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<meta http-equiv="CONTENT-TYPE" content="text/html; charset=utf-8"><title>Rakarrack Help Effects</title>
<meta name="AUTHOR" content="Josep Andreu">
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<meta name="CHANGED" content="20100112;22402500">
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<h2>EFFECTS</h2>
<p><a href="#eql">Lineal EQ</a> <a href="#comp">Compressor</a> <a href="#dist">Distortion</a>
<a href="#ovrd">Overdrive</a> <a href="#eco">Echo</a> <a href="#chor">Chorus</a> <a href="#phas">Phaser</a>
<a href="#aphas">Analog Phaser</a> <a href="#flan">Flanger</a> <a href="#reve">Reverb</a>
<a href="#eqp">Parametric EQ</a> <a href="#cabe">Cabinet Emulation</a>
<a href="#pan">AutoPan/Stereo Expander</a> <a href="#har">Harmonizer</a> <a href="#delm">Musical Delay</a>
<a href="#gate">Noise Gate</a> <a href="#wha">WahWah</a> <a href="#awha">AlienWah</a>
<a href="#dere">Derelict</a> <a href="#Valve">Valve</a> <a href="#Dual_Flange">Dual Flange</a>
<a href="#Ring">Ring</a> <a href="#Exciter">Exciter</a> <a href="#DistBand">DistBand</a>
<a href="#Arpie">Arpie</a> <a href="#Expander">Expander</a> <a href="#Shuffle">Shuffle</a>
<a href="#Synthfilter">Synthfilter</a> <a href="#VaryBand">VaryBand</a> <a href="#Convolotron">Convolotron</a>
<a href="#Looper">Looper</a> <a href="#MuTroMojo">MuTroMojo</a> <a href="#Echoverse">Echoverse</a>
<a href="#CoilCrafter">CoilCrafter</a> <a href="#ShelfBoost">ShelfBoost</a> <a href="#Vocoder">Vocoder</a>
<a href="#Sustainer">Sustainer</a> <a href="#Sequence">Sequence</a> <a href="#Shifter">Shifter</a>
<a href="#StompBox">StompBox</a> <a href="#Reverbtron">Reverbtron</a> <a href="#Echotron">Echotron</a> <a href="#StereoHarm">StereoHarm</a> <a href="#CompBand">CompBand</a> <a href="#Otrem">Opticaltrem</a> <a href="#Vibe">Vibe</a> <a
href="#Infinity">Infinity</a><br></p>
<p>The program has many effects, you can select any
of the available effects. Ten can be used simultaneously. It
cascading process, following the order that appears on the screen,
from left to right and top to bottom. The order is configurable by
the user via the button "Put in your order Rack" giving
access to this screen. The effect selected moves up or down using the
arrows. The double arrow button interchange the selected effects
between the two browsers. Effects can be selected by type clicking the type
buttons</p>
<p>The effects displayed in the main windown can be switchd draging their
label to another one, is a fast way to change the order.</p>
<p> The Hide/Show button, hide the unused effects or show all of the
chain.</p>
<p> We don't want to cut the user creativity doing effects, but they are
some good rules positioning effects, almost for the Noise Gate and the
Expander if it's used to this purpose, as we say on the FAQ, rakarrack don't
generate any noise, or not should :-), what rakarrack does is amplify the
external noises, then for better control parameters to reduce this noises is a good
rule put the Noise Gate or the Expander at the first position on the chain.
</p><p> More info about effects is available in our <a href="http://sourceforge.net/apps/mediawiki/rakarrack/index.php?title=Main_Page">wiki</a></p><p>
</p><p><img src="imagenes/order.jpg" alt="rakarrack Order Window" align="middle" border="1" height="284" width="400"></p>
<p>The effects have two common elements. The "On"
button and "Preset" input choice. These individual effect
presets are not modified by the user and in most cases are those that
Octavian Paul Nasca defined for the purposes of ZynAddSubFX. The
value of preset individual is not stored in presets general and can
not relate to the parameters in effect.</p>
<h3>Parameter Adjustment</h3>
<p> All the parameters managed with a slider widget
can be adjusted fine with the mouse wheel or the Up/Down - Left/Right
arrows in the keyboard decrease/increase value by "1",
Shift+(Left/Right Arrow) decrease/increase value by "10", Ctrl+(Left/Right
Arrow) decrease/increase value by "100" also you can navigate and adjust trough the
parameters with you computer keyboard with the Tab, Up/Down arrow and
space bar keys.</p>
<h3>MIDI Control</h3>
<p> For control the parameter values via MIDI see
the <a href="#MIDI_Implementation_Chart">MIDI Implementation Chart</a>
for the complete list of MIDI message commands recognized, to easy
control rakarrack with MIDI messages use the MIDI Learn way.</p>
<h3>Saving preset considerations</h3>
<p> All the presets can be saved as single text file but, recently we added
Convolotron, Echotron and Reverbtron, this effects can use "User" files, the
program save the paths of this "User" files, if you want to share please be
sure that this paths can also be shared and of course you will need to bring
this user files too.
</p><h3><a name="eql"></a>EQ Lineal</h3>
<table border="0" cellpadding="5" cellspacing="2">
<tbody>
<tr>
<td>
<p><img src="imagenes/eq.jpg" alt="rakarrack Lineal EQ" align="bottom" border="1" height="250" width="185"></p>
</td>
<td style="background-color: rgb(251, 251, 251); vertical-align: top;">
<p>Adapted from the ZynAddSubFX Equalizer.</p>
<p>Gain: Overall output level</p>
<p><b>Q</b>: Resonance of individual filters. Generally helps smooth
extreme settings...or make extreme settings yet more extreme.</p>
</td>
</tr>
</tbody>
</table>
<p><a href="effects.html">Effects</a></p>
<p><a href="#top">Contents</a></p>
<h3><a name="comp"></a>Compressor</h3>
<table border="0" cellpadding="5" cellspacing="2">
<tbody>
<tr>
<td style="vertical-align: top;">
<p><img src="imagenes/compresor.jpg" alt="rakarrack Compressor" align="bottom" border="1" height="250" width="185"></p>
</td>
<td bgcolor="#fbfbfb">
<p>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. </p>
<p><b>A. Time</b>: Attack Time. Time in milliseconds for attack to
settle to 64% of final amount of compression.</p>
<p><b>R. Time</b>: Release Time. Time in milliseconds for gain to return
to 64% of it's initial setting.</p>
<p>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.</p>
<p><b>Ratio</b>: 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.</p>
<p><b>Knee</b>: 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.</p>
<p><b>Thrhold</b>: Threshold. Defines the onset of compression. </p>
<p><b>Output</b>: How much gain to remove from the final stage of the
compressor.</p>
<p><b>Peek</b>: Enable/Disable peek compression.</p>
<p><b>Auto Output</b>: 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.</p>
<p><b>Stereo</b>: Process each channel separately.</p>
</td>
</tr>
</tbody>
</table>
<p><a href="effects.html">Effects</a></p>
<p><a href="help.html">Table of Contents</a></p>
<h3><a name="dist"></a>Distortion</h3>
<table border="0" cellpadding="5" cellspacing="2">
<tbody>
<tr>
<td style="vertical-align: top;">
<p><img src="imagenes/dist.jpg" alt="rakarrack Distortion" align="bottom" border="1" height="250" width="184"></p>
</td>
<td bgcolor="#fbfbfb">
<p>Adapted from the ZynAddSubFX 'Distortion'</p>
<p>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).</p>
<p>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. </p>
<p>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.</p>
<p>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. </p>
<p>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.</p>
<p>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.</p>
<p><b>Sub Octv</b>: 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). </p>
</td>
</tr>
</tbody>
</table>
<p><a href="effects.html">Effects</a></p>
<p><a href="help.html">Table of Contents</a></p>
<h3><a name="ovrd"></a>Overdrive</h3>
<table border="0" cellpadding="5" cellspacing="2">
<tbody>
<tr>
<td>
<p><img src="imagenes/ovrd.jpg" alt="rakarrack Overdrive" align="bottom" border="1" height="250" width="185"></p>
</td>
<td style="background-color: rgb(251, 251, 251); vertical-align: top;">
<p>Adapted from the ZynAddSubFX Distortion</p>
<p>Same as Distortion, but without sub-octave.</p>
</td>
</tr>
</tbody>
</table>
<p><a href="effects.html">Effects</a></p>
<p><a href="help.html">Table of Contents</a></p>
<h3><a name="eco"></a>Echo</h3>
<table border="0" cellpadding="5" cellspacing="2">
<tbody>
<tr>
<td style="vertical-align: top;">
<p><img src="imagenes/eco.jpg" alt="rakarrack Echo" align="bottom" border="1" height="250" width="185"></p>
</td>
<td bgcolor="#fbfbfb">
<p>Adapted from the ZynAddSubFX Echo</p>
<p><b>Wet/Dry</b>: Mix level of echo with original.</p>
<p><b>Reverse</b>: Mixes reverse delay with forward delay.</p>
<p>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.</p>
<p>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?</p>
<p><b>Pan</b>: Sends the delay more to the right or left. Zero balances
it in the middle.</p>
<p><b>Delay</b>: Amount of time before you hear the echo.</p>
<p><b>Lrdl.</b>: 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.</p>
<p><b>L/R Cr</b>: Left/Right Crossing. This mixes left and right
channels.</p>
<p><b>Fb</b>: 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.</p>
<p><b>Direct</b>: When is on, the effect only play the echoes.</p>
<p><b>Damp</b>: 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. </p>
</td>
</tr>
</tbody>
</table>
<p><a href="effects.html">Effects</a></p>
<p><a href="help.html">Table of Contents</a></p>
<h3><a name="chor"></a>Chorus</h3>
<table cellpadding="5" cellspacing="2">
<tbody>
<tr>
<td>
<p><img src="imagenes/chorus.jpg" alt="rakarrack Chorus" align="bottom" border="1" height="250" width="185"></p>
</td>
<td bgcolor="#fbfbfb">
<p>Adapted from the ZynAddSubFX Chorus</p>
<p>Tips</p>
<p><b>Tempo</b>: LFO Rate</p>
<p><b>Rnd</b>: Add noise to LFO to make it more natural sounding.</p>
<p><b>Intense</b>: Increase effect intensity. A digital Flanger or
Chorus uses a delay line with several hundred to several thousand
taps. When it changes taps to modulate the delay time, there is a
discontinuous change in the signal called "zipper noise". To
eliminate this, fractional delay times need to be calculated in order
to smoothly transition from one delay line tap to the next. The
default Rakarrack Phaser and Chorus use linear interpolation. This
mode makes use of a Lagrange interpolation polynomial to estimate
fractional delay times and some other tricks to maximize the flanger
depth and flatten the frequency response on the chorus. Adding the
mode switch adds the feature without breaking old presets.</p>
<p><b>St.df</b> : Stereo difference of LFO.</p>
<p><b>L/R Cr.</b> : Mix left into right, Right into left. At maximum
level, left and right channels are swapped.</p>
</td>
</tr>
</tbody>
</table>
<p><a href="effects.html">Effects</a></p>
<p><a href="help.html">Table of Contents</a></p>
<h3><a name="phas"></a>Phaser</h3>
<table cellpadding="5" cellspacing="2">
<tbody>
<tr>
<td>
<p><img src="imagenes/phaser.jpg" alt="rakarrack Phaser" align="bottom" border="1" height="250" width="185"></p>
</td>
<td bgcolor="#fbfbfb">
<p>Adapted from the ZynAddSubFX Phaser</p>
<p>More mild digital phaser with exponential LFO sweep.</p>
