PSP 84 Operation Manual

PSP 84 Operation Manual
PSP 84
Operation manual
By using this software you agree to the terms of any license agreement accompanying it. PSP,
the PSP logo, PSP 84, and Its the sound that counts! are trademarks of PSPaudioware.com
s.c.
All other trademarks are the property of their respective owners.
2002-2004 PSPaudioware.com s.c.
Table of Contents
Acknowledgments.......................................................................................................................................................... 4
End User License Agreement......................................................................................................................................... 5
Overview........................................................................................................................................................................ 6
Features ..................................................................................................................................................................... 6
Applications .............................................................................................................................................................. 7
Internal architecture ....................................................................................................................................................... 8
Controls........................................................................................................................................................................ 14
Preset management system .......................................................................................................................................... 23
Support......................................................................................................................................................................... 24
User Comments............................................................................................................................................................ 24
Acknowledgements
We would like to thank all our beta testers for their very useful comments and opinions.
Special thanks to Alan Branch, Eitan Teomi, George Daly, Hans Van Even, Joeri Vankeirsbilck,
Makoto Okabe, Mat Jarvis, Murray Seymour, Orren Merton, Peter Willems, Piotr Madziar,
Richard Winter, Sakis Anastopoulos, Sam Hocking, Scot Solida, Steven Dunston, Tammo
Trper, Ted Perlman, Tomasz Wrblewski and Tomi Liimatainen for their inexpressible support,
inspiration and friendship.
Thanks to all our customers around the world for ideas and help in development of new plug-ins.
End User License Agreement

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Overview
PSP 84 is a high-quality processor, capable of producing a wide variety of delay-based effects.
Two independent delay lines operating with variable sample rate and a precise tape saturation
algorithm with adjustable gain that allow for convincingly sounding simulation of tape delay,
including all the exciting effects resulting from tape speed instability. The Filtration section,
consisting of three 2nd order switchable resonant filter types, can be used to process input,
feedback or wet signal. The adjustable slope of the filter ranges from gentle, which is useful for
simulating high frequency absorption, typical for tape delay and wet signal equalization, to
extremely steep with high cutoff frequency peak making all the wild wah-wah and resonance
effects easily available. The Delay line sample rate and the filter cutoff can be modulated by any
of the 5 LFO waveforms that are automatically synchronized to the sequencer tempo or envelope
follower with adjustable sensitivity and attack/release. Moreover, the PSP 84 contains the
functional reverb unit with simplified setting, carefully tuned to exactly reproduce the sound of
classic spring and plate reverberators.
Features
High quality, unique signal processing algorithms,

