Page 1 Algorithm Reference October 1, 2001 ©2001 All rights reserved. Kurzweil is a product line of Young Chang Co. Kurzweil, KSP8, KDFX, LaserVerb , and Pitcher are trademarks of Young Chang Co. All other products and brand names are trademarks or registered trademarks of their respective companies.
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Page 4: Table Of Contents
KSP8 Algorithm Reference Contents • KSP8 Algorithms Listed by ID ....5 • KSP8 Algorithms Listed by Name ....9 •...
Page 5: Ksp8 Algorithms Listed By Id
KSP8 Algorithms Listed by ID Name PAUs Page Name PAUs Page Stereo Algorithms Chorus 1 Chorus 2 MiniVerb Dual Chorus 1 Dual MiniVerb Dual Chorus 2 Gated MiniVerb Flanger 1 Classic Place Flanger 2 Classic Verb LFO Phaser TQ Place...
Page 6 Name PAUs Page Name PAUs Page Mono Distortion 3 Band EQ MonoDistort+Cab 5 Band EQ MonoDistort + EQ Graphic EQ PolyDistort + EQ Dual Graphic EQ StereoDistort+EQ Dual 5 Band EQ Subtle Distort Env Follow Filt Super Shaper TrigEnvelopeFilt 3 Band Shaper LFO Sweep Filter Quantize+Alias Resonant Filter...
Page 7 Name PAUs Page Chorus+Delay Chorus+4Tap Chorus<>4Tap Chor+Dly+Reverb Chorus<>Reverb Chorus<>LasrDly St Chorus+Delay St Chorus+4Tap St Chor+Dly+Rvrb Pitcher+Chor+Dly Pitch+StChor+Dly MonoPitcher+Chor MonoPitch+StChor Chorus+Delay ms Chorus+4Tap ms Chorus<>4Tap ms Chor+Dly+Rvrb ms Chor<>LasrDly ms St Chor+Delay ms St Chor+4Tap ms StCh+Dly+Rvrb ms Ptch+Chor+Dly ms Ptch+StCh+Dly ms Flange+Delay Flange+4Tap Flange<>4Tap...
Page 8 Name PAUs Page Name PAUs Page Monaural Algorithms Mn SK Compress Mn Expander Mn MiniVerb Mn Gate Mn LaserVerb Mn Comprs/Expand MnGt+Cmp[EQ]+Rvb Mn 3 Band Comprs MnGt+Cmp<>EQ+Rvb Mn Graphic EQ Mn 6-TapDelayBPM Mn 6 Band EQ Mn 6-Tap Delay Mn 3BandEnhancer Mn Spectral 4Tap Mn HF Stimulate1 Mn Complex Echo...
Page 9: Ksp8 Algorithms Listed By Name
KSP8 Algorithms Listed by Name Name PAUs Page Name PAUs Page 2 Band Enhancer Dual AutoPanner 3 Band Compress Dual Chorus 1 3 Band Delay Dual Chorus 2 3 Band Enhancer Dual Comprs SCEQ 3 Band EQ Dual Graphic EQ...
Page 10 Name PAUs Page Name PAUs Page Gt+Tube<>MD+Flan Mn Manual Phaser HardKneeCompress Mn MiniVerb HarmonicSuppress Mn Pitcher HF Stimulate 1 Mn Res Filter HF Stimulate 3 Mn Ring Modulate LaserVerb Mn SingleLFOPhsr LaserVerb Lite Mn SK Compress LasrDly<>Reverb Mn Spectral 4Tap LasrDly<>Rvrb ms Mn Super Shaper LFO Phaser...
Page 11 Name PAUs Page Reverb<>Compress Revrse LaserVerb Ring Modulator Rotor 1 Shaper<>Reverb SingleLFO Phaser SoftKneeCompress Spectral 4-Tap Spectral 6-Tap St Chor+4Tap ms St Chor+Delay ms St Chor+Dly+Rvrb St Chorus+4Tap St Chorus+Delay St Flan+4Tap ms St Flan+Delay ms St Flan+Dly+Rvrb St Flange+4Tap St Flange+Delay StCh+Dly+Rvrb ms Stereo Analyze...
Page 12: Ksp8 Algorithm Specifications
KSP8 Algorithm Speciﬁcations MiniVerbs 1 MiniVerb 2 Dual MiniVerb 600 Mn MiniVerb Versatile, small stereo and dual mono reverbs PAUs: 1 for MiniVerb 2 for Dual MiniVerb MiniVerb is a versatile stereo reverb found in many combination algorithms, but is equally useful on its own because of its small size.
Page 13 L Output L Input MiniVerb Balance R Input MiniVerb Balance R Output Figure 2 Simpliﬁed Block Diagram of Dual 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.
Page 14 Dual MiniVerb Parameters Page 1 L Wet/Dry 0 to 100%wet R Wet/Dry 0 to 100%wet L Out Gain Off, -79.0 to 24.0 dB R Out Gain Off, -79.0 to 24.0 dB L Wet Bal -100 to 100% R Wet Bal -100 to 100% L Dry Pan -100 to 100%...
Page 15 Diff Scale 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 A multiplier which changes the size of the current room.
Page 16 3 Gated MiniVerb A reverb and gate in series PAUs: This algorithm is a small reverb followed by a gate. The main control for the reverb is the Room Type parameter. Room Type changes the structure of the algorithm to simulate many carefully crafted room types and sizes.
Page 17 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. If the gate opened and closed instantaneously, you would hear a large digital click, like a big knife switch was being thrown.
Page 18 if delayed, and thus you can get by with a dryer mix while maintaining the same subjective wet/dry level. Room Type 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.
Page 19: Reverbs
Reverbs 4 Classic Place 5 Classic Verb 6 TQ Place 7 TQ Verb 8 Diffuse Place 9 Diffuse Verb 10 OmniPlace 11 OmniVerb Reverb algorithms PAUs: 2 (Classic) or 3 (others) This set of 2- and 3-PAU algorithms can be divided into 2 groups: Verb and Place. Verb effects allow user- friendly control over medium to large spaces.
Page 20 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.
Page 21 EarRef Lvl Rvrb Time L ER Output LF Mult Absorption DiffAmtScl Late DiffLenScl HF Damping Treble L Input L Pre Dly L Output Diffusor Ambience Out Gain DiffAmtScl DiffLenScl HF Damping Treble R Input R Pre Dly R Output Diffusor Late LF Mult R ER Output...
Page 22 Page 1 (Classic Place) Wet/Dry -100 to 100% Out Gain Off; -79.0 to 24.0 dB Absorption 0 to 100 % EarRef Lvl -100 to 100% HF Damping 0 to 25088 Hz Late Lvl -100 to 100% L Pre Dly 0.0 to 230.0 ms R Pre Dly 0.0 to 230.0 ms Page 2...
Page 23 L Input Reverb Time L ER Output Absorption EarRef Lvl InjBuild DiffAmtScl InjSpread DiffLenScl Inj LP Late Lvl Treble HF Damping LF Mult L Pre Dly Injector L Output Diffuser Gain Ambience DiffAmtScl DiffLenScl Inj LP Late Lvl Treble Gain R Pre Dly Injector R Output...
Page 24 Parameters for TQ Verb and TQ Place: Page 1 (TQ Verb) Wet/Dry -100 to 100% Out Gain Off; -79.0 to 24.0 dB Rvrb Time 0.00 to 60.00 s EarRef Lvl -100 to 100% HF Damping 0 to 25088 Hz Late Lvl -100 to 100% L Pre Dly 0.0 to 230.0 ms...
Page 25 Diffuse Verb and Diffuse Place Diffuse reverbs are 3-PAU algorithms and are characterized as such because of the initial burst of diffusion inherent in the onset of the reverb. The diffusion consists of an input diffuser, ambience generator with a lopass ﬁlter, low shelving ﬁlter, and LFO moving delays, and predelay.
Page 26 Page 2 (Diffuse Verb) Room Type Hall1, ... DiffExtent 1 to 7 x Size Scale 0.01 to 2.50x Diff Cross -100 to 100 % InﬁnDecay On or Off DiffAmtScl 0.00 to 2.00 x DiffLenScl 0.01 to 2.50 x LF Split 8 to 25088 Hz LFO Rate 0.01 to 10.00 Hz...
Page 27 L Input Reverb Time Absorption Inj Build Inj Spread DiffAmtScl Inj Skew Treble Bass DiffLenScl Lopass HF Damping LF Mult Injector L Pre Dly L Output Diffuser Gain Ambience DiffAmtScl Treble Bass DiffLenScl Lopass Gain Injector R Pre Dly R Output Diffuser HF Damping LF Mult...
Page 28 Page 2 (OmniVerb) Room Type Hall1, ... Expanse -100 to 100 % Size Scale 0.00 to 2.50x InﬁnDecay On or Off DiffAmtScl 0.00 to 2.00 x DiffLenScl 0.00 to 4.50 x LF Split 8 to 25088 Hz LFO Rate 0.01 to 10.00 Hz LF Time 0.50 to 1.50 x LFO Depth...
Page 29 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.
Page 30 DiffExtent The onset diffusion duration. Higher values create longer diffuse bursts at the onset of the reverb. Diff Cross 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.
Page 31 E HF Damp The cutoff frequency of a 1 pole (6dB/oct) lowpass ﬁlter applied to the early reﬂection feedback signal. E PreDlyL, 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.
Page 32 12 Panaural Room Room reverberation algorithm PAUs: The Panaural Room reverberation is implemented using a special network arrangement of many delay lines that guarantees colorless sound. 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.
Page 33 Parameters Page 1 Wet/Dry 0 to 100%wet Out Gain Off, -79.0 to 24.0 Room Size 1.0 to 16.0 m Pre Dly 0 to 500 ms Decay Time 0.5 to 100.0 s HF Damping 8 to 25088 Hz Page 2 Bass Gain -15 to 15 dB Build Time 0 to 500 ms...
Page 34 density reverberation, and for extension of the build up period, use a setting of 50%. For an almost reverse reverberation, set Build Env to 100%. You can think of Build Env 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.
Page 35 13 Stereo Hall A stereo hall reverberation algorithm. PAUs: The Stereo Hall reverberation is implemented using a special arrangement of allpass networks and delay lines which reduces coloration and increases density. 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.
Page 36 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%.
Page 37 Pre Dly 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 Similar to predelay, but more complex, larger values of BuildTime slow down the building up of reverberation and can extend the build up process.
Page 38 14 Grand Plate A plate reverberation algorithm. PAUs: This algorithm emulates an EMT 140 steel plate reverberator. 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.
Page 39 Parameters Page 1 Wet/Dry 0 to 100%wet Out Gain Off, -79.0 to 24.0 dB Room Size 1.00 to 4.00 m Pre Dly 0 to 500 ms Decay Time 0.2 to 5.0 s HF Damping 8 to 25088 Hz LF Damping 1 to 294 Hz Page 2 Lowpass...
Page 40 15 Finite Verb Reverse reverberation algorithm. PAUs: 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.
