SSL Native v6.5 - User Guide
SSL Native v6.5 - User Guide
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SSL Native V6
User Guide
Bus Compressor
The Bus Compressor plug-in is based on the legendary centre section bus compressor found on SSL's large-format
analogue consoles. It provides high quality stereo compression, giving you critical control over the dynamic range of
audio signals.
Uses may include inserting the bus compressor over a stereo mix, which has the effect of ‘gluing’ the mix together whilst
still maintaining a big sound. The dynamics of drum overheads or whole drum kits can be controlled very effectively with
the bus compressor. As it is available as either a stereo or mono plug-in the bus compressor can be used for practically
any application that requires superior compression.
Key Features
• Dual-symmetrical knee design allows detailed shaping of the compression characteristic.
• Advanced side-chain architecture using 1st order filters delivers user-friendly frequency dependant parallel
compression.
• Amplitude Histogram and Gain Reduction history displays provide advanced real-time pre/post signal analysis.
• Max Gain Reduction control provides genuine vintage compressor characteristics.
• Intuitive user interface with drag and move graphic, mouse wheel and numeric editing.
• A/B facility for instant comparison of two different compression settings.
• Proprietary preset management functions providing compatibility between all DAW platforms.
• Global latency-free bypass.
• Superb mastering-grade audio quality delivered by SSL’s 64-bit floating point engine.
• Preset library based on settings used by industry-leading mix engineers.
Key Features
• 24-band fully parametric high quality digital EQ, featuring unique anti-cramping algorithms with no additional
CPU processing cost.
• A total of 17 different filter types: 5 different cut filters, 9 bell shapes, shelving filters, a parallel mode, and custom
HPF/LPF modes.
• Extensive control options including draggable EQ graph nodes, mouse wheel adjustment and numerical data
entry.
• Extremely low noise and low non-linear distortion filter algorithms resulting in the residual THD+N significantly
lower than 24-bit quantisation noise.
• Proprietary preset management functions providing compatibility between all major DAW platforms.
• A/B functionality for easy comparison of any two settings.
• Mid/Side and Left/Right spatial processing options.
• Individual band solo and bypass.
• Real-time FFT Analyser showing the result of the EQ processing on the audio spectrum.
• Phase & Step response graphs to show the effect of EQ processing.
• Superb mastering-grade audio quality delivered by 64-bit floating point engine.
• Preset library based on settings used by some of the world's top mix engineers.
Key Features
• Intelligent De-esser.
• Intelligent De-ploser.
• Three-band EQ.
• Compander featuring compression, downwards expansion and output drive.
• Extensive visual feedback including a real-time FFT analyser showing the result of the EQ processing on the audio
spectrum.
• Complete control over process order.
• Extensive control options including mouse wheel adjustment and numerical data entry.
• Proprietary preset management functions providing compatibility between all DAW platforms.
• A/B functionality for easy comparison of any two settings.
• Global latency-free bypass.
• Superb mastering-grade audio quality delivered by a 64-bit floating point engine.
• Preset library based on settings used by some of the world's top mixing engineers.
Key Features
• Transient shaper capable of drastically changing the attack characteristics of rhythmic tracks. An audition mode
makes for easy setup.
• Highly controllable gate featuring both open and close thresholds, attack, hold, release and range control.
• SSL Listen Mic Compressor with extra functionality.
• Separate high and low frequency enhancers provide spectral control not achievable with traditional EQ.
• Peak and RMS metering on both input and output.
• Wet/dry controls on both the main output and the LMC allow parallel processing to be easily dialled in.
• Process order control over all five sections gives complete flexibility over the serial signal chain.
• Latency-free bypass of all processing.
Key Features:
• Mode: Switches between Peak & RMS response.
• Input side-chain filtering.
• Valve In switch; introduces a tube compressor emulation for saturating the signal by adding harmonics.
• Auto make-up gain.
• Threshold, ratio, adjustable knee.
• Mix control for parallel compression.
Key Features:
• Emulates analogue gain circuits.
• The Drive controls the input level of the plug-in.
• By adjusting the Harmonics control the distortion characteristics can be shifted from 50’s valve-style overdrive (if
the Harmonics selection control is turned fully anti-clockwise for 2nd harmonic distortion) to 70’s transistor-style
grit (if the Harmonics selection control is turned fully clockwise for 3rd harmonic distortion).
• Depth and Shape control the amount of harmonics injected back into the signal, enriching the sound.
• Boost adds 6dB of headroom to avoid internal clipping.
• Dry/Wet allows parallel processing.
Mono Version
Stereo Version
Key Features
• X-Phase is an all-pass filter and time alignment tool for manually adjusting the phase.
• The gain is linear across the frequency range, but a change in phase is applied such that it ‘wraps’ at a specific
frequency and converges to 0r for all other frequencies.
• The Q-Factor of the filter will adjust the rate of phase-change across the frequency spectrum.
• 2nd Order inverts the filter.
• Delay (in various units).
• Link L/R for stereo signals.
• 180 degrees phase shift with All-pass ‘phase’ button.
Key Features:
• Four Reverb Types: Room, Hall, Plate, and Chamber.
• Early & Late Reflection types can be independently selected; for example, Small Room Early Reflections, with a
Large Plate Reverb Tail.
• 6 Band SSL EQ, with 3 bands of reverb time multipliers.
• Output compressor (applied to reverb only) side-chained by the ‘dry’ input signal; useful to help sit in the mix.
• Lockable Dry/Wet Mix control; fix the mix of dry and wet signal when switching between presets.
• Infinity reverb time switch.
• Reverb tail kill switch.
• Graphical display of EQ and reverb time multipliers, including FFT analysis with after-glow.
System Requirements
Before starting the installation process, confirm that your system meets the minimum system requirements to run SSL
Native plug-ins.
Windows
- Windows 7, (64 bit) 8 (64 bit), 8.1 (64 bit), and Windows 10 (64 bit).
- Intel Core 2 (or comparable) CPU running at 2.4GHz or higher.
- 4GB of RAM minimum, 8GB preferable.
- VST 2(64 bit), VST3 (64 bit), AAX Native (64 bit).
Mac
- OS X 10.10 Yosemite or higher (32 or 64-bit).
- Intel Dual Core Mac running at 2.4GHz or higher.
- 4GB of RAM minimum, 8GB preferable.
- AU (32 or 64bit), VST2 (32 or 64-bit), VST3 (32 or 64-bit), AAX Native (32 or 64-bit).
You will be sent a password by email. Once you have received this email, you are ready to continue.
During the purchase process you will be asked to enter your iLok User ID, then validate it (if it is not already present in
your SSL account). If you do not have an iLok account, you can create one here: https://www.ilok.com/#!registration
For further information on using iLok License Manager, please see the iLok website here: www.ilok.com/
During the installation process you can choose which plug-ins and formats you don’t require by unticking the appropriate
boxes:
In the disk image you will also find a file named iLok License Support.dmg. If you do not already have the iLok software
installed – for example if this is the first time you have used your iLok – you should double click this disk image and run
the installer. If you have a newer version of the software already installed you will be notified and prompted to quit the
installer.
