Manual - Full Compass
Model 2058
Channel Strip
Manual Track One
Model 2058
Version 1.0 – 4/2001
R & D: Ruben Tilgner
The information in this document has been carefully verified and is assumed
to be correct. However Sound Performance Lab (SPL) reserves the right to
modify the product described in this manual at any time. Changes without
notice. This document is the property of SPL and may not be copied or reproduced in any manner, in part or full without the authorization of SPL.
Limitations of Liability:
In no event will SPL be liable for any damages, including loss of data, lost
profits, cost of cover or other special, incidental, consequential or indirect
damages arising from the use of the unit, however caused and on any theory
of liability. This limitation will apply even if SPL or an authorized dealer has
been adviced of the possibility of such damage.
Sound Performance Lab
SPL electronics GmbH
P.O. Box 12 27
D- 41368 Niederkruechten, Germany
Phone +49 - 21 63 / 98 34-0
Fax +49 - 21 63 / 98 34-20
eMail: [email protected]
© 2001 SPL electronics GmbH. All Rights Reserved.
Introduction ............................................................................................... 4
Principles ................................................................................................... 4
Hookup/Security advices ........................................................................... 5
Rear panel/wiring ...................................................................................... 6
General advices, connectors & switches .................................................... 7
Control Elements
• Preamplifer stage
Microphone Gain, Line/Instrument, 48 Volt phantom power ...................... 9
Highpass, on levelling ................................................................................ 10
• De-Esser
On, S-Reduction ........................................................................................ 10
Technical information regarding the De-Esser ............................................ 11
• Compressor/Limiter
On, Limiter ................................................................................................. 11
Compression, Make-Up, technical information regarding the compressor .. 12
• Equalizer
On, Low Band, Cut/Boost (Low) ................................................................ 13
Mid-Hi Band, Cut/Boost (Mid-Hi), Air Band, Recommendations ................. 14
• Output/Display
Output ....................................................................................................... 15
Display: S-Det., Clip, Sig. ............................................................................ 15
Display: PPM-Output, Gain Reduction ....................................................... 16
Power supply .............................................................................................. 16
Specifications ............................................................................................ 17
Measurements ........................................................................................... 18
Warranty .................................................................................................... 20
SPL is mainly known for the development of highly specialized audio-tools.
Our philosophy, ”one product for one task”, is aimed at fast and simple operation in conjunction with high processing quality to ensure an excellent musical
With Track One SPL have produced a complete channel strip which for the
greater part is based on the processing concepts already successfully realized
in other products. The very complex task of a channel strip profits particularly from the innovative techniques that have always allowed the operation
of SPL equipment to be efficient and objective. The ususal recording day is
to a high degree determined by a series of opposing time limits – the ”highly
paid” singer/speaker desires a quick recording; however, if technical preparation takes a long time because of unsuitable equipment, time will be lost,
increasing the costs and souring the working environment. The Track One
in all cases however allows fast production without any loss of professional
precision and diligence.
The Track One consists of pre-amplifiers optimized for all kinds of microphones and instruments, SPL‘s acclaimed De-Esser, a compressor/limiter that
can be linked with the compressor of a second unit for stereo operation, an
equalizer (EQ) section for sound corrections or creations and an output stage
for perfectly feeding following units. A central display shows metering for
output level and gain reduction and all other status LEDs.
The Track One is a channel strip that excels primarily in two aspects: it is disarmingly easy to use and offers outstanding sound qualities.
This concept is ideally suited for alle kinds of vocal and instrument recordings in studio, broadcast or live applications as well as for „one man show“
The controls are reduced to the necessary basics to ensure highest userfriendliness. Therefore, working with the Track One can be dramatically timesaving which is most important especially in live situations.
Due to its excellent sound quality the Track One is a highly recommended
alternative to built-in console pre-amps and processing tools. All the modules
are immediately at hand for fast interactions. Recording a voice and providing
clarity, detail and intelligibility is a question of seconds.