Page 41 Page 3 Early Bass -15 to 15 dB Early Damp 8 to 25088 Hz Mid Bass -15 to 15 dB Mid Damp 8 to 25088 Hz Late Bass -15 to 15 dB Late Damp 8 to 25088 Hz Wet/Dry sets the relative amount of wet signal and dry signal. The wet signal Wet/Dry 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).
Page 42: Combination Reverbs
Combination Reverbs 50 Reverb+Compress 51 Reverb<>Compress A reverb and compressor in series. PAUs: 3 for Reverb<>Compress; 2 for Reverb+Compress Reverb<>Compress is conﬁgurable with the A->B cfg parameter as a reverb followed by a compressor Rvb->Cmp, or as a compressor followed by a reverb Cmp->Rvb. Reverb+Compress is conﬁgured only as a reverb followed by a compressor.
Page 43 To determine how much to compress the signal, the compressor must measure the signal level. Since musical signal levels will change over time, the compression amounts must change as well. You can control the rate at which compression changes in response to changing signal levels with the attack and release time controls.
Page 44 Page 3 Comp Atk 0.0 to 228.0 ms Comp Ratio 1.0:1 to 100.0:1, Inf:1 Comp Rel 0 to 3000 ms Comp Thres -79.0 to 0.0 dB CompSmooth 0.0 to 228.0 ms CompMakeUp Off, -79.0 to 24.0 dB CompSigDly 0.0 to 25.0 ms FdbkComprs In or Out ||||||||||||||||||||||||||||||...
Page 45 CompSmooth 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. CompSigDly The time in ms by which the input signal should be delayed with respect to compressor side chain processing (i.e.
Page 46 52 ClascVrb<>Comprs A reverb and compressor in series. PAUs: ClascVrb<>Comprs is conﬁgurable with the “A->B cfg” parameter as a reverb followed by a compressor “Rvb->Cmp”, or as a compressor followed by a reverb “Cmp->Rvb”. It uses the same reverb as 5 Classic Verb.
Page 47 Threshold In Amp Figure 15 Soft-Knee compression characteristics To determine how much to compress the signal, the compressor must measure the signal level. Since musical signal levels will change over time, the compression amounts must change as well. You can control the rate at which compression changes in response to changing signal levels with the attack and release time controls.
Page 48 Page 3 RvEDfDlySc 0.00 to 2.00x RvE X Blend 0 to 100 % RvEDiffAmt -100 to 100% RvEDly L 0.0 to 720.0 ms RvEDlyR 0.0 to 720.0 ms RvEDlyLX 0.0 to 720.0 ms RvEDlyRX 0.0 to 720.0 ms RvEDfDlyL 0.0 to 160.0 ms RvEDfDlyR 0.0 to 160.0 ms RvEDfDlyLX...
Page 49 CompSmooth 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. CompSigDly The time in ms by which the input signal should be delayed with respect to compressor side chain processing (i.e.
Page 50: Vocal Combination Algorithms
Vocal Combination Algorithms 53 Gate+Cmp[EQ]+Rvb 54 Gate+Cmp<>EQ+Rvb 608 MnGt+Cmp[EQ]+Rvb 609 MnGt+Cmp<>EQ+Rvb Combination algorithms designed for vocal processing. PAUs: 4 each 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.
Page 51 The gate (same gate as Algorithm 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.
Page 52 Page 3 Comp Atk 0.0 to 228.0 ms Comp Ratio 1.0:1 to 100:1, Inf:1 Comp Rel 0 to 3000 ms Comp Thres -79.0 to 0.0dB CompSmooth 0.0 to 228.0 ms CompMakeUp Off, -79.0 to 24.0 dB CompSigDly 0.0 to 25.0ms Page 4 (for Gate+Cmp[EQ]+Rvb) CmpSCBassG...
Page 53 FdbkComprs 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 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.
Page 54 CmpSCBassG, Bass Gain 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.
Page 55 Rv Type. Altering this parameter will change the diffusion from the preset amount. Rv SizeScl 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.
Page 56: More Reverbs
More Reverbs 100 LaserVerb 101 LaserVerb Lite 102 Mono LaserVerb 605 Mn LaserVerb A bizarre reverb with a falling buzz PAUs: 1 for Mono LaserVerb 2 for LaserVerb Lite 3 for 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.
Page 57 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. The output from LaserVerb can be fed back to the input.
Page 58 Wet/Dry The amount of reverbed (wet) signal relative to unaffected (dry) signal. Out Gain The overall gain or amplitude at the output of the effect. Fdbk Lvl 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.
Page 59 103 Revrse LaserVerb A bizarre reverb which runs backwards in time (uh, yeah). PAUs: Revrse LaserVerb is a mono effect that simulates the effect of running the LaserVerb (Algorithms 100–102) in reverse. When you play a sound through the algorithm, it starts out relatively diffuse then builds to the ﬁnal “hit.”...
Page 60 L Input Contour L Output Delay R Output "Dry" Out Gain R Input Figure 20 Revrse LaserVerb Parameters: Page 1 Wet/Dry 0 to 100 %wet Out Gain Off, -79.0 to 24.0 dB Rvrs W/D 0 to 100 %wet -100 to 100 % Page 2 Dly Coarse 0 to 5000 ms...
Page 61 Contour 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.”...
Page 62 104 Gated LaserVerb The LaserVerb algorithm with a gate on the output. PAUs: Gated LaserVerb is Algorithm 101 LaserVerb Lite with a gate on the output. For a detailed explanation of LaserVerb see the section for Algorithm 101 LaserVerb Lite. The gate controls are covered under Algorithm Gate.
Page 63 Page 3 Gate Thres -79.0 to 0.0 dB Gate Time 25 to 3000 ms Gate Duck On or Off Gate Atk 0.0 to 228.0 ms Gate Rel 0 to 3000 ms GateSigDly 0.0 to 25.0 ms |||||||||||||||||||||||||||||| Reduction Wet/Dry The amount of reverbed and gated (wet) signal relative to unaffected (dry) signal. The gate is on the wet signal path.
Page 64 Gate Thresh The signal level in dB required to open the gate (or close the gate if Ducking is on). Gate Duck 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.
Page 65 105 LasrDly<>Reverb A conﬁgurable combination algorithm PAUs: This algorithm is one of a group of conﬁgurable combination algorithms—that is, there’s more than one effect and you can change the sequence of those effects. With this algorithm, for example, you can have either a laser delay followed by a reverb, or vice versa.
Page 66 106 LasrDly<>Rvrb ms A conﬁgurable combination algorithm with some parameters expressed in absolute units PAUs: This algorithm is almost identical to LasrDly<>Reverb. The only difference is that Algorithm 106 uses absolute units for two features: milliseconds for delay line lengths, and Hz for LFO frequencies. Algorithm 105, on the other hand, uses the values of the Tempo parameters to determine delay line lengths and LFO rates.
Page 67: Delays
Delays 150 4-Tap Delay BPM 151 4-Tap Delay 610 Mn 6-TapDelayBPM 611 Mn 6-Tap Delay A stereo four tap delay with feedback PAUs: These are simple stereo 4-tap delay algorithms with delay lengths deﬁned in tempo beats (150 4-Tap Delay BPM) or in milliseconds (ms) (151 4-Tap Delay).
Page 68 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.
Page 69 Fdbk Level 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 The -3 dB frequency in Hz of a one pole lowpass ﬁlter (-6 dB/octave) placed in front of the delay line.
Page 70 Parameters Page 1 Wet/Dry 0 to 100%wet Out Gain Off, -79.0 to 24.0 dB Fdbk Level 0 to 100% Tempo System, 1 to 255 BPM Dry Bal -100 to 100% HF Damping 16 Hz to 25088 Hz Hold On or Off Page 2 LoopLength 0 to 32 bts...
Page 71 152 8-Tap Delay BPM 153 8-Tap Delay A stereo eight-tap delay with cross-coupled feedback PAUs: These are simple stereo 8-tap delay algorithms with delay lengths deﬁned in tempo beats (152 8-Tap Delay BPM) or in milliseconds (ms) (153 8-Tap Delay). The left and right channels are fully symmetric (all controls affect both channels).
Page 72 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. You may have to practice using the Hold parameter. Each time your sound goes through the delay, it is reduced by the feedback amount.
Page 73 Fdbk Level 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. Xcouple 8 Tap Delay is a stereo effect. The cross coupling control lets you send the feedback from a channel to its own input (0% cross coupling) or to the other channel’s input (100% cross coupling) or somewhere in between.
Page 74 the measure with interesting rhythmical patterns. Setting tap levels allows some “beats” to receive different emphasis than others. Parameters Page 1 Wet/Dry 0 to 100%wet Out Gain Off, -79.0 to 24.0 dB Fdbk Level 0 to 100% Tempo System, 1 to 255 BPM Xcouple 0 to 100% Dry Bal...
Page 75 On each output tap is a shaper. For an overview of shaper functionality, refer to the appendices in the KSP8 User’s Guide. The spectral multi-tap shapers offer four shaping loops as opposed to eight found in the V.A.S.T.
Page 76 L Dry L Output Shaper Comb (Individual Shaper, Comb and Gain for Taps 2-6) Shaper Tap 1 L Input Delay Diffuser Imaging Feedback Delay Diffuser R Input Tap 1 Shaper (Individual Shaper, Comb and Gain for Taps 2-6) R Output Shaper Comb R Dry...
Page 77 0 .1 0 x 0 .2 0 x 0 .5 0 x 1 .0 0 x 2 .0 0 x 6 .0 0 x Figure 25 Various shaper curves used in the spectral multi-taps Parameters for Spectral 4-Tap Page 1 Wet/Dry 0 to 100 % Out Gain...
Page 78 Page 3 Tap3 Delay 0 to 32 bts Tap4 Delay 0 to 32 bts Tap3 Shapr 0.10 to 6.00 x Tap4 Shapr 0.10 to 6.00 x Tap3 Pitch C-1 to C8 Tap4 Pitch C-1 to C8 Tap3 PtAmt 0 to 100% Tap4 PtAmt 0 to 100% Tap3 Level...
Page 79 Wet/Dry 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.
Page 80 156 Complex Echo 613 Mn Complex Echo Multitap delay line effect consisting of 6 independent output taps and 4 independent feedback taps PAUs: Complex Echo is an elaborate delay line with 3 independent output taps per channel, 2 independent feedback taps per channel, equal power output tap panning, feedback diffuser, and high frequency damping.
Page 81 L Tap Levels L Input Delay L Output Diffuser Out Gains Blend R Output Feedback FB2/FB1 > FB Blend Delay Diffuser R Input R Tap Levels Figure 26 Signal ﬂow of Complex Echo Parameters Page 1 Wet/Dry 0 to 100 %wet Out Gain Off, -79.0 to 24.0 dB Feedback...
Page 82 Page 3 L Tap1 Lvl 0 to 100 % R Tap1 Lvl 0 to 100 % L Tap2 Lvl 0 to 100 % R Tap2 Lvl 0 to 100 % L Tap3 Lvl 0 to 100 % R Tap3 Lvl 0 to 100 % Page 4 L Tap1 Pan...
Page 83 170 Degen Regen BPM 171 Degen Regen 614 Mn DegenRegenBPM 615 Mn Degen Regen Long delay allowing loop instability PAUs: 4 each 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.