If this is the first time you are installing the SSL Native V6 plug-ins you will need to activate the licenses
in your iLok account to your machine or a physical iLok 2/3 device. Please see the ‘Activating Plug-In
Licenses’ section of this user guide for more information.
Please note that a newer version of iLok License Manager may be available here: https://www.ilok.com/
During the installation you can choose which plug-ins and formats you don’t require by unticking the appropriate boxes:
In the zip file you will also find a file named License Support.exe. If you do not already have the iLok software installed
- for example if this is the first time you have used your iLok, you should double click this file to run the installer. If you
have a newer version of the software already installed you will be notified and prompted to quit the installer.
If this is the first time you are installing the SSL Native V6 plug-ins you will need to activate the licenses
in your iLok account to your machine or a physical iLok 2/3 device. Please see the ‘Activating Plug-In
Licenses’ section of this user guide for more information.
Please note that a newer version of iLok License Manager may be available here: https://www.ilok.com/
The SSL Native Logic Pro Essentials Installer.pkg is for Logic Pro users; it installs custom MCU maps for using an MCU
control surface with the SSL Native plug-ins in Logic Pro, as well as Logic Pro short names. These can be edited from the
Plug-in Manager in Logic if you wish: Logic Pro > Preferences > Plug-in Manager...
Introduction
The Channel Strip plug-in provides a complete SSL channel strip, based on the processing blocks of SSL's analogue
consoles, and includes high and low pass filters, a four band equaliser, compressor/limiter and gate/expander. The
channel strip can run in mono or stereo.
Before examining the plug-in in detail, the diagram below provides an operational overview.
When adjustments
Turn up input PEAK switches signal Engages Compression To activate are complete, correct
gain to a detection between Dynamics and Expansion Gate/Expander the output gain to a
suitable level. Peak and RMS modes. meters. turn up RANGE. suitable level.
Dynamics
• To press a switch in a plug-in, simply click on it. With all plug-ins, an on-screen indication adjacent to or surrounding
the switch will indicate when the switch is active.
• To turn a knob, click on it and drag it up and to the right. If your mouse has a scroll wheel, you can also turn knobs
by hovering over them and turning the scroll wheel.
• To move a knob slowly for fine adjustments, hold the To reset a knob to its default value, click on the knob
following on your keyboard whilst turning/scrolling: whilst holding the following on your keyboard:
• To view the value for any knob, hover over the knob cap with the mouse.
Automation
Every plug-in parameter can be automated in host applications that support automation. The method for recording and
editing automation varies from host to host. For specific instructions on using automation within the host, consult the
host application's documentation.
Input Section
Turn the GAIN knob to control the level of the incoming audio signal. The post-gain signal level is shown
above. Press Ø to invert the phase of the input signal.
Filter Section
There are two filters in the Filter section:
• The upper knob controls an 18dB/Octave high pass filter (20Hz to 500Hz).
• The lower knob controls a 12dB/Octave low pass filter (3kHz to 22kHz ).
Filters are inactive when turned fully anti-clockwise (OUT). Turn them clockwise to move the filter frequency
in from its extremity.
Press the E to switch the EQ emulation from G Series to E Series consoles. The diagrams below display the difference
between them:
G Series: The bell curve has a more rounded shape at low gains. G Series EQ is more subtle and is generally more
suited to instruments and vocals.
E Series: The bell curve is slightly more pointed. E Series EQ is more aggressive and is therefore better for removing
problem frequencies. It is generally more suited to drums.
Channel Equaliser Curves Amplitude (dBr) v Frequency (Hz)
25.0
5.0
0.0
-5.0
-10.0
-15.0
-20.0
-25.0
10 100 1k 10k 20k
Compressor/Limiter
To activate the Compressor/Limiter, turn the RATIO knob so that it is no longer set Input level Full scale
to 1:1.
Threshold
To turn the compressor into a ∞:1 limiter, turn the knob fully clockwise. Ratio
There is no gain makeup control as the THRESHOLD
Output level
knob controls both the level at which gain reduction
Gradient is
is introduced and the gain make-up, keeping the
shallow or flat
output level reasonably steady regardless of the for limiters
compression.
RELEASE controls how quickly the level returns to normal after the input level has
Attack
dropped below the threshold (measured in seconds). The attack time is adjusted
Expander/Gate
Input level Full scale
To activate the Expander/Gate, turn the RANGE knob so that it is no longer zero.
The green indicators in the right-hand of the two Threshold
meters in the centre of the Dynamics section show
the amount of gain reduction being introduced.
Higher ratio
By default, the Expander/Gate section functions (steeper)
as a gate. To switch to the expander, press the for gates Range
Output level
EXPAND switch. Ratio
1:1 ratio
The THRESHOLD function uses different levels to ‘open’ the gate to audio and to below and
‘close’ it again – the level at which the expander opens is higher than the level above range
at which it closes again. In other words, when the expander is opened, it stays
open until the signal level crosses the quieter ‘close’ threshold. This is known as
hysteresis and is very useful as it allows instruments to decay more naturally. The
word ‘Threshold’ normally refers to the ‘open’ threshold.
Change in 0utput level
HOLD controls the delay before the signal level starts reducing again, and RELEASE
controls how quickly the level then reduces. Note that the RELEASE interacts with
the RANGE, which determines the depth of gain reduction. Time
The attack time (the time taken for the Expander/Gate to ‘recover’ once the signal
level is above the ‘deactivate’ threshold) is normally set to 1.5ms per 40dB. Press
the FAST ATTACK switch to introduce a faster attack time of 100µs per 40dB. This is Hold
useful when gating signals with a steep rising edge, such as drums.
Attack
Release
The Process Order display at the base of the plug-in window indicates the side-chain assignments. Both EQ and filter
sections can be assigned to the side-chain together, in which case the EQ precedes the filter.
To listen to the signal feeding the side-chain, press the S/C LISTEN button in the Output section to route the side-chain
signal to the channel output.
Remember to cancel the S/C LISTEN button once you have finished auditioning the side-chain!
Please note: EQ > DYNAMICS > FILTER is not possible. This is because on SSL's analogue consoles, if the
'filter to input' switch is not activated, the filter will always occur directly after the EQ section.
Output Section
The Output section allows you to ensure that the signal retains a good level after all the signal processing.
The signal level is shown above the knob.
S/CHAIN LISTEN routes the side-chain directly to the output, so you can monitor the side-chain signal.
Peak Meter
Allows you to observe the exact level at any point in RMS Meter
time. Stereo instances display the highest level of the The thinner bar to the right shows the
left and right signals. RMS (average) level over time.
