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Manual
Channel Strip
Channel One
Model 2950
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2
Channel One
Manual Channel One
Model 2950
Version 1.1– 6/2009
This manual contains a description of the product. It in no way represents a guarantee of
particular characteristics or results of use. The information in this document has been carefully compiled and veried and, unless otherwise stated or agreed upon, correctly describes
the roduct at the time of packaging with this document.
Sound Performance Lab (SPL) continuously strives to improve its products and reserves the
right to modify the product described in this manual at any time without prior notice. This
document is the property of SPL and may not be copied or reproduced in any manner, in part
or fully, without prior authorization by SPL.
SPL electronics GmbH
Sohlweg 80, 41372 Niederkruechten, Germany
Fon +49 (0)2163 983 40
Fax +49 (0)2163 983 420
E-Mail: info@spl.info
Internet: www.spl.info
The construction of the Channel One, Model 2950, is in compliance with the standards and regulations of the European Community.
Notes on Environmental Protection
At the end of its operating life, this product must not be disposed of with regular
household waste but must be returned to a collection point for the recycling of
electrical and electronic equipment. The wheelie bin symbol on the product,
user‘s manual and packaging indicates that. The materials can be re-used in
accordance with their markings. Through re-use, recycling of raw materials, or
other forms of recycling of old products, you are making an important contribution to the
protection of our environment. Your local administrative ofce can advise you of the responsible waste disposal point.
WEEE Registration: 973 349 88
© 2009 SPL electronics GmbH. All rights reser ved. Names of other companies and their
products are trademarks of their respective owners.
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Channel One
Content
Symbols & Notes, Scope of Delivery & Packaging ............................................................ 4
Important Security Information ........................................................................................ 4
Hook Up .......................................................................................................................... 5
Introduction, Principles ................................................................................................... 6
Wiring ............................................................................................................................. 7
Sockets and switches ...................................................................................................... 8
Power ............................................................................................................................. 10
CONTROL ELEMENTS
Preamplier
MIC GAIN, 48 V, Activating phantom power, Phase reverse ............................................... 11
High pass, INSTR./LINE GAIN, INSTR./LINE IN, INSTRUMENT, Gain adjustments ............... 12
De-Esser
ON, S-REDUCTION, SPL De-Esser technology ................................................................... 13
Insert .............................................................................................................................. 14
Compressor/Limiter
ON, LIMIT ......................................................................................................................... 14
COMPRESSION, MAKE UP GAIN, NOISE GATE ................................................................... 15
SPL compressor technolog y ............................................................................................. 16
Equalizer
ON, EQ PRE COMP ............................................................................................................ 17
LMF, LMF -/+, MHF, MHF -/+, Recommendation on frequency settings for LMF and MHF ...... 18
AIR BAND, FET DISTORTION ............................................................................................. 19
Output
OUTPUT GAIN, MUTE ....................................................................................................... 20
Headphone Monitor
PLAYBACK, VOLUME, PHONES ......................................................................................... 21
Display Area
S-DETECT, CLIP, SIGNAL, WARM UP .................................................................................. 22
GAIN REDUCT., PPM-OUTPUT ........................................................................................... 23
Power supply ................................................................................................................... 23
Specications .................................................................................................................. 24
Block diagram .................................................................................................................. 25
Measurements ................................................................................................................ 26
Copy Master: recall settings ............................................................................................ 28
Options: AD coverter, I/O transformers ............................................................................ 29
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Channel One
N THIS MANUAL A LIGHTNING SYMBOL WITHIN A TRIANGLE WARNS YOU ABOUT THE
POTENTIAL FOR DANGEROUS ELECTRICAL SHOCKS – WHICH CAN ALSO OCCUR EVEN AFTER
THE MACHINE HAS BEEN DISCONNECTED FROM A POWER SOURCE.
AN EXCLAMATION MARK (!) WITHIN A TRIANGLE IS INTENDED TO MAKE YOU AWARE OF
IMPORTANT OPERATIONAL ADVICE AND/OR WARNINGS THAT MUST BE FOLLOWED. BE
ESPECIALLY ATTENTIVE TO THESE AND ALWAYS FOLLOW THE ADVICE THEY GIVE.
The symbol of a lamp directs your attention to explanations of important functions or applications.
Attention: Do not attempt any alterations to this machine without the approval or supervision
of SPL electronics GmbH. Doing so could nullify completely any and all of your warranty/guarantee rights and claims to user support.
The scope of delivery comprises the Channel One, the external power supply, the guarantee
card and this manual.
Please keep the original packaging. In case of a service procedure the original packaging
ensures a safe transport. It also serves as a safe packaging for your own transports if you do
not use special transportation cases.
Please note and retain this manual. Carefully read and follow all of the safety and operating
instructions before you use the machine. Be doubly careful to follow all warnings and special
safety instructions noted in this manual and on the unit.
Connections: Only use the connections as described. Other connections can lead to health
risks and equipment damage.
Water and humidity: Do not use this machine any where near water (for example near a wash
basin or bath, in a damp cellar, near swimming pools, or the like). In such cases there is an
extremely high risk of fatal electrical shocks!
Insertion of foreign objects or uids: Never allow a foreign object through any of the
machine‘s chassis openings. You can easily come into contact with dangerous voltage or
cause a damaging short circuit. Never allow any uids to be spilled or sprayed on the machine.
Such actions can lead to dangerous electrical shocks or re!
Opening the unit: Do not open the machine housing, as there is great risk you will damage the
machine, or – even after being disconnected – you may receive a dangerous electrical shock!
Electrical power: Run this machine only from power sources which can provide proper power
in the range from 100 to 250 volts. When in doubt about a source, contact your dealer or a
professional electrician. To be sure you have isolated the machine, do so by disconnecting all
power and signal connections. Be sure that the power supply plug is always accessible. When
not using the machine for a longer period, make sure to unplug it from your wall power socket
and from the guitar amp.