<p><b>Phase</b>: This is the offset for the "center" of the sweep. </p>
<p><b>Depth</b>: 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.</p>
<p><b>Fb</b>: Feedback. Some phasers title this "Regen". Feedback is
negative as this moves toward zero, positive from 64+. 64, strangely,
corresponds to zero feedback.</p>
</td>
</tr>
</tbody>
</table>
<p><a href="effects.html">Effects</a></p>
<p><a href="help.html">Table of Contents</a></p>
<h3><a name="aphas"></a>Analog Phaser</h3>
<table cellpadding="5" cellspacing="2">
<tbody>
<tr>
<td style="vertical-align: top;">
<p><img src="imagenes/Aphaser.jpg" alt="rakarrack Analog Phaser" align="bottom" border="1" height="250" width="185"></p>
</td>
<td bgcolor="#fbfbfb">
<p>ZynAddSubFX Phaser Seriously Hacked</p>
<p>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.</p>
<p>When should you want to use Phaser instead?</p>
<p>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.</p>
<p>B) You just plain like the original Phaser Effect and want to
use it.</p>
<p>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.</p>
<p>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.</p>
<p>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.</p>
<p><b>Wet/Dry</b>: 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.</p>
<p><b>LFO Type</b>: 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 :) .</p>
<p><b>Tempo</b>: LFO frequency.</p>
<p><b>Depth</b>: 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.</p>
<p><b>Width</b>: This is how far the LFO travels. It's the LFO amplitude.</p>
<p><b>Fb</b>: Feedback. Usually named 'Regen' on a phaser stompbox. For
deeper notches keep Fb negative for even number stages and positive for
odd number stages.</p>
<p><b>Mismatch</b>: 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.</p>
<p><b>Distort</b>: 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.</p>
<p><b>St. Diff</b>: 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.</p>
<p><b>Stages</b>: 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.</p>
<p><b>Subtract</b>: Subtract wet from dry instead of add.</p>
<p><b>Hyper</b>: 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.</p>
</td>
</tr>
</tbody>
</table>
<p><a href="effects.html">Effects</a></p>
<p><a href="help.html">Table of Contents</a></p>
<h3><a name="flan"></a>Flanger</h3>
<table cellpadding="5" cellspacing="2">
<tbody>
<tr>
<td>
<p><img src="imagenes/flanger.jpg" alt="rakarrack Flanger" align="bottom" border="1" height="250" width="185"></p>
</td>
<td style="background-color: rgb(251, 251, 251); vertical-align: top;">
<p>Adapted from the ZynAddSubFX Chorus</p>
<p><b>Intense</b>: Increase effect intensity. A digital Flanger or
Chorus uses a delay line with several hundred to several thousand
taps. When it changes taps to modulate the delay time, there is a
discontinuous change in the signal called "zipper noise". To
eliminate this, fractional delay times need to be calculated in order
to smoothly transition from one delay line tap to the next. The
default Rakarrack Phaser and Chorus use linear interpolation. This
mode makes use of a Lagrange interpolation polynomial to estimate
fractional delay times and some other tricks to maximize the flanger
depth and flatten the frequency response on the chorus. Adding the
mode switch adds the feature without breaking old presets.</p>
</td>
</tr>
</tbody>
</table>
<p><a href="effects.html">Effects</a></p>
<p><a href="help.html">Table of Contents</a></p>
<h3><a name="reve"></a>Reverb</h3>
<table border="0" cellpadding="5" cellspacing="2">
<col width="204"> <col width="487"> <tbody>
<tr>
<td>
<p><img src="imagenes/reverb.jpg" alt="rakarrack Reverb" align="bottom" border="1" height="250" width="185"></p>
</td>
<td style="background-color: rgb(251, 251, 251); vertical-align: top;">
<p>Adapted from the ZynAddSubFX Reverb</p>
</td>
</tr>
</tbody>
</table>
<p><a href="effects.html">Effects</a></p>
<p><a href="help.html">Table of Contents</a></p>
<h3><a name="eqp"></a>EQ Parametric</h3>
<table border="0" cellpadding="5" cellspacing="2">
<col width="206"> <col width="489"> <tbody>
<tr>
<td>
<p><img src="imagenes/peq.jpg" alt="rakarrack Parametric EQ" align="bottom" border="1" height="250" width="185"></p>
</td>
<td style="background-color: rgb(251, 251, 251); vertical-align: top;">
<p>Adapted from the ZynAddSubFX Equalizer</p>
</td>
</tr>
</tbody>
</table>
<p><a href="effects.html">Effects</a></p>
<p><a href="help.html">Table of Contents</a></p>
<h3><a name="cabe"></a>Cabinet Emulation</h3>
<table cellpadding="5" cellspacing="2">
<tbody>
<tr>
<td>
<br>
</td><td style="vertical-align: top;">
<p><img src="imagenes/cabinet.jpg" alt="rakarrack Cabinet" align="bottom" border="1" height="250" width="185"></p>
</td>
<td style="background-color: rgb(251, 251, 251);">
<p> Effect using the ZynAddSubFx Equalizer
engine</p>
<p>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.</p>
</td>
</tr>
</tbody>
</table>
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<h3><a name="pan"></a>AutoPan/Stereo Expander</h3>
<table cellpadding="5" cellspacing="2">
<tbody>
<tr>
<td>
<p><img src="imagenes/pan.jpg" alt="rakarrack AutoPan" align="bottom" border="1" height="250" width="185"></p>
</td>
<td style="background-color: rgb(251, 251, 251); vertical-align: top;">
<p>For the stompbox minded people, this is a
stereo tremolo effect. Extra Stereo setting creates a more spacious
stereo imaging effect.</p>
</td>
</tr>
</tbody>
</table>
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<h3><a name="har"></a>Harmonizer</h3>
<table cellpadding="5" cellspacing="2">
<tbody>
<tr>
<td style="vertical-align: top;">
<p><img src="imagenes/har.jpg" alt="rakarrack Harmonizer" align="bottom" border="1" height="250" width="185"></p>
</td>
<td style="background-color: rgb(251, 251, 251);">
<h3>Intelligent Harmonizer Explained</h3>
<p>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 <a href="#settings">Settings</a> 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:</p>
<h4>Normal Mode</h4>
<p>Is how a normal pitchshifter run, the pitch
ratio is fixed and can be selected in the <b>Interval</b> effect
parameter of the effect.</p>
<h4>Select Mode</h4>
<p>In this mode the pitch ratio is variable, this
value depends of the selected <b>Interval</b> parameter, and the <b>Note</b>
and <b>Chord</b> 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:</p>
<p>,6,Maj7,lyd,Maj(9),Maj7(9),6/9,+,m,m6,m7,m7(b5),m9,m7(9),m7(11)<br>
,m(Maj7),m(Maj7)(9),dim,dim7,7,7(Sus4),7(b5),7(9),7(#11),7(13),7(b9)<br>
,7(b13),7(#9),+Maj7,+7,1+8,1+5,(Sus4),2</p>
<h4>MIDI Mode</h4>
<p>This mode run in the same way as <b>Select
Mode</b> 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 <a href="#settings">Settings</a> 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 ... )</p>
</td>
</tr>
</tbody>
</table>
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<h3><a name="delm"></a>Musical Delay</h3>
<table cellpadding="5" cellspacing="2">
<tbody>
<tr>
<td>
<p><img src="imagenes/md.jpg" alt="rakarrack Musical Delay" align="bottom" border="1" height="250" width="185"></p>
</td>
<td bgcolor="#fbfbfb">
<h3>Musical Delay Explained</h3>
<p>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.</p>
<p>The Delay lines are measured in fractions of
quarter notes at the <b>Tempo</b> 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.</p>
<p>The center delay parameter, is the delay
between the two delay lines, and is the only one can be set equal to
zero.</p>
<p>The <b>Tempo</b> 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.</p>
<p>Off course you have Gain, Pan and Feedback
parameters for each delay line in order to adjust the desired effect.</p>
</td>
</tr>
</tbody>
</table>
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<h3><a name="dere"></a>Derelict</h3>
<table border="0" cellpadding="5" cellspacing="2">
<tbody>
<tr>
<td>
<p><img src="imagenes/derelict.jpg" alt="Derelict" align="bottom" border="0" height="250" width="185"></p>
</td>
<td style="background-color: rgb(251, 251, 251); vertical-align: top;">
<p>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.</p>
</td>
</tr>
</tbody>
</table>
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<h3><a name="gate"></a>Noise Gate</h3>
<table cellpadding="5" cellspacing="2">
<tbody>
<tr>
<td>
<p><img src="imagenes/ng.jpg" alt="rakarrack Noise Gate" align="bottom" border="1" height="250" width="185"></p>
</td>
<td style="background-color: rgb(251, 251, 251); vertical-align: top;">
<p>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.</p>
</td>
</tr>
</tbody>
</table>
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<h3><a name="wha"></a>WahWah</h3>
<table cellpadding="5" cellspacing="2">
<tbody>
<tr>
<td style="vertical-align: top;">
<p><img src="imagenes/ww.jpg" alt="rakarrack WahWah" align="bottom" border="1" height="250" width="185"></p>
</td>
<td bgcolor="#fbfbfb">
<p> Adapted from the ZynAddSubFX
DynamicFilter</p>
<p>By now you're saying, "Ok, I get the Pan,
Freq, Rnd, St. Dif, and Depth thing, but what the..."</p>
<p>In addition to use as Auto-wah (LFO-modulated
Wah-Wah), this is also an envelope filter. </p>
<p>Currently, here are the three parameters
related to envelope control:</p>
<p><b>Amp S.</b>: 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.</p>
<p><b>Amp. S.I.</b>: 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. <b>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.</b></p>
<p><b>Smooth</b>: 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.</p>
</td>
</tr>
</tbody>
</table>
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<h3><a name="awha"></a>AlienWah</h3>
<table cellpadding="5" cellspacing="2">
<tbody>
<tr>
<td>
<p><img src="imagenes/aw.jpg" alt="rakarrack AlienWah" align="bottom" border="1" height="250" width="185"></p>
</td>
<td style="background-color: rgb(251, 251, 251); vertical-align: top;">
<p>AlienWah - Created by our hero, Paul Nasca Octavian originally for the ZynAddSubFX synthesizer.<br>
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".</p>
<p>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. </p>
<p>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.</p>
<p>Here is a description of the parameters:</p>
<p><b>Wet/Dry</b>: 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.</p>
<p><b>Pan</b>: 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.</p>
<p><b>Tempo</b>: LFO rate</p>
<p><b>Rnd</b>: Add random noise to LFO to make it sound somewhat more
"analog".</p>
<p><b>LFO Type</b>: Self explanatory</p>
<p><b>Phase</b>: 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.</p>
<p><b>St. df</b>: 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.</p>
<p><b>Depth</b>: This would be better termed "width" as this parameter adjusts the amplitude of the
LFO.</p>
<p><b>Delay</b>: 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.</p>
<p><b>Fb</b>: 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.</p>
<p><b>L/R Cr.</b>: 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.
