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Effects

Lineal EQ Compressor Distorsion Overdrive Echo Chorus Phaser Analog Phaser Flanger Reverb Parametric EQ Cabinet Emulation AutoPan/Stereo Expander Harmonizer Musical Delay Noise Gate WhaWha AlienWha Derelict

MIDI Converter

Tuner

Credits

MIDI Implementation Chart

License

-h

--help

Shows the command line options and exits.

-n

--no-gui

Start the application without GUI

-l file

--load=file

Load file single Preset

-b file

--bank=file

Load a file Bank

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.

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.

Adapted from the ZynAddSubFX 'Distorsion'

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 noisey 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 Distorsion

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.

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

Freq: 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 :) .

Freq. : 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...hold on for future releases.

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.

Adapted from the ZynAddSubFX Alienwah

A cross between a Flanger, Phaser and a Wah-Wah...