Adapted from the ZynAddSubFX Equalizer.

Gain: Overall output level

Q: Resonance of individual filters. Generally helps smooth extreme settings...or make extreme settings yet more extreme.

Originally adapted from ArtsCompressor. DSP routine has been rewritten entirely as the code had been adapted at a stage in ArtsCompressor development prior to many bug fixes that occurred later in its evolution which unfortunately removed calculation of a soft-knee compression characteristic. This is mentioned to give hope to those who were aware of the bugs in compressor in Rakarrack 0.3.0. These bugs have been fixed, and this compressor has been traced using test signals to confirm it does what we think it does.

A. Time: Attack Time. Time in milliseconds for attack to settle to 64% of final amount of compression.

R. Time: Release Time. Time in milliseconds for gain to return to 64% of it's initial setting.

Note: For the curious, the strange number, 64%, relates to the RC time constant (RC means Resistor-Capacitor), which follows a natural exponential curve. The use of this behavior will make this compressor feel more natural to one who is most familiar with rack-mounted analog compressors.

Ratio: For every Ratio (dB) that the input exceeds the Threshold, the output will be allowed to increase by 1 dB. For example, using compression ratio of 2, if the signal gets 2dB above the threshold, the output will only go 1dB above the threshold.

Knee: Percentage of the region in dB of space between Threshold and 0dB (basically is Knee X -Threshold). Within this region, the ratio increases from 1 to log2(ratio). Threshold + Knee marks the point of full compression onset. For example, if knee is set to 100%, a very gradual compression characteristic will be obtained, but the maximum compression available is log2(ratio). A ratio of 32 will result in a real ratio ranging from 1 to 5. This is useful for processing cymbals as it does not crunch the dynamic attack quite so badly, but helps sustain the trailing resonance.

Thrhold: Threshold. Defines the onset of compression.

Output: How much gain to remove from the final stage of the compressor.

Peek: Enable/Disable peek compression.

Auto Output: Automatically calculate makeup gain. If this is unchecked, then adjustments of Ratio, Knee, and Thrhold will make changes to the overall output level. Unchecking may be useful to interpret Thrhold as a limiting threshold, and one would likely use a high ratio in this case.

Stereo: Process each channel separately.

Adapted from the ZynAddSubFX 'Distortion'

This is a waveshaper, and not particularly an amp or stompbox modeling effect. This must be used judiciously with EQ's as well as the LPF and HPF settings (Low Pass Filter, High Pass Filter).

Tip: Your classic green stompbox has a Pre Filter curve with cutoff at 720Hz. This corresponds to approximately 51 on the HPF setting. If trying to emulate stompbox or amp sounds, it is highly advisable to use the Pre Filter option with HPF higher than 30 and LPF lower than 70. As you increase drive, decrease LPF.

Rakarrack attempts to be friendly to computers limited in both RAM and CPU. Because of this, the waveshaper does not use oversampling to reduce the effects of digital aliasing. Without going deep into DSP theory, this means that hard clipping with a lot of drive will create harmonics that are greater than half the sampling rate. These harmonics get mirrored back into the audio range. If they are significant in magnitude, then there will be a "grainy" non-musical sound. This is what most guitarists are talking about when making reference to "digital distortion". Functions like Sine, Pow and Atan generate a lot of harmonics, and even at lower signal levels. Lmt, Clip and Zigzag are among the worst, but these are generally used for nasty noisy sounds, so a little digital aliasing may not be a bad thing.

Fortunately, for instruments such as guitars, you can significantly limit the bandwidth of the input signal without catastrophically discoloring the timbre of the instrument by applying Low Pass Pre-filtering. By reducing the high frequency content of the input, you can greatly minimize the level of aliasing harmonics present in the output.

The main point is that great sounding distortions can be obtained from Rakarrack, but the distortion module is not meant to emulate your favorite stompbox, but to off the flexibility create these sounds as well as unique flavors of distortion.

The Crunch waveshaping type is the most physically informed of the waveshaping functions currently used. The clipping characteristic is most closely related to curve produced by a JFET amplifier stage. With proper EQ settings, a sound reminiscent of high gain British stacks can be obtained.

Sub Octv: Allows you to mix some sub-octave rumble into the output. Technically speaking, this modulates the output with a square wave at half the fundamental frequency (sub octave).

Adapted from the ZynAddSubFX Distortion

Same as Distortion, but without sub-octave.

Adapted from the ZynAddSubFX Echo

Wet/Dry: Mix level of echo with original.

Reverse: Mixes reverse delay with forward delay.

More about reverse: Echo works by storing audio samples in memory and playing back samples that were stored a while ago. At a sample rate of 48kHz, the number of samples stored in memory for a 1 second delay is 48,000. To play back audio from 1 second ago, there is a reader that increments through, right in front of the write. The reader reads a sample from a second ago, and the writer writes that memory location with a new sample from the present.

Now, what if we could read this block of memory backwards? Then everything stored in memory comes out in reverse, and has an interesting reverse envelop sound. In the digital word, we can read it forward as easily as backwards, so why not keep track of both and let you mix them together?

Pan: Sends the delay more to the right or left. Zero balances it in the middle.

Delay: Amount of time before you hear the echo.