</p>
</td>
</tr>
</tbody>
</table>
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<h3><a name="Valve"></a>Valve</h3>
<table cellpadding="5" cellspacing="2">
<tbody>
<tr>
<td>
<p><img src="imagenes/val.jpg" alt="rakarrack Valve"></p>
</td>
<td style="background-color: rgb(251, 251, 251); vertical-align: top;">
<p>New Effect based on Valve, Steve Harris LADSPA
plugin, filters, harmonic enhancer and some extra distortion where
added.</p>
</td>
</tr>
</tbody>
</table>
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<h3><a name="Dual_Flange"></a>Dual Flange</h3>
<table cellpadding="5" cellspacing="2">
<tbody>
<tr>
<td style="vertical-align: top;">
<p><img src="imagenes/df.jpg" alt="rakarrack Dual Flange"></p>
</td>
<td style="background-color: rgb(251, 251, 251); vertical-align: top;">
<p>Another Flanger alternative. <br>
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.</p>
<p><b>Wet/Dry</b>: Mix delayed
signal with Dry</p>
<p><b>Pan</b>:</p>
<p><b>L/R.Cr</b>:</p>
<p><b>Depth</b>: Frequency (Hz) of
the lowest notch frequency on lowest end of sweep range</p>
<p><b>Width</b>: LFO sweep width,
deviation measured in Hz. </p>
<p><b>Offset</b>: Percent
difference in delay between delay A and delay B.</p>
<p><b>Fb</b>: Feedback. 0 is
none, negative and positive.</p>
<p><b>LPF</b>: 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</p>
<p><b>Subtract</b>: Subtract delayed
signal instead of add</p>
<p><b>Th. zero</b>: Through
Zero. Delay Dry signal to 1/2 way between high and low delay
deviations of delayed signal.</p>
<p><b>Tempo</b>: Sweep speed </p>
<p><b>St.df</b>: Stereo Difference
between LFO's</p>
<p><b>LFO Type</b>:</p>
<p><b>Rnd</b>: Random. Add
random noise to LFO to help reduce unmusical effects of linear
interpolation between samples.</p>
</td>
</tr>
</tbody>
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<h3><a name="Ring"></a>Ring</h3>
<table cellpadding="5" cellspacing="2">
<tbody>
<tr>
<td>
<p><img src="imagenes/ring.jpg" alt="rakarrack Ring"></p>
</td>
<td style="background-color: rgb(251, 251, 251); vertical-align: top;">
<p>New
Effect based on Ring, Steve Harris LADSPA
plugin. <b>Auto Freq</b> recognizes the frequency of the note played (Only
monophonic sorry ..) in order that you can use Ring as a simple
monophonic synthesizer.</p>
</td>
</tr>
</tbody>
</table>
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<h3><a name="Exciter"></a>Exciter</h3>
<table cellpadding="5" cellspacing="2">
<tbody>
<tr>
<td>
<p><img src="imagenes/ex.jpg" alt="rakarrack Exciter"></p>
</td>
<td style="background-color: rgb(251, 251, 251); vertical-align: top;">
<p>New Effect based on Harmonic, Steve Harris
LADSPA
plugin.</p>
</td>
</tr>
</tbody>
</table>
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<h3><a name="DistBand"></a>DistBand</h3>
<table cellpadding="5" cellspacing="2">
<tbody>
<tr>
<td>
<p><img src="imagenes/db.jpg" alt="rakarrack DistBand"></p>
</td>
<td style="background-color: rgb(251, 251, 251); vertical-align: top;">
<p>Multi Band Distortion, use the Cross1 and
Cross2 to select the ranges of each band.</p>
<p><br>
20Hz -> Cross1 = Low<br>
Cross1 -> Cross2 = Mid<br>
Cross2 -> 20KHz = High<br>
</p>
<p>You can select a waveshape type for each band.<br>
</p>
</td>
</tr>
</tbody>
</table>
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<h3><a name="Arpie"></a>Arpie</h3>
<table cellpadding="5" cellspacing="2">
<tbody>
<tr>
<td style="vertical-align: top;">
<p><img src="imagenes/arp.jpg" alt="rakarrack Arpie"></p>
</td>
<td style="background-color: rgb(251, 251, 251); vertical-align: top;">
<p>"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.</p>
<p><b>Wet/Dry</b>: Amount of
arpeggiated accompaniment in the mix</p>
<p><b>Arpe's</b>: Mix pitch
shifted delay line with normal delay</p>
<p><b>Pan</b>:</p>
<p><b>Tempo</b>: Measured in beats
per minute</p>
<p><b>Subdivision</b>: 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. </p>
<p><b>LRdl.</b>: Same as for Echo</p>
<p><b>L/R.Cr</b>: Same as for Echo</p>
</td>
</tr>
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<h3><a name="Expander"></a>Expander</h3>
<table cellpadding="5" cellspacing="2">
<tbody>
<tr>
<td style="vertical-align: top;">
<p><img src="imagenes/exp.jpg" alt="rakarrack Expander"></p>
</td>
<td style="background-color: rgb(251, 251, 251); vertical-align: top;">
<p>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.</p>
<p>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.</p>
<p>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.</p>
<p>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.</p>
<p>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.</p>
<p><b>A. Time</b>: Attack Time, time to open the gate</p>
<p><b>R. Time</b>: Release Time, time it takes the gate to return to off
state</p>
<p><b>Shape</b>: 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.</p>
<p><b>Thrhold</b>: Threshold.
The gate is reducing signal volume below this level. Moderate
threshold setting with smaller number for shape is useful for string
swell effects.</p>
<p><b>Level</b>: Boost the output. 1 ... 127 represents a fractional increase in volume
from 0dB to 20dB</p>
<p><b>HPF</b>: High Pass Filter</p>
<p><b>LPF</b>: Low Pass Filter</p>
</td>
</tr>
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<h3><a name="Shuffle"></a>Shuffle</h3>
<table cellpadding="5" cellspacing="2">
<tbody>
<tr>
<td>
<p><img src="imagenes/shu.jpg" alt="rakarrack Shuffle"></p>
</td>
<td style="background-color: rgb(251, 251, 251); vertical-align: top;">
<p>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.</p>
</td>
</tr>
</tbody>
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<h3><a name="Synthfilter"></a>Synthfilter</h3>
<table cellpadding="5" cellspacing="2">
<tbody>
<tr>
<td>
<p><img src="imagenes/syn.jpg" alt="rakarrack Synthfilter"></p>
</td>
<td style="background-color: rgb(251, 251, 251); vertical-align: top;">
<p>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.</p>
<p>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.</p>
<p>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.</p>
<p>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.</p>
<p><b>Wet/Dry</b>: 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.</p>
<p><b>Distort</b>: 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.</p>
<p><b>Tempo</b>: The filter can be modulated by an LFO. This sets the
rate.</p>
<p><b>LFO Type</b>: The shape of LFO you wish to use.</p>
<p><b>Subtr.</b>: Subtract. Make output of filter negative, thus changing which
parts of the signal will cancel when using Wet/Dry mix.</p>
<p><b>St. df</b>: Stereo Difference. Delay between LFO's for left and right
channels.</p>
<p><b>Width</b>: Width of the LFO, also think of this parameter as LFO
Amplitude.</p>
<p><b>Fb</b>: Feedback amount. Positive or negative values make the
feedback positive or negative. This parameter makes the filter more
resonant.</p>
<p><b>LPF Stg</b>: Number of first order low pass filter stages.</p>
<p><b>HPF Stg</b>: Number of first order high pass filter stages. HPF and LPF
are series processed, so combining the two creates a band pass
filter.</p>
<p><b>Depth</b>: 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.</p>
<p><b>E. Sens</b>: 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.</p>
<p><b>A. Time</b>: Attack Time. Sets how quickly the envelope detector responds
to input dynamics.</p>
<p><b>R. Time</b>: Release Time. Sets amount of time for envelope detector to
release the build-up from loud dynamics.</p>
<p>*Attack and Release are measured in milliseconds.</p>
<p>*Attack and Release times have no effect if E. Sens is set to
zero.</p>
<p><b>Offset</b>: 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.</p>
</td>
</tr>
</tbody>
</table>
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<h3><a name="VaryBand"></a>VaryBand</h3>
<table cellpadding="5" cellspacing="2">
<tbody>
<tr>
<td>
<p><img src="imagenes/var.jpg" alt="rakarrack VaryBand"></p>
</td>
<td style="background-color: rgb(251, 251, 251); vertical-align: top;">
<p>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.</p>
<p>20-> Cross1 = Low<br>
Cross1->Cross2 = Mid Low<br>
Cross2->Cross3 = High Low<br>
Cross3->26 KHz = High<br>
</p>
<p>The <b>Combi</b> choice select the LFO for each band</p>
<p>1 = LFO1<br>
2 = LFO2<br>
o = Bypass<br>
x = Disable Band.</p>
</td>
</tr>
</tbody>
</table>
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<h3><a name="Convolotron"></a>Convolotron</h3>
<table cellpadding="5" cellspacing="2">
<tbody>
<tr>
<td style="vertical-align: top;">
<p><img src="imagenes/con.jpg" alt="rakarrack Convolotron"></p>
</td>
<td style="background-color: rgb(251, 251, 251); vertical-align: top;">
<p>Convolotron A module optimized for physical
speaker/cabinet modeling.</p>
<p>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."</p>
<p>Some jargon associated with this effect:</p>
<p><b>IR Impulse Response</b>.