Page 84 L Output P P a a n n Level P P a a n n L Input Level R Output D D e e l l a a y y C C o o m m p p r r e e s s s s o o r r D D i i s s t t o o r r t t i i o o n n F F i i l l t t e e r r s s R Input...
Page 85 Page 2 (Degen Regen BPM) LoopLength 0 to 32 bts Mid1 Gain -79.0 to 24.0 dB LFO Period 1/24 to 32 bts Mid1 Freq 8 to 25088 Hz Bass Gain -79.0 to 24.0 dB Mid1 Width 0.010 to 5.000 oct Bass Freq 8 to 25088 Hz Mid2 Gain...
Page 86 algorithm to become unstable above 0 dB. However other parameters interact resulting in a more complex gain structure. See also Loop Lvl. Loop Lvl 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.
Page 87 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 The center frequency of the treble shelving ﬁlter in intervals of one semitone. Midn Gain The amount of boost or cut in dB that the parametric ﬁlter should apply to the speciﬁed signal band.
Page 88 CompSmooth 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 The compression ratio.
Page 89 172 Switch Loops Looped delay lines with input switching PAUs: 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. The stereo input is summed to mono and sent to any of the four delay lines.
Page 90 Parameters: Page 1 Dry In/Out In or Out Out Gain Off, -79.0 to 24.0 dB Dry Gain Off, -79.0 to 24.0 dB Tempo System, 1 to 255 BPM Fdbk Kill On or Off -100 to 100 % Max Fdbk On or Off HF Damping 8 to 25088 Hz Page 2...
Page 91 DlySelectn 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.
Page 92 173 3 Band Delay 616 Mn 3 Band Delay Three delays operating on selectable frequency bands PAUs: 2 for 3 Band Delay and 1 for Mn 3 Band Delay 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.
Page 93 Page 2 BandSelctA Low, Mid, or High BandSelctB Low, Mid, or High DelayLenA 0 to 6 bts DelayLenB 0 to 6 bts DelayLvlA 0 to 100% DelayLvlB 0 to 100% PanA -100 to 100% PanB -100 to 100% WidthA -100 to 100% WidthB -100 to 100% Page 3...
Page 94 174 Gated Delay 617 Mn Gated Delay Delay with gating and ducking PAUs: 2 each 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 Algorithm 344 Gate w/SC EQ, except this algorithm does not include side- chain EQ ﬁltering.
Page 95 Parameters: Page 1 Wet/Dry 0 to 100% Out Gain Off, -79.0 to 24.0 dB Feedback 0 to 100% Page 2 Loop Crs 0 to 5100 ms Loop Fine -20.0 to 20.0 ms L Dly Crs 0 to 5100 ms R Dly Crs 0 to 5100 ms L Dly Fine -20.0 to 20.0 ms...
Page 96 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 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.
Page 97 190 Moving Delay Generic stereo moving delay lines PAUs: Moving Delay is identical to Algorithm 191 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.
Page 98 191 Dual MovDelay 192 Dual MvDly+MvDly Generic dual mono moving delay lines PAUs: 1 for Dual MovDelay 2 for Dual MvDly+MvDly Each of these algorithms 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. Both ﬂavors of chorus pitch envelopes are offered: ChorTri for triangle, and ChorTrap for trapezoidal pitch shifting.
Page 99 In Dual MvDly+MvDly, there are 2 moving delay elements per channel distinguishable by parameters beginning with “L1,” “L2,” “R1,” and “R2.” 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.
Page 100 Page 2 L Delay 0.0 to 1000.0 ms R Delay 0.0 to 1000.0 ms L LFO Mode Flange, ... R LFO Mode Flange, ... L LFO Rate 0.00 to 10.00 Hz R LFO Rate 0.00 to 10.00 Hz L LFO Dpth 0.0 to 200.0% R LFO Dpth 0.0 to 200.0%...
Page 101 The relative amount of input signal and effected signal that is to appear L Wet/Dry R Wet/Dry 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.
Page 102: Choruses
Choruses 200 Chorus 1 201 Chorus 2 202 Dual Chorus 1 203 Dual Chorus 2 620 Mn Chorus 1 One- and three-tap stereo and dual mono choruses PAUs: 1 for Chorus 1 and Dual Chorus 1 2 for Chorus 2 and Dual Chorus 2 Chorusing is an effect that gives the illusion of multiple voices playing in unison.
Page 103 Feedback Delay L Input High Freq Damping Tap Levels L Output From Right To Right Channel Channel From Right Pans To Right Output Sum Figure 35 Block diagram of left channel of Dual Chorus 2 (right channel is similar) 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 Feedback Delay...
Page 104 Feedback Delay L Input High Freq Damping L Output Tap Level From Right To Right Channel Channel From Right Pans To Right Output Sum Figure 37 Block diagram of left channel of Dual Chorus 1 (right channel is similar) The left and right channels pass through their own chorus blocks and there may be cross-coupling between the channels.
Page 105 Range of LFO Shortest Center Longest Delay Input Delay of LFO Delay LFO Xcurs LFO Xcurs Tap Dly Figure 38 Delay for a single LFO 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).
Page 106 Page 2 Tap Lvl -100 to 100 % LFO Rate 0.01 to 10.00 Hz Tap Dly 0.0 to 1000.0 ms LFO Depth 0.0 to 50.0 ct L/R Phase 0.0 to 360.0 deg Parameters for Chorus 2 Page 1 Wet/Dry -100 to 100 %wet Out Gain Off, -79.0 to 24.0 dB Fdbk Level...
Page 107 Page 3 L PitchEnv Triangle or Trapzoid R PitchEnv Triangle or Trapzoid Parameters for Dual Chorus 2 Page 1 L Wet/Dry -100 to 100 %wet R Wet/Dry -100 to 100 %wet L Out Gain Off, -79.0 to 24.0 dB R Out Gain Off, -79.0 to 24.0 dB L Fdbk Lvl -100 to 100 %...
Page 108 Xcouple 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 The amount of high frequency content of the signal that is sent into the delay lines.
Page 109: Flangers
Flangers 225 Flanger 1 226 Flanger 2 625 Mn Flanger 1 Multi-tap ﬂangers PAUs: 1 for Flanger 1 2 for Flanger 2 Flanger 1 is a one-PAU multi-sweep thru-zero ﬂanger effect with two LFOs per channel. L Input Delay High Freq Damping Levels From Right...
Page 110 Noise L Input Delay High Freq Damping Levels From Right To Right Static Channel Channel Level LFO Feedback L Output Static Tap Feedback Out Gain Figure 41 Simpliﬁed block diagram of the left channel of Flanger 2 (right channel is similar) Flanging was originally created by summing the outputs of two un-locked tape machines while varying their sync by pressing a hand to the outside edge of one reel, thus the historic name reel-ﬂanging.
Page 111 (dB) Frequency Figure 42 Comb ﬁlters: solid line for addition; dashed line for subtraction 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 inverting). One tap is a simple static delay over which you can control the length of delay (from the input tap).
Page 112 Range of LFO Shortest Center Longest Delay Input Delay of LFO Delay LFO Xcurs LFO Xcurs Tap Dly Figure 43 Delay for a single LFO Consider a simple example where you have an LFO tap signal being subtracted from the static delay tap signal.
Page 113 Parameters for Flanger 1 Page 1 Wet/Dry -100 to 100 % wet Out Gain Off, -79.0 to 24.0 dB Fdbk Level -100 to 100 % LFO Tempo System, 1 to 255 BPM Xcouple 0 to 100 % LFO Period 1/24 to 32 bts HF Damping 8 to 25088 Hz Page 2...
Page 114 Page 3 StatDlyCrs 0.0 to 228.0 ms StatDlyFin -127 to 127 samp Xcurs1 Crs 0.0 to 228.0 ms Xcurs3 Crs 0.0 to 228.0 ms Xcurs1 Fin -127 to 127 samp Xcurs3 Fin -127 to 127 samp Xcurs2 Crs 0.0 to 228.0 ms Xcurs4 Crs 0.0 to 228.0 ms Xcurs2 Fin...
Page 115 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 A ﬁne adjustment to the static delay tap length. The resolution is one sample. StatDlyLvl The level of the static delay tap.
Page 116: Phasers
Phasers 250 LFO Phaser 251 LFO Phaser Twin 253 SingleLFO Phaser 254 VibratoPhaser 255 Manual Phaser 630 Mn LFO Phaser 631 Mn LFOPhaserTwin 632 Mn SingleLFOPhsr 633 Mn VibratoPhaser 634 Mn Manual Phaser A variety of single notch/bandpass phasers PAUs: 1 each A simple phaser is an algorithm that produces a vague swishing or phasey effect.
Page 117 Gain Gain 0 dB 0 dB 10 Hz 1000 10 Hz 1000 Freq Freq (ii) Figure 44 Response of typical phaser with (i) Wet/Dry = 50% and (ii) WetDry = -50% 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.
Page 118 Gain 0 dB 10 Hz 1000 Freq Figure 45 Response of LFO Phaser Twin with Wet/Dry set to 100%. The VibratoPhaser algorithm 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.
Page 119 Page 2 LFO Rate 0.00 to 10.00 Hz N/F Phase CenterFreq 8 to 25088 Hz NotchDepth -79.0 to 6.0 dB FLFO Depth 0 to 5400 ct NLFO Depth 0 to 100 % FLFO LRPhs 0.0 to 360.0 deg NLFO LRPhs 0.0 to 360.0 deg Wet/Dry The amount of phaser (wet) signal relative to unaffected (dry) signal as a percent.
Page 120 Notch/BP 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 The output gain in decibels (dB) to be applied to the ﬁnal output. Feedback The phaser output can be added back to its input to increase the phaser resonance (left and right).
Page 121 Wet/Dry 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 The output gain in decibels (dB) to be applied to the combined wet and dry signals.
Page 122 256 Allpass Phaser 3 257 Allpass Phaser 4 635 Mn AP Phaser 3 636 Mn AP Phaser 4 Allpass ﬁlter phasers PAUs: 3 for Allpass Phaser 3, 4 for Allpass Phaser 4, and 2 for the mono allpass phasers The allpass phasers are algorithms that use allpass ﬁlters to achieve a phaser effect. These algorithms do not have built in LFOs, so like Algorithm 255 Manual Phaser, any motion must be supplied with an...
Page 123 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.
Page 124 CenterFreq The nominal center frequency of the phaser ﬁlter. The frequency LFO modulates the phaser ﬁlter centered at this frequency. There are separate left and right controls in the stereo version. FB APNotch The number of notches the allpass ﬁlter can produce when summed with a dry signal. Used in the feedback loop.
Page 125: Comb Filters
Comb Filters 258 Barberpole Comb 637 Mn Barberpole Comb ﬁlter with constantly rising or falling frequency PAUs: 4 for Barberpole Comb and 2 for Mn Barberpole The Barberpole Comb is a comb ﬁlter with a constantly rising or falling frequency. A comb ﬁlter produces a series of evenly spaced notches or resonant peaks in the frequency response.
