Introduction

Introduction
The coreFX are a collection of several effects plug-ins with a clear user interface that ensures a uniform
operation. There are 9 core FX that can be categorized in two groups: delay-based and dynamic effects, plus
the Volume Former. Inside these groups the effects differ just from the range of the parameter values which
results in different audible effects. The Volume Former is a special case, it cannot be assigned to either of
these two groups.

Each of the plug-ins contains several hands-on presets for daily use. Just try out the presets to get an
idea of how the plug-ins work!

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 Installation

Installation
In principle, installing VST and AU plug-ins is very simple. Associated files are located in a folder that is
known to the sequencer/host software. The exact name can vary.
To use VST plug-ins with Windows, please first ensure that your software has a VST plug-in folder that is
indicated in the system or path settings. OSX always saves VST plug-ins in the "/Library/Audio/Plug-
Ins/VST" folder. AU plug-ins on Mac OS X, on the other hand, are stored in the "/Library/Audio/Plug-
Ins/Components" folder.
If you have not yet used third-party VST/AU plug-ins, you may have to consult your audio software's
documentation about how to integrate plug-ins.
To start the installation, please click the corresponding setup file in the main directory.
During the running process, you will be asked to enter a path for the installation of VST plug-ins. While a
specific installation path is recommended, it may be selected freely; you should, however, enter the above
indicated folder for the VST path in order for your host software to detect it correctly (not required for MAC OS
X).
Most current VST/AU hosts are capable of managing sub-folders via this plug-in path. You can therefore
accept the folder recommended by the installer, create a new folder or use an already existing one.
After successful installation, you can start your host/sequencer software as usual. Depending on the
program, there are different methods for scanning for new plug-ins. Some do this when starting, some only
after you have selected a menu for plug-ins.
During this scan, a VST/AU plug-in is briefly loaded ("instantiated") to request important system parameters.
Please refer to the plug-in host's manual to learn whether further steps are required to add and use a VST/AU
plug-in.

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 Delay based effects

Delay based effects

The three effects coreFX Delay, Flanger and Chorus have the same working principle: A delay line delays
the audio signal for a certain amount of time and the delayed signal is fed back to the delay input. Inside the
feedback line there is a high pass and a low pass filter that restricts the frequency spectrum of the feedback
audio signal. The delay time can be modulated (i. e. periodically changed) by an LFO (low frequency
oscillator). Depending on the delay time, the amount of modulation of this delay time and how much feedback
is given, the result is an Delay (Echo), Flanger or Chorus effect.

Delay
Delay Time: (1) Adjust the delay time with this knob between 1ms and 3 seconds. With Tap (2)you can also
tap in the delay time. Activate Sync (3) if you want to set the delay time in musical time units, the delay time
will be adjusted to the nearest musical time unit depending on the host tempo.
With Freeze (4) on, the delay input is muted and the content of the delay buffer is repeatedly played back.
The playback speed can be changed with the Delay Time control, whereby the value display now shows a
multiplication factor of the original speed instead of the delay time.

Try to automate the delay time for spacey pitch effects!

LFO: The LFO modulation is not essential to obtain a delay effect. It is just an additional parameter to add
small fluctuations of the delay time to simulate the behavior of old tape-delay machines where the tape speed
was not always constant. That's why you can switch off the LFO with the ON/OFF button (7). You can chose
from 3 different wave forms for the LFO and set the frequency (5) and depth (6)of the modulation.
Tap: (8) You can also tap in the LFO frequency with this button.
Feedback: (9) Feedback controls the amount of signal that is sent back to the delay input. The more
feedback is given, the longer the echo effect will last.

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 Delay based effects

High Pass/Low Pass/Resonance: (10) Restrict the frequency spectrum of the wet signal with the filters.
For instance, move the Low Pass fader to the left to filter out the higher frequency parts from the delayed
signal. As the filtering is applied before the feedback, each echo repetition will be a bit more dull and less
brilliant. If you move the High Pass fader to the right, you will cut out the low frequencies from the echo.
Resonance amplifies the effect of the filters, so be careful with high values, too!
Mix: (11) Controls the mix between the unprocessed (dry) and the processed (wet) audio signal. At 0%, only
the dry signal will be heard. For higher values more and more of the processed signal is added in until at 50%
both the wet and dry signals are at full volume. Similarly, for values above 50% the dry signal level decreases
until it reaches zero at 100%.

