Contents
Chapter
1: Overview
Interface
conventions
Special
buttons
The
Sample view
User-definable
Edit regions
Sample
view keyboard shortcuts
Chapter
2: File options
Loading
samples
Saving
samples
MIDI
sample transfer
Sample
analysis
Chapter
3: Clipboard editing
Copy
Paste
Cut
out
Auto-crop
Overlay
Undo
Chapter
4: Processing a Sample
Re-synthesis
tools
Audio
and Image Filters
Studio
Effects
Chapter
5: Additional tools
Stereo
<-> Mono converting
Set
playback frequency
Drum
Split
Stereo
Cross-fade
In programming Squash it!, we've tried to make the program as flexible and fully-featured as possible, whilst still retaining a sense of ease-of-use and enjoyment. Squash it! works in 100% GEM environment, so it's possible to use desk accessories (memory permitting) alongside it, and may also work under some multi-tasking operating systems. Since it is GEM-based, Squash it! works in an almost identical way to the desktop itself, with lots of windows, and a menu bar, but there are things we've added to the program that you might not be aware of, hence this section.
| [control]+[d] | Toggle sample draw |
| [l] | Toggle left channel edit (stereo samples only) |
| [r] | Toggle right channel edit (stereo samples only) |
| [m] | Toggle editable area between maximum and marked regions |
| [p] | play sample |
| [+](+[shift]) | zoom in (10 x factor) |
| [-](+[shift]) | zoom out (10x factor) |
New
Open
[control]+[o]
Squash it! will effectively load in any data of any size (up to the maximum
amount of free memory), although it has direct support for AVR, AIFF, and Intel-WAV
formats (in eight or sixteen-bits, mono or stereo). Even if the extension is
wrong, Squash it! will recognise the sample's format.
Formats such as SD1/SD2 (Sound Designer), IFF, or DVSM, and so on, can be loaded,
although you must manually set the sample's properties in the raw sample dialogue
box (which will automatically appear if the sample is not recognised). These
properties include the sample's bit depth (8/16-bit), its channel status (mono/stereo),
and sampling frequency in Hertz (from 8 to 50KHz). To edit the raw sample frequency
setting, press and hold the left or right arrow buttons to lower or raise the
sampling frequency. It must be noted that, raw sample data, if 16-bit, is assumed
to be Motorola format.
If sample load is successful, the sample window is updated and all editing processes
are activated. Note that, since raw sample import ignores any previously saved
header information, it's likely that the first one hundred or so bytes of information
in the sample will sound like noise (this is the header). This can easily be
deleted using the Sample Cut Out feature (detailed below).
Load Clipboard
[alternate]+[l]
Sample data can be loaded directly into the clipboard, effectively allowing
you to have two samples in memory. However, you cannot load raw sample data
into the clipboard directly; the sample format must be AVR, AIFF, or WAV, else
it won't load.
Save as...
[control]+[s]
Calling this menu option for the first time will open the Save window. This
can be kept open indefinitely if desired, but in order to actually save the
sample you must select a format at some time! Squash it! supports AVR, AIFF,
and Intel-WAV formats, as well as raw sample export (SPL button), and will automatically
save in 16-bit format, unless the 8-bit button is highlighted before the format
button is chosen.
To cancel the save operation, press the X button, or the closer element in the
window itself. Since the box is non-modal, it is possible to leave it open whilst
you carry on editing the sound.
Save block
[control]+[b]
User-defined sections of any sample can be saved as individual samples.
When selecting Save block, please make sure that the desired area is either
encapsulated by the user-definable flags, or in between the left and right sample
window positions, else the whole sample will be saved. The sample is automatically
saved as an AVR format file, in 16-bits.
Save Clipboard
[alternate]+[s]
Although you cannot edit the clipboard directly, the need to save it to
disk may arise at some point (particularly when using it as a scratch-pad or
alternative undo buffer), and so this option is provided. As with the block
save, the clipboard is automatically saved as an AVR format file, in 16-bits.
Copy
Paste
[control]+[v]
This will allow you to copy the clipboard data back to the sample, over-writing
the desired region and/or channel (using the [L] and [R] channel buttons). If
there is no sample in memory,then Squash it! will paste the whole clipboard
(memory permitting), else it will paste as much of the clipboard as the currently
selected range will allow. If the clipboard is empty, you will get an error,
and the sample will remain unchanged. In order to mix the clipboard with the
sample data, use the overlay mode instead.
