Kurzweil Effects

Tube and Poly Amp

These combination algorithms are provided with guitar processing in mind.

Each of the algorithms sends the signal through a gate, tone controls, tube distortion and cabinet simulation or EQ section.

Also depending on the algorithm selected, the signal may pass through chorus, flange, or moving delay.

The algorithms are mono, though the chorus or flange can provide stereo spreading at the output.

Gate: The gate (same as the Gate Algorithm) allows you to cut out noise during silence.

The gate has its own side-chain processing path, and a number of signal routing options for side-chain processing are provided.

The gate side-chain input may be taken from either the left or right channels, or the average signal magnitude of the left and right channels may be selected with the GateSCInp parameter.

Also you may choose to gate the sum of left and right channels or just one of the channels with the Gate Chan parameter.

Since the effect is mono, if you gate only one channel (left or right), then that channel will be sent to the next stage of the effect, and the channel that is not selected will be discarded.

If you choose both (L+R)/2, the sum (mix) of both channels will be used for further processing.

Gate+TubeAmp is the simplest of these algorithms.

With the exception of except Gate+TubeAmp, each of these algorithms offers a flexible chain of effects designed primarily for guitar processing.

Each chain offers a different combination of a 3-band tone control, tube-amp distortion drive, poly-amp distortion drive, cabinet simulation, chorus, flange, and a generic moving delay.

The entire algorithm is monaural with the exception of the final chorus or flange at the end of each chain, which have one input and a stereo output.

At the beginning of each chain is a 3-band tone control authentically recreating the response in many guitar preamps based on real measurements collected by Kurzweil engineers.

It is adjusted with the Bass Tone, Mid Tone, and Treb Tone controls with values ranging from 0 to 10 commonly found on many guitar amps.

The flattest frequency response is obtained by setting Mid Tone to 10.0, and both Bass and Treb Tone controls to 0.0.

The tone controls are integrated with one of two types of preamp drive circuits: TubeAmp and PolyAmp.

TubeAmp: The TubeAmp faithfully models the response and smooth distortion caused by overloading a vacuum tube circuit.
PolyAmp: PolyAmp is closely related to the PolyDistort + EQ algorithm offering a brighter sound quality with more sustain.
Distortion: The amount of distortion is controlled by adjusting the Tube Drive or Poly Drive parameter.

High frequency energy caused by distortion can be rolled off by using the Warmth parameter.
Cabinet Simulator: Following the distortion drive element is a cabinet simulator.

The cabinet simulator models the responses of various types of mic’d guitar cabinets.

The preset can be selected using the Cab Preset parameter.

The following is the list of cabinet presets and their descriptions:

Basic: Flat response from 100 Hz to 4 kHz with 24dB/oct rolloffs on each end
Lead 12: Open back hard American type with one 12” driver
2x12: Closed back classic American type with two 12” drivers
Open 12: Open back classic American type with one 12” driver
Open 10: Open back classic American type with one 10” driver
4x12: Closed back British type with four 12” drivers
Hot 2x12: Closed back hot rod type with two 12” drivers
Hot 12: Open back hot rod type with one 12” driver

Cabinet Control

The cabinet can by switched on or off with the Cab In/Out parameter.

The Cab Pan parameter adjusts the final pan position of the cabinet at the output of the algorithm, but this does not affect the cabinet signal fed into the final stereo flange or chorus.

If Ch Wet/Dry or Fl Wet/Dry is set to 100%, this pan control will not have any audible affect since the entire output of the cabinet is fed into the flange or chorus instead of the algorithm output.

Chorus or Flange

At the end of the chain is either a chorus or a flange controlled by parameters beginning with “Ch” or “Fl.”

The chorus and flange have mono inputs and stereo outputs.

Each is a standard single tap dual channel chorus (see Chorus 1) or flange (see Flanger 1) with independent controls for left and right channels found in many other 1 unit combination algorithms.

The Ch Wet/Dry or Fl Wet/Dry control determines the final output mix of the algorithm.

When set at 0%, only the cabinet simulator output is fed to the output of the algorithm.
At 100%, only the output of the chorus or flange is heard.

Balance

Left/right balance specifically for the chorus or flange can be adjusted with the Out Bal control.

Moving Delay (MD)

In addition, there is a generic monaural moving delay segment.

Its parameters begin with the letters “MD.”

The moving delay is flexible enough that it can serve as a chorus, flange, or straight delay.

For more detailed information, refer to the section describing the Dual MovDelay and Dual MvDly+MvDly algorithms.

Moving Delay Insert: Moving Delay can be inserted either before the tone controls (PreDist), or after the distortion drive (PostDist), or bypassed altogether.

This is selected with the MD Insert parameter.

Also provided is the MD Wet/Dry parameter that mixes the output of the moving delay circuit with its own input to be fed into the next effect in the chain.

Gate+TubeAmp

Effects Size : 3

Algorithm Type : Guit

Gate and Tube Amplifier emulation

Gate+TubeAmp Presets
300 Classic Gtr Dist	301 Crunch Guitar	302 SaturatedGtrDist
303 Mean 70’sFunkGtr

Figure 1. Gate+TubeAmp - Block Diagram

In/Out	In or Out When set to In the effect is active When set to Out the effect is bypassed
Out Gain	Off, -79.0 to 24.0 dB The overall gain or amplitude at the output of the entire algorithm.
GateIn/Out	In or Out When set to In the gate is active When set to Out the gate is bypassed.
GateSCInp	L, R, (L+R)/2 Select the input source channel for gate side-chain processing, left, right or both. For both (L+R)/2 the averaged magnitude is used.
Gate Chan	L, R, (L+R)/2 Select which input channel will receive gate processing, left, right or mix. This selects the mono input for the algorithm.
Gate Thres	-79.0 to 0.0 dB The signal level in dB required to open the gate (or close the gate if Ducking is On).
Gate Duck	On or Off When set to Off, the gate opens when the signal rises above threshold and closes when the gate time expires. When set to On, the gate closes when the signal rises above threshold and opens when the gate time expires.
Gate Time	25 to 3000 ms The time in seconds that the gate will stay fully on after the signal envelope rises above threshold. The gate timer is started or restarted whenever the signal envelope rises above threshold.
Gate Atk	0.0 to 228.0 ms The time for the gate to ramp from closed to open (reverse if Gate Duck is On) after the signal rises above threshold.
Gate Rel	0 to 3000 ms The time for the gate to ramp from open to closed (reverse if Gate Duck is On) after the gate timer has elapsed.
GateSigDly	0.0 to 25.0 ms The delay in milliseconds (ms) of the signal to be gated relative to the side chain signal. By delaying the main signal, the gate can be opened before the main signal rises above the gating threshold.
Bass Tone Mid Tone Treb Tone	0.0 to 10.0 Adjusts the three bands of tone control integrated with the distortion drive circuit. Flattest response is obtained by setting Mid Tone to 10.0 and both Bass Tone and Treb Tone to 0.0.
Tube Drive	Off, -79.0 to 60.0 dB Adjusts the gain into the distortion circuit. Higher values produce more distortion.
Warmth	8 to 25088 Hz Adjusts a 1 pole (6dB/oct) lowpass filter applied after distortion.
Cab Preset	Basic, Lead 12, 2x12, Open 12, Open 10, 4x12, Hot 2x12, and Hot 12 Eight preset cabinets have been created based on measurements of real guitar amplifier cabinets.

In/Out

In or Out

When set to In the effect is active
When set to Out the effect is bypassed

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the entire algorithm.

GateIn/Out

In or Out

When set to In the gate is active
When set to Out the gate is bypassed.

GateSCInp

L, R, (L+R)/2
Select the input source channel for gate side-chain processing, left, right or both.

For both (L+R)/2 the averaged magnitude is used.

Gate Chan

L, R, (L+R)/2
Select which input channel will receive gate processing, left, right or mix.

This selects the mono input for the algorithm.

Gate Thres

-79.0 to 0.0 dB
The signal level in dB required to open the gate (or close the gate if Ducking is On).

Gate Duck

On or Off

When set to Off, the gate opens when the signal rises above threshold and closes when the gate time expires.
When set to On, the gate closes when the signal rises above threshold and opens when the gate time expires.

Gate Time

Gate Atk

0.0 to 228.0 ms
The time for the gate to ramp from closed to open (reverse if Gate Duck is On) after the signal rises above threshold.

Gate Rel

0 to 3000 ms
The time for the gate to ramp from open to closed (reverse if Gate Duck is On) after the gate timer has elapsed.

GateSigDly

0.0 to 25.0 ms

The delay in milliseconds (ms) of the signal to be gated relative to the side chain signal.

By delaying the main signal, the gate can be opened before the main signal rises above the gating threshold.

Bass Tone
Mid Tone
Treb Tone

0.0 to 10.0
Adjusts the three bands of tone control integrated with the distortion drive circuit.

Flattest response is obtained by setting Mid Tone to 10.0 and both Bass Tone and Treb Tone to 0.0.

Tube Drive

Off, -79.0 to 60.0 dB
Adjusts the gain into the distortion circuit.

Higher values produce more distortion.

Warmth

8 to 25088 Hz
Adjusts a 1 pole (6dB/oct) lowpass filter applied after distortion.

Cab Preset

Basic, Lead 12, 2x12, Open 12, Open 10, 4x12, Hot 2x12, and Hot 12
Eight preset cabinets have been created based on measurements of real guitar amplifier cabinets.

TubeAmp<>MD>Chor

Effects Size : 3

Algorithm Type : Guit

Tube amplifier distortion circuits in combination with moving delays and a stereo chorus

TubeAmp<>MD>Chor Presets
313 ODriveGtrLd DlCh
314 Krazy Gtr Comper
440 Chr→GtrDst→Chr

Figure 2. Moving delay inserted pre-distortion

Figure 3. Moving delay inserted post-distortion

In/Out

In or Out

When set to In the effect is active
When set to Out the effect is bypassed

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the entire algorithm.

Input Bal

-100 to 100 %
Adjusts the ratio of left and right algorithm inputs to be summed into the monaural signal that is processed by the effect.

0% blends equal amount of left and right
Negative values blend increasing amounts of left
Positive values blend increasing amounts of right

Bass Tone
Mid Tone
Treb Tone

0.0 to 10.0
Adjusts the three bands of tone control integrated with the distortion drive circuit.

Flattest response is obtained by setting Mid Tone to 10.0 and both Bass Tone and Treb Tone to 0.0.

Tube Drive

Off, -79.0 to 60.0 dB
Adjusts the gain into the distortion circuit.

Higher values produce more distortion.

Warmth

8 to 25088 Hz
Adjusts a 1 pole (6dB/oct) lowpass filter applied after distortion.

Cab Preset

Basic, Lead 12, 2x12, Open 12, Open 10, 4x12, Hot 2x12, and Hot 12
Eight preset cabinets have been created based on measurements of real guitar amplifier cabinets.

Cab Pan

-100 to 100 %
Adjusts the output pan position of the cabinet simulator signal that is mixed at the output of the algorithm.

That when Ch Wet/Dry is set to 100%, no signal from the cabinet is mixed directly to the output, so this parameter has no affect.

MD Insert

PreDist, PostDist, Bypass
Selects where in the signal chain the moving delay is to be.

PreDist places it before the distortion and tone circuit
PostDist places it between the distortion circuit and cabinet simulator
Bypass takes it completely out of the path

MD Wet/Dry

0 to 100 %
Adjusts the ratio of the moving delay output mixed with its own input to be fed to the next effect in the chain.

MD Delay

0.0 to 1000.0 ms
Adjusts the delay time for the moving delay circuit, which is the center of LFO excursion.

MD LFOMode

ChorTri, ChorTrap, Delay, Flange
Adjusts the LFO excursion type.

In Flange mode, the LFO is optimized for flange effects and LFO Dpth adjusts the excursion amount
In ChorTri and ChorTrap modes, the LFO is optimized for triangle and trapezoidal pitch envelopes respectively, and LFO Dpth adjusts the amount of chorus detuning
In Delay mode, the LFO is turned off leaving a basic delay.

LFO Rate and LFO Dpth in Delay mode are disabled

MD LFORate

0.00 to 10.00 Hz
Adjusts the LFO speed for the moving delay circuit.

MD LFODpth

0.0 to 200.0 %

In Flange LFO mode, this adjusts an arbitrary LFO excursion amount.
In ChorTri and ChorTrap modes, this controls the chorus detune amount
In Delay mode, this is disabled

MD Fdbk

-100 to 100 %
Adjusts the level of the moving delay circuit output signal fed back into its own input.

Negative values polarity-invert the feedback signal.

Ch Wet/Dry

0 to 100 %
Adjusts the ratio of chorus signal and the cabinet simulator signal fed to the output of the algorithm.

- 0% feeds only the cabinet simulator to the output bypassing the final chorus
- 100% feeds only the chorus to the output

Ch Out Bal

-100 to 100 %
Adjusts the left/right output balance of the chorus signal.

Negative values balance toward the left while positive values balance toward the right.

TubeAmp<>MD>Flan

Effects Size : 3

Algorithm Type : DstF

Mono distortion circuits in combination with moving delays and a stereo flange

TubeAmp<>MD>Flan Presets
316 LeadGtr Dly Flng
491 MyGtrAteYo’Momma

In/Out

In or Out

When set to In the effect is active
When set to Out the effect is bypassed

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the entire algorithm.

Input Bal

-100 to 100 %
Adjusts the ratio of left and right algorithm inputs to be summed into the monaural signal that is processed by the effect.

0% blends equal amount of left and right
Negative values blend increasing amounts of left
Positive values blend increasing amounts of right

Bass Tone
Mid Tone
Treb Tone

0.0 to 10.0
Adjusts the three bands of tone control integrated with the distortion drive circuit.

Flattest response is obtained by setting Mid Tone to 10.0 and both Bass Tone and Treb Tone to 0.0.

Tube Drive

Off, -79.0 to 60.0 dB
Adjusts the gain into the distortion circuit.

Higher values produce more distortion.

Warmth

8 to 25088 Hz
Adjusts a 1 pole (6dB/oct) lowpass filter applied after distortion.

Cab Preset

Basic, Lead 12, 2x12, Open 12, Open 10, 4x12, Hot 2x12, and Hot 12
Eight preset cabinets have been created based on measurements of real guitar amplifier cabinets.

Cab Pan

-100 to 100 %
Adjusts the output pan position of the cabinet simulator signal that is mixed at the output of the algorithm.

Fl Wet/Dry is set to 100%, no signal from the cabinet is mixed directly to the output, so this parameter has no affect.

MD Insert

PreDist, PostDist, Bypass
Selects where in the signal chain the moving delay is to be.

PreDist places it before the distortion and tone circuit
PostDist places it between the distortion circuit and cabinet simulator
Bypass takes it completely out of the path

MD Wet/Dry

0 to 100 %
Adjusts the ratio of the moving delay output mixed with its own input to be fed to the next effect in the chain.

MD Delay

0.0 to 1000.0 ms
Adjusts the delay time for the moving delay circuit, which is the center of LFO excursion.

MD LFOMode

ChorTri, ChorTrap, Delay, Flange
Adjusts the LFO excursion type.

In Flange mode, the LFO is optimized for flange effects and LFO Dpth adjusts the excursion amount
In ChorTri and ChorTrap modes, the LFO is optimized for triangle and trapezoidal pitch envelopes respectively, and LFO Dpth adjusts the amount of chorus detuning
In Delay mode, the LFO is turned off leaving a basic delay.

LFO Rate and LFO Dpth in Delay mode are disabled

MD LFORate

0.00 to 10.00 Hz
Adjusts the LFO speed for the moving delay circuit.

MD LFODpth

0.0 to 200.0 %

In Flange LFO mode, this adjusts an arbitrary LFO excursion amount.
In ChorTri and ChorTrap modes, this controls the chorus detune amount
In Delay mode, this is disabled

MD Fdbk

-100 to 100 %
Adjusts the level of the moving delay circuit output signal fed back into its own input.

Negative values polarity-invert the feedback signal.

Fl Wet/Dry

0 to 100 %
Adjusts the ratio of flange signal and the cabinet simulator signal fed to the output of the algorithm.

- 0% feeds only the cabinet simulator to the output bypassing the final flange
- 100% feeds only the flange to the output

Fl Out Bal

-100 to 100 %
Adjusts the left/right output balance of the flange signal.

Negative values balance toward the left while positive values balance toward the right.

PolyAmp<>MD>Chor

Effects Size : 3

Algorithm Type : DstC

Poly amplifier distortion circuits in combination with moving delays and a stereo chorus

PolyAmp<>MD>Chor Presets
490 Flg→GtrDst→Chr

In/Out

In or Out

When set to In the effect is active
When set to Out the effect is bypassed

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the entire algorithm.

Input Bal

-100 to 100 %
Adjusts the ratio of left and right algorithm inputs to be summed into the monaural signal that is processed by the effect.

0% blends equal amount of left and right
Negative values blend increasing amounts of left
Positive values blend increasing amounts of right

Bass Tone
Mid Tone
Treb Tone

0.0 to 10.0
Adjusts the three bands of tone control integrated with the distortion drive circuit.

Flattest response is obtained by setting Mid Tone to 10.0 and both Bass Tone and Treb Tone to 0.0.

Poly Drive

Off, -79.0 to 60.0 dB
Adjusts the gain into the distortion circuit.

Higher values produce more distortion.

Warmth

8 to 25088 Hz
Adjusts a 1 pole (6dB/oct) lowpass filter applied after distortion.

Cab Preset

Basic, Lead 12, 2x12, Open 12, Open 10, 4x12, Hot 2x12, and Hot 12
Eight preset cabinets have been created based on measurements of real guitar amplifier cabinets.

Cab Pan

-100 to 100 %
Adjusts the output pan position of the cabinet simulator signal that is mixed at the output of the algorithm.

That when Ch Wet/Dry is set to 100%, no signal from the cabinet is mixed directly to the output, so this parameter has no affect.

MD Insert

PreDist, PostDist, Bypass
Selects where in the signal chain the moving delay is to be.

PreDist places it before the distortion and tone circuit
PostDist places it between the distortion circuit and cabinet simulator
Bypass takes it completely out of the path

MD Wet/Dry

0 to 100 %
Adjusts the ratio of the moving delay output mixed with its own input to be fed to the next effect in the chain.

MD Delay

0.0 to 1000.0 ms
Adjusts the delay time for the moving delay circuit, which is the center of LFO excursion.

MD LFOMode

ChorTri, ChorTrap, Delay, Flange
Adjusts the LFO excursion type.

In Flange mode, the LFO is optimized for flange effects and LFO Dpth adjusts the excursion amount
In ChorTri and ChorTrap modes, the LFO is optimized for triangle and trapezoidal pitch envelopes respectively, and LFO Dpth adjusts the amount of chorus detuning
In Delay mode, the LFO is turned off leaving a basic delay.

LFO Rate and LFO Dpth in Delay mode are disabled

MD LFORate

0.00 to 10.00 Hz
Adjusts the LFO speed for the moving delay circuit.

MD LFODpth

0.0 to 200.0 %

In Flange LFO mode, this adjusts an arbitrary LFO excursion amount.
In ChorTri and ChorTrap modes, this controls the chorus detune amount
In Delay mode, this is disabled

MD Fdbk

-100 to 100 %
Adjusts the level of the moving delay circuit output signal fed back into its own input.

Negative values polarity-invert the feedback signal.

Ch Wet/Dry

Ch Out Bal

-100 to 100 %
Adjusts the left/right output balance of the chorus signal.

Negative values balance toward the left while positive values balance toward the right.

PolyAmp<>MD>Flan

Effects Size : 3

Algorithm Type : DstF

Poly amplifier distortion circuits in combination with moving delays and a stereo flange

PolyAmp<>MD>Flan Presets
315 MildGtrOD+Dly+Fl

In/Out

In or Out

When set to In the effect is active
When set to Out the effect is bypassed

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the entire algorithm.

Input Bal

-100 to 100 %
Adjusts the ratio of left and right algorithm inputs to be summed into the monaural signal that is processed by the effect.

0% blends equal amount of left and right
Negative values blend increasing amounts of left
Positive values blend increasing amounts of right

Bass Tone
Mid Tone
Treb Tone

0.0 to 10.0
Adjusts the three bands of tone control integrated with the distortion drive circuit.

Flattest response is obtained by setting Mid Tone to 10.0 and both Bass Tone and Treb Tone to 0.0.

Poly Drive

Off, -79.0 to 60.0 dB
Adjusts the gain into the distortion circuit.

Higher values produce more distortion.

Warmth

8 to 25088 Hz
Adjusts a 1 pole (6dB/oct) lowpass filter applied after distortion.

Cab Preset

Basic, Lead 12, 2x12, Open 12, Open 10, 4x12, Hot 2x12, and Hot 12
Eight preset cabinets have been created based on measurements of real guitar amplifier cabinets.

Cab Pan

-100 to 100 %
Adjusts the output pan position of the cabinet simulator signal that is mixed at the output of the algorithm.

Fl Wet/Dry is set to 100%, no signal from the cabinet is mixed directly to the output, so this parameter has no affect.

MD Insert

PreDist, PostDist, Bypass
Selects where in the signal chain the moving delay is to be.

PreDist places it before the distortion and tone circuit
PostDist places it between the distortion circuit and cabinet simulator
Bypass takes it completely out of the path

MD Wet/Dry

0 to 100 %
Adjusts the ratio of the moving delay output mixed with its own input to be fed to the next effect in the chain.

MD Delay

0.0 to 1000.0 ms
Adjusts the delay time for the moving delay circuit, which is the center of LFO excursion.

MD LFOMode

ChorTri, ChorTrap, Delay, Flange
Adjusts the LFO excursion type.

In Flange mode, the LFO is optimized for flange effects and LFO Dpth adjusts the excursion amount
In ChorTri and ChorTrap modes, the LFO is optimized for triangle and trapezoidal pitch envelopes respectively, and LFO Dpth adjusts the amount of chorus detuning
In Delay mode, the LFO is turned off leaving a basic delay.

LFO Rate and LFO Dpth in Delay mode are disabled

MD LFORate

0.00 to 10.00 Hz
Adjusts the LFO speed for the moving delay circuit.

MD LFODpth

0.0 to 200.0 %

In Flange LFO mode, this adjusts an arbitrary LFO excursion amount.
In ChorTri and ChorTrap modes, this controls the chorus detune amount
In Delay mode, this is disabled

MD Fdbk

-100 to 100 %
Adjusts the level of the moving delay circuit output signal fed back into its own input.

Negative values polarity-invert the feedback signal.

Fl Wet/Dry

Fl Out Bal

-100 to 100 %
Adjusts the left/right output balance of the flange signal.

Negative values balance toward the left while positive values balance toward the right.

Distortion

Mono Distortion

Effects Size : 1

Small distortion

Mono Distortion Presets
45 SubtleDistortion

Mono Distortion sums its stereo input to mono, performs distortion followed by a highpass filter and sends the result as centered stereo.

Figure 4. Mono Distortion - Block Diagram

Wet/Dry

0 to 100% wet
The amount of distorted (wet) signal relative to unaffected (dry) signal.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

For distortion, it is often necessary to turn the output gain down as the distortion drive is turned up.

Dist Drive

0 to 96 dB
Applies a boost to the input signal to overdrive the distortion algorithm.

When overdriven, the distortion algorithm will soft-clip the signal.

Since distortion drive will make your signal very loud, you may have to reduce the Out Gain as the drive is increased.

Warmth

8 to 25088 Hz
A lowpass filter in the distortion control path.

This filter may be used to reduce some of the harshness of some distortion settings without reducing the bandwidth of the signal.

Highpass

8 to 25088 Hz
Allows you to reduce the bass content of the distortion content.

If you need more filtering to better simulate a speaker cabinet, you will have to choose a larger distortion algorithm.

MonoDistort+Cab

Effects Size : 2

Small distortion with cabinet emulation

MonoDistort+Cab Presets
307 Dist Cab EPiano

The distortion segment of MonoDistort+Cab is similar to Mono Distortion except the highpass is replaced by a full speaker cabinet model.

Figure 5. MonoDistort+Cab - Block Diagram

Distortion

The distortion algorithm will soft clip the input signal.

The amount of soft clipping depends on how high the distortion drive parameter is set.

Soft clipping means that there is a smooth transition from linear gain to saturated overdrive.

Higher distortion drive settings cause the transition to become progressively sharper or “harder.”

The distortion never produces hard or digital clipping, but it does approach it at high drive settings.

When you increase the distortion drive parameter you are increasing the gain of the algorithm until the signal reaches saturation.

You will have to compensate for increases in drive gain by reducing the output gain.

This algorithm will not digitally clip unless the output gain is over-driven.

Cabinet Modelling

The distortion is followed by a model of a guitar amplifier cabinet.

The model can be bypassed, or there are 8 presets which were derived from measurements of real cabinets.

8 cabinet presets:

Plain
(Flat response from 100 Hz to 4 khz with 24dB/oct rolloffs on each end)
Lead 12
(Open back hard American type with one 12” driver)
2x12
(Closed back classic American type with two 12” drivers)
Open 12
(Open back classic American type with one 12” driver)
Open 10
(Open back classic American type with one 10” driver)
4x12
(Closed back British type with four 12” drivers)
Hot 2x12
(Closed back hot rod type with two 12” drivers)
Hot 12
(Open back hot rod type with one 12” driver)

There is also a panner to route the mono signal between left and right outputs.

In/Out

0 to 100% wet
When set to In, the MonoDistort+Cab effect is active.
When set to Out, MonoDistort+Cab effect is off.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

For distortion, it is often necessary to turn the output gain down as the distortion drive is turned up.

Dist Drive

0 to 96 dB
Applies a boost to the input signal to overdrive the distortion algorithm.

When overdriven, the distortion algorithm will soft-clip the signal.

Since distortion drive will make your signal very loud, you may have to reduce the Out Gain as the drive is increased.

Warmth

8 to 25088 Hz
A lowpass filter in the distortion control path.

This filter may be used to reduce some of the harshness of some distortion settings without reducing the bandwidth of the signal.

Cab Bypass

In/Out
The guitar amplifier cabinet simulation may be bypassed.

When set to In, the cabinet simulation is active.
When set to Out, there is no cabinet filtering.

Cab Preset

Plain, Lead 12, 2x12, Open 12, Open 10, 4x12, Hot 2x12, and Hot 12
Eight preset cabinets have been created based on measurements of real guitar amplifier cabinets.

MonoDistort+EQ

Effects Size : 2

Small distortion with EQ cabinet modelling

MonoDistort+EQ Presets
308 Distortion+EQ

MonoDistort + EQ is similar to Mono Distortion except the single highpass filter is replaced with a pair of second-order highpass/lowpass filters to provide rudimentary speaker cabinet modeling.

The highpass and lowpass filters are then followed by an EQ section with bass and treble shelf filters and two parametric mid filters.

Figure 6. MonoDistort+EQ - Block Diagram

Signals that are symmetric in amplitude (they have the same shape if they are inverted, positive for negative) will usually produce odd harmonic distortion.

For example, a pure sine wave will produce smaller copies of itself at 3, 5, 7, etc. times the original frequency of the sine wave.

In MonoDistort+EQ, a DC offset may be added to the signal to break the amplitude symmetry and will cause the distortion to produce even harmonics.

This can add a “brassy” character to the distorted sound.

The DC offset added prior to distortion gets removed at a later point in the algorithm.

Figure 7. Input/Output transfer characteristic of soft clipping at various drive settings

Wet/Dry	0 to 100% wet The amount of distorted (wet) signal relative to unaffected (dry) signal.
Out Gain	Off, -79.0 to 24.0 dB The overall gain or amplitude at the output of the effect. For distortion, it is often necessary to turn the output gain down as the distortion drive is turned up.
Dist Drive	0 to 96 dB Applies a boost to the input signal to overdrive the distortion algorithm. When overdriven, the distortion algorithm will soft-clip the signal. Since distortion drive will make your signal very loud, you may have to reduce the Out Gain as the drive is increased.
Warmth	8 to 25088 Hz A lowpass filter in the distortion control path. This filter may be used to reduce some of the harshness of some distortion settings without reducing the bandwidth of the signal.
dc Offset	-100 to 100 % Amount of dc related offset in a negative or positive direction.
Cabinet HP	8 to 25088 Hz A highpass filter that controls the low frequency limit of a simulated loudspeaker cabinet.
Cabinet LP	8 to 25088 Hz A lowpass filter that controls the high frequency limit of a simulated loudspeaker cabinet.
Bass Gain	-79.0 to 24.0 dB The amount of boost or cut that the bass shelving filter should apply to the low frequency signals in dB. Every increase of 6 dB approximately doubles the amplitude of the signal. Positive values boost the bass signal below the specified frequency. Negative values cut the bass signal below the specified frequency.
Bass Freq	8 to 25088 Hz The center frequency of the bass shelving filter in intervals of one semitone.
Treb Gain	-79.0 to 24.0 dB The amount of boost or cut that the treble shelving filter should apply to the high frequency signals in dB. Every increase of 6 dB approximately doubles the amplitude of the signal. Positive values boost the treble signal above the specified frequency Negative values cut the treble signal above the specified frequency
Treb Freq	8 to 25088 Hz The center frequency of the treble shelving filter in intervals of one semitone.
Mid1 Gain Mid2 Gain	-79.0 to 24.0 dB The amount of boost or cut that the mid parametric filter should apply in dB. Every increase of 6 dB approximately doubles the amplitude of the signal. Positive values boost the signal at the specified frequency Negative values cut the signal at the specified frequency
Mid1 Freq Mid2 Freq	8 to 25088 Hz The center frequency of the mid parametric filter in intervals of one semitone. The boost or cut will be at a maximum at this frequency.
Mid1 Wid Mid2 Wid	0.010 to 5.000 oct The bandwidth of the mid parametric filter may be adjusted. You specify the bandwidth in octaves. Small values result in a very narrow filter response Large values result in a very broad response

Wet/Dry

0 to 100% wet
The amount of distorted (wet) signal relative to unaffected (dry) signal.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

For distortion, it is often necessary to turn the output gain down as the distortion drive is turned up.

Dist Drive

0 to 96 dB
Applies a boost to the input signal to overdrive the distortion algorithm.

When overdriven, the distortion algorithm will soft-clip the signal.

Since distortion drive will make your signal very loud, you may have to reduce the Out Gain as the drive is increased.

Warmth

8 to 25088 Hz
A lowpass filter in the distortion control path.

This filter may be used to reduce some of the harshness of some distortion settings without reducing the bandwidth of the signal.

dc Offset

-100 to 100 %
Amount of dc related offset in a negative or positive direction.

Cabinet HP

8 to 25088 Hz

A highpass filter that controls the low
frequency limit of a simulated loudspeaker cabinet.

Cabinet LP

8 to 25088 Hz

A lowpass filter that controls the high
frequency limit of a simulated loudspeaker cabinet.

Bass Gain

-79.0 to 24.0 dB

The amount of boost or cut that the bass
shelving filter should apply to the low frequency signals in dB.

Every increase of 6 dB approximately doubles the amplitude of the signal.

Positive values boost the bass signal below the specified frequency.
Negative values cut the bass signal below the specified frequency.

Bass Freq

8 to 25088 Hz

The center frequency of the bass shelving filter in intervals of one semitone.

Treb Gain

-79.0 to 24.0 dB

The amount of boost or cut that the treble
shelving filter should apply to the high frequency signals in dB.

Every increase of 6 dB
approximately doubles the amplitude of the signal.

Positive values boost the treble signal
above the specified frequency
Negative values cut the treble signal above the specified frequency

Treb Freq

8 to 25088 Hz

The center frequency of the treble
shelving filter in intervals of one semitone.

Mid1 Gain
Mid2 Gain

-79.0 to 24.0 dB

The amount of boost or cut that the mid parametric filter should apply in dB.

Every increase of 6 dB approximately doubles the amplitude of the signal.

Positive values boost the signal at the specified frequency
Negative values cut the signal at the specified frequency

Mid1 Freq
Mid2 Freq

8 to 25088 Hz

The center frequency of the mid parametric filter in intervals of one semitone.

The boost or cut will be at a maximum at this frequency.

Mid1 Wid
Mid2 Wid

0.010 to 5.000 oct
The bandwidth of the mid parametric filter may be adjusted.

You specify the bandwidth in octaves.

Small values result in a very narrow filter response
Large values result in a very broad response

StereoDistort+EQ

Effects Size : 3

Stereo distortion with limited EQ

StereoDistort+EQ Presets
309 Burnt Transistor

StereoDistort+EQ processes the left and right channels separately, though there is only one set of parameters for both channels.

The stereo distortion has only 1 parametric mid filter.

Figure 8. StereoDistort+EQ - Block Diagram

Wet/Dry	0 to 100% wet The amount of distorted (wet) signal relative to unaffected (dry) signal.
Out Gain	Off, -79.0 to 24.0 dB The overall gain or amplitude at the output of the effect. For distortion, it is often necessary to turn the output gain down as the distortion drive is turned up.
Dist Drive	0 to 96 dB Applies a boost to the input signal to overdrive the distortion algorithm. When overdriven, the distortion algorithm will soft-clip the signal. Since distortion drive will make your signal very loud, you may have to reduce the Out Gain as the drive is increased.
Warmth	8 to 25088 Hz A lowpass filter in the distortion control path. This filter may be used to reduce some of the harshness of some distortion settings without reducing the bandwidth of the signal.
Cabinet HP	8 to 25088 Hz A highpass filter that controls the low frequency limit of a simulated loudspeaker cabinet.
Cabinet LP	8 to 25088 Hz A lowpass filter that controls the high frequency limit of a simulated loudspeaker cabinet.
Bass Gain	-79.0 to 24.0 dB The amount of boost or cut that the bass shelving filter should apply to the low frequency signals in dB. Every increase of 6 dB approximately doubles the amplitude of the signal. Positive values boost the bass signal below the specified frequency. Negative values cut the bass signal below the specified frequency.
Bass Freq	8 to 25088 Hz The center frequency of the bass shelving filter in intervals of one semitone.
Treb Gain	-79.0 to 24.0 dB The amount of boost or cut that the treble shelving filter should apply to the high frequency signals in dB. Every increase of 6 dB approximately doubles the amplitude of the signal. Positive values boost the treble signal above the specified frequency Negative values cut the treble signal above the specified frequency
Treb Freq	8 to 25088 Hz The center frequency of the treble shelving filter in intervals of one semitone.
Mid Gain	-79.0 to 24.0 dB The amount of boost or cut that the mid parametric filter should apply in dB. Every increase of 6 dB approximately doubles the amplitude of the signal. Positive values boost the signal at the specified frequency Negative values cut the signal at the specified frequency
Mid Freq	8 to 25088 Hz The center frequency of the mid parametric filter in intervals of one semitone. The boost or cut will be at a maximum at this frequency.
Mid Wid	0.010 to 5.000 oct The bandwidth of the mid parametric filter may be adjusted. You specify the bandwidth in octaves. Small values result in a very narrow filter response Large values result in a very broad response

Wet/Dry

0 to 100% wet
The amount of distorted (wet) signal relative to unaffected (dry) signal.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

For distortion, it is often necessary to turn the output gain down as the distortion drive is turned up.

Dist Drive

Warmth

8 to 25088 Hz
A lowpass filter in the distortion control path.

This filter may be used to reduce some of the harshness of some distortion settings without reducing the bandwidth of the signal.

Cabinet HP

8 to 25088 Hz

A highpass filter that controls the low frequency limit of a simulated loudspeaker cabinet.

Cabinet LP

8 to 25088 Hz

A lowpass filter that controls the high frequency limit of a simulated loudspeaker cabinet.

Bass Gain

-79.0 to 24.0 dB

The amount of boost or cut that the bass shelving filter should apply to the low frequency signals in dB.

Every increase of 6 dB approximately doubles the amplitude of the signal.

Positive values boost the bass signal below the specified frequency.
Negative values cut the bass signal below the specified frequency.

Bass Freq

8 to 25088 Hz

The center frequency of the bass shelving filter in intervals of one semitone.

Treb Gain

-79.0 to 24.0 dB

The amount of boost or cut that the treble shelving filter should apply to the high frequency signals in dB.

Every increase of 6 dB approximately doubles the amplitude of the signal.

Positive values boost the treble signal above the specified frequency
Negative values cut the treble signal above the specified frequency

Treb Freq

8 to 25088 Hz

The center frequency of the treble shelving filter in intervals of one semitone.

Mid Gain

-79.0 to 24.0 dB

The amount of boost or cut that the mid parametric filter should apply in dB.

Every increase of 6 dB approximately doubles the amplitude of the signal.

Positive values boost the signal at the specified frequency
Negative values cut the signal at the specified frequency

Mid Freq

8 to 25088 Hz

The center frequency of the mid parametric filter in intervals of one semitone.

The boost or cut will be at a maximum at this frequency.

Mid Wid

0.010 to 5.000 oct
The bandwidth of the mid parametric filter may be adjusted.

You specify the bandwidth in octaves.

Small values result in a very narrow filter response
Large values result in a very broad response

PolyDistort+EQ

Effects Size : 2

8-stage distortion followed by EQ

PolyDistort+EQ Presets
305 Synth Distortion

Figure 9. PolyDistort+EQ - Block Diagram

PolyDistort + EQ is a distortion algorithm followed by equalization.

The algorithm consists of an input gain stage, and then 8 cascaded distortion stages.

Each stage is followed by a one pole lowpass filter.
There is also a one-pole lowpass filter in front of the first stage.
After the distortion there is a four-band EQ section: Bass, Treble, and two Parametric Mids.

PolyDistort + EQ is an unusual distortion algorithm that provides a great number of parameters to build a distortion sound from the ground up.

The eight distortion stages each add a small amount of distortion to your sound.

Taken together, you can get a very harsh heavy metal sound.

Between each distortion stage is a lowpass filter.

The lowpass filters work with the distortion stages to help mellow out the sound.

Without any lowpass filters the distortion will get very harsh and raspy.

Stages of distortion can be removed by setting the Curve parameter to 0.

You can then do a 6, 4, or 2-stage distortion algorithm.

The corresponding lowpasses should be turned off if there is no distortion in a section.

More than four stages seem necessary for lead guitar sounds
For a cleaner sound, you may want to limit yourself to only four stages

Once you have set up a distorted sound you are satisfied with, the Dist Drive parameter controls the input gain to the distortion, providing a single parameter for controlling distortion amount.

You will probably find that you will have to cut back on the output gain as you drive the distortion louder.

Post-distortion EQ is definitely needed for make things sound right.

This should be something like a guitar speaker cabinet simulator, although not exactly, since we are already doing a lot of lowpass filtering inside the distortion itself.

Possible EQ settings you can try are Treble -20 dB at 5 kHz, Bass -6 dB at 100 Hz, Mid1, wide, +6 dB at 2 kHz, Mid2, wide, +3 dB at 200 Hz, but of course you should certainly experiment to get your sound.

Treble: The Treble is helping to remove raspiness.
Bass: Is removing the extreme low end like an open-back guitar cabinet.
(Not that guitar speakers have that much low end anyway)
Mid1: Adds enough highs so that things can sound bright even in the presence of all the HF roll-off
Mid2: Adds some warmth.

Your favorite settings will probably be different.

Boosting the Treble may not be a good idea.

EQ done in front of the distortion will not be heard as simple EQ, because the distortion section makes an adjustment in one frequency range felt over a much wider range due to action of the distortion.

Simple post EQ is a bit too obvious for the ear, and it can get tired of it after a while.

Wet/Dry

0 to 100% wet
This is a simple mix of the distorted signal relative to the dry undistorted input signal.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

For distortion, it is often necessary to turn the output gain down as the distortion drive is turned up.

Dist Drive

Off, -79.0 to 48.0 dB
Applies gain to the input prior to distortion.

It is the basic “distortion drive” control.

Anything over 0 dB could clip.

Normally clipping would be bad, but the distortion algorithm tends to smooth things out.

Still, considering that for some settings of the other parameters you would have to back off the gain to -48 dB in order to get a not very distorted sound for full scale input, you should go easy on this amount.

Curve 1
Curve 2
Curve 3
Curve 4
Curve 5
Curve 6
Curve 7
Curve 8

0 to 127 %
The curvature of the individual distortion stages.

0% is no curvature (no distortion at all)
At 100%, the curve bends over smoothly and becomes perfectly flat right before it goes into clipping

LP0 Freq
LP1 Freq
LP2 Freq
LP3 Freq
LP4 Freq
LP5 Freq
LP6 Freq
LP7 Freq
LP8 Freq

8 to 25088 Hz
These are the one pole lowpass controls.

LP0 Freq handles the initial lowpass prior to the first distortion stage.

The other lowpass controls follow their respective distortion stages.

With all lowpasses out of the circuit (set to the highest frequency), the sound tends to be too bright and raspy.

With less distortion drive, less filtering is needed.

If you turn off a distortion stage (set to 0%), you should turn off the lowpass filter by setting it to the highest frequency.

Bass Gain

-79.0 to 24.0 dB
The amount of boost or cut that the bass shelving filter should apply to the low frequency signals in dB.

Every increase of 6 dB approximately doubles the amplitude of the signal.

Positive values boost the bass signal below the specified frequency.
Negative values cut the bass signal below the specified frequency.

Bass Freq

8 to 25088 Hz
The center frequency of the bass shelving filter in intervals of one semitone.

Treb Gain

-79.0 to 24.0 dB
The amount of boost or cut that the treble shelving filter should apply to the high frequency signals in dB.

Every increase of 6 dB approximately doubles the amplitude of the signal.

Positive values boost the treble signal above the specified frequency
Negative values cut the treble signal above the specified frequency

Treb Freq

8 to 25088 Hz
The center frequency of the treble shelving filter in intervals of one semitone.

Mid1 Gain
Mid2 Gain

-79.0 to 24.0 dB
The amount of boost or cut that the mid parametric filter should apply in dB.

Every increase of 6 dB approximately doubles the amplitude of the signal.

Positive values boost the signal at the specified frequency
Negative values cut the signal at the specified frequency

Mid1 Freq
Mid2 Freq

8 to 25088 Hz
The center frequency of the mid parametric filter in intervals of one semitone.

The boost or cut will be at a maximum at this frequency.

Mid1 Width
Mid2 Width

0.010 to 5.000 oct
The bandwidth of the mid parametric filter may be adjusted.

You specify the bandwidth in octaves.

Small values result in a very narrow filter response
Large values result in a very broad response

Subtle Distort

Effects Size : 1

Adds small amount of distortion to the signal

Subtle Distort Presets
311 A little dirty
312 Slight Overload

Use Subtle Distort to apply small amounts of distortion to a signal.

Distortion: The distortion characteristic is set with the Curvature and EvenOrders parameters.

Increasing Curvature increases the distortion amount while EvenOrders increases the asymmetry of the distortion, adding even distortion harmonics.

The distorted signal then is sent through two one-pole lowpass filters and added to the dry input signal.
Lowpass Filter: The lowpass filters can reduce any harshness from the raw distortion operation.

The Dry In/Out is provided as a utility to audition the distortion signal in the absence of dry signal.

Out Gain and Dist Gain can be adjusted together to match the level of the bypassed (dry only) signal.

Adding distortion to the dry signal will increase the output level unless Out Gain is reduced.

In/Out	In or Out When set to In the distortion is active When set to Out the distortion is bypassed
Dry In/Out	In or Out Utility parameter to listen to distortion without the dry signal.
Out Gain	Off, -79.0 to 24.0 dB The overall gain or amplitude at the output of the effect.
Dist Gain	Off, -79.0 to 24.0 dB The gain or amplitude of the distorted signal path prior to passing through the Out Gain adjustment.
Curvature	0 to 100 % The amount of distortion. none at 0% maximum at 100%.
EvenOrders	0 to 100 % The asymmetry of the distortion. (Number of even harmonics) none at 0% maximum at 100%.
Dist LP A	8 to 25088 Hz Frequency of Lowpass Filter A.
Dist LP B	8 to 25088 Hz Frequency of Lowpass Filter B.

In/Out

In or Out

When set to In the distortion is active
When set to Out the distortion is bypassed

Dry In/Out

In or Out
Utility parameter to listen to distortion without the dry signal.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

Dist Gain

Off, -79.0 to 24.0 dB
The gain or amplitude of the distorted signal path prior to passing through the Out Gain adjustment.

Curvature

0 to 100 %
The amount of distortion.

none at 0%
maximum at 100%.

EvenOrders

0 to 100 %
The asymmetry of the distortion.
(Number of even harmonics)

none at 0%
maximum at 100%.

Dist LP A

8 to 25088 Hz
Frequency of Lowpass Filter A.

Dist LP B

8 to 25088 Hz
Frequency of Lowpass Filter B.

Super Shaper

Effects Size : 1

8 x the gain of the V.A.S.T. shaper

Super Shaper Presets
317 Drum Shaper
319 SuperShaper

The Super Shaper algorithm packs 2-1/2 times the number of shaping loops, and 8 times the gain of the V.A.S.T. shaper.

Setting Super Shaper amount under 1.00x produces the same nonlinear curve found in the V.A.S.T. shaper.

At values above 1.00x where the V.A.S.T. shaper will pin at zero, the Super Shaper provides 6 more sine intervals before starting to zero-pin at 2.50x.

The maximum shaper amount for Super Shaper is 32.00x.

Figure 10. Four values of the Amount parameter

Wet/Dry	-100 to 100% The relative amount of input signal and effected signal that is to appear in the final effect output mix. When set to 0%, the output is taken only from the input (dry). When set to 100%, the output is all wet. Negative values polarity invert the wet signal.
Out Gain	Off, -79.0 to 24.0 dB The overall gain or amplitude at the output of the effect.
Amount	0.10 to 32.00 x Adjusts the shaper intensity.

Wet/Dry

-100 to 100%
The relative amount of input signal and effected signal that is to appear in the final effect output mix.

When set to 0%, the output is taken only from the input (dry).
When set to 100%, the output is all wet.

Negative values polarity invert the wet signal.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

Amount

0.10 to 32.00 x
Adjusts the shaper intensity.

3 Band Shaper

Effects Size : 2

3 split band shapers

3 Band Shaper Presets
318 SubtleDrumShape
320 3 Band Shaper
321 New3BandShaper

The 3 Band Shaper non-destructively splits the input signal into 3 separate bands using 1 pole (6dB/oct) filters, and applies a V.A.S.T.-type shaper to each band separately.

The cutoff frequencies for these filters are controlled with the CrossOver1 and CrossOver2 parameters.

The low band contains frequencies from 0 Hz (DC) to the lower of the two CrossOver settings.

The mid band contains frequencies between the two selected frequencies, and the hi band contains those from the higher of the two CrossOver settings, up to 24kHz.

Each frequency band has an enable switch for instantly bypassing any processing for that band, and a Mix control for adjusting the level of each band that is mixed at the output. (negative Mix values polarity invert that band)

The shaper Amt controls provide the same type of shaping as V.A.S.T. shapers, but can go to 6.00x.

Wet/Dry	-100 to 100% The relative amount of input signal and effected signal that is to appear in the final effect output mix. When set to 0%, the output is taken only from the input (dry). When set to 100%, the output is all wet. Negative values polarity invert the wet signal. Out Gain The overall gain or amplitude at the output of the effect. Amount Adjusts the shaper intensity.
Out Gain	Off, -79.0 to 24.0 dB The overall gain or amplitude at the output of the effect.
CrossOver1	17 to 25088 Hz Adjusts one of the -6dB crossover points at which the input signal will be divided into the high, mid and low bands.
CrossOver2	17 to 25088 Hz Adjusts the other -6dB crossover points at which the input signal will be divided into the high, mid and low bands.
Lo Enable Mid Enable Hi Enable	On or Off Turns processing for each band on or off. Turning each of the three bands Off results in a dry output signal.
Lo Amt Mid Amt Hi Amt	0.10 to 6.00x Adjusts the shaper intensity for each band.
Lo Mix Mid Mix Hi Mix	-100 to 100% Adjusts the level that each band is summed together as the wet signal. Negative values polarity invert the particular bands signal.

Wet/Dry

-100 to 100%
The relative amount of input signal and effected signal that is to appear in the final effect
output mix.

When set to 0%, the output is taken only from the input (dry).
When set to 100%, the output is all wet.

Negative values polarity invert the wet signal.

Out Gain The overall gain or amplitude at the output of the effect.
Amount Adjusts the shaper intensity.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

CrossOver1

17 to 25088 Hz
Adjusts one of the -6dB crossover points at which the input signal will be divided into the high, mid and low bands.

CrossOver2

17 to 25088 Hz
Adjusts the other -6dB crossover points at which the input signal will be divided into the high, mid and low bands.

Lo Enable
Mid Enable
Hi Enable

On or Off
Turns processing for each band on or off.

Turning each of the three bands Off results in a dry output signal.

Lo Amt
Mid Amt
Hi Amt

0.10 to 6.00x
Adjusts the shaper intensity for each band.