Lrdl.: Left/Right delay difference. A setting of 64 means that the delay time is the same for left and right channels. If larger than 64, the right channel is delayed longer than the left channel. Less than 64 the reverse is true. This allows you to achieve the stereo 'Ping Pong' effect.

L/R Cr: Left/Right Crossing. This mixes left and right channels.

Fb: Feedback. How much of the delayed output to add to the input. This makes the echo regenerate, like it's bouncing around in a canyon.

Direct: When is on, the effect only play the echoes.

Damp: Low pass filter on the delayed signal. This rolls off the high frequencies on every regeneration. Setting to a higher level will make the echo sound more natural.

Adapted from the ZynAddSubFX Chorus

Tips

Tempo: LFO Rate

Rnd: Add noise to LFO to make it more natural sounding.

St.df : Stereo difference of LFO.

L/R Cr. : Mix left into right, Right into left. At maximum level, left and right channels are swapped.

Adapted from the ZynAddSubFX Phaser

More mild digital phaser with exponential LFO sweep.

Phase: This is the offset for the "center" of the sweep.

Depth: LFO deviation or some may like to think of it as LFO amplitude. This is the same type of function that has been labeled 'width' in Analog Phaser.

Fb: Feedback. Some phasers title this "Regen". Feedback is negative as this moves toward zero, positive from 64+. 64, strangely, corresponds to zero feedback.

ZynAddSubFX Phaser Seriously Hacked

Overhauled DSP engine in Zyn phaser and used shell to create a "Physically Informed Digital Model of an Analog FET Phaser". Don't forget to spit after chewing that mouthful.

When should you want to use Phaser instead?

A) Next to the pitch shifter, Analog Phaser is one of the most CPU intensive FX in Rakarrack. If you find your computer can't handle it, then Phaser is much more friendly to slow processors.

B) You just plain like the original Phaser Effect and want to use it.

C) When you want to use two phasers at the same time, you can chain Analog Phaser and Phaser together for some unique filtering sounds.

A Phaser works by adding a delay to a certain group of frequencies and shifting the filter that selects the group of frequencies up and down the spectrum. When mixed with the original input, some of the delayed parts of the signal cancel with the the original (destructive interference). The technical term for this is 'notch filter'. For every two phase stages, a new notch appears in the spectrum, so for 4 stages and up we have a 'comb filter' because a plot of the frequency response begins to look like something you use to brush your hair as the number of phase stages increases.

If you don't care about what's under the hood, skip this paragraph -- To get the basic behavior of an Analog Phaser, the transfer function of a real analog phaser all-pass filter stage was computed and transformed into a discrete-time equivalent transfer function, then broken down into the resulting numerical computation algorithm. These all-pass filter stages were chained together in sequence of the overall analog circuit. Five or six different schematics were referenced in the design of this model to see what different units do to achieve a certain sound. The resulting 'Virtual Circuit' is an original creation. Finally, some of the non-perfect physical components and distortion were modeled in a simplistic way to add a certain warmth to the effect without sinking your CPU into an ice age.

Wet/Dry: At zero, both the phase shifted signal and input signal are mixed equally. This creates the deepest notch filter. Why would you want to mix all Wet? Modulating the all-pass filter stages can create frequency shifting at faster modulation rates. This produces a chorus-like effect, and you may want the frequency bending without the notch filtering. Think UniVibe.

LFO Type: More or less self explanatory. The Barber Pole setting will disappoint you if you're looking for a true barber pole phaser. This is simply an arrangement of multiple ramps modulating the phaser, so you hear the thump every time the ramp starts over. It's a pleasant enough effect at really slow rates, and creates something interesting at very high rates. In between it's annoying :) .

Tempo: LFO frequency.

Depth: How deep the phaser can sweep. It's an LFO offset. 64 is dead center. If you want it to stay in the high range, set this higher than 64. If you want the phaser to spend more time carving out the lower frequency range, set this to some thing less than 64.

Width: This is how far the LFO travels. It's the LFO amplitude.

Fb: Feedback. Usually named 'Regen' on a phaser stompbox. For deeper notches keep Fb negative for even number stages and positive for odd number stages.

Mismatch: FET Phasers suffer from the manufacturing process of JFET transistors. These things generally vary over a very wide range of properties per batch. What does this mean to a Phaser? It means that the frequency where 45 degrees phase shift occurs varies from stage to stage. This mismatch makes the notches wider, but less deep. This parameter best aligns with reality at a setting of 5 to 10. Large settings may be used to obtain a better 'Vibe sound.

Distort: FETs used as variable resistors are only linear over a certain range. This nonlinearity adds harmonics to the processed signal and somewhat warps the frequency response. This is a very subtle effect, but worthwhile to be able to zero it if you don't want it modeled.

St. Diff: Stereo Difference. Delay the LFO in the right or left channel. This combined with the panner effect can make it sound like something is twirling in an elliptical orbit around your head.

Stages: First order all-pass filter networks to be chained together. Set to one for a high pass filter (or low pass filter with subtract checked). 4 stages is very typical in the average stompbox. Bi-mode and Tri-mode phasers include switchable 6 and 8 stage filters. It's rare to see an analog phaser with more than 8 stages in stompbox form because parts get expensive, and noise becomes an expensive design problem to mitigate. In the digital world, adding more stages is just another time through the filter loop if you can spare the CPU time.