Up to 10 seconds of delay time per channel (depending on internal sampling frequency),
Continuous control over the delay time,
Cross-channel feedback and independent channel settings with channel link mode,
Tape saturation simulation with adjustable gain based on the algorithm used in our
mastering processors,
Modulation section consisting of a LFO and an envelope follower, mixed in any
proportions, used for filter cutoff and delay line sample rate modulation,
5 click-free LFO waveforms and variable LFO channel phase offset,
LP, BP, HP filter with adjustable cutoff and resonance,
Flexible filter routing capabilities,
Vintage reverberation module for faithful simulation of spring and plate reverb
characteristics,
Parameter filtration for smooth and click-free operation,
Support for sample rates of up to 192kHz,
Host and MIDI automation of all processing parameters,
Pentium IV optimization,
Over 60 presets for easy start.
Applications
The PSP 84 is primarily meant for applying to individual tracks within a mix and experimenting
with drum and synth loops. The included presets reveal its huge potential in processing vocal,
guitar and bass tracks. Though it can be successfully used for adding classic sounding delay, its
capabilities go far beyond that. The PSP 84 independent channel settings and cross-channel
feedback allow for creating animated delay effects that are more spatial and cut better through the
mix than a simple statically-panned delay. Resonant filters and a modulation section that
synchronizes to the host tempo, make it possible to create superb auto-wah-wah effects, warm
smoky dub sweeps, evolving ambient textures as well as to easily shape the wet signal's spectral
contents. The PSP 84 can create both pristine clean sounding echoes or degenerate the signal to
great extent - all depending on your needs -, thanks to the variable internal sample rate allowing
the pitch-twist effects, rich vocal doublers, flangers and unsurpassed detune effects to be
achieved. Guitarists that are not satisfied with dry and cold sounding tracks recorded using directboxes will appreciate the saturation algorithm, great for adding mild overdrive and harmonic
contents enrichment. Last but not least, the unique reverberation unit allows for adding space and
warmth that most of the native digital reverbs lack.
Internal architecture
Delay lines (DLY L, DLY R and VCO)
The core of the PSP 84 comprise of two parallel delay buffers (one for each channel), operating
with variable sample rate ranging from 0,5x to 2x of the host sample rate. There are two factors
that determine the actual (audible) delay time in the PSP 84:
actual buffer length,
internal sample rate,
The actual buffer length is set by the up/down buttons of the corresponding delay section or by
the invisible slider (please refer to the 'controls' section of this manual for further explanation). It
can also be expressed in rhythmic units (instead of milliseconds) as part of one bar which allows
for faster setting.
The Internal sample rate can be adjusted manually (manual knob in the VCO section) or
controlled by the modulating signal coming from the modulation block.
The easiest and probably the best way to explain how these two factors affect the audible delay
time is to highlight the analogy to a classic tape delay. A basic tape delay machine (picture 1) is
constructed in very similar way to an ordinary tape deck and consists of two heads, one for
recording and one for playback. The main difference is that the magnetic tape is looped, e.g. after
the audio has been recorded on the tape by the recording head it will be played back after certain
time (delay time). This time depends on how long it takes for the tape to cover the distance
between recording and playback heads, which in turn depends on the distance itself and on the
tape speed. These two factors are the exact analog equivalents of PSP 84 actual buffer length and
internal sample rate:
head distance ~ actual buffer length,
tape speed ~ internal sample rate.
The longer the actual buffer length (distance between heads) the longer resulting delay time.
The greater internal sample rate (higher tape speed) the shorter resulting delay time.
Picture 1: simple magnetic tape delay
It might seem that there is no point in having two ways of setting the delay time. This is not true.
Although both methods can be used to set the desired delay time (at least within a certain range),
they also affect other processing properties and are not equivalent. Changing the actual buffer
length in the PSP 84 is equivalent to moving the playback head to the new position. This is done
immediately and the result is that the buffer playback pointer (playback head) jumps to another
part of the audio that was recorded into the buffer. If you are familiar with audio editing
programs, you can think of it as of moving the playback pointer to another position while the
audio file is being played. Changing the internal sample rate (tape speed) is very different from
that because the playback is continuous. The Playback position does not jump to another part of
the recorded audio. Instead, the recorded audio starts to be played back faster or slower. This
results in its pitch to be altered too. If you increase the internal sample rate, which corresponds to
increasing the tape speed, sound is transposed up, and vice versa. After the whole buffer has been
played back, the pitch stabilizes because the audio is being played back at the same speed as it
was recorded.
In the digital domain the maximum frequency that can be reproduced is limited to half of the
sample rate (this is called the Nyquist frequency). That is why the side-effect of lowering the
internal sample rate is that the delayed signal is band-limited. This can be used to achieve low-fi
effects that are required for certain purposes.
The significant drawback of operating in the digital domain, that a sound engineer must be aware
of, is the aliasing phenomenon. It occurs when the original audio is being down-sampled (what
happens when the PSP 84s internal sample rate is lower than the host sample rate). If the sound
that is being down-sampled contains frequencies greater than the new Nyquist frequency (half the
new sample rate), they become audible as unwanted artifacts. Unfortunately some of the digital
processing programs that are doing sample-rate conversion somewhere in their signal path, do not
care about the aliasing problem which becomes clearly audible. There is no way to avoid aliasing
in real-time applications. The only way possible is to minimize its adverse effect by high
frequencies attenuation so that it becomes much less audible. The PSP 84 implements a 4th order
low-pass anti-alias filter prior to the sample-rate converters, that cures the aliasing problem to a
great extent. However, it cuts some of the high frequencies too, which may be unwelcome in
certain applications where a crystal-clean full-band delay is needed, such as hihat processing.
That is why, when the PSP 84s sample rate conversion module (VCO) is switched off (which
implies no aliasing), the anti-alias filter is deactivated making full-band processing possible.
Every unit making sample-rate conversions (with the exception of down-sampling with an integer
ratio) has to implement some kind of interpolation algorithm. The simple techniques of
decimation and linear interpolation that are found in many commercial products, introduce high
harmonic distortion. The PSP 84 uses high quality multi-point interpolators both at the input and
output stage of the delay buffer. These interpolators have been designed to introduce low
distortion while keeping the CPU-usage at the reasonable level at the same time. The result is a
pristine clean sound all across the allowed internal sample rate range.
The hardware tape delay machines damp the recorded sound by a certain amount every one tape
lap. This makes the sound to be quieter with each repetition. The PSP 84 contains a similar
mechanism - feedback path. The feedback value defines the attenuation of the signal and ranges
from -oo dB to 0dB, which corresponds to one signal repetition and to lossless infinite signal
