Allocation Units: 1 for Mono Distortion; 2 for MonoDistort + Cab; 2 for MonoDistort + EQ;
3 for StereoDistort + EQ
Mono Distortion sums its stereo input to mono, performs distortion followed by a hipass filter and sends the result
as centered stereo.
L Input
Distortion
R Input
Block diagram of Mono Distortion
MonoDistort + EQ is similar to Mono Distortion except the single hipass filter is replaced with a pair of second-order
hipass/lowpass filters to provide rudimentary speaker cabinet modeling. The hipass and lowpass filters are then
followed by an EQ section with bass and treble shelf filters and two parametric mid filters.
L Input
Distortion
R Input
Block diagram of MonoDistort + EQ
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 one parametric mid filter.
Cabinet
EQ
L Output
R Output
L Output
R Output
R Input
Algorithm Reference-94
Distortion
Distortion
Block diagram of StereoDistort+EQ
EQ
EQ
L OutputL Input
R Output
FXAlgs #724-6, 728: Distortion
MonoDistort + Cab is also similar to Mono Distortion except the hipass is replaced by a full speaker cabinet model.
There is also a panner to route the mono signal between left and right outputs. In MonoDistort + Cab, the distortion
is followed by a model of a guitar amplifier cabinet. The model can be bypassed, or there are eight presets which
were derived from measurements of real cabinets. (See descriptions of FXAlgs #729-732 in this book for more
information.)
L Input
R Input
Distortion
Cabinet
Filter
Pan
L Output
R Output
Block diagram of MonoDistort + Cab
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 ÒharderÓ. 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. These algorithms will not digitally
clip unless the output gain is over-driven.
Output
Input
Input/Output Transfer Characteristic of Soft Clipping at Various Drive Settings
Signals which 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 the 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
ÒbrassyÓ character to the distorted sound. The dc offset added prior to distortion gets removed at a later point in
the algorithm.
Algorithm Reference-95
FXAlgs #724-6, 728: Distortion
Parameters - Mono Distortion:
PAGE 1
Wet/Dry0 to 100%wetOut GainOff, -79.0 to 24.0 dB
Dist Drive0 to 96 dB
Warmth16 to 25088 Hz
Highpass16 to 25088 Hz
MonoDistort + Cab:
PAGE 1
Wet/Dry0 to 100%wetOut GainOff, -79.0 to 24.0 dB
Dist Drive0 to 96 dB
Warmth16 to 25088 HzCab BypassIn or Out
Cab PresetBasic
MonoDistort + EQ:
PAGE 1
Wet/Dry0 to 100%wetOut GainOff, -79.0 to 24.0 dB
Dist Drive0 to 96 dB
Warmth16 to 25088 Hzdc Offset-100 to 100%
Cabinet HP16 to 25088 HzCabinet LP16 to 25088 Hz
PAGE 2
Bass Gain-79.0 to 24.0 dBTreb Gain-79.0 to 24.0 dB
Bass Freq16 to 25088 HzTreb Freq16 to 25088 Hz
Mid1 Gain-79.0 to 24.0 dBMid2 Gain-79.0 to 24.0 dB
Mid1 Freq16 to 25088 HzMid2 Freq16 to 25088 Hz
Mid1 Width0.010 to 5.000 octMid2 Width0.010 to 5.000 oct
StereoDistort+EQ:
PAGE 1
Wet/Dry0 to 100%wetOut GainOff, -79.0 to 24.0 dB
Dist Drive0 to 96 dB
Warmth16 to 25088 Hz
Cabinet HP16 to 25088 HzCabinet LP16 to 25088 Hz
Algorithm Reference-96
FXAlgs #724-6, 728: Distortion
PAGE 2
Bass Gain-79.0 to 24.0 dBTreb Gain-79.0 to 24.0 dB
Bass Freq16 to 25088 HzTreb Freq16 to 25088 Hz
Mid Gain-79.0 to 24.0 dB
Mid Freq16 to 25088 Hz
Mid Width0.010 to 5.000 oct
Wet/DryThe amount of distorted (wet) signal relative to unaffected (dry) signal.
Out GainThe 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 DriveApplies 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.
armthA lowpass Þlter in the distortion control path. This Þlter may be used to reduce some of
W
the harshness of some distortion settings without reducing the bandwidth of the signal.
Cab BypassThe guitar ampliÞer cabinet simulation may be bypassed. When set to ÒInÓ, the cabinet
simulation is active; when set to ÒOutÓ, there is no cabinet Þltering. [MonoDistort + Cab]
Cab PresetEight preset cabinets have been created based on measurements of real guitar ampliÞer
cabinets. The presets are Basic, Lead 12, 2x12, Open 12, Open 10, 4x12, Hot 2x12, and Hot
12. See description of FX Algs #729-732 for more information. [MonoDistort + Cab]
HighpassAllows you to reduce the bass content of the distortion content. If you need more
Þltering to better simulate a speaker cabinet, you will have to choose a larger distortion
algorithm. [Mono Distortion]
MonoDistort + EQ and StereoDistort+EQ
Cabinet HPA hipass Þlter which controls the low-frequency limit of a simulated loudspeaker
cabinet.
Cabinet LPA lowpass Þlter which controls the high-frequency limit of a simulated cabinet.
Bass GainThe amount of boost or cut that the bass shelving Þlter 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 speciÞed frequency. Negative
values cut the bass signal below the speciÞed frequency.