Cord protection: Make sure that your power and guitar amplier signal cords are arranged
to avoid being stepped on or any kind of crimping and damage related to such event. Do not
allow any equipment or furniture to crimp the cords.
Power connection overloads: Avoid any kind of overload in connections to wall sockets,
extension or splitter power cords, or to signal inputs. Always keep manufacturer warnings
and instructions in mind. Overloads create re hazards and risk of dangerous shocks! >
Symbols and Notes
Scope of Delivery & Packaging
Important Security Information
Page 5
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Channel One
Lightning: Before thunderstorms or other severe weather, disconnect the machine from wall
power (but to avoid life threatening lightning strikes, not during a storm). Similarly, before
any severe weather, disconnect all the power connections of other machines and antenna and
phone/network cables which may be interconnected so that no lightning damage or overload
results from such secondary connections.
Air circulation: Chassis openings offer ventilation and serve to protect the machine from overheating. Never cover or otherwise close off these openings. Never place the machine on a soft
surface (carpet, sofa, etc.). Make sure to provide for a mounting space of 4-5 cm/2 inches to
the sides and top of the unit when mounting the unit in racks or on cabinets.
Controls and switches: Operate the controls and switches only as described in the manual.
Incorrect adjustments outside safe parameters can lead to damage and unnecessary repair
costs. Never use the switches or level controls to effect excessive or extreme changes.
Repairs: Unplug the unit from all power and signal connections and immediately contact a qualied technician when you think repairs are needed – or when moisture or foreign objects may accidentally have gotten in to the housing, or in cases when the machine may have fallen and shows
any sign of having been damaged. This also applies to any situation in which the unit has not
been subjected to any of these unusual circumstances but still is not functioning normally or its
performance is substantially altered.
In cases of damage to the power supply and cord, rst consider turning off the main circuit
breaker before unplugging the power cord.
Replacement/substitute parts: Be sure that any ser vice technician uses original replacement
parts or those with identical specications as the originals. Incorrectly substituted parts can
lead to re, electrical shock, or other dangers, including further equipment damage.
Safety inspection: Be sure always to ask a service technician to conduct a thorough safety
check and ensure that the state of the repaired machine is in all respects up to factory standards.
Cleaning: In cleaning, do not use any solvents, as these can damage the chassis nish. Use
a clean, dry cloth (if necessary, with an acid-free cleaning oil). Disconnect the machine from
your power source before cleaning.
Be very careful to check that the rear chassis power selection switch is set to the correct
local line voltage position before using the unit (230 V position: 220-240 V/50 Hz, 115 V position: 110-120 V/60 Hz)! When in doubt about a source, contact your dealer or a professional
electrician.
Before connecting any equipment make sure that any machine to be connected is turned off.
Follow all safety instructions on pages 4 and 5 and read further information about the rear
sockets and switches on pages 8, 9 an 10.
Placement
Place the unit on a level and stable surface. The unit’s enclosure is EMC-safe and effectively
shielded against HF interference. Nonetheless, you should carefully consider where you place
the unit to avoid electrical disturbances. It should be positioned so that you can easily reach
it, but there are other considerations. Try not to place it near heat sources or in direct sunlight,
and avoid exposure to vibrations, dust, heat, cold or moisture. It should also be kept away
from transformers, motors, power ampliers and digital processors. Always ensure sufcient
air circulation by keeping a distance of 4-5 cm/2 inches to the sides and top of the unit.
Important Security Information
Hook Up
Page 6
Many audio engineers know SPL’s specialized audio tools, following our philosophy “one
product for one task”. This aims at fast and simple operation in conjunction with high
processing quality to ensure highest musical performance.
With the Channel One we have produced a fully-featured channel strip which for the greater
part is based on the processing concepts already known in other products, such as the SPL
De-Esser and the DynaMaxx compressor. The very complex task of a channel strip prots
particularly from the innovative techniques that have always allowed the operation of SPL
equipment to be efcient and objective.
To a high degree the usual recording day is determined by a series of opposing time limits –
the singer/speaker desires a trouble-free and efcient 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 Channel One in all cases however allows
fast production without any loss of professional precision and diligence.
The Channel One consists of a transistor/tube pre-amplier with microphone-, line- and
instrumental inputs, a de-esser, a compressor/limiter with noise gate, an equalizer (EQ)
section and a latency-free headphone monitor.
So the Channel One has all tools on board for recording a track – along with the preamp it
offers the most needed processors for corrective and creative sound design.
To maximize user friendliness all modules have been reduced to the most important regulating and switching facilities. Fast and effective operation is in no way impeded, quite the
opposite – it is supported. And more time remains for the creative tasks.
From the outset great value was placed on high exibility. An example are the three separate
inputs for microphone, line signals or instruments, each of which has been optimized to its
function.
A twin triode tube is utilized in the process at two positions – one immediately after the
preamplier stage and the other at the end of the chain, so that the processed signal passes
the tube stage twice. This construction combines the advantages of the transistor preamplier stage (high performance with minimal distortion and low noise) with the improved
musical expression of the tone produced by tubes.
The microphone input can optionally be equipped with an input transformer from Lundahl.
The input transformer delivers a vefold amplied microphone level to the preamplier. This
additional amplication reduces the equivalent load to the preamplier electronics. The
balanced outputs can also be equipped with a Lundahl transformer.
The optional 24 bit/96 kHz AD converter module provides digital outputs. An additional input
socket on the Channel One may feed a second signal to the AD conver ter.
Displays and LEDs for signal level, gain reduction, s-detection, clip warning, warm-up status
and signal presence are combined and contained in a single display area to be monitored at
a glance.
A special feature of the PCB layout is the central star ground wiring: Disturbing inuences
that could affect the ground paths are minimized by separating audio-ground from the
remaining equipment. This leads, in the truest sense of the word ”clean”, to considerably
improved tonal quality.
The scatter free toroidal transformer supplies the equipment with the necessary voltages
and forms the basis for a clean electrical supply to all parts of the circuitry.