Cut Out
[control]+[x]
Cut out will either chop out unwanted areas of the sample, or keep just
one portion of the sample, discarding everything else. This works in conjunction
with the edit range markers. To keep a specified area, select as normal. To
cut out this region, keeping the data outside of the markers, select Cut Out
with the [shift] key held down, or press [control]+[x]+[shift].
Auto-crop
[control]+[?]
Despite our best efforts, sometimes sample data has 'dead space' in it,
that is, silence before and after the actual sound, which can result in sloppily
triggered drum sounds and wasted memory. Auto-crop is a fantastic labour-saving
device that will automatically remove this dead space (audio data below a 6dB
threshold), giving you a tighter, and noise-free sample.
Auto-crop also has a special effect mode when selected with the [shift]
key held down. This removes all zero amplitude points anywhere in the sound,
and so for example, will move delayed effects together making a contiguous sound.
Pitch bending, and speeding up drum loops without affecting pitch, can also
occur as a result of this effect, making Auto-crop doubly useful. Please note
that whilst normal auto-cropping is fine for stereo or mono sounds, the special
effect is only available for mono samples.
Insert
[control]+[i]
This mode works in a similar fashion to Paste, copying the clipboard data
to a user-defined section of the sample. However, unlike Paste, Insert does
not overwrite the data, rather it moves it to the right, allowing the clipboard
data to be 'inserted' into the sample. Choosing Insert from the Edit menu with
the [shift] key held down, or pressing [control]+[shift]+[i] will
insert the entire clipboard, starting at the left or first marker, and ending
when the clipboard finishes. Otherwise, Insert will copy into the desired region
only, ignoring the remainder of the clipboard data if the region is smaller.
If there is no sample in memory, Insert will work in exactly the same way as
Paste.
Ring Modulate ([2])
Ring modulating a sound increases the number of side-bands in a sound's spectrum,
and also their amplitudes relative to the fundamental frequency. Usually, this
results in a metallic sounding sample, and lots of synthesizers will allow you
to ring modulate with a fixed-frequency sine wave or similar. With Squash it!
though, it's possible to use the clipboard as the modulator, for some truly
expressive sounds.
Amplitude modulate ([3])
Amplitude modulation is similar to ring modulation, except that the sample's
loudness is affected rather than its frequency. The result is similar to an
amplitude LFO except of course that the clipboard contains many more amplitude
nuances than a simple waveform, and so can be very interesting.
Mix ([4])
Mixing offers a way of combining the clipboard and sample for a hybrid sound.
In order to fit both samples into the maximum headroom of 96dB allowable in
a 16-bit sample, both the samples are scaled by 50%. If the selected region
is longer than the clipboard, the remainder of the sample is left unaffected,
except for the initial section which is carefully faded back to 100% to avoid
an unpleasant sudden increase in amplitude. If however, the clipboard is longer
than the edit region, then the remainder of the clipboard is ignored.
Sum ([5])
Summing the sample and clipboard works in much the same way as mixing, except
that rather than scaled, the two samples are subjected to a hybrid filter which,
whilst reducing overall volume will brighten the loudest frequencies of both
samples. As with Mix, Sum will fade back into unaffected sample data.
Filter modulate ([6])
This last mode can be quite effective when used with the right samples.
Filter modulate uses a pseudo-low-pass filter which is then open and closed
according to the amplitude of the clipboard. Because the clipboard's amplitude
is indeterminable, sometimes the filter can become unstable and begin to accentuate
frequencies, but this more interesting than problematic.
Unlike other sample editors, Squash it! has been designed to allow you to completely re-synthesize a sample; with more than forty different tools, it's easy to drastically alter a sound's temporal and spectral characteristics beyond recognition. Generally, most processes are permissible for stereo and mono samples, although a few are limited to mono samples. And most too will allow you to edit regions as well as the entire sample (unless otherwise specified).
Reverse
[control]+[r]
This function will allow you to make a sample play backwards. Since it also
works with range and channel markers, there is a great deal more flexibility
to it than simply playing backwards! For example, one channel of a stereo sample
may be reversed, or multiple sections too.
Tile
[control]+[t]
Tiling a sample allows you to easily repeat a defined section a number of
times, making it possible to create whole new sounds from small sections of
an existing sample. Since the Tile rack-mount has its own ranges, it does not
use the markers set in the sample view (neither will it take into account channel
flags).