Bass FreqThe center frequency of the bass shelving Þlter in intervals of one semitone.
Treb GainThe amount of boost or cut that the treble shelving Þlter 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 speciÞed frequency. Negative
values cut the treble signal above the speciÞed frequency.
Treb FreqThe center frequency of the treble shelving Þlter in intervals of one semitone.
Mid GainThe amount of boost or cut that the mid parametric Þlter should apply in dB. Every
increase of 6 dB approximately doubles the amplitude of the signal. Positive values
boost the signal at the speciÞed frequency. Negative values cut the signal at the
speciÞed frequency.
Mid FreqThe center frequency of the mid parametric Þlter in intervals of one semitone. The boost
or cut will be at a maximum at this frequency.
Mid WidThe bandwidth of the mid parametric Þlter may be adjusted. You specify the bandwidth
in octaves. Small values result in a very narrow Þlter response. Large values result in a
very broad response.
Algorithm Reference-97
FXAlg #727: PolyDistort + EQ
FXAlg #727: PolyDistort + EQ
Eight-stage distortion followed by equalization
Allocation Units: 2
PolyDistort + EQ is a distortion algorithm followed by equalization. The algorithm consists of an input gain stage,
and then eight cascaded distortion stages. Each stage is followed by a one-pole LP filter. There is also a one pole
LP in front of the first stage. After the distortion there is a 4-band EQ section: Bass, Treble, and two Parametric Mids.
L Input
R Input
Dist Drive
Distort
Curve 1
Distort
Curve 3
Distort
Curve 5
Dry
LP0
Distort
Curve 2
LP1LP2
Distort
Curve 4
LP3LP4
Distort
Curve 6
LP5LP6
Algorithm Reference-98
Distort
Curve 7
LP7LP8
BassTrebleMid1Mid2
Distort
Curve 8
Parametric
Block diagram of PolyDistort + EQ
L Output
Wet
R Output
Dry
FXAlg #727: PolyDistort + EQ
PolyDistort is an unusual distortion algorithm which 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 the sound. Taken
together, they can produce a very harsh heavy metal sound. Between each distortion stage is a lopass filter. The
lopass filters work with the distortion stages to help mellow out the sound. Without any lopass 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 lopasses should be turned off if there is no distortion in a section. More than 4 stages
seem necessary for lead guitar sounds. For a cleaner sound, you may want to limit yourself to only 4 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 to 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 lopass 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. The Treble is helping to
remove raspiness, the 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, and Mid2 adds some warmth. Your favorite settings will probably be different.
Boosting the Treble may not be a good idea.
Pre-distortion EQ, available on the KDFX Studio INPUT pages, is also useful for shaping the sound. 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 tiring after a while.
Parameters:
PAGE 1
Wet/Dry0 to 100%wetOut GainOff, -79.0 to 24.0 dB
Dist DriveOff, -79.0 to 48.0 dB
PAGE 2
Curve 10 to 127%Curve 50 to 127%
Curve 20 to 127%Curve 60 to 127%
Curve 30 to 127%Curve 70 to 127%
Curve 40 to 127%Curve 80 to 127%
PAGE 2
LP0 Freq16 to 25088 Hz
LP1 Freq16 to 25088 HzLP5 Freq16 to 25088 Hz
LP2 Freq16 to 25088 HzLP6 Freq16 to 25088 Hz
LP3 Freq16 to 25088 HzLP7 Freq16 to 25088 Hz
LP4 Freq16 to 25088 HzLP8 Freq16 to 25088 Hz
Algorithm Reference-99
n
FXAlg #727: PolyDistort + EQ
PAGE 4
Bass Gain-79.0 to 24.0 dBTreb Gain-79.0 to 24.0 dB
Bass Freq16 to 25088 HzTreb Freq16 to 25088 Hz
Mid1 Gain-79.0 to 24.0 dBMid2 Gain-79.0 to 24.0 dB
Mid1 Freq16 to 25088 HzMid2 Freq16 to 25088 Hz
Mid1 Width0.010 to 5.000 octMid2 Width0.010 to 5.000 oct
Wet/DryThis is a simple mix of the distorted signal relative to the dry undistorted input signal.
Out GainThe 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 DriveApplies gain to the input prior to distortion. It is the basic Òdistortion driveÓ 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
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 ßat right before it
goes into clipping.
LP n FreqThese are the one-pole lopass controls. LP0 Freq handles the initial lopass prior to the
Þrst distortion stage. The other lopass controls follow their respective distortion stages.
With all lopasses out of the circuit (set to the highest frequency), the sound tends to be
too bright and raspy. With less distortion drive, less Þltering is needed. If you turn off a
distortion stage (set to 0%), you should turn of the lopass Þlter by setting it to the highest
frequency.
Bass GainThe amount of boost or cut that the bass-shelving Þlter 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 speciÞed frequency. Negative
values cut the bass signal below the speciÞed frequency.
Bass FreqThe center frequency of the bass shelving Þlter in intervals of one semitone.
Treb GainThe amount of boost or cut that the treble-shelving Þlter 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 speciÞed frequency. Negative
values cut the treble signal above the speciÞed frequency.
Treb FreqThe center frequency of the treble shelving Þlter in intervals of one semitone.