recirculation, respectively. In addition, the PSP 84 has controls that let you set which part of the
signal being fed back goes to the same channel and which to opposite. This allows for creation of
interesting stereo delay effects.
Filter (FLT)
The PSP 84 is not only capable of attenuating the feedback signal. It can shape its spectral
contents, making some frequencies to be attenuated more than others, thanks to the filter section.
Three filter types are available:
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low-pass,
band-pass,
high-pass.
All of them have been mathematically derived from analog 2nd order resonant filter prototypes
and, though not 100% equivalent, manage to keep most of their character. For each filter type,
cutoff frequency and resonance can be set. The resonance range let you shape the filter slope
from very soft, useful for gentle equalization, to steep with high peak around the cutoff
frequency, that makes the PSP 84 really scream. The Filter cutoff can be controlled by the
modulation section as well. The modulation depth can be both negative and positive, letting the
asymmetric LFO waveforms such as sawtooth to be inverted (up-saw becomes down-saw). When
modulation depth is not equal to 0, the manual cutoff knob controls the center frequency around
which the actual frequency oscillates.
The Filter section is not only limited to changing the spectral contents of the feedback signal.
There are three points within the PSP 84s signal path, where the filter can be plugged:
input (IN) - input signal before any other processing is filtered; filtration affects both dry
and wet signals and the filtered signal is recorded to the delay buffers. This mode is the
best for wah-wah effects and is identical to having wah-wah and delay connected in
series,
feedback (FB) - both feedback and wet signals are filtered; the result is that the first
repetition is to be filtered and every next one is to be filtered more and more. This mode is
useful for simulation of frequency absorption coming from wall reflection as well as for
exciting resonant effects.
wet (FX) - wet (processed) signal is filtered; this mode allows for degeneration-free signal
recirculation within the feedback path and controlling the output signal spectrum only.
WARNING! The loud feedback may damage your speakers and ears! When the filter is operating
in FB mode the plug-in can easily become unstable. Always start with low feedback and
resonance values while in this mode and increase them gradually to achieve the desired effect. It
is a good practice to lower the plug-in output level too.
Saturation (DRV)
In order to make the tape delay simulation sound more convincing, the PSP 84 has an advanced
tape saturation algorithm. The saturation module is located just before the delay buffer, so that
the input signal, as well as the feedback signal, go through it. The only adjustable parameter of
this section is gain. It allows for setting the desired amount of saturation - from mild harmonic
contents enrichment to heavy overdrive and distortion. The saturation algorithm is based on
PSPaudioware.com research and experience in programming the award-winning mastering
processors, and its sound is warm and pleasant unlike most digital overdrives, at least for low
gain values.
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Modulation (MOD)
The Filter cutoff and internal sampling rate can both be controlled (at the same time) with the
signal that is generated by modulation section. This section consists of two main elements:
low frequency oscillator (LFO),
envelope follower.
The LFO is capable of generating the following waveforms:
sinusoidal (SIN),
pulse (SQR) with 50% pulse width,
triangular (TRI),
sawtooth (SAW),
random (RND).
The waveforms that have discontinuities (SQR, SAW and RND), are low-pass filtered to make
the plug-in operation click-free. The random waveform is synced with the start of the bar, if the
host is capable of providing the plug-in with the necessary information about the current song
position. The LFO rate can vary within the range of 0.01 Hz to 15 Hz and can be adjusted freely
or host tempo related. The LFO waveforms that are generated for the left/right channel do not
have to be the same. In fact they can be phase-shifted which allows for very interesting fat
chorusing and panning effects when modulating the internal sample rate. The RND waveforms
behave in a different manner - if the phase offset is set to 0, the LFO signal is the same for left
and right channel; if it is not 0 then the plug-in generates two (usually) different random numbers
- one for each channel. The Filter cutoff is always modulated by the left channel LFO signal.
The LFO signal can be mixed in any proportion with the envelope follower signal, making up the
final modulation signal. The Envelope follower analyses the input signal. Its value depends on
the weighted average of the absolute signal level, calculated for a certain time window. The width
of the window is adjusted with the 'speed' knob. The signal can be preamplified before reaching
the envelope follower stage what, in turn, controls the envelope follower sensitivity.
Reverberation (RVB)
Included with the PSP 84 comes the vintage reverb module, capable of simulation of spring and
plate reverberator types. Many commercial reverb units available attempt to simulate virtual
spaces through precise early reflection shaping and generation of a dense reverberation tail.
Unfortunately they often prove to be of little use in the mix, even if sounding convincing when
listened to in isolation. PSPaudioware.com made an attempt to meet the musicians and engineers
demand for a classic mechanical reverberators simulation by analyzing their physical properties
and implementing the reverb topologies that match those mechanical constructions close. The
results produced by the two reverb types are significantly different. The Spring reverb creates a
periodic tail, something that is usually considered unwanted by reverb programmers. However,
this is close to the actual way the spring reverb acts and it turns out to sound surprisingly good in
certain recording situations. The Plate reverb has a denser and brighter tail being not as periodic
as Spring this is typical for physical plate reverberators.
The reverb module can operate in two modes - processing only the wet signal or the output
signal, consisting of dry and wet signals mixed in the proportion set by the 'mix' knob. It has an
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adjustable damping that effectively controls the reverberation time and the reverb amount, which
in fact allows for setting the internal mix proportion.
Main (MAIN)
The input signal can be either attenuated or amplified before further processing, by setting the
'input' knob. Since the PSP 84 contains non-linear elements (saturation) and an envelope follower
within its signal path, the input level does not only change the volume, but can also affect the
sound itself (its spectrum).
The dry and wet signal can be mixed in any proportions. The '-6dB method' has been used for
mix control, which means that in center position, when the proportions of the dry and wet signal
are both 50%, both signals are attenuated by -6dB. Though it may feel uncomfortable that most
of time the result of mixing is too quiet when compared to the level of the dry or wet signal only,
the point is that the PSP 84 is designed to deal with short delay times as well, and the '-6dB
method' becomes more convenient in this situations, when the alternative 'equal power method'
would cause clipping.
The outgoing signal can be attenuated or amplified allowing for final level adjustments.
The PSP 84 acts differently in bypass mode than most other plug-in processors do. Activating
this mode does not stop processing, but only copies the plug-in input to its output. All the
processing is still done in the background. Even though it eats up CPU resources, this seems to be
the only appropriate method for processors that have feedback in their signal path.
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PSP 84 block diagram