Page 126 Rising T ime Falling A mplit ude Frequency Figure 49 Barberpole comb ﬁlter responses at different instants of time Parameters: Page 1 In/Out In or Out Out Gain Off, -79.0 to 24.0 dB Rise/Fall Rise or Fall Notch/Peak Notch or Peak Rate 0.00 to 10.00 Hz Depth/Res...
Page 127 Rate The LFO rate at which the comb ﬁlters rise or fall through a complete cycle of frequencies. Comb Freq The frequency separation of the notches or peaks in the comb ﬁlter. Notch/Peak The comb ﬁlter can be constructed to produce a series of notches in the frequency response or a series on resonant peaks.
Page 128: Tremolo Effects
Tremolo Effects 270 Tremolo BPM 271 Tremolo 640 Mn Tremolo BPM 641 Mn Tremolo A stereo tremolo or auto-balance effect PAUs: Tremolo and Tremolo BPM are one-PAU stereo tremolo effects. 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.
Page 129 Page 2 LFO Rate 0 to 10.00 Hz LFO Shape Rate Scale 1 to 25088 x PulseWidth 0 to 100 % Depth 0 to 100 % 50% Weight -6 to 3 dB L/R Phase In or Out 0% 50% 100% Parameters for Tremolo BPM Page 1 In/Out...
Page 130 PulseWidth 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 The relative amount of attenuation added when the LFO is at the -6dB point.
Page 131: Panners And Stereo-Image Effects
Panners and Stereo-Image Effects 275 AutoPanner A stereo auto-panner PAUs: AutoPanner is a one-PAU 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 (Figure 51).
Page 132 Parameters Page 1 In/Out In or Out Out Gain Off, -79.0 to 24.0 dB Page 2 LFO Rate 0 to 10.00 Hz LFO Shape Rate Scale 1 to 25088 x PulseWidth 0 to 100% Origin -100 to 100 % PanWidth 0 to 100 % ImageWidth 0 to 100 %...
Page 133 276 Dual AutoPanner A dual mono auto-panner PAUs: Dual AutoPanner is a two-PAU dual 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, and parameters beginning with “R”...
Page 134 Parameters Page 1 L In/Out In or Out R In/Out In or Out L Out Gain Off, -79.0 to 24.0 dB R Out Gain Off, -79.0 to 24.0 dB Page 2 L LFO Rate 0 to 10.00 Hz L LFO Shape L RateScal 1 to 25088 x L PlseWdth...
Page 135 280 Stereo Image Stereo enhancement with stereo channel correlation metering PAUs: 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.
Page 136 Parameters Page 1 L In Gain Off, -79.0 to 24.0 dB R In Gain Off, -79.0 to 24.0 dB CenterGain Off, -79.0 to 24.0 dB Diff Gain Off, -79.0 to 24.0 dB L/R Delay -500.0 to 500.0 samp RMS Settle 0.0 to 300.0 dB/s Page 2 DiffBassG...
Page 137 281 Mono -> Stereo Stereo simulation from a mono input signal PAUs: 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.
Page 138 Page 2 Crossover1 8 to 25088 Hz Crossover2 8 to 25088 Hz Pan High -100 to 100 % Delay High 0.0 to 1000.0 ms Pan Mid -100 to 100 % Delay Mid 0.0 to 1000.0 ms Pan Low -100 to 100 % Delay Low 0.0 to 1000.0 ms In/Out...
Page 139 282 DynamicStereoize Stereo widening based on dynamic signal levels PAUs: 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.
Page 140 Localized stereo images still come from between your stereo loudspeakers, but there is an increased sense of being wrapped in the sound ﬁeld. 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. When the meter is at 100% correlation, then your signal is essentially mono.
Page 141 Page 4 Comp Ratio 1.0:1 to 100.0:1, Inf:1 Exp Ratio 1:1.0 to 1:17.0 Comp Thres -79.0 to 0.0 dB Exp Thres -79.0 to 0.0 dB MakeUpGain Off, -79.0 to 24.0 dB |||||||||||||||||||||||||||||| Reduction L In Gain The input gain of the left channel in decibels (dB). R In Gain The input gain of the right channel in decibels (dB).
Page 142 Signal Dly The time in ms by which the input signal should be delayed with respect to compressor/ expander side chain processing (i.e. side chain predelay). This allows the compression/ expansion to appear to take effect just before the signal actually rises. Comp Ratio The compression ratio in effect above compression threshold (Comp Thres).
Page 143: Guitar Cabinet Simulators
Guitar Cabinet Simulators 284 Cabinet 645 Mn Cabinet Guitar cabinet simulation ﬁlters PAUs: 3 for Cabinet and 2 for Mn Cabinet The cabinet simulator models the responses of various types of mic’d guitar cabinets. The preset can be selected using the Cab Preset parameter. Parameters: Page 1 In/Out...
Page 144: Rotary Effects
Rotary Effects 290 VibChor+Rotor 2 291 Distort + Rotary 292 VC+Dist+HiLoRotr 293 VC+Dist+1Rotor 2 294 VC+Dist+HiLoRot2 295 Rotor 1 296 VC+Dist+Rotor 4 297 VC+Tube+Rotor 4 298 Big KB3 Effect 646 Mn VC+Dist+Rotor Rotating speaker algorithms PAUs: 1 for Rotor 1 2 each for VibChor+Rotor 2, Distort + Rotary, VC+Dist+1Rotor 2, VC+Dist+HiLoRotr, Mn VC+Dist+Rotor and VC+Dist+HiLoRot2 4 each for VC+Dist+Rotor 4 and VC+Tube+Rotor 4...
Page 145 Figure 58 Rotating speaker with virtual microphones 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.
Page 146 The low frequency speaker can work in three modes: Normal, NoAccel or Stopped. 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%. For positive settings, acceleration is slow at slow speed and speeds up for fast speeds.
Page 147 Page 2 Speed Slow or Fast Brake On or Off Lo Mode Normal Lo Gain Off, -79.0 to 24.0 dB Hi Gain Off, -79.0 to 24.0 dB Lo Xover 8 to 25088 Hz Hi Xover 8 to 25088 Hz Lo HP 8 to 25088 Hz Hi LP 8 to 25088 Hz...
Page 148 Algorithm 297 VC+Tube+Rotor 4 is similar to Algorithm 296 VC+Dist+Rotor 4, except a different distortion is used—one which faithfully models the response and smooth distortion caused by overloading a vacuum tube circuit. The Tube Drive parameter replaces Dist Drive and DistWarmth and the acoustic beam width for the low driver is unavailable.
Page 149 Page 3 Lo Slow 0.00 to 2.00 Hz Hi Slow 0.00 to 2.00 Hz Lo Fast 3.00 to 10.00 Hz Hi Fast 3.00 to 10.00 Hz LoSlow>Fst 0.10 to 10.00 s HiSlow>Fst 0.10 to 10.00 s LoFst>Slow 0.10 to 10.00 s HiFst>Slow 0.10 to 10.00 s LoAccelCrv...
Page 150 Parameters (Distort + Rotary): Page 1 In/Out In or Out Out Gain Off, -79.0 to 24.0 dB Cabinet HP 8 to 25088 Hz Dist Drive 0 to 96 dB Cabinet LP 8 to 25088 Hz DistWarmth 8 to 25088 Hz Page 2 Speed Slow or Fast...
Page 151 L Output L Input P P a a n n Out Gain Mic Levels Cabinet V i i b b r r a a t t o o / / R R o o t t a a t t o o r r D i i s s t t o o r r t t i i o o n n C C h h o o r r u u s s Out Gain...
Page 152 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.
Page 153 Page 1 (VC+Dist+HiLoRot2) In/Out In or Out Out Gain Off, -79.0 to 24.0 dB VibChInOut In or Out Dist Drive 0.0 to 96.0 dB Vib/Chor Dist Warmth 8 to 25088 Hz Roto InOut In or Out Page 2 Speed Slow or Fast Xover 8 to 25088 Hz Brake...
Page 154 L Input L Output P P a a n n Out Gain R R o o t t a a t t o o r r Cabinet Levels Out Gain P P a a n n R Output R Input Figure 64 Rotor 1 Parameters (Rotor 1):...
Page 155 Vib/Chor 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 When set to In the rotary speaker is active; when set to Out the rotary speaker is bypassed.
Page 156 LoAccelCrv The shape of the acceleration curve for the speaker. 0% is a constant HiAccelCrv 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.
Page 157 LoResonate 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 The number of samples of delay in the resonator circuit in addition to the rotation excursion delay.
Page 158: Distortion
Distortion 300 Mono Distortion 301 MonoDistort+Cab 302 MonoDistort + EQ 304 StereoDistort+EQ 650 Mn Distortion 651 Mn Distort+Cab 652 Mn Distort + EQ Small distortion algorithms PAUs: 1 for Mono Distortion 2 for MonoDistort+Cab 2 for MonoDistort + EQ 3 for StereoDistort+EQ L Input L Output Distortion...
Page 159 L Input L Output Distortion R Input R Output Distortion Figure 67 Block diagram of StereoDistort+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 1 parametric mid ﬁlter. L Input L Output Cabinet...
Page 160 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.
Page 161 Page 2 Bass Gain -79.0 to 24.0 dB Treb Gain -79.0 to 24.0 dB Bass Freq 8 to 25088 Hz Treb Freq 8 to 25088 Hz Mid Gain -79.0 to 24.0 dB Mid Freq 8 to 25088 Hz Mid Width 0.010 to 5.000 oct Wet/Dry The amount of distorted (wet) signal relative to unaffected (dry) signal.
Page 162 Mid Gain For MonoDistort + EQ and StereoDistort+EQ. 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.
Page 163 303 PolyDistort + EQ Eight-stage distortion followed by equalization PAUs: 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 lowpass ﬁlter. There is also a one-pole lowpass ﬁlter in front of the ﬁrst stage.
Page 164 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.
Page 165 Page 3 LP0 Freq 8 to 25088 Hz LP1 Freq 8 to 25088 Hz LP5 Freq 8 to 25088 Hz LP2 Freq 8 to 25088 Hz LP6 Freq 8 to 25088 Hz LP3 Freq 8 to 25088 Hz LP7 Freq 8 to 25088 Hz LP4 Freq 8 to 25088 Hz...
Page 166 the signal at the speciﬁed frequency. Negative values cut the signal at the speciﬁed frequency. Mid Freq 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 The bandwidth of the mid parametric ﬁlter may be adjusted.
Page 167 305 Subtle Distort Adds small amount of distortion to signal. PAUs: Use Subtle Distort to apply small amounts of distortion to a signal. 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.
Page 168 The Super Shaperalgorithm packs 2-1/2 times the number of shaping loops, and 8 times the gain of the V.A.S.T. shaper. Refer to the appendices in the KSP8 User’s Guide for an overview of 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.
Page 169 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. Refer to the appendices in the KSP8 User’s Guide for an overview of V.A.S.T. shaping. The cutoff frequencies for these ﬁlters are controlled with the CrossOver1 and CrossOver2 parameters.
Page 170 308 Quantize+Alias 655 MnQuantize+Alias Digital quantization followed by simulated aliasing. PAUs: 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.