Mid GainThe amount of boost or cut that the mid parametric Þlter should apply in dB. Every
increase of 6 dB approximately doubles the amplitude of the signal. Positive values
boost the signal at the speciÞed frequency. Negative values cut the signal at the
speciÞed frequency.
Mid FreqThe center frequency of the mid parametric Þlter in intervals of one semitone. The boost
or cut will be at a maximum at this frequency.
Mid WidThe bandwidth of the mid parametric Þlter may be adjusted. The bandwidth is speciÞed
in octaves. Small values result in a very narrow Þlter response. Large values result in a
very broad response.
Mono distortion circuits in combination with moving delays,
and a stereo chorus or stereo flange
Allocation Units:3 each
Each of these four algorithms offer 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 re-creating 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. The
TubeAmp faithfully models the response and smooth distortion caused by overloading a vacuum tube circuit.
PolyAmp is closely related to the PolyDistort algorithm offering a brighter sound quality with more sustain. 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.
Following the distortion drive element is a cabinet simulator. The cabinet simulator models the responses of various
types of micÕd guitar cabinets. The preset can be selected using the Cab Preset parameter. The following is the list
of cabinet presets and their descriptions:
Basic
Lead 12
2x12
Open 12
Open 10
4x12
Hot 2x12
Hot 12
Flat response from 100 Hz to 4 kHz with 4th order roll-offs (24dB/oct) on each end
Open back hard American type with one 12Ó driver
Closed back classic American type with two 12Ó drivers
Open back classic American type with one 12Ó driver
Open back classic American type with one 10Ó driver
Closed back British type with four 12Ó drivers
Closed back hot rod type with two 12Ó drivers
Open back hot rod type with one 12Ó driver
Algorithm Reference-101
Tube Amp/Distortion/Delay Combinations
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.
At the end of the chain is either a chorus or a flange controlled by parameters beginning with ÒChÓ or ÒFlÓ
respectively. The chorus and flange have mono inputs and stereo outputs. Each is a standard KDFX single tap dual
channel chorus (see FXAlg #150) or flange (see FXAlg#154) with independent controls for left and right channels
found in many other 1-PAU 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. Left/right balance specifically for the chorus or flange can
be adjusted with the Out Bal control.
In addition, there is a generic monaural moving delay segment. Its parameters begin with the letters ÒMDÓ. 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 Quad MovDelay algorithms (FXAlgs #715-716).
As implemented in these four algorithms, it 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.
L Input
R Input
Chorus
Input Bal
Blend
Ch Wet/Dry
Moving
Delay
Pan
Ch Out Bal
MD Wet/Dry
Tone
Out Gain
Tube
Amp
Cab
Simulator
L Output
R Output
Algorithm Reference-102
TubeAmp<>MD>Chor with moving delay inserted PreDist
Tube Amp/Distortion/Delay Combinations
MD Wet/Dry
L Input
Input Bal
Blend
Tone
Tube
Amp
Moving
Delay
R Input
Ch Wet/Dry
Pan
Chorus
Ch Out Bal
Out Gain
TubeAmp<>MD>Chor with moving delay inserted PostDist
Parameters:
PAGE 1
In/OutIn or OutOut GainOff; -79.0 to 24.0 dB
Input Bal-100 to 100%
Cab
Simulator
L Output
R Output
PAGE 2 (TubeAmp algs)
Tube DriveOff; -79.0 to 60.0 dB
Warmth16 to 25088 Hz
Bass Tone0.0 to 10.0
Mid Tone0.0 to 10.0Cab In/OutIn or Out
Treb Tone0.0 to 10.0Cab PresetOpen 12, ...
Cab Pan-100 to 100%
PAGE 2 (PolyAmp algs)
Poly Drive0.0 to 60.0 dB
Warmth16 to 25088 Hz
Bass Tone0.0 to 10.0
Mid Tone0.0 to 10.0Cab In/OutIn or Out
Treb Tone0.0 to 10.0Cab PresetOpen 12, ...
Cab Pan-100 to 100%
Algorithm Reference-103
Tube Amp/Distortion/Delay Combinations
PAGE 3
MD InsertPost Dist, ...MD Delay0.0 to 1000.0 ms
MD Wet/Dry0 to 100%MD LFOModeFlange, ...
MD LFORate0.00 to 10.00 Hz
MD LFODpth0.0 to 200.0%
MD Fdbk-100 to 100%
PAGE 4 (Chorus algs)
Ch Rate L0.01 to 10.00 HzCh Rate R0.01 to 10.00 Hz
Ch Depth L0.0 to 100.0 ctsCh Depth R0.0 to 100.0 cts
Ch Delay L0 to 720 msCh Delay R0 to 720 ms
Ch Fdbk L-100 to 100%Ch Fdbk R-100 to 100%
Ch PtchEnvTriangle or Trapzoid
ChWet/Dry0 to 100%Ch Out Bal-100 to 100%
PAGE 4 (Flange algs)
Fl Rate0 to 32 btsFl TempoSystem; 1 to 255 BPM
Fl Xcurs L0 to 230 msFl Xcurs R0 to 230 ms
Fl Delay L0 to 230 msFl Delay R0 to 230 ms
Fl Fdbk L-100 to 100%Fl Fdbk R-100 to 100%
Fl Phase L0 to 360 degFl Phase R0 to 360 deg
Fl Wet/Dry0 to 100%Fl Out Bal-100 to 100%
In/Out
Toggles the entire effect on or off. When off, the input signal is passed.