When clipping occurs, the meter will turn red. It will remain red until the meter is reset by clicking on the meter.
Presets
Factory presets are included in the plug-in installation, installed in the following locations:
Mac: Library/Application Support/Solid State Logic/SSLNative/Presets/Channelstrip
Windows 64-bit: C:\ProgramData\Solid State Logic\SSL Native\Presets\Channelstrip
Switching between presets can be achieved by clicking the left/right arrows in the
preset management section of the plug-in GUI, and by clicking on the preset name
which will open the preset management display.
A-B Comparisons
The A B buttons at the base of the screen allows you to load two independent settings
and compare them quickly. When the plug-in is opened, setting A is selected by default.
Clicking the A or B button will switch between setting A and setting B.
UNDO and REDO functions allow undo and redo of changes made to the plug-in parameters.
Introduction
The Bus Compressor plug-in is based on the legendary centre section stereo bus compressor found on SSL's large format
analogue consoles. It provides high quality stereo compression for critical control over the dynamic range of audio
signals.
The compressor can be run in mono or stereo modes and can be used for practically any application that requires superior
compression. For example, place it over a stereo mix to ‘glue’ the mix together whilst still maintaining a big sound, or use
it on drum overheads or whole drum kits for very effective control of drum dynamics.
Interface Techniques
The interface techniques for the Stereo Bus Compressor are identical to those for the Channel Strip.
Automation
Automation support for the Stereo Bus Compressor is the same as for the Channel Strip.
Control Parameters
Below is a description of the Stereo Bus Compressor’s parameters.
RELEASE
Controls how quickly
ATTACK level returns to normal.
Controls response time Choose between
when Threshold is 0.1, 0.3, 0.6, or RATIO
crossed. 1.2 seconds, or Auto. Controls the degree
Choose between: Auto: release time of compression.
COMPRESSION METER 0.1, 0.3, 1, 3, 10 and is dependant upon Choose between:
Shows gain reduction in dB. 30ms. duration of signal peak. 2:1, 4:1 and 20:1.
S/C HPF
High-pass filter
applied to the
compressor side-
chain.
MIX CONTROL
Controls the
blend of 'dry'
uncompressed
signal and 'Wet'
compressed
signal.
A-B Comparisons
The A B buttons at the base of the screen allows you to load two independent settings
and compare them quickly. When the plug-in is opened, setting A is selected by default.
Clicking the A or B button will switch between setting A and setting B.
UNDO and REDO functions allow undo and redo of changes made to the plug-in parameters.
Introduction
The illustration below gives an overview of some of the X-Comp features which are described in full over the following
pages.
Above the main plug-in Interactive compressor I/O Difference display displays Listen button for
window is a set of controls display with draggable nodes the signal’s dynamic range. auditioning bleed signal
that are specific to your
host application.
Please refer to your host
application’s user guide
for guidelines on using
these controls
GR displays gain
reduction
GR History displays gain Switch between main and Bleed section allows LF and HF Output level metering
reduction over last second alternative plug-in settings bands to bypass the compressor. and control
Display includes draggable nodes.
Interface Techniques
The basic interface techniques for the X-Comp are largely identical to those for the Channel Strip. In addition to these
basic techniques, the following are also available:
Threshold and Ratio values can also be controlled directly within the compressor display with draggable nodes. Move
the nodes to control the Threshold and the Ratio.
The bleed bands can be adjusted in a similar way.
Automation
Automation support for X-Comp is the same as for the Channel Strip.
Knee
Output level
The Knee controls how focussed the threshold level is: Gradient is
shallow or flat
- With a hard knee (knob at minimum) the compressor’s parameters all come
for limiters
into force at precisely the point at which the threshold is crossed. In the
Compression Law diagram, this is shown by a sharp change in gradient at the
threshold. Attack
Operational tip. A hard knee allows for greater precision, but can sound more
obvious. A soft knee generally provides a more transparent result.
Max GR
The Max GR (Maximum Gain Reduction) allows you to set a limit on how much gain reduction can be introduced,
replicating the performance of older optical compressors (as shown in the upper diagram above). With a Max GR of
20dB, for example, any signals that would normally be reduced by more than 20dB will only be attenuated by 20dB. By
only compressing the middle of the dynamic range you can exert some general dynamic control whilst still preserving
the transients for impact.
Input Output
The frequencies and slope of the bleed-band filters are indicated either side of the graph; LF to the left and HF to the right.
The slope is measured as a percentage.
The bleed bands can be adjusted by clicking on the values in the displays and dragging up and right in the usual way.
Alternatively, they can be adjusted directly using the nodes in the display – brown for low frequencies and blue for high
frequencies. Move the nodes horizontally to control frequency and vertically to control percentage
Compressor Values
The table below lists the ranges of all of X-Comp’s parameters:
Peak Meter
Allows you to observe the exact level at any point in RMS Meter
time. Stereo instances display the highest level of the The thinner bar to the right shows the
left and right signals. RMS (average) level over time.
When clipping occurs, the meter will turn red. It will remain red until the meter is reset by clicking on the meter.
Turn the GAIN knob in the input section to control the level of the incoming audio signal. The post-gain
signal level is shown above.
Turn the GAIN knob in the output section to ensure that the signal retains a good signal level post-processing.
The output signal level is shown above the knob.
Bleed Graph
The bleed graph indicates the frequencies that are
bypassing the compressor, as described on the previous page.
In the graphic to the right, the input signal displays a mixture of very loud and very quiet, indicating
that the signal has a large dynamic range that is changing very quickly. The output signal indicates
that this dynamic range has been made smaller, with the loud bars being reduced in level and the
quiet bars being increased. –∞
GR History
The GR History (Gain Reduction History) meter shows the current gain reduction and how it has
fluctuating.
The thicker line towards the left of the display shows how much gain reduction is currently being used,
while the thinner lines to its right show how this has been changing over the past second.
A-B Comparisons
The A B buttons at the base of the screen allows you to load two independent settings
and compare them quickly. When the plug-in is opened, setting A is selected by default.
Clicking the A or B button will switch between setting A and setting B.
UNDO and REDO functions allow undo and redo of changes made to the plug-in parameters.
Introduction
X-EQ 2 is a highly configurable 24-band EQ plug-in, featuring unique 'anti-cramping' algorithms for an unparalleled open
and transparent sound. It’s the ultimate EQ toolkit, with up to 17 different EQ types and filter shapes available derived
from a wide variety of vintage analogue equipment. Each of the 24 bands is switchable between bell, shelf, low pass
or high pass filter types, with both classic and customisable shapes available for each band. The illustration below
introduces the X-EQ 2 plug-in features, described in full over the following pages.
Above the main plug-in Input level metering Bypass for smooth in/out Built-in FFT Output level
window is a set of and control comparisons analyser post-EQ metering and control
controls that are specific
to your host application.