Principles
6
Channel One
Introduction
Page 7
MIC IN
LINE IN
LINE IN
SEND
SEND
RETURN
RETURN
LEFT
LEF T
RIGHT
RIGHT
230V ~50Hz
Fuse: 315mA s low
115V ~60Hz
Fuse: 630 mA slow
TO REDUCE RISK OF FIRE OR EL ECTRIC SHOCK
DO NOT EXPOSE T HIS UNIT
TO RAIN OR MOIST URE.
THIS EQUIPMENT MUST BE E ARTHED.
WARNING
M AI N S IN P UT
MAINS INPUT
AVIS: RISQUE DE CHOC ÉLEC TRIQUE - NE PAS OUVRIR
RISK OF ELECTRIC SHOCK
DO NOT OPEN
CAUTION
TUBE PREAM P INPUTS
TU B E PR E A MP I N PU TS
OUTPUTS
O UT P U TS
A/D INPUT 2
A/ D I NP UT 2
INSERT
I NS E R T
PLAYBACK INP UTS
P L AY B A CK I NP U T S
S P L CO N VE R TE R S L OT – OP T IO N AL
SPL CONVERT ER SLOT – OPTIONAL
CONVERTER SERI AL #
CHANNEL ONE SERI AL # OPTIONS
GND LIFT
GND LIFT
GND
GN D L IF T
VO LT A GE | F U SE
VOLTAGE | FUSE
XLR WIRING:
PIN 1 = GND / PIN 2 = HOT (+) / PIN 3 = COLD (–)
TRS-JACK WIRING:
TIP = LEFT / RING = RIGH T / SLEEVE = GND
CONNECTOR WIRING
Sound Performance Lab
www.spl.info
MADE IN GERMANY
Microphone
Synthesizer,
sampler etc.
Effects
(delay, reverb, etc.)
InOut
2nd signal source to 2nd channel of the optional
converter (e. g. a 2nd Channel One)
Make sure that the voltage switch setting
reflects the correct local power line voltage.
Pin wiring 1/4" TRS socket (stereo jack):
tip=left, ring=right, sleeve=ground
Pin wiring XLR input sockets:
1=ground, 2=hot (+), 3=cold (-)
Pin wiring XLR output sockets:
1=ground, 2=hot (+), 3=cold (-)
PUSH
2
1
3
21
3
Pin wiring 1/4" TS socket (mono jack):
tip=left, sleeve=ground
Console/DAW/Interface
Playback
7
Channel One
Rear Panel
Wiring
Page 8
8
Channel One
Sockets and switches
Rear Panel
Signal connection
Switch off the unit before you begin the process of making the rst or any subsequent connections. Neglecting this can damage either or both your ears and your equipment.
1/4" TRS sockets
The TRS sockets LINE IN, INSERT SEND/RETURN, OUTPUTS und AD INPUT 2 support both
balanced (1/4" TRS/stereo jack connector) and unbalanced connections (1/4" TS/mono jack
connector). The PL AYBACK INPUTS sockets only support unbalanced connections.
XLR sockets
All XLR sockets are balanced inputs or outputs. Input sockets are always female for plugging
in male connectors, output sockets are always male for female connectors. All in all a comprehensible principle.
Balanced connections
It is impossible to exclude all interferences when an audio signal is transmitted through a
single wire. Shielding is effective against electric, but not against electromagnetic inuences.
Motors, transformers, and alternating current can always induce interferences. But even if the
transmission would succeed, differences in ground potentials between driver and receiver
would produce disturbances.
In balanced connections a reference signal with reversed polarity is transmitted additionally
to the audio signal through a second wire. The ground signal is routed separately through
a third wire. Input and output stages are drivers and receivers, and the receiving stage can
suppress interferences by subtracting the difference between audio and reference signal.
Unbalanced connections
Unbalanced connections from and to RCA or 1/4" TS sockets can be made without adaptors to
the balanced XLR sockets. The correct wiring is important. The diagram shows the pin conguration of the XLR sockets and how to correctly connect them for unbalanced connections:
Connections to RCA sockets are always unbalanced, a wiring to jack connectors can be both
balanced (1/4" TRS/stereo jack) or unbalanced (1/4" TS/mono jack). We recommend to use
individually congured cables from XLR to RCA or jack sockets instead of adaptors. You can
get cables in any needed conguration from audio dealers. With the diagram above, the dealer
can ensure to provide the appropriate cable for your application.
Input
Output
balanced
unbalanced
balanced
unbalanced
1=GND
2=hot (+)
3=cold (-)
1
2
3
1
2
3
1
2
3
1
2
3
Page 9
VOLTAGE
The rear panel VOLTAGE SELECTOR sets the local line voltage (115 V position: 110-120
volts/6o Hz, 230 V position: 220-240 volts/50 Hz). The diagram to the right shows the correct
switch position for 230 V power supply.
BEFORE you connect electrical power make sure that the VOLTAGE SELECTOR setting re ects
the correct local power line voltage!
MAINS INPUT – power connection and fuses
Connect the power cord to the rear MAINS INPUT socket. Transformer, power cord and case
connection conform to VDE, UL and CSA requirements.
The MAINS INPUT socket also houses the fuse. It is accessible from outside and placed right
behind the ap below the socket.
Fuse ratings are 315 mA slow blow (
230
volts) or 630 mA
slow blow (
115
volts).
GND Lift
The rear panel GND LIFT switch eliminates hum by separating the internal ground from the
unit’s housing ground. Hum can, for example, result when this unit’s housing has a common
ground connection with other devices that might have a different ground potential. The
switch is usually deactivated to retain the shielding of the housing.
MIC IN
You can connect any kind of microphone to the MIC IN socket (dynamic, condenser, tube and
ribbon microphones). 48 volts phantom power, which is required for some microphones,
can be activated with the 48V switch on the front panel. Please read the impor tant notes
in chapters “48 V” and “Activating phantom power” on page 11. The microphone input can
also be equipped with an optional input transformer (see page 29, “Information on I/O transformers”).