To define an area to be tiled, first select the offset from zero, then the actual
size of the tile. These settings will determine what part and how much of the
original sample you want to use as a tile. Next, decide how many times you would
like to repeat the sound, bearing in mind the available memory, and whether
you would like them to overlap, or not. Tiles may overlap up to 50% of their
size, thus giving the sound an increasingly complex tone, as well as making
transitions between tiles smoother. Tiles may also be placed in an alternating
fashion, or normally, where the former reverses each odd tile, just as you might
with black and white kitchen floor tiles.
LFO
[control]+[0]
LFO or low frequency oscillation can be applied to a sample to give it a
vibrato or tremolo effect. To choose between amplitude or pitch LFOs, select
the appropriate mode in the LFO rack-mount by pressing on the LED (it will light
up). Please note that pitch LFO is only available on mono samples.
The oscillating waveform can be set from 1Hz, to 50Hz, the greater the frequency,
the more rapidly changes occur in the output sound. Squash it! also has a depth
control, as well as attack and decay times, that determine the intensity of
the LFO. Greater intensities will produce greater amplitude and pitch changes
in the sample, whereas lower settings will produce a more subtle effect.
In addition, three shapes are offered (only applicable to amplitude LFO mode):
square, sawtooth, or sinusoidal. Square waveform is currently the only waveform
available for pitch modulation.
Envelope Shaper
[control]+[1]
Unlike other sample editors, Squash it! does not have any 'fade' tools,
rather it uses a four-pole amplitude envelope generator to produce the same
(and more flexible) results. In addition, the envelope shaper not only attenuates
volumes, but will also increase them!
Each of the four dials controls the volume of the envelope at each stage of
the ADSR (attack-decay-sustain-release) shape, where a setting of half-way is
equal to no attenuation; lower settings attenuate, and greater settings actually
increase the sample's original volume. Next to the four dials are three sliders
labelled attack, decay, and release. These determine the length of time for
each envelope section, up to a maximum of 33% of the total length of the defined
area. The remainder of the time is given to the sustain portion of the sound,
thus if all are set to maximum, there is effectively no sustain.
Squash/Stretch
[control]+[2]
This is a dual algorithm, and is designed for changing the temporal characteristics
of the sample. As a by-product, in 'scale' mode, the pitch of the sample will
change too (longer samples will sound lower, and shorter, higher). However,
with mono samples only, it is possible to alter the temporal characteristics
of the sound without affecting its pitch, using the 'factor' mode. Since Squash/Stretch
affects the sample's time, it can only be applied to a whole sample.
In 'scale' mode, the dial will control the amount by which the sample is stretched
or squashed (according to the mode set with the LEDs), up to a maximum of 200%
of the original length with stretch, or 10% using squash. In addition, the squash
mode has an extra option of 'stack' or 'obliterate', where the former selection
will retain all sample data (stacking it upon itself), whilst the latter discards
unwanted samples.
In 'factor' mode, the dial decides whether the sample is to be expanded or compressed,
rather than the LED switches themselves. For example, a dial setting of 0 produces
the shortest possible sound, whereas 10 produces the maximum expansion possible.
In this mode, a setting of around half-way will have the least effect. Factor
mode uses a system of time granularisation, whereby intermittent sample windows
are repeated or removed, and then smoothed together. This allows the pitch to
be largely unaffected (and is perfect for drum loops), although some artefacts
do appear as a by-product of the process.
Explode
[control]+[3]
Explode will take a mono or stereo sample (including L/R channel select),
and 'explode' it. The effect, depending on the parameters set, can range from
a variable amplitude LFO, to a disjointed 'explosion' of sound grains.
First of all, the explosion's epicentre must be set, and this can be any point
in the sample (some good effects can be generated using the start or end points).
Next, the size of the debris, and the amount of turbulence are determined using
their respective dials. Debris controls the size of the sound grain, whilst
turbulence determines the pseudo-random amount of the original sample data to
mix with the explosion.
At the explosion's epicentre, the amplitude curve is at its loudest, producing
a scale of 200% of that of the original sample volume; as the debris dissipates
from the centre, so the amplitudes reduce exponentially. Setting the debris
size to a small value will result in sound grains being spaced far apart, whereas
larger debris will join up.
Granularise
[control]+[4]
Audio grains are small sections of a sample that, when combined, create
a whole sound. Squash it!'s Granularise tool allows you to re-order the grains
in an existing sample, to create a whole new sound from the same composite parts.
in addition to the uniform granularisation (where each grain is a pre-determined
size), Granularise also has a gated feature, ideal for re-shuffling drum loops
to form new, and exciting beats.