headroom L
input L
input gain
input gain
filter
<>IN
flt mode
<>IN
=IN
flt mode
=IN
antialias
filter
antialias
filter
feedback
pan
envelope
follower
mod
source
feedback
pan
saturator
saturator
rate
converter
rate
converter
delay
buffer
z-n
z-n
feedback
gain
mod
source
delay
buffer
rate
converter
rate
converter
phase
shifter
headroom R
input R
<>FB
flt mode
<>FB
filter
filter
=FB
=FB
LFO
=FX
rvb
mode
<>FX
flt mode
=FX
flt mode
=OUT
<>FX
=OUT
=FX
rvb
mode
=FX
reverb
reverb
output L
Picture 2: PSP 84 block diagram.
output R
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Controls
The PSP 84 front panel consists of:
display,
input level meters,
controls.
The display is used to show continuous parameter's values (continuous parameters are described
below).
Input level meters (or headroom meters) indicate the peak level of the input signal. The Input
signal level can be adjusted with the INPUT knob.
Controls are used to change values of parameters. Parameters affect the way that the signal is
processed and thereby the sound. Each control has a parameter bound to it. If you change the
position of the control with your mouse (or MIDI control surface), the value of the parameter that
is bound to this control changes too. The opposite is true as well - if the value of the parameter is
changed by an automation event sent by the VST host, the position of the control will be adjusted
to reflect this change.
There are two types of controls used in the PSP 84:
continuous controls - knobs and invisible sliders
discreet controls - switches, buttons
Continuous controls are bound to the parameters that can have any value from an allowed range,
for example filter cutoff. Discreet controls are bound to the parameters for which only a few
values are allowed, such as filter type.
The PSP 84 front panel is divided into sections that correspond to the internal modules that
process the incoming audio:
DLY L - left channel delay line,
DLY R - right channel delay line,
FLT - resonant filter,
DRV/ENV - saturation module and envelope follower sensitivity,
MOD - modulation signal source,
VCO - delay line sample rate control unit,
RVB - reverberator,
MAIN - levels and mixing.
The FLT, DRV/ENV, MOD,VCO, RVB and MAIN sections can be switched off. We
recommend doing this for the sections that are not needed for current application, to save CPU
resources. The section is switched on/off by clicking on its name. The only exception is the
MAIN section that is switched by clicking on the BYPASS button.
DLY L and DLY R sections cannot be switched off and are always active.
The LINK button allows the left and right channels to share the common settings. If link mode is
active, whenever the control position (and the value of the parameter bound to this control) of one
channel is changed, the opposite channel follows this change immediately.
Most of the time the display shows the tempo. However, when the mouse is moved over any of
the PSP 84 editor controls, the current value of the parameter bound to this control is shown on
the main display.
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The PSP 84 contains the following parameters, grouped in the sections:
I. DLY (L/R) sections