LINE IN
Use the balanced LINE IN socket for high-level signals with impedances lower than 1 kOhm,
e. g. D/A converters, synthesizers or samplers. We recommend connection to a patchbay for
easier access.
The maximum input level of the LINE IN is +22 dBu.
High impedance sources (above 1 kOhm), such as e-guitars and basses, acoustic guitars with
pick-ups and so on, must be connected to the INSTRUMENT input.
IM PO RTA NT: The line input is deactivated if the instrument input is in use.
INSERT
The balanced INSERT connectors (SEND and RETURN) are used to integrate further units into
the signal path of the Channel One. The SEND connector is placed behind the de-esser, the
RETURN connector is located in front of the compressor. This also allows to record the preampli er signal via the SEND connector while another input signal can be fed into the Channel
One’s compressor or EQ sections for further processing.
GND LIFT
GND LIFT
GND
G ND L I F T
MIC IN
LINE IN
LINE IN
TUBE PREAMP INPUTS
T UB E P R E A M P I NP U T S
M A IN S I N P U T
MAINS INPUT
MIC IN
LINE IN
LINE IN
TUBE PREAMP INPUTS
T UB E P R E A M P I NP U T S
230V ~50Hz
Fuse : 315mA slow
115V ~60Hz
Fuse : 630mA slow
V OL T AG E | F U S E
VOLTAGE | FUSE
SEND
SEND
RETURN
RETURN
INSERT
I NS E RT
9
Channel One
Sockets and switches
Rear Panel
Page 10
OUTPUTS
The OUTPUTS deliver balanced output signals. An output trans former can be equipped
optionally (see page 29).
Since both connectors are working in parallel, unbalancing one connector also unbalances
the other one. If for example a mono jack connector is plugged into to the jack socket, the
XLR socket is operating unbalanced as well. Depending on the impedances of the connected
devices, a parallel use of both outputs can reduce the signal level. Therefore, we recommend
to use either the XLR or the 1/4" TRS output socket.
A/D INPUT 2
The Channel One is a mono channel strip, but the optional A/D converter card 2376 is a dualchannel device. Therefore a second (external) signal can be converted with the converter
card, if it is connected to the AD INPUT 2. If no signal is connected to the A/D INPUT 2, the
output signal of the Channel One is routed to both converter channels. The maximum input
level for the converter is +12dBu (=0 dBFS).
PLAYBACK INPUTS
The playback signal is connected to the unbalanced PLAYBACK INPUTS to create a headphone moni toring mix. If a mono playback signal is available, only the LEFT connector must
be connected. The signal will then be present on both channels. The RIGHT connector should
be used if only one channel should appear on one side of the headphones. In contrast to all
other connectors the PLAYBACK INPUTS are unbalanced.
POWER
The front panel POWER switch activates the unit, conrmed by the illuminated switch.
Switch on the unit only after you have checked the correct setting of the rear VOLTAGE
SELECTOR and 48V phantom power supply front switch.
When you activate the Channel One, the unit commences the warm-up mode to heat the tubes.
The warm-up cycle takes between 15 and 30 seconds. The WARM UP LED in the display area
illuminates during warm-up mode and the Channel One is ready to operate when the WARM UP
LED turns off.
OUTPUTS
O UT PU TS
A/D INPUT 2
A /D I N P UT 2
LEFT
LEF T
RIGHT
RIGHT
PLAYBACK INPUTS
P L AY B A C K I N PU T S
10
Channel One
Sockets and switches
POWER switch
Rear Panel
Front Panel
Page 11
MIC GAIN
The MIC GAIN control determines the level of preamplication. The preamplication values
cover a range from +8 dB up to + 68 dB. Further information on page 12, „GAIN adjustments“. If
the optional microphone input transformer is installed, the scaled values are to be increased
by ca. +14 dB (depends upon microphone, see page 29, “Information on I/O transformers”).
48 V
The 48 V switch activates phantom power for condenser microphones with built-in ampliers. Phantom power should only be activated when using microphones that require it.
VERY IMPORTANT: All microphones with balanced, ground-free outputs, can be used with
the phantom power activated. Please be sure to deactivate phantom power with all other
microphones. Unbalanced microphones may only be used with phantom power deactivated.
Activating phantom power
PLEASE ALWAYS FOLLOW THESE INSTRUCTIONS TO ACTIVATE AND DEACTIVATE PHANTOM
POWER (ALSO WHEN CHANGING MICROPHONES). THE INPUT STAGE OF THE CHANNEL ONE
CAN BE DAMAGED IF YOU IGNORE THESE PROCEDURES!
1. Connect the microphone to the Channel One.
2. Now activate phantom power to use the microphone.
3. After recording rst deactivate phantom power.
4. Wait at least one minute after deactivation of phantom power before disconnecting the
microphone! This ensures residual current will be discharged.
Phase reverse
The phase reverse function reverses the polarity of the microphone signal, inverting the
phase (by 180°) to correct phase-inverted signals caused by multiple signal sources. A voiceover artist, for example, hears himself through the headphones and simultaneously through
the bones in his head. Phase inversion will cause an unnatural sound, and even minimal
variations in distance to the microphone will cause drastic variations in the sound. Phase
inversion is also commonly encountered when using multiple microphones on a single sound
source. We recommend checking for correct polarity before recording.
11
Channel One
Control Elements
Preamplier
Page 12
High-pass filter
This switch activates the high-pass lter (often also called a “rumble lter”), which operates
from 50 Hz downwards with 12 dB/octave. The lter prevents the amplication of unwanted
low frequencies. Compared to 6 dB/octave lters, the 12 dB lter works more intensively, thus
more effectively – therefore the threshold is set to a low 50 Hz.
INSTR./LINE GAIN
This control determines the level of preamplication for signals connected either to the rear
LINE IN or to the INSTRUMENT input on the front. You activate the respective input with the
INSTR./LINE IN switch which is described below.