Please refer to your host
application’s user guide
for guidelines on using
these controls
Interactive EQ display
with draggable nodes
Switch between
main and alternative
plug-in settings
The node can be dragged around the graphical interface to adjust the gain and frequency of the band, whilst using the
mouse-wheel to adjust the band Q factor. To lock the frequency of a node, hold down shift; this is useful for changing
the gain or Q factor / roll-off of a node, without changing the frequency.
Alternatively, you can use the Puck Control.
Gain Control
(becomes roll-off on
HPF/LPF curves) Enhanced Tooltip
Double-clicking on the frequency, Q, or gain section of the Puck control will open up the numerical
entry for that parameter. The gain / roll off values can be entered on the left side of the EQ graph,
the frequency and Q on the bottom of the EQ graph.
Clicking on the node will also open an Enhanced Tooltip for that frequency band. From the Enhanced
Tooltip, the EQ shape and filter type can be selected, as well as the band solo enable and spatial
processing options. The ‘power’ icon is a band bypass for quick A/B, whilst the cross (or ‘X’ icon)
deletes the band.
When Left/Right mode is enabled (the default mode when initialising the plug-in) the following icons are displayed in
the Enhanced Tooltip:
This option applies the EQ band to both left and right (a standard stereo EQ).
This option applies the EQ band to both mid and side (a standard stereo EQ).
When a spatial processing option is selected (either left-only, right-only, or mid-only or side-only), the icon is displayed
on the node for easy reference.
Mid/Side, otherwise known as Sum and Difference, is a two channel format. Rather than each of the two
channels carrying either the Left or Right side of a stereo signal, the Mid (or Sum) carries all signals
common to both left and right channels (typically elements panned centre), and the Side (or Difference)
carries all signals ‘on the sides’. The Mid/Side format can be derived from a stereo left and right signal,
and can be turned back into a stereo signal. This allows for some really cool spatial processing techniques
when processing the Mid and Side signals differently, then recombining them back to a stereo signal - this
has been a crucial mastering technique for years!
Band solos
Each frequency band can be solo’d; this allows you to listen to only the frequencies that band is affecting and the effect
of the EQ curve on those frequencies. This is particularly useful for honing in on problem frequencies, and can be used
to great effect when combined with the Mid/Side mode!
Tip: Pressing the Shift Key whilst dragging a node will lock it to that frequency, allowing for precision
notching of resonant frequencies.
• The coloured shade areas show the impact of each individual node.
• The white line across the graph shows the resultant EQ curve from all frequency band nodes.
• With the Analyse switch enabled, an FFT display of the post-EQ signal is shown within the EQ graph. If a band solo is
enabled, you will see this on the analyser.
There are two additional responses that can be displayed on the EQ graph; the phase response and the step response.
These are shown by clicking on the icon at the bottom of the plug-in window.
All frequency bands cause some degree of phase shift; this does not audibly affect a mono audio source in isolation, but
does affect the way two sources with similar audio content combine (two drum overhead microphones, for example).
Whilst ‘phase issues’ are something to be wary of, slight phase adjustments at some frequencies can be desirable,
causing cancellation at unwanted frequencies (room mode build-up, for example), and rigidity at desirable frequencies
(the root notes of a bass instrument, for example).
The phase response graph can be used as a technical aid on critical EQ'ing tasks during mixing or mastering.
Different filters types have different effects on the step response/transients of your audio signal; some filters slow
transients down (something called ‘transient slewing’), others are more transparent. Transient slewing is not necessarily
a bad thing, it can have desirable effects; many vintage analogue audio circuits cause a great deal of transient slewing
that was undoubtedly part of the sound!
The step response graph can help you make better informed mix decisions about which filter type is better for the task
in hand.
EQ Parameter Values
High-pass filter 0 - 48dB/ octave, 6dB steps 20Hz - 20kHz 0.3 - 10.3
Low-pass filter 0 - 48dB/ octave, 6dB steps 20Hz - 20kHz 0.3 - 10.3
Low shelf +-20dB 20Hz - 20kHz 0.3 - 10.3
High shelf +-20dB 20Hz - 20kHz 0.3 - 10.3
Parametric Bell +-20dB 20Hz - 20kHz 0.3 - 10.3
• When clipping occurs, the meter will turn red. It will remain red until the meter is reset by clicking on the meter.
• Turn the GAIN knob in the input section to control the level of the incoming audio signal. The post-gain signal level
is shown on the input meter,
• Turn the GAIN knob in the output section to ensure that the signal retains a good signal level post-processing. The
output signal level is shown on the output meter.
Automation
Every plug-in parameter can be automated in host applications that support automation. The method for recording and
editing automation varies from host to host. For specific instructions on using automation within the host, consult the
host application's documentation.
Plug-in Bypass
The In/Out switch located above the Input section provides a plug-in bypass. This allows for smooth In/Out comparisons
by triggering the host application’s Bypass function.
The button must be ‘lit’ for the plug-in to be in circuit.
Presets
Factory presets are included in the plug-in installation, installed in the following locations:
Mac: Library/Application Support/Solid State Logic/SSLNative/Presets/X-EQ2
Windows 64-bit: C:\ProgramData\Solid State Logic\SSL Native\Presets\X-EQ2
Switching between presets can be achieved by clicking the left/right arrows in the preset
management section of the plug-in GUI, and by clicking on the preset name which will
open the preset management display.
A-B Comparisons
The A B buttons at the base of the screen allows you to load two independent settings and compare them quickly. When
the plug-in is opened, setting A is selected by default. Clicking the A or B button will switch between setting A and setting
B.
UNDO and REDO functions allow undo and redo of changes made to the plug-in
parameters.
Serial
Parallel
The parallel EQ is modelled on the old ‘Parallel Passive’ EQ units and the different style of band interaction means that
the EQ performs quite differently. See the EQ History section for more information.
What is anti-cramping?
You will have noticed the phrase 'anti-cramping' in our references to X-EQ 2 and may be wondering "what is this and why
should I care?" Well let us explain...
All digital audio systems have a finite frequency bandwidth dependant on the sample rate. The upper limit of this
frequency bandwidth is known as the Nyquist frequency.
At the most commonly used DAW session sample rates of 44.1 kHz and 48 kHz, the upper limit of the frequency bandwidth
is close to audible range of the human ear (22.05kHz /24kHz). In a traditional digital EQ design, as you approach this
upper limit a phenomenon called ‘cramping’ or 'wrapping' occurs, whereby bell-shaped boosts or cuts at the upper-end
of the frequency spectrum (around 15 - 20 kHz) begin to suffer and become affected. The effect of EQ cramping may not
always be immediately obvious but listen more carefully and you will hear distortion, ringing and other artefacts that
make an EQ sound unpleasant e.g. lack of smoothness in the top end. This can be particularly frustrating when trying
to get some nice 'air' out of a vocal to name one real-world example. Or perhaps, you are trying to add some overall
brightness/air to the whole mix.