Lo Mix
Mid Mix
Hi Mix

-100 to 100%
Adjusts the level that each band is summed together as the wet signal.

Negative values polarity invert the particular bands signal.

Quantize + Alias

Effects Size : 1

Digital quantization followed by simulated aliasing

Quantize + Alias Presets
329 Aliaser

The Quantize+Alias algorithm offers some of the worst artifacts that digital has to offer!

Digital audio engineers will go to great lengths to remove, or at least hide the effects of digital quantization distortion and sampling aliasing.

In Quantize+Alias we do quite the opposite, making both quantization and aliasing in-your-face effects.

The quantizer will give your sound a dirty, grundgy, perhaps industrial sound.

The aliasing component simulates the effect of having sampled a sound without adequately band limiting the signal (anti-alias filtering).

Quantization Distortion: Is a digital phenomenon caused by having only a limited number of bits with which to represent signal amplitudes (finite precision).

You are probably aware that a bit is a number which can have only one of two values: 0 or 1.

When we construct a data or signal word out of more than one bit, each additional bit will double the number of possible values.

For example a two bit number can have one of four different values: 00, 01, 10 or 11.

A three bit number can take one of eight different values, a four bit number can take one of sixteen values, etc.

An 24-bit digital-to-analog converter (DAC) like the one in the Forte & PC4 can interpret 16,777,216 different amplitude levels (2²⁴).

Let’s take a look at how finite precision of digital words affects audio signals.

The figures below are plots of a decaying sine wave with varying word lengths.

Figure 11. A decaying sine wave represented with different word lengths

Quantization

Clearly a one-bit word gives a very crude approximation to the original signal while four bits is beginning to do a good job of reproducing the original decaying sine wave.

When a good strong signal is being quantized (its word length is being shortened), quantization usually sounds like additive noise.

But notice that as the signal decays in the above figures, fewer and fewer quantization levels are being exercised until, like the one bit example, there are only two levels being toggled.

With just two levels, your signal has become a square wave.

Dynamic Range: Controlling the bit level of the quantizer is done with the DynamRange parameter (dynamic range).

At 0 dB we are at a one-bit word length.

Every 6 dB adds approximately one bit, so at 144 dB, the word length is 24 bits.

The quantizer works by cutting the gain of the input signal, making the lowest bits fall off the end of the word.

The signal is then boosted back up so we can hear it.
Headroom: At very low DynamRange settings, the step from one bit level to the next can become larger than the input signal.

The signal can still make the quantizer toggle between bit level whenever the signal crosses the zero signal level, but with the larger bit levels, the output will get louder and louder.

The Headroom parameter prevents this from happening.

When the DynamRange parameter is lower than the Headroom parameter, no more signal boost is added to counter-act the cut used to quantize the signal.

Find the DynamRange level at which the output starts to get too loud, then set Headroom to that level.

You can then change the DynamRange value without worrying about changing the signal level.

Headroom is a parameter that you set to match your signal level, then leave it alone.

DC Offset: At very low DynamRange values, the quantization becomes very sensitive to DC offset.

It affects where your signal crosses the digital zero level.

A DC offset adds a constant positive or negative level to the signal.

By adding positive DC offset, the signal will tend to quantize more often to a higher bit level than to a lower bit level.

In extreme cases (which is what we’re looking for, after all), the quantized signal will sputter, as it is stuck at one level most of the time, but occasionally toggles to another level.
Aliasing: Aliasing is an unwanted artifact (usually!) of digital sampling.

It’s an established rule in digital sampling that all signal frequency components above half the sampling frequency (the Nyquist rate) must be removed with a lowpass filter (anti-aliasing filter).

If frequencies above the Nyquist rate are not removed, you will hear aliasing.

A digital sampler cannot represent frequencies above the Nyquist rate, but rather than remove the high frequencies, the sampler folds the high frequencies back down into the lower frequencies where they are added to the original low frequencies.

If you were to play a rising pure tone through a sampler without an anti-alias filter, you would hear the tone start to fall when it past the Nyquist rate.

The pitch will continue to drop as the input tone’s frequency increases until the input tone reaches the sampling rate.

The sampled tone would then have reached dc (frequency is 0) and will start to rise again.

Usually a lowpass anti-aliasing filter is placed before the sampler to prevent this from happening.

In the Quantize+Alias algorithms, we do not actually sample the incoming signal at a lower rate.

Instead we use a special modulation algorithm to simulate the effect of pitches falling when they should be rising.

Pitch: The Pitch (coarse and fine) parameters roughly correspond to setting the Nyquist frequency.

Higher pitches result in modulating your input signal with higher frequencies.

The LFO Depth parameter changes the strength of the modulation.

Larger values of LFO Depth produce a deeper modulation which may be considered analogous to inputting a insufficiently band-limited signal for sampling.

In/Out

In or Out

When set to In, the quantizer and aliaser are active
When set to Out, the quantizer and aliaser are bypassed

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

DynamRange

0 to 144 dB
The digital dynamic range controls signal quantization, or how many bits to remove from the signal data words.

At 0 dB the hottest of signals will toggle between only two bit (or quantization) levels.

Every 6 dB added doubles the number of quantization levels.

If the signal has a lot of headroom (available signal level before digital clipping), then not all quantization levels will be reached.

Headroom

0 to 144 dB
When the signal has a lot of headroom (available signal level before digital clipping), turning down DynamRange can cause the amplitude of adjacent quantization levels to exceed the input signal level.

This causes the output to get very loud.

Set Headroom to match the amount of digital signal level still available (headroom).

This is easily done by finding the DynamRange level at which the signal starts getting louder and matching Headroom to that value.

dc Offset

-79.0 to 0.0 dB
Adds a positive DC offset to the input signal.

dc Offset is expressed in decibels (dB) relative to full-scale digital.

By adding DC offset, you can alter the position where digital zero is with respect to you signal.

At low DynamRange settings, adding DC offset may cause output sputter.

Quant W/D

0 to 100% wet
The relative amount of quantized (wet) to unaffected (dry) signal.

At 100%, you hear only the quantized signal.

Alias W/D

0 to 100% wet
Amount of aliaser output signal (wet) relative to aliaser input signal (dry) to send to the final output.

The dry signal here is taken to mean the output of the quantizer.

Lowpass

8 to 25088 Hz
Lowpass anti-aliasing filter.

Larger values produce a more extreme modulation effect.

Pitch Crs

8 to 25088 Hz
Pitch sets the frequency (coarse) at which the input signal is modulated.

Higher pitches produce a high frequency modulation.

Pitch Fine

-100 to 100 ct
Pitch sets the frequency (fine) at which the input signal is modulated.

Higher pitches produce a high frequency modulation.

LFO Depth

1 to 49 samp
The depth of the modulation, controlling how strong the modulation sounds.

Filters

Env Follow Filt

Effects Size : 2

Envelope following stereo two-pole resonant filter

Env Follow Filt Presets
356 Basic Env Filter	358 Synth Env Filter	360 EPno Env Filter
357 Phunk Env Filter	359 Bass Env Filter

The envelope following filter is a stereo resonant filter with the resonant frequency controlled by the envelope of the input signal (the maximum of left or right).

The filter type is selectable and may be

Lowpass
Highpass
Bandpass
Notch

Figure 12. Resonant Filters

The resonant frequency of the filter will remain at the minimum frequency (Min Freq) as long as the signal envelope is below the Threshold.

The Freq Sweep parameter controls how much the frequency will change with changes in envelope amplitude.

The frequency range is 0 to 8372 Hz, though the minimum setting for Min Freq is 8 Hz.

The term minimum frequency is a reference to the resonant frequency at the minimum envelope level; with a negative Freq Sweep, the filter frequency will sweep below the Min Freq.

A meter is provided to show the current resonance frequency of the filter.

The filter Resonance level may be adjusted.

The resonance is expressed in decibels (dB) of gain at the resonant frequency.

Since 50 dB of gain is available, you will have to be careful with your gain stages to avoid clipping.

The attack and release rates of the envelope follower are adjustable.

The rates are expressed in decibels per second (dB/s).

The envelope may be smoothed by a lowpass filter which can extend the attack and release times of the envelope follower.

A level meter with a threshold marker is provided.

Figure 13. Envelope Follower Filter

Wet/Dry

0 to 100%wet
The amount of modulated (wet) signal relative to unaffected (dry) signal as a percent.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

FilterType

Lowpass, Highpass, Bandpass, or Notch
The type of resonant filter to be used.

Min Freq

8 to 8372 Hz
The base frequency of the resonant filter.

The filter resonant frequency is set to the Min Freq while the signal envelope is at its minimum level or below the threshold.

Freq Sweep

-100 to 100%
How far the filter frequency can change from the Min Freq setting as the envelope amplitude changes.

Freq Sweep may be positive or negative so the filter frequency can rise above or fall below the Min Freq setting.

Resonance

0 to 50 dB
The resonance level of the resonant filter.

Resonance sets the level of the resonant peak (or the amount of cut in the case of the notch filter).

Threshold

-79.0 to 0.0 dB
The level above which signal envelope must rise before the filter begins to follow the envelope.

Below the threshold, the filter resonant frequency remains at the Min frequency.

Atk Rate

0.0 to 300.0 dB/s
Adjusts the upward slew rate of the envelope detector.

Rel Rate

0.0 to 300.0 dB/s
Adjusts the downward slew rate of the envelope detector.

Smth Rate

0.0 to 300.0 dB/s
Smooths the output of the envelope follower.

Smoothing slows down the envelope follower and can dominate the attack and release rates if set to a lower rate than either of these parameters.

LFO Sweep Filter

Effects Size : 2

LFO following stereo two-pole resonant filter

LFO Sweep Filter Presets
362 LFO Sweep Filter	365 TripFilter
363 DoubleRiseFilter
364 Circle Bandsweep

The LFO following filter is a stereo resonant filter with the resonant frequency controlled by an LFO (low-frequency oscillator).

The filter type is selectable and may be

Lowpass
Highpass
Bandpass
Notch

Figure 14. Resonant Filters

The resonant frequency of the filter will sweep between the minimum frequency (Min Freq) and the maximum frequency (Max Freq).

The minimum and maximum frequencies may be set to any combination of frequencies between 8 and 8372 Hz.

The terms minimum and maximum frequency are a reference to the resonant frequencies at the minimum and maximum envelope levels; you may set either of the frequencies to be larger than the other, though doing so will just invert the direction of the LFO.

Meters are provided to show the current resonance frequencies of the left and right channel filters.

Resonance: The filter Resonance level may be adjusted.

The resonance is expressed in decibels (dB) of gain at the resonant frequency.

Since 50 dB of gain is available, you will have to be careful with your gain stages to avoid clipping.
LFO Frequency: You can set the frequency of the LFO using the LFO Tempo and LFO Period controls.

You can explicitly set the tempo or use the system tempo from the sequencer (or MIDI clock).

The LFO Period control sets the period of the LFO (the time for one complete oscillation) in terms of the number of tempo beats per LFO period.
LFO Wave Shapes: The LFO may be configured to one of a variety of wave shapes.

Available shapes are:

Sine
Sine is simply a sinusoid waveform.
Saw+ and Saw–
Saw+ and Saw– produce rising and falling sawtooth waveforms
Pulse
Pulse produces a series of square pulses where the pulse width can be adjusted with the LFO PlsWid parameter.

When pulse width is 50%, the signal is a square wave.

The LFO PlsWid parameter is active only when the Pulse waveform is selected.

Tri.
Tri produces a triangular waveform

Smoothing

The Pulse and Saw waveforms have abrupt, discontinuous changes in amplitude which can be smoothed.

The pulse wave is implemented as a hard clipped sine wave, and, at 50% width, it turns into a sine wave when set to 100% smoothing.

The sudden change in amplitude of the sawtooth waves develops a more gradual slope with smoothing, ending up as triangle waves when set to 100% smoothing.

Figure 15. Configurable Wave Shapes

Wet/Dry

0 to 100% wet
The amount of modulated (wet) signal relative to unaffected (dry) signal as a percent.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

LFO Tempo

System, 1 to 255 BPM
Basis for the rates of the LFO, as referenced to a musical tempo in bpm (beats per minute).

When this parameter is set to System, the tempo is locked to the internal sequencer tempo or to incoming MIDI clocks.

In this case, FXMods (FUNs, LFOs, ASRs etc.) will have no effect on the Tempo parameter.

LFO Period

1/24 to 32 bts
Sets the LFO rate based on the Tempo determined above: the number of beats corresponding to one period of the LFO cycle.

For example, if the LFO Period is set to 4, the LFOs will take four beats to pass through one oscillation, so the LFO rate will be 1/4th of the Tempo setting.

If it is set to 6/24 (=1/4), the LFO will oscillate four times as fast as the Tempo.

At 0, the LFOs stop oscillating and their phase is undetermined (wherever they stopped).

LFO Shape

Sine, Saw+, Saw–, Pulse, and Tri
The waveform type for the LFO.

LFO PlsWid

0 to 100%
When the LFO Shape is set to Pulse, the PlsWid parameter sets the pulse width as a percentage of the waveform period.

The pulse is a square wave when the width is set to 50%.

This parameter is active only when the Pulse waveform is selected.

LFO Smooth

0 to 100%
Smooths the Saw+, Saw–, and Saw– waveforms.

For the sawtooth waves, smoothing makes the waveform more like a triangle wave.
For the Pulse wave, smoothing makes the waveform more like a sine wave.

FilterType

Lowpass, Highpass, Bandpass, or Notch
The type of resonant filter to be used.

Min Freq

8 to 8372 Hz
The minimum frequency of the resonant filter.

This is the resonant frequency at one of the extremes of the LFO sweep.

The resonant filter frequency will sweep between the Min Freq and Max Freq.

Max Freq

8 to 8372 Hz
The maximum frequency of the resonant filter.

This is resonant frequency at the other extreme of the LFO sweep.

The resonant filter frequency will sweep between the Min Freq and Max Freq.

Resonance

0 to 50 dB
The resonance level of the resonant filter.

Resonance sets the level of the resonant peak (or the amount of cut in the case of the notch filter).

L Phase

0.0 to 360.0 deg
The phase angle of the left channel LFO relative to the system tempo clock and the right channel phase.

R Phase

0.0 to 360.0 deg
The phase angle of the right channel LFO relative to the system tempo clock and the left channel phase.

Resonant Filter

Effects Size : 1

Stereo 2 pole resonant filter

Resonant Filter Presets
366 Resonant Filter

Resonant Filter is a stereo (linked parameters for left and right) resonating filter.

You can adjust the resonant frequency of the filter and the filter resonance level.

The filter type is selectable and may be

Lowpass
Highpass
Bandpass
Notch

Figure 16. Resonant Filters

Wet/Dry

0 to 100%wet
The amount of filtered (wet) signal relative to unaffected (dry) signal.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the filter.

FilterType

Lowpass, Highpass, Bandpass, or Notch
The type of resonant filter to be used.

Frequency

58 to 8372 Hz
The frequency of the resonant filter peak (or notch) in Hz.

The frequencies correspond to semitone increments.

Resonance

0 to 50 dB
The resonance level of the resonant filter.

Resonance sets the level of the resonant peak (or the amount of cut in the case of the notch filter).

Dual Res Filter

Effects Size : 1

Dual Mono 2 pole resonant filter

Dual Res Filter Presets
367 Dual Res Filter

Dual Res Filter is a dual mono (independent
controls for left and right) resonating filter.

You can adjust the resonant frequency of the filter and the filter resonance level.

The filter type is selectable and may be

Lowpass
Highpass
Bandpass
Notch

Figure 17. Resonant Filters

L Wet/Dry
R Wet/Dry

0 to 100%wet
The amount of filtered (wet) signal relative to unaffected (dry) signal.

L Output
R Output

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the filter.

L FiltType
R FiltType

Lowpass, Highpass, Bandpass, or Notch
The type of resonant filter to be used.

L Freq
R Freq

58 to 8372 Hz
The frequency of the resonant filter peak (or notch) in Hz.

The frequencies correspond to semitone increments.

LResonance
RResonance

0 to 50 dB
The resonance level of the resonant filter.

Resonance sets the level of the resonant peak (or the amount of cut in the case of the notch filter).

EQ Morpher

Effects Size : 4

Parallel resonant bandpass filters with parameter morphing

EQ Morpher Presets
390 EQ Morpher ah-oo
391 EQ Morpher ee-aa
392 EQ Morpher aw-er

The EQ Morpher algorithms have four parallel bandpass filters acting on the input signal and the filter results are summed for the final output.

EQ Morpher is a stereo algorithm for which the left and right channels receive separate processing using the same linked controls.

In EQ Morpher, a stereo panner that includes a width parameter to control the width of the stereo field.

Filter: For each filter, there are two sets of parameters, A and B.

The parameter Morph A>B determines which parameter set is active.

When Morph A>B is set to 0%, you are hearing the A parameters; when set to 100%, you are hearing the B parameters.

The filters may be gradually moved from A to B and back again by moving the Morph A>B parameter between 0 and 100%.

The four filters are parametric bandpass filters.

These are not standard parametric filters, which boost or cut the signal at the frequency you specify relative to the signal at other frequencies.

The bandpass filters used here pass only signals at the frequency you specify and cut all other frequencies.

The gain controls for the filters set the levels of each filter’s output.

Like the normal parametric filters, you have control of the filters’ frequencies and bandwidths.

The Freq Scale parameters may be used to adjust the A or B filters’ frequencies as a group.

This allows you to maintain a constant spectral relationship between your filters while adjusting the frequencies up and down.

The filters are arranged in parallel and their outputs summed, so the bandpass peaks are added together and the multiple resonances are audible.

Figure 18. Frequency Response

EQ Morpher can do an excellent job of simulating the resonances of the vocal tract.

Example: A buzz or sawtooth signal is a good choice of source material to experiment with the EQ Morpher.

Set the Morph A>B parameter to 0%, and find a combination of A filter settings which give an interesting vowel like sound.

It may help to start from existing ROM presets.

Next set Morph A>B to 100% and set the B parameters to a different vowel-like sound.

You can now set up some FXMods on Morph A>B to morph between the two sets of parameters, perhaps using Freq Scale to make it more expressive.

When morphing from the A parameters to the B parameters, A Filter 1 moves to B Filter 1, A Filter 2 moves to B Filter 2, and so on.

For the most normal and predictable results, it’s a good idea not to let the frequencies of the filters cross each other during the morphing.

You can ensure this doesn’t happen by making sure the four filters are arranged in ascending order of frequencies.

Descending order is okay too, provided you choose an order and stick to it.

In/Out

In or Out
When set to In the algorithm is active.
When set to Out the algorithm is bypassed.

Out Gain

Off, -79.0 to 24.0 dB
An overall level control of the EQ Morpher output.

Out Pan

-100 to 100%
Provides panning of the output signal between left and right output channels.

This is a stereo panner which pans the entire stereo image.

-100% is panned left
100% is panned right

Out Width

-100 to 100%
The width of the stereo field is controlled by this parameter.

A setting of 100% is the same full width as the input signal.
At 0% the left and right channels are narrowed to the point of being mono.
Negative values reverse the left and right channels.

Morph A>B

0 to 100%
When set to 0% the A parameters are controlling the filters, and when set to 100%, the B parameters control the filters.

Between 0 and 100%, the filters are at interpolated positions.

When morphing from the A to the B settings, A Filter 1 moves to B Filter 1, A Filter 2 moves to B Filter 2, and so on.

A FreqScale
B FreqScale

-8600 to 8600 ct
The filter frequencies for the A and B parameter sets may be offset with the FreqScale parameters.

After setting the filter parameters, the FreqScale parameters will move each of the four filter frequencies together by the same relative pitch.

A Freq 1
A Freq 2
A Freq 3
A Freq 4
B Freq 1
B Freq 2
B Freq 3
B Freq 4

8 to 25088 Hz
The center frequency of the bandpass filter peak in Hz.

This frequency may be offset by the FreqScale parameter.

A Width 1
A Width 2
A Width 3
A Width 4
B Width 1
B Width 2
B Width 3
B Width 4

0.010 to 5.000 oct
The bandwidth of the bandpass filter in octaves.

Narrow bandwidths provide the most convincing vocal sounds.

A Gain 1
A Gain 2
A Gain 3
A Gain 4
B Gain 1
B Gain 2
B Gain 3
B Gain 4

-79.0 to 24.0 dB
The level of the bandpass filter output.
At 0 dB, a sine wave at the same frequency as thefilter will be neither boost not cut.

At settings greater than 0 dB, the (hypothetical) sine wave is boosted, and below 0 dB the sine wave is cut.

Signals at frequencies other than the filter frequency are always cut more than a signal at the filter frequency.

The amount that other frequencies are cut depends on the bandwidth of the bandpass filter.

Switch Loops

Effects Size : 2

Looped tempo sync delay lines with input switching

Switch Loops Presets
188 News Update

Switch Loops allows you to run up to four parallel recirculating delay lines of different lengths, switching which delay line(s) are receiving the input signal at a given moment.

Stereo Input: The stereo input is summed to mono and sent to any of the four delay lines.

You can select which delay lines are receiving input with the DlySelect parameters.
Gain: The gain in decibels of each of the four delays can be set individually.
DecayRate: The amount of feedback to apply to each delay is set with a DecayRate parameter.

The DecayRate controls how many decibels the signal will be reduced for every second the signal is recirculating in the delay.

The length of the delays are set based on tempo (system tempo or set locally) and duration in beats.

Assuming a 4/4 time signature with tempo beats on the quarter note, 8/24 bts is an eighth triplet (8/24 equals 1/3 of a quarter note), 12/24 bts is an eighth, 16/24 bts is a quarter triplet, and 1 bts is a quarter note duration.

Dividing the quarter note into 24ths, allows delay lengths based on the most common note lengths.

To determine a delay length in seconds, divide the length of the delay (in beats) by the tempo and multiply by 60 seconds/minute.
\(T=beats/tempo*60\)
HF Damping: HF Damping controls a one pole lowpass filter on each of the delay lines.
Maximum Feedback: Max Fdbk overrides all of the DecayRate parameters and prevents the signals in the delay lines from decaying at all.
Feedback Kill: Fdbk Kill will override the DecayRate parameters and the Max Fdbk parameter by completely turning of the feedback for all the delays.

Fdbk Kill stops all the delay line recirculation.

The outputs of all the delay lines are summed, and the output gain is applied to the mono result which can be panned between the two output channels.

Figure 19. Switch Loops - Block Diagram

Dry In/Out

In or Out
If set to In, Dry In/Out allows the dry input signal to be added to the final algorithm output.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

Dry Gain

Off, -79.0 to 24.0 dB
If Dry In/Out is In, then Dry Gain controls the level of the dry input signal that is summed to the final algorithm output.

Fdbk Kill

On or Off
Forces the delay recirculation of all delay lines to stop by turning off the delay line feedback.

Fdbk Kill provides a quick way to silence the algorithm to start over with new sounds in the delays.

Fdbk Kill overrides the Max Fdbk and DecayRate parameters.

Max Fdbk

On or Off
Prevents the recirculating delay lines from decaying by turning the delay line feedback fully on.

Max Fdbk overrides the DecayRate parameters, but does not function when Fdbk Kill is On.

Tempo

System, 1 to 255 BPM
Tempo is the basis for the delay lengths, as referenced to a musical tempo in bpm (beats per minute).

When this parameter is set to System, the tempo is locked to the internal sequencer tempo or to incoming MIDI clocks.

In this case, FXMods (FUNs, LFOs, ASRs etc.) will have no effect on the Tempo parameter.

Pan

-100 to 100 %
The summed mono signal from the delay lines may be panned between left and right output channels.

-100% is panned fully left
0% is centered
100% is fully right.

HF Damping

8 to 25088 Hz
The -3 dB frequency in Hz of a one-pole lowpass filter (-6 dB/octave) placed in the feedback path of each delay line.

Multiple passes through the feedback will cause the signal to become more and more dull.

DlySelect1
DlySelect2
DlySelect3
DlySelect4

Off, A, B, C, D
You select which delay lines (A, B, C, or D) receive the mono input signal with the DlySelect (1, 2, 3, or 4) parameters.

Since there are four delay lines, you can turn on none, 1, 2, 3, or 4 of the delay lines.

All four of the DlySelect parameters are equivalent, it doesn’t matter which you use.

If you turn on a particular delay line in more than one DlySelect parameter, it’s the same as turning it on in just one DlySelect parameter.

Dly Len A
Dly Len B
Dly Len C
Dly Len D

0 to 32 bts
The delay length of the delay line.

If the DecayRate for the delay is low or Max Fdbk is On, this parameter sets the repeating delay loop length for this delay.

The delay length is specified as a fraction or multiple of the tempo, in “beats.”

The length of a delay loop in seconds can be calculated from beats as:
\(T=beats/tempo*60\)

DecayRateA
DecayRateB
DecayRateC
DecayRateD

0.0 to 230.0 dB/s
The rate at which the delay line will decay or reduce in level.

DecayRate controls a feedback level which is calculated based on DecayRate and Dly Len.

By basing the feedback gain on DecayRate, all four of the delay lines can decay at the same rate in spite of differing delay lengths.

DecayRate is expressed as decibels of signal reduction per second.

Gain A
Gain B
Gain C
Gain D

Off, -79.0 to 24.0 dB
The level of the delay output tap expressed in decibels.

Degen Regen

Degen Regen starts as a simple mono delay line with feedback.

However with the Fdbk Gain and Dist Drive parameters, the algorithm can be pushed hard into instability.

When Degen Regen is unstable, your sound gets a little louder on each pass through the delay line.

Eventually the sound will hit digital clipping when the effects processor runs out of headroom bits.

Compressor: To keep this all under control, a soft-knee compressor has been included inside the delay line loop.

With the compressor properly set, the sound never reaches digital clipping, but it does become more and more distorted as it gets pushed harder and harder into the compressor.

For details about the compressor see SoftKneeCompress.

Distortion: To make things really nasty, there’s also a distortion in the delay path.

For details about the distortion see Mono Distortion.

Degen Regen lets you set the longest mono delay line available which is just over 20 seconds.

If you want a long delay, this is the algorithm to do it.

(You don’t have to over-drive the feedback or use the distortion.)

Output Taps

The delay has two output taps in addition to the feedback tap.

Each tap may be moved along the delay line using an LFO (internal to the effects processor).

The output taps have separate controls for level and panning (in the stereo configurations).

Throw a few filters into the delay line loop, and you get a pretty versatile delay line.

The available filters are:

highpass (LF Damping)
lowpass (HF Damping)
bass shelf
treble shelf
and two parametric EQs (Mid1, Mid2).

Degen Regen BPM

Effects Size : 4

Long tempo synced delay allowing loop instability

Degen Regen BPM Presets
182 Ecko Plecks BPM	185 Nanobot Feedback	187 Wait for UFO
184 Degenerator	186 Takes a while…	306 Superphasulate

Figure 20. Degen Regen BPM - Block Diagram

Wet/Dry

-100 to 100% wet
The relative amount of input signal and delay signal that is to appear in the final effect output mix.

When set to 0%, the output is taken only from the input (dry).
When set to 100%, the output is all wet.

Out Gain

Out Gain Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

Send Gain

Out Gain Off, -79.0 to 24.0 dB
The input gain or amplitude to the Degen Regen delay loop.

Loop Gain

Out Gain Off, -79.0 to 24.0 dB
Controls the signal level of the signal which is fed back to the input of the delay line.

If other elements of Degen Regen were removed (set flat), then Loop Gain would cause the algorithm to become unstable above 0 dB.

However other parameters interact resulting in a more complex gain structure.

See also Loop Lvl.

Loop Lvl

-100 to 100%
A convenience parameter which may be used to reduce the Fdbk Gain feedback strength.

It may be helpful if you are used to dealing with feedback as a linear (percent) control.

At 100%, the feedback strength is as you have it set with Loop Gain.
Lower levels reduce the feedback signal, so at 50% the feedback signal is reduced by -6 dB from the selected Loop Gain level.
Negative values polarity invert the feedback loop signal.

Tempo

System, 1 to 255 BPM
Tempo is the basis for the delay lengths, as referenced to a musical tempo in bpm (beats per minute).

When this parameter is set to System, the tempo is locked to the internal sequencer tempo or to incoming MIDI clocks.

In this case, FXMods (FUNs, LFOs, ASRs etc.) will have no effect on the Tempo parameter.

LF Damping

8 to 25088 Hz
The -3 dB frequency in Hz of a one-pole highpass filter (6 dB/octave) placed in the feedback path of the delay line.

The signal does not go through the filter the first time through the delay line.

Multiple passes through the feedback will cause the signal to become more and more bright (removing low frequencies).

HF Damping

8 to 25088 Hz
The -3 dB frequency in Hz of a one-pole lowpass filter (-6 dB/octave) placed in the feedback path of the delay line.

The signal does not go through the filter the first time through the delay line.

Multiple passes through the feedback will cause the signal to become more and more dull.

LoopLength

0 to 32 bts
The delay length of the feedback tap.

If feedback is turned up from 0%, this parameter sets the repeating delay loop length.

The loop length is specified as a fraction or multiple of the tempo, in “beats.” The length of a delay loop in seconds can be calculated from beats as :
\(T=beats/tempo*60\)

LFO Period

1/24 to 32 bts
The feedback tap and the output taps lengths can be modulated with an LFO internal to the effects processor.

The rate at which the tap positions move are tied to a common period control (time for one complete cycle) which is expressed in beats.

The depth of modulation is specified by the LpLFODepth parameter.

Frequency in Hz can be calculated from the period in beats as:
\(F=tempo/(beats*60)\)

Since this moving delay tap is part of the feedback path through the delay, subsequent passes of the signal through the delay may result in some strange pitch modulations.

It is possible to set LFO Period with LoopLength so that alternate passes through the loop detune then retune the signal (for example, set the LFO period to double the LoopLength).

The maximum pitch shift up is not identical to the maximum pitch shift down, so the alternating detune/retune effect is not perfect.

Bass Gain

-79.0 to 24.0 dB
The amount of boost or cut in dB that the bass shelving filter should apply to the low frequency signal components.

Every increase of 6 dB approximately doubles the amplitude of the signal.

Positive values boost the bass signal below the specified frequency.
Negative values cut the bass signal below the specified frequency.

Since the filters are in the delay feedback loop, the cut or boost is cumulative on each pass the sound makes through the loop.

Bass Freq

8 to 25088 Hz
The center frequency of the bass shelving filter in intervals of one semitone.

Treb Gain

-79.0 to 24.0 dB
The amount of boost or cut in dB that the treble shelving filter should apply to the high frequency signal components.

Every increase of 6 dB approximately doubles the amplitude of the signal.

Positive values boost the treble signal above the specified frequency.
Negative values cut the treble signal above the specified frequency.

Since the filters are in the delay feedback loop, the cut or boost is cumulative on each pass the sound makes through the loop.

Treb Freq

8 to 25088 Hz
The center frequency of the treble shelving filter in intervals of one semitone.

Mid1 Gain
Mid2 Gain

-79.0 to 24.0 dB
The amount of boost or cut in dB that the parametric filter should apply to the specified signal band.

Every increase of 6 dB approximately doubles the amplitude of the signal.

Positive values boost the signal at the specified frequency.
Negative values cut the signal at the specified frequency.

Since the filters are in the delay feedback loop, the cut or boost is cumulative on each pass the sound makes through the loop.

Mid1 Freq
Mid2 Freq

Mid1 Freq
The center frequency of the parametric EQ in intervals of one semitone.

The boost or cut will be a maximum at this frequency.

Mid1 Width
Mid1 Width

0.010 to 5.000 oct
The bandwidth of the parametric EQ may be adjusted.

You specify the bandwidth in octaves.

Small values result in a very narrow (high-Q) filter response.
Large values result in a very broad response.

LpLFODepth

0.0 to 230.0 ct
The feedback (loop) delay tap will have its position modulated by an LFO (internal to the FX processor) if the LpLFODepth parameter is non-zero.

A moving tap on a delay line will result in a pitch shift, and LpLFODepth sets the maximum pitch shift (up and down) in cents.

LpLFOPhase

0.0 to 360.0 deg
Specifies the phase angle of the feedback (loop) LFO relative to the output tap LFOs and the system (or MIDI) tempo clock, if turned on (see Tempo).

Example: If one LFO is set to 0° and another is set to 180°, then when one LFO delay tap is at its shortest, the other will be at its longest.

If the system (or MIDI) tempo clock is turned on, the LFOs are synchronized to the clock with absolute phase.

T1LFODepth
T2LFODepth

0.0 to 230.0 ct
The output delay taps (1 and 2) will have their positions modulated by an LFO (internal to the FX processor) if the T1/2LFODepth parameter is non-zero.

A moving tap on a delay line will result in a pitch shift, and T1/2LFODepth sets the maximum pitch shift (up and down) in cents.

T1LFOPhase
T2LFOPhase

0.0 to 360.0 deg
Specifies the phase angle of the output LFO tap (1 or 2) relative to the other output LFO tap, the feedback (loop) LFO tap, and the system (or MIDI) tempo clock, if turned on (see Tempo).

Example: If one LFO is set to 0° and another is set to 180°, then when one LFO delay tap is at its shortest, the other will be at its longest.

If the system (or MIDI) tempo clock is turned on, the LFOs are synchronized to the clock with absolute phase.

Tap1 Delay
Tap2 Delay

0 to 32 bts
The delay length of the output tap 1 or 2.

The tap length is specified as a fraction or multiple of the tempo, in “beats.”

The length of a delay tap in seconds can be calculated from beats as
\(T=(beats/tempo)/60\)

Tap1 Level
Tap2 Level

0 to 100 %
The level of the output tap 1 or 2 expressed as a percent.

Tap1 Pan
Tap2 Pan

-100 to 100%
The output taps 1 and 2 are mono sources that can be panned to the left or right output channels.

-100% is panned fully left
100% is panned fully right.

Comp Atk

0.0 to 228.0 ms
The time for the compressor to start to cut in when there is an increase in signal level (attack) above the threshold.

Comp Rel

0 to 3000 ms
The time for the compressor to stop compressing when there is a reduction in signal level (release) from a signal level above the threshold.

CompSmooth

0.0 to 228.0 ms
A lowpass filter in the compressor control signal path.

It is intended to smooth the output of the expander’s envelope detector.

Smoothing will affect the attack or release times when the smoothing time is longer than one of the other times.

Comp Ratio

1.0:1 to 100.0:1, Inf:1
The compression ratio.

High ratios are highly compressed
Low ratios are moderately compressed

Comp Thres

-79.0 to 0.0 dB
The threshold level in dBFS (decibels relative to full scale) above which the signal begins to be compressed.

Dist Drive

0 to 96 dB
Applies a boost to the feedback signal to overdrive the distortion algorithm.

When overdriven, the distortion algorithm will soft-clip the signal.

Since distortion drive will make your signal very loud, you may have to reduce the feedback amount or turn on the compressor as the drive is increased.

DistWarmth

8 to 25088 Hz
A lowpass filter in the distortion control path.

This filter may be used to reduce some of the harshness of some distortion settings without reducing the bandwidth of the signal.

Degen Regen

Effects Size : 4

Long delay allowing loop instability

Degen RegenPresets
163 Better Tape Echo
164 Stereo Tape Slap
183 Ecko Plecks ms

Wet/Dry

-100 to 100% wet
The relative amount of input signal and delay signal that is to appear in the final effect output mix.

When set to 0%, the output is taken only from the input (dry).
When set to 100%, the output is all wet.

Out Gain

Out Gain Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

Send Gain

Out Gain Off, -79.0 to 24.0 dB
The input gain or amplitude to the Degen Regen delay loop.

Loop Gain

Loop Lvl

-100 to 100%
A convenience parameter which may be used to reduce the Fdbk Gain feedback strength.

It may be helpful if you are used to dealing with feedback as a linear (percent) control.

At 100%, the feedback strength is as you have it set with Loop Gain.
Lower levels reduce the feedback signal, so at 50% the feedback signal is reduced by -6 dB from the selected Loop Gain level.
Negative values polarity invert the feedback loop signal.

LF Damping

HF Damping

LoopLength

0.00 to 21.5 s
The delay length of the feedback tap.

If feedback is turned up from 0%, this parameter sets the repeating delay loop length.

LFO Rate

0.00 to 10.00 Hz
The feedback tap and the output taps lengths can be modulated with an LFO internal to the effects processor.

The rate at which the tap positions move are tied to a common rate control which is expressed in Hz.

The depth of modulation is specified by the LpLFODepth parameter.

Bass Gain

-79.0 to 24.0 dB
The amount of boost or cut in dB that the bass shelving filter should apply to the low frequency signal components.

Every increase of 6 dB approximately doubles the amplitude of the signal.

Positive values boost the bass signal below the specified frequency.
Negative values cut the bass signal below the specified frequency.

Since the filters are in the delay feedback loop, the cut or boost is cumulative on each pass the sound makes through the loop.

Bass Freq

8 to 25088 Hz
The center frequency of the bass shelving filter in intervals of one semitone.

Treb Gain

-79.0 to 24.0 dB
The amount of boost or cut in dB that the treble shelving filter should apply to the high frequency signal components.

Every increase of 6 dB approximately doubles the amplitude of the signal.

Positive values boost the treble signal above the specified frequency.
Negative values cut the treble signal above the specified frequency.

Since the filters are in the delay feedback loop, the cut or boost is cumulative on each pass the sound makes through the loop.

Treb Freq

8 to 25088 Hz
The center frequency of the treble shelving filter in intervals of one semitone.

Mid1 Gain
Mid2 Gain

-79.0 to 24.0 dB
The amount of boost or cut in dB that the parametric filter should apply to the specified signal band.

Every increase of 6 dB approximately doubles the amplitude of the signal.

Positive values boost the signal at the specified frequency.
Negative values cut the signal at the specified frequency.

Since the filters are in the delay feedback loop, the cut or boost is cumulative on each pass the sound makes through the loop.

Mid1 Freq
Mid2 Freq

Mid1 Freq
The center frequency of the parametric EQ in intervals of one semitone.

The boost or cut will be a maximum at this frequency.

Mid1 Width
Mid1 Width

0.010 to 5.000 oct
The bandwidth of the parametric EQ may be adjusted.

You specify the bandwidth in octaves.

Small values result in a very narrow (high-Q) filter response.
Large values result in a very broad response.

LpLFODepth

LpLFOPhase

0.0 to 360.0 deg
Specifies the phase angle of the feedback (loop) LFO relative to the output tap LFOs and the system (or MIDI) tempo clock, if turned on (see Tempo).

Example: If one LFO is set to 0° and another is set to 180°, then when one LFO delay tap is at its shortest, the other will be at its longest.

If the system (or MIDI) tempo clock is turned on, the LFOs are synchronized to the clock with absolute phase.

T1LFODepth
T2LFODepth

T1LFOPhase
T2LFOPhase

Example: If one LFO is set to 0° and another is set to 180°, then when one LFO delay tap is at its shortest, the other will be at its longest.

If the system (or MIDI) tempo clock is turned on, the LFOs are synchronized to the clock with absolute phase.

Tap1 Delay
Tap2 Delay

0.00 to 21.5 s
The delay length of the output tap 1 or 2.

Tap1 Level
Tap2 Level

0 to 100 %
The level of the output tap 1 or 2 expressed as a percent.

Tap1 Pan
Tap2 Pan

-100 to 100%
The output taps 1 and 2 are mono sources that can be panned to the left or right output channels.

-100% is panned fully left
100% is panned fully right.

Comp Atk

0.0 to 228.0 ms
The time for the compressor to start to cut in when there is an increase in signal level (attack) above the threshold.

Comp Rel

0 to 3000 ms
The time for the compressor to stop compressing when there is a reduction in signal level (release) from a signal level above the threshold.

CompSmooth

Comp Ratio

1.0:1 to 100.0:1, Inf:1
The compression ratio.

High ratios are highly compressed
Low ratios are moderately compressed

Comp Thres

-79.0 to 0.0 dB
The threshold level in dBFS (decibels relative to full scale) above which the signal begins to be compressed.

Dist Drive

DistWarmth

8 to 25088 Hz
A lowpass filter in the distortion control path.

This filter may be used to reduce some of the harshness of some distortion settings without reducing the bandwidth of the signal.

Complex Echo

Effects Size : 1

Multitap delay line effect consisting of 6 independent output taps and 4 independent feedback taps

Complex Echo Presets
170 Multitaps ms	173 Sloppy Echoes	177 Electronica Slap
171 Diffuse Slaps	175 500ms BehindSrce

Complex Echo is an elaborate delay line.

3 independent output taps per channel
2 independent feedback taps per channel
equal power output tap panning
feedback diffuser
high frequency damping

Each channel has three output taps, each of which can be delayed up to 2600ms (2.6 sec) then panned at the output.

Feedback Taps

Feedback taps can also be delayed up to 2600ms, but both feedback channels do slightly different things.

Feedback Lines

Feedback line 1 feeds the signal back to the delay input of the same channel, while feedback line 2 feeds the signal back to the opposite channel.
Feedback line 2 may also be referred to as a “ping-pong” feedback.
Relative levels for each feedback line can be set with the “FB2/FB1>FB” control where 0% only allows FB1 to be used, and 100% only allows FB2 to be used.

Diffuser

The diffuser sits at the beginning of the delay line, and consists of three controls.

Separate left and right Diff Dly parameters control the length that a signal is smeared from 0 to 100ms as it passes through these diffusers.

Diff Amt adjusts the smearing intensity.

Short diffuser delays can diffuse the sound while large delays can drastically alter the spectral flavor.

Setting all three diffuser parameters to 0 disables the diffuser.

The delay inputs have one-pole (6dB/oct) lowpass filters controlled by the HF Damping parameter.

Figure 21. Complex Echo - Signal Flow

Wet/Dry

0 to 100% wet
The relative amount of input signal and effected signal that is to appear in the final effect output mix.

When set to 0%, the output is taken only from the input (dry).
When set to 100%, the output is all wet.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

Feedback

0 to 100 %
The amplitude of the feedback tap(s) fed back to the beginning of the delay.

FB2 / FB1>FB

0 to 100 %
Balance control between feedback line 1 and line 2.

0% turns off feedback line 2 only allowing use of feedback line 1
50% is an even mix of both lines
100% turns off line 1

HF Damping

8 to 25088 Hz
The amount of high frequency content of the signal to the input of the delay.

This control determines the cutoff frequency of the one-pole (-6dB/octave) lowpass filters.

L Diff Dly
R Diff Dly

0 to 100 ms
Adjusts delay length of the diffusers.

Diff Amt

0 to 100 %
Adjusts the diffuser intensity.

L Fdbk1 Dly

0 to 2600 ms
Adjusts the delay length of the left channel’s feedback tap fed back to the left channel’s delay input.

L Fdbk2 Dly

0 to 2600 ms
Adjusts the delay length of the left channel’s feedback tap fed back to the right channel’s delay input.

R Fdbk1 Dly

0 to 2600 ms
Adjusts the delay length of the right channel’s feedback tap fed back to the right channel’s delay input.

R Fdbk2 Dly

0 to 2600 ms
Adjusts the delay length of the right channel’s feedback tap fed back to the left channel’s delay input.

L Tap1 Dly
L Tap2 Dly
L Tap3 Dly
R Tap1 Dly
R Tap2 Dly
R Tap3 Dly

0 to 2600 ms
Adjusts the delay length of the left and right channel’s three output taps.

L Tap1 Lvl
L Tap2 Lvl
L Tap3 Lvl
R Tap1 Lvl
R Tap2 Lvl
R Tap3 Lvl

0 to 100 %
Adjusts the listening level of the left and right channel’s three output taps.

L Tap1 Pan
L Tap2 Pan
L Tap3 Pan
R Tap1 Pan
R Tap2 Pan
R Tap3 Pan

-100 to 100 %
Adjusts the equal power pan position of the left and right channel’s three output taps.

0% is center pan
negative values pan to the left
positive values pan to the right

Spectral Taps

Spectral Taps are not available for use on the PC3 series

Spectral 4-Tap (2 unit) and Spectral 6-Tap (3 unit) tempo-based multi-tap delay effects.

They are similar to a simple 4 and 8-tap BPM delays with feedback, but have their feedback and output taps modified with shapers and filters.

Feedback Path

In the feedback path of each are a diffuser, highpass filter, lowpass filter, and imager.

Delay Tap Configuration

Each delay tap has a shaper, comb filter, balance and level controls with the exception of Tap 1, which does not have a comb filter.

Diffusion

Diffusers add a quality that can be described as “smearing” the feedback signal.

The more a signal has been regenerated through feedback and consequently fed through the diffuser, the more it is smeared.

It requires two parameters, one for the duration a signal is smeared labeled Diff Delay, and the other for the amount it is smeared labeled Diff Amt.

To disable the diffuser, both Diff Delay and Diff Amt should be set to zero.

Short Diff Delay settings have subtle smearing effects.
Increasing Diff Delay will be more noticeable, and long delay settings will take on a ringy resonant quality.
Positive diffusion settings will add diffusion while maintaining image integrity.
Negative diffusion amounts will cause the feedback image to lose image integrity and become wide.

Highpass and Lowpass Filters

Two 1 pole 6dB/oct filters are also in the feedback path: highpass and lowpass.

Highpass filter roll-off frequency is controlled with LF Damping
Lowpass filter roll-off frequency is controlled by HF Damping

Imaging

The imager shifts the stereo input image when fed through feedback.

Small positive or negative values shift the image to the right or left respectively
Larger values shift the image so much that the image gets scrambled through each feedback generation.

Shapers

On each output tap is a shaper.

The spectral multi-tap shapers offer four shaping loops as opposed to eight found in the V.A.S.T. shapers, but can allow up to 6.00x intensity.

Figure 22. Various shaper curves used in the spectral multi-taps

Resonant Comb Filters: Immediately following the shapers on taps 2 and above are resonant comb filters tuned in semitones.

These comb filters make the taps become pitched.

When a comb filter is in use, the shaper before it can be used to intensify these pitched qualities.
Balance / Level Controls: Each tap also has separate balance and level controls.
Tempo Control: Since these are tempo based effects, tap delay values and feedback delay (labeled LoopLength) values are set relative to a beat.

The beat duration is set by adjusting Tempo in BPM.

The tempo can be synced to the system clock by setting Tempo to System.

Each tap’s delay is adjusted relative to one beat, in 1/24 beat increments.

Notice that 24 is a musically useful beat division because it can divide a beat into halves, 3rds, 4ths, 6ths, 8ths, 12ths, and of course 24ths.

Tempo Example

For example, setting LoopLength to 1-12/24 bts will put the feedback tap at 1-1/2 beats (dotted quarter note in 4/4 time) of delay making the feedback repetition occur every one and a half beats.

This is equivalent to 3/4 of a second at 120 BPM.

When Tempo is set to 60 BPM, each 1/24th of a beat is equivalent to 1/24th of a second.
When tempo is set to 250 BPM, each 1/24th of a beat is equivalent to 10ms of delay.

Spectral 4-Tap

Effects Size : 2

Tempo based 4 tap delay with added shapers and resonant comb filters on each tap

Spectral 4-Tap Presets
176 Dub Skanque Dly
178 Spectral 4-Tap
179 Astral Taps

Wet/Dry

0 to 100 %
The relative amount of input signal and effected signal that is to appear in the final effect output mix.

When set to 0%, the output is taken only from the input (dry).
When set to 100%, the output is all wet.
Negative values polarity invert the wet signal.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

Fdbk Level

0 to 100 %
The amount that the feedback tap is fed to the input of the delay.

HF Damping

8 to 25088 Hz
The amount of high frequency content of the signal to the input of the delay.

This control determines the cutoff frequency of the one-pole (-6dB/octave) lowpass filters.

LF Damping

8 to 25088 Hz
The amount of low frequency content of the signal to the input of the delay.

This control determines the cutoff frequency of the one-pole (-6dB/octave) lowpass filters.

Tempo

System, 0 to 255 BPM
Basis for the rates of the delay times, as referenced to a musical tempo in BPM (beats per minute).

When this parameter is set to System, the tempo is locked to the internal sequencer tempo or to incoming MIDI clocks.
In this case, FXMods (FUNs, LFOs, ASRs etc.) will have no effect on the Tempo parameter.

Diff Dly

0 to 20.0 ms
The length that the diffuser smears the signal sent to the input of the delay.

Diff Amt

-100 to 100 %
The intensity that the diffuser smears the signal sent to the input of the delay. Negative
values decorrelate the stereo signal.

LoopLength

On or Off
The delay length of the feedback tap in 24ths of a beat.

Fdbk Image

-100 to 100 %
Sets the amount the stereo image is shifted each time it passes through the feedback line.

Tap1 Delay
Tap2 Delay
Tap3 Delay
Tap4 Delay

0 to 32 bts
Adjusts the length of time in 24ths of a beat each output tap is delayed.