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.</p>
<p><b>Convolution</b>.
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.</p>
<p><b>Convolotron</b>. 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...</p>
<p>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.</p>
<p> <b>Parameters</b></p>
<p><b>Wet/Dry</b>: Self explanatory.
For amp responses, most of the time you will want this all wet.</p>
<p><b>Pan</b>: The effect is processed in mono (L & R are mixed
on the
input). Pan routes the output left or right.</p>
<p><b>Level</b>: The actual IR file will determine
the gain of the effect. This helps you adjust to a good
level.</p>
<p><b>Damp</b>: Darkens the sound of the IR.
</p>
<p><b>Fb</b>: Feedback. Pretty
self explanatory.</p>
<p><b>Length</b>: 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.</p>
<p><b></b><b>Safe Mode</b>: Automatically
limits Length internally based on sample rate and info from your
/proc/cpuinfo to help prevent freezing your computer.</p>
<p><b>Preset</b>: Several amp IR's are available by
default for your convenience.</p>
<p><b>User</b>:
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.</p>
<br>
</td>
</tr>
</tbody>
</table>
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<h3><a name="Looper"></a>Looper</h3>
<table cellpadding="5" cellspacing="2">
<tbody>
<tr>
<td style="vertical-align: top;">
<p><img src="imagenes/loo.jpg" alt="rakarrack Looper"></p>
</td>
<td style="background-color: rgb(251, 251, 251); vertical-align: top;">
<p>Looper</p>
<p>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 <b> presence of repetitive,
stereotyped behaviors.</b><br>
(The American Heritage® Dictionary of the English Language, Fourth
Edition copyright ©2000 by Houghton Mifflin Company)<br>
<br>
It's a joke...<br>
</p>
<p><b>Wet/Dry</b>: Balance the mix of the recorded loop with the
incoming signal to the output.</p>
<p><b>Level 1</b>: Volume level of track 1. '64' is unity.
Higher is louder than the original signal, lower is more quiet.</p>
<p><b>Level 2</b>: Volume level of track 2.</p>
<p><b>Tempo</b>: 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.</p>
<p><b>Time Sig</b>: Time Signature. Use this to set the beat of the song. The
loop will be quantized to the nearest measure.</p>
<p><b>MS</b>: 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.</p>
<p><b>Reverse</b>: Play loop in reverse.</p>
<p><b>Auto Play</b>: 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</p>
<p><b>Play/Stop</b>: What more needs to be said?</p>
<p><b>Pause</b>: Pauses whatever is happening.</p>
<p><b>Record</b>: 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.</p>
<p><b>R1, R2</b>: Enable recording on track 1
and track 2, respectively. If neither are checked, then the
record button will not do anything.</p>
<p><b>Track 1, 2</b>: 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.</p>
<p><b>Clear</b>: Erase the selected track and set
length to zero.</p>
<p><b>Lnk</b>: 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.</p>
<p><b>M</b>: 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.</p>
<p>Example sequence of operation:<br>
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.</p>
</td>
</tr>
</tbody>
</table>
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<h3><a name="MuTroMojo"></a>MutroMojo</h3>
<table cellpadding="5" cellspacing="2">
<tbody>
<tr>
<td style="vertical-align: top;">
<p><img src="imagenes/mut.jpg" alt="rakarrack MuTroMojo"></p>
</td>
<td style="background-color: rgb(251, 251, 251); vertical-align: top;">
<p>MutroMojo</p>
<p>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.</p>
<p>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. </p>
<p>Additionally, you have the envelope follower applied with the E. Sens
parameter, and smoothed by Smooth. Here's the breakdown:</p>
<p><b>Wet/Dry</b>: 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".</p>
<p><b>LP, BP, HP</b>: 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.</p>
<p><b>Stg</b>: 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.</p>
<p><b>Width</b>: Deviation of the LFO. This describes how far it sweeps.</p>
<p><b>Tempo</b>: LFO Rate expressed in beats per minute. This is synchronized when Tap Tempo is active.</p>
<p><b>Res.</b>: Resonance. Increasing this makes a higher and more narrow peak at the filter resonant frequency.</p>
<p><b>Range</b>: Upper bound on the filter sweep range.</p>
<p><b>Wah</b>: 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.</p>
<p><b>E. Sens</b>: 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.</p>
<p><b>Smooth</b>: 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.</p>
</td>
</tr>
</tbody>
</table>
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<h3><a name="Echoverse"></a>Echoverse</h3>
<table cellpadding="5" cellspacing="2">
<tbody><tr><td style="vertical-align: top;">
<p><img src="imagenes/ecv.jpg" alt="rakarrack Echoverse"></p>
</td>
<td style="background-color: rgb(251, 251, 251); vertical-align: top;">
<p>Echoverse</p>
<p>This effect is intended to be an "in your face" wall of sound.<br>
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.</p>
<p>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).</p>
<p><b>Wet/Dry</b>: Direct/Reverberant mix. It's like
changing the distance you are located from the sound source.</p>
<p><b>Reverse</b>: Same as Echo</p>
<p><b>Pan</b>: Move the effect to the left or right</p>
<p><b>Tempo</b>: Beats per minute. This is synchronized to TapTempo when
active.</p>
<p><b>LRdl.</b>: Difference between left channel and right channel delay
times.</p>
<p><b>Fb.</b>: Feedback. Other descriptive words are Repeat, Regen or
Tail.</p>
<p><b>SubDivision</b>: 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. </p>
<p><b>Damp</b>: High frequency damping.</p>
<p><b>E.S.</b>: Extra Stereo. This is the amount of the effect
applied.</p>
<p><b>Angle</b>: This is the angle from which the sound appears to be coming.</p>
</td></tr></tbody>
</table>
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<h3><a name="CoilCrafter"></a>Coil Crafter</h3>
<table cellpadding="5" cellspacing="2">
<tbody><tr><td style="vertical-align: top;">
<p><img src="imagenes/coi.jpg" alt="rakarrack CoilCrafter"></p>
</td>
<td style="background-color: rgb(251, 251, 251); vertical-align: top;">
<p>Coil Crafter - Pickup Emulation/Converter</p>
<p>
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.</p>
<p>Two caveats:</p>
<p>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.</p>
<p>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.</p>
<p><b>Gain</b>: Volume control</p>
<p><b>Tone</b>: Simple low cut</p>
<p><b>Origin</b>: Pickups you are using. This will "undo" your pickup's
response.</p>
<p><b>Freq1</b>: Resonant frequency of origin pickup.</p>
<p><b>Q1</b>: Resonance of origin pickup.</p>
<p><b>Destiny</b>: This is the type of pickup you wish to emulate.</p>
<p><b>Freq2</b>: Resonant frequency of desired pickup response.</p>
<p><b>Q2</b>: Resonance of desired pickup.</p>
<p><b>Pos</b>: Uses harmonic exciter to emulate switching to neck position pickup.</p>
</td></tr></tbody>
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<h3><a name="ShelfBoost"></a>ShelfBoost</h3>
<table cellpadding="5" cellspacing="2">
<tbody><tr><td style="vertical-align: top;">
<p><img src="imagenes/she.jpg" alt="rakarrack ShelfBoost"></p>
</td>
<td style="background-color: rgb(251, 251, 251); vertical-align: top;">
<p>Low Shelf Tone Booster</p>
<p>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.</p>
</td></tr></tbody>
</table>
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<h3><a name="Vocoder"></a>Vocoder</h3>
<table cellpadding="5" cellspacing="2">
<tbody><tr><td style="vertical-align: top;">
<p><img src="imagenes/voc.jpg" alt="rakarrack Vocoder"></p>
</td>
<td style="background-color: rgb(251, 251, 251); vertical-align: top;">
<p>Vocoder</p>
<p>For those not already familiar with vocoders, this is the effect responsible for many robot sounds on several popular
songs.</p>
<p>The basic structure is this:</p>
<p>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.</p>
<p>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.</p>
<p>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.</p>
<p>Additionally there are some basic hard-coded utilities to help in processing the voice channel:<br>
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
:-(. </p>
<p>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 soft-knee compressor. Because of the way this compressor
behaves, increasing the Input level is tantamount to decreasing the
threshold and increasing ratio simultaneously. It is designed
this way so it works effectively without the user needing to know
anything about it. Incidentally, this same compressor is used to
create the Sustainer effect.</p>
<p>The take-home message with regard to the "invisible" utilities is that
the Input level parameter gives you a lot of power over the dynamics of
the microphone input, and can help in getting a good consistent vocal
sound onto the instrument being processed.</p>
<p>Here is the list of parameters:</p>
<p><b>Wet/Dry</b>: We have been over this many times.</p>
<p><b>Pan</b>: Move the processed signal to the left or right</p>
<p><b>Input</b>: Microphone input level. Explained in detail above.</p>
<p><b>Muf.</b>: "Muffle". Maybe more like "blur" or
"smear". This increases the averaging time on the power coming
into each frequency band. Adjusting to a large level has an
effect of making it sound like the voice is in a very large reverberant
room. Lower levels achieve a more articulate sound.</p>
<p><b>Q</b>: Resonance of the
filters. Increasing this number makes the filter bands more
narrow. Extremely high values make the filters so resonant it
sounds like metallic reverberation. This also "smears" the
vocals. Unless you are going for extreme sounds, the best range
is 65 to 90.</p>
<p><b>Ring</b>: If you are going for extreme sounds, this
is the ticket. This causes the voice from each mic band to be
multiplied with each instrument band. The adjustment is basically
modulation depth.</p>
<p><b>Level</b>: Final output level. </p>
</td></tr></tbody>
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<h3><a name="Sustainer"></a>Sustainer</h3>
<table cellpadding="5" cellspacing="2">
<tbody><tr><td style="vertical-align: top;">
<p><img src="imagenes/sus.jpg" alt="rakarrack Sustainer"></p>
</td>
<td style="background-color: rgb(251, 251, 251); vertical-align: top;">
<p>Sustainer</p>
<p>A very simple "no frills" soft knee compressor
good for making notes sustain. All you have to think about is
"sustain" and output volume.</p>
<p>This module was inspired by, and is built upon, the vocal compressor
used in the Vocoder. This compressor has a more "bright"