Input BalAdjusts 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, while positive values blend increasing amounts
of right.
Out GainThe overall gain or amplitude at the output of the effect.
Bass Tone, Mid Tone, Treb Tone Adjusts the 3 bands of the 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, Poly DriveAdjusts the gain into each distortion circuit. Higher values produce more distortion.
WarmthAdjusts a 1-pole (6dB/oct) lopass Þlter applied after distortion.
Cab In/OutTurns the cabinet simulator on or off.
Cab PresetSelects the preset cabinet type.
Cab PanAdjusts the output pan position of the cabinet simulator signal that is mixed at the
output of the algorithm. Note that when Ch Wet/Dry or 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 InsertSelects 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, and Bypass takes it completely out of the path.
MD Wet/DryAdjusts the ratio of the moving delay output mixed with its own input to be fed to the
next effect in the chain.
Algorithm Reference-104
Tube Amp/Distortion/Delay Combinations
MD DelayAdjusts the delay time for the moving delay circuit, which is the center of LFO
excursion.
MD LFOModeAdjusts the LFO excursion type. In Flange mode, the LFO is optimized for ßange 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 LFORateAdjusts the LFO speed for the moving delay circuit.
MD LFODpthIn 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 FdbkAdjusts 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, Fl Wet/DryAdjusts the ratio of ßange or 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 Þnal chorus or ßange. 100% feeds only the ßange or chorus to the output.
Ch Out Bal, Fl Out BalAdjusts the left/right output balance of the chorus or ßange signal. Negative values
balance toward the left while positive values balance toward the right.
Vibrato/chorus, through optional distortion, into rotating speaker
Allocation Units: 2 for VibChor+Rotor 2; 4 for VibChor+Rotor 4
The VibChor+Rotor algorithms contain multiple effects designed for the Hammond B3¨ emulation (KB3 mode).
These effects are the Hammond¨ vibrato/chorus, amplifier distortion, and rotating speaker (Leslie¨). Each of these
effects may be turned off or bypassed, or the entire algorithm may be bypassed.
L Input
R Output
Vibrato/
Chorus
Distortion
(Optional)
Pan
Rotator
Pan
Mic Levels
Pan
Rotator
Pan
Cabinet
Cabinet
L Output
Out Gain
R Output
Block diagram of VibChor+Rotor
The first effect in the chain is the Hammond vibrato/chorus algorithm. The vibrato/chorus has six settings which
are the same as those used in the Hammond B3: three vibrato (V1, V2, V3) and three chorus (C1, C2, C3) settings. In
VibChor+Rotor 4, the vibrato chorus has been carefully modeled after the electro-mechanical vibrato/chorus in the
B3. The vibrato/chorus in VibChor+Rotor 2 uses 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 VibChor+Rotor 4 vibrato/chorus.
In VibChor+Rotor 4 an amplifier distortion algorithm follows the vibrato/chorus. See the section in this book on
FXAlg #724 for more information about the distortion algorithm.
Finally, the signal passes through a rotating speaker routine. 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 pair of virtual microphones which can be
positioned at varying positions (angles) around the rotors. An angle of 0° is loosely defined as the front. You can
also control the levels and left-right panning of each virtual microphone. The signal is then passed through a final
lowpass filter to simulate the band-limiting effect of the speaker cabinet.
For the rotating speakers, you can control the crossover 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 rate, the effective driver size and tremolo can be set. The
rotation rate sets how fast the rotating speaker is spinning. The effective driver size is the radius of the path followed
by the speaker relative to its center of rotation. 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. 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 ÒwidthÓ. 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. Note that it can take up to one full speaker rotation before you hear changes to
tremolo when parameter values are changed. Negative microphone angles take a longer time to respond to tremolo
changes than positive microphone angles.
(i)(ii)
Acoustic beams for (i) low frequency driver and (ii) high frequency driver.
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.
Lo GainOff, -79.0 to 24.0 dBHi GainOff, -79.0 to 24.0 dB
Lo Rate-10.00 to 10.00 HzHi Rate-10.00 to 10.00 Hz
Lo Size0 to 250 mmHi Size0 to 250 mm
Lo Trem0 to 100%Hi Trem0 to 100%
Lo Beam W45.0 to 360.0 degHi Beam W45.0 to 360.0 deg
PAGE 3
LoMicA Pos-180.0 to 180.0 degLoMicB Pos-180.0 to 180.0 deg
LoMicA Lvl0 to 100%LoMicB Lvl0 to 100%
LoMicA Pan-100 to 100%LoMicB Pan-100 to 100%
HiMicA Pos-180.0 to 180.0 degHiMicB Pos-180.0 to 180.0 deg
HiMicA Lvl0 to 100%HiMicB Lvl0 to 100%
HiMicA Pan-100 to 100%HiMicB Pan-100 to 100%
PAGE 4
LoResonate0 to 100%HiResonate0 to 100%
Lo Res Dly10 to 2550 sampHi Res Dly10 to 2550 samp
LoResXcurs0 to 510 sampHiResXcurs0 to 510 samp
ResH/LPhase 0.0 to 360.0 deg
In/Out
Out GainThe overall gain or amplitude at the output of the effect. For distortion, it is often
VibChInOutWhen set to ÒInÓ the vibrato/chorus is active; when set to ÒOutÓ the vibrato/chorus is
Vib/ChorThis control sets the Hammond B3¨ vibrato/chorus. There are six settings for this effect:
Roto InOutWhen set to ÒInÓ the rotary speaker is active; when set to ÒOutÓ the rotary speaker is
Dist DriveApplies a boost to the input signal to overdrive the distortion algorithm. When
When set to ÒInÓ, the algorithm is active; when set to ÒOutÓ the algorithm is bypassed.
necessary to turn the output gain down as the distortion drive is turned up.
bypassed.
three vibratos ÒV1Ó, ÒV2Ó, ÒV3Ó, and three choruses ÒC1Ó, ÒC2Ó, ÒC3Ó
bypassed.
