rec level (for tape saturation), putting these highly flexible and
creative parameters at the user’s disposal.

Controls:

• INPUT LEVEL: this control sets the input level of the signal going into the module. Boosting the level to +15
clips the input stage electronics creating analog saturation.
• VOLUME A/B: these controls regulate the level of the delayed signal independently for the two channels
(dry/wet ratio). When A or B control is set to minimum the signal is completely dry, meaning that the tape
effect is bypassed and the signal goes through preamp electronics only.
• SUSTAIN A/B: these controls regulate the decay /repeats of the delayed signal independently for the two
channels. Setting decay to maximum sends the unit into feedback, which can be used to create some wild
effects.
• TIME Bar: this control sets the delay time for the A and B channels. When the module is set for A/B
independent operation, the bar will switch on the A or B setting according to what channel is selected by the
MODE switch. When the BPM sync switch is set to ON, the Time Bar will switch from continuously variable
to fixed steps that correspond to the time divisions described in the Echo Time display. Moving the Time Bar
during playback creates a “glide” effect of tape speeding up and slowing down.
• ECHO TIME Display: this display shows the time set for each channel in milleseconds (ms), moving with

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the time bar. When the Mode switch is set to A or B the value shown will correspond to the one set for that
channel. When the BPM Sync switch is set to ON, the shown value with change from ms to time divisions.
Possible values are: 1/1, 1/2d, 1/2t, 1/2, 1/4d, 1/4t, 1/4, 1/8d, 1/8t, 1/8, 1/16d, 1/16t, 1/16, 1/32.
• MODE Switch: this 3-position switch (A, B, STEREO) determines the behavior of the module with relation
to the delay time. When set to STEREO, delay time for both channels is the same, and moving the time bar
will change both. When set to A or B the delay times are independent for each channels. When set to A, the
Time Bar will show the A-channel delay time; setting it to B will show the corresponding B-channel delay
time.
• BPM SYNC Switch: this switch will make the delay times sync to the BPM set in the host DAW session.
• ECHO Switch: this switch is the master bypass of the module.
• Meter Select Swicth (IN/OUT): this control will make the meter show the input or output level according to
the position selected.
• RESET Switch: this control will reset the module back to its default state. Clicking a second time will bring
the module to its last setting.
• ECHOFLEX Logo: clicking on this logo will open the hatch underneath showing the trim pots. Clicking on
the border of the open hatch will close it.

• REC LEVEL Trim Pot: this control adjusts the level of the signal hitting the tape. At the highest settings
the tape will start to saaturate in a more evident manner, an effect which can be used to create interesting
results.
• TAPE WEAR Trim Pot: this control adjusts the high frequency damping due to tape wear and use.
• WOW/FLUTTER Trim Pot: this control adjusts the global wow and flutter effect typical of an out aligned
tape machine. At the most extreme setting the effect is particularly aggressive with pitch and speed changes
(just like in the real unit), while at the minimum setting the effect is that of a perfectly aligned tape.
• NOISE Trim Pot: this control adjusts the amount of tape hiss and preamp noise in the signal. At the most
extreme setting also AC line hum will became very audible; the hum’s frequency can be selected to reflect
either US 60Hz or EU 50Hz.

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24 – Precision Comp/Limiter

The Precision Comp/Limiter is based on one of the legendary British compressors found in top-level studios
worldwide. Originally produced in the ‘70’s, it quickly became a much sought after unit for many audio
applications, including music recording/mixing, mastering, post-production and broadcasting. Its sonic
character is not as aggressive as that of many other popular units of the same era. Even when the unit is
pushed to extreme settings and despite its fixed 1ms attack time, the compression is fairly gentle and retains
a musical quality that doesn’t “choke” or squash program material. Nevertheless, the compression effect is
clearly distinguishable. When the limiter is kicked in, the gain reduction becomes more pronounced, especially
with lower limiter thresholds. The compressor and limiter can operate independently from each other. The
compression is achieved via a set of diodes, thus the original hardware’s name “discreet”. The overall effect is
a nice “fat and thick” sound (a trademark of this British manufacturer), which doesn’t lose focus or detail. These
characteristics are also aided by the set of transformers used in the signal path.

Controls:

COMPRESSOR:
• INPUT dB: sets the input level in dB, and is useful for fine-tuning the level hitting the threshold. The
minimum resolution for each step is 1.5 dB.
• THRESHOLD dBU: sets the level above which compression takes place; the lower the threshold the more
compressed the signal.
• RECOVERY ms: sets the time it takes the compressor to release the signal back to its uncompressed state;
available values are: 100ms, 400ms, 800ms, 1500ms, Auto 1 and Auto 2.
• GAIN: sets the amount of makeup gain applied to the compressor’s circuit output. Use it to compensate for
gain loss from the compression effect.
• RATIO: sets the amount of dBs the signal is reduced by once it gets past the set threshold. A ratio of 3:1
means that if the signal is exceeding the threshold by 3 dB it will be reduced to 1 dB.
• COMP IN: this switch engages the compression circuit.

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LIMITER:
• THRESHOLD dBU: sets the level above which limiting takes place; the lower the threshold the more
compressed the signal.
• RECOVERY ms: sets the time it takes the limiter to release the signal when its level falls below the set
threshold; available values are: 50ms, 100ms, 200ms, 800ms, Auto 1 and Auto 2.
• OUTPUT dB: sets the output level of the limiter section. It’s useful for raising the signal level after gain
reduction.
• ATTACK FAST/SLOW: this switch allows you to select between 2 different attack times, a slower 2ms and a
faster 1ms.
• LIMIT IN: this switch engages the limiter circuit.

MASTER Section:
• POWER: turns the module on or off, effectively bypassing it.
• RESET: resets all parameters to their default status once clicked.
• L=R: links the two channels for stereo operation: all parameters affect the two channels in the same way to
maintain a consistent stereo image.
• L/R: click on the L or R icons if you want to dial in specific compression settings for L and R channels
independently.
• M/S: this option uses a mid-side matrix to insert the stereo compressor on your processing chain. This
means that you will be able to independently compress the center or the sides of your stereo program. The
L and R icons will be replaced by M and S when in this mode. Click M to set the compression for the center
(mid) of your stereo image or S to set compression for the sides.

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25 – Master EQ 432

This module is based on what is considered to be the “golden reference” standard in mastering equalizers. This
piece of hardware was the brainchild of two of the most ingenious minds in music recording technology, true
pioneers who joined forces to create what was the first parametric EQ ever invented. The design quickly became
the basis for subsequent parametric EQs from manufacturers the world over. On top of the fully parametric and
totally independent bands with almost no interaction (something that was not available until this circuit topology
invention), its success came from its no-compromise design based on discrete, high-performance, custom
op-amps and a transformerless signal path yielding pristine, transparent audio. This was a key factor in defining
it as the first-call EQ among mastering engineers which it still is today. This module is truly “as good as it gets,”
the secret weapon to give mixes that master-quality final polish with unrivaled high end air, solid low end and
extreme detail in the critical midrange spectrum.

Controls:

Shelving Bands:
• HI SHELVING: determines the gain of the HI shelving band. The frequency of the band is fixed at 10 KHz.
• LO SHELVING: determines the gain of the LOW shelving band. The frequency of the band is determined by
the selector.
• LO SHELVING Frequency Selector: sets the frequency of the low shelving band. The settings are 50Hz or
100Hz.

LO Parametric Band:
• Level: sets the boost or cut (in dB) of the low parametric band.
• Frequency: sets the frequency of the low parametric band.
• Shape: sets the Q of the LO parametric band. The settings are 5, 6, 9, 11, 15 dB/Octave.

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MID Parametric Band:

• Level: sets the boost or cut (in dB) of the mid parametric band.
• Frequency: sets the frequency of the mid parametric band.
• Shape: sets the Q of the mid parametric band. The settings are 5, 6, 9, 11, 15 dB/Octave.

HI Parametric Band:
• Level: sets the boost or cut (in dB) of the high parametric band.
• Frequency: sets the frequency of the high parametric band.
• Shape: sets the Q of the high parametric band. The settings are 5, 6, 9, 11, 15 dB/Octave.

Other:
• PUSH EQ: works as the global ON/OFF switch for the module.
• RANGE dB: sets the range of the gain knobs for all frequency bands; possible vaules are +/- 6, +/- 9, +/-
12 dB. This allows to have a different resolutions per step of the gain knobs in case finer adjustments are
needed.
• OUT: sets the boost or cut to the output level (in dB) of the module.
• RESET: when clicked, this control brings the settings of all of the parameters back to their default.
• L=R: links the two channels for stereo operation: all parameters affect the two channels in the same way to
maintain a consistent stereo image.
• L/R: click on the L or R icons if you want to dial in specific EQ settings for L and R channels independently.
• M/S: this option uses a Mid-Side matrix to insert the stereo EQ in your processing chain. This means that
you can independently equalize the Center or the Sides of your stereo program. The L and R icons are
replaced by M and S in this mode. Click M to set the EQ for the Center (Mid) of your stereo image or S to
set EQ for the sides.

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26 – EQ 73

The EQ 73 is based on what is probably acknowledged as the most universally used preamp in music recording
history. This class A device has a sonic signature that is almost instantly recognizable to engineers and recording
enthusiasts the world over. Thanks to this particular circuitry the signal passing through it becomes thick and
bold making every instrument stand out with fullness. To bridge the classic sound with modern functionalities
the control layout was improved over the original design, and a few features were added like the preamp circuit
saturation.

Controls:

• HI SHELVING: determines the gain of the HI shelving band. The frequency of the band is fixed at 10 KHz.
• IN: sets the input level of the modeled mic preamp section. Increasing gain will result in more level and
progressively more saturation of the processed signal, up to a gritty level of distortion. The range is +/-20dB.
• OUT: sets the output level of the module. It can be used to compensate the variability of the input level in
both positive or negative. The range is +/-20dB.
• Hi-Pass Filter: sets the frequency of the Hi-Pass filter. The slope of the filter is 12 dB/Oct. Available
frequencies are: 50Hz, 80Hz, 160Hz, 300Hz. OFF position switches the band out of the signal path.
• Low Shelf Band Freq: sets the frequency of the Low Shelf Band. Possible frequencies are: 35Hz, 60Hz,
110Hz, 220Hz. OFF position switches the band out of the signal path.
• Low Shelf Gain: sets the gain of the Low Shelf Band. Range is +/- 16 dB.
• Presence Band Freq: sets the frequency of the Presence (mid) band. Possible frequencies are: 360Hz,
700Hz, 1.6KHz, 3.2Khz, 4.8KHz, 7.2KHz. OFF position switches the band out of the signal path.
• Presence Band Gain: sets the gain of the Presence (mid) band. Range is +/- 16 dB.
• High Shelf Gain: sets the gain of the HI Shelf band. Range is +/- 16 dB. Frequency is fixed to 12KHz.
• Line/Mic: select the operative level of the input section.

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• EQ: this switch activates the EQ controls. When EQ is not active the signal is still passing through the
preamp section.
• Phase: inverts the phase of the signal being processed by the module.
• ON: turns on or off the whole module.
• RESET: when clicked, this control brings the settings of all of the parameters back to their default.
• L=R: links the two channels for stereo operation: all parameters affect the two channels in the same way to
maintain a consistent stereo image.
• L/R: click on the L or R icons if you want to dial in specific EQ settings for L and R channels independently.
• M/S: this option uses a Mid-Side matrix to insert the stereo EQ in your processing chain. This means that
you can independently equalize the Center or the Sides of your stereo program. The L and R icons are
replaced by M and S in this mode. Click M to set the EQ for the Center (Mid) of your stereo image or S to
set EQ for the sides.

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27 – EQ 81

This module is based on another classic piece of hardware whose character was instrumental in shaping the
sound of many hit songs from iconic musicians and producers; the EQ 81 can be considered the evolution of
its older brother, the EQ 73. There are a few twists which make the EQ 81 capable of delivering a voice of its
own. The most evident difference here is in the versatility of the EQ section, sporting four independent bands
(high / low shelving, plus two midrange peaking filters) rounded out by high and low pass filters; this module
is particularly useful when a more precise intervention on a track is needed, still maintaining tone and organic,
analog warmth throughout the signal path. As for the EQ 73, a special attention was given to the perfect
matching of the EQ curves, but a step further was taken into the modelling of the analog preamp section in all of
its colorful nuances.

Controls:

• IN: Sets the input level of the modeled mic preamp section. Increasing gain will result in more level and
progressively more saturation of the processed signal, up to a gritty level of distortion. The range is +/-20dB.
• OUT: sets the output level of the module. It can be used to compensate the variability of the input level in
both positive or negative. The range is +/-20dB.
• Hi-Pass Filter: sets the frequency of the Hi-Pass filter. The slope of the filter is 18 dB/Oct. Available
frequencies are: 27Hz, 47Hz, 82Hz, 100Hz, 270Hz. OFF position switches the filter out of the signal path.
• Low-Pass Filter: sets the frequency of the Low-Pass FIlter. The slope of the filter is 12 dB/Oct. Available
frequencies are: 3.9KHz, 5.6KHz, 8.2KHz, 12KHz, 18KHz. OFF position switches the filter out of the signal
path.
• Low Band Freq: sets the frequency of the Low Shelf Band. Possible values are: 33Hz, 56Hz, 100Hz, 180Hz,
330Hz. OFF position switches the band out of the signal path.
• Low Band Shelf/Bell: switches the Low Band from shelving to peaking mode with a constant 0.2 Q.
• Low Band Gain: sets the gain of the Low Band. Available Range is +/- 15 dB.

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• Low Mid Band Freq: sets the frequency of the Low Mid Band. Possible values are: 220Hz, 270Hz, 330Hz,
390Hz, 470Hz, 560Hz, 680Hz, 820Hz, 1000Hz, 1200Hz. OFF position switches the band out of the signal
path.
• Low Mid band Gain: sets the gain of the Low Mid Band. Available range is +/-10 dB.
• Low Mid Band Hi-Q: switches the bandwidth of the Low Mid Band from a constant 0.5 to a gain dependant
1.0 with a slightly more pronounced gain range.
• High Mid Band Freq: sets the frequency of the High Mid Band. Possible values are: 1.5KHz, 1.8KHz,
2.2KHz, 2.7KHz, 3.3KHz, 3.9KHz,4.7KHz, 5.6KHz, 6.8KHz, 8.2KHz. OFF switches the band out of the signal
chain.
• High Mid Band Gain: sets the gain of the High Mid Band. Available range is +/- 10 dB.
• High Mid Band Hi-Q: Switches the bandwidth of the High Mid Band from constant 0.3 to a gain dependent
1.0 with a slightly more pronounced gain range.
• High Band Frequency: sets the frequency of the High Band. Possible values are: 3.3KHz, 4.7KHz, 6.8KHz,
10KHz, 15KHz. OFF position switches the band out of the signal path.
• High Band Gain: sets the gain of the High Band. Available range is +/- 15 dB.
• High Band Shelf/Bell: switches the Low Band from shelving to peaking mode with a constant 0.2 Q.
• Phase: inverts the phase of the processed signal.
• EQ: turns the EQ section on or off; when EQ is off signal is still passing through the preamp section.
• ON: turns the whole module on or off.
• RESET: when clicked, this control brings the settings of all of the parameters back to their default.
• L=R: links the two channels for stereo operation: all parameters affect the two channels in the same way to
maintain a consistent stereo image.
• L/R: click on the L or R icons if you want to dial in specific EQ settings for L and R channels independently.
• M/S: this option uses a Mid-Side matrix to insert the stereo EQ in your processing chain. This means that
you can independently equalize the Center or the Sides of your stereo program. The L and R icons are
replaced by M and S in this mode. Click M to set the EQ for the Center (Mid) of your stereo image or S to
set EQ for the sides.

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28 – EQ PA

The EQ PA is based on one of the most famous classic American EQ circuit designs. This three band delivers
a thick sound and excellent tone shaping capabilities thanks to it’s proportional Q design and added preamp
stage which will impart a distinctly “fat and punchy” sonic character.