sound than the normal compressor. Read up on Vocoder for more
information about this compressor.</p>
<p>The Attack time is equal to the Release time, and this time is set to
50ms.</p>
<p><b>Gain</b>: Gain recovery after the compressor. This is a simple volume
control.</p>
<p><b>Sustain</b>: This lowers the threshold and increases the ratio as the number
increases. Adjust anywhere from mild compression to a more extreme
"breathing" compression sound.</p>
</td></tr></tbody>
</table>
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<h3><a name="Sequence"></a>Sequence</h3>
<table cellpadding="5" cellspacing="2">
<tbody><tr><td style="vertical-align: top;">
<p><img src="imagenes/seq.jpg" alt="rakarrack Sequence"></p>
</td><td style="background-color: rgb(251, 251, 251); vertical-align: top;">
<p>Sequence</p>
<p><br>
</p>
<p>8-step sequencer with multiple modes of
operation. The sequencer controls effects that are hard-coded
internally which can be activated by the Mode selector. To date
the sequencer includes various modes for modulating a band-pass filter,
amplitude, and there is also a pitch bending mode based on the code
used in Harmonizer and Shifter.</p>
<p>Some of the uses include a sequenced tremolo (try Tremor mode) or
something akin to the Sample/Hold filter (Stepper mode), or a sequenced
wah wah (lineal).</p>
<p><b>Preset</b>: Several built-in preset configurations exist to showcase the possibilities of the
effect.</p>
<p><b>Wet/Dry</b>: Nothing special here.</p>
<p><b>Sliders 1 - 8</b>: Level of effect at each step in
the sequence. If a filter, sliders set filter center
frequency. If Tremor, sets volume of signal at the corresponding
step in the sequence.</p>
<p><b>Tempo</b>: Rate in Beats Per Minute at which the sequencer changes to the next
step.</p>
<p><b>Q</b>: Filter resonance. This may or may not do anything
with modes that don't use a filter. For certain modes, it may be
used to control a "hidden" parameter, so it is worthwhile to try it to
see if it does something.</p>
<p><b>St. df</b>: Stereo difference. This delays the right
channel sequence from the left by the number of steps selected
(1-8). This makes it sound as though the sound source is moving
all around the room.</p>
<p><b>Range</b> For most modes, it provides a choice ranges over which the
sequence sliders control the parameter. For example, on a filter,
it controls the maximum and minimum center frequencies.</p>
<p><b>Mode</b> Select different modes of operation. The modes
include:</p>
<p><i>Lineal</i>: Filter & optional amplitude. Moves smoothly from one step to the
next.</p>
<p><i>Up Down</i>: Filter & optional amplitude. Returns to zero between
steps, and steps up to the level adjusted in the sequence sliders.</p>
<p><i>Stepper</i>: Filter & optional amplitude. Steps abruptly from
one level to the next. This is the most like a Sample/Hold
modulated filter.</p>
<p><i>Shifter</i>: Frequency shifter. Bends smoothly from one frequency
to the next</p>
<p><i>Tremor</i>: Amplitude. This is a sequenced Tremolo that can be
switched between two different stepping modes by the "Amp" switch.</p>
<p><i>Arpeggiator</i>: Frequency shifter. Pass directly from one frequency to
the next one, the semitone is adjusted by the sequence sliders,
semitone = "slider value / 10".</p>
<p><i>Chorus</i>: Frequency Shifter, this mode use small amount of pitch
shifter in order to generate a chorus effect.With St.df=0 normal
operation, with St.df=1 added Extra Stereo, with St.df=2, Q can
control the pan.</p>
<p><b>Amp</b>: Invoke amplitude modulation when available in the mode. In
Frequency shifter modes Amp is used to switch down the frequency.</p>
<p>Finally, here is a reward for those who read the Help:</p>
<p>What is a sequencer effect without a random mode? There is an
"Easter Egg" hidden in this effect that invokes a random
sequence. Pull all the step sliders (1-8) to zero. If the
sum of the values on these sliders is less than 4, then the Sequencer
will step randomly. To return to a sequenced operation, bring one
of the sliders to something greater than 4. At this point, the
Sequence queue contains the randomly generated levels from the last
complete cycle through the queue. In random mode, these are
refreshed on each cycle so the sequence does not repeat, but once you
leave random mode the current random sequence remains in the
queue. You can "erase" each of these one-by-one by adjusting the
corresponding slider.</p>
</td></tr></tbody>
</table>
<p><a href="effects.html">Effects</a></p>
<p><a href="help.html">Table of Contents</a></p>
<h3><a name="Shifter"></a>Shifter</h3>
<table cellpadding="5" cellspacing="2">
<tbody><tr><td style="vertical-align: top;">
<p><img src="imagenes/shi.jpg" alt="rakarrack Shifter"></p>
</td><td style="background-color: rgb(251, 251, 251); vertical-align: top;">
<p>Shifter Pitch Shifter</p>
<p>Pitch shifting effect built on the Harmonizer
DSP engine. This flows with the general concept of a synthesizer
pitch wheel or a whammy pedal.</p>
<p>This effect has the ability to shift pitch up or down an octave. <br>
In Trigger mode, the effect will cause the pitch to bend up (or down)
when the signal has exceeded the threshold. The rate for upward
bending is set by Attack. Time to return to normal pitch after
the signal has fallen below the threshold is set by Decay. A
second mode is available to make use of the Whammy slider for a pedal
controlled digital whammy effect.</p>
<p><b>Wet/Dry</b>: Mix some of the unaffected signal to the output.</p>
<p><b>Int</b>: Musical interval setting the maximum pitch deviation. In
Trigger Mode with interval value set to "0", the real interval is set
to "1" and can be controlled by the Whammy slider.</p>
<p><b>Gain</b>: Overall effect volume.</p>
<p><b>Pan</b>: Pan effect to left or right channel.</p>
<p><b>Attack</b>: Sets how quickly the pitch bends upward when input has exceeded the
threshold.</p>
<p><b>Decay</b>: Sets how quickly the pitch returns to "zero" after signal is below
threshold.</p>
<p><b>Thrshold</b>: Level at which the signal will trigger pitch bending.</p>
<p><b>Down</b>: Pitch bends up by
default. Check this box to make pitch bend downward.</p>
<p><b>Whammy</b>: If Whammy mode is selected, this acts as a smooth pitch
bend from "zero" (in tune) to a maximum set by Interval. This is
intended primarily for external MIDI controllers, whether it may be a
MIDI foot pedal, pitch wheel, Ardour automation, or a MIDI sequencer
software.</p>
<p><b>Mode</b>: Sets mode of operation.</p>
</td></tr></tbody>
</table>
<p><a href="effects.html">Effects</a></p>
<p><a href="help.html">Table of Contents</a></p>
<h3><a name="StompBox"></a>StompBox</h3>
<table cellpadding="5" cellspacing="2">
<tbody><tr><td style="vertical-align: top;">
<p><img src="imagenes/sto.jpg" alt="rakarrack StompBox"></p>
</td>
<td style="background-color: rgb(251, 251, 251); vertical-align: top;">
<p>StompBox</p>
<p>A stand-alone stompbox emulator.</p>
<p>Between the various equalization tools and distortion modules,
Rakarrack offers great flexibility to get a wide range of distortion
sounds, but this method requires much experimentation, while some
things are simply not possible. What if you are a person who
wants to dial in a good sound quickly, and don't enjoy tweaking
seemingly endless possibilities?</p>
<p>StompBox aims to simplify the process by giving the user a familiar
"StompBox" interface with only "Level", "Gain", and a 3-band EQ, which
generally covers all the features presented by a typical stompbox.</p>
<p>Additionally, most of these "Stomp Box" modes are physically informed
models of actual stompboxes, with the most notable features derived
from the circuit's schematic. Please enjoy these models as a
unique creation of their own, and not as something expected to be a
part-for-part replacement. It is my belief (as the developer)
that these models have some convincing characteristics, but I am aware
of many subtle things that are not taken into account. Further,
there are many variants of a specific type of pedal on the market; for
example, an Overdrive pedal is basically the same circuit for almost
all models marketed as "overdrive". The differences are minor
variations that the specific designer determined produced a better
sound. Even though the names of the modes indicate one model or
the other, the internal signal processing in Rakarrack includes
variations of its own: some to counteract unfortunate digital artifacts
and some applied as common "mods" that tweakers apply to their physical
stompboxes.</p>
<p>Perhaps a good analogy would be that of an artistic painting versus a
photograph. The Rakarrack stompbox models are more like an
artistic painting, where the artist (programmer) looks at the object
(schematic), "paint" (write code) the prominent features, then add
details that make it become an identifiable image of the original
object. In contrast, a photograph captures things exactly how
they appear, with artifacts such as resolution and lighting/color
distortions. Many commercial products attempt more of a
"photographic" approach to stompbox modeling, where each part and
component is modeled in detail. This of course, makes the CPU
usage increase dramatically while IMHO little value is added in terms
of making a *GOOD* sound (however they do make it much more true to the
original). I think there would also be risk of legal issues if we
attempted to model commercial stompboxes exactly and (especially)
present them to our users in that form. For the most part you
will be able to tell what stompboxes were used as a standard for a
specific type of sound, as the names are relatively tell-tale.
Hopefully we have kept things generic enough to avoid aggravating the
industry. If any person of such authority takes issue, let it be
known we are happy to make changes & remove things you feel are
infringing on your patent or trademark rights. However, being
free software, this is probably greater benefit in the form of free
marketing for these industries.</p>
<p>Last, not all of the stompboxes modeled actually have a 3-band
EQ. For these cases, one or two of the EQ bands is configured to
emulate the EQ control on the pedal while the others have a
configuration that matches the "theme" of the pedal, and they have no
effect at "0" setting. Thus, any pedal model will have EQ
controls that behave similarly to the original.</p>
<p>Now with all the explanation and disclaimers set aside, let's give some
insight to what this thing is able to do. The first important
parameter to understand is the "Mode" selector. This is where you
get to select the flavor of stompbox distortion you wish to model:</p>
<p><i>Amp</i>: This is a model twice derived from a tube preamp.