Gain range for line signals reaches from -12 dB to +22 dB. The attenuation allows to also
process very high levels. The 0 dB mark is highlighted in the line gain scale – this facilitates
to find the setting where a line level signal is processed at unity gain.
Instrument signals can be amplied between +4 dB and +39 dB.
INSTR./LINE IN
With this switch you select between the microphone (off) and line or instrument inputs (on).
The rear mic and line inputs can remain connected, regardless of which input is selected. You
can choose the line input as source as long as the instrument input is not being used.
INSTRUMENT
The INSTRUMENT input jack is placed on the front for easy access. It should be used to connect
instruments like e-bass and guitars, acoustic guitars with pick-ups, etc. The INSTRUMENT
input features a 1 MOhm (one mega Ohm) input impedance. Line signals with lower impedances, such as from D/A converters, samplers, synths, etc. should be connected to the rear
LINE IN socket.
IMPORTANT: As long as an instrument is plugged into front INSTRUMENT input, the rear
panel LINE IN input is deactivated.
Gain adjustments
For perfect levelling of the preamplier rstly switch off all other modules (de-esser,
compressor, 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 +3 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 gain value is to be
reduced accordingly.
12
Channel One
Preamplier
Control Elements
Page 13
ON
The rst processing module is the de-esser, which removes disturbing sibilants when required.
The de-esser module is activated with the ON button. The S-DETECT LED in the display area
shows that S-sounds are being detected. It is independent from the S-REDUCTION control
and always informs about detected sibilants – attracting your attention to a possible need for
regulation (also see „S-DETECT“ on page 22).
S-REDUCTION
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) the processing is more intensive with extreme S-sound levels than with
those of lower levels. This processing method achieves a consistent level of the remaining
sibilants in the output signal.
SPL De-Esser technology
In contrast to common de-essers based upon compressor techniques the SPL De-Esser makes
use of the phase cancellation principle. It employs lters that process only the reducible
”S-frequencies” but do not interfere with the remainder of the spectrum. The S-frequencies
are detected automatically, the phase is inverted and mixed with the original signal. 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 average 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 certain S-portion
are not processed whereas those that are too loud, or do not effectively contribute to the
sound, are reduced – but 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 reduced. As a result, the SPL De-Esser does not need to be
monitored and re-adjusted permanently to keep processing constant – and it can always be
applied before the compressor, as changing its position would not be an advantage. That is
why an accordant switching function is not necessary.
13
Channel One
De-Esser
Control Elements
Page 14
INSERT
The insert button activates any attachments of external equipment for effects, such as delay
or reverb, that are connected to the INSERT SEND/RETURN loop on the rear of the Channel
One. This binds them into the signal chain, thereby enhancing the processing capabilities ad
innitum.
The INSERT point is located between the de-esser and compressor. This allows to use the preamplier stage/de-esser combination of the Channel One separately from the compressor/
EQ combination. This broadens the range of uses enormously, because in this manner the
Channel One can be used as two independent units.
As long as units are not connected to the INSERT loop, the signal ow is not interrupted,
even when the INSERT button is pressed. The most exible method of use with the balanced
designed insert sockets is to be achieved by connection to a patch bay.
ON
The ON button activates the compressor/limiter/noise gate module. At the same time the
GAIN REDUCT. display shows the processing intensity (see “GAIN REDUCT.” on page 23).
Usually the signal ow follows the design of the Channel One and for this reason the input
signal normally arises from the de-esser or, when activated, from the INSERT. However, with
the EQ PRE COMP. switch the EQ module can be switched in front of the compressor module.
This allows it to be used either as a nal compressor or limiter (further information in the
section “EQ PRE COMP.” on page 17).
LIMIT
The LIMIT switch turns the compressor into a limiter. 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 included. It is therefore advisable when modulating a subsequent unit that a headroom of 2 to 4 dB remains. Now peaks can be intercepted
very well and unobtrusively. Peak limiters have a system-based disadvantage in producing
audible distortions considerably sooner, so with regard to both sound quality and recording
safety, we think the soft limiter mode is the better choice for a recording channel strip.
Control Elements
14
Channel One
Insert
Compressor
Control Elements
Page 15
COMPRESSION
The COMPRESSION control sets the intensity of compression. Turning the control clockwise
increases compression. The working area spans between 0 dB (fully left) and -32 dB (fully
right).
The compressor applies the so-called “soft-knee” characteristic, which means it starts
processing earlier than with hard-knee curve (see page 26, diagram 1, curve B). Hard-knee
compressors can sometimes gain more loudness, but they process abruptly and the danger
to ruin a recording with compression artifacts is much higher. On the other hand the sof tknee compressor always helps very well to keep levels under control and ensures highest
recording safety – and if there is a desire to gain further loudness, the signal can still be
processed after recording.
At maximal compression it operates with a ratio of 1:2.5 between input and output signal –
very effective dynamic processings are achievable with unobtrusive sound characteristics.
The exact development of the compressor curve is portrayed in the diagram 1 on page 26.
When setting the COMPRESSION rate the GAIN REDUCT. display in the display area is of
great assistance. The ef fect 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.
MAKE UP GAIN
With the MAKE UP GAIN control you can restore the overall level reduction caused by
compression or limiting. With assistance of the GAIN REDUCT. display setting the MAKE UP
GAIN control is very easy: If the maximal reduction value caused by the loudest tone amounts
to -9 dB, for instance, the MAKE UP GAIN 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.
NOISE GATE
The NOISE GATE control reduces soft disturbances during signal pauses. When turned fully
counter clockwise the noise gate is switched of f. By turning the control in a clockwise direction the threshold value increases. This means that the noise gate closes relatively earlier.
The processing span of the NOISE GATE control is between -100 dB/OFF (gate control turned
fully counter clockwise) and + 15 dB (gate control turned fully clockwise). The noise gate is
therefore operable over the complete dynamic range.