Controls:

• LF Gain: controls the amount of gain of the Low band. Range is +/-12 dBs. The gain is stepped so that
resolution is 1dB from 0 to +/-6 dBs and 1.5 dBs from +/-6 to +/-12 dBs. Intermediate steps are not shown
on labels but available to the user.
• LF Frequency: sets the frequency of the Low band; available values are 30, 40, 50, 100, 200, 300, 400 Hz.
• LF SHELF: sets the Low band in peaking or shelving mode.
• MF Gain: controls the amount of gain of the Mid band. Range is +/-12 dBs. The gain is stepped so that
resolution is 1dB from 0 to +/-6 dBs and 1.5 dBs from +/-6 to +/-12 dBs. Intermediate steps are not shown
on labels but available to the user.
• MF Frequency: sets the frequency of the Low band; available values are 200, 400, 600, 800, 1500, 3000,
5000 Hz.
• HF Gain: controls the amount of gain of the Hi band. Range is +/-12 dBs. The gain is stepped so that
resolution is 1dB from 0 to +/-6 dBs and 1.5 dBs from +/-6 to +/-12 dBs. Intermediate steps are not shown
on labels but available to the user.
• HF Frequency: sets the frequency of the Hi band; available values are 2.5, 5, 7, 10, 12.5, 15, 20 KHz.
• HF SHELF: sets the Hi band in peaking or shelving mode.
• PREAMP: engages the preamp stage. Adjusting the gain when the preamp is engaged will add the
harmonic saturation from this part of the circuitry, whose results can be either a subtle harmonic
enhancement to the track or a very pronounced distortion. Use it to taste for many creative results.
• GAIN: controls the amount of preamp gain when the “Preamp” stage is engaged. Range is +/- 20 dB. When

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the Preamp stage is not engaged the Gain knob act as an input level control for the module with the same
+/- 20 dB of range.
• PHASE: inverts the phase of the signal going through the module by 180°.
• OUT: sets the overall output level of the module; range is +/- 20 dB.
• BAND PASS: engages the band-pass filter; the filter high-passes the signal with a broad 4th order high pass
at 100 Hz, while the low pass filter starts the 4th order slope at 5 KHz.
• EQ: turns the EQ section of the module on or off. GAIN and OUT controls remain active and preamp stage
can be engaed as well.
• RESET: when clicked, this control brings the settings of all of the parameters back to their default position.
• L=R: links the two channels for stereo operation: all parameters affect the two channels in the same way to
maintain a consistent stereo image.
• L/R: click on the L or R icons if you want to dial in specific EQ settings for L and R channels independently.
• M/S: this option uses a Mid-Side matrix to insert the stereo EQ in your processing chain. This means that
you can independently equalize the Center or the Sides of your stereo program. The L and R icons are
replaced by M and S in this mode. Click M to set the EQ for the Center (Mid) of your stereo image or S to
set EQ for the sides.

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29 – EQ PB

Similar to its EQ PA brother, this module adds a fourth band for increased flexibility, particulary in the midrange
part of the spectrum, where the two overlapping bands can create complex shapes, allowing deep sculpting of
a track. Combined with the preamp stage this can be a toneful machine capable to deliver a vast array of tones.

Controls:

• LF Gain: controls the amount of gain of the Low band. Range is +/-12 dBs. The gain is stepped so that
resolution is 1dB from 0 to +/-6 dBs and 1.5 dBs from +/-6 to +/-12 dBs. Intermediate steps are not shown
on labels but available to the user.
• LF Frequency: sets the frequency of the Low band; available values are 30, 40, 50, 100, 200, 300, 400 Hz.
• LF SHELF: sets the Low band in peaking or shelving mode.
• LMF Gain: controls the amount of gain of the Low Mid band. Range is +/-12 dBs. The gain is stepped so
that resolution is 1dB from 0 to +/-6 dBs and 1.5 dBs from +/-6 to +/-12 dBs. Intermediate steps are not
shown on labels but available to the user.
• LMF Frequency: sets the frequency of the Low band; available values are 75, 150, 180, 240, 500, 700, 1000
Hz.
• HMF Gain: controls the amount of gain of the Hi Mid band. Range is +/-12 dBs. The gain is stepped so that
resolution is 1dB from 0 to +/-6 dBs and 1.5 dBs from +/-6 to +/-12 dBs. Intermediate steps are not shown
on labels but available to the user.
• HMF Frequency: sets the frequency of the Low band; available values are 800, 1500, 3000, 5000, 8000,
1000, 12500 Hz.
• HF Gain: controls the amount of gain of the Hi band. Range is +/-12 dBs. The gain is stepped so that
resolution is 1dB from 0 to +/-6 dBs and 1.5 dBs from +/-6 to +/-12 dBs. Intermediate steps are not shown
on labels but available to the user.
• HF Frequency: sets the frequency of the Hi band; available values are 2.5, 5, 7, 10, 12.5, 15, 20 KHz.

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• HF SHELF: sets the Hi band in peaking or shelving mode

• PREAMP: engages the preamp stage. Adjusting the gain when the preamp is engaged will add the
harmonic saturation from this part of the circuitry, whose results can be either a subtle harmonic
enhancement to the track or a very pronounced distortion. Use it to taste for many creative results.
• GAIN: controls the amount of preamp gain when the “Preamp” stage is engaged. Range is +/- 20 dB. When
the Preamp stage is not engaged the Gain knob act as an input level control for the module with the same
+/- 20 dB of range.
• PHASE: inverts the phase of the signal going through the module by 180°.
• OUT: sets the overall output level of the module; range is +/- 20 dB.
• EQ: turns the EQ section of the module on or off. GAIN and OUT controls remain active and preamp stage
can be engaed as well.
• RESET: when clicked, this control brings the settings of all of the parameters back to their default position.
• L=R: links the two channels for stereo operation: all parameters affect the two channels in the same way to
maintain a consistent stereo image.
• L/R: click on the L or R icons if you want to dial in specific EQ settings for L and R channels independently.
• M/S: this option uses a Mid-Side matrix to insert the stereo EQ in your processing chain. This means that
you can independently equalize the Center or the Sides of your stereo program. The L and R icons are
replaced by M and S in this mode. Click M to set the EQ for the Center (Mid) of your stereo image or S to
set EQ for the sides.

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30 – EQ PG

This module uses the same proportional Q approach and all discrete based signal path but in a ten bands
graphic fashion instead of a parametric type. This makes for several differences in that the equalization can
be applied on larger parts of the spectrum by acting on several bands at once. This, on top of the clear visual
shape of the equalization applied, results in massive yet fast tonal shaping capabilities, which combined with the
onboard preamp stage makes for a rather powerful tool, beyond simple equalization.

Controls:

• EQ Bands: these controls set the gain for the ten frequency bands. Range is +/- 12 dB. Available
frequencies are: 30Hz, 63Hz, 125Hz, 250Hz, 500Hz, 1KHz, 2KHz, 4KHz, 8KHz, 16Khz.
• PREAMP: engages the preamp stage. Adjusting the gain when the preamp is engaged will add the
harmonic saturation from this part of the circuitry, whose results can be either a subtle harmonic
enhancement to the track or a very pronounced distortion. Use it to taste for many creative results.
• GAIN: controls the amount of preamp gain when the “Preamp” stage is engaged. Range is +/- 20 dB. When
the Preamp stage is not engaged the Gain knob act as an input level control for the module with the same
+/- 20 dB of range.
• PHASE: inverts the phase of the signal going through the module by 180°.
• OUT: sets the overall output level of the module; range is +/- 20 dB.
• EQ: turns the EQ section of the module on or off. GAIN and OUT controls remain active and preamp stage
can be engaed as well.
• RESET: when clicked, this control brings the settings of all of the parameters back to their default position.
• L=R: links the two channels for stereo operation: all parameters affect the two channels in the same way to
maintain a consistent stereo image.
• L/R: click on the L or R icons if you want to dial in specific EQ settings for L and R channels independently.
• M/S: this option uses a Mid-Side matrix to insert the stereo EQ in your processing chain. This means that

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you can independently equalize the Center or the Sides of your stereo program. The L and R icons are
replaced by M and S in this mode. Click M to set the EQ for the Center (Mid) of your stereo image or S to
set EQ for the sides.

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31 – Stealth Limiter

The Stealth Limiter is a mastering peak limiter whose main feature is the extreme transparency of the
processing. The technology used under the hood makes it so clean and capable of raising the level of a track
without introducing any side effects or distortion, even when very high amounts of gain reduction are applied
and where other limiters start to fail.

Controls:

• INPUT: sets the input gain that will be added to the program material; basically, how loud you want to be.
• OUT CEILING: sets the maximum output level of the processed audio. The default setting for this control
is -0.1 dB. We suggest that you never exceed -0.2 dB or -0.1 dB here to preserve the audio and avoid
distortion by the audio playback system.

MAIN Section Controls:

• Bypass: bypasses the processing.
• Unity Gain Monitor: when engaged this control allows you to compare processed and unprocessed sounds
at the same level; whatever input gain is added to the audio, it will be compensated in the opposite direction,
thus avoiding the pitfall of “louder sounds better” and allowing for a more precise and objective evaluation of
the processing applied to the audio.
• Infrasonic Filter: this control engages a 4th order high pass filter with a 22Hz cutoff frequency. This is very
useful to get rid of infrasonic and very low frequency rumbles which could put excessive workload on the
limiter.

METERING Section:
The Stealth Plug limiter includes a full featured level metering system to help evaluating the correct amount
of limiting being applied, by just looking at the Limiter interface.

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• INPUT: this meter shows both Peak and RMS level of the signal that is entering into the limiter. Peak level is
shown with the brigther bar and RMS is shown on the darker bar. You can also read instantaneous numeric
values for both Peak and RMS levels right above and at the bottom of the meters.
• OUTPUT: this meter shows both Peak and RMS level of the signal once boosted by the limiter. Peak level
is shown with the brighter bar and RMS is shown on the darker bar. You can also read instantaneous
numeric values for both Peak and RMS levels right above and at the bottom of the meters. As a very rough
indication, loud masters will show RMS values of -9 / -8 dB RMS. Higher levels than this might result in
distorted audio. Consider this as a very rough indication that only applies to commercial music.
• REDUCTION: this meter indicated the Peak reduciton amout. The more the limiter works the more this
meter will illuminate.
• INTERSAMPLE PEAK LIMITER: when this sign is displayed the limiter is working in ISPL mode.

MODE Section Controls:

The controls in this section allow you to set the operating mode of the limiter and the kind of processing
going on under the hood.
• Tight: this is T-RackS Stealth Limiter’s default mode, and its most neutral. It respects the source material’s
natural sonic dynamics the most. Even when set to the highest limiting setting, the music will never sound
“compressed” or “constrained.” With this setting, you can increase loudness while maintaining a neutral
effect on sonic timbre and dynamics.
• Balanced: like its name suggests, this is a more balanced processing that can be used when a small
amount of dynamic compression is acceptable (or desirable). The difference between this mode and TIGHT
is so subtle that the two are only subtly distinguishable at high loudness levels and with extreme limiting.
• Harmonics 1: this mode adds the slight harmonic saturation of a Class A tube circuitry to the party, so that
some color is added to the music program. Still, being the harmonic enhancement stage “pre” limiting, the
transparency of the limiting process is maintained. The amount of harmonic enhancement is determined
by the amount of signal that enters into the plug-in. There is no control over this on the limiter itself, so to
experiment with more or less “coloration” just adjust the signal level before the Stealth Limiter plug-in.
• Harmonics 2: like the Harmonics 1 mode, this mode adds a nice solid state harmonic saturation for a
different flavor and slight more “edge” that can be useful to impart a bit more character to a weaker mix.
Also for Harmonics 2 mode, the amount of harmonic enhancement is determined by the amount of signal
that enters into the plug-in. There is no control over this on the limiter itself, so to experiment with more or
less “coloration” just adjust the signal level before the Stealth Limiter plug-in.

ISPL Section Controls (Inter Sample Peak Limiting)

When engaged, this control allows the module to run at an higher sampling rate and enables a special
limiting mode that prevents unexpected possible distortion on playback devices even if the output samples
level never exceed 0dBFS.
The ISPL limiter can work at two different rates, 4X and 16X.
Use 4X for an optimal balance between intersample peak limiting performance and CPU usage.
16X gives a slighly more safer intersample peaks protection, at the expense of a very high CPU load.
When ISPL is not engaged the limiter runs at the current session’s sample rate and it is not guaranteed to
protect the program against intersample peaks..

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Note on Intersample Peak Limiting

With traditional mastering processes, it’s normal to push the volume level towards a -0.1dB or even 0.0dB
ceiling. Spikes above this ceiling are suppressed via the use of hard limiters or clippers. Yet these methods
only ensure a stable digital level, they do not guarantee undistorted playback once the music is translated
into an analog signal.
This is because the process of playback involves a number of conversion processes that, though the digital
waveform may be below 0.0dB, can cause the analog waveform to exceed this threshold and introduce
unwanted distortion and fuzz.
T-RackS Stealth Limiter’s inter-sample peak limiting algorithm gets around this issue by predicting the
converted analog waveform of the digital source. Through this process it’s able to limit the sound in a way
that ensures the final converted audio will never or very rarely exceed the set ceiling. This is useful in the
modern era where music is often heard on different devices and via different services that all deliver their
music via compressed formats, like MP3 or AAC.

DITHERING Section Controls:

The Stealth Limiter can add a dithering noise to the audio program to make the bit depth reduction that will
eventually be applied at a later stage to sound best.
The controls in this section allow to set the level of the dithering noise depending on the bit depth the
material will be reduced to (16 or 24 bits).
Please note: the Stealth Limiter does not apply any bit reduction or truncation, which has to be applied
elsewhere later in the processing chain by the T-RackS standalone application or by the DAW of choice

General Controls:
• RESET: brings all controls to the default state.
• L=R: links the two channels for stereo operation: all parameters affect the two channels in the same way to
maintain a consistent stereo image. Click on the L or R icons if you want to dial in specific settings for L and
R channels independently.

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32 – Mic Room

Mic Room is a microphone-modeling module based on IK’s Multimedia patented technology. It gives you
the power to process and “re-mic” your physical microphones to sound and perform like some of the most
coveted mic models of all time. A vast selection of the most popular timeless classic and contemporary
microphones used in recording studios and live stages around the world has been squeezed into this virtual
“mic locker” and made ready for DAW use. A simple and effective approach allows you to have “any mic
to any mic” in both source and model fields. This allows for tremendous flexibility when used in the DAW
environment. Now, crafting the perfect vocal or instrument sound is possible, even when the original track has
been recorded with a less than optimal dynamic, or vice versa, when you need to “grunge up” a too hi-fi track.

Controls:

• SOURCE: selects the mic that was used to record the track you are processing with Mic Room. If such
a mic is not in the list, try using a similar kind of microphone (e.g. same diaphragm size, same mic type,
condenser, ribbon or dynamic), for a compatible result. Selection can be performed via the microphone field
and popup menu, or via the selection arrows.
• TARGET: use this control to select the microphone you want the source track to sound like. Selection can be
performed via the microphone field and popup menu, or via the selection arrows.
• INPUT GAIN: this knob controls the gain applied to the input of the plug-in. Use this to adjust the level of the
processed track so that the meter peaks to a ballpark between halfway to three quarters of its length. Range
is +/- 24 dBs.
• PROXIMITY: this knob controls the amount of the proximity effect which is typical of directional
microphones. The closer a source is to the microphone, the more the low end increases. Turn this control
clockwise to augment the proximity effect. Counterclockwise to diminish it. This can be used to fine tune
the bass amount in an previously recorded track without having to use an external EQ, and by naturally
intervening on the selected mics frequency response curve. Range is +/- 5 dBs.
• HARMONICS: this knob controls the amount of harmonic non linearity which can be added to Target

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microphones. It can be used to add subtle character to a flat or dull track, or when exaggerated to add a
perceivable “grit”. Range is 0 to 10 dBs.
• IN: this button engages the module’s processing.
• L=R: links the two channels for stereo operation: all parameters affect the two channels in the same way to
maintain a consistent stereo image.
• L/R: click on the L or R icons if you want to dial in specific mic type settings for L and R channels
independently.
• M/S: this option uses a Mid-Side matrix on your processing chain. This means that you will be able to
independently process the Center or the Sides of your stereo program. The L and R icons will be replaced
by M and S when in this mode. Click M to set the mic types for the Center (Mid) of your stereo image or S to
set mic types for the Sides.