Tap1 Shapr
Tap2 Shapr
Tap3 Shapr
Tap4 Shapr

0.10 to 6.00 x
Adjusts the intensity of the shaper at each output tap.

Tap2 Pitch
Tap3 Pitch
Tap4 Pitch

C-1 to C8
Adjusts the frequency in semitones of the comb filter at each output tap.

Tap2 PtAmt
Tap3 PtAmt
Tap4 PtAmt

0 to 100%
Adjusts the intensity of the comb filter at each output tap.

Tap1 Level
Tap2 Level
Tap3 Level
Tap4 Level

0 to 100%
Adjusts the relative amplitude that each output tap is heard.

Tap1 Bal
Tap2 Bal
Tap3 Bal
Tap4 Bal

-100 to 100%
Adjusts the left/right balance of each output tap.

Negative values bring down the right channel
Positive values bring down the left channel.

Spectral 6-Tap

Effects Size : 3

Tempo based 6 tap delay with added shapers and resonant comb filters on each tap

Spectral 6-Tap Presets
180 SpectraShapeTaps
181 Fanfare In Gmaj

Figure 23. Spectral 6-Tap

Wet/Dry

0 to 100 %
The relative amount of input signal and effected signal that is to appear in the final effect output mix.

When set to 0%, the output is taken only from the input (dry).
When set to 100%, the output is all wet.
Negative values polarity invert the wet signal.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

Fdbk Level

0 to 100 %
The amount that the feedback tap is fed to the input of the delay.

HF Damping

8 to 25088 Hz
The amount of high frequency content of the signal to the input of the delay.

This control determines the cutoff frequency of the one-pole (-6dB/octave) lowpass filters.

LF Damping

8 to 25088 Hz
The amount of low frequency content of the signal to the input of the delay.

This control determines the cutoff frequency of the one-pole (-6dB/octave) lowpass filters.

Tempo

Diff Dly

0 to 20.0 ms
The length that the diffuser smears the signal sent to the input of the delay.

Diff Amt

-100 to 100 %
The intensity that the diffuser smears the signal sent to the input of the delay. Negative
values decorrelate the stereo signal.

LoopLength

On or Off
The delay length of the feedback tap in 24ths of a beat.

Fdbk Image

-100 to 100 %
Sets the amount the stereo image is shifted each time it passes through the feedback line.

Tap1 Delay
Tap2 Delay
Tap3 Delay
Tap4 Delay
Tap5 Delay
Tap6 Delay

0 to 32 bts
Adjusts the length of time in 24ths of a beat each output tap is delayed.

Tap1 Shapr
Tap2 Shapr
Tap3 Shapr
Tap4 Shapr
Tap5 Shapr
Tap6 Shapr

0.10 to 6.00 x
Adjusts the intensity of the shaper at each output tap.

Tap2 Pitch
Tap3 Pitch
Tap4 Pitch
Tap5 Pitch
Tap6 Pitch

C-1 to C8
Adjusts the frequency in semitones of the comb filter at each output tap.

Tap2 PtAmt
Tap3 PtAmt
Tap4 PtAmt
Tap5 PtAmt
Tap6 PtAmt

0 to 100%
Adjusts the intensity of the comb filter at each output tap.

Tap1 Level
Tap2 Level
Tap3 Level
Tap4 Level
Tap5 Level
Tap6 Level

0 to 100%
Adjusts the relative amplitude that each output tap is heard.

Tap1 Bal
Tap2 Bal
Tap3 Bal
Tap4 Bal
Tap5 Bal
Tap6 Bal

-100 to 100%
Adjusts the left/right balance of each output tap.

Negative values bring down the right channel
Positive values bring down the left channel.

Moving Delay

Dual MovDelay

Effects Size : 1

Generic dual mono moving delay line

Dual MovDelay Presets
156 Wide Slapbk 76ms	159 17ms Ambience	196 Warped Echoes
157 TiteSlapAmb 50ms	161 StereoFlamDelay	198 L:Flange R:Delay
158 33ms Ambience	174 Pad Psychosis

This algorithm offers generic moving delay lines in a dual mono configuration.

Each separate moving delay can be used as a flanger, chorus, or static delay line selectable by the LFO Mode parameter.

Chorus

Both flavors of chorus pitch envelopes are offered for pitch shifting:

ChorTri for triangle
ChorTrap for trapezoidal

Refer to Choruses for more information on these envelope shapes.

The value functions much like a wet/dry mix where:

0% means that only the algorithm input dry signal is fed into effect B (putting the effects in parallel)
100% means only the output of effect A is fed into effect B (putting the effects in series).

Moving Delay Control

Each moving delay offers control over center delay length, LFO excursion, LFO rate, feedback, and high frequency damping.

LFO Excursion

The delay length, in milliseconds, is the center of LFO excursion.

LFO excursion is controlled by the LFO Dpth parameter in percent.

LFO Depth is an arbitrary value, and is the percentage of available excursion.

LFO Mode

Flange: When using LFO Mode Flange, this adjusts the range that the LFO will move the delay tap.
ChorTri / ChorTrap: When in LFO Mode ChorTri or ChorTrap, this controls the maximum pitch depth caused by the moving delay tap, and is constant regardless of LFO Rate.

Figure 24. Dual MovDelay - Signal Flow

L Wet/Dry
R Wet/Dry

0 to 100%wet
The relative amount of input signal and effected signal that is to appear in the final effect output mix for each input channel.

When set to 0%, the output is taken only from the corresponding input (dry) signal.
When set to 100%, the output is all wet.

L Out Gain
R Out Gain

Off; -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect for each input channel.

L Pan
R Pan

-100 to 100%
The output panning position of each moving delay circuit.

0% is center
Negative values pan left
Positive values pan right

L Delay
R Delay

0.0 to 1000.0 ms
Adjusts the delay time for each moving delay circuit, which is the center of LFO excursion.

L LFO Mode
R LFO Mode

Flange, ChorTri, ChorTrap, Delay
Adjusts the LFO excursion type.

In Flange mode, the LFO is optimized for flange effects and LFO Dpth adjusts the excursion amount.
In ChorTri and ChorTrap modes, the LFO is optimized for triangle and trapezoidal pitch envelopes respectively, and LFO Dpth adjusts the amount of chorus detuning.
In Delay mode, the LFO is turned off leaving a basic delay.

LFO Rate and LFO Dpth in Delay mode are disabled.

L LFO Rate
R LFO Rate

0.00 to 10.00 Hz
Adjusts the LFO speed for each moving delay circuit.

In Delay mode, this is disabled.

L LFO Dpth
R LFO Dpth

0.0 to 200.0%

In Flange LFO mode, this adjusts an arbitrary LFO excursion amount.
In ChorTri and ChorTrap modes, this controls the chorus detune amount.

In Delay mode, this is disabled.

L Feedback
R Feedback

-100 to 100%
Adjusts the level of each moving delay circuits output signal fed back into their own inputs.

Negative values polarity-invert the feedback signal.

L HF Damp
R HF Damp

8 to 25088 Hz
Adjusts the cutoff frequency of a 1-pole (6dB/oct) lowpass filter in each moving delay circuit.

MovDelay

Effects Size : 1

Generic stereo moving delay line

MovDelay Presets
151 Basic Dly 250ms	165 Dub Delay ms	154 MedSlapback 76ms
152 Simple Slap 60ms	197 Ween-vox
153 TightSlapbk 30ms	220 Rich Noodle

Moving Delay is identical to Dual MovDelay except that the algorithm now has stereo controls rather than dual mono.

This means all the controls except L Pan and R Pan are no longer dual left and right but are ganged into single controls controlling both left and right channels.

Wet/Dry

0 to 100 %
The relative amount of input signal and effected signal that is to appear in the final effect output mix for each input channel.

When set to 0%, the output is taken only from the input (dry) signal.
When set to 100%, the output is all wet.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

L Pan
R Pan

-100 to 100 %
The output panning position of each moving delay circuit.

0% is center
Negative values pan left
Positive values pan right.

Delay

0.0 to 1000.0 ms
Adjusts the delay time for the moving delay circuits, which is the center of LFO excursion.

LFO Mode

ChorTri, ChorTrap, Delay, Flange
Adjusts the LFO excursion type.

Flange mode, the LFO is optimized for flange effects and LFO Dpth adjusts the excursion amount.
In ChorTri and ChorTrap modes, the LFO is optimized for triangle and trapezoidal pitch envelopes respectively, and LFO Dpth adjusts the amount of chorus detuning.
Delay mode, the LFO is turned off leaving a basic delay.

LFO Rate and LFO Dpth in Delay mode are disabled.

LFO Rate

0.00 to 10.00 Hz
Adjusts the LFO speed for the moving delay circuits.

LFO Depth

0.0 to 200.0 %
- Flange LFO mode, this adjusts an arbitrary LFO excursion amount.
- In ChorTri and ChorTrap modes, this controls the chorus detune amount.

In delay mode, this is disabled.

Feedback

-100 to 100 %
Adjusts the level of the moving delay circuits output signal fed back into their own inputs.

Negative values polarity invert the feedback signal.

HF Damping

8 to 25088 Hz
Adjusts the cutoff frequency of a 1-pole (6dB/oct) lowpass filter in the moving delay circuits.

Dual MvDly+MvDly

Effects Size : 2

Generic dual mono moving delay lines

Dual MvDly+MvDly Presets
199 2Dlys 1Chr 1Flng

In Dual MvDly+MvDly, there are 2 moving delay elements per channel distinguishable by parameters beginning with “L1,” “L2,” “R1,” and “R2.”

Channel Mix: The second moving delay on each channel is fed with a mix of the first delays and the input dry signal for that particular channel.

These mixes are controlled by L1/Dry→L2 and R1/Dry→R2.

Each of the four moving delays have separate Mix and Pan levels.

The input dry signal for each channel can also be panned.

The Wet/Dry parameter controls the ratio between the sum of both moving delay elements on that channel regardless of pan position, and the input dry signal.

Out Gain, like Wet/Dry, adjusts the output level for each channel regardless of pan position.

Figure 25. Dual MvDly+MvDly - Signal Flow

L Wet/Dry
R Wet/Dry

-100 to 100%wet
The relative amount of input signal and effected signal that is to appear in the final effect output mix for each input channel.

When set to 0%, the output is taken only from the corresponding input (dry) signal.
When set to 100%, the output is all wet.

L Out Gain
R Out Gain

Off; -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect for each input channel.

L1 Mix
L2 Mix
R1 Mix
R2 Mix

-100 to 100%
Adjusts the mix levels for each moving delay circuit.

The resulting sum makes up the wet signal.

Negative values polarity-invert the signal.

L1 Pan
L2 Pan
R1 Pan
R2 Pan

-100 to 100%
The output panning position of each moving delay circuit.

0% is center
Negative values pan left
Positive values pan right

L Dry Pan
R Dry Pan

-100 to 100%
Adjusts the output pan position of the input dry signals.

The dry level is controlled with Wet/Dry.

0% pans to center
Negative values pan left
Positive values pan right

L1/Dry→L2
R1/Dry→R2

0 to 100%
Adjusts the input mix into the second pair of moving delay circuits.

The value represents a ratio of the output of the first moving delay circuit and the input dry signal.

A value of 0% allows only the input dry signal to be fed into the second delay
A value of 100% only allows the first delay to be fed into the second

L1 Delay
R2 Delay

0.0 to 1000.0 ms
Adjusts the delay time for each moving delay circuit, which is the center of LFO excursion.

L1 LFO Mode
L2 LFO Mode
R1 LFO Mode
R2 LFO Mode

Flange, ChorTri, ChorTrap, Delay
Adjusts the LFO excursion type.

In Flange mode, the LFO is optimized for flange effects and LFO Dpth adjusts the excursion amount.
In ChorTri and ChorTrap modes, the LFO is optimized for triangle and trapezoidal pitch envelopes respectively, and LFO Dpth adjusts the amount of chorus detuning.
In Delay mode, the LFO is turned off leaving a basic delay.

LFO Rate and LFO Dpth in Delay mode are disabled.

L1 LFO Rate
L2 LFO Rate
R1 LFO Rate
R2 LFO Rate

0.00 to 10.00 Hz
Adjusts the LFO speed for each moving delay circuit.

L1 LFO Dpth
L2 LFO Dpth
R1 LFO Dpth
R2 LFO Dpth

0.0 to 200.0%

In Flange LFO mode, this adjusts an arbitrary LFO excursion amount.
In ChorTri and ChorTrap modes, this controls the chorus detune amount.
In delay mode, this is disabled.

L1 Fdbk
L2 Fdbk
R1 Fdbk
R2 Fdbk

-100 to 100%
Adjusts the level of each moving delay circuits output signal fed back into their own inputs.

Negative values polarity-invert the feedback signal.

L1 HF Damp
L2 HF Damp
R1 HF Damp
R2 HF Damp

8 to 25088 Hz
Adjusts the cutoff frequency of a 1-pole (6dB/oct) lowpass filter in each moving delay circuit.

Gated Delay

Effects Size : 2

Delay with gating and ducking

Gated Delay Presets
193 Ducked Delay
194 Drum+Bass Zapper

Gated Delay is a delay with feedback which has its output and feedback controlled by a gate.

The gate side-chain is the same as in Gate w/SC EQ, except this algorithm does not include side-chain EQ filtering.

Gating a delay is not particularly interesting until the sense of the gate is reversed by turning on the Ducking parameter.

With ducking, the gate passes signal only when the side-chain input signal is below the gate threshold.

The with ducking turned on, Gated Delay could also be called the “Monster Truck Effect.”

Set Wet/Dry to about 50%.

What happens is that as long as a signal is coming in that is above the gate threshold, all you will hear is the dry signal.

When the input signal stops, then the gate opens up, and suddenly the delay takes over.

Example: For example, if you sent the speech phrase “Welcome to the monster truck rally” through the effect, what you would hear is “Welcome to the monster truck rally, rally, rally…” Of course to really get the desired effect, you may need to adjust the gate, the delay and the feedback.

See Gate w/SC EQ for details on controlling the gate.

The loop delay length (for feedback) is the same for both left and right channels to keep timing constant.

The output delay lengths may be different for the two channels to give a syncopated or “ping-pong” feel.

The Feedback parameter controls how long it will take for the looping delay sound to decay.

Figure 26. Gated Delay - Block Diagram

Wet/Dry

0 to 100%
The amount of gated delay signal (wet) relative to the input dry signal to send to the output.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

Feedback

0 to 100%
The amount of the loop delay signal to add to the input of the delay.

Feedback controls how long the looped delay takes to decay.

Loop Crs

0 to 5100 ms
The length of the delay loop in milliseconds (ms).

The loop time controls the duration of the repeated “snippet” of sound.

Loop Fine

-20.0 to 20.0 ms
The length of the delay loop in milliseconds (ms).

The loop time controls the duration of the repeated “snippet” of sound.

L Dly Crs
R Dly Crs

0 to 5100 ms
The length of the delay for the final output taps in milliseconds (ms).

L Dly Fine
R Dly Fine

-20.0 to 20.0 ms
The length of the delay for the final output taps in milliseconds (ms).

Threshold

-79.0 to 24.0 dB
The signal level in dB required to open the gate (or close the gate if Ducking is on).

Ducking

On or Off

When set to Off, the gate opens when the signal rises above threshold and closes when the gate time expires.
When set to On, the gate closes when the signal rises above threshold and opens when the gate time expires.

This effect is most interesting when Ducking is on.

Retrigger

On or Off
If Retrigger is On, the gate timer is constantly restarted (retriggered) as long as the side chain signal is above the threshold.

The gate then remains open (assuming Ducking is Off) until the signal falls below the threshold and the gate timer has elapsed.

If Retrigger is Off, then the gate timer starts at the moment the signal rises above the threshold and the gate closes after the timer elapses, whether or not the signal is still above threshold.

With Retrigger off, use the Env Time to control how fast the side chain signal envelope drops below the threshold.

With Retrigger set to Off, the side chain envelope must fall below threshold before the gate can open again.

Env Time

0 to 3000 ms
Envelope time is for use when Retrigger is set to Off.

The envelope time controls the time for the side chain signal envelope to drop below the threshold.

At short times, the gate can reopen rapidly after it has closed, and you may find the gate opening unexpectedly due to an amplitude modulation of the side chain signal.
For long times, the gate will remain closed until the envelope has a chance to fall, and you may miss gating events.

Gate Time

0 to 3000 ms
The time in seconds that the gate will stay fully on after the signal envelope rises above threshold.

The gate timer is started or restarted whenever the signal envelope rises above threshold.

If Retrigger is On, the gate timer is continually reset while the side chain signal is above the threshold.

Atk Time

0.0 to 228.0 ms
The time for the gate to ramp from closed to open (reverse if Ducking is On) after the signal rises above threshold.

Rel Time

0 to 3000 ms
The time for the gate to ramp from open to closed (reverse if Ducking is On) after the gate timer has elapsed.

3 Band Delay

Effects Size : 2

Three delays operating on selectable frequency bands

3 Band Delay Presets
195 3BandDly Drums

3 Band Delay uses a band splitting filter to divide the input signal into 3 frequency bands.

The filtered bands of the signal are then passed through 3 parallel delay lines.

You can select the frequencies at which the bands are split.

You can select which frequency band (Low, Mid, or High) gets passed through a particular delay line.

You can choose to pass the same band through all 3 delay lines, or you can send each band through its own delay line.

Delay line lengths are tempo based.

Tempo is expressed in beats per minute (BPM) and the delay lengths are expressed as the number of beats (bts) at the tempo.

The delay length beats are adjustable in increments of 1/24th of a beat, which is a useful fraction because it can divide beats into 2, 3, 4, 6, 8, or 12 parts.

The length of a delay in seconds can be calculated as:
\(T=(beats/tempo)*60\)

The outputs of each stereo delay line can be panned to the final stereo output.

The full stereo field is moved with this panner, and the width of the stereo field can be reduced with the Width parameter.

Figure 27. 3 Band Delay - Block Diagram

Wet/Dry

0 to 100%wet
The relative amount of input signal and delay signal that is to appear at the final effect output mix.

0% only the dry input is heard
100% only the delayed (wet) signal is heard.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

Tempo

System, 1 to 255 BPM
Basis for the delay lengths, as referenced to a musical tempo in bpm (beats per minute).

When this parameter is set to System, the tempo is locked to the internal system tempo or to incoming MIDI clocks.

In this case, FXMods (FUNs, LFOs, ASRs etc.) will have no effect on the Tempo parameter.

Crossover1
Crossover2

8 to 25088 Hz
The Crossover parameters set the frequencies which divide the three frequency bands.

The two Crossover parameters are interchangeable, so either may contain the higher frequency value.

BandSelctA
BandSelctB
BandSelctC

Low, Mid, or High
Selects which of the three frequency bands (Low, Mid, or High) is to pass through the particular delay (A, B, or C).

DelayLenA
DelayLenB
DelayLenC

0 to 6 bts
The delay lengths (for delays A, B, and C) as tempo beat durations.

The delay length is specified as a fraction or multiple of the tempo in “beats.”

The length of a delay in seconds can be calculated as:
\(T=(beats/tempo)*60\)

DelayLvlA
DelayLvlB
DelayLvlC

0 to 100%
The amount of signal from the delays (A, B, and C) which gets sent to the final wet/dry mix.

PanA
PanB
PanC

-100 to 100%
Each stereo delay (A, B, and C) has a stereo panner.

The stereo image is maintained but is “tilted” to the right or left.

0% there is no change to the signal
100% both input signals are sent to the right channel
50%, what had been hard left at the input will now be in the center, and what had been in the center at the input will now be halfway between center and right

Negative values tilt the signal to the left.

WidthA
WidthB
WidthC

-100 to 100%
The stereo image width of each panner can be controlled with the Width parameter.

100% is full stereo width, so the left input is sent to the left output channel while the right input is sent to the right output channel
0% Width, the stereo width is narrowed to mono (left and right are summed), and the panner behaves like a mono-to-stereo panner

Negative Width settings swap the left and right channels.

4 Tap Delays

These are simple stereo 4-tap delay algorithms:

Delay lengths are defined in:

tempo beats (4-Tap Delay BPM)
milliseconds (ms) (4-Tap Delay).

The left and right channels are fully symmetric (all controls affect both channels).

The duration of each stereo delay tap (length of the delay) and the signal level from each stereo tap may be set.

Prior to output each delay tap passes through a level and left-right balance control.

The taps are summed and added to the dry input signal through a Wet/Dry control.

The delayed signal from the “Loop” tap may be fed back to the delay input.

Figure 28. Left channel of 4-Tap Delay

The delay length for non-BPM tap delays is the sum of the coarse and fine parameters for the tap multiplied by the DelayScale parameter which is common to all non-BPM taps.

The DelayScale parameter allows you to change the lengths of all the taps together.

A repetitive loop delay is created by turning up the Fdbk Level parameter.

Only the Loop tap is fed back to the input of the delay, so this is the tap which controls the loop rate.

Usually you will want the Loop delay length to be longer than the other tap lengths.

Set the Loop delay length to the desired length then set the other taps to fill in the measure with interesting rhythmical patterns.

Setting tap levels allows some “beats” to receive different emphasis than others.

The delay lengths for 4-Tap Delay are in units of milliseconds (ms).

If you want to base delay lengths on tempo, then the 4-Tap Delay BPM algorithm may be more convenient.

At 100%, your sound will be repeated indefinitely.

HF Damping selectively removes high frequency content from your delayed signal and will also cause your sound to eventually disappear.

Hold: The Hold parameter is a switch which controls signal routing.

When turned on, Hold will play whatever signal is in the delay line indefinitely.

Hold overrides the feedback parameter and prevents any incoming signal from entering the delay.

The Hold parameter has no effect on the Wet/Dry or HF Damping parameters, which continue to work as usual, so if there is some HF Damping, the delay will eventually die out.

You may have to practice using the Hold parameter.

Feedback: The feedback (Fdbk Level) controls how long a sound in the delay line takes to die out.

Each time your sound goes through the delay, it is reduced by the feedback amount.

If feedback is fairly low and you turn on Hold at the wrong moment, you can get a disconcerting jump in level at some point in the loop.

4-Tap Delay

Effects Size : 1

A stereo four tap delay in (ms) with feedback

4-Tap Delay Presets
155 LongishSlap 95ms	160 Stereo Delay ms	167 4-Tap Dly Pan ms
168 SemiCircle 4-Tap

Wet/Dry

0 to 100% wet
The relative amount of input signal and delay signal that is to appear in the final effect output mix.

When set to 0%, the output is taken only from the input (dry).
When set to 100%, the output is all wet.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

Fdbk Level

0 to 100%
The percentage of the delayed signal to feed back or return to the delay input.

Turning up the feedback will cause the effect to repeatedly echo or act as a crude reverb.

HF Damping

16 Hz to 25088 Hz
The -3 dB frequency in Hz of a one pole lowpass filter (-6 dB/octave) placed in front of the delay line.

The filter is specified for a signal passing through the filter once.

Multiple passes through the feedback will cause the signal to become more and more dull.

Dry Bal

-100 to 100%
The left-right balance of the dry signal.

-100% allows only the left dry signal to pass
100% lets only the right dry signal pass
0% lets equal amounts of the left and right dry signals pass to their respective outputs

Hold

On or Off
A switch which when turned on, locks any signal currently in the delay to play until Hold is turned off.

When Hold is on, no signal can enter the delay and Feedback is set to 100% behind the scenes.

Hold does not affect the HF Damping or Wet/Dry mix.

Loop Crs

0 to 2540 ms
The coarse delay length of the Loop tap.

If the feedback is turned up, this parameter sets the repeating delay loop length.

The resolution of the coarse adjust is 20 milliseconds, but finer resolution can be obtained using the Loop Fine parameter.

Loop Fine

-20 to 20 ms
A fine adjustment to the Loop tap delay length.

The delay resolution is 0.2 milliseconds (ms).

Loop Fine is added to Loop Crs (coarse) to get the actual delay length.

DelayScale

0.00x to 10.00x
The DelayScale parameter allows you to change the lengths of all the taps together.

Tap1 Crs
Tap2 Crs
Tap3 Crs
Tap4 Crs

0 to 2540 ms
The coarse delay lengths of the output taps.

The resolution of the coarse adjust is 20 milliseconds, but finer resolution can be obtained using the Tapn Fine parameters.

Tap1 Fine
Tap2 Fine
Tap3 Fine
Tap4 Fine

-20 to 20 ms
The delay resolution is 0.2 milliseconds (ms).

Tap1/2/3/4 Fine is added to Tap1/2/3/4 Crs (coarse) to get actual delay lengths.

Tap1 Level
Tap2 Level
Tap3 Level
Tap4 Level

0 to 100 %
The amount of signal from each of the taps which get sent to the output.

With the Loop Lvl control, you can give different amounts of emphasis to various taps in the loop.

Tap1 Bal
Tap2 Bal
Tap3 Bal
Tap4 Bal

-100 to 100 %
The left-right balance of each of the stereo taps.

-100% allows only the left tap to pass to the left
100% lets only the right tap pass to the right output
At 0%, equal amounts of the left and right taps pass to their respective outputs

4-Tap Delay BPM

Effects Size : 1

A stereo four tap tempo delay with feedback

4-Tap Delay BPM Presets
150 Basic Delay 1/8
166 4-Tap Delay BPM
172 OffbeatFlamDelay

In this Algorithm, the delay length for any given tap is determined by the tempo:

expressed in beats per minute (BPM)
delay length of the tap expressed in beats (bts).

The tempo alters all tap lengths together.

With the tempo in beats per minute and delay lengths in beats, you can calculate the length of a delay in seconds as:
\(beats/tempo*60 (sec/min)\)

There is a limited amount of delay memory available (over 2.5 seconds for 4-Tap Delay BPM).

When slow tempos and/or long lengths are specified, you may run out of delay memory, at which point the delay length will be cut in half.

When you slow down the tempo, you may find the delays suddenly getting shorter.

A repetitive loop delay is created by turning up the feedback parameter (Fdbk Level).

Only the Loop tap is fed back to the input of the delay, so this is the tap which controls the loop rate.

Usually you will want the Loop tap (LoopLength parameter) to be longer than the other tap lengths.
To repeat a pattern on a 4/4 measure (4 beats per measure) simply set LoopLength to 4 bts.

The output taps can then be used to fill in the measure with interesting rhythmical patterns.

Setting tap levels allows some “beats” to receive different emphasis than others.

Wet/Dry

0 to 100% wet
The relative amount of input signal and delay signal that is to appear in the final effect output mix.

When set to 0%, the output is taken only from the input (dry).
When set to 100%, the output is all wet.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

Fdbk Level

0 to 100%
The percentage of the delayed signal to feed back or return to the delay input.

Turning up the feedback will cause the effect to repeatedly echo or act as a crude reverb.

Tempo

System, 1 to 255 BPM
Basis for the delay lengths, as referenced to a musical tempo in bpm (beats per minute).

When this parameter is set to System, the tempo is locked to the internal sequencer tempo or to incoming MIDI clocks.
In this case, FXMods (FUNs, LFOs, ASRs etc.) will have no effect on the Tempo parameter.

HF Damping

Dry Bal

-100 to 100%
The left-right balance of the dry signal.

-100% allows only the left dry signal to pass
100% lets only the right dry signal pass
0% lets equal amounts of the left and right dry signals pass to their respective outputs

Hold

On or Off
A switch which when turned on, locks any signal currently in the delay to play until Hold is turned off.

When Hold is on, no signal can enter the delay and Feedback is set to 100% behind the scenes.

Hold does not affect the HF Damping or Wet/Dry mix.

LoopLength

0 to 32 bts
The delay length of the Loop tap.

If the feedback is turned up, this parameter sets the repeating delay loop length. LoopLength sets the loop delay length as a tempo beat duration.

The tempo is specified with the Tempo parameter and the delay length is given in beats (bts).

The delay length in seconds is calculated as:
\(beats/tempo*60 (sec/min)\)

Tap1 Delay
Tap2 Delay
Tap3 Delay
Tap4 Delay

0 to 32 bts
The delay lengths of the taps as tempo beat durations.

The tempo is specified with the Tempo parameter and the delay length is given in beats (bts).

The delay length in seconds is calculated as:
\(beats/tempo*60 (sec/min)\)

Use the output taps to create interesting rhythmic patterns within the repeating loop.

Tap1 Level
Tap2 Level
Tap3 Level
Tap4 Level

0 to 100 %
The amount of signal from each of the taps which get sent to the output.

Tap1 Bal
Tap2 Bal
Tap3 Bal
Tap4 Bal

-100 to 100 %
The left-right balance of each of the stereo taps.

-100% allows only the left tap to pass to the left
100% lets only the right tap pass to the right output
At 0%, equal amounts of the left and right taps pass to their respective outputs

8 Tap Delay BPMs

This is a simple stereo tempo 8-Tap Delay BPM algorithm.

Delay lengths are defined in tempo beats.

The duration of each stereo delay tap (length of the delay) and the signal level from each stereo tap may be set.

Prior to output each delay tap passes through a level and left-right balance control.

Pairs of stereo taps are tied together with balance controls acting with opposite left-right sense.

The taps are summed and added to the dry input signal through a Wet/Dry control.

The delayed signal from the “Loop” tap may be fed back to the delay input.

The sum of the input signal and the feedback signal may be mixed or swapped with the input/feedback signal from the other channel (cross- coupling).

When used with feedback, cross-coupling can achieve a ping-pong effect between the left and right channels.

Figure 29. Left channel of 8-Tap Delay

A repetitive loop delay is created by turning up the Fdbk Level parameter.

Only the Loop tap is fed back to the input of the delay, so this is the tap which controls the loop rate.

Usually you will want the Loop delay length to be longer than the other tap lengths.

Set the Loop delay length to the desired length then set the other taps to fill in the measure with interesting rhythmical patterns.

Setting tap levels allows some “beats” to receive different emphasis than others.

At 100%, your sound will be repeated indefinitely.

HF Damping selectively removes high frequency content from your delayed signal and will also cause your sound to eventually disappear.

Hold: The Hold parameter is a switch which controls signal routing.

When turned on, Hold will play whatever signal is in the delay line indefinitely.

Hold overrides the feedback parameter and prevents any incoming signal from entering the delay.

The Hold parameter has no effect on the Wet/Dry or HF Damping parameters, which continue to work as usual, so if there is some HF Damping, the delay will eventually die out.

You may have to practice using the Hold parameter.

Feedback: The feedback (Fdbk Level) controls how long a sound in the delay line takes to die out.

Each time your sound goes through the delay, it is reduced by the feedback amount.

If feedback is fairly low and you turn on Hold at the wrong moment, you can get a disconcerting jump in level at some point in the loop.

Effects Size : 2

A stereo eight tap tempo delay with feedback

8-Tap Delay BPM Presets
169 8-Tap Delay BPM

The delay length for any given tap is determined by the tempo, expressed in beats per minute (BPM), and the delay length of the tap expressed in beats (bts).

The tempo alters all tap lengths together.

With the tempo in beats per minute and delay lengths in beats, you can calculate the length of a delay in seconds as:
\(beats/tempo*60 (sec/min)\)

There is a limited amount of delay memory available (over 5 seconds for 8-Tap Delay BPM)

When slow tempos and/or long lengths are specified, you may run out of delay memory, at which point the delay length will be cut in half.

When you slow down the tempo, you may find the delays suddenly getting shorter.

A repetitive loop delay is created by turning up the feedback parameter (Fdbk Level).

Only the Loop tap is fed back to the input of the delay, so this is the tap which controls the loop rate.

Usually you will want the Loop tap (LoopLength parameter) to be longer than the other tap lengths.

To repeat a pattern on a 4/4 measure (4 beats per measure) simply set LoopLength to 4 bts.

The output taps can then be used to fill in the measure with interesting rhythmical patterns.

Setting tap levels allows some “beats” to receive different emphasis than others.

Wet/Dry

0 to 100% wet
The relative amount of input signal and delay signal that is to appear in the final effect output mix.

When set to 0%, the output is taken only from the input (dry).
When set to 100%, the output is all wet.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

Fdbk Level

0 to 100%
The percentage of the delayed signal to feed back or return to the delay input.

Turning up the feedback will cause the effect to repeatedly echo or act as a crude reverb.

Tempo

Xcouple

0 to 100%

HF Damping

Dry Bal

-100 to 100%
The left-right balance of the dry signal.

-100% allows only the left dry signal to pass
100% lets only the right dry signal pass
0% lets equal amounts of the left and right dry signals pass to their respective outputs

Hold

On or Off
A switch which when turned on, locks any signal currently in the delay to play until Hold is turned off.

When Hold is on, no signal can enter the delay and Feedback is set to 100% behind the scenes.

Hold does not affect the HF Damping or Wet/Dry mix.

LoopLength

Tap1 Delay
Tap2 Delay
Tap3 Delay
Tap4 Delay
Tap5 Delay
Tap6 Delay
Tap7 Delay
Tap8 Delay

0 to 32 bts
The delay lengths of the taps as tempo beat durations.

The tempo is specified with the Tempo parameter and the delay length is given in beats (bts).

The delay length in seconds is calculated as:
\(beats/tempo*60 (sec/min)\)

Use the output taps to create interesting rhythmic patterns within the repeating loop.

Tap1 Level
Tap2 Level
Tap3 Level
Tap4 Level
Tap5 Level
Tap6 Level
Tap7 Level
Tap8 Level

0 to 100 %
The amount of signal from each of the taps which get sent to the output.

Tap1 Bal
Tap2 Bal
Tap3 Bal
Tap4 Bal
Tap5 Bal
Tap6 Bal
Tap7 Bal
Tap8 Bal

-100 to 100 %
The left-right balance of each of the stereo taps.

-100% allows only the left tap to pass to the left
100% lets only the right tap pass to the right output
At 0%, equal amounts of the left and right taps pass to their respective outputs

Modulators

Ring Modulator

Effects Size : 1

A configurable ring modulator

Ring Modulator Presets
372 Ring Modulator

Ring modulation is a simple effect in which two signals are multiplied together.

Typically, an input signal is modulated with a simple carrier waveform such as a sine wave or a sawtooth.

Since the modulation is symmetric \((a*b = b*a)\), deciding which signal is the carrier and which is the modulation signal is a question of perspective.

A simple, unchanging waveform is generally considered the carrier.

How a Ring Modulator works: To see how the ring modulator works, we’ll have to go through a little high school math and trigonometry.

Let’s look at the simple case of two equal amplitude sine waves modulating each other.

Real signals will be more complex, but they will be much more difficult to analyze.

The two sine waves generally will be oscillating at different frequencies.

A sine wave signal at any time \(t\) having a frequency \(f\) is represented as \(sin(f_1t + \phi)\) where \(\phi\) is constant phase angle to correct for the sine wave not being \(0\) at \(t = 0\).

The sine wave could also be represented with a cosine function which is a sine function with a 90° phase shift.

To simply matters, we will write \(A = f_1t + \phi_1\) for one of the sine waves and \(B = f_2 t + \phi_2\) for the other sine wave.

The ring modulator multiplies the two signals to produce \(sin A sin B\). We can try to find a trigonometric identity for this, or we can just look it up in a trigonometry book:

\(2 sin A sin B = cos(A - B) - cos(A + B)\)

This equation tells us that multiplying two sine waves produces two new sine waves (or cosine waves) at the sum and difference of the original frequencies.

The following figure shows the output frequencies (solid lines) for a given input signal pair (dashed lines):

Figure 30. Result of modulating two sine waves A and B

This algorithm has two operating modes which is set with the Mod Mode parameter.

\(L*R\) Mode: In \(L*R\) mode, you supply the modulation and carrier signals as two mono signals on the left and right inputs.

The output in \(L*R\) mode is also mono and you may use the \(L*R\) Pan parameter to pan the output.

The oscillator parameters on parameter pages 2 and 3 will be inactive while in \(L*R\) mode.

The figure below shows the signal flow when in \(L*R\) mode:

Figure 31. L*R mode Ring Modulator

Osc mode: The other modulation mode is Osc.

In Osc mode, the algorithm inputs and outputs are stereo, and the carrier signal for both channels is generated inside the algorithm.

The carrier signal is the sum of 5 oscillators.

4 of the oscillators are simple sine waves and a fifth may be configured to one of a variety of wave shapes.

With all oscillators, you can set level and frequency.

Figure 32. Osc mode Ring Modulator

The configurable oscillator also lets you set the wave shape.

Figure 33. Available Wave Shapes

Sine is simply another sine waveform.
Tri produces a triangular waveform.
Expon produces a waveform with narrow, sharp peaks which seems to rise exponentially from 0.
Pulse produces a series of square pulses where the pulse width can be adjusted with the Osc1PlsWid parameter. When pulse width is 50%, the signal is a square wave. The Osc1PlsWid parameter is active only when the Pulse waveform is selected.

The pulse wave is implemented as a hard clipped sine wave, and, at 50% width, it turns into a sine wave when set to 100% smoothing.

Saw+ and Saw– produce rising and falling sawtooth waveforms.

The sudden change in amplitude of the sawtooth waves develops a more gradual slope with smoothing, ending up as triangle waves when set to 100% smoothing.

The Pulse and Saw waveforms have abrupt, discontinuous changes in amplitude which can be smoothed.

Wet/Dry

0 to 100%wet
The amount of modulated (wet) signal relative to unaffected (dry) signal as a percent.

When in L * R mode, the left input will be used as the dry signal.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

Mod Mode

L * R or Osc
Switches between the two operating modes of the algorithm (L * R and Osc).

The L * R mode treats the left and right inputs as the modulator and carrier signals.

It does not matter which input is left and which is right except to note that only the left signal will be passed through as dry.

L * R Pan

-100 to 100%
The output panning of the both wet and dry signals.

-100% is panned fully left
0% is panned center
100% is panned right.

This control is active only in L * R mode.

Osc1 Lvl

0 to 100%
The level of the configurable oscillator.

0% is off
100% is maximum.

This parameter is active only in Osc mode.

Osc1 Freq

8 to 25088 Hz
The fundamental frequency of the configurable oscillator.

The oscillators can be set through the audible frequencies 8-25088 Hz with one-semitone resolution.

This parameter is active only in Osc mode.

Osc1Shape

Sine, Saw+, Saw–, Pulse, Tri, and Expon
Shape selects the waveform type for the configurable oscillator.

This parameter is active only in Osc mode.

Osc1PlsWid

0 to 100%
When the configurable oscillator is set to Pulse, the PlsWid parameter sets the pulse width as a percentage of the waveform period.

The pulse is a square wave when the width is set to 50%.

This parameter is active only in Osc mode and when the Pulse waveform is selected.

Osc1Smooth

0 to 100%
Smooths the Saw+, Saw–, and Pulse waveforms.

For the sawtooth waves, smoothing makes the waveform more like a triangle wave.
For the Pulse wave, smoothing makes the waveform more like a sine wave.

Sine2 Lvl
Sine3 Lvl
Sine4 Lvl
Sine5 Lvl

0 to 100%
The four sine wave oscillators may have their levels set between 0% (off) and 100% (maximum).

This parameter is active only in Osc mode.

Sine2 Freq
Sine3 Freq
Sine4 Freq
Sine5 Freq

8 to 25088 Hz
The four sine wave oscillators may have their frequencies set with this parameter.

The oscillators can be set through the audible frequencies 8–25088 Hz with one-semitone resolution.

This parameter is active only in Osc mode.

Pitcher

Effects Size : 1

Creates pitch from pitched or non-pitched signal

Pitcher Presets
373 PitcherA
374 PitcherB

The Pitcher algorithm applies a filter which has a series of peaks in the frequency response to the input signal.

The peaks may be adjusted so that their frequencies are all multiples of a selectable frequency, all the way up to 24 kHz.

Applied to Noise

When applied to a sound with a noise-like spectrum (white noise, with a flat spectrum, or cymbals, with a very dense spectrum of many individual components), an output is produced which sounds very pitched, since most of its spectral energy ends up concentrated around multiples of a fundamental frequency.

If the original signal has no significant components at the desired pitch or harmonics, the output level remains low.

If there are enough peaks in the input spectrum (obtained by using sounds with noise components, or combining lots of different simple sounds, especially low pitched ones, or severely distorting a simple sound) then Pitcher can do a good job of imposing its pitch on the sound.

Applied to Sawtooth

Applying Pitcher to sounds such as a single sawtooth wave will tend to not produce much output, unless the sawtooth frequency and the Pitcher frequency match or are harmonically related. Otherwise the peaks in the input spectrum won’t line up with the peaks in the Pitcher filter.

Left and Right Inputs

The left and right inputs are processed independently with common controls of pitch and weighting.

Weight Parameters

The four weight parameters named Odd Wts, Pair Wts, Quartr Wts and Half Wts control the exact shape of the frequency response of Pitcher.

An exact description of what each one does is, unfortunately, impossible, since there is a great deal of interaction between them.

Examples

Pitch setting of 1 kHz, which is close to a value of C6.
Weight settings are listed in brackets following this format: [Odd, Pair, Quartr, Half].

Pitcher at [100, 100, 100, 100]

Figure 34. Pitcher at [100, 100, 100, 100]

Pitcher at [-100, 100, 100, 100]
All peaks are exact multiples of the fundamental frequency set by the Pitch parameter.

This setting gives the most “pitchiness” to the output.

Figure 35. Pitcher at [-100, 100, 100, 100]

Pitcher at [100, 0, 0, 0]
Peaks are odd multiples of a frequency one octave down from the Pitch setting.

This gives a hollow, square-wave-like sound to the output.

Figure 36. Pitcher at [100, 0, 0, 0]

Pitcher at [-100, 0, 0, 0]
There are deeper notches between wider peaks.

Figure 37. Pitcher at [-100, 0, 0, 0]

Pitcher at [0, 100, 100, 100]
There are peaks on odd harmonic multiples and notches on even harmonic multiples of a frequency one octave down from the Pitch setting.

Figure 38. Pitcher at [0, 100, 100, 100]

Pitcher at [50,100,100,100]
Is like [100,100,100,100], except that all the peaks are at (all) multiples of half the Pitch frequency.

Figure 39. Pitcher at [50,100,100,100]

Pitcher at [-50,100,100,100]
Is halfway between [0,100,100,100] and [100,100,100,100].

Figure 40. Pitcher at [-50,100,100,100]

Pitcher at [100, -100, 100, 100]
Is halfway between [0,100,100,100] and [-100,100,100,100].

If the Odd parameter is modulated with an FXMOD, then you can morph smoothly between the [100,100,100,100] and [-100,100,100,100] curves.

Figure 41. Pitcher at [100, -100, 100, 100]

Wet/Dry

0 to 100 %wet
The relative amount of input signal and effected signal that is to appear in the final effect output mix.

When set to 0%, the output is taken only from the input (dry).
When set to 100%, the output is all wet.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

Pitch

C-1 to G9
The fundamental pitch imposed upon the input.

Ptch Offst

-12.0 to 12.0 ST
An offset from the pitch frequency in semitones.

This is also available for adding an additional continuous controller mod like pitch bend.

Odd Wts

-100 to 100 %

Quartr Wts

-100 to 100 %

Pair Wts

-100 to 100 %

Half Wts

-100 to 100 %

Poly Pitcher

Effects Size : 2

Creates pitch from pitched or non-pitched signal—twice.

Poly Pitcher Presets
375 PolyPtVoxChanger
376 HollowPolyPitchr

Poly Pitcher is closely based on Pitcher, and most of the features of Poly Pitcher are covered in the section on Pitcher.

Poly Pitcher is really just a pair of Pitcher algorithms (A and B) using the same inputs and summing to the same outputs.

There is one set of weight parameters (Odd Wts, Pair Wts, Quartr Wts, and Half Wts), which are applied to both pitcher sections. However, the actual pitch settings for the two pitchers can be set independently.

You can also set the relative level of the two pitchers with the A/B Mix parameter.

One last difference from Pitcher is that there are separate pitch offset parameters for left and right channels for both pitchers.

With separate left/right controls for the pitch offset, you can produce a greater sense of stereo separation.

Wet/Dry

0 to 100 %wet
The relative amount of input signal and effected signal that is to appear in the final effect output mix.

When set to 0%, the output is taken only from the input (dry).
When set to 100%, the output is all wet.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

Odd Wts

-100 to 100 %

Quartr Wts

-100 to 100 %

Pair Wts

-100 to 100 %

Half Wts

-100 to 100 %

A/B Mix

0 to 100 %
The relative amount of pitcher A and pitcher B to mix to the final output.

At 0%, only pitcher A can be heard at the output, and at 100%, you can hear only pitcher B.
50% produces equal amounts of both.

Pitch A
Pitch B

C-1 to G9
The fundamental pitch imposed upon the input expressed in semitone scale intervals.

Pitcher A and Pitcher B may be set independently.

PchOff AL
PchOff AR
PchOff BL
PchOff BR

-12.0 to 12.0 ST
An offset from the pitch frequency in semitones.

Not only are the A and B pitchers treated separately, the left and right channels have their own controls for increased stereo separation.

Pitch offset may be useful as a modifiable control resembling pitch bend.

Frequency Offset

Effects Size : 2

Single Side Band Modulation

Frequency Offset Presets
269 Westward Waves	386 Frequency Offset	387 Drum Loosener
388 Drum Tightener

Frequency Offset and MutualFreqOffset perform single side band (SSB) modulation.

Essentially what this means is that every frequency component of your input sound will be offset (in frequency) or modulated by the same amount.

In the Frequency Offset algorithm, if you have the OffsetFreq and Offs Scale parameters set to a frequency of 100 Hz, then all frequencies in your sound will be offset up (or down) by 100 Hz.

Both algorithms produce modulation both up and down and you can control the relative amount of up and down modulation with separate level and pan controls.

The Frequency Offset algorithms are very similar to Ring Modulator, which is a dual side band modulator.

If you set the up and down level parameters to match, the output will be quite close to the Ring Modulator output.

Unlike Ring Modulator however, you can choose to listen to just the up modulation or the down modulation, and not necessarily both.

In addition, you can pan the up and down modulation outputs in different directions (left or right).

Figure 42. Single Side Band Modulation (Frequency Offset)

Frequency Offset is a mono algorithm that modulates your input signal with a pure sine wave.

A sine wave contains a single frequency, so your input signal will be offset in frequency by the frequency of the sine wave.
Provides panning with width of the dry input signals directly to the output

Figure 43. Frequency Offset - Block Diagram

Wet/Dry

0 to 100% wet
The amount of modulated (wet) signal relative to the unaffected (dry) signal as a percent.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

In Lowpass

8 to 25088 Hz
A first-order lowpass filter is provided to reduce the bandwidth of the input signal.

Considering the many new frequency components that will be created, the lowpass filter may help tame the sound.

OffsetFreq

0.00 to 10.00 Hz
The frequency when multiplied with Offs Scale which is the modulation frequency.

The offset or modulation frequency is the frequency in Hz which is added to and/or subtract from all the frequencies of the input signal.

Offs Scale

1 to 25088x
A scale factor which is multiplied with the OffsetFreq parameter to produce the offset or modulation frequency.

DwnOffsLvl

0 to 100%
The level of the down modulated signal.

Negative values polarity invert the signal.

UpOffsLvl

0 to 100%
The level of the up modulated signal.

Negative values polarity invert the signal.

DwnOffsPan

-100 to 100%
The down modulated signal may be panned to the left or right algorithm outputs.

-100% sends the signal to the left output
100% sends the signal to the right output

UpOffsPan

-100 to 100%
The up modulated signal may be panned to the left or right algorithm outputs.

-100% sends the signal to the left output
100% sends the signal to the right output

MutualFreqOffset

Effects Size : 2

Mutual Band Modulation

MutualFreqOffset Presets
389 Vox Honker

MutualFreqOffset modulates the two input signals (left and right) with each other.

If one of the signals is a sine wave, the algorithm behaves like Frequency Offset.

Now imagine that one of the input signals is the sum of two sine waves. Both of the two sine waves will modulate the signal on the other input.

Example

For example, if the two sine waves are at 100 Hz and 200 Hz, upward modulation of another signal at 1000 Hz will produce pitches at 1100 Hz and 1200 Hz.

Obviously this is going to get very complicated to work out when the inputs are more than simple sine waves.

Downward Modulation

With downward modulation, you will hear the pitch drop as you increase the frequency of the input sound.

The downward modulation is a difference (subtraction) in frequencies.

If the difference drops to negative values, the frequency will start to rise again.

It doesn’t matter which frequency gets subtracted from the other, since the result will sound the same.

Example: For example 1000 Hz - 100 Hz = 900 Hz will produce the same pitch as 100 Hz - 1000 Hz = -900 Hz.

Upward Modulation

Similarly, upward modulation is a sum of frequencies and pitch will rise as you increase the frequency of input sound.

Summed frequencies passing the Nyquist rate

In a digital sampled system, frequencies higher than half the sample rate (the Nyquist rate, 24 kHz in Kurzweils) cannot be represented.

