Quartet Chorus Ensemble | Reason Studios

Reason 13.1 Operation Manual

DAW | Rack Plugin

Quartet Chorus Ensemble

Quartet Chorus Ensemble is a fabulous sounding chorus device, with four different characteristic chorus/ensemble algorithms. Each of the four algorithms can have their own unique parameter settings - including the Dry/Wet parameter - so you could switch between the algorithms and get the exact result you are looking for.

Quartet is designed to be used mainly as an insert effect, for spicing up individual instrument sounds with nice dense choruses and modulations.

#Panel reference

#Global controls

#Loading and saving patches

Loading and saving patches is done in the same way as with any other internal Reason device, see “Loading patches” and “Saving patches” for details.

Click the Routing selector to select “Stereo” or Dual Mono” from the pop-up menu.

Stereo: With this selected the L+R input signals are mixed before being sent into the stereo effect. This means you can connect a mono input signal and get stereo output signals.

Dual Mono: The L+R input channels are processed independently.

The Width control can be used for setting the stereo width - from mono to nice and wide stereo.

Note that the Width parameter can be set individually for each of the four chorus algorithms.

This controls the mix of the dry and processed signals.

Note that the Dry/Wet parameter can be set individually for each of the four chorus algorithms.

Note that all chorus/ensemble effects require some amount of dry signal to produce the desired effect. Therefore, 100% Wet also includes a certain amount of dry signal.

If you are using Quartet as a send effect you would probably want to have the Dry/Wet knob set to 100%.

The Chorus effect algorithm simulates multiple detuned “copies” of the input signal. The Chorus is basically a delay line with adjustable feedback. The principle is to split the input signal in two, run one signal dry and the other through the delay line, and then sum the two signals.

The picture below shows the basic principle of the chorus:

Here you set the delay time between the dry and processed signals. In practice, this determines where the notches/peaks will appear in the frequency spectrum.

Range: 1.00-30.00 ms

This determines the depth of the LFO modulation, i.e. by how much the delay time should be modulated. If you set this to 0, the delay time will be static (most effective if you add some feedback).

This determines the frequency of the LFO modulating the delay time. The higher the value, the faster the sound will oscillate.

Range: 0.10-5.00 Hz

This governs the amount of effect signal fed back to the input, which in turn affects the intensity and character of the chorus effect. Turning this towards 100% produces a flanger type of effect with a pronounced resonance “tone”, while keeping it around 50% produces a more gentle chorus effect.

The BBD is a bucket brigade delay line which simulates vintage ensemble effects. Historically, the bucket brigade delay line was built up by a series of (analog) capacitors, that were clocked to consecutively transmit signals, via one capacitor at a time, thus creating a delayed signal. The BBD algorithm in Quartet features three chorus effects in parallel, and therefore provides a much richer and denser effect than the Chorus algorithm.

The picture below shows the basic principle of the BBD algorithm:

Here you set the delay time between the dry and processed signals. The delay is preset scaled between the three delay lines. In practice, this determines where the notches/peaks will appear in the frequency spectrum.

Range: 1.00-30.00 ms

This determines the depth of the LFO modulation, i.e. by how much the delay time should be modulated. If you set this to 0, the delay time will be static (unless you are using Noise Mod, see “Noise Mod”).

If Mod Depth and Noise Mod (see “Noise Mod”) are both set to 0, the Width control (see “Width”) has no effect.

This determines the frequency of the LFO modulating the delay time.

Range: 0.20-10.00 Hz

This amplitude-modulates the signal with lowpass filtered noise, and generates a kind of “sparkling” effect.

If Noise Mod and Mod Depth (see “Mod Depth”) are both set to 0, the Width control (see “Width”) has no effect.

The FFT algorithm simulates a type of chorus/ensemble effect by utilizing noise modulation of the signal partials. First the signal is analyzed using FFT (Fast Fourier Transform) and converted to a representation in the frequency domain. Then, the partials are modulated by noise to achieve a very nice and dense ensemble effect.

This sets the accuracy (and speed) of the frequency analysis. “1” is the fastest detection and preserves transients in the signal - but this also leaves out detection of low frequencies. “4” is the most accurate detection. However, it’s also slower since it also detects low-frequency material (which takes a little longer to detect).

This determines the depth of the noise modulation of the signal’s partials. The parameter controls a combination of noise amplitude and bandwidth. The result also depends on the Frequency Range parameter (see “Frequency Range”).

The picture below shows how the Mod Depth parameter affects the partials at full Frequency Range:

#Frequency Range

The Frequency Range parameter determines which part of the frequency range should be noise-modulated and which part should be left unaffected.

Set the desired Frequency Range by dragging either “handle” sideways.

To move the Frequency Range while maintaining the currently set bandwidth, drag the area between the “handles” sideways.

The picture below shows how the Mod Depth parameter affects the partials in the signal at two different Frequency Range settings:

The first example shows the modulation of the partials at full bandwidth. The second example shows the partial modulation with the lower Frequency set to a higher value. In the second example, only the upper partials are modulated. The lower partials are left unaffected.

The Grain algorithm generates an ensemble effect by “extracting” grains from the input signal in real-time and then cross-fading through the grains in various ways. The method is similar to the “Long Grains” algorithm used in the Grain Sample Manipulator device in Reason. The picture below shows the principle for the Grain algorithm:

An example of a signal generated from 5 grains of the input signal.

The Random Phase function randomly alters the phase of the grains to create a “bubbly” kind of effect, caused by phase cancellation. This is especially noticeable when the Jitter parameter (see “Jitter”) is set to a low value.

This controls the grain length. High values produce a more smooth effect, whereas low values generate more of a “stuttering” effect.

This randomly changes the initial pitch of the grains.

The Jitter function modulates the grain playback position randomly. The Jitter function can be great for generating chorus-like effects and to make a sound more “alive”, depending on the other settings.

The Density function is a combination of grain size, playback rate and the amount of grain overlap. High values produce a really fat and dense chorus/ensemble effect, whereas low values generate a “thinner” effect.

This CV input can be used for modulating the Mod Depth parameter in the different algorithms. The input accepts bipolar control signals. The input signal can be attenuated with the corresponding attenuation knob.

Note that the CV Modulation is global for the Mod Depth control in all algorithms.

This CV input can be used for modulating the Width parameter in the different algorithms. The input accepts bipolar control signals. The input signal can be attenuated with the corresponding attenuation knob.

Note that the CV Modulation is global for the Width control in all algorithms.

This CV input can be used for modulating the Dry/Wet parameter in the different algorithms. The input accepts bipolar control signals. The input signal can be attenuated with the corresponding attenuation knob.

Note that the CV Modulation is global for the Dry/Wet control in all algorithms.

#Input Left & Right

Patch the audio signals you want to process here.

If your input signal is in mono, connect only to the L (left) input.

Note that the result also depends on the current Routing setting (see “Routing”).

#Output Left & Right

These are the audio outputs.

Note that the result also depends on the current Routing setting (see “Routing”).