Subtract: Subtract wet from dry instead of add.

Hyper: Flattens out the lower end of the LFO. Used with a Tri wave, this emulates the "Hyper Sine" found in some analog phaser pedals. It was a clever way that analog filter designers contrived to make the LFO behave according to the human's perception of frequency. To our ears, musical pitch increases exponentially with frequency.

Adapted from the ZynAddSubFX Chorus

Adapted from the ZynAddSubFX Reverb

Adapted from the ZynAddSubFX Equalizer

Effect using the ZynAddSubFx Equalizer engine

This applies an approximation of certain speakers, cabinets,speaker/cabinet combinations according to publicly available frequency plots. For those wanting convolution-based cabinet modeling you can use Convolotron, but this cost a little bit of CPU resources.

For the stompbox minded people, this is a stereo tremolo effect. Extra Stereo setting creates a more spacious stereo imaging effect.

Intelligent Harmonizer Explained

Rakarrack harmonizer effect use the audio engine of the smbPtichShifter.cpp located at http://www.dspdimension.com. In order to save CPU use only a mono pitch-shifter in the lowest quality available, you can change this on the program Settings window, but only for a low quality values because high quality ones use too much CPU. The audio signal converted to mono is send it to the pitch-shifter and returned to the two pole Peak filter, panned and send it to both channels L/R. The rakarrack Harmonizer has 3 modes:

Normal Mode

Is how a normal pitchshifter run, the pitch ratio is fixed and can be selected in the Interval effect parameter of the effect.

Select Mode

In this mode the pitch ratio is variable, this value depends of the selected Interval parameter, and the Note and Chord effect parameters. Rakarrack then recognize the audio note played (Only "melodies" monophonic data can be played in this mode) and modifies the pitch ratio in order to do a musical harmonization with the Tonality/Chord selected in the parameters, of course the twelve tonality's are available and 33 chords:

,6,Maj7,lyd,Maj(9),Maj7(9),6/9,+,m,m6,m7,m7(b5),m9,m7(9),m7(11)
,m(Maj7),m(Maj7)(9),dim,dim7,7,7(Sus4),7(b5),7(9),7(#11),7(13),7(b9)
,7(b13),7(#9),+Maj7,+7,1+8,1+5,(Sus4),2

MIDI Mode

This mode run in the same way as Select Mode but the Tonality/Chord is recognized via MIDI notes, the MIDI chord recognizer recognizes all the above chords plus all the inversions and bass changed chords, also has memory, they use the chord just another chord is send it and recognized. The MIDI channel can be selected in the Settings window, then the Harmonizer adjust the pitch ratio with the audio note recognized, played by the user, and the Tonality/Chord received via MIDI. (Sequencer track ... )

Musical Delay Explained

The musical delay effect is a dual line delay, the word "musical" is due that you can adjust the delay time in both lines in a musical terms, with Tempo effect parameter, and the Delay1, Delay2, Delay3 effect parameters.

The Delay lines are measured in fractions of quarter notes at the Tempo selected. That means 1/2 is an Eighth note and 1/4 is a Sixteenth note. (1,1/2,1/3,1/4,1/5,1/6) are the possible values, that include eighth triplets, etc.

The center delay parameter, is the delay between the two delay lines, and is the only one can be set equal to zero.

The Tempo effect parameter value range is big (10~480) that's for admit half or double song Tempo in order to obtain largest or shortest delays.

Off course you have Gain, Pan and Feedback parameters for each delay line in order to adjust the desired effect.

Distortion using resonant filter engine and standard Distortion waveshaping functions. Controls are mostly like Distortion module, but includes "Color" to add resonance to the waveshaper signal. This allows for a "bigger" sound that is useful for fuzzes and so on. Your imagination will take it where you want. Addition of this module also allows the user to have up to 3 simultaneous distortion modules, 2 of with can have sub-octave modulation.

New Effect based on Gate, Steve Harris LADSPA plugin. Only use the noise gate when you really need it. This can be a life saver, but it can also be a source of great frustration if improperly configured.

Adapted from the ZynAddSubFX DynamicFilter

By now you're saying, "Ok, I get the Pan, Freq, Rnd, St. Dif, and Depth thing, but what the..."

In addition to use as Auto-wah (LFO-modulated Wah-Wah), this is also an envelope filter.

Currently, here are the three parameters related to envelope control:

Amp S.: Amplitude sensitivity. This is like the sweep range. If sensitivity is set low, you have to pluck really hard to make the filter move. If set high, then the filter will sweep to the limit.

Amp. S.I.: This stands for "Amplitude Sensitivity Inverse", but is applied in a somewhat interesting way. This slider offsets the base frequency for the wah wah filter resonance. At the same time, it acts as a logical check for which direction to sweep. When this is set less than 64, the filter responds to plucking by sweeping upward in frequency. When this is set higher than 64, then the filter responds to plucking by sweeping downward in frequency. Tip: If you want to control wah-wah with a MIDI foot pedal, this is the parameter you will want to map. For pure pedal-controlled wah-wah, set Amp. S and Depth to zero.

Smooth: How much to smooth the signal envelope. At a low level, the sound is like a bubble maker. At a high level, the filter responds very slowly to signal attack.