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. [VibChor+Rotor 4 only]
DistWarmthA lowpass Þlter in the distortion control path. This Þlter may be used to reduce some of
the harshness of some distortion settings without reducing the bandwidth of the signal.
[VibChor+Rotor 4 only]
Cabinet LPA lowpass Þlter to simulate the band-limiting of a speaker cabinet. The Þlter controls the
upper frequency limit of the output.
XoverThe frequency at which high and low frequency bands are split and sent to separate
rotating drivers.
Lo GainThe gain or amplitude of the signal passing through the rotating woofer (low-frequency
driver.
Lo RateThe rotation rate of the rotating woofer (low-frequency driver). The woofer can rotate
clockwise or counter-clockwise. The direction of rotation depends on the sign of the rate
parameter. Assuming microphone angles are set toward the front (between -90° and 90°)
and microphones at positive angles are panned to the right (positive pan values), then
positive rates correspond to clockwise rotation when viewed from the top.
Lo Size
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 TremControls the depth of tremolo of the low frequency signal. Expressed as a percentage of
full scale tremolo.
Lo Beam WThe 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 GainThe gain or amplitude of the signal passing through the rotating tweeter (high-
frequency driver.
Hi RateThe rotation rate of the rotating tweeter (high-frequency driver). The tweeter can rotate
clockwise or counter-clockwise. The direction of rotation depends on the sign of the rate
parameter. Assuming microphone angles are set toward the front (between -90° and 90°)
and microphones at positive angles are panned to the right (positive pan values), then
positive rates correspond to clockwise rotation when viewed from the top.
Hi SizeThe 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 TremControls the depth of tremolo of the high frequency signal. Expressed as a percentage of
full scale tremolo.
Hi Beam WThe 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).
Mic PosThe angle of the virtual microphones in degrees from the ÒfrontÓ of the rotating speaker.
This parameter is not well suited to modulation because adjustments to it will result in
large sample skips (audible as clicks when signal is passing through the effect). There are
four of these parameters to include 2 pairs (A and B) for high and low frequency drivers.
Mic LvlThe level of the virtual microphone signal being sent to the output. There are four of
these parameters to include 2 pairs (A and B) for high and low frequency drivers.
Mic PanLeft-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.
LoResonateA 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 DlyThe number of samples of delay in the resonator circuit in addition to the rotation
excursion delay. This is for the low frequency signal path.
LoResXcursThe 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.
HiResonateA simulation of cabinet resonant modes expressed as a percentage. For realism, you
should use very low settings. This is for the high frequency signal path.
Hi Res DlyThe number of samples of delay in the resonator circuit in addition to the rotation
excursion delay. This is for the high frequency signal path.
HiResXcursThe 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/LPhsThis 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.
Algorithm Reference-110
FXAlg #734: Distort + Rotary
FXAlg #734: Distort + Rotary
Small distortion followed by rotary speaker effect
Allocation Units: 2
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. 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.
L Input
Distortion
R Input
Block diagram of Distort + Rotary
The first part of Distort + Rotary is a distortion algorithm. See the section of this book on FXAlg #723 for details.
Next the signal passes through a rotating speaker routine. See the section of this book on FXAlg #733 for details.
Rotator
L Output
Out GainCabinet
R Output
Rotator
Parameters:
PAGE 1
In/OutIn or OutOut GainOff, -79.0 to 24.0 dB
Cabinet HP16 to 25088 HzDist Drive0 to 96 dB
Cabinet LP16 to 25088 HzDistWarmth16 to 25088 Hz
PAGE 2
Xover16 to 25088 HzMic Angle0.0 to 360.0 deg
Lo GainOff, -79.0 to 24.0 dBHi GainOff, -79.0 to 24.0 dB
Lo Rate-10.00 to 10.00 HzHi Rate-10.00 to 10.00 Hz
Lo Size0 to 250 mmHi Size0 to 250 mm
Lo Trem0 to 100%Hi Trem0 to 100%
PAGE 3
ResH/LPhs0.0 to 360.0 deg
LoResonate0 to 100%HiResonate0 to 100%
Lo Res Dly10 to 2550 sampHi Res Dly10 to 2550 samp
LoResXcurs0 to 510 sampHiResXcurs0 to 510 samp
Algorithm Reference-111
FXAlg #734: Distort + Rotary
In/OutWhen set to ÒInÓ, the algorithm is active; when set to ÒOffÓ the algorithm is bypassed.
Out GainThe 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 DriveApplies 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.
DistWarmthA lowpass Þlter in the distortion control path. This Þlter may be used to reduce some of
the harshness of some distortion settings without reducing the bandwidth of the signal.
Cabinet HPA hipass Þlter to simulate the band-limiting of a speaker cabinet. The Þlter controls the
lower frequency limit of the output.