Page 171 (ii) (iii) (iv) Figure 72 A decaying sine wave represented with different word lengths: (i) 1-bit, (ii) 2-bit, (iii) 3-bit, (iv) 4-bit. 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.
Page 172 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 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).
Page 173 In/Out When set to In, the quantizer and aliaser are active; when set to Out, the quantizer and aliaser are bypassed. Out Gain The overall gain or amplitude at the output of the effect. DynamRange The digital dynamic range controls signal quantization, or how many bits to remove from the signal data words.
Page 174 309 Quantize+Flange Digital quantization followed by ﬂanger PAUs: Digital audio engineers will go to great lengths to remove, or at least hide the effects of digital quantization distortion. In Quantize+Flange we do quite the opposite, making quantization an in-your-face effect. The quantizer will give your sound a dirty, grundgy, perhaps industrial sound.
Page 175 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.
Page 176 Page 2 Fl Tempo System, 1 to 255 BPM Fl Fdbk -100 to 100 % Fl Period 0 to 32 bts Fl L Phase 0.0 to 360.0 deg Fl R Phase 0.0 to 360.0 deg Fl StatLvl -100 to 100 % Fl LFO Lvl -100 to 100 % Page 3...
Page 177 Tempo. At 0, the LFOs stop oscillating and their phase is undetermined (wherever they stopped). Fl Fdbk The level of the ﬂanger feedback signal into the ﬂanger delay line. The feedback signal is taken from the LFO delay tap. Negative values polarity invert the feedback signal. Fl L/R Phase The phase angles of the left and right LFOs relative to each other and to the system tempo clock, if turned on (see Fl Tempo).
Page 178: Guitar Combination Algorithms
Guitar Combination Algorithms 310 Gate+TubeAmp 311 Gate+Tube+Reverb 312 Gt+Tube<>MD+Chor 313 Gt+Tube<>MD+Flan 314 Gt+Tube<>2MD 315 Gt+Cmp+Dst+EQ+Ch 316 Gt+Cmp+Dst+EQ+Fl 656 Mn Gate+TubeAmp Combination algorithms designed for guitar processing. PAUs: 3 for Gate+TubeAmp, 4 for the others 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.
Page 179 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, and both Bass and Treb Tone controls to 0.
Page 180 ﬁltered output into the stereo chorus or ﬂanger. For a description of the chorus see Algorithm Chorus 1, and for the ﬂanger see Algorithm 225 Flanger L Input L Out put Tube Moving Moving Gate Tone Cabinet Delay Delay R Input R Out put L Input L Out put...
Page 181 Page 2 Gate Thres -79.0 to 0.0 dB Gate Time 25 to 3000 ms Gate Duck On or Off Gate Atk 0.0 to 228.0 ms Gate Rel 0 to 3000 ms GateSigDly 0.0 to 25.0 ms |||||||||||||||||||||||||||||| Reduction Page 3 Bass Tone 0.0 to 10.0 Tube Drive...
Page 182 Page 4 Rv Type Hall 1, ... Rv Time 0.5 to 30.0 s, Inf Rv DiffScl 0.00 to 2.00 x Rv Density 0.00 to 4.00 x Rv SizeScl 0.00 to 4.00 x Rv HF Damp 8 to 25088 Hz Rv PreDly 0 to 620 ms Rv PreDlyR 0 to 620 ms...
Page 183 Page 5 (Gt+Tube<>MD+Chor) Ch Rate L 0.01 to 10.00 Hz Ch Rate R 0.01 to 10.00 Hz Ch Depth L 0.0 to 100.0 ct Ch Depth R 0.0 to 100.0 ct Ch Delay L 0.0 to 720.0 ms Ch Delay R 0.0 to 720.0 ms Ch Fdbk L -100 to 100%...
Page 184 Page 4 MD1Delay 0.0 to 1000.0 ms MD2Delay 0.0 to 1000.0 ms MD1LFOMode 8 to 25088 Hz MD2LFOMode 8 to 25088 Hz MD1LFORate 0.00 to 10.00 Hz MD2LFORate 0.00 to 10.00 Hz MD1LFODpth 0.0 to 200.0% MD2LFODpth 0.0 to 200.0% MD1Fdbk -100 to 100% MD2Fdbk...
Page 185 Page 5 Bass Gain -79.0 to 24.0 dB Treb Gain -79.0 to 24.0 dB Bass Freq 8 to 25088 Hz Treb Freq 8 to 25088 Hz Mid Gain -79.0 to 24.0 dB Mid Freq 8 to 25088 Hz Mid Width 0.010 to 5.000 oct Page 6 (Gt+Cmp+Dst+EQ+Ch)
Page 186 FdbkComprs A switch to set whether the compressor side-chain is conﬁgured for feed-forward (Out) or feedback (In). Gate Thres The signal level in dB required to open the gate (or close the gate if Ducking is On). Gate Duck When set to Off, the gate opens when the signal rises above threshold and closes when the gate time expires.
Page 187 Bass Gain 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.
Page 188 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.
Page 189 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 “MD.”...
Page 190 MD Wet/Dry L Input Input Bal Tube Moving Tone Blend Delay Simulator R Input Ch Wet/Dry L Output Chorus R Output Ch Out Bal Out Gain Figure 83 TubeAmp<>MD>Chor with moving delay inserted post-distortion Parameters: Page 1 In/Out In or Out Out Gain Off, -79.0 to 24.0 dB Input Bal...
Page 191 Page 3 MD Insert PreDist, PostDist, Bypass MD Delay 0.0 to 1000.0 ms MD Wet/Dry 0 to 100 % MD LFOMode ChorTri, ChorTrap, Delay, Flange MD LFORate 0.00 to 10.00 Hz MD LFODpth 0.0 to 200.0 % MD Fdbk -100 to 100 % Page 4 (Chorus Algorithms) Ch Rate L...
Page 192 MD Insert 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, and Bypass takes it completely out of the path. MD Wet/Dry Adjusts the ratio of the moving delay output mixed with its own input to be fed to the next effect in the chain.
Page 193 321 Flange<>Shaper This algorithm is one of a group of conﬁgurable combination algorithms—that is, there’s more than one effect in the algorithm, and you can change the sequence of those effects. With this algorithm, for example, you can have either a ﬂanger followed by a shaper, or vice versa. The combination algorithms are organized in groups, with IDs predominantly in the 400s (there are a few exceptions, of course).
Page 194 322 Shaper<>Reverb This algorithm is one of a group of conﬁgurable combination algorithms—that is, there’s more than one effect in the algorithm and you can change the sequence of those effects. With this algorithm, for example, you can have either a shaper followed by a reverb, or vice versa. The combination algorithms are organized in groups, with IDs predominantly in the 400s (there are a few exceptions, of course).
Page 195: Compressors And Expanders
Compressors and Expanders 330 HardKneeCompress 331 SoftKneeCompress 347 Dual SKCompress 660 Mn HK Compress 661 Mn SK Compress Stereo hard- and soft-knee signal compression algorithms PAUs: 2 for Dual SKCompress; 1 for the others The stereo hard- and soft-knee compressors are very similar algorithms and provide identical parameters and user interface.
Page 196 Threshold Threshold In Amp In Amp Figure 85 Hard- and Soft-Knee compression characteristics To determine how much to compress the signal, the compressor must measure the signal level. Since musical signal levels will change over time, the compression amounts must change as well. You can control the rate at which compression changes in response to changing signal levels with the attack and release time controls.
Page 197 Parameters: Page 1 In/Out In or Out Out Gain Off, -79.0 to 24.0 dB FdbkComprs In or Out SC Input L, R, (L+R)/2 ComprsChan L, R, L & R Page 2 Atk Time 0.0 to 228.0 ms Ratio 1.0:1 to 100:1, Inf:1 Rel Time 0 to 3000 ms Threshold...
Page 198 332 Compress w/SC EQ 348 Dual Comprs SCEQ Soft-knee compression algorithm with ﬁltering in the side chain. PAUs: 2 for Compress w/SC EQ and 3 for Dual Comprs SCEQ The Compress w/SC EQ algorithm is the same as the SoftKneeCompress algorithm (331, page 195)except...
Page 199 Page 2 Atk Time 0.0 to 228.0 ms Ratio 1.0:1 to 100.0:1, Inf:1 Rel Time 0 to 3000 ms Threshold -79.0 to 24.0 dB SmoothTime 0.0 to 228.0 ms MakeUpGain Off, -79.0 to 24.0 dB Signal Dly 0.0 to 25.0 ms |||||||||||||||||||||||||||||| Reduction Page 3...
Page 200 MakeUpGain 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 compression. SCBassGain The amount of boost or cut that the side chain bass shelving ﬁlter should apply to the low frequency signals in dB.
Page 201 333 Opto Compress 334 Opto Comprs SCEQ Compression with 2 release time constants PAUs: 2 for Opto Compress and 3 for Opto Comprs SCEQ Opto Compress is a basic compressor with two different release rates, which change from one rate to another as the compression gain reduction crosses a threshold set by the Rel Thres (release threshold) parameter.
Page 202 M M a a x x i i m m u u m m M M a a g g n n i i t t u u d d e e M M a a x x i i m m u u m m C C o o m m p p r r e e s s s s o o r r M M a a g g n n i i t t u u d d e e C C o o m m p p u u t t a a t t i i o o n n s s...
Page 203 main signal path does, it can tame down an attack transient by compressing the attack before it actually happens. A meter is provided to display the amount of gain reduction that is applied to the signal as a result of compression.
Page 204 Rel Time A The time for the compressor to stop compressing when there is a reduction in signal level (release) from a signal level above the threshold. This release time is active while the signal is reduced by more than the release threshold setting. Rel Time B The time for the compressor to stop compressing when there is a reduction in signal level (release) from a signal level above the threshold.
Page 205 335 Band Compress Stereo algorithm to compress a single frequency band PAUs: Band Compress is in most respects identical to Algorithm 331 SoftKneeCompress on page 195. However, Band Compress compresses only on a single band of frequencies. Frequency band selection is based on a parametric ﬁlter.
Page 206 Filter Gain Select Select Compressor L, R or L, R or Computation Max L, R Max L, R L, R, Bandpass Bandpass or L&R Compress Filters Filters Channel L Input L Output Delay Notch Gain Filters Delay R Input R Output Figure 90 Band Compress block diagram The soft-knee compressor is used which has a more gradual transition from compressed to unity gain.
Page 207 the attack and release times, although the effect is signiﬁcant only when its time is longer than the attack or release time. Generally the smoothing time should be kept at or shorter than the attack time. You have the choice of using the compressors conﬁgured as feed-forward or feedback compressors. For feed-forward, set the FdbkComprs parameter to Out;...
Page 208 ComprsChan Select which input channel will receive compression processing—left, right or both. If you select left or right, the opposite channel will pass through unaffected. FdbkComprs A switch to set whether the compressor side chain is conﬁgured for feed-forward (Out) or feedback (In).
Page 209 336 3 Band Compress 349 Dual 3 Band Comp 665 Mn 3 Band Comprs Soft-knee 3 frequency band compression algorithm PAUs: 3 for Mn 3 Band Comprs 4 for 3 Band Compress 8 for Dual 3 Band Comp The three-band compressor divides the input stereo signal into 3 frequency bands and runs each band through its own soft-knee compressor.