In either uniform or gated modes, there can be up to 100 grains which are either
determined by the number of grains setting (for uniform), or by the gate threshold
control. Squash it! will then take each of these grains and re-order them randomly,
applying an amplitude envelope to each one to make transitions between grains
seamless.
The way in which the grains are re-ordered although random, is determinable
using a user-defined behaviour pattern. There are three buttons: attraction
to sample start and end, and uniform granularisation. The former two modes will
mix many grains together, overlapping them to produce a rich hybrid tone, whereas
the latter is more useful for atonal sounds and effects (such as percussion
loops).
Ring modulation
[control]+[5]
Unlike the ring modulation in the Overlay section, this module self-oscillates
the sample using one of two modes. By selecting the option from the menu bar
as normal, Squash it! will ring modulate the selected region and/or channel,
with a reversed copy of itself. However, when the menu is selected with the
[shift] key held down (or pressing [control]+[shift]+[5]), it
will ring modulate the sample with a phase-shifted copy. The degree of phase
shifting is user-definable too. When the option is selected, the mouse cursor
will initially turn to a text marker, indicating that it needs further input.
Use the keys [1] through to [9] (not on the keypad), to control
the phase shift.
Phase Modulation
[control]+[6]
This function will allow a sample to self-modulate using a time-variable
phase-shifted copy of itself. The results can vary from simple sum and cancellation
effects to the amplitude (caused by overlapping out-of-phase waveforms), to
complex frequency partial additions.
Phase modulation has just three controls: Offset determines the initial amount
that the phase-shifting starts, expressed as a percentage of the estimated cycle
length. Thus, setting the dial half-way, should produce a phase shifted copy
180° out of phase with the original sample. Depth controls the maximum phase
shift of the waveform, and rate controls the speed at which the maximum depth
is reached. It is this rate control that is responsible for adding very pronounced
harmonics to the original sample (up to around twice and half the fundamental
frequency, depending of course on the rate's speed).
Squared Modulation
[control]+[6]
This is an odd modulator, in that whilst it adds harmonics, it actually
reduces the differences in amplitude. As with the other modulators, it self-modulates
the desired sample, with provisions for ranges and channels. Repeated use of
the squared modulator will result in a square wave sound, regardless of the
original sample's waveform properties.
Optimise
[alternate]+[o]
This menu option offers a fast-track to volume optimising (making the peak
amplitude near or at 100% of the maximum allowable amplitude). Further volume
control can be sought in the volume tools rack-mount box.
More volume tools...
[alternate]+[v]
The Volume Tools rack-mount contains four processes for affecting the sound's
amplitude, including a duplication of the Optimise option in the Tools menu.
Three of the modes are selectable using the keys [1] through [3] (or
by pressing the radio button on the rack-mount), whilst the fourth, gain, is
the default tool, and is automatically used if no other is selected.
Zero is perhaps the simplest of them all, since all this does is clear the selected
region of any sample data. (Of course, this module only has a use when used
in conjunction with the range markers, and then small sections can be cleared
of noise or glitches.) Overdrive, like Optimise will increase the sound's amplitude
content, although it will increase its volume beyond the threshold for 16-bit
samples (96dB), eventually hard clipping all sample data. However, with lower
settings, Overdrive can be more useful than the Optimise tool for gaining overall
optimum volume. Overdrive is used in conjunction with the left-most dial in
the Volume Tools rack-mount.
In its simplest form, Gain can be used to amplify or accentuate the sample's
volume, by a scale of ±10%. However, in addition to the gain control itself,
there are also minimum and maximum threshold controls, which determine the amplitude
area to be affected. Using these it is possible to amplify the louder sections,
whilst keeping the quieter (that which is likely to be noise) sections un-amplified.
Likewise, quieter amplitudes can be boosted, whilst loud sections are kept the
same, giving a similar effect as an expander.
Erode
[control]+[e]
Erode contains a number of sub-processes that will allow you to distort
and corrode the sound in some interesting ways. The first dial does just that;
erodes parts of the sound away by a user-definable amount, adding an amount
of noise as a by-product of the process. If however, you just want to add white
noise, then there's a special dial just for that (the dial for this controls
the volume of noise added)!.
Erode will also allow you to add record crackle and low-frequency hum (like
a dodgy pre-amp), to make drum loops and other noises sound as if they've been
sampled from vinyl! The frequency of the hum and its level can be controlled,
as can the degree of crackling, so it's possible to emulate a new 12" record,
or one that's been sitting around in a dusty room for years. Combined with noise,
and a low-pass filter, some truly authentic grungy loops can be created from
otherwise perfectly sourced sample data.