The general circuit flow follows that of a typical overdrive pedal
(Tubescreamer type), however the waveshaper uses dynamically modulated
symmetry with a soft clipping function somewhat emulating the behavior
of a vacuum tube. The reason it is considered twice derived is
because the overdrive pedal originally used the frequency contouring of
a tube preamp along with a soft diode clipper to (poorly) emulate the
characteristics of an overdriven tube. Rakarrack "Amp" model
therefore is a model of a model, but the clipping routine is much more
dynamic and "alive" than the average diode clipper. The
pre-emphasis filter was modified somewhat to pick up more of the lower
end to make it better suited to fat blues lead tones. The 3-band
EQ is constructed based on criteria derived partially from a typical
tube amp tone stack blended with some of the character of the overdrive
pedal tone control. If Low and Mid bands are left flat (zero),
then the high band behaves like the OD pedal Tone control.</p>
<p><i>Grunge</i>: This will explain itself to those who are familiar
with the source model. The original has only high and low EQ
bands. The Rakarrack model implements the mid band to allow for
some mid scooping, or a mid boost if desired. Set mid at "zero"
to get the most true behavior of the original.</p>
<p><i>Rat</i></p>
<p><i>Fat Cat</i>: Rat and Fat Cat are minor variations on the same
thing. In truth, the difference between the two is subtle.
The EQ behaves like the original if low and mid bands are left flat
(zero). The high band behaves as the original tone control.
The mid band EQ is designed to work on frequencies associated with the
nasal sound. The low band is very broad and can be adjusted to
make the pedal sound very "beefy".</p>
<p><i>Dist+</i>: The name says it all for any who know the
original. The original has no tone control, so all three bands
are fabricated. The EQ on this is of the same type as for the Rat
model. To get the sound of the original pedal, leave mid and low
flat, then push high to its maximum.</p>
<p><i>Death</i>: A good mode for a raw-edged chainsaw guitar sound and death
metal enthusiasts. One may easily guess the original model if it
is stated that the internal circuit is identical to "Grunge", only some
contouring modifications and 3 bands of EQ. This model, unlike
the original, has a gain control.</p>
<p><i>Mid Elves Own</i>: Say it out loud and it will will sound
similar to how the original stompbox name sounds. This model
excludes the sweepable mid-band EQ. Instead, the mid-band picks a
good place and leaves it fixed there. It is an intentional design
decision not to include the extra slider for mid sweep. This is
to maintain the simplicity of a generic interface for all models. The
Rakarrack Parametric EQ as well as MuTroMojo modules are capable of
providing a sweepable mid scoop or boost if desired.</p>
<p><i>Fuzz</i>:
The first kind of distortion to come in a stompbox. It makes
everything sound as you hear things while experiencing mild
electrocution from the chassis of a poorly grounded amplifier.
This does not capture any particular circuit, but rather summarizes an
era (although elements of the various different fuzz boxes were
considered in the design). The controls on this unit do not
follow the pattern of the stompbox models as fuzz originally was a 1 or
2 knob wonder. Here is the guide:</p>
<p><b>Level</b>: This does what it says</p>
<p><b>Gain</b>: This is mostly consistent with the name. This makes it
more distorted but at the same time the lows get cut from the input
increasingly as gain is increased. This is similar to how the
classic Fuzz Face behaves.</p>
<p><b>Low</b>: This has nothing to do with the low frequency EQ. This
controls the bias in the virtual "circuit". As you adjust up or
down, the waveform gets clipped increasingly asymmetrically in one
direction or the other. It has no effect at "0" setting, although the
clipping characteristic is naturally asymmetric. This setting can
make the fuzz sound rather "nasty", and is a good parameter to map to
the ACI to change dynamically.</p>
<p><b>Mid</b>: A mid-band EQ, but not like a normal mid-band EQ. This
one operates on the input before the distortion, and is tuned to act
like a stuck wah wah pedal. Settings less than zero cut mids from
the input mix, making the fuzz sound a little more "woolly".</p>
<p><b>High</b> This is a tone knob modeled according to the Big Muff tone
control. Positive numbers cut lows while boosting highs.
Negative numbers cut highs while boosting lows.</p>
</td></tr></tbody>
</table>
<p><a href="effects.html">Effects</a></p>
<p><a href="help.html">Table of Contents</a></p>
<h3><a name="Reverbtron"></a>Reverbtron</h3>
<table cellpadding="5" cellspacing="2">
<tbody><tr><td style="vertical-align: top;">
<p><img src="imagenes/rvt.jpg" alt="rakarrack Reverbtron"></p>
</td><td style="background-color: rgb(251, 251, 251); vertical-align: top;">
<p>Reverbtron</p>
<p>A convolution Reverb and delay processor.</p>
<p>Reverbtron is based upon the engine of Convolotron (thus the name), but
takes into consideration that time-domain convolution has expensive
taste for CPU cycles. Convolving a reverb impulse response in
real time is impossible without the use of special hardware dedicated
to intensive DSP functions unless something in the impulse response is
sacrificed for lower processing requirements.</p>
<p>In the case of Reverbtron, the full shape of the response it preserved
from the head to the tail, but true replication of the frequency
response is sacrificed. In short, the "magic" actually happens in the
IR conversion utility used to convert the .wav IR file to a .rvb file
for Rakarrack. Various elements of the waveform are considered,
then the .rvb file is constructed as a list of pairs: Time, Reflection
Amplitude. This file is actually a plain text file, and you may
edit this file or generate your own custom files. In fact you
could use plain utilities such as bash scripting, perl, octave, scilab,
etc. Internally Reverbtron convolves this directly with the
signal using the time index to determine at what delay amount to
perform the multiplication. When Length is reduce, Reverbtron
simply subsamples the file at even intervals. As one can see, the
quality of the resultant reverb relies 100% on the construction of the
.rvb file. This has the side effect for enabling future
improvements to Reverbtron for users who install Rakarrack from their
distribution repository. Then they only need to download .rvb
files as the Rakarrack team improves the IR file processing.</p>
<p>A tip to reduce CPU usage to an amount that even some slower processors
can easily manage is open the Rakarrack Preferences and set the internal
sampling rate to a low rate, down to 8kHz. It may seem a low quality
sound will result from using a samplerate of 8kHz, but with some magical
mathematics in the downsampling and upsampling process it is possible to
replicate the input signal perfectly in the upsampling routine, only
band-limited to 1/2 the sample rate. In other words, the loss in
quality is no worse than applying a steep low-pass filter at slightly
below 1/2 the sample rate, assuming you are using one of the sinc
interpolation up/down sampling selections. There will be non-musical
artifacts resulting from zero-order hold or linear interpolation,
although this may be acceptable to you if your CPU is too slow to handle
the higher quality selections. Almost all selections of resampling will
be faster than processing at a higher samplerate.</p>
<p>Here are the parameters:</p>
<p><b>Wet/Dry</b>: Amount of dry signal to mix with the convolved
signal. Notice that the IR file defines the maximum
wetness. The Fade parameter fades the head end of the IR where
the amount of dry is defined. Usually this works well by mixing
all wet and setting fade until it appears there is silence at the
beginning for a duration equal to the incoming instrument's note attack
time.</p>
<p><b>Pan</b>: Fade the output more to the left or right.</p>
<p><b>Level</b>: Helps to compensate for louder and softer IR volume
levels.</p>
<p><b>Damp</b>: Reduces high frequency content in the signal. This can help to tame poorly behaved IR
files, or simply to impose a different character on the IR.</p>
<p><b>Fb</b>: Feedback. Feed output to input. Be careful as this
has the same sensitivity and character as microphone feedback.
Setting initial delay or fade to a modest level helps to create more of
a regenerative sound instead of feedback. Damp can be used to tame the
feedback as well.</p>
<p><b>Length</b>: This is measured in the number of points to process.
A length of 800 means that literally 800 reflections will be processed,
but it also means 800 multiplication and addition instructions will
happen for every frame. This is an indication of the amount of
subsampling performed on the IR file. Contrary to intuition, a
high number does not always make the reverb sound better. There
can be multiple "sweet spots" along the range, even with numbers less
than 300.</p>
<p><b>Stretch</b>: Want your IR to reverberate for a longer period of
time? Have a long one and want to shorten it? Stretch
changes the time base. A negative number shortens the IR, a
positive number makes it longer. You can stretch some IRs long
enough to create good discrete echoes; and very complex ones at that.</p>
<p><b>I. Del</b>: Initial Delay. This parameter delays the entire
reverb response. This is a way of thumbing your nose at
physics. You can create reverb effects that don't happen in the
real physical world.</p>
<p><b>Fade</b>: Reduce the level of the initial response of the IR.
This can be used to increase the wet level on an IR that contains much
of the direct response.</p>
<p><b>Diffusion</b>: Diffuse the percussive echoes. Synthesizes an
artificial Head Related Transfer Function (HRTF) and performs a second convolution
on the output. Increasing the slider value increases the number of
reflections in the HRTF, and will better diffuse the discrete echo
sounds. For instruments with fast or percussive attacks this can be a
very necessary control to help make the reverb more smooth and natural
sounding.</p>
<p><b>ES</b>: Extra Stereo. Emulate a spatial effect in stereo. This works
approximately like this:
Signal->LPF->Right->LPF->Delay->Left->FeedbackToSignal....and it just
keeps feeding back on itself in a loop until it decays. The delay time
is set internally by a guess at the room size based on properties of the
IR file. The method assumes in a real room your ears will be most
sensitive to reflections from the walls on the left and right side. Each
time a sound reflects off a wall, generally it loses more high frequency
content than low frequency content, and a reflection will be delayed and
filtered by the time it passes your left ear, bounces off the right wall
and returns to your right ear. Anybody with technical understanding
will know this is a very general approximation, but the result is
pleasant, and certainly creates an illusion of space when listening in
stereo (particularly with headphones).</p>
<p><b>LPF</b>: Low Pass Filter. Sets the cutoff point where high
frequencies are damped in the ES option.</p>
<p><b>Safe</b>: Limits Length regardless of the setting.</p>
<p><b>User</b>: Browse for your own .rbv file. These files are
generated by invoking the rakverb command on a .wav format IR file.