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33 – Saturator X

The new T-RackS Saturator X delivers that classic analog saturation and takes you on a trip back in time
to the birth of “modern” recording before computers and opens your recordings to the mysterious, elusive
and warm world of analog saturation. The effect of Saturation, or the introduction of harmonic distortion,
on an audio track or mix can add that certain special “flavor” and “warmth” that only magnetic tape, tubes,
transformers or transistors can deliver.

Each of the ten saturation modes in the Saturator X reproduce the behavior of the electronic hardware
component and circuitry (or recording medium, in the case of tape modes) that has been modeled for its
peculiar character and different harmonic patterns it produces when driven hard. This makes for a very versatile
and comprehensive saturation device as the palette of achievable tones varies from the very subtle “color” of
the transformers (which offer a super light change in frequency response and only a hint of breakup towards the
very top end of the gain range) to more pronounced and downright “nasty” distortion as in tube or solid states
circuit designs.

Thanx to this versatile structure it can be freely used as a creative tool to add dimension, depth and a “warm
and fuzzy” analog feel to either individual tracks or on the master fader of the DAW. On single tracks, distortion
amounts can eventually be pushed to extreme limits for artistic purposes, whereas on a full mix gain should be
applied more judiciously, in order to impart the character of an analog stage, be it tube, solid state or a tape
deck, unless the distortion effect is desired on the master 2 bus too.

Controls:

• GAIN: this control sets the level of the signal going into the module. The higher the level, the more
pronounced the saturation effect will be as the electronics of the engaged mode will reach their limit and
start to distort the signal. The available gain range is from 0 to 40dBs.
• OUTPUT: this control determines the output level of the module. It works in the opposite direction as the

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Gain control so that it can be used to compensate the output volume increase as more level (and thus
saturation) is generated. The available range is from 0dB (no attenuation) to -40 dB.
• LINK: this control links the Gain and Output controls in an inverse way: when active as Gain is raised the
Output is lowered by the same amount in order to compensate the volume increase and keep the output
level consistent. Knobs can be set independently even when linked: they will move together maintaining the
relative difference.
• MODE: this control lets you select the electronics type whose saturation behavior is modeled.
Available modes are:
• TAPE 1: this mode reproduces the classic character and high frequency compression of an analog
tape machine reaching saturation when pushed above normal recording levels. It can also an provide a
good amount of added RMS and loudness level to finished mixes to make them sound fuller and “more
complete” if slightly overdriven. More pronounced gain can provide a beautiful overload effect on many
kinds of individual tracks including drums and vocals, and it can also keep the high frequency content
under control.
• TAPE 2: similar to Tape 1 in terms of typical uses, this mode reproduces a more “Lo-Fi” analog tape
response with more pronounced distortion and high frequency roll-off. It can be very effective in creating
bold sonic enhancements to individual tracks or stems.
• MASTER +6dB: this is a digitally-recreated soft saturation curve that can be very useful when mastering
finished tracks to add RMS level and “body” to a somewhat weak-sounding mix. If used gently, it adds
a nice shine to every mix. If pushed harder, it becomes more suitable for individual tracks to make them
sound bigger.
• MASTER + 12dB: this is similar to Master +6dB but with a more pronounced soft-saturation curve to
make the effect even more prominent.
• TUBE - PUSH PULL: this mode produces classic tube saturation with odd harmonics added by the
symmetrical clipping of the Push/Pull electronics. This creates a punchy distortion that can make
everything sound warmer if used lightly or with an aggressive edge when pushed.
• TUBE - CLASS A: this mode produces tube saturation with more even harmonics added by the
asymmetry of the Class A design. When used gently on stems or a finished mix, it adds a beautiful
“halo” to the audio that is processed, definitely making it sound more analog. When pushed, the
asymmetrical characteristic of Class A produce an uncontrolled distortion that sometimes sounds
more interesting than Push/Pull.
• SOLID STATE - PUSH PULL: same as the Tube mode, but with the more aggressive character of the
solid state circuitry. This mode is best used when transients need to be preserved.
• SOLID STATE - CLASS A: same as the Tube mode but with the more aggressive character of the solid
state circuitry. This mode adds the same kind of magic “halo” the Tube Class A provides but with a more
defined transient response.
• TRANSFORMER - IRON: this mode reproduces the very subtle color of iron core transformer harmonic
enhancement which features less even harmonics. This mode adds a very gentle and subtle saturation
adding an slight amount of distortion. For this reason. this mode is not useful for creative distortion
effects and is best used for adding just a touch of color to tracks, stems or finished mixes. Since this
mode is not clipping the audio, it can also be used to get a very nice pumping effect from the brickwall
limiter when pushed really hard.
• TRANSFORMER - STEEL: this mode reproduces steel core transformer behavior. The difference with
the iron is that it provides almost equal amount of odd and even harmonic enhancement. This mode
adds a very gentle saturation introducing only a slight amount of distortion. For this reason, this mode

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is not useful for creative distortion effects and is best used for adding just a touch of color to tracks,
stems or finished mixes. Since this mode is not clipping the audio, it can also be used to get a very nice
pumping effect from the brick wall limiter when pushed really hard.

• IN: engages or bypasses the processing.

• BRICKWALL: this control engages a brickwall limiter post output control. It can be enabled to prevent
overshoots of the module’s output when the input gain is turned higher to achieve distortion effects. Limiter’s
ceiling is set to -0.2 dBs. The brick wall limiter included with Saturator-X is used to keep the output level
under control, but it can also be used to create a specific sonic character. In fact, this limiter can have a nice,
intentional “pumping” effect that is very desirable on rhythmic material. To drive the Brickwall limiter hard,
always use the Transformer modes since they don’t clip the audio, leaving all the compression duties to the
limiter.
• OVERSAMPLING: this control sets the oversampling mode of the module. The values are 1X (no
oversampling), 2X and 4X. Higher values of oversampling will allow higher fidelity of the harmonic pattern
reproduction and an overall higher quality sound with less (if any) aliasing artifacts. The tradeoff is that
here the module is run at double or four times the session’s sample rate, so its CPU consumption will
be higher. Since it can be set independently for each module in a session, you can decide which tracks
need less or no oversampling versus which ones need to be run higher (like the master fader) according
to the available power of your CPU.
• RESET: when clicked, this control brings the settings of all of the parameters back to their default.
• L=R: links the two channels for stereo operation: all parameters affect the two channels in the same way to
maintain a consistent stereo image.
• L/R: click on the L or R icons if you want to dial in specific settings for L and R channels independently.
• M/S: this option uses a Mid-Side matrix to insert the stereo saturator in your processing chain. This means
that you can independently process the Center or the Sides of your stereo program. The L and R icons are
replaced by M and S in this mode. Click M to adjust settings for the Center (Mid) of your stereo image or S
for the sides.

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T-RackS 6 Module Processors

34 – ONE

The new ONE mastering processor was designed as an “all-in-one” solution for powerful and easy to use
adjustments in creating final masters. It includes an EQ, compressor, harmonic exciter, low-end enhancer and
limiter that will let anyone achieve professional quality results quickly and easily.

The module has 9 controls:

• AIR: gently controls the boost or cut of the High Frequencies.

• FOCUS: let you adjust the intelligibility of the mids.
• BODY: lets you fine tune the amount of low frequencies without upsetting the overall frequency balance.
• BASS PUNCH: lets you control the low-end boost and tightness of the sound at the same time.
• ANALOG: adds in a touch of harmonic enhancement to help give the track dimension and color.
• TRANSIENTS: can enhance or smooth out the attacks, to achieve the ideal balance of impact for your mix.
• WIDTH: lets you fine tune the stereo image of your track.
• PUSH: is a macro that controls all the settings of the compressor from gentle to hard.
• VOLUME: the one knob that makes your music sound louder without worrying about artifacts.

The ONE module will let anyone achieve professional quality results in the blink of an eye. Just turn its knobs,
experiment and hear the mojo.

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T-RackS 6 Module Processors

35 – Dyna-Mu

The DYNA-MU is an astonishing emulation of a well-known tube compressor celebrated for its warm, creamy
sound and vivid colored tone, which thanks to the highest level of expertise from IK Multimedia, the pioneer in
virtual circuit modeling, is perfectly recreated and indistinguishable from the hardware.
This modern-classic work of art has the characteristic of gluing together your mix with incredible quality, while it
shines on individual tracks and is perfect as part of a professional mastering chain. Additionally, the saturation it
produces when pushed to its limits can be used in many creative ways.

The DYNA MU, has 2 rows of controls that work for both L/R channels or Mid-Side, which are:

• Input: controls the amount of signal entering the processor.

• Threshold: sets the audio level for when the processor is engaged.
• Attack: the time in ms for the start processing after the threshold is exceeded.
• Release: how fast the processor reduces the compression ratio after the level falls below the threshold.
• Output: controls the level of signal coming out of the processor.

As you might notice, a common control not shown is the Ratio. The Ratio in this processor depends the amount
of gain reduction required and on the mode you select. Normally the Dyna-Mu will have a somewhat softer
ratio of around 1.5:1 but when the HARD switch is engaged a medium ratio of 4:1 is used. The HARD setting is
perfect for single tracks while the normal setting is best suited for mixes.
The other 3 controls are GANG, LINK and Side Chain:
The GANG switch lets you control both rows at the same time while the LINK switch controls the sidechain
detection mode for the two channels which can be either linked or independent.
The Side Chain switch inserts a gentle high pass filter so the processor is less sensitive to frequencies below
100Hz, ideal for heavy kick drums or deep basses in the mix.
The sound palette of the DYNA-MU is perfect to give cohesiveness and warmth to your mixes or tracks.

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T-RackS 6 Module Processors

36 – EQual

The T-RackS EQual digital equalizer is a 10-band, ultra-clear, high-end parametric equalizer with an extremely
transparent sound, ultra-precise editing and a vast array of filter shapes that are based on the typical curves of
classic analog EQ’s like Neve, SSL, API and other sought-after hardware.
This incredibly flexible “hybrid monster” gives you the best of both worlds - digital precision with transparent
clarity and on-demand analog character that works perfectly for high end mastering or individual track work.
EQual opens with a flat white EQ curve and displays the spectrum of your audio source in the background and
by clicking anywhere on the curve, the first band is created. From there you can select the shape and types of
equalization based on your needs with up to 10 total bands.
The result is an ultra-high resolution processor that is the most transparent equalization tool digital audio
technology allows today, offering the most wanted EQ curves and crystal clear processing for the most musical
adjustment to any tracks or an entire mix.
For each band you will have different parameters, depending on the shape and type of filter applied, which are:

BELL:
• Standard
• UK Modern 1 (Neve 1073/81)
• UK Modern 2
• UK Classic (SSL 4000 modeling)
• USA Classic 1 (API 550A/B)
• USA Classic 2 (API 560)
• T-RackS Classic
• Mastering SS
• Mastering Tube
• Classic Passive

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T-RackS 6 Module Processors

LOW SHELF:
• Standard
• UK Modern 1 (Neve 1073/81)
• UK Modern 2
• UK Classic (SSL 4000 modeling)
• USA Classic 1 (API 550A/B)
• T-RackS Classic
• Mastering SS
• Mastering Tube
• Classic Passive

HI SHELF:
• Standard
• UK Modern 1 (Neve 1073/81)
• UK Modern 2
• UK Classic (SSL 4000 modeling)
• USA Classic 1 (API 550A/B)
• T-RackS Classic
• Mastering SS
• Mastering Tube
• Classic Passive

HI PASS:
• Standard
• UK Modern 1 (Neve 1073/81)
• UK Modern 2
• UK Classic (SSL 4000 modeling)
• T-RackS Classic

LOW PASS:
• Standard
• UK Modern 1 (Neve 1073/81)
• UK Modern 2
• UK Classic (SSL 4000 modeling)
• T-RackS Classic

­­­­­­­­­95 EQual
T-RackS 6 Module Processors

BAND PASS:
• Standard

BAND STOP:
• Standard

TILT SHELF:
• Standard

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T-RackS 6 Module Processors

37 – Master Match X

The Master Match is a built in module of the new T-RackS 5, which, differently from the other modules, it has a
fixed position in the chain: the last module.
This innovative module has the aim of easily give to your mix, the round and sonic characteristics of the
references you had in mind already while recording and mixing, or at least the closest possible.
Also it gives you more sonic ideas, that you might never thought of.
The main point is find the perfect EQ that will give your track the sound of your reference, without interfere with
your musical expression.

The technic is to analyze up to 3 references tracks equalization and RMS level, analyze your track and apply an
EQ curve based on the results.

The Module is dividend in 4:

• Reference Upper Screen
• Curve Graphic Screen
• Matching Edit
• Matching Steps

To match your track with your ideas there are 3 simple steps to do:
Learn References, Learn Source and Match.

LEARN REFERENCES
To add a reference, drag and drop the track you choose in the upper screen or press the “PLUS” icon.
After insert at least 1 reference track, use the “PLAY” button to preview or the X to delete.

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T-RackS 6 Module Processors

It is possible to analyze only a piece of the reference by selecting a range with the mouse; the selection turn
orange, and 2 handlers appears at the sides of the selection.
Press LEARN REFERENCE to analyze the track; the curve will appear on the Graphic and the LEARN SOURCE
will blink to indicate the suggested next step.

LEARN SOURCE
The Learn Source start blinking suggesting to analyze your track.
Press LEARN SOURCE and the button will lights up, waiting you to play your track in your DAW or in the
standalone.
Re-press the button to stop the analysis and your track curve will appear in the screen.

MATCHING
It is now time to match your track with your reference.
To do so, simply press MATCHING and the resulting curve applied will appear on the screen and the EQ
MATCHING fader will automatically move to the suggested position for the best result.

The best musical result is an EQ that gives you the best sounding song with the character of your references.

It is possible to create up to 8 Filters on the resulting curves now, simply clicking on the screen, each of them
with individual FREQ, GAIN and Q, editable with the mouse or the values on the screen.

Also you can control other parameters to make your master sound perfect:

• EQ MATCHING: is the percentage of matching equalization resulted from the analisis. It goes from no
matching at 0% to exagerate matching at 100%.
• LEVEL MATCHING: is the percentage go matching RMS level resulted from the analisis. It goes from no
matching at 0% to full match at 100%.
• LF LIMIT: set the limit until the matching is applied in the low frequency.
• HF LIMIT: set the limit until the matching is applied in the high frequency.
• MODE A/B: this mode gives you the possibility to choose between a more sharp or a more smoothed
matching.

­­­­­­­­­98 Master Match X

T-RackS 6 Module Processors

38 – Leslie®

IK Multimedia and Hammond USA and Suzuki Music Corp. of Japan, teamed up to bring you the best sounding
Leslie in the digital world, giving you 5 legendary Leslie® amps and 6 cabinets in one powerful module to use
inside the T-RackS 5 mixing and mastering workstation to further shape the sound of your tracks.

The amps were recreated using IK’s innovative Dynamic Interaction Modeling Technology which can analyze
and model the behaviour of every electronic component in the circuit as well as their mutual interactions to
reproduce the exact behavior of the entire system in a non-linear and dynamic way.

Officially Certified Leslie® Models

Leslie 122 Leslie 122A Leslie 147

­­­­­­­­­99 Leslie®
T-RackS 6 Module Processors

Leslie 3300w Leslie G37 Leslie Studio 12

The interface
The module is divided into 3 parts, one of which has 3 different “pages” to it.
The first part of the module is dedicated to the controls. This is where the majority of adjustments to amps and
cabinets happens. It is divided into 3 pages: Setup, Amp and EQ. The Slow/Fast control is always visible at the
bottom of the controls for easy access with the Brake as an external control.