Each erosion process can be applied as many times as necessary. To disable a
particular erosion algorithm, set its depth or volume control to 0, and it will
be ignored.
Smooth/Expand
[alternate]+[m]
This is another dual rack-mount processor, and has been combined since each
process does the opposite of the other. To choose either processor (since both
cannot be applied simultaneously), click on the appropriate LED in the half-rack
panel (or press [1] for Smooth, and [2] for Expand). Smooth has
a similar effect to a low-pass filter, smoothing differences in amplitudes.
However, unlike a normal audio filter, the process can be applied to certain
amplitude regions (using the lower and upper threshold controls). Also, Smooth
can be made to feed-back on itself (much like an unstable filter), producing
very 'unsmoothed' surfaces! This is achieved by setting the depth control dial
to greater than half-way.
Expand does the opposite. Rather than smoothing the amplitudes between juxtaposed
samples, it accentuates them, resulting in a boost in the high frequency content
of the sample. Like Smooth, Expand can also become unstable, and is also prone
to sharp peaks in the audio (which are automatically compressed to save digital
distortion), when the depth setting rises above half-way. Using the threshold
controls it is also possible to use Expand as a pseudo-EQ, as only some parts,
and thus some frequencies, will be expanded.
Logic filter
[control]+[l]
This collection of filters is so-called since it just deals with samples
as if they were huge blocks of numeric data (which really they are). Each sub-process
can be selected by clicking on the appropriate button or pressing keys [1]
through to [4] on the keyboard. The two simplest modes are De-click,
and Flip, where the latter simply produces a phase-reversed version of itself
(negative values become positive, and vice versa), and the former attempts to
subdue any audio anomalies that it finds (more drastic de-clicking can actually
be achieved with blurring). By flipping only certain regions or channels, it's
possible to create some very interesting phase-relationship effects in stereo
samples.
Shuffle and Shift are more complex in that they both have three sub-modes of
their own. These sub-modes determine what happens to the sample data once it
exceeds the maximum allowable value.
Shift allows you to move the sample data up and down in the amplitude domain,
by a user-definable amount (to a maximum of -32,768. and +32,767). In terms
of repairing samples, Shift allows you to re-set the equilibrium, making all
peaks evenly distributed in the positive and negative domains, and can also
'fix' unsigned sample data (since Squash it! assumes all sample data to be signed,
regardless of any settings in the file header). To achieve this, shift the sample
upwards by 32,767, using the 'wrap' sub-mode. Wrap actually allows you to keep
shifting waveform data up and down without actually destroying it. Once the
upper threshold has been breached, sample data will begin to appear on the bottom,
so it is actually possible to turn a sample inside-out! In 'destroy' mode, all
sample data that goes beyond the amplitude threshold is rounded off, resulting
in hard compression. 'Squash' mode does a similar thing except that instead
of rounding off data, it is folded back on itself.
Shuffle does the same thing as Shift, only in the time domain. As with Shift,
the amount to 'move' the sample by can be a positive or negative integer, but
this time up to a maximum of the sample frame size. Using the wrap sub-mode,
shuffling audio data to the left, past the sample end, will result in it appearing
at the start of the sample (shuffling to the right has the reverse effect).
In destroy mode, all sample data that is 'shuffled' off the end of a sound is
lost, which can actually be beneficial when trying to remove start or end portions
of a sound. Finally, in 'squash' mode, the sample data is folded back on itself,
resulting in a mix of reversed and non-reversed sample.
Comb filter
[control]+[b]
Comb filters are an interesting variety of filter since they can be used
to emphasise a sound's fundamental and all its sub-harmonics, by filtering out
all the others. Squash it! actually has two varieties of comb filter, FIR and
IIR, the latter of which produces a more resonant sounding tone (since it is
based on a recursive or feed-back filter loop). The FIR Comb filter can be chosen
by selecting the menu option as normal; for the IIR filter, select Comb filter
with the [shift] key held down (or press [control]+[shift]+[b]).
Whichever filter is used, the size of the filter bands are user-definable. When
the option is selected, the mouse cursor will initially turn to a text marker,
indicating that it needs further input. Use the keys [1] through to [9]
(not on the keypad), to control the bandwidth size.