Flanger
Delay Time: The typical delay time for a flanger effect is between 1ms and 7ms. The short delay times in
combination with high feedback values result in the typical tonal effects of the delay.
LFO: As with the delay, you can chose from 3 different wave forms for the LFO and set the frequency and
depth of the modulation. You will notice that there are deeper modulations possible as the LFO modulation is
crucial for a flanger effect.
Feedback: Feedback controls the amount of signal that is sent back to the delay input. The more feedback is
given, the stronger the "ringing" of the flanger is.
High Pass/Low Pass/Resonance: Although not essential for a flanger effect, the filter can help to shape the
overall sound of the flanger. Also be careful with the resonance value!
Mix: Controls the ratio between the unprocessed and the processed audio signal. For an audible flanger effect
this has to be higher than a typical delay mix.

Chorus
Delay Time: The typical delay time for a chorus effect is between 3.5ms and 14ms. This results in a
perceived doubling of the audio signal.
LFO:There are also 3 different wave forms for the LFO with an high amount of modulation possible. The
modulation of the delay time causes a slightly different pitch of the delayed signal which strengthens the
doubling effect.
Feedback: Feedback controls the amount of signal that is sent back to the delay input. For a chorus effect
this is not really necessary (in fact, a flanger is just a chorus with shorter delay times and feedback). But it is
not forbidden to use feedback anyway.
High Pass/Low Pass/Resonance:Although not essential for a chorus effect, the filter can help to shape the
overall sound of the chorus. This is especially useful to keep the bass unaltered and apply the effect to higher
frequencies only.
Mix: As the chorus effect needs a mixture between the delayed and un-delayed signal, you will achieve the
best results at around 50%.

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 Dynamic effects

Dynamic effects
These five effects have a common working principle, they are all dynamic effects: There is an amplifier with a
variable amount of amplification that is controlled by the level of the input signal. When the level goes over or
under a certain point (the treshold) the amplification changes to obtain the desired effect: So, for instance, you
can realize compression when the signal level is reduced above a threshold, or you get a noise gate by
suppressing audio below the threshold and so on.
A secondary stereo input channel pair can be used to side-chain the dynamic effects (except the Limiter). The
dynamic effect is controlled by another audio signal then to realize classical "pumping" effects where a kick
drum on one track controls compression of another track to make that little extra headroom for the kick.

There are useful presets for each effect, giving you a great starting point for your processing needs.

Common Controls
Waveform Display: (1) The display at the upper right corner of the FX shows the waveform of the processed
signal. The colored line shows the amount of the amplification or attenuation of the signal. Move the mouse
over the waveforms to see the scaling of the display. You can change the scaling of the audio waveform and
the gain reduction graph independently by right clicking on the display.
Peak/RMS: (2) There are two modes of input detection, Peak or RMS. Peak uses the peak level while RMS
uses an average over a short period of time. Peak is therefore faster, while RMS is more accurate. Which
mode is better depends on the audio material.
Attack/Release: (3) These time values specify how quickly the effect kicks in (Attack) or how long it takes
to return to become inactive again (Release).
Soft: (4) The characteristic of a dynamic effect has a typical fold, the "knee" (or two, in case of the 2-Point
Compressor) at the theshold point(s). With the Soft value you can make the fold more or less round to let the
effect sound more natural (this is called "soft knee").
Out Gain: (5) After the variable amplifier has processed the signal according the volume of the input signal, it
may be lower in volume. The Out Gain slider can be used to compensate this.

This is also known as Makeup Gain.