DLY L/DLY R are always lit.
MODE - switches the delay time setting mode; the delay time can be set in ms
(TIME) using INVISIBLE SLIDERS and DOWN/UP buttons or as the tempodependent rhythmic value (NOTE mode), expressed as the numerator and
denominator of the fraction of one bar, using DOWN/UP buttons. If the host
application is capable of providing the plug-in with tempo information, the
PSP 84 will get in-sync automatically when in NOTE mode and will follow
every tempo change. If you use a host that does not have such functionality,
you can still set the tempo manually by clicking on the main display area and
dragging left/right or up/down.
DOWN/UP:
- when in TIME mode used for setting the delay time in miliseconds. Press
shift or ctrl key to increase the step. Please note that the step depends on the
delay time RANGE set. The minimum step is 1/1000 of the current range.
- when in NOTE mode used for changing numerator/denominator values. The
result is obtained by dividing the numerator by the denominator, defining the
tempo-related delay time expressed as the fraction of one bar. Press shift or
ctrl to decrease the value.
INVISIBLE SLIDERS - two sliders (one for each channel) located between
DOWN and UP buttons used for setting the delay time in ms. Simply click on
them and drag your mouse while keeping the mouse button down. They do
nothing when operating in NOTE mode.
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RANGE - changes the delay time range; the available ranges are:
100 ms,
1000 ms,
5000 ms.
Setting the range to be narrower allows for finer delay time setting. The Range
can only be set manually when the plug-in is in TIME mode. When in NOTE
mode, the range is automatically set to be as narrow as possible with the actual
delay time resulting from numerator and denominator settings and the current
tempo. In NOTE mode, the resulting delay time may exceed the maximum
allowed value of 5000 ms. If this happens the display starts to blink and the
actual audible delay time is set to 5000 ms.
FB - sets the feedback attenuation. This can be any value from -oo (no
feedback) to 0dB (lossless signal recirculation) with -6dB in the center
position.
FB PAN - controls the feedback panning for the channel. The signal can be
fed back to the same channel and to the opposite channel in the proportion that
is controlled by this knob.
DLY INV - inverts the wet (delayed) signal phase.
FB INV - inverts the feedback signal phase.
LINK - links the channels.
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II.FLT section
FLT - activates/deactivates the filter section.
MODE - controls where the filter is plugged in within the PSP 84s signal
path. There are three modes available:
IN - input signal is filtered prior to any other processing,
FB - both feedback and wet (effect) signals are filtered. This results in
the first occurence of delayed signal to be affected and each
consecutive repetition to be filtered more and more.
FX - wet (effect) signal is filtered. In this mode the signal can
recirculate in the feedback loop unaffected (without degradation) and
only the wet signal, just before it is mixed with the dry signal, is
filtered.
TYPE - controls the filter type. Can be any of the following:
LP - low pass resonant filter,
BP - band pass resonant filter,
HP - high pass resonant filter.
CUTOFF - filter cutoff frequency. When the cutoff modulation is not equal to
0, then the value set using this knob is the center frequency around of which
the actual cutoff frequency oscillates. The allowed range is from 50Hz to
16kHz, with 1kHz in the center position.
RES - filter resonance. Sets the height of the peak at the cutoff frequency. In 0
position there is no resonance at all, in maximum position the filter is close to
self-oscillation. Be careful when turning this knob up, particularly when the
filter is in FB mode, since higher resonance values may result in instability.
MOD - cutoff modulation. The Filter cutoff can be modulated within the range
of -2...+2 octaves. This knob allows the modulation depth to be set. If turned
counter-clockwise from the center position, then the modulating signal is
phase-reversed, allowing for example to change up-saw waveform into downsaw.
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III. DRV/ENV section