Optionally the Track One can be equipped with a 24 bit/96 kHz Ad converter
for direct digital recording. An additional connector allows to insert a second
signal - if for example two Track Ones are used to process stereo material,
only one converter is needed.
The second option is a Lundahl transformer input stage. These transformers
already amplify the microphone signals by factor 5 – the preamps are relieved
by this factor, which improves noise values and linearity of the amplification
in the high frequency spectrum.
A special feature of the printed circuitry layout is the central star ground
wiring scheme: Disturbing influences that could affect the ground paths are
minimized in that the audio-ground is separated from the ground of the
remaining equipment. This leads, in the truest sense of the word ”clean”, to
considerably improved tonal quality. The scatter free toroidal transformer,
manufactured to SPL tolerances, supplies the equipment with the necessary
voltages and forms the basis for a clean electrical supply to all parts of the
Hookup & Security Advices
Carefully select a place for setting up the Track One. The unit should be
situated away from heat sources and direct sunlight. Avoid installation in environments exposed to vibrations, dust, heat, cold or moisture. Keep the unit
away from transformers or motors or any other unit that could generate large
variations in power supply or cause electrical interferences. Do not install the
unit in proximity to power amplifiers or digital processors. You may consider
placing it in a rack containing other analog gear. Such placement can prevent
interference from Word Clock, Smpte, MIDI, etc.
• Do not open the case. You may risk electric shock and may damage your
• Leave repairs and maintenance to a qualified service technician. Should
foreign objects fall inside the case, contact your authorized dealer or support
• To avoid electric shock or fire hazards do not expose your unit to rain or
• In case of lightning unplug the unit. Please unplug the cable by pulling on
the plug only; never pull on the cable.
• Never force a switch or knob.
• To clean the case use a lint-free cloth. Avoid cleaning agents as they may
damage the chassis. Manufactured in standard 19“ EIA format, it utilises
two rack units.
• Please support the back of the unit whenever it is being mounted into a
19 inch rack (especially important when touring).
(use stereo jack)
1=GND 2=hot (+)
3=cold (-)
pin wiring input XLR
Compressor-Link with 2nd Track One
Fuse: 315mA - 15 watts
Sleeve = GND
Pin 1 = GND
Pin 2 = Hot (+)
Tip = Hot (+)
line signals
Unbal. only:
use mono jack
pin wiring stereo jack plug
Tip = hot (+), ring = cold (-), sleeve = GND
Balanced operation:
use TRS jack
Unbal. operation:
use mono jack
(HD) recorder
or console
Ring = Cold (-)
Pin 3 = Cold (-)
pin wiring XLR output
1 = GND 2 = hot (+) 3 = cold (-)
2nd signal input for
optional AD converter
(with converter only)
Rear Panel/Wiring
General Advices
Again, while Channel One’s housing is EMV-proof and protects against
HF-interference, placement of the unit is very important since it amplifies
microphone signals as well as other unwanted signals. Before connecting the
Channel One or any other equipment turn off all power. Adjust the voltage
setting on the back so that it corresponds with the power conditions.
The following graph shows the correct wiring for connecting unbalanced
signals to the balanced XLR connectors:
1= GND
2 = hot (+)
3 = cold (-)
The Line/Instr. connector accepts unbalanced signals only. The output jack
connector accepts both balanced and unbalanced signals (use a mono jack
plug for connecting unbelanced signals). Please also refer to „Analog Outputs“
on page 8.
Connectors & Switches
Mic connector
Unbal. only:
use mono jack
The Mic connector is used to plug in microphones of any type (dynamic,
condenser or tube microphones etc.). If 48 V phantom power is required for
some mics, switch on the 48 V button. For further information please read
“48V phantom power“ on page XXX.
Line/Instr. connector & switch
This connector is used to connect line and instruments signals. A switch
ensures to set the appropriate level and impedance values.