When the summed frequencies pass the Nyquist rate, the pitch starts coming back down.

MutualFreqOffset may require extra gain compensation so separate left, right input gain controls and a gain control for the final (wet) output are provided.

MutualFreqOffset provides panning with width of the dry input signals directly to the output

Wet/Dry

0 to 100% wet
The amount of modulated (wet) signal relative to the unaffected (dry) signal as a percent.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

InLowpassL
InLowpassR

In Gain L
In Gain R

Off, -79.0 to 24.0 dB
Two independent gain controls (left and right) to adjust the amplitude of the input signals. (See Wet Gain.)

Wet Gain

Off, -79.0 to 24.0 dB
The gain or amplitude of the modulated (wet) signal.
Produces an output based on multiplying the left and right inputs.

This is very different from adding signals, and controlling levels can be tricky.

Ideally you would set the input gains and the wet gain so that the signal level remains flat when you adjust Wet/Dry while ensuring you hear no internal clipping.

Use Out Gain for overall level control.

DwnOffsLvl

0 to 100%
The level of the down modulated signal.

Negative values polarity invert the signal.

UpOffsLvl

0 to 100%
The level of the up modulated signal.

Negative values polarity invert the signal.

DwnOffsPan

-100 to 100%
The down modulated signal may be panned to the left or right algorithm outputs.

-100% sends the signal to the left output
100% sends the signal to the right output

UpOffsPan

-100 to 100%
The up modulated signal may be panned to the left or right algorithm outputs.

-100% sends the signal to the left output
100% sends the signal to the right output

WackedPitchLFO

Effects Size : 3

LFO based pitch shifter

WackedPitchLFO Presets
219 Nickel Chorus	396 Drum Frightener	398 Fallout
395 Contact	397 Mad Hatter	399 Ascension

Okay, it ain’t pretty, but WackedPitchLFO uses LFO modulated delay lines with cross fades to produce a shift of signal pitch.

You can set the amount of shift in coarse steps of semitones or fine steps of cents (hundredths of a semitone).

This shifter works using the same concepts used to detune a sound in a chorus algorithm.

In a chorus algorithm, an LFO is used to change the length of a delay line.

By smoothly changing a delay line length from long to short to long, the signal is effectively resampled at a new rate causing the pitch to rise and fall.

In the WackedPitchLFO algorithm, the signal level is made to rise and fall in time with the delay line movement so that we only hear signal from the delay line when the pitch is rising (or falling).

By overlapping and adding several delay taps moved by several LFOs, we can then produce a relatively smooth pitch shifted signal.

Tremolo effect

It is possible for sounds coming out of the delay lines to be out of phase, which means that a certain amount of cancellation can occur.

The result sounds like there is a certain amount of tremolo in the pitch shifted signal.

The depth of the tremolo will depend on the pitch of the signal, the rate of the LFO and the amount of pitch shifting—it will be different for every pitch.

The rate of the tremolo is the rate of the LFO.

At higher rates the tremolo can be objectionable.
At slow LFO rates, the pitch shifting is quite clean, though you will hear some flanging.

However longer delay line lengths are needed at slower LFO rates for a given amount of pitch shift. The delays can get quite long, and it is possible to run out of available delay (in which case you will get less pitch shift than you request).

The trade-off is tremolo for delay.

Higher frequency signals will sound better when pitch shifted than lower frequency signals.

Increasing the amount of pitch shift will increase both the amount of tremolo and the amount of delay.

Feedback: You can introduce feedback in WackedPitchLFO.

When you do, the signal can be made to continuously rise (or fall) as it repeatedly passes through the feedback loop.
Pitch Shifter: The pitch shifter is based on delay lines.

Changing the amount of pitch shift will produce large jumps in delay line lengths, and you will hear the jumps as clicks if you are playing a sound while changing the shift amount.

For this reason, the shift amount parameters will not work well as modifiable parameters on an FXMOD page.

Wet/Dry

-100 to 100 %wet
The relative amount of input signal and pitch shifted signal that is to appear in the final effect output mix.

When set to 0%, the output is taken only from the input (dry).
When set to 100%, the output is all wet.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

Feedback

0 to 100 %
By introducing feedback, the pitch can be made to continually rise or fall as the signal makes successive passes through the pitch shifter.

LFO Rate

0.01 to 10.00 Hz
The frequency of the LFOs that drive the pitch shifter.

The pitch shifter produces a certain amount of tremolo that will oscillate based on this rate.

However reducing the rate will increase the delay lengths needed by the pitch shifter.

Shift Crs

-24 to 24 ST
A coarse adjust to the pitch shift amount from -24 to +24 semitones.

The algorithm performs best when the amount of pitch shift is small.

Shift Fine

-100 to 100 ct
A fine adjust to the pitch shift amount from -100 to +100 cents (hundredths of a semitone).

Lowpass

8 to 25088 Hz
A lowpass filter in the algorithm feedback loop.

Use the lowpass to tame some of the higher frequency artifacts.

This is especially important when using feedback.

Highpass

8 to 25088 Hz
A highpass filter in the algorithm feedback loop.

Use the highpass to tame the lower frequencies when using feedback.

Chaos!

Effects Size : 2

Fun with chaos and instability

Chaos! Presets
304 Blown Speaker
381 Ring Linger

The moment you scroll to the Chaos! algorithm, you will discover it is wildly unstable.

Chaos! is a delay feedback algorithm which includes lots of gain with distortion plus plenty of filters tweaking the sound.

Modifying the parameters will often cause the algorithm to jump from one chaotic instability state to another, often unpredictably.

For the most part Chaos! howls and resonates on its own, and while an input signal can affect the output, the effect of the input signal on the output is usually small.

When self- resonating, the sound you can get can be very strange.

It is particularly interesting if you keep modifying the parameters.

What do you use this effect for?
Well, that’s the creative challenge!

You should be very careful with the Out Gain or Drive Cut settings with Chaos!

Best Starting Point: If you start the algorithm in a stable state (not self resonating) and start increasing gains (in the distortion drive or filters), the output level can build.

The feedback can be every bit as unpleasant as putting a microphone next to a loudspeaker!
(There’s an application: simulating PA system feedback!)

Figure 44. Chaos! - Block Diagram

Chaos! is a feedback loop with delay, distortion and lots of filters.

Most of the Kurzweil effects carefully manage levels on feedback loops to prevent instability.

In a digital system, uncontrolled instability will usually rapidly enter digital clipping with full scale signal output. Very nasty.

Chaos! also keeps a lid on levels, preventing digital clipping but allowing instability.

You will still need to cut back on Out Gain (or Drive Cut) to bring the signal down to reasonable levels.

The distortion drive when turned up, will push Chaos! into instability unless Drive Cut is used to hold the level down.

As the sound starts becoming unstable, your input signal will still have a strong effect on the output.

As more and more drive is applied, the self-resonance dominates the output.

The delay length is expressed as a frequency where the length of the delay in seconds is 1/frequency.

Why do this?

A short delay line with a lot of feedback will resonate at a frequency of 1/length of the delay.

It is the resonant behavior of Chaos! which is particularly interesting, which make the delay more naturally expressed as a frequency.

Not only will the delay resonate at its natural frequency (1/length), but you may also hear many overtones (or harmonics).

There is a switch to invert the feedback (FB Invert).

When set to In, FB Invert will cause the natural frequency and its harmonics to be suppressed while frequencies between the harmonics now resonate.

In this case the frequency one octave down and its odd harmonics are resonating.

Filters

In addition to the distortion warmth filter, there are six filters built into the delay line loop:

highpass
lowpass
treble shelf
bass shelf
2 x parametric midrange

Boosting the shelves or mids increases the strength of instability at the boosted frequencies.

Since overall level is controlled, the net effect is to reduce the level of the other frequencies.

Using filters to cut frequencies is similar, but with cut it is possible to remove so much signal that the algorithm drops into stability and stops self-resonating.

The individual elements of Chaos! (filters and so forth) are fairly basic, and you may understand them well.

When put together as the Chaos! algorithm, the interactions become very complex and many of the old rules don’t seem to apply. Keep experimenting!

In/Out

In or Out

When set to In, the effect is active.
When set to Out, the effect is bypassed.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

The output gain is outside and after the feedback loop.

Drive

0 to 96 dB
Sets how high the distortion is to be driven.

The distortion and its drive gain are inside the feedback loop.

Drive Cut

Off, -79.0 to 0.0 dB
Reduces the signal level after the distortion.

By reducing the signal level after the distortion,
Chaos! can be returned to stability while still producing a lot of distortion.

Drive Cut is also inside the feedback loop.

Warmth

8 to 25088 Hz
Warmth affects the character of the distortion.

Warmth reduces (at low settings) the higher frequency distortion components without making the overall signal dull.

Dly FreqC

8 to 25088 Hz
The feedback signal path includes a short delay line which will tend to resonate at a frequency of 1/length of the delay.

The delay length is therefore expressed as the resonant frequency.

All the filters in the feedback loop also add delay, so with more filtering, the resonance tuning will drift flat.

Dly FreqF

-100 to 100 ct
The resonant frequency of the feedback delay line can be tuned sharp or flat in one cent (hundredths of a semitone) increments.

FB Invert

In or Out
The feedback signal can be inverted (subtracted instead of added) so that instead of resonance at the specified frequency and its harmonics, the resonance occurs between those frequencies.

This is like setting resonance one octave lower, but using only the odd harmonics.

Highpass

8 to 25088 Hz
The highpass filter removes frequencies below the specified cut-off frequency.

The filter is first order, cutting signal level at 6 dB per octave of frequency.

When set to the lowest frequency, the filter is performing very little cut of the low frequencies.

When Chaos! is self-resonating, turning up the highpass frequency will cause high frequencies to be emphasized.

Lowpass

8 to 25088 Hz
The lowpass filter removes frequencies above the specified cut-off frequency.

The filter is first order, cutting signal level at 6 dB per octave of frequency.

When set to the highest frequency, the filter is performing very little cut of the high frequencies.

When Chaos! is self-resonating, turning down the lowpass frequency will cause low frequencies to be emphasized.

Bass Gain

-79.0 to 24.0 dB
The amount of boost or cut in decibels to apply to the bass shelf filter inside the feedback loop.

Boost will emphasize frequencies below the filter frequency, while cut will emphasize frequencies above the filter frequency.

Bass Freq

8 to 25088 Hz
The frequency in Hz below which the bass shelf filter performs boost or cut.

Treb Gain

-79.0 to 24.0 dB
The amount of boost or cut in decibels to apply to the treble shelf filter inside the feedback loop.

Boost will emphasize frequencies above the filter frequency, while cut will emphasize frequencies below the filter frequency.

Treb Freq

8 to 25088 Hz
The frequency in Hz above which the treble shelf filter performs boost or cut.

Mid1 Gain
Mid2 Gain

-79.0 to 24.0 dB
The amount of boost or cut in decibels to apply to the midrange parametric filter inside the feedback loop.

Boost will emphasize the specified filter frequency while cut will emphasize all other frequencies.

Mid1 Freq
Mid2 Freq

8 to 25088 Hz
The frequency in Hz at which the midrange parametric filter performs boost or cut.

Mid1 Width
Mid2 Width

0.010 to 5.000 oct
The width of the frequency band in octaves of the midrange parametric filter.

When the filter is set for boost, a narrow band (low settings) will cause the resonating output to approach a pure tone more rapidly.

MonoPitcher

These algorithms each apply a filter that has a series of peaks in the frequency response to the input signal.

The peaks may be adjusted so that their frequencies are all multiples of a selectable frequency, all the way up to 24 kHz.

The graphs below show Pt PkSplit going from 0% to 100%, for a Pt Pitch of 1 kHz (roughly. C6), and Pt PkShape set to 0.

Figure 45. Response of Pitcher with different PkSplit settings.

That a Pt PkSplit of 100% gives only odd multiples of a fundamental that is one octave down from no splitting.

The presence of only odd multiples will produce a hollow sort of sound, like a square wave (which also only has odd harmonics).

Curiously enough, at a Pt PkSplit of 50% we also get odd multiples of a frequency that is now two octaves below the original Pitch parameter.

In general, most values of PkSplit will give peak positions that are not harmonically related.

The figures below show Pt PkShape of -1.0, 0.0, and 1.0, for a Pitch of C6 and a PkSplit of 0%.

Figure 46. Response of Pitcher with different PkShape settings

Sawtooth: Applying Pitcher to sounds such as a single sawtooth wave will tend to not produce much output, unless the sawtooth frequency and the Pitcher frequency match or are harmonically related, because otherwise the peaks in the input spectrum won’t line up with the peaks in the Pitcher filter.
Best Pitcher Results: If there are enough peaks in the input spectrum (obtained by using sounds with noise components, or combining lots of different simple sounds, especially low pitched ones, or severely distorting a simple sound) then Pitcher can do a good job of imposing its pitch on the sound.
Other Settings: Multiple Pitcher algorithms can be run to produce chordal output.

At extremely low Pitch settings, the effect begins to sound more like a multi-tap delay, but this can be pretty cool, too.
Vocoder-like Effect: A vocoder-like effect can be produced, although in some sense it works in exactly an opposite way to a real vocoder.

A real vocoder will superimpose the spectrum of one signal (typically speech) onto a musical signal (which has only a small number of harmonically related spectral peaks).

Pitcher takes an input such as speech, and then picks out only the components that match a harmonic series, as though they were from a musical note.

MonoPitcher+Chor

Effects Size : 2

Pitcher and Chorus combination

MonoPitcher+Chor Presets
377 Pitcher+Chorus

See the Chorus section for the Chorus parameters.

Wet/Dry

100 to 100 %wet
This is a simple mix of the pitched chorused signal relative to the dry input signal.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

Mix Pitchr

-100 to 100 %
The amount of the pitcher signal to be sent directly to the output as a percent.

Any signal that this parameter sends to the output does not get sent to the chorus.

Mix Chorus

-100 to 100 %
The amount chorus signal to send to the output as a percent.

Pt/Dry→Ch

0 to 100 %
The relative amount of Pitcher signal to dry signal to send to the chorus or flanger.

- At 0% the dry input signal is routed to the chorus or flanger.
- At 100%, the chorus receives its input entirely from the Pitcher.

Pt Inp Bal

-100 to 100 %
Since this is a mono algorithm, an input balance control is provided to mix the left and right inputs to the Pitcher.

-100% is left only
0% is left
100% is right only

Pt Out Pan

-100 to 100 %
Pans the output from:

left (-100%)
center (0%)
right (100%).

Pt Pitch

C-1 to G 9
The “fundamental” frequency of the Pitcher output.

This sets the frequency of the lowest peak in terms of standard note names.

All the other peaks will be at multiples of this pitch.

Pt PkSplit

0 to 100 %
Splits the pitcher peaks into two peaks, which both move away from their original unsplit position, one going up and the other down in frequency.

At 0% there is no splitting; all peaks are at multiples of the fundamental.
At 100% the peak going up merges with the peak going down from the next higher position.

Pt Offset

-12.0 to 12.0 ST
An offset in semitones from the frequency specified in Pitch.

Pt PkShape

-1.0 to 1.0
Controls the shape of the pitcher spectral peaks.

0.0 gives the most “pitchiness” to the output, in that the peaks are narrow, with not much energy between them.
-1.0 makes the peaks wider.
1.0 brings up the level between the peaks.

MonoPitcher+Flan

Effects Size : 2

Pitcher and Flanger combination

MonoPitcher+Flan Presets
378 Pitcher+Flange

Go to the Flanger section for the Flanger parameters.

Wet/Dry

100 to 100 %wet
This is a simple mix of the pitched flanged signal relative to the dry input signal.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

Mix Pitchr

-100 to 100 %
The amount of the pitcher signal to be sent directly to the output as a percent.

Any signal that this parameter sends to the output does not get sent to the flanger.

Mix Flange

-100 to 100 %
The amount of the flanger signal to send to the output as a percent.

Pt/Dry→Fl

0 to 100 %
The relative amount of Pitcher signal to dry signal to send to the flanger.

At 0% the dry input signal is routed to the flanger.
At 100%, the flanger receives its input entirely from the Pitcher.

Pt Inp Bal

-100 to 100 %
Since this is a mono algorithm, an input balance control is provided to mix the left and right inputs to the Pitcher.

-100% is left only
0% is left
100% is right only

Pt Out Pan

-100 to 100 %
Pans the output from:

left (-100%)
center (0%)
-right (100%).

Pt Pitch

Pt PkSplit

0 to 100 %
Splits the pitcher peaks into two peaks, which both move away from their original unsplit position, one going up and the other down in frequency.

At 0% there is no splitting; all peaks are at multiples of the fundamental.
At 100% the peak going up merges with the peak going down from the next higher position.

Pt Offset

-12.0 to 12.0 ST
An offset in semitones from the frequency specified in Pitch.

Pt PkShape

-1.0 to 1.0
Controls the shape of the pitcher spectral peaks.

0.0 gives the most “pitchiness” to the output, in that the peaks are narrow, with not much
energy between them.
-1.0 makes the peaks wider.
1.0 brings up the level between the peaks.

Fl LFO cfg

Sets the user interface mode for controlling each of the four flange LFOs.

Fl LRPhase

0.0 to 360.0 deg
Controls the relative phase between left channel LFOs and right channel LFOs.

In Dual1Tap mode, however, this parameter is accurate only when Fl Rate 1 and Fl Rate 2 are set to the same speed, and only after the Fl LFO cfg parameter is moved, or the algorithm is called up.

Fl Phase 1
Fl Phase 2

0.0 to 360.0 deg
These adjust the corresponding LFO phase relationships between themselves and the internal beat clock.

MiniVerbs

MiniVerb

Effects Size : 1

Algorithm Type : Rvrb

Compact reverb used in many combination algorithms.

Rvrb

MiniVerb Presets
4 NiceLittleBooth	17 Percussive Room	51 Medium Hall
11 Viewing Booth	18 SmallStudioRoom	55 Grandiose Hall
13 Add Ambience	41 Brass Chamber	56 Elegant Hall
15 BrightSmallRoom	42 Sax Chamber	57 Bright Hall
16 Bassy Room	43 Plebe Chamber	58 Ball Room

MiniVerb is a versatile stereo reverb found in many combination algorithms, but is equally useful on its own because of its small size.

The main control for this effect is the Room Type parameter.

Room Type changes the structure of the algorithm to simulate many carefully crafted room types and sizes.

Spaces characterised as booths, small rooms, chambers, halls and large spaces can be selected.

Figure 47. MiniVerb - Simplified Block Diagram

Room Types: Each Room Type incorporates different diffusion, room size and reverb density settings.

Room Types were designed to sound best when Diff Scale, Size Scale and Density are set to the default values of 1.00x.

If you want a reverb to sound perfect immediately, set the Diff Scale, Size Scale and Density parameters to 1.00x, pick a Room Type and you’ll be on the way to a great sounding reverb.

But if you want to experiment with new reverb ﬂavors, changing the scaling parameters away from 1.00x can cause a subtle (or drastic!) coloring of the carefully crafted Room Types.

Diffusion: Diffusion characterizes how the reverb spreads the early reﬂections out in time.

At very low settings of Diff Scale, the early reﬂections start to sound quite discrete, and at higher settings the early reﬂections are seamless.

Density: Density controls how tightly the early reﬂections are packed in time.

Low Density settings have the early reﬂections grouped close together, and higher values spread the reflections for a smoother reverb.

Dual MiniVerb

Effects Size : 2

Algorithm Type : Rvrb

Panning can be maintained in the reverb tails.

Rvrb

Dual MiniVerb Presets
119 L:SmlRm R:Hall

Dual MiniVerb gives you independent reverbs on both channels which has obvious beneﬁts for mono material.

With stereo material, any panning or image placement can be maintained, even in the reverb tails!

This is pretty unusual behavior for a reverb, since even real halls will rapidly delocalize acoustic images in the reverberation.

Since maintaining image placement in the reverberation is so unusual, you will have to carefully consider whether it is appropriate for your particular situation.

To use Dual MiniVerb to maintain stereo signal placement, set the reverb parameters for both channels to the same values. The Dry Pan and Wet Bal parameters should be fully left (-100%) for the left MiniVerb and fully right (100%) for the right MiniVerb.

Dual MiniVerb has a full MiniVerb, including Wet/Dry, Pre Delay and Out Gain controls, dedicated to both the left and right channels.

In Figure 2, the two blocks labeled MiniVerb contain a complete copy of the contents of Figure 1.

Figure 48. Dual MiniVerb - Simplified Block Diagram

Wet/Dry

0 to 100% wet
A simple mix of the reverb sound with the dry .

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

Rvrb Time

0.5 to 30.0 s, Inf
The reverb time displayed is accurate for normal settings of the other parameters.

Normal Settings

HF Damping = 25088kHz
Diff Scale = 1.00x
Room Scale = 1.00x
Density = 1.00x

Changing Rvrb Time to Inf creates an infinitely sustaining reverb.

HF Damping

8 to 25088 Hz
Reduces high frequency components of the reverb above the displayed cutoff frequency.

Removing higher reverb frequencies can often make rooms sound more natural.

L/R Pre Dly

0 to 620 ms
The delay between the start of a sound and the output of the first reverb reflections from that sound.

Longer predelays can help make larger spaces sound more realistic.

Longer times can also help improve the clarity of a mix by separating the reverb signal from the dry signal, so the dry signal is not obscured.

Likewise, the wet signal will be more audible if delayed, and thus you can get by with a dryer mix while maintaining the same subjective wet/dry level.

Room Type

Booth1, Booth2, Booth3, Room1, Room2, Room3, Chamber1, Chamber2, Chamber 3, Plate1, Plate2, Hall1, Hall2, Hall3, Hall4, Large1, Large 2, Large3, Large4
Changes the configuration of the reverb algorithm to simulate a wide array of carefully designed room types and sizes.

This parameter effectively allows you to have several different reverb algorithms only a parameter change away.

Smaller Room Types will sound best with shorter Rvrb Times, and vice versa.

This parameter changes the structure of the reverb algorithm, you do not want to modulate it.

Diff Scale

0.00 to 2.00x
A multiplier which affects the diffusion of the reverb.

At 1.00x, the diffusion will be the normal, carefully adjusted amount for the current Room Type.

Altering this parameter will change the diffusion from the preset amount.

Size Scale

0.00 to 4.00x
A multiplier which changes the size of the current room.

At 1.00x, the room will be the normal, carefully tweaked size of the current Room Type.

Altering this parameter will change the size of the room, and thus will cause a subtle coloration of the reverb (since the room’s dimensions are changing).

Density

0.00 to 4.00x
A multiplier which affects the density of the reverb.

At 1.00x, the room density will be the normal, a carefully set amount for the current Room Type.

Altering this parameter will change the density of the reverb, which may color the room slightly.

Wet Bal

-100 to 100%
In Dual MiniVerb, two mono signals (left and right) are fed into two separate stereo reverbs.

If you center the wet balance (0%), the left and right outputs of the reverb will be sent to the final output in equal amounts. This will add a sense of spaciousness.

Dry Pan

-100 to 100%
Adjusts the output pan position of the input dry signals.

The dry level is controlled with Wet/Dry.

0% pans to center
Negative values pan left
Positive values pan right

Gated MiniVerb

Effects Size : 2

Algorithm Type : GRvb

Phil Collins "In The Air Tonight" drum effect

Gated MiniVerb Presets
108 Gated Reverb
109 Gate Plate

Gated MiniVerb uses the compact MiniVerb reverb algorithm followed by a gate.

The reverb component of this algorithm is covered in MiniVerb

A gate behaves like an on off switch for a signal.

The gate turns the output of the reverb on and off based on the amplitude of the input signal.

One or both input channels is used to control whether the switch is on (gate is open) or off (gate is closed).

The on/off control is called “side chain” processing. You select which of the two input channels or both is used for side chain processing. When you select both channels, the sum of the left and right input amplitudes is used.

Gate Threshold: The gate is opened when the side chain amplitude rises above a level that you specify with the Gate Thres parameter.

The gate will stay open for as long as the side chain signal is above the threshold.

Signal Below Threshold
When the signal drops below the threshold, the gate will remain open for the time set with the Gate Time parameter.
At the end of the Gate Time, the gate closes.
Signal Above Threshold
When the signal rises above threshold, it opens again. What is happening is that the gate timer is being constantly retriggered while the signal is above threshold.

Gate Duck: If Gate Duck is turned on, then the behavior of the gate is reversed.

The gate is open while the side chain signal is below threshold, and it closes when the signal rises above threshold.
Gate Atk and Gate Rel: If the gate opened and closed instantaneously, you would hear a large digital click, like a big knife switch was being thrown.

Obviously that’s not a good idea, so Gate Atk (attack) and Gate Rel (release) parameters are use to set the times for the gate to open and close.

More precisely, depending on whether Gate Duck is Off or On, Gate Atk sets how fast the gate opens or closes when the side chain signal rises above the threshold.

The Gate Rel sets how fast the gate closes or opens after the gate timer has elapsed.
Signal Delay: The Signal Dly parameter delays the signal being gated, but does not delay the side chain signal.

By delaying the main signal relative to the side chain signal, you can open the gate just before the main signal rises above threshold.

It’s a little like being able to pick up the telephone before it rings.

Wet/Dry

0 to 100% wet
A simple mix of the reverb sound with the dry sound.

When set fully dry (0%), the gate is still active.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

An overall level control of the effect’s output (applied after the gate)

Rvrb Time

0.5 to 30.0 s, Inf
The reverb time displayed is accurate for normal settings of the other parameters.

Normal Settings

HF Damping = 25088kHz
Diff Scale = 1.00x
Room Scale = 1.00x
Density = 1.00x

Changing Rvrb Time to Inf creates an infinitely sustaining reverb.

HF Damping

8 to 25088 Hz
Reduces high frequency components of the reverb above the displayed cutoff frequency.

Removing higher reverb frequencies can often make rooms sound more natural.

L/R Pre Dly

0 to 620 ms
The delay between the start of a sound and the output of the first reverb reflections from that sound.

Longer predelays can help make larger spaces sound more realistic.

Longer times can also help improve the clarity of a mix by separating the reverb signal from the dry signal, so the dry signal is not obscured.

Likewise, the wet signal will be more audible if delayed, and thus you can get by with a dryer mix while maintaining the same subjective wet/dry level.

Room Type

This parameter effectively allows you to have several different reverb algorithms only a parameter change away.

Smaller Room Types will sound best with shorter Rvrb Times, and vice versa.

This parameter changes the structure of the reverb algorithm, you do not want to modulate it.

Diff Scale

Size Scale

Density

Gate Thres

-79.0 to 0.0 dB
The input signal level in dB required to open the gate (or close the gate if Gate Duck is on).

Gate Duck

In or Out
When set to Off, the gate opens when the signal rises above threshold and closes when the gate time expires. When set to On, the gate closes when the signal rises above threshold and opens when the gate time expires.

Gate Time

Gate Atk

0.0 to 228.0 ms
The attack time for the gate to ramp from closed to open (reverse if Gate Duck is On) after the signal rises above threshold.

Gate Rel

0 to 3000 ms
The release time for the gate to ramp from open to closed (reverse if Gate Duck is On) after the gate timer has elapsed.

GateSigDly

0.0 to 25.0 ms
The delay in milliseconds (ms) of the reverb signal relative to the side chain signal. By delaying the reverb signal, the gate can be opened before the reverb signal rises above the gating threshold.

Suppressor

HarmonicSuppress

Effects Size : 2

Removes harmonically related bands of frequencies.

HarmonicSuppress Presets
345 OddHarmSupress
346 60Hz KillBuzz

HarmonicSuppress is based on comb filtering.

A simple filter which removes harmonically related frequency bands with a spectrum which looks like a comb.

Harmonics: With the Harmonics parameter, you can choose to expand the odd harmonics (including the fundamental) or even harmonics (not including the fundamental) or all harmonics.

Choosing all harmonics is the same as choosing even harmonics at half the frequency.

Figure 49. HarmonicSuppress - Filtering

The algorithm expand the signal in the specified band(s) (reduce the signal’s gain) when the signal falls below the expansion threshold in the specified band(s).

Side Chain Input: You can select which channel, left (L), right (R) or the larger of the two (L & R) is used to control the expansion (side chain processing) with the SC Input parameter.
Expansion Channel: You can also select which channel is actually expanded, again left (L), right (R) or both (L & R) using the ExpandChan parameter.
Ratio: The amount of expansion is expressed as an expansion ratio.

Expanding a signal reduces its level below the threshold. The expansion ratio is the inverse of the slope of the expander input/output characteristic.

An expansion ratio of 1:1 will have no effect on the signal.

A zero ratio (1:infinity), will expand all signal levels below the threshold level to the null or zero level. (This expander expands to 1:17 at most, but that’s a lot.)
Threshold: Thresholds are expressed as a decibel level relative to digital full-scale (dBFS) where 0 dBFS is digital full-scale and all other available values are negative.

Figure 50. HarmonicSuppress - Simplified Block Diagram

To determine how much to expand the signal, the expander must measure the signal level.

Since musical signal levels will change over time, the expansion amounts must change as well.

You can control how fast the expansion changes in response to changing signal levels with the attack and release time controls.

Attack Time: The attack time is defined as the time for the expansion to turn off when the signal rises above the threshold. This time should be very short for most applications.
Release Time: The expander release time is the time for the signal to expand down after the signal drops below threshold. The expander release time may be set quite long.

An expander may be used to suppress background noise in the absence of signal, thus typical expander settings use a fast attack (to avoid losing real signal), slow release (to gradually fade out the noise), and the threshold set just above the noise level. You can set just how far to drop the noise with the expansion ratio.

The signal being expanded may be delayed relative to the side chain processing. The delay allows the signal to stop being expanded just before an attack transient arrives. Since the side chain processing “knows” what the input signal is going to be before the main signal path does, it can tame down an attack transient by releasing the expander before the attack actually happens.

A meter is provided to display the amount of gain reduction that is applied to the signal as a result of expansion.

In/Out

In or Out
When set to In the expander is active.
When set to Out the band suppressor is bypassed.

Out Gain

Off, -79.0 to 24.0 dB
The output gain parameter may be used to increase the gain by as much as 24 dB, or reduce the gain to nothing.

The Out Gain parameter does not control the signal level when the algorithm is set to Out.

Fund FreqC

8 to 25088 Hz
The fine frequency control sets the fundamental frequency of the harmonic structure to be expanded.

Since the filter is a comb filter, the separation between harmonically related expansion bands is also controlled.

Fund FreqF

-100 to 100 ct
The fine frequency control sets the fundamental frequency of the harmonic structure to be expanded.

Since the filter is a comb filter, the separation between harmonically related expansion bands is also controlled.

Harmonics

Even, Odd, All
Sets the harmonic structure of the expansion comb filter.

When set to Even, only the even harmonics of the specified fundamental frequency (including any dc signal level) are expanded, with no expansion of the fundamental.
When set to Odd, the odd harmonics, including the fundamental, are expanded.
The All setting expands all even and odd harmonics including any dc signal level. The All setting is the same as the Even when the Even frequency is set to half the value of All.

SC Input

L, R, L & R
Select the input source channel for side-chain processing—left (L), right ® or both
(L & R).

When set to L & R, the maximum of left and right amplitudes is used.

ExpandChan

L, R, L & R
Select which input channel will receive expander processing—left, right or both.

If you select left or right, the opposite channel will pass through unaffected.

Atk Time

0.0 to 228.0 ms
The time for the expander to increase the gain of the signal (turns off the expander) after the signal rises above threshold.

Rel Time

0 to 3000 ms
The time for the expander to reduce the signal level when the signal drops below the threshold (turning on expansion).

Smooth Time

0.0 to 228.0 ms
A lowpass filter in the control signal path. It is intended to smooth the output of the expander’s envelope detector.

Smoothing will affect the attack or release times when the smoothing time is longer than one of the other times.

Signal Dly

0.0 to 25.0 ms
The time in ms by which the input signal should be delayed with respect to expander side chain processing (i.e. side chain predelay).
This allows the expansion to appear to turn off just before the signal actually rises.

Ratio

1:1.0 to 1:17.0
The expansion ratio.

High values (1:17 max) are highly expanded, low values (1:1 min) are moderately expanded.

Threhold

-79.0 to 0.0 dB
The expansion threshold level in dBFS (decibels relative to full scale) below which the signal begins to be expanded.

MakeUpGain

Off, -79.0 to 24.0 dB
Provides an additional control of the output gain.

The Out Gain and MakeUpGain controls are additive (in decibels) and together may provide a maximum of 24 dB boost to offset gain reduction due to expansion.

Stereo Tremolo

In the classical sense, a tremolo is the rapid repetition of a single note created by an instrument.

Early music synthesists imitated this by using an LFO to modulate the amplitude of a tone. This is the same concept as amplitude modulation, except that a tremolo usually implies that the modulation rate is much slower.

Tremolo and Tremolo BPM are 1 unit sized stereo tremolo effects.

Tremolo BPM

Effects Size : 1

Tempo synced tremolo

Tremolo BPM Presets
270 Tremolo BPM
271 Fast Tremolo BPM

Tremolo

Effects Size : 1

Frequency controlled tremolo

Tremolo Presets
272 Tremolo in Hz

LFO Shapes: Tremolo and Tremolo BPM provide six different LFO shapes.

Figure 51. LFO Shapes

L/R Phase: L/R Phase flips the LFO phase of the left channel for auto-balancing applications.
50% Weight: The 50% Weight parameter bends the LFO shape up or down relative to its -6dB point.

At 0dB, there is no change to the LFO shape.

Positive values will bend the LFO up towards unity, while negative values will bend it down towards full attenuation.

LFO metering: LFO metering can be viewed on the bottom of the Para2 page.

Figure 52. Action of the "50% Weight" parameter

Tremolo also includes an LFO rate scale for AM synthesis, and Tremolo BPM provides tempo based LFO syncing including system syncing.

In/Out

In or Out
When set to In the algorithm is active.
When set to Out the algorithm is bypassed.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

Tempo

System, 0 to 255 BPM
For Tremolo BPM.
Basis for the rate of the LFO, as referenced to a musical tempo in BPM (beats per minute).

When this parameter is set to System, the tempo is locked to the internal sequencer tempo or to incoming MIDI clocks. In this case, FXMods (FUNs, LFOs, ASRs etc.) will have no effect on the Tempo parameter.

LFO Rate

0 to 10.00 Hz
For Tremolo.
The speed of the tremolo LFO in cycles per second.

LFO Rate

0 to 12.00 x
For Tremolo BPM.
The number of LFO cycles in one beat relative to the selected Tempo.

For example, 1.00x means the LFO repeats once per beat; 2.00x twice per beat, and so on.

Rate Scale

1 to 25088 x
For Tremolo.
This multiplies the speed of the LFO rate into the audio range. When above 19x, the values increment in semitone steps. These steps are accurate when LFO Rate is set to 1.00 Hz.

LFO Phase

For Tremolo BPM.
This parameter shifts the phase of the tremolo LFO relative to an internal beat reference.

It is most useful when Tempo is set to System and LFO Phase controls the phase of the LFO relative to MIDI clock.

Depth

0 to 100 %
This controls the amount of attenuation applied when the LFO is at its deepest excursion point.

LFO Shape

Sine, Saw+, Saw–, Pulse, Tri, Expon
The waveform type for the LFO.

PulseWidth

0 to 100 %
When the LFO Shape is set to Pulse, this parameter sets the pulse width as a percentage of the waveform period.

The pulse is a square wave when the width is set to 50%.

This parameter is active only when the Pulse waveform is selected.

50% Weight

-6 to 3 dB
The relative amount of attenuation added when the LFO is at the -6dB point.

This causes the LFO shape to bow up or down depending on whether this parameter is set positive or negative

L/R Phase

In or Out
LFO phase relationship of the left channel.

Flipping the left channel’s LFO out of phase causes the effect to become an auto-balancer.

Allpass Filter Phaser

The allpass phasers are algorithms that use allpass filters to achieve a phaser effect.

These algorithms do not have built in LFOs, so like Manual Phaser, any motion must be supplied with an FXMod.

Allpass Filter: A phaser uses a special filter called an allpass filter to modify the phase response of a signal’s spectrum without changing the amplitude of the spectrum.

As the term “allpass filter” suggests, the filter by itself does not change the amplitude response of a signal passing through it. An allpass filter does not cut or boost any frequencies. An allpass filter does cause some frequencies to be delayed a little in time, and this small time shift is also known as a phase change. The frequency where the phase change has its greatest effect is a parameter that you can control.

By modulating the frequency of the phaser, you get the swishy phaser sound. With a modulation rate of around 6 Hz, an effect similar to vibrato may be obtained, but only in a limited range of filter frequencies.

By adding the phaser output to the dry input using, for example, a Wet/Dry parameter, you can produced peaks and notches in the frequency response.

At frequencies where the phaser is “in phase” with the dry signal, the signal level doubles (or there is a 6 dB level increase approximately).
At frequencies where the phaser and dry signals are “out of phase,” the two signals cancel each other out and there is a notch in the frequency response.

You can get a complete notch when Wet/Dry is set to 50%.

If subtraction is used instead of addition by setting Wet/Dry to -50%, then the notches become peaks and the peaks become notches.

High Order Allpass Filters: Unlike the other phasers, the allpass phasers use high order allpass filters.

The order of the allpass filters sets the number of notches that will appear in the frequency response when the dry and filtered signals are mixed.

The number of notches in the frequency response ranges from 3 to 6 for Allpass Phaser 3 and 7 to 10 for Allpass Phaser 4.

Allpass Phaser 3 and Allpass Phaser 4 are identical except for the number of notches and effect size usage.
Phaser motion: As mentioned earlier, allpass phasers leave the phaser motion up to you, so they have no built in LFOs.

To get phaser motion, you have to change the filter center frequencies (left and right channels) yourself. The best way to do this is with an FXMod. (eg. FXLFO)
Feedback: When feedback is used, it can greatly exaggerate the peaks and notches, producing a much more resonant sound with notches and peaks that are not harmonically related.
Cross-Coupling: Cross-coupling (XCouple) the feedback between the left an right channels increases the complexity of the frequency response.

When a lot of feedback is used, the non-harmonic structure produces very bell-like tones, particularly with XCouple set to 100%. (Don’t modulate the frequencies to get this effect.)

Try experiments using different allpass orders for the feedback, different frequency arrangements, changing the sign (+/-) of the feedback (Fdbk Level) parameter, and different input sources.
eg. drums are a good starting point

Figure 53. Allpass Phaser

The allpass phaser algorithms use a typical signal routing with wet/dry and cross-coupled feedback. A different number of notches may be chosen for the feedback path than for the direct output.

Allpass Phaser 3

Effects Size : 3

xxxxx

Allpass Phaser 3 Presets
383 Hip Hop Aura
384 Woodenize
385 Marimbafication

Allpass Phaser 4

Effects Size : 4

xxxxx

Allpass Phaser 3 Presets
264 Static Phaser 5

Wet/Dry

-100 to 100% wet
The amount of phaser (wet) signal relative to unaffected (dry) signal as a percent.

Out Gain

Off, -79.0 to 24.0 dB
The output gain in decibels (dB) to be applied to the combined wet and dry signals.

Fdbk Level

-100 to 100%
The phaser output can be added back to its input to increase the phaser resonance.

Negative values polarity invert the feedback signal.

XCouple

0 to 100%
Determines how much of the right feedback signal to feed into the left input channel and how much left feedback to feed into the right input channel.

When increasing cross-coupling, the amount of feedback from one channel into its own input is reduced, so that at 100% the left feeds back entirely to the right channel and vice versa.

L/R CenterFreq

8 to 25088 Hz
The nominal center frequency of the phaser filter.

The frequency LFO modulates the phaser filter centered at this frequency.

FB APNotch

3 to 6 or 7 to 10
The number of notches the allpass filter can produce when summed with a dry signal.

Used in the feedback loop. Higher values produce more resonant peaks, for a more complex resonant structure.

OutAPNotch

3 to 6 or 7 to 10
The number of notches the allpass filter can produce when summed with a dry signal.

Used on the algorithm output.

Higher values produce a steeper, longer phase response resulting in more peaks and notches when combined with the dry signal.

Comb Filter

A comb filter produces a series of evenly spaced notches or resonant peaks in the frequency response.

The comb filter gets its name from the comb-like appearance of the frequency response.

Figure 54. Frequency response of comb filter

Notches: A comb filter producing notches is created by adding the input signal to a delayed (and possibly attenuated) version of the input signal.

Figure 55. Frequency response of comb filter

Peaks: To produce peaks, the output signal is passed through a short delay and attenuation (level reduction) and added to the input signal to produce a delay feedback.

Barberpole Comb

Effects Size : 4

Constantly rising or falling frequency

Barberpole Comb Presets
265 Slow Riser	268 All The Way Down
266 BarberPole Notch
267 BarberPole Peak

The Barberpole Comb is a comb filter with a constantly rising or falling frequency.

The Barberpole Comb can be configured to produce either notches or peaks. There is a twist to the Barberpole Comb algorithm in that the notches or peaks can be made to shift up (or down) in frequency. As the notches or peaks shift up, the highest notches or peaks go away while new notches or peaks appear at the lowest frequencies.

Figure 56. Barberpole Comb Filter - at different instances in time

In/Out

In or Out
When set to In the algorithm is active.

When set to Out the algorithm is bypassed.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

Rise/Fall

Rise or Fall
When set to Rise, the comb filter frequencies rotate to higher frequencies.

When set to Fall, the comb filter frequencies rotate to lower frequencies.

Rate

0.00 to 10.00 Hz
The LFO rate at which the comb filters rise or fall through a complete cycle of
frequencies.

Comb Freq

8 to 25088 Hz
The frequency separation of the notches or peaks in the comb filter.

Notch/Peak

Notch or Peak
The comb filter can be constructed to produce a series of notches in the frequency response or a series on resonant peaks.

Depth/Res

0 to 100%
The depth of the notches (set Notch/Peak to Notch) or the height of the resonant peaks (set Notch/Peak to Peak) can be set with Depth/Res.

A setting of 100% produces the deepest notches or the highest peaks.

L/R Phase

-180.0 to 178.5 deg
The left and right channels may place the comb filter notches or peaks at different frequencies, controlled by the relative phase angles of the LFOs controlling the comb frequency rotations for the left and right channels.

Enhancers

2 Band Enhancer

Effects Size : 1

Boost bass energy and brighter high frequencies

2 Band Enhancer Presets
368 2 Band Enhancer

The 2 Band Enhancer modifies the spectral content of the input signal primarily by brightening signals with little or no high frequency content, and boosting pre-existing bass energy.

First, the input is non-destructively split into 2 frequency bands using 6 dB/oct highpass and lowpass filters.
The highpassed band is processed to add additional high frequency content by using a nonlinear transfer function in combination with a high shelving filter.
Each band can then be separately delayed to sample accuracy and mixed back together in varying amounts.
One sample of delay is approximately equivalent to 20 microseconds, or 180 degrees of phase shift at 24 khz.

Using what we know about psychoacoustics, phase shifting, or delaying certain frequency bands relative to others can have useful affects without adding any gain.

Hi Delay / Lo Delay: In this algorithm, delaying the lowpassed signal relative to the highpass signal brings out the high frequency transient of the input signal giving it more definition.

Conversely, delaying the highpass signal relative to the lowpass signal brings out the low frequency transient information which can provide punch.
Hi Xfer: The transfer applied to the highpass signal can be used to generate additional high frequency content when set to a non-zero value. As the value is scrolled away from 0, harmonic content is added in increasing amounts to brighten the signal.

In addition to adding harmonics, positive values impose a dynamically compressed quality, while negative values sound dynamically expanded. This type of compression can bring out frequencies in a particular band even more.
The expanding quality is particularly useful when trying to restore transient information.

In/Out

In or Out
When set to In the effect is active.
When set to Out the effect is bypassed.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

CrossOver

17 to 25088 Hz
Adjusts the -6dB crossover point at which the input signal will be divided into the
highpass band and a lowpass bands.

Hi Drive

Off, -79.0 to 24.0 dB
Adjusts the gain into the transfer function. The affect of the transfer can be intensified
or reduced by respectively increasing or decreasing this value.

Hi Xfer

-100 to 100 %
The intensity of the transfer function.

Hi Shelf F

8 to 25088 Hz
The frequency of where the high shelving filter starts to boost or attenuate.

Hi Shelf G

-96 to 24 dB
The boost or cut of the high shelving filter.

Hi Delay

0 to 500 samp
Adjusts the number of samples the highpass signal is delayed.

Hi Mix

Off, -79.0 to 24.0 dB
Adjusts the output gain of the highpass signal.

Lo Delay

0 to 500 samp
Adjusts the number of samples the lowpass signal is delayed.

Lo Mix

Off, -79.0 to 24.0 dB
Adjusts the output gain of the lowpass signal.

3 Band Enhancer

Effects Size : 2

Boost the high, mid, and low frequencies

3 Band Enhancer Presets
369 3 Band Enhancer
370 Extreem Enhancer

The 3 Band *Enhancer modifies the spectral content of the input signal by boosting existing spectral content, or stimulating new ones.

First, the input is non-destructively split into 3 frequency bands using 6 dB/oct highpass and lowpass filters
The high and mid bands are separately processed to add additional high frequency content by using two nonlinear transfer functions.
The low band is processed by a single nonlinear transfer to enhance low frequency energy.
Each band can also be separately delayed to sample accuracy and mixed back together in varying amounts.
One sample of delay is approximately equivalent to 20 microseconds, or 180 degrees of phase shift with the 24 khz sampling rate.

Using what we know about psychoacoustics, phase shifting, or delaying certain frequency bands relative to others can have useful affects without adding any gain.

Hi Delay / Lo Delay: In this algorithm, delaying the lower bands relative to higher bands brings out the high frequency transient of the input signal giving it more definition.

Conversely, delaying the higher bands relative to the lower bands brings out the low frequency transient information which can provide punch.

Figure 57. 3 Band Enhancer - Left Channel

Lo Xfer / Mid Xfer /Hi Xfer: The nonlinear transfers applied to the high and mid bands can be used to generate additional high and mid frequency content when Xfer1 and Xfer2 are set to non-zero values. As the value is scrolled away from 0, harmonic content is added in increasing amounts.

In addition, setting both positive or negative will respectively impose a dynamically compressed or expanded quality. This type of compression can bring out frequencies in a particular band even more.

The expanding quality is useful when trying to restore transient information.

The low band has a nonlinear transfer that requires only one parameter. Its affect is controlled similarly.

More complex dynamic control can be obtained by setting these independent of each other.

Setting one positive and the other negative can even reduce the noise floor in some applications.

In/Out

In or Out
When set to In the effect is active.
When set to Out the effect is bypassed.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

CrossOver1

17 to 25088 Hz
Adjusts one of the -6dB crossover points at which the input signal will be divided into the
high, mid and low bands.

CrossOver2

17 to 25088 Hz
Adjusts the other -6dB crossover points at which the input signal will be divided into the high, mid and low bands.

Enable

On or Off
Low, Mid, and High.
Turns processing for each band on or off. Turning each of the 3 bands off results in a dry output signal.

Drive

Off, -79.0 to 24.0 dB
Low. Mid, and High.
Adjusts the input into each transfer. Increasing the drive will increase the effects.

Xfer

-100 to 100 %
Low, Mid, and High; Xfer1 and Xfer2 for Mid and High.
Adjusts the intensity of the transfer curves.

Delay

Lo Delay 0 to 1000 samp
Mid Delay / Hi Delay 0 to 500 samp
Adjusts the number of samples the each signal is delayed.

Mix

Off, -79.0 to 24.0 dB
Low, Mid, and High.
Adjusts the output gain of each band.

HF Stimulate 1

Effects Size : 1

Boost and add high frequency partials

HF Stimulate 1 Presets
371 HF Stimulator

The HF Stimulate 1 algorithm is a close copy of the V.A.S.T. High Frequency Stimulator DSP function, giving control of the first highpass filter frequency, the distortion drive and the amplitude of the result (Stim Gain). As a bonus, the distortion curve can also be adjusted.

Figure 58. HF Stimulate 1 - Left Channel

The overall effect of a high-frequency stimulator is to boost the high frequency partials of the signal, and depending on the settings of the parameters, it can add high-frequency partials to the signal as well.

It’s useful for building sounds that cut through the mix, and have a bright crisp nature.

The high-frequency stimulator works like this:

The signal is run through a highpass filter, then through a distortion function, then through a second highpass filter.
It is then mixed with the original signal after passing through the final Stim Gain level control of the algorithm.

Stim Gain

Off, -79.0 to 24.0 dB
The gain of the high frequency stimulated signal applied prior to being added to the original input signal.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

Dist Drive

-79.0 to 48.0 dB
The amount to boost (or cut) the signal level to drive the distortion.

Higher values will increase the distortion of high frequency signal components.

Dist Curve

0 to 127%
The curvature of the distortion.