DRV - activates/deactivates the saturation section.
DRIVE - has two bound parameters:

saturation gain ranging from 0dB to +24dB with +12dB in center position.
This affects the saturation depth. Please note that even if this knob is set to
0dB, there is some saturation unless the section is switched off.
IV. MOD section

MOD - activates/deactivates modulation section. If it is switched off, neither
filter nor vco is modulated, even if the modulation depth controls are set to the
values other than zero.
MODE - switches the LFO rate setting mode:

FREQ mode - the LFO rate is set in Hz,
NOTE mode - the LFO rate is expressed in tempo-related rhythmic
value.
DOWN/UP:
in FREQ mode - these buttons changes the LFO rate expressed in
Hertz. The allowed range is 0.01Hz to 15Hz with 3Hz in the center
position.
in NOTE mode - they set the numerator and denominator values (press
shift or ctrl to decrease the value).
19
WAVE - clicking on the corresponding symbol changes the LFO waveform.

Clicking on the already active waveform symbol restarts the LFO. There are 5
LFO waveforms available:
sinusoidal,
pulse with 50% width (square),
triangular,
sawtooth,
random (synced to the bar start if the VST host is capable of sending
current song position information).
PHASE - controls the phase offset between the LFO signal for the L/R
channel within the range of -90 degrees to +90 degrees. Only the VCO can be
modulated for each channel independently. The filter is common for both
channels and is always modulated by the left channel modulation signal.
SOURCE - sets the proportion of the LFO and ENVELOPE FOLLOWER

signals, comprising the final modulation signal.
SENS sets the envelope follower sensitivity
SPEED - integrated control of the ENVELOPE FOLLOWER attack/release

time. It changes the width of the time-window for which the average absolute
input signal value is calculated. The allowed range is 1 ms to 1000 ms.
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V. VCO section
VCO - activates/deactivates the section. If this is off, then the delay lines are
operating with the same sample rate that the host does. Antialias filters are
then deactivated too, allowing for full band-width processing.
MANUAL - manual control of delay line sample rate within the range of 0.5x
- 2x of the host sample rate.
MOD - delay line sample rate modulation. If the LFO phase is not equal to 0,
then each channel delay line is modulated separately by its own modulation
signal.
VI. RVB section

