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Operation manual
By using this software you agree to the terms of any license agreement accompanying it. PSP, the PSP logo,
PSP 85, and Its the sound that counts! are trademarks of PSPaudioware.com s.c.
VST is a trademark of Steinberg Soft und Hardware GmbH.
All other trademarks are the property of their respective owners.
2010 PSPaudioware.com s.c.
SYSTEM REQUIREMENTS
MAC:
PowerPC or Intel Mac
- AudioUnit:
Mac OSX 10.5.0 or later
Host application capable of running AudioUnits with Cocoa view
- VST:
Mac OSX 10.4.0 or later
VST 2.4 compatible host application
- RTAS:
Mac OSX 10.4.0 or later
ProTools 8.0 or later
PC:
Windows XP Service Pack 2, Windows Vista or Windows 7 operating system
- VST:
VST 2.4 compatible host application
- RTAS:
ProTools 8.0 or later
OVERVIEW
PSP 85 is the product of our unfading fascination over the endless possibilities offered by
variable sample rate delay line. It is a high-quality processor, capable of producing an
extremely wide variety of delay-based effects from simple echoes to complex rhythmic
patterns, from faithful magnetic tape delay simulations to twisted out-of-this-world pitchshifting and resonant filtering effects.
FEATURES
High quality, 64-bit signal path and unique proprietary processing algorithms,
Up to 20 seconds of delay time per channel (depending on internal sampling
frequency),
Pre-delay for each channel
Continuous control over the delay time,
Cross-channel feedback and independent channel settings with channel link mode,
Delay gating for improved control over number and level of echoes,
Wet signal ducking for improved sound clarity,
Modulation section consisting of tempo-synced LFO and envelope follower sources,
mixed in any proportions,
5 click-free LFO waveforms and variable LFO channel phase offset,
LP, BP, HP filter with adjustable cutoff and resonance as well as saturation
Flexible filter routing capabilities,
Vintage reverberation module for faithful simulation of spring and plate reverb
characteristics,
Parameter edge-filtering for smooth and click-free operation,
Support for sample rates of up to 192kHz,
MIDI and VST automation of all parameters,
60 brand new built-in factory presets plus supplementary PSP 84 bank.
APPLICATIONS
PSP 85 is primarily meant to be applied to individual tracks and to be used for live
performances. It is a great tool for experimenting with drum loops and instrumental patterns,
too. The included presets reveal its significant potential in processing all kinds of tracks, from
vocals to synth and guitar. It can be successfully used for creating classic sounding delays, but
its capabilities go far beyond that. PSP 85 independent channel settings and cross-channel
feedback allow for creating spatial delay effects that cut better through the mix than a simple
statically-panned delay. The built-in ducking, which can be controlled by an external sidechain signal can greatly improve the mix clarity. Resonant filters, envelope detector and low
freqency oscillator which follows song position and tempo changes give enormous
modulation possibilities. Delay gating offers extended control over the number and level of
echoes, something that was usually only possible with multi-tap delay.
PSP 85 can create both pristine clean sounding echoes or dirty distorted delay effects - all
depending on your needs - thanks to variable internal sample rate allowing the pitch-twist
effects, rich vocal doublers, flangers and unsurpassed detune effects to be achieved. Last but
not least, the reverberation unit will diffuse the sound even more, adding more space and
warmth whenever requred.
INTERNAL ARCHITECTURE
VARIABLE SAMPLE RATE DELAY LINES
The heart of the PSP 85 consists of two delay buffers operating at variable sample rate
ranging from half to twice the host sample rate. There are two factors that determine the
actual (audible) delay time of the plug-in:
the length of the delay line (buffer size in samples),
the sample rate the delay line operates at.
The actual buffer length is set by the up/down buttons located in the delay section or by the
invisible slider (please refer to the 'controls' section of this manual for further explanation). It
can be expressed in either milliseconds as the rhythmic value.
The internal sample rate can be adjusted manually (manual knob in the VCO section) or
controlled by the modulation 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 - the audio, that has been recorded would be played back over and over again at time
intervals which depend on tape length and speed it is running at. These two factors are the
exact analog equivalents of PSP 85 delay buffer length and delay line sample rate:
tape length ~ delay line (buffer) length,
tape speed ~ delay line sample rate.
Increasing the buffer length results in longer delay time. In contrast, increasing the sample
rate shortens it.
It might seem there is no point in having two ways of setting the delay time, but close
consideration of the properties of each one reveals the significant differences. Although both
methods can be used to set the desired delay time (within a certain range), they also affect
other processing properties and are not equivalent. Changing the buffer length is immediately
heard as a sudden jump to another part of the audio stored in it. Changing the sample rate
(tape speed) is very different because the playback remains continuous. The playback position
will not jump to another part of the recorded audio. Instead, it is played back faster or slower.
The side effect is an obvious pitch change. If you increase the delay line 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 at.