Page 210 Threshold In Amp Figure 93 Soft-Knee compression characteristics To determine how much to compress the signal, the compressor must measure the signal level. Since musical signal levels will change over time, the compression amounts must change as well. You can control the rate at which compression changes in response to changing signal levels with the attack and release time controls.
Page 211 Parameters: Page 1 In/Out In or Out Out Gain Off, -79.0 to 24.0 dB SC Input L, R, (L+R)/2 ComprsChan L, R, L&R FdbkComprs In or Out Crossover1 8 to 25088 Hz Signal Dly 0.0 to 25.0 ms Crossover2 8 to 25088 Hz Page 2 Atk Low 0.0 to 228.0 ms...
Page 212 FdbkComprs A switch to set whether the compressor side chain is conﬁgured for feed-forward (Out) or feedback (In). Signal Dly 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.
Page 213 340 Expander 662 Mn Expander A stereo expansion algorithm PAUs: This is a stereo expander algorithm. The algorithms expands the signal (reduced the signal’s gain) when the signal falls below the expansion threshold. The amount of expansion is based on the larger magnitude of the left and right channels.
Page 214 Threshold In Amp Figure 95 Expansion transfer characteristic 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”...
Page 215 Ratio The expansion ratio. High values (1:17 max) are highly expanded, low values (1:1 min) are moderately expanded. Threshold The expansion threshold level in dBFS (decibels relative to full scale) below which the signal begins to be expanded. MakeUpGain 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.
Page 216 341 Compress/Expand 342 Comp/Exp + EQ 664 Mn Comprs/Expand A stereo soft-knee compression and expansion algorithm with and without equalization PAUs: 2 for Compress/Expand 3 for Comp/Exp + EQ These are stereo compressor and expander algorithms. One version is followed by equalization and the other is not.
Page 217 To determine how much to compress or expand the signal, the compressor/expander must measure the signal level. Since musical signal levels will change over time, the compression and expansion amounts must change as well. You can control how fast the compression or expansion changes in response to changing signal levels with the attack and release time controls.
Page 218 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.
Page 219 Page 4 Bass Gain -79.0 to 24.0 dB Treb Gain -79.0 to 24.0 dB Bass Freq 8 to 25088 Hz Treb Freq 8 to 25088 Hz Mid Gain -79.0 to 24.0 dB Mid Freq 8 to 25088 Hz Mid Wid 0.010 to 5.000 oct In/Out When set to In the compressor/expander is active;...
Page 220 MakeUpGain 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 compression or expansion. Bass Gain For Comp/Exp + EQ.
Page 221: Gates
Gates 343 Gate 344 Gate w/SC EQ 663 Mn Gate Signal gate algorithms PAUs: 1 for Gate 2 for Gate w/SC EQ Gate and Gate w/SC EQ perform stand-alone gate processing and can be conﬁgured as a stereo or mono effects.
Page 222 attack gate release time time time signal rises signal falls above threshold below threshold Figure 100 Signal envelope for Gate and Gate w/SC EQ when Retrigger is On If Retrigger is Off (Gate w/SC EQ only), then the gate will open when the side chain signal rises above threshold as before.
Page 223 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. If the gate opened and closed instantaneously, you would hear a large digital click, like a big knife switch was being thrown.
Page 224 hear one of the input channels, but you want your mono output panned to stereo. -100% is panned to the left, and 100% is panned to the right. SC Input 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.
Page 225 SCMidGain 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.
Page 226: Eqs
350 3 Band EQ 351 5 Band EQ 354 Dual 5 Band EQ 671 Mn 6 Band EQ Bass and treble shelving ﬁlters and parametric EQs PAUs: 1 for 3 Band EQ 2 for Mn 6 Band EQ 3 for 5 Band EQ and Dual 5 Band EQ These algorithms are multi-band equalizers with 1–4 bands of parametric EQ and with bass and treble tone controls.
Page 227 Page 3 Mid3 Gain -79.0 to 24.0 dB Mid3 Freq 8 to 25088 Hz Mid3 Width 0.010 to 5.000 oct Parameters for Mn 6 Band EQ Page 1 In/Out In or Out Out Gain Off, -79.0 to 24.0 dB Bass Gain -79.0 to 24.0 dB Treb Gain -79.0 to 24.0 dB...
Page 228 Page 3 LMid2Gain -79.0 to 24.0 dB RMid2Gain -79.0 to 24.0 dB LMid2Freq 8 to 25088 Hz RMid2Freq 8 to 25088 Hz LMid2Width 0.010 to 5.000 oct RMid2Width 0.010 to 5.000 oct LMid3Gain -79.0 to 24.0 dB RMid3Gain -79.0 to 24.0 dB LMid3Freq 8 to 25088 Hz RMid3Freq...
Page 229 352 Graphic EQ 353 Dual Graphic EQ 670 Mn Graphic EQ Dual mono 10 band graphic equalizer PAUs: 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.
Page 230 (dB) 1000 2000 4000 8000 16000 Freq (Hz) Figure 103 Overall response with all gains set to +12 dB, 0 dB and -6 dB Parameters for Graphic EQ Page 1 In/Out In or Out Page 2 31Hz G -12.0 to 24.0 dB 1000Hz G -12.0 to 24.0 dB 62Hz G...
Page 231 Page 3 R 31Hz G -12.0 to 24.0 dB R 1000Hz G -12.0 to 24.0 dB R 62Hz G -12.0 to 24.0 dB R 2000Hz G -12.0 to 24.0 dB R 125Hz G -12.0 to 24.0 dB R 4000Hz G -12.0 to 24.0 dB R 250Hz G -12.0 to 24.0 dB...
Page 232: Miscellaneous Filters
Miscellaneous Filters 360 Env Follow Filt 675 Mn Env Filter Envelope following stereo two-pole resonant ﬁlter PAUs: 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, or notch.
Page 233 Envelope Follower L Input L Input Resonant Filter R Input R Input Figure 105 Block diagram of envelope following ﬁlter Parameters Page 1 Wet/Dry 0 to 100%wet Out Gain Off, -79.0 to 24.0 dB FilterType Lowpass Min Freq 8 to 8372 Hz Freq Sweep -100 to 100% 0Hz 2k 4k 6k...
Page 234 361 TrigEnvelopeFilt 676 Mn Trig Env Filt Triggered envelope following stereo two-pole resonant ﬁlter PAUs: 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.
Page 235 Triggered Envelope Trigger Envelope Follower Generator Generator L Input L Input Resonant Filter R Input R Input Figure 107 Block diagram of triggered envelope ﬁlter The time constant of the envelope follower may be set (Env Rate) as well as the decay rate of the generated envelope (Rel Rate).
Page 236 Retrigger The 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 useful only when it is below the value of Trigger. Env Rate The envelope detector decay rate which can be used to prevent false triggering.
Page 237 362 LFO Sweep Filter 677 Mn LFOSweepFilt LFO following stereo two-pole resonant ﬁlter PAUs: 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, or notch. (ii) (iii) (iv)
Page 238 PulseWidth Sine Saw+ Saw- Pulse Figure 109 Conﬁgurable wave shapes Parameters Page 1 Wet/Dry 0 to 100%wet Out Gain Off, -79.0 to 24.0 dB LFO Tempo System, 1 to 255 BPM LFO Shape Sine LFO Period 1/24 to 32 bts LFO PlsWid 0 to 100% LFO Smooth...
Page 239 LFO Smooth 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 The type of resonant ﬁlter to be used. May be one of Lowpass, Highpass, Bandpass, or Notch.
Page 240 363 Resonant Filter 364 Dual Res Filter 678 Mn Res Filter Stereo and dual mono 2 pole resonant ﬁlters PAUs: The resonant ﬁlter is available as a stereo (linked parameters for left and right) or dual mono (independent controls for left and right). The ﬁlter type is selectable and may be lowpass, highpass, bandpass, or notch. (ii) (iii) (iv)
Page 241 Parameters for Dual Res Filter Page 1 L Wet/Dry 0 to 100%wet R Wet/Dry 0 to 100%wet L Output Off, -79.0 to 24.0 dB R Output Off, -79.0 to 24.0 dB Page 2 L FiltType Lowpass R FiltType Highpass L Freq 58 to 8372 Hz R Freq 58 to 8372 Hz...
Page 242 365 EQ Morpher 366 Mono EQ Morpher 679 Mn EQ Morpher Parallel resonant bandpass ﬁlters with parameter morphing PAUs: 2 for Mono EQ Morpher 4 for EQ Morpher The EQ Morpher algorithms have four parallel bandpass ﬁlters acting on the input signal and the ﬁlter results are summed for the ﬁnal output.
Page 243 0 dB 0 dB Bandwidth Freq Freq (ii) Figure 112 Frequency response of (i) single bandpass ﬁlter; (ii) sum of two bandpass ﬁlters EQ Morpher can do an excellent job of simulating the resonances of the vocal tract. A buzz or sawtooth signal is a good choice of source material to experiment with the EQ Morpher.
Page 244 Page 3 A Freq 3 8 to 25088 Hz B Freq 3 8 to 25088 Hz A Width 3 0.010 to 5.000 oct B Width 3 0.010 to 5.000 oct A Gain 3 -79.0 to 24.0 dB B Gain 3 -79.0 to 24.0 dB A Freq 4 8 to 25088 Hz...
Page 245: Enhancers, Suppressors, And Modulators
Enhancers, Suppressors, and Modulators 370 2 Band Enhancer Two-band spectral modiﬁer PAUs: 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.
Page 246 Hi Drive 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 The intensity of the transfer function. Hi Shelf F The frequency of where the high shelving ﬁlter starts to boost or attenuate. Hi Shelf G The boost or cut of the high shelving ﬁlter.
Page 247 371 3 Band Enhancer 672 Mn 3BandEnhancer Three-band spectral modiﬁer PAUs: 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 (Figure 113).
Page 248 Page 2 Lo Enable On or Off Mid Enable On or Off Lo Drive Off, -79.0 to 24.0 dB Mid Drive Off, -79.0 to 24.0 dB Lo Xfer -100 to 100 % Mid Xfer1 -100 to 100 % Mid Xfer2 -100 to 100 % Lo Delay 0 to 1000 samp...
Page 249 372 HF Stimulate 1 373 HF Stimulate 3 673 Mn HF Stimulate1 674 Mn HF Stimulate2 High-frequency stimulators PAUs: 1 for HF Stimulate 1 and Mn HF Stimulate1 2 for Mn HF Stimulate2 3 for HF Stimulate 3 The high-frequency stimulator algorithms are stereo and mono algorithms closely based on the V.A.S.T. High Frequency Stimulator DSP function, and the manual description is repeated here (edited for KDFX speciﬁcs).
Page 250 Page 1 (HF Stimulate 1 and Mn HF Stimulate1) Stim Gain Off, -79.0 to 24.0 dB Out Gain Off, -79.0 to 24.0 dB Dist Drive -79.0 to 48.0 dB Highpass 8 to 25088 Hz Dist Curve 0 to 127% Stim Gain The gain of the high frequency stimulated signal applied prior to being added to the original input signal.