The noise gate functions very stably: the point at which it opens lies 6 dB above the point at
which it closes again (hysteresis of 6 dB). Denite closure and opening is therefore assured
– the most feared characteristic of “uttering” is excluded. Even critical signals are cleanly
processed.
The noise gate’s release time is set automatically. The automation, which depends upon
the program, adjusts itself to the musical piece, thereby ensuring optimal (undetectable)
opening and closing.
15
Channel One
Compressor
Control Elements
Page 16
SPL compressor technology
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 nal 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 (e. g. distorted sounds, “pumping”, etc).
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 regulates the level of large uctuations faster than smaller ones;
tones of longer duration are automatically processed with a longer attack time to prevent
distortions.
The automatic setting of the release times is dependent on the input signal, too. Fast and
large level uctuations are correspondingly processed with shorter time constants than
minor uctuations 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.
A further technical specialty of the circuitry contributes to the high audio quality of the
compressor in the Channel One: SPL’s double VCA drive. One VCA receives the in-phase,
the other the out-of-phase signal. Subsequently the signal is passed through a differential
amplier. The effect of this circuitry is that distortion products and offset uctuations can be
removed – the product of the differential of both signals means that possible interference is
canceled out. The original information is however further amplied by 6 dB. In addition the
VCAs provide relief to each other because they share their loads. They do not even run the
danger of operating in the saturation range, which ensures to avoid offset noises, audible as
clicks or pops.
SPL’s double VCA drive circuitry overall displays vastly improved distortion values so that a
distinctly clearer and more transparent sound impression is achieved than with conventional
circuitry. Voices and instruments are given a considerably more natural and dynamic timbre
whereas ”mufed” tones are not audible.
The compressor characteristics are portrayed on page 26.
16
Channel One
Compressor
Control Elements
Page 17
ON
The ON button inserts the equalizer/FET distortion module into the signal path. Under normal
circumstances the input signal comes from the compressor. With the EQ PRE COMP. button
the equalizer can be switched in before the compressor so that the input signal is received
from the de-esser or insert.
IM PO RTA NT: To avoid irritations at the beginning of a recording it is recommended to deactivate the equalizer/FET distortion module. If not, tonal changes could occur immediately and
furthermore, in the case of the DISTORTION control, additional distortions.
EQ PRE COMP.
The EQ PRE COMP. switch reverses the sequence of compressor/limiter and equalizer (EQ).
When the switch is pressed the equalizer operates in front of the compressor/limiter; when
not pressed the succession remains unchanged. This function permits very exible operation
with the Channel One when it is necessary to resolve recurring problems or to create special
sounds.
The following examples describe when the equalizer is to be switched in front of the
compressor/limiter.
When over-accentuation of instruments or voices is registered within certain frequency
ranges these ranges can rst be reduced with the EQ. The signal can subsequently be
compressed more easily.
A further sensible application is the use of the compressor module as a nal limiter to maintain a stable output level. If the EQ was to be used again after limiting it could not be guaranteed that the output level would not alter.
17
Channel One
Control Elements
Equalizer
Page 18
LMF
The center frequency of the half-parametric bass lter is set with the LMF control (low/mid
frequencies).
The adjustable frequency range lies between 30 Hz and 700 Hz so that this lter 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 MHF lter ensures that the entire frequency spectrum is covered.
LMF -/+
The LMF-/+ control determines the boost or cut of the LMF lter; the maximum values lie
between +/- 14 dB. The LMF lter also operates to the proportional-Q-principle, in other
words the bandwidth is dependent on the selected boost or cut. This lter characteristic
permits a musically more sensible processing of the frequency spectrum than with constantQ lters: if a more thorough setting has been chosen this will lead to far preciser denition
of the frequency range to be processed. This in turn minimizes inuences from adjacent
ranges.
The boost or cut values, in relation to the bandwidth, lie somewhat higher than with the MHF
lter. The bandwidth is therefore narrower at maximum boost than with the MHF lter for
even more precise ltering. The exact curve of the LMF lter is shown in diagram 4 on page
27.
The LMF lter 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 during recording or subsequently when mixing etc.
MHF
The center frequency of the semi-parametric mid/high frequency lter is set with the MHF
control.
The frequency range can be set between 680 Hz and 15 kHz so that this lter covers a range of
4.5 octaves and can be equally employed in the lower mid as well as the high range.
This together with the LMF lter ensures that the entire frequency spectrum is covered.
MHF -/+
This control determines the boost, or cut of the MHF lter; the maximum values lie between
+/- 12 dB. The MHF lter utilizes the proportional-Q-principle, too: 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 curves of the MHF lter can be seen in diagram 3 on page 27).
The lter 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.
Recommendation on frequency settings for LMF and MHF
To nd the frequency which is to be processed as quickly and accurately as possible, rstly
adjust the MHF -/+ control to the maximum position. Subsequently the relevant frequency
should be sought. Because the lter at maximum setting works with the smallest bandwidth,
the frequencies can be heard most distinctly at this setting, making them easier to locate.
Finally the desired MHF +/- setting can be applied after the frequency is determined with
MH F.
18
Channel One
Equalizer
Control Elements
Page 19
AIR BAND
The high frequency lter in the equalizer module is described as the AIR BAND. A coilcapacitor-lter 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 soft and natural tonal property, characteristic of the coil-capacitor lter, lends itself
extremely well to provide clarity and ... well, air, 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.
The characteristics of the AIR BAND lter are shown in diagram 2 on page 26.
FET DISTORTION
The DISTORTION control offers the capability of applying distortions to signals. The distortions are innitely variable from 0% through to distinctly perceptible harmonics. The distortion stage is located in front of the equalizer so that the newly created spectrums can also be
processed with the EQ.
A (more or less over-driven) eld-effect transistor forms a part of the distortion circuitry. It’s
characteristic curve is similar to a tube and sounds distinctly “warmer” than a pure diodedistortioner.
The signal level is of utmost importance to the operating mode of the DISTORTION module.