AlienWah - Created by our hero, Paul Nasca Octavian originally for the ZynAddSubFX synthesizer.
AlienWah fits into the comb filter category, similar to phasers and flangers.  AlienWah amplitude modulates two delay lines each with a phase separation of "Phase", then feeds them into each other in a somewhat circular fashion making each delay line a cascade of low order digital filters. In this manner it is similar to a Phaser.  It's sort of like a digital low-pass filter that spirals down a pair of delay lines causing an interesting interference pattern.  Some people may think of this effect as a "Flaser".

For the less technically minded, the AlienWah configuration produces mostly vocal sounds with a strange timbre sounding much like sci-fi depictions of a language spoken by invading space aliens, or some strange creature found near the center of the earth.

Even though some of the more extreme settings create weird alien sounds, it can also be configured for interesting subtle comb filtering sounds for shaping a rather beautiful timbre onto an instrument.  It is a truly amazing effect.

Here is a description of the parameters:

Wet/Dry: Being a delay-based effect, Wet/Dry does more than simply mix Wet/Dry sounds.  Destructive interference occurs between the effected signal and clean so a proper blend can be used also to voice the effect.

Pan: Put effect more to the left or right channel.  Used in conjunction with Wet/Dry mix, one can voice the effect differently between left right.

Tempo: LFO rate

Rnd: Add random noise to LFO to make it sound somewhat more "analog".

LFO Type: Self explanatory

Phase: This is a very interesting parameter.  It sets the modulation offset between the two delay lines.  On either extreme it has a lower "closed mouth" sound, while in the middle it creates more of an "open mouth" sound.  This parameter "centers" the LFO around the point it is set.

St. df: Stereo Difference between left & right LFO.  A setting of 64 has both LFOs in phase between left/right.  This parameter used with L/R Cr. is like having a second "Phase" parameter, so more interesting interaction can be achieved by using this control.  It's more than what meets the eye.

Depth: This would be better termed "width" as this parameter adjusts the amplitude of the LFO.

Delay: Length of the delay line in samples.  In a sense, this also is the filter order.  The longer the delay time, the deeper the sound of the filter.

Fb: Feedback.  Less than 64 is negative feedback, more than 64 is positive feedback.  64 corresponds to zero feedback. Without feedback, the effect is rather dull.  It's the perpetual regeneration along the delay line that creates  the interesting phase cancellations and resonances that sound alien.

L/R Cr.: Left/Right Crossing.  Amount of left to mix into right and right into left.  Less than zero subtracts left from right, right from left, while greater than zero adds them.

New Effect based on Valve, Steve Harris LADSPA plugin, filters, harmonic enhancer and some extra distortion where added.

Another Flanger alternative.
Basic operation of a flanger: Delay the incoming signal and add it to the input signal. This creates destructive interference in a similar manner to a Phaser, but the notch filter frequencies are evenly spaced from the lowest notch up to infinity (in theory). Dual Flange picks two different delays and adds them to the input signal. The two separately delayed paths have a spacing relationship that follows the other by a percentage, and not a constant offset. At maximum wet, these two delay "taps" are added equally, thus the notches between the two are infinitely deep, and the effect is at its maximum severity. Mixing in the clean signal reduces the depth of the notches, but also creates a complex and interesting frequency response. The result is a unique color to the tone not available by a standard flanger.

Wet/Dry: Mix delayed signal with Dry

Pan:

L/R.Cr:

Depth: Frequency (Hz) of the lowest notch frequency on lowest end of sweep range

Width: LFO sweep width, deviation measured in Hz.

Offset: Percent difference in delay between delay A and delay B.

Fb: Feedback. 0 is none, negative and positive.

LPF: Damp the delay line. This makes the high frequency notches more shallow, thus the sound approaches that of a Phaser or other mellow sweeping comb filter

Subtract: Subtract delayed signal instead of add

Th. zero: Through Zero. Delay Dry signal to 1/2 way between high and low delay deviations of delayed signal.

Tempo: Sweep speed

St.df: Stereo Difference between LFO's

LFO Type:

Rnd: Random. Add random noise to LFO to help reduce unmusical effects of linear interpolation between samples.

New Effect based on Ring, Steve Harris LADSPA plugin. Auto Freq recognizes the frequency of the note played (Only monophonic sorry ..) in order that you can use Ring as a simple monophonic synthesizer.

New Effect based on Harmonic, Steve Harris LADSPA plugin.

Multi Band Distortion, use the Cross1 and Cross2 to select the ranges of each band.

20Hz -> Cross1 = Low
Cross1 -> Cross2 = Mid
Cross2 -> 20KHz = High

You can select a waveshape type for each band.

"Arpie" is a pet name for Arpeggiator. This effect is a modified version of Echo. The delay line is played back at higher rates to create arpeggios at octave intervals. You can select a preset sequence, then adjust the number of steps to repeat from the sequence to gain a wide variety of patterns. Arpie can be adjusted to achieve thick and beautiful soundscapes as well as very edgy electronic trance and techno sounds. Of course, with slower tempos such that several phrases are repeated one can get the tempo-doubling side effect to lay out paint-pealing leads without hardly moving your fingers.