Page 251 374 HarmonicSuppress 375 Tone Suppressor Stereo algorithms to expand a single frequency band or harmonic bands. PAUs: Tone Suppressor and HarmonicSuppress are special expander algorithms. In most respects they are identical to Algorithm Expander. However, Tone Suppressor expands only on a single band of frequencies.
Page 252 linear Freq ( i ) ( i i ) ( i i i ) Figure 116 HarmonicSuppress ﬁltering at full expansion F marks fundamental Harmonics are Even (i), Odd (ii), All (iii) The algorithms 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).
Page 253 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. The attack time is deﬁned as the time for the expansion to turn off when the signal rises above the threshold.
Page 254 Page 2 Atk Time 0.0 to 228.0 ms Ratio 1:1.0 to 1:17.0 Rel Time 0 to 3000 ms Threshold -79.0 to 0.0 dB SmoothTime 0.0 to 228.0 ms MakeUpGain Off, -79.0 to 24.0 dB Signal Dly 0.0 to 25.0 ms |||||||||||||||||||||||||||||| Reduction In/Out...
Page 255 Signal Dly 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 The expansion ratio.
Page 256 380 Ring Modulator 680 Mn Ring Modulate A conﬁgurable ring modulator PAUs: 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.
Page 257 Out Gain L Input L Output R Output R Input Figure 120 L R mode ring modulator 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 ﬁve oscillators (see Figure 122).
Page 258 PulseWidth Sine Saw+ Saw- Pulse Expon Figure 122 Conﬁgurable wave shapes Parameters Page 1 Wet/Dry 0 to 100%wet Out Gain Off, -79.0 to 24.0 dB Mod Mode R or Osc R Gain Off, -79.0 to 48.0 dB R Pan -100 to 100% Page 2 Osc1 Lvl 0 to 100%...
Page 259 Osc1Shape Shape selects the waveform type for the conﬁgurable oscillator. Choices are Sine, Saw+, Saw–, Pulse, Tri, and Expon. This parameter is active only in Osc mode. Osc1PlsWid When the conﬁgurable oscillator is set to Pulse, the PlsWid parameter sets the pulse width as a percentage of the waveform period.
Page 260 381 Pitcher 681 Mn Pitcher Creates pitch from pitched or non-pitched signal PAUs: This 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.
Page 261 In Figure 125, 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 125 Pitcher at [100, 0, 0, 0] In Figure 126, there are deeper notches between wider peaks Figure 126 Pitcher at [-100, 0, 0, 0] In Figure 127, there are peaks on odd harmonic multiples and notches on even harmonic multiples of a...
Page 262 Figure 128 Pitcher at [50,100,100,100] Figure 129 is halfway between [0,100,100,100] and [100,100,100,100]. Figure 129 Pitcher at [-50,100,100,100] Figure 130 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.
Page 263 Figure 131 Pitcher at [100, 100, -100, 100] Figure 132 Pitcher at [100, 100, 100, -100] To save space, we’ve left out the other 1,632,240,792 response curves. Parameters Wet/Dry 0 to 100 %wet Out Gain Off, -79.0 to 24.0 dB Pitch C-1 to G9 Ptch Offst...
Page 264 382 Poly Pitcher Creates pitch from pitched or non-pitched signal—twice. PAUs: Poly Pitcher is closely based on Algorithm 381 Pitcher, and most of the features of Poly Pitcher are covered in the section on Algorithm Pitcher. Poly Pitcher is really just a pair of Pitcher algorithms (A and B) using the same inputs and summing to the same outputs.
Page 265 Pitch A, B The fundamental pitch imposed upon the input expressed in semitone scale intervals. Pitcher A and pitcher B may be set independently. PchOff AL 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 PchOff AR PchOff BL...
Page 266 383 Pitcher+MiniVerb Combination algorithm of Pitcher followed by MiniVerb PAUs: Pitcher+MiniVerbis Algorithm 381 Pitcher followed by Algorithm MiniVerb. Pitcher applies a ﬁlter to the signal, the ﬁlter having a regular series of peaks in its frequency response which generally imposes a pitch on the input signal.
Page 267 Page 3 Pch/Dry>Rv 0 to 100 % 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 HFDamp 8 to 25088 Hz Rv PreDlyL 0 to 620 ms Rv PreDlyR 0 to 620 ms...
Page 268 Rv HFDamp Reduces high frequency components of the reverb above the displayed cutoff frequency. Removing higher reverb frequencies can often make rooms sound more natural. Rv PreDlyL/R The delay between the start of a sound and the output of the ﬁrst reverb reﬂections from that sound.
Page 269 384 Flange<>Pitcher This algorithm is one of a group of conﬁgurable combination algorithms—that is, there’s more than one effect in the algorithm, and you can change the sequence of those effects. With this algorithm, for example, you can have either a ﬂanger followed by a pitcher, or vice versa. The combination algorithms are organized in groups, with IDs predominantly in the 400s (there are a few exceptions, of course).
Page 270 385 Frequency Offset 386 MutualFreqOffset 682 Mn Freq Offset Single Side Band Modulation PAUs: 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.
Page 271 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. 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.
Page 272 Page 2 OffsetFreq 0.00 to 10.00 Hz Offs Scale 1 too25088x DwnOffsLvl 0 to 100 % UpOffsLvl 0 to 100 % DwnOffsPan -100 to 100 % UpOffsPan -100 to 100 % Parameters (MutualFreqOffset): Page 1 Wet/Dry 0 to 100 %wet Out Gain Off, -79.0 to 24.0 dB In Gain L...
Page 273 DwnOffsPan The down modulated signal may be panned to the left or right algorithm outputs. -100% sends the signal to the left output and 100% sends the signal to the right output. UpOffsPan The up modulated signal may be panned to the left or right algorithm outputs. -100% sends the signal to the left output and 100% sends the signal to the right output.
Page 274 387 WackedPitchLFO An LFO based pitch shifter. PAUs: 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).
Page 275 LFO Rate 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 A coarse adjust to the pitch shift amount from -24 to +24 semitones.
Page 276 390 Chaos! 690 Mn Chaos! Fun with chaos and instability PAUs: 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.
Page 277 ( i ) ( i i ) Figure 138 Resonating frequencies with FB Invert set to (i) Out and set to (ii) In. In addition to the distortion warmth ﬁlter, there are six ﬁlters built into the delay line loop: a highpass, a lowpass, a treble and a bass shelf, and two parametric midrange ﬁlters.
Page 278 Drive Cut 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 Warmth affects the character of the distortion. Warmth reduces (at low settings) the higher frequency distortion components without making the overall signal dull.
Page 279 391 ADSR Synth 392 Env Synth 691 Mn ADSR Synth 692 Mn Env Synth Synthetic signal generation and control PAUs: 3 for Env Synth and Mn Env Synth 4 for ADSR Synth and Mn ADSR Synth The Synth algorithms produce synthetic waveforms (saw tooth, triangle, sine, pulse, exponential). The algorithms include an envelope following ﬁlter which is a lowpass, highpass, lowpass or notch ﬁlter with cut-off frequency controlled by the envelope of the input signal.
Page 280 global key tracking on the FXMod pages (program or setup editors). To do this, ﬁnd Wave Freq on one of the FXMod pages by scrolling to the appropriate bus and scrolling to Wave Freq in the Param column. Set Adjust to the minimum frequency (8 Hz), Source to GKeyNum (1, 2, 9, Enter) and Depth to the maximum 12800 ct.
Page 281 100% sust ain level at t ack decay sust ain release t ime t ime t ime t ime signal rises signal falls above t hreshold below t hreshold Figure 141 ADSR amplitude control Env Synth amplitude control is slightly simpler conceptually. The amplitude of the synthesized signal simply follows the amplitude of the selected input (left, right or both).
Page 282 Page 3 FilterType Lowpass, ... Filt Thres -79.0 to 0.0 dB FiltMinFreq 8 to 8372 Hz FltAtkRate 0.0 to 300.0 dB/s Freq Sweep -100 to 100% FltRelRate 0.0 to 300.0 dB/s Filt Res 0 to 50 dB FiltSmooth 0.0 to 300 dB/s 60 40 8 dB 0 0 Hz...
Page 283 Wave Freq The fundamental frequency of the synthesized waveform. For key tracking, this parameter is usually modulated using the global key number (GKeyNum) in the FXMod page of the program or setup editor—set Adjust to minimum (8 Hz) and Depth to maximum (12800 ct).
Page 284 ADSR Rel For ADSR Synth only. The release time for the generated signal to drop from the sustain level to off. EnvAtkRate For Env Synth only. Adjusts the upward slew rate of the envelope detector for amplitude envelope following. EnvRelRate For Env Synth only.
Page 285: Combination Algorithms
Combination Algorithms 400 Chorus+Delay 401 Chorus+4Tap 403 Chor+Dly+Reverb 409 Pitcher+Chor+Dly 450 Flange+Delay 451 Flange+4Tap 453 Flan+Dly+Reverb 459 Pitcher+Flan+Dly A family of combination effect algorithms (combination indicated by “+”) PAUs: 1 or 2 Signal Routing (algorithms containing 2 effects) The algorithms listed above 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.
Page 286 Parameters for Two-effect Routing Page 1 Wet/Dry -100 to 100 % Out Gain Off; -79.0 to 24.0 dB Mix Effect A -100 to 100 % Mix Effect B -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.
Page 287 Page 2 A/Dry>B -100 to 100 % A/Dry>B -100 to 100 % A/B -> -100 to 100 % A/B -> -100 to 100 % Mix Effect Left and Right. Adjusts the amount of each effect that is mixed together as the algorithm wet signal.
Page 288 Flangers 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.
Page 289 maximum possible time. Because of this, when you slow down the tempo, you may ﬁnd the delays lose their sync. 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.
Page 290 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...
Page 291 411 MonoPitcher+Chor 461 MonoPitcher+Flan Monaural pitching algorithm (ﬁlter with harmonically related resonant peaks) with chorus or ﬂanger PAUs: 2 each 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.
Page 292 The ﬁgures below show Pt PkShape of -1.0 and 1.0, for a Pitch of C6 and a PkSplit of 0%. PeakShape = 1.0 PeakShape = -1.0 PeakSplit = 0% PeakSplit = 0% Figure 144 Response of Pitcher with different PkShape settings 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.
Page 293 Page 2 Pt Inp Bal -100 to 100 % Pt Out Pan -100 to 100 % Pt Pitch C-1 to G 9 Pt Offset -12.0 to 12.0 ST Pt PkSplit 0 to 100 % Pt PkShape -1.0 to 1.0 Page 3 ChPtchEnvL Triangle or Trapzoid ChPtchEnvL...
Page 294 Mix Chorus, Mix Flange The amount of the ﬂanger or chorus signal to send to the output as a percent. Pt/Dry->Ch, Pt/Dry->Fl 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 or ﬂanger receives its input entirely from the Pitcher.