The maximum frequency that can be reproduced by digital system is limited to half its sample
rate (called Nyquist frequency). The significant drawback of working in the digital domain a
sound engineer must be aware of, is the aliasing phenomenon. It occurs when the original
audio is being down-sampled (which happens when PSP 85s 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
very pronounced, usually unwanted anharmonic artifacts. There is no way to avoid aliasing in
real-time applications unless very high sample rate is used. It doesn't mean it can't be
minimized, though. PSP 85 implements a 4th order low-pass anti-alias filter prior to samplerate 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. To address this need when PSP 85 sample rate conversion module
(VCO) is switched off (which implies no aliasing), the anti-alias filter is deactivated making
full-band processing possible.
Every system doing sample rate conversion has to implement some kind of interpolation
algorithm. The simple techniques of decimation and linear interpolation, found in many
commercial products, introduce high harmonic distortion. PSP 85 uses high quality multipoint 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 and latency at
the reasonable level. The result is a pristine clean sound all across the allowed internal sample
rate range.
FILTER
PSP 85 filter section can be used to shape the spectral content of the input, feedback or wet
(processed) signal. Three filter types are available:
low-pass,
band-pass,
high-pass.
All of them have been mathematically derived from analog 2nd order resonant filter
prototypes. Their frequency response faithfully emulates that of their analog counterparts over
entire audible range thanks to oversampling.
For each filter type, cutoff frequency and resonance can be set. Both can be modulated. The
modulation depth can be negative and positive, letting the asymmetric LFO waveforms such
as sawtooth to be inverted (up-saw becomes down-saw). The phase lag between cutoff and
resonance modulation can be set by OFFSET LFO parameter.
The filter implements the internal saturator which can be used to simulate magnetic tape
properties or to achieve more pronounced distortion effects.
As mentioned before, there are three locations within PSP 85 signal path the filter can be
applied at:
input (IN) - input signal is subject to filtering before any other processing is done;
filtration affects both dry and wet signals and the filtered signal is recorded into the
delay buffers. This mode is the best for delayed wah-wah effects and is similar to
having wah-wah and delay connected in series,
feedback (FB) - both feedback and wet signals are filtered; the result is that every
consecutive delay repetition is 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) only wet (processed) signal is filtered; this mode allows for degenerationfree signal recirculation within the feedback path while controlling the output signal
spectrum.
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 or put the limiter at your
master bus.
MODULATION
Filter cutoff frequency and resonance as well as delay line sample rate can be modulated (at
the same time) by the signal that is generated by the modulation section. It consists of two
main functional blocks:
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 having abrupt changes (SQR, SAW and RND), are low-pass filtered to make
the plug-in operation click-free. All five can be synced to host tempo and song position.
The LFO rate can vary within the range of 0.01 Hz to 15 Hz. The phase offset between right
and left LFO channels can be set to non-zero value for fat chorusing and panning effects when
applied to delay line sample rate. When modulating the filter parameters, the left LFO channel
always controls the cutoff frequency, while right channel modulates the resonance. This
allows for setting some phase lag between those two parameters.
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 and extracts its
temporal envelope. 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, while its sensitivity to input signal level is determined by SENS knob.
The PSP 85 ducker and gate are controlled by the envelope follower as well. The position of
SPEED and SENS knobs will affect their operation as well.
REVERBERATOR
Included with PSP 85 is the vintage reverb module, capable of simulating spring and plate
reverberators. 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. At PSP, we 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
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constructions close. The results produced by the two reverb types are significantly different.
The Spring reverb creates a somehow periodic tail, something that is usually considered
unwanted by reverb designers. However, this is close to the actual way the spring reverb acts
and it turns out to sound surprisingly good in certain recording situations. Plate reverb has a
denser, more diffused and brighter tail being not as periodic as spring.
The reverb module can operate in two modes - processing wet signal or the overall (summed)
output signal. It has an adjustable damping that effectively controls the reverberation time and
the reverb amount, which in fact allows for setting the internal mix proportion.
MASTER SECTION
The input and output signals can be attenuated or amplified as set by INPUT and OUTPUT
knobs. Since PSP 85 contains non-linear elements (saturation) and an envelope follower
within its signal path, the INPUT knob position does not only change the volume, but can also
affect the sound itself.
Dry and wet signals can be mixed in any proportions. The '-6dB' mixing law is used. In center
position both dry and wet signal are attenuated by 6dB prior to summing. Unless very short
delay time is set, this might require some gain boost at either input or output to compensate
for volume loss.
To overcome the problems arising when using delay effects in real-world mixing situations
PSP 85 can duck wet signal automatically. Quite often the delay tail gets masked by a busy
mix. Its rhythmic attributes are no longer perceived. Instead, it only increases the overall hum
and mess. The usual workaround is to use host automation to attenuate the delay when the
mix gets busy and increase its level when there's place for it. The ducker does the same, but
automatically. It can respond to PSP 85 main input signal or separate sidechain signal. The
BLOCK DIAGRAM
The diagram on the next page shows entire PSP 85 audio and modulation (control) signal
paths.