Wet/Dry: Amount of arpeggiated accompaniment in the mix

Arpe's: Mix pitch shifted delay line with normal delay

Pan:

Tempo: Measured in beats per minute

Subdivision: Ratio to subdivide measure. A setting of 1 is equal to 1 measure at Tempo. For example, 1/4 means the length of the delay line is equal to a quarter note at the selected tempo. The sequence will jump 4 steps per measure.

LRdl.: Same as for Echo

L/R.Cr: Same as for Echo

The primary purpose for this effect is to offer a differently natured Noise Gate than the "Noise Gate" module. The turn-on characteristic is modeled after that of a BJT transistor, thus it behaves more similarly to an analog noise gate. It is likely that Expander uses less CPU resources than Noise Gate, but neither are very resource intensive modules.

The basic behavior is the louder you play, the louder it gets until it gets pinned at the maximum "Level". As a noise gate, the volume is exponentially increased in the lower volume levels until Threshold, at which point the gate is entirely open.

To use the effect as an expander, it is best to set the threshold at higher levels, and shape at moderate levels (less than 10). Short attack/release times in this mode will cause it to work much like one would expect from a dynamics expander. In this mode it is essentially a compressor in reverse.

To take the idea of expander a little further, one can set long attack times and short release times to achieve a string swell effect akin to adjusting the volume pot on the guitar...only the effect does it automatically for you.

Finally, the filters may be used for a bit of frequency shaping before a distortion effect, or used in front of the Compressor to help brighten the tone.

A. Time: Attack Time, time to open the gate

R. Time: Release Time, time it takes the gate to return to off state

Shape: Gate transition shape. Large number means gate turns on suddenly at threshold. Small number means gate gradually turns on Noise suppression is best at higher numbers, but the gradual expansion above the threshold can be a more musical way for gate to turn on.

Thrhold: Threshold. The gate is reducing signal volume below this level. Moderate threshold setting with smaller number for shape is useful for string swell effects.

Level: Boost the output. 1 ... 127 represents a fractional increase in volume from 0dB to 20dB

HPF: High Pass Filter

LPF: Low Pass Filter

Effect based on Stereo Shuffling paper by Michael Gerzon. That convert L/R signals to M/S, Mid and Side. You can equalize one of this bands with a parametric four band, the Rev selector select the M/S band to equalize, ON = S, OFF = M. That produce interesting spatial sound, also can be used to remove certain frequencies on the M/S channels.

A different type of view of the Analog Phaser effect revealed a basic structure supporting high-order filters with adjustable number of filter stages ...most of the work was already done, so thanks again to Paul Nasca for this piece of code from the original ZynAddSubFX Phaser.

Starting with a phaser (as opposed to wah-wah or EQ) was inspired by a phaser circuit modification suggested by Mark Hammer, a frequent forum poster at diystompboxes.com/smfforum. This modification allowed two phase stages to be converted to low-pass stages with a switch in an analog phaser. This is the type of effect marketed as a "phase-wah". Of course, converting all of these phase stages to low pass stages makes this circuit look an awful lot like what is found inside the old Korg Delta DL-50 synthesizer.

Converting a normal OTA phaser into a synth filter would be a relatively ugly hack in the circuit bending world, but a simple exercise in software bending, and with a more elegant final product. With a little bit of experimentation, I had a filter sounding much like something found in an analog synthesizer. Thinking back to an analog envelope filter I once built, I determined high pass filter stages would also be of great utility. The final result: A very flexible filter module able to accomplish many flavors of low pass, band pass and high pass filter shapes. Possibilities range anywhere from simple wah-wah sounds to definite mushy filters from synthland.

You can operate up to 12 high-pass and 12 low pass stages simultaneously. At present, the user is protected from resonance near instability. If you are a synth filter lover and want to be able to make your filter unstable, then it is a very minor adjustment to the source code if you wish to internally amp up the feedback range to allow this. Finally, the SynthFilter comes with wet/dry mix so you can accomplish interesting phase-wah sounds.

Wet/Dry: Like many other filtering effects, it is not as simple as Wet/Dry. Destructive interference happens between the wet and dry signals to varying degrees as this slider is adjusted, so new & interesting notch and comb filtering effects evolve as you move this from wet to dry.

Distort: This adds some nonlinear response to the filter. In addition to adding audible distortion, this parameter has the effect of making the filter less resonant. Set to zero for a "clean" filter sound.

Tempo: The filter can be modulated by an LFO. This sets the rate.

LFO Type: The shape of LFO you wish to use.

Subtr.: Subtract. Make output of filter negative, thus changing which parts of the signal will cancel when using Wet/Dry mix.

St. df: Stereo Difference. Delay between LFO's for left and right channels.

Width: Width of the LFO, also think of this parameter as LFO Amplitude.

Fb: Feedback amount. Positive or negative values make the feedback positive or negative. This parameter makes the filter more resonant.

LPF Stg: Number of first order low pass filter stages.

HPF Stg: Number of first order high pass filter stages. HPF and LPF are series processed, so combining the two creates a band pass filter.

Depth: Filter "start frequency". Set Width and E. Sens to zero then use this to manually sweep the filter with a MIDI message. More simply, use this parameter with a MIDI expression pedal to make Synthfilter behave as an interesting type of wahwah.