Page 295: Configurable Combination Algorithms
Conﬁgurable Combination Algorithms 105 LasrDly<>Reverb 321 Flange<>Shaper 322 Shaper<>Reverb 384 Flange<>Pitcher 402 Chorus<>4Tap 404 Chorus<>Reverb 405 Chorus<>LasrDly 452 Flange<>4Tap 454 Flange<>Reverb 455 Flange<>LasrDly A family of conﬁgurable combination effect algorithms (conﬁgurability indicated by “<>”) PAUs: Signal Routing Each of these combination algorithms offers two separate effects combined with ﬂexible signal routing mechanism.
Page 296 A/Dry->B Mix Chorus Input 2-Tap Blend 4-Tap Chorus Delay Mix 4 Tap Wet/Dry Output Blend Out Gain Configured as Ch -> 4T A/Dry->B Mix 4 Tap Input 4-Tap Blend 2-Tap Delay Chorus Mix Chorus Wet/Dry Output Blend Out Gain Configured as 4T -> Ch Figure 145 Chorus<>4Tap with A->B cfg set to Ch->4T and 4T->Ch Bidirectional Routing...
Page 297 Individual Effect Components Conﬁgurable Chorus and Flange 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.” General descriptions of chorus and ﬂange functionality can be found in the Choruses Flangers sections.
Page 298 Left Right Cont r ol Set 1 Cont r ol Set 2 LFOL LFO R Figure 147 LFO control in Dual1Tap mode Left Right Cont r ol Set 1 LFOL LFO R Figure 148 LFO control in Link1Tap mode Left Right Cont r ol Set 1 LFO1L...
Page 299 Parameters for Choruses Page 1 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...
Page 300 All other Flange parameters Flangers on page 109. Parameters with a 1 or 2 correspond to LFO taps organized as described above. Laser Delay 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.
Page 301 Laser Delay L Input L Output Element XCouple From Right To Right Channel Channel Imaging Delay Feedback To Right From Right Channel Channel Figure 150 Laser Delay (left channel) Parameters for Laser Delay Dly Time L 0 to 6 bts Dly Time R 0 to 6 bts Dly Fdbk L...
Page 302 channels to swap each regeneration, which is also referred to as “ping-ponging.” The regeneration affects of cross-coupling are not heard when LsrCntour is used by itself. LsrCntour Left and Right. Controls the overall envelope shape of the laser regeneration. When set to a high value, sounds passing through will start at a high level and slowly decay.
Page 303 Shaper The shaper offered in these combination effects have the same sonic qualities as those found in V.A.S.T. Refer to the appendices in the KSP8 User’s Guide for an overview. Parameters associated with this effect begin with “Shp.” This KDFX shaper also offers input and output 1-pole (6dB/oct) lowpass ﬁlters controlled by the Shp Inp LP and Shp Out LP respectively.
Page 304 Shp Out LP Adjusts the cutoff frequency of the 1-pole (6dB/oct) lowpass ﬁlter at the output of the shaper. Shp Amount Adjusts the shaper intensity. This is exactly like the one in V.A.S.T. Shp OutPad Adjusts the output gain at the output of the shaper to compensate for added gain caused by the shaper.
Page 305: More Combination Algorithms
More Combination Algorithms 406 St Chorus+Delay 407 St Chorus+4Tap 408 St Chor+Dly+Rvrb 410 Pitch+StChor+Dly 412 MonoPitch+StChor 420 Chorus+Delay ms 421 Chorus+4Tap ms 422 Chorus<>4Tap ms 423 Chor+Dly+Rvrb ms 425 Chor<>LasrDly ms 426 St Chor+Delay ms 427 St Chor+4Tap ms 428 StCh+Dly+Rvrb ms 429 Ptch+Chor+Dly ms 430 Ptch+StCh+Dly ms 456 St Flange+Delay...
Page 306 For full details on using these algorithms use the following links to jump to the appropriate section. Combination Algorithms on page 285 Conﬁgurable Combination Algorithms on page 295 More Combination Algorithms on page 305...
Page 307 498 FXMod Diagnostic FXMod source metering utility algorithm PAUs: The FXMod diagnostic algorithm is used to obtain a metered display of FXMod sources. This algorithm allows you to view the current levels of any data sliders, MIDI controls, switches, or internally generated V.A.S.T.
Page 308 499 Stereo Analyze 699 Mn Analyze Signal metering and channel summation utility algorithm PAUs: Stereo Analyze is a utility algorithm that provides metering of stereo signals as its primary function. In addition to metering, the gains of the two channels are separately controllable, either channel may be inverted, and sum and differences to the two channels may be metered and monitored.
Page 309 difference channels. If the entire mix seems to have a relative left/right delay, you can use the L/R Delay parameter to attempt to correct the problem. Positive delays are delaying the left channel, while negative delays are delaying the right channel. By inverting one channel with respect to the other, you can hear what is characterized as “phasey-ness.”...
Page 310: Monaural (Mono) Algorithms
Monaural (Mono) Algorithms The algorithms with IDs in the 600s are monaural versions of stereo algorithms that are described elsewhere in this manual. They are almost identical to their stereo counterparts; the only difference is that they provide one channel of processing instead of two. For descriptions of these algorithms, follow the links below to the stereo counterparts of the algorithms you want to review.
Page 311: 5.1 And Surround Algorithms
5.1 and Surround Algorithms 700 OmniPlace 5.1 701 OmniVerb 5.1 702 TQ Place 5.1 703 TQ Verb 5.1 Reverb algorithms for 5.1 PAUs: 12 PAUs each These reverbs are modeled after the stereo versions with the same names OmniPlace, OmniVerb, Verb).
Page 312 from the same direction, there will typically be a difference in reverb character when they are fed into the reverb. This Cen Couple control provides a way to reduce this difference. When set to 0, the center channel input is treated independently of the other channels. As the value increases, the center channel input effectively turns into a phantom source between the left and right inputs.
Page 313 ERDfAmtScl ERDfAmtScl ERDfAmtScl ERDfLenScl ER HF Damp ERDfLenScl ER HF Damp ERDfLenScl ER HF Damp Send LS Input Diffuser Diffuser Diffuser Delay L Input Diffuser Diffuser Diffuser Delay C Input Diffuser Diffuser Delay Diffuser Diffuser Diffuser Delay R Input LS ER Output RS Input Diffuser Diffuser...
Page 314 Page 2 (TQ Place 5.1) Room Type Booth1, ... LateSource EarRef, Dry Size Scale 0.00 to 5.00x Cen Couple 0 to 100% DiffAmtScl 0.00 to 2.00 x DiffLenScl 0.00 to 4.50 x LF Split 8 to 25088 Hz LFO Rate 0.01 to 10.00 Hz LF Time 0.50 to 2.00 x...
Page 315 pattern is scattered from its source. A value of 0% concentrates the reﬂections close to the source, while 100% scatters reﬂections away from its source. Each input into the late reverb is combined, diffused, lowpass ﬁltered, delayed, then injected once into the ambience generator.
Page 316 ERDfAmtScl ERDfAmtScl ERDfAmtScl ER HF Damp ER HF Damp ERDfLenScl ERDfLenScl ERDfLenScl ER HF Damp Send Diffuser Diffuser Diffuser Delay LS Input L Input Diffuser Diffuser Diffuser Delay Diffuser Diffuser Delay C Input R Input Diffuser Diffuser Diffuser Delay LS ER Output Diffuser Diffuser Diffuser...
Page 317 Page 2 (OmniPlace 5.1) Room Type Booth1, ... Size Scale 0.00 to 5.00x DiffAmtScl 0.00 to 2.00 x DiffLenScl 0.00 to 4.50 x LF Split 8 to 25088 Hz LFO Rate 0.01 to 10.00 Hz LF Time 0.50 to 2.00 x LFO Depth 0.0 to 100.0 ct Page 3...
Page 318 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.
Page 319 LFO Rate Adjusts the rate at which certain reverb delay lines move. See LFO Depth for more information. LFO Depth 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.
Page 320 ERDifAmtScl Scales the amount of diffusion within the early reﬂection network. ERDfLenScl Adjusts the length of diffusion within the early reﬂection network. ER HF Damp This adjusts the cutoff frequency of a 1-pole (6dB/oct) lowpass ﬁlter applied to the early reﬂections FrntPreDly Adds additional predelay to front, rear, left, right, or center wet output channels.
Page 321 704 Surround Surround reverberation algorithm for six sources PAUs: Surround allows you to place each of six dry mono sources in a surround reverberation space. “Global” controls act in common on the overall simulated space, while “source” controls are replicated for each of the six sources.
Page 322 To allow control over the early growth of reverberation, the inputs are passed through an “injector” that can stretch the source before it drives the reverberator. Only when Build Envelope is set to 0% is the reverberator driven by the pure dry signal. For settings of Build Envelope greater than 0%, the reverberator is “stroked”...
Page 323 Drct Level 0 to 100 % Drct > Ctr On / Off ER Level 0 to 100 % Rvrb > Ctr On / Off Rvrb Level 0 to 100 % Page 2 The second page of global controls manage the details of the global reverberator. Room Size 2.5 to 40 m Build Time...
Page 324 Global Controls (Pages 1 and 2) Drct Level The level of the submix of all six sources’ direct component. It acts equally upon the direct mixes for L, R, C, LS, and RS. This is the component that implements the distance and direction controls. Drct > Ctr acts as a switch, enabling the mixing of the six sources’...
Page 325 Source Controls There are six sets of source controls, one each for Sources 1–6. Each set of source controls is on its own page. On each page, the parameter names indicate the corresponding source. On Page 3 (the control page for Source 1), the parameters are named 1 Direction, 1 Distance, and so on.
Page 326 720 Compress 5.1 5.1 channel soft-knee signal compression algorithm PAUs: Compress 5.1 contains four independent soft-knee compressors with ﬂexible side chain routing. Each compressor side chain may take its input from any or all of the 5.1 input channels. For each of the 5.1 channels, you can select which of the four compressor side chains will be used to compress the channel (or you can have channels bypass any compression).
Page 327 release times, there is another time parameter: SmoothTime. The smoothing parameter will increase both the attack and release times, although the effect is signiﬁcant only when its time is longer than the attack or release time. Generally the smoothing time should be kept at or shorter than the attack time. The signal being compressed may be delayed relative to the side chain compression processing using the SigDly parameters.
Page 328 the surround channels together, and the center and LFE channels separately. Of course very unusual compressor routings are possible. Parameters: Page 1 In/Out In or Out Out Gain Off, -79.0 to 24.0 dB L-> Bypass, Comp1, ... R-> Bypass, Comp1, ... C->...
Page 329 Page 6 Comp2Atk 0.0 to 228.0 ms Comp2Ratio 1.0:1 to 100:1, Inf:1 Comp2Rel 0 to 3000 ms Comp2Thres -79.0 to 0.0dB SmthTime2 0.0 to 228.0 ms MakeUp2 Off, -79.0 to 24.0 dB SigDly2 0.0 to 25.0ms |||||||||||||||||||||||||||||| Reduction Page 7 Comp3Atk 0.0 to 228.0 ms Comp3Ratio...
Page 330 SmthTimen A lowpass ﬁlter in the control 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. SigDlyn The time in ms by which the input signal should be delayed with respect to compressor side chain processing (i.e.

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