The solution that most manufacturers use to get around this issue is to implement ‘oversampling’; the audio entering
the digital processor (in this case, the plug-in) is upsampled to a higher sample rate, processing is applied without the
audible cramping issue as the upper bandwidth limit has moved further away from the audible frequency range, and then
downsampled back to the original sample rate. Whilst this works, it’s often at the expense of CPU usage for the user in
their DAW and therefore ‘costs’ a lot more processing power.
X-EQ 2 uses SSL's own proprietary 'anti-cramping' algorithms to prevent the unpleasant artefacts of EQ cramping,
particularly those which cause asymmetrical response curves for bell boosts at around 15 -20kHz. Unlike standard digital
anti-cramping, SSL’s proprietary solution achieves this without any additional CPU cost. How do we do this? Well, you'd
have to ask our DSP experts and I think they are quite keen to keep this one a secret...
P – 3dB Classical definition where Q is measured 3dB below peak for boost or 3dB above peak for cut.
0 – 3dB Definition used for bells in some US products, the bandwidth for Q calculation is measured 3dB below
0dB line for cut or 3dB above for boost. Q in both definitions correct only for +6dB boost/cut.
P/2 New ‘musical’ definition based on the bandwidth measurement in the middle of a bell filter, between
peak and 0dB line.
Normalisation 12 equalisers are normalised to have exactly the same bell shape for +6dB boost.
Classic Symmetrical (Classic Sym)
The most popular parametric EQ shape used in various mixing
consoles and outboard gear. Almost constant Q characteristic.
P – 3dB
Proportional 1 (Prop 1)
Proportional equalisers are recognised as being more ‘musical’ than
Constant Q. The bells are wider below +6dB and narrower above +6dB
(or –6dB for cut).
Proportional 2 (Prop 2)
Like Proportional 1 but with larger changes below and above +6dB
peak (or –6dB for cut).
P –3dB
Proportional 3 (Prop 3)
Like Proportional 1 but with extra widened bells between 0 and +3dB
(or –3dB for cut).
P – 3dB
Parallel
Note this is accessed via the Parallel switch, not the drop-down menu.
Recreates the passive LC parallel equaliser with all its advantages
(sound) and disadvantages (band interaction, asymmetry). As found
in graphic equalisers.
3dB boost, 0 – 3dB (x2) cut
Shelving Bands
Low and High shelves
Q value is used to control overshoot characteristic. On the left is the
Low shelf filter with a low Q value, on the right the High shelf filter
exhibits overshoot with a large Q value.
Note that there are no alternative shapes for shelving bands
Filter Shapes
Critical
‘Critical Damped’ filters simulate a chain of passive analogue RC (for
high cut) and CR (for low cut) stages fixing a behaviour similar to a
series of RC elements in vintage analogue equipment.
Bessel
Linear phase behaviour leads to no overshoot or ringing resulting from
a sudden transition between signal levels. The drawback is a sluggish
roll-off rate.
Butterworth
Characterised by having a maximally flat magnitude response, i.e. no
amplitude ripple in the passband.
Chebychev
Characterised by having an equiripple magnitude response, meaning
the magnitude increases and decreases regularly from DC to the cutoff
frequency.
EQ History
An Audio Engineer’s Best Friend
The equaliser is the oldest and the most popular sound processing tool. From the earliest days, its main function has
been to correct or enhance sound by cutting or boosting certain frequency ranges. Engineers have developed countless
equalisers for over 50 years, some of which became legendary and were considered bench mark equipment in studios.
The most popular type of EQ in recording and post-production studios is the parametric equaliser or PEQ. It offers maximal
flexibility due to direct access to all relevant filter parameters. Properly used, the PEQ is a very powerful tool and the best
friend of every sound engineer in the battle for perfect sound. If misused, it can be the greatest enemy of any recording.
Parallel Passive EQ
Parallel EQ exhibits quite different sonic properties to the familiar serial parametric EQ. We are generally used to hearing
the effect of one EQ band superimposed on another, as opposed to the band interaction inherent to a parallel EQ.
Because the bands are placed in a parallel configuration, phase cancellations and re-enforcements happen that are not
always obvious when first encountered.
Passive EQ is something that is found in old equaliser units and is generally known for its transparent and natural sound,
but has some problems associated with it. However, in the digital domain these shortcomings do not have such an
influence.
A passive EQ does not have any gain elements, but can still have controls to seemingly boost frequencies as well as cut.
What actually happens is that the entire signal is cut by an amount, but the frequencies that are apparently ‘boosted’ are
simply not cut as much. Therefore the unit must attenuate either the input, the output, or both to allow enough headroom.
Unfortunately, in the analogue domain, a 20dB reduction in signal level produces a 20dB increase in the noise floor.
Luckily in the digital domain, with a 64-bit floating point DSP, these issues do not remain.
In X-EQ 2 – when the ‘parallel’ button is engaged – you are presented with a parallel passive EQ model that the original
designers of these devices could only have dreamed of. The noise floor can be disregarded due to the huge resolution
available in SSL Native plug-ins.
You may find yourself entering this mode more and more as you become familiar with the sonic signature. Larger gain
changes are possible without colouration, and boost starts to become something that is usable to a significant degree
in a digital EQ!
Parallel EQ does exhibit asymmetry in its boost and cut characteristics. But this is not such a bad thing as most engineers
would agree that boost is best done with low (wide) Q values and cut with a higher (narrower) Q.
Introduction
Vocalstrip is a collection of refined tools for superior vocal processing. The illustration below introduces the main
Vocalstrip plug-in features which are described in full over the following pages.
Compander
Displays
Preset Processing
Management Order
Interface Overview
The basic interface techniques for the Vocalstrip are largely identical to those for the Channel Strip.
Plug-in Bypass
The switch located above the Input section provides a plug-in bypass. This allows for smooth In/Out
comparisons by triggering the host application’s Bypass function.
The button must be ‘lit’ for the plug-in to be in circuit.
Presets
Factory presets are included in the plug-in installation, installed in the following locations:
Mac: Library/Application Support/Solid State Logic/SSLNative/Presets/Vocalstrip2
Windows 64-bit: C:\ProgramData\Solid State Logic\SSL Native\Presets\Vocalstrip2
Switching between presets can be achieved by clicking the left/right arrows in the
preset management section of the plug-in GUI, and by clicking on the preset name
which will open the preset management display.
A-B Comparisons
The A B buttons at the base of the screen allows you to load two independent settings
and compare them quickly. When the plug-in is opened, setting A is selected by default.
Clicking the A or B button will switch between setting A and setting B.
UNDO and REDO functions allow undo and redo of changes made to the plug-in parameters.
Automation
Automation support for Vocalstrip is the same as for the Channel Strip.
• When clipping occurs, the meter will turn red. It will remain red until the meter is reset by clicking on the meter.