Setup page

In the setup page, you have control over these essential parameters:
• SPRING: adds spring reverb to the signal before going in the amplifier. This simulates the on-board spring

­­­­­­­­­100 Leslie®
T-RackS 6 Module Processors

reverb tank found on many vintage Hammond® organs.

• MIC DISTANCE: you can decide how far from the cabinet you want the mic to be placed.
• SLOW/FAST SPEED: This sets the rotation speed of the horn and the drum.
• ACCELERATION/DECELERATION: sets the amount of time in the transition from fast rotation to slow and
vice-versa.

Amp page

Leslie 122 Leslie 3300w Leslie G37 / Studio 12

The “Amp” button will bring up the specific amplifier controls associated to each included model:
• GAIN: controls the amount of signal going to the amp and the level of saturation.
• VOLUME: controls the amount of signal going out of the amp into the cabinet.
• INPUT HPF: controls the frequency of the high pass filtering of the input signal from OFF to 200Hz.
• TUBE DRIVE (3300W amp only): Controls the amount of signal being “warmed up” by the tube section.
When at minimum, the amp is in the clean channel. As soon as the knob is turned up, it automatically
switches to the drive channel.
• TUBE MODE (3300W amp only): Controls the amount of tube saturation from Soft to Hard.
• CONTOUR (G37/Studio12 amp only): controls the amount of “warmth” created by the tube of the preamp.
• EQ: A guitar amp-style equalizer with controls for Low, Mid and High frequencies.

­­­­­­­­­101 Leslie®
T-RackS 6 Module Processors

EQ page

The EQ button shows a full parametric EQ to let you further shape the sound with adjustable frequency and +/-
15dB of Gain control.

The center part of the module lets you select the amp and cabinet models and vary the mic positions which
affects the perception of the rotation, or “doppler effect.” You can switch between a 90° position, which is the
most common position that gives a non-symmetric perception of the rotation, to 180°, which is less common
but still heard on a lot of famous recordings.

­­­­­­­­­102 Leslie®
T-RackS 6 Module Processors

The third section of the module is for the mixer where you have full mix control of the four microphone channels
used when capturing the Leslie speakers. Each channel has its own level and pan control so that you can set
the Leslie mix as you prefer. You can, for example, set the drum to use just one channel panned center to have
solid monaural low frequencies, and then use the horn in stereo. Or you can get a 70s sonic flavour by using just
one mic on the top and one on the bottom panned just a bit apart, as another example.

­­­­­­­­­103 Leslie®
T-RackS 6 Module Processors

39 – Tape Machine 24

A model of the MCI JH24. First produced in 1980, the JH24 was a staple in US studios during the 1980s. This
completely transformerless, high-performance op-amp-based design delivers a pristine, phase-coherent and
remarkably true-to-the-source audio performance. This results in a moderate and elegant polish to your music.

The interface controls are:

• INPUT LEVEL: this control sets the level of the signal going into the module.
• INPUT/REPRO: with INPUT the signal passes only through the input and output electronic stages of the
recorders, bypassing the tape. Depending on how the recorder is designed the “Input” path can sound
very very similar to the plug-in input, or slightly more colored. REPRO passes your signal through the
entire analog recording system: input-> recording amplifier -> recording head -> tape -> playback head ->
playback preamplifier -> output stage.
• TRUE STEREO: even a perfectly aligned tape machine will have a slight level, EQ and distortion differences
between the left and right channels. This is good and part of the analog recording mojo! If a perfectly
identical L and R processing is a must, simply disable this feature.
• TAPE FORMULA: selects between the 4 available tape formulas. Each tape formula has its own harmonic
characteristic and the way it characterizes the sound will changes. This dynamic modeling is important in
order to have an even broader palette of sounds:
• 250: modelled after the 3M/Scotch “250 Audio Recording Tape”. it delivers a warm tone with slightly
more distortion and saturation than the other formulas included in the package.
• 456: modelled after the Ampex “456 High-Output Mastering Tape”. It offers a warm, round tone with a
slight touch of saturation that greatly responds to the recording level.
• GP9: modelled after the Quantegy “Quantegy GP9 Grand Master Platinum”. It’s very punchy and it’s
perfect for high-quality analog recordings of nowadays.
• 499: modelled after the Ampex “499 Grand Master Gold Studio Mastering Audio Tape”. This tape is

­­­­­­­­­104 Tape Machine 24

T-RackS 6 Module Processors

designed to handle a great amount of level with minimal distortion and compression. It also exhibits
a high frequency added definition that makes it perfect for printing “digital-like” recordings, while still
maintaining an analog sound.
• TAPE SPEED: selects the speed in inches per second (IPS) of the tape transport between 15 ips and 30 ips.
The higher speed gives you more fidelity, and the lower speed delivers a warmer rounder tone.
• TRANSPORT MODELING: precisely models the behavior of the mechanical transport. Small irregularities
in the movement of the tape create various degrees of sonic alteration to the audio program, especially
between the two channels. Keep it on for the magic to happen! In case a perfectly steady performance is
needed, just disable the control.
• AUTO CALIBRATION: the Tape Machines are calibrated by default at their factory calibration and it possible
edit the calibration parameters at your will. With the auto-calibration, it is possible to reset the edits and go
back to the factory calibration.
• RECORD BIAS: this control adjusts the amount of bias in the recorded signal. Ideal bias voltage settings
provide maximum sensitivity and low distortion. Intentionally overbiasing is a common technique to produce
a warmer, gently saturated sound. Underbiasing can be used to nicely boost the high frequencies in a truly
unique way and add distortion and other nonlinear effects.
• RECORD LEVEL: when in REPRO it adjusts the level of the recorded signal after the Input.
• RECORD HF EQ: adjusts the high-frequency shelving added to the recorded signal on the tape.
• PLAY LEVEL: when in REPRO it adjusts the level of the recorded signal before the output after the entire
tape. It is used to calibrate the machine.
• PLAY HF EQ: adjusts the high-frequency shelving added to the playback signal from the tape.
• PLAY LF EQ: adjusts the low-frequency shelving added to the playback signal from the tape.
• OUTPUT LEVEL: this control determines the output level of the module.
• ON: activates or bypasses the processing.
• RESET: when clicked, this control brings the settings of all of the parameters back to their default.

­­­­­­­­­105 Tape Machine 24

T-RackS 6 Module Processors

40 – Tape Machine 80

The legendary Studer A80 Mk II, in its various revisions, is an essential part of countless influential records.
Engineered and manufactured in Switzerland from 1970 to 1988, it became the de facto standard for
professional high-end multi-tracking. The A80’s sonic signature is a perfect blend between transparency and
subtle harmonic enhancement.

The interface controls are:

• INPUT LEVEL: this control sets the level of the signal going into the module.
• INPUT/REPRO: with INPUT the signal passes only through the input and output electronic stages of the
recorders, bypassing the tape. Depending on how the recorder is designed the “Input” path can sound
very very similar to the plug-in input, or slightly more colored. REPRO passes your signal through the
entire analog recording system: input-> recording amplifier -> recording head -> tape -> playback head ->
playback preamplifier -> output stage.
• TRUE STEREO: even a perfectly aligned tape machine will have a slight level, EQ and distortion differences
between the left and right channels. This is good and part of the analog recording mojo! If a perfectly
identical L and R processing is a must, simply disable this feature.
• TAPE FORMULA: selects between the 4 available tape formulas. Each tape formula has its own harmonic
characteristic and the way it characterizes the sound will changes. This dynamic modeling is important in
order to have an even broader palette of sounds:
• 250: modelled after the 3M/Scotch “250 Audio Recording Tape”. it delivers a warm tone with slightly
more distortion and saturation than the other formulas included in the package.
• 456: modelled after the Ampex “456 High-Output Mastering Tape”. It offers a warm, round tone with a
slight touch of saturation that greatly responds to the recording level.
• GP9: modelled after the Quantegy “Quantegy GP9 Grand Master Platinum”. It’s very punchy and it’s
perfect for high-quality analog recordings of nowadays.

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T-RackS 6 Module Processors

• 499: modelled after the Ampex “499 Grand Master Gold Studio Mastering Audio Tape”. This tape is
designed to handle a great amount of level with minimal distortion and compression. It also exhibits
a high frequency added definition that makes it perfect for printing “digital-like” recordings, while still
maintaining an analog sound.
• TAPE SPEED: selects the speed in inches per second (IPS) of the tape transport between 15 ips and 30 ips.
The higher speed gives you more fidelity, and the lower speed delivers a warmer rounder tone.
• TRANSPORT MODELING: precisely models the behavior of the mechanical transport. Small irregularities
in the movement of the tape create various degrees of sonic alteration to the audio program, especially
between the two channels. Keep it on for the magic to happen! In case a perfectly steady performance is
needed, just disable the control.
• AUTO CALIBRATION: the Tape Machines are calibrated by default at their factory calibration and it possible
edit the calibration parameters at your will. With the auto-calibration, it is possible to reset the edits and go
back to the factory calibration.
• RECORD BIAS: this control adjusts the amount of bias in the recorded signal. Ideal bias voltage settings
provide maximum sensitivity and low distortion. Intentionally overbiasing is a common technique to produce
a warmer, gently saturated sound. Underbiasing can be used to nicely boost the high frequencies in a truly
unique way and add distortion and other nonlinear effects.
• RECORD LEVEL: when in REPRO it adjusts the level of the recorded signal after the Input.
• RECORD HF EQ: adjusts the high-frequency shelving added to the recorded signal on the tape.
• PLAY LEVEL: when in REPRO it adjusts the level of the recorded signal before the output after the entire
tape. It is used to calibrate the machine.
• PLAY HF EQ: adjusts the high-frequency shelving added to the playback signal from the tape.
• PLAY LF EQ: adjusts the low-frequency shelving added to the playback signal from the tape.
• OUTPUT LEVEL: this control determines the output level of the module.
• ON: activates or bypasses the processing.
• RESET: when clicked, this control brings the settings of all of the parameters back to their default.

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T-RackS 6 Module Processors

41 – Tape Machine 440

The Ampex 440B series dates from the late 1960s and delivers a typical, unmistakably soulful color that made
this professional and historic mixing and mastering machine a legend. This superb blend of art and technology
has the power to turn your mixes into finished tracks.

The interface controls are:

• INPUT LEVEL: this control sets the level of the signal going into the module.
• INPUT/REPRO: with INPUT the signal passes only through the input and output electronic stages of the
recorders, bypassing the tape. Depending on how the recorder is designed the “Input” path can sound
very very similar to the plug-in input, or slightly more colored. REPRO passes your signal through the
entire analog recording system: input-> recording amplifier -> recording head -> tape -> playback head ->
playback preamplifier -> output stage.
• TRUE STEREO: even a perfectly aligned tape machine will have a slight level, EQ and distortion differences
between the left and right channels. This is good and part of the analog recording mojo! If a perfectly
identical L and R processing is a must, simply disable this feature.
• TAPE FORMULA: selects between the 4 available tape formulas. Each tape formula has its own harmonic
characteristic and the way it characterizes the sound will changes. This dynamic modeling is important in
order to have an even broader palette of sounds:
• 250: modelled after the 3M/Scotch “250 Audio Recording Tape”. it delivers a warm tone with slightly
more distortion and saturation than the other formulas included in the package.
• 456: modelled after the Ampex “456 High-Output Mastering Tape”. It offers a warm, round tone with a
slight touch of saturation that greatly responds to the recording level.
• GP9: modelled after the Quantegy “Quantegy GP9 Grand Master Platinum”. It’s very punchy and it’s
perfect for high-quality analog recordings of nowadays.
• 499: modelled after the Ampex “499 Grand Master Gold Studio Mastering Audio Tape”. This tape is

­­­­­­­­­108 Tape Machine 440

T-RackS 6 Module Processors

designed to handle a great amount of level with minimal distortion and compression. It also exhibits
a high frequency added definition that makes it perfect for printing “digital-like” recordings, while still
maintaining an analog sound.
• TAPE SPEED: selects the speed in inches per second (IPS) of the tape transport between 7.5 ips and 15
ips. The higher speed gives you more fidelity, and the lower speed delivers a warmer rounder tone.
• TRANSPORT MODELING: precisely models the behavior of the mechanical transport. Small irregularities
in the movement of the tape create various degrees of sonic alteration to the audio program, especially
between the two channels. Keep it on for the magic to happen! In case a perfectly steady performance is
needed, just disable the control.
• AUTO CALIBRATION: the Tape Machines are calibrated by default at their factory calibration and it possible
edit the calibration parameters at your will. With the auto-calibration, it is possible to reset the edits and go
back to the factory calibration.
• RECORD BIAS: this control adjusts the amount of bias in the recorded signal. Ideal bias voltage settings
provide maximum sensitivity and low distortion. Intentionally overbiasing is a common technique to produce
a warmer, gently saturated sound. Underbiasing can be used to nicely boost the high frequencies in a truly
unique way and add distortion and other nonlinear effects.
• RECORD LEVEL: when in REPRO it adjusts the level of the recorded signal after the Input.
• RECORD HF EQ: adjusts the high-frequency shelving added to the recorded signal on the tape.
• PLAY LEVEL: when in REPRO it adjusts the level of the recorded signal before the output after the entire
tape. It is used to calibrate the machine.
• PLAY HF EQ: adjusts the high-frequency shelving added to the playback signal from the tape.
• PLAY LF EQ: adjusts the low-frequency shelving added to the playback signal from the tape.
• OUTPUT LEVEL: this control determines the output level of the module.
• ON: activates or bypasses the processing.
• RESET: when clicked, this control brings the settings of all of the parameters back to their default.

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T-RackS 6 Module Processors

42 – Tape Machine 99

The Revox PR99 Mk II is a stereo professional recorder produced by Studer in the 1980s. Engineered around
a hybrid design that combines the best of discrete and op-amp topologies, this model offers a very smooth
frequency response, especially on bass, with almost no trace of the typical “head bump” phenomenon. A
standard for broadcast and classical music, this machine was appreciated by audiophiles for its exceptional
sonic quality.

The interface controls are:

• INPUT LEVEL: this control sets the level of the signal going into the module.
• INPUT/REPRO: with INPUT the signal passes only through the input and output electronic stages of the
recorders, bypassing the tape. Depending on how the recorder is designed the “Input” path can sound
very very similar to the plug-in input, or slightly more colored. REPRO passes your signal through the
entire analog recording system: input-> recording amplifier -> recording head -> tape -> playback head ->
playback preamplifier -> output stage.
• TRUE STEREO: even a perfectly aligned tape machine will have a slight level, EQ and distortion differences
between the left and right channels. This is good and part of the analog recording mojo! If a perfectly
identical L and R processing is a must, simply disable this feature.
• TAPE FORMULA: selects between the 4 available tape formulas. Each tape formula has its own harmonic
characteristic and the way it characterizes the sound will changes. This dynamic modeling is important in
order to have an even broader palette of sounds:
• 250: modelled after the 3M/Scotch “250 Audio Recording Tape”. it delivers a warm tone with slightly
more distortion and saturation than the other formulas included in the package.
• 456: modelled after the Ampex “456 High-Output Mastering Tape”. It offers a warm, round tone with a
slight touch of saturation that greatly responds to the recording level.
• GP9: modelled after the Quantegy “Quantegy GP9 Grand Master Platinum”. It’s very punchy and it’s
perfect for high-quality analog recordings of nowadays.