Usually, the switch can be set to INSTR. also for line signals, except they are
delivering very high levels - in this case, the switch should be set to LINE.
Analog outputs
Balanced operation:
use TRS jack
The Analog Outputs deliver balanced output signals. Since both connectors
are working in parallel, unbalancing one connector also unbalances the other
one. If for example a mono jack plug is connected to the jack connector, the
XLR connector is switched to unbalanced operation as well.
Unbal. operation:
use mono jack
Connectors & Switches
A/D Input 2
This connector serves to feed a further signal to the optional AD converter.
Two different signals can be converted at the same time. The maximum input
level should not exceed +12dBu to avoid clipping of the converter (+12dBu
represents the digital full scale level, 0 dBfs).
(with converter only)
GND Lift
The GND Lift switch separates internal ground from chassis ground. The
switch should be activated to eliminate ground loop humming which may
occur if the Channel One is connected to units with another ground potential.
Comp. Link
The Comp. Link function allows to drive two Track One compressor stages
from one master control unit, e. g. for coherent stereo processing.
The units are to be connected via a stereo wiring with jack connectors.
(use stereo jack)
The switch serves to determine which of the units is the master to control
the COMPRESSION, MAKE UP and LIMIT controls from. The respective controls
on the slave unit are deactivated in link mode. All other controls and switches
(including the ON switch) must be taken care off on both units. The GAIN
REDUCTION metering of the master unit now is the master display for both
If the two units are to be used separately again, the wiring must be disconnected and the SLAVE unit must be set to MASTER.
IMPORTANT: If two units are connected for master/slave operation, NEVER
EVER switch both units to MASTER! Both units would try to control each
other - in worst cases, damaging the units can not be excluded.
Preamplifier Stage
Control Elements
Microphone Gain
The Microphone Gain control determines the preamplification of the microphone signal. The preamplification values extend up to + 65 dB. If Lundahl
input transformers are fitted the scale values are to be increased by + 14 dB.
Please refer to „About levelling“ on page 10 for further information.
If the LINE/INSTR. switch is activated, the MICROPHONE GAIN control determines the preamplification of the respective signal. Please note that the
control‘s scale is only valid for microphone signals.
Instrument/Line On – Mic Off
This button allows selection of the input source. The microphone signal is
available for processing when the button is not pressed; pressing this button
activates the Instrument/Line signal which is to be connected on th rear
48 Volt phantom power
The 48 Volt phantom power in the Channel One serves to supply condenser
microphones which are equipped with in-built preamplifiers. A precise
construction and disturbance free electrical supply are the main requirements
for their trouble free operation. In the Channel One the voltage is maintained
at a precise 48 V and delivers a maximum current of 14 mA. This is sufficient
for all types of microphones.
WARNING: All microphones with balanced, ground free output (including
tube microphones) can be operated with phantom power switched on. The
following procedure is to be adhered to: Firstly connect the microphone to the
Channel One, then switch on the phantom power – you can now commence
work. When recording has been completed firstly switch off the phantom
power then wait 30 seconds before disconnecting the microphone from the
Channel One. This allows residual voltages to be discharged.
Phantom powering is only used with condenser microphones. With any
other type of microphone it is to be switched off ! An unbalanced microphone
is not to be used with phantom power switched on!
Control Elements
Preamplifer Stage
The Highpass filter is used to eliminate disturbing low frequencies.
These disturbances could impair the following processing or AD conversion.
The cut-off frequency at 50 Hz avoids influences to vocals. The roll-off is
12 dB/octave. The highpass filter is placed behind the preamplifier to enable
its use also for line or instrument signals.
About levelling
For perfect levelling of the preamplifier firstly switch off all other modules
(De-esser, Compressor/Limiter, EQ) and set the Output control to 0 dB. The
signal can now be levelled with the assistance of the PPM output display.