• Turn the GAIN knob in the input section to control the level of the incoming audio signal. The post-gain signal level
is shown on the input meter,
• Turn the GAIN knob in the output section to ensure that the signal retains a good signal level post-processing. The
output signal level is shown on the output meter.
Vocalstrip Modules
De-esser
Sibilance is often found in vocal recordings whereby the ‘S’ consonants are too pronounced. The Vocalstrip de-esser can
detect and remove sibilance.
• Switch the de-esser on by clicking on the power button in its top left-hand corner.
• Adjust the Threshold control to set the detection level. You will see the gain reduction meter
above the de-Esser controls change depending on where this is set.
• Adjust the Frequency knob to change the frequency range that triggers the de-esser.
• To listen to the sibilance that is being removed, press the Listen button. This can be
particularly helpful in ensuring that the threshold is correctly set.
• Turn on the Lookahead to detect incoming sibilance before it reaches the de-esser block -
this increases latency, but results in a smoother sound. Ensure Lookahead is turned
off if monitoring live inputs through Vocalstrip.
• The de-esser runs in SPLIT band mode by default (gain reduction is applied to the frequency
range that triggers it), but can be switched to BROADBAND mode (gain reduction is
applied to the full frequency range when triggered).
• Switch the de-ploser on by clicking on the power button in its top left-hand corner.
• Adjust the Threshold control to set the detection level. You will see the gain reduction meter
above the de-ploser controls change depending on where this is set.
• Adjust the Frequency knob to change the frequency range that triggers the de-ploser.
• To listen to the plosives being removed, press the Listen button. This can be
particularly helpful in ensuring that the threshold is correctly set.
• Turn on the Lookahead to detect incoming plosives before they reach the de-ploser block -
this increases latency, but results in a smoother sound. Ensure Lookahead is turned
off if monitoring live inputs through Vocalstrip.
• The de-ploser runs in SPLIT band mode by default (gain reduction is applied to the frequency
range that triggers it), but can be switched to BROADBAND mode (gain reduction is
applied to the full frequency range when triggered).
Equaliser
Switch the EQ on by clicking on the power button. Three popular EQ types are provided to allow you to cut low frequencies,
find and remove resonances, and shape the top end of the signal. Switch each band on by clicking on the corresponding
power switch.
• The high pass filter operates in the range of 30Hz to 300Hz , with a slight
boost around its cut-off frequency.
• The notch filter operates in the range 200Hz to 10kHz and offers 12dB of
boost and 36dB of attenuation with a high Q value.
• The high band EQ offers 12dB of boost/attenuation over the range of 1kHz
to 20kHz with a low Q value.
EQ Display
When any controls within the EQ section are moved, the display in the top right-hand quarter of the window becomes
an EQ graph:
• The line across the graph shows the frequency response of the current EQ
settings.
• The shaded area represents the impact of each of the three EQ bands.
• A display of the signal’s frequency response is shown in green. The FFT signal
displayed can be switched between the plug-in input and output and
can be switched off, all using the FFT buttons above the graph.
Expander
First the signal enters a fixed ratio downwards expander, designed to reduce
room ambience, spill or breath noise that is often brought up by the subsequent
compression.
To set the level at which the level reduction is introduced, turn the Exp Thresh
(Expander Threshold). Turn the threshold to 0dB to stop the expander affecting
the input signal. The amount of level reduction introduced is shown in the green
bar above the Exp Thresh knob.
Compressor
The compressor offers fully variable Ratio, Thresh (threshold), Attack, Release and Makeup controls, along with a choice
of Hard or Soft Knee.
The output stage, which is sourced after the make-up gain, features an optional Drive feature that introduces harmonic
characteristics to the signal. Its intensity is increased with the Makeup gain. If you use a lot of make-up gain to drive the
circuit, the level can be reduced again using the Output level control. The amount of level reduction introduced is shown
in the red bar above the compressor controls.
Compander Display
When any controls within the Compander section are moved, the display in the top of the window
shows two different graphs:
• The left-hand graph is a gain-law display, showing the relationship between input and output
levels.
• The right-hand graph is an I/O difference display, showing how often each level occurs within the
input and output signals. The input is shown on the left, and the output on the right.
• The vertical scale is amplitude, with 0dB at the top and –∞ at the bottom. The length of each line
protruding from the centre represents the number of incidents of that amplitude over a period of
seconds.
• To the right of the I/O difference display are the gain reduction meters; an orange meter for the
compressor, a green meter for the expander.
Processing Order
The processing order is controlled at the base of the plug-in window.
Click on the left or right arrows to move a module earlier or later in the plug-in signal path.
Introduction
Drumstrip is a one-stop solution for superior drum processing, providing tailor-made tools for fixing and polishing your
drum sounds. The illustration below introduces its features which are described in full over the following pages.
Gate activity metering Gate for removing spill Transient Shaper for increasing/reducing
Above the main plug-in section transient attack
window is a set of controls
that are specific to your
host application.
Please refer to your host
application’s user guide
for guidelines on using
these controls
Internal bypass
for smooth in/out
comparisons
HF and LF Enhancers, for enriching Processing Compressor with Listen Output metering
high and low frequencies order controls Mic characteristics and control
Interface Overview
The basic interface techniques for the Drumstrip are largely identical to those for the Channel Strip.
Plug-in Bypass
The power switch located above the Input section provides an internal plug-in bypass. This allows for
smoother In/Out comparisons by avoiding the latency issues associated with the host application’s
Bypass function. The button must be ‘lit’ for the plug-in to be in circuit.
Automation
Automation support for Drumstrip is the same as for the Channel Strip.
A-B Comparisons
The A B buttons at the base of the screen allows you to load two independent settings
and compare them quickly. When the plug-in is opened, setting A is selected by default.
Clicking the A or B button will switch between setting A and setting B.
UNDO and REDO functions allow undo and redo of changes made to the plug-in parameters.
Automation
Automation support for Vocalstrip 2 is the same as for the Channel Strip.
Peak Meter
Allows you to observe the exact level at any point in RMS Meter
time. Stereo instances display the highest level of the The thinner bar to the right shows the
left and right signals. RMS (average) level over time.
When clipping occurs, the meter will turn red. It will remain red until the meter is reset by clicking on the meter.
Turn the GAIN knob in the input section to control the level of the incoming audio signal. The post-gain
signal level is shown above.
Turn the GAIN knob in the output section to ensure that the signal retains a good signal level post-
processing. The output signal level is shown above the knob.
Transient Shaper
The Transient Shaper allows you to add attack to the start of a drum hit by increasing the amplitude of the attack portion
of the signal whilst leaving the decay unchanged. The right hand waveform is a processed version of the one on the left.
It has been passed through the transient shaper where the amplitude of the attack portion
has been increased.
Switch the Shaper on by clicking on the ‘power’ button.