RVB - activates/deactivates the reverberation section.
MODE - lets you apply reverberation to the wet signal (FX) only or to the
final output signal (OUT) (after the dry signal has added to the processed
signal).
TYPE - switches between two classic (mechanical) reverb types. SPRING

simulates the spring reverb. In this case the reflections density does not
increase with the time and reverberation pattern is periodic. PLATE recreates
the plate reverberator characteristics that are brighter than a spring reverb and
the reflections density increases with time.
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DAMP - controls the reflection damping, effectively determining the

reverberation time.
AMT - sets the internal proportion of the dry and wet signal. Please note, that
this is internal to the reverb section so the dry signal here can mean either the
PSP 84 wet signal (if the reverb mode is FX) or its output signal (if the reverb
mode is OUT).
VII. MAIN section

MAIN - unlike all other section indicators, this one cannot be switched by
clicking on it. It only shows whether the unit is in bypass mode or not. Use
BYPASS switch to control this.
INPUT - controls the input signal attenuation/amplification.
MIX - sets the proportion of dry (unaffected) and wet (processed) signal in the
output signal.
OUTPUT - attenuates/amplifies the output signal.
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BYPASS - activates the bypass mode. When bypass is on, all the processing is
still done in the background even though the PSP 84s output signal is the
same as its input signal.
VIII. Other
There is an additional INVISIBLE SLIDER located on the main display area
used to set the BPM tempo manually if the host is not capable of providing the
plug-in with tempo information
Clicking on PSP 84 logo brings up the BACK PANEL showing the registered
owner name and version information.
23
Preset management system

The plug-in has been equipped with its own Preset Management System that is totally host- and
platform-independent. This allows the presets, bank and MIDI setup files to be easily exchanged
between the users of different computer systems and host applications. The preset management
system bar is located below the main plug-in gui and consists of the following sections:
- PRESET contains controls for saving and loading preset files, the combo-box for
restoring the preset as well as changing its name and the left/right arrows which allow
the previous/next preset to be quickly restored,
- BANK contains controls for saving and loading bank files,
- MIDI contains controls for loading and saving the MIDI configuration file (the
numbers of MIDI Control Change messages and parameters assigned to them), the
default button which stores the current configuration in registry or preference file so
that whenever the plug-in is instantiated this configuration is restored. The text box
shows the MIDI CC number assigned to the plug-in editor control being under the
mouse cursor while the last two buttons allow the incoming MIDI CC message to be
assigned to the plug-in editor control or to erase the already-existing assignment. To
assign a MIDI CC number to a editor control press the red dot button and then move
the editor control of your choice with your mouse. The first MIDI CC message
received will be automatically assigned. To erase the assignment press the black cross
button and then point the editor control by moving it with your mouse. The
assignment will be immediately canceled.
24
Support
If you have any questions about the principles or operation of our plug-ins, please visit our web
site www.PSPaudioware.com where you can find the latest product information, free software
updates and answers to the most frequently asked questions.
You can also contact us by e-mail: support@PSPaudioware.com We will gladly answer all of
your questions. As a rule we respond within 24 hours.
PSPaudioware.com s.c.
Dzikiej Ry 11/8 Jozefoslaw
05-500 Piaseczno
Poland.
ph. +48 601 96 31 73
fax.: +48 22 711 25 69
http://www.PSPaudioware.com
contact@PSPaudioware.com
User Comments
We welcome any opinions and comments related to PSP 84. We would also be grateful if you
shared with us your experiences using PSP 84. For example, if youve created a useful preset, let
us know.
Please, contact us at: contact@PSPaudioware.com

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