"rakverb -i foo.wav" will generate a file called "foo.rbv" in the same
directory. This is the file you would want to load into
Reverbtron.</p>
<p><b>Preset</b>: Rakarrack comes with several IR files already processed
and ready to go so you don't need to search the web looking for impulse
responses.</p>
</td></tr></tbody>
</table>
<p><a href="effects.html">Effects</a></p>
<p><a href="help.html">Table of Contents</a></p>
<h3><a name="Echotron"></a>Echotron</h3>
<table cellpadding="5" cellspacing="2">
<tbody><tr><td style="vertical-align: top;">
<p><img style="width: 175px; height: 250px;" src="imagenes/ect.jpg" alt="rakarrack Echotron"></p>
</td><td style="background-color: rgb(251, 251, 251); vertical-align: top;">
<p>Echotron</p>
<p>Multi-tap delay with virtually unlimited taps (limited to 127,
but if more than that is desired, use Reverbtron). In the most
simple case, assign the timing and spacing of delay taps in a simple
text file, then load as a custom "user" file. Tempo is normalized
to 1 measure = 1 Second at 60 beats per minute. Simply think of a
"1.0" in the time column of the text file as "one measure". For
example, if you want 8th note echoes at Tempo, assign delays in
multiples of 0.125. Potential musical subdivisions are limited
only by your creativity and ability to do fractional mathematics as
pertains to musical rhythm and timing. If you don't like math,
but you like to experiment, then simply try a bunch of numbers less
than 6 and see what happens. Maybe someday we will make a GUI
editor to generate the text files</p>
<p>A more advanced feature of Echotron is the assignment of filters in each
delay line tap. In the text file, you can configure a state-variable
filter with center frequency, resonance, and individually mix the 3
bands, as with MuTroMojo. Up to 32 filters may be configured in the text
file. If you wish to bypass a filter in any line of the file, set
Stages to 0 (zero). The first 32 occurrences of filters with stages
greater than 0 will be processed. Any filter parameters defined after
the first 32 are ignored.</p>
<p>You can assign left/right panning in the text file to create interesting rotational
patterns.</p>
<p>There is a maximum delay time of 6 seconds, but there is no minimum
delay time; for example, a delay time of zero is perfectly acceptable
if using the filters to create a phaser or wah wah (be careful with the
feedback parameter). Other small delay times are possible for
creating chorus and flanger effects. You may also set the delay
times the same for multiple taps if you wish to create a comb filter
with the state variable filter, or even an 8-band equalizer is possible.</p>
<p>From the GUI, you can enable/disable the filters, enable/disable filter
modulation, enable/disable delay line modulation, and all is
synchronized to Tempo. You may also limit the number of Taps
processed from the file.</p>
<p>This effect is the swiss army tool for stereo ping-pong and rotational
delays, flangers, phasers (thanks to state variable filter) and even
stereo spatialization techniques. Hopefully we have done a good
job demonstrating the most noteworthy possibilities in the default
files installed with Rakarrack.</p>
<p>Here is a description of the parameters:</p>
<p><b>Wet/Dry</b>: Mix unprocessed signal with processed output.</p>
<p><b>Pan</b>: Pan processed output to left or right channel.</p>
<p><b>Tempo</b>: Beats per minute. Synchronizes with master TapTempo.</p>
<p><b>Damp</b>: High frequency damping in the feedback loop.</p>
<p><b>Fb</b>: Amount of Feedback (regeneration)</p>
<p><b>L/R Cr.</b>: Amount of blending left
& right channels. Less than zero means subtract left from right
& right from left. Greater than zero means adding left to
right, right to left. At +/-32 left & right are mixed
50/50. At +/-64, left/right are completely swapped.</p>
<p><b>Width</b>: Width of the LFO. This adjusts the LFO amplitude.</p>
<p><b>Depth</b>: Filter center
frequency. "0" means it is centered on the frequency designated
in the text file. >0 Shifts the filter up in frequency, <0
shifts them down. This is a good parameter to assign to a MIDI
expression pedal.</p>
<p><b>St. df</b>: Sets stereo time difference between LFO right and left
channels.</p>
<p><b>LFO Type</b>: Select the modulation shape.</p>
<p><b>AF</b>: Activate Filters. If the box is checked, filters defined in the
text file will be applied to the delay taps.</p>
<p><b>MF</b>: Modulate Filters. If the box is checked, modulation will be
applied to the filters' cut-off frequencies.</p>
<p><b>MD</b>: Modulate Delays. If the box is checked, the delay line will be
modulated (like a chorus or flanger).</p>
<p><b>#</b>: Sets the number of taps to process sequentially from the text
file. For example, if you have a text file that defines 20
echoes, you can limit it to use the first 2 or 3 or....whatever you
want up to 127.</p>
<p><b>File</b>: Select from several files supplied by the Rakarrack team. This
will include a broad spectrum of uses so you can get started using
Echotron without ever touching a text file.</p>
<p><b>User</b>: So you want to do
something you can't do with one of the default files. You edit
your own text file with the desired delay times, levels, and filter
pattern, browse to this file, select, and voila! Your Echotron has
morphed into a completely unique effect.</p>
<p><b>Text Files</b></p><p>
</p><p>Below is an example text file with explanation of each field and any caveats you may need to
know:</p><img style="width: 844px; height: 271px;" alt="rakarrack file .dly" src="imagenes/dlyfile.png"><br>
<br>
<p><b>Filter</b>: This field multiplies the tempo for the filter
LFO modulation. For example, if you are playing a song at 80 beats per
minute, but you wish to make the LFO half as fast as the echoes, you
enter a 0.5 in this field to adjust the filter LFO speed to 40 beats
per minute.</p>
<p><b>Delay</b>: This field multiplies tempo for the delay line
modulation. Same concept as for filter, but this is the LFO applied to
the delay line. Note there is not a field for delay time subdivision.
This is because you define this by the times you put into the Time
column, so such a field would be redundant.</p>
<p><b>Pan</b>: Ranges from -1.0 to 1.0. less than 0.0 is pan left,
greater is pan right. 0.0 puts the delay equally to left and
right. Anything magnitude of +/-1.0 or greater will be treated as
extreme left or right.</p>
<p><b>Time</b>: This is the real time at 60 beats per minute. An
easier way to think of this as a musician is increments of
measures. "1.0" means 1 measure at the tempo selected in the
effect GUI. The above indicates a quarter note in the first line,
and a 1/2 note in the second line.Ranges from -6.0 to 6.0</p>
<p><b>Level</b>: how loud you want the echo to return. 1.0 returns it
exactly as loud as it came it. More makes it come back louder,
less makes it softer. You can use both positive and negative
numbers.Ranges from -2.0 to 2.0</p>
<p><b>LP</b>: Mix State variable Low Pass Filter amount.</p>
<p><b>BP</b>: Band pass filter amount.</p>
<p><b>HP</b>: High Pass Filter amount.</p>
<p>*For LP, BP, and HP when filters are activated, these also adjust the
level. If you don't want the filter to have an effect at a
certain time, then set LP, BP, and HP all to 1.0.Ranges -2.0 to 2.0</p>
<p><b>Freq</b>: Filter center frequency.Ranges 20.0 to 20000.0</p>
<p><b>Q</b>: Filter resonance.Ranges 0.0 to 300.0 </p>
<p><b>Stages</b>: Number of filter stages. This is virtually
unlimited, but you can crash your CPU with the processing requirement
if you aren't careful.Ranges 1 to 16</p>
<p>Caveats:</p>
<p>Always separate the fields with a <TAB>, and not spaces. Echotron looks specifically for TAB
separation.</p>
<p>You can enter as many lines as you want, but only the first 127 will be used by the program.
</p>
<p>There are only 8 filters available. After the 8th line, filter parameters are ignored by the
program.</p>
<p>It is not wise to set Q values greater than 300, but there is no
arbitrary limit. At some high range the filter is likely to go
unstable.</p>
<p>It is not wise to set Level greater in magnitude to +/-1.0. There
is some normalization internal to Echotron, but it is best to think of
1.0 as the volume knob's maximum range.</p>
<p>It is best to copy one of the default files distributed with Rakarrack
and edit it as desired. Always save with an extension .dly so you
don't confuse it with other kinds of text files. </p>
</td></tr></tbody>
</table>
<p><a href="effects.html">Effects</a></p>
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<h3><a name="StereoHarm"></a>StereoHarm</h3>
<br>
<table cellpadding="5" cellspacing="2">
<tbody><tr><td style="vertical-align: top;">
<p><img style="width: 185px; height: 250px;" src="imagenes/sth.jpg" alt="rakarrack StereoHarm"></p>
</td><td style="background-color: rgb(251, 251, 251); vertical-align: top;">
<p>Stereo Harm</p>
<p>This is a Stereo Harmonizer, two voices in stereo, the SEL and MIDI
functions are same as Harmonizer, please read the Harmonizer
help, in this effect Chrm L and Chrm R where added, that is
Chroma, you can use Interval L/R set to "0" and modify this Chromas to
reach a "Open" stereo chorus effect. This Chorma parameters can be used
also in normal Mode. </p>
<p>In SEL and MIDI modes Chroma parameter doesent has effect, because the frequency is selected internally by the
program.</p>
</td></tr></tbody>
</table>
<p><a href="effects.html">Effects</a></p>
<p><a href="help.html">Table of Contents</a></p>
<h3><a name="CompBand"></a>CompBand</h3>
<br>
<table cellpadding="5" cellspacing="2">
<tbody><tr><td style="vertical-align: top;">
<p><img style="width: 185px; height: 250px;" src="imagenes/cpb.jpg" alt="rakarrack CompBand"></p>
</td><td style="background-color: rgb(251, 251, 251); vertical-align: top;">
<p>CompBand</p>
<p>Is Four band Compressor, using the Compressor effect, the Wet/Dry
parameter is included for mastering purposes,
probably if you use with a single instrument you will want to hear the
Wet. </p>
<p>Four bands are availables L(Low), ML (Mid Low), MH (Mid High), H
(High). You can control the ratio and threshold for each one of this
bands.</p>
<p>The Cross Sliders is to determine the frequency range of each band in the
form:</p>
<p>0-> Cross1 = Low</p>
<p>Cross1->Cross2 = Mid Low</p>
<p>Cross2->Cross3 = High Low</p>
<p>Cross3->26 KHz = High</p>
</td></tr></tbody>
</table>
<p><a href="effects.html">Effects</a></p>
<p><a href="help.html">Table of Contents</a></p>
<h3><a name="Otrem"></a>Opticaltrem</h3>
<br>
<table cellpadding="5" cellspacing="2">
<tbody><tr><td style="vertical-align: top;">
<p><img style="width: 185px; height: 250px;" src="imagenes/opt.jpg" alt="rakarrack Optocaltrem"></p>
</td><td style="background-color: rgb(251, 251, 251); vertical-align: top;">
<p>Opticaltrem</p>
<p>Short explanation of Tremolo, for those who are "newbies" to guitar
effects: A tremolo is like an automatic volume knob. You
set the rate and depth, and it's like having somebody change the volume
knob up and down at the set rate and amount set by depth. No
effects processor would be complete without a tremolo. In prior
versions of Rakarrack, there is the AutoPan function in the stereo Pan
effect. In Auto mode, the Pan is a tremolo, but is not a
"classic" sounding tremolo.</p>
<p>This module emulates the tremolo found in classic guitar amplifiers
(which also can be found in several stompbox tremolo units). The
reason for the name "Opticaltrem" is because of the photo-electric
device that was used to change the electrical resistance of the signal
path in the amplifier circuit. The device in that configuration
makes a tremolo effect when exposed to a pulsing light source.