The meter gives visual feedback on how much attack is
being added using the Gain and Amount controls. Gain
controls the detection level of the controller signal, and
should be set so that only the transients you want to shape
are detected. If this is set too low then the Shaper will do
nothing; if it is set too high then the Shaper will detect too
many transients, resulting in an exaggerated process, and
the attack appearing too long. The default setting of 0dB
should be a good starting point.
Amount controls the amount of the processed signal added to the unprocessed signal. This process can increase the
peak level of a signal significantly, so watch the output meter carefully.
Speed controls the length of time the added attack takes to fall back down to
the normal signal level once it has reached the top of the attack phase. Turn the
knob clockwise for a slower speed, and longer transient.
The Invert switch inverts the processed signal so that it is subtracted from the
Speed=0 Speed=1
unprocessed signal. This has the effect of softening the attack, resulting in more
body in the drum sound.
The Listen switch allows you to listen to the processed signal, to assist in the set up process.
When the Invert and Listen buttons are both pressed, the signal will not be inverted.
HF and LF Enhancers
The HF and LF enhancers respectively enrich the high and low frequencies of the input signal.
Whereas a standard EQ simply raises the level of certain frequencies, the Enhancer adds a
combination of 2nd and 3rd harmonics to those frequencies, producing a more pleasing effect.
Switch each Enhancer on by clicking on the power button in its top left-hand corner. No effect is
heard until an Enhancer’s Drive and Amount are turned up.
HF Cutoff sets the frequency above which the HF Enhancer generates harmonics. It ranges from
2kHz up to 20kHz – To add air or sparkle to a signal, push this frequency towards the higher end
of the range. To give more presence to a signal, use the lower end of the range. Note that the effect is barely audible in
the 15kHz to 20kHz range.
LF Turnover sets the frequency below which the LF Enhancer generates harmonics. It ranges from 20Hz up to 250Hz. The
LF Enhancer is great for adding depth and weight to kick drums, snare or toms.
The Listen Mic Compressor features very quick fixed time constants. This means it is easily capable of
producing distortion on low frequency material.
Processing Order
The five processing blocks in Drumstrip can be configured in any order, as defined by the Process Order blocks at the
base of the plug-in window.
To move a module
within the order
press either the left
arrow or the right arrow.
By default the gate is first in the chain so that it is able to act on the full dynamic range of the signal before the signal is
processed by the Listen Mic Compressor.
Introduction
The illustration below gives an overview of some of the X-ValveComp features which are described in full over the
following pages.
Internal bypass
for smooth in/out
comparisons
Valve switch for Compressor Mix level for parallel Makeup Output metering
activating the valve Time compression gain level and control
emulation Constants control
Interface Overview
The basic interface techniques for the X-ValveComp are largely identical to those for the Channel Strip. In addition to
these basic techniques, the following are also available:
To control Input and Output levels, click and drag upwards or downwards.
To enter a precise value for any parameter, double-click on its value display, enter a value on your computer keyboard
and press the Return key.
Threshold and Ratio values can also be controlled directly within the Compression Law display. Move the left hand node
along its diagonal line to control the Threshold, and move the right hand node up and down to control the ratio.
Automation
Automation support for X-ValveComp is the same as for the Channel Strip.
Presets
Factory presets are included in the plug-in installation, installed in the following locations:
Mac: Library/Application Support/Solid State Logic/SSLNative/Presets/XValveComp
Windows 64-bit: C:\ProgramData\Solid State Logic\SSL Native\Presets\XValveComp
Switching between presets can be achieved by clicking the left/right arrows in the
preset management section of the plug-in GUI, and by clicking on the preset name
which will open the preset management display.
A-B Comparisons
The A B buttons at the base of the screen allows you to load two independent settings
and compare them quickly. When the plug-in is opened, setting A is selected by default.
Clicking the A or B button will switch between setting A and setting B.
UNDO and REDO functions allow undo and redo of changes made to the plug-in parameters.
Peak Meter
Allows you to observe the exact level at any point in RMS Meter
time. Stereo instances display the highest level of the The thinner bar to the right shows the
left and right signals. RMS (average) level over time.
When clipping occurs, the meter will turn red. It will remain red until the meter is reset by clicking on the meter.
Turn the GAIN knob in the input section to control the level of the incoming audio signal. The post-gain
signal level is shown above.
Turn the GAIN knob in the output section to ensure that the signal retains a good signal level post-processing.
The output signal level is shown above the knob.
Valve In
Valve in introduces a tube compressor emulation for saturating the signal by adding harmonics.
Side-chain
Hi- and Lo-pass filters are available for the Compressor side-chain. To switch the filters into the compressor
side chain, switch the S/Chain in button on. To listen to the Compressor side-chain, click on the Listen
button. Adjusting the filter controls will change the signal that triggers the gain reduction; removing low-end
frequencies from the side-chain can reduce unwanted pumping effects, for example.
Mix
Dry/Wet option for parallel compression.
Auto Gain
You can allow the plug-in to set the make-up gain automatically by clicking the Auto button (towards the
bottom of the screen).
Introduction
The illustration below gives an overview of some of the X-Saturator features which are described in full over the following
pages.
Internal bypass
for smooth in/out
comparisons
Depth determines the level Harmonics control for Drive control for adjusting Shape control affects Output metering
of the harmonics injected adding or subtracting the amount of overtones the balance of and control
back to the signal 2nd order harmonics added to the signal harmonic content
Interface Overview
The basic interface techniques for the X-Saturator are largely identical to those for the Channel Strip.
Automation
Automation support for X-Saturator is the same as for the Channel Strip.
Plug-in Bypass
The power switch located above the Input section provides an internal plug-in bypass. This allows for
smoother In/Out comparisons by avoiding any latency issues associated with the host application’s
Bypass function. The button must be ‘lit’ for the saturator to be in circuit (as shown).
Peak Meter
Allows you to observe the exact level at any point in RMS Meter
time. Stereo instances display the highest level of the The thinner bar to the right shows the
left and right signals. RMS (average) level over time.
When clipping occurs, the meter will turn red. It will remain red until the meter is reset by clicking on the meter.
Turn the GAIN knob in the input section to control the level of the incoming audio signal. The post-gain
signal level is shown above.
Turn the GAIN knob in the output section to ensure that the signal retains a good signal level post-
processing. The output signal level is shown above the knob.
Boost +6dB
Will add 6dB of Headroom above your saturation point.
Drive
Dialling in the drive will determine the level of saturation that will be added.
Harmonics
By adjusting the Harmonics control the distortion characteristics can be shifted from ‘50s valve-style overdrive (if the
control is turned fully anti-clockwise for 2nd harmonic distortion) to ‘70s transistor-style grit (if the control is turned fully
clockwise for 3rd harmonic distortion).