0% is no curvature (no distortion at all).
At 100%, the curve bends over smoothly and becomes perfectly flat right before it goes into digital clipping.

Highpass

8 to 25088 Hz
A first order highpass filter that removes low frequencies prior to being distorted.

Verb and Place Reverbs

The reverb algorithms can be divided into 2 groups: Verb and Place.

Verb reverbs cover medium to large spaces

Verb

Verb effects allow user-friendly control over medium to large spaces. Their decay times are controlled by Rvrb Time or LateRvbTim parameters, and Room Types range from rooms to large areas.

Place reverbs are optimized for small spaces

Place

Place algorithms on the other hand are optimized for small spaces. Decay time is controlled by the Absorption parameter, and Room Types offers several booths.

Reverb components

Each reverb algorithm consists of a several components.
These components provide sonic building blocks for both the body of the reverb and the early reflection portions.

a diffuser,
an injector,
predelay,
an ambience generator with feedback, and
various filters.

Ambience generator

The ambience generator is the heart of each reverb algorithm and creates most of the “late” reverb in algorithms with an Early Reflections circuit. It consists of a complex arrangement of delay lines to disperse the sound.

By using feedback in conjunction with the ambience generator, a reverb tail is produced. The length of this reverb tail is controlled by the Rvrb Time parameter in the Verb algorithms, or the Absorption parameter in Place algorithms.

In the feedback loop of the ambience generator are filters that further enhance the sonic properties of each reverb. A lowpass filter is controlled by HF Damping and mimics high frequency energy that is absorbed as the sound travels around a room. A low shelving filter is controlled by LF Split and LF Time, which are used to shorten or lengthen the decay time of low frequency energy.

Smoother reverbs

In order to create reverbs that are smoother and richer, some of the delays in the ambience generator are moved by LFOs.

The LFOs are adjusted by using the LFO Rate and LFO Depth controls. When used subtly, unwanted artifacts such as flutteriness and ringiness that are inherent in digital reverbs can be reduced.

Diffusers

At the beginning of each algorithm are diffusers. A diffuser creates an initial “smearing” quality on input
signals usually before the signal enters the ambience generating loop. The DiffAmtScl and DiffLenScl
parameters change the amount and the length of time that the sound is smeared. The Diffuse reverbs, however, implement diffusion a little differently.
(See the sections on Diffuse Verb and Diffuse Place for
more detailed information)

Injection

Some algorithms use injector mechanisms when feeding a signal into the ambience generator.

An injector creates copies of the input signal at different delay intervals and feeds each copy into the ambience
generator at different points. This results in finer control over the onset of the reverb. By tapering the amplitudes of early copies vs. late copies, the initial build of the reverb can be controlled. Inj Build controls this taper.

Inj Build: Negative values create a slower build, while positive values create a faster build.
Inj Spread: Inj Spread scales the time intervals that the copies are made.
Inj Skew: Inj Skew (Omni reverbs) delays one channel relative to the other before injecting into the ambience generator.

Negative values delay the left side while positive values delay the right side.
Inj LP: Inj LP controls the cutoff frequency of a 1-pole (6dB/oct) lowpass filter associated with the injector.
Predelay: Predelay can give the illusion that a space is more voluminous. Separate control over left and right predelay
is provided that can be used to de-correlate the center image, increasing reverb envelopment.

In addition to filters inside the ambience feedback loop, there also may be filters placed at the output of the reverb including a low shelf, high shelf, and/or lowpass.

Early Reflection: Algorithms that use Early Reflection circuits employ a combination of delays, diffusers, and filters to create ambience that is sparser than the late portion of the reverb. These early reflections model the initial near-discrete echoes rebounding directly off of near field surfaces before the reverb has a chance to become diffuse. They add realism when emulating real rooms and halls.
Room Type: Due to the inherent complexity of reverb algorithms and the sheer number of variables responsible for their character, the Room Type parameter provides condensed preset collections of these variables.

Each Room Type collection has been painstakingly selected by Kurzweil engineers to provide the best sounding combination of mutually complementary variables modeling an assortment of reverb families.

When you select a room type, an entire incorporated set of delay lengths and diffusion settings are established within the algorithm. By using the Size Scale, DiffAmtScl, DiffLenScl, and Inj Spread parameters, you may scale individual elements away from their pre-defined value. When set to 1.00x, each of these elements is equivalent to its preset value as determined by the current Room Type.

Room Types with similar names in different reverb algorithms do not sound the same. For example, Hall1 in Diffuse Verb does not sound the same as Hall1 in TQ Verb.

The Size Scale parameter scales the inherent size of the reverb chosen by Room Type. For a true representation of the selected Room Type size, set this to 1.00x. Scaling the size below this will create smaller spaces, while larger scale factors will create large spaces.

Your starting point when creating a new reverb preset should be the Room Type parameter.This parameter selects the basic type of reverb being used.

Room Types
	Classic Place	Classic Verb	TQ Place	TQ Verb	Diffuse Place	Diffuse Verb	Omni Place	Omni Verb
Booth1
Booth2
Booth3
Booth4
Booth5
Room1
Room2
Room3
Room4
Gate1
Gate2
Chamber
Chamber1
Chamber2
Plate
Plate1
Plate2
Plate3
Hall1
Hall2
Hall3
Hall4
Hall5
Large
Large1
Large2
Delay
NonLin

InfinDecay: The InfinDecay switch is designed to override the Rvrb Time parameter and create a reverb tail with an infinite decay time when On. However, certain HF Damping settings may reduce this effect, and cause the tail to taper away.

Classic Verb and Classic Place

Classic reverbs are 2 unit sized algorithms with early reflections.

Figure 59. Classic Verb / Classic Place - Signal flow

Late Reverb: The late portion consists of an input diffuser, ambience generator with low shelving filters, lopass filters, and LFO moving delays, and predelay.
Early Reflection: The early reflection portion consists of one delay per channel sent to its own output channel controlled by E Dly L and E Dly R, and one delay per channel sent to its opposite output channel controlled be E Dly LX and E Dly RX.

Each of these delays also use a Diffuser. Diffusion lengths are separately controlled by E DifDly L, E DifDly R, E DifDly LX, and E DifDly RX while diffusion amounts are all adjusted with E DiffAmt.

The late reverb and early reflection portions are independently mixed together with the Late Lvl and EarRef Lvl controls. The wet signal is passed through a final high shelving filter before being mixed with the dry signal.

Figure 60. Classic Verb / Classic Place - Early Reflection portion

Classic Place

Effects Size : 2

xxxx

Classic Place Presets
1 Small Wood Booth	18 SmallStudioRoom	89 School Stairwell
3 PrettySmallPlace	47 In The Studio	14 With A Mic
48 My Garage

Classic Verb

Effects Size : 2

Classic Verb Presets
2 Natural Room	35 Real Room	63 Reflective Hall
5 Sun Room	39 Sizzly Drum Room	64 Smooth Hall
19 ClassicRoom	50 Small Hall	85 Sweet Hall
20 Utility Room	52 Real Niceverb	97 Classic Plate
21 Thick Room	58 Opera House	98 Weighty Platey
22 The Real Room	59 Spacious Hall	102 Splendid Palace
24 Real Big Room	60 Classic Chapel	32 Bathroom
61 Semisweet Hall

TQ Verb and TQ Place

TQ reverbs are 3 unit sized algorithms with early reflections.

Figure 61. TQ Verb / TQ Place - Signal flow

Late Reverb

The late portion consists of an input diffuser, injector, ambience generator with a lopass filter, low shelving filter, and LFO moving delays, and predelay.

Early Reflection

The early reflection portion combines a combination of delays, diffusers, and feedback.

The relative delay lengths are all fixed but are scalable with the E Dly Scl parameter.
Relative diffusion lengths are also fixed, and are scalable with the E DfLenScl parameter.
Diffusion amount are adjusted with E DiffAmt.
The E Build parameter ramps the gains associated with each delay line in a way that changes the characteristic of the onset of the early reflections. Negative amounts create a slower onset while positive amount create a faster onset.

Figure 62. TQ Verb / TQ Place - Early Reflection portion

TQ Place

Effects Size : 3

TQ Place Presets
A Distance Away
Non-Linear 3

TQ Verb

Effects Size : 3

TQ Verb Presets
Soundboard	Bloom Chamber	Real Dense Hall
Spitty Drum room	ClassicalChamber	Flinty Hall
Stall One	Mosque Room	HighSchool Gym
Green Room	Empty Stage	Long & Narrow
Large Room	Abbey Piano Hall	Medm Warm Plate
Brt Empty Room	The Long Haul	Immense Mosque
Bigger Perc Room	Sweeter Hall	JudgeJudyChamber
The Piano Hall

Diffuse Verb and Diffuse Place

Diffuse reverbs are 3 effect sized algorithms.

They are characterized as such because of the initial burst of diffusion inherent in the onset of the reverb.

Figure 63. Diffuse Verb / Diffuse Place - Signal flow

Diffusion

The diffusion consists of an input diffuser, ambience generator with a lopass filter, low shelving filter, and LFO moving delays, and predelay.

Diffusor

In the diffuse reverbs, the diffuser is implemented a little differently. The diffuser is just inside the ambience generation loop, so changes in diffusion create changes the reverb decay.

DiffExtent and Diff Cross

The diffuse reverbs also offer DiffExtent and Diff Cross parameters.

DiffExtent selects one of seven arbitrary gate time lengths of the initial diffusion burst.
Diff Cross adjusts the combination of left and right channels that are diffused.

Diffuse Place

Effects Size : 3

Diffuse Place Presets
Standard Booth
Live Place

Diffuse Verb

Effects Size : 3

Diffuse Verb Presets
The Comfy Club	My Dreamy 481!!	Diffuse Gate
Bob’sDiffuseHall	Deep Hall	Far Bloom
Predelay Hall	Bloom Plate	Furbelows
Bloom Hall	Clean Plate	Festoons
Burst Space	RealSmoothPlate	Concert Hall

OmniVerb and OmniPlace

Omni reverbs are 3 effect sized algorithms.

They consist of an input diffuser, injector, ambience generator with a lopass filter, low shelving filter, and LFO moving delays, and predelay.

Expanse: The Expanse parameter adjusts the amount of reverb energy that is fed to the edges of the stereo image. A value of 0% concentrates energy in the center of the image, while non-zero values spread it out. Positive and negative values impose different characteristics on the reverb image.

At the output of the reverb are a pair each of low shelving and high shelving filters.

Figure 64. OmniVerb / OmniPlace - Signal flow

OmniPlace

Effects Size : 3

OmniPlace Presets
Add More Air	Non-Linear 1	Drum Latch 2
Small Closet	Exponent Booth	Acid Trip Room
Dreamverb	Drum Latch 1

OmniVerb

Effects Size : 3

OmniVerb Presets
Live Chamber	Standing Ovation	Plate Mail
Cool Dark Place	Soundbrd/rvb	Big Gym
Pad Space	Long PreDly Hall	Slapverb

Absorption

0 to 100 %
This controls the amount of reflective material that is in the space being emulated, much like an acoustical absorption coefficient. The lower the setting, the longer it will take for the sound to die away. A setting of 0% will cause an infinite decay time.

Rvrb Time

0.00 to 60.00 s
Adjusts the basic decay time of the late portion of the reverb.

LateRvbTim

0.00 to 60.00 s
Adjusts the basic decay time of the late portion of the reverb after diffusion.

HF Damping

0 to 25088 Hz
This controls the amount of high frequency energy that is absorbed as the reverb decays. The values set the cutoff frequency of the 1 pole (6dB/oct) lowpass filter within the reverb feedback loop.

L Pre Dly and R Pre Dly

0.0 to 230.0 ms
These control the amount that each channel of the reverb is delayed relative to the dry signal. Setting different lengths for both channels can de-correlate the center portion of the reverb image and make it seem wider. This only affects the late reverb in algorithms that have early reflections.

Lopass

8 to 25088 Hz
Controls the cutoff frequency of a 1 pole (6dB/oct) lowpass filter at the output of the reverb. This only affects the late reverb in algorithms that have early reflections.

EarRef Lvl

-100 to 100%
The mix level of the early reflection portion of algorithms offering early reflections.

Late Lvl

-100 to 100%
The mix level of the late reverb portion of algorithms offering early reflections.

Room Type

Booth1, Booth2, Booth3, Room1, Room2, Room3, Chamber1, Chamber2, Chamber 3, Plate1, Plate2, Hall1, Hall2, Hall3, Hall4, Large1, Large 2, Large3, Large4
This parameter selects the basic type of reverb being emulated, and should be your starting point when creating your own reverb presets. Due to the inherent complexity of reverb algorithms and the sheer number of variables responsible for their character, the Room Type parameter provides condensed preset collections of these variables. Each Room Type preset has been painstakingly selected by Kurzweil engineers to provide the best sounding collection of mutually complementary variables modeling an assortment of reverb families. When a room type is selected, an entire incorporated set of delay lengths and diffusion settings are established within the algorithm. By using the Size Scale, DiffAmtScl, DiffLenScl, and Inj Spread parameters, you may scale individual elements away from their preset value. When set to 1.00x, each of these elements are accurately representing their preset values determined by the current Room Type. Room Types with similar names in different reverb algorithms do not sound the same. For example, Hall1 in Diffuse Verb does not sound the same as Hall1 in TQ Verb.

Size Scale

0.01 to 2.50x
Scales the inherent size of the reverb chosen by Room Type. For a true representation of the selected Room Type size, set this to 1.00x. Scaling the size below this will create smaller spaces, while larger scale factors will create large spaces. See Room Type for more detailed information.

InfinDecay

On or Off
Found in “Verb” algorithms. When turned On, the reverb tail will decay indefinitely. When turned Off, the decay time is determined by the Rvrb Time or LateRvbTim parameters.

LF Split

8 to 25088 Hz
Used in conjunction with LF Time. This controls the upper frequency limit of the low frequency decay time multiplier. Energy below this frequency will decay faster or slower depending on the LF Time parameter.

LF Time

0.50 to 1.50 x
Used in conjunction with LF Split. This modifies the decay time of the energy below the LF Split frequency. A setting of 1.00x will make low frequency energy decay at the rate determined by the decay time. Higher values will cause low frequency energy to decay slower, and lower values will cause it to decay more quickly.

TrebShlf F

8 to 25088 Hz
The frequency of a high shelving filter at the output of the late reverb.

TrebShlf G

-79.0 to 24.0 dB
The gain of a high shelving filter at the output of the late reverb.

BassShlf F

8 to 25088 Hz
The frequency of a low shelving filter at the output of the late reverb.

BassShlf G

-79.0 to 24.0 dB
The gain of a low shelving filter at the output of the late reverb.

DiffAmtScl

0.00 to 2.00 x
The amount of diffusion at the onset of the reverb. For true representation of the selected Room Type diffusion amount, set to 1.00x.

DiffLenScl

0.00 to 4.50 x
The length of the diffusion at the onset of the reverb. For true representation of the selected Room Type diffusion length, set to 1.00x

DiffExtent

1 to 7 x
The onset diffusion duration. Higher values create longer diffuse bursts at the onset of the reverb.

Diff Cross

-100 to 100 %
The onset diffusion cross-coupling character. Although subtle, this parameter bleeds left and right channels into each other during onset diffusion, and also in the body of the reverb. 0% setting will disable this. Increasing this value in either the positive or negative direction will increase its affect.

Expanse

-100 to 100 %
Amount of late reverb energy biased toward the edges of the stereo image. A setting of 0% will bias energy towards the center. Moving away from 0% will bias energy towards the sides. Positive and negative values will have a different character.

LFO Rate

0.01 to 10.00 Hz
The rate at which certain reverb delay lines move. See LFO Depth for more information.

LFO Depth

0.0 to 100.0 ct
Adjusts the detuning depth in cents caused by a moving reverb delay line. Moving delay lines can imitate voluminous flowing air currents and reduce unwanted artifacts like ringing and flutter when used properly. Depth settings under 1.5ct with LFO Rate settings under 1.00Hz are recommended for modeling real spaces. High depth settings can create chorusing qualities, which won’t be unsuitable for real acoustic spaces, but can nonetheless create interesting effects. Instruments that have little if no inherent pitch fluctuation (like piano) are much more sensitive to this LFO than instruments that normally have a lot of vibrato (like voice) or non-pitched instruments (like snare drum).

Inj Build

-100 to 100 %
Used in conjunction with Inj Spread, this adjusts the envelope of the onset of the reverb. Specifically, it tapers the amplitudes of a series of delayed signals injected into the body of the reverb. Values above 0% will produce a faster build, while values below 0% will cause the build to be more gradual.

Inj Spread

0.00 to 4.50 x
Used in conjunction with Inj Build, this scales the length of the series of delays injected into the body of the reverb. For a true representation of the selected Room Type injector spread, set this to 1.00x.

Inj LP

8 to 25088 Hz
The cutoff frequency of a 1 pole (6dB/oct) lowpass filter applied to the signal being injected into the body of the reverb.

Inj Skew

-200 to 200 ms
The amount of delay applied to either the left or right channel of the reverb injector. Positive values delay the right channel while negative values delay the left channel.

E DiffAmt

-100 to 100 %
The amount of diffusion applied to the early reflection network.

E DfLenScl

0.00 to 2.50 x
The length of diffusion applied to the early reflection network. This is influenced by E PreDlyL and E PreDlyR.

E Dly Scl

0.00 to 2.50 x
Scales the delay lengths inherent in the early reflection network.

E Build

-100 to 100 %
The envelope of the onset of the early reflections. Values above 0% will create a faster attack while values below 0% will create a slower attack.

E Fdbk Amt

-100 to 100 %
The amount of the output of an early reflection portion that is fed back into the input of the opposite channel in front of the early predelays. Overall, it lengthens the decay rate of the early reflection network. Negative values polarity invert the feedback signal.

E HF Damp

8 to 25088 Hz
The cutoff frequency of a 1 pole (6dB/oct) lowpass filter applied to the early reflection feedback signal.

E PreDlyL and E PreDlyR

The amount of delay in early reflections relative to the dry signal. These are independent of the late reverb predelay times, but will influence E Dly Scl.

E Dly L and E Dly R

0.0 to 150.0 ms
The left and right early reflection delays fed to the same output channels. E Dly LX E Dly RX The left and right early reflection delays fed to the opposite output channels.

E DifDlyL and E DifDlyR

0.0 to 160.0 ms
The diffusion delays of the diffusers on delay taps fed to the same output channels

E DifDlyLX and DifDlyRX

0.0 to 230.0 ms
The diffusion delays of the diffusers on delay taps fed to the opposite output channels.

E X Blend

0 to 100 %
The balance between early reflection delay tap signals with diffusers fed to their same output channel, and those fed to opposite channels. 0% will only allow delay taps being fed to opposite output channels to be heard, while 100% allows only delay taps going to the same channels to be heard.

Panaural Reverb

Panaural Room

Effects Size : 3

xxxx

Panaural Room Presets
29 Tabla Room	37 Med Large Room	Huge Batcave
33 Drum Room	75 Recital Hall	Big Gym
34 Small Dark Room	76 Generic Hall	145 Slapverb

The Panaural Room reverberation is implemented using a special network arrangement of many delay lines that guarantees colorless sound.

Figure 65. Panaural Room - Simplified block diagram

Reverberator: The reverberator is inherently stereo with each input injected into the “room” at multiple locations. The signals entering the reverberator first pass through a shelving bass equalizer with a range of +/-15dB.

To shorten the decay time of high frequencies relative to mid frequencies, lowpass filters controlled by HF Damping are distributed throughout the network.
Room Size: Room Size scales all the delay times of the network (but not the Pre Dly or Build Time), to change the simulated room dimension over a range of 1 to 16m.
Decay Time: Decay Time varies the feedback gains to achieve decay times from 0.5 to 100 seconds.

The Room Size and Decay Time controls are interlocked so that a chosen Decay Time will be maintained while Room Size is varied.

Output: A two input stereo mixer, controlled by Wet/Dry and Out Gain, feeds the output.
Early Reflections: The duration and spacing of the early reflections are influenced by Room Size and Build Time, while the number and relative loudness of the individual reflections are influenced by Build Env.

When Build Env is near 0% or 100%, fewer reflections are created.

The maximum number of important early reflections, 13, is achieved at a setting of 50%.

Controll Over Reverb Growth

To get control over the growth of reverberation, the left and right inputs each are passed through an “injector” that can extend the source before it drives the reverberator.

Build Env: Only when Build Env is set to 0% is the reverberator driven in pure stereo by the pure dry signal.

For settings of Build Env greater than 0%, the reverberator is fed multiple times.

Build Env controls the injector so that the reverberation begins abruptly (0%), builds immediately to a sustained level (50%), or builds gradually to a maximum (100%).
Build Time: Build Time varies the injection length over a range of 0 to 500ms.

At a Build Time of 0ms, there is no extension of the build time. In this case, the Build Env control adjusts the density of the reverberation, with maximum density at a setting of 50%.

In addition to the two build controls, there is an overall Pre Dly control that can delay the entire reverberation process by up to 500ms.

Wet/Dry

0 to 100%wet
The amount of the stereo reverberator (wet) signal relative to the original input (dry) signal to be output.

The dry signal is not affected by the HF Roll control.

The wet signal is affected by the HF Roll control and by all the other reverberator controls.

The balance between wet and dry signals is an extremely important factor in achieving a good mix.

Emphasizing the wet signal gives the effect of more reverberation and of greater distance from the source.

Out Gain

Off, -79.0 to 24.0
The overall output level for the reverberation effect, and controls the level for both the wet and dry signal paths.

Decay Time

0.5 to 100.0 s
The reverberation decay time (mid-band “RT60”), the time required before the reverberation has died away to 60dB below its “running” level.

Adjust decay time according to the tempo and articulation of the music and to taste.

HF Damping

8 to 25088 Hz
Adjusts lowpass filters in the reverberator so that high frequencies die away more quickly than mid and low frequencies.

This shapes the reverberation for a more natural, more acoustically accurate sound.

Bass Gain

-15 to 15 dB
Adjusts bass equalizers in the reverberator so that low frequencies die away more quickly than mid and high frequencies.

This can be used to make the reverberation less muddy.

Room Size

1.0 to 16.0 m
Choosing an appropriate room size is very important in getting a good reverberation effect.

For impulsive sources, such as percussion instruments or plucked strings, increase the size setting until discrete early reflections become audible, and then back it off slightly.

For slower, softer music, use the largest size possible.

At lower settings, RoomSize leads to coloration, especially if the DecayTime is set too high.

Pre Dly

0 to 500 ms
Introducing predelay creates a gap of silence between that allows the dry signal to stand out with greater clarity and intelligibility against the reverberant background.

This is especially helpful with vocal or classical music.

Build Time

0 to 500 ms
Similar to predelay, but more complex, larger values of BuildTime slow down the building up of reverberation and can extend the build up process.

Experiment with BuildTime and BuildEnv and use them to optimize the early details of reverberation.
====
A BuildTime of 0ms and a BuildEnv of 0% is a good default setting that yields fast arriving, natural reverberation.
====

Build Env

0 to 100%
When BuildTime has been set to greater than about 80ms, BuildEnv begins to have an audible influence on the early unfolding of the reverberation process.

For lower density reverberation that starts cleanly and impulsively, use a setting of 0%.
For the highest density reverberation, and for extension of the build up period, use a setting of 50%.

For an almost reverse reverberation, set BuildEnv to 100%.

You can think of BuildEnv as setting the position of a see-saw.

The left end of the see-saw represents the driving of the reverberation at the earliest time, the pivot point as driving the reverberation at mid-point in the time sequence, and the right end as the last signal to drive the reverberator.

At settings near 0%, the see-saw is tilted down on the right: the reverberation starts abruptly and the drive drops with time.
Near 50%, the see- saw is level and the reverberation is repetitively fed during the entire build time.
At settings near 100%, the see-saw is tilted down on the left, so that the reverberation is hit softly at first, and then at increasing level until the end of the build time.

Hall Reverb

Stereo Hall

Effects Size : 3

xxxx

Stereo Hall Presets
69 Short Hall

The Stereo Hall reverberation is implemented using a special arrangement of allpass networks and delay lines which reduces coloration and increases density.

Figure 66. Stereo Hall - Simplified block diagram

Reverberator

The reverberator is inherently stereo with each input injected into the “room” at multiple locations.

To shorten the decay time of low and high frequencies relative to mid frequencies, bass equalizers and lowpass filters, controlled by Bass Gain and by HF Damping, are placed within the network.

Room Size: Room Size scales all the delay times of the network (but not the Pre Dly or Build Time), to change the simulated room dimension over a range of 10 to 75m.
Decay Time: Decay Time varies the feedback gains to achieve decay times from 0.5 to 100 seconds.

The Room Size and Decay Time controls are interlocked so that a chosen Decay Time will be maintained while Room Size is varied. At smaller sizes, the reverb becomes quite colored and is useful only for special effects.
Stereo Mixer: A two input stereo mixer, controlled by Wet/Dry and Out Gain, feeds the output. The Lowpass control acts only on the wet signal and can be used to smooth out the reverb high end without modifying the reverb decay time at high frequencies.
Reverb Tail: Within the reverberator, certain delays can be put into a time varying motion to break up patterns and to increase density in the reverb tail.

Using the LFO Rate and Depth controls carefully with longer decay times can be beneficial. But beware of the pitch shifting artifacts which can accompany randomization when it is used in greater amounts.
Diffusion: Also within the reverberator, the Diffusion control can reduce the diffusion provided by some allpass networks. While the reverb will eventually reach full diffusion regardless of the Diffusion setting, the early reverb diffusion can be reduced, which sometimes is useful to help keep the dry signal “in the clear.”
Mono Source Signals: The reverberator structure is stereo and requires that the dry source be applied to both left and right inputs. If the source is mono, it should still be applied (pan centered) to both left and right inputs.

Failure to drive both inputs will result in offset initial reverb images and later ping-ponging of the reverberation. Driving only one input will also increase the time required to build up reverb density.
Controlling Growth of Reverberation: To gain control over the growth of reverberation, the left and right inputs each are passed through an “injector” that can extend the source before it drives the reverberator. Only when Build Env is set to 0% is the reverberator driven in pure stereo by the pure dry signal.

For settings of Build Env greater than 0%, the reverberator is fed multiple times. Build Env controls the injector so that the reverberation begins abruptly (0%), builds immediately to a sustained level (50%), or builds gradually to a maximum (100%).

Build Time varies the injection length over a range of 0 to 500ms. At a Build Time of 0ms, there is no extension of the build time. In this case, the Build Env control adjusts the density of the reverberation, with maximum density at a setting of 50%.

In addition to the two build controls, there is an overall Pre Dly control that can delay the entire reverberation process by up to 500ms.

Wet/Dry

0 to 100%wet
The amount of the stereo reverberator (wet) signal relative to the original input (dry) signal to be output.

The dry signal is not affected by the HF Roll control.
The wet signal is affected by the HF Roll control and by all the other reverberator controls.

The balance between wet and dry signals is an extremely important factor in achieving a good mix. Emphasizing the wet signal gives the effect of more reverberation and of greater distance from the source.

Out Gain

Off, -79.0 to 24.0 dB
The overall output level for the reverberation effect, and controls the level for both the wet and dry signal paths.

Decay Time

0.5 to 100.0 ms
The reverberation decay time (mid-band “RT60”), the time required before the reverberation has died away to 60dB below its “running” level.

Adjust decay time according to the tempo and articulation of the music and to taste.

HF Damping

Bass Gain

-15 to 0 dB
Adjusts bass equalizers in the reverberator so that low frequencies die away more quickly than mid and high frequencies.

This can be used to make the reverberation less muddy.

Lowpass

8 to 25088 Hz
Used to shape the overall reverberation signal’s treble content, but does not modify the decay time.

Reduce the treble for a softer, more acoustic sound.

Room Size

2.0 to 15.0 m
Choosing an appropriate room size is very important in getting a good reverberation effect.

For impulsive sources, such as percussion instruments or plucked strings, increase the size setting until discrete early reflections become audible, and then back it off slightly.

For slower, softer music, use the largest size possible.

At lower settings, RoomSize leads to coloration, especially if the DecayTime is set too high.

Pre Dly

0 to 500 ms
Introducing predelay creates a gap of silence between that allows the dry signal to stand out with greater clarity and intelligibility against the reverberant background.

This is especially helpful with vocal or classical music.

Build Time

0 to 500 ms
Similar to predelay, but more complex, larger values of BuildTime slow down the building up of reverberation and can extend the build up process. Experiment with BuildTime and BuildEnv and use them to optimize the early details of reverberation.

A BuildTime of 0ms and a BuildEnv of 0% is a good default setting that yields fast arriving, natural reverberation.

Build Env

0 to 100%
When BuildTime has been set to greater than about 80ms, BuildEnv begins to have an audible influence on the early unfolding of the reverberation process.

For lower density reverberation that starts cleanly and impulsively, use a setting of 0%.
For the highest density reverberation, and for extension of the build up period, use a setting of 50%.
For an almost reverse reverberation, set BuildEnv to 100%.

You can think of BuildEnv as setting the position of a see-saw. The left end of the see-saw represents the driving of the reverberation at the earliest time, the pivot point as driving the reverberation at mid-point in the time sequence, and the right end as the last signal to drive the reverberator.

At settings near 0%, the see-saw is tilted down on the right: the reverberation starts abruptly and the drive drops with time.
Near 50%, the see- saw is level and the reverberation is repetitively fed during the entire build time.
At settings near 100%, the see-saw is tilted down on the left, so that the reverberation is hit softly at first, and then at increasing level until the end of the build time.

LFO Rate and LFO Depth

0.00 to 5.10 Hz (LFO Rate) and 0.00 to 10.20 ct (LFO Depth)
Within the reverberator, the certain delay values can be put into a time varying motion to break up patterns and to increase density in the reverb tail. Using the LFO Rate and Depth controls carefully with longer decay times can be beneficial. But beware of the pitch shifting artifacts which can accompany randomization when it is used in greater amounts.

Diffusion

0 to 100%
Within the reverberator, the Diffusion control can reduce the diffusion provided some of the allpass networks. While the reverb will eventually reach full diffusion regardless of the Diffusion setting, the early reverb diffusion can be reduced, which sometimes is useful to help keep the dry signal “in the clear.”

Plate Reverb

Plate History

Plate reverberators were manufactured during the 1950s, ‘60s, ‘70s, and perhaps into the ‘80s. By the end of the 1980s, they had been supplanted in the marketplace by digital reverberators, which first appeared in 1976. While a handful of companies made plate reverberators, EMT (Germany) was the best known and most popular.

How a Plate Works

A plate reverberator is generally quite heavy and large, perhaps 4 feet high by 7 feet long and a foot thick. They were only slightly adjustable, with controls for high frequency damping and decay time. Some were stereo in, stereo out, others mono in, mono out.

A plate reverb begins with a sheet of plate steel suspended by its edges, leaving the plate free to vibrate.

At one (or two) points on the plate, an electromagnetic driver (sort of a small loudspeaker without a cone) is arranged to couple the dry signal into the plate, sending out sound vibrations into the plate in all directions.
At one or two other locations, a pickup is placed, sort of like a dynamic microphone whose diaphragm is the plate itself, to pick up the reverberation. Since the sound waves travel very rapidly in steel (faster than they do in air), and since the dimensions of the plate are not large, the sound quickly reaches the plate edges and reflects from them. This results in a very rapid build up of the reverberation, essentially free of early reflections and with no distinguishable gap before the onset of reverb.

Plates offered a wonderful sound of their own, easily distinguished from other reverberators in the pre-digital reverb era, such as springs or actual “echo” chambers.

Plates were bright and diffused (built up echo density) rapidly. Curiously, when we listen to a vintage plate today, we find that the much vaunted brightness is nothing like what we can accomplish digitally; we actually have to deliberately reduce the brightness of a plate emulation to match the sound of a real plate. Similarly, we find that we must throttle back on the low frequency content as well.

Grand Plate

Effects Size : 3

Emulates an EMT 140 steel plate reverberator

Grand Plate Presets
90 Real Plate
91 Bright Plate

This algorithm emulates an EMT 140 steel plate reverberator.

The algorithm developed for Grand Plate was carefully crafted for rapid diffusion, low coloration, freedom from discrete early reflections, and “brightness.” We also added some controls that were never present in real plates: size, pre delay of up to 500ms, LF damping, lowpass roll off, and bass roll off. Furthermore, we allow a wider range of decay time adjustment than a conventional plate.

Once the algorithm was complete, we tuned it by presenting the original EMT reverb on one channel and the Grand Plate emulation on the other. A lengthy and careful tuning of Grand Plate (tuning at the micro detail level of each delay and gain in the algorithm) was carried out until the stereo spread of this reverb was matched in all the time periods: early, middle, and late.

Reverberator: The heart of this reverb is the plate simulation network, with its two inputs and two outputs. It is a full stereo reverberation network, which means that the left and right inputs get slightly different treatment in the reverberator. This yields a richer, more natural stereo image from stereo sources.

The incoming left source is passed through predelay, lowpass (Lowpass), and bass shelf (Bass Gain) blocks. The right source is treated similarly. There are lowpass filters (HF Damping) and highpass filters (LF Damping) embedded in the plate simulation network to modify the decay times. The reverb network also accommodates the Room Size and Decay Time controls. An output mixer assembles dry and wet signals.

If you have a mono source, assign it to both inputs for best results.

Wet/Dry

0 to 100%wet
The amount of the stereo reverberator (wet) signal relative to the original input (dry) signal sent to the output.

The dry signal is not affected by the Lowpass or Bass Gain controls.
The wet signal is affected by the Lowpass and Bass Gain controls and by all the other reverberator controls.

Out Gain

Off, -79.0 to 24.0 dB
The overall output level for the reverberation effect and controls the level for both the wet and dry signal paths.

Room Size

1.00 to 4.00 m
Choosing an appropriate room size is very important in getting a good reverberation effect.

For impulsive sources, such as percussion instruments or plucked strings, increase the size setting until discrete reflections become audible, and then back it off slightly.

For slower, softer music, use the largest size possible.

At lower settings, Room Size leads to coloration, especially if the Decay Time is set too high.

To emulate a plate reverb, this control is typically set to 1.9m.

Pre Dly

0 to 500 ms
Introducing predelay creates a gap of silence between the dry sound and the reverberation, allowing the dry signal to stand out with greater clarity and intelligibility against the reverberant background.

Especially helpful with vocals or classical music.

Decay Time

0.2 to 5.0 s
The reverberation decay time (mid-band “RT60”), the time required before the reverberation has died away to 60dB below its “running” level.

Adjust decay time according to the tempo and articulation of the music.

To emulate a plate reverb, this control is typically set in the range of 1 to 5 seconds.

HF Damping

LF Damping

1 to 294 Hz
Adjusts highpass filters in the reverberator so that low frequencies die away more quickly than mid and high frequencies.

This shapes the reverberation for a more natural, more acoustically accurate sound.

To emulate a plate reverb, this control is typically set to 52 Hz.

Lowpass

8 to 25088 Hz
Shapes the overall reverberation signal’s treble content, but does not modify the decay time.

Reduce the treble for a duller, more natural acoustic effect.

To emulate a plate reverb, this control is typically set to 3951 Hz.

Bass Gain

-15 to 0 dB
Shapes the overall reverberation signal’s bass content, but does not modify the decay time.

Reduce the bass for a less muddy sound.

To emulate a plate reverb, this control is typically set to -12 dB.

Reverse Reverb

Finite Verb

Effects Size : 3

Reverse reverb

Finite Verb Presets
105 Reverse Reverb 1
106 Reverse Reverb 2
107 Reverse Reverb 3

The left and right sources are summed before being fed into a tapped delay line which directly simulates the impulse response of a reverberator. The taps are placed in sequence from zero delay to a maximum delay value, at quasi-regular spacings. By varying the coefficients with which these taps are summed, one can create the effect of a normal rapidly building/slowly decaying reverb or a reverse reverb which builds slowly then stops abruptly.

A special tap is picked off the tapped delay line and its length is controlled by Dly Length. It can be summed into the output wet mix (Dly Lvl) to serve as the simulated dry source that occurs after the reverse reverb sequence has built up and ended. It can also be fed back for special effects. Fdbk Lvl and HF Damping tailor the gain and spectrum of the feedback signal. Despite the complex reverb-like sound of the tapped delay line, the Feedback tap is a pure delay. Feeding it back is like reapplying the source, as in a simple tape echo.

Dly Length and Rvb Length range from 300 to 3000 milliseconds. With the R1 Rvb Env variants, Rvb Length corresponds to a decay time (RT60).

To make things a little more interesting, the tapped delay line mixer is actually broken into three mixers, an early, middle, and late mixer. Each mixes its share of taps and then applies the submix to a lowpass filter (cut only) and a simple bass control (boost and cut). Finally, the three equalized sub mixes are mixed into one signal. The Bass and Damp controls allow special effects such as a reverb that begins dull and increases in two steps to a brighter sound.

Rvb Env

The Rvb Env control selects 27 cases of envelope gains for the taps. Nine cases emulate a normal forward evolving reverb, but with some special twists.

R1 build to a single peak.
R2 build to two peaks.
R3 build to three peaks.
S1/S2/S3 is dullest (S1) to sharpest (S3).

FWD R1xx

Cases FWD R1xx have a single reverb peak, with a fast attack and slower decay.

FWD R1Sx

The sub cases FWD R1Sx vary the sharpness of the envelope, from dullest (S1) to sharpest (S3).

FWD R2xx

The sub cases FWD R2xx have two peaks; that is, the reverb builds, decays, builds again, and decays again.

FWD R3xx

The sub cases FWD R3xx have three peaks.

SYM R1xx

The sub cases SYM have a symmetrical build and decay time.

SYM R1Sx / SYM R2Sx / SYM R3Sx

The cases R1 build to a single peak, while R2 and R3 have two and three peaks, respectively.

REV R1xx

The sub cases REV simulate a reverse reverb effect.

REV R1xx / REV R2Sx / REV R3Sx

Imitates a backward running reverb, with a long rising “tail” ending abruptly (followed, optionally, by the “dry” source mixed by Dly Lvl). Once again, the number of peaks and the sharpness are variable.

Rvb Env parameter settings
FWD R1S1	SYM R1S1	REV R1S1
FWD R1S2	SYM R1S2	REV R1S2
FWD R1S3	SYM R1S3	REV R1S3
FWD R2S1	SYM R2S1	REV R2S1
FWD R2S2	SYM R2S2	REV R2S2
FWD R2S3	SYM R2S3	REV R2S3
FWD R3S1	SYM R3S1	REV R3S1
FWD R3S2	SYM R3S2	REV R3S2
FWD R3S3	SYM R3S3	REV R3S3

The usual Wet/Dry and Output Gain controls are provided.

Wet/Dry

0 to 100% wet
Wet/Dry sets the relative amount of wet signal and dry signal.

The wet signal consists of the reverb itself (stereo) and the delayed mono signal arriving after the reverb has ended (simulating the dry source in the reverse reverb sequence). The amount of the delayed signal mixed to the Wet signal is separately adjustable with the Dly Lvl control.

The Dry signal is the stereo input signal.

Out Gain

Off, -79.0 to 24.0 dB
This controls the level of the output mix, wet and dry, sent back into the sound source.

Fdbk Lvl

0 to 100%
This controls the feedback gain of the separate, (mono) delay tap.

INFO: A high value contributes a long repeating echo character to the reverb sound.

HF Damping

8 to 25088 Hz
HF Damping adjusts a lowpass filter in the late delay tap feedback path so that high frequencies die away more quickly than mid and low frequencies.

Dly Lvl

0 to 100%
This adjusts the level of the separate, (mono) delay tap used to simulate the dry source of a reverse reverb effect.

This same tap is used for feedback.

Dly Length

300 to 3000 ms
Sets the length (in milliseconds), of the separate, (mono) delay tap used to simulate the dry source of a reverse reverb effect.

This same tap is used for feedback.

Rvb Env

refer to Rvb Env parameter settings table
The Rvb Env control selects 27 cases of envelope gains for the taps.

Nine cases emulate a normal forward evolving reverb, another nine emulate a reverb building symmetrically to a peak at the mid point, while the last nine cases emulate a reverse building reverb.

For each major shape, there are three variants of one, two, and three repetitions and three variants of envelope sharpness.

Rvb Length

300 to 3000 ms
Sets the length (in milliseconds), from start to finish, of the reverberation process.

This parameter is essentially the decay time or RT60 for the Rvb Env cases ..R1.. where there is only one repetition.

Early Bass / Mid Bass / Late Bass

-15 to 15 dB

These bass controls shape the frequency response (boost or cut) of the three periods of the finite reverb sequence.

Use them to tailor the way the reverb bass content changes with time.

Early Damp / Mid Damp / Late Damp

8 to 25088 Hz
These treble controls shape the frequency response (cut only) of the three periods of the finite reverb sequence.

Use them to tailor the way the reverb treble content changes with time.

Parametric EQ

These algorithms are multi-band equalizers with 1–4 bands of parametric EQ and with bass and treble tone controls.

You can control the gain, frequency and bandwidth of each band of parametric EQ and control of the gain and frequencies of the bass and treble tone controls.

The small 3 Band EQ does not provide control of the bandwidth for the parametric Mid filter.

The algorithms 5 Band EQ and 3 Band EQ are stereo, meaning the parameters for the left and right channels are ganged—the parameters have the same effect on both channels.

Dual 5 Band EQ provides separate control for left and right channels.

3 Band EQ

Effects Size : 1

Bass and treble shelving filters and parametric EQs

3 Band EQ Presets
350 AM Radio
351 U-Shaped EQ

5 Band EQ

Effects Size : 3

Bass and treble shelving filters and parametric EQs

5 Band EQ Presets
352 5 Band EQ Flat

Dual 5 Band EQ

Effects Size : 3

Bass and treble shelving filters and parametric EQs

Dual 5 Band EQ Presets
355 Dual 5 Band EQ

In/Out

In or Out
When set to In the tone controls are active.
When set to Out the tone controls are bypassed.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

Bass Gain

-79.0 to 24.0 dB
The amount of boost or cut that the filter should apply to the low frequency signals in dB.

Every increase of 6 dB approximately doubles the amplitude of the signal.
Positive values boost the bass signal below the specified frequency.
Negative values cut the bass signal below the specified frequency.

Bass Freq

8 to 25088 Hz
The center frequency of the bass shelving filter in intervals of one semitone.

Treb Gain

-79.0 to 24.0 dB
The amount of boost or cut that the filter should apply to the high frequency signals in dB.

Every increase of 6 dB approximately doubles the amplitude of the signal.
Positive values boost the treble signal above the specified frequency.
Negative values cut the treble signal above the specified frequency.

Treb Freq

8 to 25088 Hz
The center frequency of the treble shelving filter in intervals of one semitone.

Mid1/Mid2 Gain

-79.0 to 24.0 dB
The amount of boost or cut that the filter should apply in dB.

Every increase of 6 dB approximately doubles the amplitude of the signal.
Positive values boost the signal at the specified frequency.
Negative values cut the signal at the specified frequency.

Mid1/Mid2 Freq

8 to 25088 Hz
The center frequency of the EQ in intervals of one semitone.

The boost or cut will be at a maximum at this frequency.

Mid1/Mid2 Width

0.010 to 5.000 oct
The bandwidth of the EQ may be adjusted.

You specify the bandwidth in octaves.
Small values result in a very narrow filter response.
Large values result in a very broad response.

Graphic EQ

The graphic equalizer is available as stereo (linked parameters for left and right) or dual mono (independent controls for left and right).

The graphic equalizer has ten bandpass filters per channel.

For each band the gain may be adjusted from -12 dB to +24 dB.

The frequency response of all the bands is shown in Figure 102 below.

The dual graphic equalizer has a separate set of controls for the two mono channels.

Figure 67. Filter response of each Bandpass filter

Like all graphic equalizers, the filter response is not perfectly flat when all gains are set to the same level (except at 0 dB), but rather has ripple from band to band. To minimize the EQ ripple, you should attempt to center the overall settings around 0 dB.

Figure 68. Overall response with all gains set to +12 dB, 0 dB and -6 dB

Graphic EQ

Effects Size : 3

xxx

Graphic EQ Presets
353 Graphic EQ Flat

Dual Graphic EQ

Effects Size : 3

xxx

Dual Graphic EQ Presets
354 Dual Graphic EQ

In/Out

In or Out
When set to In the left channel equalizer is active.
When set to Out the left channel equalizer is bypassed.

31Hz G

-12.0 to 24.0 dB
Gain of the left 31 Hz band in dB.

62Hz G

-12.0 to 24.0 dB
Gain of the left 62 Hz band in dB.

125Hz G

-12.0 to 24.0 dB
Gain of the left 125 Hz band in dB.

250Hz G

-12.0 to 24.0 dB
Gain of the left 250 Hz band in dB.

500Hz G

-12.0 to 24.0 dB
Gain of the left 500 Hz band in dB.

1000Hz G

-12.0 to 24.0 dB
Gain of the left 1000 Hz band in dB.

2000Hz G

-12.0 to 24.0 dB
Gain of the left 2000 Hz band in dB.

4000Hz G

-12.0 to 24.0 dB
Gain of the left 4000 Hz band in dB.

8000Hz G

-12.0 to 24.0 dB
Gain of the left 8000 Hz band in dB.

16000Hz G

-12.0 to 24.0 dB
Gain of the left 16000 Hz band in dB.

LaserVerb

LaserVerb has to be heard to be believed!

Feed it an impulsive sound such as a snare drum, and LaserVerb plays the impulse back as a delayed train of closely spaced impulses, and as time passes, the spacing between the impulses gets wider.

The close spacing of the impulses produces a discernible buzzy pitch which gets lower as the impulse spacing increases.

The following figure is a simplified representation of the LaserVerb impulse response.

Figure 69. LaserVerb - Simplified Impulse Response

An impulse response of a system is what you would see if you had an oscilloscope on the system output and you gave the system an impulse or a spike for an input.

With appropriate parameter settings this effect produces a descending buzz or whine somewhat like a diving airplane or a siren being turned off.

The descending buzz is most prominent when given an impulsive input such as a drum hit.

When used as a reverb, it tends to be highly metallic and has high pitched tones at certain parameter settings.

To get the descending buzz, start with about half a second of delay, set the Contour parameter to a high value (near 1), and set the HF Damping to a low value (at or near 0).

Contour

The Contour parameter controls the overall shape of the LaserVerb impulse response.

At high values the response builds up very quickly decays slowly.

As the Contour value is reduced, the decay becomes shorter and the sound takes longer to build up.

At a setting of 0, the response degenerates to a simple delay.

Spacing

The Spacing parameter controls the initial separation of impulses in the impulse response and the rate of their subsequent separation.

Low values result in a high initial pitch (impulses are more closely spaced) and takes longer for the pitch to lower.

Feedback

The output from LaserVerb can be fed back to the input.

By turning up the feedback, the duration of the LaserVerb sound can be greatly extended.

XCouple

Cross-coupling may also be used to move the signal between left and right channels, producing a left/right ping-pong effect at the most extreme settings.

Effects Unit Size Differences

The 2 unit version is a sparser version than the 3 unit version. Its buzzing is somewhat coarser.
The 1 unit version is like the 2 unit version except the two input channels are summed and run through a single mono LaserVerb.
The 1 unit version does not have the cross-coupling control but does have output panning.

Figure 70. LaserVerb - Simplified Block Diagram of the left channel

LaserVerb

Effects Size : 3

A bizarre reverb with a falling buzz

LaserVerb Presets
135 LaserVerb
136 LaserWaves

LaserVerb Lite

Effects Size : 2

xxxxxxxx

LaserVerb Presets
137 Cheap LaserVerb
143 Spry Young Boy

Mono LaserVerb

Effects Size : 1

LaserVerb Presets
133 Drum Neurezonate
140 Lazerfazer Echoes
141 Simple LaserVerb

Wet/Dry

0 to 100% wet
The amount of reverbed (wet) signal relative to unaffected (dry) signal.

Out Gain

Off, -79.0 to 24.0dB
The overall gain or amplitude at the output of the effect.

Fdbk Lvl

0 to 100%
The percentage of the reverb output to feed back or return to the reverb input.

Turning up the feedback is a way to stretch out the duration of the reverb, or, if the reverb is set to behave as a delay, to repeat the delay.

The higher feedback is set, the longer the decay or echo will last.

Xcouple

0 to 100%

LaserVerb and LaserVerb Lite are stereo effects.

The cross-coupling control lets you send the sum of the input and feedback from one channel to its own LaserVerb effect (0% cross coupling) or to the other channel’s effect (100% cross coupling) or somewhere in between.

This control is not available in Mono LaserVerb.

HF Damping

8 to 25088Hz
The damping of high frequencies relative to low frequencies.

When set to the highest frequency (25088 Hz), there is no damping and all frequencies decay at the samerate.