Cabinet LPA lowpass Þlter to simulate the band-limiting of a speaker cabinet. The Þlter controls the
upper frequency limit of the output.
XoverThe frequency at which high and low frequency bands are split and sent to separate
rotating drivers.
For details on the rest of the parameters see the previous section (FXAlg #733) of this book.
Algorithm Reference-112
FXAlg #735/6: KB3 FX
FXAlg #735/6: KB3 FX
Vibrato/chorus into distortion into rotating speaker into cabinet
Allocation Units: 7 for full working effect (4 for KB3 FXBus, 3 for KB3 AuxFX)
The KB3 FXBus and KB3 AuxFX algorithms contain multiple effects designed for the Hammond B3 emulation (KB3
mode). For correct operation, both effects must be running at the same time, with the output of KB3 FXBus feeding
the input of KB3 AuxFX. The two algorithms work as one algorithm which use all the available KDFX resources.
While the input to KB3 FXBus is stereo (which gets summed to mono) and the output from KB3 AuxFX is stereo,
the signals between the two algorithms are the low frequency (left) and high frequency (right) signal bands used to
drive the lower and upper rotary speakers. It is possible to run these two algorithms as independent effects, but it
is recommended.
These effects are the Hammond vibrato/chorus, amplifier distortion, and rotating speaker (Leslie) emulations.
Each of these effects may be turned off or bypassed, or the entire algorithm may be bypassed. To bypass the rotary,
the switches in both KB3 FXBus and KB3 AuxFX must be set to ÒOutÓ.
Hi Gain
L Input
R Input
Vibrato/
Chorus
Distortion
Cabinet
Filter
Lo Gain
L Output
R Output
Block diagram of KB3 FXBus
Pan
L Input
R Input
Rotator
Rotator
L Output
Pan
Mic LevelsOut Gain
Pan
R Output
Pan
Block diagram of KB3 AuxFX
The first effect in the chain is the Hammond vibrato/chorus algorithm. The vibrato/chorus has six settings which
are the same as those used in the Hammond B3: three vibrato (V1, V2, V3) and three chorus (C1, C2, C3) settings.
The vibrato chorus has been carefully modeled after the electro-mechanical vibrato/chorus in the B3.
An amplifier distortion algorithm follows the vibrato/chorus. For details, see the section in this book on FXAlg
#723.
The distorted signal is next passed to a cabinet emulation filter and a pair of crossover filters for band splitting. The
measurements of a real Leslie¨ speaker was used in the design of these filters. Default parameter values reflect
these measurements, but you may alter them if you like. The Lo HP parameter controls a hipass filter which defines
the lowest frequency to pass through the speaker. Likewise the Hi LP parameter is a lowpass filter controlling the
Algorithm Reference-113
FXAlg #735/6: KB3 FX
highest frequency. The crossover filters for the lower and upper drivers may be set independently. A small amount
of overlap seems to work well. The gains of the high and low band signals may also be separately controlled.
At this point KB3 FXBus has finished its processing and passes the high and low signals to the KB3 AuxFX algorithm
which contains the rotating-speaker routine. See the section in this book on FXAlg #733 for details.
Parameters (KB3 FXBus):
PAGE 1
In/OutIn or OutOut GainOff, -79.0 to 24.0 dB
VibChInOutIn or OutDist Drive0 to 96 dB
Vib/ChorV1DistWarmth16 to 25088 Hz
PAGE 2
RotoInOutIn or Out
Lo GainOff, -79.0 to 24.0 dBHi GainOff, -79.0 to 24.0 dB
Lo Xover16 to 25088 HzHi Xover16 to 25088 Hz
Lo HP16 to 25088 HzHi LP16 to 25088 Hz
In/OutWhen set to ÒInÓ, the algorithm is active; when set to ÒOutÓ the algorithm is bypassed.
For the entire algorithm to be active, KB3 AuxFX must also be active.
Out GainThe 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.
VibChInOutWhen set to ÒInÓ the vibrato/chorus is active; when set to ÒOutÓ the vibrato/chorus is
bypassed.
Vib/ChorThis control sets the Hammond B3¨ vibrato/chorus. There are six settings for this effect:
three vibratos ÒV1Ó, ÒV2Ó, ÒV3Ó, and three choruses ÒC1Ó, ÒC2Ó, ÒC3Ó
Roto InOutWhen set to ÒInÓ the rotary speaker is active; when set to ÒOutÓ the rotary speaker is
bypassed. By bypassing the rotary effect in KB3 FXBus, only the crossover Þlters are
bypassed. You must also bypass KB3 AuxFX to completely bypass the rotary speakers.
Likewise, for the entire rotary to be active, KB3 AuxFX must also be active.
Dist DriveApplies 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.
WarmthA lowpass Þlter in the distortion control path. This Þlter may be used to reduce some of
the harshness of some distortion settings without reducing the bandwidth of the signal.
Lo GainThe gain or amplitude of the signal passing through the rotating woofer (low frequency
driver. The control is also available in KB3 AuxFX.
Lo XoverThe crossover frequency for the low frequency driver. Lo Xover controls a lowpass Þlter.
Lo HPA hipass Þlter to simulate the band-limiting of a speaker cabinet. The Þlter controls the
lower frequency limit of the output.
Hi GainThe gain or amplitude of the signal passing through the rotating tweeter (high frequency
driver. The control is also available in KB3 AuxFX.