Low/High-pass filters
[alternate]+[f]
Both types of filters are combined into one rack-mount, and can be toggled
using either keys [1] and [2], or by pressing the appropriate
LED on the rack-mount itself. Each filter section contains two dials that set
the cut-off frequency (fine and coarse settings), and a bar graph that displays
the roll-off slope. The default is 6dB (single-pole filter), but up to 24dB
per octave (four-pole) filter cut-off slopes are possible, although the greater
they are, the longer it takes to calculate. To increase or decrease the roll-off
slope, press and hold the up or down arrows next to the four segment LED bar
graph (the number of lit LEDs denote an increase in 6dB per octave).
The low-pass filter has an 'ideal' range of 0 - 5KHz, and the high-pass from
1KHz to 20KHz, although of course the roll-off slope has a profound influence
on the actual frequencies that are affected. For example, with a slower roll-off
slope, more frequencies above or below the cut-off will pass through, whereas
a 24dB slope will allow only the frequencies above or below the cut-off point.
Digital Delay
[alternate]+[d]
Digital delays have many uses as a studio effects processor; not only can
they be used for their intended purpose, but they can also be used to create
pitch-related effects (when short delay times are used), and pseudo-reverberation.
Delays can be applied to either left or right, or both channels (when used with
stereo samples), but cannot be applied to user-defined ranges.
Squash it!'s digital delay has a set of four dials: two are used to set the
delay time itself; one for fine tuning (values between 0 and 25ms), and one
for coarse settings (0 - 625ms). Used together, it is possible to set any delay
time from anywhere in between 0 and 650ms.
The other dials control the quality of the delay, allowing you to control rate
and feedback characteristics. Feedback determines how much of the original volume
is mixed in with the delayed signal, where the maximum setting will actually
cause the delays never to reduce amplitude (a setting of 0 is normal). Rate
determines how many delay-taps there are in the effect; larger values obviously
produce longer delay times and a more pronounced effect, though low rates can
be very useful for generating pitch effects, when a short decay time is set
too.
Harmoniser
[alternate]+[h]
Harmonisers add a pitched up or down effect to an input signal, and are
frequently used in studios for embellishing vocals to making chorusing effects
and so on. Squash it!'s harmoniser works only with mono samples, and can only
be applied to entire sounds (ignoring any edit ranges set). However, it does
have two modes of operation (toggled by pressing the 'scale to fit' LED), making
it very flexible. Also, the maximum range of pitch harmonising is ±1 octave.
With 'scale to fit' disabled then the length of time of the effect is variable.
This depends on whether it is pitched upwards, in which case, the sample effect
is shorter than the original sample, or downwards, making the effect longer
(and thus truncating it). 'Scale to fit' ensures that, to the best of Squash
it!'s time compression/expansion algorithm, the pitched effect is the same duration
as the original sample. For sounds such as drum loops and vocal passages, where
the timing is critical, this is essential.
The Harmoniser rack-mount may also be used as a standard pitch-shifter, since
there is a control for the dry:effect balance. Setting this dial to the maximum
'effect' bias will mean that none of the original signal is mixed with the pitched
effect. Of course, it's primary use is to control the mix of original pitch
and affected pitch, so you can create more subtle harmonising effects.
Ping-pong Delay
[alternate]+[p]
Ping-pong delay has some similarities to the standard digital delay except
that, like the Rotary Speaker Simulator, it is a special stereo effect. Thus,
neither the range markers or the edit channel flags are used, and of course,
you will end up with a stereo sample at the end of processing! Ping-pong delay
differs from the standard digital delay in that the decay-taps are positioned
alternately on the left and right channels. Also, although all the other controls
for decay rate and delay times are the same as the Digital Delay rack-mount,
Ping-pong delay does not have provisions for feedback control.
Rotary Speaker Simulator
[alternate]+[r]
This effect automatically generates a stereo sample from a mono sound (unless
additional memory is unavailable), but will utilise a stereo sample too, if
one is already created. Naturally, as it is a stereo effect, the channel edit
select buttons are ineffective, although it is possible to perform the actual
effect on user-definable ranges.
Rotary Speaker Simulator creates an effect whereby a sound is constantly moved
around the stereo image. There are two controls, depth, and rate (in Hertz),
where the depth determines the degree of stereo separation (at its greatest
setting, sound rotation is panned hard left or right). The rate sets how quickly
the sound pans from left to right, with lower values producing much more subtle
fades. As well as controlling the attributes of the speaker rotation, it's also
possible to change its shape from the more usual sinusoidal waveform, to square
or sawtooth, for some interesting effects.