Depth / Shape
Depth and Shape control the number of harmonics injected back into the signal, enriching the sound. Depth sets the
amplitude of the added harmonics; at 0% no harmonics are introduced to the input signal, at 100% the full amount of
generated harmonic content is introduced to the input signal. Shape will add overtones in the high frequency spectrum;
-50% is subtle and smooth, +50 will sound more aggressive.
A-B Comparisons
The A B buttons at the base of the screen allows you to load two independent settings
and compare them quickly. When the plug-in is opened, setting A is selected by default.
Clicking the A or B button will switch between setting A and setting B.
UNDO and REDO functions allow undo and redo of changes made to the plug-in parameters.
Introduction
The illustration below gives an overview of some of the X-Phase features which are described in full over the following
pages.
Above the main plug-in Delay control for Left/Right switch to independently edit the
window is a set of controls adjusting the delay left and right signal (stereo version only)
that are specific to your
host application.
Please refer to your host
application’s user guide
for guidelines on using
these controls
Internal bypass
for smooth in/out
comparisons
Interface Overview
The basic interface techniques for the X-Phase are largely identical to those for the Channel Strip.
Automation
Automation support for X-Phase is the same as for the Channel Strip.
Plug-in Bypass
The In/Out switch located above the Input section provides a plug-in bypass. This allows for
smooth In/Out comparisons by triggering the host application’s Bypass function. The button
must be ‘lit’ for the plug-in to be in circuit.
Peak Meter
Allows you to observe the exact level at any point in RMS Meter
time. There are two meters in stereo and one in mono. The thinner bar to the right shows the
RMS (average) level over time.
When clipping occurs, the meter will turn red. It will remain red until the meter is reset by clicking on the meter.
Delay
Activate or deactivate a delay for the whole signal with the Delay In switch. Use
the dial to set the delay in seconds, milliseconds and samples, meters or feet. The
single +/– switches will give you smaller increments than the double +/– switches.
All-pass In
Activates the All-pass Filter.
All-pass Phase
Inverts the phase of the All-pass Filter.
Frequency
Sets the Frequency of the filter.
Second Order
Switches the filter between first and second order.
Q Factor
Sets the Q of the Filter, thus it will adjust the rate of phase change across the frequency spectrum.
Stereo Version
Stereo versions have an additional L-R-Switch to edit stereo signals.
A-B Comparisons
The A B buttons at the base of the screen allows you to load two independent settings
and compare them quickly. When the plug-in is opened, setting A is selected by default.
Clicking the A or B button will switch between setting A and setting B.
UNDO and REDO functions allow undo and redo of changes made to the plug-in parameters.
This line shows the phase relationship between different frequencies in the signal sent through the X-Desk. SSL products
are known for their “width” and “depth” in the mix, and much of this is down to the phase relationships between different
parts of the signal. You can see that there is a fairly large “flat” section of phase response between 80Hz and 2kHz. When
audio passes through the X-Phase, the phase of these frequencies remains largely unaffected by passing through the
circuit. However, looking to the right hand side of the chart you can see that a frequency of 20kHz is 20˚ out of phase with
a frequency of 1kHz. This will affect how the frequencies in that signal interact by reinforcing and cancelling each other
depending on their phase, which can lead to comb filtering.
You’d be amazed at some of the phase differences that happen when you run a signal through a device, with some
particularly expensive devices having a +/- 70˚ degree phase difference between the low and the high frequencies;
meaning that different parts of the audio signal could be 140˚ out of phase with each other.
This is what X-Phase helps to solve; not only getting your signals in phase with each other, but also the frequencies within
your signal to be in phase with themselves.
Here’s the science…
Introduction
The illustration below gives an overview of the FlexVerb features which are described in full over the following pages.
Early Reflections and Reverb Tail control 6-Band EQ knob and 3-Band Reverb Time Lockable Output level
Reverb Tail selection section. numerical control Multipliers Wet/Dry Mix metering and
and blend. section. Control control.
Interface Overview
The basic interface techniques for the FlexVerb are largely identical to those for the Channel Strip.
Automation
Automation support for FlexVerb is the same as for the Channel Strip.
Plug-in Bypass
The In/Out switch located above the Input section provides a plug-in bypass. This allows for smooth
In/Out comparisons by triggering the host application’s Bypass function. The button must be ‘lit’ for
the plug-in to be in circuit.
A-B Comparisons
The A B buttons at the base of the screen allows you to load two independent settings
and compare them quickly. When the plug-in is opened, setting A is selected by default.
Clicking the A or B button will switch between setting A and setting B.
UNDO and REDO functions allow undo and redo of changes made to the plug-in parameters.
Input signal
HF Multiplier MF Multiplier LF Multiplier to compressor
side-chain
Reverb Tail
There are four reverb types: Room, Hall, Plate and Chamber. Each reverb type has three sizes: Small,
Medium, and Large.
Select the Early Reflection and Reverb Tail type and size by clicking on the drop-down selection boxes.
With the LINK engaged, selecting the Early Reflection Type and Size will allocate the same settings for the
Reverb Tail. With the LINK disengaged, the Reverb Tail Type and Size is independent of the Early Reflections.
The INFINITE switch sets an infinite reverb time value, the KILL switch kills the reverb tail.
Parameter Description
Blend ER & Tail The mix between the early reflections and the reverb tail.
Pre-Delay The delay between the input signal and the reverb.
Modulation Rate & Depth Applies light frequency modulation to the reverb, giving a more
natural sounding reverb tail.
6-Band EQ
Six bands of EQ can be applied to the reverb tail to shape overall reverb tone. This includes two parametric bands, high-
shelf and low shelf bands, as well as high-pass and low-pass filters. The EQ only affects the reverb; it does not affect the
input source signal.
Graphical Display
The graphical display shows:
• The resultant curve of the EQ section that can be adjusted from the
draggable circular nodes.
• The reverb time multiplier crossover points and values that can be
adjusted from the draggable square nodes.
• An FFT analysis with after-glow.
The FFT can be switched off by right/CTRL-clicking the graphical display, and selecting 'off'.
Mix
A Dry/Wet control for applying reverb directly to channel processing, rather than via an aux send.
The lock function excludes the mix control parameter from the preset management system.
The lock function is useful, for example, when switching between multiple presets; set the mix control
to the preferred blend of 'Dry' input signal and 'Wet' reverb signal, then audition different reverb presets
without affecting the mix.
82BDNM01E
SSL® and Solid State Logic® are ® registered trademarks of Solid State Logic. SSL Native™ is a
trademark of Solid State Logic.
All other product names and trademarks are the property of their respective owners and are hereby
acknowledged.
No part of this publication may be reproduced in any form or by any means, whether mechanical or
electronic, without the written permission of Solid State Logic, Oxford, OX5 1RU, England
As research and development is a continual process, Solid State Logic reserves the right to change
the features and specifications described herein without notice or obligation.
Solid State Logic cannot be held responsible for any loss or damage arising directly or indirectly
from any error or omission in this manual.
E&OE