To achieve a good working level the values should range between 0 and
+6 dB. At these levels an optimal drive level and enough headroom for further
processing (e. g. adding level in the EQ stage) is guaranteed. The Clip LED will
warn you of potential peaks; if during recording the Clip LED illuminates, the
preamplifying value is to be reduced accordingly.
Control Elements
The first module behind the pre-amplifier stage is the De-Esser, which immediately removes disturbing S-sounds when required. The De-Esser module is
activated when the button is on. The S-Detect. LED in the display will show that
S-sounds are being detected regardless of the selected S-Reduction value, in
other words even when the button is switched off detection is still shown in
the display.
With the S-Reduction control you can determine the intensity of S-sound
reduction. Because processing is undertaken from comparison with the level
of the entire frequency spectrum ( see next section ”Technical information ...”)
the processing is more intensive with extreme S-sound levels than with those
of lower levels. After processing the output signal has a consistent S-sound
Control Elements
Technical information regarding the De-Esser
In contrast to common de-essers that influence a frequency band of about
2 octaves with compressor techniques the Auto-Dynamic De-Esser utilizes
filters that process only the reducible ”S-frequencies” but do not interfere
with the remainder of the spectrum. The S-frequencies that lie in the unpleasant range are automatically recognized, the phase is inverted and mixed
with the original signal. In this manner the disturbing frequency is quenched
and the hissing noise reduced. This method of operation has distinct advantages because it is unobtrusive and helps retain the original tonal quality.
Compressor-typical side effects such as lisping or nasal tones do not occur.
Finally its operation is as simple as pulling on the hand brake.
The reduction is accomplished by comparing the entire level with the individual S-sounds: the De-Esser functions only when the S-noise level exceeds the
average level of the entire frequency spectrum. This means for example that
original S-sounds with a determinate S-portion are not processed whereas
those that are too loud, or do not effectively contribute to the sound, are
reduced – the character of the voice remains unchanged.
A further specialty is the integrated auto-threshold-function which makes
processing independent of the input level. Even when the speaker or singer
does not maintain a constant distance to the microphone, processing is retained at the pre-set S-reduction value. Conventional systems are dependent on
the input level and work more intensively as the distance to the microphone is
Control Elements
The On button activates the Compressor/Limiter module. At the same time
the Gain Reduction display shows the processing intensity (see section ”Gain
Reduction” on page 16).
The Limit button switches the Compressor to limiter mode. The Compression
control serves the purpose of controlling the threshold. The Limiter does not
function as a peak limiter, in other words there is no guarantee that all peaks
are intercepted. It is therefore advisable when modulating a subsequent unit
that a headroom of 2 to 4 dB remains. Peak limiters have a system-based
disadvantage in that audible distortions are heard considerably sooner, which
Control Elements
is particularly undesirable for live recording – a „destroyed“ signal must be
recorded again, but a clean signal can still be processed, if for example further
compression/limiting is needed.
The Compression control sets the intensity of compression. Turning the
control clockwise increases compression. The working area spans between +
20 dB (counter clockwise limit) and -50 dB (clockwise limit).
The compressor applies the so-called ”soft-knee” characteristic, which
means that quiet passages are processed at a lower compression ratio than
louder passages. At maximal compression it operates with a ratio of 1:3 –
very effective dynamic limits are achievable when inconspicuous characteristics are to be processed. The exact development of the compressor curve is
portrayed in the diagram 1 on page 18. When setting the compression rate the
Gain Reduction display in the display field is of great assistance. The effect on
the selected compression rate is scaled in 1.5 dB steps. Depending on signal
source and dynamic structure the reduction values should lie between 4 and
8 dB to restrict higher peaks and to optimize the operation of the subsequent
recording system.
IMPORTANT: Make sure that the Comp. Link switch on the rear panel is set
to MASTER, otherwise the compressor will not work. Only if two devices are
operated in master/slave mode, this switch may be set to SLAVE on one unit.
Please also refer to „Connections“ on page 8.