Incidentally, this is the same type of component used in "Optical
Compressors". If you have ever heard that term, now you
know what it means.</p>
<p>Here is a more technical explanation:</p>
<p>The volume knob is nothing more than a variable resistor. The
volume knob varies the amount of signal current going into the next
stage of the circuit. In many old guitar amplifiers this effect
is emulated by a variable resistance that can be changed by exposing it
to light. Usually the variable resistance element consisted of a
light-proof container (opaque box) containing a Cds (Cadmium Sulfide)
cell and a lamp, or LED for more modern units. When Cds is
exposed to light, its electrical resistance decreases, and thus its
terminals behave like the resistance between the wiper and outer lug on
a potentiometer (volume control). </p>
<p>If the response of the pulsing lamp was perfectly linear with the
voltage from the LFO, and if the Cds cell response was perfectly linear
with the light, then the effect would sound exactly the same as
Pan. As it is with most physical electrical devices, most
things have interesting nonlinear properties. Even more, there is
a time dependency in the system. It takes time for a lamp to heat
up and turn on. When current is removed, there is still some heat
in the filament, and light fades out in a time-dependent way.
Then, the Cds cell itself has some "memory". The resistance does
not change instantly when light is applied and removed, nor does it
work the same for charging and discharging.</p>
<p>Perhaps that explains why there is a need for a special "Opticaltrem"
module in Rakarrack to emulate this behavior. We hope this will
bring to mind the sound of the tremolo in an old tube amp.</p>
<p><b>Depth</b>: Amount of effect. At full depth it is almost turning off entirely between
pulses.</p>
<p><b>Tempo</b>: Speed of pulsating sound.</p>
<p><b>Rnd</b>: Adds some random noise to the LFO to emulate the imperfections of an analog LFO.</p>
<p><b>LFO Type</b>: The shape you want it to use.</p>
<p><b>St.df</b>: Stereo difference between left and right LFO's.</p>
<p><b>Pan</b>: The effect can be panned to the left or to the right.</p>
</td></tr></tbody>
</table>
<p><a href="effects.html">Effects</a></p>
<p><a href="help.html">Table of Contents</a></p>
<h3><a name="Vibe"></a>Vibe</h3>
<br>
<table cellpadding="5" cellspacing="2">
<tbody><tr><td style="vertical-align: top;">
<p><img style="width: 180px; height: 250px;" src="imagenes/vib.jpg" alt="rakarrack Vibe"></p>
</td><td style="background-color: rgb(251, 251, 251); vertical-align: top;">
<p>Vibe</p>
<p> Chorus/Phaser effect emulating a rotating speaker.</p>
<p>This effect was added at the request of a Rakarrack user wishing to have a
software UniVibe. This effect is based on the original UniVibe circuit and
mathematically models components most likely to contribute to the overall
effect. Of course, the originals varied widely from one unit to the next
due to some components being manufactured under less advanced process
control. On these components we made our best guess and tuned by ear. This
is the same situation for those attempting the same kind of thing with
modern analog clones.</p>
<p>The sound of the original type of unit can be easily found by searching the
internet for "Vibe effect demo". Its popularity among guitar players makes
for the proliferation of clones and audio clip demonstrations, and
ultimately an attempt by the Rakarrack team to emulate this. We hope you
will find our version satisfactory and useful.</p>
<p>Things not available in the original units are the inclusion of stereo
processing paths and feedback, not to mention the variety of LFO's standard
to all Rakarrack modulation effects. Consequently, this effect can be taken
to sonic destinations the original was never able to achieve. The feedback
scheme applied in software, for example, would require a messy modification
using several extra electronics components to properly apply to a real
analog unit. In software this is more elegant and produces a lovely phasing
effect.</p>
<p><b>Wet/Dry</b>: The original 'Vibe had a switch allowing you to select "Chorus" or
"Vibrato". Internal to the circuit, this switch only selected between
different amounts of wet/dry in the final mix. The "Chorus" setting is a
bit of a misnomer because it is actually more of a Phaser. A Wet/Dry of 0
(50/50) corresponds to the "Chorus" setting. A wet/dry setting of -64 (all
wet) corresponds to the "Vibrato" setting of the original analog unit.
Later clones of the circuit added the wet/dry mix as a pot so you could mix
anything in between. Rakarrack applies this philosophy.</p>
<p><b>Width</b>: Width of modulation (LFO) sweep.</p>
<p><b>Depth</b>: How deep the modulation can go on the lowest end of the sweep. A
small number will introduce more "thump" in the response.</p>
<p><b>Tempo</b>: Speed of the LFO sweep</p>
<p><b>Rnd</b>: Adds some randomized "noise" to the LFO to help make it sound less
mechanical.</p>
<p><b>LFO</b>: Type Modulation Shape</p>
<p><b>St. df</b>: Amount of delay between left and right channel LFO's. At 0 or 127
the LFO's are 180 degrees out of phase. At 32 or 96 means there is a
quadrature relationship between the two. If you don't know what that means,
then you only need to know this setting to makes left and right channels
sound different by adding stereo width to the effect. Anything near 64 will
have minimal stereo spreading.</p>
<p><b>Fb</b>: Feedback. 0 is no feedback. The original Vibe does not use feedback. A
setting of -64 is extreme negative feedback and causes a more intense phaser
sound. +64 makes the whole effect sound more "full".</p>
<p><b>L/R.Cr</b>: Left/Right channel crossing. Mix left channel into right channel
and right channel into left channel. Less than zero mixes the channels out
of phase from each other, greater than zero mixes left and right like a
normal mixer. This parameter can have an interesting outcome depending on
the setting on St.df. In some cases the interference between left and right
can be used to make a tremolo effect, particularly when St.df is set to 32
(quadrature).</p>
<p><b>Pan</b>: Pan the effect to the left or right. This can be used to change the
color of the effect between left and right channels if Wet/Dry mix is set to
a certain amount dry.</p>
</td></tr></tbody>
</table>
<br>
<p><a href="effects.html">Effects</a></p>
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<h3><a name="Infinity"></a>Infinity</h3>
<br>
<table cellpadding="5" cellspacing="2">
<tbody><tr><td style="vertical-align: top;">
<p><img style="width: 180px; height: 250px;" src="imagenes/inf.jpg" alt="rakarrack Infinity"></p>
</td><td style="background-color: rgb(251, 251, 251); vertical-align: top;">
<p>Ininity</p>
<p>This is an implementation of and audio counterpart to the classic barberpole
illusion: the filters appear to be continuously ascending (or descending)
akin to the manner in which the colored strips on the barberpole appear to
have no beginning or end. There are other illusions that come out of this
effect model as well when you increase the rate.</p>
<p>One such illusion is the apparent beat frequency. Set the "Rev" switch to
reverse left and right channel directions and mix all wet. Turn up the rate
until you hear a beat frequency. If you are wearing headphones, remove one
from the left or right ear: the beat frequency is gone. With stereo
speakers, mute one speaker and the beat frequency goes away. It is most
interesting with headphones because the left and right channel signals are
completely isolated. The perception of a beat frequency is entirely in your
head!</p>
<p>The infinity is also capable of frequency shifting, as in Single Sideband
Modulation, not pitch shifting. This does crazy robotic things to vocals,
or in a more subtle form may be used to slightly detune your instrument.
You may also use it to detune a delay then use jack to create a feedback
loop so you have an echo which is perpetually ascending or descending.</p>
<p>Apart from illusions and crazy sound effects, there is a wide range of
subtle swirls or stereo panning and doppler shift effects possible.</p>
<p>Think of this as an 8-band EQ where each band perpetually moves up the
spectrum to the top of the range, then starts over at the bottom. This will
give an understanding for adjusting the level on the bands #1-8.</p>
<p><b>Wet/Dry</b>: Mix un-filtered signal with the filtered signal.</p>
<p><b>Res</b>: Filter resonance, also known as an adjustment of the Q factor. More
positive is more resonance, negative is less resonance. It is suggested to
set Res to 0 or less when using Infinity for frequency shifting or doppler
effects.</p>
<p><b>1-8</b>: Level of filter bands. Infinity uses 8-bandpass filters spaced evenly
on a logarithmic scale from Start to End. The mix level of each filter band
is adjusted by these controls. If all are set to the same value, they have
practically no effect except when resonance is set quite high. It is
suggested to use alternating +/- settings. A varying volume (like tremolo)
effect can be obtained by setting them at different levels and low Res with
some 0 in between.</p>
<p><b>Rev</b>: Reverse the direction between left and right channels. If Right channel
filters are sweeping upward, left will be falling when this is
activated.</p>
<p><b>Stages</b>: This adds up to 12 phaser stages to each filter band. A strong
phaser effect can be obtained by setting this to something greater than 4
and alternating the bands #1-8 by 64, -64, 64, -64... There is an
interesting switch internally that interacts with the Pan parameter when
Stages is greater than 8. This makes pan into a variable rate parameter,
which allows for rotating doppler effects.</p>
<p><b>AutoPan</b>: As long as stages is, this acts only as an auto-panning effect
that alternates amplitude on left and right channels. When stages then
it acts as a variable rate control allowing doppler frequency bending to
create the illusion of rotation.</p>
<p><b>St. df</b>: Stereo offset between left and right filters.</p>
<p><b>Start</b>: Where the filters start. Set lower than end if you want filters to
sweep upward. Set greater than End if you want filters to sweep
downward.</p>
<p><b>End</b>: Where the filters end.</p>
<p><b>Tempo</b>: The rate the filters sweep up or down.</p>
<p><b>Subdiv</b>: Tempo frequency subdivision. Numbers greater than zero make the
rate slower. Numbers less than 0 make the rate faster, even up into audible
range frequencies for the crazy pitch shifting and ring modulation type
sounds. Notice this has no effect on the AutoPan frequency. AutoPan Tempo
is hard-coded to cycle every full measure at Tempo.</p>
</td></tr></tbody>
</table>
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