E. Sens: Envelope Sensitivity. This adjusts how much the filter sweep responds to the dynamics of the input signal. Numbers greater than zero cause the filter to sweep upward as your playing gets louder. Numbers less than zero cause the filter to sweep downward as your playing gets louder. Set to zero for no effect. Use "Depth" to adjust the filter start frequency. For example, if filter is to sweep downward, set Depth to a large number and E. Sens to a negative number.

A. Time: Attack Time. Sets how quickly the envelope detector responds to input dynamics.

R. Time: Release Time. Sets amount of time for envelope detector to release the build-up from loud dynamics.

*Attack and Release are measured in milliseconds.

*Attack and Release times have no effect if E. Sens is set to zero.

Offset: Separation between High Pass and Low Pass filters. If both HP and LP filters have stages 1 or more, this adjusts the bandwidth of the resulting bandpass filter. Set to zero, a bandpass filter with a narrow peak will result, or set large, the bandpass filter will be wider.

Four Bands volume modulated with two LFOs. You can select for each LFO the type, Tempo(speed) and LR difference (St.df). Cross points are to defining the band frequency range.

20-> Cross1 = Low
Cross1->Cross2 = Mid Low
Cross2->Cross3 = High Low
Cross3->26 KHz = High

The Combi choice select the LFO for each band

1 = LFO1
2 = LFO2
o = Bypass
x = Disable Band.

Convolotron A module optimized for physical speaker/cabinet modeling.

A lame EE joke: "If you do that once, I will convolve your face with a Dirac Delta. If you do it again, I will convolve your head with an impulse train."

Some jargon associated with this effect:

IR Impulse Response. Filters are often classified according to their impulse response. An impulse response is the output of the filter (or physical system) resulting from application of a short-duration, one-time, "impulse" of energy. An audio signal can be constructed mathematically as a series of such impulses separated by an infinitesimal period of time, and scaled by a certain factor representing the real-time amplitude of the signal. If one impulse is applied to a filter, the filter begins to react with its characteristic impulse response. Then if another impulse is applied a little later, the same impulse response is invoked and is added to the first. If we think of the audio signal as a rapid succession of impulses, then the result of adding up all the impulse responses will reconstruct what the physical system will do in response to the input signal. This process is called convolution.

Convolution. A process of successively adding the impulse responses for any signal that has been or will be applied to the system. In the case of realtime audio processing the impulse responses are causal in nature, so we can ignore the future and apply the effects only for what has already occurred.

Convolotron. A rakarrack module that takes an audio file as an IR, and convolves it with the input signal. This signal may be an IR recorded from an amplifier, a microphone, a cheap computer speaker system, horns, bells, gongs, reverb...kicking a door...

Some caveats: Convolotron is very CPU hungry. Rakarrack does straight-forward time-domain convolution. Another Linux software application to compare is JConv, which is not as CPU hungry because there are some frequency domain math tricks which allow the user to sacrifice some quality to reduce processing load. As a result, Rakarrack requires some "horsepower", but is better suited to amplifier/cabinet modeling assuming your CPU can handle it.

Parameters

Wet/Dry: Self explanatory. For amp responses, most of the time you will want this all wet.

Pan: The effect is processed in mono (L & R are mixed on the input). Pan routes the output left or right.

Level: The actual IR file will determine the gain of the effect. This helps you adjust to a good level.

Damp: Darkens the sound of the IR.

Fb: Feedback. Pretty self explanatory.

Length: Measured in milliseconds. This tells Convolotron where to truncate the IR. Set this to the maximum value your CPU can afford without Xruns. A longer Length will result in a more true model of the system being convolved.

Safe Mode: Automatically limits Length internally based on sample rate and info from your /proc/cpuinfo to help prevent freezing your computer.

Preset: Several amp IR's are available by default for your convenience.

User: You may download or generate your own IR's. You can even do silly things like loading ring tones or segments of a song (if you like to experiment). Must be wav format, and keep in mind that convolotron only uses the first "Length" milliseconds of the file...so if you have a file with 100ms silence at the beginning, you will get nothing but silence on the output.

Looper

Autism: A pervasive developmental disorder characterized by severe deficits in social interaction and communication, by an extremely limited range of activities and interests, and often by the presence of repetitive, stereotyped behaviors.
(The American Heritage® Dictionary of the English Language, Fourth Edition copyright ©2000 by Houghton Mifflin Company)

It's a joke...

Wet/Dry: Balance the mix of the recorded loop with the incoming signal to the output.

Level 1: Volume level of track 1. '64' is unity. Higher is louder than the original signal, lower is more quiet.

Level 2: Volume level of track 2.

Tempo: Set the tempo of the song. This control responds to TapTempo. However, time stretching is not used in the Looper, so changing Tempo during recording, or on a recorded loop will have no effect on the recorded tempo.

Time Sig: Time Signature. Use this to set the beat of the song. The loop will be quantized to the nearest measure.

MS: Metronome Sound. You can select N (Normal mode) to hear a "mark" on every first beat of the measure, then a lower "tick" for the remaining beats, or H (High tick) to hear the "mark" sound on every beat, or L (Low tick) to hear the lower tick sound on every beat.

Reverse: Play loop in reverse.

Auto Play: If this is checked, the looper will begin to play automatically after recording the first time. It also will cause the Record button to start play automatically. Disable this option to synchronize Looper with Jack Transport

Play/Stop: What more needs to be said?

Pause: Pauses whatever is happening.