Note that the control signal consists of two channels. In this manual we usually refer to them
as left and right, but it is important to keep in mind that this naming is correct for delay line
resamplers only. The filter cutoff, in contrast, is always modulated by left control signal while
the resonance is always controlled by right control signal. The result is that the cutoff and
resonance values are always equal for both left and right audio channels (same filtering is
applied to both channels). This solution has significant advantage, since you can control the
LFO phase lag between cutoff and resonance by tweaking the OFFSET knob.
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meter L
meter R
input R
input L
input gain
input gain
filter
<>IN
flt mode
<>IN
flt mode
=IN
=IN
antialias
filter
antialias
filter
mute in
mute in
resampler
resampler
feedback
pan
z-n
pre delay
envelope
follower
resampler
z-n
+
resampler
+
fb gain
-n
main delay
pre delay
z-n
main delay
resampler
resampler
<>FB
<>FB
flt mode
phase
offset
=FB
filter
LFO
<>FX
=FB
=FX
=FX
flt mode
flt mode
filter
<>FX
dly pan
gate
gate
ducking
ducking
rvb
mode
OUT
OUT
reverb
rvb
mode
FX
FX
+
output L
+
output R
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USER INTERFACE
MASTER SECTION
DISPLAY when parameter is being edited, its value is displayed.
Otherwise shows the current tempo in BPM. When host doesn't provide
plug-in with tempo information, it can be set manually by clicking and
dragging over display area.
BYPASS bypasses the plug-in processing. Note that the processing is still
performed in the background, even though the results are not heard.
INPUT input gain; either attenuates or boosts the signal prior to any
processing.
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DELAY SECTION
CHANNEL INDICATORS show selected delay channel. Unless LINK is
on, any delay section adjustment will only affect the selected channel.
LINK - allows both delay channels to be set at a time. When on, any
adjustment will affect both delay channels.
Right-clicking LINK copies all delay section parameter values from the
selected channel to the other one.
METERS show left and right channel input signal levels
PRE TIME initial (pre) delay time. It is applied to the input signal prior to
feedback return. Its time is always expressed in relation to the main delay
time, as a fraction of the latter.
RANGE delay time setting range. The smaller range allow for finer delay
time adjustments. Only active when in TIME mode.
MODE delay time mode. Two modes are available:
- TIME the time is expressed in milliseconds and doesn't follow tempo
changes
- BEAT the time is expressed as tempo-related rhythmic figure
MAIN TIME main delay time. It is applied after feedback return. The hot
area between arrow buttons allows for fast coarse setting.
FEEDBACK PAN feedback return panning. Can be used to route part or
entire feedback signal to the other delay channel.
DELAY PAN delay output panning. Each delay line (channel) can be
panned independently.
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DELAY GATE noise gate threshold. Signal below this level will be further
attenuated. Gating offers additional, more flexible control over number and
level of echoes, closing the gap between standard and multitap delays.
PHASE INVERT switches the phase of the feeback return and/or delay
signals. The effect is usually only audible when delay time is short. Useful
for chorus/flanger effects.
MUTE IN mutes the delay line input. On mono tracks, clever muting and
feedback panning allows the two delay lines to be used in series.
MODULATION SECTION
MODE LFO operating mode; three different modes are available:
- FRQ free-running; rate set in Hertz
- BEAT free-running; rate expressed as tempo-related rhythmic value
- POS like BEAT but phase stays locked to song position
RATE LFO rate; use hot area between arrow buttons for fast coarse setting
SHAPE LFO waveform; five are available:
- sine
- square
- triangular
- sawtooth
- random (outputs different values even if POS mode is on!)
OFFSET phase lag between LFO channels. When modulating delay line
sample rate (VCOMOD) this determines the left to right channel phase
offset. When modulating filter, sets phase offset between cutoff and
resonance modulation
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VCO SECTION
ON/OFF toggles the variable sample rate delay line on and off. When on,
the highest frequencies will be slightly attenuated by the additional antialias
filter designed to reduce the anharmonic distortion.
MANUAL delay line sample rate control. Will change the delay time and
result in transient pitch shift, audible as long as the audio in the delay buffer
is played back at the speed different from the one it was captured at.
FILTER SECTION
ON/OFF toggles the filter on and off
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MODE filter configuration switch. Sets the location of the filter within
plug-in signal chain:
- IN filtering is applied to the input signal prior to any further processing,
- FBCK filtering is applied to the feedback signal, just before its return,
- FX filtering is applied to wet (processed) signal, after all processing but
before wet/dry mixing.
TYPE filter type: low pass, band pass and high pass are available
CUTOFF filter cutoff frequency. When band-pass type is selected, controls
center frequency.
RESO filter resonance. Determines the level of the response peak around
cutoff frequency. High values will result in self-oscillation.
DRIVE the amount of internal filter saturation. Low values will add some
harmonics, higher will result in audible distortion.
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REVERB SECTION
ON/OFF toggles the reverberator on and off
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