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T-RackS 6 Module Processors

• 499: modelled after the Ampex “499 Grand Master Gold Studio Mastering Audio Tape”. This tape is
designed to handle a great amount of level with minimal distortion and compression. It also exhibits
a high frequency added definition that makes it perfect for printing “digital-like” recordings, while still
maintaining an analog sound.
• TAPE SPEED: selects the speed in inches per second (IPS) of the tape transport between 7.5 ips and 15 ips.
The higher speed gives you more fidelity, and the lower speed delivers a warmer rounder tone.
• TRANSPORT MODELING: precisely models the behavior of the mechanical transport. Small irregularities
in the movement of the tape create various degrees of sonic alteration to the audio program, especially
between the two channels. Keep it on for the magic to happen! In case a perfectly steady performance is
needed, just disable the control.
• AUTO CALIBRATION: the Tape Machines are calibrated by default at their factory calibration and it possible
edit the calibration parameters at your will. With the auto-calibration, it is possible to reset the edits and go
back to the factory calibration.
• RECORD BIAS: this control adjusts the amount of bias in the recorded signal. Ideal bias voltage settings
provide maximum sensitivity and low distortion. Intentionally overbiasing is a common technique to produce
a warmer, gently saturated sound. Underbiasing can be used to nicely boost the high frequencies in a truly
unique way and add distortion and other nonlinear effects.
• RECORD LEVEL: when in REPRO it adjusts the level of the recorded signal after the Input.
• RECORD HF EQ: adjusts the high-frequency shelving added to the recorded signal on the tape.
• PLAY LEVEL: when in REPRO it adjusts the level of the recorded signal before the output after the entire
tape. It is used to calibrate the machine.
• PLAY HF EQ: adjusts the high-frequency shelving added to the playback signal from the tape.
• OUTPUT LEVEL: this control determines the output level of the module.
• ON: activates or bypasses the processing.
• RESET: when clicked, this control brings the settings of all of the parameters back to their default.

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T-RackS 6 Module Processors

43 – Sunset Sound Studio Reverb

Sunset Sound Studio Reverb is an all-new hybrid convolution/modelling engine made to deliver the highest
quality reverbs. With the right amount of control, we captured each space in the studio using Sunset’s own
microphone collection and each studio’s own console.
IK’s award-winning physical modeling technology recreates each studio’s preamps and console signal path to
achieve a perfectly accurate amount of warmth and tone
With Sunset Sound Studio Reverb it is possible to use all the rooms, chambers, and mechanical reverbs of the
physical studio in one single plug-in, giving to your track that magical vibe and ambiance it deserves.

The interface controls are:

• POWER: This control enables or disables the processor.

• RESET: This control resets the processor to its default state.
• INPUT: This control adjusts the amount of signal going into the processor. The signal can be lowered until
reaching complete silence ( -∞ ) or boosted up to 6dB from its nominal level. The input state is saved in each
preset.
• MONO: Since Sunset Sound Studio Reverb is a true stereo plugin, this controls sums the L and R channels
of the input to create a MONO input for the wet signal.
• PRE-DELAY: This control determines the amount of time between the initial input signal and the
reverberation. Values from the pre-delay range from 0ms to 1sec. All values are shown as ms (milliseconds).
• DECAY: This control adjusts the length of the reverb, to shorten or elongate the time that the reverb tail lasts.
Because of the dynamic nature of the control, the displayed value is from 0% to 100%. To maintain the
integrity of each space, the length of 100% will depend on the selected room, while the 0% will be a decay
of 1 ms.
• HPF (Hi pass filter): This control sets the frequency of the high pass filter before the reverberation path. It is
a 6dB/octave filter with a range from 10Hz to 20 kHz, with an OFF option set under 10Hz.

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The default setting is OFF. All values are shown in Hz and kHz.
• LPF (Low pass filter): This control set the frequency of the low pass filter before the reverberation path. It is
a 6dB/octave filter with a range from 40Hz to 20 kHz, with an OFF option set above 20kHz.
The default setting is OFF. All values are shown in Hz and kHz.
• LOW (Low Shelf): This control changes the gain of the low shelf filter after the reverberation path. The
frequency is set at 200Hz with a Q of 0.4 and its range is from -12dB to 12dB.
The default setting is 0dB. All values are showed in dB.
This control can be used by clicking & dragging on the vertical axis or by typing in a numerical value in the
space where the value is displayed.
• HIGH (High shelf): This control changes the gain of the high shelf filter after the reverberation path. The
frequency is set at 5kHz with a Q of 0.4 and its range is from -12dB to 12dB.
The default setting is 0dB.
• MIC: Selector control with 3 fixed positions. Set to “L” only the left microphone will be used, in “R” only the
Right microphone will be used, and in STEREO both mics will be used.
• WIDTH: When STEREO is selected, this control lets you adjust width of the stereo image of the reverb.
• DRY: Controls the amount of original sound.
• WET: Controls the amount of processed sound.
• SOLO: Solos just the WET or DRY signal to let you hear only the processed signal or the original signal
without needing to adjust the WET/DRY mix.
• REVERB TYPE: These controls select the reverb to be used.
The first row selects the room type:
• Chamber
• Live room
• ISO booth
• Plate/Spring
The second row selects which of the included reverbs to be used:
• Chamber: Studio 1, Studio 2 or Studio 3
• Live room: Studio 1, Studio 2 or Studio 3
• Iso Booth: Studio 1, Studio 2 or Studio 3
• PLATE: Plate 1, Plate 2 and Spring.

The screen default and main view are a 360-degree image which used to indicate the selected room. Each time
a reverb is selected, the matching image is loaded. To navigate the screen simply click and drag the image.
When a Plate or Spring is selected, the display will show a fixed photo of the Plate or Spring.

There are two controls that are placed at the bottom of the screen:

• INFO: Covers the image and displays the description and general info about the selected room or plate. If
you want to know more about the Sunset Sound Recorders, visit the website http://www.sunsetsound.com
• OPTION: Some secondary controls for each room are found in this menu, adding interesting possibilities to
shape the sound of the selected reverb.

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The available options are:

• Chamber of Studio 1, Studio 2 and Studio 3:
• Phase selection between 0º and 180º.
• Live Room Studio 1:
• Dampening: None, Half and Full. This control alters the response of the room.
• Phase selection between 0º and 180º.
• Live Room Studio 2:
• Position: Position 1, Position 2 and Position 3. This controls lets users select which of the 3 main
positions the room is recorded in.
• Phase selection between 0º and 180º.
• Live Room Studio 3:
• Dampening: None, Half and Full. This controls alters the response of the room.
• Phase selection between 0º and 180º.
• Iso Booth Studio 1, Studio 2 and Studio 3:
• Phase selection between 0º and 180º.
• Plate 1:
• Decay: Low, Mid, High. Select between the shortest, intermediate and longest decays available on
this vintage machine.
• Phase selection between 0º and 180º.
• Plate 2:
• Decay: select different length of decay between 300ms, 600ms, 1000ms, 1500ms, 1800ms, 2300ms,
4000ms, 5000ms and 6000ms.
• Phase selection between 0º and 180º.
• Spring:
• Decay: Low, Mid, High. Select between the shortest, intermediate and longest decays available on
this vintage machine.
• Phase selection between 0º and 180º.

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44 – Space Delay

Tape echo offers a unique sonic signature, and perhaps no unit offers a more recognizable sound than
the Roland® RE-201 Space Echo™. Now you can enjoy the endlessly flexible controls and the instantly
recognizable organic sound of this classic delay unit, captured with an unprecedented level of detail in the
T-RackS Space Delay.

Using the same mix of dynamic convolution and physical modeling as IK’s Tape Machine Collection, the
T-RackS Space Delay captures the sound of this iconic effect, down to the finest detail.

The interface controls are:

• POWER: enables or disables the processor.

• RESET: resets the processor to its default state.
• PEAK/LEVEL: the PEAK LED and the VU meters show the delay output level. When in use, the PEAK LED
should blink when at the correct playing level.
• INPUT: adjusts the signal level going into the processor. On the original hardware, overdriving the input
affects the tone of the processed signal and is often used as part of the effect. If a cleaner sound is desired,
reduce the input volume below unity and increase the output volume to compensate.
• DRY/WET: adjusts the mix between the DRY and WET signal, with the 12 o’clock position being 50%/50%.
• MODE: determines which effects are active, since the processor is combining tape echo and spring reverb.
The tape echo section has three tape playback heads which change position, and in combination with the
spring reverb create the 12 modes available, the same of the original hardware. The following table shows
the modes and combinations:

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• REVERB VOL: adjusts the level of the Spring reverb applied to the processed signal between -∞ and +6dB.
The control is available from mode 5 to 12 and inactive from 1 to 4.
• REVERB PAN: adjusts the placement of the reverb in the stereo panorama. Center placement is at the 12
o’clock position. The control is active only when used in mono-to-stereo or stereo.
• BASS: adjusts the level of the low frequency shelf EQ applied to the tape echo part of the processor. No
effect when at 12 o’clock position.
• TREBLE: adjusts the level of the high frequency shelf EQ applied to the tape echo part of the processor. No
effect when at 12 o’clock position.
• RATE: adjusts the lenth of the delay depending on the mode, the active heads and the relation between
them. Rate controls the speed of the tape: when at 12 o’clock position it is at nominal speed; rotating
clockwise speeds up the tape for shorter delays and rotating counter-clockwise slows down the tape for
longer delays.
The Delay times are usually:
• Head 1: between 69ms and 177ms
• Head 2: between 137ms and 337ms
• Head 3: between 189ms and 489ms
Rate also controls the general EQ of the processed signal: lowering the speed of the tape darkens the
signal while increasing the speed of the tape brightens the signal.
• FEEDBACK: adjusts the level of the delay repeats. The default is the 11 o’clock position. Rotating clockwise
increases the level of the feedback and starts auto-oscillating at around 2 o’clock. Decrease the level of the
feedback by rotating counter-clockwise.
• DELAY VOL: adjusts the level of the delay applied to the processed signal between -∞ and +6dB. The
control is available from mode 1 to 11 and inactive at 12 (reverb only).DELAY PAN: Active only when used in
mono-to-stereo or stereo. Adjusts the placement of the Delay in the stereo panorama. Center placement is
at the 12 o’clock position.
• FILTERS: the filters affect the input signal going into the wet channel of the processor:
• HPF (high-pass filter): This control changes the frequency of the low cut filter. It is a 24dB/octave filter
with a range from 10Hz to 20 kHz.
• LPF (Low pass filter): This control changes the frequency of the low cut filter before the reverberation
path. It is a 24dB/octave filter with a range from 40Hz to 20 kHz.

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• DUCKING: controls the amount of volume loss in the tape emulation in relation to the dry signal from 0% to
100%.
• STEREO REV: the original unit’s reverb is a mono 3-spring box. Switching the control to ON switches the
spring reverb to stereo operation. The control is only available when the plugin is used in mono-to-stereo or
stereo, and disabled when used in mono.
• FX FEED: in the original unit, it is possible to disable the signal sent to the effects and it is commonly called
“DUB-switch”. When the FX FEED is ON, the signal is sent to the effects; when OFF it is not.
• NOISE AMOUNT: adjusts the level of tape hiss noise between 0 and 10.
• TAPE AGE: this control emulates the effect of aging analog tape and: the resulting impact on sound.
0 is the newest, clearest sound, while 10 emulates a very used and dirty tape. The effects is based on
measurements of the impact on EQ, the wow/flutter amount and saturation of tapes of various ages and
condition.
• SYNC: switch control. When ON, the delay syncs with the host/DAW’s BPM.
• HEAD TIME: here it is possible to manually enter the heads’ playback delay in milliseconds, or in note
divisions when SYNC is ON.
• HEAD INDICATOR/PAN: this control indicates via the LEDs which heads are active depending on the
MODE, and let you adjust the stereo panorama of each head independently.

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45 – Comprexxor

T-RackS Comprexxor is based on one of the most flexible and creative analog compressors ever made. With
eight selectable curves, selectable 2nd or 3rd order harmonic saturation, high-pass/band-pass filters on the
sidechain path and much more, Comprexxor delivers endless sonic flexibility, whether that’s emulating classic
sounds of the past or inspiring new ones for tomorrow.

Main Controls

• INPUT: Adjusts the amount of signal going into the processor.

• ATTACK: Sets the time it takes the compressor to start its action once the signal has exceeded the
threshold.
• RELEASE: Sets the time it takes the compressor to release the signal back to its uncompressed state once
its level falls below the set threshold.
• OUTPUT: Adjusts the output volume and is used to compensate for the loss of level in the program that can
be produced by particularly heavy compression settings.
• DRY/WET: This control adjusts the mix between the processed and the unprocessed signal. The values go
from 0% (DRY) to 100% (WET).
• RATIO: Sets the behavior of the compressor related to the signal hitting the threshold. A setting of 4:1
means that any signal exceeding the set threshold will be reduced by four times (4 dB over the threshold will
be reduced to 1dB).
The ratio can be selected by clicking the button or clicking on the relative LED.
• OPTICAL: This control automatically switches the RATIO to 10:1, the ATTACK at 10 and the RELEASE
at 0. This control is meant as a macro to switch the machine to a starting point to achieve a particular
compression behavior, with its ranges highlighted by the yellow dashed line.
• GB MODE: This control enables the GB MODE, which acts as the ALL-IN mode of the 1176. The switch
also automatically switches the RATION to 1:1 and ATTACK to 0, which is the starting point for using this
mode. The GB MODE affects also the other ratio values.

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Sidechain Controls

Sidechain controls will affect the signal going into the processor.

• HP (Hi-Pass filter): Changes the frequency of the low-cut filter applied to the signal going to the detector of
the processor. It is a 6dB/octave filter with a range from 10Hz to 20 kHz, with an OFF setting at before 10Hz.
• BAND/FREQ (Peak filter): These two controls enable and adjust a peak filter to enhance a desired
frequency going to the detector of the processor. It is a 3dB peak with Q-1 and a range from 10Hz to 20
kHz.

Color Controls

• DIST2/3/OFF: This control enables the harmonic emphasis of the 2nd harmonic (DIST2) or the 3rd Harmonic
(DIST3) added at the signal going out of the processor. When no LED is lit, the emphasis is OFF
• HP: Enables or disables the Hi-Pass filter applied to the signal going out of the processor.
• TONE: This control adjusts the global distortion level on each distortion mode. The control is bipolar and
turning clockwise it will increase the level, while counterclockwise it will decrease it.
Values goes from -5 to 0 to 5, where 0 is the standard “color.”
• COLOR METER: This meter shows when the harmonic emphasis starts working and if 2nd, 3rd or both
harmonics are emphasized.

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46 – Teac A-6100 MKII

Main Controls

• INPUT LEVEL: this control sets the level of the signal going into the module.
• INPUT/REPRO: with INPUT the signal passes only through the input and output electronic stages of the
recorders, bypassing the tape. REPRO passes your signal through the entire analog recording system: input-
> recording amplifier -> recording head -> tape -> playback head -> playback preamplifier -> output stage.
• TRUE STEREO: even a perfectly aligned tape machine will have a slight level, EQ and distortion differences
between the left and right channels.
• TAPE FORMULA: selects between the 4 available tape formulas.
• SM911: modelled after the BASF SM 911
• 456: modelled after the Ampex “456 High-Output Mastering Tape”.
• GP9: modelled after the Quantegy “Quantegy GP9 Grand Master Platinum”.
• 499: modelled after the Ampex “499 Grand Master Gold Studio Mastering Audio Tape”
• TAPE SPEED: selects the speed in inches per second (IPS) of the tape transport between LOW (7,5 ips)
and HIGH (15 ips). The higher speed gives you more fidelity, and the lower speed delivers a warmer rounder
tone.
• TRANSPORT MODELING: precisely models the behavior of the mechanical transport. Small irregularities
in the movement of the tape create various degrees of sonic alteration to the audio program, especially
between the two channels. In case a perfectly steady performance is needed, just disable the control.
• AUTO CALIBRATION: the Tape Machines are calibrated by default at their factory calibration and it possible
edit the calibration parameters at your will. With the auto-calibration, it is possible to reset the edits and go
back to the factory calibration.
• RECORD BIAS: this control adjusts the amount of bias in the recorded signal. Ideal bias voltage settings
provide maximum sensitivity and low distortion. Intentionally overbiasing is a common technique to produce
a warmer, gently saturated sound. Underbiasing can be used to nicely boost the high frequencies in a truly

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unique way and add distortion and other nonlinear effects.

• RECORD LEVEL: when in REPRO it adjusts the level of the recorded signal after the Input. RECORD HF
EQ: adjusts the high-frequency shelving added to the recorded signal on the tape.
• PLAY LEVEL: when in REPRO it adjusts the level of the recorded signal before the output after the entire
tape. It is used to calibrate the machine.
• PLAY HF EQ: adjusts the high-frequency shelving added to the playback signal from the tape.
• PLAY LF EQ: adjusts the low-frequency shelving added to the playback signal from the tape.
• OUTPUT LEVEL: this control determines the output level of the module.
• POWER: activates or bypasses the processing.
• RESET: when clicked, this control brings the settings of all of the parameters back to their default.