At lower frequency settings, high frequency signal components will decay faster than low frequency components.

If set too low, everything will decay almost immediately.

Pan

-100 to 100%

The Pan control is available in the Mono LaserVerb.

The left and right inputs get summed to mono, the mono signal passes through the LaserVerb, and the final mono output is panned to the left and right outputs.

Panning ranges from:

Fully Left -100%
Centered 0%
Fully Right 100%

Dly Coarse

0 to 5000ms
You can set the overall delay length from 0 to 2 seconds (3 unit effects) or 0 to 1.3 seconds (2 unit effects).

Lengthening the delay will increase the duration or decay time of the reverb.

To reduce LaserVerb to a simple delay, set the Contour and Feedback controls to 0.

TIP: Use a delay of about half a second as a starting point.

Dly Fine

-20.0 to 20.0ms
The delay fine adjust is added to the delay coarse adjust to provide a delay resolution down to 0.1 ms.

Spacing

0.0 to 40.0samp
Determines the starting pitch of the descending buzz and how fast it descends.

The Spacing parameter sets the initial separation of impulses in the impulse response and subsequent rate of increasing impulse separation.

The spacing between impulses is given in samples and may be a fraction of a sample.
(A sample is the time between successive digital words which is 20.8 µs or 1/48000 seconds.)

For low values, the buzz starts at high Frequencies and drops slowly.
At high values the buzz starts at a lower pitch and drops rapidly.

Contour

0.0 to 100.0%
Controls the overall envelope shape of the reverb.

When set to a high value, sounds passed through the reverb start at a high level and slowly decay.

As the control value is reduced, it takes some time for the effect to build up before decaying.

TIP: At a value of around 34, the reverb is behaving like a reverse reverb, building up to a hit.

When the Contour is set to 0, LaserVerb is reduced to a simple delay.

Revrse LaserVerb

Effects Size : 4

A bizarre reverb which runs backwards in time (uh, yeah).

Revrse LaserVerb Presets
Rvrs LaserVerb
Waterford

Revrse LaserVerb is a mono effect that simulates the effect of running the LaserVerb in reverse.

When you play a sound through the algorithm, it starts out relatively diffuse then builds to the final “hit.”

Since Kurzweil effects cannot break the universal rules of causality (sorry, your Kurzweil instrument doesn’t know what you are about to play!), there can be a significant delay between what you play and when you hear it.

In addition to the normal Wet/Dry control, with the Rvrs W/D, the dry signal is considered to be the delayed “hit” signal.

Revrse LaserVerb is LaserVerb in reverse, so when it is fed an impulsive sound such as a snare drum, it plays the impulse back as a delayed train of closely spaced impulses, and as time passes, the spacing between the impulses gets closer until they coalesce at the “hit.”

The close spacing of the impulses produces a discernible buzzy pitch which gets higher as the impulse spacing decreases.

The following figure is a simplified representation of the Revrse LaserVerb impulse response.

Figure 71. Revrse LaserVerb - Simplified Impulse Response

An impulse response of a system is what you would see if you had an oscilloscope on the system output and you gave the system an impulse or a spike for an input.

With appropriate parameter settings this effect produces an ascending buzz or whine. The ascending buzz is most prominent when given an impulsive input such as a drum hit.

To get the ascending buzz, start with about half a second of delay and set the Contour parameter to a high value (near 100%).

Contour: The Contour parameter controls the overall shape of the LaserVerb impulse response. At high values the response builds up slowly to the “hit.” As the Contour value is reduced, the response starts out lower and rises more rapidly to the “hit.”
Spacing: The Spacing parameter controls the initial separation of impulses in the impulse response and the rate of their subsequent separation. Low values result in a high initial pitch (impulses are more closely spaced) and takes longer for the pitch to lower.

Figure 72. Revrse LaserVerb - Simplified Block Diagram

Wet/Dry

0 to 100 % wet
The amount of reverbed (wet) signal relative to unaffected (dry) signal.

Rvrs W/D

0 to 100 % wet
A special wet/dry control in which the “dry” signal is in fact delayed so that it is the last sound to be sent to the output, as if the LaserVerb is being played in reverse.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

Pan

-100 to 100 %
The left and right inputs get summed to mono, the mono signal passes through the Revrse LaserVerb, and the final mono output is panned to the left and right outputs.

Panning ranges from:

Fully Left -100%
Centered 0%
Fully Right 100%

Dly Coarse

0 to 5000 ms
You can set the overall delay length from 0 to 5 seconds.
Lengthening the delay will increase the duration or decay time of the reverb.

Dly Fine

-20.0 to 20.0 ms
The delay fine adjust is added to the delay coarse adjust to provide a delay resolution down to 0.2 ms.

Spacing

0 to 200 samp
Determines the starting pitch of the ascending buzz and how fast it ascends.
The Spacing parameter sets the initial separation of impulses in the impulse response and subsequent rate of decreasing impulse separation.
The spacing between impulses is given in samples and may be a fraction of a sample.
(A sample is the time between successive digital words which is 20.8 µs or 1/48000 seconds.)
For low values, the buzz builds to a higher frequency than for higher Spacing settings.

Contour

0.0 to 100.0 %
Controls the overall envelope shape of the reverb. When set to a high value, sounds start at a high level and build slowly to the final “hit.”
As the control value is reduced, sounds start lower and build rapidly to the final “hit.

Gated LaserVerb

Effects Size : 3

LaserVerb Lite with a gate on the output.

Gated LaserVerb Presets
Ringy Drum Plate	Growler	Oil Tank
Gated LaserVerb	Wobbly Plate

Gated LaserVerb is LaserVerb Lite with a gate on the output.

The gate controls are covered under Gate. Signal routings between the inputs, the LaserVerb, the gate, and the outputs are described in the following diagram.

Figure 73. Gated LaserVerb - Signal Flow

LaserVerb is a stereo algorithm that produces interesting sounds in the reverb decay. However, the decay often lasts longer than desired. The gate may be used to cut the output signal after the input signal drops below a threshold.

You may select whether to gate the LaserVerb output based on the input signal level or the signal level at the output of the LaserVerb. In most cases the gate would be based on the input signal.

When you gate on the output signal, you must wait for the LaserVerb tail to drop below the threshold before the gate will close. Whether you gate based on the input or the output signal strength, you can select which input or output channel to use as the gating side chain signal.

Gating Side Chain Signal selections:

Left
Right
Average of the left and right magnitudes.

Wet/Dry

0 to 100 % wet
The amount of reverbed (wet) signal relative to unaffected (dry) signal.
The gate is on the wet signal path.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

Fdbk Lvl

0 to 100
The percentage of the reverb output to feed back or return to the reverb input.
Turning up the feedback is a way to stretch out the duration of the reverb, or, if the reverb is set to behave as a delay, to repeat the delay.
The higher feedback is set, the longer the decay or echo will last.

Xcouple

0 to 100

LaserVerb Lite is a stereo effect.

HF Damping

8 to 25088 Hz
The damping of high frequencies relative to low frequencies.

When set to the highest frequency (25088 Hz), there is no damping and all frequencies decay at the same rate.

At lower frequency settings, high frequency signal components will decay faster than low frequency components.

If set too low, everything will decay almost immediately.

GateIn/Out

Enables (On) or disables (Off) the gate.

NOTE: Not affected by Wet/Dry.

GateSClnp

L, R, (L+R)/2
Select whether the gate side chain signal should use the left (L) channel, right ® channel or the average magnitude of left and right channels ((L+R)/2) to control the gate.

GateSCSrc

Input or Output
Select whether the gate side chain signal should be taken from the algorithm input or from the LaserVerb Lite output.

Dly Coarse

0 to 5000 ms
You can set the overall delay length from 0 to 5 seconds.
To reduce LaserVerb to a simple delay, set the Contour and Feedback controls to 0%.

Use a delay of about half a second as a starting point.

Dly Fine

-20.0 to 20.0 ms
The delay fine adjust is added to the delay coarse adjust to provide a delay resolution down to 0.2 ms.

Spacing

0.0 to 40.0 samp
Determines the starting pitch of the ascending buzz and how fast it ascends.
The Spacing parameter sets the initial separation of impulses in the impulse response and subsequent rate of decreasing impulse separation.
The spacing between impulses is given in samples and may be a fraction of a sample.
(A sample is the time between successive digital words which is 20.8 µs or 1/48000 seconds.)
For low values, the buzz builds to a higher frequency than for higher Spacing settings.

Contour

Gate Thresh

-79.0 to 0.0 dB
The signal level in dB required to open the gate (or close the gate if Ducking is on).

Gate Duck

On or Off
When set to Off, the gate opens when the signal rises above threshold and closes when the gate time expires.

When set to On, the gate closes when the signal rises above threshold and opens when the gate time expires.

Gate Time

Gate Atk

0.0 to 228.0 ms
The time for the gate to ramp from closed to open (reverse if Ducking is on) after the signal rises above threshold.

Gate Rel

0 to 3000 ms
The time for the gate to ramp from open to closed (reverse if Ducking is on) after the gate timer has elapsed.

GateSigDly

0.0 to 25.0 ms
The delay in milliseconds (ms) of the signal to be gated relative to the side chain signal.

By delaying the main signal, the gate can be opened before the main signal rises above the gating threshold.

Panning

AutoPanner

Effects Size : 1

A stereo auto-panning effect

AutoPanner Presets
274 Simple Panner

AutoPanner is a 1 unit stereo auto pan effect.

The process of panning a stereo image consists of shrinking the image width of the input program then cyclically moving this smaller image from side to side while maintaining relative distances between program point sources.

This effect provides six different LFO shapes, variable center attenuation, and a rate scaler that scales LFO rate into the audible range for a new flavor of amplitude modulation effects.

Figure 74. AutoPanner - LFO Shapes

Final image placement can be monitored on the lower right of the Para 2 page. The top meter labeled “L” shows the left edge of the image while the second meter labeled “R” shows the right edge.

The entire image will fall between the “L” and “R” meter marks.

In the following diagram

ImageWidth is 50%
LFO Shape is set to Sine
Origin is 0%
PanWidth is 100%

Figure 75. Stereo Auto-panning

In/Out

In or Out

When set to In the auto-panner is active.
When set to Out auto-panner is bypassed.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

LFO Rate

0 to 10.00 Hz
The speed of the panning motion.

Rate Scale

1 to 25088 x
Multiplies the speed of the LFO rate into the audio range.

When above 19x, the values increment in semitone steps.

These steps are accurate when LFO Rate is set to 1.00 Hz.

Origin

-100 to 100 %
The axis for the panning motion.

At 0%, panning excursion is centered between the listening speakers.
Positive values shift the axis to the right
Negative values shift it to the left.

At -100% or +100%, there is no room for panning excursion.

Pan Width

0 to 100 %
The amount of auto pan excursion.

This value represents the percentage of total panning motion available after Origin and ImageWidth are set.

ImageWidth

0 to 100 %
The width of the original input program material before it is auto panned.

At 0%, the input image is shrunk to a single point source allowing maximum panning excursion
At 100%, the original width is maintained leaving no room for panning excursion

CentrAtten

-12 to 0 dB
Amount the signal level is dropped as it is panned through the center of the listening stereo speaker array.

For the smoothest tracking, a widely accepted subjective reference is -3dB.

Values above -3dB will cause somewhat of a bump in level as an image passes through the center.
Values below -3dB will cause a dip in level at the center.

LFO Shape

Sine, Saw+, Saw–, Pulse, Tri, and Expon
The waveform type for the LFO.

PulseWidth

0 to 100%

This parameter is active only when the Pulse waveform is selected.

When the LFO Shape is set to Pulse, this parameter sets the pulse width as a percentage of the waveform period.

The pulse is a square wave when the width is set to 50%.

Dual AutoPanner

Effects Size : 2

A dual mono auto-panner

Dual AutoPanner Presets
275 Dual Panner

Dual AutoPanner is a 2 unit sized mono auto pan effect.

Left and right inputs are treated as two mono signals, which can each be independently auto-panned.

Parameters beginning with “L” control the left input channel.
Parameters beginning with “R” control the right input channel.

Autopanning a mono signal consists of choosing an axis offset, or Origin, as the center of LFO excursion, then adjusting the desired excursion amount, or PanWidth.

The PanWidth parameter is a percentage of the available excursion space after Origin is adjusted.

If Origin is set to full left (-100%) or full right (100%) then there will be no room for LFO excursion.

Control of six different LFO shapes, variable center attenuation, and a rate scaler that scales LFO rate into the audible range for a new flavor of amplitude modulation effects are also provided for each channel.

Figure 76. Dual AutoPanner - LFO Shapes

Final image placement can be seen on the bottom right of the Para 2 and Para 3 pages respectively for left and right input channels. The moving mark represents the location of each channel within the stereo field.

In the following diagram

LFO Shape is set to Sine
Origin is 15%
PanWidth is 100%

Figure 77. Mono Auto-Panning

In/Out

In or Out

When set to In the auto-panner is active
When set to Out auto-panner is bypassed

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

LFO Rate

0 to 10.00 Hz
The speed of the panning motion.

Origin

-100 to 100 %
The axis for the panning motion.

At 0%, panning excursion is centered between the listening speakers
Positive values shift the axis to the right
Negative values shift it to the left

At -100% or +100%, there is no room for panning excursion.

Pan Width

0 to 100 %
The amount of auto pan excursion.

This value represents the percentage of total panning motion available after Origin and ImageWidth are set.

CentrAtten

-12 to 0 dB
Amount the signal level is dropped as it is panned through the center of the listening stereo speaker array.
====
For the smoothest tracking, a widely accepted subjective reference is -3dB.
====

- Values above -3dB will cause somewhat of a bump in level as an image passes through the center
- Values below -3dB will cause a dip in level at the center

LFO Shape

Sine, Saw+, Saw–, Pulse, Tri, and Expon
The waveform type for the LFO.

PulseWidth

0 to 100%

This parameter is active only when the Pulse waveform is selected.

When the LFO Shape is set to Pulse, this parameter sets the pulse width as a percentage of the waveform period.

The pulse is a square wave when the width is set to 50%.

Stereo Imaging

Stereo Image

Effects Size : 1

Stereo enhancement with stereo channel correlation metering

Stereo Image Presets
276 Widespread

Stereo Image is a stereo enhancement algorithm with metering for stereo channel correlation.

The stereo enhancement performs simple manipulations of the sum and difference of the left and right input channels to allow widening of the stereo field and increased sound field envelopment.

After manipulating sum and difference signals, the signals are recombined (a sum and difference of the sum and difference) to produce final left and right output.

Figure 78. Stereo Image - Block Diagram

The sum of left and right channels represents the mono or center mix of your stereo signal.

The Difference Signal: The difference of left and right channels contains the part of the signal that contains stereo spatial information.

The Stereo Image algorithm has controls to change the relative amounts of sum (or center) versus difference signals.

By increasing the difference signal, you can broaden the stereo image.

Be warned, though, that too much difference signal will make your stereo image sound “phasey.”

With phasey stereo, acoustic images become difficult to localize and can sound like they are coming from all around or from within your head.

Bass

A bass shelf filter on the difference signal is also provided.

By boosting only the low frequencies of the difference signal, you can greatly improve your sense of stereo envelopment without destroying your stereo sound field. (Envelopment is the feeling of being surrounded by your acoustic environment.)

Localized stereo images still come from between your stereo loudspeakers, but there is an increased sense of being wrapped in the sound field.

Stereo Correlation Meter

The Stereo Image algorithm contains a stereo correlation meter.

The stereo correlation meter tells you how alike or how different your output stereo channels are from each other.

The correlation meter can give you an indication of how well a recording will mix to mono.

When the meter is at 100% correlation, then your signal is essentially mono.
At 0% correlation, your left and right channels are the same, but polarity inverted (there is only the difference signal).

RMS Signal Levels

The meter follows RMS signal levels (root-mean- square) and the RMS Settle parameter controls how responsive the meter is to changing signals.

The ‘M’ part of RMS is “mean” or average of the squared signal.

Since a mean over all time is neither practical or useful, we must calculate the mean over shorter periods of time.

If the time is too short we are simply following the signal wave form, which is not helpful either, since the meter would constantly bounce around.

The RMS Settle parameter provides a range of useful time scales.

L In Gain

Off, -79.0 to 24.0 dB
The input gain of the left channel in decibels (dB).

R In Gain

Off, -79.0 to 24.0 dB
The input gain of the right channel in decibels (dB).

CenterGain

Off, -79.0 to 24.0 dB
The level of the sum of left and right channels in decibels (dB).

The summed stereo signal represents the mono or center mix.

Diff Gain

Off, -79.0 to 24.0 dB
The level of the difference of left and right channels in decibels (dB).

The difference signal contains the spatial component of the stereo signal.

L/R Delay

-500.0 to 500.0 samp
If this parameter is positive, the left signal is delayed by the indicated amount.

If it is negative, the right channel is delayed. You can use this parameter to try to improve cancellation of the difference signal if you suspect one channel is delayed with respect to the other.

RMS Settle

0.0 to 300.0 dB/s
Controls how fast the RMS meters can rise or fall with changing signal levels.

DiffBassG

-79.0 to 24.0 dB
DiffBassG is the gain parameter of a bass shelf filter on the difference signal.

DiffBassG sets how many decibels (dB) to boost or cut the low frequencies.

By boosting the low frequency components of the difference signal you can increase the sense of acoustic envelopment, the sense of being surrounded by an acoustic space.

DiffBassF

8 to 25088 Hz
The transition frequency in Hertz (Hz) of the difference signal bass shelf filter is set by DiffBassF.

Mono→Stereo

Effects Size : 1

Stereo simulation from a mono input signal

Mono->Stereo Presets
277 Widener Mn→St

Mono → Stereo creates a stereo signal from a mono input signal. The algorithm works by combining a number of band-splitting, panning and delay tricks.

The In Select parameter lets you choose the left or right channel for you mono input, or you may choose to sum the left and right inputs.

Figure 79. Mono→Stereo - Block Diagram

The mono input signal is split into three frequency bands (Low, Mid, and High).

The frequencies at which the bands get split are set with the Crossover parameters.

Each band can then be delayed and panned to some position within your stereo field.

The final step manipulates the sum and difference signals of the pseudo-stereo signal created by recombining the split frequency bands.

The sum of left and right channels represents the mono or center mix of your stereo signal.

The Difference Signal

The difference of left and right channels contains the part of the signal that contains stereo spatial information.

The Mono → Stereo algorithm has controls to change the relative amounts of sum (or center) versus difference signals.

By increasing the difference signal, you can broaden the stereo image.

Be warned, though, that too much difference signal will make your stereo image sound “phasey.”

With phasey stereo, acoustic images become difficult to localize and can sound like they are coming from all around you or from within your head.

Bass: A bass shelf filter on the difference signal is also provided.

By boosting only the low frequencies of the difference signal, you can greatly improve your sense of stereo envelopment without destroying your stereo sound field. (Envelopment is the feeling of being surrounded by your acoustic environment.)

Localized stereo images still come from between your stereo loudspeakers, but there is an increased sense of being wrapped in the sound field.

In/Out

In or Out
The algorithm is functioning when In/Out is set to In.

If set to Out, whatever is on the input channels gets passed to the output unaltered.

Out Gain

Off, -79.0 to 24.0 dB
The output gain of the pseudo-stereo signal in decibels (dB).

CenterGain

Off, -79.0 to 24.0 dB
The level of the sum of the intermediate left and right stereo channels in decibels (dB).

The summed stereo signal represents the mono or center mix.

Diff Gain

Off, -79.0 to 24.0 dB
The level of the difference of the intermediate left and right stereo channels in decibels (dB).

The difference signal contains the spatial component of the stereo signal.

In Select

L, R, or (L+R)/2
The input signal may come from the left L or right R input channel, or the left and right channels may be summed to obtain the mono signal (L+R)/2.

DiffBassG

-79.0 to 24.0 dB
By boosting the low frequency components of the difference signal of the intermediate stereo result, you can increase the sense of acoustic envelopment, the sense of being surrounded by an acoustic space.

DiffBassG is the gain parameter of a bass shelf filter on the difference signal.

DiffBassG sets how many decibels (dB) to boost or cut the low frequencies.

DiffBassF

8 to 25088 Hz
The transition frequency in Hertz (Hz) of the difference signal bass shelf filter is set by DiffBassF.

Crossover 1
Crossover 2

8 to 25088 Hz
The two Crossover parameters set the frequencies at which the band-split filters split the mono signal into three bands.

The two parameters are interchangeable: either may have a higher frequency than the other.

Pan Low
Pan Mid
Pan High

-100 to 100 %
The panning of each band is separately controllable.
-100% is fully left and 100% is fully right.

Delay Low
Delay Mid
Delay High

0.0 to 1000.0 ms
The delays are set in milliseconds (ms).

DynamicStereoize

Effects Size : 2

Stereo widening based on dynamic signal levels

DynamicStereoize Presets
278 Dynam Stereoizer

DynamicStereoize is a stereo enhancement (or reduction) algorithm.

By increasing the level of the difference signal between left and right input channels relative to the summed left and right channels (mono), you get an increased sense of stereo separation.

Likewise, reducing the difference signal relative to the summed signal makes the sound more mono or centered. So far this description differs little from the algorithm Stereo Image.

Now if we place dynamic range controls (compressor and/or expander) on either the summed or difference signal paths, some interesting things happen.

A compressor reduces output signal level when the input signal level gets louder.

An expander reduces output signal level when the input signal gets softer. With a compressor or expander on one of the sum or difference signal paths, your sound can be made very spacious at low signal levels but centered at higher levels.

You can also achieve the opposite effect with low level signal centered and high signal levels wide.

Figure 80. DynamicStereoize - Block Diagram

Compressor/Expander

The compressor/expander switching in the figure above looks a little complicated, but conceptually it is very simple.

Using the Comp +/- parameter you select whether to compress or expand the summed left and right signal ((L+R)/2) or the difference signal ((L-R)/2).

The final sum and difference calculation reconstructs the original left and right signals (assuming you turned off the intermediate processing).

You can prove this to yourself by solving the equations (L+R)/2 + (L-R)/2 and (L+R)/2 - (L-R)/2.

Example using Compressor/Expander

Let’s look at an example using compression and expansion on the sum and difference signals.

Mono loud, spacious soft: We want to make the sound mono when it is loud and spacious when it is soft.

There are two approaches: you can expand the summed signal, or compress the difference signal.

By expanding the summed signal, the mono component gets reduced as the sound gets quieter, producing a more spacious sound.

By compressing the difference signal, the out of phase components are reduced as the signal gets louder for a more mono signal at higher levels.

You will have to work with the CenterGain and Diff Gain parameters to achieve the balance of spaciousness and mono you are looking for.

The difference between Compress/Expand and the compressor/expander used here is that this compressor/expander is mono, working on a single (sum or difference) channel.

Bass

Stereo Correlation Meter

The DynamicStereoize algorithm contains a stereo correlation meter.

The stereo correlation meter tells you how alike or how different your output stereo channels are from each other.

The correlation meter can give you an indication of how well a recording will mix to mono.

When the meter is at 100% correlation, then your signal is essentially mono.
At 0% correlation, your left and right channels are the same, but polarity inverted (there is only the difference signal).

RMS Signal Levels

Left/Right Delay

The left input can be delayed with respect to the right input channel (or the other way around).

You can use the L/R Delay to realign your inputs in time or to experiment with the precedence effect.

With the precedence effect, when we hear a sound first with the left ear then with the right ear, it sounds like the sound is coming from the left side.

Flanger

Flanging was originally created by summing the outputs of two un-locked tape machines while varying their sync by pressing a hand on the outside edge of one reels thus the name "reel-flanging".

Comb Filter: The key to achieving the flanging effect is the summing of a signal with a time-displaced replica of itself. The result is a series of notches in the frequency spectrum.

These notches are equally spaced in (linear) frequency at multiples whose wavelengths are equal to the time delay.

The result is generally referred to as a comb .
(The name arising from the resemblance of the spectrum to a comb.)

If the levels of the signals being added or subtracted are the same, the notches will be of infinite depth (in dB) and the peaks will be up 6 dB.
Delay Length: Flanging is achieved by time-varying the delay length, thus changing the frequencies of the notches.

The shorter the delay time, the greater the notch separation.

This delay time variation imparts a sense of motion to the sound.

Typically the delay times are on the order of 0-5 ms.

Longer times begin to get into the realm of chorusing, where the ear begins to perceive the audio output as nearly two distinct signals, but with a variable time displacement.

Figure 81. Comb Filter - solid line for addition; dashed line for subtraction

Multi-Tap-Delay

The heart of the flanger implemented here is a multi-tap delay line.

You can set the level of each tap as a percentage of the input level, and the level may be negative (phase-inverted).

One tap is a simple static delay over which you can control the length of delay (from the input tap).

LFO: Four of the taps can have their lengths modulated up and down by a low frequency oscillator (LFO).

You are given control of the rate of the LFOs, how far each LFO can sweep through the delay line, and the relative phases of the LFOs (i.e., whether the LFO is taking the taps from the input tap or bringing them toward it).
LFO Tempo / LFO Period: The flanger uses tempo units (based on the sequencer tempo or MIDI clock if you like), together with the number of tempo beats per LFO cycle.

Thus if the tempo is 120 bpm (beats per minute) and the LFO Period is set to 1 beat, the LFOs will pass through 120 complete cycles in a minute or 2 cycles per second (2 Hz).

Increasing the LFO Period increases the period of the LFOs (slows them down). An LFO Period setting of 16 beats will take 4 measures (in 4/4 time) for a complete LFO oscillation.

Excursion

You can set how far each LFO can sweep through the delay line with the excursion controls (Xcurs).

The excursion is the maximum distance an LFO will move from the center of its sweep.

The total range of an LFO is twice the excursion.

You set the delay to the center of LFO excursion with the Dly parameters.

The excursion and delay controls both have coarse and fine adjustments. By setting the excursion to zero length, the LFO delay tap becomes a simple static tap.

Modifying the delay to the center of LFO excursion will result in a sudden change of delay length and consequently, a discontinuity in the signal being read from the delay line.

This can produce a characteristic zippering effect.

The Dly parameters should be as long as the Xcurs parameters or longer, or else changing (or modulating) the excursion will force the center of LFO excursion to move, with the resulting signal discontinuities.

Static Delay Tap

The static delay tap does not suffer the zippering problem, and changes to its length will occur smoothly.

You can assign the static delay tap to an FX Mod, and use the source controller to do manual flanging.

Figure 82. Delay for a single LFO

Example - Classic Thru→Zero Flanger Effect.: Consider a simple example where you have an LFO tap signal being subtracted from the static delay tap signal.

If the delays are set such that at certain times both taps are the same length, then both taps have the same signal and the subtraction produces a null or zero output.

The effect is most pronounced when the static tap is set at one of the ends of the LFO excursion where the LFO tap motion is the slowest.

This is the classic Thru-Zero flanger effect.

Adding other LFO taps to the mix increases the complexity of the final sound, and obtaining a true Thru-Zero effect may take some careful setting of delays and LFO phases.
Wet / Dry Control: The flanger has a Wet/Dry control as well, which can further add complexity to the output as the dry signal is added to various delayed wet components for more comb filtering.
LFO Phase Relationship: When using more than one LFO, you can set up the phase relationships between each of the LFOs.

The LFOs of the left channel and those of the right channel will be set up in the same phase relationship except that you may offset the phases of the right channel as a group relative to the left channel (L/R Phase).

L/R Phase is the only control which treats left and right channels differently and has a significant effect on the stereo image.

If you have tempo set to the system tempo, the phases will maintain their synchronization with the tempo clock.

At the beat of the tempo clock, a phase set to 0° will be at the center of the LFO excursion and moving away from the delay input.
Feedback: Regenerative feedback has been incorporated in order to produce a more intense resonant effect.

The signal is fed back is from the first LFO delay tap (LFO1), and has its own level control (Fdbk Level).

In-phase spectral components arriving at the summer add together, introducing a series of resonant peaks in the frequency spectrum between the notches.

The amplitude of these peaks depends on the degree of feedback, and they can be made very resonant.
Cross-coupling: Cross-coupling (Xcouple) allows the signals of the right and left channels to be mixed or swapped.

The cross- coupling is placed after the summation of the feedback to the input signal.

When feedback and cross-coupling are turned up, you will get a ping-pong effect between right and left channels.
Lowpass Filter: A lowpass filter (HF Damping) right before the input to the delay line is effective in emulating the classic sounds of older analog flangers with their limited bandwidths (typically 5-6kHz).
Notch Density: As stated earlier, it is the movement of the notches created in the frequency spectrum that give the flanger its unique sound.

It should be obvious that sounds with a richer harmonic structure will be effected in a much more dramatic way than harmonically starved sounds.

Having more notches, i.e. a greater "notch-density", should produce an even more intense effect.

This increase in notch density may be achieved by having a number of modulating delay lines, all set at the same rate, but different depths. Setting the depths proportionately results in a more pleasing effect.
Analog Noise: An often characteristic effect of flanging is the sound of system noise being flanged.

Various pieces of analog gear add noise to the signal, and when this noise passes through a flanger, you can hear the noise "whooshing".

In Kurzweil instruments, the noise level is very low, and in fact if no sound is being played, there is no noise at all at this point in the signal chain.

To recreate the effect of system noise flanging, white noise may be added to the input of the flanger signal (Flanger 2 only).

Since white noise has a lot of high frequency content and may sound too bright, it may be tamed with a first-order lowpass filter.

Flanger 1

Effects Size : 1

xxxxxxxxxxx

Flanger 1 Presets
225 Big Slow Flange	231 Wetlip Flange	249 CacophonousFlng
226 Squeeze Flange	235 Ned Flangers	495 Dr. Who
228 Throaty Flange	236 Wispy Flange	229 PsuedoAnaGtrFlng
246 Gulp Flange	230 Flanger Double	247 Splat Flange

Figure 83. Flanger 1 - Simplified Block Diagram of Left Channel

Flanger 2

Effects Size : 2

xxxxxxxxxxx

Flanger 2 Presets
232 Simply Flange	234 Soft Edge Flange	233 Analog Flanger
245 Stereo Flanger

Figure 84. Flanger 2 - Simplified Block Diagram of Left Channel

Wet/Dry

-100 to 100 % wet
The relative amount of input signal and flanger signal that is to appear in the final effect output mix.

When set to 0%, the output is taken only from the input (dry).
When set to 100%, the output is all wet.

Negative values polarity invert the wet signal.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

Fdbk Level

-100 to 100 %
The level of the feedback signal into the delay line.

The feedback signal is taken from the LFO1 delay tap.

Negative values polarity invert the feedback signal.

Xcouple

-100 to 100 %
How much of the left channel input and feedback signals are sent to the right channel delay line and vice versa.

At 50%, equal amounts from both channels are sent to both delay lines.
At 100%, the left feeds the right delay and vice versa. Xcouple has no effect if Fdbk Level is set to 0%.

HF Damping

8 to 25088 Hz
The amount of high frequency content of the signal sent into the delay lines.

This control determines the cutoff frequency of the one-pole (-6dB/octave) lowpass filters.

LFO Tempo

System, 1 to 255 BPM
Basis for the rates of the LFOs, as referenced to a musical tempo in bpm (beats per minute).

When this parameter is set to System, the tempo is locked to the internal sequencer tempo or to incoming MIDI clocks.

In this case, FXMods (FUNs, LFOs, ASRs etc.) will have no effect on the Tempo parameter.

LFO Period

1/24 to 32 bts
Sets the LFO rate based on the Tempo determined above: the number of beats corresponding to one period of the LFO cycle.

Example: For example, if the LFO Period is set to 4, the LFOs will take four beats to pass through one oscillation, so the LFO rate will be 1/4th of the Tempo setting.

If it is set to 6/24 (=1/4), the LFO will oscillate four times as fast as the Tempo.

At 0, the LFOs stop oscillating and their phase is undetermined (wherever they stopped).

Noise Gain

Off, -79.0 to -30.0 dB
NOTE: Flanger 2 only.

The amount of noise (dB relative to full scale) to add to the input signal.

In many flangers, you can hear the noise floor of the signal being flanged, but with low-noise instruments, if there is no input signal, there is no noise floor unless it is explicitly added.

Noise LP

8 to 25088 Hz

Flanger 2 only.

The cut-off frequency of a one pole lowpass filter acting on the noise injection signal.

The lowpass removes high frequencies from an otherwise pure white noise signal.

StatDlyCrs

0.0 to 228.0 ms
The length of the static delay tap.

The name suggests the tap is stationary, but it can be connected to a control source such as a data slider, a ribbon, or a V.A.S.T. function to smoothly vary the delay length.

The range for all delays and excursions is 0 to 230 ms, but for flanging the range 0 to 5 ms is most effective.

StatDlyFin

-127 to 127 samp
A fine adjustment to the static delay tap length.

The resolution is one sample.

StatDlyLvl

-100 to 100 %
The level of the static delay tap.

Negative values polarity invert the signal.

Setting any tap level to 0% turns off the delay tap.

Xcurs1 Crs
Xcurs2 Crs

0.0 to 228.0 ms
The LFO excursion controls set how far the LFO modulated delay taps can move from the center of their ranges.

The total range of the LFO sweep is twice the excursion.

If the excursion is set to 0, the LFO does not move and the tap behaves like a simple delay line set to the minimum delay.

The excursion cannot be made longer than the delay to the center of excursion (see Dly Crs and Dly Fin) because delays cannot be made horter than 0.

If you attempt longer excursions, the length of the Dly Crs/Fin will be forced to increase (though you will not see the increased length displayed in the Dly Crs/Fin parameters).

The range for all delays and excursions is 0 to 230 ms, but for flanging the range 0 to 5 ms is most effective.

This parameter is a coarse adjustment for the excursion.

Xcurs1 Fin
Xcurs2 Fin

-127 to 127 samp
A fine adjustment for the LFO excursions.

The resolution is one sample.

Dly1 Crs
Dly2 Crs

0.0 to 228.0 ms
The delay to the center of LFO tap range.

The maximum delay will be this delay plus the LFO excursion delay.
The minimum delay will be this delay minus the LFO excursion delay.

Since delays cannot be less than 0 ms in length, the this delay length will be increased if LFO excursion is larger than this delay length.

TIP : For flanging the range 0 to 5 ms is most effective.

This parameter is a coarse adjustment for the delay.

Dly1 Fin
Dly2 Fin

-127 to 127 samp
A fine adjustment to the minimum delay tap lengths.
The resolution is one sample.

LFO1 Level
LFO2 Level

-100 to 100 %
The levels of the LFO modulated delay taps.

Negative values polarity invert the signal.

Setting any tap level to 0% turns off the delay tap.

LFO1 Phase
LFO2 Phase

0.0 to 360.0 deg
The phase angles of the LFOs relative to each other and to the system tempo clock, if turned on (see Tempo).
Example::
For example, if one LFO is set to 0° and another is set to 180°, then when one LFO delay tap is at its shortest, the other will be at its longest.

If the system tempo clock is on, the LFOs are synchronized to the clock with absolute phase.

A phase of 0° will put an LFO tap at the center of its range and its lengthening.

L/R Phase

0.0 to 360.0 deg
Adds the specified phase angle to the right channel LFOs.

In all other respects the right and left channels are symmetric.

By moving this control away from 0°, the stereo sound field is broken up and a stereo image becomes difficult to spatially locate.

The effect is usually described as “phasey.”

It tends to impart a greater sense of motion.

Vocal Combination

Cut out noise during vocal silence.

Two combination algorithms are provided with vocal processing in mind. Both include a gate followed by a compressor and a reverb.

In Gate+Cmp[EQ]+Rvb, equalization is included as part of the compressor’s side-chain processing.

Side-chain equalization allows some interesting processing possibilities including “de-essing” (by boosting the treble in the side-chain).
In Gate+Cmp<>EQ+Rvb, the equalization can be configured before or after the compressor.

EQ: The EQ includes bass, treble and mid controls (gain and frequency for each plus width for the mid EQ).
Gate: The gate allows you to cut out noise during vocal silence.

You must decide whether to gate based on left or right channels or to gate based on both channels (average magnitude).

Both the gate and compressor have their own side-chain processing paths.

For both the gate and compressor, side-chain input may be taken from either the left or right channels, or the average signal magnitude of the left and right channels may be selected using the GateSCInp or CompSCInp parameters.

The gate is the same as used in Gate.
Reverb: The reverb is the same as used in MiniVerb.
You will find all the same controls and room settings.

Gate+Cmp[EQ]+Rvb

Effects Size : 4

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Gate+Cmp[EQ]+Rvb Presets
110 Vocal Room
111 Vocal Stage

Figure 85. Gate+Cmp[EQ]+Rvb - Simplified Block Diagram

Gate+Cmp<>EQ+Rvb

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Gate+Cmp<>EQ+Rvb Presets
- - - - - - - - - - - -

Figure 86. Gate+Cmp<>EQ+Rvb set to Cmp→EQ

Figure 87. Gate+Cmp<>EQ+Rvb set to EQ→Cmp

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the entire algorithm.

GateIn/Out

In or Out
When set to In the gate is active.
When set to Out the gate is bypassed.

GateSCInp

L, R, (L+R)/2
Select the input source channel for gate side-chain processing: left, right or both.

For both (L+R)/2 the averaged magnitude is used.

CompIn/Out

In or Out
When set to In the compressor is active.
When set to Out the compressor is bypassed.

CompSCInp

L, R, (L+R)/2
Select the input source channel for compressor side-chain processing—Left, Right or Both.

For both (L+R)/2 the averaged magnitude is used.

FdbkComprs

In or Out
A switch to set whether the compressor side-chain is configured for feed-forward (Out) or feedback (In).

Feedback compression is not available in the Gate+Cmp<>EQ+Rvb algorithm.

A→B cfg

Cmp→EQ or EQ→Cmp
Controls the routing order of the compressor and EQ in Gate+Cmp<>EQ+Rvb.

When set to Cmp→EQ, the output of the compressor feeds into the EQ.

When set to EQ→Cmp, the EQ feeds into the compressor.

A compressor is a non-linear, time-variant effect, so the relative order can make a difference, particularly when the compression is extreme enough to behave as distortion.

Gate Thres

-79.0 to 0.0 dB
The signal level in dB required to open the gate (or close the gate if Gate Duck is on).

Gate Duck

Gate Time

25 to 3000 ms
The time in seconds that the gate will stay fully on after the signal envelope rises above threshold.

The gate timer is started or restarted whenever the signal envelope rises above threshold.

If Retrigger is On, the gate timer is continually reset while the side chain signal is above the threshold.

Gate Atk

0.0 to 228.0 ms
The time for the gate to ramp from closed to open (reverse if Gate Duck is on) after the signal rises above threshold.

Gate Rel

0 to 3000 ms
The time for the gate to ramp from open to closed (reverse if Gate Duck is On) after the gate timer has elapsed.

GateSigDly

0.0 to 25.0 ms
The delay in milliseconds (ms) of the signal to be gated relative to the side chain signal.

By delaying the main signal, the gate can be opened before the main signal rises above the gating threshold.

Comp Atk

0.0 to 228.0 ms
The time for the compressor to start to cut in when there is an increase in signal level (attack) above the threshold.

Comp Rel

0 to 3000 ms
The time for the compressor to stop compressing when there is a reduction in signal level (release) from a signal level above the threshold.

CompSmooth

0.0 to 228.0 ms
A lowpass filter in the compressor side-chain signal path.

It is intended to smooth the output of the compressor’s envelope detector.

Smoothing will affect the attack or release times when the smoothing time is longer than one of the other times.

CompSigDly

0.0 to 25.0ms
The time in ms by which the input signal should be delayed with respect to compressor side-chain processing (i.e. side-chain predelay).

This allows the compression to appear to take effect just before the signal actually rises.

Comp Ratio

1.0:1 to 100:1, Inf:1
The compression ratio.

High ratios are highly compressed
Low ratios are moderately compressed

Comp Thres

-79.0 to 0.0dB
The compressor threshold level in dBFS (decibels relative to full scale) above which the signal begins to be compressed.

CompMakeUp

Off, -79.0 to 24.0 dB
A gain or amplitude control provided to offset gain reduction due to compression.

The EQ parameters with names starting with CmpSC refer to EQ filters in the side-chain processing path of Gate+Cmp[EQ]+Rvb.

The prefix is not used in Gate+Cmp<>EQ+Rvb where the EQ is in the main signal path.

CmpSCBassG
Bass Gain

-79.0 to 24.0 dB
The amount of boost or cut that the bass shelving filter should apply to the low frequency signals in dB.

Every increase of 6 dB approximately doubles the amplitude of the signal.

Positive values boost the bass signal below the specified frequency.
Negative values cut the bass signal below the specified frequency.

CmpSCBassF
Bass Freq

8 to 25088 Hz
The center frequency of the bass shelving filter in intervals of one semitone.

CmpSCTrebG
Treb Gain

-79.0 to 24.0 dB
The amount of boost or cut that the treble shelving filter should apply to the high frequency signals in dB.

Every increase of 6 dB approximately doubles the amplitude of the signal.

Positive values boost the treble signal above the specified frequency.
Negative values cut the treble signal above the specified frequency.

CmpSCTrebF
Treb Freq

8 to 25088 Hz
The center frequency of the treble shelving filters in intervals of one semitone.

CmpSCMidG
Mid Gain

-79.0 to 24.0 dB
The amount of boost or cut that the parametric mid filter should apply in dB to the specified frequency band.

Every increase of 6 dB approximately doubles the amplitude of the signal.

Positive values boost the signal at the specified frequency.
Negative values cut the signal at the specified frequency.

CmpSCMidF
Mid Freq

8 to 25088 Hz
The center frequency of the parametric mid filter in intervals of one semitone.

The boost or cut will be at a maximum at this frequency.

CmpSCMidW
Mid Width

0.010 to 5.000 oct
The bandwidth of the side chain parametric mid filter may be adjusted.

You specify the bandwidth in octaves.

Small values result in a very narrow filter response.
Large values result in a very broad response.

Reverb W/D

0 to 100 %wet
A simple mix of the reverb sound with the dry (compressed) sound.

Rv PreDly L/R

0 to 620 ms
The delay between the start of a sound and the output of the first reverb reflections from that sound.

Longer predelays can help make larger spaces sound more realistic.

Longer times can also help improve the clarity of a mix by separating the reverb signal from the dry signal, so the dry signal is not obscured.

Likewise, the wet signal will be more audible if delayed, and thus you can get by with a dryer mix while maintaining the same subjective wet/dry level.

Rv Time

0.5 to 30.0 s, Inf
The reverb time displayed is accurate for normal settings of the other parameters (HF Damping = 25088 kHz, and Rv DiffScl, Rv SizeScl and Rv Density = 1.00x).

Changing Rv Time to Inf creates an infinitely sustaining reverb.

Rv Type

This parameter effectively allows you to have several different reverb algorithms only a parameter change away.

Smaller Rv Types will sound best with shorter Rv Times, and vice versa.

That since this parameter changes the structure of the reverb algorithm, you may not modulate it.)

Rv HF Damp

8 to 25088 Hz
Reduces high frequency components of the reverb above the displayed cutoff frequency.

Removing higher reverb frequencies can often make rooms sound more natural.

Rv DiffScl

0.00 to 2.00x
A multiplier which affects the diffusion of the reverb.

At 1.00x, the diffusion will be the normal, carefully adjusted amount for the current Rv Type.

Altering this parameter will change the diffusion from the preset amount.

Rv SizeScl

0.00 to 4.00x
A multiplier which changes the reverb size of the current room.

At 1.00x, the room will be the normal, carefully tweaked size of the current Rv Type.

Altering this parameter will change the size of the room, and thus will cause a subtle coloration of the reverb (since the room’s dimensions are changing).

Rv Density

0.00 to 4.00x
A multiplier which affects the density of the reverb.

At 1.00x, the room density will be the normal, carefully set amount for the current Rv Type.

Altering this parameter will change the density of the reverb, which may color the room slightly.

Gate

Signal gating

A gate behaves like an on off switch for a signal.

One or both input channels is used to control whether the switch is on (gate is open) or off (gate is closed).

The on/off control is called “side chain” processing. You select which of the two input channels or both is used for side chain processing.

When you select both channels, the sum of the left and right input amplitudes is used.

The gate is opened when the side chain amplitude rises above a level that you specify with the Threshold parameter.

Stereo / Mono Effect: Gate and Gate w/SC EQ perform stand-alone gate processing and can be configured as a stereo or mono effects.

As a stereo effect, the stereo signal gates itself based on its amplitude. As a mono effect, you can use one mono input signal to gate a second mono input signal (or one channel can gate itself).

Separate output gain and panning for both channels is provided for improved mono processing flexibility.

Figure 88. Gate - Simplified Block Diagram

ReTrigger

Gate w/SC EQ will behave differently depending on whether the Retrigger parameter is set to Off or On.

For the simpler Gate, there is no Retrigger parameter, and it is as if Retrigger is always On.

On: If Retrigger is On, the gate will stay open for as long as the side chain signal is above the threshold.

When the signal drops below the threshold, the gate will remain open for the time set with the Gate Time parameter.

At the end of the Gate Time, the gate closes.

When the signal rises above threshold, it opens again.

What is happening is that the gate timer is being constantly retriggered while the signal is above threshold.

You will typically use the gate with Retrigger set to on for percussive sounds.

Figure 89. Gate and Gate w/SC EQ when Retrigger is On

Off: If Retrigger is Off (Gate w/SC EQ only), then the gate will open when the side chain signal rises above threshold as before.

The gate will then close as soon as the gate time has elapsed, whether or not the signal is still above threshold.

The gate will not open again until the envelope of the side chain signal falls below the threshold and rises above threshold again.

Since an envelope follower is used, you can control how fast the envelope follows the signal with the Env Time parameter.

Retrigger set to Off is useful for gating sustained sounds or where you need precise control of how long the gate should remain open.

Figure 90. Gate w/SC EQ signal envelope when Retrigger is Off

Ducking: If Ducking is turned On, then the behavior of the gate is reversed.

The gate is open while the side chain signal is below threshold, and it closes when the signal rises above threshold.
Atk Time and Rel Time: If the gate opened and closed instantaneously, you would hear a large digital click, like a big knife switch was being thrown.

Obviously that’s not a good idea, so Atk Time (attack) and Rel Time (release) parameters are use to set the times for the gate to open and close.

More precisely, depending on whether Ducking is off or on, Atk Time sets how fast the gate opens or closes when the side chain signal rises above the threshold.

The Rel Time sets how fast the gate closes or opens after the gate timer has elapsed.
Signal Dly: The Signal Dly parameter delays the signal being gated, but does not delay the side chain signal.

By delaying the main signal relative to the side chain signal, you can open the gate just before the main signal rises above threshold.

It’s a little like being able to pick up the telephone before it rings!
Bass / Parametric & Treble Filter: For Gate w/SC EQ (not the simpler Gate), filtering can be done on the side chain signal.

There are controls for a bass shelf filter, a treble shelf filter and a parametric (mid) filter.

By filtering the side chain, you can control the sensitivity of the gate to different frequencies.

For example, you can have the gate open only if high frequencies are present, or only if low frequencies are present.

Gate

Effects Size : 1

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Gate Presets
340 Simple Gate

Gate w/SC EQ

Effects Size : 2

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Gate w/SC EQ Presets
341 Gate w/SC EQ

In/Out

In or Out
When set to In the gate is active.
When set to Out the gate is bypassed.

L/R Out Gain

Off, -79.0 to 24.0 dB
The separate output signal levels in dB for the left and right channels.

The output gains are calculated before the final output panning.

L/R Pan

-100 to 100 %
Both of the gated signal channels can be panned between left and right prior to final output.

This can be useful when the gate is used as a mono effect, and you don’t want to hear one of the input channels, but you want your mono output panned to stereo.

-100% is panned to the left
100% is panned to the right.

SC Input

L, R or (L+R)/2
The side chain input may be the amplitude of the left L input channel, the right R input channel, or the sum of the amplitudes of left and right (L+R)/2.

You can gate a stereo signal with itself by using the sum, a mono signal with itself, or you can gate a mono signal using a second mono signal as the side chain.

Threshold

-79.0 to 0.0 dB
The signal level in dB required to open the gate (or close the gate if Ducking is On).

Ducking

Env Time

[blue]_ 0 to 3000 ms_
Envelope time is for use when Retrigger is set to Off.

The envelope time controls the time for the side chain signal envelope to drop below the threshold.

At short times, the gate can reopen rapidly after it has closed, and you may find the gate opening unexpectedly due to an amplitude modulation of the side chain signal.

For long times, the gate will remain closed until the envelope has a chance to fall, and you may miss gating events.

Gate Time

0 to 3000 ms
The time in seconds that the gate will stay fully on after the signal envelope rises above threshold.

The gate timer is started or restarted whenever the signal envelope rises above threshold.

If Retrigger is On, the gate timer is continually reset while the side chain signal is above the threshold.