Record: Unless AutoPlay is selected, Record only arms recording. Play/Stop will set the tape rolling. First time, it begins recording and will continue to record until you push record again. At this point you have defined the length of the loop. If you press record again, everything you play will be overdubbed onto the active track(s). To erase the track, you need to have the track active and select "clear". This operates on the active track only if R1 or R2 (corresponding to the track) is selected. This allows you to record on track 2 while hearing track 1.

R1, R2: Enable recording on track 1 and track 2, respectively. If neither are checked, then the record button will not do anything.

Track 1, 2: Enable Play, Stop, Record on the selected track. Again, both the track and the R button need to be selected to record on the specified track.

Clear: Erase the selected track and set length to zero.

Lnk: Link track 1 and track 2 with the same length. Activate this if you want track 1 and 2 to play at the same time. Deactivate if you want something like a verse-chorus-verse-chorus song structure where track 1 is the chorus part, and track 2 is the verse part.

M: Enable Metronome. Note Looper has its own metronome so it is able to synchronize with your play/pause commands. Pressing play resets the metronome to beat 1, no matter where it is in the measure. You will probably confuse yourself if you try to use the master metronome and Looper metronome simultaneously.

Example sequence of operation:
Set Wet/Dry to 0, Level 1 to 64, Auto Play Active, Track 1 active, R1 active. Press record and begin to play into the loop. When finished, press record again, and your loop will immediately start up from the beginning. Play along, or press record again to add more parts to the loop. Harmonizer with Octave down for bass simulator can be of assistance. MuTroMojo and Synthfilter with reverse envelope sweep can be of assistance for bongo drum sounds.

MutroMojo

The name is derived from two well known effects using the same filter topology: Mutron, Mojotron. For those not well accustomed to classic analog pedal effects, these were envelope controlled filters built upon the state-variable-filter. This filter structure has three "states" which can be extracted: High pass, Low pass, Band pass. This is easy to implement digitally, and is much more efficient for the processor than to compute the three filter bands as separate filters.

MuTroMojo allows you to adjust the three bands at whatever levels you want. This flexibility allows anything from various wahwah effects to synth filter effects, and possibly even sounds of the moogerfooger (though Synthfilter structure is better suited to "moog-like" sounds). Like any good Rakarrack effect, it has an LFO synchronized to Tempo. This means Tap Tempo will set the rate of the LFO effect.

Additionally, you have the envelope follower applied with the E. Sens parameter, and smoothed by Smooth. Here's the breakdown:

Wet/Dry: It's obvious that this mixes wet and dry signals, but this interacts like a phaser or flanger. By mixing some Dry into higher order filters (increase Stages) you can create notches in the frequency response. With individualized control over HP, BP, and LP filter bands, you can create some interesting comb filter responses. This is more than simply "amount of effect".

LP, BP, HP: Low Pass, Band Pass, High Pass, respectively. Greater than zero mixes the output in phase, less than zero mixes the output out of phase. This can be alternated to create different types of destructive interference between filter bands. For the more technically minded, you can dust off your filter cookbook and tune in to a wide variety of filter responses. This can also be used as a sort of parametric EQ or tone stack.

Stg: Stages. Each filter stage is a second order filter. "Stg" sets the number of filters in series. The number of stages will improve the amount of rejection above/below the cutoff. A tip: You can use this module as a compromise for up sampling quality when using up sampling. For example set Wah and Range to maximum levels with LP at 32 and BP, HP at 0. With this configuration you have a 12dB/Octave low pass filter set at about 6kHz. For clean electric guitar, you won't lose much of the original signal because most guitar pickups form such a filter at 5kHz. Then the upsampling can be set to Zero Order Hold, and this filter will smooth it out. To improve quality, increase the MuTroMojo Stages. In this mode it acts as an "Interpolating Filter", which is able to reconstruct the missing information between samples. The higher quality upsampling options do the same thing, but they use much higher order Sinc interpolation filters and try to maximize bandwidth. Unfortunately these higher order filters use more CPU, and they also have bad transient responses. If you can sacrifice the bandwidth, you can use lower order filters for nearly the same result. This is where the MuTroMojo can give you more choice with whether you want to trade bandwidth or CPU usage while balancing transient characteristics. If none of that made sense, then maybe you can ask on the rakarrack IRC channel for a better explanation...or look up sample rate conversion and interpolation for discussion of the trade-offs and benefits. Otherwise, just experiment with settings that sound good to you, and forget about the technical details of why.

Width: Deviation of the LFO. This describes how far it sweeps.

Tempo: LFO Rate expressed in beats per minute. This is synchronized when Tap Tempo is active.

Res.: Resonance. Increasing this makes a higher and more narrow peak at the filter resonant frequency.

Range: Upper bound on the filter sweep range.

Wah: Sweeps the filter resonant frequency from the minimum bound (hard coded in each preset) up to the upper bound (set by Range). This is the parameter you would want to map to a MIDI expression pedal for a wah wah effect.

E. Sens: Amount that filter sweep responds to input signal dynamics. Positive number means the filter center frequency sweeps upwared, negative means it sweeps downward. Zero means the filter does not respond to dynamics of the input.