Make Up
With the Make Up control the level reduction caused by compression or limiting can be restored. With assistance of the Gain Reduction display in the
display field setting the Make Up control is very easy: If the maximal reduction
value caused by the loudest tone amounts to -9 dB, for instance, the Make Up
control is also to be set to the value +9 dB. If the Compressor/Limiter is now
switched off the achieved gain in loudness will be audible.
Technical Information regarding the Compressor/Limiter
In the Compressor/Limiter section of the Channel One the parameters for the
time constants (Attack and Release) are set automatically and adapt themselves to the changing conditions of the input signal, far better than can ever
be achieved by manual adjustments. The transient and final oscillation behavior of voices and instruments are constantly changing and at times are so
erratic that a manual control will only achieve good average values, which at
critical moments can produce disadvantageous effects (distortion and artifacts).
If for example the compressor has to react very quickly to harsh P or T noises
it must also be capable of reacting slowly to softer tones – otherwise distortion occurs. Accordingly the Channel One Compressor/Limiter regulates the
level of large fluctuations faster than smaller ones; tones of longer duration
are automatically processed with a longer attack time to prevent distortions.
Control Elements
Even the control of the release time is dependent on the input signal. Fast
and large level fluctuations are correspondingly processed with shorter time
constants than minor fluctuations in order to limit the distortion of the audio
signal as far as possible. Overall this technique provides the optimal solution
between fast, unobtrusive control response and the least distortion of the
audio signal. The result is a natural and transparent sound impression.
The Compressor/Limiter characteristics are portrayed on page 18.
Control Elements
The On button inserts the Equalizer module into the signal path.
Low Band
The center frequency of the half-parametric bass filter is set with the Low
Band control. The adjustable frequency range lies between 30 Hz and 720 Hz
so that this filter covers a range of about 4.5 octaves, allowing it to be used
from the deepest bass to the lower mid range. This together with the Mid-Hi
filter ensures that the entire frequency spectrum is covered.
Cut/Boost (Low)
The Cut/Boost control determines the boost or cut of the Low Band filter;
the maximum values lie between +/- 14 dB. The Low Band filter also operates
to the proportional-Q-principle, in other words the bandwidth is dependent
on the selected boost or cut. With the Low Band filter the factor with which
the relationship of the boost or cut values, in relation to the bandwidth, is
determined lies somewhat higher than with the Mid-Hi filter. The bandwidth is
therefore marginally narrower at maximum boost than with the Mid-Hi filter.
The exact curve of the Low Band filter is shown in diagram 2 on page 18.
The Low Band filter can be applied in many ways. Examples are; to accentuate the fundamental sound of a voice, to cut ”boom frequencies” and for
placement of bass emphasized instruments such as bass guitar, bass drumsor synthesizers during recording or subsequently when mixing etc.
Control Elements
Mid-Hi Band
The center frequency of the semi-parametric mid high frequency filter is set
with the Mid-Hi Band control. The frequency range can be set between 650 Hz
and 13.7 kHz so that this filter covers a range of 4.5 octaves and can be equally
employed in the lower mid as well as the high range.
Cut/Boost (Mid-Hi)
The Cut/Boost control determines the boost, or cut of the Mid-Hi filter; the
maximum values lie between +/- 12 dB. The Mid-Hi filter utilizes the proportional-Q-principle. In other words the bandwidth is dependent on the selected
boost or cut. The higher the boost or cut values are set, so the bandwidth
becomes narrower; by low boost or cut values the bandwidth increases (the
exact curve of the Mid-Hi filter can be seen in diagram 3 on page 19). This filter
characteristic permits a musically more sensible processing of the frequency
spectrum than with constant-Q filters: if a more thorough setting has been
chosen this will lead to far preciser definition of the frequency range to be
processed. This in turn minimizes influences from adjacent ranges.
This filter construction permits the complete scope, from selective removal
of accentuated frequencies through to character giving accentuations of an
instrument, to be effectively and quickly covered.