Atk Time

0.0 to 228.0 ms
The time for the gate to ramp from closed to open (reverse if Ducking is On) after the signal rises above threshold.

Rel Time

0 to 3000 ms
The time for the gate to ramp from open to closed (reverse if Ducking is On) after the gate timer has elapsed.

Signal Dly

0.0 to 25.0 ms
The delay in milliseconds (ms) of the signal to be gated relative to the side chain signal.

By delaying the main signal, the gate can be opened before the main signal rises above the gating threshold.

Gate w/SC EQ Parameters

Retrigger

On or Off
If Retrigger is On, the gate timer is constantly restarted (retriggered) as long as the side chain signal is above the threshold.

The gate then remains open (assuming Ducking is Off) until the signal falls below the threshold and the gate timer has elapsed.

If Retrigger is Off, then the gate timer starts at the moment the signal rises above the threshold and the gate closes after the timer elapses, whether or not the signal is still above threshold.

With Retrigger off, use the Env Time to control how fast the side chain signal envelope drops below the threshold.

With Retrigger set to off, the side chain envelope must fall below threshold before the gate can open again.

SCBassGain

-79.0 to 24.0 dB
The amount of boost or cut that the side chain bass shelving filter should apply to the low frequency signals in dB.

Every increase of 6 dB approximately doubles the amplitude of the signal.

Positive values boost the bass signal below the specified frequency.
Negative values cut the bass signal below the specified frequency.

SCBassFreq

8 to 25088 Hz
The center frequency of the side chain bass shelving filters in intervals of one semitone.

SCTrebGain

-79.0 to 24.0 dB
The amount of boost or cut that the side chain treble shelving filter should apply to the high frequency signals in dB.

Every increase of 6 dB approximately doubles the amplitude of the signal.

Positive values boost the treble signal above the specified frequency.
Negative values cut the treble signal above the specified frequency.

SCTrebFreq

8 to 25088 Hz
The center frequency of the side chain treble shelving filters in intervals of one semitone.

SCMidGain

-79.0 to 24.0 dB
The amount of boost or cut that the side chain parametric mid filter should apply in dB to the specified frequency band.

Every increase of 6 dB approximately doubles the amplitude of the signal.

- Positive values boost the signal at the specified frequency.
- Negative values cut the signal at the specified frequency.

SCMidFreq

8 to 25088 Hz
The center frequency of the side chain parametric mid filter in intervals of one semitone. The boost or cut will be at a maximum at this frequency.

SCMidWidth

0.010 to 5.000 oct
The bandwidth of the side chain parametric mid filter may be adjusted.

You specify the bandwidth in octaves.

Small values result in a very narrow filter response.
Large values result in a very broad response.

Phasers

A variety of single notch/bandpass phasers

A simple phaser is an algorithm that produces a vague swishing or phasey effect.

When the phaser signal is combined with the dry input signal or the phaser is fed back on itself, peaks and/or notches can be produced in the filter response making the effect much more pronounced.

Allpass Filter: A phaser uses a special filter called an allpass filter to modify the phase response of a signal’s spectrum without changing the amplitude of the spectrum.

Okay, that was a bit of a mouthful—so what does it mean? As the term “allpass filter” suggests, the filter by itself does not change the amplitude response of a signal passing through it.

An allpass filter does not cut or boost any frequencies.

An allpass filter does cause some frequencies to be delayed a little in time, and this small time shift is also known as a phase change.

The frequency where the phase change has its greatest effect is a parameter that you can control.

By modulating the frequency of the phaser, you get the swishy phaser sound.

With a modulation rate of around 6 Hz, an effect similar to vibrato may be obtained, but only in a limited range of filter frequencies.

By adding the phaser output to the dry input using, for example, a Wet/Dry parameter, you can produced peaks and notches in the frequency response.

At frequencies where the phaser is “in phase” with the dry signal, the signal level doubles (or there is a 6 dB level increase approximately).

At frequencies where the phaser and dry signals are “out of phase,” the two signals cancel each other out and there is a notch in the frequency response. You can get a complete notch when Wet/Dry is set to 50%.

If subtraction is used instead of addition by setting Wet/Dry to -50%, then the notches become peaks and the peaks become notches.

Figure 91. Response of a typical phaser

LFOs: Most of the phaser algorithms presented here have built in low frequency oscillators (LFOs) to generate the motion of the phasers.

In the case of Manual Phaser, the phaser motion is left to you.
Feedback: Some of the phaser algorithms have feedback.

When feedback is used, it can greatly exaggerate the peaks and notches, producing a much more resonant sound.

LFO Phaser

Effects Size : 1

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LFO Phaser Presets
251 Circles
256 Fast&Slow Phaser

LFO Phaser is a simple phaser algorithm with Wet/Dry and Fdbk Level parameters.

LFOs: Two LFOs are used to control the filter frequency and the depth of the resulting notch.

You can control the depths, rates, and phases of both the LFOs.

The algorithm is stereo so the relative phases of the LFOs for the left and right channels can be set.
Filter Frequency: When setting the LFO which controls the filter frequency, you specify the center frequency around which the LFO will modulate and the depth of the LFO.

The depth specifies how many cents (hundredths of a semitone) to move the filter frequency up and down.
NotchDepth: The NotchDepth parameter provides an alternative way of combining wet and dry phaser signals to produce a notch.

In this case the parameter specifies the depth of the notch in decibels (dB).

The depth of the notch can be modulated with the notch LFO.

The notch LFO is completely independent of the frequency LFO.
The rates of the LFOs may be different.

The relative phases of the notch and frequency LFOs (N/F Phase) only has meaning when the LFOs are running at the same rate.

As with all LFO phases, it is not recommended to directly modulate the phase settings with an FXMod.

LFO Phaser Twin

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LFO Phaser Twin Presets
250 Slow Deep Phaser
254 Fast Phaser

LFO Phaser Twin produces a pair of notches separated by a spectral peak.

Figure 92. Response LFO Phaser Twin

The center frequency parameter sets the frequency of the center peak. Like LFO Phaser, the filter frequency can be modulated with a built in LFO.

The Notch/Dry parameter produces a pair of notches when set to 100%.

The output signal is dry when set to 0% and at 200%, the signal is a pure (wet) allpass response.

LFO Phaser Twin does not have Out Gain or feedback parameters.

SingleLFO Phaser

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SingleLFO Phaser Presets
252 Saucepan Phaser
257 Wawawawawawawawa
258 Slow Swish Phase

SingleLFO Phaser is identical to LFO Phaser except that the notch and frequency LFOs always run at the same rate.

Wet/Dry

0 to 100 %wet
The amount of phaser (wet) signal relative to unaffected (dry) signal as a percent.

Out Gain

Off, -79.0 to 24.0 dB
The output gain in decibels (dB) to be applied to the combined wet and dry signals.

Fdbk Level

-100 to 100 %
The phaser output can be added back to its input to increase the phaser resonance.

Negative values polarity invert the feedback signal.

LFO Rate

0.00 to 10.00 Hz
The rate of both the center frequency LFO and the notch depth LFO for the SingleLFO Phaser algorithm.

CenterFreq

8 to 25088 Hz
The nominal center frequency of the phaser filter.

The frequency LFO modulates the phaser filter centered at this frequency.

FLFO Depth

0 to 5400 ct
The depth in cents that the frequency LFO sweeps the phaser filter above and below the center frequency.

FLFO Rate

0.00 to 10.00 Hz
The rate of the center frequency LFO for the LFO Phaser algorithm.

FLFO LRPhs

0.0 to 360.0 deg
Sets the phase difference between the left and right channels of the center frequency LFO.

A setting of 180 degrees results in one being at a at the minimum frequency while the other channel is at the maximum.

NotchDepth

-79.0 to 6.0 dB
The nominal depth of the notch.

The notch depth LFO modulates the depth of the notch.

For maximum LFO depth, set NotchDepth to 0 dB and NLFO Depth to 100%.

NLFO Depth

0 to 100 %
The excursion of the notch depth LFO in units of percentage of the total range.

The depth of the LFO is limited to the range of the NotchDepth parameter such that a full 100% modulation is only possible with the NotchDepth is at the center of its range (0 dB).

NLFO Rate

0.00 to 10.00 Hz
The rate of the notch depth LFO for the LFO Phaser algorithm.

NLFO LRPhs

0.0 to 360.0 deg
The phase difference between the left and right channels of the notch depth LFO.

A setting of 180 degrees results in one channel being at highest amplitude while the other channel is at lowest amplitude.

N/F Phase

0.0 to 360.0 deg
The phase difference between the notch depth and center frequency LFOs.

For LFO Phaser, this parameter is largely meaningless unless the FMod Rate and NMod Rate are set identically.

VibratoPhaser

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VibratoPhaser Presets
253 ThunderPhaser
255 Vibrato Phaser

VibratoPhaser has a couple of interesting twists.

The bandwidth of the phaser filter can be adjusted exactly like a parametric EQ filter.

The In Width controls how the stereo input signal is routed through the effect.

At 100% In Width, left input is processed to the left output, and right to right.

Lower In Width values narrow the input stereo field until at 0%, the processing is mono.

Negative values reverse left and right channels.

The dry signal is not affected by In Width.

50% Wet/Dry will produce a full notch.
-50% Wet/Dry you get a bandpass.

Wet/Dry

-100 to 100 %wet
The amount of phaser (wet) signal relative to unaffected (dry) signal as a percent.

When set to 50% you get a complete notch.
When set to -50%, the response is a bandpass filter.

100% is a pure allpass filter (no amplitude changes, but a strong phase response).

Out Gain

Off, -79.0 to 24.0 dB
The output gain in decibels (dB) to be applied to the combined wet and dry signals.

CenterFreq

8 to 25088 Hz
The nominal center frequency of the phaser filter.

The frequency LFO modulates the phaser filter centered at this frequency.

Bandwidth

0.010 to 5.000 oct
If the phaser is set to behave as a sweeping notch or bandpass, the bandwidth of the notch or bandpass is set with Bandwidth.

This parameter works the same as for parametric EQ filter bandwidths.

LFO Depth

0 to 100 %
The depth that the frequency LFO sweeps the phaser filter above and below the center frequency as a percent.

LFO Rate

0.00 to 10.00 Hz
The rate of the LFO in Hz.

The LFO Rate may be scaled up by the Rate Scale parameter.

L/R Phase

In Width

-100 to 100 %
The width of the stereo field that passes through the stereo phaser filtering.

This parameter does not affect the dry signal.

When set to 100%, the left and right channels are processed to their respective outputs.

Smaller values narrow the stereo image until at 0% the input channels are summed to mono and set to left and right outputs.

Negative values interchange the left and right channels.

Manual Phaser

Effects Size : 1

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Manual Phaser Presets
260 Static Phaser 1	261 Static Phaser 2	262 Static Phaser 3
263 Static Phaser 4

Manual Phaser leaves the phaser motion up to you, so it has no built in LFOs.

Manual Phaser has a Notch/BP parameter

Notch at the center frequency when Wet/Dry is set to - 100%
Resonant Bandpass at the center frequency when Wet/Dry is set to 100%.
At 0% the signal is dry.

To get phaser motion, you have to change the filter center frequencies (left and right channels) yourself. The best way to do this is with an FXMod.

There are also feedback parameters for the left and right channels.

Notch/BP

0 to 200 %
The amount of notch depth or bandpass.

- At -100% there is a complete notch at the center frequency.
- At 100% the filter response is a peak at the center frequency.
- 0% is the dry unaffected signal.

Out Gain

Off, -79.0 to 24.0 dB
The output gain in decibels (dB) to be applied to the final output.

L Feedback
R Feedback

-100 to 100 %
The phaser output can be added back to its input to increase the phaser resonance (left and right).

Negative values polarity invert the feedback signal.

L Ctr Freq
R Ctr Freq

8 to 25088 Hz
The nominal center frequency of the phaser filter (left and right).

For a true phaser effect you may want to modulate these parameters by setting up FX Mods.

Rotary

Rotating speaker emulations

Hammond B3®, Hammond®, Leslie® are all Registered Trade Marks of Suzuki Musical Instrument Corporation

The rotary algorithms contain multiple effects designed for the Hammond B3 emulation (KB3 mode).

These effects may include the Hammond vibrato/chorus, amplifier distortion, cabinet emulation and rotating speaker (Leslie).

A variety of rotating speaker algorithms have been designed to deal with different circumstances. Some of the algorithms are designed to trade off features or model quality to allow the rotating speaker model to work in fewer effects units.

Figure 93. Traditional Vibrato / Chorus Settings

Vibrato / Chorus

The first effect in the chain is often the Hammond vibrato/chorus algorithm.

The vibrato/chorus has six settings which are the same as those used in the Hammond B3:

Vibrato settings (V1, V2, V3)
Chorus settings (C1, C2, C3)

Vibrato / Chorus Modelling

In VC+Dist+Rotor 4 and VC+Tube+Rotor 4, the vibrato chorus has been carefully modeled after the electromechanical vibrato/chorus in the B3.

The vibrato/chorus in the other smaller algorithms use a conventional design, which has been set to match the B3 sound as closely as possible, but does not quite have the same character as the fully modeled vibrato/chorus.

Rotating Speaker

The final section of each of the rotary algorithms is the rotating speaker routine.

The various algorithms may trade off some features of the rotating speaker routine and the tradeoffs will be discussed for each algorithm separately.

How a Rotating Speaker works: The rotating speaker has separately controllable tweeter and woofer drivers.

The signal is split into high and low frequency bands and the two bands are run through separate rotors.

The upper and lower rotors each have a one or two virtual microphones which can be positioned at varying positions (angles) around the rotors.

An angle of 0° is loosely defined as the front of the speaker.

You can also control the levels and left-right panning of each virtual microphone.

Figure 94. Rotating speaker with virtual microphones

Cabinet Filter

The signal is then passed through a final cabinet filter to simulate the band-limiting effect of the speaker cabinet.

The cabinet filter is often a simple lowpass filter.

Cross-over Frequency: For the rotating speakers, you can control the cross-over frequency of the high and low frequency bands (the frequency where the high and low frequencies get separated).

The rotating speakers for the high and low frequencies have their own controls.

For both, the rotation speed, the effective driver size and tremolo can be set.

The effective driver size is the radius of the path followed by the speaker relative to its center of rotation.

Doppler Shift: This parameter is used to calculate the resulting Doppler shift of the moving speaker.

Doppler shift is the pitch shift that occurs when a sound source moves toward or away from you the listener.

In a rotating speaker, the Doppler shift will sound like vibrato.

Acoustic Shadowing: As well as Doppler shift, there will be some acoustic shadowing as the speaker is alternately pointed away from you and toward you.

The shadowing is simulated with a tremolo over which you can control the tremolo depth and “width.”

The high frequency driver (rotating horn) will have a narrower acoustic beam width (dispersion) than the low frequency driver, and the widths of both may be adjusted.

Negative microphone angles take a longer time to respond to tremolo changes than positive microphone angles.

It can take up to one full speaker rotation before you hear changes to tremolo when parameter values are changed.

Figure 95. Acoustic Beams

Resonance

You can control resonant modes within the rotating speaker cabinet with the Lo and Hi Resonate parameters.

For a realistic rotating speaker, the resonance level and delay excursion should be set quite low.

High levels will give wild pitch shifting.

Rotation Speeds

The rotating speaker algorithms give you a great deal of control over the rotation speeds.

The direction of rotation (clockwise or counter-clockwise) can be set.

A rotating speaker generally has two rotation speeds: fast or slow.

There is also a Brake parameter to stop the speakers from rotating.

You can set the fast and slow rotation rates in Hz for both the high and low frequency speakers.

When you switch between the fast speed and slow speed, the rotating speaker takes time to ramp the speed up or down, just like a real rotating speaker.

The time to ramp from slow to fast can be set and a different time can be set for fast to slow.

The low frequency speaker can work in three modes:

Normal
NoAccel
Stopped

Acceleration Curve

Finally, the shape of the acceleration itself can be controlled.

The acceleration curve parameter produces a constant acceleration (linear change of speed) when set to 0%.

At the most negative settings, the speed will overshoot the fast rate before settling down (acceleration goes negative when approaching the fast speed).

Positive settings, acceleration is slow at slow speed and speeds up for fast speeds.
Negative settings, acceleration is fast at slow speeds, then slows down at fast speeds.

Figure 96. Effect of acceleration curve on speed

Rotary Quick Reference Chart
	Vibrato Chorus	Distortion	Mic Positions	Cabinet High/Low Filter	Acoustic Beam Width	Effects Units Used
VibChor+Rotor 2	Simplified	No	4	Low	Yes	2
Distort + Rotary	No	Yes	-	Both	No	2
VC+Dist+HiLoRotr	Yes	Simple	2 *No High/Low	-	Yes *Fixed Low	2
VC+Dist+1Rotor 2	Yes	Yes	2 *No High/Low	Low	Yes *Paired	2
VC+Dist+HiLoRot2	Yes	Yes	2 *No High/Low	-	No	2
Rotor 1	No	No	2 *No High/Low	Low	Yes *Paired	1
VC+Dist+Rotor 4	Yes	Yes	4	Low	Yes	4
KB3a	Yes	No	-	Both	No	4
KB3b	No	Yes	4	-	Yes	4

VibChor+Rotor 2

Effects Size : 2

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VibChor+Rotor 2 Presets
280 CleanRotors fast	282 CleanRotors f C1	284 CleanRotors f Hi
281 CleanRotors slow	283 CleanRotors f V1	285 CleanRotors s Hi

Algorithm VibChor+Rotor 2 is similar in design to VC+Dist+Rotor 4 but lacks the distortion section and uses a simplified vibrato/chorus model.

Distort + Rotary

Effects Size : 2

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Distort + Rotary Presets
286 SlightDstRotor f
287 SlightDstRotor s

Algorithm Distort + Rotary models an amplifier distortion followed by a rotating speaker.

The rotating speaker has separately controllable tweeter and woofer drivers.

The algorithm has three main sections.

First, the input stereo signal is summed to mono and may be distorted by a tube amplifier simulation (see Mono Distortion for details).
The signal is then passed into the rotator section where it is split into high and low frequency bands and the two bands are run through separate rotators.

The two bands are recombined and measured at two positions, spaced by a controllable relative angle (microphone simulation) to obtain a stereo signal again.
Finally the signal is passed through a speaker cabinet simulation.

Figure 97. Distort + Rotary

VC+Dist+HiLoRotr

Effects Size : 2

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VC+Dist+HiLoRotr Presets
288 DirtyRotors fast
289 DirtyRotors slow

Algorithm VC+Dist+HiLoRotr gives you a model of the Hammond vibrato/chorus, distortion and the band splitting for high and low frequency drivers.

To pack all this into a 2 unit algorithm, a few sacrifices had to be made to the list of parameters for the rotating speaker model.

So what’s missing?

The resonance controls for the low frequency driver are gone.
There is no control of the acoustic beam width for the low driver.
The microphone panning is gone and there is a single microphone level control for the A and B microphones.
The distortion used is a smaller version of PolyDistort + EQ.

Even with fewer features, this algorithm gives a convincing Leslie effect while allowing space for more algorithms.

Figure 98. VC+Dist+HiLoRotr

VC+Dist+1Rotor 2

Effects Size : 2

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VC+Dist+1Rotor 2 Presets
290 MoreDistRotor f
291 MoreDistRotor s

Algorithm VC+Dist+1Rotor 2 models a single rotating speaker in a 2 unit algorithm.

In other respects the algorithm is quite full featured and includes:

Hammond vibrato/chorus model
distortion
full control of the rotating speaker model (speed, size for Doppler shift, tremolo, acoustic beam width, cabinet resonance)
microphone positions
panning

You get all the features, but only for one driver.

The signal does not get split into a high band and low band and passed through separate drivers.

Figure 99. VC+Dist+1Rotor 2

VC+Dist+HiLoRotr2

Effects Size : 2

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VC+Dist+HiLoRotr2 Presets
292 HeavyDistRotor f
293 HeavyDistRotor s

Algorithm VC+Dist+HiLoRot2 makes different tradeoffs than Algorithm VC+Dist+HiLoRotr.

The distortion is the same as used in Algorithm Mono Distortion.

This distortion uses more processor resources than the PolyDistort + EQ, so VC+Dist+HiLoRot2 does not include the acoustic beam width control for either the high or low frequency drivers.

The signal flow is the same as VC+Dist+HiLoRotr.

Figure 100. VC+Dist+HiLoRotr2

Rotor 1

Effects Size : 1

Rotating speaker model on a budget

Rotor 1 Presets
294 Res Rotor1 fast
295 Res Rotor1 slow

Rotor 1 most attractive feature is its compact small size.

However, a few things had to be scaled back:

No vibrato/chorus model
No distortion control
There is only a single rotating driver rather than a pair for high and low frequency bands.

Aside from these omissions, the rotating speaker model is quite full featured.

It includes full control of the rotating speaker including speed, size for Doppler shift, tremolo, acoustic beam width, cabinet lowpass filter and resonance and full microphone control for two microphone positions.

Figure 101. Rotor 1

VC+Dist+Rotor 4

Effects Size : 4

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VC+Dist+Rotor 4 Presets
296 FullRotors4 fast
297 FullRotors4 slow

VC+Dist+Rotor 4 begins with the full vibrato/chorus model and is followed by amplifier distortion (see Mono Distortion).

The distortion is followed by the rotating speaker model and a cabinet lowpass filter.

KB3a

Previously known as "Big KB3" in K25/K26/KSP8 models
Effects Size : 4

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KB3a Presets
298 VibChorStortCab
580 VibChorCabT

KB3b

Effects Size : 3

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KB3b Presets
299 Hi Lo Roto KB3

In/Out

In or Out
When set to In, the algorithm is active.
When set to Out the algorithm is bypassed.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

For distortion, it is often necessary to turn the output gain down as the distortion drive is turned up.

VibChInOut

In or Out
When set to In the vibrato/chorus is active.
When set to Out the vibrato/chorus is bypassed.

Vib/Chor

V1, V2, V3, C1, C2, C3.
This control sets the Hammond B3 vibrato/chorus.

There are six settings for this effect: three vibratos V1, V2, and V3, and three choruses C1, C2, and C3.

Roto InOut

In or Out
When set to In the rotary speaker is active.
When set to Out the rotary speaker is bypassed.

Dist Curve

0.0 to 96.0 dB
Controls the curvature of the distortion.

0% is no curvature (no distortion at all).
At 100%, the curve bends over smoothly and becomes perfectly flat right before it goes into clipping.

DistWarmth

8 to 25088 Hz
A lowpass filter in the distortion control path.

This filter may be used to reduce some of the harshness of some distortion settings without reducing the bandwidth of the signal.

DistLPFreq

8 to 25088 Hz
Controls one-pole lowpass filters in the PolyDistort + EQ (in VC+Dist+HiLoRotr).

Without the lowpass filters, the sound tends to be too bright and raspy.

With less distortion drive, less filtering is needed.

If you turn off the distortion curve (set to 0%), you should turn off the lowpass filter by setting it to the highest frequency.

Cabinet LP

8 to 25088 Hz
A lowpass filter to simulate the band-limiting of a speaker cabinet.

The filter controls the upper frequency limit of the output.

Xover

8 to 25088 Hz
The frequency at which high and low frequency bands are split and sent to separate rotating drivers.

Lo Xover

8 to 25088 Hz
Separate high and low crossover frequency controls.

Lo Xover controls a lowpass filter.

Hi Xover

8 to 25088 Hz
Separate high and low crossover frequency controls.

Hi Xover controls a highpass filter.

Speed

Slow or Fast
Sets the rotating speakers to run at either the slow rate or the fast rate.

Brake

On or Off
When set to On, the rotating speakers will slow to a halt.

Lo Mode

Normal, NoAccel, Stopped

Normal setting, you will have full control of the low frequency speaker with the Speed parameter.
NoAccel setting will hold the low frequency speaker at the slow speed, and the Speed parameter will have no effect on its speed, though Brake will still work.
Stopped position, the low frequency speaker will not spin at all.

Dist Drive
Tube Drive

Lo Slow
Hi Slow

0.00 to 2.00 Hz
The rotation rate in hertz (Hz) of the speaker when Speed is set to Slow.

Lo Fast
Hi Fast

3.00 to 10.00 Hz
The rotation rate in hertz (Hz) of the speaker when Speed is set to Fast.

LoSlow>Fst
HiSlow>Fst

0.10 to 10.00 s
The time for the speaker to accelerate from slow speed to fast speed.

LoFst>Slow
HiFst>Slow

0.10 to 10.00 s
The time for the speaker to decelerate from fast speed to slow speed.

Lo Gain

Off, -79.0 to 24.0 dB
The gain or amplitude of the signal passing through the rotating woofer (low frequency driver).

Lo Size

0 to 250 mm
The effective size (radius of rotation) of the rotating woofer in millimeters.

Affects the amount of Doppler shift or vibrato of the low frequency signal.

Lo Trem

0 to 100%
Controls the depth of tremolo of the low frequency signal.

Expressed as a percentage of full scale tremolo.

Lo Beam W

45.0 to 360.0 deg
The rotating speaker effect attempts to model a rotating woofer for the low frequency driver.

The acoustic radiation pattern of a woofer tends to range from omnidirectional (radiates in directions in equal amounts) to a wide beam.

You may adjust the beam width from 45° to 360°.
If you imagine looking down on the rotating speaker, the beam angle is the angle between the -6 dB levels of the beam.

At 360°, the woofer is omnidirectional.

Hi Gain

Off, -79.0 to 24.0 dB
The gain or amplitude of the signal passing through the rotating tweeter (high frequency driver).

Hi Size

0 to 250 mm
The effective size (radius of rotation) of the rotating tweeter in millimeters.

Affects the amount of Doppler shift or vibrato of the high frequency signal.

Hi Trem

0 to 100%
Controls the depth of tremolo of the high frequency signal.

Expressed as a percentage of full scale tremolo.

Hi Beam W

45.0 to 360.0 deg
The rotating speaker effect attempts to model a rotating horn for the high frequency driver.

The acoustic radiation pattern of a horn tends to be a narrow beam.

You may adjust the beam width from 45° to 360°.

If you imagine looking down on the rotating speaker, the beam angle is the angle between the -6 dB levels of the beam.

At 360°, the horn is omnidirectional (radiates in all directions equally).

Lo HP

8 to 25088 Hz
A highpass filter to simulate the band-limiting of a speaker cabinet.

The filter controls the lower frequency limit of the output.

Hi LP

8 to 25088 Hz
A lowpass filter to simulate the band-limiting of a speaker cabinet.

The filter controls the upper frequency limit of the output.

Mic Pos

??
The angle of the virtual microphones in degrees from the “front” of the rotating speaker.

This parameter is not well suited to modulation because adjustments to it result in large sample skips (audible as clicks when signal is passing through the effect).

There are four of these parameters to include two pairs (A and B) for high and low frequency drivers.

Mic A Lvl
Mic B Lvl
LoMic Lvls
HiMic Lvls

0 to 100%
The level of the virtual microphone signal being sent to the output.

There are four of these parameters to include two pairs (A and B) for high and low frequency drivers.

Mic A Pos
Mic B Pos
LoMicA Pos
LoMicB Pos
HiMicA Pos
HiMicB Pos

-180.0 to 180.0 deg
Left-right panning of the virtual microphone signals.

A setting of -100% is panned fully left, and 100% is panned fully right.

There are four of these parameters to include two pairs (A and B) for high and low frequency drivers.

LoAccelCrv
HiAccelCrv

-100 to 100%
The shape of the acceleration curve for the speaker.

0% is a constant acceleration.
Positive values cause the speaker to speed up slowly at first then quickly reach the fast rate.
Negative values cause a quick initial speed-up then slowly settle in to the fast speed—overshoot is possible.

LoSpinDir
HiSpinDir

CW or CCW
The direction of rotation of the speaker.

The choice is clockwise (CW) or counter-clockwise (CCW).

LoResonate

0 to 100%
A simulation of cabinet resonant modes express as a percentage.

For realism, you should use very low settings.

This is for the low frequency signal path.

Lo Res Dly

10 to 2550 samp
The number of samples of delay in the resonator circuit in addition to the rotation excursion delay.

This is for the low frequency signal path.

LoResXcurs

0 to 510 samp
The number of samples of delay to sweep through the resonator at the rotation rate of the rotating speaker.

This is for the low frequency signal path.

HiResonate

0 to 100%
A simulation of cabinet resonant modes express as a percentage.

For realism, you should use very low settings.

This is for the high frequency signal path.

Hi Res Dly

10 to 2550 samp
The number of samples of delay in the resonator circuit in addition to the rotation excursion delay.

This is for the high frequency signal path.

HiResXcurs

0 to 510 samp
The number of samples of delay to sweep through the resonator at the rotation rate of the rotating speaker.

This is for the high frequency signal path.

ResH/LPhs

0.0 to 360.0 deg
This parameter sets the relative phases of the high and low resonators.

The angle value in degrees is somewhat arbitrary and you can expect the effect of this parameter to be rather subtle.

Chorus

Chorusing is an effect that gives the illusion of multiple voices playing in unison.

The effect is achieved by detuning copies of the original signal and summing the detuned copies back with the original. Low-frequency oscillators (LFOs) are used modulate the positions of output taps from a delay line. The delay line tap modulation causes the pitch of the signal to shift up and down, producing the required detuning.

The choruses are available as stereo or dual mono.

Figure 102. Chorus 2 - Block diagram of left channel

The stereo choruses have the parameters for the left and right channels ganged.

Chorus 2 is multi-tapped delay (3 taps) based chorus effect with cross-coupling and individual output tap panning.

Figure 103. Dual Chorus 2 - Block diagram of left channel

The dual mono choruses are like the stereo choruses but have separate left and right controls. Dual mono choruses also allow you to pan the delay taps between left or right outputs. The left and right channels pass through their own chorus blocks and there may be cross-coupling between the channels.

Figure 104. Chorus 1 - Block diagram of left channel

Chorus 1 has one delay tap.

Figure 105. Dual Chorus 1 - Block diagram of left channel

For Chorus 2 and Dual Chorus 2, each channel has three moving taps which are summed, while Chorus 1 and Dual Chorus 1 have one moving tap for both channels. For the dual mono choruses you can pan the taps to left or right. The summed taps (or the single tap of Chorus 1) is used for the wet output signal. The summed tap outputs, weighted by their level controls, are used for feedback back to the delay line input. The input and feedback signals go through a one pole lowpass filter (HF Damping) before going entering the delay line.

The Wet/Dry control is an equal power crossfade.

The Output Gain parameters affects both wet and dry signals.

For each of the LFO tapped delay lines, you may set the tap levels, the left/right pan position, delays of the modulating delay lines, the rates of the LFO cycles, and the maximum depths of the pitch detuning. The LFOs detune the pitch of signal copies above and below the original pitch. The depth units are in cents, and there are 100 cents in a semitone.

In the stereo Chorus 1 and Chorus 2, the relative phases of the LFOs modulating the left and right channels may be adjusted. The settings of the LFO rates and the LFO depths determine how far the LFOs will sweep across their delay lines from the shortest delays to the longest delays (the LFO excursions). The Tap delays specify the average amount of delay of the LFO modulated delay lines, or in other words the delay to the center of the LFO excursion. The center of LFO excursion can not move smoothly. Changing the center of LFO excursion creates discontinuities in the tapped signal. It is therefore a good idea to adjust the Tap Dly parameter to a reasonable setting (one which gives enough delay for the maximum LFO excursion), then leave it.

Modulating Tap Dly will produce unwanted zipper noise. If you increase the LFO modulation depth or reduce the LFO rate to a point where the LFO excursion exceeds the specified Tap Dly, the center of LFO excursion will be moved up, and again cause signal discontinuities. However, if enough Tap Dly is specified, Depth and Rate will be modulated smoothly.

As the LFOs sweep across the delay lines, the signal will change pitch. The pitch will change with a triangular envelope (rise-fall-rise-fall) or with a trapezoidal envelope (rise-hold-fall-hold). You can choose the pitch envelope with the Pitch Env parameter. Unfortunately rate and depth cannot be smoothly modulated when Pitch Env is set to Trapzoid.

Chorus 1

Effects Size : 1

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Chorus 1 Presets
210 Sm Stereo Chorus

Chorus 2

Effects Size : 2

Chorus 2 Presets
211 Lg Stereo Chorus
213 Dense Gtr Chorus

Dual Chorus 1

Effects Size : 1

Dual Chorus 1 Presets
200 Basic Chorus	203 Ordinary Chorus	206 Everyday Chorus
201 Smooth Chorus	204 SlowSpinChorus	215 Bass Chorus
202 Chorusier	205 Chorus Morris

Dual Chorus 2

Effects Size : 2

Dual Chorus 2 Presets
207 Thick Chorus	216 Stereo Chorus	208 Soft Chorus
218 Wide Chorus	209 Rock Chorus

Wet/Dry

-100 to 100%wet
The relative amount of input (dry) signal and chorus (wet) signal that is to appear in the final effect output mix.

When set to 0%, the output is taken only from the input.
When set to 100%, the output is all wet.

Negative values polarity invert the wet signal.

Out Gain

Off, -79.0 to 24.0 dB
The overall gain or amplitude at the output of the effect.

Fdbk Lvl

-100 to 100%
The level of the feedback signal into the delay line.

The feedback signal is taken from the LFO1 delay tap.
Negative values polarity invert the feedback signal.

Xcouple

0 to 100%
Controls how much of the left channel input and feedback signals are sent to the right channel delay line and vice versa.

At 50%, equal amounts from both channels are sent to both delay lines.
At 100%, the left feeds the right delay and vice versa.

HF Damping

16 Hz to 25088 Hz
The amount of high frequency content of the signal that is sent into the delay lines.

This control determines the cutoff frequency of the one-pole (-6dB/octave) lowpass filter.

Pitch Env

Triangle or Trapzoid
The pitch of the chorus modulation can be made to follow a triangular (Triangle) envelope (rise-fall-rise-fall) or a trapezoidal (Trapzoid) envelope (rise-hold-fall-hold).

Tap Lvl

-100 to 100 %
The pitch of the chorus modulation can be made to follow a triangular (Triangle) envelope (rise-fall-rise-fall) or a trapezoidal (Trapzoid) envelope (rise-hold-fall- hold).

Levels of the LFO modulated delay taps.

Negative values polarity invert the signal.

Setting any tap level to 0% effectively turns off the delay tap. Since these controls allow the full input level to pass through all the delay taps, a 100% setting on all the summed taps will significantly boost the wet signal relative to dry. A 50% setting may be more reasonable.

Tap Pan

-100 to 100 %
The left or right output panning of the delay taps.

The range is -100% for fully left to 100% for fully right.

Setting the pan to 0% sends equal amounts to both left and right channels for center or mono panning.
Dual Chorus 1 and Dual Chorus 2 only.

LFO Rate

0.01 to 10.00 Hz
Used to set the speeds of modulation of the delay lines.

Low rates increase LFO excursion (see LFO Dpth).
If Pitch Env is set to Trapzoid, you will be unable to put the rate on an FXMod or otherwise change the rate without introducing discontinuities (glitches or zippering) to your output signal.

The triangular Triangle Pitch Env setting does allow smooth rate modulation, provided you’ve specified enough delay.

LFO Depth

0.0 to 50.0 ct
The maximum depths of detuning of the LFO modulated delay lines.

The depth controls range from 0 to 50 cents.

(There are 100 cents in a semitone.)

If you do not have enough delay specified with Tap Dly to get the depth you’ve dialed up, then Tap Dly will be forced to increase (with signal discontinuities if signal is present).

The LFOs move a tap back and forth across the delay lines to shift the pitch of the tapped signal.

The maximum distance the taps get moved from the center position of the LFO is called the LFO excursion.

Excursion is calculated from both the LFO depth and rate settings.

Large depths and low rates produce large excursions.

If Pitch Env is set to Trapzoid, you will be unable to put the depth on an FXMod or otherwise change the depth without introducing discontinuities (glitches or zippering) to your output signal.

The Triangle Pitch Env setting does allow smooth depth modulation, provided you’ve specified enough delay.

Tap Dly

0.0 to 1000.0 ms
The average delay length, or the delay to the center of the LFO sweep.

If the delay is shorter than the LFO excursion, then the Tap Dly will be forced to a longer length equal to the amount of required excursion (the parameter display will not change though).

Changing this parameter while signal is present will cause signal discontinuities.

It’s best to set and forget this one.

Set it long enough so that there are no discontinuities with the largest Depth and lowest Rates that you will be using.

L/R Phase

0.0 to 360.0 deg
(Or LFOn LRPhs) In the stereo Chorus 1 and Chorus 2, the relative phases of the LFOs for the left and right channels may be adjusted.

Kurzweil String Resonance (KSR)

Introducing Sympathetic Vibration

When you play a chord on an acoustic piano and hold down the keys while the notes decay, the dampers on the corresponding strings remain up, and you hear a particular set of harmonics that evolve from the undamped strings.

You don’t hear any significant harmonics from the other strings.

If you play the same chord and hold it with the sustain pedal, you’ll hear a much different, richer set of harmonics as the notes decay.

That’s because all the strings are undamped, and each string gradually begins to vibrate at its resonant frequency, in response to the vibrations of the strings struck by the hammers.

This phenomenon is called sympathetic vibration, and is an important component of the sound of an acoustic piano.

The most noticeable of these sympathetic vibrations come from the strings whose fundamental pitches match the harmonics of the strings that were struck by the hammers.

To create a more realistic acoustic piano sound, Kurzweil sound engineers have developed special effects settings that imitate sympathetic vibrations called Kurzweil String Resonance (KSR).

Kurzweil String Resonance (KSR)

Kurzweil String Resonance (KSR) is the new piano string resonance emulation.

The String Resonance parameter works in conjunction with the FX preset 600 String Resonance to emulate the sound of strings resonating in an acoustic piano.

When combined, these two components create KSR (Kurzweil String Resonance).

When a layer has the String Resonance parameter set to On, the FX preset 600 String Resonance monitors which keys are being held on that layer and uses them to tune the algorithm in the FX preset.

Any audio that passes though the FX preset while these keys are held will cause emulated strings to resonate based on this tuning.

String Resonance

Effects Size : 4

Sympathetic vibration to emulate an acoustic piano

String Resonance Presets
600 String Resonance

Combination Effects "+"

Serial or Parallel 2/3 unit combination effects

These algorithms are built from a combination of 2 or 3 algorithms allowing serial or parallel routings between any 2 effects.

Signal Routing for 2 effects

Chorus+Delay
Chorus+4Tap
Flange+Delay
Flange+4Tap

The algorithms with 2 effects can be arranged in series or parallel.

Effect A and B are respectively designated as the first and second listed effects in the algorithm name.

The output of effect A is wired to the input of effect B, and the input into effect B is a mix of effect A and the algorithm input dry signal.

The effect B input mix is controlled by a parameter A/Dry>B. where A is effect A, and B is effect B.

For example, in Chorus+Delay, the parameter name is Ch/Dry>Dly.

The value functions much like a wet/dry mix where 0% means that only the algorithm input dry signal is fed into effect B (putting the effects in parallel), and 100% means only the output of effect A is fed into effect B (putting the effects in series).

Figure 106. An example of routing using Chorus+4Tap

Mix: Both effect A and B outputs are mixed at the algorithm output to become the wet signal.

These mix levels are controlled with the 2 parameters that begin with “Mix.”

These allow only one or both effect outputs to be heard.

Negative mix amounts polarity invert the signal which can change the character of each effect when mixed together or with the dry signal.
Wet/Dry: The Wet/Dry parameter adjusts the balance between the sum of both effects determined by the Mix parameters, and the input dry signal.

Negative Wet/Dry values polarity invert the summed wet signal relative to dry.

Two-Effect Routing
Wet/Dry	-100 to 100%	Out Gain	Off; -79.0 to 24.0 dB
Mix Effect	-100 to 100%
Mix Effect	-100 to 100%
		A/Dry→B	0 to 100%

Mix Effect

Adjusts the amount of each effect that is mixed together as the algorithm wet signal.

Negative values polarity-invert that particular signal.

A/Dry→B

This parameter controls how much of the A effect is mixed with dry and fed into the B effect.

A and B are designated in the algorithm name.

This control functions like a wet/dry mix

0% is completely dry
100% is effect A only

Signal Routing for 3 effects

Chor+Dly+Reverb
Flan+Dly+Reverb
Pitcher+Chor+Dly
Pitcher+Flan+Dly

The algorithms listed above with three effects allow serial or parallel routing between any three effects.

Effect A is wired to the input of effect B and C, and effect B is wired into effect C.

The input of effect B is a mix between effect A and the algorithm dry input.

The input into effect C is a three-way mix between effect A, effect B, and the dry signal.

As in the two-effect routing, the input of effect B is controlled by a parameter A/Dry>B.

For example, in Chor+Dly+Reverb, the parameter name is Ch/Dry>Dly.

The input into effect C is controlled by two parameters named A/B →* and */Dry→C where A, B, and C correspond to the names of effects A, B, and C.

The first parameter mixes effect A and B into a temporary buffer represented by the symbol "*".

The second parameter mixes this temporary buffer "*" with the dry signal to be fed into effect C.

These mixing controls function similarly to Wet/Dry parameters.

A setting of 0% only mixes the effect to the right of the "/" in the parameter name, while 100% only mixes the effect to the left of the "/".

Negative values polarity-invert the numerator’s signal.

Figure 107. An example of routing using Chorus+Delay+Reverb

Mix: Effects A, B, and C outputs are mixed at the algorithm output to become the wet signal.

Separate mixing levels are provided for left and right channels, and are named L Mix or R Mix.

Negative mix amounts polarity-invert the signal which can change the character of each effect when mixed together or with the dry signal.
Wet/Dry: The Wet/Dry parameter adjusts the balance between the sum of all effects determined by the Mix parameters, and the input dry signal.

Negative Wet/Dry values polarity-invert the summed wet signal relative to dry.

Three-Effect Routing
Wet/Dry	-100 to 100%	Out Gain	Off; -79.0 to 24.0 dB
L Mix Effect A	-100 to 100%	R Mix Effect A	-100 to 100%
L Mix Effect B	-100 to 100%	R Mix Effect B	-100 to 100%
L Mix Effect C	-100 to 100%	R Mix Effect C	-100 to 100%

A/Dry>B

-100 to 100%

A/B →*

-100 to 100%

*/Dry→C

-100 to 100%

L Mix Effect, R Mix Effect

Adjusts the amount of each effect that is mixed together as the algorithm wet signal.

Separate left and right controls are provided.

Negative values polarity-invert that particular signal.

A/Dry>B

This parameter controls how much of the A effect is mixed with dry and fed into the B effect.

A and B are designated in the algorithm name.

This control functions like a wet/dry mix:

0% is completely dry
100% is effect A only

A/B →*

This parameter is first of two parameters that control what is fed into effect C.

This adjusts how much of the effect A is mixed with effect B, the result of which is represented as the symbol "*".

Negative values polarity-invert the A effect.

0% is completely B effect
100% is completely A effect

*/Dry→C

This parameter is the second of two parameters that control whet is fed into effect C.

This adjusts how much of the "*" signal (sum of effects A and B determined by A/B →*) is mixed with the dry signal and fed into effect C.

0% is completely dry signal
100% is completely "*" signal.

Choruses

A general description of chorus functionality can be found in the Choruses section.

The choruses are basic 1-tap dual choruses.
Separate LFO controls are provided for each channel.
Slight variations between algorithms may exist.
Some algorithms offer separate left and right feedback controls,while some offer only one for both channels.
Also, cross-coupling and high frequency damping may be
offered in some and not in others.
Parameters associated with chorus control begin with “Ch” in the parameter name.

Parameters for Choruses
Ch PtchEnv	Triangle or Trapzoid	Ch LRPhase	0.0 to 360.0 deg
Ch Rate L	0.01 to 10.00 Hz	Ch Rate R	0.01 to 10.00 Hz
Ch Depth L	0.0 to 100 ct	Ch Depth R	0.0 to 100 ct
Ch Delay L	0.0 to 360.0 ms	Ch Delay R	0.0 to 360.0 ms
Ch Fdbk	-100 to 100 %
Ch Xcouple	0 to 100 %	Ch HF Damp	8 to 25088 Hz

Flangers

A general description of chorus functionality can be found in the Flangers section.

The flangers are basic 1-tap dual flangers.
Separate LFO controls are provided for each channel.
Slight variations between algorithms may exist.
Some algorithms offer separate left and right feedback controls, while some offer only one for both channels.
Also, cross-coupling and high frequency damping may be
offered in some and not in others.
Parameters associated with chorus control begin with “Fl” in the parameter name.
In addition to the LFO delay taps, some flangers may offer a static delay tap for creating through-zero flange effects. The maximum delay time for this tap is 230ms and is controlled by the Fl StatDly parameter. Its level is controlled by the Fl StatLvl parameter.

Parameters for Flangers
Fl Tempo	System; 1 to 255 BPM	Fl HF Damp	8 to 25088 Hz
Fl Rate	0.01 to 10.00 Hz
Fl Xcurs L	0 to 230 ms	Fl Xcurs R	0 to 230 ms
Fl Delay L	0 to 230 ms	Fl Delay R	0 to 230 ms
Fl Fdbk L	-100 to 100 %	Fl Fdbk R	-100 to 100 %
Fl Phase L	0 to 360 deg	Fl Phase R	0 to 360 deg

Fl HF Damp

8 to 25088 Hz

Fl Xcouple

0 to 100 %

Fl StatDly

0 to 230 ms

Fl StatLvl

-100 to 100 %

Delays

There is a limited amount of delay memory available (usually 1.5 seconds for these delays), selecting slow tempos and/or long delay lengths may cause you to run out of delay memory.

At this point, each delay will pin at its maximum possible time.

Because of this, when you slow down the tempo, you may find the delays lose their sync.

Delay (Dly) is a basic tempo-based dual channel delay with added functionality, including image shifting, and high frequency damping.
Separate left and right controls are generally provided for delay time and feedback, and laser controls.
Parameters associated with delay in a combination algorithm begin with Dly.
The delay length for each channel is determined by Dly Tempo, expressed in beats per minute (BPM), and the delay length (Dly Time L and Dly Time R) of each channel is expressed in beats (bts).
The tempo alters both channel delay lengths together. With the tempo in beats per minute and delay lengths in beats, you can calculate the length of a delay in seconds as \(beats/tempo*60 (sec/min)\)
Delay regeneration is controlled by Dly Fdbk.
Separate left and right feedback control is generally
provided, but due to resource allocation, some delays in combinations may have a single control for both channels.
Dly FBImag and Dly HFDamp are just like the HFDamp and Image parameters found in other algorithms. Not all delays in combination algorithms will have both of these parameters due to resource allocation.

Parameters for Delays
Dly Time L	0 to 32 bts	Dly Time R	0 to 32 bts
Dly Fdbk L	-100 to 100 %	Dly Fdbk R	-100 to 100 %
Dly HFDamp	0 to 32 bts	Dly Imag	-100 to 100 %

Combination 4-Tap

Combination 4-Tap is a tempo based 4 tap delay with feedback used in combination algorithms.
Parameters associated with the 4 tap effect start with “4T.”
The control over the feedback tap and individual output taps is essentially the same as the 4-Tap Delay BPM algorithm, with the exception that the delay times will pin at the maximum delay time instead of automatically cutting their times in half.

Parameters for combination 4-Tap
4T Tempo	System; 1 to 255 BPM
4T LoopLen	0 to 8 bts
4T FB Lvl	-100 to 100 %

Tap1 Delay

0 to 8 bts

Tap3 Delay

0 to 8 bts

Tap1 Level

-100 to 100 %

Tap3 Level

-100 to 100 %

Tap1 Bal

-100 to 100 %

Tap3 Bal

-100 to 100 %

Tap2 Delay

0 to 8 bts

Tap4 Delay

0 to 8 bts

Tap2 Level

-100 to 100 %

Tap4 Level

-100 to 100 %

Tap2 Bal

-100 to 100 %

Tap4 Bal

-100 to 100 %

Reverbs

The reverbs offered in these combination effects is MiniVerb.

Information about it can be found in the MiniVerb documentation.

Parameters associated with this reverb begin with Rv.