To achieve useful results the level should lie in the range 0 to + 6 dB. Over and above this
the results are strongly dependent on the condition of the input signal and its spectrum.
The processing of sinewave-like signals (e. g. e-piano, vocal, guitar) is audible much earlier
than signals with predominant harmonical contents (e. g. snare drum, hi hat etc). It is recommended that time and effort is taken to nd the correct setting.
IM PO RTA NT: To avoid irritations at the beginning of a recording it is recommended to deactivate the EQ, and in particular set the DISTORTION control to 0%. If not, tonal changes will
occur immediately and fur thermore, in the case of the DISTORTION control, additional distortions.
19
Channel One
Equalizer/FET Distortion
Control Elements
Page 20
OUTPUT GAIN
The outgoing signal can either be dampened to -20 dB or further amplied by +5.5 dB with the
OUTPUT control to provide optimal drive to the subsequent units or the optional AD converter.
The selected output level is shown on the PPM OUTPUT display in the display eld.
Before a recording commences the OUTPUT control should be set to 0 dB (12 o’clock position): the uninuenced values are then legible and available for adjustment of the preampliers levels.
MUTE
The Mute switch mutes the output signal; when activated, the PPM OUTPUT display does not
show any values.
An instance of a sensible application could possibly be when the output signal of the Channel
One, together with the playback signal, are reproduced via the studio monitors during a
recording session. When subsequently the recorded take is monitored it becomes possible to
hear extraneous singing or comments arising from the singer. It is therefore advised to press
the MUTE switch to permit listening to a clean recording. Do not forget to deactivate the Mute
switch before continuing recording.
Another instance could be allowing the musician to rehearse for a while and then, when ready,
freeing the signal path and commencing recording by deactivating the MUTE switch.
20
Channel One
Output
Control Elements
Page 21
An individual monitoring mix is important for best working conditions and especially a good
intonation. That is why the Channel One is equipped with a headphone monitor section,
providing a headphone amplier and a mixing stage to generate an individual mix for the
musician with playback and recording signals.
In general, the headphone monitor section can of course always serve for direct monitoring
of the recording via headphones. Another practical use of the headphone monitor module is
to monitor the signal quality directly to locate and eliminate possible interference rapidly.
PLAYBACK
The PLAYBACK control regulates the volume of the playback signal which is passed to the
musician. There are two methods of passing the mono playback signal: The rst is to pass
the music to both ear pieces of the headphone in which case PLAYBACK INPUT LEFT must
be connected. On the other hand some musicians want to hear the playback signal through
only one ear piece so they can hear themselves directly with the other ear (playback signal
only without microphone signal). In this instance connect PLAYBACK INPUT RIGHT and set the
VOLUME control to off (also see page 10, PLAYBACK INPUTS).
VOLUME
The VOLUME control regulates the volume adjustment of the microphone, instrument or
line signal. The setting is independent to that of the OUTPUT control or MUTE switch, which
means the volume in the headphones does not alter although the output value of a modulation has changed.
TIP: When working with hard disc systems or digital mixing consoles latency may be present.
Flanging or phasing effects occur if the musician receives the monitor signal with a time lag.
It is therefore recommended, to obviate latency, that the monitor signal passes directly from
the headphone monitor to the headphones. It should be remembered that the recording
signal has not been picked up again by the playback signal because phase quenching can
occur when the same signal is mixed by both the PLAYBACK and VOLUME controls.
PHONES
Connect headphones to the standard 1/4" (TRS) stereo jack plug on the front panel. The lowresistance input allows for connection of all usual headphones.
The layout is: Tip =left channel, Ring = right channel, Sleeve = ground.
Make sure that the plug is rmly seated for a solid connection. Reduce volume level before
you remove or plug in a headphone (or when switching headphones). NEVER plug in a mono
1/4" jack (TS) to the headphone output. The use of a mono 1/4" will lead to a short-circuit
that will destroy the amplier stage. Standard headphone connectors always have stereo
plugs, and thus a correct connection will be assured when you only connect headphones
directly. Double check that you use stereo 1/4" TRS plugs when you connect headphones via
patchbays or extension cables etc.
21
Channel One
Headphone Monitor
Control Elements
Page 22
All status and level displays are concentrated in a central display area, so all important information can be perceived at a glance.
S-DETECT
The S-DETECT LED shows when sibilants have been detected. It is only active when the
de-esser is switched on, but it is independent from the S-REDUCTION control. So if you turn
on the De-Esser, you are always informed about detected sibilants and a possible need for
regulation.
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 ash as seldom as
possible.
At all relevant points of the signal ow the display gets read off: behind the preamplier,
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 respective gain control in the preamplier.
SIGNAL
The SIGNAL LED illuminates when a signal is being received at the preamplier. This provides
a quick method of checking that a signal source is correctly connected. All levels above -50 dB
are covered.
WARM UP
The WARM UP LED gives an indication regarding the warm up phase of the tube stage. When
the LED is extinguished the Channel One is ready for operation; during warm-up the output
signal level is low and sounds distorted.
22
Channel One
Display area
Control Elements
Page 23
Power Supply
GAIN REDUCT.
The GAIN REDUCT. display provides information about the processing being undertaken with
the compressor/limiter or the noise gate. The level changes, perhaps caused by compression, are scaled in 1.5 dB steps. The display is activated when the compressor/limiter module
is switched on.
Noise gate operation is visualized by illumination of all GAIN REDUCT. LEDs when the signal
level lies under the gate threshold setting.
PPM OUTPUT
The PPM OUTPUT display shows the peak reading of the output level (calibrated to 0 dB) and
is present at the rear outputs. This display also serves to level the preamplier. The value
“0dBFS” marked on the lef t side represents the maximum level of the optional AD converter
which should not be exceeded (further information is given in the manual of the AD converter
and on page 29).
Although the values of the PPM OUTPUT display only cover up to + 12 dB sufcient 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.