Smooth: Smooth the envelope detector shape. This has no effect unless you have E. Sens set to something other than zero. If E. Sens is set to a useful level, then the Smooth parameter adjusts how quickly the filter center frequency responds to changes in signal dynamics.

Echoverse

This effect is intended to be an "in your face" wall of sound.
This has been adapted from the normal "Echo" effect, but with some modifications that help improve the sound and behavior. With Wet/Dry mix, you will notice it is possible to get 100% wet (echoes only). The reverse is a little more "clean". More importantly, this effect synchronizes to master Tempo, and is responsive to TapTempo control.

SubDivision allows you to divide delay times to 2,3,4... beats per measure. Also added is "Extra Stereo" effect that makes the effect come alive in 3D space (assuming you're amplifying or recording in stereo).

Wet/Dry: Direct/Reverberant mix. It's like changing the distance you are located from the sound source.

Reverse: Same as Echo

Pan: Move the effect to the left or right

Tempo: Beats per minute. This is synchronized to TapTempo when active.

LRdl.: Difference between left channel and right channel delay times.

Fb.: Feedback. Other descriptive words are Repeat, Regen or Tail.

SubDivision: Delay time can by synchronized to the length of a musical measure subdivision. For example, the image to the left indicates delay time is set to 1/2 notes of a song played at a Tempo of 90 beats per minute.

Damp: High frequency damping.

E.S.: Extra Stereo. This is the amount of the effect applied.

Angle: This is the angle from which the sound appears to be coming.

Coil Crafter - Pickup Emulation/Converter

This unit contains two filters tuned to typical pickup frequency responses. Origin is used to match the pickup you're using. This filter "undoes", or reverses, the frequency response of your source pickup. "Destiny" is what kind of pickups you want it to sound like is in your guitar. This applies the frequency response of the pickups selected.

Two caveats:

A) This does not make your Gibson Les Paul sound like a strat. It makes it sound like you put single coil pickups in your Les Paul. The Tone control helps to take off some of the lows so you can somewhat emulate different guitar body types.

B) You have to adjust parameters for the "Origin" pickup in order to cancel the effects of the pickups you are actually using. Without good electronics lab test equipment, this is largely guesswork tuned into range by your ears. If it sounds good, then who cares if it's technically right.

Gain: Volume control

Tone: Simple low cut

Origin: Pickups you are using. This will "undo" your pickup's response.

Freq1: Resonant frequency of origin pickup.

Q1: Resonance of origin pickup.

Destiny: This is the type of pickup you wish to emulate.

Freq2: Resonant frequency of desired pickup response.

Q2: Resonance of desired pickup.

Pos: Uses harmonic exciter to emulate switching to neck position pickup.

Low Shelf Tone Booster

This is a simple Low Shelf Filter that can be used as Tone Booster, is good to put after Distortion effects in order to adjust the distortion tone or cut undesired freqs. Also with a really hard gain you can obtain a nice distortion in combination with the "invisible" Compressor/Limiter that rakarrack has on the end of the chain, if you try that be sure to low the main Output volume.

Vocoder

For those not already familiar with vocoders, this is the effect responsible for many robot sounds on several popular songs.

The basic structure is this:

Your voice from a microphone needs to be connected to the "Aux" input of Rakarrack. Second, you will have some sound source coming through Rakarrack's normal processing chain. This is likely a guitar or synthesizer. When you speak into the mic, you will hear the sound of your instrument with a robot-like voice. The best types of signals are constant and rich in frequency content. For example, distorted guitar chords are a good "carrier" for the Vocoder. Various synthesizer strings sounds or brass emulations are also good sources for the voice.

This particular implementation uses the analog Vocoder model. These units were implemented with several parallel filter banks. One filter bank was used on the voice, and the other on the instrument. The bank on the voice was used to "find out" what frequency bands are being used by the voice, then the power from each of these bands was used to scale the corresponding bands on the instrument so it has more or less power in these bands. Briefly put, the power in the signal of each frequency band from the mic was used to control the volume on each corresponding frequency band for the incoming instrument. Think of two 32 band equalizers. The first 32 band EQ has a mic plugged into it. As you speak into the mic, you can see the VU meter bars bounce on each corresponding band (if this is the kind of EQ that displays VU bars for every band). Imagine something being linked to the VU meter bars that can push the sliders up and down on the Equalizer used to process the instrument. This is what a vocoder does.

Rakarrack uses a 32-band filter bank to implement the effect. These are all second-order band pass filters spaced evenly on a logarithmic scale from 300Hz to 3600Hz, thus covering most of the important formant frequencies.

Additionally there are some basic hard-coded utilities to help in processing the voice channel:
Gate: Anything below -75dB is gated. You can "cheat" the gate with the Input level adjustment. If too much noise is getting through to the vocoder when you're not talking, you can decrease the input level until the noise floor disappears. Of course, if you have a noisy preamp, mic, lots of background noise...etc, this gives you poor signal/noise ratio, then there is little we can program into the application to help improve the dynamics :-(.

Compressor: A simple compressor is applied to the vocals with a ratio of approximately 2:1. It is implemented using add/subtract/multiply/divide instructions to avoid CPU costly log and pow functions. As a result, the compression curve looks more like saturation on a vacuum tube or JFET, and the compression ratio increases steadily with the input signal level. Consequently, it is a soft-knee compressor and lends itself well to vocal processing as a easy