Parameters for Reverb (MiniVerb)
Rv Type	Hall1
Rv Time	0.5 to 30.0 s; Inf
Rv DiffScl	0.00 to 2.00x	Rv Density	0.00 to 4.00x
Rv SizeScl	0.00 to 4.00x	Rv HF Damp	8 to 25088 Hz
Rv PreDlyL	0 to 620 ms	Rv PreDlyR	0 to 620 ms

Chorus+Delay

Effects Size : 1

Combination Chorus and tempo Delay

Chorus+Delay Presets
217 Chorus Fastback	402 Chorus & Echo	428 SpeeChorusDeep
400 BasicChorusDelay	406 Doubler & Echo
401 Chorus PanDelay	418 FastChorusDouble

Chorus+4Tap

Effects Size : 1

Combination Chorus and 4 tap tempo Delay

Chorus+4Tap Presets
416 CrackedPorcelain

Chor+Dly+Reverb

Effects Size : 2

Combination Chorus, tempo Delay and reverb

Chor+Dly+Reverb Presets
224 CDR for Lead Gtr	417 Rich Delay	444 Chor-Delay Booth
403 CDR Lead	422 DeepChorDlyHall	445 Chor Tin Room
404 CDR Lead 2	423 ClassicEP ChorRm	446 Boiler Plate
407 Chorus Booth	437 Into The Abyss	447 O.T.T. Pad
408 ChorusSmallRoom	438 BroadRevSlapback	448 TheChorusCloset
414 Flam Dly Bckgrnd	439 Carlsbad Cavern
415 CDHall Halo	443 Cut it out!! CDR

Pitcher+Chor+Dly

Effects Size : 2

Combination Pitcher, Chorus and tempo Delay

Pitcher+Chor+Dly Presets
379 Pitcher+Chor+Dly

Flange+Delay

Effects Size : 1

Combination Flanger and tempo Delay

Flange+Delay Presets
450 Flange + Delay	452 Slapback Flange	486 FD Lead Madness
451 ThroatyFlangeDly	471 Flange Delay

Flange+4Tap

Effects Size : 1

Combination Flange and 4-tap tempo Delay

Flange+4Tap Presets
459 Flange + 4Tap	472 Mecha-Godzilla	475 Drum&Bass FlgDly

Flan+Dly+Reverb

Effects Size : 2

Combination Flange, tempo Delay and Reverb

Flan+Dly+Reverb Presets
456 Flange Dly 3-D	466 Flange Room	480 SyntheticRmFlg
457 Fl Dl Large Hall	470 Flange-Dly Hall	487 Brite Rippleverb
460 FlangeDelayHall	473 Industro-Flange	488 Rotary Club
461 SloFlangeDlyRoom	474 Panning FDRoom	489 Flangey Hall
463 FlangeDlyBigHall	479 F-D Hall

Pitcher+Flan+Dly

Effects Size : 2

Combination Pitcher, Flange and tempo Delay

Pitcher+Flan+Dly Presets
380 Pitcher+Flng+Dly

More Combination Effects "+"

Combination effect algorithms using time/frequency units instead of tempo

The algorithms listed here are identical in most respects to combination effects elsewhere documented.

The difference is that they do not use tempo features for setting delay line lengths or LFO rates.

Instead, delay line lengths are set in units of milliseconds (ms) for time, and LFO rates are set in units of Hertz (Hz) for frequency.

The difference for algorithms with “St” in the name is that they use stereo controls (ganged controls) rather than dual mono controls for the chorus and flange components of the algorithms.

St Chorus+Delay

Effects Size : 1

Stereo Chorus and Delay (ms)

St Chorus+Delay Presets
221 PinchChorusDelay	222 StChorus+Delay	223 StChor+3vs2Delay
406 St Chorus+Delay

St Flange+Delay

Effects Size : 1

Stereo Flange and Delay (ms)

St Flange+Delay Presets
237 Crystal Flange	242 StFlng+3vs2Delay	244 DampedEchoFlange
241 StFlange+Delay	243 Singing Flanger	492 Glacial Canyon

Configurable Combinations "<>"

Signal Routing

Each of these combination algorithms offers two separate effects combined with flexible signal routing mechanism.

This mechanism allows the two effects to be either in series bidirectionally or in parallel.

This is done by first designating one effect “A,” and the other “B” where the output of effect A is always wired to effect B.

A and B are assigned with the A→B cfg parameter.

For example, when A→B cfg is set to Ch→Dly, then effect A is the chorus, and effect B is the delay, and the output of the chorus is wired to the input of the delay.

The amount of effect A fed into effect B is controlled by the A/Dry→B parameter.

This controls the balance between effect A output, and the algorithm dry input signal fed into effect B behaving much like a wet/dry mix.

When set to 0%, only the dry signal is fed into B allowing parallel effect routing.
At 100%, only the A output is fed into B, and at 50%, there is an equal mix of both.

Mix

Both effect A and B outputs are mixed at the algorithm output to become the wet signal.

These mix levels are controlled with the 2 parameters that begin with “Mix.”

These allow only one or both effect outputs to be heard.

Negative mix amounts polarity invert the signal which can change the character of each effect when mixed together or with the dry signal.

Wet/Dry

The Wet/Dry parameter adjusts the balance between the sum of both effects determined by the Mix parameters, and the input dry signal.

Negative Wet/Dry values polarity invert the summed wet signal relative to dry.

Figure 108. Chorus<>4Tap with A→B cfg set to Ch→4T and 4T→Ch

Bidirectional Routing
Wet/Dry	-100 to 100 %	Out Gain	Off; -79.0 to 24.0 dB
Mix Effect	-100 to 100 %
Mix Effect	-100 to 100 %
A→B cfg	EffectA→EffectB	A/Dry→B	0 to 100 %

Chorus and Flange

General descriptions of chorus and flange functionality can be found in the Choruses or Flangers sections.

The configurable chorus and flange have two moving delay taps per channel.
Parameters associated with chorus control begin with “Ch” in the parameter name, and those associated with flange begin with “Fl.”
Since these effects have 2 taps per channel, control over 4 LFOs is necessary, but with a minimum number of user parameters.
- This is accomplished by offering 2 sets of LFO controls with three user interface modes: Dual1Tap, Link1Tap, or Link2Tap.
- These are selectable with the LFO cfg parameter and affect the functionality of the two sets of rate, depth and delay controls (and also phase and feedback controls for the Flange).
Each parameter is labeled with a 1 or a 2 in the parameter name to indicate to which control set it belongs.
Control set 1 consists of controls whose name ends with a 1, and control set 2 consists of controls whose name ends with a 2.

Dual1Tap Mode: In Dual1Tap mode, each control set independently controls one tap in each channel.

This is useful for dual mono applications where separate control over left and right channels is desired.

Control Set 1 controls the left channel, and Control Set 2 controls the right channel.

The second pair of moving delay taps are disabled in this mode.

LRPhase is unpredictable unless both rates are set to the same speed.

Then, the phase value is accurate only after the LFOs are reset.

LFOs can be reset by either changing the LFO cfg parameter, or loading in the algorithm by selecting a preset or studio that uses it.

For user-friendly LRPhase control, use either the Link1Tap or Link2Tap modes.

Link1Tap Mode

In Link1Tap mode, Control Set 1 controls 1 tap in both the left and right channels.

Control Set 2 has no affect, and the second pair of LFO delay taps are disabled.

This mode is optimized for an accurate LRPhase relationship between the left and right LFOs.

Link2Tap Mode

In Link2Tap mode, Control Set 1 controls the first left and right pair of LFOs, while Control Set 2 controls the second pair.

This mode uses all four LFOs for a richer sound, and is optimized for LRPhase relationships.

Each of the two taps per channel are summed together at the output, and the Fdbk parameters control the sum of both LFO taps on each channel fed back to the input.

Static Delay Tap

In addition to the LFO delay taps, the Flange offers a static delay tap for creating through-zero flange effects.

The maximum delay time for this tap is 230ms and is controlled by the Fl StatDly parameter.
Its feedback amount is controlled by the Fl StatFB.
Separate mix levels for the LFO taps and the static tap are then controlled by the Fl StatLvl and Fl LFO Lvl controls.
The feedback and level controls can polarity invert each signal be setting them to negative values

Parameters for Choruses
Ch LFO cfg	Dual1Tap…	Ch LRPhase	0 to 360 deg
Ch Rate 1	0.01 to 10.00 Hz	Ch Rate 2	0.01 to 10.00 Hz
Ch Depth 1	0.0 to 100 ct	Ch Depth 2	0.0 to 100 ct
Ch Delay 1	0 to 1000 ms	Ch Delay 2	0 to 1000 ms
Ch Fdbk L	-100 to 100 %	Ch Fdbk R	-100 to 100 %
Ch Xcouple	0 to 100 %	Ch HF Damp	8 to 25088 Hz

Parameters for Flangers Page 1
Fl LFO cfg	Dual1Tap…	Fl LRPhase	0 to 360 deg
Fl Rate 1	0.01 to 10.00 Hz	Fl Rate 2	0.01 to 10.00 Hz
Fl Xcurs 1	0 to 230 ms	Fl Xcurs 2	0 to 230 ms
Fl Delay 1	0 to 1000 ms	Fl Delay 2	0 to 1000 ms
Fl Fdbk 1	-100 to 100 %	Fl Fdbk 2	-100 to 100 %
Fl Phase 1	0 to 360 deg	Fl Phase 2	0 to 360 deg

Parameters for Flangers Page 2
Fl HF Damp	8 to 25088 Hz
Fl Xcouple	0 to 100 %
Fl StatDly	0 to 230 ms
Fl StatFB	-100 to 100 %
Fl StatLvl	-100 to 100 %
Fl LFO Lvl	-100 to 100 %

Laser Delay

There is a limited amount of delay memory available (usually 1.5 seconds for Laser Delay), selecting slow tempos and/or long delay lengths may cause you to run out of delay memory.

At this point, each delay will pin at its maximum possible time.

When you slow down the tempo, you may find the delays lose their sync.

Laser Delay (LasrDly) is a tempo-based delay with added functionality, including image shifting, cross-coupling, high frequency damping, low frequency damping, and a LaserVerb element.
Separate left and right controls are provided for delay time, feedback, and laser controls.
Parameters associated with LaserVerb in a combination algorithm begin with “Dly” or “Lsr.”
The delay length for each channel is determined by Dly Tempo, expressed in beats per minute (BPM), and the delay length (Dly Time L and Dly Time R) of each channel is expressed in beats (bts).
The tempo alters both channel delay lengths together. With the tempo in beats per minute and delay lengths in beats, you can calculate the length of a delay in seconds as \(beats/tempo*60 (sec/min)\).
The laser controls perform similarly to those found in LaserVerb, and affect the laser element of the effect.
The LsrCntour changes the laser regeneration envelope shape.
- Higher values increase the regeneration amount, and setting it to 0% will disable the Laser Delay portion completely turning the effect into a basic delay.
LsrSpace controls the impulse spacing of each regeneration.
- Low values create a strong initial pitched quality with slow descending resonances, while higher values cause the resonance to descend faster through each regeneration.

See the LaserVerb section for more detailed information.

Delay regeneration is controlled collectively by the Dly Fdbk and LsrCntour parameters since the laser element contains feedback within itself.
- Setting both to 0% defeats all regeneration, including the laser element entirely.
- Increasing either one will increase regeneration overall, but with different qualities.
Dly Fdbk is a feedback control in the classic sense, feeding the entire output of the effect back into the input, with negative values polarity inverting the signal.
The LsrCntour parameter adds only the Laser Delay portion of the effect, including its own regeneration.

For the most intense laser-ness, keep Dly Fdbk at 0% while LsrCntour is enabled.

Dly FBImag, Dly Xcouple, Dly HFDamp, and Dly LFDamp are just like those found in other algorithms.
Due to resource allocation limits, not all Laser Delays in combination algorithms will have all four of these parameters.

Parameters for Laser Delay
Dly Time L	0 to 6 bts	Dly Time R	0 to 6 bts
Dly Fdbk L	-100 to 100 %	Dly Fdbk R	-100 to 100 %
Dly HFDamp	0 to 32 bts	Dly FBImag	-100 to 100 %
Dly LFDamp	0.10 to 6.00	Dly Xcple	0 to 100 %
LsrCntourL	0 to 100 %	LsrCntourR	0 to 100 %
LsrSpace L	0 to 100 samp	LsrSpace R	0 to 100 samp

Combination 4-Tap

Combination 4-Tap is a tempo based 4 tap delay with feedback used in combination algorithms.
Parameters associated with the 4 tap effect start with “4T.”
The control over the feedback tap and individual output taps is essentially the same as the 4-Tap Delay BPM algorithm, with the exception that the delay times will pin at the maximum delay time instead of automatically cutting their times in half.
Additionally, the feedback path may also offer cross-coupling, an imager, a highpass filter, and/or a lowpass filter.

Parameters for combination 4-Tap Page 1
4T LoopLen	0 to 32 bts
4T FB Lvl	-100 to 100 %
4T FB Imag	-100 to 100 %
4T FB XCpl	0 to 100 %
4T HF Damp	8 to 25088 Hz
4T LF Damp	8 to 25088 Hz

Parameters for combination 4-Tap Page 2
Tap1 Delay	0 to 8 bts	Tap3 Delay	0 to 8 bts
Tap1 Level	-100 to 100 %	Tap3 Level	-100 to 100 %
Tap1 Bal	-100 to 100 %	Tap3 Bal	-100 to 100 %
Tap2 Delay	0 to 8 bts	Tap4 Delay	0 to 8 bts
Tap2 Level	-100 to 100 %	Tap4 Level	-100 to 100 %
Tap2 Bal	-100 to 100 %	Tap4 Bal	-100 to 100 %

Reverbs

The reverbs offered in these combination effects is MiniVerb.

Information about it can be found in the MiniVerb documentation.

Parameters associated with this reverb begin with Rv.

Parameters for Reverb (MiniVerb)
Rv Type	Hall1
Rv Time	0.5 to 30.0 s; Inf
Rv DiffScl	0.00 to 2.00x	Rv Density	0.00 to 4.00x
Rv SizeScl	0.00 to 4.00x	Rv HF Damp	8 to 25088 Hz
Rv PreDlyL	0 to 620 ms	Rv PreDlyR	0 to 620 ms

Pitcher

The pitchers offered in these effects are the same as that found in its stand alone version.

Information about it can be found in the Pitcher documentation.

Parameters associated with this effect begin with Pt.

Parameters for Pitcher
Pt Pitch	C-1 to G9
Pt Offset	-12.0 to 12.0 ST
Pt Odd Wts	-100 to 100 %
Pt PairWts	-100 to 100 %
Pt 1/4 Wts	-100 to 100 %
Pt 1/2 Wts	-100 to 100 %

Shaper

The shaper offered in these combination effects have the same sonic qualities as those found in the V.A.S.T. Shaper DSP.

Parameters associated with this effect begin with “Shp.”
This shaper also offers input and output 1-pole (6dB/oct) lowpass filters controlled by the Shp Inp LP and Shp Out LP respectively.
There is an additional output gain labeled Shp OutPad to compensate for the added gain caused by shaping a signal.

Parameters for Shaper
Shp Inp LP	8 to 25088 Hz
Shp Amt	0.10 to 6.00 x
Shp Out LP	8 to 25088 Hz
Shp OutPad	Off; -79.0 to 0.0 dB

LasrDly<>Reverb

Effects Size : 2

Laser Tempo Delay and Reverb.

LasrDly<>Reverb Presets
144 Gunshot Verb	189 Timbre Taps	147 Room + Delay
190 LaserDelay ->Rvb

Flange<>Shaper

Effects Size : 2

Flange and Shaper.

Flange<>Shaper Presets
322 Shaper->Flange	458 Flanged Edge	484 Flange->Shaper

Shaper<>Reverb

Effects Size : 2

Shaper and Reverb

Shaper<>Reverb Presets
323 Shaper→Reverb

Flange<>Pitcher

Effects Size : 2

Flange and Pitcher

Flange<>Pitcher Presets
483 Flange→Pitcher
485 Pitch Spinner

Chorus<>4Tap

Effects Size : 2

Chorus and 4 Tap

Chorus<>4Tap Presets
212 Full Chorus	420 Chorused Taps	433 Chorus Echoverb
449 C-D

Chorus<>Reverb

Effects Size : 2

Chorus and Reverb

Chorus<>Reverb Presets
62 Pipes Hall	424 Chorus Slow Hall	431 Chorus Med Hall
409 ChorusMedChamber	425 SoftChorus Hall	432 Chorus Big Hall
410 Chorus MiniHall	426 Chorus Air	434 Chorus Bass Room
411 Chorus HiCeiling	427 PsiloChorusHall	435 New Chorus Hall
412 ChorBigBrtPlate	429 Chorus Room	436 Floyd Hall
413 CathedralChorus	430 Chorus Smallhall

Chorus<>LasrDly

Effects Size : 2

Chorus and Laser Delay

Chorus<>LasrDly Presets
421 MultiEchoChorus
442 Laser Amalgam

Flange<>4Tap

Effects Size : 2

Flange and 4 Tap

Flange<>4Tap Presets
465 FlangeTap Synth	468 Flange 4 Tap	238 NarrowResFlange
467 Flange Echo	481 Space Flanger

Flange<>Reverb

Effects Size : 2

Flange and Reverb

Flange<>Reverb Presets
453 Flange Booth	462 Flange Hall	477 Pewter FlangeVrb
454 FlangeVerb Clav	464 Flange Theatre	478 WeirdFlangePlate
455 Flange Amb Smack	469 Flange Hall 2

Flange<>LasrDly

Effects Size : 2

Flange and Laser Delay

Flange<>LasrDly Presets
36 Flanged Taps
482 Lazertag Flange

Instrument FX Presets / FX Algorithms

FX
Type

FX
Preset

FX
Algorithm

FX
Units

Rvrb

1 Small Wood Booth

4 Classic Place

Rvrb

2 Natural Room

5 Classic Verb

Rvrb

3 PrettySmallPlace

4 Classic Place

Rvrb

4 NiceLittleBooth

1 MiniVerb

Rvrb

5 Sun Room

5 Classic Verb

Rvrb

6 Soundboard

7 TQ Verb

Rvrb

7 Add More Air

10 OmniPlace

Rvrb

8 Standard Booth

8 Diffuse Place

Rvrb

9 A Distance Away

6 TQ Place

Rvrb

10 Live Place

8 Diffuse Place

Rvrb

11 Viewing Booth

1 MiniVerb

Rvrb

12 Small Closet

10 OmniPlace

Rvrb

13 Add Ambience

1 MiniVerb

Rvrb

14 With A Mic

4 Classic Place

Rvrb

15 BrightSmallRoom

1 MiniVerb

Rvrb

16 Bassy Room

1 MiniVerb

Rvrb

17 Percussive Room

1 MiniVerb

Rvrb

18 SmallStudioRoom

4 Classic Place

Rvrb

19 ClassRoom

5 Classic Verb

Rvrb

20 Utility Room

5 Classic Verb

Rvrb

21 Thick Room

5 Classic Verb

Rvrb

22 The Real Room

5 Classic Verb

Rvrb

23 Small Drum Room

1 MiniVerb

Rvrb

24 Real Big Room

5 Classic Verb

Rvrb

25 The Comfy Club

9 Diffuse Verb

Rvrb

26 Spitty Drum Room

7 TQ Verb

Rvrb

27 Stall One

7 TQ Verb

Rvrb

28 Green Room

7 TQ Verb

Rvrb

29 Tabla Room

12 Panaural Room

Rvrb

30 Large Room

7 TQ Verb

Rvrb

31 Platey Room

14 Grand Plate

Rvrb

32 Bathroom

5 Classic Verb

Rvrb

33 Drum Room

12 Panaural Room

Rvrb

34 Small Dark Room

12 Panaural Room

Rvrb

35 Real Room

5 Classic Verb

Rvrb

36 Brt Empty Room

7 TQ Verb

Rvrb

37 Med Large Room

12 Panaural Room

Rvrb

38 Bigger Perc Room

7 TQ Verb

Rvrb

39 Sizzly Drum Room

5 Classic Verb

Rvrb

40 Live Chamber

11 OmniVerb

Rvrb

41 Brass Chamber

1 MiniVerb

Rvrb

42 Sax Chamber

1 MiniVerb

Rvrb

43 Plebe Chamber

1 MiniVerb

Rvrb

44 JudgeJudyChamber

7 TQ Verb

Rvrb

45 Bloom Chamber

7 TQ Verb

Rvrb

46 ClassicalChamber

7 TQ Verb

Rvrb

47 In The Studio

4 Classic Place

Rvrb

48 My Garage

4 Classic Place

Rvrb

49 Cool Dark Place

11 OmniVerb

Rvrb

50 Small Hall

5 Classic Verb

Rvrb

51 Medium Hall

1 MiniVerb

Rvrb

52 Real Niceverb

5 Classic Verb

Rvrb

53 Opera House

5 Classic Verb

Rvrb

54 Mosque Room

7 TQ Verb

Rvrb

55 Grandiose Hall

1 MiniVerb

Rvrb

56 Elegant Hall

1 MiniVerb

Rvrb

57 Bright Hall

1 MiniVerb

Rvrb

58 Ballroom

1 MiniVerb

Rvrb

59 Spacious Hall

5 Classic Verb

Rvrb

60 Classic Chapel

5 Classic Verb

Rvrb

61 Semisweet Hall

5 Classic Verb

CRvb

62 Pipes Hall

404 Chorus<>Reverb

Rvrb

63 Reflective Hall

5 Classic Verb

Rvrb

64 Smoooth Hall

5 Classic Verb

Rvrb

65 Empty Stage

7 TQ Verb

Rvrb

66 Pad Space

11 OmniVerb

Rvrb

67 Bob’sDiffuseHall

9 Diffuse Verb

Rvrb

68 Abbey Piano Hall

7 TQ Verb

Rvrb

69 Short Hall

13 Stereo Hall

Rvrb

70 The Long Haul

7 TQ Verb

Rvrb

71 Predelay Hall

9 Diffuse Verb

Rvrb

72 Sweeter Hall

7 TQ Verb

Rvrb

73 The Piano Hall

7 TQ Verb

Rvrb

74 Bloom Hall

9 Diffuse Verb

Rvrb

75 Recital Hall

12 Panaural Room

Rvrb

76 Generic Hall

12 Panaural Room

Rvrb

77 Burst Space

9 Diffuse Verb

Rvrb

78 Real Dense Hall

7 TQ Verb

Rvrb

79 Concert Hall

9 Diffuse Verb

Rvrb

80 Standing Ovation

11 OmniVerb

Rvrb

81 Flinty Hall

7 TQ Verb

Rvrb

82 HighSchool Gym

7 TQ Verb

Rvrb

83 My Dreamy 481!!

9 Diffuse Verb

Rvrb

84 Deep Hall

9 Diffuse Verb

Rvrb

85 Sweet Hall

5 Classic Verb

Rvrb

86 Soundbrd/rvb

11 OmniVerb

Rvrb

87 Long & Narrow

7 TQ Verb

Rvrb

88 Long PreDly Hall

11 OmniVerb

Rvrb

89 School Stairwell

4 Classic Place

Rvrb

90 Real Plate

14 Grand Plate

Rvrb

91 Bright Plate

14 Grand Plate

Rvrb

92 Medm Warm Plate

7 TQ Verb

Rvrb

93 Bloom Plate

9 Diffuse Verb

Rvrb

94 Clean Plate

9 Diffuse Verb

Rvrb

95 Plate Mail

11 OmniVerb

Rvrb

96 RealSmoothPlate

9 Diffuse Verb

Rvrb

97 Classic Plate

5 Classic Verb

Rvrb

98 Weighty Platey

5 Classic Verb

Rvrb

99 Huge Tight Plate

9 Diffuse Verb

Rvrb

100 Immense Mosque

7 TQ Verb

Rvrb

101 Dreamverb

10 OmniPlace

Rvrb

102 Splendid Palace

5 Classic Verb

Rvrb

103 Big Gym

11 OmniVerb

Rvrb

104 Huge Batcave

12 Panaural Room

Rvrb

105 Reverse Reverb 1

15 Finite Verb

Rvrb

106 Reverse Reverb 2

15 Finite Verb

Rvrb

107 Reverse Reverb 3

15 Finite Verb

GRvb

108 Gated Reverb

3 Gated MiniVerb

GRvb

109 Gate Plate

3 Gated MiniVerb

Vocl

110 Vocal Room

53 Gate+Cmp[EQ]+Rvb

Vocl

111 Vocal Stage

53 Gate+Cmp[EQ]+Rvb

RvCm

112 Reverb>Compress

51 Reverb<>Compress

RvCm

113 Reverb>Compress2

51 Reverb<>Compress

RvCm

114 Drum Comprs>Rvb

51 Reverb<>Compress

RvCm

115 Rvrb Compression

50 Reverb+Compress

RvCm

116 Snappy Drum Room

50 Reverb+Compress

RvCm

117 Roomitizer

50 Reverb+Compress

RvCm

118 Live To Tape

50 Reverb+Compress

Rvrb

119 L:SmlRm R:Hall

2 Dual MiniVerb

Rvrb

120 Non-Linear 1

10 OmniPlace

Rvrb

121 Non-Linear 2

15 Finite Verb

Rvrb

122 Non-Linear 3

6 TQ Place

Rvrb

123 Exponent Booth

10 OmniPlace

Rvrb

124 Drum Latch 1

10 OmniPlace

Rvrb

125 Drum Latch 2

10 OmniPlace

Rvrb

126 Diffuse Gate

9 Diffuse Verb

Rvrb

127 Acid Trip Room

10 OmniPlace

GLvb

128 Ringy Drum Plate

104 Gated LaserVerb

GLvb

129 Oil Tank

104 Gated LaserVerb

GLvb

130 Wobbly Plate

104 Gated LaserVerb

PchR

131 Pitcher Hall

383 Pitcher+Miniverb

PchR

132 DistantTVRoom

383 Pitcher+Miniverb

Lvrb

133 Drum Neurezonate

102 Mono LaserVerb

GLvb

134 Growler

104 Gated LaserVerb

Lvrb

135 LaserVerb

100 LaserVerb

Lvrb

136 Laserwaves

100 LaserVerb

Lvrb

137 Cheap LaserVerb

101 LaserVerb Lite

GLvb

138 Gated LaserVerb

104 Gated LaserVerb

Lvrb

139 Rvrs LaserVerb

103 Revrse LaserVerb

Lvrb

140 LazerfazerEchoes

102 Mono LaserVerb

Lvrb

141 Simple LaserVerb

102 Mono LaserVerb

Lvrb

142 Crystallizer

100 LaserVerb

Lvrb

143 Spry Young Boy

101 LaserVerb Lite

DlyR

144 Gunshot Verb

105 LasrDly<>Reverb

Rvrb

145 Slapverb

11 OmniVerb

Rvrb

146 Far Bloom

9 Diffuse Verb

DlyR

147 Room + Delay

105 LasrDly<>Reverb

CDRv

148 New Hall w/Delay

403 Chor+Dly+Reverb

CDRv

149 Delay Big Hall

403 Chor+Dly+Reverb

Dly

150 Basic Delay 1/8

150 4-Tap Delay BPM

Dly

151 Basic Dly 250ms

190 Moving Delay

Dly

152 Simple Slap 60ms

190 Moving Delay

Dly

153 TightSlapbk 30ms

190 Moving Delay

Dly

154 MedSlapback 76ms

190 Moving Delay

Dly

155 LongishSlap 95ms

151 4-Tap Delay

Dly

156 Wide Slapbk 76ms

191 Dual MovDelay

Dly

157 TiteSlapAmb 50ms

191 Dual MovDelay

Dly

158 33ms Ambience

191 Dual MovDelay

Dly

159 17ms Ambience

191 Dual MovDelay

Dly

160 Stereo Delay ms

151 4-Tap Delay

Dly

161 StereoFlamDelay

191 Dual MovDelay

Dly

163 Better Tape Echo

171 Degen Regen

Dly

164 Stereo Tape Slap

171 Degen Regen

Dly

165 Dub Delay ms

190 Moving Delay

Dly

166 4-Tap Delay BPM

150 4-Tap Delay BPM

Dly

167 4-Tap Dly Pan ms

151 4-Tap Delay

Dly

168 SemiCircle 4-Tap

151 4-Tap Delay

Dly

169 8-Tap Delay BPM

152 8-Tap Delay BPM

Dly

170 Multitaps ms

156 Complex Echo

Dly

171 Diffuse Slaps

156 Complex Echo

Dly

172 OffbeatFlamDelay

150 4-Tap Delay BPM

Dly

173 Sloppy Echoes

156 Complex Echo

Dly

174 Pad Psychosis

191 Dual MovDelay

Dly

175 500ms BehindSrce

156 Complex Echo

Dly

176 Dub Skanque Dly

154 Spectral 4-Tap

Dly

177 Electronica Slap

156 Complex Echo

Dly

178 Spectral 4-Tap

154 Spectral 4-Tap

Dly

179 Astral Taps

154 Spectral 4-Tap

Dly

180 SpectraShapeTaps

155 Spectral 6-Tap

Dly

181 Fanfare In Gmaj

155 Spectral 6-Tap

Dly

182 Ecko Plecks BPM

170 Degen Regen BPM

Dly

183 Ecko Plecks ms

171 Degen Regen

Dly

184 Degenerator

170 Degen Regen BPM

Dly

185 Nanobot Feedback

170 Degen Regen BPM

Dly

186 Takes a while…

170 Degen Regen BPM

Dly

187 Wait for UFO

170 Degen Regen BPM

Dly

188 News Update

172 Switch Loops

DlyR

189 Timbre Taps

105 LasrDly<>Reverb

DlyR

190 LaserDelay→Rvb

105 LasrDly<>Reverb

Rvrb

191 Furbelows

9 Diffuse Verb

Rvrb

192 Festoons

9 Diffuse Verb

Dly

193 Ducked Delay

174 Gated Delay

Dly

194 Drum+Bass Zapper

174 Gated Delay

Dly

195 3BandDly Drums

173 3 Band Delay

Dly

196 Warped Echoes

191 Dual MovDelay

Dly

197 Ween-vox

190 Moving Delay

Dly

198 L:Flange R:Delay

191 Dual MovDelay

Dly

199 2Dlys 1Chr 1Flng

192 Dual MvDly+MvDly

Chor

200 Basic Chorus

202 Dual Chorus 1

Chor

201 Smooth Chorus

202 Dual Chorus 1

Chor

202 Chorusier

202 Dual Chorus 1

Chor

203 Ordinary Chorus

202 Dual Chorus 1

Chor

204 SlowSpinChorus

202 Dual Chorus 1

Chor

205 Chorus Morris

202 Dual Chorus 1

Chor

206 Everyday Chorus

202 Dual Chorus 1

Chor

207 Thick Chorus

203 Dual Chorus 2

Chor

208 Soft Chorus

203 Dual Chorus 2

Chor

209 Rock Chorus

203 Dual Chorus 2

Chor

210 Sm Stereo Chorus

200 Chorus 1

Chor

211 Lg Stereo Chorus

201 Chorus 2

CDly

212 Full Chorus

402 Chorus<>4Tap

Chor

213 Dense Gtr Chorus

201 Chorus 2

CDly

214 Standrd Gtr Chor

406 St Chorus+Delay

Chor

215 Bass Chorus

202 Dual Chorus 1

Chor

216 Stereo Chorus

203 Dual Chorus 2

CDly

217 Chorus Fastback

400 Chorus+Delay

Chor

218 Wide Chorus

203 Dual Chorus 2

PLFO

219 Nickel Chorus

387 WackedPitchLFO

Dly

220 Rich Noodle

190 Moving Delay

CDly

221 PinchChorusDelay

406 St Chorus+Delay

CDly

222 StChorus+Delay

406 St Chorus+Delay

CDly

223 StChor+3vs2Delay

406 St Chorus+Delay

CDRv

224 CDR for Lead Gtr

403 Chor+Dly+Reverb

Flng

225 Big Slow Flange

225 Flanger 1

Flng

226 Squeeze Flange

225 Flanger 1

Flng

227 Sweet Flange

225 Flanger 1

Flng

228 Throaty Flange

225 Flanger 1

Flng

229 PseudoAnaGtrFlng

225 Flanger 1

Flng

230 Flanger Double

225 Flanger 1

Flng

231 Wetlip Flange

225 Flanger 1

Flng

232 Simply Flange

226 Flanger 2

Flng

233 Analog Flanger

226 Flanger 2

Flng

234 Soft Edge Flange

226 Flanger 2

Flng

235 Ned Flangers

225 Flanger 1

Flng

236 Wispy Flange

225 Flanger 1

FDly

237 Crystal Flange

456 St Flange+Delay

FDly

238 NarrowResFlange

452 Flange<>4Tap

FDly

239 TightSlapFlange

450 Flange+Delay

FDly

240 Flanged Taps

455 Flange<>LasrDly

FDly

241 StFlange+Delay

456 St Flange+Delay

FDly

242 StFlng+3vs2Delay

456 St Flange+Delay

FDly

243 Singing Flanger

456 St Flange+Delay

FDly

244 DampedEchoFlange

456 St Flange+Delay

Flng

245 Stereo Flanger

226 Flanger 2

Flng

246 Gulp Flange

225 Flanger 1

Flng

247 Splat Flange

225 Flanger 1

Flng

248 Spread Flange

225 Flanger 1

Flng

249 CacophonousFlng

225 Flanger 1

Phsr

250 Slow Deep Phaser

251 LFO Phaser Twin

Phsr

251 Circles

250 LFO Phaser

Phsr

252 Saucepan Phaser

253 SingleLFO Phaser

Phsr

253 ThunderPhaser

254 VibratoPhaser

Phsr

254 Fast Phaser

251 LFO Phaser Twin

Phsr

255 Vibrato Phaser

254 VibratoPhaser

Phsr

256 Fast&Slow Phaser

250 LFO Phaser

Phsr

257 Wawawawawawawawa

253 SingleLFO Phaser

Phsr

258 Slow Swish Phase

253 SingleLFO Phaser

Freq

259 Slippery Slope

385 Frequency Offset

Phsr

260 Static Phaser 1

255 Manual Phaser

Phsr

261 Static Phaser 2

255 Manual Phaser

Phsr

262 Static Phaser 3

255 Manual Phaser

Phsr

263 Static Phaser 4

255 Manual Phaser

Phsr

264 Static Phaser 5

257 Allpass Phaser 4

Phsr

265 Slow Riser

258 Barberpole Comb

Phsr

266 BarberPole Notch

258 Barberpole Comb

Phsr

267 BarberPole Peak

258 Barberpole Comb

Phsr

268 All The Way Down

258 Barberpole Comb

Freq

269 Westward Waves

385 Frequency Offset

Trem

270 Tremolo BPM

Trem

271 Fast Tremolo BPM

270 Tremolo BPM

Trem

272 Tremolo in Hz

271 Tremolo

Trem

273 FastPulseTremolo

270 Tremolo BPM

Pan

274 Simple Panner

275 AutoPanner

Pan

275 Dual Panner

276 Dual AutoPanner

Ster

276 Widespread

280 Stereo Image

Ster

277 Widener Mn→St

281 Mono → Stereo

Ster

278 Dynam Stereoizer

282 DynamicStereoize

KB3

280 CleanRotors fast

290 VibChor+Rotor 2

KB3

281 CleanRotors slow

290 VibChor+Rotor 2

KB3

282 CleanRotors f C1

290 VibChor+Rotor 2

KB3

283 CleanRotors f V1

290 VibChor+Rotor 2

KB3

284 CleanRotors f Hi

290 VibChor+Rotor 2

KB3

285 CleanRotors s Hi

290 VibChor+Rotor 2

KB3

286 SlightDstRotor

291 Distort + Rotary

KB3

287 SlightDstRotor

291 Distort + Rotary

KB3

288 DirtyRotors fast

292 VC+Dist+HiLoRotr

KB3

289 DirtyRotors slow

292 VC+Dist+HiLoRotr

KB3

290 MoreDistRotor

293 VC+Dist+1Rotor 2

KB3

291 MoreDistRotor

293 VC+Dist+1Rotor 2

KB3

292 HeavyDistRotor

294 VC+Dist+HiLoRot2

KB3

293 HeavyDistRotor

294 VC+Dist+HiLoRot2

KB3

294 Res Rotor1 fast

295 Rotor 1

KB3

295 Res Rotor1 slow

295 Rotor 1

KB3

296 FullRotors4 fast

296 VC+Dist+Rotor 4

KB3

297 FullRotors4 slow

296 VC+Dist+Rotor 4

KB3a

298 VibChorStortCab

298 KB3 FXBus

KB3b

299 Hi Lo Roto KB3

Guit

300 Classic Gtr Dist

310 Gate+TubeAmp

Guit

301 Crunch Guitar

310 Gate+TubeAmp

Guit

302 SaturatedGtrDist

310 Gate+TubeAmp

Guit

303 Mean 70’sFunkGtr

310 Gate+TubeAmp

Kaos

304 Blown Speaker

390 Chaos!

Dist

305 Synth Distortion

303 PolyDistort + EQ

Dly

306 Superphasulate

170 Degen Regen BPM

Dist

307 Dist Cab EPiano

301 MonoDistort+Cab

Dist

308 Distortion+EQ

302 MonoDistort + EQ

Dist

309 Burnt Transistor

304 StereoDistort+EQ

Dist

310 SubtleDistortion

300 Mono Distortion

Dist

311 A little dirty

305 Subtle Distort

Dist

312 Slight Overload

305 Subtle Distort

DstC

313 ODriveGtrLd DlCh

317 TubeAmp<>MD>Chor

DstC

314 Krazy Gtr Comper

317 TubeAmp<>MD>Chor

DstF

315 MildGtrOD+Dly+Fl

320 PolyAmp<>MD>Flan

DstF

316 LeadGtr Dly Flng

318 TubeAmp<>MD>Flan

Shpr

317 Drum Shaper

306 Super Shaper

Shpr

318 SubtleDrumShape

307 3 Band Shaper

Shpr

319 SuperShaper

306 Super Shaper

Shpr

320 3 Band Shaper

307 3 Band Shaper

Shpr

321 New3BandShaper

307 3 Band Shaper

FShp

322 Shaper→Flange

321 Flange<>Shaper

ShpR

323 Shaper→Reverb

322 Shaper<>Reverb

QntA

329 Aliaser

308 Quantize+Alias

Cmpr

330 HKCompressor 3:1

330 HardKneeCompress

Cmpr

331 HKCompressor 5:1

330 HardKneeCompress

Cmpr

332 SK FB Comprs 6:1

331 SoftKneeCompress

Cmpr

333 SKCompressor 9:1

331 SoftKneeCompress

Cmpr

334 SKCompressr 12:1

331 SoftKneeCompress

Cmpr

336 Compress w/SC EQ

332 Compress w/SC EQ

CmpX

337 Compress/Expand

341 Compress/Expand

CmpX

338 Comprs/Expnd +EQ

342 Comp/Exp + EQ

Expd

339 Expander

340 Expander

Gate

340 Simple Gate

343 Gate

Gate

341 Gate w/ SC EQ

344 Gate w/SC EQ

Cmpr

342 3Band Compressor

336 3 Band Compress

Cmpr

343 3Band Compress2

336 3 Band Compress

Cmpr

344 Mid Compressor

335 Band Compress

Expd

345 OddHarmSuppress

374 HarmonicSuppress

Expd

346 60Hz Buzz Kill

374 HarmonicSuppress

Cmpr

347 Dual SK Compress

347 Dual SKCompress

Cmpr

348 Dual Comprs SCEQ

350 AM Radio

350 3 Band EQ

351 U-Shaped EQ

350 3 Band EQ

352 5 Band EQ Flat

351 5 Band EQ

353 Graphic EQ Flat

352 Graphic EQ

354 Dual Graphic EQ

353 Dual Graphic EQ

355 Dual 5 Band EQ

354 Dual 5 Band EQ

Filt

356 Basic Env Filter

360 Env Follow Filt

Filt

357 Phunk Env Filter

360 Env Follow Filt

Filt

358 Synth Env Filter

360 Env Follow Filt

Filt

359 Bass Env Filter

360 Env Follow Filt

Filt

360 EPno Env Filter

360 Env Follow Filt

Filt

362 LFO Sweep Filter

Filt

363 DoubleRiseFilter

362 LFO Sweep Filter

Filt

364 Circle Bandsweep

362 LFO Sweep Filter

Filt

365 TripFilter

362 LFO Sweep Filter

Filt

366 Resonant Filter

363 Resonant Filter

Filt

367 Dual Res Filter

364 Dual Res Filter

Enhc

368 2 Band Enhancer

370 2 Band Enhancer

Enhc

369 3 Band Enhancer

371 3 Band Enhancer

Enhc

370 Extreem Enhancer

371 3 Band Enhancer

Stim

371 HF Stimulator

372 HF Stimulate 1

RMod

372 Ring Modulator

380 Ring Modulator

Pchr

373 PitcherA

381 Pitcher

Pchr

374 PitcherB

381 Pitcher

Pchr

375 PolyPtVoxChanger

382 Poly Pitcher

Pchr

376 HollowPolyPitchr

382 Poly Pitcher

PchC

377 Pitcher+Chorus

411 MonoPitcher+Chor

PchF

378 Pitcher+Flange

461 MonoPitcher+Flan

PchC

379 Pitcher+Chor+Dly

409 Pitcher+Chor+Dly

PchF

380 Pitcher+Flng+Dly

459 Pitcher+Flan+Dly

Kaos

381 Ring Linger

390 Chaos!

Lvrb

382 Waterford

103 Revrse LaserVerb

Phsr

383 Hip Hop Aura

256 Allpass Phaser 3

Phsr

384 Woodenize

256 Allpass Phaser 3

Phsr

385 Marimbafication

256 Allpass Phaser 3

Freq

386 Frequency Offset

385 Frequency Offset

Freq

387 Drum Loosener

385 Frequency Offset

Freq

388 Drum Tightener

385 Frequency Offset

Freq

389 Vox Honker

386 MutualFreqOffset

Filt

390 EQ Morpher ah-oo

365 EQ Morpher

Filt

391 EQ Morpher ee-aa

365 EQ Morpher

Filt

392 EQ Morpher aw-er

365 EQ Morpher

PLFO

395 Contact

387 WackedPitchLFO

PLFO

396 Drum Frightener

387 WackedPitchLFO

PLFO

397 Mad Hatter

387 WackedPitchLFO

PLFO

398 Fallout

387 WackedPitchLFO

PLFO

399 Ascension

387 WackedPitchLFO

CDly

400 BasicChorusDelay

400 Chorus+Delay

CDly

401 Chorus PanDelay

400 Chorus+Delay

CDly

402 Chorus & Echo

400 Chorus+Delay

CDRv

403 CDR Lead

403 Chor+Dly+Reverb

CDRv

404 CDR Lead 2

403 Chor+Dly+Reverb

CDly

405 Chorus Delay 2

400 Chorus+Delay

CDly

406 Doubler & Echo

400 Chorus+Delay

CDRv

407 Chorus Booth

403 Chor+Dly+Reverb

CDRv

408 ChorusSmallRoom

403 Chor+Dly+Reverb

CRvb

409 ChorusMedChamber

404 Chorus<>Reverb

CRvb

410 Chorus MiniHall

404 Chorus<>Reverb

CRvb

411 Chorus HiCeiling

404 Chorus<>Reverb

CRvb

412 ChorBigBrtPlate

404 Chorus<>Reverb

CRvb

413 CathedralChorus

404 Chorus<>Reverb

CDRv

414 Flam Dly Bckgrnd

403 Chor+Dly+Reverb

CDRv

415 CDHall Halo

403 Chor+Dly+Reverb

CDly

416 CrackedPorcelain

401 Chorus+4Tap

CDRv

417 Rich Delay

403 Chor+Dly+Reverb

CDly

418 FastChorusDouble

400 Chorus+Delay

CDly

419 MultiTap Chorus

401 Chorus+4Tap

CDly

420 Chorused Taps

402 Chorus<>4Tap

CDly

421 MultiEchoChorus

405 Chorus<>LasrDly

CDRv

422 DeepChorDlyHall

403 Chor+Dly+Reverb

CDRv

423 ClassicEP ChorRm

403 Chor+Dly+Reverb

CRvb

424 Chorus Slow Hall

404 Chorus<>Reverb

CRvb

425 SoftChorus Hall

404 Chorus<>Reverb

CRvb

426 Chorus Air

404 Chorus<>Reverb

CRvb

427 PsiloChorusHall

404 Chorus<>Reverb

CDly

428 SpeeChorusDeep

400 Chorus+Delay

CRvb

429 Chorus Room

404 Chorus<>Reverb

CRvb

430 Chorus Smallhall

404 Chorus<>Reverb

CRvb

431 Chorus Med Hall

404 Chorus<>Reverb

CRvb

432 Chorus Big Hall

404 Chorus<>Reverb

CDly

433 Chorus Echoverb

402 Chorus<>4Tap

CRvb

434 Chorus Bass Room

404 Chorus<>Reverb

CRvb

435 New Chorus Hall

404 Chorus<>Reverb

CRvb

436 Floyd Hall

404 Chorus<>Reverb

CDRv

437 Into The Abyss

403 Chor+Dly+Reverb

CDRv

438 BroadRevSlapback

403 Chor+Dly+Reverb

CDRv

439 Carlsbad Cavern

403 Chor+Dly+Reverb

DstC

440 Chr→GtrDst→Chr

317 TubeAmp<>MD>Chor

CDRv

441 That’s No Moon!!

403 Chor+Dly+Reverb

CDly

442 Laser Amalgam

405 Chorus<>LasrDly

CDRv

443 Cut it out!! CDR

403 Chor+Dly+Reverb

CDRv

444 Chor-Delay Booth

403 Chor+Dly+Reverb

CDRv

445 Chor Tin Room

403 Chor+Dly+Reverb

CDRv

446 Boiler Plate

403 Chor+Dly+Reverb

CDRv

447 O.T.T. Pad

403 Chor+Dly+Reverb

CDRv

448 TheChorusCloset

403 Chor+Dly+Reverb

CDly

449 C-D

402 Chorus<>4Tap

FDly

450 Flange + Delay

450 Flange+Delay

FDly

451 ThroatyFlangeDly

450 Flange+Delay

FDly

452 Slapback Flange

450 Flange+Delay

FRvb

453 Flange Booth

454 Flange<>Reverb

FRvb

454 FlangeVerb Clav

454 Flange<>Reverb

FRvb

455 Flange Amb Smack

454 Flange<>Reverb

FDRv

456 Flange Dly 3-D

453 Flan+Dly+Reverb

FDRv

457 Fl Dl Large Hall

453 Flan+Dly+Reverb

FShp

458 Flanged Edge

321 Flange<>Shaper

FDly

459 Flange + 4Tap

451 Flange+4Tap

FDRv

460 FlangeDelayHall

453 Flan+Dly+Reverb

FDRv

461 SloFlangeDlyRoom

453 Flan+Dly+Reverb

FRvb

462 Flange Hall

454 Flange<>Reverb

FDRv

463 FlangeDlyBigHall

453 Flan+Dly+Reverb

FRvb

464 Flange Theatre

454 Flange<>Reverb

FDly

465 FlangeTap Synth

452 Flange<>4Tap

FDRv

466 Flange Room

453 Flan+Dly+Reverb

FDly

467 Flange Echo

452 Flange<>4Tap

FDly

468 Flange 4 Tap

452 Flange<>4Tap

FRvb

469 Flange Hall 2

454 Flange<>Reverb

FDRv

470 Flange-Dly Hall

453 Flan+Dly+Reverb

FDly

471 Flange Delay

450 Flange+Delay

FDly

472 Mecha-Godzilla

451 Flange+4Tap

FDRv

473 Industro-Flange

453 Flan+Dly+Reverb

FDRv

474 Panning FDRoom

453 Flan+Dly+Reverb

FDly

475 Drum&Bass FlgDly

451 Flange+4Tap

FDly

476 Laserflange

455 Flange<>LasrDly

FRvb

477 Pewter FlangeVrb

454 Flange<>Reverb

FRvb

478 WeirdFlangePlate

454 Flange<>Reverb

FDRv

479 F-D Hall

453 Flan+Dly+Reverb

FDRv

480 SyntheticRmFlg

453 Flan+Dly+Reverb

FDly

481 Space Flanger

452 Flange<>4Tap

FDly

482 Lazertag Flange

455 Flange<>LasrDly

FPch

483 Flange→Pitcher

384 Flange<>Pitcher

FShp

484 Flange→Shaper

321 Flange<>Shaper

FPch

485 Pitch Spinner

384 Flange<>Pitcher

FDly

486 FD Lead Madness

450 Flange+Delay

FDRv

487 Brite Rippleverb

453 Flan+Dly+Reverb

FDRv

488 Rotary Club

453 Flan+Dly+Reverb

FDRv

489 Flangey Hall

453 Flan+Dly+Reverb

DstC

490 Flg→GtrDst→Chr

319 PolyAmp<>MD>Chor

DstF

491 MyGtrAteYo’Momma

318 TubeAmp<>MD>Flan

FDly

492 Glacial Canyon

456 St Flange+Delay

CDRv

494 Ultima Thule Pad

403 Chor+Dly+Reverb

Flng

495 Dr. Who

225 Flanger 1

TDst

500 Gate+TTubeAmp3

TDst

501 MonoDistT + EQ

TDst

502 TTubeAmp3

KB3a

580 VibChorCabT

Reso

600 String Resonance