Mix: (6) Adjusts the dry/wet mix between unprocessed and processed audio signal.
Peak meters: (7) The Peak meters show the input and output signals, the CLIP indicator get red when
clipping occurs. Click on it to reset these indicators.
Limiter: (8) To prevent the output signal of the effect from clipping, you can activate an additional Limiter
here. This is the same limiter used as Hard Limiter in the "Limiter" on the next page

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 Dynamic effects

Compressor
The Compressor is used to compress the audio to balance out the differences between louder and softer
signals. If the signal level goes over the treshold, the amplification is reduced and so the dynamic range of the
signal is reduced by this.
Threshold: This sets the response threshold for regulation. For example, -20 dB indicates that the input signal
is only compressed once it reaches -20 dB; below this level, no change will take place (please note that this is
only strictly true with the Soft control at 0).
Ratio: This parameter controls the compression ratio. A value of 10 means that, above the threshold level,
any additional 10dB of input level will result in a 1dB increase in the output level.
Out Gain: As the reduced amplification of the signals over the threshold result in an overall reduction of the
audio level you can compensate it with the Out Gain fader.

Expander
This is the opposite effect to a compressor. If the signal level goes below the threshold, the amplification is
reduced, so low signal becomes even lower, increasing the dynamic range of the signal. You could see this a
softer form of a "Gate" below, where a signal below the threshold is not cut out completely but faded out..
Threshold: Above the threshold the Expander does nothing, below the threshold the Expander reduces the
level according to the set ratio.
Ratio: Controls the expansion ratio. A value of 10 means that, below the threshold level, decreasing the input
level by 1dB will reduce the output level by 10dB.

Limiter
This is a special case of a compressor with a very high ratio to make sure that the signal stays under a certain
level. This is used to raise the overall loudness of a signal without clipping it. While the use of compressors or
expanders is to shape the sound the best limiter is the one you don't notice at all, apart from the desired
perceived loudness increase.
In Gain: The In Gain slider can be used to increase the amplitude of the signal before it hits the limiter.
Ceiling: The limiter prevents the signal from getting over the ceiling level, so this is typically set just below
0dB, like -0.1 dB.
Hard Limiter: Due to the attack time, it takes a couple of milliseconds before the limiter catches up with
sudden changes in the input level. This can lead to short transients exceeding the ceiling level. Activate the
Hard Limiter to add a second limiter into the signal chain that catches the remaining transients and makes
sure that not a single sample exceeds the ceiling level. This hard limiter uses a look-ahead time of 10ms to
observe incoming transients. Due to the look-ahead time, it can start lowering the gain ahead of the transient,
greatly reducing audible artifacts. However, this introduces an additional delay of 10ms to the entire plug-in.
This is usually automatically compensated in the host software and is only relevant if the overall processing
latency must be low, e.g. when processing audio live.

There is no Mix control in the Limiter (plausible, when you think about it)!

Gate
A gate is the corresponding special case of the expander with a very high ratio, so essentially the signal is
entirely suppressed when it falls below the treshold. The complete designation of the effect is "Noise gate"
and this also indicates the most common application of this effect: To filter out unwanted background noises
in the pauses of a vocal or instrumental performance.
Level: The Level controls the threshold, below which the signal will be suppressed.

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 Dynamic effects

Attack/Release: The attack time can be used to control the transients of the output signal: With long attack
times, it takes a while before the gate opens - cutting away the transient of a sound. Similarly, the release time
can be used to control the sustain of a sound, making it more or less percussive.

2-Point Compressor
This is a dual Compressor/Expander offering you an even bigger flexibility in dynamic processing. Unlike the
cFX Compressor, the ratios can also be smaller than 1, making it an expander to increase the dynamic range
of the signal. For instance, you can combine the two sections to a Compressor/Expander combination to
shape drum sounds: Expand the sound at lower levels to remove background noises and compress the loud
hits in one go. Check out the presets!
Ratio: This parameter controls the compression ratio of the point. Values above 1 will apply compression
above the according threshold level, with values below 1 it works as an expander for signal levels below the
treshold level.

Since it is not determined at which point with which threshold compression or expansion is applied,
extreme levels can occur under certain value combinations. We strongly recommend the use of the
Limiter option!

Sidechain controls and filter

A side-chain is a secondary stereo input channel pair that can be used to control the dynamic effect by a
different signal as the one being processed. For instance, to realize the classical "pumping" effects where a
kick drum on one track controls compression of another track to make that little extra headroom for the kick
drum to be really in front.