Hi XoverThe crossover frequency for the high frequency driver. Hi Xover controls a hipass Þlter.
Hi LPA lowpass Þlter to simulate the band-limiting of a speaker cabinet. The Þlter controls the
upper frequency limit of the output.
Algorithm Reference-114
Parameters (KB3 AuxFX):
PAGE 1
In/OutIn or OutOut GainOff, -79.0 to 24.0 dB
PAGE 2
Lo GainOff, -79.0 to 24.0 dBHi GainOff, -79.0 to 24.0 dB
Lo Rate-10.00 to 10.00 HzHi Rate-10.00 to 10.00 Hz
Lo Size0 to 250 mmHi Size0 to 250 mm
Lo Trem0 to 100%Hi Trem0 to 100%
Lo Beam W45.0 to 360.0 degHi Beam W45.0 to 360.0 deg
PAGE 3
LoMicA Pos-180.0 to 180.0 degLoMicB Pos-180.0 to 180.0 deg
LoMicA Lvl0 to 100%LoMicB Lvl0 to 100%
LoMicA Pan-100 to 100%LoMicB Pan-100 to 100%
HiMicA Pos-180.0 to 180.0 degHiMicB Pos-180.0 to 180.0 deg
HiMicA Lvl0 to 100%HiMicB Lvl0 to 100%
HiMicA Pan-100 to 100%HiMicB Pan-100 to 100%
FXAlg #735/6: KB3 FX
PAGE 4
LoResonate0 to 100%HiResonate0 to 100%
Lo Res Dly10 to 2550 sampHi Res Dly10 to 2550 samp
LoResXcurs0 to 510 sampHiResXcurs0 to 510 samp
ResH/LPhs0.0 to 360.0 deg
In/OutWhen set to ÒInÓ, the algorithm is active; when set to ÒOffÓ the algorithm is bypassed.
For the entire algorithm to be active, KB3 FXBus must also be active with its Roto InOut
parameter set to ÒInÓ. To completely bypass the rotary, one or both of the In/Out or Roto
InOut parameters in KB3 FXBus must also be bypassed.
Out GainThe overall gain or amplitude at the output of the effect.
Lo GainThe gain or amplitude of the signal passing through the rotating woofer (low frequency
driver. The control is also available in KB3 FXBus.
Hi GainThe gain or amplitude of the signal passing through the rotating tweeter (high frequency
driver. The control is also available in KB3 FXBus.
For details on the rest of the parameters see the section of this book on FXAlg #733.
Algorithm Reference-115
FXAlg #900: Env Follow Filt
FXAlg #900: Env Follow Filt
Envelope-following stereo 2-pole resonant filter
Allocation Units: 2
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 one of low pass (i), highpass
(ii), band pass (iii), or notch (iv).
(i)(ii)
(iii)(iv)
Resonant Filter Types: (i) lowpass, (ii) highpass, (iii) bandpass, and (iv) notch.
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 16 Hz. Note
that 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.
Envelope
Follower
L Input
Resonant Filter
R Input
Block diagram of envelope-following 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.
L Input
R Input
Algorithm Reference-116
FXAlg #900: Env Follow Filt
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 lopass filter which can extend the attack and release times of the
envelope follower. A level meter with a threshold marker is provided.
Parameters:
PAGE 1
Wet/Dry0 to 100%wetOut GainOff, -79.0 to 24.0 dB
FilterTypeLowpassMin Freq16 to 8372 Hz
FFreq Sweep-100 to 100%
0Hz 2k 4k 6kResonance0 to 50 dB
PAGE 2
Threshold-79.0 to 0.0 dBAtk Rate0.0 to 300.0 dB/s
Rel Rate0.0 to 300.0 dB/s
Smth Rate0.0 to 300.0 dB/s
E
-dB 60 40 * 16 * 8 4 0
Wet/DryThe amount of modulated (wet) signal relative to unaffected (dry) signal as a percent.
Out GainThe overall gain or amplitude at the output of the effect.
FilterTypeThe type of resonant Þlter to be used. May be one of ÒLowpassÓ, ÒHighpassÓ,
ÒBandpassÓ, or ÒNotchÓ.
Min FreqThe base frequency of the resonant Þlter. The Þlter resonant frequency is set to the Min
Freq while the signal envelope is at its minimum level or below the threshold.
Freq SweepHow far the Þlter frequency can change from the Min Freq setting as the envelope
amplitude changes. Freq Sweep may be positive or negative so the Þlter frequency can
rise above or fall below the Min Freq setting.
ResonanceThe resonance level of the resonant Þlter. Resonance sets the level of the resonant peak.
In the notch Þlter, this sets the amount of cut, so 0 dB provides the highest, widest notch,
and higher levels make the notch increasingly narrower and shallower.
ThresholdRepresents the level above which signal envelope must rise before the Þlter begins to
follow the envelope. Below the threshold, the Þlter resonant frequency will remain at the
Min frequency.
Atk RateAdjusts the upward slew rate of the envelope detector.
Rel RateAdjusts the downward slew rate of the envelope detector.
Smth RateSmooths 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.
The triggered envelope-following filter is used to produce a filter sweep when the input rises above a trigger level.