Built around a toroidal transformer, the power supply ensures a minimal electromagnetic
eld with no hum or mechanical noise. The power supply‘s output side is ltered by an RC
circuit to extract noise and hums caused by your power service. 6000µf capacitors smooth
out the positive and negative half waves.
The phantom power is derived from a separate winding in the transformer, a precise current
regulator 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 250 Volt power supply for the tube stage is ltered with 300µF to minimize hum.
Further information on page 9.
23
Channel One
Display area
Control Elements
Page 24
Microphone input
Frequency range: 10 Hz-100 kHz
(100 kHz = -3dB)
Common mode rejection: 1 kHz: -80 dB/10 kHz: -78 dB
(@ -20 dBu)
THD & N: Gain: A-weighted:
20 dB -97,1 dBu
40 dB -91,1 dBu
65 dB -69,4 dBu
Dynamic range: 118 dB
Line/instrument input
Frequency range: 10 Hz-100 kHz
(-3 dB)
Common mode rejection: 1 kHz: -80 dB/10 kHz: -78 dB
(@ 0 dBu, LINE IN only)
THD & N: Gain: A-weighted:
5 dB -99,4 dBu
20 dB -97,2 dBu
42 dB -79,4 dBu
Input impedance: Line: 20 kOhm / Instrument: 1 MOhm
Maximum input level: Line: +22 dBu / Instrument: +14 dBu
Dynamic range: 119 dB
Output
Maximum output level XLR/TRS: +20dBu
Output impedance: ‹50 Ohm
Dimensions & weight
Standard-EIA-19"/2U housing 482 x 88 x 210 mm
Weight 4,15 kg/ca. 9,15 lbs
Note: 0 dBu = 0,775 V. Specications are subject to change without notice.
24
Channel One
Specifications
Page 25
25
Channel One
Block diagram
Page 26
A
B
C
Diagram 1:
compressor characteristics
Reference curve A displays the
relation between input and
output.
Curve B shows the curve characteristics of the compressor. The
soft knee characteristic is clearly
visible.
Curve C portrays the limiter’s
curve characteristics.
Diagram 2 shows various cut
and boost settings of the air
band lter.
26
Channel One
Measurements
Compressor/Limiter, Air Band
Page 27
Diagram 3 displays various cut
and boost settings of the MHF
lter at 3 kHz.
The proportional-Q characteristic
is distinctly visible.
Diagram 4 displays the curves
of the LMF lter.
Various cut and boost settings
at 150 Hz.
Again the proportional-Q characteristic is clearly visible.
27
Channel One
Mid-Hi Filter, Low Filter
Measurements
Page 28
Copy master: recall settings
Artist:
Album/Gig:
Title:
Engineer:
Track(s)/Groups:
Date:
Page 29
29
Channel One
Please note that you can order products with optional equipment from all dealers, even if they
do only list standard product versions, for example in an online store. Please contact your
dealer or SPL before you place an order. Optional equipment can also be installed after sales.
Available option for the Channel One, model 2950:
• 24 bit/96 kHz A/D converter (user installation possible).
• Lundahl input and output transformers (installation only by qualied technicians or SPL).
24/96 AD converter, model 2376
The optional converter card provides a digital output. Output signals are delivered via a
S/P-DIF output through two sockets: one RCA socket and one optical socket. The converter
provides 24 bit signals. All common sample rates can be selected (see below). Highly accurate
quartz oscillators ensure a clean, low-jitter master clock.
SAMPLE RATE:
The A/D converter allows you to select among the four most common sample
rates of 44.1, 48, 88.2 and 96 kHz. The 44.1/48 button selects one of the two basic sample
rates (out: 44.1 kHz; in: 48 kHz). The x2 button doubles these sample rates to select 88.2 or
96 kHz respectively.
DIGITAL OUTPUTS: The converted S/P-DIF signal is routed in parallel to the RCA and optical
outputs. The signal is in professional format with no sample rate data in the status block.
SYNC INPUT: Since this is an AD converter, the SYNC INPUT is no audio signal input. The
SYNC INPUT allows you to feed the converter with an external sample rate. Connect an S/P-DIF
output from your master source (e.g. DAW interface) to the SYNC input. The AD converter will
automatically switch to the same sample rate that is received. The A/D converter 2376 is not
equipped to accept Word Clock synchronization.
The yellow Sync Lock LED illuminates when a valid sync signal is present at the SYNC INPUT
and the converter is automatically synchronized to the external sample rate.
To prevent interference, the internal oscillators are automatically disabled when an external
clock signal is present. If the sync signal is no longer present (e.g. in the case of a dropout),
the converter automatically reverts to the sample rate selected via the converter’s control
switches.
Information on I/O transformers
We think a good par t of the “warmth” that is commonly associated with vintage gear comes
from transformers. With transformers the low end and lower mids sound rounder, full-bodied
with more punch. The top end gets a silky touch and benets from improved presence without
sounding boosted. Reasons are reduced odd harmonics (which produce harsh top end impressions) and a slower characteristic compared to electronic stages which causes a more voluminous sound. We recommend transformers especially for vocals while electronic stages can be
better for highest precision in signal transmission (transients), but in the end it’s a question of
personal taste, applications or for example which microphones are in use.
Used in SPL preamps or channel strips, the input transformers add ca. 14dB gain (depending
on the microphone). This must be added to the scaled values. The additional passive gain
relieves the complete unit permanently at any gain level. The higher gain levels are also
benecial with ribbon microphones. That’s why the input transformer is more important in
preamps, but to benet from all possible sonic effects and full operational safety, both input
and output should be equipped with transformers.
Options
A/D converter, I/O transformers
OPTICAL
SPDIF SPDIF
x2
48
44,1
24/96 AD
Converter
DIGITAL OUTPUTSSYNC INPUT
SAMPLE RATE
INPUT OUTPUT OUTPUT
SYNC
LOCK
Model
2376
Made in
Taiwan
Page 30
Manual Channel One
Model 2950
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