Air Band
The high frequency filter in the equalizer module is described as the ”Air
Band” and serves the processing of the frequency range of 2 and 20 kHz. A
coil-capacitor-filter with so called bell characteristics and a center frequency
of 17.5 kHz comes into operation here. At this frequency the maximum possible
accentuation is +10 dB, the maximum possible damping is -10 dB. The characteristics of the Air Band filter are shown in diagram 4 on page 19.
The ”soft” and natural tonal property, characteristic of the coil-capacitor
filter, lends itself extremely well to provide clarity to vocals in the upper
frequency range thereby improving their presence. On the other hand harsh
sounds can be lent a more pleasant sound characteristic through damping.
Recommendation on frequency settings
To find the frequency which is to be processed as quickly and accurately
as possible the Cut/Boost control should firstly be adjusted to the maximum
position. Subsequently the relevant frequency should be sought. Following
this the required boost or cut can be set with the Cut/Boost control. Because
the filter at maximum setting works with the smallest bandwidth the frequencies can be heard most distinctly at this setting, making them easier to locate.
Control Elements
The outgoing signal can either be dampened to –20 dB or further amplified
by +6 dB with the Output control to provide optimal drive to the subsequent
units or the optional AD converter. The individually selected output level is
shown on the PPM-Output display in the display field. Before a recording
commences the Output control should be set to zero: The uninfluenced values
from the Output control are then legible and available for adjustment of the
pre-amplifiers levels.
Display: S-Det.
The S-Det. LED shows when S-sounds have been detected. It is only active
when the De-Esser is switched on and is independent from the selected
S-Reduction setting.
Display: Clip
The Clip LED shows overload in the unit. The clipping level of the LED lies
approximately 2 dB below the internal full scale (conforms to + 19 dBu). The
Clip LED should flash as seldom as possible.
At all relevant points of the signal flow the display gets read off: behind the
Pre-amplifier, behind the Compressor/Limiter, behind the EQ and behind the
Output control. All possible causes for overload can be directly checked (overdriven Microphone/Instrument/Line Gain, an excessive Make Up value in the
Compressor/Limiter, too much boost in the EQs or too high output level).
Possible causes of overload can be quickly detected by simply switching
off the modules individually. If overloads occur during recording the quickest
remedy is to gradually reduce the Gain control in the Pre-amplifier.
Display: Sig.
The Sig. (signal) LED illuminates when a signal is being received at the
preamplifier. This provides a quick method of checking that a signal source is
correctly connected. All levels above -50 dB are covered.
Control Elements
Display: PPM Output
The PPM Output display shows the peak reading of the output level (calibrated to 0 dB) and is present at the analog outputs on the rear of the unit. This
display also serves to the pre-amplifying Gain. The value ”+12 dBu” marked on
the right side represents the maximum level of the optional AD/DA converter
which should not be exceeded. (Further information is given in the directions
to the AD/DA converter).
Although the values of the PPM Output display only cover up to + 12 dB
sufficient headroom remains internally (approximately 6 dB) so that the output
value can exceed this limit without causing clipping. The range of optimal
noise performance lies between 0 and + 9 dB.
Display: Gain Reduction
The Gain Reduction display provides information about the processing being
undertaken with the Compressor/Limiter or the Noise Gate. The level change,
perhaps caused by compression, are scaled in 1.5 dB steps. The display is activated when the Compressor/Limiter module is switched on.
Power Supply
Built around a torroidal transformer, the power supply allows for a minimal
electromagnetic field with no hum or mechanical noise. The power supply‘s
output side is filtered by an RC circuit to extract noise and hums caused by
your power service. 6000 µf capacitors smooth out the positive and negative
half waves.
The phantom power is derived with a precise current regulator to provide a
clean phantom power of 48 volts. Our high quality 0.1 % / 6,81 kOhm resistors
ensure the pristine quality of the phantom power supply.