The triggered envelope-following filter is a stereo resonant filter with the resonant frequency controlled by a
triggered envelope follower. The filter type is selectable and may be one of low pass (i), high pass (ii), band pass
(iii), or notch (iv). See the previous section of this book, FXAlg #900, for diagrams of the filter actions.
Envelope
Follower
L Input
R Input
Block diagram of Triggered Envelope Filter
The resonant frequency of the filter will remain at the minimum frequency (Min Freq) prior to being triggered. On
a trigger, the resonant frequency will sweep to the maximum frequency (Max Freq). The minimum and maximum
frequencies may be set to any combination of frequencies between 16 and 8372 Hz. Note that 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. 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.
When the input signal envelope rises above the trigger level, an envelope generator is started which has an instant
attack and exponential decay. The generated attack may be lengthened with the smoothing parameter. The
smoothing parameter can also lengthen the generated decay if the smoothing rate is lower than the decay. The
generated envelope is then used to control the resonant frequency of the filter.
Trigger
Generator
Triggered
Envelope
Generator
L Input
Resonant Filter
R Input
The time constant of the envelope follower may be set (Env Rate) as well as the decay rate of the generated envelope
(Rel Rate). After the detected envelope rises above the Trigger level, a trigger event cannot occur again until the
signal drops below the Retrigger level. In general, Retrigger should be set lower than the Trigger level. A level meter
with a trigger marker is provided.
Parameters:
PAGE 1
Wet/Dry0 to 100%wetOut GainOff, -79.0 to 24.0 dB
FilterTypeLowpassMin Freq16 to 8372 Hz
FMax Freq16 to 8372 Hz
0Hz 2k 4k 6kResonance0 to 50 dB
Algorithm Reference-118
FXAlg #901: TrigEnvelopeFilt
PAGE 2
Trigger-79.0 to 0.0 dBEnv Rate0.0 to 300.0 dB/s
Retrigger-79.0 to 0.0 dBRel Rate0.0 to 300.0 dB/s
Smth Rate0.0 to 300.0 dB/s
E
-dB 60 40 * 16 * 8 4 0
Wet/DryThe amount of modulated (wet) signal relative to unaffected (dry) signal as a percent.
Out GainThe overall gain or amplitude at the output of the effect.
FilterTypeThe type of resonant Þlter to be used. May be one of ÒLowpassÓ, ÒHighpassÓ,
ÒBandpassÓ, or ÒNotchÓ.
Min FreqThe base frequency of the resonant Þlter. The Þlter resonant frequency is set to the base
frequency while the signal envelope is below the threshold.
Max FreqThe frequency of the resonant Þlter that can be reached when the envelope follower
output reaches full-scale. The resonant frequency will sweep with the envelope from the
base frequency, approaching the limit frequency with rising amplitudes.
ResonanceThe resonance level of the resonant Þlter. Resonance sets the level of the resonant peak.
In the notch Þlter, this sets the amount of cut, so 0 dB provides the highest, widest notch,
and higher levels make the notch increasingly narrower and shallower.
TriggerThe threshold at which the envelope detector triggers in fractions of full scale where 0dB
is full scale.
RetriggerThe threshold at which the envelope detector resets such that it can trigger again in
fractions of full scale where 0dB is full scale. This value is only useful when it is below
the value of Trigger.
Env RateThe envelope detector decay rate which can be used to prevent false triggering. When
the signal envelope falls below the retrigger level, the Þlter can be triggered again when
the signal rises above the trigger level. Since the input signal can ßuctuate rapidly, it is
necessary to adjust the rate at which the signal envelope can fall to the retrigger level.
The rate is provided in decibels per second (dB/s).
Rel RateThe downward slew rate of the triggered envelope generator. The rate is provided in
decibels per second (dB/s).
Smth RateSmooths the output of the envelope generator. Smoothing slows down the envelope
follower and can dominate the release rate if set lower rate than this parameter. You can
use the smoothing rate to lengthen the attack of the generated envelope which would
otherwise have an instant attack. The rate is provided in decibels per second (dB/s)
Algorithm Reference-119
FXAlg #902: LFO Sweep Filter
FXAlg #902: LFO Sweep Filter
LFO-following stereo 2-pole resonant filter
Allocation Units: 2
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 one of low pass (i), high pass (ii), band pass (iii), or notch (iv). See
the section of this book on FXAlg #900 for diagrams of the filter actions.
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 16 and 8372 Hz. Note that 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.
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.
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.
The LFO may be configured to one of a variety of wave shapes. Available shapes are Sine, Saw+, Saw-, Pulse and
Tri. Sine is simply a sinusoid waveform. Tri produces a triangular waveform, and Pulse produces a series of square
pulses where the pulse width can be adjusted with the ÒLFO PlsWidÓ parameter. When pulse width is 50%, the
signal is a square wave. The ÒLFO PlsWidÓ parameter is only active when the Pulse waveform is selected. Saw+ and
Saw- produce rising and falling sawtooth waveforms. 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 sawtooths
develops a more gradual slope with smoothing, ending up as triangle waves when set to 100% smoothing.
PulseWidth
SineSaw+Saw-PulseTri
Configurable Wave Shapes
Parameters:
PAGE 1
Wet/Dry0 to 100%wetOut GainOff, -79.0 to 24.0 dB
LFO TempoSystem, 1 to 255 BPM LFO ShapeSine
LFO Period1/24 to 32 btsLFO PlsWid0 to 100%
LFO Smooth0 to 100%
Algorithm Reference-120
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