Your host software has to support side-chaining, i.e. it has to be capable of routing audio from one track
(or bus) into the additional inputs (3/4) of the plug-in on another track. Consult the manual of your host
software if and how you can do this in your host software!
If side-chaining is not possible, you can simulate this effect with the "Volume Former" on the next page

To use a mix between the internal and the side-chain signal, move the Int/Ext: (2) slider towards Ext. With the
High Pass/ Low Pass (3) filter you can use just certain frequency components to control the effect, for
instance lower the low pass filter value to let only the bass components of the side-chain signal control the
dynamic effect. For control purposes you can click the Listen button (1) to listen to the side-chain signal.

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 Volume Former

Volume Former
The Volume Former is an automated volume envelope, that is controlled by a beat-synched or free running
LFO. For the volume control you can alternatively manipulate the volume level (VCA) or use a low pass (LPG)
or high pass gate (HPG). With the Volume Former you can apply a Tremolo effect, transform pads into
pumping chords (for hosts that doesn't support side-chaining) or accentuate rhythmical aspects of a drum
loop, like isolating the kick drum. Just try the presets to get the idea!

Trigger LFO
The Trigger LFO (low frequency oscillator) periodically triggers the envelope. So, for instance, set the time to
1/4 to trigger the envelope every beat.
When Sync(1) is active the trigger frequency is set dependent from the host
tempo and you can chose a Time(2) between 1/1 (whole note or once a
measure with a 4/4 beat) and 1/64 note (a quite fast stutter effect). The
envelope tempo and phase always follows the host tempo and beat position.
When Free(3) is active, you can set a trigger frequency between 0.2 and 25
Hz. This is independent from the host tempo and also from the playback
position, the trigger oscillator is free running, that means, it just runs once the
plug-in is loaded and the actual trigger time is more or less random.

Some hosts may reset the LFO on playback start due to technical
reasons.

The Time value for the Sync and Free modes are independent
parameters. When you switch from either mode to the other and change
the time, the former mode's time is still set when you switch back.

Move the Phase of the LFO with the Shift value (4). You can shift the starting
point of the envelope from -50% to +50% of the selected time.
Manual trigger: For more complex rhythms, you can switch the LFO trigger
off with the Trigger switch and assign a parameter automation
to the Manual Trigger parameter in your host.

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 Volume Former

Envelope

The envelope is a AHD envelope, thus consisting of three phases Attack, Hold and Decay. The Envelope can
be used in Gate or Pump mode (1). In Gate mode the Envelope rises the volume over the attack time, keeps
it at maximum level for the hold time and lowers it over the decay time. In Pump mode the envelope is
mirrored horizontally and works vice versa: starting from maximum level, it fells to zero over the attack time,
stays there for the hold time and rises back to maximum over the decay time.
(2)Adjust the time spans and curve shapes by dragging the circular handles at the envelope. Below the
graphic there's also a numeric readout of the values (3). You set the time spans in percent of the whole
LFO cycle.

You'll notice, that the values don't have to add up to 100%. That means, you can set envelopes that are
longer or shorter than the LFO cycle. By this you can use only the parts of the envelopes or open/close
the gate for just a part of the LFO cycle.

(4) You can see the effect of the Volume Former on the volume directly in the waveform display, that runs
through behind the envelope.

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 Volume Former

Gate

Choose between VCA, LPG and HPG with the gate type switch (1).
VCA: The envelope controls the volume with a Voltage Controlled
Amplifier, so all frequencies are affected equally.
LPG: The envelope controls the volume with a Low Pass Gate. The low
pass gate comes from the "west coast style" synthesizer design and
works differently: The volume reduction is accomplished by a 1-pole low
pass filter where the envelope controls the cutoff frequency (higher
envelope level = higher cutoff frequency = more of the whole signal
coming trough the gate). This results in a more natural sounding volume
reduction, where the higher frequencies are reduced earlier as the lower
ones when the level turns down. (With a departing sound source, the
higher frequencies also disappear first.)
HPG: With the High Pass Gate a high pass filter is used for gating, here
the lower frequencies disappear first.

The Amount slider (2) controls the overall effect of the volume former: When set to 0% it has no effect at all,
at 100% it turns the volume to silence when the envelope level is 0. You can check the input and output levels
with the peak meters (3)
Bypass the whole effect with the Bypass check box (4).

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