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47 – Teac A-3340S

Main Controls

• INPUT LEVEL: this control sets the level of the signal going into the module.
• INPUT/REPRO: with INPUT the signal passes only through the input and output electronic stages of the
recorders, bypassing the tape. REPRO passes your signal through the entire analog recording system: input-
> recording amplifier -> recording head -> tape -> playback head -> playback preamplifier -> output stage.
• TRUE STEREO: even a perfectly aligned tape machine will have a slight level, EQ and distortion differences
between the left and right channels.
• TAPE FORMULA: selects between the 4 available tape formulas.
• SM911: modelled after the BASF SM 911
• 456: modelled after the Ampex “456 High-Output Mastering Tape”.
• GP9: modelled after the Quantegy “Quantegy GP9 Grand Master Platinum”.
• 499: modelled after the Ampex “499 Grand Master Gold Studio Mastering Audio Tape”
• TAPE SPEED: selects the speed in inches per second (IPS) of the tape transport between LOW (7,5 ips)
and HIGH (15 ips). The higher speed gives you more fidelity, and the lower speed delivers a warmer rounder
tone.
• TRANSPORT MODELING: precisely models the behavior of the mechanical transport. Small irregularities
in the movement of the tape create various degrees of sonic alteration to the audio program, especially
between the two channels. In case a perfectly steady performance is needed, just disable the control.
• AUTO CALIBRATION: the Tape Machines are calibrated by default at their factory calibration and it possible
edit the calibration parameters at your will. With the auto-calibration, it is possible to reset the edits and go
back to the factory calibration.
• RECORD BIAS: this control adjusts the amount of bias in the recorded signal. Ideal bias voltage settings
provide maximum sensitivity and low distortion. Intentionally overbiasing is a common technique to produce
a warmer, gently saturated sound. Underbiasing can be used to nicely boost the high frequencies in a truly

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unique way and add distortion and other nonlinear effects.

• RECORD LEVEL: when in REPRO it adjusts the level of the recorded signal after the Input. RECORD HF
EQ: adjusts the high-frequency shelving added to the recorded signal on the tape.
• PLAY LEVEL: when in REPRO it adjusts the level of the recorded signal before the output after the entire
tape. It is used to calibrate the machine.
• PLAY HF EQ: adjusts the high-frequency shelving added to the playback signal from the tape.
• PLAY LF EQ: adjusts the low-frequency shelving added to the playback signal from the tape.
• OUTPUT LEVEL: this control determines the output level of the module.
• POWER: activates or bypasses the processing.
• RESET: when clicked, this control brings the settings of all of the parameters back to their default.

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48 – TASCAM 388

Main Controls

• INPUT LEVEL: this control sets the level of the signal going into the module.
• INPUT/REPRO: with INPUT the signal passes only through the input and output electronic stages of the
recorders, bypassing the tape. REPRO passes your signal through the entire analog recording system: input-
> recording amplifier -> recording head -> tape -> playback head -> playback preamplifier -> output stage.
• TRUE STEREO: even a perfectly aligned tape machine will have a slight level, EQ and distortion differences
between the left and right channels.
• TAPE FORMULA: selects between the 4 available tape formulas.
• 35: Modelled after the BASF LPR35.
• 456: modelled after the Ampex “456 High-Output Mastering Tape”.
• GP9: modelled after the Quantegy “Quantegy GP9 Grand Master Platinum”.
• 499: modelled after the Ampex “499 Grand Master Gold Studio Mastering Audio Tape”
• TRANSPORT MODELING: precisely models the behavior of the mechanical transport. Small irregularities
in the movement of the tape create various degrees of sonic alteration to the audio program, especially
between the two channels. In case a perfectly steady performance is needed, just disable the control.
• AUTO CALIBRATION: the Tape Machines are calibrated by default at their factory calibration and it possible
edit the calibration parameters at your will. With the auto-calibration, it is possible to reset the edits and go
back to the factory calibration.
• RECORD BIAS: this control adjusts the amount of bias in the recorded signal. Ideal bias voltage settings
provide maximum sensitivity and low distortion. Intentionally overbiasing is a common technique to produce
a warmer, gently saturated sound. Underbiasing can be used to nicely boost the high frequencies in a truly
unique way and add distortion and other nonlinear effects.
• RECORD LEVEL: when in REPRO it adjusts the level of the recorded signal after the Input. RECORD HF
EQ: adjusts the high-frequency shelving added to the recorded signal on the tape.

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• EQ: 3 band Equalizer, positioned right before the output (hence after the tape). Those are 3 bells of +/-15dB.
• LOW: goes from 50Hz to 1kHz
• MID: goes from 500Hz to 5kHz
• HIGH: goes from 2.5kHz to 15kHz
• OUTPUT LEVEL: this control determines the output level of the module.
• POWER: activates or bypasses the processing.
• RESET: when clicked, this control brings the settings of all of the parameters back to their default.

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49 – TASCAM PORTA ONE

Main Controls

• INPUT LEVEL: this control sets the level of the signal going into the module.
• INPUT/REPRO: with INPUT the signal passes only through the input and output electronic stages of the
recorders, bypassing the tape. REPRO passes your signal through the entire analog recording system: input->
recording amplifier -> recording head -> tape -> playback head -> playback preamplifier -> output stage.
• TRUE STEREO: even a perfectly aligned tape machine will have a slight level, EQ and distortion differences
between the left and right channels.
• TAPE FORMULA: selects between the 2 available tape formulas: Type II and Type I.
• TRANSPORT MODELING: precisely models the behavior of the mechanical transport. Small irregularities
in the movement of the tape create various degrees of sonic alteration to the audio program, especially
between the two channels. In case a perfectly steady performance is needed, just disable the control.
• AUTO CALIBRATION: the Tape Machines are calibrated by default at their factory calibration and it possible
edit the calibration parameters at your will. With the auto-calibration, it is possible to reset the edits and go
back to the factory calibration.
• RECORD BIAS: this control adjusts the amount of bias in the recorded signal. Ideal bias voltage settings
provide maximum sensitivity and low distortion. Intentionally overbiasing is a common technique to produce
a warmer, gently saturated sound. Underbiasing can be used to nicely boost the high frequencies in a truly
unique way and add distortion and other nonlinear effects.
• RECORD LEVEL: when in REPRO it adjusts the level of the recorded signal after the Input. RECORD HF
EQ: adjusts the high-frequency shelving added to the recorded signal on the tape.
• EQ: it is a dual shelving eq positioned right before the output (hence after the tape), with LF +/- 10dB at
100Hz and HF +/- 10dB at 1000Hz
• OUTPUT LEVEL: this control determines the output level of the module.

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• POWER: activates or bypasses the processing.

• RESET: when clicked, this control brings the settings of all of the parameters back to their default.

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50 – FAME Studio Reverb

FAME Studio Reverb is the second product in the T-RackS Studio Reverb family featuring a hybrid convolution/
modelling engine made to deliver the highest quality reverbs possible.
We captured each space in the studio using FAME’s own microphone collection and each studio’s own console
for an umatched level of authenticity.
With FAME Studio Reverb it’s possible to use all the rooms, chambers, and mechanical reverbs of the physical
studio in one single plug-in, giving your track the magical vibe and ambiance it deserves.

• POWER: Enables or disables the processor.

• INPUT: Adjusts the amount of signal going into the processor. The signal can be lowered until reaching -15
dB or boosted up to 15 dB from its nominal level. The input state is saved in each preset.
• MONO/STEREO: Selects between a stereo input or a mono input. In Mono, stereo sources are summed to
mono before entering the processor.
• MODERN/VINTAGE: Selects between a clean modern sound or a more vintage one modelled after the
original Universal Audio 610 mixing console housed in Studio B.
• OUTPUT: Adjusts the amount of signal output by the processor. The signal can be lowered until reaching
-15 dB or boosted up to 15 dB from its nominal level. The output state is saved in each preset.
• RESET: Resets the processor to its default state.

The screen default and main view are a 360-degree image, which is used to indicate the selected room. Each
time a reverb is selected, the matching image is loaded. To navigate the screen, simply click and drag the
image. When Plate reverb is selected, the display will show a static photo of the Plate unit.
The TEXT BOX over the main view selects the reverb type:
• Plate
• Chamber

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• Studio A Booths
• Studio A Live
• Studio B Booths
• Studio B Live

The KNOB on the right of the text box selects the current reverb options:
• Plate: Adjusts the decay time (0.5s, 1.5s, 2.5s, 3.5s, 4.5s , 5s).
• Chamber: Selects which microphones are used (Dynamic, Condenser).
• Studio A Booths: Selects which booth will be used (Vocal, Amps, Leslie/Drums).
• Studio A Live: Selects the position in the room (Bass, Drums, Organ/El.Keys, Vocal/Guitars, Piano).
• Studio B Booths: Selects which booth will be used (Vocal, Amps).
• Studio B Live: Selects the position in the room (Piano/Keys, Bass/Guitars, Drums, Slapback).

On the top right of the main view you’ll find an info button:
• INFO: Covers the image and displays the description and general info about the selected room or plate. If
you want to know more about the FAME Studios, visit the website
https://famestudios.com

To modify your reverb:

• SPREAD: Choose which model to use for the sound source. Use FOCUSED for single sound sources like a
guitar amp and BROAD for wide sources like a whole drum kit.
• PRE-DELAY: Determines the amount of time between the initial input signal and the reverberation. Values
from the pre-delay range from 0 ms to 1 sec. All values are shown as ms (milliseconds).
• DECAY: Adjusts the length of the reverb, to shorten the time that the reverb tail lasts. Because of the
dynamic nature of the control, the displayed value is from 0% to 100%. To maintain the integrity of each
space, the length of 100% will depend on the selected room, while the 0% will be a decay of 1 ms.
• HPF (Hi pass filter): Sets the frequency of the high-pass filter before the reverberation path. It is a 6 dB/
octave filter with a range from 10 Hz to 20 kHz. The default setting is 10 Hz. All values are shown in Hz and
kHz.
• LOW (Low Shelf): Changes the gain of the low-shelf filter after the reverberation path. The frequency is set
at 200 Hz with a Q of 0.4 and its range is from -12 dB to 12 dB. The default setting is 0 dB. All values are
showed in dB.
• HIGH (High shelf): Changes the gain of the high-shelf filter after the reverberation path. The frequency is set
at 5 kHz with a Q of 0.4 and its range is from -12 dB to 12 dB. The default setting is 0 dB.
• LPF (Low pass filter): Sets the frequency of the low-pass filter before the reverberation path. It is a 6 dB/
octave filter with a range from 10 Hz to 20 kHz. The default setting is 20 kHz. All values are shown in Hz and
kHz.

On the right side of the interface you’ll find a mixer for adjusting the volume of each component of the plugin.
Each fader can adjust the volume from minus infinite all the way up to 0 dB.
• PAN: Adjusts the stereo field position of the relative sound signal.

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• WET LEFT/RIGHT: Adjusts the volume of the reverb effect.

• DRY: Adjusts the volume of the original sound.
• SOLO: Solos just the relative signal to let you hear only the processed signal or the original signal without
needing to adjust the WET/DRY mix.
• LINK: Allows the WET LEFT and WET RIGHT channels to be edited simultaneously.

When Studio A or Studio B Live Room is selected, two additional faders appear:
• MONO 1/2: Adjusts the volume of the MONO room microphones. MONO 1 is a ribbon mic and MONO 2 is
a condenser mic.

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51 – Joe Chiccarelli Vocal Strip

Joe Chiccarelli Vocal Strip is a compression chain developed in collaboration with Joe Chiccarelli. The goal is
to finely reproduce every detail of Joe’s signature vocal tracks processing within a single and easy to use plug-
in. The core of Joe’s workflow consists of three different compressors connected in parallel with the dry signal,
together with a de-essing unit, a high-pass filter and a 3-band EQ. Last, but not least, a set of 6 high-quality FX
units allow the user to get a professional sounding vocal track with just a single plug-in.

On the left side of the module you can find the input stage featuring an input level fader, a High-pass Filter and a
De-esser:
• INPUT LEVEL: fader control. Adjusts the amount of signal going into the processor. The signal can be set
from +10 dB until complete silence (-∞). The input state is saved in the presets. Default value is -1.5 dB. A
multicolor led in the top right corner of the input fader allows you to visualize the input signal level. The led
will turn to yellow at -6 dbFS and red at 0 dBFS.
• HPF: knob control. Adjusts the cutoff frequency for the high-pass filter. Values range from 20 Hz to 250 Hz.
Default value is 30 Hz.
• DE-ESS AMOUNT: knob control. Adjusts the amount of de-essing applied to the input signal. Values range
from 0 (off) to 10 (max). Default value is 3.0. A green led in the top-right corner of the De-esser module will
light up every time the De-esser is engaged.
• DE-ESS TYPE: switch control. Selects between two different frequencies for the lower de-essing band. “A”
corresponds to 6.4 khz while “B” corresponds to 7.6 khz. Default value is A.

After the input stage there is the main block of this processor: the parallel compression module.
This module features 3 unique compressors, each one featuring an analog-style Gain Reduction meter and
an overload LED that turns on when the compressor is clipping. Each compressor is in parallel and working
independently from the others.
• COMP 1 AMOUNT: knob control. Adjusts the amount of compression for the Main Parallel Compressor. The
compressor is based on the UREI 1176 and the COMP 1 knob is a macro of the available controls on the

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hardware counterpart. Values ranges from 0 to 10. Default value is 3.3.

• COMP 1 OUTPUT: knob control. Adjusts the output level of the Main Parallel Compressor. Values ranges
from -inf to +12 dB. Default value is 1.5 dB.
• COMP 1 MUTE: button control. Enables or disables the Main Parallel Compressor. Default value is OFF
(Processor Enabled).
• COMP 2 AMOUNT: knob control. Adjusts the amount of compression for the Retro Parallel Compressor.
The compressor is based on the Fairchild 670 and the COMP 2 knob is a macro of the available controls on
the hardware counterpart. Values ranges from 0 to 10. Default value is 3.1.
• COMP 2 OUTPUT: knob control. Adjusts the output level of the Retro Parallel Compressor. Values ranges
from -inf to +12 dB. Default value is 1.5 dB.
• COMP 2 MUTE: button control. Enables or disables the Retro Parallel Compressor. Default value is OFF
(Processor Enabled).
• COMP 3 AMOUNT: knob control. Adjusts the amount of compression for the Heavy Parallel Compressor.
The compressor is based on the Universal Audio 176 Tube Compressor and the COMP 3 knob is a macro of
the available controls on the hardware counterpart. Values ranges from 0 to 10. Default value is 3.3.
• COMP3 OUTPUT: knob control. Adjusts the output level of the Heavy Parallel Compressor. Values ranges
from -inf to +12 dB. Default value is -14.5 dB.
• COMP 3 MUTE: button control. Enables or disables the Heavy Parallel Compressor. Default value is OFF
(Processor Enabled).

After the compressors you can find an EQ section featuring various analog-modelled filters:
• BODY: knob control. Adjusts bass boost amount of an analog-modeled EQ. Values range from 0 (off) to 10
(max boost). Default value is 2.5..
• PRESENCE: knob control. This knob is a macro: in the lower half of the range it applies a mid cut
accordingly to the PRE/POST selection, while in the upper half of the range it applies a mid boost right after
the Main Parallel Compressor. Values range from -5 to +5. Default value is 0.5.
• AIR: knob control. Adjusts air boost amount on the EQ. Values range from 0 (off) to 10 (max boost). Default
value is 4.0.
• PRE/POST: switch control. Selects whether the EQ is inserted PRE or POST the Parallel Compressors
block. Default value is PRE.