The supply voltage can be set to 230 V/50 Hz or 115 V/60 Hz. Check your
country‘s power requirements for the appropriate setting. An AC power cord
is included to feed the IEC-spec, 3-prong connector. Transformer, AC cord and
IEC-receptacle are VDE, UL and CSA approved. The main fuse is rated at 315 mA
for 230 v and 630 mA for 115 v.
Chassis ground and AC ground can be physically disconnected by the
“Ground Lift” switch (Gnd Lift). This helps to eliminate hums.
Microphone Input
Frequency response:
(200 kHz = -3 dB)
10 Hz-200 kHz
Common mode rejection:
(at -20 dBu)
1 kHz: -80 dB / 10 kHz: -68 dB
20 dB
40 dB
65 dB
-97,5 dBu
-91,0 dBu
-69,6 dBu
Dynamic response:
Instrument Input
Frequency response:
(180 kHz = -3 dB)
10 Hz-180 kHz
7 dB
20 dB
42 dB
-98,4 dBu
-95,8 dBu
-77,2 dBu
Input impedance Line:
12kOhm / Instr.: 1 MOhm
Max. input level Line:
+25 dBu / Instr.: +13 dBu
Dynamic response:
115 dB
Max. output level XLR/jack:
+20 dBu
Output impedance:
‹ 50 Ohm
Power supply
Toroidal transformer:
15 VA
315 mA (230 V/50 Hz)
630 mA (115 V/60 Hz)
Stand.-EIA-19 inch/1 U housing:
482 x 44 x 210 mm
3.1 kg
Compressor/Limiter, Low Band
Diagram 1 portrays the
compressor and limiter
Line A displays the relation
between input and output
Line B shows the curve
characteristics of the compressor
Line C protrays the limiter‘s
curve characteristics
Diagram 2 shows the curves
of the Low Band filter
200 Hz
Mid-Hi Band, Air Band
Diagram 3 displays various
cut and boost settings of
the Mid-Hi filter at around
1,8 kHz
Diagram 4 portrays various
cut and boost settings of
the Air Band filter
SPL electronics GmbH (hereafter called SPL) products are warranted only in
the country where purchased, through the authorized SPL distributor in that
country, against defects in material or workmanship. The specific period of
this limited warranty shall be that which is described to the original retail
purchaser by the authorized SPL dealer or distributor at the time of purchase.
SPL does not, however, warrant its products against any and all defects:
1) arising out of materials or workmanship not provided or furnished by SPL,
2) resulting from abnormal use of the product or use in violation of instructions, or
3) in products repaired or serviced by other than authorized SPL repair
facilities, or
4) in products with removed or defaced serial numbers, or 5) in components
or parts or products expressly warranted by another manufacturer.
SPL agrees, through the applicable authorized distributor, to repair or
replace defects covered by this limited warranty with parts or products of
original or improved design, at its option in each respect, if the defective
product is shipped prior to the end of the warranty period to the designated
authorized SPL warranty repair facility in the country where purchased, or
to the SPL factory in Germany, in the original packaging or a replacement
supplied by SPL, with all transportation costs and full insurance paid each
way by the purchaser or owner.
All remedies and the measure of damages are limited to the above services.
It is possible that economic loss or injury to person or property may result
from the failure of the product; however, even if SPL has been advised of this
possibility, this limited warranty does not cover any such consequential or
incidental damages. Some states or countries do not allow the limitations or
exclusion of incidental or consequential damages, so the above limitation may
not apply to you.
Any and all warranties, express or implied, arising by law, course of dealing,
course of performance, usage of trade, or otherwise, including but not limited
to implied warranties of merchantability and fitness for particular, are limited
to a period of 1 (one) year from either the date of manufacture. Some states or
countries do not allow limitations on how long an implied warranty lasts, so
the above limitations may not apply to you.
This limited warranty gives you specific legal rights, and you may also have
other rights which vary from state to state, country to country.
SPL electronics GmbH
41372 Niederkruechten, Germany
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