Right under the EQ section you’ll find a simple FX section featuring various high-quality effects for your vocal
tracks.
• REVERB: knob control. Adjusts the output level of the reverb effect. Values range from 0 to 10.
• Default value is 0.0.
• REVERB A/B: latch buttons. Selects the reverb effect type. A corresponds to Plate, B corresponds to
Chamber. Only one button can be enabled at a time and when the other one is pressed the first one is
automatically switched off. Default value is A.
• SIZE: knob control. Adjusts the decay of the reverb effect. Value ranges from 0 to 10. Default value is 2.5.
• MOD: knob control. Adjusts the output level of the modulation effect processor. Values range from 0 to 10.
Default value is 2.5.
• MOD A/B: latch buttons. Selects the modulation effect type. A corresponds to Doubler, B corresponds

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to Chorus. Only one button can be enabled at a time and when the other one is pressed the first one is
automatically switched off. Default value is A.
• ECHO: knob control. Macro that adjusts both the delay effect level and its feedback value. Values range
from 0 to 10. Default value is 4.0.
• ECHO A/B: latch buttons. Selects the delay effect type. A corresponds to Tape, B corresponds to Ping
Pong. Only one button can be enabled at a time and when the other one is pressed the first one is
automatically switched off. Default value is A.
• TIME: knob control. Adjusts the time division for the delay effect. The delay effect take its base tempo from
the Host and then creates subdivisions of this tempo according to the TIME value. Value ranges from 1/64 to
1/1. Default value is 1/16.

Last, but not least, on the far right you can find the output stage featuring a distortion module, a global dry/wet
control and an output fader.
• DRIVE: knob control. Adjusts the amount of distortion applied to the output signal. Values range from 0 (no
distortion) to 10 (max distortion). Default value is 0.0.
• COLOR: knob control. Adjusts the tone of the distortion processor. Values range from -5 to +5.
• Default value is 0.0.
• OUTPUT LEVEL: fader control. Adjusts the amount of signal outputted by the processor. The signal can be
set from +10 dB until the complete silence (-∞). The output state it is saved in the presets. Default value is
-5.0 dB. A multicolor led in the top right corner of the output fader allows you to visualize the output signal
level. The led will turn to yellow at -6 dbFS and red at 0 dBFS.
• DRY/WET: knob control. Adjusts the balance between processed signal and dry signal. Values range from
DRY to WET. Default value is 75%. It’s worth noting that this dry/wet knob doesn’t alter the FX section level.

Other controls available are the following:

• RESET: Momentary switch control. This control resets the processor to its default state. The reset is always
in the UNPRESSED position, hence disabled, and it is enabled when the switch is in the PRESSED position.
The switch will go in the PRESSED position when the control is clicked.
• POWER: Switch control. This control enables or disables the processor. When enabled the switch is in the
PRESSED position and the LED is lit. When disabled the switch is in the UNPRESSED position and the LED
is unlit.

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52 – The Farm Stone Room

The Farm Stone Room is the third product in the “Studio Reverb” convolution reverb family, following “FAME”
and “Sunset Sound.”
This new reverb plug-in aims at faithfully reproducing the rich and characteristic sound of the Stone Room, the
main drum recording room located at The Farm Studio, Fisher Lane, England.

The left half of the interface features the main controls for customizing the reverb parameters and a render
based on the real The Farm Stone Room.
Available controls are:
• PRE-DELAY: Knob control. Determines the amount of time between the dry signal and the reverberation.
Values for the PRE-DELAY range from 0 ms to 999 ms.
• DECAY: Knob control. Adjusts the length of the reverb. This is used to shorten and lengthen the time that
the reverb tail lasts. The values depend on the recorded space’s IRs. Because of the dynamic nature of the
control, the shown value will be from 0% to 100%: To maintain the integrity of the room space, the 100%
will depend on the IR length, while the 0% will be the IR shortened to 1 ms.
• SIZE: Knob control. Adjusts the size of the room. This is achieved by resampling the IR in order to increase
or decrease the length without cutting any part. Values range from 0% to 200%.
• HP (High-Pass Filter): Knob control. Changes the frequency of the high-pass filter before the reverberation
engine. It is a 6 dB/octave filter with a range from 10 Hz to 22 kHz.
• CURTAIN CLOSED: Switch control. Selects which position is used for the rear wall curtains. If enabled, the
curtains will be closed resulting in a more dampened sound.

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The right half of the interface features a mixing desk inspired set of controls that allows the user to fully
customize the sound and the character of the plug-in.
The controls are divided into 4 vertical sections, each one controlling different signals. The first one controls the
input stage and the dry signal.
Available controls are:
• INPUT GAIN: Knob control. Adjusts the amount of signal going into the processor. The signal can be
adjusted between -20 dB and + 20dB from its nominal level. The input state is saved in the presets.
• INPUT MONO: Switch control. Selects whether the program audio is passed to the reverb processor as True
Stereo or Summed Mono.
• DRY PAN: Knob control. Adjusts the dry signal position in the stereo field. Values range from -100 (Left) to
+100 (Right).
• DRY LEVEL: Fader control. Adjusts the amount of dry signal getting into the output mix. Values range from
-∞ to +6 dB.
• DRY SOLO: Button control. If enabled, isolates the dry signal by muting everything else. Solo buttons work
in an additive way, meaning that more than one can be enabled at the same time.

The second section controls the STEREO ROOM signal, a faithful model of The Farm
Stone Room captured through a set of Neumann U87 microphones.
A stereo compressor and a 3-band EQ are available in this section.
Available controls are:
• STEREO ROOM COMP THRESH: Knob control. Adjusts the threshold of the compressor applied to the
stereo room signal. Values range from -30 dB to 0 dB.
• STEREO ROOM COMP MIX: Knob control. Adjusts the dry/wet balance for the stereo room compressor.
Values range from 0% (dry only) to 100% (wet only).
• STEREO ROOM PAN: Knob control. Adjusts the stereo room signal position in the stereo field. Values range
from -100 (Left) to +100 (Right).
• STEREO ROOM WIDTH: Knob control. Adjusts the stereo field width of the stereo room signal between 0%
(mono sum) and 100% (L and R full open).
• STEREO ROOM LEVEL: Fader control. Adjusts the amount of stereo room signal getting into the output
mix. Values range from -∞ to +6 dB.
• STEREO ROOM SOLO: Button control. If enabled, isolates the stereo room signal by muting everything
else. Solo buttons work in an additive way, meaning that more than one can be enabled at the same time.
• HIGH (STEREO ROOM): Knob control. Adjusts the gain of the high-shelf filter on the stereo room signal.
The filter is set at a frequency of 5 kHz and its gain ranges from -12 dB to +12 dB.
• MID (STEREO ROOM): Knob control. Adjusts the gain of the mid-bell filter on the stereo room signal. The
filter is set at a frequency of 2 kHz with a Q value of 0.4 and its gain ranges from -12 dB to +12 dB.
• LOW (STEREO ROOM): Knob control. Adjusts the gain of the low-shelf filter on the stereo room signal. The
filter is set at a frequency of 200 Hz and its gain ranges from -12 dB to +12 dB.

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The third section controls the MONO CRUSH signal, a punchy and dirty mono room captured using a vintage
STC4035 microphone.
In order to faithfully reproduce the world-famous sound from The Farm, an SSL Listen
Mic Compressor (LMC) has been modeled and inserted in this section, together with a 3- band EQ.
Available controls are:
• MONO CRUSH LMC THRESH: Knob control. Adjusts the threshold of the LMC compressor applied to the
mono crush signal. Values range from -30 dB to 0 dB.
• MONO CRUSH LMC MIX: Knob control. Adjusts the dry/wet balance for the LMC compressor. Values
range from 0% (dry only) to 100% (wet only).
• MONO CRUSH PAN: Knob control. Adjusts the mono crush signal position in the stereo field. Values range
from -100 (Left) to +100 (Right).
• MONO CRUSH LEVEL: Fader control. Adjusts the amount of mono crush signal getting into the output mix.
Values range from -∞ to +6 dB.
• MONO CRUSH SOLO: Button control. If enabled, isolates the mono crush signal by muting everything else.
Solo buttons work in an additive way, meaning that more than one can be enabled at the same time.
• HIGH (MONO CRUSH): Knob control. Adjusts the gain of the high-shelf filter for the mono crush signal. The
filter is set at a frequency of 5 kHz and its gain ranges from -12 dB to +12 dB.
• MID FREQ (MONO CRUSH): Knob control. Adjusts the frequency of the mid-bell filter on the mono crush
signal. The filter Q is fixed at 0.4. Values range from 200 Hz to 3 kHz.
• MID GAIN (MONO CRUSH): Knob control. Adjusts the gain of the mid-bell filter on the mono crush signal.
Values range from -12 dB to +12 dB.
• LOW (MONO CRUSH): Knob control. Adjusts the gain of the low-shelf filter on the mono crush signal. The
filter is set at a frequency of 200 Hz and its gain ranges from -12 dB to +12 dB.

Finally, the fourth section is the master section and features the following controls:
• POWER: Switch control. Enables or disables the processor. When enabled the switch is in the PRESSED
position and the LED is lit. When disabled the switch is in the LIFTED position and the LED is unlit. This
control can be used by a single click.
• RESET: Momentary switch control. Resets the processor to its default state. The reset is always in the
LIFTED position, hence disabled, and it is enabled when the switch is in the PRESSED position. The switch
will go in the PRESSED position when the control is clicked. This control can be used by a single click.
• OUTPUT LEVEL: Knob control. Adjusts the amount of signal going into the output bus. The signal can be
adjusted between -12 dB and 12 dB from its nominal level. The output state is saved in the presets.

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53 – Bass One

Bass One allows the user to have full control over the low frequencies spectrum of the track.
The GUI is inspired by the ONE processor since the concept behind both processors is the same: having a
single plugin that can enhance a track in many different ways.
While ONE was aimed at adjusting the full frequencies spectrum, Bass One is focused towards low frequencies
only since handling those correctly is a critical and not so straightforward step for basically every modern music
production.

Available controls are:

• LO-FREQ: Adjusts the frequency at which the processor will start to operate. Frequencies below this value
will pass unaltered. Values range from 20Hz to 200Hz.
• HI-FREQ: Adjusts the frequency at which the processor will stop to operate. Frequencies over this value will
pass unaltered. Values range from 200Hz to 400Hz.
• ENHANCE: knob control. Adjusts the threshold for the expander. Increasing this value will lead to a higher
amount of expansion in the processed band, that translates to a cleaner and more focused low-end. Values
ranges from 0 to 10.
• MODE (ENHANCE): Selects the enhancing expander hardness. Options are SOFT or HARD.
• RESET: This control resets the processor to its default state.
• BYPASS: This control enables or disables the processor.
• PUSH: Adjusts the amount of compression for the processed band. It’s a bipolar knob, when used in the
positive range it applies upward compression while on the negative range it applies downward compression.
This allow for a natural increase and decrease of RMS value depending on the PUSH knob value. Values
ranges from -15 to 15.
• FOCUS: Adjusts the stereo width of the processed band. When set at 0 the stereo image is left untouched
while at 10 the signal is entirely summed in mono, which can help in making bass frequencies more clear
and focused.

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• OUTPUT: Adjusts the volume of the processed band. Values ranges from -15dB to +15dB.
• DRIVE: Adjusts the amount of distortion applied to the processed signal. Values range from 0 (no distortion)
to 10 (max distortion).
• ANALOG: Adjusts the amount of saturation applied to the processed band. Adding saturation allow the low
frequencies to better “cut through the mix”. Values ranges from 0 to 10.
• HARMONICS: Adjusts the harmonic content of the drive and saturation processors.. Values range from 0 to
10.
• SUB: Enables or disables the pitch shifter set an octave below the selected range.
• MULTI METER: In the center of the GUI there is a round display that acts as a multi meter. It features two
different operating modes: MULTI and SPECTRUM.

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54 – Dual Spring

Dual Spring is a versatile, highly customisable convolution spring reverb processor.

The core of this plugin are 10 state-of-the-art spring IRs captured from iconic amplifiers and spring units.
Thanks to a powerful double spring IR engine the user can unleash all his creativity and create virtually any
spring sound he can think of.
The interface is pretty self explanatory. Together with all the controls, there are IN and OUT meters that display
the real-time level of the signals entering and exiting from the processor.
Behind the meters a vacuum tube can be seen glowing when the input/output stage is saturating. (Preamp-like
saturation).
At the center of the GUI a slim display shows a graphical representation of the two springs currently used for the
reverb.

Available controls are:

• INPUT: Adjusts the amount of signal going into the processor. Values range from -15dB to +15dB.
• DRY/WET: Adjusts the balance between processed signal and dry signal. Values range from 0% (DRY) to
100% (WET).
• OUTPUT: Adjusts the level of the signal outputted by the processor. Values range from -15dB to +15dB.
• RESET: This control resets the processor to its default state.
• POWER: Enables or disables the processor.

On the lower part of the GUI there are two identical spring modules. Each one features the following controls:
• MATERIAL: This knob is the main tone modifier of this reverb. Different spring materials result in a different
sound and character. Technically speaking, each material consist of a different spring IR. Available materials,
from mellow to bright, are the following: Pb (Lead), Fe (Iron), Cu (Copper), Ag (Silver), Au (Gold). Default: Fe
for Spring A, Ag for spring B.

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• TONE: Adjusts the overall tone of the spring. Values range from -5 (MELLOW) to +5 (BRIGHT).
• LENGTH: Adjusts the decay of the reverberation. Values range from 10% up to 200%.
• STRETCH: When enabled, the LENGTH will control the re-size of the IR based, rather than the added fade
out.
• MONO: Selects the output mode. A different set of mono IR is used if MONO is enabled.
• PAN: Adjusts the stereo positioning for the reverb signal. Values range from 10L (LEFT) to 10R (RIGHT) with
C as CENTER.

At the center between the two springs there is a blend control.

• BLEND: Adjusts the mix ratio between the two springs. Values range from -100 (A only) to +100 (B only).
• SPRINGS: The two springs can be clicked and a sound is played simulating a hit on the spring. This can be
useful both for quickly hearing the spring settings or for artistic purposes.

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55 – Lo-Fi Punch

Lo-Fi Punch is a creative-oriented processor that can intuitively deliver Lo-Fi vibes to your tracks.
Providing the users a fun and fast way of adding Lo-Fi distortion, modulations and compression is mandatory
considering the higher and higher popularity that the Lo-Fi genre is facing in the last years.
This plugin can be considered an extended version of the Lo-Fi available on Syntronik.

Available controls are:

• INPUT: Adjusts the level of the signal entering in the processor. Values range from -15db to +15dB.
• LO-FI: Adjusts the overall amount of Lo-Fi distortion. Values range from 0 to 10.
• LO-FI TYPE: Selects the algorithm for the Lo-Fi control. TYPE 1 is Saturation (Vintage Crush) while TYPE 2
is bit-crusher (8-bit vibes).
• FILTER: Adjusts the tone of the output signal. When set at minimum value the signal can pass unaltered
while at maximum value only a small band centered at around 700Hz will be outputted. Values ranges from 0
to 10.
• DRIVE: Adjusts the amount of distortion. Values ranges from 0 to 10.
• WIDTH: Adjusts the stereo field of the processed band. When the knob is at its center value the signal is not
altered. Reducing this value will make the signal more “mono-summed” in the center. Increasing this value
will add stereo spread effect. Values ranges from 0 to 10.
• NOISE: Adjusts the amount of noise introduced into the LoFi algorithms. Values ranges from 0 to 10.
• WOW: Adjusts the amount of wow effect. This is a low-frequency cyclical wobble typically caused by
unstable or worn out vinyl discs. Values ranges from 0 to 10.
• FLUTTER: Adjusts the amount of flutter effect. This is a high-frequency cyclical wobble typically caused by
bad calibrated or worn out tape machines. Values ranges from 0 to 10.
• PITCH: Adjusts the pitch of the processed audio, useful for detuning effects. The pitch changes continuously
from -5 semitones up to +5 semitones. Values range from -5 to +5.

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• COMPRESSION: Adjusts the amount of compression applied to the signal. Values ranges from 0 to 10.
• SOFT/HARD: Selects the type of compression.
• OUTPUT: Adjusts the level of the signal outputted by the processor. Values ranges from -15dB to +15dB.
• DRY/WET: Adjusts the balance between processed signal and dry signal. Values range from 0% to 100%.
• RESET: This control resets the processor to its default state.
• POWER: This control enables or disables the processor.

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56 – Delay Lab

Delay Lab is a light and simple delay processor characterised by a clear and detailed sound bundled with a
series of digital effects that can add a great amount of depth and impact to virtually any sound.

Available controls are:

• POWER: This control enables or disables the processor.
• RESET: This control resets the processor to its default state.
• INPUT: Adjusts the amount of signal going into the processor. Values range from -15dB to +15dB.
• MIX: Adjust the dry/wet balance of the output signal. Values range from 0% (DRY) to 100% (WET).
• OUTPUT: Adjusts the level of the signal outputted by the processor. Values range from -15dB to +15dB.
• STEREO MODE: Select the stereo operating mode for the delay. Available modes are MONO, STEREO and
PING-PONG.
• PAN: Controls the position of the delay signal in the stereo field. Values range from 10L (LEFT) to 10R
(RIGHT) with C as CENTER.
• TIME: Configures the delay time, which is the interval between each repetition. Values range from 4/1 to
1/512 when BPM SYNC is ON and from 2 seconds to 0.001 seconds when BPM SYNC is OFF.
• SYNC: Decides wether the delay is synced with the host BPM or not. If set to ON the time scale will be
displayed in bar divisions. If set to OFF the time scale will be in milliseconds.
• FEEDBACK: Configures the delay feedback. Its value is expressed as a percentage and represent the
amount of output signal that is fed back to the input stage. Setting this value over a certain level will cause a
feedback loop to build up. Values range from 0% to 100%.
• GATE: Adjusts the threshold level for the wet signal. Values ranges from 0 to -50dB.
• HIGH PASS: Adjusts the low frequency extension of the delay. Frequencies below this value are cut off.
Values range from 20Hz to 900Hz.

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• LOW PASS: Adjusts the high frequency extension of the delay. Frequencies over this value are cut off.
Values range from 0.5kHz to 20kHz.
• DOUBLING: Key feature of the digital delay, this knob selects the amount of doubling effect applied to the
processed signal. Values range from 0 to 10.
• ON/OFF (DOUBLING): Enables or disables the doubling effect.
• BIT REDUCTION: Key feature of the digital delay, this knob selects the amount of bit reduction applied to
the processed signal. Values range from 0 to 10.
• ON/OFF (BIT REDUCTION): Enables or disables the bit reduction effect.
• TO DRY (BIT REDUCTION): If enabled the bit reduction effect will be applied both to the dry signal and the
processed one. If disabled the effect will be applied only on the processed signal.
• DOWNSAMPLING: Selects the amount of downsampling applied to the processed signal. Values range
from 0 to 10.
• ON/OFF (DOWNSAMPLING): Enables or disables the downsampling effect.
• TO DRY (DOWNSAMPLING): If enabled the downsampling effect will be applied both to the dry signal and
the processed one. If disabled the effect will be applied only on the processed signal.
• CLIPPING: Selects the level at which the signal will get digitally clipped at. Values range from -20dB to
-0.1dB but on the GUI is expressed as 0-10.
• ON/OFF (CLIPPING): Enables or disables the clipping effect.

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57 – Pusher

The Pusher is a brand new saturation plug in for T-RackS. Multiband processing at the core of this plug-in splits
the incoming signal into up to four frequency bands and treats each one separately with a dedicated set of
controls. The width of each band can be finely adjusted so that the processing can be very selective (in case of
a narrow band) or spread across a wider range. Each band is loaded with a powerful saturation engine, capable
of emulating from the slightest analog color all the way up to heavy distortion.

Available controls are:

• POWER: Enables or disables the processor.
• RESET: This control resets the processor to its default state.
• BAND SPLITTING SLIDERS: The three sliders allow the user to select the frequencies used by the plugin
to split the audio signal into 4 bands. Each slider can move until it hits a limit (slider 1 can move from 20Hz
up to slider 2 value, slider 2 can move from slider 1 value up to slider 3 value, slider 3 can move from slider 2
value up until 20000Hz).
• INPUT: Adjusts the level of the signal entering in the processor. Values range from -15db to +15dB.
• ATK: Adjusts the attack time for the envelope follower used as modulation source for all the ENVELOPE
knobs in the 4 bands. Values range from 0.1ms to 50.0ms.
• REL: Adjusts the release time for the envelope follower used as modulation source for all the ENVELOPE
knobs in the 4 bands. Values range from 10.0ms to 1000.0ms.
• OUTPUT: Adjusts the level of the signal outputted by the processor. Values range from -15db to +15dB.
• DRY/WET: Adjusts the balance between processed signal and dry signal. Values range from 0% to 100%.
• BAND BLEED: Adjusts how much the overtones generated by the distortion algorithm bleed outside the
band that they come from. Values range from 0% to 100%.

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Each of the 4 bands features 7 parameters that controls how each band saturation behaves:
• DISTORTION TYPE: Selects which distortion algorithm for the band. A selection of 5 algorithms is available.
Values range from A to E.
• DRIVE: Adjusts the amount of distortion applied to the signal. Values range from 0 (no distortion) to 10 (max
distortion).
• COLOR: . Adjusts the tone of the distortion algorithm, lower values results in mellow distortion while higher
values means brighter distiortion. Values range from 0 to 10.
• LEVEL: Adjusts the output level for the selected band. Values range from -6dB to +6dB.
• DRIVE ENVELOPE: Controls the amount and direction of the DRIVE knob modulation. Values range from
-1.0 to +1.0.
• COLOR ENVELOPE: Controls the amount and direction of the COLOR knob modulation. Values range from
• -1.0 to +1.0.
• LEVEL ENVELOPE: knob control. Controls the amount and direction of the LEVEL knob modulation. Values
range from -1.0 to +1.0.
• LINK: When this lever switch is engaged, the controls of the two bands on its sides is linked, meaning that
moving any knob on one of the two linked bands results in the same movement on the linked band knobs.
• BAND POWER: Enables or disables the processing on the band. When disengaged, the relative band
processing is bypassed and the audio of that band is left unaltered. This switch also controls the two states
of the background image for the relative band.
• SOLO: Allow the user to isolate the relative band, to listen precisely how the effect is sounding without any
distraction.

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58 – Filter Fusion

Filter Fusion is a plugin filter processor capable of answering to virtually any filtering needing thanks to its
different fully-customisable filter types, an envelope follower, a LFO and a step sequencer.
No matter if you need a lush sounding 6dB/oct high pass filter or a EDM style heavy side chain pumping effect,
this plugin will never let you down!
The heart is right at the center of the interface and is the M-Type filter, based on the Moog® transistor ladder
filter. Submenus include LP, HP, BP, PEAK, NOTCH, with the available slopes in the range 6-12-18-24dB/oct.

MAIN controls are the following:

• RESET: Resets the plugin to its initial state.
• POWER: Enables or disables the audio processor.
• INPUT: Adjusts the signal level entering the processor. Values range from -15 to +15dB. A multicolor LED
on top of this knob displays the input signal level. The LED colors thresholds are set as follows: -60dBFS =
green,-9dBFS = yellow, -0.1dBFS = red.
• OUTPUT: Adjusts the signal level outputted by the processor. Values range from -15 to +15dB. A multicolor
LED on top of this knob displays the output signal level. The LED colors thresholds are set as follows:
-60dBFS = green, -9dBFS = yellow, -0.1dBFS = red.
• DRY/WET: Adjust the balance between the unprocessed (dry) and processed signal (wet). Available values
are in percents and range from 0% (DRY) to 100% (WET).
• FILTER TYPE: Selects the type of filter. Available types include: Low Pass (LP), High Pass (HP), Band Pass
(BP), PEAK, NOTCH.
• FILTER SLOPE: Selects the slope of the filter. Available slopes include: 6, 12, 18, 24dB/oct.
• CUTOFF: Selects the cutoff frequency for the filter. Values range from 20Hz to 20000Hz. The slider located
under this knob controls the amount of modulation applied to this knob and ranges from -1 to +1.
• RESONANCE: Adjusts the resonance of the filter. Values range from 0 to 10.The slider located under this
knob controls the amount of modulation applied to this knob and ranges from -1 to +1.

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• DRIVE: Adjusts the amount of distortion applied to the filtered signal. Values range from 0 to 10. SPREAD:
Controls the stereo spreading effect of the filter.
• MODULATION: Controls the balance between the 3 different modulation sources. Each of the 3 parameters
is calculated depending on the area of the triangle formed between the opposing side and the current
joystick point. For example, considering the STEP SEQUENCER parameter (which is on the top vertex of
the triangle), its value is based on the area between the bottom side and the current point so that when the
joystick is positioned on the bottom side the area is 0 and the STEP SEQUENCER value is 0. When the
joystick is positioned on the top vertex, the area is maximised and the STEP SEQUENCER value is 1.

ENVELOPE FOLLOWER controls are the following:

• SENSITIVITY: Adjusts the gain of the envelope follower circuit. Higher values make the envelope follower
react to weak signals but this could introduce more noise. Values range from 0 to 10.
• ATTACK: Adjusts the attack time for the envelope follower circuit. Values range from 0ms to 150ms .
• RELEASE: Adjusts the release time for the envelope follower circuit. Values range from 10ms to 1000ms.

STEP SEQUENCER controls are the following:

• STEP SLIDERS 1-16: slider control. Each step have an individual slider that adjusts how each step modify
the modulation, allowing to adjust intensity from 0 to 1 in both positive and negative directions. Values range
from -1 to +1.
• SYNC: Enables or disables tempo sync with host.
• DIVISION: Allow the user to define the divition for the selected BPM between 2/1 to 1/32.
• BPM: Allow the user to define the tempo for the step sequencer. This field is locked if the SYNC button is
engaged. Values range from 5 to 500.
• STEP: Allow the user to define the number of steps before restarting from step 1. Available values are integer
numbers from 1 to 16.
• MOD: Allow the user to choose the sequencer operating mode. Available modes are: Forward (FWR),
Backwards (BWR), Back and Forth (BNF) and Random (RND).

LFO controls are the following:

• WAVE: Selects the waveform for the LFO modulator. Available waveforms are: Sine, Triangle, Square, Saw L,
Saw R, Random.
• RATE: Adjusts the frequency of the LFO modulator. Values range from 0.1 to 100Hz SYNC: Enables or
disables tempo sync with host for the rate parameter
• RETRIG: Enables the receipt of the Start and Stop signal from the DAW, and consequentialy retrigger the
phase of the LFO.
• SMOOTH: Adjust the smoothness of the LFO wave. Values goes from 0 to 10.

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59 – Channel Strip X

The Channel Strip X is an all-in-one solution for fast mixing. It is a channel strip with all the necessities for mixing
an instrument or an ensemble. It includes a 6 curves parametric EQ, a De-Esser, a multiband compressor, a
transient designer, and a stereo enhancer.
Instead of loading a series of processors for mixing, load Channel Strip X and by selecting the instrument from
the target list, you will load an EQ preset tailored for that instrument, and with just a couple of macros adjust it to
perfectly fit the program.
Effects are taken out of the picture, in order to give you more time be creative with those.

Available controls are:

• TARGET LIST: This is the list of EQ preset curves made for instruments or ensemble. User can save EQ
settings as targets at any time.
• INPUT: Adjust the level of the audio going into the processor. By default is set to 0 (no gain, no reduction, 12
o’clock) but it will range between +/- 15 dB.
• VINTAGE: When engaged, the button will enable the pre-amp emulation of the API.
• DE-ESSER: Adjust the amount of de-essing processing, with A/B modes. Range goes from 0 to 10.
• EQ Curves: It is a six band bi-quad filter. Band 1 and 6 will always be Shelfs, while band 2 to 5 will be
bells. All with adjustable Freq, Gain and Q. Regarding the Q, there will be an inverse relation with the
Gain: at higher gain, the Q will increase his value.
• EQUALIZER: Adjust the EQ impact. Increase or decrease the effect of the EQ on a signal, by adjusting the
gaining of the various bands. When decreasing, value 0 of the knob would mean that the EQ isn’t effecting
the signal. When increasing, the value 10 of the know would mean the EQ is “double”. Range goes from 0 to
200% and default is 100% (EQ curve default).
• COMPRESSION: Control the amount of multiband compression The output will have an auto-gain, to
compensate. The FAST button, will change the Attack ad Release time of all the bands
• TRANSIENT RECOVER: Adjust the amount transient enhancing applied to the signal.

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• OUTPUT: The output knob will adjust the output signal by +/-15dB.
• STEREO ENHANCE: The stereo enhance control, by default will not increase or decrease the stereo image
of the track. Range is +/- 10, where -10 is Mono summing, while +10 is the maximum stereo opening
available. This control will work from the MONO MAKER upper limit setting to 20kHz.
• MONO MAKER: Different from the stereo enhancer, the mono maker tries to decrease the stereo image of
the track until a certain frequency. the starting frequency is 20Hz, while the upper frequency can be decided
by the user from 20Hz to 300Hz.

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60 – Support
For any questions you may have, please refer to the FAQ webpage at:
ikmultimedia.com/faq
Here you will find answers to the most commonly asked questions.

To submit a Technical Support Form, go to:

ikmultimedia.com/support

For other requests such as Product, Sales, or Web info, please go to:
ikmultimedia.com/contact-us

60.1 – IK Product Manager

The IK Product Manager gives you one central location to manage all the latest IK Multimedia products. It’s your
central command for registration, downloads, installation, authorization, checking for updates, and much more.

• Register hardware and software in one location

• Download, install, and authorize software
• Simplified sounds download with pause/resume
• Authorize and de-authorize computers
• Update software, sounds and firmware
• Easily manage your entire IK collection

Download IK Product manager at:

ikmultimedia.com/pm

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IK Multimedia Production Srl IK Multimedia US, LLC IK Multimedia Asia
Via dell’Industria, 46, 590 Sawgrass Corporate Pkwy. TB Tamachi Bldg. 1F, MBE #709
41122 Modena Sunrise, FL 33325 4-11-1 Shiba
Italy USA Minato-ku, Tokyo 108-0014
Japan

www.ikmultimedia.com

T-RackS®, SCC™, DSM™ are trademarks or registered trademarks property of IK Multimedia Production. Leslie® is a trademark
property of Hammond Organ/Leslie, wholly-owned brand subsidiaries of Suzuki Musical Instruments Mfg., Ltd. of Hamamatsu, Japan,
used under license. TASCAM® and TEAC® are trademarks property of TEAC Corporation Japan used under license.
All other product names and images, trademarks and artists names are the property of their respective owners, which are in no way
associated or affiliated with IK Multimedia. Product names are used solely for the purpose of identifying the specific products that were
studied during IK Multimedia’s sound model development and for describing certain types of tones produced with IK Multimedia’s digital
modeling technology. Use of these names does not imply any cooperation or endorsement.
Fairchild® is a registered trademark property of Avid Technology, Inc. Pultec® is a registered trademark property of Pulse Techniques,
LLC. Ampex® is a registered trademark property of Ampex Corp. Studer® is a registered trademark property of Studer International AG.
Revox® is a registered trademark property of Revox AG Corporation.
LA-2A and 1176 LN are registered trademarks of Universal Audio, Inc. IK Multimedia’s products, specifically, the T-RackS Black 76
Limiting Amplifier and T-RackS White 2A Leveling Amplifier, are manufactured and developed by IK Multimedia based on its own
modeling techniques. Universal Audio has neither endorsed nor sponsored IK Multimedia’s products in any manner, nor licensed any
intellectual property for use in this product.
Mac, the Mac logo, the Audio Units logo, iPhone®, iPod touch®, iPad® are trademarks of Apple Computer, Inc., registered in the U.S.
and other countries. App Store is a service mark of Apple Inc. Windows and the Windows logo are trademarks or registered trademarks
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Avid, the Avid logo, Pro Tools, and RTAS are trademarks or registered trademarks of Avid Technology, Inc. in the U.S. and other countries.

All specifications are subject to change without further notice.

Document Version: 6.0.0

Latest Update: 2024/07/12

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